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 && use_client_settings)
1252 gst_rtsp_address_pool_reserve_address (pool, ct->destination,
1253 ct->port.min, 2, ct->ttl, &addr);
1255 addr = gst_rtsp_address_pool_acquire_address (pool, flags, 2);
1260 tmp_rtp = addr->port;
1262 g_clear_object (&inetaddr);
1263 inetaddr = g_inet_address_new_from_string (addr->address);
1265 /* Don't bind to multicast addresses, this does not work on
1266 * Windows. You're supposed to bind to ANY and then join the
1267 * multicast group, which udpsrc/sink does for us already.
1269 if (g_inet_address_get_is_multicast (inetaddr)) {
1270 g_object_unref (inetaddr);
1271 inetaddr = g_inet_address_new_any (family);
1280 if (inetaddr == NULL)
1281 inetaddr = g_inet_address_new_any (family);
1284 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
1285 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
1286 g_object_unref (rtp_sockaddr);
1289 g_object_unref (rtp_sockaddr);
1291 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
1292 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
1293 g_clear_object (&rtp_sockaddr);
1298 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
1299 g_object_unref (rtp_sockaddr);
1301 /* check if port is even */
1302 if ((tmp_rtp & 1) != 0) {
1303 /* port not even, close and allocate another */
1305 g_clear_object (&rtp_socket);
1310 tmp_rtcp = tmp_rtp + 1;
1312 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
1313 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
1314 g_object_unref (rtcp_sockaddr);
1315 g_clear_object (&rtp_socket);
1318 g_object_unref (rtcp_sockaddr);
1321 addr_str = g_inet_address_to_string (inetaddr);
1323 addr_str = addr->address;
1324 g_clear_object (&inetaddr);
1326 if (!create_and_configure_udpsources_one_family (udpsrc_out, rtp_socket,
1327 rtcp_socket, family, addr_str, tmp_rtp, tmp_rtcp, transport)) {
1330 goto no_udp_protocol;
1336 play_udpsources_one_family (stream, udpsrc_out, family);
1338 g_object_get (G_OBJECT (udpsrc_out[0]), "port", &rtpport, NULL);
1339 g_object_get (G_OBJECT (udpsrc_out[1]), "port", &rtcpport, NULL);
1341 /* this should not happen... */
1342 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
1345 /* set RTP and RTCP sockets */
1346 set_sockets_for_udpsinks (stream, rtp_socket, rtcp_socket, family);
1348 server_port_out->min = rtpport;
1349 server_port_out->max = rtcpport;
1351 *server_addr_out = addr;
1352 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
1354 g_object_unref (rtp_socket);
1355 g_object_unref (rtcp_socket);
1379 g_object_unref (inetaddr);
1380 g_list_free_full (rejected_addresses,
1381 (GDestroyNotify) gst_rtsp_address_free);
1383 gst_rtsp_address_free (addr);
1385 g_object_unref (rtp_socket);
1387 g_object_unref (rtcp_socket);
1393 * gst_rtsp_stream_allocate_udp_sockets:
1394 * @stream: a #GstRTSPStream
1395 * @family: protocol family
1396 * @transport_method: transport method
1398 * Allocates RTP and RTCP ports.
1400 * Returns: %TRUE if the RTP and RTCP sockets have been succeccully allocated.
1403 gst_rtsp_stream_allocate_udp_sockets (GstRTSPStream * stream,
1404 GSocketFamily family, GstRTSPTransport *ct, gboolean use_client_settings)
1406 GstRTSPStreamPrivate *priv;
1407 gboolean result = FALSE;
1408 GstRTSPLowerTrans transport = ct->lower_transport;
1410 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1411 priv = stream->priv;
1412 g_return_val_if_fail (priv->is_joined, FALSE);
1414 g_mutex_lock (&priv->lock);
1416 if (family == G_SOCKET_FAMILY_IPV4) {
1417 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1418 if (priv->have_ipv4_mcast)
1420 priv->have_ipv4_mcast =
1421 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_mcast_v4,
1422 &priv->server_port_v4, ct, &priv->addr_v4, use_client_settings);
1425 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV4, priv->udpsrc_v4,
1426 &priv->server_port_v4, ct, &priv->server_addr_v4, use_client_settings);
1429 if (transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1430 if (priv->have_ipv6_mcast)
1432 priv->have_ipv6_mcast =
1433 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_mcast_v6,
1434 &priv->server_port_v6, ct, &priv->addr_v6, use_client_settings);
1436 if (priv->have_ipv6)
1439 alloc_ports_one_family (stream, G_SOCKET_FAMILY_IPV6, priv->udpsrc_v6,
1440 &priv->server_port_v6, ct, &priv->server_addr_v6, use_client_settings);
1445 result = priv->have_ipv4 || priv->have_ipv4_mcast || priv->have_ipv6 ||
1446 priv->have_ipv6_mcast;
1448 g_mutex_unlock (&priv->lock);
1454 * gst_rtsp_stream_set_client_side:
1455 * @stream: a #GstRTSPStream
1456 * @client_side: TRUE if this #GstRTSPStream is running on the 'client' side of
1457 * an RTSP connection.
1459 * Sets the #GstRTSPStream as a 'client side' stream - used for sending
1460 * streams to an RTSP server via RECORD. This has the practical effect
1461 * of changing which UDP port numbers are used when setting up the local
1462 * side of the stream sending to be either the 'server' or 'client' pair
1463 * of a configured UDP transport.
1466 gst_rtsp_stream_set_client_side (GstRTSPStream * stream, gboolean client_side)
1468 GstRTSPStreamPrivate *priv;
1470 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1471 priv = stream->priv;
1472 g_mutex_lock (&priv->lock);
1473 priv->client_side = client_side;
1474 g_mutex_unlock (&priv->lock);
1478 * gst_rtsp_stream_set_client_side:
1479 * @stream: a #GstRTSPStream
1481 * See gst_rtsp_stream_set_client_side()
1483 * Returns: TRUE if this #GstRTSPStream is client-side.
1486 gst_rtsp_stream_is_client_side (GstRTSPStream * stream)
1488 GstRTSPStreamPrivate *priv;
1491 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1493 priv = stream->priv;
1494 g_mutex_lock (&priv->lock);
1495 ret = priv->client_side;
1496 g_mutex_unlock (&priv->lock);
1502 * gst_rtsp_stream_get_server_port:
1503 * @stream: a #GstRTSPStream
1504 * @server_port: (out): result server port
1505 * @family: the port family to get
1507 * Fill @server_port with the port pair used by the server. This function can
1508 * only be called when @stream has been joined.
1511 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
1512 GstRTSPRange * server_port, GSocketFamily family)
1514 GstRTSPStreamPrivate *priv;
1516 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1517 priv = stream->priv;
1518 g_return_if_fail (priv->is_joined);
1520 g_mutex_lock (&priv->lock);
1521 if (family == G_SOCKET_FAMILY_IPV4) {
1523 *server_port = priv->server_port_v4;
1526 *server_port = priv->server_port_v6;
1528 g_mutex_unlock (&priv->lock);
1532 * gst_rtsp_stream_get_rtpsession:
1533 * @stream: a #GstRTSPStream
1535 * Get the RTP session of this stream.
1537 * Returns: (transfer full): The RTP session of this stream. Unref after usage.
1540 gst_rtsp_stream_get_rtpsession (GstRTSPStream * stream)
1542 GstRTSPStreamPrivate *priv;
1545 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1547 priv = stream->priv;
1549 g_mutex_lock (&priv->lock);
1550 if ((session = priv->session))
1551 g_object_ref (session);
1552 g_mutex_unlock (&priv->lock);
1558 * gst_rtsp_stream_get_ssrc:
1559 * @stream: a #GstRTSPStream
1560 * @ssrc: (out): result ssrc
1562 * Get the SSRC used by the RTP session of this stream. This function can only
1563 * be called when @stream has been joined.
1566 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
1568 GstRTSPStreamPrivate *priv;
1570 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1571 priv = stream->priv;
1572 g_return_if_fail (priv->is_joined);
1574 g_mutex_lock (&priv->lock);
1575 if (ssrc && priv->session)
1576 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
1577 g_mutex_unlock (&priv->lock);
1581 * gst_rtsp_stream_set_retransmission_time:
1582 * @stream: a #GstRTSPStream
1583 * @time: a #GstClockTime
1585 * Set the amount of time to store retransmission packets.
1588 gst_rtsp_stream_set_retransmission_time (GstRTSPStream * stream,
1591 GST_DEBUG_OBJECT (stream, "set retransmission time %" G_GUINT64_FORMAT, time);
1593 g_mutex_lock (&stream->priv->lock);
1594 stream->priv->rtx_time = time;
1595 if (stream->priv->rtxsend)
1596 g_object_set (stream->priv->rtxsend, "max-size-time",
1597 GST_TIME_AS_MSECONDS (time), NULL);
1598 g_mutex_unlock (&stream->priv->lock);
1602 * gst_rtsp_stream_get_retransmission_time:
1603 * @stream: a #GstRTSPStream
1605 * Get the amount of time to store retransmission data.
1607 * Returns: the amount of time to store retransmission data.
1610 gst_rtsp_stream_get_retransmission_time (GstRTSPStream * stream)
1614 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1616 g_mutex_lock (&stream->priv->lock);
1617 ret = stream->priv->rtx_time;
1618 g_mutex_unlock (&stream->priv->lock);
1624 * gst_rtsp_stream_set_retransmission_pt:
1625 * @stream: a #GstRTSPStream
1628 * Set the payload type (pt) for retransmission of this stream.
1631 gst_rtsp_stream_set_retransmission_pt (GstRTSPStream * stream, guint rtx_pt)
1633 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
1635 GST_DEBUG_OBJECT (stream, "set retransmission pt %u", rtx_pt);
1637 g_mutex_lock (&stream->priv->lock);
1638 stream->priv->rtx_pt = rtx_pt;
1639 if (stream->priv->rtxsend) {
1640 guint pt = gst_rtsp_stream_get_pt (stream);
1641 gchar *pt_s = g_strdup_printf ("%d", pt);
1642 GstStructure *rtx_pt_map = gst_structure_new ("application/x-rtp-pt-map",
1643 pt_s, G_TYPE_UINT, rtx_pt, NULL);
1644 g_object_set (stream->priv->rtxsend, "payload-type-map", rtx_pt_map, NULL);
1646 gst_structure_free (rtx_pt_map);
1648 g_mutex_unlock (&stream->priv->lock);
1652 * gst_rtsp_stream_get_retransmission_pt:
1653 * @stream: a #GstRTSPStream
1655 * Get the payload-type used for retransmission of this stream
1657 * Returns: The retransmission PT.
1660 gst_rtsp_stream_get_retransmission_pt (GstRTSPStream * stream)
1664 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
1666 g_mutex_lock (&stream->priv->lock);
1667 rtx_pt = stream->priv->rtx_pt;
1668 g_mutex_unlock (&stream->priv->lock);
1674 * gst_rtsp_stream_set_buffer_size:
1675 * @stream: a #GstRTSPStream
1676 * @size: the buffer size
1678 * Set the size of the UDP transmission buffer (in bytes)
1679 * Needs to be set before the stream is joined to a bin.
1684 gst_rtsp_stream_set_buffer_size (GstRTSPStream * stream, guint size)
1686 g_mutex_lock (&stream->priv->lock);
1687 stream->priv->buffer_size = size;
1688 g_mutex_unlock (&stream->priv->lock);
1692 * gst_rtsp_stream_get_buffer_size:
1693 * @stream: a #GstRTSPStream
1695 * Get the size of the UDP transmission buffer (in bytes)
1697 * Returns: the size of the UDP TX buffer
1702 gst_rtsp_stream_get_buffer_size (GstRTSPStream * stream)
1706 g_mutex_lock (&stream->priv->lock);
1707 buffer_size = stream->priv->buffer_size;
1708 g_mutex_unlock (&stream->priv->lock);
1713 /* executed from streaming thread */
1715 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
1717 GstRTSPStreamPrivate *priv = stream->priv;
1718 GstCaps *newcaps, *oldcaps;
1720 newcaps = gst_pad_get_current_caps (pad);
1722 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
1725 g_mutex_lock (&priv->lock);
1726 oldcaps = priv->caps;
1727 priv->caps = newcaps;
1728 g_mutex_unlock (&priv->lock);
1731 gst_caps_unref (oldcaps);
1735 dump_structure (const GstStructure * s)
1739 sstr = gst_structure_to_string (s);
1740 GST_INFO ("structure: %s", sstr);
1744 static GstRTSPStreamTransport *
1745 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
1747 GstRTSPStreamPrivate *priv = stream->priv;
1749 GstRTSPStreamTransport *result = NULL;
1754 if (rtcp_from == NULL)
1757 tmp = g_strrstr (rtcp_from, ":");
1761 port = atoi (tmp + 1);
1762 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1764 g_mutex_lock (&priv->lock);
1765 GST_INFO ("finding %s:%d in %d transports", dest, port,
1766 g_list_length (priv->transports));
1768 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1769 GstRTSPStreamTransport *trans = walk->data;
1770 const GstRTSPTransport *tr;
1773 tr = gst_rtsp_stream_transport_get_transport (trans);
1775 if (priv->client_side) {
1776 /* In client side mode the 'destination' is the RTSP server, so send
1778 min = tr->server_port.min;
1779 max = tr->server_port.max;
1781 min = tr->client_port.min;
1782 max = tr->client_port.max;
1785 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
1791 g_object_ref (result);
1792 g_mutex_unlock (&priv->lock);
1799 static GstRTSPStreamTransport *
1800 check_transport (GObject * source, GstRTSPStream * stream)
1802 GstStructure *stats;
1803 GstRTSPStreamTransport *trans;
1805 /* see if we have a stream to match with the origin of the RTCP packet */
1806 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1807 if (trans == NULL) {
1808 g_object_get (source, "stats", &stats, NULL);
1810 const gchar *rtcp_from;
1812 dump_structure (stats);
1814 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1815 if ((trans = find_transport (stream, rtcp_from))) {
1816 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1818 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
1821 gst_structure_free (stats);
1829 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1831 GstRTSPStreamTransport *trans;
1833 GST_INFO ("%p: new source %p", stream, source);
1835 trans = check_transport (source, stream);
1838 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1842 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
1844 GST_INFO ("%p: new SDES %p", stream, source);
1848 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
1850 GstRTSPStreamTransport *trans;
1852 trans = check_transport (source, stream);
1855 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1856 gst_rtsp_stream_transport_keep_alive (trans);
1860 GstStructure *stats;
1861 g_object_get (source, "stats", &stats, NULL);
1863 dump_structure (stats);
1864 gst_structure_free (stats);
1871 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1873 GST_INFO ("%p: source %p bye", stream, source);
1877 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1879 GstRTSPStreamTransport *trans;
1881 GST_INFO ("%p: source %p bye timeout", stream, source);
1883 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1884 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1885 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1890 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
1892 GstRTSPStreamTransport *trans;
1894 GST_INFO ("%p: source %p timeout", stream, source);
1896 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1897 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
1898 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
1903 on_new_sender_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
1905 GST_INFO ("%p: new sender source %p", stream, source);
1908 GstStructure *stats;
1909 g_object_get (source, "stats", &stats, NULL);
1911 dump_structure (stats);
1912 gst_structure_free (stats);
1919 on_sender_ssrc_active (GObject * session, GObject * source,
1920 GstRTSPStream * stream)
1924 GstStructure *stats;
1925 g_object_get (source, "stats", &stats, NULL);
1927 dump_structure (stats);
1928 gst_structure_free (stats);
1935 clear_tr_cache (GstRTSPStreamPrivate * priv, gboolean is_rtp)
1938 g_list_foreach (priv->tr_cache_rtp, (GFunc) g_object_unref, NULL);
1939 g_list_free (priv->tr_cache_rtp);
1940 priv->tr_cache_rtp = NULL;
1942 g_list_foreach (priv->tr_cache_rtcp, (GFunc) g_object_unref, NULL);
1943 g_list_free (priv->tr_cache_rtcp);
1944 priv->tr_cache_rtcp = NULL;
1948 static GstFlowReturn
1949 handle_new_sample (GstAppSink * sink, gpointer user_data)
1951 GstRTSPStreamPrivate *priv;
1955 GstRTSPStream *stream;
1958 sample = gst_app_sink_pull_sample (sink);
1962 stream = (GstRTSPStream *) user_data;
1963 priv = stream->priv;
1964 buffer = gst_sample_get_buffer (sample);
1966 is_rtp = GST_ELEMENT_CAST (sink) == priv->appsink[0];
1968 g_mutex_lock (&priv->lock);
1970 if (priv->tr_cache_cookie_rtp != priv->transports_cookie) {
1971 clear_tr_cache (priv, is_rtp);
1972 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1973 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1974 priv->tr_cache_rtp =
1975 g_list_prepend (priv->tr_cache_rtp, g_object_ref (tr));
1977 priv->tr_cache_cookie_rtp = priv->transports_cookie;
1980 if (priv->tr_cache_cookie_rtcp != priv->transports_cookie) {
1981 clear_tr_cache (priv, is_rtp);
1982 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
1983 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1984 priv->tr_cache_rtcp =
1985 g_list_prepend (priv->tr_cache_rtcp, g_object_ref (tr));
1987 priv->tr_cache_cookie_rtcp = priv->transports_cookie;
1990 g_mutex_unlock (&priv->lock);
1993 for (walk = priv->tr_cache_rtp; walk; walk = g_list_next (walk)) {
1994 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
1995 gst_rtsp_stream_transport_send_rtp (tr, buffer);
1998 for (walk = priv->tr_cache_rtcp; walk; walk = g_list_next (walk)) {
1999 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
2000 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
2003 gst_sample_unref (sample);
2008 static GstAppSinkCallbacks sink_cb = {
2009 NULL, /* not interested in EOS */
2010 NULL, /* not interested in preroll samples */
2015 get_rtp_encoder (GstRTSPStream * stream, guint session)
2017 GstRTSPStreamPrivate *priv = stream->priv;
2019 if (priv->srtpenc == NULL) {
2022 name = g_strdup_printf ("srtpenc_%u", session);
2023 priv->srtpenc = gst_element_factory_make ("srtpenc", name);
2026 g_object_set (priv->srtpenc, "random-key", TRUE, NULL);
2028 return gst_object_ref (priv->srtpenc);
2032 request_rtp_encoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
2034 GstRTSPStreamPrivate *priv = stream->priv;
2035 GstElement *oldenc, *enc;
2039 if (priv->idx != session)
2042 GST_DEBUG_OBJECT (stream, "make RTP encoder for session %u", session);
2044 oldenc = priv->srtpenc;
2045 enc = get_rtp_encoder (stream, session);
2046 name = g_strdup_printf ("rtp_sink_%d", session);
2047 pad = gst_element_get_request_pad (enc, name);
2049 gst_object_unref (pad);
2052 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTP_ENCODER], 0,
2059 request_rtcp_encoder (GstElement * rtpbin, guint session,
2060 GstRTSPStream * stream)
2062 GstRTSPStreamPrivate *priv = stream->priv;
2063 GstElement *oldenc, *enc;
2067 if (priv->idx != session)
2070 GST_DEBUG_OBJECT (stream, "make RTCP encoder for session %u", session);
2072 oldenc = priv->srtpenc;
2073 enc = get_rtp_encoder (stream, session);
2074 name = g_strdup_printf ("rtcp_sink_%d", session);
2075 pad = gst_element_get_request_pad (enc, name);
2077 gst_object_unref (pad);
2080 g_signal_emit (stream, gst_rtsp_stream_signals[SIGNAL_NEW_RTCP_ENCODER], 0,
2087 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
2089 GstRTSPStreamPrivate *priv = stream->priv;
2092 GST_DEBUG ("request key %08x", ssrc);
2094 g_mutex_lock (&priv->lock);
2095 if ((caps = g_hash_table_lookup (priv->keys, GINT_TO_POINTER (ssrc))))
2096 gst_caps_ref (caps);
2097 g_mutex_unlock (&priv->lock);
2103 request_rtp_rtcp_decoder (GstElement * rtpbin, guint session,
2104 GstRTSPStream * stream)
2106 GstRTSPStreamPrivate *priv = stream->priv;
2108 if (priv->idx != session)
2111 if (priv->srtpdec == NULL) {
2114 name = g_strdup_printf ("srtpdec_%u", session);
2115 priv->srtpdec = gst_element_factory_make ("srtpdec", name);
2118 g_signal_connect (priv->srtpdec, "request-key",
2119 (GCallback) request_key, stream);
2121 return gst_object_ref (priv->srtpdec);
2125 * gst_rtsp_stream_request_aux_sender:
2126 * @stream: a #GstRTSPStream
2127 * @sessid: the session id
2129 * Creating a rtxsend bin
2131 * Returns: (transfer full): a #GstElement.
2136 gst_rtsp_stream_request_aux_sender (GstRTSPStream * stream, guint sessid)
2140 GstStructure *pt_map;
2145 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
2147 pt = gst_rtsp_stream_get_pt (stream);
2148 pt_s = g_strdup_printf ("%u", pt);
2149 rtx_pt = stream->priv->rtx_pt;
2151 GST_INFO ("creating rtxsend with pt %u to %u", pt, rtx_pt);
2153 bin = gst_bin_new (NULL);
2154 stream->priv->rtxsend = gst_element_factory_make ("rtprtxsend", NULL);
2155 pt_map = gst_structure_new ("application/x-rtp-pt-map",
2156 pt_s, G_TYPE_UINT, rtx_pt, NULL);
2157 g_object_set (stream->priv->rtxsend, "payload-type-map", pt_map,
2158 "max-size-time", GST_TIME_AS_MSECONDS (stream->priv->rtx_time), NULL);
2160 gst_structure_free (pt_map);
2161 gst_bin_add (GST_BIN (bin), gst_object_ref (stream->priv->rtxsend));
2163 pad = gst_element_get_static_pad (stream->priv->rtxsend, "src");
2164 name = g_strdup_printf ("src_%u", sessid);
2165 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2167 gst_object_unref (pad);
2169 pad = gst_element_get_static_pad (stream->priv->rtxsend, "sink");
2170 name = g_strdup_printf ("sink_%u", sessid);
2171 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
2173 gst_object_unref (pad);
2179 * gst_rtsp_stream_set_pt_map:
2180 * @stream: a #GstRTSPStream
2184 * Configure a pt map between @pt and @caps.
2187 gst_rtsp_stream_set_pt_map (GstRTSPStream * stream, guint pt, GstCaps * caps)
2189 GstRTSPStreamPrivate *priv = stream->priv;
2191 g_mutex_lock (&priv->lock);
2192 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps));
2193 g_mutex_unlock (&priv->lock);
2197 request_pt_map (GstElement * rtpbin, guint session, guint pt,
2198 GstRTSPStream * stream)
2200 GstRTSPStreamPrivate *priv = stream->priv;
2201 GstCaps *caps = NULL;
2203 g_mutex_lock (&priv->lock);
2205 if (priv->idx == session) {
2206 caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (pt));
2208 GST_DEBUG ("Stream %p, pt %u: caps %" GST_PTR_FORMAT, stream, pt, caps);
2209 gst_caps_ref (caps);
2211 GST_DEBUG ("Stream %p, pt %u: no caps", stream, pt);
2215 g_mutex_unlock (&priv->lock);
2221 pad_added (GstElement * rtpbin, GstPad * pad, GstRTSPStream * stream)
2223 GstRTSPStreamPrivate *priv = stream->priv;
2225 GstPadLinkReturn ret;
2228 GST_DEBUG ("Stream %p added pad %s:%s for pad %s:%s", stream,
2229 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2231 name = gst_pad_get_name (pad);
2232 if (sscanf (name, "recv_rtp_src_%u", &sessid) != 1) {
2238 if (priv->idx != sessid)
2241 if (gst_pad_is_linked (priv->sinkpad)) {
2242 GST_WARNING ("Stream %p: Pad %s:%s is linked already", stream,
2243 GST_DEBUG_PAD_NAME (priv->sinkpad));
2247 /* link the RTP pad to the session manager, it should not really fail unless
2248 * this is not really an RTP pad */
2249 ret = gst_pad_link (pad, priv->sinkpad);
2250 if (ret != GST_PAD_LINK_OK)
2252 priv->recv_rtp_src = gst_object_ref (pad);
2259 GST_ERROR ("Stream %p: Failed to link pads %s:%s and %s:%s", stream,
2260 GST_DEBUG_PAD_NAME (pad), GST_DEBUG_PAD_NAME (priv->sinkpad));
2265 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc,
2266 GstRTSPStream * stream)
2268 /* TODO: What to do here other than this? */
2269 GST_DEBUG ("Stream %p: Got EOS", stream);
2270 gst_pad_send_event (stream->priv->sinkpad, gst_event_new_eos ());
2273 /* must be called with lock */
2275 create_sender_part (GstRTSPStream * stream, GstBin * bin,
2278 GstRTSPStreamPrivate *priv;
2279 GstPad *pad, *sinkpad = NULL;
2280 gboolean is_tcp = FALSE, is_udp = FALSE;
2283 priv = stream->priv;
2285 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2286 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2287 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2289 if (is_udp && !create_and_configure_udpsinks (stream))
2290 goto no_udp_protocol;
2292 for (i = 0; i < 2; i++) {
2293 GstPad *teepad, *queuepad;
2294 /* For the sender we create this bit of pipeline for both
2295 * RTP and RTCP. Sync and preroll are enabled on udpsink so
2296 * we need to add a queue before appsink and udpsink to make
2297 * the pipeline not block. For the TCP case, we want to pump
2298 * client as fast as possible anyway. This pipeline is used
2299 * when both TCP and UDP are present.
2301 * .--------. .-----. .---------. .---------.
2302 * | rtpbin | | tee | | queue | | udpsink |
2303 * | send->sink src->sink src->sink |
2304 * '--------' | | '---------' '---------'
2305 * | | .---------. .---------.
2306 * | | | queue | | appsink |
2307 * | src->sink src->sink |
2308 * '-----' '---------' '---------'
2310 * When only UDP or only TCP is allowed, we skip the tee and queue
2311 * and link the udpsink (for UDP) or appsink (for TCP) directly to
2314 /* Only link the RTP send src if we're going to send RTP, link
2315 * the RTCP send src always */
2316 if (priv->srcpad || i == 1) {
2319 gst_bin_add (bin, priv->udpsink[i]);
2320 sinkpad = gst_element_get_static_pad (priv->udpsink[i], "sink");
2325 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
2326 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
2327 gst_bin_add (bin, priv->appsink[i]);
2328 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
2329 &sink_cb, stream, NULL);
2332 if (is_udp && is_tcp) {
2333 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2335 /* make tee for RTP/RTCP */
2336 priv->tee[i] = gst_element_factory_make ("tee", NULL);
2337 gst_bin_add (bin, priv->tee[i]);
2339 /* and link to rtpbin send pad */
2340 pad = gst_element_get_static_pad (priv->tee[i], "sink");
2341 gst_pad_link (priv->send_src[i], pad);
2342 gst_object_unref (pad);
2344 priv->udpqueue[i] = gst_element_factory_make ("queue", NULL);
2345 g_object_set (priv->udpqueue[i], "max-size-buffers",
2346 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2348 gst_bin_add (bin, priv->udpqueue[i]);
2349 /* link tee to udpqueue */
2350 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2351 pad = gst_element_get_static_pad (priv->udpqueue[i], "sink");
2352 gst_pad_link (teepad, pad);
2353 gst_object_unref (pad);
2354 gst_object_unref (teepad);
2356 /* link udpqueue to udpsink */
2357 queuepad = gst_element_get_static_pad (priv->udpqueue[i], "src");
2358 gst_pad_link (queuepad, sinkpad);
2359 gst_object_unref (queuepad);
2360 gst_object_unref (sinkpad);
2363 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
2364 g_object_set (priv->appqueue[i], "max-size-buffers",
2365 1, "max-size-bytes", 0, "max-size-time", G_GINT64_CONSTANT (0),
2367 gst_bin_add (bin, priv->appqueue[i]);
2368 /* and link tee to appqueue */
2369 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
2370 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
2371 gst_pad_link (teepad, pad);
2372 gst_object_unref (pad);
2373 gst_object_unref (teepad);
2375 /* and link appqueue to appsink */
2376 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
2377 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2378 gst_pad_link (queuepad, pad);
2379 gst_object_unref (pad);
2380 gst_object_unref (queuepad);
2381 } else if (is_tcp) {
2382 /* only appsink needed, link it to the session */
2383 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
2384 gst_pad_link (priv->send_src[i], pad);
2385 gst_object_unref (pad);
2387 /* when its only TCP, we need to set sync and preroll to FALSE
2388 * for the sink to avoid deadlock. And this is only needed for
2389 * sink used for RTCP data, not the RTP data. */
2391 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
2393 /* else only udpsink needed, link it to the session */
2394 gst_pad_link (priv->send_src[i], sinkpad);
2395 gst_object_unref (sinkpad);
2399 /* check if we need to set to a special state */
2400 if (state != GST_STATE_NULL) {
2401 if (priv->udpsink[i] && (priv->srcpad || i == 1))
2402 gst_element_set_state (priv->udpsink[i], state);
2403 if (priv->appsink[i] && (priv->srcpad || i == 1))
2404 gst_element_set_state (priv->appsink[i], state);
2405 if (priv->appqueue[i] && (priv->srcpad || i == 1))
2406 gst_element_set_state (priv->appqueue[i], state);
2407 if (priv->udpqueue[i] && (priv->srcpad || i == 1))
2408 gst_element_set_state (priv->udpqueue[i], state);
2409 if (priv->tee[i] && (priv->srcpad || i == 1))
2410 gst_element_set_state (priv->tee[i], state);
2423 /* must be called with lock */
2425 create_receiver_part (GstRTSPStream * stream, GstBin * bin,
2428 GstRTSPStreamPrivate *priv;
2429 GstPad *pad, *selpad;
2433 priv = stream->priv;
2435 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2437 for (i = 0; i < 2; i++) {
2438 /* Only connect recv RTP sink if we expect to receive RTP. Connect recv
2439 * RTCP sink always */
2440 if (priv->sinkpad || i == 1) {
2441 /* For the receiver we create this bit of pipeline for both
2442 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
2443 * and it is all funneled into the rtpbin receive pad.
2445 * .--------. .--------. .--------.
2446 * | udpsrc | | funnel | | rtpbin |
2447 * | src->sink src->sink |
2448 * '--------' | | '--------'
2452 * '--------' '--------'
2454 /* make funnel for the RTP/RTCP receivers */
2455 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
2456 gst_bin_add (bin, priv->funnel[i]);
2458 pad = gst_element_get_static_pad (priv->funnel[i], "src");
2459 gst_pad_link (pad, priv->recv_sink[i]);
2460 gst_object_unref (pad);
2462 if (priv->udpsrc_v4[i]) {
2464 /* we set and keep these to playing so that they don't cause NO_PREROLL return
2465 * values. This is only relevant for PLAY pipelines */
2466 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_PLAYING);
2467 gst_element_set_locked_state (priv->udpsrc_v4[i], TRUE);
2470 gst_bin_add (bin, priv->udpsrc_v4[i]);
2472 /* and link to the funnel v4 */
2473 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2474 pad = gst_element_get_static_pad (priv->udpsrc_v4[i], "src");
2475 gst_pad_link (pad, selpad);
2476 gst_object_unref (pad);
2477 gst_object_unref (selpad);
2480 if (priv->udpsrc_v6[i]) {
2482 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_PLAYING);
2483 gst_element_set_locked_state (priv->udpsrc_v6[i], TRUE);
2485 gst_bin_add (bin, priv->udpsrc_v6[i]);
2487 /* and link to the funnel v6 */
2488 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2489 pad = gst_element_get_static_pad (priv->udpsrc_v6[i], "src");
2490 gst_pad_link (pad, selpad);
2491 gst_object_unref (pad);
2492 gst_object_unref (selpad);
2496 /* make and add appsrc */
2497 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
2498 priv->appsrc_base_time[i] = -1;
2499 g_object_set (priv->appsrc[i], "format", GST_FORMAT_TIME, NULL);
2500 gst_bin_add (bin, priv->appsrc[i]);
2501 /* and link to the funnel */
2502 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
2503 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
2504 gst_pad_link (pad, selpad);
2505 gst_object_unref (pad);
2506 gst_object_unref (selpad);
2510 /* check if we need to set to a special state */
2511 if (state != GST_STATE_NULL) {
2512 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2513 gst_element_set_state (priv->funnel[i], state);
2514 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2515 gst_element_set_state (priv->appsrc[i], state);
2521 * gst_rtsp_stream_join_bin:
2522 * @stream: a #GstRTSPStream
2523 * @bin: (transfer none): a #GstBin to join
2524 * @rtpbin: (transfer none): a rtpbin element in @bin
2525 * @state: the target state of the new elements
2527 * Join the #GstBin @bin that contains the element @rtpbin.
2529 * @stream will link to @rtpbin, which must be inside @bin. The elements
2530 * added to @bin will be set to the state given in @state.
2532 * Returns: %TRUE on success.
2535 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
2536 GstElement * rtpbin, GstState state)
2538 GstRTSPStreamPrivate *priv;
2541 GstPadLinkReturn ret;
2543 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2544 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2545 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2547 priv = stream->priv;
2549 g_mutex_lock (&priv->lock);
2550 if (priv->is_joined)
2553 /* create a session with the same index as the stream */
2556 GST_INFO ("stream %p joining bin as session %u", stream, idx);
2558 if (priv->profiles & GST_RTSP_PROFILE_SAVP
2559 || priv->profiles & GST_RTSP_PROFILE_SAVPF) {
2561 g_signal_connect (rtpbin, "request-rtp-encoder",
2562 (GCallback) request_rtp_encoder, stream);
2563 g_signal_connect (rtpbin, "request-rtcp-encoder",
2564 (GCallback) request_rtcp_encoder, stream);
2565 g_signal_connect (rtpbin, "request-rtp-decoder",
2566 (GCallback) request_rtp_rtcp_decoder, stream);
2567 g_signal_connect (rtpbin, "request-rtcp-decoder",
2568 (GCallback) request_rtp_rtcp_decoder, stream);
2571 if (priv->sinkpad) {
2572 g_signal_connect (rtpbin, "request-pt-map",
2573 (GCallback) request_pt_map, stream);
2576 /* get pads from the RTP session element for sending and receiving
2579 /* get a pad for sending RTP */
2580 name = g_strdup_printf ("send_rtp_sink_%u", idx);
2581 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
2584 /* link the RTP pad to the session manager, it should not really fail unless
2585 * this is not really an RTP pad */
2586 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
2587 if (ret != GST_PAD_LINK_OK)
2590 name = g_strdup_printf ("send_rtp_src_%u", idx);
2591 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
2594 /* Need to connect our sinkpad from here */
2595 g_signal_connect (rtpbin, "pad-added", (GCallback) pad_added, stream);
2597 g_signal_connect (rtpbin, "on-npt-stop", (GCallback) on_npt_stop, stream);
2599 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
2600 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
2604 name = g_strdup_printf ("send_rtcp_src_%u", idx);
2605 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
2607 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
2608 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
2611 /* get the session */
2612 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
2614 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
2616 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
2618 g_signal_connect (priv->session, "on-ssrc-active",
2619 (GCallback) on_ssrc_active, stream);
2620 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
2622 g_signal_connect (priv->session, "on-bye-timeout",
2623 (GCallback) on_bye_timeout, stream);
2624 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
2627 /* signal for sender ssrc */
2628 g_signal_connect (priv->session, "on-new-sender-ssrc",
2629 (GCallback) on_new_sender_ssrc, stream);
2630 g_signal_connect (priv->session, "on-sender-ssrc-active",
2631 (GCallback) on_sender_ssrc_active, stream);
2633 if (!create_sender_part (stream, bin, state))
2634 goto no_udp_protocol;
2636 create_receiver_part (stream, bin, state);
2639 /* be notified of caps changes */
2640 priv->caps_sig = g_signal_connect (priv->send_src[0], "notify::caps",
2641 (GCallback) caps_notify, stream);
2644 priv->is_joined = TRUE;
2645 g_mutex_unlock (&priv->lock);
2652 g_mutex_unlock (&priv->lock);
2657 GST_WARNING ("failed to link stream %u", idx);
2658 gst_object_unref (priv->send_rtp_sink);
2659 priv->send_rtp_sink = NULL;
2660 g_mutex_unlock (&priv->lock);
2665 GST_WARNING ("failed to allocate ports %u", idx);
2666 gst_object_unref (priv->send_rtp_sink);
2667 priv->send_rtp_sink = NULL;
2668 gst_object_unref (priv->send_src[0]);
2669 priv->send_src[0] = NULL;
2670 gst_object_unref (priv->send_src[1]);
2671 priv->send_src[1] = NULL;
2672 gst_object_unref (priv->recv_sink[0]);
2673 priv->recv_sink[0] = NULL;
2674 gst_object_unref (priv->recv_sink[1]);
2675 priv->recv_sink[1] = NULL;
2676 if (priv->udpsink[0])
2677 gst_element_set_state (priv->udpsink[0], GST_STATE_NULL);
2678 if (priv->udpsink[1])
2679 gst_element_set_state (priv->udpsink[1], GST_STATE_NULL);
2680 if (priv->udpsrc_v4[0]) {
2681 gst_element_set_state (priv->udpsrc_v4[0], GST_STATE_NULL);
2682 gst_object_unref (priv->udpsrc_v4[0]);
2683 priv->udpsrc_v4[0] = NULL;
2685 if (priv->udpsrc_v4[1]) {
2686 gst_element_set_state (priv->udpsrc_v4[1], GST_STATE_NULL);
2687 gst_object_unref (priv->udpsrc_v4[1]);
2688 priv->udpsrc_v4[1] = NULL;
2690 if (priv->udpsrc_mcast_v4[0]) {
2691 gst_element_set_state (priv->udpsrc_mcast_v4[0], GST_STATE_NULL);
2692 gst_object_unref (priv->udpsrc_mcast_v4[0]);
2693 priv->udpsrc_mcast_v4[0] = NULL;
2695 if (priv->udpsrc_mcast_v4[1]) {
2696 gst_element_set_state (priv->udpsrc_mcast_v4[1], GST_STATE_NULL);
2697 gst_object_unref (priv->udpsrc_mcast_v4[1]);
2698 priv->udpsrc_mcast_v4[1] = NULL;
2700 if (priv->udpsrc_v6[0]) {
2701 gst_element_set_state (priv->udpsrc_v6[0], GST_STATE_NULL);
2702 gst_object_unref (priv->udpsrc_v6[0]);
2703 priv->udpsrc_v6[0] = NULL;
2705 if (priv->udpsrc_v6[1]) {
2706 gst_element_set_state (priv->udpsrc_v6[1], GST_STATE_NULL);
2707 gst_object_unref (priv->udpsrc_v6[1]);
2708 priv->udpsrc_v6[1] = NULL;
2710 if (priv->udpsrc_mcast_v6[0]) {
2711 gst_element_set_state (priv->udpsrc_mcast_v6[0], GST_STATE_NULL);
2712 gst_object_unref (priv->udpsrc_mcast_v6[0]);
2713 priv->udpsrc_mcast_v6[0] = NULL;
2715 if (priv->udpsrc_mcast_v6[1]) {
2716 gst_element_set_state (priv->udpsrc_mcast_v6[1], GST_STATE_NULL);
2717 gst_object_unref (priv->udpsrc_mcast_v6[1]);
2718 priv->udpsrc_mcast_v6[1] = NULL;
2720 g_mutex_unlock (&priv->lock);
2726 * gst_rtsp_stream_leave_bin:
2727 * @stream: a #GstRTSPStream
2728 * @bin: (transfer none): a #GstBin
2729 * @rtpbin: (transfer none): a rtpbin #GstElement
2731 * Remove the elements of @stream from @bin.
2733 * Return: %TRUE on success.
2736 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
2737 GstElement * rtpbin)
2739 GstRTSPStreamPrivate *priv;
2741 gboolean is_tcp, is_udp;
2743 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2744 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
2745 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
2747 priv = stream->priv;
2749 g_mutex_lock (&priv->lock);
2750 if (!priv->is_joined)
2751 goto was_not_joined;
2753 /* all transports must be removed by now */
2754 if (priv->transports != NULL)
2755 goto transports_not_removed;
2757 clear_tr_cache (priv, TRUE);
2758 clear_tr_cache (priv, FALSE);
2760 GST_INFO ("stream %p leaving bin", stream);
2763 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
2765 g_signal_handler_disconnect (priv->send_src[0], priv->caps_sig);
2766 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
2767 gst_object_unref (priv->send_rtp_sink);
2768 priv->send_rtp_sink = NULL;
2769 } else if (priv->recv_rtp_src) {
2770 gst_pad_unlink (priv->recv_rtp_src, priv->sinkpad);
2771 gst_object_unref (priv->recv_rtp_src);
2772 priv->recv_rtp_src = NULL;
2775 is_tcp = priv->protocols & GST_RTSP_LOWER_TRANS_TCP;
2777 is_udp = ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
2778 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST));
2781 for (i = 0; i < 2; i++) {
2782 if (priv->udpsink[i])
2783 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
2784 if (priv->appsink[i])
2785 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
2786 if (priv->appqueue[i])
2787 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
2788 if (priv->udpqueue[i])
2789 gst_element_set_state (priv->udpqueue[i], GST_STATE_NULL);
2791 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
2792 if (priv->funnel[i])
2793 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
2794 if (priv->appsrc[i])
2795 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
2797 if (priv->udpsrc_v4[i]) {
2798 if (priv->sinkpad || i == 1) {
2799 /* and set udpsrc to NULL now before removing */
2800 gst_element_set_locked_state (priv->udpsrc_v4[i], FALSE);
2801 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2802 /* removing them should also nicely release the request
2803 * pads when they finalize */
2804 gst_bin_remove (bin, priv->udpsrc_v4[i]);
2806 /* we need to set the state to NULL before unref */
2807 gst_element_set_state (priv->udpsrc_v4[i], GST_STATE_NULL);
2808 gst_object_unref (priv->udpsrc_v4[i]);
2812 if (priv->udpsrc_mcast_v4[i]) {
2813 if (priv->sinkpad || i == 1) {
2814 /* and set udpsrc to NULL now before removing */
2815 gst_element_set_locked_state (priv->udpsrc_mcast_v4[i], FALSE);
2816 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2817 /* removing them should also nicely release the request
2818 * pads when they finalize */
2819 gst_bin_remove (bin, priv->udpsrc_mcast_v4[i]);
2821 gst_element_set_state (priv->udpsrc_mcast_v4[i], GST_STATE_NULL);
2822 gst_object_unref (priv->udpsrc_mcast_v4[i]);
2826 if (priv->udpsrc_v6[i]) {
2827 if (priv->sinkpad || i == 1) {
2828 gst_element_set_locked_state (priv->udpsrc_v6[i], FALSE);
2829 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2830 gst_bin_remove (bin, priv->udpsrc_v6[i]);
2832 gst_element_set_state (priv->udpsrc_v6[i], GST_STATE_NULL);
2833 gst_object_unref (priv->udpsrc_v6[i]);
2836 if (priv->udpsrc_mcast_v6[i]) {
2837 if (priv->sinkpad || i == 1) {
2838 gst_element_set_locked_state (priv->udpsrc_mcast_v6[i], FALSE);
2839 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2840 gst_bin_remove (bin, priv->udpsrc_mcast_v6[i]);
2842 gst_element_set_state (priv->udpsrc_mcast_v6[i], GST_STATE_NULL);
2843 gst_object_unref (priv->udpsrc_mcast_v6[i]);
2847 if (priv->udpsink[i] && is_udp && (priv->srcpad || i == 1))
2848 gst_bin_remove (bin, priv->udpsink[i]);
2849 if (priv->appsrc[i] && (priv->sinkpad || i == 1))
2850 gst_bin_remove (bin, priv->appsrc[i]);
2851 if (priv->appsink[i] && is_tcp && (priv->srcpad || i == 1))
2852 gst_bin_remove (bin, priv->appsink[i]);
2853 if (priv->appqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2854 gst_bin_remove (bin, priv->appqueue[i]);
2855 if (priv->udpqueue[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2856 gst_bin_remove (bin, priv->udpqueue[i]);
2857 if (priv->tee[i] && is_tcp && is_udp && (priv->srcpad || i == 1))
2858 gst_bin_remove (bin, priv->tee[i]);
2859 if (priv->funnel[i] && (priv->sinkpad || i == 1))
2860 gst_bin_remove (bin, priv->funnel[i]);
2862 if (priv->sinkpad || i == 1) {
2863 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
2864 gst_object_unref (priv->recv_sink[i]);
2865 priv->recv_sink[i] = NULL;
2868 priv->udpsrc_v4[i] = NULL;
2869 priv->udpsrc_v6[i] = NULL;
2870 priv->udpsrc_mcast_v4[i] = NULL;
2871 priv->udpsrc_mcast_v6[i] = NULL;
2872 priv->udpsink[i] = NULL;
2873 priv->appsrc[i] = NULL;
2874 priv->appsink[i] = NULL;
2875 priv->appqueue[i] = NULL;
2876 priv->udpqueue[i] = NULL;
2877 priv->tee[i] = NULL;
2878 priv->funnel[i] = NULL;
2882 gst_object_unref (priv->send_src[0]);
2883 priv->send_src[0] = NULL;
2886 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
2887 gst_object_unref (priv->send_src[1]);
2888 priv->send_src[1] = NULL;
2890 g_object_unref (priv->session);
2891 priv->session = NULL;
2893 gst_caps_unref (priv->caps);
2897 gst_object_unref (priv->srtpenc);
2899 gst_object_unref (priv->srtpdec);
2901 priv->is_joined = FALSE;
2902 g_mutex_unlock (&priv->lock);
2908 g_mutex_unlock (&priv->lock);
2911 transports_not_removed:
2913 GST_ERROR_OBJECT (stream, "can't leave bin (transports not removed)");
2914 g_mutex_unlock (&priv->lock);
2920 * gst_rtsp_stream_get_rtpinfo:
2921 * @stream: a #GstRTSPStream
2922 * @rtptime: (allow-none): result RTP timestamp
2923 * @seq: (allow-none): result RTP seqnum
2924 * @clock_rate: (allow-none): the clock rate
2925 * @running_time: (allow-none): result running-time
2927 * Retrieve the current rtptime, seq and running-time. This is used to
2928 * construct a RTPInfo reply header.
2930 * Returns: %TRUE when rtptime, seq and running-time could be determined.
2933 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
2934 guint * rtptime, guint * seq, guint * clock_rate,
2935 GstClockTime * running_time)
2937 GstRTSPStreamPrivate *priv;
2938 GstStructure *stats;
2939 GObjectClass *payobjclass;
2941 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
2943 priv = stream->priv;
2945 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
2947 g_mutex_lock (&priv->lock);
2949 /* First try to extract the information from the last buffer on the sinks.
2950 * This will have a more accurate sequence number and timestamp, as between
2951 * the payloader and the sink there can be some queues
2953 if (priv->udpsink[0] || priv->appsink[0]) {
2954 GstSample *last_sample;
2956 if (priv->udpsink[0])
2957 g_object_get (priv->udpsink[0], "last-sample", &last_sample, NULL);
2959 g_object_get (priv->appsink[0], "last-sample", &last_sample, NULL);
2964 GstSegment *segment;
2965 GstRTPBuffer rtp_buffer = GST_RTP_BUFFER_INIT;
2967 caps = gst_sample_get_caps (last_sample);
2968 buffer = gst_sample_get_buffer (last_sample);
2969 segment = gst_sample_get_segment (last_sample);
2971 if (gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp_buffer)) {
2973 *seq = gst_rtp_buffer_get_seq (&rtp_buffer);
2977 *rtptime = gst_rtp_buffer_get_timestamp (&rtp_buffer);
2980 gst_rtp_buffer_unmap (&rtp_buffer);
2984 gst_segment_to_running_time (segment, GST_FORMAT_TIME,
2985 GST_BUFFER_TIMESTAMP (buffer));
2989 GstStructure *s = gst_caps_get_structure (caps, 0);
2991 gst_structure_get_int (s, "clock-rate", (gint *) clock_rate);
2993 if (*clock_rate == 0 && running_time)
2994 *running_time = GST_CLOCK_TIME_NONE;
2996 gst_sample_unref (last_sample);
3000 gst_sample_unref (last_sample);
3005 if (g_object_class_find_property (payobjclass, "stats")) {
3006 g_object_get (priv->payloader, "stats", &stats, NULL);
3011 gst_structure_get_uint (stats, "seqnum", seq);
3014 gst_structure_get_uint (stats, "timestamp", rtptime);
3017 gst_structure_get_clock_time (stats, "running-time", running_time);
3020 gst_structure_get_uint (stats, "clock-rate", clock_rate);
3021 if (*clock_rate == 0 && running_time)
3022 *running_time = GST_CLOCK_TIME_NONE;
3024 gst_structure_free (stats);
3026 if (!g_object_class_find_property (payobjclass, "seqnum") ||
3027 !g_object_class_find_property (payobjclass, "timestamp"))
3031 g_object_get (priv->payloader, "seqnum", seq, NULL);
3034 g_object_get (priv->payloader, "timestamp", rtptime, NULL);
3037 *running_time = GST_CLOCK_TIME_NONE;
3041 g_mutex_unlock (&priv->lock);
3048 GST_WARNING ("Could not get payloader stats");
3049 g_mutex_unlock (&priv->lock);
3055 * gst_rtsp_stream_get_caps:
3056 * @stream: a #GstRTSPStream
3058 * Retrieve the current caps of @stream.
3060 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
3064 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
3066 GstRTSPStreamPrivate *priv;
3069 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3071 priv = stream->priv;
3073 g_mutex_lock (&priv->lock);
3074 if ((result = priv->caps))
3075 gst_caps_ref (result);
3076 g_mutex_unlock (&priv->lock);
3082 * gst_rtsp_stream_recv_rtp:
3083 * @stream: a #GstRTSPStream
3084 * @buffer: (transfer full): a #GstBuffer
3086 * Handle an RTP buffer for the stream. This method is usually called when a
3087 * message has been received from a client using the TCP transport.
3089 * This function takes ownership of @buffer.
3091 * Returns: a GstFlowReturn.
3094 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
3096 GstRTSPStreamPrivate *priv;
3098 GstElement *element;
3100 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3101 priv = stream->priv;
3102 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3103 g_return_val_if_fail (priv->is_joined, FALSE);
3105 g_mutex_lock (&priv->lock);
3106 if (priv->appsrc[0])
3107 element = gst_object_ref (priv->appsrc[0]);
3110 g_mutex_unlock (&priv->lock);
3113 if (priv->appsrc_base_time[0] == -1) {
3114 /* Take current running_time. This timestamp will be put on
3115 * the first buffer of each stream because we are a live source and so we
3116 * timestamp with the running_time. When we are dealing with TCP, we also
3117 * only timestamp the first buffer (using the DISCONT flag) because a server
3118 * typically bursts data, for which we don't want to compensate by speeding
3119 * up the media. The other timestamps will be interpollated from this one
3120 * using the RTP timestamps. */
3121 GST_OBJECT_LOCK (element);
3122 if (GST_ELEMENT_CLOCK (element)) {
3124 GstClockTime base_time;
3126 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3127 base_time = GST_ELEMENT_CAST (element)->base_time;
3129 priv->appsrc_base_time[0] = now - base_time;
3130 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[0];
3131 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3132 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3133 GST_TIME_ARGS (base_time));
3135 GST_OBJECT_UNLOCK (element);
3138 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3139 gst_object_unref (element);
3147 * gst_rtsp_stream_recv_rtcp:
3148 * @stream: a #GstRTSPStream
3149 * @buffer: (transfer full): a #GstBuffer
3151 * Handle an RTCP buffer for the stream. This method is usually called when a
3152 * message has been received from a client using the TCP transport.
3154 * This function takes ownership of @buffer.
3156 * Returns: a GstFlowReturn.
3159 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
3161 GstRTSPStreamPrivate *priv;
3163 GstElement *element;
3165 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
3166 priv = stream->priv;
3167 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
3169 if (!priv->is_joined) {
3170 gst_buffer_unref (buffer);
3171 return GST_FLOW_NOT_LINKED;
3173 g_mutex_lock (&priv->lock);
3174 if (priv->appsrc[1])
3175 element = gst_object_ref (priv->appsrc[1]);
3178 g_mutex_unlock (&priv->lock);
3181 if (priv->appsrc_base_time[1] == -1) {
3182 /* Take current running_time. This timestamp will be put on
3183 * the first buffer of each stream because we are a live source and so we
3184 * timestamp with the running_time. When we are dealing with TCP, we also
3185 * only timestamp the first buffer (using the DISCONT flag) because a server
3186 * typically bursts data, for which we don't want to compensate by speeding
3187 * up the media. The other timestamps will be interpollated from this one
3188 * using the RTP timestamps. */
3189 GST_OBJECT_LOCK (element);
3190 if (GST_ELEMENT_CLOCK (element)) {
3192 GstClockTime base_time;
3194 now = gst_clock_get_time (GST_ELEMENT_CLOCK (element));
3195 base_time = GST_ELEMENT_CAST (element)->base_time;
3197 priv->appsrc_base_time[1] = now - base_time;
3198 GST_BUFFER_TIMESTAMP (buffer) = priv->appsrc_base_time[1];
3199 GST_DEBUG ("stream %p: first buffer at time %" GST_TIME_FORMAT
3200 ", base %" GST_TIME_FORMAT, stream, GST_TIME_ARGS (now),
3201 GST_TIME_ARGS (base_time));
3203 GST_OBJECT_UNLOCK (element);
3206 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
3207 gst_object_unref (element);
3210 gst_buffer_unref (buffer);
3215 /* must be called with lock */
3217 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
3220 GstRTSPStreamPrivate *priv = stream->priv;
3221 const GstRTSPTransport *tr;
3223 tr = gst_rtsp_stream_transport_get_transport (trans);
3225 switch (tr->lower_transport) {
3226 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
3227 case GST_RTSP_LOWER_TRANS_UDP:
3233 dest = tr->destination;
3234 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3238 } else if (priv->client_side) {
3239 /* In client side mode the 'destination' is the RTSP server, so send
3241 min = tr->server_port.min;
3242 max = tr->server_port.max;
3244 min = tr->client_port.min;
3245 max = tr->client_port.max;
3250 GST_INFO ("setting ttl-mc %d", ttl);
3251 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
3252 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
3254 GST_INFO ("adding %s:%d-%d", dest, min, max);
3255 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
3256 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
3257 priv->transports = g_list_prepend (priv->transports, trans);
3259 GST_INFO ("removing %s:%d-%d", dest, min, max);
3260 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
3261 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
3262 priv->transports = g_list_remove (priv->transports, trans);
3264 priv->transports_cookie++;
3267 case GST_RTSP_LOWER_TRANS_TCP:
3269 GST_INFO ("adding TCP %s", tr->destination);
3270 priv->transports = g_list_prepend (priv->transports, trans);
3272 GST_INFO ("removing TCP %s", tr->destination);
3273 priv->transports = g_list_remove (priv->transports, trans);
3275 priv->transports_cookie++;
3278 goto unknown_transport;
3285 GST_INFO ("Unknown transport %d", tr->lower_transport);
3292 * gst_rtsp_stream_add_transport:
3293 * @stream: a #GstRTSPStream
3294 * @trans: (transfer none): a #GstRTSPStreamTransport
3296 * Add the transport in @trans to @stream. The media of @stream will
3297 * then also be send to the values configured in @trans.
3299 * @stream must be joined to a bin.
3301 * @trans must contain a valid #GstRTSPTransport.
3303 * Returns: %TRUE if @trans was added
3306 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
3307 GstRTSPStreamTransport * trans)
3309 GstRTSPStreamPrivate *priv;
3312 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3313 priv = stream->priv;
3314 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3315 g_return_val_if_fail (priv->is_joined, FALSE);
3317 g_mutex_lock (&priv->lock);
3318 res = update_transport (stream, trans, TRUE);
3319 g_mutex_unlock (&priv->lock);
3325 * gst_rtsp_stream_remove_transport:
3326 * @stream: a #GstRTSPStream
3327 * @trans: (transfer none): a #GstRTSPStreamTransport
3329 * Remove the transport in @trans from @stream. The media of @stream will
3330 * not be sent to the values configured in @trans.
3332 * @stream must be joined to a bin.
3334 * @trans must contain a valid #GstRTSPTransport.
3336 * Returns: %TRUE if @trans was removed
3339 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
3340 GstRTSPStreamTransport * trans)
3342 GstRTSPStreamPrivate *priv;
3345 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3346 priv = stream->priv;
3347 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
3348 g_return_val_if_fail (priv->is_joined, FALSE);
3350 g_mutex_lock (&priv->lock);
3351 res = update_transport (stream, trans, FALSE);
3352 g_mutex_unlock (&priv->lock);
3358 * gst_rtsp_stream_update_crypto:
3359 * @stream: a #GstRTSPStream
3361 * @crypto: (transfer none) (allow-none): a #GstCaps with crypto info
3363 * Update the new crypto information for @ssrc in @stream. If information
3364 * for @ssrc did not exist, it will be added. If information
3365 * for @ssrc existed, it will be replaced. If @crypto is %NULL, it will
3366 * be removed from @stream.
3368 * Returns: %TRUE if @crypto could be updated
3371 gst_rtsp_stream_update_crypto (GstRTSPStream * stream,
3372 guint ssrc, GstCaps * crypto)
3374 GstRTSPStreamPrivate *priv;
3376 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3377 g_return_val_if_fail (crypto == NULL || GST_IS_CAPS (crypto), FALSE);
3379 priv = stream->priv;
3381 GST_DEBUG_OBJECT (stream, "update key for %08x", ssrc);
3383 g_mutex_lock (&priv->lock);
3385 g_hash_table_insert (priv->keys, GINT_TO_POINTER (ssrc),
3386 gst_caps_ref (crypto));
3388 g_hash_table_remove (priv->keys, GINT_TO_POINTER (ssrc));
3389 g_mutex_unlock (&priv->lock);
3395 * gst_rtsp_stream_get_rtp_socket:
3396 * @stream: a #GstRTSPStream
3397 * @family: the socket family
3399 * Get the RTP socket from @stream for a @family.
3401 * @stream must be joined to a bin.
3403 * Returns: (transfer full) (nullable): the RTP socket or %NULL if no
3404 * socket could be allocated for @family. Unref after usage
3407 gst_rtsp_stream_get_rtp_socket (GstRTSPStream * stream, GSocketFamily family)
3409 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3413 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3414 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3415 family == G_SOCKET_FAMILY_IPV6, NULL);
3416 g_return_val_if_fail (priv->udpsink[0], NULL);
3418 if (family == G_SOCKET_FAMILY_IPV6)
3423 g_object_get (priv->udpsink[0], name, &socket, NULL);
3429 * gst_rtsp_stream_get_rtcp_socket:
3430 * @stream: a #GstRTSPStream
3431 * @family: the socket family
3433 * Get the RTCP socket from @stream for a @family.
3435 * @stream must be joined to a bin.
3437 * Returns: (transfer full) (nullable): the RTCP socket or %NULL if no
3438 * socket could be allocated for @family. Unref after usage
3441 gst_rtsp_stream_get_rtcp_socket (GstRTSPStream * stream, GSocketFamily family)
3443 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
3447 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3448 g_return_val_if_fail (family == G_SOCKET_FAMILY_IPV4 ||
3449 family == G_SOCKET_FAMILY_IPV6, NULL);
3450 g_return_val_if_fail (priv->udpsink[1], NULL);
3452 if (family == G_SOCKET_FAMILY_IPV6)
3457 g_object_get (priv->udpsink[1], name, &socket, NULL);
3463 * gst_rtsp_stream_set_seqnum:
3464 * @stream: a #GstRTSPStream
3465 * @seqnum: a new sequence number
3467 * Configure the sequence number in the payloader of @stream to @seqnum.
3470 gst_rtsp_stream_set_seqnum_offset (GstRTSPStream * stream, guint16 seqnum)
3472 GstRTSPStreamPrivate *priv;
3474 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
3476 priv = stream->priv;
3478 g_object_set (G_OBJECT (priv->payloader), "seqnum-offset", seqnum, NULL);
3482 * gst_rtsp_stream_get_seqnum:
3483 * @stream: a #GstRTSPStream
3485 * Get the configured sequence number in the payloader of @stream.
3487 * Returns: the sequence number of the payloader.
3490 gst_rtsp_stream_get_current_seqnum (GstRTSPStream * stream)
3492 GstRTSPStreamPrivate *priv;
3495 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
3497 priv = stream->priv;
3499 g_object_get (G_OBJECT (priv->payloader), "seqnum", &seqnum, NULL);
3505 * gst_rtsp_stream_transport_filter:
3506 * @stream: a #GstRTSPStream
3507 * @func: (scope call) (allow-none): a callback
3508 * @user_data: (closure): user data passed to @func
3510 * Call @func for each transport managed by @stream. The result value of @func
3511 * determines what happens to the transport. @func will be called with @stream
3512 * locked so no further actions on @stream can be performed from @func.
3514 * If @func returns #GST_RTSP_FILTER_REMOVE, the transport will be removed from
3517 * If @func returns #GST_RTSP_FILTER_KEEP, the transport will remain in @stream.
3519 * If @func returns #GST_RTSP_FILTER_REF, the transport will remain in @stream but
3520 * will also be added with an additional ref to the result #GList of this
3523 * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each transport.
3525 * Returns: (element-type GstRTSPStreamTransport) (transfer full): a #GList with all
3526 * transports for which @func returned #GST_RTSP_FILTER_REF. After usage, each
3527 * element in the #GList should be unreffed before the list is freed.
3530 gst_rtsp_stream_transport_filter (GstRTSPStream * stream,
3531 GstRTSPStreamTransportFilterFunc func, gpointer user_data)
3533 GstRTSPStreamPrivate *priv;
3534 GList *result, *walk, *next;
3535 GHashTable *visited = NULL;
3538 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
3540 priv = stream->priv;
3544 visited = g_hash_table_new_full (NULL, NULL, g_object_unref, NULL);
3546 g_mutex_lock (&priv->lock);
3548 cookie = priv->transports_cookie;
3549 for (walk = priv->transports; walk; walk = next) {
3550 GstRTSPStreamTransport *trans = walk->data;
3551 GstRTSPFilterResult res;
3554 next = g_list_next (walk);
3557 /* only visit each transport once */
3558 if (g_hash_table_contains (visited, trans))
3561 g_hash_table_add (visited, g_object_ref (trans));
3562 g_mutex_unlock (&priv->lock);
3564 res = func (stream, trans, user_data);
3566 g_mutex_lock (&priv->lock);
3568 res = GST_RTSP_FILTER_REF;
3570 changed = (cookie != priv->transports_cookie);
3573 case GST_RTSP_FILTER_REMOVE:
3574 update_transport (stream, trans, FALSE);
3576 case GST_RTSP_FILTER_REF:
3577 result = g_list_prepend (result, g_object_ref (trans));
3579 case GST_RTSP_FILTER_KEEP:
3586 g_mutex_unlock (&priv->lock);
3589 g_hash_table_unref (visited);
3594 static GstPadProbeReturn
3595 pad_blocking (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
3597 GstRTSPStreamPrivate *priv;
3598 GstRTSPStream *stream;
3601 priv = stream->priv;
3603 GST_DEBUG_OBJECT (pad, "now blocking");
3605 g_mutex_lock (&priv->lock);
3606 priv->blocking = TRUE;
3607 g_mutex_unlock (&priv->lock);
3609 gst_element_post_message (priv->payloader,
3610 gst_message_new_element (GST_OBJECT_CAST (priv->payloader),
3611 gst_structure_new_empty ("GstRTSPStreamBlocking")));
3613 return GST_PAD_PROBE_OK;
3617 * gst_rtsp_stream_set_blocked:
3618 * @stream: a #GstRTSPStream
3619 * @blocked: boolean indicating we should block or unblock
3621 * Blocks or unblocks the dataflow on @stream.
3623 * Returns: %TRUE on success
3626 gst_rtsp_stream_set_blocked (GstRTSPStream * stream, gboolean blocked)
3628 GstRTSPStreamPrivate *priv;
3630 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3632 priv = stream->priv;
3634 g_mutex_lock (&priv->lock);
3636 priv->blocking = FALSE;
3637 if (priv->blocked_id == 0) {
3638 priv->blocked_id = gst_pad_add_probe (priv->srcpad,
3639 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3640 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocking,
3641 g_object_ref (stream), g_object_unref);
3644 if (priv->blocked_id != 0) {
3645 gst_pad_remove_probe (priv->srcpad, priv->blocked_id);
3646 priv->blocked_id = 0;
3647 priv->blocking = FALSE;
3650 g_mutex_unlock (&priv->lock);
3656 * gst_rtsp_stream_is_blocking:
3657 * @stream: a #GstRTSPStream
3659 * Check if @stream is blocking on a #GstBuffer.
3661 * Returns: %TRUE if @stream is blocking
3664 gst_rtsp_stream_is_blocking (GstRTSPStream * stream)
3666 GstRTSPStreamPrivate *priv;
3669 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3671 priv = stream->priv;
3673 g_mutex_lock (&priv->lock);
3674 result = priv->blocking;
3675 g_mutex_unlock (&priv->lock);
3681 * gst_rtsp_stream_query_position:
3682 * @stream: a #GstRTSPStream
3684 * Query the position of the stream in %GST_FORMAT_TIME. This only considers
3685 * the RTP parts of the pipeline and not the RTCP parts.
3687 * Returns: %TRUE if the position could be queried
3690 gst_rtsp_stream_query_position (GstRTSPStream * stream, gint64 * position)
3692 GstRTSPStreamPrivate *priv;
3696 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3698 priv = stream->priv;
3700 g_mutex_lock (&priv->lock);
3701 /* depending on the transport type, it should query corresponding sink */
3702 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3703 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3704 sink = priv->udpsink[0];
3706 sink = priv->appsink[0];
3709 gst_object_ref (sink);
3710 g_mutex_unlock (&priv->lock);
3715 ret = gst_element_query_position (sink, GST_FORMAT_TIME, position);
3716 gst_object_unref (sink);
3722 * gst_rtsp_stream_query_stop:
3723 * @stream: a #GstRTSPStream
3725 * Query the stop of the stream in %GST_FORMAT_TIME. This only considers
3726 * the RTP parts of the pipeline and not the RTCP parts.
3728 * Returns: %TRUE if the stop could be queried
3731 gst_rtsp_stream_query_stop (GstRTSPStream * stream, gint64 * stop)
3733 GstRTSPStreamPrivate *priv;
3738 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
3740 priv = stream->priv;
3742 g_mutex_lock (&priv->lock);
3743 /* depending on the transport type, it should query corresponding sink */
3744 if ((priv->protocols & GST_RTSP_LOWER_TRANS_UDP) ||
3745 (priv->protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST))
3746 sink = priv->udpsink[0];
3748 sink = priv->appsink[0];
3751 gst_object_ref (sink);
3752 g_mutex_unlock (&priv->lock);
3757 query = gst_query_new_segment (GST_FORMAT_TIME);
3758 if ((ret = gst_element_query (sink, query))) {
3761 gst_query_parse_segment (query, NULL, &format, NULL, stop);
3762 if (format != GST_FORMAT_TIME)
3765 gst_query_unref (query);
3766 gst_object_unref (sink);