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_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
355 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
357 goto no_udp_protocol;
359 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
360 "send-duplicates")) {
361 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
362 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
365 ("old multiudpsink version found without send-duplicates property");
368 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
370 g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size,
373 GST_WARNING ("multiudpsink version found without buffer-size property");
376 g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
377 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
378 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
379 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
380 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
381 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
382 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
383 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
384 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
386 /* we keep these elements, we will further configure them when the
387 * client told us to really use the UDP ports. */
388 stream->udpsrc[0] = udpsrc0;
389 stream->udpsrc[1] = udpsrc1;
390 stream->udpsink[0] = udpsink0;
391 stream->udpsink[1] = udpsink1;
392 stream->server_port.min = rtpport;
393 stream->server_port.max = rtcpport;
406 no_udp_rtcp_protocol:
417 gst_element_set_state (udpsrc0, GST_STATE_NULL);
418 gst_object_unref (udpsrc0);
421 gst_element_set_state (udpsrc1, GST_STATE_NULL);
422 gst_object_unref (udpsrc1);
425 gst_element_set_state (udpsink0, GST_STATE_NULL);
426 gst_object_unref (udpsink0);
429 gst_element_set_state (udpsink1, GST_STATE_NULL);
430 gst_object_unref (udpsink1);
436 /* executed from streaming thread */
438 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
440 GstCaps *newcaps, *oldcaps;
442 newcaps = gst_pad_get_current_caps (pad);
444 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
447 g_mutex_lock (&stream->lock);
448 oldcaps = stream->caps;
449 stream->caps = newcaps;
450 g_mutex_unlock (&stream->lock);
453 gst_caps_unref (oldcaps);
457 dump_structure (const GstStructure * s)
461 sstr = gst_structure_to_string (s);
462 GST_INFO ("structure: %s", sstr);
466 static GstRTSPStreamTransport *
467 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
470 GstRTSPStreamTransport *result = NULL;
475 if (rtcp_from == NULL)
478 tmp = g_strrstr (rtcp_from, ":");
482 port = atoi (tmp + 1);
483 dest = g_strndup (rtcp_from, tmp - rtcp_from);
485 g_mutex_lock (&stream->lock);
486 GST_INFO ("finding %s:%d in %d transports", dest, port,
487 g_list_length (stream->transports));
489 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
490 GstRTSPStreamTransport *trans = walk->data;
493 min = trans->transport->client_port.min;
494 max = trans->transport->client_port.max;
496 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
502 g_mutex_unlock (&stream->lock);
509 static GstRTSPStreamTransport *
510 check_transport (GObject * source, GstRTSPStream * stream)
513 GstRTSPStreamTransport *trans;
515 /* see if we have a stream to match with the origin of the RTCP packet */
516 trans = g_object_get_qdata (source, ssrc_stream_map_key);
518 g_object_get (source, "stats", &stats, NULL);
520 const gchar *rtcp_from;
522 dump_structure (stats);
524 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
525 if ((trans = find_transport (stream, rtcp_from))) {
526 GST_INFO ("%p: found transport %p for source %p", stream, trans,
529 /* keep ref to the source */
530 trans->rtpsource = source;
532 g_object_set_qdata (source, ssrc_stream_map_key, trans);
534 gst_structure_free (stats);
543 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
545 GstRTSPStreamTransport *trans;
547 GST_INFO ("%p: new source %p", stream, source);
549 trans = check_transport (source, stream);
552 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
556 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
558 GST_INFO ("%p: new SDES %p", stream, source);
562 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
564 GstRTSPStreamTransport *trans;
566 trans = check_transport (source, stream);
569 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
570 gst_rtsp_stream_transport_keep_alive (trans);
575 g_object_get (source, "stats", &stats, NULL);
577 dump_structure (stats);
578 gst_structure_free (stats);
585 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
587 GST_INFO ("%p: source %p bye", stream, source);
591 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
593 GstRTSPStreamTransport *trans;
595 GST_INFO ("%p: source %p bye timeout", stream, source);
597 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
598 trans->rtpsource = NULL;
599 trans->timeout = TRUE;
604 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
606 GstRTSPStreamTransport *trans;
608 GST_INFO ("%p: source %p timeout", stream, source);
610 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
611 trans->rtpsource = NULL;
612 trans->timeout = TRUE;
617 handle_new_sample (GstAppSink * sink, gpointer user_data)
622 GstRTSPStream *stream;
624 sample = gst_app_sink_pull_sample (sink);
628 stream = (GstRTSPStream *) user_data;
629 buffer = gst_sample_get_buffer (sample);
631 g_mutex_lock (&stream->lock);
632 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
633 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
635 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
636 gst_rtsp_stream_transport_send_rtp (tr, buffer);
638 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
641 g_mutex_unlock (&stream->lock);
643 gst_sample_unref (sample);
648 static GstAppSinkCallbacks sink_cb = {
649 NULL, /* not interested in EOS */
650 NULL, /* not interested in preroll samples */
655 * gst_rtsp_stream_join_bin:
656 * @stream: a #GstRTSPStream
657 * @bin: a #GstBin to join
658 * @rtpbin: a rtpbin element in @bin
659 * @state: the target state of the new elements
661 * Join the #Gstbin @bin that contains the element @rtpbin.
663 * @stream will link to @rtpbin, which must be inside @bin. The elements
664 * added to @bin will be set to the state given in @state.
666 * Returns: %TRUE on success.
669 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
670 GstElement * rtpbin, GstState state)
674 GstPad *pad, *teepad, *queuepad, *selpad;
675 GstPadLinkReturn ret;
677 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
678 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
679 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
681 g_mutex_lock (&stream->lock);
682 if (stream->is_joined)
685 /* create a session with the same index as the stream */
688 GST_INFO ("stream %p joining bin as session %d", stream, idx);
690 if (!alloc_ports (stream))
693 /* get a pad for sending RTP */
694 name = g_strdup_printf ("send_rtp_sink_%u", idx);
695 stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
697 /* link the RTP pad to the session manager, it should not really fail unless
698 * this is not really an RTP pad */
699 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
700 if (ret != GST_PAD_LINK_OK)
703 /* get pads from the RTP session element for sending and receiving
705 name = g_strdup_printf ("send_rtp_src_%u", idx);
706 stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
708 name = g_strdup_printf ("send_rtcp_src_%u", idx);
709 stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
711 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
712 stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
714 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
715 stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
718 /* get the session */
719 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
721 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
723 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
725 g_signal_connect (stream->session, "on-ssrc-active",
726 (GCallback) on_ssrc_active, stream);
727 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
729 g_signal_connect (stream->session, "on-bye-timeout",
730 (GCallback) on_bye_timeout, stream);
731 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
734 for (i = 0; i < 2; i++) {
735 /* For the sender we create this bit of pipeline for both
736 * RTP and RTCP. Sync and preroll are enabled on udpsink so
737 * we need to add a queue before appsink to make the pipeline
738 * not block. For the TCP case, we want to pump data to the
739 * client as fast as possible anyway.
741 * .--------. .-----. .---------.
742 * | rtpbin | | tee | | udpsink |
743 * | send->sink src->sink |
744 * '--------' | | '---------'
745 * | | .---------. .---------.
746 * | | | queue | | appsink |
747 * | src->sink src->sink |
748 * '-----' '---------' '---------'
750 /* make tee for RTP/RTCP */
751 stream->tee[i] = gst_element_factory_make ("tee", NULL);
752 gst_bin_add (bin, stream->tee[i]);
754 /* and link to rtpbin send pad */
755 pad = gst_element_get_static_pad (stream->tee[i], "sink");
756 gst_pad_link (stream->send_src[i], pad);
757 gst_object_unref (pad);
760 gst_bin_add (bin, stream->udpsink[i]);
762 /* link tee to udpsink */
763 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
764 pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
765 gst_pad_link (teepad, pad);
766 gst_object_unref (pad);
767 gst_object_unref (teepad);
770 stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
771 gst_bin_add (bin, stream->appqueue[i]);
772 /* and link to tee */
773 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
774 pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
775 gst_pad_link (teepad, pad);
776 gst_object_unref (pad);
777 gst_object_unref (teepad);
780 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
781 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
782 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
783 gst_bin_add (bin, stream->appsink[i]);
784 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
785 &sink_cb, stream, NULL);
786 /* and link to queue */
787 queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
788 pad = gst_element_get_static_pad (stream->appsink[i], "sink");
789 gst_pad_link (queuepad, pad);
790 gst_object_unref (pad);
791 gst_object_unref (queuepad);
793 /* For the receiver we create this bit of pipeline for both
794 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
795 * and it is all funneled into the rtpbin receive pad.
797 * .--------. .--------. .--------.
798 * | udpsrc | | funnel | | rtpbin |
799 * | src->sink src->sink |
800 * '--------' | | '--------'
804 * '--------' '--------'
806 /* make funnel for the RTP/RTCP receivers */
807 stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
808 gst_bin_add (bin, stream->funnel[i]);
810 pad = gst_element_get_static_pad (stream->funnel[i], "src");
811 gst_pad_link (pad, stream->recv_sink[i]);
812 gst_object_unref (pad);
815 gst_bin_add (bin, stream->udpsrc[i]);
816 /* and link to the funnel */
817 selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
818 pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
819 gst_pad_link (pad, selpad);
820 gst_object_unref (pad);
821 gst_object_unref (selpad);
823 /* make and add appsrc */
824 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
825 gst_bin_add (bin, stream->appsrc[i]);
826 /* and link to the funnel */
827 selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
828 pad = gst_element_get_static_pad (stream->appsrc[i], "src");
829 gst_pad_link (pad, selpad);
830 gst_object_unref (pad);
831 gst_object_unref (selpad);
833 /* check if we need to set to a special state */
834 if (state != GST_STATE_NULL) {
835 gst_element_set_state (stream->udpsink[i], state);
836 gst_element_set_state (stream->appsink[i], state);
837 gst_element_set_state (stream->appqueue[i], state);
838 gst_element_set_state (stream->tee[i], state);
839 gst_element_set_state (stream->funnel[i], state);
840 gst_element_set_state (stream->appsrc[i], state);
842 /* we set and keep these to playing so that they don't cause NO_PREROLL return
844 gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
845 gst_element_set_locked_state (stream->udpsrc[i], TRUE);
848 /* be notified of caps changes */
849 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
850 (GCallback) caps_notify, stream);
852 stream->is_joined = TRUE;
853 g_mutex_unlock (&stream->lock);
860 g_mutex_unlock (&stream->lock);
865 g_mutex_unlock (&stream->lock);
866 GST_WARNING ("failed to allocate ports %d", idx);
871 GST_WARNING ("failed to link stream %d", idx);
872 gst_object_unref (stream->send_rtp_sink);
873 stream->send_rtp_sink = NULL;
874 g_mutex_unlock (&stream->lock);
880 * gst_rtsp_stream_leave_bin:
881 * @stream: a #GstRTSPStream
883 * @rtpbin: a rtpbin #GstElement
885 * Remove the elements of @stream from @bin. @bin must be set
886 * to the NULL state before calling this.
888 * Return: %TRUE on success.
891 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
896 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
897 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
898 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
900 g_mutex_lock (&stream->lock);
901 if (!stream->is_joined)
904 /* all transports must be removed by now */
905 g_return_val_if_fail (stream->transports == NULL, FALSE);
907 GST_INFO ("stream %p leaving bin", stream);
909 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
910 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
911 gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
912 gst_object_unref (stream->send_rtp_sink);
913 stream->send_rtp_sink = NULL;
915 for (i = 0; i < 2; i++) {
916 /* and set udpsrc to NULL now before removing */
917 gst_element_set_locked_state (stream->udpsrc[i], FALSE);
918 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
920 /* removing them should also nicely release the request
921 * pads when they finalize */
922 gst_bin_remove (bin, stream->udpsrc[i]);
923 gst_bin_remove (bin, stream->udpsink[i]);
924 gst_bin_remove (bin, stream->appsrc[i]);
925 gst_bin_remove (bin, stream->appsink[i]);
926 gst_bin_remove (bin, stream->appqueue[i]);
927 gst_bin_remove (bin, stream->tee[i]);
928 gst_bin_remove (bin, stream->funnel[i]);
930 gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
931 gst_object_unref (stream->recv_sink[i]);
932 stream->recv_sink[i] = NULL;
934 stream->udpsrc[i] = NULL;
935 stream->udpsink[i] = NULL;
936 stream->appsrc[i] = NULL;
937 stream->appsink[i] = NULL;
938 stream->appqueue[i] = NULL;
939 stream->tee[i] = NULL;
940 stream->funnel[i] = NULL;
942 gst_object_unref (stream->send_src[0]);
943 stream->send_src[0] = NULL;
945 gst_element_release_request_pad (rtpbin, stream->send_src[1]);
946 gst_object_unref (stream->send_src[1]);
947 stream->send_src[1] = NULL;
949 g_object_unref (stream->session);
951 gst_caps_unref (stream->caps);
953 stream->is_joined = FALSE;
954 g_mutex_unlock (&stream->lock);
965 * gst_rtsp_stream_get_rtpinfo:
966 * @stream: a #GstRTSPStream
967 * @rtptime: result RTP timestamp
968 * @seq: result RTP seqnum
970 * Retrieve the current rtptime and seq. This is used to
971 * construct a RTPInfo reply header.
973 * Returns: %TRUE when rtptime and seq could be determined.
976 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
977 guint * rtptime, guint * seq)
979 GObjectClass *payobjclass;
981 payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
983 if (!g_object_class_find_property (payobjclass, "seqnum") ||
984 !g_object_class_find_property (payobjclass, "timestamp"))
987 g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
993 * gst_rtsp_stream_recv_rtp:
994 * @stream: a #GstRTSPStream
995 * @buffer: (transfer full): a #GstBuffer
997 * Handle an RTP buffer for the stream. This method is usually called when a
998 * message has been received from a client using the TCP transport.
1000 * This function takes ownership of @buffer.
1002 * Returns: a GstFlowReturn.
1005 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1008 GstElement *element;
1010 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1011 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1012 g_return_val_if_fail (stream->is_joined, FALSE);
1014 g_mutex_lock (&stream->lock);
1015 element = gst_object_ref (stream->appsrc[0]);
1016 g_mutex_unlock (&stream->lock);
1018 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1020 gst_object_unref (element);
1026 * gst_rtsp_stream_recv_rtcp:
1027 * @stream: a #GstRTSPStream
1028 * @buffer: (transfer full): a #GstBuffer
1030 * Handle an RTCP buffer for the stream. This method is usually called when a
1031 * message has been received from a client using the TCP transport.
1033 * This function takes ownership of @buffer.
1035 * Returns: a GstFlowReturn.
1038 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1041 GstElement *element;
1043 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1044 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1045 g_return_val_if_fail (stream->is_joined, FALSE);
1047 g_mutex_lock (&stream->lock);
1048 element = gst_object_ref (stream->appsrc[1]);
1049 g_mutex_unlock (&stream->lock);
1051 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1053 gst_object_unref (element);
1058 /* must be called with lock */
1060 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1063 GstRTSPTransport *tr;
1068 tr = trans->transport;
1070 switch (tr->lower_transport) {
1071 case GST_RTSP_LOWER_TRANS_UDP:
1072 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1078 dest = tr->destination;
1079 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1084 min = tr->client_port.min;
1085 max = tr->client_port.max;
1088 if (add && !trans->active) {
1089 GST_INFO ("adding %s:%d-%d", dest, min, max);
1090 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1091 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1093 GST_INFO ("setting ttl-mc %d", ttl);
1094 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
1095 g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
1097 stream->transports = g_list_prepend (stream->transports, trans);
1098 trans->active = TRUE;
1100 } else if (trans->active) {
1101 GST_INFO ("removing %s:%d-%d", dest, min, max);
1102 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1103 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1104 stream->transports = g_list_remove (stream->transports, trans);
1105 trans->active = FALSE;
1110 case GST_RTSP_LOWER_TRANS_TCP:
1111 if (add && !trans->active) {
1112 GST_INFO ("adding TCP %s", tr->destination);
1113 stream->transports = g_list_prepend (stream->transports, trans);
1114 trans->active = TRUE;
1116 } else if (trans->active) {
1117 GST_INFO ("removing TCP %s", tr->destination);
1118 stream->transports = g_list_remove (stream->transports, trans);
1119 trans->active = FALSE;
1124 GST_INFO ("Unknown transport %d", tr->lower_transport);
1132 * gst_rtsp_stream_add_transport:
1133 * @stream: a #GstRTSPStream
1134 * @trans: a #GstRTSPStreamTransport
1136 * Add the transport in @trans to @stream. The media of @stream will
1137 * then also be send to the values configured in @trans.
1139 * @stream must be joined to a bin.
1141 * @trans must contain a valid #GstRTSPTransport.
1143 * Returns: %TRUE if @trans was added
1146 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1147 GstRTSPStreamTransport * trans)
1151 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1152 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1153 g_return_val_if_fail (stream->is_joined, FALSE);
1154 g_return_val_if_fail (trans->transport != NULL, FALSE);
1156 g_mutex_lock (&stream->lock);
1157 res = update_transport (stream, trans, TRUE);
1158 g_mutex_unlock (&stream->lock);
1164 * gst_rtsp_stream_remove_transport:
1165 * @stream: a #GstRTSPStream
1166 * @trans: a #GstRTSPStreamTransport
1168 * Remove the transport in @trans from @stream. The media of @stream will
1169 * not be sent to the values configured in @trans.
1171 * @stream must be joined to a bin.
1173 * @trans must contain a valid #GstRTSPTransport.
1175 * Returns: %TRUE if @trans was removed
1178 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1179 GstRTSPStreamTransport * trans)
1183 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1184 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1185 g_return_val_if_fail (stream->is_joined, FALSE);
1186 g_return_val_if_fail (trans->transport != NULL, FALSE);
1188 g_mutex_lock (&stream->lock);
1189 res = update_transport (stream, trans, FALSE);
1190 g_mutex_unlock (&stream->lock);