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;
75 /* TRUE if this stream is running on
76 * the client side of an RTSP link (for RECORD) */
80 GstRTSPProfile profiles;
81 GstRTSPLowerTrans protocols;
83 /* pads on the rtpbin */
84 GstPad *send_rtp_sink;
89 /* the RTPSession object */
92 /* SRTP encoder/decoder */
97 /* sinks used for sending and receiving RTP and RTCP over ipv4, they share
99 GstElement *udpsrc_v4[2];
100 /* UDP sources for UDP multicast transports */
101 GstElement *udpsrc_mcast_v4[2];
103 /* sinks used for sending and receiving RTP and RTCP over ipv6, they share
105 GstElement *udpsrc_v6[2];
106 /* UDP sources for UDP multicast transports */
107 GstElement *udpsrc_mcast_v6[2];
109 GstElement *udpqueue[2];
110 GstElement *udpsink[2];
112 /* for TCP transport */
113 GstElement *appsrc[2];
114 GstClockTime appsrc_base_time[2];
115 GstElement *appqueue[2];
116 GstElement *appsink[2];
119 GstElement *funnel[2];
124 GstClockTime rtx_time;
126 /* server ports for sending/receiving over ipv4 */
127 GstRTSPRange server_port_v4;
128 GstRTSPAddress *server_addr_v4;
131 /* server ports for sending/receiving over ipv6 */
132 GstRTSPRange server_port_v6;
133 GstRTSPAddress *server_addr_v6;
136 /* multicast addresses */
137 GstRTSPAddressPool *pool;
138 GstRTSPAddress *addr_v4;
139 GstRTSPAddress *addr_v6;
140 gboolean have_ipv4_mcast;
141 gboolean have_ipv6_mcast;
143 /* the caps of the stream */
147 /* transports we stream to */
150 guint transports_cookie;
152 GList *tr_cache_rtcp;
153 guint tr_cache_cookie_rtp;
154 guint tr_cache_cookie_rtcp;
159 /* stream blocking */
163 /* pt->caps map for RECORD streams */
167 #define DEFAULT_CONTROL NULL
168 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
169 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
170 GST_RTSP_LOWER_TRANS_TCP
183 SIGNAL_NEW_RTP_ENCODER,
184 SIGNAL_NEW_RTCP_ENCODER,
188 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
189 #define GST_CAT_DEFAULT rtsp_stream_debug
191 static GQuark ssrc_stream_map_key;
193 static void gst_rtsp_stream_get_property (GObject * object, guint propid,
194 GValue * value, GParamSpec * pspec);
195 static void gst_rtsp_stream_set_property (GObject * object, guint propid,
196 const GValue * value, GParamSpec * pspec);
198 static void gst_rtsp_stream_finalize (GObject * obj);
200 static guint gst_rtsp_stream_signals[SIGNAL_LAST] = { 0 };
202 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
205 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
207 GObjectClass *gobject_class;
209 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
211 gobject_class = G_OBJECT_CLASS (klass);
213 gobject_class->get_property = gst_rtsp_stream_get_property;
214 gobject_class->set_property = gst_rtsp_stream_set_property;
215 gobject_class->finalize = gst_rtsp_stream_finalize;
217 g_object_class_install_property (gobject_class, PROP_CONTROL,
218 g_param_spec_string ("control", "Control",
219 "The control string for this stream", DEFAULT_CONTROL,
220 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
222 g_object_class_install_property (gobject_class, PROP_PROFILES,
223 g_param_spec_flags ("profiles", "Profiles",
224 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
225 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
227 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
228 g_param_spec_flags ("protocols", "Protocols",
229 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
230 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
232 gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER] =
233 g_signal_new ("new-rtp-encoder", G_TYPE_FROM_CLASS (klass),
234 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
235 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
237 gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER] =
238 g_signal_new ("new-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
239 G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
240 G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
242 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
244 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
248 gst_rtsp_stream_init (GstRTSPStream * stream)
250 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
252 GST_DEBUG ("new stream %p", stream);
257 priv->control = g_strdup (DEFAULT_CONTROL);
258 priv->profiles = DEFAULT_PROFILES;
259 priv->protocols = DEFAULT_PROTOCOLS;
261 g_mutex_init (&priv->lock);
263 priv->keys = g_hash_table_new_full (g_direct_hash, g_direct_equal,
264 NULL, (GDestroyNotify) gst_caps_unref);
265 priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
266 (GDestroyNotify) gst_caps_unref);
270 gst_rtsp_stream_finalize (GObject * obj)
272 GstRTSPStream *stream;
273 GstRTSPStreamPrivate *priv;
275 stream = GST_RTSP_STREAM (obj);
278 GST_DEBUG ("finalize stream %p", stream);
280 /* we really need to be unjoined now */
281 g_return_if_fail (!priv->is_joined);
284 gst_rtsp_address_free (priv->addr_v4);
286 gst_rtsp_address_free (priv->addr_v6);
287 if (priv->server_addr_v4)
288 gst_rtsp_address_free (priv->server_addr_v4);
289 if (priv->server_addr_v6)
290 gst_rtsp_address_free (priv->server_addr_v6);
292 g_object_unref (priv->pool);
294 g_object_unref (priv->rtxsend);
296 gst_object_unref (priv->payloader);
298 gst_object_unref (priv->srcpad);
300 gst_object_unref (priv->sinkpad);
301 g_free (priv->control);
302 g_mutex_clear (&priv->lock);
304 g_hash_table_unref (priv->keys);
305 g_hash_table_destroy (priv->ptmap);
307 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
311 gst_rtsp_stream_get_property (GObject * object, guint propid,
312 GValue * value, GParamSpec * pspec)
314 GstRTSPStream *stream = GST_RTSP_STREAM (object);
318 g_value_take_string (value, gst_rtsp_stream_get_control (stream));
321 g_value_set_flags (value, gst_rtsp_stream_get_profiles (stream));
324 g_value_set_flags (value, gst_rtsp_stream_get_protocols (stream));
327 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
332 gst_rtsp_stream_set_property (GObject * object, guint propid,
333 const GValue * value, GParamSpec * pspec)
335 GstRTSPStream *stream = GST_RTSP_STREAM (object);
339 gst_rtsp_stream_set_control (stream, g_value_get_string (value));
342 gst_rtsp_stream_set_profiles (stream, g_value_get_flags (value));
345 gst_rtsp_stream_set_protocols (stream, g_value_get_flags (value));
348 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
353 * gst_rtsp_stream_new:
356 * @payloader: a #GstElement
358 * Create a new media stream with index @idx that handles RTP data on
359 * @pad and has a payloader element @payloader if @pad is a source pad
360 * or a depayloader element @payloader if @pad is a sink pad.
362 * Returns: (transfer full): a new #GstRTSPStream
365 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * pad)
367 GstRTSPStreamPrivate *priv;
368 GstRTSPStream *stream;
370 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
371 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
373 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
376 priv->payloader = gst_object_ref (payloader);
377 if (GST_PAD_IS_SRC (pad))
378 priv->srcpad = gst_object_ref (pad);
380 priv->sinkpad = gst_object_ref (pad);
386 * gst_rtsp_stream_get_index:
387 * @stream: a #GstRTSPStream
389 * Get the stream index.
391 * Return: the stream index.
394 gst_rtsp_stream_get_index (GstRTSPStream * stream)
396 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
398 return stream->priv->idx;
402 * gst_rtsp_stream_get_pt:
403 * @stream: a #GstRTSPStream
405 * Get the stream payload type.
407 * Return: the stream payload type.
410 gst_rtsp_stream_get_pt (GstRTSPStream * stream)
412 GstRTSPStreamPrivate *priv;
415 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
419 g_object_get (G_OBJECT (priv->payloader), "pt", &pt, NULL);
425 * gst_rtsp_stream_get_srcpad:
426 * @stream: a #GstRTSPStream
428 * Get the srcpad associated with @stream.
430 * Returns: (transfer full): the srcpad. Unref after usage.
433 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
435 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
437 if (!stream->priv->srcpad)
440 return gst_object_ref (stream->priv->srcpad);
444 * gst_rtsp_stream_get_sinkpad:
445 * @stream: a #GstRTSPStream
447 * Get the sinkpad associated with @stream.
449 * Returns: (transfer full): the sinkpad. Unref after usage.
452 gst_rtsp_stream_get_sinkpad (GstRTSPStream * stream)
454 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
456 if (!stream->priv->sinkpad)
459 return gst_object_ref (stream->priv->sinkpad);
463 * gst_rtsp_stream_get_control:
464 * @stream: a #GstRTSPStream
466 * Get the control string to identify this stream.
468 * Returns: (transfer full): the control string. g_free() after usage.
471 gst_rtsp_stream_get_control (GstRTSPStream * stream)
473 GstRTSPStreamPrivate *priv;
476 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
480 g_mutex_lock (&priv->lock);
481 if ((result = g_strdup (priv->control)) == NULL)
482 result = g_strdup_printf ("stream=%u", priv->idx);
483 g_mutex_unlock (&priv->lock);
489 * gst_rtsp_stream_set_control:
490 * @stream: a #GstRTSPStream
491 * @control: a control string
493 * Set the control string in @stream.
496 gst_rtsp_stream_set_control (GstRTSPStream * stream, const gchar * control)
498 GstRTSPStreamPrivate *priv;
500 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
504 g_mutex_lock (&priv->lock);
505 g_free (priv->control);
506 priv->control = g_strdup (control);
507 g_mutex_unlock (&priv->lock);
511 * gst_rtsp_stream_has_control:
512 * @stream: a #GstRTSPStream
513 * @control: a control string
515 * Check if @stream has the control string @control.
517 * Returns: %TRUE is @stream has @control as the control string
520 gst_rtsp_stream_has_control (GstRTSPStream * stream, const gchar * control)
522 GstRTSPStreamPrivate *priv;
525 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
529 g_mutex_lock (&priv->lock);
531 res = (g_strcmp0 (priv->control, control) == 0);
535 if (sscanf (control, "stream=%u", &streamid) > 0)
536 res = (streamid == priv->idx);
540 g_mutex_unlock (&priv->lock);
546 * gst_rtsp_stream_set_mtu:
547 * @stream: a #GstRTSPStream
550 * Configure the mtu in the payloader of @stream to @mtu.
553 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
555 GstRTSPStreamPrivate *priv;
557 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
561 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
563 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
567 * gst_rtsp_stream_get_mtu:
568 * @stream: a #GstRTSPStream
570 * Get the configured MTU in the payloader of @stream.
572 * Returns: the MTU of the payloader.
575 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
577 GstRTSPStreamPrivate *priv;
580 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
584 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
589 /* Update the dscp qos property on the udp sinks */
591 update_dscp_qos (GstRTSPStream * stream)
593 GstRTSPStreamPrivate *priv;
595 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
599 if (priv->udpsink[0]) {
600 g_object_set (G_OBJECT (priv->udpsink[0]), "qos-dscp", priv->dscp_qos,
604 if (priv->udpsink[1]) {
605 g_object_set (G_OBJECT (priv->udpsink[1]), "qos-dscp", priv->dscp_qos,
611 * gst_rtsp_stream_set_dscp_qos:
612 * @stream: a #GstRTSPStream
613 * @dscp_qos: a new dscp qos value (0-63, or -1 to disable)
615 * Configure the dscp qos of the outgoing sockets to @dscp_qos.
618 gst_rtsp_stream_set_dscp_qos (GstRTSPStream * stream, gint dscp_qos)
620 GstRTSPStreamPrivate *priv;
622 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
626 GST_LOG_OBJECT (stream, "set DSCP QoS %d", dscp_qos);
628 if (dscp_qos < -1 || dscp_qos > 63) {
629 GST_WARNING_OBJECT (stream, "trying to set illegal dscp qos %d", dscp_qos);
633 priv->dscp_qos = dscp_qos;
635 update_dscp_qos (stream);
639 * gst_rtsp_stream_get_dscp_qos:
640 * @stream: a #GstRTSPStream
642 * Get the configured DSCP QoS in of the outgoing sockets.
644 * Returns: the DSCP QoS value of the outgoing sockets, or -1 if disbled.
647 gst_rtsp_stream_get_dscp_qos (GstRTSPStream * stream)
649 GstRTSPStreamPrivate *priv;
651 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
655 return priv->dscp_qos;
659 * gst_rtsp_stream_is_transport_supported:
660 * @stream: a #GstRTSPStream
661 * @transport: (transfer none): a #GstRTSPTransport
663 * Check if @transport can be handled by stream
665 * Returns: %TRUE if @transport can be handled by @stream.
668 gst_rtsp_stream_is_transport_supported (GstRTSPStream * stream,
669 GstRTSPTransport * transport)
671 GstRTSPStreamPrivate *priv;
673 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
677 g_mutex_lock (&priv->lock);
678 if (transport->trans != GST_RTSP_TRANS_RTP)
679 goto unsupported_transmode;
681 if (!(transport->profile & priv->profiles))
682 goto unsupported_profile;
684 if (!(transport->lower_transport & priv->protocols))
685 goto unsupported_ltrans;
687 g_mutex_unlock (&priv->lock);
692 unsupported_transmode:
694 GST_DEBUG ("unsupported transport mode %d", transport->trans);
695 g_mutex_unlock (&priv->lock);
700 GST_DEBUG ("unsupported profile %d", transport->profile);
701 g_mutex_unlock (&priv->lock);
706 GST_DEBUG ("unsupported lower transport %d", transport->lower_transport);
707 g_mutex_unlock (&priv->lock);
713 * gst_rtsp_stream_set_profiles:
714 * @stream: a #GstRTSPStream
715 * @profiles: the new profiles
717 * Configure the allowed profiles for @stream.
720 gst_rtsp_stream_set_profiles (GstRTSPStream * stream, GstRTSPProfile profiles)
722 GstRTSPStreamPrivate *priv;
724 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
728 g_mutex_lock (&priv->lock);
729 priv->profiles = profiles;
730 g_mutex_unlock (&priv->lock);
734 * gst_rtsp_stream_get_profiles:
735 * @stream: a #GstRTSPStream
737 * Get the allowed profiles of @stream.
739 * Returns: a #GstRTSPProfile
742 gst_rtsp_stream_get_profiles (GstRTSPStream * stream)
744 GstRTSPStreamPrivate *priv;
747 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_RTSP_PROFILE_UNKNOWN);
751 g_mutex_lock (&priv->lock);
752 res = priv->profiles;
753 g_mutex_unlock (&priv->lock);
759 * gst_rtsp_stream_set_protocols:
760 * @stream: a #GstRTSPStream
761 * @protocols: the new flags
763 * Configure the allowed lower transport for @stream.
766 gst_rtsp_stream_set_protocols (GstRTSPStream * stream,
767 GstRTSPLowerTrans protocols)
769 GstRTSPStreamPrivate *priv;
771 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
775 g_mutex_lock (&priv->lock);
776 priv->protocols = protocols;
777 g_mutex_unlock (&priv->lock);
781 * gst_rtsp_stream_get_protocols:
782 * @stream: a #GstRTSPStream
784 * Get the allowed protocols of @stream.
786 * Returns: a #GstRTSPLowerTrans
789 gst_rtsp_stream_get_protocols (GstRTSPStream * stream)
791 GstRTSPStreamPrivate *priv;
792 GstRTSPLowerTrans res;
794 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream),
795 GST_RTSP_LOWER_TRANS_UNKNOWN);
799 g_mutex_lock (&priv->lock);
800 res = priv->protocols;
801 g_mutex_unlock (&priv->lock);
807 * gst_rtsp_stream_set_address_pool:
808 * @stream: a #GstRTSPStream
809 * @pool: (transfer none): a #GstRTSPAddressPool
811 * configure @pool to be used as the address pool of @stream.
814 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
815 GstRTSPAddressPool * pool)
817 GstRTSPStreamPrivate *priv;
818 GstRTSPAddressPool *old;
820 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
824 GST_LOG_OBJECT (stream, "set address pool %p", pool);
826 g_mutex_lock (&priv->lock);
827 if ((old = priv->pool) != pool)
828 priv->pool = pool ? g_object_ref (pool) : NULL;
831 g_mutex_unlock (&priv->lock);
834 g_object_unref (old);
838 * gst_rtsp_stream_get_address_pool:
839 * @stream: a #GstRTSPStream
841 * Get the #GstRTSPAddressPool used as the address pool of @stream.
843 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
847 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
849 GstRTSPStreamPrivate *priv;
850 GstRTSPAddressPool *result;
852 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
856 g_mutex_lock (&priv->lock);
857 if ((result = priv->pool))
858 g_object_ref (result);
859 g_mutex_unlock (&priv->lock);
865 * gst_rtsp_stream_get_multicast_address:
866 * @stream: a #GstRTSPStream
867 * @family: the #GSocketFamily
869 * Get the multicast address of @stream for @family.
871 * Returns: (transfer full) (nullable): the #GstRTSPAddress of @stream
872 * or %NULL when no address could be allocated. gst_rtsp_address_free()
876 gst_rtsp_stream_get_multicast_address (GstRTSPStream * stream,
877 GSocketFamily family)
879 GstRTSPStreamPrivate *priv;
880 GstRTSPAddress *result;
881 GstRTSPAddress **addrp;
882 GstRTSPAddressFlags flags;
884 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
888 if (family == G_SOCKET_FAMILY_IPV6) {
889 flags = GST_RTSP_ADDRESS_FLAG_IPV6;
890 addrp = &priv->addr_v6;
892 flags = GST_RTSP_ADDRESS_FLAG_IPV4;
893 addrp = &priv->addr_v4;
896 g_mutex_lock (&priv->lock);
897 if (*addrp == NULL) {
898 if (priv->pool == NULL)
901 flags |= GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
903 *addrp = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
907 result = gst_rtsp_address_copy (*addrp);
908 g_mutex_unlock (&priv->lock);
915 GST_ERROR_OBJECT (stream, "no address pool specified");
916 g_mutex_unlock (&priv->lock);
921 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
922 g_mutex_unlock (&priv->lock);
928 * gst_rtsp_stream_reserve_address:
929 * @stream: a #GstRTSPStream
930 * @address: an address
935 * Reserve @address and @port as the address and port of @stream.
937 * Returns: (nullable): the #GstRTSPAddress of @stream or %NULL when
938 * the address could be reserved. gst_rtsp_address_free() after usage.
941 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
942 const gchar * address, guint port, guint n_ports, guint ttl)
944 GstRTSPStreamPrivate *priv;
945 GstRTSPAddress *result;
947 GSocketFamily family;
948 GstRTSPAddress **addrp;
950 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
951 g_return_val_if_fail (address != NULL, NULL);
952 g_return_val_if_fail (port > 0, NULL);
953 g_return_val_if_fail (n_ports > 0, NULL);
954 g_return_val_if_fail (ttl > 0, NULL);
958 addr = g_inet_address_new_from_string (address);
960 GST_ERROR ("failed to get inet addr from %s", address);
961 family = G_SOCKET_FAMILY_IPV4;
963 family = g_inet_address_get_family (addr);
964 g_object_unref (addr);
967 if (family == G_SOCKET_FAMILY_IPV6)
968 addrp = &priv->addr_v6;
970 addrp = &priv->addr_v4;
972 g_mutex_lock (&priv->lock);
973 if (*addrp == NULL) {
974 GstRTSPAddressPoolResult res;
976 if (priv->pool == NULL)
979 res = gst_rtsp_address_pool_reserve_address (priv->pool, address,
980 port, n_ports, ttl, addrp);
981 if (res != GST_RTSP_ADDRESS_POOL_OK)
984 if (strcmp ((*addrp)->address, address) ||
985 (*addrp)->port != port || (*addrp)->n_ports != n_ports ||
986 (*addrp)->ttl != ttl)
987 goto different_address;
989 result = gst_rtsp_address_copy (*addrp);
990 g_mutex_unlock (&priv->lock);
997 GST_ERROR_OBJECT (stream, "no address pool specified");
998 g_mutex_unlock (&priv->lock);
1003 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
1005 g_mutex_unlock (&priv->lock);
1010 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
1011 " reserved", address);
1012 g_mutex_unlock (&priv->lock);
1017 /* must be called with lock */
1019 set_sockets_for_udpsinks (GstRTSPStream * stream, GSocket * rtp_socket,
1020 GSocket * rtcp_socket, GSocketFamily family)
1022 GstRTSPStreamPrivate *priv = stream->priv;
1023 const gchar *multisink_socket;
1025 if (family == G_SOCKET_FAMILY_IPV6)
1026 multisink_socket = "socket-v6";
1028 multisink_socket = "socket";
1030 g_object_set (G_OBJECT (priv->udpsink[0]), multisink_socket, rtp_socket,
1032 g_object_set (G_OBJECT (priv->udpsink[1]), multisink_socket, rtcp_socket,
1036 /* must be called with lock */
1038 create_and_configure_udpsinks (GstRTSPStream * stream)
1040 GstRTSPStreamPrivate *priv = stream->priv;
1041 GstElement *udpsink0, *udpsink1;
1046 if (priv->udpsink[0])
1047 udpsink0 = priv->udpsink[0];
1049 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
1052 goto no_udp_protocol;
1054 if (priv->udpsink[1])
1055 udpsink1 = priv->udpsink[1];
1057 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
1060 goto no_udp_protocol;
1062 /* configure sinks */
1064 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
1065 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
1067 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
1068 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
1070 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
1072 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
1073 /* Needs to be async for RECORD streams, otherwise we will never go to
1074 * PLAYING because the sinks will wait for data while the udpsrc can't
1075 * provide data with timestamps in PAUSED. */
1077 g_object_set (G_OBJECT (udpsink0), "async", FALSE, NULL);
1078 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
1080 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
1081 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
1083 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
1084 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
1086 /* update the dscp qos field in the sinks */
1087 update_dscp_qos (stream);
1089 priv->udpsink[0] = udpsink0;
1090 priv->udpsink[1] = udpsink1;
1101 /* must be called with lock */
1103 play_udpsources_one_family (GstRTSPStream * stream, GstElement * udpsrc_out[2],
1104 GSocketFamily family)
1106 GstRTSPStreamPrivate *priv;
1107 GstPad *pad, *selpad;
1111 priv = stream->priv;
1112 bin = GST_BIN (gst_object_get_parent (GST_OBJECT (priv->funnel[1])));
1114 for (i = 0; i < 2; i++) {
1115 if (priv->sinkpad || i == 1) {
1117 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1118 * values. This is only relevant for PLAY pipelines */
1119 gst_element_set_state (udpsrc_out[i], GST_STATE_PLAYING);
1120 gst_element_set_locked_state (udpsrc_out[i], TRUE);
1123 gst_bin_add (bin, udpsrc_out[i]);
1125 /* and link to the funnel */
1126 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1127 pad = gst_element_get_static_pad (udpsrc_out[i], "src");
1128 gst_pad_link (pad, selpad);
1129 gst_object_unref (pad);
1130 gst_object_unref (selpad);
1134 gst_object_unref (bin);
1137 /* must be called with lock */
1139 create_and_configure_udpsources_one_family (GstElement * udpsrc_out[2],
1140 GSocket * rtp_socket, GSocket * rtcp_socket, GSocketFamily family,
1141 const gchar * address, gint rtpport, gint rtcpport,
1142 GstRTSPLowerTrans transport)
1144 GstStateChangeReturn ret;
1146 udpsrc_out[0] = gst_element_factory_make ("udpsrc", NULL);
1147 udpsrc_out[1] = gst_element_factory_make ("udpsrc", NULL);
1149 if (udpsrc_out[0] == NULL || udpsrc_out[1] == NULL)
1152 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1153 g_object_set (G_OBJECT (udpsrc_out[0]), "address", address, NULL);
1154 g_object_set (G_OBJECT (udpsrc_out[1]), "address", address, NULL);
1155 g_object_set (G_OBJECT (udpsrc_out[0]), "port", rtpport, NULL);
1156 g_object_set (G_OBJECT (udpsrc_out[1]), "port", rtcpport, NULL);
1157 g_object_set (G_OBJECT (udpsrc_out[0]), "loop", FALSE, NULL);
1158 g_object_set (G_OBJECT (udpsrc_out[1]), "loop", FALSE, NULL);
1161 g_object_set (G_OBJECT (udpsrc_out[0]), "socket", rtp_socket, NULL);
1162 g_object_set (G_OBJECT (udpsrc_out[1]), "socket", rtcp_socket, NULL);
1164 ret = gst_element_set_state (udpsrc_out[0], GST_STATE_READY);
1165 if (ret == GST_STATE_CHANGE_FAILURE)
1167 ret = gst_element_set_state (udpsrc_out[1], GST_STATE_READY);
1168 if (ret == GST_STATE_CHANGE_FAILURE)
1178 gst_object_unref (udpsrc_out[0]);
1180 gst_object_unref (udpsrc_out[1]);
1186 alloc_ports_one_family (GstRTSPStream * stream, GSocketFamily family,
1187 GstElement * udpsrc_out[2], GstRTSPRange * server_port_out,
1188 GstRTSPTransport * ct, GstRTSPAddress ** server_addr_out,
1189 gboolean use_client_settings)
1191 GstRTSPStreamPrivate *priv = stream->priv;
1192 GSocket *rtp_socket = NULL;
1193 GSocket *rtcp_socket;
1194 gint tmp_rtp, tmp_rtcp;
1196 gint rtpport, rtcpport;
1197 GList *rejected_addresses = NULL;
1198 GstRTSPAddress *addr = NULL;
1199 GInetAddress *inetaddr = NULL;
1201 GSocketAddress *rtp_sockaddr = NULL;
1202 GSocketAddress *rtcp_sockaddr = NULL;
1203 GstRTSPAddressPool *pool;
1204 GstRTSPLowerTrans transport;
1208 transport = ct->lower_transport;
1210 /* Start with random port */
1213 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1214 G_SOCKET_PROTOCOL_UDP, NULL);
1216 goto no_udp_protocol;
1217 g_socket_set_multicast_loopback (rtcp_socket, FALSE);
1219 if (*server_addr_out)
1220 gst_rtsp_address_free (*server_addr_out);
1222 /* try to allocate 2 UDP ports, the RTP port should be an even
1223 * number and the RTCP port should be the next (uneven) port */
1226 if (rtp_socket == NULL) {
1227 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1228 G_SOCKET_PROTOCOL_UDP, NULL);
1230 goto no_udp_protocol;
1231 g_socket_set_multicast_loopback (rtp_socket, FALSE);
1234 if (pool && ((transport == GST_RTSP_LOWER_TRANS_UDP &&
1235 gst_rtsp_address_pool_has_unicast_addresses (pool))
1236 || transport == GST_RTSP_LOWER_TRANS_UDP_MCAST)) {
1237 GstRTSPAddressFlags flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT;
1239 if (transport == GST_RTSP_LOWER_TRANS_UDP)
1240 flags |= GST_RTSP_ADDRESS_FLAG_UNICAST;
1242 flags |= GST_RTSP_ADDRESS_FLAG_MULTICAST;
1245 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1247 if (family == G_SOCKET_FAMILY_IPV6)
1248 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1250 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1252 if (ct->destination && transport == GST_RTSP_LOWER_TRANS_UDP_MCAST
1253 && use_client_settings)
1254 gst_rtsp_address_pool_reserve_address (pool, ct->destination,
1255 ct->port.min, 2, ct->ttl, &addr);
1257 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1262 tmp_rtp = addr->port;
1264 g_clear_object (&inetaddr);
1265 inetaddr = g_inet_address_new_from_string (addr->address);
1267 /* On Windows it's not possible to bind to a multicast address
1268 * but the OS will make sure to filter out all packets that
1269 * arrive not for the multicast address the socket joined.
1271 * On Linux and others it is necessary to bind to a multicast
1272 * address to let the OS filter out all packets that are received
1273 * on the same port but for different addresses than the multicast
1277 if (g_inet_address_get_is_multicast (inetaddr)) {
1278 g_object_unref (inetaddr);
1279 inetaddr = g_inet_address_new_any (family);
1289 if (inetaddr == NULL)
1290 inetaddr = g_inet_address_new_any (family);
1293 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1294 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1295 g_object_unref (rtp_sockaddr);
1298 g_object_unref (rtp_sockaddr);
1300 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1301 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1302 g_clear_object (&rtp_sockaddr);
1307 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1308 g_object_unref (rtp_sockaddr);
1310 /* check if port is even */
1311 if ((tmp_rtp & 1) != 0) {
1312 /* port not even, close and allocate another */
1314 g_clear_object (&rtp_socket);
1319 tmp_rtcp = tmp_rtp + 1;
1321 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1322 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1323 g_object_unref (rtcp_sockaddr);
1324 g_clear_object (&rtp_socket);
1327 g_object_unref (rtcp_sockaddr);
1330 addr_str = g_inet_address_to_string (inetaddr);
1332 addr_str = addr->address;
1333 g_clear_object (&inetaddr);
1335 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1336 rtcp_socket, family, addr_str, tmp_rtp, tmp_rtcp, transport)) {
1339 goto no_udp_protocol;
1345 play_udpsources_one_family (stream, udpsrc_out, family);
1347 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1348 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1350 /* this should not happen... */
1351 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1354 /* set RTP and RTCP sockets */
1355 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1357 server_port_out->min = rtpport;
1358 server_port_out->max = rtcpport;
1360 *server_addr_out = addr;
1361 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1363 g_object_unref (rtp_socket);
1364 g_object_unref (rtcp_socket);
1388 g_object_unref (inetaddr);
1389 g_list_free_full (rejected_addresses,
1390 (GDestroyNotify) gst_rtsp_address_free);
1392 gst_rtsp_address_free (addr);
1394 g_object_unref (rtp_socket);
1396 g_object_unref (rtcp_socket);
1402 * gst_rtsp_stream_allocate_udp_sockets:
1403 * @stream: a #GstRTSPStream
1404 * @family: protocol family
1405 * @transport_method: transport method
1407 * Allocates RTP and RTCP ports.
1409 * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
1412 gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
1413 GSocketFamily family, GstRTSPTransport * ct, gboolean use_client_settings)
1415 GstRTSPStreamPrivate *priv;
1416 gboolean result = FALSE;
1417 GstRTSPLowerTrans transport = ct->lower_transport;
1419 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1420 priv = stream->priv;
1421 g_return_val_if_fail (priv->is_joined, FALSE);
1423 g_mutex_lock (&priv->lock);
1425 if (family == G_SOCKET_FAMILY_IPV4) {
1426 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1427 if (priv->have_ipv4_mcast)
1429 priv->have_ipv4_mcast =
1430 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
1431 priv->udpsrc_mcast_v4, &priv->server_port_v4, ct, &priv->addr_v4,
1432 use_client_settings);
1435 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1436 &priv->server_port_v4, ct, &priv->server_addr_v4,
1437 use_client_settings);
1440 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1441 if (priv->have_ipv6_mcast)
1443 priv->have_ipv6_mcast =
1444 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
1445 priv->udpsrc_mcast_v6, &priv->server_port_v6, ct, &priv->addr_v6,
1446 use_client_settings);
1448 if (priv->have_ipv6)
1451 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1452 &priv->server_port_v6, ct, &priv->server_addr_v6,
1453 use_client_settings);
1458 result = priv->have_ipv4 || priv->have_ipv4_mcast || priv->have_ipv6 ||
1459 priv->have_ipv6_mcast;
1461 g_mutex_unlock (&priv->lock);
1467 * gst_rtsp_stream_set_client_side:
1468 * @stream: a #GstRTSPStream
1469 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1470 * an RTSP connection.
1472 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1473 * streams to an RTSP server via RECORD. This has the practical effect
1474 * of changing which UDP port numbers are used when setting up the local
1475 * side of the stream sending to be either the 'server' or 'client' pair
1476 * of a configured UDP transport.
1479 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1481 GstRTSPStreamPrivate *priv;
1483 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1484 priv = stream->priv;
1485 g_mutex_lock (&priv->lock);
1486 priv->client_side = client_side;
1487 g_mutex_unlock (&priv->lock);
1491 * gst_rtsp_stream_set_client_side:
1492 * @stream: a #GstRTSPStream
1494 * See gst_rtsp_stream_set_client_side()
1496 * Returns: TRUE if this #GstRTSPStream is client-side.
1499 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1501 GstRTSPStreamPrivate *priv;
1504 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1506 priv = stream->priv;
1507 g_mutex_lock (&priv->lock);
1508 ret = priv->client_side;
1509 g_mutex_unlock (&priv->lock);
1515 * gst_rtsp_stream_get_server_port:
1516 * @stream: a #GstRTSPStream
1517 * @server_port: (out): result server port
1518 * @family: the port family to get
1520 * Fill @server_port with the port pair used by the server. This function can
1521 * only be called when @stream has been joined.
1524 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1525 GstRTSPRange * server_port, GSocketFamily family)
1527 GstRTSPStreamPrivate *priv;
1529 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1530 priv = stream->priv;
1531 g_return_if_fail (priv->is_joined);
1533 g_mutex_lock (&priv->lock);
1534 if (family == G_SOCKET_FAMILY_IPV4) {
1536 *server_port = priv->server_port_v4;
1539 *server_port = priv->server_port_v6;
1541 g_mutex_unlock (&priv->lock);
1545 * gst_rtsp_stream_get_rtpsession:
1546 * @stream: a #GstRTSPStream
1548 * Get the RTP session of this stream.
1550 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1553 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1555 GstRTSPStreamPrivate *priv;
1558 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1560 priv = stream->priv;
1562 g_mutex_lock (&priv->lock);
1563 if ((session = priv->session))
1564 g_object_ref (session);
1565 g_mutex_unlock (&priv->lock);
1571 * gst_rtsp_stream_get_ssrc:
1572 * @stream: a #GstRTSPStream
1573 * @ssrc: (out): result ssrc
1575 * Get the SSRC used by the RTP session of this stream. This function can only
1576 * be called when @stream has been joined.
1579 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1581 GstRTSPStreamPrivate *priv;
1583 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1584 priv = stream->priv;
1585 g_return_if_fail (priv->is_joined);
1587 g_mutex_lock (&priv->lock);
1588 if (ssrc && priv->session)
1589 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1590 g_mutex_unlock (&priv->lock);
1594 * gst_rtsp_stream_set_retransmission_time:
1595 * @stream: a #GstRTSPStream
1596 * @time: a #GstClockTime
1598 * Set the amount of time to store retransmission packets.
1601 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1604 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1606 g_mutex_lock (&stream->priv->lock);
1607 stream->priv->rtx_time = time;
1608 if (stream->priv->rtxsend)
1609 g_object_set (stream->priv->rtxsend, "max-size-time",
1610 GST_TIME_AS_MSECONDS (time), NULL);
1611 g_mutex_unlock (&stream->priv->lock);
1615 * gst_rtsp_stream_get_retransmission_time:
1616 * @stream: a #GstRTSPStream
1618 * Get the amount of time to store retransmission data.
1620 * Returns: the amount of time to store retransmission data.
1623 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1627 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1629 g_mutex_lock (&stream->priv->lock);
1630 ret = stream->priv->rtx_time;
1631 g_mutex_unlock (&stream->priv->lock);
1637 * gst_rtsp_stream_set_retransmission_pt:
1638 * @stream: a #GstRTSPStream
1641 * Set the payload type (pt) for retransmission of this stream.
1644 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1646 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1648 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1650 g_mutex_lock (&stream->priv->lock);
1651 stream->priv->rtx_pt = rtx_pt;
1652 if (stream->priv->rtxsend) {
1653 guint pt = gst_rtsp_stream_get_pt (stream);
1654 gchar *pt_s = g_strdup_printf ("%d", pt);
1655 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1656 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1657 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1659 gst_structure_free (rtx_pt_map);
1661 g_mutex_unlock (&stream->priv->lock);
1665 * gst_rtsp_stream_get_retransmission_pt:
1666 * @stream: a #GstRTSPStream
1668 * Get the payload-type used for retransmission of this stream
1670 * Returns: The retransmission PT.
1673 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1677 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1679 g_mutex_lock (&stream->priv->lock);
1680 rtx_pt = stream->priv->rtx_pt;
1681 g_mutex_unlock (&stream->priv->lock);
1687 * gst_rtsp_stream_set_buffer_size:
1688 * @stream: a #GstRTSPStream
1689 * @size: the buffer size
1691 * Set the size of the UDP transmission buffer (in bytes)
1692 * Needs to be set before the stream is joined to a bin.
1697 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1699 g_mutex_lock (&stream->priv->lock);
1700 stream->priv->buffer_size = size;
1701 g_mutex_unlock (&stream->priv->lock);
1705 * gst_rtsp_stream_get_buffer_size:
1706 * @stream: a #GstRTSPStream
1708 * Get the size of the UDP transmission buffer (in bytes)
1710 * Returns: the size of the UDP TX buffer
1715 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1719 g_mutex_lock (&stream->priv->lock);
1720 buffer_size = stream->priv->buffer_size;
1721 g_mutex_unlock (&stream->priv->lock);
1726 /* executed from streaming thread */
1728 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1730 GstRTSPStreamPrivate *priv = stream->priv;
1731 GstCaps *newcaps, *oldcaps;
1733 newcaps = gst_pad_get_current_caps (pad);
1735 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1738 g_mutex_lock (&priv->lock);
1739 oldcaps = priv->caps;
1740 priv->caps = newcaps;
1741 g_mutex_unlock (&priv->lock);
1744 gst_caps_unref (oldcaps);
1748 dump_structure (const GstStructure * s)
1752 sstr = gst_structure_to_string (s);
1753 GST_INFO ("structure: %s", sstr);
1757 static GstRTSPStreamTransport *
1758 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1760 GstRTSPStreamPrivate *priv = stream->priv;
1762 GstRTSPStreamTransport *result = NULL;
1767 if (rtcp_from == NULL)
1770 tmp = g_strrstr (rtcp_from, ":");
1774 port = atoi (tmp + 1);
1775 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1777 g_mutex_lock (&priv->lock);
1778 GST_INFO ("finding %s:%d in %d transports", dest, port,
1779 g_list_length (priv->transports));
1781 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1782 GstRTSPStreamTransport *trans = walk->data;
1783 const GstRTSPTransport *tr;
1786 tr = gst_rtsp_stream_transport_get_transport (trans);
1788 if (priv->client_side) {
1789 /* In client side mode the 'destination' is the RTSP server, so send
1791 min = tr->server_port.min;
1792 max = tr->server_port.max;
1794 min = tr->client_port.min;
1795 max = tr->client_port.max;
1798 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1804 g_object_ref (result);
1805 g_mutex_unlock (&priv->lock);
1812 static GstRTSPStreamTransport *
1813 check_transport (GObject * source, GstRTSPStream * stream)
1815 GstStructure *stats;
1816 GstRTSPStreamTransport *trans;
1818 /* see if we have a stream to match with the origin of the RTCP packet */
1819 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1820 if (trans == NULL) {
1821 g_object_get (source, "stats", &stats, NULL);
1823 const gchar *rtcp_from;
1825 dump_structure (stats);
1827 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1828 if ((trans = find_transport (stream, rtcp_from))) {
1829 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1831 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1834 gst_structure_free (stats);
1842 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1844 GstRTSPStreamTransport *trans;
1846 GST_INFO ("%p: new source %p", stream, source);
1848 trans = check_transport (source, stream);
1851 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1855 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1857 GST_INFO ("%p: new SDES %p", stream, source);
1861 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1863 GstRTSPStreamTransport *trans;
1865 trans = check_transport (source, stream);
1868 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1869 gst_rtsp_stream_transport_keep_alive (trans);
1873 GstStructure *stats;
1874 g_object_get (source, "stats", &stats, NULL);
1876 dump_structure (stats);
1877 gst_structure_free (stats);
1884 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1886 GST_INFO ("%p: source %p bye", stream, source);
1890 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1892 GstRTSPStreamTransport *trans;
1894 GST_INFO ("%p: source %p bye timeout", stream, source);
1896 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1897 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1898 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1903 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1905 GstRTSPStreamTransport *trans;
1907 GST_INFO ("%p: source %p timeout", stream, source);
1909 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1910 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1911 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1916 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1918 GST_INFO ("%p: new sender source %p", stream, source);
1921 GstStructure *stats;
1922 g_object_get (source, "stats", &stats, NULL);
1924 dump_structure (stats);
1925 gst_structure_free (stats);
1932 on_sender_ssrc_active (GObject * session, GObject * source,
1933 GstRTSPStream * stream)
1937 GstStructure *stats;
1938 g_object_get (source, "stats", &stats, NULL);
1940 dump_structure (stats);
1941 gst_structure_free (stats);
1948 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1951 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1952 g_list_free (priv->tr_cache_rtp);
1953 priv->tr_cache_rtp = NULL;
1955 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1956 g_list_free (priv->tr_cache_rtcp);
1957 priv->tr_cache_rtcp = NULL;
1961 static GstFlowReturn
1962 handle_new_sample (GstAppSink * sink, gpointer user_data)
1964 GstRTSPStreamPrivate *priv;
1968 GstRTSPStream *stream;
1971 sample = gst_app_sink_pull_sample (sink);
1975 stream = (GstRTSPStream *) user_data;
1976 priv = stream->priv;
1977 buffer = gst_sample_get_buffer (sample);
1979 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1981 g_mutex_lock (&priv->lock);
1983 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1984 clear_tr_cache (priv, is_rtp);
1985 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1986 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1987 priv->tr_cache_rtp =
1988 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1990 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1993 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1994 clear_tr_cache (priv, is_rtp);
1995 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1996 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1997 priv->tr_cache_rtcp =
1998 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
2000 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
2003 g_mutex_unlock (&priv->lock);
2006 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
2007 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2008 gst_rtsp_stream_transport_send_rtp (tr, buffer);
2011 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
2012 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2013 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
2016 gst_sample_unref (sample);
2021 static GstAppSinkCallbacks sink_cb = {
2022 NULL, /* not interested in EOS */
2023 NULL, /* not interested in preroll samples */
2028 get_rtp_encoder (GstRTSPStream * stream, guint session)
2030 GstRTSPStreamPrivate *priv = stream->priv;
2032 if (priv->srtpenc == NULL) {
2035 name = g_strdup_printf ("srtpenc_%u", session);
2036 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
2039 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
2041 return gst_object_ref (priv->srtpenc);
2045 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2047 GstRTSPStreamPrivate *priv = stream->priv;
2048 GstElement *oldenc, *enc;
2052 if (priv->idx != session)
2055 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
2057 oldenc = priv->srtpenc;
2058 enc = get_rtp_encoder (stream, session);
2059 name = g_strdup_printf ("rtp_sink_%d", session);
2060 pad = gst_element_get_request_pad (enc, name);
2062 gst_object_unref (pad);
2065 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
2072 request_rtcp_encoder (GstElement * rtpbin, guint session,
2073 GstRTSPStream * stream)
2075 GstRTSPStreamPrivate *priv = stream->priv;
2076 GstElement *oldenc, *enc;
2080 if (priv->idx != session)
2083 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
2085 oldenc = priv->srtpenc;
2086 enc = get_rtp_encoder (stream, session);
2087 name = g_strdup_printf ("rtcp_sink_%d", session);
2088 pad = gst_element_get_request_pad (enc, name);
2090 gst_object_unref (pad);
2093 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
2100 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2102 GstRTSPStreamPrivate *priv = stream->priv;
2105 GST_DEBUG ("request key %08x", ssrc);
2107 g_mutex_lock (&priv->lock);
2108 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
2109 gst_caps_ref (caps);
2110 g_mutex_unlock (&priv->lock);
2116 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
2117 GstRTSPStream * stream)
2119 GstRTSPStreamPrivate *priv = stream->priv;
2121 if (priv->idx != session)
2124 if (priv->srtpdec == NULL) {
2127 name = g_strdup_printf ("srtpdec_%u", session);
2128 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2131 g_signal_connect (priv->srtpdec, "request-key",
2132 (GCallback) request_key, stream);
2134 return gst_object_ref (priv->srtpdec);
2138 * gst_rtsp_stream_request_aux_sender:
2139 * @stream: a #GstRTSPStream
2140 * @sessid: the session id
2142 * Creating a rtxsend bin
2144 * Returns: (transfer full): a #GstElement.
2149 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2153 GstStructure *pt_map;
2158 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2160 pt = gst_rtsp_stream_get_pt (stream);
2161 pt_s = g_strdup_printf ("%u", pt);
2162 rtx_pt = stream->priv->rtx_pt;
2164 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2166 bin = gst_bin_new (NULL);
2167 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2168 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2169 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2170 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2171 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2173 gst_structure_free (pt_map);
2174 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2176 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2177 name = g_strdup_printf ("src_%u", sessid);
2178 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2180 gst_object_unref (pad);
2182 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2183 name = g_strdup_printf ("sink_%u", sessid);
2184 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2186 gst_object_unref (pad);
2192 * gst_rtsp_stream_set_pt_map:
2193 * @stream: a #GstRTSPStream
2197 * Configure a pt map between @pt and @caps.
2200 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2202 GstRTSPStreamPrivate *priv = stream->priv;
2204 g_mutex_lock (&priv->lock);
2205 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2206 g_mutex_unlock (&priv->lock);
2210 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2211 GstRTSPStream * stream)
2213 GstRTSPStreamPrivate *priv = stream->priv;
2214 GstCaps *caps = NULL;
2216 g_mutex_lock (&priv->lock);
2218 if (priv->idx == session) {
2219 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2221 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2222 gst_caps_ref (caps);
2224 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2228 g_mutex_unlock (&priv->lock);
2234 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2236 GstRTSPStreamPrivate *priv = stream->priv;
2238 GstPadLinkReturn ret;
2241 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2242 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2244 name = gst_pad_get_name (pad);
2245 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2251 if (priv->idx != sessid)
2254 if (gst_pad_is_linked (priv->sinkpad)) {
2255 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2256 GST_DEBUG_PAD_NAME (priv->sinkpad));
2260 /* link the RTP pad to the session manager, it should not really fail unless
2261 * this is not really an RTP pad */
2262 ret = gst_pad_link (pad, priv->sinkpad);
2263 if (ret != GST_PAD_LINK_OK)
2265 priv->recv_rtp_src = gst_object_ref (pad);
2272 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2273 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2278 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2279 GstRTSPStream * stream)
2281 /* TODO: What to do here other than this? */
2282 GST_DEBUG ("Stream %p: Got EOS", stream);
2283 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2286 /* must be called with lock */
2288 create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2290 GstRTSPStreamPrivate *priv;
2291 GstPad *pad, *sinkpad = NULL;
2292 gboolean is_tcp = FALSE, is_udp = FALSE;
2295 priv = stream->priv;
2297 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2298 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2299 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2301 if (is_udp && !create_and_configure_udpsinks (stream))
2302 goto no_udp_protocol;
2304 for (i = 0; i < 2; i++) {
2305 GstPad *teepad, *queuepad;
2306 /* For the sender we create this bit of pipeline for both
2307 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2308 * we need to add a queue before appsink and udpsink to make
2309 * the pipeline not block. For the TCP case, we want to pump
2310 * client as fast as possible anyway. This pipeline is used
2311 * when both TCP and UDP are present.
2313 * .--------. .-----. .---------. .---------.
2314 * | rtpbin | | tee | | queue | | udpsink |
2315 * | send->sink src->sink src->sink |
2316 * '--------' | | '---------' '---------'
2317 * | | .---------. .---------.
2318 * | | | queue | | appsink |
2319 * | src->sink src->sink |
2320 * '-----' '---------' '---------'
2322 * When only UDP or only TCP is allowed, we skip the tee and queue
2323 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2326 /* Only link the RTP send src if we're going to send RTP, link
2327 * the RTCP send src always */
2328 if (priv->srcpad || i == 1) {
2331 gst_bin_add (bin, priv->udpsink[i]);
2332 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2337 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2338 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2339 gst_bin_add (bin, priv->appsink[i]);
2340 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2341 &sink_cb, stream, NULL);
2344 if (is_udp && is_tcp) {
2345 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2347 /* make tee for RTP/RTCP */
2348 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2349 gst_bin_add (bin, priv->tee[i]);
2351 /* and link to rtpbin send pad */
2352 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2353 gst_pad_link (priv->send_src[i], pad);
2354 gst_object_unref (pad);
2356 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2357 g_object_set (priv->udpqueue[i], "max-size-buffers",
2358 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2360 gst_bin_add (bin, priv->udpqueue[i]);
2361 /* link tee to udpqueue */
2362 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2363 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2364 gst_pad_link (teepad, pad);
2365 gst_object_unref (pad);
2366 gst_object_unref (teepad);
2368 /* link udpqueue to udpsink */
2369 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2370 gst_pad_link (queuepad, sinkpad);
2371 gst_object_unref (queuepad);
2372 gst_object_unref (sinkpad);
2375 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2376 g_object_set (priv->appqueue[i], "max-size-buffers",
2377 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2379 gst_bin_add (bin, priv->appqueue[i]);
2380 /* and link tee to appqueue */
2381 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2382 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2383 gst_pad_link (teepad, pad);
2384 gst_object_unref (pad);
2385 gst_object_unref (teepad);
2387 /* and link appqueue to appsink */
2388 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2389 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2390 gst_pad_link (queuepad, pad);
2391 gst_object_unref (pad);
2392 gst_object_unref (queuepad);
2393 } else if (is_tcp) {
2394 /* only appsink needed, link it to the session */
2395 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2396 gst_pad_link (priv->send_src[i], pad);
2397 gst_object_unref (pad);
2399 /* when its only TCP, we need to set sync and preroll to FALSE
2400 * for the sink to avoid deadlock. And this is only needed for
2401 * sink used for RTCP data, not the RTP data. */
2403 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2405 /* else only udpsink needed, link it to the session */
2406 gst_pad_link (priv->send_src[i], sinkpad);
2407 gst_object_unref (sinkpad);
2411 /* check if we need to set to a special state */
2412 if (state != GST_STATE_NULL) {
2413 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2414 gst_element_set_state (priv->udpsink[i], state);
2415 if (priv->appsink[i] && (priv->srcpad || i == 1))
2416 gst_element_set_state (priv->appsink[i], state);
2417 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2418 gst_element_set_state (priv->appqueue[i], state);
2419 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2420 gst_element_set_state (priv->udpqueue[i], state);
2421 if (priv->tee[i] && (priv->srcpad || i == 1))
2422 gst_element_set_state (priv->tee[i], state);
2435 /* must be called with lock */
2437 create_receiver_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2439 GstRTSPStreamPrivate *priv;
2440 GstPad *pad, *selpad;
2444 priv = stream->priv;
2446 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2448 for (i = 0; i < 2; i++) {
2449 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2450 * RTCP sink always */
2451 if (priv->sinkpad || i == 1) {
2452 /* For the receiver we create this bit of pipeline for both
2453 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2454 * and it is all funneled into the rtpbin receive pad.
2456 * .--------. .--------. .--------.
2457 * | udpsrc | | funnel | | rtpbin |
2458 * | src->sink src->sink |
2459 * '--------' | | '--------'
2463 * '--------' '--------'
2465 /* make funnel for the RTP/RTCP receivers */
2466 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2467 gst_bin_add (bin, priv->funnel[i]);
2469 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2470 gst_pad_link (pad, priv->recv_sink[i]);
2471 gst_object_unref (pad);
2473 if (priv->udpsrc_v4[i]) {
2475 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2476 * values. This is only relevant for PLAY pipelines */
2477 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2478 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2481 gst_bin_add (bin, priv->udpsrc_v4[i]);
2483 /* and link to the funnel v4 */
2484 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2485 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2486 gst_pad_link (pad, selpad);
2487 gst_object_unref (pad);
2488 gst_object_unref (selpad);
2491 if (priv->udpsrc_v6[i]) {
2493 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2494 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2496 gst_bin_add (bin, priv->udpsrc_v6[i]);
2498 /* and link to the funnel v6 */
2499 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2500 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2501 gst_pad_link (pad, selpad);
2502 gst_object_unref (pad);
2503 gst_object_unref (selpad);
2507 /* make and add appsrc */
2508 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2509 priv->appsrc_base_time[i] = -1;
2510 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2511 gst_bin_add (bin, priv->appsrc[i]);
2512 /* and link to the funnel */
2513 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2514 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2515 gst_pad_link (pad, selpad);
2516 gst_object_unref (pad);
2517 gst_object_unref (selpad);
2521 /* check if we need to set to a special state */
2522 if (state != GST_STATE_NULL) {
2523 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2524 gst_element_set_state (priv->funnel[i], state);
2525 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2526 gst_element_set_state (priv->appsrc[i], state);
2532 * gst_rtsp_stream_join_bin:
2533 * @stream: a #GstRTSPStream
2534 * @bin: (transfer none): a #GstBin to join
2535 * @rtpbin: (transfer none): a rtpbin element in @bin
2536 * @state: the target state of the new elements
2538 * Join the #GstBin @bin that contains the element @rtpbin.
2540 * @stream will link to @rtpbin, which must be inside @bin. The elements
2541 * added to @bin will be set to the state given in @state.
2543 * Returns: %TRUE on success.
2546 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2547 GstElement * rtpbin, GstState state)
2549 GstRTSPStreamPrivate *priv;
2552 GstPadLinkReturn ret;
2554 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2555 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2556 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2558 priv = stream->priv;
2560 g_mutex_lock (&priv->lock);
2561 if (priv->is_joined)
2564 /* create a session with the same index as the stream */
2567 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2569 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2570 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2572 g_signal_connect (rtpbin, "request-rtp-encoder",
2573 (GCallback) request_rtp_encoder, stream);
2574 g_signal_connect (rtpbin, "request-rtcp-encoder",
2575 (GCallback) request_rtcp_encoder, stream);
2576 g_signal_connect (rtpbin, "request-rtp-decoder",
2577 (GCallback) request_rtp_rtcp_decoder, stream);
2578 g_signal_connect (rtpbin, "request-rtcp-decoder",
2579 (GCallback) request_rtp_rtcp_decoder, stream);
2582 if (priv->sinkpad) {
2583 g_signal_connect (rtpbin, "request-pt-map",
2584 (GCallback) request_pt_map, stream);
2587 /* get pads from the RTP session element for sending and receiving
2590 /* get a pad for sending RTP */
2591 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2592 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2595 /* link the RTP pad to the session manager, it should not really fail unless
2596 * this is not really an RTP pad */
2597 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2598 if (ret != GST_PAD_LINK_OK)
2601 name = g_strdup_printf ("send_rtp_src_%u", idx);
2602 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2605 /* Need to connect our sinkpad from here */
2606 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2608 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2610 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2611 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2615 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2616 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2618 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2619 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2622 /* get the session */
2623 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2625 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2627 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2629 g_signal_connect (priv->session, "on-ssrc-active",
2630 (GCallback) on_ssrc_active, stream);
2631 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2633 g_signal_connect (priv->session, "on-bye-timeout",
2634 (GCallback) on_bye_timeout, stream);
2635 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2638 /* signal for sender ssrc */
2639 g_signal_connect (priv->session, "on-new-sender-ssrc",
2640 (GCallback) on_new_sender_ssrc, stream);
2641 g_signal_connect (priv->session, "on-sender-ssrc-active",
2642 (GCallback) on_sender_ssrc_active, stream);
2644 if (!create_sender_part (stream, bin, state))
2645 goto no_udp_protocol;
2647 create_receiver_part (stream, bin, state);
2650 /* be notified of caps changes */
2651 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2652 (GCallback) caps_notify, stream);
2655 priv->is_joined = TRUE;
2656 g_mutex_unlock (&priv->lock);
2663 g_mutex_unlock (&priv->lock);
2668 GST_WARNING ("failed to link stream %u", idx);
2669 gst_object_unref (priv->send_rtp_sink);
2670 priv->send_rtp_sink = NULL;
2671 g_mutex_unlock (&priv->lock);
2676 GST_WARNING ("failed to allocate ports %u", idx);
2677 gst_object_unref (priv->send_rtp_sink);
2678 priv->send_rtp_sink = NULL;
2679 gst_object_unref (priv->send_src[0]);
2680 priv->send_src[0] = NULL;
2681 gst_object_unref (priv->send_src[1]);
2682 priv->send_src[1] = NULL;
2683 gst_object_unref (priv->recv_sink[0]);
2684 priv->recv_sink[0] = NULL;
2685 gst_object_unref (priv->recv_sink[1]);
2686 priv->recv_sink[1] = NULL;
2687 if (priv->udpsink[0])
2688 gst_element_set_state (priv->udpsink[0], GST_STATE_NULL);
2689 if (priv->udpsink[1])
2690 gst_element_set_state (priv->udpsink[1], GST_STATE_NULL);
2691 if (priv->udpsrc_v4[0]) {
2692 gst_element_set_state (priv->udpsrc_v4[0], GST_STATE_NULL);
2693 gst_object_unref (priv->udpsrc_v4[0]);
2694 priv->udpsrc_v4[0] = NULL;
2696 if (priv->udpsrc_v4[1]) {
2697 gst_element_set_state (priv->udpsrc_v4[1], GST_STATE_NULL);
2698 gst_object_unref (priv->udpsrc_v4[1]);
2699 priv->udpsrc_v4[1] = NULL;
2701 if (priv->udpsrc_mcast_v4[0]) {
2702 gst_element_set_state (priv->udpsrc_mcast_v4[0], GST_STATE_NULL);
2703 gst_object_unref (priv->udpsrc_mcast_v4[0]);
2704 priv->udpsrc_mcast_v4[0] = NULL;
2706 if (priv->udpsrc_mcast_v4[1]) {
2707 gst_element_set_state (priv->udpsrc_mcast_v4[1], GST_STATE_NULL);
2708 gst_object_unref (priv->udpsrc_mcast_v4[1]);
2709 priv->udpsrc_mcast_v4[1] = NULL;
2711 if (priv->udpsrc_v6[0]) {
2712 gst_element_set_state (priv->udpsrc_v6[0], GST_STATE_NULL);
2713 gst_object_unref (priv->udpsrc_v6[0]);
2714 priv->udpsrc_v6[0] = NULL;
2716 if (priv->udpsrc_v6[1]) {
2717 gst_element_set_state (priv->udpsrc_v6[1], GST_STATE_NULL);
2718 gst_object_unref (priv->udpsrc_v6[1]);
2719 priv->udpsrc_v6[1] = NULL;
2721 if (priv->udpsrc_mcast_v6[0]) {
2722 gst_element_set_state (priv->udpsrc_mcast_v6[0], GST_STATE_NULL);
2723 gst_object_unref (priv->udpsrc_mcast_v6[0]);
2724 priv->udpsrc_mcast_v6[0] = NULL;
2726 if (priv->udpsrc_mcast_v6[1]) {
2727 gst_element_set_state (priv->udpsrc_mcast_v6[1], GST_STATE_NULL);
2728 gst_object_unref (priv->udpsrc_mcast_v6[1]);
2729 priv->udpsrc_mcast_v6[1] = NULL;
2731 g_mutex_unlock (&priv->lock);
2737 * gst_rtsp_stream_leave_bin:
2738 * @stream: a #GstRTSPStream
2739 * @bin: (transfer none): a #GstBin
2740 * @rtpbin: (transfer none): a rtpbin #GstElement
2742 * Remove the elements of @stream from @bin.
2744 * Return: %TRUE on success.
2747 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2748 GstElement * rtpbin)
2750 GstRTSPStreamPrivate *priv;
2752 gboolean is_tcp, is_udp;
2754 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2755 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2756 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2758 priv = stream->priv;
2760 g_mutex_lock (&priv->lock);
2761 if (!priv->is_joined)
2762 goto was_not_joined;
2764 /* all transports must be removed by now */
2765 if (priv->transports != NULL)
2766 goto transports_not_removed;
2768 clear_tr_cache (priv, TRUE);
2769 clear_tr_cache (priv, FALSE);
2771 GST_INFO ("stream %p leaving bin", stream);
2774 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2776 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2777 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2778 gst_object_unref (priv->send_rtp_sink);
2779 priv->send_rtp_sink = NULL;
2780 } else if (priv->recv_rtp_src) {
2781 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2782 gst_object_unref (priv->recv_rtp_src);
2783 priv->recv_rtp_src = NULL;
2786 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2788 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2789 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2792 for (i = 0; i < 2; i++) {
2793 if (priv->udpsink[i])
2794 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2795 if (priv->appsink[i])
2796 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2797 if (priv->appqueue[i])
2798 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2799 if (priv->udpqueue[i])
2800 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2802 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2803 if (priv->funnel[i])
2804 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2805 if (priv->appsrc[i])
2806 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2808 if (priv->udpsrc_v4[i]) {
2809 if (priv->sinkpad || i == 1) {
2810 /* and set udpsrc to NULL now before removing */
2811 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2812 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2813 /* removing them should also nicely release the request
2814 * pads when they finalize */
2815 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2817 /* we need to set the state to NULL before unref */
2818 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2819 gst_object_unref (priv->udpsrc_v4[i]);
2823 if (priv->udpsrc_mcast_v4[i]) {
2824 if (priv->sinkpad || i == 1) {
2825 /* and set udpsrc to NULL now before removing */
2826 gst_element_set_locked_state (priv->udpsrc_mcast_v4[i], FALSE);
2827 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2828 /* removing them should also nicely release the request
2829 * pads when they finalize */
2830 gst_bin_remove (bin, priv->udpsrc_mcast_v4[i]);
2832 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2833 gst_object_unref (priv->udpsrc_mcast_v4[i]);
2837 if (priv->udpsrc_v6[i]) {
2838 if (priv->sinkpad || i == 1) {
2839 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2840 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2841 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2843 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2844 gst_object_unref (priv->udpsrc_v6[i]);
2847 if (priv->udpsrc_mcast_v6[i]) {
2848 if (priv->sinkpad || i == 1) {
2849 gst_element_set_locked_state (priv->udpsrc_mcast_v6[i], FALSE);
2850 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2851 gst_bin_remove (bin, priv->udpsrc_mcast_v6[i]);
2853 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2854 gst_object_unref (priv->udpsrc_mcast_v6[i]);
2858 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
2859 gst_bin_remove (bin, priv->udpsink[i]);
2860 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2861 gst_bin_remove (bin, priv->appsrc[i]);
2862 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
2863 gst_bin_remove (bin, priv->appsink[i]);
2864 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2865 gst_bin_remove (bin, priv->appqueue[i]);
2866 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2867 gst_bin_remove (bin, priv->udpqueue[i]);
2868 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2869 gst_bin_remove (bin, priv->tee[i]);
2870 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2871 gst_bin_remove (bin, priv->funnel[i]);
2873 if (priv->sinkpad || i == 1) {
2874 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2875 gst_object_unref (priv->recv_sink[i]);
2876 priv->recv_sink[i] = NULL;
2879 priv->udpsrc_v4[i] = NULL;
2880 priv->udpsrc_v6[i] = NULL;
2881 priv->udpsrc_mcast_v4[i] = NULL;
2882 priv->udpsrc_mcast_v6[i] = NULL;
2883 priv->udpsink[i] = NULL;
2884 priv->appsrc[i] = NULL;
2885 priv->appsink[i] = NULL;
2886 priv->appqueue[i] = NULL;
2887 priv->udpqueue[i] = NULL;
2888 priv->tee[i] = NULL;
2889 priv->funnel[i] = NULL;
2893 gst_object_unref (priv->send_src[0]);
2894 priv->send_src[0] = NULL;
2897 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2898 gst_object_unref (priv->send_src[1]);
2899 priv->send_src[1] = NULL;
2901 g_object_unref (priv->session);
2902 priv->session = NULL;
2904 gst_caps_unref (priv->caps);
2908 gst_object_unref (priv->srtpenc);
2910 gst_object_unref (priv->srtpdec);
2912 priv->is_joined = FALSE;
2913 g_mutex_unlock (&priv->lock);
2919 g_mutex_unlock (&priv->lock);
2922 transports_not_removed:
2924 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2925 g_mutex_unlock (&priv->lock);
2931 * gst_rtsp_stream_get_rtpinfo:
2932 * @stream: a #GstRTSPStream
2933 * @rtptime: (allow-none): result RTP timestamp
2934 * @seq: (allow-none): result RTP seqnum
2935 * @clock_rate: (allow-none): the clock rate
2936 * @running_time: (allow-none): result running-time
2938 * Retrieve the current rtptime, seq and running-time. This is used to
2939 * construct a RTPInfo reply header.
2941 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2944 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2945 guint * rtptime, guint * seq, guint * clock_rate,
2946 GstClockTime * running_time)
2948 GstRTSPStreamPrivate *priv;
2949 GstStructure *stats;
2950 GObjectClass *payobjclass;
2952 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2954 priv = stream->priv;
2956 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2958 g_mutex_lock (&priv->lock);
2960 /* First try to extract the information from the last buffer on the sinks.
2961 * This will have a more accurate sequence number and timestamp, as between
2962 * the payloader and the sink there can be some queues
2964 if (priv->udpsink[0] || priv->appsink[0]) {
2965 GstSample *last_sample;
2967 if (priv->udpsink[0])
2968 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2970 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2975 GstSegment *segment;
2976 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2978 caps = gst_sample_get_caps (last_sample);
2979 buffer = gst_sample_get_buffer (last_sample);
2980 segment = gst_sample_get_segment (last_sample);
2982 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2984 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2988 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2991 gst_rtp_buffer_unmap (&rtp_buffer);
2995 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2996 GST_BUFFER_TIMESTAMP (buffer));
3000 GstStructure *s = gst_caps_get_structure (caps, 0);
3002 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
3004 if (*clock_rate == 0 && running_time)
3005 *running_time = GST_CLOCK_TIME_NONE;
3007 gst_sample_unref (last_sample);
3011 gst_sample_unref (last_sample);
3016 if (g_object_class_find_property (payobjclass, "stats")) {
3017 g_object_get (priv->payloader, "stats", &stats, NULL);
3022 gst_structure_get_uint (stats, "seqnum", seq);
3025 gst_structure_get_uint (stats, "timestamp", rtptime);
3028 gst_structure_get_clock_time (stats, "running-time", running_time);
3031 gst_structure_get_uint (stats, "clock-rate", clock_rate);
3032 if (*clock_rate == 0 && running_time)
3033 *running_time = GST_CLOCK_TIME_NONE;
3035 gst_structure_free (stats);
3037 if (!g_object_class_find_property (payobjclass, "seqnum") ||
3038 !g_object_class_find_property (payobjclass, "timestamp"))
3042 g_object_get (priv->payloader, "seqnum", seq, NULL);
3045 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
3048 *running_time = GST_CLOCK_TIME_NONE;
3052 g_mutex_unlock (&priv->lock);
3059 GST_WARNING ("Could not get payloader stats");
3060 g_mutex_unlock (&priv->lock);
3066 * gst_rtsp_stream_get_caps:
3067 * @stream: a #GstRTSPStream
3069 * Retrieve the current caps of @stream.
3071 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
3075 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
3077 GstRTSPStreamPrivate *priv;
3080 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3082 priv = stream->priv;
3084 g_mutex_lock (&priv->lock);
3085 if ((result = priv->caps))
3086 gst_caps_ref (result);
3087 g_mutex_unlock (&priv->lock);
3093 * gst_rtsp_stream_recv_rtp:
3094 * @stream: a #GstRTSPStream
3095 * @buffer: (transfer full): a #GstBuffer
3097 * Handle an RTP buffer for the stream. This method is usually called when a
3098 * message has been received from a client using the TCP transport.
3100 * This function takes ownership of @buffer.
3102 * Returns: a GstFlowReturn.
3105 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
3107 GstRTSPStreamPrivate *priv;
3109 GstElement *element;
3111 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3112 priv = stream->priv;
3113 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3114 g_return_val_if_fail (priv->is_joined, FALSE);
3116 g_mutex_lock (&priv->lock);
3117 if (priv->appsrc[0])
3118 element = gst_object_ref (priv->appsrc[0]);
3121 g_mutex_unlock (&priv->lock);
3124 if (priv->appsrc_base_time[0] == -1) {
3125 /* Take current running_time. This timestamp will be put on
3126 * the first buffer of each stream because we are a live source and so we
3127 * timestamp with the running_time. When we are dealing with TCP, we also
3128 * only timestamp the first buffer (using the DISCONT flag) because a server
3129 * typically bursts data, for which we don't want to compensate by speeding
3130 * up the media. The other timestamps will be interpollated from this one
3131 * using the RTP timestamps. */
3132 GST_OBJECT_LOCK (element);
3133 if (GST_ELEMENT_CLOCK (element)) {
3135 GstClockTime base_time;
3137 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3138 base_time = GST_ELEMENT_CAST (element)->base_time;
3140 priv->appsrc_base_time[0] = now - base_time;
3141 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
3142 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3143 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3144 GST_TIME_ARGS (base_time));
3146 GST_OBJECT_UNLOCK (element);
3149 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3150 gst_object_unref (element);
3158 * gst_rtsp_stream_recv_rtcp:
3159 * @stream: a #GstRTSPStream
3160 * @buffer: (transfer full): a #GstBuffer
3162 * Handle an RTCP buffer for the stream. This method is usually called when a
3163 * message has been received from a client using the TCP transport.
3165 * This function takes ownership of @buffer.
3167 * Returns: a GstFlowReturn.
3170 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
3172 GstRTSPStreamPrivate *priv;
3174 GstElement *element;
3176 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3177 priv = stream->priv;
3178 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3180 if (!priv->is_joined) {
3181 gst_buffer_unref (buffer);
3182 return GST_FLOW_NOT_LINKED;
3184 g_mutex_lock (&priv->lock);
3185 if (priv->appsrc[1])
3186 element = gst_object_ref (priv->appsrc[1]);
3189 g_mutex_unlock (&priv->lock);
3192 if (priv->appsrc_base_time[1] == -1) {
3193 /* Take current running_time. This timestamp will be put on
3194 * the first buffer of each stream because we are a live source and so we
3195 * timestamp with the running_time. When we are dealing with TCP, we also
3196 * only timestamp the first buffer (using the DISCONT flag) because a server
3197 * typically bursts data, for which we don't want to compensate by speeding
3198 * up the media. The other timestamps will be interpollated from this one
3199 * using the RTP timestamps. */
3200 GST_OBJECT_LOCK (element);
3201 if (GST_ELEMENT_CLOCK (element)) {
3203 GstClockTime base_time;
3205 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3206 base_time = GST_ELEMENT_CAST (element)->base_time;
3208 priv->appsrc_base_time[1] = now - base_time;
3209 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3210 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3211 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3212 GST_TIME_ARGS (base_time));
3214 GST_OBJECT_UNLOCK (element);
3217 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3218 gst_object_unref (element);
3221 gst_buffer_unref (buffer);
3226 /* must be called with lock */
3228 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3231 GstRTSPStreamPrivate *priv = stream->priv;
3232 const GstRTSPTransport *tr;
3234 tr = gst_rtsp_stream_transport_get_transport (trans);
3236 switch (tr->lower_transport) {
3237 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3238 case GST_RTSP_LOWER_TRANS_UDP:
3244 dest = tr->destination;
3245 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3249 } else if (priv->client_side) {
3250 /* In client side mode the 'destination' is the RTSP server, so send
3252 min = tr->server_port.min;
3253 max = tr->server_port.max;
3255 min = tr->client_port.min;
3256 max = tr->client_port.max;
3261 GST_INFO ("setting ttl-mc %d", ttl);
3262 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3263 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3265 GST_INFO ("adding %s:%d-%d", dest, min, max);
3266 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3267 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3268 priv->transports = g_list_prepend (priv->transports, trans);
3270 GST_INFO ("removing %s:%d-%d", dest, min, max);
3271 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3272 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3273 priv->transports = g_list_remove (priv->transports, trans);
3275 priv->transports_cookie++;
3278 case GST_RTSP_LOWER_TRANS_TCP:
3280 GST_INFO ("adding TCP %s", tr->destination);
3281 priv->transports = g_list_prepend (priv->transports, trans);
3283 GST_INFO ("removing TCP %s", tr->destination);
3284 priv->transports = g_list_remove (priv->transports, trans);
3286 priv->transports_cookie++;
3289 goto unknown_transport;
3296 GST_INFO ("Unknown transport %d", tr->lower_transport);
3303 * gst_rtsp_stream_add_transport:
3304 * @stream: a #GstRTSPStream
3305 * @trans: (transfer none): a #GstRTSPStreamTransport
3307 * Add the transport in @trans to @stream. The media of @stream will
3308 * then also be send to the values configured in @trans.
3310 * @stream must be joined to a bin.
3312 * @trans must contain a valid #GstRTSPTransport.
3314 * Returns: %TRUE if @trans was added
3317 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3318 GstRTSPStreamTransport * trans)
3320 GstRTSPStreamPrivate *priv;
3323 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3324 priv = stream->priv;
3325 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3326 g_return_val_if_fail (priv->is_joined, FALSE);
3328 g_mutex_lock (&priv->lock);
3329 res = update_transport (stream, trans, TRUE);
3330 g_mutex_unlock (&priv->lock);
3336 * gst_rtsp_stream_remove_transport:
3337 * @stream: a #GstRTSPStream
3338 * @trans: (transfer none): a #GstRTSPStreamTransport
3340 * Remove the transport in @trans from @stream. The media of @stream will
3341 * not be sent to the values configured in @trans.
3343 * @stream must be joined to a bin.
3345 * @trans must contain a valid #GstRTSPTransport.
3347 * Returns: %TRUE if @trans was removed
3350 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3351 GstRTSPStreamTransport * trans)
3353 GstRTSPStreamPrivate *priv;
3356 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3357 priv = stream->priv;
3358 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3359 g_return_val_if_fail (priv->is_joined, FALSE);
3361 g_mutex_lock (&priv->lock);
3362 res = update_transport (stream, trans, FALSE);
3363 g_mutex_unlock (&priv->lock);
3369 * gst_rtsp_stream_update_crypto:
3370 * @stream: a #GstRTSPStream
3372 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3374 * Update the new crypto information for @ssrc in @stream. If information
3375 * for @ssrc did not exist, it will be added. If information
3376 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3377 * be removed from @stream.
3379 * Returns: %TRUE if @crypto could be updated
3382 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3383 guint ssrc, GstCaps * crypto)
3385 GstRTSPStreamPrivate *priv;
3387 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3388 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3390 priv = stream->priv;
3392 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3394 g_mutex_lock (&priv->lock);
3396 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3397 gst_caps_ref (crypto));
3399 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3400 g_mutex_unlock (&priv->lock);
3406 * gst_rtsp_stream_get_rtp_socket:
3407 * @stream: a #GstRTSPStream
3408 * @family: the socket family
3410 * Get the RTP socket from @stream for a @family.
3412 * @stream must be joined to a bin.
3414 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3415 * socket could be allocated for @family. Unref after usage
3418 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3420 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3424 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3425 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3426 family == G_SOCKET_FAMILY_IPV6, NULL);
3427 g_return_val_if_fail (priv->udpsink[0], NULL);
3429 if (family == G_SOCKET_FAMILY_IPV6)
3434 g_object_get (priv->udpsink[0], name, &socket, NULL);
3440 * gst_rtsp_stream_get_rtcp_socket:
3441 * @stream: a #GstRTSPStream
3442 * @family: the socket family
3444 * Get the RTCP socket from @stream for a @family.
3446 * @stream must be joined to a bin.
3448 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3449 * socket could be allocated for @family. Unref after usage
3452 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3454 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3458 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3459 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3460 family == G_SOCKET_FAMILY_IPV6, NULL);
3461 g_return_val_if_fail (priv->udpsink[1], NULL);
3463 if (family == G_SOCKET_FAMILY_IPV6)
3468 g_object_get (priv->udpsink[1], name, &socket, NULL);
3474 * gst_rtsp_stream_set_seqnum:
3475 * @stream: a #GstRTSPStream
3476 * @seqnum: a new sequence number
3478 * Configure the sequence number in the payloader of @stream to @seqnum.
3481 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3483 GstRTSPStreamPrivate *priv;
3485 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3487 priv = stream->priv;
3489 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3493 * gst_rtsp_stream_get_seqnum:
3494 * @stream: a #GstRTSPStream
3496 * Get the configured sequence number in the payloader of @stream.
3498 * Returns: the sequence number of the payloader.
3501 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3503 GstRTSPStreamPrivate *priv;
3506 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3508 priv = stream->priv;
3510 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3516 * gst_rtsp_stream_transport_filter:
3517 * @stream: a #GstRTSPStream
3518 * @func: (scope call) (allow-none): a callback
3519 * @user_data: (closure): user data passed to @func
3521 * Call @func for each transport managed by @stream. The result value of @func
3522 * determines what happens to the transport. @func will be called with @stream
3523 * locked so no further actions on @stream can be performed from @func.
3525 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3528 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3530 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3531 * will also be added with an additional ref to the result #GList of this
3534 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3536 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3537 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3538 * element in the #GList should be unreffed before the list is freed.
3541 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3542 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3544 GstRTSPStreamPrivate *priv;
3545 GList *result, *walk, *next;
3546 GHashTable *visited = NULL;
3549 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3551 priv = stream->priv;
3555 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3557 g_mutex_lock (&priv->lock);
3559 cookie = priv->transports_cookie;
3560 for (walk = priv->transports; walk; walk = next) {
3561 GstRTSPStreamTransport *trans = walk->data;
3562 GstRTSPFilterResult res;
3565 next = g_list_next (walk);
3568 /* only visit each transport once */
3569 if (g_hash_table_contains (visited, trans))
3572 g_hash_table_add (visited, g_object_ref (trans));
3573 g_mutex_unlock (&priv->lock);
3575 res = func (stream, trans, user_data);
3577 g_mutex_lock (&priv->lock);
3579 res = GST_RTSP_FILTER_REF;
3581 changed = (cookie != priv->transports_cookie);
3584 case GST_RTSP_FILTER_REMOVE:
3585 update_transport (stream, trans, FALSE);
3587 case GST_RTSP_FILTER_REF:
3588 result = g_list_prepend (result, g_object_ref (trans));
3590 case GST_RTSP_FILTER_KEEP:
3597 g_mutex_unlock (&priv->lock);
3600 g_hash_table_unref (visited);
3605 static GstPadProbeReturn
3606 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3608 GstRTSPStreamPrivate *priv;
3609 GstRTSPStream *stream;
3612 priv = stream->priv;
3614 GST_DEBUG_OBJECT (pad, "now blocking");
3616 g_mutex_lock (&priv->lock);
3617 priv->blocking = TRUE;
3618 g_mutex_unlock (&priv->lock);
3620 gst_element_post_message (priv->payloader,
3621 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3622 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3624 return GST_PAD_PROBE_OK;
3628 * gst_rtsp_stream_set_blocked:
3629 * @stream: a #GstRTSPStream
3630 * @blocked: boolean indicating we should block or unblock
3632 * Blocks or unblocks the dataflow on @stream.
3634 * Returns: %TRUE on success
3637 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3639 GstRTSPStreamPrivate *priv;
3641 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3643 priv = stream->priv;
3645 g_mutex_lock (&priv->lock);
3647 priv->blocking = FALSE;
3648 if (priv->blocked_id == 0) {
3649 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3650 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3651 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3652 g_object_ref (stream), g_object_unref);
3655 if (priv->blocked_id != 0) {
3656 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3657 priv->blocked_id = 0;
3658 priv->blocking = FALSE;
3661 g_mutex_unlock (&priv->lock);
3667 * gst_rtsp_stream_is_blocking:
3668 * @stream: a #GstRTSPStream
3670 * Check if @stream is blocking on a #GstBuffer.
3672 * Returns: %TRUE if @stream is blocking
3675 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3677 GstRTSPStreamPrivate *priv;
3680 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3682 priv = stream->priv;
3684 g_mutex_lock (&priv->lock);
3685 result = priv->blocking;
3686 g_mutex_unlock (&priv->lock);
3692 * gst_rtsp_stream_query_position:
3693 * @stream: a #GstRTSPStream
3695 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3696 * the RTP parts of the pipeline and not the RTCP parts.
3698 * Returns: %TRUE if the position could be queried
3701 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3703 GstRTSPStreamPrivate *priv;
3707 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3709 priv = stream->priv;
3711 g_mutex_lock (&priv->lock);
3712 /* depending on the transport type, it should query corresponding sink */
3713 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3714 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3715 sink = priv->udpsink[0];
3717 sink = priv->appsink[0];
3720 gst_object_ref (sink);
3721 g_mutex_unlock (&priv->lock);
3726 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3727 gst_object_unref (sink);
3733 * gst_rtsp_stream_query_stop:
3734 * @stream: a #GstRTSPStream
3736 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3737 * the RTP parts of the pipeline and not the RTCP parts.
3739 * Returns: %TRUE if the stop could be queried
3742 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3744 GstRTSPStreamPrivate *priv;
3749 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3751 priv = stream->priv;
3753 g_mutex_lock (&priv->lock);
3754 /* depending on the transport type, it should query corresponding sink */
3755 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3756 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3757 sink = priv->udpsink[0];
3759 sink = priv->appsink[0];
3762 gst_object_ref (sink);
3763 g_mutex_unlock (&priv->lock);
3768 query = gst_query_new_segment (GST_FORMAT_TIME);
3769 if ((ret = gst_element_query (sink, query))) {
3772 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3773 if (format != GST_FORMAT_TIME)
3776 gst_query_unref (query);
3777 gst_object_unref (sink);