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))
65 struct _GstRTSPStreamPrivate
69 /* Only one pad is ever set */
70 GstPad *srcpad, *sinkpad;
71 GstElement *payloader;
76 /* TRUE if this stream is running on
77 * the client side of an RTSP link (for RECORD) */
81 GstRTSPProfile profiles;
82 GstRTSPLowerTrans protocols;
84 /* pads on the rtpbin */
85 GstPad *send_rtp_sink;
90 /* the RTPSession object */
93 /* SRTP encoder/decoder */
98 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
100 GstElement *udpsrc_v4[2];
101 /* UDP sources for UDP multicast transports */
102 GstElement *udpsrc_mcast_v4[2];
104 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
106 GstElement *udpsrc_v6[2];
107 /* UDP sources for UDP multicast transports */
108 GstElement *udpsrc_mcast_v6[2];
110 GstElement *udpqueue[2];
111 GstElement *udpsink[2];
113 /* for TCP transport */
114 GstElement *appsrc[2];
115 GstClockTime appsrc_base_time[2];
116 GstElement *appqueue[2];
117 GstElement *appsink[2];
120 GstElement *funnel[2];
125 GstClockTime rtx_time;
127 /* server ports for sending/receiving over ipv4 */
128 GstRTSPRange server_port_v4;
129 GstRTSPAddress *server_addr_v4;
132 /* server ports for sending/receiving over ipv6 */
133 GstRTSPRange server_port_v6;
134 GstRTSPAddress *server_addr_v6;
137 /* multicast addresses */
138 GstRTSPAddressPool *pool;
139 GstRTSPAddress *addr_v4;
140 GstRTSPAddress *addr_v6;
141 gboolean have_ipv4_mcast;
142 gboolean have_ipv6_mcast;
144 gchar *multicast_iface;
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;
162 /* stream blocking */
166 /* pt->caps map for RECORD streams */
169 GstRTSPPublishClockMode publish_clock_mode;
172 #define DEFAULT_CONTROL NULL
173 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
174 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
175 GST_RTSP_LOWER_TRANS_TCP
188 SIGNAL_NEW_RTP_ENCODER,
189 SIGNAL_NEW_RTCP_ENCODER,
193 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
194 #define GST_CAT_DEFAULT rtsp_stream_debug
196 static GQuark ssrc_stream_map_key;
198 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
199 GValue * value, GParamSpec * pspec);
200 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
201 const GValue * value, GParamSpec * pspec);
203 static void gst_rtsp_stream_finalize (GObject * obj);
205 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
207 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
210 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
212 GObjectClass *gobject_class;
214 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
216 gobject_class = G_OBJECT_CLASS (klass);
218 gobject_class->get_property = gst_rtsp_stream_get_property;
219 gobject_class->set_property = gst_rtsp_stream_set_property;
220 gobject_class->finalize = gst_rtsp_stream_finalize;
222 g_object_class_install_property (gobject_class, PROP_CONTROL,
223 g_param_spec_string ("control", "Control",
224 "The control string for this stream", DEFAULT_CONTROL,
225 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
227 g_object_class_install_property (gobject_class, PROP_PROFILES,
228 g_param_spec_flags ("profiles", "Profiles",
229 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
230 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
232 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
233 g_param_spec_flags ("protocols", "Protocols",
234 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
235 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
237 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
238 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
239 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
240 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
242 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
243 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
244 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
245 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
247 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
249 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
253 gst_rtsp_stream_init (GstRTSPStream * stream)
255 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
257 GST_DEBUG ("new stream %p", stream);
262 priv->control = g_strdup (DEFAULT_CONTROL);
263 priv->profiles = DEFAULT_PROFILES;
264 priv->protocols = DEFAULT_PROTOCOLS;
265 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
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 g_free (priv->multicast_iface);
304 gst_object_unref (priv->payloader);
306 gst_object_unref (priv->srcpad);
308 gst_object_unref (priv->sinkpad);
309 g_free (priv->control);
310 g_mutex_clear (&priv->lock);
312 g_hash_table_unref (priv->keys);
313 g_hash_table_destroy (priv->ptmap);
315 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
319 gst_rtsp_stream_get_property (GObject * object, guint propid,
320 GValue * value, GParamSpec * pspec)
322 GstRTSPStream *stream = GST_RTSP_STREAM (object);
326 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
329 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
332 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
335 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
340 gst_rtsp_stream_set_property (GObject * object, guint propid,
341 const GValue * value, GParamSpec * pspec)
343 GstRTSPStream *stream = GST_RTSP_STREAM (object);
347 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
350 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
353 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
356 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
361 * gst_rtsp_stream_new:
364 * @payloader: a #GstElement
366 * Create a new media stream with index @idx that handles RTP data on
367 * @pad and has a payloader element @payloader if @pad is a source pad
368 * or a depayloader element @payloader if @pad is a sink pad.
370 * Returns: (transfer full): a new #GstRTSPStream
373 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
375 GstRTSPStreamPrivate *priv;
376 GstRTSPStream *stream;
378 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
379 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
381 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
384 priv->payloader = gst_object_ref (payloader);
385 if (GST_PAD_IS_SRC (pad))
386 priv->srcpad = gst_object_ref (pad);
388 priv->sinkpad = gst_object_ref (pad);
394 * gst_rtsp_stream_get_index:
395 * @stream: a #GstRTSPStream
397 * Get the stream index.
399 * Return: the stream index.
402 gst_rtsp_stream_get_index (GstRTSPStream * stream)
404 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
406 return stream->priv->idx;
410 * gst_rtsp_stream_get_pt:
411 * @stream: a #GstRTSPStream
413 * Get the stream payload type.
415 * Return: the stream payload type.
418 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
420 GstRTSPStreamPrivate *priv;
423 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
427 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
433 * gst_rtsp_stream_get_srcpad:
434 * @stream: a #GstRTSPStream
436 * Get the srcpad associated with @stream.
438 * Returns: (transfer full): the srcpad. Unref after usage.
441 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
443 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
445 if (!stream->priv->srcpad)
448 return gst_object_ref (stream->priv->srcpad);
452 * gst_rtsp_stream_get_sinkpad:
453 * @stream: a #GstRTSPStream
455 * Get the sinkpad associated with @stream.
457 * Returns: (transfer full): the sinkpad. Unref after usage.
460 gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
462 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
464 if (!stream->priv->sinkpad)
467 return gst_object_ref (stream->priv->sinkpad);
471 * gst_rtsp_stream_get_control:
472 * @stream: a #GstRTSPStream
474 * Get the control string to identify this stream.
476 * Returns: (transfer full): the control string. g_free() after usage.
479 gst_rtsp_stream_get_control (GstRTSPStream * stream)
481 GstRTSPStreamPrivate *priv;
484 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
488 g_mutex_lock (&priv->lock);
489 if ((result = g_strdup (priv->control)) == NULL)
490 result = g_strdup_printf ("stream=%u", priv->idx);
491 g_mutex_unlock (&priv->lock);
497 * gst_rtsp_stream_set_control:
498 * @stream: a #GstRTSPStream
499 * @control: a control string
501 * Set the control string in @stream.
504 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
506 GstRTSPStreamPrivate *priv;
508 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
512 g_mutex_lock (&priv->lock);
513 g_free (priv->control);
514 priv->control = g_strdup (control);
515 g_mutex_unlock (&priv->lock);
519 * gst_rtsp_stream_has_control:
520 * @stream: a #GstRTSPStream
521 * @control: a control string
523 * Check if @stream has the control string @control.
525 * Returns: %TRUE is @stream has @control as the control string
528 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
530 GstRTSPStreamPrivate *priv;
533 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
537 g_mutex_lock (&priv->lock);
539 res = (g_strcmp0 (priv->control, control) == 0);
543 if (sscanf (control, "stream=%u", &streamid) > 0)
544 res = (streamid == priv->idx);
548 g_mutex_unlock (&priv->lock);
554 * gst_rtsp_stream_set_mtu:
555 * @stream: a #GstRTSPStream
558 * Configure the mtu in the payloader of @stream to @mtu.
561 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
563 GstRTSPStreamPrivate *priv;
565 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
569 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
571 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
575 * gst_rtsp_stream_get_mtu:
576 * @stream: a #GstRTSPStream
578 * Get the configured MTU in the payloader of @stream.
580 * Returns: the MTU of the payloader.
583 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
585 GstRTSPStreamPrivate *priv;
588 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
592 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
597 /* Update the dscp qos property on the udp sinks */
599 update_dscp_qos (GstRTSPStream * stream)
601 GstRTSPStreamPrivate *priv;
603 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
607 if (priv->udpsink[0]) {
608 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
612 if (priv->udpsink[1]) {
613 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
619 * gst_rtsp_stream_set_dscp_qos:
620 * @stream: a #GstRTSPStream
621 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
623 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
626 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
628 GstRTSPStreamPrivate *priv;
630 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
634 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
636 if (dscp_qos < -1 || dscp_qos > 63) {
637 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
641 priv->dscp_qos = dscp_qos;
643 update_dscp_qos (stream);
647 * gst_rtsp_stream_get_dscp_qos:
648 * @stream: a #GstRTSPStream
650 * Get the configured DSCP QoS in of the outgoing sockets.
652 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
655 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
657 GstRTSPStreamPrivate *priv;
659 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
663 return priv->dscp_qos;
667 * gst_rtsp_stream_is_transport_supported:
668 * @stream: a #GstRTSPStream
669 * @transport: (transfer none): a #GstRTSPTransport
671 * Check if @transport can be handled by stream
673 * Returns: %TRUE if @transport can be handled by @stream.
676 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
677 GstRTSPTransport * transport)
679 GstRTSPStreamPrivate *priv;
681 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
685 g_mutex_lock (&priv->lock);
686 if (transport->trans != GST_RTSP_TRANS_RTP)
687 goto unsupported_transmode;
689 if (!(transport->profile & priv->profiles))
690 goto unsupported_profile;
692 if (!(transport->lower_transport & priv->protocols))
693 goto unsupported_ltrans;
695 g_mutex_unlock (&priv->lock);
700 unsupported_transmode:
702 GST_DEBUG ("unsupported transport mode %d", transport->trans);
703 g_mutex_unlock (&priv->lock);
708 GST_DEBUG ("unsupported profile %d", transport->profile);
709 g_mutex_unlock (&priv->lock);
714 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
715 g_mutex_unlock (&priv->lock);
721 * gst_rtsp_stream_set_profiles:
722 * @stream: a #GstRTSPStream
723 * @profiles: the new profiles
725 * Configure the allowed profiles for @stream.
728 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
730 GstRTSPStreamPrivate *priv;
732 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
736 g_mutex_lock (&priv->lock);
737 priv->profiles = profiles;
738 g_mutex_unlock (&priv->lock);
742 * gst_rtsp_stream_get_profiles:
743 * @stream: a #GstRTSPStream
745 * Get the allowed profiles of @stream.
747 * Returns: a #GstRTSPProfile
750 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
752 GstRTSPStreamPrivate *priv;
755 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
759 g_mutex_lock (&priv->lock);
760 res = priv->profiles;
761 g_mutex_unlock (&priv->lock);
767 * gst_rtsp_stream_set_protocols:
768 * @stream: a #GstRTSPStream
769 * @protocols: the new flags
771 * Configure the allowed lower transport for @stream.
774 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
775 GstRTSPLowerTrans protocols)
777 GstRTSPStreamPrivate *priv;
779 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
783 g_mutex_lock (&priv->lock);
784 priv->protocols = protocols;
785 g_mutex_unlock (&priv->lock);
789 * gst_rtsp_stream_get_protocols:
790 * @stream: a #GstRTSPStream
792 * Get the allowed protocols of @stream.
794 * Returns: a #GstRTSPLowerTrans
797 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
799 GstRTSPStreamPrivate *priv;
800 GstRTSPLowerTrans res;
802 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
803 GST_RTSP_LOWER_TRANS_UNKNOWN);
807 g_mutex_lock (&priv->lock);
808 res = priv->protocols;
809 g_mutex_unlock (&priv->lock);
815 * gst_rtsp_stream_set_address_pool:
816 * @stream: a #GstRTSPStream
817 * @pool: (transfer none): a #GstRTSPAddressPool
819 * configure @pool to be used as the address pool of @stream.
822 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
823 GstRTSPAddressPool * pool)
825 GstRTSPStreamPrivate *priv;
826 GstRTSPAddressPool *old;
828 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
832 GST_LOG_OBJECT (stream, "set address pool %p", pool);
834 g_mutex_lock (&priv->lock);
835 if ((old = priv->pool) != pool)
836 priv->pool = pool ? g_object_ref (pool) : NULL;
839 g_mutex_unlock (&priv->lock);
842 g_object_unref (old);
846 * gst_rtsp_stream_get_address_pool:
847 * @stream: a #GstRTSPStream
849 * Get the #GstRTSPAddressPool used as the address pool of @stream.
851 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
855 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
857 GstRTSPStreamPrivate *priv;
858 GstRTSPAddressPool *result;
860 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
864 g_mutex_lock (&priv->lock);
865 if ((result = priv->pool))
866 g_object_ref (result);
867 g_mutex_unlock (&priv->lock);
873 * gst_rtsp_stream_set_multicast_iface:
874 * @stream: a #GstRTSPStream
875 * @multicast_iface: (transfer none): a multicast interface
877 * configure @multicast_iface to be used for @stream.
880 gst_rtsp_stream_set_multicast_iface (GstRTSPStream * stream,
881 const gchar * multicast_iface)
883 GstRTSPStreamPrivate *priv;
886 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
890 GST_LOG_OBJECT (stream, "set multicast iface %s",
891 GST_STR_NULL (multicast_iface));
893 g_mutex_lock (&priv->lock);
894 if ((old = priv->multicast_iface) != multicast_iface)
895 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
898 g_mutex_unlock (&priv->lock);
905 * gst_rtsp_stream_get_multicast_iface:
906 * @stream: a #GstRTSPStream
908 * Get the multicast interface used for @stream.
910 * Returns: (transfer full): the multicast interface for @stream. g_free() after
914 gst_rtsp_stream_get_multicast_iface (GstRTSPStream * stream)
916 GstRTSPStreamPrivate *priv;
919 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
923 g_mutex_lock (&priv->lock);
924 if ((result = priv->multicast_iface))
925 result = g_strdup (result);
926 g_mutex_unlock (&priv->lock);
932 * gst_rtsp_stream_get_multicast_address:
933 * @stream: a #GstRTSPStream
934 * @family: the #GSocketFamily
936 * Get the multicast address of @stream for @family.
938 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
939 * or %NULL when no address could be allocated. gst_rtsp_address_free()
943 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
944 GSocketFamily family)
946 GstRTSPStreamPrivate *priv;
947 GstRTSPAddress *result;
948 GstRTSPAddress **addrp;
949 GstRTSPAddressFlags flags;
951 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
955 if (family == G_SOCKET_FAMILY_IPV6) {
956 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
957 addrp = &priv->addr_v6;
959 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
960 addrp = &priv->addr_v4;
963 g_mutex_lock (&priv->lock);
964 if (*addrp == NULL) {
965 if (priv->pool == NULL)
968 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
970 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
974 result = gst_rtsp_address_copy (*addrp);
975 g_mutex_unlock (&priv->lock);
982 GST_ERROR_OBJECT (stream, "no address pool specified");
983 g_mutex_unlock (&priv->lock);
988 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
989 g_mutex_unlock (&priv->lock);
995 * gst_rtsp_stream_reserve_address:
996 * @stream: a #GstRTSPStream
997 * @address: an address
1002 * Reserve @address and @port as the address and port of @stream.
1004 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
1005 * the address could be reserved. gst_rtsp_address_free() after usage.
1008 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
1009 const gchar * address, guint port, guint n_ports, guint ttl)
1011 GstRTSPStreamPrivate *priv;
1012 GstRTSPAddress *result;
1014 GSocketFamily family;
1015 GstRTSPAddress **addrp;
1017 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1018 g_return_val_if_fail (address != NULL, NULL);
1019 g_return_val_if_fail (port > 0, NULL);
1020 g_return_val_if_fail (n_ports > 0, NULL);
1021 g_return_val_if_fail (ttl > 0, NULL);
1023 priv = stream->priv;
1025 addr = g_inet_address_new_from_string (address);
1027 GST_ERROR ("failed to get inet addr from %s", address);
1028 family = G_SOCKET_FAMILY_IPV4;
1030 family = g_inet_address_get_family (addr);
1031 g_object_unref (addr);
1034 if (family == G_SOCKET_FAMILY_IPV6)
1035 addrp = &priv->addr_v6;
1037 addrp = &priv->addr_v4;
1039 g_mutex_lock (&priv->lock);
1040 if (*addrp == NULL) {
1041 GstRTSPAddressPoolResult res;
1043 if (priv->pool == NULL)
1046 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
1047 port, n_ports, ttl, addrp);
1048 if (res != GST_RTSP_ADDRESS_POOL_OK)
1051 if (strcmp ((*addrp)->address, address) ||
1052 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
1053 (*addrp)->ttl != ttl)
1054 goto different_address;
1056 result = gst_rtsp_address_copy (*addrp);
1057 g_mutex_unlock (&priv->lock);
1064 GST_ERROR_OBJECT (stream, "no address pool specified");
1065 g_mutex_unlock (&priv->lock);
1070 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
1072 g_mutex_unlock (&priv->lock);
1077 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
1078 " reserved", address);
1079 g_mutex_unlock (&priv->lock);
1084 /* must be called with lock */
1086 set_sockets_for_udpsinks (GstRTSPStream * stream, GSocket * rtp_socket,
1087 GSocket * rtcp_socket, GSocketFamily family)
1089 GstRTSPStreamPrivate *priv = stream->priv;
1090 const gchar *multisink_socket;
1092 if (family == G_SOCKET_FAMILY_IPV6)
1093 multisink_socket = "socket-v6";
1095 multisink_socket = "socket";
1097 g_object_set (G_OBJECT (priv->udpsink[0]), multisink_socket, rtp_socket,
1099 g_object_set (G_OBJECT (priv->udpsink[1]), multisink_socket, rtcp_socket,
1103 /* must be called with lock */
1105 create_and_configure_udpsinks (GstRTSPStream * stream)
1107 GstRTSPStreamPrivate *priv = stream->priv;
1108 GstElement *udpsink0, *udpsink1;
1113 if (priv->udpsink[0])
1114 udpsink0 = priv->udpsink[0];
1116 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1119 goto no_udp_protocol;
1121 if (priv->udpsink[1])
1122 udpsink1 = priv->udpsink[1];
1124 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1127 goto no_udp_protocol;
1129 /* configure sinks */
1131 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1132 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1134 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1135 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1137 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
1139 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1140 /* Needs to be async for RECORD streams, otherwise we will never go to
1141 * PLAYING because the sinks will wait for data while the udpsrc can't
1142 * provide data with timestamps in PAUSED. */
1144 g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
1145 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1147 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1148 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1150 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1151 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1153 /* update the dscp qos field in the sinks */
1154 update_dscp_qos (stream);
1156 priv->udpsink[0] = udpsink0;
1157 priv->udpsink[1] = udpsink1;
1168 /* must be called with lock */
1170 play_udpsources_one_family (GstRTSPStream * stream, GstElement * udpsrc_out[2],
1171 GSocketFamily family)
1173 GstRTSPStreamPrivate *priv;
1174 GstPad *pad, *selpad;
1178 priv = stream->priv;
1179 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[1])));
1181 for (i = 0; i < 2; i++) {
1182 if (priv->sinkpad || i == 1) {
1184 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1185 * values. This is only relevant for PLAY pipelines */
1186 gst_element_set_state (udpsrc_out[i], GST_STATE_PLAYING);
1187 gst_element_set_locked_state (udpsrc_out[i], TRUE);
1190 gst_bin_add (bin, udpsrc_out[i]);
1192 /* and link to the funnel */
1193 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1194 pad = gst_element_get_static_pad (udpsrc_out[i], "src");
1195 gst_pad_link (pad, selpad);
1196 gst_object_unref (pad);
1197 gst_object_unref (selpad);
1199 /* otherwise sync state with parent in case it's running already
1201 if (!priv->srcpad) {
1202 gst_element_sync_state_with_parent (udpsrc_out[i]);
1207 gst_object_unref (bin);
1210 /* must be called with lock */
1212 create_and_configure_udpsources_one_family (GstElement * udpsrc_out[2],
1213 GSocket * rtp_socket, GSocket * rtcp_socket, GSocketFamily family,
1214 const gchar * address, gint rtpport, gint rtcpport,
1215 const gchar * multicast_iface, GstRTSPLowerTrans transport)
1217 GstStateChangeReturn ret;
1219 udpsrc_out[0] = gst_element_factory_make ("udpsrc", NULL);
1220 udpsrc_out[1] = gst_element_factory_make ("udpsrc", NULL);
1222 if (udpsrc_out[0] == NULL || udpsrc_out[1] == NULL)
1225 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1226 g_object_set (G_OBJECT (udpsrc_out[0]), "address", address, NULL);
1227 g_object_set (G_OBJECT (udpsrc_out[1]), "address", address, NULL);
1228 g_object_set (G_OBJECT (udpsrc_out[0]), "port", rtpport, NULL);
1229 g_object_set (G_OBJECT (udpsrc_out[1]), "port", rtcpport, NULL);
1230 g_object_set (G_OBJECT (udpsrc_out[0]), "multicast-iface", multicast_iface,
1232 g_object_set (G_OBJECT (udpsrc_out[1]), "multicast-iface", multicast_iface,
1234 g_object_set (G_OBJECT (udpsrc_out[0]), "loop", FALSE, NULL);
1235 g_object_set (G_OBJECT (udpsrc_out[1]), "loop", FALSE, NULL);
1238 g_object_set (G_OBJECT (udpsrc_out[0]), "socket", rtp_socket, NULL);
1239 g_object_set (G_OBJECT (udpsrc_out[1]), "socket", rtcp_socket, NULL);
1241 ret = gst_element_set_state (udpsrc_out[0], GST_STATE_READY);
1242 if (ret == GST_STATE_CHANGE_FAILURE)
1244 ret = gst_element_set_state (udpsrc_out[1], GST_STATE_READY);
1245 if (ret == GST_STATE_CHANGE_FAILURE)
1255 gst_object_unref (udpsrc_out[0]);
1257 gst_object_unref (udpsrc_out[1]);
1263 alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
1264 GstElement * udpsrc_out[2], GstRTSPRange * server_port_out,
1265 GstRTSPTransport * ct, GstRTSPAddress ** server_addr_out,
1266 gboolean use_client_settings)
1268 GstRTSPStreamPrivate *priv = stream->priv;
1269 GSocket *rtp_socket = NULL;
1270 GSocket *rtcp_socket;
1271 gint tmp_rtp, tmp_rtcp;
1273 gint rtpport, rtcpport;
1274 GList *rejected_addresses = NULL;
1275 GstRTSPAddress *addr = NULL;
1276 GInetAddress *inetaddr = NULL;
1278 GSocketAddress *rtp_sockaddr = NULL;
1279 GSocketAddress *rtcp_sockaddr = NULL;
1280 GstRTSPAddressPool *pool;
1281 GstRTSPLowerTrans transport;
1282 const gchar *multicast_iface = priv->multicast_iface;
1286 transport = ct->lower_transport;
1288 /* Start with random port */
1291 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1292 G_SOCKET_PROTOCOL_UDP, NULL);
1294 goto no_udp_protocol;
1295 g_socket_set_multicast_loopback (rtcp_socket, FALSE);
1297 if (*server_addr_out)
1298 gst_rtsp_address_free (*server_addr_out);
1300 /* try to allocate 2 UDP ports, the RTP port should be an even
1301 * number and the RTCP port should be the next (uneven) port */
1304 if (rtp_socket == NULL) {
1305 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1306 G_SOCKET_PROTOCOL_UDP, NULL);
1308 goto no_udp_protocol;
1309 g_socket_set_multicast_loopback (rtp_socket, FALSE);
1312 if (pool && ((transport == GST_RTSP_LOWER_TRANS_UDP &&
1313 gst_rtsp_address_pool_has_unicast_addresses (pool))
1314 || transport == GST_RTSP_LOWER_TRANS_UDP_MCAST)) {
1315 GstRTSPAddressFlags flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
1317 if (transport == GST_RTSP_LOWER_TRANS_UDP)
1318 flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
1320 flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
1323 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1325 if (family == G_SOCKET_FAMILY_IPV6)
1326 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1328 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1330 if (ct->destination && transport == GST_RTSP_LOWER_TRANS_UDP_MCAST
1331 && use_client_settings)
1332 gst_rtsp_address_pool_reserve_address (pool, ct->destination,
1333 ct->port.min, 2, ct->ttl, &addr);
1335 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1340 tmp_rtp = addr->port;
1342 g_clear_object (&inetaddr);
1343 inetaddr = g_inet_address_new_from_string (addr->address);
1345 /* If we're supposed to bind to a multicast address, instead bind
1346 * to ANY and let udpsrc later join the relevant multicast group
1348 if (g_inet_address_get_is_multicast (inetaddr)) {
1349 g_object_unref (inetaddr);
1350 inetaddr = g_inet_address_new_any (family);
1359 if (inetaddr == NULL)
1360 inetaddr = g_inet_address_new_any (family);
1363 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1364 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1365 g_object_unref (rtp_sockaddr);
1368 g_object_unref (rtp_sockaddr);
1370 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1371 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1372 g_clear_object (&rtp_sockaddr);
1377 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1378 g_object_unref (rtp_sockaddr);
1380 /* check if port is even */
1381 if ((tmp_rtp & 1) != 0) {
1382 /* port not even, close and allocate another */
1384 g_clear_object (&rtp_socket);
1389 tmp_rtcp = tmp_rtp + 1;
1391 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1392 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1393 g_object_unref (rtcp_sockaddr);
1394 g_clear_object (&rtp_socket);
1397 g_object_unref (rtcp_sockaddr);
1400 addr_str = g_inet_address_to_string (inetaddr);
1402 addr_str = addr->address;
1403 g_clear_object (&inetaddr);
1405 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1406 rtcp_socket, family, addr_str, tmp_rtp, tmp_rtcp, multicast_iface,
1410 goto no_udp_protocol;
1416 play_udpsources_one_family (stream, udpsrc_out, family);
1418 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1419 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1421 /* this should not happen... */
1422 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1425 /* set RTP and RTCP sockets */
1426 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1428 server_port_out->min = rtpport;
1429 server_port_out->max = rtcpport;
1431 *server_addr_out = addr;
1432 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1434 g_object_unref (rtp_socket);
1435 g_object_unref (rtcp_socket);
1459 g_object_unref (inetaddr);
1460 g_list_free_full (rejected_addresses,
1461 (GDestroyNotify) gst_rtsp_address_free);
1463 gst_rtsp_address_free (addr);
1465 g_object_unref (rtp_socket);
1467 g_object_unref (rtcp_socket);
1473 * gst_rtsp_stream_allocate_udp_sockets:
1474 * @stream: a #GstRTSPStream
1475 * @family: protocol family
1476 * @transport_method: transport method
1478 * Allocates RTP and RTCP ports.
1480 * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
1483 gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
1484 GSocketFamily family, GstRTSPTransport * ct, gboolean use_client_settings)
1486 GstRTSPStreamPrivate *priv;
1487 gboolean result = FALSE;
1488 GstRTSPLowerTrans transport = ct->lower_transport;
1490 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1491 priv = stream->priv;
1492 g_return_val_if_fail (priv->is_joined, FALSE);
1494 g_mutex_lock (&priv->lock);
1496 if (family == G_SOCKET_FAMILY_IPV4) {
1497 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1498 if (priv->have_ipv4_mcast)
1500 priv->have_ipv4_mcast =
1501 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
1502 priv->udpsrc_mcast_v4, &priv->server_port_v4, ct, &priv->addr_v4,
1503 use_client_settings);
1506 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1507 &priv->server_port_v4, ct, &priv->server_addr_v4,
1508 use_client_settings);
1511 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1512 if (priv->have_ipv6_mcast)
1514 priv->have_ipv6_mcast =
1515 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
1516 priv->udpsrc_mcast_v6, &priv->server_port_v6, ct, &priv->addr_v6,
1517 use_client_settings);
1519 if (priv->have_ipv6)
1522 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1523 &priv->server_port_v6, ct, &priv->server_addr_v6,
1524 use_client_settings);
1529 result = priv->have_ipv4 || priv->have_ipv4_mcast || priv->have_ipv6 ||
1530 priv->have_ipv6_mcast;
1532 g_mutex_unlock (&priv->lock);
1538 * gst_rtsp_stream_set_client_side:
1539 * @stream: a #GstRTSPStream
1540 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1541 * an RTSP connection.
1543 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1544 * streams to an RTSP server via RECORD. This has the practical effect
1545 * of changing which UDP port numbers are used when setting up the local
1546 * side of the stream sending to be either the 'server' or 'client' pair
1547 * of a configured UDP transport.
1550 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1552 GstRTSPStreamPrivate *priv;
1554 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1555 priv = stream->priv;
1556 g_mutex_lock (&priv->lock);
1557 priv->client_side = client_side;
1558 g_mutex_unlock (&priv->lock);
1562 * gst_rtsp_stream_is_client_side:
1563 * @stream: a #GstRTSPStream
1565 * See gst_rtsp_stream_set_client_side()
1567 * Returns: TRUE if this #GstRTSPStream is client-side.
1570 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1572 GstRTSPStreamPrivate *priv;
1575 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1577 priv = stream->priv;
1578 g_mutex_lock (&priv->lock);
1579 ret = priv->client_side;
1580 g_mutex_unlock (&priv->lock);
1586 * gst_rtsp_stream_get_server_port:
1587 * @stream: a #GstRTSPStream
1588 * @server_port: (out): result server port
1589 * @family: the port family to get
1591 * Fill @server_port with the port pair used by the server. This function can
1592 * only be called when @stream has been joined.
1595 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1596 GstRTSPRange * server_port, GSocketFamily family)
1598 GstRTSPStreamPrivate *priv;
1600 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1601 priv = stream->priv;
1602 g_return_if_fail (priv->is_joined);
1604 g_mutex_lock (&priv->lock);
1605 if (family == G_SOCKET_FAMILY_IPV4) {
1607 *server_port = priv->server_port_v4;
1610 *server_port = priv->server_port_v6;
1612 g_mutex_unlock (&priv->lock);
1616 * gst_rtsp_stream_get_rtpsession:
1617 * @stream: a #GstRTSPStream
1619 * Get the RTP session of this stream.
1621 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1624 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1626 GstRTSPStreamPrivate *priv;
1629 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1631 priv = stream->priv;
1633 g_mutex_lock (&priv->lock);
1634 if ((session = priv->session))
1635 g_object_ref (session);
1636 g_mutex_unlock (&priv->lock);
1642 * gst_rtsp_stream_get_encoder:
1643 * @stream: a #GstRTSPStream
1645 * Get the SRTP encoder for this stream.
1647 * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
1650 gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
1652 GstRTSPStreamPrivate *priv;
1653 GstElement *encoder;
1655 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1657 priv = stream->priv;
1659 g_mutex_lock (&priv->lock);
1660 if ((encoder = priv->srtpenc))
1661 g_object_ref (encoder);
1662 g_mutex_unlock (&priv->lock);
1668 * gst_rtsp_stream_get_ssrc:
1669 * @stream: a #GstRTSPStream
1670 * @ssrc: (out): result ssrc
1672 * Get the SSRC used by the RTP session of this stream. This function can only
1673 * be called when @stream has been joined.
1676 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1678 GstRTSPStreamPrivate *priv;
1680 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1681 priv = stream->priv;
1682 g_return_if_fail (priv->is_joined);
1684 g_mutex_lock (&priv->lock);
1685 if (ssrc && priv->session)
1686 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1687 g_mutex_unlock (&priv->lock);
1691 * gst_rtsp_stream_set_retransmission_time:
1692 * @stream: a #GstRTSPStream
1693 * @time: a #GstClockTime
1695 * Set the amount of time to store retransmission packets.
1698 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1701 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1703 g_mutex_lock (&stream->priv->lock);
1704 stream->priv->rtx_time = time;
1705 if (stream->priv->rtxsend)
1706 g_object_set (stream->priv->rtxsend, "max-size-time",
1707 GST_TIME_AS_MSECONDS (time), NULL);
1708 g_mutex_unlock (&stream->priv->lock);
1712 * gst_rtsp_stream_get_retransmission_time:
1713 * @stream: a #GstRTSPStream
1715 * Get the amount of time to store retransmission data.
1717 * Returns: the amount of time to store retransmission data.
1720 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1724 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1726 g_mutex_lock (&stream->priv->lock);
1727 ret = stream->priv->rtx_time;
1728 g_mutex_unlock (&stream->priv->lock);
1734 * gst_rtsp_stream_set_retransmission_pt:
1735 * @stream: a #GstRTSPStream
1738 * Set the payload type (pt) for retransmission of this stream.
1741 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1743 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1745 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1747 g_mutex_lock (&stream->priv->lock);
1748 stream->priv->rtx_pt = rtx_pt;
1749 if (stream->priv->rtxsend) {
1750 guint pt = gst_rtsp_stream_get_pt (stream);
1751 gchar *pt_s = g_strdup_printf ("%d", pt);
1752 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1753 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1754 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1756 gst_structure_free (rtx_pt_map);
1758 g_mutex_unlock (&stream->priv->lock);
1762 * gst_rtsp_stream_get_retransmission_pt:
1763 * @stream: a #GstRTSPStream
1765 * Get the payload-type used for retransmission of this stream
1767 * Returns: The retransmission PT.
1770 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1774 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1776 g_mutex_lock (&stream->priv->lock);
1777 rtx_pt = stream->priv->rtx_pt;
1778 g_mutex_unlock (&stream->priv->lock);
1784 * gst_rtsp_stream_set_buffer_size:
1785 * @stream: a #GstRTSPStream
1786 * @size: the buffer size
1788 * Set the size of the UDP transmission buffer (in bytes)
1789 * Needs to be set before the stream is joined to a bin.
1794 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1796 g_mutex_lock (&stream->priv->lock);
1797 stream->priv->buffer_size = size;
1798 g_mutex_unlock (&stream->priv->lock);
1802 * gst_rtsp_stream_get_buffer_size:
1803 * @stream: a #GstRTSPStream
1805 * Get the size of the UDP transmission buffer (in bytes)
1807 * Returns: the size of the UDP TX buffer
1812 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1816 g_mutex_lock (&stream->priv->lock);
1817 buffer_size = stream->priv->buffer_size;
1818 g_mutex_unlock (&stream->priv->lock);
1823 /* executed from streaming thread */
1825 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1827 GstRTSPStreamPrivate *priv = stream->priv;
1828 GstCaps *newcaps, *oldcaps;
1830 newcaps = gst_pad_get_current_caps (pad);
1832 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1835 g_mutex_lock (&priv->lock);
1836 oldcaps = priv->caps;
1837 priv->caps = newcaps;
1838 g_mutex_unlock (&priv->lock);
1841 gst_caps_unref (oldcaps);
1845 dump_structure (const GstStructure * s)
1849 sstr = gst_structure_to_string (s);
1850 GST_INFO ("structure: %s", sstr);
1854 static GstRTSPStreamTransport *
1855 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1857 GstRTSPStreamPrivate *priv = stream->priv;
1859 GstRTSPStreamTransport *result = NULL;
1864 if (rtcp_from == NULL)
1867 tmp = g_strrstr (rtcp_from, ":");
1871 port = atoi (tmp + 1);
1872 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1874 g_mutex_lock (&priv->lock);
1875 GST_INFO ("finding %s:%d in %d transports", dest, port,
1876 g_list_length (priv->transports));
1878 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1879 GstRTSPStreamTransport *trans = walk->data;
1880 const GstRTSPTransport *tr;
1883 tr = gst_rtsp_stream_transport_get_transport (trans);
1885 if (priv->client_side) {
1886 /* In client side mode the 'destination' is the RTSP server, so send
1888 min = tr->server_port.min;
1889 max = tr->server_port.max;
1891 min = tr->client_port.min;
1892 max = tr->client_port.max;
1895 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1901 g_object_ref (result);
1902 g_mutex_unlock (&priv->lock);
1909 static GstRTSPStreamTransport *
1910 check_transport (GObject * source, GstRTSPStream * stream)
1912 GstStructure *stats;
1913 GstRTSPStreamTransport *trans;
1915 /* see if we have a stream to match with the origin of the RTCP packet */
1916 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1917 if (trans == NULL) {
1918 g_object_get (source, "stats", &stats, NULL);
1920 const gchar *rtcp_from;
1922 dump_structure (stats);
1924 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1925 if ((trans = find_transport (stream, rtcp_from))) {
1926 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1928 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1931 gst_structure_free (stats);
1939 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1941 GstRTSPStreamTransport *trans;
1943 GST_INFO ("%p: new source %p", stream, source);
1945 trans = check_transport (source, stream);
1948 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1952 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1954 GST_INFO ("%p: new SDES %p", stream, source);
1958 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1960 GstRTSPStreamTransport *trans;
1962 trans = check_transport (source, stream);
1965 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1966 gst_rtsp_stream_transport_keep_alive (trans);
1970 GstStructure *stats;
1971 g_object_get (source, "stats", &stats, NULL);
1973 dump_structure (stats);
1974 gst_structure_free (stats);
1981 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1983 GST_INFO ("%p: source %p bye", stream, source);
1987 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1989 GstRTSPStreamTransport *trans;
1991 GST_INFO ("%p: source %p bye timeout", stream, source);
1993 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1994 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1995 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
2000 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2002 GstRTSPStreamTransport *trans;
2004 GST_INFO ("%p: source %p timeout", stream, source);
2006 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
2007 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
2008 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
2013 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2015 GST_INFO ("%p: new sender source %p", stream, source);
2018 GstStructure *stats;
2019 g_object_get (source, "stats", &stats, NULL);
2021 dump_structure (stats);
2022 gst_structure_free (stats);
2029 on_sender_ssrc_active (GObject * session, GObject * source,
2030 GstRTSPStream * stream)
2034 GstStructure *stats;
2035 g_object_get (source, "stats", &stats, NULL);
2037 dump_structure (stats);
2038 gst_structure_free (stats);
2045 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
2048 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
2049 g_list_free (priv->tr_cache_rtp);
2050 priv->tr_cache_rtp = NULL;
2052 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
2053 g_list_free (priv->tr_cache_rtcp);
2054 priv->tr_cache_rtcp = NULL;
2058 static GstFlowReturn
2059 handle_new_sample (GstAppSink * sink, gpointer user_data)
2061 GstRTSPStreamPrivate *priv;
2065 GstRTSPStream *stream;
2068 sample = gst_app_sink_pull_sample (sink);
2072 stream = (GstRTSPStream *) user_data;
2073 priv = stream->priv;
2074 buffer = gst_sample_get_buffer (sample);
2076 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
2078 g_mutex_lock (&priv->lock);
2080 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
2081 clear_tr_cache (priv, is_rtp);
2082 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2083 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2084 priv->tr_cache_rtp =
2085 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
2087 priv->tr_cache_cookie_rtp = priv->transports_cookie;
2090 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
2091 clear_tr_cache (priv, is_rtp);
2092 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2093 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2094 priv->tr_cache_rtcp =
2095 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
2097 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
2100 g_mutex_unlock (&priv->lock);
2103 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
2104 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2105 gst_rtsp_stream_transport_send_rtp (tr, buffer);
2108 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
2109 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2110 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
2113 gst_sample_unref (sample);
2118 static GstAppSinkCallbacks sink_cb = {
2119 NULL, /* not interested in EOS */
2120 NULL, /* not interested in preroll samples */
2125 get_rtp_encoder (GstRTSPStream * stream, guint session)
2127 GstRTSPStreamPrivate *priv = stream->priv;
2129 if (priv->srtpenc == NULL) {
2132 name = g_strdup_printf ("srtpenc_%u", session);
2133 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
2136 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
2138 return gst_object_ref (priv->srtpenc);
2142 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2144 GstRTSPStreamPrivate *priv = stream->priv;
2145 GstElement *oldenc, *enc;
2149 if (priv->idx != session)
2152 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
2154 oldenc = priv->srtpenc;
2155 enc = get_rtp_encoder (stream, session);
2156 name = g_strdup_printf ("rtp_sink_%d", session);
2157 pad = gst_element_get_request_pad (enc, name);
2159 gst_object_unref (pad);
2162 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
2169 request_rtcp_encoder (GstElement * rtpbin, guint session,
2170 GstRTSPStream * stream)
2172 GstRTSPStreamPrivate *priv = stream->priv;
2173 GstElement *oldenc, *enc;
2177 if (priv->idx != session)
2180 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
2182 oldenc = priv->srtpenc;
2183 enc = get_rtp_encoder (stream, session);
2184 name = g_strdup_printf ("rtcp_sink_%d", session);
2185 pad = gst_element_get_request_pad (enc, name);
2187 gst_object_unref (pad);
2190 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
2197 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2199 GstRTSPStreamPrivate *priv = stream->priv;
2202 GST_DEBUG ("request key %08x", ssrc);
2204 g_mutex_lock (&priv->lock);
2205 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
2206 gst_caps_ref (caps);
2207 g_mutex_unlock (&priv->lock);
2213 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
2214 GstRTSPStream * stream)
2216 GstRTSPStreamPrivate *priv = stream->priv;
2218 if (priv->idx != session)
2221 if (priv->srtpdec == NULL) {
2224 name = g_strdup_printf ("srtpdec_%u", session);
2225 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2228 g_signal_connect (priv->srtpdec, "request-key",
2229 (GCallback) request_key, stream);
2231 return gst_object_ref (priv->srtpdec);
2235 * gst_rtsp_stream_request_aux_sender:
2236 * @stream: a #GstRTSPStream
2237 * @sessid: the session id
2239 * Creating a rtxsend bin
2241 * Returns: (transfer full): a #GstElement.
2246 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2250 GstStructure *pt_map;
2255 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2257 pt = gst_rtsp_stream_get_pt (stream);
2258 pt_s = g_strdup_printf ("%u", pt);
2259 rtx_pt = stream->priv->rtx_pt;
2261 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2263 bin = gst_bin_new (NULL);
2264 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2265 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2266 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2267 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2268 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2270 gst_structure_free (pt_map);
2271 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2273 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2274 name = g_strdup_printf ("src_%u", sessid);
2275 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2277 gst_object_unref (pad);
2279 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2280 name = g_strdup_printf ("sink_%u", sessid);
2281 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2283 gst_object_unref (pad);
2289 * gst_rtsp_stream_set_pt_map:
2290 * @stream: a #GstRTSPStream
2294 * Configure a pt map between @pt and @caps.
2297 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2299 GstRTSPStreamPrivate *priv = stream->priv;
2301 g_mutex_lock (&priv->lock);
2302 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2303 g_mutex_unlock (&priv->lock);
2307 * gst_rtsp_stream_set_publish_clock_mode:
2308 * @stream: a #GstRTSPStream
2309 * @mode: the clock publish mode
2311 * Sets if and how the stream clock should be published according to RFC7273.
2316 gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream,
2317 GstRTSPPublishClockMode mode)
2319 GstRTSPStreamPrivate *priv;
2321 priv = stream->priv;
2322 g_mutex_lock (&priv->lock);
2323 priv->publish_clock_mode = mode;
2324 g_mutex_unlock (&priv->lock);
2328 * gst_rtsp_stream_get_publish_clock_mode:
2329 * @factory: a #GstRTSPStream
2331 * Gets if and how the stream clock should be published according to RFC7273.
2333 * Returns: The GstRTSPPublishClockMode
2337 GstRTSPPublishClockMode
2338 gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream)
2340 GstRTSPStreamPrivate *priv;
2341 GstRTSPPublishClockMode ret;
2343 priv = stream->priv;
2344 g_mutex_lock (&priv->lock);
2345 ret = priv->publish_clock_mode;
2346 g_mutex_unlock (&priv->lock);
2352 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2353 GstRTSPStream * stream)
2355 GstRTSPStreamPrivate *priv = stream->priv;
2356 GstCaps *caps = NULL;
2358 g_mutex_lock (&priv->lock);
2360 if (priv->idx == session) {
2361 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2363 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2364 gst_caps_ref (caps);
2366 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2370 g_mutex_unlock (&priv->lock);
2376 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2378 GstRTSPStreamPrivate *priv = stream->priv;
2380 GstPadLinkReturn ret;
2383 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2384 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2386 name = gst_pad_get_name (pad);
2387 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2393 if (priv->idx != sessid)
2396 if (gst_pad_is_linked (priv->sinkpad)) {
2397 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2398 GST_DEBUG_PAD_NAME (priv->sinkpad));
2402 /* link the RTP pad to the session manager, it should not really fail unless
2403 * this is not really an RTP pad */
2404 ret = gst_pad_link (pad, priv->sinkpad);
2405 if (ret != GST_PAD_LINK_OK)
2407 priv->recv_rtp_src = gst_object_ref (pad);
2414 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2415 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2420 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2421 GstRTSPStream * stream)
2423 /* TODO: What to do here other than this? */
2424 GST_DEBUG ("Stream %p: Got EOS", stream);
2425 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2428 /* must be called with lock */
2430 create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2432 GstRTSPStreamPrivate *priv;
2433 GstPad *pad, *sinkpad = NULL;
2434 gboolean is_tcp = FALSE, is_udp = FALSE;
2437 priv = stream->priv;
2439 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2440 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2441 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2443 if (is_udp && !create_and_configure_udpsinks (stream))
2444 goto no_udp_protocol;
2446 for (i = 0; i < 2; i++) {
2447 GstPad *teepad, *queuepad;
2448 /* For the sender we create this bit of pipeline for both
2449 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2450 * we need to add a queue before appsink and udpsink to make
2451 * the pipeline not block. For the TCP case, we want to pump
2452 * client as fast as possible anyway. This pipeline is used
2453 * when both TCP and UDP are present.
2455 * .--------. .-----. .---------. .---------.
2456 * | rtpbin | | tee | | queue | | udpsink |
2457 * | send->sink src->sink src->sink |
2458 * '--------' | | '---------' '---------'
2459 * | | .---------. .---------.
2460 * | | | queue | | appsink |
2461 * | src->sink src->sink |
2462 * '-----' '---------' '---------'
2464 * When only UDP or only TCP is allowed, we skip the tee and queue
2465 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2468 /* Only link the RTP send src if we're going to send RTP, link
2469 * the RTCP send src always */
2470 if (priv->srcpad || i == 1) {
2473 gst_bin_add (bin, priv->udpsink[i]);
2474 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2479 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2480 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2481 gst_bin_add (bin, priv->appsink[i]);
2482 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2483 &sink_cb, stream, NULL);
2486 if (is_udp && is_tcp) {
2487 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2489 /* make tee for RTP/RTCP */
2490 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2491 gst_bin_add (bin, priv->tee[i]);
2493 /* and link to rtpbin send pad */
2494 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2495 gst_pad_link (priv->send_src[i], pad);
2496 gst_object_unref (pad);
2498 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2499 g_object_set (priv->udpqueue[i], "max-size-buffers",
2500 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2502 gst_bin_add (bin, priv->udpqueue[i]);
2503 /* link tee to udpqueue */
2504 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2505 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2506 gst_pad_link (teepad, pad);
2507 gst_object_unref (pad);
2508 gst_object_unref (teepad);
2510 /* link udpqueue to udpsink */
2511 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2512 gst_pad_link (queuepad, sinkpad);
2513 gst_object_unref (queuepad);
2514 gst_object_unref (sinkpad);
2517 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2518 g_object_set (priv->appqueue[i], "max-size-buffers",
2519 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2521 gst_bin_add (bin, priv->appqueue[i]);
2522 /* and link tee to appqueue */
2523 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2524 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2525 gst_pad_link (teepad, pad);
2526 gst_object_unref (pad);
2527 gst_object_unref (teepad);
2529 /* and link appqueue to appsink */
2530 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2531 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2532 gst_pad_link (queuepad, pad);
2533 gst_object_unref (pad);
2534 gst_object_unref (queuepad);
2535 } else if (is_tcp) {
2536 /* only appsink needed, link it to the session */
2537 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2538 gst_pad_link (priv->send_src[i], pad);
2539 gst_object_unref (pad);
2541 /* when its only TCP, we need to set sync and preroll to FALSE
2542 * for the sink to avoid deadlock. And this is only needed for
2543 * sink used for RTCP data, not the RTP data. */
2545 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2547 /* else only udpsink needed, link it to the session */
2548 gst_pad_link (priv->send_src[i], sinkpad);
2549 gst_object_unref (sinkpad);
2553 /* check if we need to set to a special state */
2554 if (state != GST_STATE_NULL) {
2555 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2556 gst_element_set_state (priv->udpsink[i], state);
2557 if (priv->appsink[i] && (priv->srcpad || i == 1))
2558 gst_element_set_state (priv->appsink[i], state);
2559 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2560 gst_element_set_state (priv->appqueue[i], state);
2561 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2562 gst_element_set_state (priv->udpqueue[i], state);
2563 if (priv->tee[i] && (priv->srcpad || i == 1))
2564 gst_element_set_state (priv->tee[i], state);
2577 /* must be called with lock */
2579 create_receiver_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2581 GstRTSPStreamPrivate *priv;
2582 GstPad *pad, *selpad;
2586 priv = stream->priv;
2588 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2590 for (i = 0; i < 2; i++) {
2591 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2592 * RTCP sink always */
2593 if (priv->sinkpad || i == 1) {
2594 /* For the receiver we create this bit of pipeline for both
2595 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2596 * and it is all funneled into the rtpbin receive pad.
2598 * .--------. .--------. .--------.
2599 * | udpsrc | | funnel | | rtpbin |
2600 * | src->sink src->sink |
2601 * '--------' | | '--------'
2605 * '--------' '--------'
2607 /* make funnel for the RTP/RTCP receivers */
2608 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2609 gst_bin_add (bin, priv->funnel[i]);
2611 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2612 gst_pad_link (pad, priv->recv_sink[i]);
2613 gst_object_unref (pad);
2615 if (priv->udpsrc_v4[i]) {
2617 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2618 * values. This is only relevant for PLAY pipelines */
2619 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2620 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2623 gst_bin_add (bin, priv->udpsrc_v4[i]);
2625 /* and link to the funnel v4 */
2626 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2627 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2628 gst_pad_link (pad, selpad);
2629 gst_object_unref (pad);
2630 gst_object_unref (selpad);
2633 if (priv->udpsrc_v6[i]) {
2635 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2636 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2638 gst_bin_add (bin, priv->udpsrc_v6[i]);
2640 /* and link to the funnel v6 */
2641 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2642 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2643 gst_pad_link (pad, selpad);
2644 gst_object_unref (pad);
2645 gst_object_unref (selpad);
2649 /* make and add appsrc */
2650 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2651 priv->appsrc_base_time[i] = -1;
2653 gst_element_set_state (priv->appsrc[i], GST_STATE_PLAYING);
2654 gst_element_set_locked_state (priv->appsrc[i], TRUE);
2656 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, "is-live",
2658 gst_bin_add (bin, priv->appsrc[i]);
2659 /* and link to the funnel */
2660 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2661 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2662 gst_pad_link (pad, selpad);
2663 gst_object_unref (pad);
2664 gst_object_unref (selpad);
2668 /* check if we need to set to a special state */
2669 if (state != GST_STATE_NULL) {
2670 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2671 gst_element_set_state (priv->funnel[i], state);
2677 * gst_rtsp_stream_join_bin:
2678 * @stream: a #GstRTSPStream
2679 * @bin: (transfer none): a #GstBin to join
2680 * @rtpbin: (transfer none): a rtpbin element in @bin
2681 * @state: the target state of the new elements
2683 * Join the #GstBin @bin that contains the element @rtpbin.
2685 * @stream will link to @rtpbin, which must be inside @bin. The elements
2686 * added to @bin will be set to the state given in @state.
2688 * Returns: %TRUE on success.
2691 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2692 GstElement * rtpbin, GstState state)
2694 GstRTSPStreamPrivate *priv;
2697 GstPadLinkReturn ret;
2699 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2700 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2701 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2703 priv = stream->priv;
2705 g_mutex_lock (&priv->lock);
2706 if (priv->is_joined)
2709 /* create a session with the same index as the stream */
2712 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2714 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2715 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2717 g_signal_connect (rtpbin, "request-rtp-encoder",
2718 (GCallback) request_rtp_encoder, stream);
2719 g_signal_connect (rtpbin, "request-rtcp-encoder",
2720 (GCallback) request_rtcp_encoder, stream);
2721 g_signal_connect (rtpbin, "request-rtp-decoder",
2722 (GCallback) request_rtp_rtcp_decoder, stream);
2723 g_signal_connect (rtpbin, "request-rtcp-decoder",
2724 (GCallback) request_rtp_rtcp_decoder, stream);
2727 if (priv->sinkpad) {
2728 g_signal_connect (rtpbin, "request-pt-map",
2729 (GCallback) request_pt_map, stream);
2732 /* get pads from the RTP session element for sending and receiving
2735 /* get a pad for sending RTP */
2736 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2737 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2740 /* link the RTP pad to the session manager, it should not really fail unless
2741 * this is not really an RTP pad */
2742 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2743 if (ret != GST_PAD_LINK_OK)
2746 name = g_strdup_printf ("send_rtp_src_%u", idx);
2747 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2750 /* Need to connect our sinkpad from here */
2751 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2753 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2755 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2756 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2760 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2761 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2763 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2764 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2767 /* get the session */
2768 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2770 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2772 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2774 g_signal_connect (priv->session, "on-ssrc-active",
2775 (GCallback) on_ssrc_active, stream);
2776 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2778 g_signal_connect (priv->session, "on-bye-timeout",
2779 (GCallback) on_bye_timeout, stream);
2780 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2783 /* signal for sender ssrc */
2784 g_signal_connect (priv->session, "on-new-sender-ssrc",
2785 (GCallback) on_new_sender_ssrc, stream);
2786 g_signal_connect (priv->session, "on-sender-ssrc-active",
2787 (GCallback) on_sender_ssrc_active, stream);
2789 if (!create_sender_part (stream, bin, state))
2790 goto no_udp_protocol;
2792 create_receiver_part (stream, bin, state);
2795 /* be notified of caps changes */
2796 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2797 (GCallback) caps_notify, stream);
2800 priv->joined_bin = bin;
2801 priv->is_joined = TRUE;
2802 g_mutex_unlock (&priv->lock);
2809 g_mutex_unlock (&priv->lock);
2814 GST_WARNING ("failed to link stream %u", idx);
2815 gst_object_unref (priv->send_rtp_sink);
2816 priv->send_rtp_sink = NULL;
2817 g_mutex_unlock (&priv->lock);
2822 GST_WARNING ("failed to allocate ports %u", idx);
2823 gst_object_unref (priv->send_rtp_sink);
2824 priv->send_rtp_sink = NULL;
2825 gst_object_unref (priv->send_src[0]);
2826 priv->send_src[0] = NULL;
2827 gst_object_unref (priv->send_src[1]);
2828 priv->send_src[1] = NULL;
2829 gst_object_unref (priv->recv_sink[0]);
2830 priv->recv_sink[0] = NULL;
2831 gst_object_unref (priv->recv_sink[1]);
2832 priv->recv_sink[1] = NULL;
2833 if (priv->udpsink[0])
2834 gst_element_set_state (priv->udpsink[0], GST_STATE_NULL);
2835 if (priv->udpsink[1])
2836 gst_element_set_state (priv->udpsink[1], GST_STATE_NULL);
2837 if (priv->udpsrc_v4[0]) {
2838 gst_element_set_state (priv->udpsrc_v4[0], GST_STATE_NULL);
2839 gst_object_unref (priv->udpsrc_v4[0]);
2840 priv->udpsrc_v4[0] = NULL;
2842 if (priv->udpsrc_v4[1]) {
2843 gst_element_set_state (priv->udpsrc_v4[1], GST_STATE_NULL);
2844 gst_object_unref (priv->udpsrc_v4[1]);
2845 priv->udpsrc_v4[1] = NULL;
2847 if (priv->udpsrc_mcast_v4[0]) {
2848 gst_element_set_state (priv->udpsrc_mcast_v4[0], GST_STATE_NULL);
2849 gst_object_unref (priv->udpsrc_mcast_v4[0]);
2850 priv->udpsrc_mcast_v4[0] = NULL;
2852 if (priv->udpsrc_mcast_v4[1]) {
2853 gst_element_set_state (priv->udpsrc_mcast_v4[1], GST_STATE_NULL);
2854 gst_object_unref (priv->udpsrc_mcast_v4[1]);
2855 priv->udpsrc_mcast_v4[1] = NULL;
2857 if (priv->udpsrc_v6[0]) {
2858 gst_element_set_state (priv->udpsrc_v6[0], GST_STATE_NULL);
2859 gst_object_unref (priv->udpsrc_v6[0]);
2860 priv->udpsrc_v6[0] = NULL;
2862 if (priv->udpsrc_v6[1]) {
2863 gst_element_set_state (priv->udpsrc_v6[1], GST_STATE_NULL);
2864 gst_object_unref (priv->udpsrc_v6[1]);
2865 priv->udpsrc_v6[1] = NULL;
2867 if (priv->udpsrc_mcast_v6[0]) {
2868 gst_element_set_state (priv->udpsrc_mcast_v6[0], GST_STATE_NULL);
2869 gst_object_unref (priv->udpsrc_mcast_v6[0]);
2870 priv->udpsrc_mcast_v6[0] = NULL;
2872 if (priv->udpsrc_mcast_v6[1]) {
2873 gst_element_set_state (priv->udpsrc_mcast_v6[1], GST_STATE_NULL);
2874 gst_object_unref (priv->udpsrc_mcast_v6[1]);
2875 priv->udpsrc_mcast_v6[1] = NULL;
2877 g_mutex_unlock (&priv->lock);
2883 * gst_rtsp_stream_leave_bin:
2884 * @stream: a #GstRTSPStream
2885 * @bin: (transfer none): a #GstBin
2886 * @rtpbin: (transfer none): a rtpbin #GstElement
2888 * Remove the elements of @stream from @bin.
2890 * Return: %TRUE on success.
2893 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2894 GstElement * rtpbin)
2896 GstRTSPStreamPrivate *priv;
2898 gboolean is_tcp, is_udp;
2900 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2901 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2902 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2904 priv = stream->priv;
2906 g_mutex_lock (&priv->lock);
2907 if (!priv->is_joined)
2908 goto was_not_joined;
2910 priv->joined_bin = NULL;
2912 /* all transports must be removed by now */
2913 if (priv->transports != NULL)
2914 goto transports_not_removed;
2916 clear_tr_cache (priv, TRUE);
2917 clear_tr_cache (priv, FALSE);
2919 GST_INFO ("stream %p leaving bin", stream);
2922 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2924 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2925 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2926 gst_object_unref (priv->send_rtp_sink);
2927 priv->send_rtp_sink = NULL;
2928 } else if (priv->recv_rtp_src) {
2929 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2930 gst_object_unref (priv->recv_rtp_src);
2931 priv->recv_rtp_src = NULL;
2934 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2936 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2937 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2940 for (i = 0; i < 2; i++) {
2941 if (priv->udpsink[i])
2942 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2943 if (priv->appsink[i])
2944 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2945 if (priv->appqueue[i])
2946 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2947 if (priv->udpqueue[i])
2948 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2950 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2951 if (priv->funnel[i])
2952 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2953 if (priv->appsrc[i])
2954 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2956 if (priv->udpsrc_v4[i]) {
2957 if (priv->sinkpad || i == 1) {
2958 /* and set udpsrc to NULL now before removing */
2959 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2960 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2961 /* removing them should also nicely release the request
2962 * pads when they finalize */
2963 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2965 /* we need to set the state to NULL before unref */
2966 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2967 gst_object_unref (priv->udpsrc_v4[i]);
2971 if (priv->udpsrc_mcast_v4[i]) {
2972 if (priv->sinkpad || i == 1) {
2973 /* and set udpsrc to NULL now before removing */
2974 gst_element_set_locked_state (priv->udpsrc_mcast_v4[i], FALSE);
2975 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2976 /* removing them should also nicely release the request
2977 * pads when they finalize */
2978 gst_bin_remove (bin, priv->udpsrc_mcast_v4[i]);
2980 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2981 gst_object_unref (priv->udpsrc_mcast_v4[i]);
2985 if (priv->udpsrc_v6[i]) {
2986 if (priv->sinkpad || i == 1) {
2987 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2988 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2989 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2991 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2992 gst_object_unref (priv->udpsrc_v6[i]);
2995 if (priv->udpsrc_mcast_v6[i]) {
2996 if (priv->sinkpad || i == 1) {
2997 gst_element_set_locked_state (priv->udpsrc_mcast_v6[i], FALSE);
2998 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2999 gst_bin_remove (bin, priv->udpsrc_mcast_v6[i]);
3001 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
3002 gst_object_unref (priv->udpsrc_mcast_v6[i]);
3006 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
3007 gst_bin_remove (bin, priv->udpsink[i]);
3008 if (priv->appsrc[i]) {
3009 if (priv->sinkpad || i == 1) {
3010 gst_element_set_locked_state (priv->appsrc[i], FALSE);
3011 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
3012 gst_bin_remove (bin, priv->appsrc[i]);
3014 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
3015 gst_object_unref (priv->appsrc[i]);
3018 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
3019 gst_bin_remove (bin, priv->appsink[i]);
3020 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3021 gst_bin_remove (bin, priv->appqueue[i]);
3022 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3023 gst_bin_remove (bin, priv->udpqueue[i]);
3024 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3025 gst_bin_remove (bin, priv->tee[i]);
3026 if (priv->funnel[i] && (priv->sinkpad || i == 1))
3027 gst_bin_remove (bin, priv->funnel[i]);
3029 if (priv->sinkpad || i == 1) {
3030 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
3031 gst_object_unref (priv->recv_sink[i]);
3032 priv->recv_sink[i] = NULL;
3035 priv->udpsrc_v4[i] = NULL;
3036 priv->udpsrc_v6[i] = NULL;
3037 priv->udpsrc_mcast_v4[i] = NULL;
3038 priv->udpsrc_mcast_v6[i] = NULL;
3039 priv->udpsink[i] = NULL;
3040 priv->appsrc[i] = NULL;
3041 priv->appsink[i] = NULL;
3042 priv->appqueue[i] = NULL;
3043 priv->udpqueue[i] = NULL;
3044 priv->tee[i] = NULL;
3045 priv->funnel[i] = NULL;
3049 gst_object_unref (priv->send_src[0]);
3050 priv->send_src[0] = NULL;
3053 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
3054 gst_object_unref (priv->send_src[1]);
3055 priv->send_src[1] = NULL;
3057 g_object_unref (priv->session);
3058 priv->session = NULL;
3060 gst_caps_unref (priv->caps);
3064 gst_object_unref (priv->srtpenc);
3066 gst_object_unref (priv->srtpdec);
3068 priv->is_joined = FALSE;
3069 g_mutex_unlock (&priv->lock);
3075 g_mutex_unlock (&priv->lock);
3078 transports_not_removed:
3080 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
3081 g_mutex_unlock (&priv->lock);
3087 * gst_rtsp_stream_get_joined_bin:
3088 * @stream: a #GstRTSPStream
3090 * Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL.
3092 * Return: (transfer full): the joined bin or NULL.
3095 gst_rtsp_stream_get_joined_bin (GstRTSPStream * stream)
3097 GstRTSPStreamPrivate *priv;
3100 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3102 priv = stream->priv;
3104 g_mutex_lock (&priv->lock);
3105 bin = priv->joined_bin ? gst_object_ref (priv->joined_bin) : NULL;
3106 g_mutex_unlock (&priv->lock);
3112 * gst_rtsp_stream_get_rtpinfo:
3113 * @stream: a #GstRTSPStream
3114 * @rtptime: (allow-none): result RTP timestamp
3115 * @seq: (allow-none): result RTP seqnum
3116 * @clock_rate: (allow-none): the clock rate
3117 * @running_time: (allow-none): result running-time
3119 * Retrieve the current rtptime, seq and running-time. This is used to
3120 * construct a RTPInfo reply header.
3122 * Returns: %TRUE when rtptime, seq and running-time could be determined.
3125 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
3126 guint * rtptime, guint * seq, guint * clock_rate,
3127 GstClockTime * running_time)
3129 GstRTSPStreamPrivate *priv;
3130 GstStructure *stats;
3131 GObjectClass *payobjclass;
3133 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3135 priv = stream->priv;
3137 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
3139 g_mutex_lock (&priv->lock);
3141 /* First try to extract the information from the last buffer on the sinks.
3142 * This will have a more accurate sequence number and timestamp, as between
3143 * the payloader and the sink there can be some queues
3145 if (priv->udpsink[0] || priv->appsink[0]) {
3146 GstSample *last_sample;
3148 if (priv->udpsink[0])
3149 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
3151 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
3156 GstSegment *segment;
3157 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
3159 caps = gst_sample_get_caps (last_sample);
3160 buffer = gst_sample_get_buffer (last_sample);
3161 segment = gst_sample_get_segment (last_sample);
3163 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
3165 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
3169 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
3172 gst_rtp_buffer_unmap (&rtp_buffer);
3176 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
3177 GST_BUFFER_TIMESTAMP (buffer));
3181 GstStructure *s = gst_caps_get_structure (caps, 0);
3183 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
3185 if (*clock_rate == 0 && running_time)
3186 *running_time = GST_CLOCK_TIME_NONE;
3188 gst_sample_unref (last_sample);
3192 gst_sample_unref (last_sample);
3197 if (g_object_class_find_property (payobjclass, "stats")) {
3198 g_object_get (priv->payloader, "stats", &stats, NULL);
3203 gst_structure_get_uint (stats, "seqnum", seq);
3206 gst_structure_get_uint (stats, "timestamp", rtptime);
3209 gst_structure_get_clock_time (stats, "running-time", running_time);
3212 gst_structure_get_uint (stats, "clock-rate", clock_rate);
3213 if (*clock_rate == 0 && running_time)
3214 *running_time = GST_CLOCK_TIME_NONE;
3216 gst_structure_free (stats);
3218 if (!g_object_class_find_property (payobjclass, "seqnum") ||
3219 !g_object_class_find_property (payobjclass, "timestamp"))
3223 g_object_get (priv->payloader, "seqnum", seq, NULL);
3226 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
3229 *running_time = GST_CLOCK_TIME_NONE;
3233 g_mutex_unlock (&priv->lock);
3240 GST_WARNING ("Could not get payloader stats");
3241 g_mutex_unlock (&priv->lock);
3247 * gst_rtsp_stream_get_caps:
3248 * @stream: a #GstRTSPStream
3250 * Retrieve the current caps of @stream.
3252 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
3256 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
3258 GstRTSPStreamPrivate *priv;
3261 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3263 priv = stream->priv;
3265 g_mutex_lock (&priv->lock);
3266 if ((result = priv->caps))
3267 gst_caps_ref (result);
3268 g_mutex_unlock (&priv->lock);
3274 * gst_rtsp_stream_recv_rtp:
3275 * @stream: a #GstRTSPStream
3276 * @buffer: (transfer full): a #GstBuffer
3278 * Handle an RTP buffer for the stream. This method is usually called when a
3279 * message has been received from a client using the TCP transport.
3281 * This function takes ownership of @buffer.
3283 * Returns: a GstFlowReturn.
3286 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
3288 GstRTSPStreamPrivate *priv;
3290 GstElement *element;
3292 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3293 priv = stream->priv;
3294 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3295 g_return_val_if_fail (priv->is_joined, FALSE);
3297 g_mutex_lock (&priv->lock);
3298 if (priv->appsrc[0])
3299 element = gst_object_ref (priv->appsrc[0]);
3302 g_mutex_unlock (&priv->lock);
3305 if (priv->appsrc_base_time[0] == -1) {
3306 /* Take current running_time. This timestamp will be put on
3307 * the first buffer of each stream because we are a live source and so we
3308 * timestamp with the running_time. When we are dealing with TCP, we also
3309 * only timestamp the first buffer (using the DISCONT flag) because a server
3310 * typically bursts data, for which we don't want to compensate by speeding
3311 * up the media. The other timestamps will be interpollated from this one
3312 * using the RTP timestamps. */
3313 GST_OBJECT_LOCK (element);
3314 if (GST_ELEMENT_CLOCK (element)) {
3316 GstClockTime base_time;
3318 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3319 base_time = GST_ELEMENT_CAST (element)->base_time;
3321 priv->appsrc_base_time[0] = now - base_time;
3322 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
3323 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3324 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3325 GST_TIME_ARGS (base_time));
3327 GST_OBJECT_UNLOCK (element);
3330 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3331 gst_object_unref (element);
3339 * gst_rtsp_stream_recv_rtcp:
3340 * @stream: a #GstRTSPStream
3341 * @buffer: (transfer full): a #GstBuffer
3343 * Handle an RTCP buffer for the stream. This method is usually called when a
3344 * message has been received from a client using the TCP transport.
3346 * This function takes ownership of @buffer.
3348 * Returns: a GstFlowReturn.
3351 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
3353 GstRTSPStreamPrivate *priv;
3355 GstElement *element;
3357 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3358 priv = stream->priv;
3359 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3361 if (!priv->is_joined) {
3362 gst_buffer_unref (buffer);
3363 return GST_FLOW_NOT_LINKED;
3365 g_mutex_lock (&priv->lock);
3366 if (priv->appsrc[1])
3367 element = gst_object_ref (priv->appsrc[1]);
3370 g_mutex_unlock (&priv->lock);
3373 if (priv->appsrc_base_time[1] == -1) {
3374 /* Take current running_time. This timestamp will be put on
3375 * the first buffer of each stream because we are a live source and so we
3376 * timestamp with the running_time. When we are dealing with TCP, we also
3377 * only timestamp the first buffer (using the DISCONT flag) because a server
3378 * typically bursts data, for which we don't want to compensate by speeding
3379 * up the media. The other timestamps will be interpollated from this one
3380 * using the RTP timestamps. */
3381 GST_OBJECT_LOCK (element);
3382 if (GST_ELEMENT_CLOCK (element)) {
3384 GstClockTime base_time;
3386 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3387 base_time = GST_ELEMENT_CAST (element)->base_time;
3389 priv->appsrc_base_time[1] = now - base_time;
3390 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3391 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3392 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3393 GST_TIME_ARGS (base_time));
3395 GST_OBJECT_UNLOCK (element);
3398 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3399 gst_object_unref (element);
3402 gst_buffer_unref (buffer);
3407 /* must be called with lock */
3409 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3412 GstRTSPStreamPrivate *priv = stream->priv;
3413 const GstRTSPTransport *tr;
3415 tr = gst_rtsp_stream_transport_get_transport (trans);
3417 switch (tr->lower_transport) {
3418 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3419 case GST_RTSP_LOWER_TRANS_UDP:
3425 dest = tr->destination;
3426 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3430 } else if (priv->client_side) {
3431 /* In client side mode the 'destination' is the RTSP server, so send
3433 min = tr->server_port.min;
3434 max = tr->server_port.max;
3436 min = tr->client_port.min;
3437 max = tr->client_port.max;
3442 GST_INFO ("setting ttl-mc %d", ttl);
3443 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3444 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3446 GST_INFO ("adding %s:%d-%d", dest, min, max);
3447 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3448 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3449 priv->transports = g_list_prepend (priv->transports, trans);
3451 GST_INFO ("removing %s:%d-%d", dest, min, max);
3452 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3453 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3454 priv->transports = g_list_remove (priv->transports, trans);
3456 priv->transports_cookie++;
3459 case GST_RTSP_LOWER_TRANS_TCP:
3461 GST_INFO ("adding TCP %s", tr->destination);
3462 priv->transports = g_list_prepend (priv->transports, trans);
3464 GST_INFO ("removing TCP %s", tr->destination);
3465 priv->transports = g_list_remove (priv->transports, trans);
3467 priv->transports_cookie++;
3470 goto unknown_transport;
3477 GST_INFO ("Unknown transport %d", tr->lower_transport);
3484 * gst_rtsp_stream_add_transport:
3485 * @stream: a #GstRTSPStream
3486 * @trans: (transfer none): a #GstRTSPStreamTransport
3488 * Add the transport in @trans to @stream. The media of @stream will
3489 * then also be send to the values configured in @trans.
3491 * @stream must be joined to a bin.
3493 * @trans must contain a valid #GstRTSPTransport.
3495 * Returns: %TRUE if @trans was added
3498 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3499 GstRTSPStreamTransport * trans)
3501 GstRTSPStreamPrivate *priv;
3504 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3505 priv = stream->priv;
3506 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3507 g_return_val_if_fail (priv->is_joined, FALSE);
3509 g_mutex_lock (&priv->lock);
3510 res = update_transport (stream, trans, TRUE);
3511 g_mutex_unlock (&priv->lock);
3517 * gst_rtsp_stream_remove_transport:
3518 * @stream: a #GstRTSPStream
3519 * @trans: (transfer none): a #GstRTSPStreamTransport
3521 * Remove the transport in @trans from @stream. The media of @stream will
3522 * not be sent to the values configured in @trans.
3524 * @stream must be joined to a bin.
3526 * @trans must contain a valid #GstRTSPTransport.
3528 * Returns: %TRUE if @trans was removed
3531 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3532 GstRTSPStreamTransport * trans)
3534 GstRTSPStreamPrivate *priv;
3537 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3538 priv = stream->priv;
3539 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3540 g_return_val_if_fail (priv->is_joined, FALSE);
3542 g_mutex_lock (&priv->lock);
3543 res = update_transport (stream, trans, FALSE);
3544 g_mutex_unlock (&priv->lock);
3550 * gst_rtsp_stream_update_crypto:
3551 * @stream: a #GstRTSPStream
3553 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3555 * Update the new crypto information for @ssrc in @stream. If information
3556 * for @ssrc did not exist, it will be added. If information
3557 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3558 * be removed from @stream.
3560 * Returns: %TRUE if @crypto could be updated
3563 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3564 guint ssrc, GstCaps * crypto)
3566 GstRTSPStreamPrivate *priv;
3568 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3569 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3571 priv = stream->priv;
3573 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3575 g_mutex_lock (&priv->lock);
3577 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3578 gst_caps_ref (crypto));
3580 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3581 g_mutex_unlock (&priv->lock);
3587 * gst_rtsp_stream_get_rtp_socket:
3588 * @stream: a #GstRTSPStream
3589 * @family: the socket family
3591 * Get the RTP socket from @stream for a @family.
3593 * @stream must be joined to a bin.
3595 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3596 * socket could be allocated for @family. Unref after usage
3599 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3601 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3605 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3606 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3607 family == G_SOCKET_FAMILY_IPV6, NULL);
3608 g_return_val_if_fail (priv->udpsink[0], NULL);
3610 if (family == G_SOCKET_FAMILY_IPV6)
3615 g_object_get (priv->udpsink[0], name, &socket, NULL);
3621 * gst_rtsp_stream_get_rtcp_socket:
3622 * @stream: a #GstRTSPStream
3623 * @family: the socket family
3625 * Get the RTCP socket from @stream for a @family.
3627 * @stream must be joined to a bin.
3629 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3630 * socket could be allocated for @family. Unref after usage
3633 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3635 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3639 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3640 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3641 family == G_SOCKET_FAMILY_IPV6, NULL);
3642 g_return_val_if_fail (priv->udpsink[1], NULL);
3644 if (family == G_SOCKET_FAMILY_IPV6)
3649 g_object_get (priv->udpsink[1], name, &socket, NULL);
3655 * gst_rtsp_stream_set_seqnum:
3656 * @stream: a #GstRTSPStream
3657 * @seqnum: a new sequence number
3659 * Configure the sequence number in the payloader of @stream to @seqnum.
3662 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3664 GstRTSPStreamPrivate *priv;
3666 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3668 priv = stream->priv;
3670 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3674 * gst_rtsp_stream_get_seqnum:
3675 * @stream: a #GstRTSPStream
3677 * Get the configured sequence number in the payloader of @stream.
3679 * Returns: the sequence number of the payloader.
3682 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3684 GstRTSPStreamPrivate *priv;
3687 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3689 priv = stream->priv;
3691 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3697 * gst_rtsp_stream_transport_filter:
3698 * @stream: a #GstRTSPStream
3699 * @func: (scope call) (allow-none): a callback
3700 * @user_data: (closure): user data passed to @func
3702 * Call @func for each transport managed by @stream. The result value of @func
3703 * determines what happens to the transport. @func will be called with @stream
3704 * locked so no further actions on @stream can be performed from @func.
3706 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3709 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3711 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3712 * will also be added with an additional ref to the result #GList of this
3715 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3717 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3718 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3719 * element in the #GList should be unreffed before the list is freed.
3722 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3723 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3725 GstRTSPStreamPrivate *priv;
3726 GList *result, *walk, *next;
3727 GHashTable *visited = NULL;
3730 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3732 priv = stream->priv;
3736 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3738 g_mutex_lock (&priv->lock);
3740 cookie = priv->transports_cookie;
3741 for (walk = priv->transports; walk; walk = next) {
3742 GstRTSPStreamTransport *trans = walk->data;
3743 GstRTSPFilterResult res;
3746 next = g_list_next (walk);
3749 /* only visit each transport once */
3750 if (g_hash_table_contains (visited, trans))
3753 g_hash_table_add (visited, g_object_ref (trans));
3754 g_mutex_unlock (&priv->lock);
3756 res = func (stream, trans, user_data);
3758 g_mutex_lock (&priv->lock);
3760 res = GST_RTSP_FILTER_REF;
3762 changed = (cookie != priv->transports_cookie);
3765 case GST_RTSP_FILTER_REMOVE:
3766 update_transport (stream, trans, FALSE);
3768 case GST_RTSP_FILTER_REF:
3769 result = g_list_prepend (result, g_object_ref (trans));
3771 case GST_RTSP_FILTER_KEEP:
3778 g_mutex_unlock (&priv->lock);
3781 g_hash_table_unref (visited);
3786 static GstPadProbeReturn
3787 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3789 GstRTSPStreamPrivate *priv;
3790 GstRTSPStream *stream;
3793 priv = stream->priv;
3795 GST_DEBUG_OBJECT (pad, "now blocking");
3797 g_mutex_lock (&priv->lock);
3798 priv->blocking = TRUE;
3799 g_mutex_unlock (&priv->lock);
3801 gst_element_post_message (priv->payloader,
3802 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3803 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3805 return GST_PAD_PROBE_OK;
3809 * gst_rtsp_stream_set_blocked:
3810 * @stream: a #GstRTSPStream
3811 * @blocked: boolean indicating we should block or unblock
3813 * Blocks or unblocks the dataflow on @stream.
3815 * Returns: %TRUE on success
3818 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3820 GstRTSPStreamPrivate *priv;
3822 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3824 priv = stream->priv;
3826 g_mutex_lock (&priv->lock);
3828 priv->blocking = FALSE;
3829 if (priv->blocked_id == 0) {
3830 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3831 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3832 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3833 g_object_ref (stream), g_object_unref);
3836 if (priv->blocked_id != 0) {
3837 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3838 priv->blocked_id = 0;
3839 priv->blocking = FALSE;
3842 g_mutex_unlock (&priv->lock);
3848 * gst_rtsp_stream_is_blocking:
3849 * @stream: a #GstRTSPStream
3851 * Check if @stream is blocking on a #GstBuffer.
3853 * Returns: %TRUE if @stream is blocking
3856 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3858 GstRTSPStreamPrivate *priv;
3861 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3863 priv = stream->priv;
3865 g_mutex_lock (&priv->lock);
3866 result = priv->blocking;
3867 g_mutex_unlock (&priv->lock);
3873 * gst_rtsp_stream_query_position:
3874 * @stream: a #GstRTSPStream
3876 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3877 * the RTP parts of the pipeline and not the RTCP parts.
3879 * Returns: %TRUE if the position could be queried
3882 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3884 GstRTSPStreamPrivate *priv;
3888 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3890 priv = stream->priv;
3892 g_mutex_lock (&priv->lock);
3893 /* depending on the transport type, it should query corresponding sink */
3894 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3895 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3896 sink = priv->udpsink[0];
3898 sink = priv->appsink[0];
3901 gst_object_ref (sink);
3902 g_mutex_unlock (&priv->lock);
3907 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3908 gst_object_unref (sink);
3914 * gst_rtsp_stream_query_stop:
3915 * @stream: a #GstRTSPStream
3917 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3918 * the RTP parts of the pipeline and not the RTCP parts.
3920 * Returns: %TRUE if the stop could be queried
3923 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3925 GstRTSPStreamPrivate *priv;
3930 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3932 priv = stream->priv;
3934 g_mutex_lock (&priv->lock);
3935 /* depending on the transport type, it should query corresponding sink */
3936 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3937 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3938 sink = priv->udpsink[0];
3940 sink = priv->appsink[0];
3943 gst_object_ref (sink);
3944 g_mutex_unlock (&priv->lock);
3949 query = gst_query_new_segment (GST_FORMAT_TIME);
3950 if ((ret = gst_element_query (sink, query))) {
3953 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3954 if (format != GST_FORMAT_TIME)
3957 gst_query_unref (query);
3958 gst_object_unref (sink);