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 | GST_RTSP_ADDRESS_FLAG_MULTICAST, 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 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
525 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
526 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
528 g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
529 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
530 g_object_unref (socket);
531 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
532 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
533 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
534 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
535 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
536 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
537 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
539 /* we keep these elements, we will further configure them when the
540 * client told us to really use the UDP ports. */
541 priv->udpsrc[0] = udpsrc0;
542 priv->udpsrc[1] = udpsrc1;
543 priv->udpsink[0] = udpsink0;
544 priv->udpsink[1] = udpsink1;
545 priv->server_port.min = rtpport;
546 priv->server_port.max = rtcpport;
559 no_udp_rtcp_protocol:
570 gst_element_set_state (udpsrc0, GST_STATE_NULL);
571 gst_object_unref (udpsrc0);
574 gst_element_set_state (udpsrc1, GST_STATE_NULL);
575 gst_object_unref (udpsrc1);
578 gst_element_set_state (udpsink0, GST_STATE_NULL);
579 gst_object_unref (udpsink0);
582 gst_element_set_state (udpsink1, GST_STATE_NULL);
583 gst_object_unref (udpsink1);
590 * gst_rtsp_stream_get_server_port:
591 * @stream: a #GstRTSPStream
592 * @server_port: (out): result server port
594 * Fill @server_port with the port pair used by the server. This function can
595 * only be called when @stream has been joined.
598 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
599 GstRTSPRange * server_port)
601 GstRTSPStreamPrivate *priv;
603 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
605 g_return_if_fail (priv->is_joined);
607 g_mutex_lock (&priv->lock);
609 *server_port = priv->server_port;
610 g_mutex_unlock (&priv->lock);
614 * gst_rtsp_stream_get_ssrc:
615 * @stream: a #GstRTSPStream
616 * @ssrc: (out): result ssrc
618 * Get the SSRC used by the RTP session of this stream. This function can only
619 * be called when @stream has been joined.
622 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
624 GstRTSPStreamPrivate *priv;
626 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
628 g_return_if_fail (priv->is_joined);
630 g_mutex_lock (&priv->lock);
631 if (ssrc && priv->session)
632 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
633 g_mutex_unlock (&priv->lock);
636 /* executed from streaming thread */
638 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
640 GstRTSPStreamPrivate *priv = stream->priv;
641 GstCaps *newcaps, *oldcaps;
643 newcaps = gst_pad_get_current_caps (pad);
645 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
648 g_mutex_lock (&priv->lock);
649 oldcaps = priv->caps;
650 priv->caps = newcaps;
651 g_mutex_unlock (&priv->lock);
654 gst_caps_unref (oldcaps);
658 dump_structure (const GstStructure * s)
662 sstr = gst_structure_to_string (s);
663 GST_INFO ("structure: %s", sstr);
667 static GstRTSPStreamTransport *
668 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
670 GstRTSPStreamPrivate *priv = stream->priv;
672 GstRTSPStreamTransport *result = NULL;
677 if (rtcp_from == NULL)
680 tmp = g_strrstr (rtcp_from, ":");
684 port = atoi (tmp + 1);
685 dest = g_strndup (rtcp_from, tmp - rtcp_from);
687 g_mutex_lock (&priv->lock);
688 GST_INFO ("finding %s:%d in %d transports", dest, port,
689 g_list_length (priv->transports));
691 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
692 GstRTSPStreamTransport *trans = walk->data;
693 const GstRTSPTransport *tr;
696 tr = gst_rtsp_stream_transport_get_transport (trans);
698 min = tr->client_port.min;
699 max = tr->client_port.max;
701 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
706 g_mutex_unlock (&priv->lock);
713 static GstRTSPStreamTransport *
714 check_transport (GObject * source, GstRTSPStream * stream)
717 GstRTSPStreamTransport *trans;
719 /* see if we have a stream to match with the origin of the RTCP packet */
720 trans = g_object_get_qdata (source, ssrc_stream_map_key);
722 g_object_get (source, "stats", &stats, NULL);
724 const gchar *rtcp_from;
726 dump_structure (stats);
728 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
729 if ((trans = find_transport (stream, rtcp_from))) {
730 GST_INFO ("%p: found transport %p for source %p", stream, trans,
732 g_object_set_qdata (source, ssrc_stream_map_key, trans);
734 gst_structure_free (stats);
742 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
744 GstRTSPStreamTransport *trans;
746 GST_INFO ("%p: new source %p", stream, source);
748 trans = check_transport (source, stream);
751 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
755 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
757 GST_INFO ("%p: new SDES %p", stream, source);
761 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
763 GstRTSPStreamTransport *trans;
765 trans = check_transport (source, stream);
768 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
769 gst_rtsp_stream_transport_keep_alive (trans);
774 g_object_get (source, "stats", &stats, NULL);
776 dump_structure (stats);
777 gst_structure_free (stats);
784 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
786 GST_INFO ("%p: source %p bye", stream, source);
790 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
792 GstRTSPStreamTransport *trans;
794 GST_INFO ("%p: source %p bye timeout", stream, source);
796 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
797 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
802 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
804 GstRTSPStreamTransport *trans;
806 GST_INFO ("%p: source %p 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 handle_new_sample (GstAppSink * sink, gpointer user_data)
816 GstRTSPStreamPrivate *priv;
820 GstRTSPStream *stream;
822 sample = gst_app_sink_pull_sample (sink);
826 stream = (GstRTSPStream *) user_data;
828 buffer = gst_sample_get_buffer (sample);
830 g_mutex_lock (&priv->lock);
831 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
832 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
834 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
835 gst_rtsp_stream_transport_send_rtp (tr, buffer);
837 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
840 g_mutex_unlock (&priv->lock);
842 gst_sample_unref (sample);
847 static GstAppSinkCallbacks sink_cb = {
848 NULL, /* not interested in EOS */
849 NULL, /* not interested in preroll samples */
854 * gst_rtsp_stream_join_bin:
855 * @stream: a #GstRTSPStream
856 * @bin: a #GstBin to join
857 * @rtpbin: a rtpbin element in @bin
858 * @state: the target state of the new elements
860 * Join the #Gstbin @bin that contains the element @rtpbin.
862 * @stream will link to @rtpbin, which must be inside @bin. The elements
863 * added to @bin will be set to the state given in @state.
865 * Returns: %TRUE on success.
868 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
869 GstElement * rtpbin, GstState state)
871 GstRTSPStreamPrivate *priv;
874 GstPad *pad, *teepad, *queuepad, *selpad;
875 GstPadLinkReturn ret;
877 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
878 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
879 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
883 g_mutex_lock (&priv->lock);
887 /* create a session with the same index as the stream */
890 GST_INFO ("stream %p joining bin as session %d", stream, idx);
892 if (!alloc_ports (stream))
895 /* get a pad for sending RTP */
896 name = g_strdup_printf ("send_rtp_sink_%u", idx);
897 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
899 /* link the RTP pad to the session manager, it should not really fail unless
900 * this is not really an RTP pad */
901 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
902 if (ret != GST_PAD_LINK_OK)
905 /* get pads from the RTP session element for sending and receiving
907 name = g_strdup_printf ("send_rtp_src_%u", idx);
908 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
910 name = g_strdup_printf ("send_rtcp_src_%u", idx);
911 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
913 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
914 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
916 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
917 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
920 /* get the session */
921 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
923 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
925 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
927 g_signal_connect (priv->session, "on-ssrc-active",
928 (GCallback) on_ssrc_active, stream);
929 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
931 g_signal_connect (priv->session, "on-bye-timeout",
932 (GCallback) on_bye_timeout, stream);
933 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
936 for (i = 0; i < 2; i++) {
937 /* For the sender we create this bit of pipeline for both
938 * RTP and RTCP. Sync and preroll are enabled on udpsink so
939 * we need to add a queue before appsink to make the pipeline
940 * not block. For the TCP case, we want to pump data to the
941 * client as fast as possible anyway.
943 * .--------. .-----. .---------.
944 * | rtpbin | | tee | | udpsink |
945 * | send->sink src->sink |
946 * '--------' | | '---------'
947 * | | .---------. .---------.
948 * | | | queue | | appsink |
949 * | src->sink src->sink |
950 * '-----' '---------' '---------'
952 /* make tee for RTP/RTCP */
953 priv->tee[i] = gst_element_factory_make ("tee", NULL);
954 gst_bin_add (bin, priv->tee[i]);
956 /* and link to rtpbin send pad */
957 pad = gst_element_get_static_pad (priv->tee[i], "sink");
958 gst_pad_link (priv->send_src[i], pad);
959 gst_object_unref (pad);
962 gst_bin_add (bin, priv->udpsink[i]);
964 /* link tee to udpsink */
965 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
966 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
967 gst_pad_link (teepad, pad);
968 gst_object_unref (pad);
969 gst_object_unref (teepad);
972 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
973 gst_bin_add (bin, priv->appqueue[i]);
974 /* and link to tee */
975 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
976 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
977 gst_pad_link (teepad, pad);
978 gst_object_unref (pad);
979 gst_object_unref (teepad);
982 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
983 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
984 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
985 gst_bin_add (bin, priv->appsink[i]);
986 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
987 &sink_cb, stream, NULL);
988 /* and link to queue */
989 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
990 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
991 gst_pad_link (queuepad, pad);
992 gst_object_unref (pad);
993 gst_object_unref (queuepad);
995 /* For the receiver we create this bit of pipeline for both
996 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
997 * and it is all funneled into the rtpbin receive pad.
999 * .--------. .--------. .--------.
1000 * | udpsrc | | funnel | | rtpbin |
1001 * | src->sink src->sink |
1002 * '--------' | | '--------'
1006 * '--------' '--------'
1008 /* make funnel for the RTP/RTCP receivers */
1009 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1010 gst_bin_add (bin, priv->funnel[i]);
1012 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1013 gst_pad_link (pad, priv->recv_sink[i]);
1014 gst_object_unref (pad);
1016 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1018 gst_element_set_state (priv->udpsrc[i], GST_STATE_PLAYING);
1019 gst_element_set_locked_state (priv->udpsrc[i], TRUE);
1021 gst_bin_add (bin, priv->udpsrc[i]);
1022 /* and link to the funnel */
1023 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1024 pad = gst_element_get_static_pad (priv->udpsrc[i], "src");
1025 gst_pad_link (pad, selpad);
1026 gst_object_unref (pad);
1027 gst_object_unref (selpad);
1029 /* make and add appsrc */
1030 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1031 gst_bin_add (bin, priv->appsrc[i]);
1032 /* and link to the funnel */
1033 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1034 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1035 gst_pad_link (pad, selpad);
1036 gst_object_unref (pad);
1037 gst_object_unref (selpad);
1039 /* check if we need to set to a special state */
1040 if (state != GST_STATE_NULL) {
1041 gst_element_set_state (priv->udpsink[i], state);
1042 gst_element_set_state (priv->appsink[i], state);
1043 gst_element_set_state (priv->appqueue[i], state);
1044 gst_element_set_state (priv->tee[i], state);
1045 gst_element_set_state (priv->funnel[i], state);
1046 gst_element_set_state (priv->appsrc[i], state);
1050 /* be notified of caps changes */
1051 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1052 (GCallback) caps_notify, stream);
1054 priv->is_joined = TRUE;
1055 g_mutex_unlock (&priv->lock);
1062 g_mutex_unlock (&priv->lock);
1067 g_mutex_unlock (&priv->lock);
1068 GST_WARNING ("failed to allocate ports %d", idx);
1073 GST_WARNING ("failed to link stream %d", idx);
1074 gst_object_unref (priv->send_rtp_sink);
1075 priv->send_rtp_sink = NULL;
1076 g_mutex_unlock (&priv->lock);
1082 * gst_rtsp_stream_leave_bin:
1083 * @stream: a #GstRTSPStream
1085 * @rtpbin: a rtpbin #GstElement
1087 * Remove the elements of @stream from @bin. @bin must be set
1088 * to the NULL state before calling this.
1090 * Return: %TRUE on success.
1093 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1094 GstElement * rtpbin)
1096 GstRTSPStreamPrivate *priv;
1099 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1100 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1101 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1103 priv = stream->priv;
1105 g_mutex_lock (&priv->lock);
1106 if (!priv->is_joined)
1107 goto was_not_joined;
1109 /* all transports must be removed by now */
1110 g_return_val_if_fail (priv->transports == NULL, FALSE);
1112 GST_INFO ("stream %p leaving bin", stream);
1114 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1115 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1116 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1117 gst_object_unref (priv->send_rtp_sink);
1118 priv->send_rtp_sink = NULL;
1120 for (i = 0; i < 2; i++) {
1121 /* and set udpsrc to NULL now before removing */
1122 gst_element_set_locked_state (priv->udpsrc[i], FALSE);
1123 gst_element_set_state (priv->udpsrc[i], GST_STATE_NULL);
1125 /* removing them should also nicely release the request
1126 * pads when they finalize */
1127 gst_bin_remove (bin, priv->udpsrc[i]);
1128 gst_bin_remove (bin, priv->udpsink[i]);
1129 gst_bin_remove (bin, priv->appsrc[i]);
1130 gst_bin_remove (bin, priv->appsink[i]);
1131 gst_bin_remove (bin, priv->appqueue[i]);
1132 gst_bin_remove (bin, priv->tee[i]);
1133 gst_bin_remove (bin, priv->funnel[i]);
1135 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1136 gst_object_unref (priv->recv_sink[i]);
1137 priv->recv_sink[i] = NULL;
1139 priv->udpsrc[i] = NULL;
1140 priv->udpsink[i] = NULL;
1141 priv->appsrc[i] = NULL;
1142 priv->appsink[i] = NULL;
1143 priv->appqueue[i] = NULL;
1144 priv->tee[i] = NULL;
1145 priv->funnel[i] = NULL;
1147 gst_object_unref (priv->send_src[0]);
1148 priv->send_src[0] = NULL;
1150 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1151 gst_object_unref (priv->send_src[1]);
1152 priv->send_src[1] = NULL;
1154 g_object_unref (priv->session);
1156 gst_caps_unref (priv->caps);
1158 priv->is_joined = FALSE;
1159 g_mutex_unlock (&priv->lock);
1170 * gst_rtsp_stream_get_rtpinfo:
1171 * @stream: a #GstRTSPStream
1172 * @rtptime: result RTP timestamp
1173 * @seq: result RTP seqnum
1175 * Retrieve the current rtptime and seq. This is used to
1176 * construct a RTPInfo reply header.
1178 * Returns: %TRUE when rtptime and seq could be determined.
1181 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1182 guint * rtptime, guint * seq)
1184 GstRTSPStreamPrivate *priv;
1185 GObjectClass *payobjclass;
1187 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1188 g_return_val_if_fail (rtptime != NULL, FALSE);
1189 g_return_val_if_fail (seq != NULL, FALSE);
1191 priv = stream->priv;
1193 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1195 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1196 !g_object_class_find_property (payobjclass, "timestamp"))
1199 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1205 * gst_rtsp_stream_get_caps:
1206 * @stream: a #GstRTSPStream
1208 * Retrieve the current caps of @stream.
1210 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1214 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1216 GstRTSPStreamPrivate *priv;
1219 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1221 priv = stream->priv;
1223 g_mutex_lock (&priv->lock);
1224 if ((result = priv->caps))
1225 gst_caps_ref (result);
1226 g_mutex_unlock (&priv->lock);
1232 * gst_rtsp_stream_recv_rtp:
1233 * @stream: a #GstRTSPStream
1234 * @buffer: (transfer full): a #GstBuffer
1236 * Handle an RTP buffer for the stream. This method is usually called when a
1237 * message has been received from a client using the TCP transport.
1239 * This function takes ownership of @buffer.
1241 * Returns: a GstFlowReturn.
1244 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1246 GstRTSPStreamPrivate *priv;
1248 GstElement *element;
1250 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1251 priv = stream->priv;
1252 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1253 g_return_val_if_fail (priv->is_joined, FALSE);
1255 g_mutex_lock (&priv->lock);
1256 element = gst_object_ref (priv->appsrc[0]);
1257 g_mutex_unlock (&priv->lock);
1259 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1261 gst_object_unref (element);
1267 * gst_rtsp_stream_recv_rtcp:
1268 * @stream: a #GstRTSPStream
1269 * @buffer: (transfer full): a #GstBuffer
1271 * Handle an RTCP buffer for the stream. This method is usually called when a
1272 * message has been received from a client using the TCP transport.
1274 * This function takes ownership of @buffer.
1276 * Returns: a GstFlowReturn.
1279 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1281 GstRTSPStreamPrivate *priv;
1283 GstElement *element;
1285 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1286 priv = stream->priv;
1287 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1288 g_return_val_if_fail (priv->is_joined, FALSE);
1290 g_mutex_lock (&priv->lock);
1291 element = gst_object_ref (priv->appsrc[1]);
1292 g_mutex_unlock (&priv->lock);
1294 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1296 gst_object_unref (element);
1301 /* must be called with lock */
1303 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1306 GstRTSPStreamPrivate *priv = stream->priv;
1307 const GstRTSPTransport *tr;
1309 tr = gst_rtsp_stream_transport_get_transport (trans);
1311 switch (tr->lower_transport) {
1312 case GST_RTSP_LOWER_TRANS_UDP:
1313 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1319 dest = tr->destination;
1320 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1325 min = tr->client_port.min;
1326 max = tr->client_port.max;
1330 GST_INFO ("adding %s:%d-%d", dest, min, max);
1331 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1332 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1334 GST_INFO ("setting ttl-mc %d", ttl);
1335 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1336 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1338 priv->transports = g_list_prepend (priv->transports, trans);
1340 GST_INFO ("removing %s:%d-%d", dest, min, max);
1341 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1342 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1343 priv->transports = g_list_remove (priv->transports, trans);
1347 case GST_RTSP_LOWER_TRANS_TCP:
1349 GST_INFO ("adding TCP %s", tr->destination);
1350 priv->transports = g_list_prepend (priv->transports, trans);
1352 GST_INFO ("removing TCP %s", tr->destination);
1353 priv->transports = g_list_remove (priv->transports, trans);
1357 goto unknown_transport;
1364 GST_INFO ("Unknown transport %d", tr->lower_transport);
1371 * gst_rtsp_stream_add_transport:
1372 * @stream: a #GstRTSPStream
1373 * @trans: a #GstRTSPStreamTransport
1375 * Add the transport in @trans to @stream. The media of @stream will
1376 * then also be send to the values configured in @trans.
1378 * @stream must be joined to a bin.
1380 * @trans must contain a valid #GstRTSPTransport.
1382 * Returns: %TRUE if @trans was added
1385 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1386 GstRTSPStreamTransport * trans)
1388 GstRTSPStreamPrivate *priv;
1391 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1392 priv = stream->priv;
1393 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1394 g_return_val_if_fail (priv->is_joined, FALSE);
1396 g_mutex_lock (&priv->lock);
1397 res = update_transport (stream, trans, TRUE);
1398 g_mutex_unlock (&priv->lock);
1404 * gst_rtsp_stream_remove_transport:
1405 * @stream: a #GstRTSPStream
1406 * @trans: a #GstRTSPStreamTransport
1408 * Remove the transport in @trans from @stream. The media of @stream will
1409 * not be sent to the values configured in @trans.
1411 * @stream must be joined to a bin.
1413 * @trans must contain a valid #GstRTSPTransport.
1415 * Returns: %TRUE if @trans was removed
1418 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1419 GstRTSPStreamTransport * trans)
1421 GstRTSPStreamPrivate *priv;
1424 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1425 priv = stream->priv;
1426 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1427 g_return_val_if_fail (priv->is_joined, FALSE);
1429 g_mutex_lock (&priv->lock);
1430 res = update_transport (stream, trans, FALSE);
1431 g_mutex_unlock (&priv->lock);