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;
65 GstRTSPAddress *server_addr;
67 /* multicast addresses */
68 GstRTSPAddressPool *pool;
71 /* the caps of the stream */
75 /* transports we stream to */
87 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
88 #define GST_CAT_DEFAULT rtsp_stream_debug
90 static GQuark ssrc_stream_map_key;
92 static void gst_rtsp_stream_finalize (GObject * obj);
94 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
97 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
99 GObjectClass *gobject_class;
101 g_type_class_add_private (klass, sizeof (GstRTSPStreamPrivate));
103 gobject_class = G_OBJECT_CLASS (klass);
105 gobject_class->finalize = gst_rtsp_stream_finalize;
107 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
109 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
113 gst_rtsp_stream_init (GstRTSPStream * stream)
115 GstRTSPStreamPrivate *priv = GST_RTSP_STREAM_GET_PRIVATE (stream);
117 GST_DEBUG ("new stream %p", stream);
121 g_mutex_init (&priv->lock);
125 gst_rtsp_stream_finalize (GObject * obj)
127 GstRTSPStream *stream;
128 GstRTSPStreamPrivate *priv;
130 stream = GST_RTSP_STREAM (obj);
133 GST_DEBUG ("finalize stream %p", stream);
135 /* we really need to be unjoined now */
136 g_return_if_fail (!priv->is_joined);
139 gst_rtsp_address_free (priv->addr);
140 if (priv->server_addr)
141 gst_rtsp_address_free (priv->server_addr);
143 g_object_unref (priv->pool);
144 gst_object_unref (priv->payloader);
145 gst_object_unref (priv->srcpad);
146 g_mutex_clear (&priv->lock);
148 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
152 * gst_rtsp_stream_new:
155 * @payloader: a #GstElement
157 * Create a new media stream with index @idx that handles RTP data on
158 * @srcpad and has a payloader element @payloader.
160 * Returns: a new #GstRTSPStream
163 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
165 GstRTSPStreamPrivate *priv;
166 GstRTSPStream *stream;
168 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
169 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
170 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
172 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
175 priv->payloader = gst_object_ref (payloader);
176 priv->srcpad = gst_object_ref (srcpad);
182 * gst_rtsp_stream_get_index:
183 * @stream: a #GstRTSPStream
185 * Get the stream index.
187 * Return: the stream index.
190 gst_rtsp_stream_get_index (GstRTSPStream * stream)
192 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), -1);
194 return stream->priv->idx;
198 * gst_rtsp_stream_get_srcpad:
199 * @stream: a #GstRTSPStream
201 * Get the srcpad associated with @stream.
203 * Return: the srcpad. Unref after usage.
206 gst_rtsp_stream_get_srcpad (GstRTSPStream * stream)
208 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
210 return gst_object_ref (stream->priv->srcpad);
214 * gst_rtsp_stream_set_mtu:
215 * @stream: a #GstRTSPStream
218 * Configure the mtu in the payloader of @stream to @mtu.
221 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
223 GstRTSPStreamPrivate *priv;
225 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
229 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
231 g_object_set (G_OBJECT (priv->payloader), "mtu", mtu, NULL);
235 * gst_rtsp_stream_get_mtu:
236 * @stream: a #GstRTSPStream
238 * Get the configured MTU in the payloader of @stream.
240 * Returns: the MTU of the payloader.
243 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
245 GstRTSPStreamPrivate *priv;
248 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
252 g_object_get (G_OBJECT (priv->payloader), "mtu", &mtu, NULL);
258 * gst_rtsp_stream_set_address_pool:
259 * @stream: a #GstRTSPStream
260 * @pool: a #GstRTSPAddressPool
262 * configure @pool to be used as the address pool of @stream.
265 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
266 GstRTSPAddressPool * pool)
268 GstRTSPStreamPrivate *priv;
269 GstRTSPAddressPool *old;
271 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
275 GST_LOG_OBJECT (stream, "set address pool %p", pool);
277 g_mutex_lock (&priv->lock);
278 if ((old = priv->pool) != pool)
279 priv->pool = pool ? g_object_ref (pool) : NULL;
282 g_mutex_unlock (&priv->lock);
285 g_object_unref (old);
289 * gst_rtsp_stream_get_address_pool:
290 * @stream: a #GstRTSPStream
292 * Get the #GstRTSPAddressPool used as the address pool of @stream.
294 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
298 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
300 GstRTSPStreamPrivate *priv;
301 GstRTSPAddressPool *result;
303 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
307 g_mutex_lock (&priv->lock);
308 if ((result = priv->pool))
309 g_object_ref (result);
310 g_mutex_unlock (&priv->lock);
316 * gst_rtsp_stream_get_address:
317 * @stream: a #GstRTSPStream
319 * Get the multicast address of @stream.
321 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
322 * allocated. gst_rtsp_address_free() after usage.
325 gst_rtsp_stream_get_address (GstRTSPStream * stream)
327 GstRTSPStreamPrivate *priv;
328 GstRTSPAddress *result;
330 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
334 g_mutex_lock (&priv->lock);
335 if (priv->addr == NULL) {
336 if (priv->pool == NULL)
339 priv->addr = gst_rtsp_address_pool_acquire_address (priv->pool,
340 GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_MULTICAST, 2);
341 if (priv->addr == NULL)
344 result = gst_rtsp_address_copy (priv->addr);
345 g_mutex_unlock (&priv->lock);
352 GST_ERROR_OBJECT (stream, "no address pool specified");
353 g_mutex_unlock (&priv->lock);
358 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
359 g_mutex_unlock (&priv->lock);
365 * gst_rtsp_stream_reserve_address:
366 * @stream: a #GstRTSPStream
368 * Get a specific multicast address of @stream.
370 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
371 * allocated. gst_rtsp_address_free() after usage.
374 gst_rtsp_stream_reserve_address (GstRTSPStream * stream,
375 const gchar * address, guint port, guint n_ports, guint ttl)
377 GstRTSPStreamPrivate *priv;
378 GstRTSPAddress *result;
380 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
381 g_return_val_if_fail (address != NULL, NULL);
382 g_return_val_if_fail (port > 0, NULL);
383 g_return_val_if_fail (n_ports > 0, NULL);
384 g_return_val_if_fail (ttl > 0, NULL);
388 g_mutex_lock (&priv->lock);
389 if (priv->addr == NULL) {
390 if (priv->pool == NULL)
393 priv->addr = gst_rtsp_address_pool_reserve_address (priv->pool, address,
395 if (priv->addr == NULL)
398 if (strcmp (priv->addr->address, address) ||
399 priv->addr->port != port || priv->addr->n_ports != n_ports ||
400 priv->addr->ttl != ttl)
401 goto different_address;
403 result = gst_rtsp_address_copy (priv->addr);
404 g_mutex_unlock (&priv->lock);
411 GST_ERROR_OBJECT (stream, "no address pool specified");
412 g_mutex_unlock (&priv->lock);
417 GST_ERROR_OBJECT (stream, "failed to acquire address %s from pool",
419 g_mutex_unlock (&priv->lock);
424 GST_ERROR_OBJECT (stream, "address %s is not the same that was already"
425 " reserved", address);
426 g_mutex_unlock (&priv->lock);
431 /* must be called with lock */
433 alloc_ports (GstRTSPStream * stream)
435 GstRTSPStreamPrivate *priv = stream->priv;
436 GstStateChangeReturn ret;
437 GstElement *udpsrc0, *udpsrc1;
438 GstElement *udpsink0, *udpsink1;
439 GSocket *rtp_socket = NULL;
440 GSocket *rtcp_socket;
441 gint tmp_rtp, tmp_rtcp;
443 gint rtpport, rtcpport;
444 GList *rejected_addresses = NULL;
445 GstRTSPAddress *addr = NULL;
446 GSocketFamily family;
447 GInetAddress *inetaddr = NULL;
448 GSocketAddress *rtp_sockaddr = NULL;
449 GSocketAddress *rtcp_sockaddr = NULL;
457 /* Start with random port */
461 family = G_SOCKET_FAMILY_IPV6;
463 family = G_SOCKET_FAMILY_IPV4;
466 rtcp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
467 G_SOCKET_PROTOCOL_UDP, NULL);
469 goto no_udp_protocol;
471 if (priv->server_addr)
472 gst_rtsp_address_free (priv->server_addr);
474 /* try to allocate 2 UDP ports, the RTP port should be an even
475 * number and the RTCP port should be the next (uneven) port */
478 if (rtp_socket == NULL) {
479 rtp_socket = g_socket_new (family, G_SOCKET_TYPE_DATAGRAM,
480 G_SOCKET_PROTOCOL_UDP, NULL);
482 goto no_udp_protocol;
485 if (priv->pool && gst_rtsp_address_pool_has_unicast_addresses (priv->pool)) {
486 GstRTSPAddressFlags flags;
489 rejected_addresses = g_list_prepend (rejected_addresses, addr);
491 flags = GST_RTSP_ADDRESS_FLAG_EVEN_PORT | GST_RTSP_ADDRESS_FLAG_UNICAST;
493 flags |= GST_RTSP_ADDRESS_FLAG_IPV6;
495 flags |= GST_RTSP_ADDRESS_FLAG_IPV4;
497 addr = gst_rtsp_address_pool_acquire_address (priv->pool, flags, 2);
502 tmp_rtp = addr->port;
504 g_clear_object (&inetaddr);
505 inetaddr = g_inet_address_new_from_string (addr->address);
513 if (inetaddr == NULL)
514 inetaddr = g_inet_address_new_any (family);
517 rtp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtp);
518 if (!g_socket_bind (rtp_socket, rtp_sockaddr, FALSE, NULL)) {
519 g_object_unref (rtp_sockaddr);
522 g_object_unref (rtp_sockaddr);
524 rtp_sockaddr = g_socket_get_local_address (rtp_socket, NULL);
525 if (rtp_sockaddr == NULL || !G_IS_INET_SOCKET_ADDRESS (rtp_sockaddr)) {
526 g_clear_object (&rtp_sockaddr);
531 g_inet_socket_address_get_port (G_INET_SOCKET_ADDRESS (rtp_sockaddr));
532 g_object_unref (rtp_sockaddr);
534 /* check if port is even */
535 if ((tmp_rtp & 1) != 0) {
536 /* port not even, close and allocate another */
538 g_clear_object (&rtp_socket);
543 tmp_rtcp = tmp_rtp + 1;
545 rtcp_sockaddr = g_inet_socket_address_new (inetaddr, tmp_rtcp);
546 if (!g_socket_bind (rtcp_socket, rtcp_sockaddr, FALSE, NULL)) {
547 g_object_unref (rtcp_sockaddr);
548 g_clear_object (&rtp_socket);
551 g_object_unref (rtcp_sockaddr);
553 g_clear_object (&inetaddr);
555 udpsrc0 = gst_element_factory_make ("udpsrc", NULL);
556 udpsrc1 = gst_element_factory_make ("udpsrc", NULL);
558 if (udpsrc0 == NULL || udpsrc1 == NULL)
559 goto no_udp_protocol;
561 g_object_set (G_OBJECT (udpsrc0), "socket", rtp_socket, NULL);
562 g_object_set (G_OBJECT (udpsrc1), "socket", rtcp_socket, NULL);
564 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
565 if (ret == GST_STATE_CHANGE_FAILURE)
567 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
568 if (ret == GST_STATE_CHANGE_FAILURE)
571 /* all fine, do port check */
572 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
573 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
575 /* this should not happen... */
576 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
579 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
581 goto no_udp_protocol;
583 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
584 g_object_set (G_OBJECT (udpsink0), "socket", rtp_socket, NULL);
586 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
588 goto no_udp_protocol;
590 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
591 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
592 g_object_set (G_OBJECT (udpsink0), "buffer-size", priv->buffer_size, NULL);
594 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
595 g_object_set (G_OBJECT (udpsink1), "socket", rtcp_socket, NULL);
596 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
597 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
598 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
599 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
600 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
601 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
603 /* we keep these elements, we will further configure them when the
604 * client told us to really use the UDP ports. */
605 priv->udpsrc[0] = udpsrc0;
606 priv->udpsrc[1] = udpsrc1;
607 priv->udpsink[0] = udpsink0;
608 priv->udpsink[1] = udpsink1;
609 priv->server_port.min = rtpport;
610 priv->server_port.max = rtcpport;
612 priv->server_addr = addr;
613 g_list_free_full (rejected_addresses, (GDestroyNotify) gst_rtsp_address_free);
615 g_object_unref (rtp_socket);
616 g_object_unref (rtcp_socket);
644 gst_element_set_state (udpsrc0, GST_STATE_NULL);
645 gst_object_unref (udpsrc0);
648 gst_element_set_state (udpsrc1, GST_STATE_NULL);
649 gst_object_unref (udpsrc1);
652 gst_element_set_state (udpsink0, GST_STATE_NULL);
653 gst_object_unref (udpsink0);
656 gst_element_set_state (udpsink1, GST_STATE_NULL);
657 gst_object_unref (udpsink1);
660 g_object_unref (inetaddr);
661 g_list_free_full (rejected_addresses,
662 (GDestroyNotify) gst_rtsp_address_free);
664 gst_rtsp_address_free (addr);
666 g_object_unref (rtp_socket);
668 g_object_unref (rtcp_socket);
674 * gst_rtsp_stream_get_server_port:
675 * @stream: a #GstRTSPStream
676 * @server_port: (out): result server port
678 * Fill @server_port with the port pair used by the server. This function can
679 * only be called when @stream has been joined.
682 gst_rtsp_stream_get_server_port (GstRTSPStream * stream,
683 GstRTSPRange * server_port)
685 GstRTSPStreamPrivate *priv;
687 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
689 g_return_if_fail (priv->is_joined);
691 g_mutex_lock (&priv->lock);
693 *server_port = priv->server_port;
694 g_mutex_unlock (&priv->lock);
698 * gst_rtsp_stream_get_ssrc:
699 * @stream: a #GstRTSPStream
700 * @ssrc: (out): result ssrc
702 * Get the SSRC used by the RTP session of this stream. This function can only
703 * be called when @stream has been joined.
706 gst_rtsp_stream_get_ssrc (GstRTSPStream * stream, guint * ssrc)
708 GstRTSPStreamPrivate *priv;
710 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
712 g_return_if_fail (priv->is_joined);
714 g_mutex_lock (&priv->lock);
715 if (ssrc && priv->session)
716 g_object_get (priv->session, "internal-ssrc", ssrc, NULL);
717 g_mutex_unlock (&priv->lock);
720 /* executed from streaming thread */
722 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
724 GstRTSPStreamPrivate *priv = stream->priv;
725 GstCaps *newcaps, *oldcaps;
727 newcaps = gst_pad_get_current_caps (pad);
729 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
732 g_mutex_lock (&priv->lock);
733 oldcaps = priv->caps;
734 priv->caps = newcaps;
735 g_mutex_unlock (&priv->lock);
738 gst_caps_unref (oldcaps);
742 dump_structure (const GstStructure * s)
746 sstr = gst_structure_to_string (s);
747 GST_INFO ("structure: %s", sstr);
751 static GstRTSPStreamTransport *
752 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
754 GstRTSPStreamPrivate *priv = stream->priv;
756 GstRTSPStreamTransport *result = NULL;
761 if (rtcp_from == NULL)
764 tmp = g_strrstr (rtcp_from, ":");
768 port = atoi (tmp + 1);
769 dest = g_strndup (rtcp_from, tmp - rtcp_from);
771 g_mutex_lock (&priv->lock);
772 GST_INFO ("finding %s:%d in %d transports", dest, port,
773 g_list_length (priv->transports));
775 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
776 GstRTSPStreamTransport *trans = walk->data;
777 const GstRTSPTransport *tr;
780 tr = gst_rtsp_stream_transport_get_transport (trans);
782 min = tr->client_port.min;
783 max = tr->client_port.max;
785 if ((strcmp (tr->destination, dest) == 0) && (min == port || max == port)) {
791 g_object_ref (result);
792 g_mutex_unlock (&priv->lock);
799 static GstRTSPStreamTransport *
800 check_transport (GObject * source, GstRTSPStream * stream)
803 GstRTSPStreamTransport *trans;
805 /* see if we have a stream to match with the origin of the RTCP packet */
806 trans = g_object_get_qdata (source, ssrc_stream_map_key);
808 g_object_get (source, "stats", &stats, NULL);
810 const gchar *rtcp_from;
812 dump_structure (stats);
814 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
815 if ((trans = find_transport (stream, rtcp_from))) {
816 GST_INFO ("%p: found transport %p for source %p", stream, trans,
818 g_object_set_qdata_full (source, ssrc_stream_map_key, trans,
821 gst_structure_free (stats);
829 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
831 GstRTSPStreamTransport *trans;
833 GST_INFO ("%p: new source %p", stream, source);
835 trans = check_transport (source, stream);
838 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
842 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
844 GST_INFO ("%p: new SDES %p", stream, source);
848 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
850 GstRTSPStreamTransport *trans;
852 trans = check_transport (source, stream);
855 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
856 gst_rtsp_stream_transport_keep_alive (trans);
861 g_object_get (source, "stats", &stats, NULL);
863 dump_structure (stats);
864 gst_structure_free (stats);
871 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
873 GST_INFO ("%p: source %p bye", stream, source);
877 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
879 GstRTSPStreamTransport *trans;
881 GST_INFO ("%p: source %p bye timeout", stream, source);
883 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
884 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
885 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
890 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
892 GstRTSPStreamTransport *trans;
894 GST_INFO ("%p: source %p timeout", stream, source);
896 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
897 gst_rtsp_stream_transport_set_timed_out (trans, TRUE);
898 g_object_set_qdata (source, ssrc_stream_map_key, NULL);
903 handle_new_sample (GstAppSink * sink, gpointer user_data)
905 GstRTSPStreamPrivate *priv;
909 GstRTSPStream *stream;
911 sample = gst_app_sink_pull_sample (sink);
915 stream = (GstRTSPStream *) user_data;
917 buffer = gst_sample_get_buffer (sample);
919 g_mutex_lock (&priv->lock);
920 for (walk = priv->transports; walk; walk = g_list_next (walk)) {
921 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
923 if (GST_ELEMENT_CAST (sink) == priv->appsink[0]) {
924 gst_rtsp_stream_transport_send_rtp (tr, buffer);
926 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
929 g_mutex_unlock (&priv->lock);
931 gst_sample_unref (sample);
936 static GstAppSinkCallbacks sink_cb = {
937 NULL, /* not interested in EOS */
938 NULL, /* not interested in preroll samples */
943 * gst_rtsp_stream_join_bin:
944 * @stream: a #GstRTSPStream
945 * @bin: a #GstBin to join
946 * @rtpbin: a rtpbin element in @bin
947 * @state: the target state of the new elements
949 * Join the #Gstbin @bin that contains the element @rtpbin.
951 * @stream will link to @rtpbin, which must be inside @bin. The elements
952 * added to @bin will be set to the state given in @state.
954 * Returns: %TRUE on success.
957 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
958 GstElement * rtpbin, GstState state)
960 GstRTSPStreamPrivate *priv;
963 GstPad *pad, *teepad, *queuepad, *selpad;
964 GstPadLinkReturn ret;
966 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
967 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
968 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
972 g_mutex_lock (&priv->lock);
976 /* create a session with the same index as the stream */
979 GST_INFO ("stream %p joining bin as session %d", stream, idx);
981 if (!alloc_ports (stream))
984 /* get a pad for sending RTP */
985 name = g_strdup_printf ("send_rtp_sink_%u", idx);
986 priv->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
988 /* link the RTP pad to the session manager, it should not really fail unless
989 * this is not really an RTP pad */
990 ret = gst_pad_link (priv->srcpad, priv->send_rtp_sink);
991 if (ret != GST_PAD_LINK_OK)
994 /* get pads from the RTP session element for sending and receiving
996 name = g_strdup_printf ("send_rtp_src_%u", idx);
997 priv->send_src[0] = gst_element_get_static_pad (rtpbin, name);
999 name = g_strdup_printf ("send_rtcp_src_%u", idx);
1000 priv->send_src[1] = gst_element_get_request_pad (rtpbin, name);
1002 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
1003 priv->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
1005 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
1006 priv->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
1009 /* get the session */
1010 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &priv->session);
1012 g_signal_connect (priv->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1014 g_signal_connect (priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1016 g_signal_connect (priv->session, "on-ssrc-active",
1017 (GCallback) on_ssrc_active, stream);
1018 g_signal_connect (priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1020 g_signal_connect (priv->session, "on-bye-timeout",
1021 (GCallback) on_bye_timeout, stream);
1022 g_signal_connect (priv->session, "on-timeout", (GCallback) on_timeout,
1025 for (i = 0; i < 2; i++) {
1026 /* For the sender we create this bit of pipeline for both
1027 * RTP and RTCP. Sync and preroll are enabled on udpsink so
1028 * we need to add a queue before appsink to make the pipeline
1029 * not block. For the TCP case, we want to pump data to the
1030 * client as fast as possible anyway.
1032 * .--------. .-----. .---------.
1033 * | rtpbin | | tee | | udpsink |
1034 * | send->sink src->sink |
1035 * '--------' | | '---------'
1036 * | | .---------. .---------.
1037 * | | | queue | | appsink |
1038 * | src->sink src->sink |
1039 * '-----' '---------' '---------'
1041 /* make tee for RTP/RTCP */
1042 priv->tee[i] = gst_element_factory_make ("tee", NULL);
1043 gst_bin_add (bin, priv->tee[i]);
1045 /* and link to rtpbin send pad */
1046 pad = gst_element_get_static_pad (priv->tee[i], "sink");
1047 gst_pad_link (priv->send_src[i], pad);
1048 gst_object_unref (pad);
1051 gst_bin_add (bin, priv->udpsink[i]);
1053 /* link tee to udpsink */
1054 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1055 pad = gst_element_get_static_pad (priv->udpsink[i], "sink");
1056 gst_pad_link (teepad, pad);
1057 gst_object_unref (pad);
1058 gst_object_unref (teepad);
1061 priv->appqueue[i] = gst_element_factory_make ("queue", NULL);
1062 gst_bin_add (bin, priv->appqueue[i]);
1063 /* and link to tee */
1064 teepad = gst_element_get_request_pad (priv->tee[i], "src_%u");
1065 pad = gst_element_get_static_pad (priv->appqueue[i], "sink");
1066 gst_pad_link (teepad, pad);
1067 gst_object_unref (pad);
1068 gst_object_unref (teepad);
1071 priv->appsink[i] = gst_element_factory_make ("appsink", NULL);
1072 g_object_set (priv->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1073 g_object_set (priv->appsink[i], "emit-signals", FALSE, NULL);
1074 gst_bin_add (bin, priv->appsink[i]);
1075 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (priv->appsink[i]),
1076 &sink_cb, stream, NULL);
1077 /* and link to queue */
1078 queuepad = gst_element_get_static_pad (priv->appqueue[i], "src");
1079 pad = gst_element_get_static_pad (priv->appsink[i], "sink");
1080 gst_pad_link (queuepad, pad);
1081 gst_object_unref (pad);
1082 gst_object_unref (queuepad);
1084 /* For the receiver we create this bit of pipeline for both
1085 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
1086 * and it is all funneled into the rtpbin receive pad.
1088 * .--------. .--------. .--------.
1089 * | udpsrc | | funnel | | rtpbin |
1090 * | src->sink src->sink |
1091 * '--------' | | '--------'
1095 * '--------' '--------'
1097 /* make funnel for the RTP/RTCP receivers */
1098 priv->funnel[i] = gst_element_factory_make ("funnel", NULL);
1099 gst_bin_add (bin, priv->funnel[i]);
1101 pad = gst_element_get_static_pad (priv->funnel[i], "src");
1102 gst_pad_link (pad, priv->recv_sink[i]);
1103 gst_object_unref (pad);
1105 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1107 gst_element_set_state (priv->udpsrc[i], GST_STATE_PLAYING);
1108 gst_element_set_locked_state (priv->udpsrc[i], TRUE);
1110 gst_bin_add (bin, priv->udpsrc[i]);
1111 /* and link to the funnel */
1112 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1113 pad = gst_element_get_static_pad (priv->udpsrc[i], "src");
1114 gst_pad_link (pad, selpad);
1115 gst_object_unref (pad);
1116 gst_object_unref (selpad);
1118 /* make and add appsrc */
1119 priv->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1120 gst_bin_add (bin, priv->appsrc[i]);
1121 /* and link to the funnel */
1122 selpad = gst_element_get_request_pad (priv->funnel[i], "sink_%u");
1123 pad = gst_element_get_static_pad (priv->appsrc[i], "src");
1124 gst_pad_link (pad, selpad);
1125 gst_object_unref (pad);
1126 gst_object_unref (selpad);
1128 /* check if we need to set to a special state */
1129 if (state != GST_STATE_NULL) {
1130 gst_element_set_state (priv->udpsink[i], state);
1131 gst_element_set_state (priv->appsink[i], state);
1132 gst_element_set_state (priv->appqueue[i], state);
1133 gst_element_set_state (priv->tee[i], state);
1134 gst_element_set_state (priv->funnel[i], state);
1135 gst_element_set_state (priv->appsrc[i], state);
1139 /* be notified of caps changes */
1140 priv->caps_sig = g_signal_connect (priv->send_rtp_sink, "notify::caps",
1141 (GCallback) caps_notify, stream);
1143 priv->is_joined = TRUE;
1144 g_mutex_unlock (&priv->lock);
1151 g_mutex_unlock (&priv->lock);
1156 g_mutex_unlock (&priv->lock);
1157 GST_WARNING ("failed to allocate ports %d", idx);
1162 GST_WARNING ("failed to link stream %d", idx);
1163 gst_object_unref (priv->send_rtp_sink);
1164 priv->send_rtp_sink = NULL;
1165 g_mutex_unlock (&priv->lock);
1171 * gst_rtsp_stream_leave_bin:
1172 * @stream: a #GstRTSPStream
1174 * @rtpbin: a rtpbin #GstElement
1176 * Remove the elements of @stream from @bin.
1178 * Return: %TRUE on success.
1181 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
1182 GstElement * rtpbin)
1184 GstRTSPStreamPrivate *priv;
1187 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1188 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
1189 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
1191 priv = stream->priv;
1193 g_mutex_lock (&priv->lock);
1194 if (!priv->is_joined)
1195 goto was_not_joined;
1197 /* all transports must be removed by now */
1198 g_return_val_if_fail (priv->transports == NULL, FALSE);
1200 GST_INFO ("stream %p leaving bin", stream);
1202 gst_pad_unlink (priv->srcpad, priv->send_rtp_sink);
1203 g_signal_handler_disconnect (priv->send_rtp_sink, priv->caps_sig);
1204 gst_element_release_request_pad (rtpbin, priv->send_rtp_sink);
1205 gst_object_unref (priv->send_rtp_sink);
1206 priv->send_rtp_sink = NULL;
1208 for (i = 0; i < 2; i++) {
1209 gst_element_set_state (priv->udpsink[i], GST_STATE_NULL);
1210 gst_element_set_state (priv->appsink[i], GST_STATE_NULL);
1211 gst_element_set_state (priv->appqueue[i], GST_STATE_NULL);
1212 gst_element_set_state (priv->tee[i], GST_STATE_NULL);
1213 gst_element_set_state (priv->funnel[i], GST_STATE_NULL);
1214 gst_element_set_state (priv->appsrc[i], GST_STATE_NULL);
1215 /* and set udpsrc to NULL now before removing */
1216 gst_element_set_locked_state (priv->udpsrc[i], FALSE);
1217 gst_element_set_state (priv->udpsrc[i], GST_STATE_NULL);
1219 /* removing them should also nicely release the request
1220 * pads when they finalize */
1221 gst_bin_remove (bin, priv->udpsrc[i]);
1222 gst_bin_remove (bin, priv->udpsink[i]);
1223 gst_bin_remove (bin, priv->appsrc[i]);
1224 gst_bin_remove (bin, priv->appsink[i]);
1225 gst_bin_remove (bin, priv->appqueue[i]);
1226 gst_bin_remove (bin, priv->tee[i]);
1227 gst_bin_remove (bin, priv->funnel[i]);
1229 gst_element_release_request_pad (rtpbin, priv->recv_sink[i]);
1230 gst_object_unref (priv->recv_sink[i]);
1231 priv->recv_sink[i] = NULL;
1233 priv->udpsrc[i] = NULL;
1234 priv->udpsink[i] = NULL;
1235 priv->appsrc[i] = NULL;
1236 priv->appsink[i] = NULL;
1237 priv->appqueue[i] = NULL;
1238 priv->tee[i] = NULL;
1239 priv->funnel[i] = NULL;
1241 gst_object_unref (priv->send_src[0]);
1242 priv->send_src[0] = NULL;
1244 gst_element_release_request_pad (rtpbin, priv->send_src[1]);
1245 gst_object_unref (priv->send_src[1]);
1246 priv->send_src[1] = NULL;
1248 g_object_unref (priv->session);
1249 priv->session = NULL;
1251 gst_caps_unref (priv->caps);
1254 priv->is_joined = FALSE;
1255 g_mutex_unlock (&priv->lock);
1266 * gst_rtsp_stream_get_rtpinfo:
1267 * @stream: a #GstRTSPStream
1268 * @rtptime: result RTP timestamp
1269 * @seq: result RTP seqnum
1271 * Retrieve the current rtptime and seq. This is used to
1272 * construct a RTPInfo reply header.
1274 * Returns: %TRUE when rtptime and seq could be determined.
1277 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
1278 guint * rtptime, guint * seq)
1280 GstRTSPStreamPrivate *priv;
1281 GObjectClass *payobjclass;
1283 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1284 g_return_val_if_fail (rtptime != NULL, FALSE);
1285 g_return_val_if_fail (seq != NULL, FALSE);
1287 priv = stream->priv;
1289 payobjclass = G_OBJECT_GET_CLASS (priv->payloader);
1291 if (!g_object_class_find_property (payobjclass, "seqnum") ||
1292 !g_object_class_find_property (payobjclass, "timestamp"))
1295 g_object_get (priv->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
1301 * gst_rtsp_stream_get_caps:
1302 * @stream: a #GstRTSPStream
1304 * Retrieve the current caps of @stream.
1306 * Returns: (transfer full): the #GstCaps of @stream. use gst_caps_unref()
1310 gst_rtsp_stream_get_caps (GstRTSPStream * stream)
1312 GstRTSPStreamPrivate *priv;
1315 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
1317 priv = stream->priv;
1319 g_mutex_lock (&priv->lock);
1320 if ((result = priv->caps))
1321 gst_caps_ref (result);
1322 g_mutex_unlock (&priv->lock);
1328 * gst_rtsp_stream_recv_rtp:
1329 * @stream: a #GstRTSPStream
1330 * @buffer: (transfer full): a #GstBuffer
1332 * Handle an RTP buffer for the stream. This method is usually called when a
1333 * message has been received from a client using the TCP transport.
1335 * This function takes ownership of @buffer.
1337 * Returns: a GstFlowReturn.
1340 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1342 GstRTSPStreamPrivate *priv;
1344 GstElement *element;
1346 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1347 priv = stream->priv;
1348 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1349 g_return_val_if_fail (priv->is_joined, FALSE);
1351 g_mutex_lock (&priv->lock);
1352 element = gst_object_ref (priv->appsrc[0]);
1353 g_mutex_unlock (&priv->lock);
1355 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1357 gst_object_unref (element);
1363 * gst_rtsp_stream_recv_rtcp:
1364 * @stream: a #GstRTSPStream
1365 * @buffer: (transfer full): a #GstBuffer
1367 * Handle an RTCP buffer for the stream. This method is usually called when a
1368 * message has been received from a client using the TCP transport.
1370 * This function takes ownership of @buffer.
1372 * Returns: a GstFlowReturn.
1375 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1377 GstRTSPStreamPrivate *priv;
1379 GstElement *element;
1381 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1382 priv = stream->priv;
1383 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1384 g_return_val_if_fail (priv->is_joined, FALSE);
1386 g_mutex_lock (&priv->lock);
1387 element = gst_object_ref (priv->appsrc[1]);
1388 g_mutex_unlock (&priv->lock);
1390 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1392 gst_object_unref (element);
1397 /* must be called with lock */
1399 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1402 GstRTSPStreamPrivate *priv = stream->priv;
1403 const GstRTSPTransport *tr;
1405 tr = gst_rtsp_stream_transport_get_transport (trans);
1407 switch (tr->lower_transport) {
1408 case GST_RTSP_LOWER_TRANS_UDP:
1409 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1415 dest = tr->destination;
1416 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1421 min = tr->client_port.min;
1422 max = tr->client_port.max;
1426 GST_INFO ("adding %s:%d-%d", dest, min, max);
1427 g_signal_emit_by_name (priv->udpsink[0], "add", dest, min, NULL);
1428 g_signal_emit_by_name (priv->udpsink[1], "add", dest, max, NULL);
1430 GST_INFO ("setting ttl-mc %d", ttl);
1431 g_object_set (G_OBJECT (priv->udpsink[0]), "ttl-mc", ttl, NULL);
1432 g_object_set (G_OBJECT (priv->udpsink[1]), "ttl-mc", ttl, NULL);
1434 priv->transports = g_list_prepend (priv->transports, trans);
1436 GST_INFO ("removing %s:%d-%d", dest, min, max);
1437 g_signal_emit_by_name (priv->udpsink[0], "remove", dest, min, NULL);
1438 g_signal_emit_by_name (priv->udpsink[1], "remove", dest, max, NULL);
1439 priv->transports = g_list_remove (priv->transports, trans);
1443 case GST_RTSP_LOWER_TRANS_TCP:
1445 GST_INFO ("adding TCP %s", tr->destination);
1446 priv->transports = g_list_prepend (priv->transports, trans);
1448 GST_INFO ("removing TCP %s", tr->destination);
1449 priv->transports = g_list_remove (priv->transports, trans);
1453 goto unknown_transport;
1460 GST_INFO ("Unknown transport %d", tr->lower_transport);
1467 * gst_rtsp_stream_add_transport:
1468 * @stream: a #GstRTSPStream
1469 * @trans: a #GstRTSPStreamTransport
1471 * Add the transport in @trans to @stream. The media of @stream will
1472 * then also be send to the values configured in @trans.
1474 * @stream must be joined to a bin.
1476 * @trans must contain a valid #GstRTSPTransport.
1478 * Returns: %TRUE if @trans was added
1481 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1482 GstRTSPStreamTransport * trans)
1484 GstRTSPStreamPrivate *priv;
1487 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1488 priv = stream->priv;
1489 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1490 g_return_val_if_fail (priv->is_joined, FALSE);
1492 g_mutex_lock (&priv->lock);
1493 res = update_transport (stream, trans, TRUE);
1494 g_mutex_unlock (&priv->lock);
1500 * gst_rtsp_stream_remove_transport:
1501 * @stream: a #GstRTSPStream
1502 * @trans: a #GstRTSPStreamTransport
1504 * Remove the transport in @trans from @stream. The media of @stream will
1505 * not be sent to the values configured in @trans.
1507 * @stream must be joined to a bin.
1509 * @trans must contain a valid #GstRTSPTransport.
1511 * Returns: %TRUE if @trans was removed
1514 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1515 GstRTSPStreamTransport * trans)
1517 GstRTSPStreamPrivate *priv;
1520 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1521 priv = stream->priv;
1522 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1523 g_return_val_if_fail (priv->is_joined, FALSE);
1525 g_mutex_lock (&priv->lock);
1526 res = update_transport (stream, trans, FALSE);
1527 g_mutex_unlock (&priv->lock);