2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
25 #include <gst/app/gstappsrc.h>
26 #include <gst/app/gstappsink.h>
28 #include "rtsp-stream.h"
30 #define GST_RTSP_STREAM_GET_PRIVATE(obj) \
31 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_STREAM, GstRTSPStreamPrivate))
33 struct _GstRTSPStreamPrivate
38 GstElement *payloader;
43 /* pads on the rtpbin */
44 GstPad *send_rtp_sink;
48 /* the RTPSession object */
51 /* sinks used for sending and receiving RTP and RTCP, they share
53 GstElement *udpsrc[2];
54 GstElement *udpsink[2];
55 /* for TCP transport */
56 GstElement *appsrc[2];
57 GstElement *appqueue[2];
58 GstElement *appsink[2];
61 GstElement *funnel[2];
63 /* server ports for sending/receiving */
64 GstRTSPRange server_port;
66 /* multicast addresses */
67 GstRTSPAddressPool *pool;
70 /* the caps of the stream */
74 /* transports we stream to */
86 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
87 #define GST_CAT_DEFAULT rtsp_stream_debug
89 static GQuark ssrc_stream_map_key;
91 static void gst_rtsp_stream_finalize (GObject * obj);
93 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
96 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
98 GObjectClass *gobject_class;
100 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
102 gobject_class = G_OBJECT_CLASS (klass);
104 gobject_class->finalize = gst_rtsp_stream_finalize;
106 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
108 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
112 gst_rtsp_stream_init (GstRTSPStream * stream)
114 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
116 GST_DEBUG ("new stream %p", stream);
120 g_mutex_init (&priv->lock);
124 gst_rtsp_stream_finalize (GObject * obj)
126 GstRTSPStream *stream;
127 GstRTSPStreamPrivate *priv;
129 stream = GST_RTSP_STREAM (obj);
132 GST_DEBUG ("finalize stream %p", stream);
134 /* we really need to be unjoined now */
135 g_return_if_fail (!priv->is_joined);
138 gst_rtsp_address_free (priv->addr);
140 g_object_unref (priv->pool);
141 gst_object_unref (priv->payloader);
142 gst_object_unref (priv->srcpad);
143 g_mutex_clear (&priv->lock);
145 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
149 * gst_rtsp_stream_new:
152 * @payloader: a #GstElement
154 * Create a new media stream with index @idx that handles RTP data on
155 * @srcpad and has a payloader element @payloader.
157 * Returns: a new #GstRTSPStream
160 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
162 GstRTSPStreamPrivate *priv;
163 GstRTSPStream *stream;
165 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
166 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
167 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
169 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
172 priv->payloader = gst_object_ref (payloader);
173 priv->srcpad = gst_object_ref (srcpad);
179 * gst_rtsp_stream_get_index:
180 * @stream: a #GstRTSPStream
182 * Get the stream index.
184 * Return: the stream index.
187 gst_rtsp_stream_get_index (GstRTSPStream * stream)
189 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
191 return stream->priv->idx;
195 * gst_rtsp_stream_set_mtu:
196 * @stream: a #GstRTSPStream
199 * Configure the mtu in the payloader of @stream to @mtu.
202 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
204 GstRTSPStreamPrivate *priv;
206 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
210 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
212 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
216 * gst_rtsp_stream_get_mtu:
217 * @stream: a #GstRTSPStream
219 * Get the configured MTU in the payloader of @stream.
221 * Returns: the MTU of the payloader.
224 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
226 GstRTSPStreamPrivate *priv;
229 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
233 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
239 * gst_rtsp_stream_set_address_pool:
240 * @stream: a #GstRTSPStream
241 * @pool: a #GstRTSPAddressPool
243 * configure @pool to be used as the address pool of @stream.
246 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
247 GstRTSPAddressPool * pool)
249 GstRTSPStreamPrivate *priv;
250 GstRTSPAddressPool *old;
252 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
256 GST_LOG_OBJECT (stream, "set address pool %p", pool);
258 g_mutex_lock (&priv->lock);
259 if ((old = priv->pool) != pool)
260 priv->pool = pool ? g_object_ref (pool) : NULL;
263 g_mutex_unlock (&priv->lock);
266 g_object_unref (old);
270 * gst_rtsp_stream_get_address_pool:
271 * @stream: a #GstRTSPStream
273 * Get the #GstRTSPAddressPool used as the address pool of @stream.
275 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
279 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
281 GstRTSPStreamPrivate *priv;
282 GstRTSPAddressPool *result;
284 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
288 g_mutex_lock (&priv->lock);
289 if ((result = priv->pool))
290 g_object_ref (result);
291 g_mutex_unlock (&priv->lock);
297 * gst_rtsp_stream_get_address:
298 * @stream: a #GstRTSPStream
300 * Get the multicast address of @stream.
302 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
303 * allocated. gst_rtsp_address_free() after usage.
306 gst_rtsp_stream_get_address (GstRTSPStream * stream)
308 GstRTSPStreamPrivate *priv;
309 GstRTSPAddress *result;
311 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
315 g_mutex_lock (&priv->lock);
316 if (priv->addr == NULL) {
317 if (priv->pool == NULL)
320 priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
321 GST_RTSP_ADDRESS_FLAG_EVEN_PORT, 2);
322 if (priv->addr == NULL)
325 result = gst_rtsp_address_copy (priv->addr);
326 g_mutex_unlock (&priv->lock);
333 GST_ERROR_OBJECT (stream, "no address pool specified");
334 g_mutex_unlock (&priv->lock);
339 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
340 g_mutex_unlock (&priv->lock);
346 * gst_rtsp_stream_reserve_address:
347 * @stream: a #GstRTSPStream
349 * Get a specific multicast address of @stream.
351 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
352 * allocated. gst_rtsp_address_free() after usage.
355 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
356 const gchar * address, guint port, guint n_ports, guint ttl)
358 GstRTSPStreamPrivate *priv;
359 GstRTSPAddress *result;
361 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
362 g_return_val_if_fail (address != NULL, NULL);
363 g_return_val_if_fail (port > 0, NULL);
364 g_return_val_if_fail (n_ports > 0, NULL);
365 g_return_val_if_fail (ttl > 0, NULL);
369 g_mutex_lock (&priv->lock);
370 if (priv->addr == NULL) {
371 if (priv->pool == NULL)
374 priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
376 if (priv->addr == NULL)
379 if (strcmp (priv->addr->address, address) ||
380 priv->addr->port != port || priv->addr->n_ports != n_ports ||
381 priv->addr->ttl != ttl)
382 goto different_address;
384 result = gst_rtsp_address_copy (priv->addr);
385 g_mutex_unlock (&priv->lock);
392 GST_ERROR_OBJECT (stream, "no address pool specified");
393 g_mutex_unlock (&priv->lock);
398 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
400 g_mutex_unlock (&priv->lock);
405 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
406 " reserved", address);
407 g_mutex_unlock (&priv->lock);
412 /* must be called with lock */
414 alloc_ports (GstRTSPStream * stream)
416 GstRTSPStreamPrivate *priv = stream->priv;
417 GstStateChangeReturn ret;
418 GstElement *udpsrc0, *udpsrc1;
419 GstElement *udpsink0, *udpsink1;
420 gint tmp_rtp, tmp_rtcp;
422 gint rtpport, rtcpport;
432 /* Start with random port */
436 host = "udp://[::0]";
438 host = "udp://0.0.0.0";
440 /* try to allocate 2 UDP ports, the RTP port should be an even
441 * number and the RTCP port should be the next (uneven) port */
443 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
445 goto no_udp_protocol;
446 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
448 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
449 if (ret == GST_STATE_CHANGE_FAILURE) {
455 gst_element_set_state (udpsrc0, GST_STATE_NULL);
456 gst_object_unref (udpsrc0);
460 goto no_udp_protocol;
463 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
465 /* check if port is even */
466 if ((tmp_rtp & 1) != 0) {
467 /* port not even, close and allocate another */
471 gst_element_set_state (udpsrc0, GST_STATE_NULL);
472 gst_object_unref (udpsrc0);
478 /* allocate port+1 for RTCP now */
479 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
481 goto no_udp_rtcp_protocol;
484 tmp_rtcp = tmp_rtp + 1;
485 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
487 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
488 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
489 if (ret == GST_STATE_CHANGE_FAILURE) {
494 gst_element_set_state (udpsrc0, GST_STATE_NULL);
495 gst_object_unref (udpsrc0);
497 gst_element_set_state (udpsrc1, GST_STATE_NULL);
498 gst_object_unref (udpsrc1);
503 /* all fine, do port check */
504 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
505 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
507 /* this should not happen... */
508 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
511 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
513 goto no_udp_protocol;
515 g_object_get (G_OBJECT (udpsrc0), "used-socket", &socket, NULL);
516 g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
517 g_object_unref (socket);
518 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
520 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
522 goto no_udp_protocol;
524 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
525 "send-duplicates")) {
526 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
527 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
530 ("old multiudpsink version found without send-duplicates property");
533 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
535 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
537 GST_WARNING ("multiudpsink version found without buffer-size property");
540 g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
541 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
542 g_object_unref (socket);
543 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
544 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
545 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
546 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
547 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
548 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
549 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
551 /* we keep these elements, we will further configure them when the
552 * client told us to really use the UDP ports. */
553 priv->udpsrc[0] = udpsrc0;
554 priv->udpsrc[1] = udpsrc1;
555 priv->udpsink[0] = udpsink0;
556 priv->udpsink[1] = udpsink1;
557 priv->server_port.min = rtpport;
558 priv->server_port.max = rtcpport;
571 no_udp_rtcp_protocol:
582 gst_element_set_state (udpsrc0, GST_STATE_NULL);
583 gst_object_unref (udpsrc0);
586 gst_element_set_state (udpsrc1, GST_STATE_NULL);
587 gst_object_unref (udpsrc1);
590 gst_element_set_state (udpsink0, GST_STATE_NULL);
591 gst_object_unref (udpsink0);
594 gst_element_set_state (udpsink1, GST_STATE_NULL);
595 gst_object_unref (udpsink1);
602 * gst_rtsp_stream_get_server_port:
603 * @stream: a #GstRTSPStream
604 * @server_port: (out): result server port
606 * Fill @server_port with the port pair used by the server. This function can
607 * only be called when @stream has been joined.
610 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
611 GstRTSPRange * server_port)
613 GstRTSPStreamPrivate *priv;
615 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
617 g_return_if_fail (priv->is_joined);
619 g_mutex_lock (&priv->lock);
621 *server_port = priv->server_port;
622 g_mutex_unlock (&priv->lock);
626 * gst_rtsp_stream_get_ssrc:
627 * @stream: a #GstRTSPStream
628 * @ssrc: (out): result ssrc
630 * Get the SSRC used by the RTP session of this stream. This function can only
631 * be called when @stream has been joined.
634 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
636 GstRTSPStreamPrivate *priv;
638 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
640 g_return_if_fail (priv->is_joined);
642 g_mutex_lock (&priv->lock);
643 if (ssrc && priv->session)
644 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
645 g_mutex_unlock (&priv->lock);
648 /* executed from streaming thread */
650 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
652 GstRTSPStreamPrivate *priv = stream->priv;
653 GstCaps *newcaps, *oldcaps;
655 newcaps = gst_pad_get_current_caps (pad);
657 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
660 g_mutex_lock (&priv->lock);
661 oldcaps = priv->caps;
662 priv->caps = newcaps;
663 g_mutex_unlock (&priv->lock);
666 gst_caps_unref (oldcaps);
670 dump_structure (const GstStructure * s)
674 sstr = gst_structure_to_string (s);
675 GST_INFO ("structure: %s", sstr);
679 static GstRTSPStreamTransport *
680 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
682 GstRTSPStreamPrivate *priv = stream->priv;
684 GstRTSPStreamTransport *result = NULL;
689 if (rtcp_from == NULL)
692 tmp = g_strrstr (rtcp_from, ":");
696 port = atoi (tmp + 1);
697 dest = g_strndup (rtcp_from, tmp - rtcp_from);
699 g_mutex_lock (&priv->lock);
700 GST_INFO ("finding %s:%d in %d transports", dest, port,
701 g_list_length (priv->transports));
703 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
704 GstRTSPStreamTransport *trans = walk->data;
705 const GstRTSPTransport *tr;
708 tr = gst_rtsp_stream_transport_get_transport (trans);
710 min = tr->client_port.min;
711 max = tr->client_port.max;
713 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
718 g_mutex_unlock (&priv->lock);
725 static GstRTSPStreamTransport *
726 check_transport (GObject * source, GstRTSPStream * stream)
729 GstRTSPStreamTransport *trans;
731 /* see if we have a stream to match with the origin of the RTCP packet */
732 trans = g_object_get_qdata (source, ssrc_stream_map_key);
734 g_object_get (source, "stats", &stats, NULL);
736 const gchar *rtcp_from;
738 dump_structure (stats);
740 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
741 if ((trans = find_transport (stream, rtcp_from))) {
742 GST_INFO ("%p: found transport %p for source %p", stream, trans,
744 g_object_set_qdata (source, ssrc_stream_map_key, trans);
746 gst_structure_free (stats);
754 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
756 GstRTSPStreamTransport *trans;
758 GST_INFO ("%p: new source %p", stream, source);
760 trans = check_transport (source, stream);
763 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
767 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
769 GST_INFO ("%p: new SDES %p", stream, source);
773 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
775 GstRTSPStreamTransport *trans;
777 trans = check_transport (source, stream);
780 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
781 gst_rtsp_stream_transport_keep_alive (trans);
786 g_object_get (source, "stats", &stats, NULL);
788 dump_structure (stats);
789 gst_structure_free (stats);
796 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
798 GST_INFO ("%p: source %p bye", stream, source);
802 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
804 GstRTSPStreamTransport *trans;
806 GST_INFO ("%p: source %p bye timeout", stream, source);
808 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
809 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
814 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
816 GstRTSPStreamTransport *trans;
818 GST_INFO ("%p: source %p timeout", stream, source);
820 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
821 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
826 handle_new_sample (GstAppSink * sink, gpointer user_data)
828 GstRTSPStreamPrivate *priv;
832 GstRTSPStream *stream;
834 sample = gst_app_sink_pull_sample (sink);
838 stream = (GstRTSPStream *) user_data;
840 buffer = gst_sample_get_buffer (sample);
842 g_mutex_lock (&priv->lock);
843 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
844 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
846 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
847 gst_rtsp_stream_transport_send_rtp (tr, buffer);
849 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
852 g_mutex_unlock (&priv->lock);
854 gst_sample_unref (sample);
859 static GstAppSinkCallbacks sink_cb = {
860 NULL, /* not interested in EOS */
861 NULL, /* not interested in preroll samples */
866 * gst_rtsp_stream_join_bin:
867 * @stream: a #GstRTSPStream
868 * @bin: a #GstBin to join
869 * @rtpbin: a rtpbin element in @bin
870 * @state: the target state of the new elements
872 * Join the #Gstbin @bin that contains the element @rtpbin.
874 * @stream will link to @rtpbin, which must be inside @bin. The elements
875 * added to @bin will be set to the state given in @state.
877 * Returns: %TRUE on success.
880 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
881 GstElement * rtpbin, GstState state)
883 GstRTSPStreamPrivate *priv;
886 GstPad *pad, *teepad, *queuepad, *selpad;
887 GstPadLinkReturn ret;
889 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
890 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
891 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
895 g_mutex_lock (&priv->lock);
899 /* create a session with the same index as the stream */
902 GST_INFO ("stream %p joining bin as session %d", stream, idx);
904 if (!alloc_ports (stream))
907 /* get a pad for sending RTP */
908 name = g_strdup_printf ("send_rtp_sink_%u", idx);
909 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
911 /* link the RTP pad to the session manager, it should not really fail unless
912 * this is not really an RTP pad */
913 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
914 if (ret != GST_PAD_LINK_OK)
917 /* get pads from the RTP session element for sending and receiving
919 name = g_strdup_printf ("send_rtp_src_%u", idx);
920 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
922 name = g_strdup_printf ("send_rtcp_src_%u", idx);
923 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
925 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
926 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
928 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
929 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
932 /* get the session */
933 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
935 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
937 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
939 g_signal_connect (priv->session, "on-ssrc-active",
940 (GCallback) on_ssrc_active, stream);
941 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
943 g_signal_connect (priv->session, "on-bye-timeout",
944 (GCallback) on_bye_timeout, stream);
945 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
948 for (i = 0; i < 2; i++) {
949 /* For the sender we create this bit of pipeline for both
950 * RTP and RTCP. Sync and preroll are enabled on udpsink so
951 * we need to add a queue before appsink to make the pipeline
952 * not block. For the TCP case, we want to pump data to the
953 * client as fast as possible anyway.
955 * .--------. .-----. .---------.
956 * | rtpbin | | tee | | udpsink |
957 * | send->sink src->sink |
958 * '--------' | | '---------'
959 * | | .---------. .---------.
960 * | | | queue | | appsink |
961 * | src->sink src->sink |
962 * '-----' '---------' '---------'
964 /* make tee for RTP/RTCP */
965 priv->tee[i] = gst_element_factory_make ("tee", NULL);
966 gst_bin_add (bin, priv->tee[i]);
968 /* and link to rtpbin send pad */
969 pad = gst_element_get_static_pad (priv->tee[i], "sink");
970 gst_pad_link (priv->send_src[i], pad);
971 gst_object_unref (pad);
974 gst_bin_add (bin, priv->udpsink[i]);
976 /* link tee to udpsink */
977 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
978 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
979 gst_pad_link (teepad, pad);
980 gst_object_unref (pad);
981 gst_object_unref (teepad);
984 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
985 gst_bin_add (bin, priv->appqueue[i]);
986 /* and link to tee */
987 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
988 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
989 gst_pad_link (teepad, pad);
990 gst_object_unref (pad);
991 gst_object_unref (teepad);
994 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
995 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
996 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
997 gst_bin_add (bin, priv->appsink[i]);
998 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
999 &sink_cb, stream, NULL);
1000 /* and link to queue */
1001 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1002 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1003 gst_pad_link (queuepad, pad);
1004 gst_object_unref (pad);
1005 gst_object_unref (queuepad);
1007 /* For the receiver we create this bit of pipeline for both
1008 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1009 * and it is all funneled into the rtpbin receive pad.
1011 * .--------. .--------. .--------.
1012 * | udpsrc | | funnel | | rtpbin |
1013 * | src->sink src->sink |
1014 * '--------' | | '--------'
1018 * '--------' '--------'
1020 /* make funnel for the RTP/RTCP receivers */
1021 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1022 gst_bin_add (bin, priv->funnel[i]);
1024 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1025 gst_pad_link (pad, priv->recv_sink[i]);
1026 gst_object_unref (pad);
1028 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1030 gst_element_set_state (priv->udpsrc[i], GST_STATE_PLAYING);
1031 gst_element_set_locked_state (priv->udpsrc[i], TRUE);
1033 gst_bin_add (bin, priv->udpsrc[i]);
1034 /* and link to the funnel */
1035 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1036 pad = gst_element_get_static_pad (priv->udpsrc[i], "src");
1037 gst_pad_link (pad, selpad);
1038 gst_object_unref (pad);
1039 gst_object_unref (selpad);
1041 /* make and add appsrc */
1042 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1043 gst_bin_add (bin, priv->appsrc[i]);
1044 /* and link to the funnel */
1045 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1046 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1047 gst_pad_link (pad, selpad);
1048 gst_object_unref (pad);
1049 gst_object_unref (selpad);
1051 /* check if we need to set to a special state */
1052 if (state != GST_STATE_NULL) {
1053 gst_element_set_state (priv->udpsink[i], state);
1054 gst_element_set_state (priv->appsink[i], state);
1055 gst_element_set_state (priv->appqueue[i], state);
1056 gst_element_set_state (priv->tee[i], state);
1057 gst_element_set_state (priv->funnel[i], state);
1058 gst_element_set_state (priv->appsrc[i], state);
1062 /* be notified of caps changes */
1063 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1064 (GCallback) caps_notify, stream);
1066 priv->is_joined = TRUE;
1067 g_mutex_unlock (&priv->lock);
1074 g_mutex_unlock (&priv->lock);
1079 g_mutex_unlock (&priv->lock);
1080 GST_WARNING ("failed to allocate ports %d", idx);
1085 GST_WARNING ("failed to link stream %d", idx);
1086 gst_object_unref (priv->send_rtp_sink);
1087 priv->send_rtp_sink = NULL;
1088 g_mutex_unlock (&priv->lock);
1094 * gst_rtsp_stream_leave_bin:
1095 * @stream: a #GstRTSPStream
1097 * @rtpbin: a rtpbin #GstElement
1099 * Remove the elements of @stream from @bin. @bin must be set
1100 * to the NULL state before calling this.
1102 * Return: %TRUE on success.
1105 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1106 GstElement * rtpbin)
1108 GstRTSPStreamPrivate *priv;
1111 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1112 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1113 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1115 priv = stream->priv;
1117 g_mutex_lock (&priv->lock);
1118 if (!priv->is_joined)
1119 goto was_not_joined;
1121 /* all transports must be removed by now */
1122 g_return_val_if_fail (priv->transports == NULL, FALSE);
1124 GST_INFO ("stream %p leaving bin", stream);
1126 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1127 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1128 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1129 gst_object_unref (priv->send_rtp_sink);
1130 priv->send_rtp_sink = NULL;
1132 for (i = 0; i < 2; i++) {
1133 /* and set udpsrc to NULL now before removing */
1134 gst_element_set_locked_state (priv->udpsrc[i], FALSE);
1135 gst_element_set_state (priv->udpsrc[i], GST_STATE_NULL);
1137 /* removing them should also nicely release the request
1138 * pads when they finalize */
1139 gst_bin_remove (bin, priv->udpsrc[i]);
1140 gst_bin_remove (bin, priv->udpsink[i]);
1141 gst_bin_remove (bin, priv->appsrc[i]);
1142 gst_bin_remove (bin, priv->appsink[i]);
1143 gst_bin_remove (bin, priv->appqueue[i]);
1144 gst_bin_remove (bin, priv->tee[i]);
1145 gst_bin_remove (bin, priv->funnel[i]);
1147 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1148 gst_object_unref (priv->recv_sink[i]);
1149 priv->recv_sink[i] = NULL;
1151 priv->udpsrc[i] = NULL;
1152 priv->udpsink[i] = NULL;
1153 priv->appsrc[i] = NULL;
1154 priv->appsink[i] = NULL;
1155 priv->appqueue[i] = NULL;
1156 priv->tee[i] = NULL;
1157 priv->funnel[i] = NULL;
1159 gst_object_unref (priv->send_src[0]);
1160 priv->send_src[0] = NULL;
1162 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1163 gst_object_unref (priv->send_src[1]);
1164 priv->send_src[1] = NULL;
1166 g_object_unref (priv->session);
1168 gst_caps_unref (priv->caps);
1170 priv->is_joined = FALSE;
1171 g_mutex_unlock (&priv->lock);
1182 * gst_rtsp_stream_get_rtpinfo:
1183 * @stream: a #GstRTSPStream
1184 * @rtptime: result RTP timestamp
1185 * @seq: result RTP seqnum
1187 * Retrieve the current rtptime and seq. This is used to
1188 * construct a RTPInfo reply header.
1190 * Returns: %TRUE when rtptime and seq could be determined.
1193 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1194 guint * rtptime, guint * seq)
1196 GstRTSPStreamPrivate *priv;
1197 GObjectClass *payobjclass;
1199 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1200 g_return_val_if_fail (rtptime != NULL, FALSE);
1201 g_return_val_if_fail (seq != NULL, FALSE);
1203 priv = stream->priv;
1205 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1207 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1208 !g_object_class_find_property (payobjclass, "timestamp"))
1211 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1217 * gst_rtsp_stream_get_caps:
1218 * @stream: a #GstRTSPStream
1220 * Retrieve the current caps of @stream.
1222 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1226 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1228 GstRTSPStreamPrivate *priv;
1231 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1233 priv = stream->priv;
1235 g_mutex_lock (&priv->lock);
1236 if ((result = priv->caps))
1237 gst_caps_ref (result);
1238 g_mutex_unlock (&priv->lock);
1244 * gst_rtsp_stream_recv_rtp:
1245 * @stream: a #GstRTSPStream
1246 * @buffer: (transfer full): a #GstBuffer
1248 * Handle an RTP buffer for the stream. This method is usually called when a
1249 * message has been received from a client using the TCP transport.
1251 * This function takes ownership of @buffer.
1253 * Returns: a GstFlowReturn.
1256 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1258 GstRTSPStreamPrivate *priv;
1260 GstElement *element;
1262 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1263 priv = stream->priv;
1264 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1265 g_return_val_if_fail (priv->is_joined, FALSE);
1267 g_mutex_lock (&priv->lock);
1268 element = gst_object_ref (priv->appsrc[0]);
1269 g_mutex_unlock (&priv->lock);
1271 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1273 gst_object_unref (element);
1279 * gst_rtsp_stream_recv_rtcp:
1280 * @stream: a #GstRTSPStream
1281 * @buffer: (transfer full): a #GstBuffer
1283 * Handle an RTCP buffer for the stream. This method is usually called when a
1284 * message has been received from a client using the TCP transport.
1286 * This function takes ownership of @buffer.
1288 * Returns: a GstFlowReturn.
1291 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1293 GstRTSPStreamPrivate *priv;
1295 GstElement *element;
1297 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1298 priv = stream->priv;
1299 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1300 g_return_val_if_fail (priv->is_joined, FALSE);
1302 g_mutex_lock (&priv->lock);
1303 element = gst_object_ref (priv->appsrc[1]);
1304 g_mutex_unlock (&priv->lock);
1306 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1308 gst_object_unref (element);
1313 /* must be called with lock */
1315 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1318 GstRTSPStreamPrivate *priv = stream->priv;
1319 const GstRTSPTransport *tr;
1321 tr = gst_rtsp_stream_transport_get_transport (trans);
1323 switch (tr->lower_transport) {
1324 case GST_RTSP_LOWER_TRANS_UDP:
1325 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1331 dest = tr->destination;
1332 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1337 min = tr->client_port.min;
1338 max = tr->client_port.max;
1342 GST_INFO ("adding %s:%d-%d", dest, min, max);
1343 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1344 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1346 GST_INFO ("setting ttl-mc %d", ttl);
1347 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1348 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1350 priv->transports = g_list_prepend (priv->transports, trans);
1352 GST_INFO ("removing %s:%d-%d", dest, min, max);
1353 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1354 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1355 priv->transports = g_list_remove (priv->transports, trans);
1359 case GST_RTSP_LOWER_TRANS_TCP:
1361 GST_INFO ("adding TCP %s", tr->destination);
1362 priv->transports = g_list_prepend (priv->transports, trans);
1364 GST_INFO ("removing TCP %s", tr->destination);
1365 priv->transports = g_list_remove (priv->transports, trans);
1369 goto unknown_transport;
1376 GST_INFO ("Unknown transport %d", tr->lower_transport);
1383 * gst_rtsp_stream_add_transport:
1384 * @stream: a #GstRTSPStream
1385 * @trans: a #GstRTSPStreamTransport
1387 * Add the transport in @trans to @stream. The media of @stream will
1388 * then also be send to the values configured in @trans.
1390 * @stream must be joined to a bin.
1392 * @trans must contain a valid #GstRTSPTransport.
1394 * Returns: %TRUE if @trans was added
1397 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1398 GstRTSPStreamTransport * trans)
1400 GstRTSPStreamPrivate *priv;
1403 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1404 priv = stream->priv;
1405 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1406 g_return_val_if_fail (priv->is_joined, FALSE);
1408 g_mutex_lock (&priv->lock);
1409 res = update_transport (stream, trans, TRUE);
1410 g_mutex_unlock (&priv->lock);
1416 * gst_rtsp_stream_remove_transport:
1417 * @stream: a #GstRTSPStream
1418 * @trans: a #GstRTSPStreamTransport
1420 * Remove the transport in @trans from @stream. The media of @stream will
1421 * not be sent to the values configured in @trans.
1423 * @stream must be joined to a bin.
1425 * @trans must contain a valid #GstRTSPTransport.
1427 * Returns: %TRUE if @trans was removed
1430 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1431 GstRTSPStreamTransport * trans)
1433 GstRTSPStreamPrivate *priv;
1436 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1437 priv = stream->priv;
1438 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1439 g_return_val_if_fail (priv->is_joined, FALSE);
1441 g_mutex_lock (&priv->lock);
1442 res = update_transport (stream, trans, FALSE);
1443 g_mutex_unlock (&priv->lock);