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"
36 GST_DEBUG_CATEGORY_STATIC (rtsp_stream_debug);
37 #define GST_CAT_DEFAULT rtsp_stream_debug
39 static GQuark ssrc_stream_map_key;
41 static void gst_rtsp_stream_finalize (GObject * obj);
43 G_DEFINE_TYPE (GstRTSPStream, gst_rtsp_stream, G_TYPE_OBJECT);
46 gst_rtsp_stream_class_init (GstRTSPStreamClass * klass)
48 GObjectClass *gobject_class;
50 gobject_class = G_OBJECT_CLASS (klass);
52 gobject_class->finalize = gst_rtsp_stream_finalize;
54 GST_DEBUG_CATEGORY_INIT (rtsp_stream_debug, "rtspstream", 0, "GstRTSPStream");
56 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
60 gst_rtsp_stream_init (GstRTSPStream * stream)
62 g_mutex_init (&stream->lock);
66 gst_rtsp_stream_finalize (GObject * obj)
68 GstRTSPStream *stream;
70 stream = GST_RTSP_STREAM (obj);
72 /* we really need to be unjoined now */
73 g_return_if_fail (!stream->is_joined);
76 gst_rtsp_address_free (stream->addr);
78 g_object_unref (stream->pool);
79 gst_object_unref (stream->payloader);
80 gst_object_unref (stream->srcpad);
81 g_mutex_clear (&stream->lock);
83 G_OBJECT_CLASS (gst_rtsp_stream_parent_class)->finalize (obj);
87 * gst_rtsp_stream_new:
90 * @payloader: a #GstElement
92 * Create a new media stream with index @idx that handles RTP data on
93 * @srcpad and has a payloader element @payloader.
95 * Returns: a new #GstRTSPStream
98 gst_rtsp_stream_new (guint idx, GstElement * payloader, GstPad * srcpad)
100 GstRTSPStream *stream;
102 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
103 g_return_val_if_fail (GST_IS_PAD (srcpad), NULL);
104 g_return_val_if_fail (GST_PAD_IS_SRC (srcpad), NULL);
106 stream = g_object_new (GST_TYPE_RTSP_STREAM, NULL);
108 stream->payloader = gst_object_ref (payloader);
109 stream->srcpad = gst_object_ref (srcpad);
115 * gst_rtsp_stream_set_mtu:
116 * @stream: a #GstRTSPStream
119 * Configure the mtu in the payloader of @stream to @mtu.
122 gst_rtsp_stream_set_mtu (GstRTSPStream * stream, guint mtu)
124 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
126 GST_LOG_OBJECT (stream, "set MTU %u", mtu);
128 g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL);
132 * gst_rtsp_stream_get_mtu:
133 * @stream: a #GstRTSPStream
135 * Get the configured MTU in the payloader of @stream.
137 * Returns: the MTU of the payloader.
140 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
144 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
146 g_object_get (G_OBJECT (stream->payloader), "mtu", &mtu, NULL);
152 * gst_rtsp_stream_set_address_pool:
153 * @stream: a #GstRTSPStream
154 * @pool: a #GstRTSPAddressPool
156 * configure @pool to be used as the address pool of @stream.
159 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
160 GstRTSPAddressPool * pool)
162 GstRTSPAddressPool *old;
164 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
166 GST_LOG_OBJECT (stream, "set address pool %p", pool);
168 g_mutex_lock (&stream->lock);
169 if ((old = stream->pool) != pool)
170 stream->pool = pool ? g_object_ref (pool) : NULL;
173 g_mutex_unlock (&stream->lock);
176 g_object_unref (old);
180 * gst_rtsp_stream_get_address_pool:
181 * @stream: a #GstRTSPStream
183 * Get the #GstRTSPAddressPool used as the address pool of @stream.
185 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
189 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
191 GstRTSPAddressPool *result;
193 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
195 g_mutex_lock (&stream->lock);
196 if ((result = stream->pool))
197 g_object_ref (result);
198 g_mutex_unlock (&stream->lock);
204 * gst_rtsp_stream_get_address:
205 * @stream: a #GstRTSPStream
207 * Get the multicast address of @stream.
209 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
210 * allocated. gst_rtsp_address_free() after usage.
213 gst_rtsp_stream_get_address (GstRTSPStream * stream)
215 GstRTSPAddress *result;
217 g_mutex_lock (&stream->lock);
218 if (stream->addr == NULL) {
219 if (stream->pool == NULL)
222 stream->addr = gst_rtsp_address_pool_acquire_address (stream->pool,
223 GST_RTSP_ADDRESS_FLAG_EVEN_PORT, 2);
224 if (stream->addr == NULL)
227 result = gst_rtsp_address_copy (stream->addr);
228 g_mutex_unlock (&stream->lock);
235 GST_ERROR_OBJECT (stream, "no address pool specified");
236 g_mutex_unlock (&stream->lock);
241 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
242 g_mutex_unlock (&stream->lock);
247 /* must be called with lock */
249 alloc_ports (GstRTSPStream * stream)
251 GstStateChangeReturn ret;
252 GstElement *udpsrc0, *udpsrc1;
253 GstElement *udpsink0, *udpsink1;
254 gint tmp_rtp, tmp_rtcp;
256 gint rtpport, rtcpport;
260 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
268 /* Start with random port */
272 host = "udp://[::0]";
274 host = "udp://0.0.0.0";
276 /* try to allocate 2 UDP ports, the RTP port should be an even
277 * number and the RTCP port should be the next (uneven) port */
279 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
281 goto no_udp_protocol;
282 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
284 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
285 if (ret == GST_STATE_CHANGE_FAILURE) {
291 gst_element_set_state (udpsrc0, GST_STATE_NULL);
292 gst_object_unref (udpsrc0);
296 goto no_udp_protocol;
299 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
301 /* check if port is even */
302 if ((tmp_rtp & 1) != 0) {
303 /* port not even, close and allocate another */
307 gst_element_set_state (udpsrc0, GST_STATE_NULL);
308 gst_object_unref (udpsrc0);
314 /* allocate port+1 for RTCP now */
315 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
317 goto no_udp_rtcp_protocol;
320 tmp_rtcp = tmp_rtp + 1;
321 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
323 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
324 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
325 if (ret == GST_STATE_CHANGE_FAILURE) {
330 gst_element_set_state (udpsrc0, GST_STATE_NULL);
331 gst_object_unref (udpsrc0);
333 gst_element_set_state (udpsrc1, GST_STATE_NULL);
334 gst_object_unref (udpsrc1);
339 /* all fine, do port check */
340 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
341 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
343 /* this should not happen... */
344 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
347 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
349 goto no_udp_protocol;
351 g_object_get (G_OBJECT (udpsrc0), "used-socket", &socket, NULL);
352 g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
353 g_object_unref (socket);
354 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
356 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
358 goto no_udp_protocol;
360 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
361 "send-duplicates")) {
362 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
363 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
366 ("old multiudpsink version found without send-duplicates property");
369 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
371 g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size,
374 GST_WARNING ("multiudpsink version found without buffer-size property");
377 g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
378 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
379 g_object_unref (socket);
380 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
381 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
382 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
383 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
384 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
385 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
386 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
388 /* we keep these elements, we will further configure them when the
389 * client told us to really use the UDP ports. */
390 stream->udpsrc[0] = udpsrc0;
391 stream->udpsrc[1] = udpsrc1;
392 stream->udpsink[0] = udpsink0;
393 stream->udpsink[1] = udpsink1;
394 stream->server_port.min = rtpport;
395 stream->server_port.max = rtcpport;
408 no_udp_rtcp_protocol:
419 gst_element_set_state (udpsrc0, GST_STATE_NULL);
420 gst_object_unref (udpsrc0);
423 gst_element_set_state (udpsrc1, GST_STATE_NULL);
424 gst_object_unref (udpsrc1);
427 gst_element_set_state (udpsink0, GST_STATE_NULL);
428 gst_object_unref (udpsink0);
431 gst_element_set_state (udpsink1, GST_STATE_NULL);
432 gst_object_unref (udpsink1);
438 /* executed from streaming thread */
440 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
442 GstCaps *newcaps, *oldcaps;
444 newcaps = gst_pad_get_current_caps (pad);
446 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
449 g_mutex_lock (&stream->lock);
450 oldcaps = stream->caps;
451 stream->caps = newcaps;
452 g_mutex_unlock (&stream->lock);
455 gst_caps_unref (oldcaps);
459 dump_structure (const GstStructure * s)
463 sstr = gst_structure_to_string (s);
464 GST_INFO ("structure: %s", sstr);
468 static GstRTSPStreamTransport *
469 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
472 GstRTSPStreamTransport *result = NULL;
477 if (rtcp_from == NULL)
480 tmp = g_strrstr (rtcp_from, ":");
484 port = atoi (tmp + 1);
485 dest = g_strndup (rtcp_from, tmp - rtcp_from);
487 g_mutex_lock (&stream->lock);
488 GST_INFO ("finding %s:%d in %d transports", dest, port,
489 g_list_length (stream->transports));
491 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
492 GstRTSPStreamTransport *trans = walk->data;
495 min = trans->transport->client_port.min;
496 max = trans->transport->client_port.max;
498 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
504 g_mutex_unlock (&stream->lock);
511 static GstRTSPStreamTransport *
512 check_transport (GObject * source, GstRTSPStream * stream)
515 GstRTSPStreamTransport *trans;
517 /* see if we have a stream to match with the origin of the RTCP packet */
518 trans = g_object_get_qdata (source, ssrc_stream_map_key);
520 g_object_get (source, "stats", &stats, NULL);
522 const gchar *rtcp_from;
524 dump_structure (stats);
526 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
527 if ((trans = find_transport (stream, rtcp_from))) {
528 GST_INFO ("%p: found transport %p for source %p", stream, trans,
531 /* keep ref to the source */
532 trans->rtpsource = source;
534 g_object_set_qdata (source, ssrc_stream_map_key, trans);
536 gst_structure_free (stats);
545 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
547 GstRTSPStreamTransport *trans;
549 GST_INFO ("%p: new source %p", stream, source);
551 trans = check_transport (source, stream);
554 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
558 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
560 GST_INFO ("%p: new SDES %p", stream, source);
564 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
566 GstRTSPStreamTransport *trans;
568 trans = check_transport (source, stream);
571 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
572 gst_rtsp_stream_transport_keep_alive (trans);
577 g_object_get (source, "stats", &stats, NULL);
579 dump_structure (stats);
580 gst_structure_free (stats);
587 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
589 GST_INFO ("%p: source %p bye", stream, source);
593 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
595 GstRTSPStreamTransport *trans;
597 GST_INFO ("%p: source %p bye timeout", stream, source);
599 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
600 trans->rtpsource = NULL;
601 trans->timeout = TRUE;
606 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
608 GstRTSPStreamTransport *trans;
610 GST_INFO ("%p: source %p timeout", stream, source);
612 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
613 trans->rtpsource = NULL;
614 trans->timeout = TRUE;
619 handle_new_sample (GstAppSink * sink, gpointer user_data)
624 GstRTSPStream *stream;
626 sample = gst_app_sink_pull_sample (sink);
630 stream = (GstRTSPStream *) user_data;
631 buffer = gst_sample_get_buffer (sample);
633 g_mutex_lock (&stream->lock);
634 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
635 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
637 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
638 gst_rtsp_stream_transport_send_rtp (tr, buffer);
640 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
643 g_mutex_unlock (&stream->lock);
645 gst_sample_unref (sample);
650 static GstAppSinkCallbacks sink_cb = {
651 NULL, /* not interested in EOS */
652 NULL, /* not interested in preroll samples */
657 * gst_rtsp_stream_join_bin:
658 * @stream: a #GstRTSPStream
659 * @bin: a #GstBin to join
660 * @rtpbin: a rtpbin element in @bin
661 * @state: the target state of the new elements
663 * Join the #Gstbin @bin that contains the element @rtpbin.
665 * @stream will link to @rtpbin, which must be inside @bin. The elements
666 * added to @bin will be set to the state given in @state.
668 * Returns: %TRUE on success.
671 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
672 GstElement * rtpbin, GstState state)
676 GstPad *pad, *teepad, *queuepad, *selpad;
677 GstPadLinkReturn ret;
679 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
680 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
681 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
683 g_mutex_lock (&stream->lock);
684 if (stream->is_joined)
687 /* create a session with the same index as the stream */
690 GST_INFO ("stream %p joining bin as session %d", stream, idx);
692 if (!alloc_ports (stream))
695 /* get a pad for sending RTP */
696 name = g_strdup_printf ("send_rtp_sink_%u", idx);
697 stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
699 /* link the RTP pad to the session manager, it should not really fail unless
700 * this is not really an RTP pad */
701 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
702 if (ret != GST_PAD_LINK_OK)
705 /* get pads from the RTP session element for sending and receiving
707 name = g_strdup_printf ("send_rtp_src_%u", idx);
708 stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
710 name = g_strdup_printf ("send_rtcp_src_%u", idx);
711 stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
713 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
714 stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
716 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
717 stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
720 /* get the session */
721 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
723 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
725 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
727 g_signal_connect (stream->session, "on-ssrc-active",
728 (GCallback) on_ssrc_active, stream);
729 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
731 g_signal_connect (stream->session, "on-bye-timeout",
732 (GCallback) on_bye_timeout, stream);
733 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
736 for (i = 0; i < 2; i++) {
737 /* For the sender we create this bit of pipeline for both
738 * RTP and RTCP. Sync and preroll are enabled on udpsink so
739 * we need to add a queue before appsink to make the pipeline
740 * not block. For the TCP case, we want to pump data to the
741 * client as fast as possible anyway.
743 * .--------. .-----. .---------.
744 * | rtpbin | | tee | | udpsink |
745 * | send->sink src->sink |
746 * '--------' | | '---------'
747 * | | .---------. .---------.
748 * | | | queue | | appsink |
749 * | src->sink src->sink |
750 * '-----' '---------' '---------'
752 /* make tee for RTP/RTCP */
753 stream->tee[i] = gst_element_factory_make ("tee", NULL);
754 gst_bin_add (bin, stream->tee[i]);
756 /* and link to rtpbin send pad */
757 pad = gst_element_get_static_pad (stream->tee[i], "sink");
758 gst_pad_link (stream->send_src[i], pad);
759 gst_object_unref (pad);
762 gst_bin_add (bin, stream->udpsink[i]);
764 /* link tee to udpsink */
765 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
766 pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
767 gst_pad_link (teepad, pad);
768 gst_object_unref (pad);
769 gst_object_unref (teepad);
772 stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
773 gst_bin_add (bin, stream->appqueue[i]);
774 /* and link to tee */
775 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
776 pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
777 gst_pad_link (teepad, pad);
778 gst_object_unref (pad);
779 gst_object_unref (teepad);
782 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
783 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
784 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
785 gst_bin_add (bin, stream->appsink[i]);
786 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
787 &sink_cb, stream, NULL);
788 /* and link to queue */
789 queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
790 pad = gst_element_get_static_pad (stream->appsink[i], "sink");
791 gst_pad_link (queuepad, pad);
792 gst_object_unref (pad);
793 gst_object_unref (queuepad);
795 /* For the receiver we create this bit of pipeline for both
796 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
797 * and it is all funneled into the rtpbin receive pad.
799 * .--------. .--------. .--------.
800 * | udpsrc | | funnel | | rtpbin |
801 * | src->sink src->sink |
802 * '--------' | | '--------'
806 * '--------' '--------'
808 /* make funnel for the RTP/RTCP receivers */
809 stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
810 gst_bin_add (bin, stream->funnel[i]);
812 pad = gst_element_get_static_pad (stream->funnel[i], "src");
813 gst_pad_link (pad, stream->recv_sink[i]);
814 gst_object_unref (pad);
816 /* we set and keep these to playing so that they don't cause NO_PREROLL return
818 gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
819 gst_element_set_locked_state (stream->udpsrc[i], TRUE);
821 gst_bin_add (bin, stream->udpsrc[i]);
822 /* and link to the funnel */
823 selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
824 pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
825 gst_pad_link (pad, selpad);
826 gst_object_unref (pad);
827 gst_object_unref (selpad);
829 /* make and add appsrc */
830 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
831 gst_bin_add (bin, stream->appsrc[i]);
832 /* and link to the funnel */
833 selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
834 pad = gst_element_get_static_pad (stream->appsrc[i], "src");
835 gst_pad_link (pad, selpad);
836 gst_object_unref (pad);
837 gst_object_unref (selpad);
839 /* check if we need to set to a special state */
840 if (state != GST_STATE_NULL) {
841 gst_element_set_state (stream->udpsink[i], state);
842 gst_element_set_state (stream->appsink[i], state);
843 gst_element_set_state (stream->appqueue[i], state);
844 gst_element_set_state (stream->tee[i], state);
845 gst_element_set_state (stream->funnel[i], state);
846 gst_element_set_state (stream->appsrc[i], state);
850 /* be notified of caps changes */
851 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
852 (GCallback) caps_notify, stream);
854 stream->is_joined = TRUE;
855 g_mutex_unlock (&stream->lock);
862 g_mutex_unlock (&stream->lock);
867 g_mutex_unlock (&stream->lock);
868 GST_WARNING ("failed to allocate ports %d", idx);
873 GST_WARNING ("failed to link stream %d", idx);
874 gst_object_unref (stream->send_rtp_sink);
875 stream->send_rtp_sink = NULL;
876 g_mutex_unlock (&stream->lock);
882 * gst_rtsp_stream_leave_bin:
883 * @stream: a #GstRTSPStream
885 * @rtpbin: a rtpbin #GstElement
887 * Remove the elements of @stream from @bin. @bin must be set
888 * to the NULL state before calling this.
890 * Return: %TRUE on success.
893 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
898 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
899 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
900 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
902 g_mutex_lock (&stream->lock);
903 if (!stream->is_joined)
906 /* all transports must be removed by now */
907 g_return_val_if_fail (stream->transports == NULL, FALSE);
909 GST_INFO ("stream %p leaving bin", stream);
911 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
912 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
913 gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
914 gst_object_unref (stream->send_rtp_sink);
915 stream->send_rtp_sink = NULL;
917 for (i = 0; i < 2; i++) {
918 /* and set udpsrc to NULL now before removing */
919 gst_element_set_locked_state (stream->udpsrc[i], FALSE);
920 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
922 /* removing them should also nicely release the request
923 * pads when they finalize */
924 gst_bin_remove (bin, stream->udpsrc[i]);
925 gst_bin_remove (bin, stream->udpsink[i]);
926 gst_bin_remove (bin, stream->appsrc[i]);
927 gst_bin_remove (bin, stream->appsink[i]);
928 gst_bin_remove (bin, stream->appqueue[i]);
929 gst_bin_remove (bin, stream->tee[i]);
930 gst_bin_remove (bin, stream->funnel[i]);
932 gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
933 gst_object_unref (stream->recv_sink[i]);
934 stream->recv_sink[i] = NULL;
936 stream->udpsrc[i] = NULL;
937 stream->udpsink[i] = NULL;
938 stream->appsrc[i] = NULL;
939 stream->appsink[i] = NULL;
940 stream->appqueue[i] = NULL;
941 stream->tee[i] = NULL;
942 stream->funnel[i] = NULL;
944 gst_object_unref (stream->send_src[0]);
945 stream->send_src[0] = NULL;
947 gst_element_release_request_pad (rtpbin, stream->send_src[1]);
948 gst_object_unref (stream->send_src[1]);
949 stream->send_src[1] = NULL;
951 g_object_unref (stream->session);
953 gst_caps_unref (stream->caps);
955 stream->is_joined = FALSE;
956 g_mutex_unlock (&stream->lock);
967 * gst_rtsp_stream_get_rtpinfo:
968 * @stream: a #GstRTSPStream
969 * @rtptime: result RTP timestamp
970 * @seq: result RTP seqnum
972 * Retrieve the current rtptime and seq. This is used to
973 * construct a RTPInfo reply header.
975 * Returns: %TRUE when rtptime and seq could be determined.
978 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
979 guint * rtptime, guint * seq)
981 GObjectClass *payobjclass;
983 payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
985 if (!g_object_class_find_property (payobjclass, "seqnum") ||
986 !g_object_class_find_property (payobjclass, "timestamp"))
989 g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
995 * gst_rtsp_stream_recv_rtp:
996 * @stream: a #GstRTSPStream
997 * @buffer: (transfer full): a #GstBuffer
999 * Handle an RTP buffer for the stream. This method is usually called when a
1000 * message has been received from a client using the TCP transport.
1002 * This function takes ownership of @buffer.
1004 * Returns: a GstFlowReturn.
1007 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1010 GstElement *element;
1012 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1013 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1014 g_return_val_if_fail (stream->is_joined, FALSE);
1016 g_mutex_lock (&stream->lock);
1017 element = gst_object_ref (stream->appsrc[0]);
1018 g_mutex_unlock (&stream->lock);
1020 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1022 gst_object_unref (element);
1028 * gst_rtsp_stream_recv_rtcp:
1029 * @stream: a #GstRTSPStream
1030 * @buffer: (transfer full): a #GstBuffer
1032 * Handle an RTCP buffer for the stream. This method is usually called when a
1033 * message has been received from a client using the TCP transport.
1035 * This function takes ownership of @buffer.
1037 * Returns: a GstFlowReturn.
1040 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1043 GstElement *element;
1045 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1046 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1047 g_return_val_if_fail (stream->is_joined, FALSE);
1049 g_mutex_lock (&stream->lock);
1050 element = gst_object_ref (stream->appsrc[1]);
1051 g_mutex_unlock (&stream->lock);
1053 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1055 gst_object_unref (element);
1060 /* must be called with lock */
1062 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1065 GstRTSPTransport *tr;
1070 tr = trans->transport;
1072 switch (tr->lower_transport) {
1073 case GST_RTSP_LOWER_TRANS_UDP:
1074 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1080 dest = tr->destination;
1081 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1086 min = tr->client_port.min;
1087 max = tr->client_port.max;
1090 if (add && !trans->active) {
1091 GST_INFO ("adding %s:%d-%d", dest, min, max);
1092 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1093 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1095 GST_INFO ("setting ttl-mc %d", ttl);
1096 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
1097 g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
1099 stream->transports = g_list_prepend (stream->transports, trans);
1100 trans->active = TRUE;
1102 } else if (trans->active) {
1103 GST_INFO ("removing %s:%d-%d", dest, min, max);
1104 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1105 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1106 stream->transports = g_list_remove (stream->transports, trans);
1107 trans->active = FALSE;
1112 case GST_RTSP_LOWER_TRANS_TCP:
1113 if (add && !trans->active) {
1114 GST_INFO ("adding TCP %s", tr->destination);
1115 stream->transports = g_list_prepend (stream->transports, trans);
1116 trans->active = TRUE;
1118 } else if (trans->active) {
1119 GST_INFO ("removing TCP %s", tr->destination);
1120 stream->transports = g_list_remove (stream->transports, trans);
1121 trans->active = FALSE;
1126 GST_INFO ("Unknown transport %d", tr->lower_transport);
1134 * gst_rtsp_stream_add_transport:
1135 * @stream: a #GstRTSPStream
1136 * @trans: a #GstRTSPStreamTransport
1138 * Add the transport in @trans to @stream. The media of @stream will
1139 * then also be send to the values configured in @trans.
1141 * @stream must be joined to a bin.
1143 * @trans must contain a valid #GstRTSPTransport.
1145 * Returns: %TRUE if @trans was added
1148 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1149 GstRTSPStreamTransport * trans)
1153 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1154 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1155 g_return_val_if_fail (stream->is_joined, FALSE);
1156 g_return_val_if_fail (trans->transport != NULL, FALSE);
1158 g_mutex_lock (&stream->lock);
1159 res = update_transport (stream, trans, TRUE);
1160 g_mutex_unlock (&stream->lock);
1166 * gst_rtsp_stream_remove_transport:
1167 * @stream: a #GstRTSPStream
1168 * @trans: a #GstRTSPStreamTransport
1170 * Remove the transport in @trans from @stream. The media of @stream will
1171 * not be sent to the values configured in @trans.
1173 * @stream must be joined to a bin.
1175 * @trans must contain a valid #GstRTSPTransport.
1177 * Returns: %TRUE if @trans was removed
1180 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1181 GstRTSPStreamTransport * trans)
1185 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1186 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1187 g_return_val_if_fail (stream->is_joined, FALSE);
1188 g_return_val_if_fail (trans->transport != NULL, FALSE);
1190 g_mutex_lock (&stream->lock);
1191 res = update_transport (stream, trans, FALSE);
1192 g_mutex_unlock (&stream->lock);