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)
1254 if (udpsrc_out[0]) {
1255 gst_element_set_state (udpsrc_out[0], GST_STATE_NULL);
1256 g_clear_object (&udpsrc_out[0]);
1258 if (udpsrc_out[1]) {
1259 gst_element_set_state (udpsrc_out[1], GST_STATE_NULL);
1260 g_clear_object (&udpsrc_out[1]);
1267 alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
1268 GstElement * udpsrc_out[2], GstRTSPRange * server_port_out,
1269 GstRTSPTransport * ct, GstRTSPAddress ** server_addr_out,
1270 gboolean use_client_settings)
1272 GstRTSPStreamPrivate *priv = stream->priv;
1273 GSocket *rtp_socket = NULL;
1274 GSocket *rtcp_socket;
1275 gint tmp_rtp, tmp_rtcp;
1277 gint rtpport, rtcpport;
1278 GList *rejected_addresses = NULL;
1279 GstRTSPAddress *addr = NULL;
1280 GInetAddress *inetaddr = NULL;
1282 GSocketAddress *rtp_sockaddr = NULL;
1283 GSocketAddress *rtcp_sockaddr = NULL;
1284 GstRTSPAddressPool *pool;
1285 GstRTSPLowerTrans transport;
1286 const gchar *multicast_iface = priv->multicast_iface;
1290 transport = ct->lower_transport;
1292 /* Start with random port */
1295 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1296 G_SOCKET_PROTOCOL_UDP, NULL);
1298 goto no_udp_protocol;
1299 g_socket_set_multicast_loopback (rtcp_socket, FALSE);
1301 if (*server_addr_out)
1302 gst_rtsp_address_free (*server_addr_out);
1304 /* try to allocate 2 UDP ports, the RTP port should be an even
1305 * number and the RTCP port should be the next (uneven) port */
1308 if (rtp_socket == NULL) {
1309 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1310 G_SOCKET_PROTOCOL_UDP, NULL);
1312 goto no_udp_protocol;
1313 g_socket_set_multicast_loopback (rtp_socket, FALSE);
1316 if (pool && ((transport == GST_RTSP_LOWER_TRANS_UDP &&
1317 gst_rtsp_address_pool_has_unicast_addresses (pool))
1318 || transport == GST_RTSP_LOWER_TRANS_UDP_MCAST)) {
1319 GstRTSPAddressFlags flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
1321 if (transport == GST_RTSP_LOWER_TRANS_UDP)
1322 flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
1324 flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
1327 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1329 if (family == G_SOCKET_FAMILY_IPV6)
1330 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1332 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1334 if (ct->destination && transport == GST_RTSP_LOWER_TRANS_UDP_MCAST
1335 && use_client_settings)
1336 gst_rtsp_address_pool_reserve_address (pool, ct->destination,
1337 ct->port.min, 2, ct->ttl, &addr);
1339 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1344 tmp_rtp = addr->port;
1346 g_clear_object (&inetaddr);
1347 inetaddr = g_inet_address_new_from_string (addr->address);
1349 /* If we're supposed to bind to a multicast address, instead bind
1350 * to ANY and let udpsrc later join the relevant multicast group
1352 if (g_inet_address_get_is_multicast (inetaddr)) {
1353 g_object_unref (inetaddr);
1354 inetaddr = g_inet_address_new_any (family);
1363 if (inetaddr == NULL)
1364 inetaddr = g_inet_address_new_any (family);
1367 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1368 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1369 g_object_unref (rtp_sockaddr);
1372 g_object_unref (rtp_sockaddr);
1374 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1375 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1376 g_clear_object (&rtp_sockaddr);
1381 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1382 g_object_unref (rtp_sockaddr);
1384 /* check if port is even */
1385 if ((tmp_rtp & 1) != 0) {
1386 /* port not even, close and allocate another */
1388 g_clear_object (&rtp_socket);
1393 tmp_rtcp = tmp_rtp + 1;
1395 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1396 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1397 g_object_unref (rtcp_sockaddr);
1398 g_clear_object (&rtp_socket);
1401 g_object_unref (rtcp_sockaddr);
1404 addr_str = g_inet_address_to_string (inetaddr);
1406 addr_str = addr->address;
1407 g_clear_object (&inetaddr);
1409 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1410 rtcp_socket, family, addr_str, tmp_rtp, tmp_rtcp, multicast_iface,
1414 goto no_udp_protocol;
1420 play_udpsources_one_family (stream, udpsrc_out, family);
1422 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1423 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1425 /* this should not happen... */
1426 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1429 /* set RTP and RTCP sockets */
1430 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1432 server_port_out->min = rtpport;
1433 server_port_out->max = rtcpport;
1435 *server_addr_out = addr;
1436 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1438 g_object_unref (rtp_socket);
1439 g_object_unref (rtcp_socket);
1463 g_object_unref (inetaddr);
1464 g_list_free_full (rejected_addresses,
1465 (GDestroyNotify) gst_rtsp_address_free);
1467 gst_rtsp_address_free (addr);
1469 g_object_unref (rtp_socket);
1471 g_object_unref (rtcp_socket);
1477 * gst_rtsp_stream_allocate_udp_sockets:
1478 * @stream: a #GstRTSPStream
1479 * @family: protocol family
1480 * @transport_method: transport method
1482 * Allocates RTP and RTCP ports.
1484 * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
1487 gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
1488 GSocketFamily family, GstRTSPTransport * ct, gboolean use_client_settings)
1490 GstRTSPStreamPrivate *priv;
1491 gboolean result = FALSE;
1492 GstRTSPLowerTrans transport = ct->lower_transport;
1494 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1495 priv = stream->priv;
1496 g_return_val_if_fail (priv->is_joined, FALSE);
1498 g_mutex_lock (&priv->lock);
1500 if (family == G_SOCKET_FAMILY_IPV4) {
1501 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1502 if (priv->have_ipv4_mcast)
1504 priv->have_ipv4_mcast =
1505 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
1506 priv->udpsrc_mcast_v4, &priv->server_port_v4, ct, &priv->addr_v4,
1507 use_client_settings);
1510 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1511 &priv->server_port_v4, ct, &priv->server_addr_v4,
1512 use_client_settings);
1515 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1516 if (priv->have_ipv6_mcast)
1518 priv->have_ipv6_mcast =
1519 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
1520 priv->udpsrc_mcast_v6, &priv->server_port_v6, ct, &priv->addr_v6,
1521 use_client_settings);
1523 if (priv->have_ipv6)
1526 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1527 &priv->server_port_v6, ct, &priv->server_addr_v6,
1528 use_client_settings);
1533 result = priv->have_ipv4 || priv->have_ipv4_mcast || priv->have_ipv6 ||
1534 priv->have_ipv6_mcast;
1536 g_mutex_unlock (&priv->lock);
1542 * gst_rtsp_stream_set_client_side:
1543 * @stream: a #GstRTSPStream
1544 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1545 * an RTSP connection.
1547 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1548 * streams to an RTSP server via RECORD. This has the practical effect
1549 * of changing which UDP port numbers are used when setting up the local
1550 * side of the stream sending to be either the 'server' or 'client' pair
1551 * of a configured UDP transport.
1554 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1556 GstRTSPStreamPrivate *priv;
1558 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1559 priv = stream->priv;
1560 g_mutex_lock (&priv->lock);
1561 priv->client_side = client_side;
1562 g_mutex_unlock (&priv->lock);
1566 * gst_rtsp_stream_is_client_side:
1567 * @stream: a #GstRTSPStream
1569 * See gst_rtsp_stream_set_client_side()
1571 * Returns: TRUE if this #GstRTSPStream is client-side.
1574 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1576 GstRTSPStreamPrivate *priv;
1579 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1581 priv = stream->priv;
1582 g_mutex_lock (&priv->lock);
1583 ret = priv->client_side;
1584 g_mutex_unlock (&priv->lock);
1590 * gst_rtsp_stream_get_server_port:
1591 * @stream: a #GstRTSPStream
1592 * @server_port: (out): result server port
1593 * @family: the port family to get
1595 * Fill @server_port with the port pair used by the server. This function can
1596 * only be called when @stream has been joined.
1599 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1600 GstRTSPRange * server_port, GSocketFamily family)
1602 GstRTSPStreamPrivate *priv;
1604 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1605 priv = stream->priv;
1606 g_return_if_fail (priv->is_joined);
1608 g_mutex_lock (&priv->lock);
1609 if (family == G_SOCKET_FAMILY_IPV4) {
1611 *server_port = priv->server_port_v4;
1614 *server_port = priv->server_port_v6;
1616 g_mutex_unlock (&priv->lock);
1620 * gst_rtsp_stream_get_rtpsession:
1621 * @stream: a #GstRTSPStream
1623 * Get the RTP session of this stream.
1625 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1628 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1630 GstRTSPStreamPrivate *priv;
1633 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1635 priv = stream->priv;
1637 g_mutex_lock (&priv->lock);
1638 if ((session = priv->session))
1639 g_object_ref (session);
1640 g_mutex_unlock (&priv->lock);
1646 * gst_rtsp_stream_get_encoder:
1647 * @stream: a #GstRTSPStream
1649 * Get the SRTP encoder for this stream.
1651 * Returns: (transfer full): The SRTP encoder for this stream. Unref after usage.
1654 gst_rtsp_stream_get_srtp_encoder (GstRTSPStream * stream)
1656 GstRTSPStreamPrivate *priv;
1657 GstElement *encoder;
1659 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1661 priv = stream->priv;
1663 g_mutex_lock (&priv->lock);
1664 if ((encoder = priv->srtpenc))
1665 g_object_ref (encoder);
1666 g_mutex_unlock (&priv->lock);
1672 * gst_rtsp_stream_get_ssrc:
1673 * @stream: a #GstRTSPStream
1674 * @ssrc: (out): result ssrc
1676 * Get the SSRC used by the RTP session of this stream. This function can only
1677 * be called when @stream has been joined.
1680 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1682 GstRTSPStreamPrivate *priv;
1684 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1685 priv = stream->priv;
1686 g_return_if_fail (priv->is_joined);
1688 g_mutex_lock (&priv->lock);
1689 if (ssrc && priv->session)
1690 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1691 g_mutex_unlock (&priv->lock);
1695 * gst_rtsp_stream_set_retransmission_time:
1696 * @stream: a #GstRTSPStream
1697 * @time: a #GstClockTime
1699 * Set the amount of time to store retransmission packets.
1702 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1705 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1707 g_mutex_lock (&stream->priv->lock);
1708 stream->priv->rtx_time = time;
1709 if (stream->priv->rtxsend)
1710 g_object_set (stream->priv->rtxsend, "max-size-time",
1711 GST_TIME_AS_MSECONDS (time), NULL);
1712 g_mutex_unlock (&stream->priv->lock);
1716 * gst_rtsp_stream_get_retransmission_time:
1717 * @stream: a #GstRTSPStream
1719 * Get the amount of time to store retransmission data.
1721 * Returns: the amount of time to store retransmission data.
1724 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1728 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1730 g_mutex_lock (&stream->priv->lock);
1731 ret = stream->priv->rtx_time;
1732 g_mutex_unlock (&stream->priv->lock);
1738 * gst_rtsp_stream_set_retransmission_pt:
1739 * @stream: a #GstRTSPStream
1742 * Set the payload type (pt) for retransmission of this stream.
1745 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1747 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1749 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1751 g_mutex_lock (&stream->priv->lock);
1752 stream->priv->rtx_pt = rtx_pt;
1753 if (stream->priv->rtxsend) {
1754 guint pt = gst_rtsp_stream_get_pt (stream);
1755 gchar *pt_s = g_strdup_printf ("%d", pt);
1756 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1757 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1758 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1760 gst_structure_free (rtx_pt_map);
1762 g_mutex_unlock (&stream->priv->lock);
1766 * gst_rtsp_stream_get_retransmission_pt:
1767 * @stream: a #GstRTSPStream
1769 * Get the payload-type used for retransmission of this stream
1771 * Returns: The retransmission PT.
1774 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1778 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1780 g_mutex_lock (&stream->priv->lock);
1781 rtx_pt = stream->priv->rtx_pt;
1782 g_mutex_unlock (&stream->priv->lock);
1788 * gst_rtsp_stream_set_buffer_size:
1789 * @stream: a #GstRTSPStream
1790 * @size: the buffer size
1792 * Set the size of the UDP transmission buffer (in bytes)
1793 * Needs to be set before the stream is joined to a bin.
1798 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1800 g_mutex_lock (&stream->priv->lock);
1801 stream->priv->buffer_size = size;
1802 g_mutex_unlock (&stream->priv->lock);
1806 * gst_rtsp_stream_get_buffer_size:
1807 * @stream: a #GstRTSPStream
1809 * Get the size of the UDP transmission buffer (in bytes)
1811 * Returns: the size of the UDP TX buffer
1816 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1820 g_mutex_lock (&stream->priv->lock);
1821 buffer_size = stream->priv->buffer_size;
1822 g_mutex_unlock (&stream->priv->lock);
1827 /* executed from streaming thread */
1829 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1831 GstRTSPStreamPrivate *priv = stream->priv;
1832 GstCaps *newcaps, *oldcaps;
1834 newcaps = gst_pad_get_current_caps (pad);
1836 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1839 g_mutex_lock (&priv->lock);
1840 oldcaps = priv->caps;
1841 priv->caps = newcaps;
1842 g_mutex_unlock (&priv->lock);
1845 gst_caps_unref (oldcaps);
1849 dump_structure (const GstStructure * s)
1853 sstr = gst_structure_to_string (s);
1854 GST_INFO ("structure: %s", sstr);
1858 static GstRTSPStreamTransport *
1859 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1861 GstRTSPStreamPrivate *priv = stream->priv;
1863 GstRTSPStreamTransport *result = NULL;
1868 if (rtcp_from == NULL)
1871 tmp = g_strrstr (rtcp_from, ":");
1875 port = atoi (tmp + 1);
1876 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1878 g_mutex_lock (&priv->lock);
1879 GST_INFO ("finding %s:%d in %d transports", dest, port,
1880 g_list_length (priv->transports));
1882 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1883 GstRTSPStreamTransport *trans = walk->data;
1884 const GstRTSPTransport *tr;
1887 tr = gst_rtsp_stream_transport_get_transport (trans);
1889 if (priv->client_side) {
1890 /* In client side mode the 'destination' is the RTSP server, so send
1892 min = tr->server_port.min;
1893 max = tr->server_port.max;
1895 min = tr->client_port.min;
1896 max = tr->client_port.max;
1899 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1905 g_object_ref (result);
1906 g_mutex_unlock (&priv->lock);
1913 static GstRTSPStreamTransport *
1914 check_transport (GObject * source, GstRTSPStream * stream)
1916 GstStructure *stats;
1917 GstRTSPStreamTransport *trans;
1919 /* see if we have a stream to match with the origin of the RTCP packet */
1920 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1921 if (trans == NULL) {
1922 g_object_get (source, "stats", &stats, NULL);
1924 const gchar *rtcp_from;
1926 dump_structure (stats);
1928 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1929 if ((trans = find_transport (stream, rtcp_from))) {
1930 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1932 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1935 gst_structure_free (stats);
1943 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1945 GstRTSPStreamTransport *trans;
1947 GST_INFO ("%p: new source %p", stream, source);
1949 trans = check_transport (source, stream);
1952 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1956 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1958 GST_INFO ("%p: new SDES %p", stream, source);
1962 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1964 GstRTSPStreamTransport *trans;
1966 trans = check_transport (source, stream);
1969 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1970 gst_rtsp_stream_transport_keep_alive (trans);
1974 GstStructure *stats;
1975 g_object_get (source, "stats", &stats, NULL);
1977 dump_structure (stats);
1978 gst_structure_free (stats);
1985 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1987 GST_INFO ("%p: source %p bye", stream, source);
1991 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1993 GstRTSPStreamTransport *trans;
1995 GST_INFO ("%p: source %p bye timeout", stream, source);
1997 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1998 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1999 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
2004 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2006 GstRTSPStreamTransport *trans;
2008 GST_INFO ("%p: source %p timeout", stream, source);
2010 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
2011 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
2012 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
2017 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2019 GST_INFO ("%p: new sender source %p", stream, source);
2022 GstStructure *stats;
2023 g_object_get (source, "stats", &stats, NULL);
2025 dump_structure (stats);
2026 gst_structure_free (stats);
2033 on_sender_ssrc_active (GObject * session, GObject * source,
2034 GstRTSPStream * stream)
2038 GstStructure *stats;
2039 g_object_get (source, "stats", &stats, NULL);
2041 dump_structure (stats);
2042 gst_structure_free (stats);
2049 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
2052 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
2053 g_list_free (priv->tr_cache_rtp);
2054 priv->tr_cache_rtp = NULL;
2056 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
2057 g_list_free (priv->tr_cache_rtcp);
2058 priv->tr_cache_rtcp = NULL;
2062 static GstFlowReturn
2063 handle_new_sample (GstAppSink * sink, gpointer user_data)
2065 GstRTSPStreamPrivate *priv;
2069 GstRTSPStream *stream;
2072 sample = gst_app_sink_pull_sample (sink);
2076 stream = (GstRTSPStream *) user_data;
2077 priv = stream->priv;
2078 buffer = gst_sample_get_buffer (sample);
2080 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
2082 g_mutex_lock (&priv->lock);
2084 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
2085 clear_tr_cache (priv, is_rtp);
2086 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2087 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2088 priv->tr_cache_rtp =
2089 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
2091 priv->tr_cache_cookie_rtp = priv->transports_cookie;
2094 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
2095 clear_tr_cache (priv, is_rtp);
2096 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
2097 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2098 priv->tr_cache_rtcp =
2099 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
2101 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
2104 g_mutex_unlock (&priv->lock);
2107 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
2108 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2109 gst_rtsp_stream_transport_send_rtp (tr, buffer);
2112 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
2113 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2114 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
2117 gst_sample_unref (sample);
2122 static GstAppSinkCallbacks sink_cb = {
2123 NULL, /* not interested in EOS */
2124 NULL, /* not interested in preroll samples */
2129 get_rtp_encoder (GstRTSPStream * stream, guint session)
2131 GstRTSPStreamPrivate *priv = stream->priv;
2133 if (priv->srtpenc == NULL) {
2136 name = g_strdup_printf ("srtpenc_%u", session);
2137 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
2140 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
2142 return gst_object_ref (priv->srtpenc);
2146 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2148 GstRTSPStreamPrivate *priv = stream->priv;
2149 GstElement *oldenc, *enc;
2153 if (priv->idx != session)
2156 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
2158 oldenc = priv->srtpenc;
2159 enc = get_rtp_encoder (stream, session);
2160 name = g_strdup_printf ("rtp_sink_%d", session);
2161 pad = gst_element_get_request_pad (enc, name);
2163 gst_object_unref (pad);
2166 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
2173 request_rtcp_encoder (GstElement * rtpbin, guint session,
2174 GstRTSPStream * stream)
2176 GstRTSPStreamPrivate *priv = stream->priv;
2177 GstElement *oldenc, *enc;
2181 if (priv->idx != session)
2184 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
2186 oldenc = priv->srtpenc;
2187 enc = get_rtp_encoder (stream, session);
2188 name = g_strdup_printf ("rtcp_sink_%d", session);
2189 pad = gst_element_get_request_pad (enc, name);
2191 gst_object_unref (pad);
2194 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
2201 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2203 GstRTSPStreamPrivate *priv = stream->priv;
2206 GST_DEBUG ("request key %08x", ssrc);
2208 g_mutex_lock (&priv->lock);
2209 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
2210 gst_caps_ref (caps);
2211 g_mutex_unlock (&priv->lock);
2217 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
2218 GstRTSPStream * stream)
2220 GstRTSPStreamPrivate *priv = stream->priv;
2222 if (priv->idx != session)
2225 if (priv->srtpdec == NULL) {
2228 name = g_strdup_printf ("srtpdec_%u", session);
2229 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2232 g_signal_connect (priv->srtpdec, "request-key",
2233 (GCallback) request_key, stream);
2235 return gst_object_ref (priv->srtpdec);
2239 * gst_rtsp_stream_request_aux_sender:
2240 * @stream: a #GstRTSPStream
2241 * @sessid: the session id
2243 * Creating a rtxsend bin
2245 * Returns: (transfer full): a #GstElement.
2250 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2254 GstStructure *pt_map;
2259 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2261 pt = gst_rtsp_stream_get_pt (stream);
2262 pt_s = g_strdup_printf ("%u", pt);
2263 rtx_pt = stream->priv->rtx_pt;
2265 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2267 bin = gst_bin_new (NULL);
2268 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2269 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2270 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2271 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2272 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2274 gst_structure_free (pt_map);
2275 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2277 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2278 name = g_strdup_printf ("src_%u", sessid);
2279 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2281 gst_object_unref (pad);
2283 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2284 name = g_strdup_printf ("sink_%u", sessid);
2285 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2287 gst_object_unref (pad);
2293 * gst_rtsp_stream_set_pt_map:
2294 * @stream: a #GstRTSPStream
2298 * Configure a pt map between @pt and @caps.
2301 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2303 GstRTSPStreamPrivate *priv = stream->priv;
2305 g_mutex_lock (&priv->lock);
2306 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2307 g_mutex_unlock (&priv->lock);
2311 * gst_rtsp_stream_set_publish_clock_mode:
2312 * @stream: a #GstRTSPStream
2313 * @mode: the clock publish mode
2315 * Sets if and how the stream clock should be published according to RFC7273.
2320 gst_rtsp_stream_set_publish_clock_mode (GstRTSPStream * stream,
2321 GstRTSPPublishClockMode mode)
2323 GstRTSPStreamPrivate *priv;
2325 priv = stream->priv;
2326 g_mutex_lock (&priv->lock);
2327 priv->publish_clock_mode = mode;
2328 g_mutex_unlock (&priv->lock);
2332 * gst_rtsp_stream_get_publish_clock_mode:
2333 * @factory: a #GstRTSPStream
2335 * Gets if and how the stream clock should be published according to RFC7273.
2337 * Returns: The GstRTSPPublishClockMode
2341 GstRTSPPublishClockMode
2342 gst_rtsp_stream_get_publish_clock_mode (GstRTSPStream * stream)
2344 GstRTSPStreamPrivate *priv;
2345 GstRTSPPublishClockMode ret;
2347 priv = stream->priv;
2348 g_mutex_lock (&priv->lock);
2349 ret = priv->publish_clock_mode;
2350 g_mutex_unlock (&priv->lock);
2356 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2357 GstRTSPStream * stream)
2359 GstRTSPStreamPrivate *priv = stream->priv;
2360 GstCaps *caps = NULL;
2362 g_mutex_lock (&priv->lock);
2364 if (priv->idx == session) {
2365 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2367 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2368 gst_caps_ref (caps);
2370 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2374 g_mutex_unlock (&priv->lock);
2380 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2382 GstRTSPStreamPrivate *priv = stream->priv;
2384 GstPadLinkReturn ret;
2387 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2388 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2390 name = gst_pad_get_name (pad);
2391 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2397 if (priv->idx != sessid)
2400 if (gst_pad_is_linked (priv->sinkpad)) {
2401 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2402 GST_DEBUG_PAD_NAME (priv->sinkpad));
2406 /* link the RTP pad to the session manager, it should not really fail unless
2407 * this is not really an RTP pad */
2408 ret = gst_pad_link (pad, priv->sinkpad);
2409 if (ret != GST_PAD_LINK_OK)
2411 priv->recv_rtp_src = gst_object_ref (pad);
2418 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2419 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2424 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2425 GstRTSPStream * stream)
2427 /* TODO: What to do here other than this? */
2428 GST_DEBUG ("Stream %p: Got EOS", stream);
2429 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2432 /* must be called with lock */
2434 create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2436 GstRTSPStreamPrivate *priv;
2437 GstPad *pad, *sinkpad = NULL;
2438 gboolean is_tcp = FALSE, is_udp = FALSE;
2441 priv = stream->priv;
2443 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2444 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2445 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2447 if (is_udp && !create_and_configure_udpsinks (stream))
2448 goto no_udp_protocol;
2450 for (i = 0; i < 2; i++) {
2451 GstPad *teepad, *queuepad;
2452 /* For the sender we create this bit of pipeline for both
2453 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2454 * we need to add a queue before appsink and udpsink to make
2455 * the pipeline not block. For the TCP case, we want to pump
2456 * client as fast as possible anyway. This pipeline is used
2457 * when both TCP and UDP are present.
2459 * .--------. .-----. .---------. .---------.
2460 * | rtpbin | | tee | | queue | | udpsink |
2461 * | send->sink src->sink src->sink |
2462 * '--------' | | '---------' '---------'
2463 * | | .---------. .---------.
2464 * | | | queue | | appsink |
2465 * | src->sink src->sink |
2466 * '-----' '---------' '---------'
2468 * When only UDP or only TCP is allowed, we skip the tee and queue
2469 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2472 /* Only link the RTP send src if we're going to send RTP, link
2473 * the RTCP send src always */
2474 if (priv->srcpad || i == 1) {
2477 gst_bin_add (bin, priv->udpsink[i]);
2478 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2483 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2484 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2485 gst_bin_add (bin, priv->appsink[i]);
2486 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2487 &sink_cb, stream, NULL);
2490 if (is_udp && is_tcp) {
2491 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2493 /* make tee for RTP/RTCP */
2494 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2495 gst_bin_add (bin, priv->tee[i]);
2497 /* and link to rtpbin send pad */
2498 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2499 gst_pad_link (priv->send_src[i], pad);
2500 gst_object_unref (pad);
2502 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2503 g_object_set (priv->udpqueue[i], "max-size-buffers",
2504 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2506 gst_bin_add (bin, priv->udpqueue[i]);
2507 /* link tee to udpqueue */
2508 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2509 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2510 gst_pad_link (teepad, pad);
2511 gst_object_unref (pad);
2512 gst_object_unref (teepad);
2514 /* link udpqueue to udpsink */
2515 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2516 gst_pad_link (queuepad, sinkpad);
2517 gst_object_unref (queuepad);
2518 gst_object_unref (sinkpad);
2521 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2522 g_object_set (priv->appqueue[i], "max-size-buffers",
2523 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2525 gst_bin_add (bin, priv->appqueue[i]);
2526 /* and link tee to appqueue */
2527 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2528 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2529 gst_pad_link (teepad, pad);
2530 gst_object_unref (pad);
2531 gst_object_unref (teepad);
2533 /* and link appqueue to appsink */
2534 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2535 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2536 gst_pad_link (queuepad, pad);
2537 gst_object_unref (pad);
2538 gst_object_unref (queuepad);
2539 } else if (is_tcp) {
2540 /* only appsink needed, link it to the session */
2541 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2542 gst_pad_link (priv->send_src[i], pad);
2543 gst_object_unref (pad);
2545 /* when its only TCP, we need to set sync and preroll to FALSE
2546 * for the sink to avoid deadlock. And this is only needed for
2547 * sink used for RTCP data, not the RTP data. */
2549 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2551 /* else only udpsink needed, link it to the session */
2552 gst_pad_link (priv->send_src[i], sinkpad);
2553 gst_object_unref (sinkpad);
2557 /* check if we need to set to a special state */
2558 if (state != GST_STATE_NULL) {
2559 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2560 gst_element_set_state (priv->udpsink[i], state);
2561 if (priv->appsink[i] && (priv->srcpad || i == 1))
2562 gst_element_set_state (priv->appsink[i], state);
2563 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2564 gst_element_set_state (priv->appqueue[i], state);
2565 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2566 gst_element_set_state (priv->udpqueue[i], state);
2567 if (priv->tee[i] && (priv->srcpad || i == 1))
2568 gst_element_set_state (priv->tee[i], state);
2581 /* must be called with lock */
2583 create_receiver_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2585 GstRTSPStreamPrivate *priv;
2586 GstPad *pad, *selpad;
2590 priv = stream->priv;
2592 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2594 for (i = 0; i < 2; i++) {
2595 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2596 * RTCP sink always */
2597 if (priv->sinkpad || i == 1) {
2598 /* For the receiver we create this bit of pipeline for both
2599 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2600 * and it is all funneled into the rtpbin receive pad.
2602 * .--------. .--------. .--------.
2603 * | udpsrc | | funnel | | rtpbin |
2604 * | src->sink src->sink |
2605 * '--------' | | '--------'
2609 * '--------' '--------'
2611 /* make funnel for the RTP/RTCP receivers */
2612 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2613 gst_bin_add (bin, priv->funnel[i]);
2615 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2616 gst_pad_link (pad, priv->recv_sink[i]);
2617 gst_object_unref (pad);
2619 if (priv->udpsrc_v4[i]) {
2621 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2622 * values. This is only relevant for PLAY pipelines */
2623 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2624 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2627 gst_bin_add (bin, priv->udpsrc_v4[i]);
2629 /* and link to the funnel v4 */
2630 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2631 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2632 gst_pad_link (pad, selpad);
2633 gst_object_unref (pad);
2634 gst_object_unref (selpad);
2637 if (priv->udpsrc_v6[i]) {
2639 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2640 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2642 gst_bin_add (bin, priv->udpsrc_v6[i]);
2644 /* and link to the funnel v6 */
2645 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2646 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2647 gst_pad_link (pad, selpad);
2648 gst_object_unref (pad);
2649 gst_object_unref (selpad);
2653 /* make and add appsrc */
2654 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2655 priv->appsrc_base_time[i] = -1;
2657 gst_element_set_state (priv->appsrc[i], GST_STATE_PLAYING);
2658 gst_element_set_locked_state (priv->appsrc[i], TRUE);
2660 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, "is-live",
2662 gst_bin_add (bin, priv->appsrc[i]);
2663 /* and link to the funnel */
2664 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2665 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2666 gst_pad_link (pad, selpad);
2667 gst_object_unref (pad);
2668 gst_object_unref (selpad);
2672 /* check if we need to set to a special state */
2673 if (state != GST_STATE_NULL) {
2674 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2675 gst_element_set_state (priv->funnel[i], state);
2681 * gst_rtsp_stream_join_bin:
2682 * @stream: a #GstRTSPStream
2683 * @bin: (transfer none): a #GstBin to join
2684 * @rtpbin: (transfer none): a rtpbin element in @bin
2685 * @state: the target state of the new elements
2687 * Join the #GstBin @bin that contains the element @rtpbin.
2689 * @stream will link to @rtpbin, which must be inside @bin. The elements
2690 * added to @bin will be set to the state given in @state.
2692 * Returns: %TRUE on success.
2695 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2696 GstElement * rtpbin, GstState state)
2698 GstRTSPStreamPrivate *priv;
2701 GstPadLinkReturn ret;
2703 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2704 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2705 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2707 priv = stream->priv;
2709 g_mutex_lock (&priv->lock);
2710 if (priv->is_joined)
2713 /* create a session with the same index as the stream */
2716 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2718 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2719 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2721 g_signal_connect (rtpbin, "request-rtp-encoder",
2722 (GCallback) request_rtp_encoder, stream);
2723 g_signal_connect (rtpbin, "request-rtcp-encoder",
2724 (GCallback) request_rtcp_encoder, stream);
2725 g_signal_connect (rtpbin, "request-rtp-decoder",
2726 (GCallback) request_rtp_rtcp_decoder, stream);
2727 g_signal_connect (rtpbin, "request-rtcp-decoder",
2728 (GCallback) request_rtp_rtcp_decoder, stream);
2731 if (priv->sinkpad) {
2732 g_signal_connect (rtpbin, "request-pt-map",
2733 (GCallback) request_pt_map, stream);
2736 /* get pads from the RTP session element for sending and receiving
2739 /* get a pad for sending RTP */
2740 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2741 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2744 /* link the RTP pad to the session manager, it should not really fail unless
2745 * this is not really an RTP pad */
2746 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2747 if (ret != GST_PAD_LINK_OK)
2750 name = g_strdup_printf ("send_rtp_src_%u", idx);
2751 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2754 /* Need to connect our sinkpad from here */
2755 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2757 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2759 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2760 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2764 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2765 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2767 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2768 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2771 /* get the session */
2772 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2774 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2776 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2778 g_signal_connect (priv->session, "on-ssrc-active",
2779 (GCallback) on_ssrc_active, stream);
2780 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2782 g_signal_connect (priv->session, "on-bye-timeout",
2783 (GCallback) on_bye_timeout, stream);
2784 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2787 /* signal for sender ssrc */
2788 g_signal_connect (priv->session, "on-new-sender-ssrc",
2789 (GCallback) on_new_sender_ssrc, stream);
2790 g_signal_connect (priv->session, "on-sender-ssrc-active",
2791 (GCallback) on_sender_ssrc_active, stream);
2793 if (!create_sender_part (stream, bin, state))
2794 goto no_udp_protocol;
2796 create_receiver_part (stream, bin, state);
2799 /* be notified of caps changes */
2800 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2801 (GCallback) caps_notify, stream);
2804 priv->joined_bin = bin;
2805 priv->is_joined = TRUE;
2806 g_mutex_unlock (&priv->lock);
2813 g_mutex_unlock (&priv->lock);
2818 GST_WARNING ("failed to link stream %u", idx);
2819 gst_object_unref (priv->send_rtp_sink);
2820 priv->send_rtp_sink = NULL;
2821 g_mutex_unlock (&priv->lock);
2826 GST_WARNING ("failed to allocate ports %u", idx);
2827 gst_object_unref (priv->send_rtp_sink);
2828 priv->send_rtp_sink = NULL;
2829 gst_object_unref (priv->send_src[0]);
2830 priv->send_src[0] = NULL;
2831 gst_object_unref (priv->send_src[1]);
2832 priv->send_src[1] = NULL;
2833 gst_object_unref (priv->recv_sink[0]);
2834 priv->recv_sink[0] = NULL;
2835 gst_object_unref (priv->recv_sink[1]);
2836 priv->recv_sink[1] = NULL;
2837 if (priv->udpsink[0])
2838 gst_element_set_state (priv->udpsink[0], GST_STATE_NULL);
2839 if (priv->udpsink[1])
2840 gst_element_set_state (priv->udpsink[1], GST_STATE_NULL);
2841 if (priv->udpsrc_v4[0]) {
2842 gst_element_set_state (priv->udpsrc_v4[0], GST_STATE_NULL);
2843 gst_object_unref (priv->udpsrc_v4[0]);
2844 priv->udpsrc_v4[0] = NULL;
2846 if (priv->udpsrc_v4[1]) {
2847 gst_element_set_state (priv->udpsrc_v4[1], GST_STATE_NULL);
2848 gst_object_unref (priv->udpsrc_v4[1]);
2849 priv->udpsrc_v4[1] = NULL;
2851 if (priv->udpsrc_mcast_v4[0]) {
2852 gst_element_set_state (priv->udpsrc_mcast_v4[0], GST_STATE_NULL);
2853 gst_object_unref (priv->udpsrc_mcast_v4[0]);
2854 priv->udpsrc_mcast_v4[0] = NULL;
2856 if (priv->udpsrc_mcast_v4[1]) {
2857 gst_element_set_state (priv->udpsrc_mcast_v4[1], GST_STATE_NULL);
2858 gst_object_unref (priv->udpsrc_mcast_v4[1]);
2859 priv->udpsrc_mcast_v4[1] = NULL;
2861 if (priv->udpsrc_v6[0]) {
2862 gst_element_set_state (priv->udpsrc_v6[0], GST_STATE_NULL);
2863 gst_object_unref (priv->udpsrc_v6[0]);
2864 priv->udpsrc_v6[0] = NULL;
2866 if (priv->udpsrc_v6[1]) {
2867 gst_element_set_state (priv->udpsrc_v6[1], GST_STATE_NULL);
2868 gst_object_unref (priv->udpsrc_v6[1]);
2869 priv->udpsrc_v6[1] = NULL;
2871 if (priv->udpsrc_mcast_v6[0]) {
2872 gst_element_set_state (priv->udpsrc_mcast_v6[0], GST_STATE_NULL);
2873 gst_object_unref (priv->udpsrc_mcast_v6[0]);
2874 priv->udpsrc_mcast_v6[0] = NULL;
2876 if (priv->udpsrc_mcast_v6[1]) {
2877 gst_element_set_state (priv->udpsrc_mcast_v6[1], GST_STATE_NULL);
2878 gst_object_unref (priv->udpsrc_mcast_v6[1]);
2879 priv->udpsrc_mcast_v6[1] = NULL;
2881 g_mutex_unlock (&priv->lock);
2887 * gst_rtsp_stream_leave_bin:
2888 * @stream: a #GstRTSPStream
2889 * @bin: (transfer none): a #GstBin
2890 * @rtpbin: (transfer none): a rtpbin #GstElement
2892 * Remove the elements of @stream from @bin.
2894 * Return: %TRUE on success.
2897 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2898 GstElement * rtpbin)
2900 GstRTSPStreamPrivate *priv;
2902 gboolean is_tcp, is_udp;
2904 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2905 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2906 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2908 priv = stream->priv;
2910 g_mutex_lock (&priv->lock);
2911 if (!priv->is_joined)
2912 goto was_not_joined;
2914 priv->joined_bin = NULL;
2916 /* all transports must be removed by now */
2917 if (priv->transports != NULL)
2918 goto transports_not_removed;
2920 clear_tr_cache (priv, TRUE);
2921 clear_tr_cache (priv, FALSE);
2923 GST_INFO ("stream %p leaving bin", stream);
2926 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2928 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2929 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2930 gst_object_unref (priv->send_rtp_sink);
2931 priv->send_rtp_sink = NULL;
2932 } else if (priv->recv_rtp_src) {
2933 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2934 gst_object_unref (priv->recv_rtp_src);
2935 priv->recv_rtp_src = NULL;
2938 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2940 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2941 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2944 for (i = 0; i < 2; i++) {
2945 if (priv->udpsink[i])
2946 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2947 if (priv->appsink[i])
2948 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2949 if (priv->appqueue[i])
2950 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2951 if (priv->udpqueue[i])
2952 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2954 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2955 if (priv->funnel[i])
2956 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2957 if (priv->appsrc[i])
2958 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2960 if (priv->udpsrc_v4[i]) {
2961 if (priv->sinkpad || i == 1) {
2962 /* and set udpsrc to NULL now before removing */
2963 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2964 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2965 /* removing them should also nicely release the request
2966 * pads when they finalize */
2967 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2969 /* we need to set the state to NULL before unref */
2970 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2971 gst_object_unref (priv->udpsrc_v4[i]);
2975 if (priv->udpsrc_mcast_v4[i]) {
2976 if (priv->sinkpad || i == 1) {
2977 /* and set udpsrc to NULL now before removing */
2978 gst_element_set_locked_state (priv->udpsrc_mcast_v4[i], FALSE);
2979 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2980 /* removing them should also nicely release the request
2981 * pads when they finalize */
2982 gst_bin_remove (bin, priv->udpsrc_mcast_v4[i]);
2984 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2985 gst_object_unref (priv->udpsrc_mcast_v4[i]);
2989 if (priv->udpsrc_v6[i]) {
2990 if (priv->sinkpad || i == 1) {
2991 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2992 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2993 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2995 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2996 gst_object_unref (priv->udpsrc_v6[i]);
2999 if (priv->udpsrc_mcast_v6[i]) {
3000 if (priv->sinkpad || i == 1) {
3001 gst_element_set_locked_state (priv->udpsrc_mcast_v6[i], FALSE);
3002 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
3003 gst_bin_remove (bin, priv->udpsrc_mcast_v6[i]);
3005 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
3006 gst_object_unref (priv->udpsrc_mcast_v6[i]);
3010 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
3011 gst_bin_remove (bin, priv->udpsink[i]);
3012 if (priv->appsrc[i]) {
3013 if (priv->sinkpad || i == 1) {
3014 gst_element_set_locked_state (priv->appsrc[i], FALSE);
3015 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
3016 gst_bin_remove (bin, priv->appsrc[i]);
3018 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
3019 gst_object_unref (priv->appsrc[i]);
3022 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
3023 gst_bin_remove (bin, priv->appsink[i]);
3024 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3025 gst_bin_remove (bin, priv->appqueue[i]);
3026 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3027 gst_bin_remove (bin, priv->udpqueue[i]);
3028 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
3029 gst_bin_remove (bin, priv->tee[i]);
3030 if (priv->funnel[i] && (priv->sinkpad || i == 1))
3031 gst_bin_remove (bin, priv->funnel[i]);
3033 if (priv->sinkpad || i == 1) {
3034 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
3035 gst_object_unref (priv->recv_sink[i]);
3036 priv->recv_sink[i] = NULL;
3039 priv->udpsrc_v4[i] = NULL;
3040 priv->udpsrc_v6[i] = NULL;
3041 priv->udpsrc_mcast_v4[i] = NULL;
3042 priv->udpsrc_mcast_v6[i] = NULL;
3043 priv->udpsink[i] = NULL;
3044 priv->appsrc[i] = NULL;
3045 priv->appsink[i] = NULL;
3046 priv->appqueue[i] = NULL;
3047 priv->udpqueue[i] = NULL;
3048 priv->tee[i] = NULL;
3049 priv->funnel[i] = NULL;
3053 gst_object_unref (priv->send_src[0]);
3054 priv->send_src[0] = NULL;
3057 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
3058 gst_object_unref (priv->send_src[1]);
3059 priv->send_src[1] = NULL;
3061 g_object_unref (priv->session);
3062 priv->session = NULL;
3064 gst_caps_unref (priv->caps);
3068 gst_object_unref (priv->srtpenc);
3070 gst_object_unref (priv->srtpdec);
3072 priv->is_joined = FALSE;
3073 g_mutex_unlock (&priv->lock);
3079 g_mutex_unlock (&priv->lock);
3082 transports_not_removed:
3084 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
3085 g_mutex_unlock (&priv->lock);
3091 * gst_rtsp_stream_get_joined_bin:
3092 * @stream: a #GstRTSPStream
3094 * Get the previous joined bin with gst_rtsp_stream_join_bin() or NULL.
3096 * Return: (transfer full): the joined bin or NULL.
3099 gst_rtsp_stream_get_joined_bin (GstRTSPStream * stream)
3101 GstRTSPStreamPrivate *priv;
3104 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3106 priv = stream->priv;
3108 g_mutex_lock (&priv->lock);
3109 bin = priv->joined_bin ? gst_object_ref (priv->joined_bin) : NULL;
3110 g_mutex_unlock (&priv->lock);
3116 * gst_rtsp_stream_get_rtpinfo:
3117 * @stream: a #GstRTSPStream
3118 * @rtptime: (allow-none): result RTP timestamp
3119 * @seq: (allow-none): result RTP seqnum
3120 * @clock_rate: (allow-none): the clock rate
3121 * @running_time: (allow-none): result running-time
3123 * Retrieve the current rtptime, seq and running-time. This is used to
3124 * construct a RTPInfo reply header.
3126 * Returns: %TRUE when rtptime, seq and running-time could be determined.
3129 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
3130 guint * rtptime, guint * seq, guint * clock_rate,
3131 GstClockTime * running_time)
3133 GstRTSPStreamPrivate *priv;
3134 GstStructure *stats;
3135 GObjectClass *payobjclass;
3137 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3139 priv = stream->priv;
3141 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
3143 g_mutex_lock (&priv->lock);
3145 /* First try to extract the information from the last buffer on the sinks.
3146 * This will have a more accurate sequence number and timestamp, as between
3147 * the payloader and the sink there can be some queues
3149 if (priv->udpsink[0] || priv->appsink[0]) {
3150 GstSample *last_sample;
3152 if (priv->udpsink[0])
3153 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
3155 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
3160 GstSegment *segment;
3161 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
3163 caps = gst_sample_get_caps (last_sample);
3164 buffer = gst_sample_get_buffer (last_sample);
3165 segment = gst_sample_get_segment (last_sample);
3167 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
3169 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
3173 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
3176 gst_rtp_buffer_unmap (&rtp_buffer);
3180 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
3181 GST_BUFFER_TIMESTAMP (buffer));
3185 GstStructure *s = gst_caps_get_structure (caps, 0);
3187 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
3189 if (*clock_rate == 0 && running_time)
3190 *running_time = GST_CLOCK_TIME_NONE;
3192 gst_sample_unref (last_sample);
3196 gst_sample_unref (last_sample);
3201 if (g_object_class_find_property (payobjclass, "stats")) {
3202 g_object_get (priv->payloader, "stats", &stats, NULL);
3207 gst_structure_get_uint (stats, "seqnum", seq);
3210 gst_structure_get_uint (stats, "timestamp", rtptime);
3213 gst_structure_get_clock_time (stats, "running-time", running_time);
3216 gst_structure_get_uint (stats, "clock-rate", clock_rate);
3217 if (*clock_rate == 0 && running_time)
3218 *running_time = GST_CLOCK_TIME_NONE;
3220 gst_structure_free (stats);
3222 if (!g_object_class_find_property (payobjclass, "seqnum") ||
3223 !g_object_class_find_property (payobjclass, "timestamp"))
3227 g_object_get (priv->payloader, "seqnum", seq, NULL);
3230 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
3233 *running_time = GST_CLOCK_TIME_NONE;
3237 g_mutex_unlock (&priv->lock);
3244 GST_WARNING ("Could not get payloader stats");
3245 g_mutex_unlock (&priv->lock);
3251 * gst_rtsp_stream_get_caps:
3252 * @stream: a #GstRTSPStream
3254 * Retrieve the current caps of @stream.
3256 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
3260 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
3262 GstRTSPStreamPrivate *priv;
3265 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3267 priv = stream->priv;
3269 g_mutex_lock (&priv->lock);
3270 if ((result = priv->caps))
3271 gst_caps_ref (result);
3272 g_mutex_unlock (&priv->lock);
3278 * gst_rtsp_stream_recv_rtp:
3279 * @stream: a #GstRTSPStream
3280 * @buffer: (transfer full): a #GstBuffer
3282 * Handle an RTP buffer for the stream. This method is usually called when a
3283 * message has been received from a client using the TCP transport.
3285 * This function takes ownership of @buffer.
3287 * Returns: a GstFlowReturn.
3290 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
3292 GstRTSPStreamPrivate *priv;
3294 GstElement *element;
3296 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3297 priv = stream->priv;
3298 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3299 g_return_val_if_fail (priv->is_joined, FALSE);
3301 g_mutex_lock (&priv->lock);
3302 if (priv->appsrc[0])
3303 element = gst_object_ref (priv->appsrc[0]);
3306 g_mutex_unlock (&priv->lock);
3309 if (priv->appsrc_base_time[0] == -1) {
3310 /* Take current running_time. This timestamp will be put on
3311 * the first buffer of each stream because we are a live source and so we
3312 * timestamp with the running_time. When we are dealing with TCP, we also
3313 * only timestamp the first buffer (using the DISCONT flag) because a server
3314 * typically bursts data, for which we don't want to compensate by speeding
3315 * up the media. The other timestamps will be interpollated from this one
3316 * using the RTP timestamps. */
3317 GST_OBJECT_LOCK (element);
3318 if (GST_ELEMENT_CLOCK (element)) {
3320 GstClockTime base_time;
3322 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3323 base_time = GST_ELEMENT_CAST (element)->base_time;
3325 priv->appsrc_base_time[0] = now - base_time;
3326 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
3327 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3328 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3329 GST_TIME_ARGS (base_time));
3331 GST_OBJECT_UNLOCK (element);
3334 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3335 gst_object_unref (element);
3343 * gst_rtsp_stream_recv_rtcp:
3344 * @stream: a #GstRTSPStream
3345 * @buffer: (transfer full): a #GstBuffer
3347 * Handle an RTCP buffer for the stream. This method is usually called when a
3348 * message has been received from a client using the TCP transport.
3350 * This function takes ownership of @buffer.
3352 * Returns: a GstFlowReturn.
3355 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
3357 GstRTSPStreamPrivate *priv;
3359 GstElement *element;
3361 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3362 priv = stream->priv;
3363 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3365 if (!priv->is_joined) {
3366 gst_buffer_unref (buffer);
3367 return GST_FLOW_NOT_LINKED;
3369 g_mutex_lock (&priv->lock);
3370 if (priv->appsrc[1])
3371 element = gst_object_ref (priv->appsrc[1]);
3374 g_mutex_unlock (&priv->lock);
3377 if (priv->appsrc_base_time[1] == -1) {
3378 /* Take current running_time. This timestamp will be put on
3379 * the first buffer of each stream because we are a live source and so we
3380 * timestamp with the running_time. When we are dealing with TCP, we also
3381 * only timestamp the first buffer (using the DISCONT flag) because a server
3382 * typically bursts data, for which we don't want to compensate by speeding
3383 * up the media. The other timestamps will be interpollated from this one
3384 * using the RTP timestamps. */
3385 GST_OBJECT_LOCK (element);
3386 if (GST_ELEMENT_CLOCK (element)) {
3388 GstClockTime base_time;
3390 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3391 base_time = GST_ELEMENT_CAST (element)->base_time;
3393 priv->appsrc_base_time[1] = now - base_time;
3394 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3395 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3396 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3397 GST_TIME_ARGS (base_time));
3399 GST_OBJECT_UNLOCK (element);
3402 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3403 gst_object_unref (element);
3406 gst_buffer_unref (buffer);
3411 /* must be called with lock */
3413 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3416 GstRTSPStreamPrivate *priv = stream->priv;
3417 const GstRTSPTransport *tr;
3419 tr = gst_rtsp_stream_transport_get_transport (trans);
3421 switch (tr->lower_transport) {
3422 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3423 case GST_RTSP_LOWER_TRANS_UDP:
3429 dest = tr->destination;
3430 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3434 } else if (priv->client_side) {
3435 /* In client side mode the 'destination' is the RTSP server, so send
3437 min = tr->server_port.min;
3438 max = tr->server_port.max;
3440 min = tr->client_port.min;
3441 max = tr->client_port.max;
3446 GST_INFO ("setting ttl-mc %d", ttl);
3447 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3448 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3450 GST_INFO ("adding %s:%d-%d", dest, min, max);
3451 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3452 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3453 priv->transports = g_list_prepend (priv->transports, trans);
3455 GST_INFO ("removing %s:%d-%d", dest, min, max);
3456 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3457 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3458 priv->transports = g_list_remove (priv->transports, trans);
3460 priv->transports_cookie++;
3463 case GST_RTSP_LOWER_TRANS_TCP:
3465 GST_INFO ("adding TCP %s", tr->destination);
3466 priv->transports = g_list_prepend (priv->transports, trans);
3468 GST_INFO ("removing TCP %s", tr->destination);
3469 priv->transports = g_list_remove (priv->transports, trans);
3471 priv->transports_cookie++;
3474 goto unknown_transport;
3481 GST_INFO ("Unknown transport %d", tr->lower_transport);
3488 * gst_rtsp_stream_add_transport:
3489 * @stream: a #GstRTSPStream
3490 * @trans: (transfer none): a #GstRTSPStreamTransport
3492 * Add the transport in @trans to @stream. The media of @stream will
3493 * then also be send to the values configured in @trans.
3495 * @stream must be joined to a bin.
3497 * @trans must contain a valid #GstRTSPTransport.
3499 * Returns: %TRUE if @trans was added
3502 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3503 GstRTSPStreamTransport * trans)
3505 GstRTSPStreamPrivate *priv;
3508 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3509 priv = stream->priv;
3510 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3511 g_return_val_if_fail (priv->is_joined, FALSE);
3513 g_mutex_lock (&priv->lock);
3514 res = update_transport (stream, trans, TRUE);
3515 g_mutex_unlock (&priv->lock);
3521 * gst_rtsp_stream_remove_transport:
3522 * @stream: a #GstRTSPStream
3523 * @trans: (transfer none): a #GstRTSPStreamTransport
3525 * Remove the transport in @trans from @stream. The media of @stream will
3526 * not be sent to the values configured in @trans.
3528 * @stream must be joined to a bin.
3530 * @trans must contain a valid #GstRTSPTransport.
3532 * Returns: %TRUE if @trans was removed
3535 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3536 GstRTSPStreamTransport * trans)
3538 GstRTSPStreamPrivate *priv;
3541 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3542 priv = stream->priv;
3543 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3544 g_return_val_if_fail (priv->is_joined, FALSE);
3546 g_mutex_lock (&priv->lock);
3547 res = update_transport (stream, trans, FALSE);
3548 g_mutex_unlock (&priv->lock);
3554 * gst_rtsp_stream_update_crypto:
3555 * @stream: a #GstRTSPStream
3557 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3559 * Update the new crypto information for @ssrc in @stream. If information
3560 * for @ssrc did not exist, it will be added. If information
3561 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3562 * be removed from @stream.
3564 * Returns: %TRUE if @crypto could be updated
3567 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3568 guint ssrc, GstCaps * crypto)
3570 GstRTSPStreamPrivate *priv;
3572 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3573 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3575 priv = stream->priv;
3577 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3579 g_mutex_lock (&priv->lock);
3581 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3582 gst_caps_ref (crypto));
3584 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3585 g_mutex_unlock (&priv->lock);
3591 * gst_rtsp_stream_get_rtp_socket:
3592 * @stream: a #GstRTSPStream
3593 * @family: the socket family
3595 * Get the RTP socket from @stream for a @family.
3597 * @stream must be joined to a bin.
3599 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3600 * socket could be allocated for @family. Unref after usage
3603 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3605 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3609 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3610 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3611 family == G_SOCKET_FAMILY_IPV6, NULL);
3612 g_return_val_if_fail (priv->udpsink[0], NULL);
3614 if (family == G_SOCKET_FAMILY_IPV6)
3619 g_object_get (priv->udpsink[0], name, &socket, NULL);
3625 * gst_rtsp_stream_get_rtcp_socket:
3626 * @stream: a #GstRTSPStream
3627 * @family: the socket family
3629 * Get the RTCP socket from @stream for a @family.
3631 * @stream must be joined to a bin.
3633 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3634 * socket could be allocated for @family. Unref after usage
3637 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3639 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3643 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3644 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3645 family == G_SOCKET_FAMILY_IPV6, NULL);
3646 g_return_val_if_fail (priv->udpsink[1], NULL);
3648 if (family == G_SOCKET_FAMILY_IPV6)
3653 g_object_get (priv->udpsink[1], name, &socket, NULL);
3659 * gst_rtsp_stream_set_seqnum:
3660 * @stream: a #GstRTSPStream
3661 * @seqnum: a new sequence number
3663 * Configure the sequence number in the payloader of @stream to @seqnum.
3666 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3668 GstRTSPStreamPrivate *priv;
3670 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3672 priv = stream->priv;
3674 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3678 * gst_rtsp_stream_get_seqnum:
3679 * @stream: a #GstRTSPStream
3681 * Get the configured sequence number in the payloader of @stream.
3683 * Returns: the sequence number of the payloader.
3686 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3688 GstRTSPStreamPrivate *priv;
3691 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3693 priv = stream->priv;
3695 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3701 * gst_rtsp_stream_transport_filter:
3702 * @stream: a #GstRTSPStream
3703 * @func: (scope call) (allow-none): a callback
3704 * @user_data: (closure): user data passed to @func
3706 * Call @func for each transport managed by @stream. The result value of @func
3707 * determines what happens to the transport. @func will be called with @stream
3708 * locked so no further actions on @stream can be performed from @func.
3710 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3713 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3715 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3716 * will also be added with an additional ref to the result #GList of this
3719 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3721 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3722 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3723 * element in the #GList should be unreffed before the list is freed.
3726 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3727 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3729 GstRTSPStreamPrivate *priv;
3730 GList *result, *walk, *next;
3731 GHashTable *visited = NULL;
3734 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3736 priv = stream->priv;
3740 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3742 g_mutex_lock (&priv->lock);
3744 cookie = priv->transports_cookie;
3745 for (walk = priv->transports; walk; walk = next) {
3746 GstRTSPStreamTransport *trans = walk->data;
3747 GstRTSPFilterResult res;
3750 next = g_list_next (walk);
3753 /* only visit each transport once */
3754 if (g_hash_table_contains (visited, trans))
3757 g_hash_table_add (visited, g_object_ref (trans));
3758 g_mutex_unlock (&priv->lock);
3760 res = func (stream, trans, user_data);
3762 g_mutex_lock (&priv->lock);
3764 res = GST_RTSP_FILTER_REF;
3766 changed = (cookie != priv->transports_cookie);
3769 case GST_RTSP_FILTER_REMOVE:
3770 update_transport (stream, trans, FALSE);
3772 case GST_RTSP_FILTER_REF:
3773 result = g_list_prepend (result, g_object_ref (trans));
3775 case GST_RTSP_FILTER_KEEP:
3782 g_mutex_unlock (&priv->lock);
3785 g_hash_table_unref (visited);
3790 static GstPadProbeReturn
3791 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3793 GstRTSPStreamPrivate *priv;
3794 GstRTSPStream *stream;
3797 priv = stream->priv;
3799 GST_DEBUG_OBJECT (pad, "now blocking");
3801 g_mutex_lock (&priv->lock);
3802 priv->blocking = TRUE;
3803 g_mutex_unlock (&priv->lock);
3805 gst_element_post_message (priv->payloader,
3806 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3807 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3809 return GST_PAD_PROBE_OK;
3813 * gst_rtsp_stream_set_blocked:
3814 * @stream: a #GstRTSPStream
3815 * @blocked: boolean indicating we should block or unblock
3817 * Blocks or unblocks the dataflow on @stream.
3819 * Returns: %TRUE on success
3822 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3824 GstRTSPStreamPrivate *priv;
3826 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3828 priv = stream->priv;
3830 g_mutex_lock (&priv->lock);
3832 priv->blocking = FALSE;
3833 if (priv->blocked_id == 0) {
3834 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3835 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3836 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3837 g_object_ref (stream), g_object_unref);
3840 if (priv->blocked_id != 0) {
3841 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3842 priv->blocked_id = 0;
3843 priv->blocking = FALSE;
3846 g_mutex_unlock (&priv->lock);
3852 * gst_rtsp_stream_is_blocking:
3853 * @stream: a #GstRTSPStream
3855 * Check if @stream is blocking on a #GstBuffer.
3857 * Returns: %TRUE if @stream is blocking
3860 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3862 GstRTSPStreamPrivate *priv;
3865 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3867 priv = stream->priv;
3869 g_mutex_lock (&priv->lock);
3870 result = priv->blocking;
3871 g_mutex_unlock (&priv->lock);
3877 * gst_rtsp_stream_query_position:
3878 * @stream: a #GstRTSPStream
3880 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3881 * the RTP parts of the pipeline and not the RTCP parts.
3883 * Returns: %TRUE if the position could be queried
3886 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3888 GstRTSPStreamPrivate *priv;
3892 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3894 priv = stream->priv;
3896 g_mutex_lock (&priv->lock);
3897 /* depending on the transport type, it should query corresponding sink */
3898 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3899 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3900 sink = priv->udpsink[0];
3902 sink = priv->appsink[0];
3905 gst_object_ref (sink);
3906 g_mutex_unlock (&priv->lock);
3911 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3912 gst_object_unref (sink);
3918 * gst_rtsp_stream_query_stop:
3919 * @stream: a #GstRTSPStream
3921 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3922 * the RTP parts of the pipeline and not the RTCP parts.
3924 * Returns: %TRUE if the stop could be queried
3927 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3929 GstRTSPStreamPrivate *priv;
3934 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3936 priv = stream->priv;
3938 g_mutex_lock (&priv->lock);
3939 /* depending on the transport type, it should query corresponding sink */
3940 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3941 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3942 sink = priv->udpsink[0];
3944 sink = priv->appsink[0];
3947 gst_object_ref (sink);
3948 g_mutex_unlock (&priv->lock);
3953 query = gst_query_new_segment (GST_FORMAT_TIME);
3954 if ((ret = gst_element_query (sink, query))) {
3957 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3958 if (format != GST_FORMAT_TIME)
3961 gst_query_unref (query);
3962 gst_object_unref (sink);