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;
1218 if (*server_addr_out)
1219 gst_rtsp_address_free (*server_addr_out);
1221 /* try to allocate 2 UDP ports, the RTP port should be an even
1222 * number and the RTCP port should be the next (uneven) port */
1225 if (rtp_socket == NULL) {
1226 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
1227 G_SOCKET_PROTOCOL_UDP, NULL);
1229 goto no_udp_protocol;
1233 GstRTSPAddressFlags flags;
1235 if (transport == GST_RTSP_LOWER_TRANS_UDP &&
1236 gst_rtsp_address_pool_has_unicast_addresses (pool))
1237 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
1238 else if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST)
1239 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST;
1244 rejected_addresses = g_list_prepend (rejected_addresses, addr);
1246 if (family == G_SOCKET_FAMILY_IPV6)
1247 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
1249 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
1251 if (ct->destination && transport == GST_RTSP_LOWER_TRANS_UDP_MCAST
1252 && use_client_settings)
1253 gst_rtsp_address_pool_reserve_address (pool, ct->destination,
1254 ct->port.min, 2, ct->ttl, &addr);
1256 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1261 tmp_rtp = addr->port;
1263 g_clear_object (&inetaddr);
1264 inetaddr = g_inet_address_new_from_string (addr->address);
1266 /* Don't bind to multicast addresses, this does not work on
1267 * Windows. You're supposed to bind to ANY and then join the
1268 * multicast group, which udpsrc/sink does for us already.
1270 if (g_inet_address_get_is_multicast (inetaddr)) {
1271 g_object_unref (inetaddr);
1272 inetaddr = g_inet_address_new_any (family);
1281 if (inetaddr == NULL)
1282 inetaddr = g_inet_address_new_any (family);
1285 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1286 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1287 g_object_unref (rtp_sockaddr);
1290 g_object_unref (rtp_sockaddr);
1292 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1293 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1294 g_clear_object (&rtp_sockaddr);
1299 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1300 g_object_unref (rtp_sockaddr);
1302 /* check if port is even */
1303 if ((tmp_rtp & 1) != 0) {
1304 /* port not even, close and allocate another */
1306 g_clear_object (&rtp_socket);
1311 tmp_rtcp = tmp_rtp + 1;
1313 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1314 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1315 g_object_unref (rtcp_sockaddr);
1316 g_clear_object (&rtp_socket);
1319 g_object_unref (rtcp_sockaddr);
1322 addr_str = g_inet_address_to_string (inetaddr);
1324 addr_str = addr->address;
1325 g_clear_object (&inetaddr);
1327 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1328 rtcp_socket, family, addr_str, tmp_rtp, tmp_rtcp, transport)) {
1331 goto no_udp_protocol;
1337 play_udpsources_one_family (stream, udpsrc_out, family);
1339 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1340 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1342 /* this should not happen... */
1343 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1346 /* set RTP and RTCP sockets */
1347 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1349 server_port_out->min = rtpport;
1350 server_port_out->max = rtcpport;
1352 *server_addr_out = addr;
1353 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1355 g_object_unref (rtp_socket);
1356 g_object_unref (rtcp_socket);
1380 g_object_unref (inetaddr);
1381 g_list_free_full (rejected_addresses,
1382 (GDestroyNotify) gst_rtsp_address_free);
1384 gst_rtsp_address_free (addr);
1386 g_object_unref (rtp_socket);
1388 g_object_unref (rtcp_socket);
1394 * gst_rtsp_stream_allocate_udp_sockets:
1395 * @stream: a #GstRTSPStream
1396 * @family: protocol family
1397 * @transport_method: transport method
1399 * Allocates RTP and RTCP ports.
1401 * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
1404 gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
1405 GSocketFamily family, GstRTSPTransport * ct, gboolean use_client_settings)
1407 GstRTSPStreamPrivate *priv;
1408 gboolean result = FALSE;
1409 GstRTSPLowerTrans transport = ct->lower_transport;
1411 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1412 priv = stream->priv;
1413 g_return_val_if_fail (priv->is_joined, FALSE);
1415 g_mutex_lock (&priv->lock);
1417 if (family == G_SOCKET_FAMILY_IPV4) {
1418 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1419 if (priv->have_ipv4_mcast)
1421 priv->have_ipv4_mcast =
1422 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4,
1423 priv->udpsrc_mcast_v4, &priv->server_port_v4, ct, &priv->addr_v4,
1424 use_client_settings);
1427 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1428 &priv->server_port_v4, ct, &priv->server_addr_v4,
1429 use_client_settings);
1432 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1433 if (priv->have_ipv6_mcast)
1435 priv->have_ipv6_mcast =
1436 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6,
1437 priv->udpsrc_mcast_v6, &priv->server_port_v6, ct, &priv->addr_v6,
1438 use_client_settings);
1440 if (priv->have_ipv6)
1443 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1444 &priv->server_port_v6, ct, &priv->server_addr_v6,
1445 use_client_settings);
1450 result = priv->have_ipv4 || priv->have_ipv4_mcast || priv->have_ipv6 ||
1451 priv->have_ipv6_mcast;
1453 g_mutex_unlock (&priv->lock);
1459 * gst_rtsp_stream_set_client_side:
1460 * @stream: a #GstRTSPStream
1461 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1462 * an RTSP connection.
1464 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1465 * streams to an RTSP server via RECORD. This has the practical effect
1466 * of changing which UDP port numbers are used when setting up the local
1467 * side of the stream sending to be either the 'server' or 'client' pair
1468 * of a configured UDP transport.
1471 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1473 GstRTSPStreamPrivate *priv;
1475 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1476 priv = stream->priv;
1477 g_mutex_lock (&priv->lock);
1478 priv->client_side = client_side;
1479 g_mutex_unlock (&priv->lock);
1483 * gst_rtsp_stream_set_client_side:
1484 * @stream: a #GstRTSPStream
1486 * See gst_rtsp_stream_set_client_side()
1488 * Returns: TRUE if this #GstRTSPStream is client-side.
1491 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1493 GstRTSPStreamPrivate *priv;
1496 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1498 priv = stream->priv;
1499 g_mutex_lock (&priv->lock);
1500 ret = priv->client_side;
1501 g_mutex_unlock (&priv->lock);
1507 * gst_rtsp_stream_get_server_port:
1508 * @stream: a #GstRTSPStream
1509 * @server_port: (out): result server port
1510 * @family: the port family to get
1512 * Fill @server_port with the port pair used by the server. This function can
1513 * only be called when @stream has been joined.
1516 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1517 GstRTSPRange * server_port, GSocketFamily family)
1519 GstRTSPStreamPrivate *priv;
1521 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1522 priv = stream->priv;
1523 g_return_if_fail (priv->is_joined);
1525 g_mutex_lock (&priv->lock);
1526 if (family == G_SOCKET_FAMILY_IPV4) {
1528 *server_port = priv->server_port_v4;
1531 *server_port = priv->server_port_v6;
1533 g_mutex_unlock (&priv->lock);
1537 * gst_rtsp_stream_get_rtpsession:
1538 * @stream: a #GstRTSPStream
1540 * Get the RTP session of this stream.
1542 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1545 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1547 GstRTSPStreamPrivate *priv;
1550 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1552 priv = stream->priv;
1554 g_mutex_lock (&priv->lock);
1555 if ((session = priv->session))
1556 g_object_ref (session);
1557 g_mutex_unlock (&priv->lock);
1563 * gst_rtsp_stream_get_ssrc:
1564 * @stream: a #GstRTSPStream
1565 * @ssrc: (out): result ssrc
1567 * Get the SSRC used by the RTP session of this stream. This function can only
1568 * be called when @stream has been joined.
1571 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1573 GstRTSPStreamPrivate *priv;
1575 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1576 priv = stream->priv;
1577 g_return_if_fail (priv->is_joined);
1579 g_mutex_lock (&priv->lock);
1580 if (ssrc && priv->session)
1581 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1582 g_mutex_unlock (&priv->lock);
1586 * gst_rtsp_stream_set_retransmission_time:
1587 * @stream: a #GstRTSPStream
1588 * @time: a #GstClockTime
1590 * Set the amount of time to store retransmission packets.
1593 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1596 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1598 g_mutex_lock (&stream->priv->lock);
1599 stream->priv->rtx_time = time;
1600 if (stream->priv->rtxsend)
1601 g_object_set (stream->priv->rtxsend, "max-size-time",
1602 GST_TIME_AS_MSECONDS (time), NULL);
1603 g_mutex_unlock (&stream->priv->lock);
1607 * gst_rtsp_stream_get_retransmission_time:
1608 * @stream: a #GstRTSPStream
1610 * Get the amount of time to store retransmission data.
1612 * Returns: the amount of time to store retransmission data.
1615 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1619 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1621 g_mutex_lock (&stream->priv->lock);
1622 ret = stream->priv->rtx_time;
1623 g_mutex_unlock (&stream->priv->lock);
1629 * gst_rtsp_stream_set_retransmission_pt:
1630 * @stream: a #GstRTSPStream
1633 * Set the payload type (pt) for retransmission of this stream.
1636 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1638 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1640 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1642 g_mutex_lock (&stream->priv->lock);
1643 stream->priv->rtx_pt = rtx_pt;
1644 if (stream->priv->rtxsend) {
1645 guint pt = gst_rtsp_stream_get_pt (stream);
1646 gchar *pt_s = g_strdup_printf ("%d", pt);
1647 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1648 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1649 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1651 gst_structure_free (rtx_pt_map);
1653 g_mutex_unlock (&stream->priv->lock);
1657 * gst_rtsp_stream_get_retransmission_pt:
1658 * @stream: a #GstRTSPStream
1660 * Get the payload-type used for retransmission of this stream
1662 * Returns: The retransmission PT.
1665 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1669 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1671 g_mutex_lock (&stream->priv->lock);
1672 rtx_pt = stream->priv->rtx_pt;
1673 g_mutex_unlock (&stream->priv->lock);
1679 * gst_rtsp_stream_set_buffer_size:
1680 * @stream: a #GstRTSPStream
1681 * @size: the buffer size
1683 * Set the size of the UDP transmission buffer (in bytes)
1684 * Needs to be set before the stream is joined to a bin.
1689 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1691 g_mutex_lock (&stream->priv->lock);
1692 stream->priv->buffer_size = size;
1693 g_mutex_unlock (&stream->priv->lock);
1697 * gst_rtsp_stream_get_buffer_size:
1698 * @stream: a #GstRTSPStream
1700 * Get the size of the UDP transmission buffer (in bytes)
1702 * Returns: the size of the UDP TX buffer
1707 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1711 g_mutex_lock (&stream->priv->lock);
1712 buffer_size = stream->priv->buffer_size;
1713 g_mutex_unlock (&stream->priv->lock);
1718 /* executed from streaming thread */
1720 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1722 GstRTSPStreamPrivate *priv = stream->priv;
1723 GstCaps *newcaps, *oldcaps;
1725 newcaps = gst_pad_get_current_caps (pad);
1727 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1730 g_mutex_lock (&priv->lock);
1731 oldcaps = priv->caps;
1732 priv->caps = newcaps;
1733 g_mutex_unlock (&priv->lock);
1736 gst_caps_unref (oldcaps);
1740 dump_structure (const GstStructure * s)
1744 sstr = gst_structure_to_string (s);
1745 GST_INFO ("structure: %s", sstr);
1749 static GstRTSPStreamTransport *
1750 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1752 GstRTSPStreamPrivate *priv = stream->priv;
1754 GstRTSPStreamTransport *result = NULL;
1759 if (rtcp_from == NULL)
1762 tmp = g_strrstr (rtcp_from, ":");
1766 port = atoi (tmp + 1);
1767 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1769 g_mutex_lock (&priv->lock);
1770 GST_INFO ("finding %s:%d in %d transports", dest, port,
1771 g_list_length (priv->transports));
1773 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1774 GstRTSPStreamTransport *trans = walk->data;
1775 const GstRTSPTransport *tr;
1778 tr = gst_rtsp_stream_transport_get_transport (trans);
1780 if (priv->client_side) {
1781 /* In client side mode the 'destination' is the RTSP server, so send
1783 min = tr->server_port.min;
1784 max = tr->server_port.max;
1786 min = tr->client_port.min;
1787 max = tr->client_port.max;
1790 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1796 g_object_ref (result);
1797 g_mutex_unlock (&priv->lock);
1804 static GstRTSPStreamTransport *
1805 check_transport (GObject * source, GstRTSPStream * stream)
1807 GstStructure *stats;
1808 GstRTSPStreamTransport *trans;
1810 /* see if we have a stream to match with the origin of the RTCP packet */
1811 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1812 if (trans == NULL) {
1813 g_object_get (source, "stats", &stats, NULL);
1815 const gchar *rtcp_from;
1817 dump_structure (stats);
1819 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1820 if ((trans = find_transport (stream, rtcp_from))) {
1821 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1823 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1826 gst_structure_free (stats);
1834 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1836 GstRTSPStreamTransport *trans;
1838 GST_INFO ("%p: new source %p", stream, source);
1840 trans = check_transport (source, stream);
1843 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1847 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1849 GST_INFO ("%p: new SDES %p", stream, source);
1853 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1855 GstRTSPStreamTransport *trans;
1857 trans = check_transport (source, stream);
1860 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1861 gst_rtsp_stream_transport_keep_alive (trans);
1865 GstStructure *stats;
1866 g_object_get (source, "stats", &stats, NULL);
1868 dump_structure (stats);
1869 gst_structure_free (stats);
1876 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1878 GST_INFO ("%p: source %p bye", stream, source);
1882 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1884 GstRTSPStreamTransport *trans;
1886 GST_INFO ("%p: source %p bye timeout", stream, source);
1888 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1889 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1890 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1895 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1897 GstRTSPStreamTransport *trans;
1899 GST_INFO ("%p: source %p timeout", stream, source);
1901 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1902 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1903 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1908 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1910 GST_INFO ("%p: new sender source %p", stream, source);
1913 GstStructure *stats;
1914 g_object_get (source, "stats", &stats, NULL);
1916 dump_structure (stats);
1917 gst_structure_free (stats);
1924 on_sender_ssrc_active (GObject * session, GObject * source,
1925 GstRTSPStream * stream)
1929 GstStructure *stats;
1930 g_object_get (source, "stats", &stats, NULL);
1932 dump_structure (stats);
1933 gst_structure_free (stats);
1940 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1943 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1944 g_list_free (priv->tr_cache_rtp);
1945 priv->tr_cache_rtp = NULL;
1947 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1948 g_list_free (priv->tr_cache_rtcp);
1949 priv->tr_cache_rtcp = NULL;
1953 static GstFlowReturn
1954 handle_new_sample (GstAppSink * sink, gpointer user_data)
1956 GstRTSPStreamPrivate *priv;
1960 GstRTSPStream *stream;
1963 sample = gst_app_sink_pull_sample (sink);
1967 stream = (GstRTSPStream *) user_data;
1968 priv = stream->priv;
1969 buffer = gst_sample_get_buffer (sample);
1971 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1973 g_mutex_lock (&priv->lock);
1975 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1976 clear_tr_cache (priv, is_rtp);
1977 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1978 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1979 priv->tr_cache_rtp =
1980 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1982 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1985 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1986 clear_tr_cache (priv, is_rtp);
1987 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1988 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1989 priv->tr_cache_rtcp =
1990 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1992 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1995 g_mutex_unlock (&priv->lock);
1998 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1999 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2000 gst_rtsp_stream_transport_send_rtp (tr, buffer);
2003 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
2004 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2005 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
2008 gst_sample_unref (sample);
2013 static GstAppSinkCallbacks sink_cb = {
2014 NULL, /* not interested in EOS */
2015 NULL, /* not interested in preroll samples */
2020 get_rtp_encoder (GstRTSPStream * stream, guint session)
2022 GstRTSPStreamPrivate *priv = stream->priv;
2024 if (priv->srtpenc == NULL) {
2027 name = g_strdup_printf ("srtpenc_%u", session);
2028 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
2031 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
2033 return gst_object_ref (priv->srtpenc);
2037 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2039 GstRTSPStreamPrivate *priv = stream->priv;
2040 GstElement *oldenc, *enc;
2044 if (priv->idx != session)
2047 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
2049 oldenc = priv->srtpenc;
2050 enc = get_rtp_encoder (stream, session);
2051 name = g_strdup_printf ("rtp_sink_%d", session);
2052 pad = gst_element_get_request_pad (enc, name);
2054 gst_object_unref (pad);
2057 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
2064 request_rtcp_encoder (GstElement * rtpbin, guint session,
2065 GstRTSPStream * stream)
2067 GstRTSPStreamPrivate *priv = stream->priv;
2068 GstElement *oldenc, *enc;
2072 if (priv->idx != session)
2075 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
2077 oldenc = priv->srtpenc;
2078 enc = get_rtp_encoder (stream, session);
2079 name = g_strdup_printf ("rtcp_sink_%d", session);
2080 pad = gst_element_get_request_pad (enc, name);
2082 gst_object_unref (pad);
2085 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
2092 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2094 GstRTSPStreamPrivate *priv = stream->priv;
2097 GST_DEBUG ("request key %08x", ssrc);
2099 g_mutex_lock (&priv->lock);
2100 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
2101 gst_caps_ref (caps);
2102 g_mutex_unlock (&priv->lock);
2108 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
2109 GstRTSPStream * stream)
2111 GstRTSPStreamPrivate *priv = stream->priv;
2113 if (priv->idx != session)
2116 if (priv->srtpdec == NULL) {
2119 name = g_strdup_printf ("srtpdec_%u", session);
2120 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2123 g_signal_connect (priv->srtpdec, "request-key",
2124 (GCallback) request_key, stream);
2126 return gst_object_ref (priv->srtpdec);
2130 * gst_rtsp_stream_request_aux_sender:
2131 * @stream: a #GstRTSPStream
2132 * @sessid: the session id
2134 * Creating a rtxsend bin
2136 * Returns: (transfer full): a #GstElement.
2141 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2145 GstStructure *pt_map;
2150 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2152 pt = gst_rtsp_stream_get_pt (stream);
2153 pt_s = g_strdup_printf ("%u", pt);
2154 rtx_pt = stream->priv->rtx_pt;
2156 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2158 bin = gst_bin_new (NULL);
2159 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2160 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2161 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2162 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2163 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2165 gst_structure_free (pt_map);
2166 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2168 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2169 name = g_strdup_printf ("src_%u", sessid);
2170 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2172 gst_object_unref (pad);
2174 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2175 name = g_strdup_printf ("sink_%u", sessid);
2176 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2178 gst_object_unref (pad);
2184 * gst_rtsp_stream_set_pt_map:
2185 * @stream: a #GstRTSPStream
2189 * Configure a pt map between @pt and @caps.
2192 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2194 GstRTSPStreamPrivate *priv = stream->priv;
2196 g_mutex_lock (&priv->lock);
2197 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2198 g_mutex_unlock (&priv->lock);
2202 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2203 GstRTSPStream * stream)
2205 GstRTSPStreamPrivate *priv = stream->priv;
2206 GstCaps *caps = NULL;
2208 g_mutex_lock (&priv->lock);
2210 if (priv->idx == session) {
2211 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2213 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2214 gst_caps_ref (caps);
2216 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2220 g_mutex_unlock (&priv->lock);
2226 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2228 GstRTSPStreamPrivate *priv = stream->priv;
2230 GstPadLinkReturn ret;
2233 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2234 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2236 name = gst_pad_get_name (pad);
2237 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2243 if (priv->idx != sessid)
2246 if (gst_pad_is_linked (priv->sinkpad)) {
2247 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2248 GST_DEBUG_PAD_NAME (priv->sinkpad));
2252 /* link the RTP pad to the session manager, it should not really fail unless
2253 * this is not really an RTP pad */
2254 ret = gst_pad_link (pad, priv->sinkpad);
2255 if (ret != GST_PAD_LINK_OK)
2257 priv->recv_rtp_src = gst_object_ref (pad);
2264 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2265 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2270 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2271 GstRTSPStream * stream)
2273 /* TODO: What to do here other than this? */
2274 GST_DEBUG ("Stream %p: Got EOS", stream);
2275 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2278 /* must be called with lock */
2280 create_sender_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2282 GstRTSPStreamPrivate *priv;
2283 GstPad *pad, *sinkpad = NULL;
2284 gboolean is_tcp = FALSE, is_udp = FALSE;
2287 priv = stream->priv;
2289 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2290 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2291 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2293 if (is_udp && !create_and_configure_udpsinks (stream))
2294 goto no_udp_protocol;
2296 for (i = 0; i < 2; i++) {
2297 GstPad *teepad, *queuepad;
2298 /* For the sender we create this bit of pipeline for both
2299 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2300 * we need to add a queue before appsink and udpsink to make
2301 * the pipeline not block. For the TCP case, we want to pump
2302 * client as fast as possible anyway. This pipeline is used
2303 * when both TCP and UDP are present.
2305 * .--------. .-----. .---------. .---------.
2306 * | rtpbin | | tee | | queue | | udpsink |
2307 * | send->sink src->sink src->sink |
2308 * '--------' | | '---------' '---------'
2309 * | | .---------. .---------.
2310 * | | | queue | | appsink |
2311 * | src->sink src->sink |
2312 * '-----' '---------' '---------'
2314 * When only UDP or only TCP is allowed, we skip the tee and queue
2315 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2318 /* Only link the RTP send src if we're going to send RTP, link
2319 * the RTCP send src always */
2320 if (priv->srcpad || i == 1) {
2323 gst_bin_add (bin, priv->udpsink[i]);
2324 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2329 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2330 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2331 gst_bin_add (bin, priv->appsink[i]);
2332 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2333 &sink_cb, stream, NULL);
2336 if (is_udp && is_tcp) {
2337 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2339 /* make tee for RTP/RTCP */
2340 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2341 gst_bin_add (bin, priv->tee[i]);
2343 /* and link to rtpbin send pad */
2344 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2345 gst_pad_link (priv->send_src[i], pad);
2346 gst_object_unref (pad);
2348 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2349 g_object_set (priv->udpqueue[i], "max-size-buffers",
2350 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2352 gst_bin_add (bin, priv->udpqueue[i]);
2353 /* link tee to udpqueue */
2354 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2355 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2356 gst_pad_link (teepad, pad);
2357 gst_object_unref (pad);
2358 gst_object_unref (teepad);
2360 /* link udpqueue to udpsink */
2361 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2362 gst_pad_link (queuepad, sinkpad);
2363 gst_object_unref (queuepad);
2364 gst_object_unref (sinkpad);
2367 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2368 g_object_set (priv->appqueue[i], "max-size-buffers",
2369 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2371 gst_bin_add (bin, priv->appqueue[i]);
2372 /* and link tee to appqueue */
2373 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2374 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2375 gst_pad_link (teepad, pad);
2376 gst_object_unref (pad);
2377 gst_object_unref (teepad);
2379 /* and link appqueue to appsink */
2380 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2381 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2382 gst_pad_link (queuepad, pad);
2383 gst_object_unref (pad);
2384 gst_object_unref (queuepad);
2385 } else if (is_tcp) {
2386 /* only appsink needed, link it to the session */
2387 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2388 gst_pad_link (priv->send_src[i], pad);
2389 gst_object_unref (pad);
2391 /* when its only TCP, we need to set sync and preroll to FALSE
2392 * for the sink to avoid deadlock. And this is only needed for
2393 * sink used for RTCP data, not the RTP data. */
2395 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2397 /* else only udpsink needed, link it to the session */
2398 gst_pad_link (priv->send_src[i], sinkpad);
2399 gst_object_unref (sinkpad);
2403 /* check if we need to set to a special state */
2404 if (state != GST_STATE_NULL) {
2405 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2406 gst_element_set_state (priv->udpsink[i], state);
2407 if (priv->appsink[i] && (priv->srcpad || i == 1))
2408 gst_element_set_state (priv->appsink[i], state);
2409 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2410 gst_element_set_state (priv->appqueue[i], state);
2411 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2412 gst_element_set_state (priv->udpqueue[i], state);
2413 if (priv->tee[i] && (priv->srcpad || i == 1))
2414 gst_element_set_state (priv->tee[i], state);
2427 /* must be called with lock */
2429 create_receiver_part (GstRTSPStream * stream, GstBin * bin, GstState state)
2431 GstRTSPStreamPrivate *priv;
2432 GstPad *pad, *selpad;
2436 priv = stream->priv;
2438 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2440 for (i = 0; i < 2; i++) {
2441 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2442 * RTCP sink always */
2443 if (priv->sinkpad || i == 1) {
2444 /* For the receiver we create this bit of pipeline for both
2445 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2446 * and it is all funneled into the rtpbin receive pad.
2448 * .--------. .--------. .--------.
2449 * | udpsrc | | funnel | | rtpbin |
2450 * | src->sink src->sink |
2451 * '--------' | | '--------'
2455 * '--------' '--------'
2457 /* make funnel for the RTP/RTCP receivers */
2458 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2459 gst_bin_add (bin, priv->funnel[i]);
2461 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2462 gst_pad_link (pad, priv->recv_sink[i]);
2463 gst_object_unref (pad);
2465 if (priv->udpsrc_v4[i]) {
2467 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2468 * values. This is only relevant for PLAY pipelines */
2469 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2470 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2473 gst_bin_add (bin, priv->udpsrc_v4[i]);
2475 /* and link to the funnel v4 */
2476 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2477 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2478 gst_pad_link (pad, selpad);
2479 gst_object_unref (pad);
2480 gst_object_unref (selpad);
2483 if (priv->udpsrc_v6[i]) {
2485 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2486 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2488 gst_bin_add (bin, priv->udpsrc_v6[i]);
2490 /* and link to the funnel v6 */
2491 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2492 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2493 gst_pad_link (pad, selpad);
2494 gst_object_unref (pad);
2495 gst_object_unref (selpad);
2499 /* make and add appsrc */
2500 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2501 priv->appsrc_base_time[i] = -1;
2502 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2503 gst_bin_add (bin, priv->appsrc[i]);
2504 /* and link to the funnel */
2505 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2506 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2507 gst_pad_link (pad, selpad);
2508 gst_object_unref (pad);
2509 gst_object_unref (selpad);
2513 /* check if we need to set to a special state */
2514 if (state != GST_STATE_NULL) {
2515 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2516 gst_element_set_state (priv->funnel[i], state);
2517 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2518 gst_element_set_state (priv->appsrc[i], state);
2524 * gst_rtsp_stream_join_bin:
2525 * @stream: a #GstRTSPStream
2526 * @bin: (transfer none): a #GstBin to join
2527 * @rtpbin: (transfer none): a rtpbin element in @bin
2528 * @state: the target state of the new elements
2530 * Join the #GstBin @bin that contains the element @rtpbin.
2532 * @stream will link to @rtpbin, which must be inside @bin. The elements
2533 * added to @bin will be set to the state given in @state.
2535 * Returns: %TRUE on success.
2538 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2539 GstElement * rtpbin, GstState state)
2541 GstRTSPStreamPrivate *priv;
2544 GstPadLinkReturn ret;
2546 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2547 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2548 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2550 priv = stream->priv;
2552 g_mutex_lock (&priv->lock);
2553 if (priv->is_joined)
2556 /* create a session with the same index as the stream */
2559 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2561 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2562 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2564 g_signal_connect (rtpbin, "request-rtp-encoder",
2565 (GCallback) request_rtp_encoder, stream);
2566 g_signal_connect (rtpbin, "request-rtcp-encoder",
2567 (GCallback) request_rtcp_encoder, stream);
2568 g_signal_connect (rtpbin, "request-rtp-decoder",
2569 (GCallback) request_rtp_rtcp_decoder, stream);
2570 g_signal_connect (rtpbin, "request-rtcp-decoder",
2571 (GCallback) request_rtp_rtcp_decoder, stream);
2574 if (priv->sinkpad) {
2575 g_signal_connect (rtpbin, "request-pt-map",
2576 (GCallback) request_pt_map, stream);
2579 /* get pads from the RTP session element for sending and receiving
2582 /* get a pad for sending RTP */
2583 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2584 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2587 /* link the RTP pad to the session manager, it should not really fail unless
2588 * this is not really an RTP pad */
2589 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2590 if (ret != GST_PAD_LINK_OK)
2593 name = g_strdup_printf ("send_rtp_src_%u", idx);
2594 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2597 /* Need to connect our sinkpad from here */
2598 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2600 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2602 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2603 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2607 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2608 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2610 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2611 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2614 /* get the session */
2615 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2617 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2619 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2621 g_signal_connect (priv->session, "on-ssrc-active",
2622 (GCallback) on_ssrc_active, stream);
2623 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2625 g_signal_connect (priv->session, "on-bye-timeout",
2626 (GCallback) on_bye_timeout, stream);
2627 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2630 /* signal for sender ssrc */
2631 g_signal_connect (priv->session, "on-new-sender-ssrc",
2632 (GCallback) on_new_sender_ssrc, stream);
2633 g_signal_connect (priv->session, "on-sender-ssrc-active",
2634 (GCallback) on_sender_ssrc_active, stream);
2636 if (!create_sender_part (stream, bin, state))
2637 goto no_udp_protocol;
2639 create_receiver_part (stream, bin, state);
2642 /* be notified of caps changes */
2643 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2644 (GCallback) caps_notify, stream);
2647 priv->is_joined = TRUE;
2648 g_mutex_unlock (&priv->lock);
2655 g_mutex_unlock (&priv->lock);
2660 GST_WARNING ("failed to link stream %u", idx);
2661 gst_object_unref (priv->send_rtp_sink);
2662 priv->send_rtp_sink = NULL;
2663 g_mutex_unlock (&priv->lock);
2668 GST_WARNING ("failed to allocate ports %u", idx);
2669 gst_object_unref (priv->send_rtp_sink);
2670 priv->send_rtp_sink = NULL;
2671 gst_object_unref (priv->send_src[0]);
2672 priv->send_src[0] = NULL;
2673 gst_object_unref (priv->send_src[1]);
2674 priv->send_src[1] = NULL;
2675 gst_object_unref (priv->recv_sink[0]);
2676 priv->recv_sink[0] = NULL;
2677 gst_object_unref (priv->recv_sink[1]);
2678 priv->recv_sink[1] = NULL;
2679 if (priv->udpsink[0])
2680 gst_element_set_state (priv->udpsink[0], GST_STATE_NULL);
2681 if (priv->udpsink[1])
2682 gst_element_set_state (priv->udpsink[1], GST_STATE_NULL);
2683 if (priv->udpsrc_v4[0]) {
2684 gst_element_set_state (priv->udpsrc_v4[0], GST_STATE_NULL);
2685 gst_object_unref (priv->udpsrc_v4[0]);
2686 priv->udpsrc_v4[0] = NULL;
2688 if (priv->udpsrc_v4[1]) {
2689 gst_element_set_state (priv->udpsrc_v4[1], GST_STATE_NULL);
2690 gst_object_unref (priv->udpsrc_v4[1]);
2691 priv->udpsrc_v4[1] = NULL;
2693 if (priv->udpsrc_mcast_v4[0]) {
2694 gst_element_set_state (priv->udpsrc_mcast_v4[0], GST_STATE_NULL);
2695 gst_object_unref (priv->udpsrc_mcast_v4[0]);
2696 priv->udpsrc_mcast_v4[0] = NULL;
2698 if (priv->udpsrc_mcast_v4[1]) {
2699 gst_element_set_state (priv->udpsrc_mcast_v4[1], GST_STATE_NULL);
2700 gst_object_unref (priv->udpsrc_mcast_v4[1]);
2701 priv->udpsrc_mcast_v4[1] = NULL;
2703 if (priv->udpsrc_v6[0]) {
2704 gst_element_set_state (priv->udpsrc_v6[0], GST_STATE_NULL);
2705 gst_object_unref (priv->udpsrc_v6[0]);
2706 priv->udpsrc_v6[0] = NULL;
2708 if (priv->udpsrc_v6[1]) {
2709 gst_element_set_state (priv->udpsrc_v6[1], GST_STATE_NULL);
2710 gst_object_unref (priv->udpsrc_v6[1]);
2711 priv->udpsrc_v6[1] = NULL;
2713 if (priv->udpsrc_mcast_v6[0]) {
2714 gst_element_set_state (priv->udpsrc_mcast_v6[0], GST_STATE_NULL);
2715 gst_object_unref (priv->udpsrc_mcast_v6[0]);
2716 priv->udpsrc_mcast_v6[0] = NULL;
2718 if (priv->udpsrc_mcast_v6[1]) {
2719 gst_element_set_state (priv->udpsrc_mcast_v6[1], GST_STATE_NULL);
2720 gst_object_unref (priv->udpsrc_mcast_v6[1]);
2721 priv->udpsrc_mcast_v6[1] = NULL;
2723 g_mutex_unlock (&priv->lock);
2729 * gst_rtsp_stream_leave_bin:
2730 * @stream: a #GstRTSPStream
2731 * @bin: (transfer none): a #GstBin
2732 * @rtpbin: (transfer none): a rtpbin #GstElement
2734 * Remove the elements of @stream from @bin.
2736 * Return: %TRUE on success.
2739 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2740 GstElement * rtpbin)
2742 GstRTSPStreamPrivate *priv;
2744 gboolean is_tcp, is_udp;
2746 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2747 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2748 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2750 priv = stream->priv;
2752 g_mutex_lock (&priv->lock);
2753 if (!priv->is_joined)
2754 goto was_not_joined;
2756 /* all transports must be removed by now */
2757 if (priv->transports != NULL)
2758 goto transports_not_removed;
2760 clear_tr_cache (priv, TRUE);
2761 clear_tr_cache (priv, FALSE);
2763 GST_INFO ("stream %p leaving bin", stream);
2766 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2768 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2769 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2770 gst_object_unref (priv->send_rtp_sink);
2771 priv->send_rtp_sink = NULL;
2772 } else if (priv->recv_rtp_src) {
2773 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2774 gst_object_unref (priv->recv_rtp_src);
2775 priv->recv_rtp_src = NULL;
2778 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2780 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2781 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2784 for (i = 0; i < 2; i++) {
2785 if (priv->udpsink[i])
2786 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2787 if (priv->appsink[i])
2788 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2789 if (priv->appqueue[i])
2790 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2791 if (priv->udpqueue[i])
2792 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2794 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2795 if (priv->funnel[i])
2796 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2797 if (priv->appsrc[i])
2798 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2800 if (priv->udpsrc_v4[i]) {
2801 if (priv->sinkpad || i == 1) {
2802 /* and set udpsrc to NULL now before removing */
2803 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2804 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2805 /* removing them should also nicely release the request
2806 * pads when they finalize */
2807 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2809 /* we need to set the state to NULL before unref */
2810 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2811 gst_object_unref (priv->udpsrc_v4[i]);
2815 if (priv->udpsrc_mcast_v4[i]) {
2816 if (priv->sinkpad || i == 1) {
2817 /* and set udpsrc to NULL now before removing */
2818 gst_element_set_locked_state (priv->udpsrc_mcast_v4[i], FALSE);
2819 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2820 /* removing them should also nicely release the request
2821 * pads when they finalize */
2822 gst_bin_remove (bin, priv->udpsrc_mcast_v4[i]);
2824 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2825 gst_object_unref (priv->udpsrc_mcast_v4[i]);
2829 if (priv->udpsrc_v6[i]) {
2830 if (priv->sinkpad || i == 1) {
2831 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2832 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2833 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2835 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2836 gst_object_unref (priv->udpsrc_v6[i]);
2839 if (priv->udpsrc_mcast_v6[i]) {
2840 if (priv->sinkpad || i == 1) {
2841 gst_element_set_locked_state (priv->udpsrc_mcast_v6[i], FALSE);
2842 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2843 gst_bin_remove (bin, priv->udpsrc_mcast_v6[i]);
2845 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2846 gst_object_unref (priv->udpsrc_mcast_v6[i]);
2850 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
2851 gst_bin_remove (bin, priv->udpsink[i]);
2852 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2853 gst_bin_remove (bin, priv->appsrc[i]);
2854 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
2855 gst_bin_remove (bin, priv->appsink[i]);
2856 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2857 gst_bin_remove (bin, priv->appqueue[i]);
2858 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2859 gst_bin_remove (bin, priv->udpqueue[i]);
2860 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2861 gst_bin_remove (bin, priv->tee[i]);
2862 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2863 gst_bin_remove (bin, priv->funnel[i]);
2865 if (priv->sinkpad || i == 1) {
2866 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2867 gst_object_unref (priv->recv_sink[i]);
2868 priv->recv_sink[i] = NULL;
2871 priv->udpsrc_v4[i] = NULL;
2872 priv->udpsrc_v6[i] = NULL;
2873 priv->udpsrc_mcast_v4[i] = NULL;
2874 priv->udpsrc_mcast_v6[i] = NULL;
2875 priv->udpsink[i] = NULL;
2876 priv->appsrc[i] = NULL;
2877 priv->appsink[i] = NULL;
2878 priv->appqueue[i] = NULL;
2879 priv->udpqueue[i] = NULL;
2880 priv->tee[i] = NULL;
2881 priv->funnel[i] = NULL;
2885 gst_object_unref (priv->send_src[0]);
2886 priv->send_src[0] = NULL;
2889 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2890 gst_object_unref (priv->send_src[1]);
2891 priv->send_src[1] = NULL;
2893 g_object_unref (priv->session);
2894 priv->session = NULL;
2896 gst_caps_unref (priv->caps);
2900 gst_object_unref (priv->srtpenc);
2902 gst_object_unref (priv->srtpdec);
2904 priv->is_joined = FALSE;
2905 g_mutex_unlock (&priv->lock);
2911 g_mutex_unlock (&priv->lock);
2914 transports_not_removed:
2916 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2917 g_mutex_unlock (&priv->lock);
2923 * gst_rtsp_stream_get_rtpinfo:
2924 * @stream: a #GstRTSPStream
2925 * @rtptime: (allow-none): result RTP timestamp
2926 * @seq: (allow-none): result RTP seqnum
2927 * @clock_rate: (allow-none): the clock rate
2928 * @running_time: (allow-none): result running-time
2930 * Retrieve the current rtptime, seq and running-time. This is used to
2931 * construct a RTPInfo reply header.
2933 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2936 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2937 guint * rtptime, guint * seq, guint * clock_rate,
2938 GstClockTime * running_time)
2940 GstRTSPStreamPrivate *priv;
2941 GstStructure *stats;
2942 GObjectClass *payobjclass;
2944 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2946 priv = stream->priv;
2948 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2950 g_mutex_lock (&priv->lock);
2952 /* First try to extract the information from the last buffer on the sinks.
2953 * This will have a more accurate sequence number and timestamp, as between
2954 * the payloader and the sink there can be some queues
2956 if (priv->udpsink[0] || priv->appsink[0]) {
2957 GstSample *last_sample;
2959 if (priv->udpsink[0])
2960 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2962 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2967 GstSegment *segment;
2968 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2970 caps = gst_sample_get_caps (last_sample);
2971 buffer = gst_sample_get_buffer (last_sample);
2972 segment = gst_sample_get_segment (last_sample);
2974 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2976 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2980 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2983 gst_rtp_buffer_unmap (&rtp_buffer);
2987 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2988 GST_BUFFER_TIMESTAMP (buffer));
2992 GstStructure *s = gst_caps_get_structure (caps, 0);
2994 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
2996 if (*clock_rate == 0 && running_time)
2997 *running_time = GST_CLOCK_TIME_NONE;
2999 gst_sample_unref (last_sample);
3003 gst_sample_unref (last_sample);
3008 if (g_object_class_find_property (payobjclass, "stats")) {
3009 g_object_get (priv->payloader, "stats", &stats, NULL);
3014 gst_structure_get_uint (stats, "seqnum", seq);
3017 gst_structure_get_uint (stats, "timestamp", rtptime);
3020 gst_structure_get_clock_time (stats, "running-time", running_time);
3023 gst_structure_get_uint (stats, "clock-rate", clock_rate);
3024 if (*clock_rate == 0 && running_time)
3025 *running_time = GST_CLOCK_TIME_NONE;
3027 gst_structure_free (stats);
3029 if (!g_object_class_find_property (payobjclass, "seqnum") ||
3030 !g_object_class_find_property (payobjclass, "timestamp"))
3034 g_object_get (priv->payloader, "seqnum", seq, NULL);
3037 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
3040 *running_time = GST_CLOCK_TIME_NONE;
3044 g_mutex_unlock (&priv->lock);
3051 GST_WARNING ("Could not get payloader stats");
3052 g_mutex_unlock (&priv->lock);
3058 * gst_rtsp_stream_get_caps:
3059 * @stream: a #GstRTSPStream
3061 * Retrieve the current caps of @stream.
3063 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
3067 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
3069 GstRTSPStreamPrivate *priv;
3072 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3074 priv = stream->priv;
3076 g_mutex_lock (&priv->lock);
3077 if ((result = priv->caps))
3078 gst_caps_ref (result);
3079 g_mutex_unlock (&priv->lock);
3085 * gst_rtsp_stream_recv_rtp:
3086 * @stream: a #GstRTSPStream
3087 * @buffer: (transfer full): a #GstBuffer
3089 * Handle an RTP buffer for the stream. This method is usually called when a
3090 * message has been received from a client using the TCP transport.
3092 * This function takes ownership of @buffer.
3094 * Returns: a GstFlowReturn.
3097 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
3099 GstRTSPStreamPrivate *priv;
3101 GstElement *element;
3103 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3104 priv = stream->priv;
3105 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3106 g_return_val_if_fail (priv->is_joined, FALSE);
3108 g_mutex_lock (&priv->lock);
3109 if (priv->appsrc[0])
3110 element = gst_object_ref (priv->appsrc[0]);
3113 g_mutex_unlock (&priv->lock);
3116 if (priv->appsrc_base_time[0] == -1) {
3117 /* Take current running_time. This timestamp will be put on
3118 * the first buffer of each stream because we are a live source and so we
3119 * timestamp with the running_time. When we are dealing with TCP, we also
3120 * only timestamp the first buffer (using the DISCONT flag) because a server
3121 * typically bursts data, for which we don't want to compensate by speeding
3122 * up the media. The other timestamps will be interpollated from this one
3123 * using the RTP timestamps. */
3124 GST_OBJECT_LOCK (element);
3125 if (GST_ELEMENT_CLOCK (element)) {
3127 GstClockTime base_time;
3129 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3130 base_time = GST_ELEMENT_CAST (element)->base_time;
3132 priv->appsrc_base_time[0] = now - base_time;
3133 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
3134 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3135 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3136 GST_TIME_ARGS (base_time));
3138 GST_OBJECT_UNLOCK (element);
3141 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3142 gst_object_unref (element);
3150 * gst_rtsp_stream_recv_rtcp:
3151 * @stream: a #GstRTSPStream
3152 * @buffer: (transfer full): a #GstBuffer
3154 * Handle an RTCP buffer for the stream. This method is usually called when a
3155 * message has been received from a client using the TCP transport.
3157 * This function takes ownership of @buffer.
3159 * Returns: a GstFlowReturn.
3162 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
3164 GstRTSPStreamPrivate *priv;
3166 GstElement *element;
3168 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3169 priv = stream->priv;
3170 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3172 if (!priv->is_joined) {
3173 gst_buffer_unref (buffer);
3174 return GST_FLOW_NOT_LINKED;
3176 g_mutex_lock (&priv->lock);
3177 if (priv->appsrc[1])
3178 element = gst_object_ref (priv->appsrc[1]);
3181 g_mutex_unlock (&priv->lock);
3184 if (priv->appsrc_base_time[1] == -1) {
3185 /* Take current running_time. This timestamp will be put on
3186 * the first buffer of each stream because we are a live source and so we
3187 * timestamp with the running_time. When we are dealing with TCP, we also
3188 * only timestamp the first buffer (using the DISCONT flag) because a server
3189 * typically bursts data, for which we don't want to compensate by speeding
3190 * up the media. The other timestamps will be interpollated from this one
3191 * using the RTP timestamps. */
3192 GST_OBJECT_LOCK (element);
3193 if (GST_ELEMENT_CLOCK (element)) {
3195 GstClockTime base_time;
3197 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3198 base_time = GST_ELEMENT_CAST (element)->base_time;
3200 priv->appsrc_base_time[1] = now - base_time;
3201 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3202 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3203 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3204 GST_TIME_ARGS (base_time));
3206 GST_OBJECT_UNLOCK (element);
3209 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3210 gst_object_unref (element);
3213 gst_buffer_unref (buffer);
3218 /* must be called with lock */
3220 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3223 GstRTSPStreamPrivate *priv = stream->priv;
3224 const GstRTSPTransport *tr;
3226 tr = gst_rtsp_stream_transport_get_transport (trans);
3228 switch (tr->lower_transport) {
3229 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3230 case GST_RTSP_LOWER_TRANS_UDP:
3236 dest = tr->destination;
3237 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3241 } else if (priv->client_side) {
3242 /* In client side mode the 'destination' is the RTSP server, so send
3244 min = tr->server_port.min;
3245 max = tr->server_port.max;
3247 min = tr->client_port.min;
3248 max = tr->client_port.max;
3253 GST_INFO ("setting ttl-mc %d", ttl);
3254 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3255 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3257 GST_INFO ("adding %s:%d-%d", dest, min, max);
3258 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3259 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3260 priv->transports = g_list_prepend (priv->transports, trans);
3262 GST_INFO ("removing %s:%d-%d", dest, min, max);
3263 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3264 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3265 priv->transports = g_list_remove (priv->transports, trans);
3267 priv->transports_cookie++;
3270 case GST_RTSP_LOWER_TRANS_TCP:
3272 GST_INFO ("adding TCP %s", tr->destination);
3273 priv->transports = g_list_prepend (priv->transports, trans);
3275 GST_INFO ("removing TCP %s", tr->destination);
3276 priv->transports = g_list_remove (priv->transports, trans);
3278 priv->transports_cookie++;
3281 goto unknown_transport;
3288 GST_INFO ("Unknown transport %d", tr->lower_transport);
3295 * gst_rtsp_stream_add_transport:
3296 * @stream: a #GstRTSPStream
3297 * @trans: (transfer none): a #GstRTSPStreamTransport
3299 * Add the transport in @trans to @stream. The media of @stream will
3300 * then also be send to the values configured in @trans.
3302 * @stream must be joined to a bin.
3304 * @trans must contain a valid #GstRTSPTransport.
3306 * Returns: %TRUE if @trans was added
3309 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3310 GstRTSPStreamTransport * trans)
3312 GstRTSPStreamPrivate *priv;
3315 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3316 priv = stream->priv;
3317 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3318 g_return_val_if_fail (priv->is_joined, FALSE);
3320 g_mutex_lock (&priv->lock);
3321 res = update_transport (stream, trans, TRUE);
3322 g_mutex_unlock (&priv->lock);
3328 * gst_rtsp_stream_remove_transport:
3329 * @stream: a #GstRTSPStream
3330 * @trans: (transfer none): a #GstRTSPStreamTransport
3332 * Remove the transport in @trans from @stream. The media of @stream will
3333 * not be sent to the values configured in @trans.
3335 * @stream must be joined to a bin.
3337 * @trans must contain a valid #GstRTSPTransport.
3339 * Returns: %TRUE if @trans was removed
3342 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3343 GstRTSPStreamTransport * trans)
3345 GstRTSPStreamPrivate *priv;
3348 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3349 priv = stream->priv;
3350 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3351 g_return_val_if_fail (priv->is_joined, FALSE);
3353 g_mutex_lock (&priv->lock);
3354 res = update_transport (stream, trans, FALSE);
3355 g_mutex_unlock (&priv->lock);
3361 * gst_rtsp_stream_update_crypto:
3362 * @stream: a #GstRTSPStream
3364 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3366 * Update the new crypto information for @ssrc in @stream. If information
3367 * for @ssrc did not exist, it will be added. If information
3368 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3369 * be removed from @stream.
3371 * Returns: %TRUE if @crypto could be updated
3374 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3375 guint ssrc, GstCaps * crypto)
3377 GstRTSPStreamPrivate *priv;
3379 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3380 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3382 priv = stream->priv;
3384 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3386 g_mutex_lock (&priv->lock);
3388 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3389 gst_caps_ref (crypto));
3391 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3392 g_mutex_unlock (&priv->lock);
3398 * gst_rtsp_stream_get_rtp_socket:
3399 * @stream: a #GstRTSPStream
3400 * @family: the socket family
3402 * Get the RTP socket from @stream for a @family.
3404 * @stream must be joined to a bin.
3406 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3407 * socket could be allocated for @family. Unref after usage
3410 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3412 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3416 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3417 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3418 family == G_SOCKET_FAMILY_IPV6, NULL);
3419 g_return_val_if_fail (priv->udpsink[0], NULL);
3421 if (family == G_SOCKET_FAMILY_IPV6)
3426 g_object_get (priv->udpsink[0], name, &socket, NULL);
3432 * gst_rtsp_stream_get_rtcp_socket:
3433 * @stream: a #GstRTSPStream
3434 * @family: the socket family
3436 * Get the RTCP socket from @stream for a @family.
3438 * @stream must be joined to a bin.
3440 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3441 * socket could be allocated for @family. Unref after usage
3444 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3446 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3450 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3451 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3452 family == G_SOCKET_FAMILY_IPV6, NULL);
3453 g_return_val_if_fail (priv->udpsink[1], NULL);
3455 if (family == G_SOCKET_FAMILY_IPV6)
3460 g_object_get (priv->udpsink[1], name, &socket, NULL);
3466 * gst_rtsp_stream_set_seqnum:
3467 * @stream: a #GstRTSPStream
3468 * @seqnum: a new sequence number
3470 * Configure the sequence number in the payloader of @stream to @seqnum.
3473 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3475 GstRTSPStreamPrivate *priv;
3477 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3479 priv = stream->priv;
3481 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3485 * gst_rtsp_stream_get_seqnum:
3486 * @stream: a #GstRTSPStream
3488 * Get the configured sequence number in the payloader of @stream.
3490 * Returns: the sequence number of the payloader.
3493 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3495 GstRTSPStreamPrivate *priv;
3498 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3500 priv = stream->priv;
3502 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3508 * gst_rtsp_stream_transport_filter:
3509 * @stream: a #GstRTSPStream
3510 * @func: (scope call) (allow-none): a callback
3511 * @user_data: (closure): user data passed to @func
3513 * Call @func for each transport managed by @stream. The result value of @func
3514 * determines what happens to the transport. @func will be called with @stream
3515 * locked so no further actions on @stream can be performed from @func.
3517 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3520 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3522 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3523 * will also be added with an additional ref to the result #GList of this
3526 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3528 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3529 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3530 * element in the #GList should be unreffed before the list is freed.
3533 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3534 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3536 GstRTSPStreamPrivate *priv;
3537 GList *result, *walk, *next;
3538 GHashTable *visited = NULL;
3541 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3543 priv = stream->priv;
3547 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3549 g_mutex_lock (&priv->lock);
3551 cookie = priv->transports_cookie;
3552 for (walk = priv->transports; walk; walk = next) {
3553 GstRTSPStreamTransport *trans = walk->data;
3554 GstRTSPFilterResult res;
3557 next = g_list_next (walk);
3560 /* only visit each transport once */
3561 if (g_hash_table_contains (visited, trans))
3564 g_hash_table_add (visited, g_object_ref (trans));
3565 g_mutex_unlock (&priv->lock);
3567 res = func (stream, trans, user_data);
3569 g_mutex_lock (&priv->lock);
3571 res = GST_RTSP_FILTER_REF;
3573 changed = (cookie != priv->transports_cookie);
3576 case GST_RTSP_FILTER_REMOVE:
3577 update_transport (stream, trans, FALSE);
3579 case GST_RTSP_FILTER_REF:
3580 result = g_list_prepend (result, g_object_ref (trans));
3582 case GST_RTSP_FILTER_KEEP:
3589 g_mutex_unlock (&priv->lock);
3592 g_hash_table_unref (visited);
3597 static GstPadProbeReturn
3598 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3600 GstRTSPStreamPrivate *priv;
3601 GstRTSPStream *stream;
3604 priv = stream->priv;
3606 GST_DEBUG_OBJECT (pad, "now blocking");
3608 g_mutex_lock (&priv->lock);
3609 priv->blocking = TRUE;
3610 g_mutex_unlock (&priv->lock);
3612 gst_element_post_message (priv->payloader,
3613 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3614 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3616 return GST_PAD_PROBE_OK;
3620 * gst_rtsp_stream_set_blocked:
3621 * @stream: a #GstRTSPStream
3622 * @blocked: boolean indicating we should block or unblock
3624 * Blocks or unblocks the dataflow on @stream.
3626 * Returns: %TRUE on success
3629 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3631 GstRTSPStreamPrivate *priv;
3633 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3635 priv = stream->priv;
3637 g_mutex_lock (&priv->lock);
3639 priv->blocking = FALSE;
3640 if (priv->blocked_id == 0) {
3641 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3642 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3643 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3644 g_object_ref (stream), g_object_unref);
3647 if (priv->blocked_id != 0) {
3648 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3649 priv->blocked_id = 0;
3650 priv->blocking = FALSE;
3653 g_mutex_unlock (&priv->lock);
3659 * gst_rtsp_stream_is_blocking:
3660 * @stream: a #GstRTSPStream
3662 * Check if @stream is blocking on a #GstBuffer.
3664 * Returns: %TRUE if @stream is blocking
3667 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3669 GstRTSPStreamPrivate *priv;
3672 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3674 priv = stream->priv;
3676 g_mutex_lock (&priv->lock);
3677 result = priv->blocking;
3678 g_mutex_unlock (&priv->lock);
3684 * gst_rtsp_stream_query_position:
3685 * @stream: a #GstRTSPStream
3687 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3688 * the RTP parts of the pipeline and not the RTCP parts.
3690 * Returns: %TRUE if the position could be queried
3693 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3695 GstRTSPStreamPrivate *priv;
3699 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3701 priv = stream->priv;
3703 g_mutex_lock (&priv->lock);
3704 /* depending on the transport type, it should query corresponding sink */
3705 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3706 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3707 sink = priv->udpsink[0];
3709 sink = priv->appsink[0];
3712 gst_object_ref (sink);
3713 g_mutex_unlock (&priv->lock);
3718 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3719 gst_object_unref (sink);
3725 * gst_rtsp_stream_query_stop:
3726 * @stream: a #GstRTSPStream
3728 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3729 * the RTP parts of the pipeline and not the RTCP parts.
3731 * Returns: %TRUE if the stop could be queried
3734 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3736 GstRTSPStreamPrivate *priv;
3741 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3743 priv = stream->priv;
3745 g_mutex_lock (&priv->lock);
3746 /* depending on the transport type, it should query corresponding sink */
3747 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3748 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3749 sink = priv->udpsink[0];
3751 sink = priv->appsink[0];
3754 gst_object_ref (sink);
3755 g_mutex_unlock (&priv->lock);
3760 query = gst_query_new_segment (GST_FORMAT_TIME);
3761 if ((ret = gst_element_query (sink, query))) {
3764 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3765 if (format != GST_FORMAT_TIME)
3768 gst_query_unref (query);
3769 gst_object_unref (sink);