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 g_object_set (G_OBJECT (stream->payloader), "mtu", mtu, NULL);
130 * gst_rtsp_stream_get_mtu:
131 * @stream: a #GstRTSPStream
133 * Get the configured MTU in the payloader of @stream.
135 * Returns: the MTU of the payloader.
138 gst_rtsp_stream_get_mtu (GstRTSPStream * stream)
142 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), 0);
144 g_object_get (G_OBJECT (stream->payloader), "mtu", &mtu, NULL);
150 * gst_rtsp_stream_set_address_pool:
151 * @stream: a #GstRTSPStream
152 * @pool: a #GstRTSPAddressPool
154 * configure @pool to be used as the address pool of @stream.
157 gst_rtsp_stream_set_address_pool (GstRTSPStream * stream,
158 GstRTSPAddressPool * pool)
160 GstRTSPAddressPool *old;
162 g_return_if_fail (GST_IS_RTSP_STREAM (stream));
164 g_mutex_lock (&stream->lock);
165 if ((old = stream->pool) != pool)
166 stream->pool = pool ? g_object_ref (pool) : NULL;
169 g_mutex_unlock (&stream->lock);
172 g_object_unref (old);
176 * gst_rtsp_stream_get_address_pool:
177 * @stream: a #GstRTSPStream
179 * Get the #GstRTSPAddressPool used as the address pool of @stream.
181 * Returns: (transfer full): the #GstRTSPAddressPool of @stream. g_object_unref() after
185 gst_rtsp_stream_get_address_pool (GstRTSPStream * stream)
187 GstRTSPAddressPool *result;
189 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), NULL);
191 g_mutex_lock (&stream->lock);
192 if ((result = stream->pool))
193 g_object_ref (result);
194 g_mutex_unlock (&stream->lock);
200 * gst_rtsp_stream_get_address:
201 * @stream: a #GstRTSPStream
203 * Get the multicast address of @stream.
205 * Returns: the #GstRTSPAddress of @stream or %NULL when no address could be
206 * allocated. gst_rtsp_address_free() after usage.
209 gst_rtsp_stream_get_address (GstRTSPStream * stream)
211 GstRTSPAddress *result;
213 g_mutex_lock (&stream->lock);
214 if (stream->addr == NULL) {
215 if (stream->pool == NULL)
218 stream->addr = gst_rtsp_address_pool_acquire_address (stream->pool,
219 GST_RTSP_ADDRESS_FLAG_EVEN_PORT, 2);
220 if (stream->addr == NULL)
223 result = gst_rtsp_address_copy (stream->addr);
224 g_mutex_unlock (&stream->lock);
231 GST_ERROR_OBJECT (stream, "no address pool specified");
232 g_mutex_unlock (&stream->lock);
237 GST_ERROR_OBJECT (stream, "failed to acquire address from pool");
238 g_mutex_unlock (&stream->lock);
243 /* must be called with lock */
245 alloc_ports (GstRTSPStream * stream)
247 GstStateChangeReturn ret;
248 GstElement *udpsrc0, *udpsrc1;
249 GstElement *udpsink0, *udpsink1;
250 gint tmp_rtp, tmp_rtcp;
252 gint rtpport, rtcpport;
256 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
264 /* Start with random port */
268 host = "udp://[::0]";
270 host = "udp://0.0.0.0";
272 /* try to allocate 2 UDP ports, the RTP port should be an even
273 * number and the RTCP port should be the next (uneven) port */
275 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
277 goto no_udp_protocol;
278 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
280 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
281 if (ret == GST_STATE_CHANGE_FAILURE) {
287 gst_element_set_state (udpsrc0, GST_STATE_NULL);
288 gst_object_unref (udpsrc0);
292 goto no_udp_protocol;
295 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
297 /* check if port is even */
298 if ((tmp_rtp & 1) != 0) {
299 /* port not even, close and allocate another */
303 gst_element_set_state (udpsrc0, GST_STATE_NULL);
304 gst_object_unref (udpsrc0);
310 /* allocate port+1 for RTCP now */
311 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
313 goto no_udp_rtcp_protocol;
316 tmp_rtcp = tmp_rtp + 1;
317 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
319 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
320 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
321 if (ret == GST_STATE_CHANGE_FAILURE) {
326 gst_element_set_state (udpsrc0, GST_STATE_NULL);
327 gst_object_unref (udpsrc0);
329 gst_element_set_state (udpsrc1, GST_STATE_NULL);
330 gst_object_unref (udpsrc1);
335 /* all fine, do port check */
336 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
337 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
339 /* this should not happen... */
340 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
343 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
345 goto no_udp_protocol;
347 g_object_get (G_OBJECT (udpsrc0), "used-socket", &socket, NULL);
348 g_object_set (G_OBJECT (udpsink0), "socket", socket, NULL);
349 g_object_set (G_OBJECT (udpsink0), "close-socket", FALSE, NULL);
351 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
353 goto no_udp_protocol;
355 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
356 "send-duplicates")) {
357 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
358 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
361 ("old multiudpsink version found without send-duplicates property");
364 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
366 g_object_set (G_OBJECT (udpsink0), "buffer-size", stream->buffer_size,
369 GST_WARNING ("multiudpsink version found without buffer-size property");
372 g_object_get (G_OBJECT (udpsrc1), "used-socket", &socket, NULL);
373 g_object_set (G_OBJECT (udpsink1), "socket", socket, NULL);
374 g_object_set (G_OBJECT (udpsink1), "close-socket", FALSE, NULL);
375 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
376 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
377 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
378 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
379 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
380 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
382 /* we keep these elements, we will further configure them when the
383 * client told us to really use the UDP ports. */
384 stream->udpsrc[0] = udpsrc0;
385 stream->udpsrc[1] = udpsrc1;
386 stream->udpsink[0] = udpsink0;
387 stream->udpsink[1] = udpsink1;
388 stream->server_port.min = rtpport;
389 stream->server_port.max = rtcpport;
402 no_udp_rtcp_protocol:
413 gst_element_set_state (udpsrc0, GST_STATE_NULL);
414 gst_object_unref (udpsrc0);
417 gst_element_set_state (udpsrc1, GST_STATE_NULL);
418 gst_object_unref (udpsrc1);
421 gst_element_set_state (udpsink0, GST_STATE_NULL);
422 gst_object_unref (udpsink0);
425 gst_element_set_state (udpsink1, GST_STATE_NULL);
426 gst_object_unref (udpsink1);
432 /* executed from streaming thread */
434 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPStream * stream)
436 GstCaps *newcaps, *oldcaps;
438 newcaps = gst_pad_get_current_caps (pad);
440 GST_INFO ("stream %p received caps %p, %" GST_PTR_FORMAT, stream, newcaps,
443 g_mutex_lock (&stream->lock);
444 oldcaps = stream->caps;
445 stream->caps = newcaps;
446 g_mutex_unlock (&stream->lock);
449 gst_caps_unref (oldcaps);
453 dump_structure (const GstStructure * s)
457 sstr = gst_structure_to_string (s);
458 GST_INFO ("structure: %s", sstr);
462 static GstRTSPStreamTransport *
463 find_transport (GstRTSPStream * stream, const gchar * rtcp_from)
466 GstRTSPStreamTransport *result = NULL;
471 if (rtcp_from == NULL)
474 tmp = g_strrstr (rtcp_from, ":");
478 port = atoi (tmp + 1);
479 dest = g_strndup (rtcp_from, tmp - rtcp_from);
481 g_mutex_lock (&stream->lock);
482 GST_INFO ("finding %s:%d in %d transports", dest, port,
483 g_list_length (stream->transports));
485 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
486 GstRTSPStreamTransport *trans = walk->data;
489 min = trans->transport->client_port.min;
490 max = trans->transport->client_port.max;
492 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
498 g_mutex_unlock (&stream->lock);
505 static GstRTSPStreamTransport *
506 check_transport (GObject * source, GstRTSPStream * stream)
509 GstRTSPStreamTransport *trans;
511 /* see if we have a stream to match with the origin of the RTCP packet */
512 trans = g_object_get_qdata (source, ssrc_stream_map_key);
514 g_object_get (source, "stats", &stats, NULL);
516 const gchar *rtcp_from;
518 dump_structure (stats);
520 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
521 if ((trans = find_transport (stream, rtcp_from))) {
522 GST_INFO ("%p: found transport %p for source %p", stream, trans,
525 /* keep ref to the source */
526 trans->rtpsource = source;
528 g_object_set_qdata (source, ssrc_stream_map_key, trans);
530 gst_structure_free (stats);
539 on_new_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
541 GstRTSPStreamTransport *trans;
543 GST_INFO ("%p: new source %p", stream, source);
545 trans = check_transport (source, stream);
548 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
552 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPStream * stream)
554 GST_INFO ("%p: new SDES %p", stream, source);
558 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
560 GstRTSPStreamTransport *trans;
562 trans = check_transport (source, stream);
565 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
566 gst_rtsp_stream_transport_keep_alive (trans);
571 g_object_get (source, "stats", &stats, NULL);
573 dump_structure (stats);
574 gst_structure_free (stats);
581 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
583 GST_INFO ("%p: source %p bye", stream, source);
587 on_bye_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
589 GstRTSPStreamTransport *trans;
591 GST_INFO ("%p: source %p bye timeout", stream, source);
593 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
594 trans->rtpsource = NULL;
595 trans->timeout = TRUE;
600 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
602 GstRTSPStreamTransport *trans;
604 GST_INFO ("%p: source %p timeout", stream, source);
606 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
607 trans->rtpsource = NULL;
608 trans->timeout = TRUE;
613 handle_new_sample (GstAppSink * sink, gpointer user_data)
618 GstRTSPStream *stream;
620 sample = gst_app_sink_pull_sample (sink);
624 stream = (GstRTSPStream *) user_data;
625 buffer = gst_sample_get_buffer (sample);
627 g_mutex_lock (&stream->lock);
628 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
629 GstRTSPStreamTransport *tr = (GstRTSPStreamTransport *) walk->data;
631 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
632 gst_rtsp_stream_transport_send_rtp (tr, buffer);
634 gst_rtsp_stream_transport_send_rtcp (tr, buffer);
637 g_mutex_unlock (&stream->lock);
639 gst_sample_unref (sample);
644 static GstAppSinkCallbacks sink_cb = {
645 NULL, /* not interested in EOS */
646 NULL, /* not interested in preroll samples */
651 * gst_rtsp_stream_join_bin:
652 * @stream: a #GstRTSPStream
653 * @bin: a #GstBin to join
654 * @rtpbin: a rtpbin element in @bin
655 * @state: the target state of the new elements
657 * Join the #Gstbin @bin that contains the element @rtpbin.
659 * @stream will link to @rtpbin, which must be inside @bin. The elements
660 * added to @bin will be set to the state given in @state.
662 * Returns: %TRUE on success.
665 gst_rtsp_stream_join_bin (GstRTSPStream * stream, GstBin * bin,
666 GstElement * rtpbin, GstState state)
670 GstPad *pad, *teepad, *queuepad, *selpad;
671 GstPadLinkReturn ret;
673 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
674 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
675 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
677 g_mutex_lock (&stream->lock);
678 if (stream->is_joined)
681 /* create a session with the same index as the stream */
684 GST_INFO ("stream %p joining bin as session %d", stream, idx);
686 if (!alloc_ports (stream))
689 /* get a pad for sending RTP */
690 name = g_strdup_printf ("send_rtp_sink_%u", idx);
691 stream->send_rtp_sink = gst_element_get_request_pad (rtpbin, name);
693 /* link the RTP pad to the session manager, it should not really fail unless
694 * this is not really an RTP pad */
695 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
696 if (ret != GST_PAD_LINK_OK)
699 /* get pads from the RTP session element for sending and receiving
701 name = g_strdup_printf ("send_rtp_src_%u", idx);
702 stream->send_src[0] = gst_element_get_static_pad (rtpbin, name);
704 name = g_strdup_printf ("send_rtcp_src_%u", idx);
705 stream->send_src[1] = gst_element_get_request_pad (rtpbin, name);
707 name = g_strdup_printf ("recv_rtp_sink_%u", idx);
708 stream->recv_sink[0] = gst_element_get_request_pad (rtpbin, name);
710 name = g_strdup_printf ("recv_rtcp_sink_%u", idx);
711 stream->recv_sink[1] = gst_element_get_request_pad (rtpbin, name);
714 /* get the session */
715 g_signal_emit_by_name (rtpbin, "get-internal-session", idx, &stream->session);
717 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
719 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
721 g_signal_connect (stream->session, "on-ssrc-active",
722 (GCallback) on_ssrc_active, stream);
723 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
725 g_signal_connect (stream->session, "on-bye-timeout",
726 (GCallback) on_bye_timeout, stream);
727 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
730 for (i = 0; i < 2; i++) {
731 /* For the sender we create this bit of pipeline for both
732 * RTP and RTCP. Sync and preroll are enabled on udpsink so
733 * we need to add a queue before appsink to make the pipeline
734 * not block. For the TCP case, we want to pump data to the
735 * client as fast as possible anyway.
737 * .--------. .-----. .---------.
738 * | rtpbin | | tee | | udpsink |
739 * | send->sink src->sink |
740 * '--------' | | '---------'
741 * | | .---------. .---------.
742 * | | | queue | | appsink |
743 * | src->sink src->sink |
744 * '-----' '---------' '---------'
746 /* make tee for RTP/RTCP */
747 stream->tee[i] = gst_element_factory_make ("tee", NULL);
748 gst_bin_add (bin, stream->tee[i]);
750 /* and link to rtpbin send pad */
751 pad = gst_element_get_static_pad (stream->tee[i], "sink");
752 gst_pad_link (stream->send_src[i], pad);
753 gst_object_unref (pad);
756 gst_bin_add (bin, stream->udpsink[i]);
758 /* link tee to udpsink */
759 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
760 pad = gst_element_get_static_pad (stream->udpsink[i], "sink");
761 gst_pad_link (teepad, pad);
762 gst_object_unref (pad);
763 gst_object_unref (teepad);
766 stream->appqueue[i] = gst_element_factory_make ("queue", NULL);
767 gst_bin_add (bin, stream->appqueue[i]);
768 /* and link to tee */
769 teepad = gst_element_get_request_pad (stream->tee[i], "src_%u");
770 pad = gst_element_get_static_pad (stream->appqueue[i], "sink");
771 gst_pad_link (teepad, pad);
772 gst_object_unref (pad);
773 gst_object_unref (teepad);
776 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
777 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
778 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
779 gst_bin_add (bin, stream->appsink[i]);
780 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
781 &sink_cb, stream, NULL);
782 /* and link to queue */
783 queuepad = gst_element_get_static_pad (stream->appqueue[i], "src");
784 pad = gst_element_get_static_pad (stream->appsink[i], "sink");
785 gst_pad_link (queuepad, pad);
786 gst_object_unref (pad);
787 gst_object_unref (queuepad);
789 /* For the receiver we create this bit of pipeline for both
790 * RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
791 * and it is all funneled into the rtpbin receive pad.
793 * .--------. .--------. .--------.
794 * | udpsrc | | funnel | | rtpbin |
795 * | src->sink src->sink |
796 * '--------' | | '--------'
800 * '--------' '--------'
802 /* make funnel for the RTP/RTCP receivers */
803 stream->funnel[i] = gst_element_factory_make ("funnel", NULL);
804 gst_bin_add (bin, stream->funnel[i]);
806 pad = gst_element_get_static_pad (stream->funnel[i], "src");
807 gst_pad_link (pad, stream->recv_sink[i]);
808 gst_object_unref (pad);
811 gst_bin_add (bin, stream->udpsrc[i]);
812 /* and link to the funnel */
813 selpad = gst_element_get_request_pad (stream->funnel[i], "sink_%u");
814 pad = gst_element_get_static_pad (stream->udpsrc[i], "src");
815 gst_pad_link (pad, selpad);
816 gst_object_unref (pad);
817 gst_object_unref (selpad);
819 /* make and add appsrc */
820 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
821 gst_bin_add (bin, stream->appsrc[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->appsrc[i], "src");
825 gst_pad_link (pad, selpad);
826 gst_object_unref (pad);
827 gst_object_unref (selpad);
829 /* check if we need to set to a special state */
830 if (state != GST_STATE_NULL) {
831 gst_element_set_state (stream->udpsink[i], state);
832 gst_element_set_state (stream->appsink[i], state);
833 gst_element_set_state (stream->appqueue[i], state);
834 gst_element_set_state (stream->tee[i], state);
835 gst_element_set_state (stream->funnel[i], state);
836 gst_element_set_state (stream->appsrc[i], state);
838 /* we set and keep these to playing so that they don't cause NO_PREROLL return
840 gst_element_set_state (stream->udpsrc[i], GST_STATE_PLAYING);
841 gst_element_set_locked_state (stream->udpsrc[i], TRUE);
844 /* be notified of caps changes */
845 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
846 (GCallback) caps_notify, stream);
848 stream->is_joined = TRUE;
849 g_mutex_unlock (&stream->lock);
856 g_mutex_unlock (&stream->lock);
861 g_mutex_unlock (&stream->lock);
862 GST_WARNING ("failed to allocate ports %d", idx);
867 GST_WARNING ("failed to link stream %d", idx);
868 gst_object_unref (stream->send_rtp_sink);
869 stream->send_rtp_sink = NULL;
870 g_mutex_unlock (&stream->lock);
876 * gst_rtsp_stream_leave_bin:
877 * @stream: a #GstRTSPStream
879 * @rtpbin: a rtpbin #GstElement
881 * Remove the elements of @stream from @bin. @bin must be set
882 * to the NULL state before calling this.
884 * Return: %TRUE on success.
887 gst_rtsp_stream_leave_bin (GstRTSPStream * stream, GstBin * bin,
892 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
893 g_return_val_if_fail (GST_IS_BIN (bin), FALSE);
894 g_return_val_if_fail (GST_IS_ELEMENT (rtpbin), FALSE);
896 g_mutex_lock (&stream->lock);
897 if (!stream->is_joined)
900 /* all transports must be removed by now */
901 g_return_val_if_fail (stream->transports == NULL, FALSE);
903 GST_INFO ("stream %p leaving bin", stream);
905 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
906 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
907 gst_element_release_request_pad (rtpbin, stream->send_rtp_sink);
908 gst_object_unref (stream->send_rtp_sink);
909 stream->send_rtp_sink = NULL;
911 for (i = 0; i < 2; i++) {
912 /* and set udpsrc to NULL now before removing */
913 gst_element_set_locked_state (stream->udpsrc[i], FALSE);
914 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
916 /* removing them should also nicely release the request
917 * pads when they finalize */
918 gst_bin_remove (bin, stream->udpsrc[i]);
919 gst_bin_remove (bin, stream->udpsink[i]);
920 gst_bin_remove (bin, stream->appsrc[i]);
921 gst_bin_remove (bin, stream->appsink[i]);
922 gst_bin_remove (bin, stream->appqueue[i]);
923 gst_bin_remove (bin, stream->tee[i]);
924 gst_bin_remove (bin, stream->funnel[i]);
926 gst_element_release_request_pad (rtpbin, stream->recv_sink[i]);
927 gst_object_unref (stream->recv_sink[i]);
928 stream->recv_sink[i] = NULL;
930 stream->udpsrc[i] = NULL;
931 stream->udpsink[i] = NULL;
932 stream->appsrc[i] = NULL;
933 stream->appsink[i] = NULL;
934 stream->appqueue[i] = NULL;
935 stream->tee[i] = NULL;
936 stream->funnel[i] = NULL;
938 gst_object_unref (stream->send_src[0]);
939 stream->send_src[0] = NULL;
941 gst_element_release_request_pad (rtpbin, stream->send_src[1]);
942 gst_object_unref (stream->send_src[1]);
943 stream->send_src[1] = NULL;
945 g_object_unref (stream->session);
947 gst_caps_unref (stream->caps);
949 stream->is_joined = FALSE;
950 g_mutex_unlock (&stream->lock);
961 * gst_rtsp_stream_get_rtpinfo:
962 * @stream: a #GstRTSPStream
963 * @rtptime: result RTP timestamp
964 * @seq: result RTP seqnum
966 * Retrieve the current rtptime and seq. This is used to
967 * construct a RTPInfo reply header.
969 * Returns: %TRUE when rtptime and seq could be determined.
972 gst_rtsp_stream_get_rtpinfo (GstRTSPStream * stream,
973 guint * rtptime, guint * seq)
975 GObjectClass *payobjclass;
977 payobjclass = G_OBJECT_GET_CLASS (stream->payloader);
979 if (!g_object_class_find_property (payobjclass, "seqnum") ||
980 !g_object_class_find_property (payobjclass, "timestamp"))
983 g_object_get (stream->payloader, "seqnum", seq, "timestamp", rtptime, NULL);
989 * gst_rtsp_stream_recv_rtp:
990 * @stream: a #GstRTSPStream
991 * @buffer: (transfer full): a #GstBuffer
993 * Handle an RTP buffer for the stream. This method is usually called when a
994 * message has been received from a client using the TCP transport.
996 * This function takes ownership of @buffer.
998 * Returns: a GstFlowReturn.
1001 gst_rtsp_stream_recv_rtp (GstRTSPStream * stream, GstBuffer * buffer)
1004 GstElement *element;
1006 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1007 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1008 g_return_val_if_fail (stream->is_joined, FALSE);
1010 g_mutex_lock (&stream->lock);
1011 element = gst_object_ref (stream->appsrc[0]);
1012 g_mutex_unlock (&stream->lock);
1014 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1016 gst_object_unref (element);
1022 * gst_rtsp_stream_recv_rtcp:
1023 * @stream: a #GstRTSPStream
1024 * @buffer: (transfer full): a #GstBuffer
1026 * Handle an RTCP buffer for the stream. This method is usually called when a
1027 * message has been received from a client using the TCP transport.
1029 * This function takes ownership of @buffer.
1031 * Returns: a GstFlowReturn.
1034 gst_rtsp_stream_recv_rtcp (GstRTSPStream * stream, GstBuffer * buffer)
1037 GstElement *element;
1039 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), GST_FLOW_ERROR);
1040 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1041 g_return_val_if_fail (stream->is_joined, FALSE);
1043 g_mutex_lock (&stream->lock);
1044 element = gst_object_ref (stream->appsrc[1]);
1045 g_mutex_unlock (&stream->lock);
1047 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (element), buffer);
1049 gst_object_unref (element);
1054 /* must be called with lock */
1056 update_transport (GstRTSPStream * stream, GstRTSPStreamTransport * trans,
1059 GstRTSPTransport *tr;
1064 tr = trans->transport;
1066 switch (tr->lower_transport) {
1067 case GST_RTSP_LOWER_TRANS_UDP:
1068 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1074 dest = tr->destination;
1075 if (tr->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1080 min = tr->client_port.min;
1081 max = tr->client_port.max;
1084 if (add && !trans->active) {
1085 GST_INFO ("adding %s:%d-%d", dest, min, max);
1086 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1087 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1089 GST_INFO ("setting ttl-mc %d", ttl);
1090 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl-mc", ttl, NULL);
1091 g_object_set (G_OBJECT (stream->udpsink[1]), "ttl-mc", ttl, NULL);
1093 stream->transports = g_list_prepend (stream->transports, trans);
1094 trans->active = TRUE;
1096 } else if (trans->active) {
1097 GST_INFO ("removing %s:%d-%d", dest, min, max);
1098 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1099 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1100 stream->transports = g_list_remove (stream->transports, trans);
1101 trans->active = FALSE;
1106 case GST_RTSP_LOWER_TRANS_TCP:
1107 if (add && !trans->active) {
1108 GST_INFO ("adding TCP %s", tr->destination);
1109 stream->transports = g_list_prepend (stream->transports, trans);
1110 trans->active = TRUE;
1112 } else if (trans->active) {
1113 GST_INFO ("removing TCP %s", tr->destination);
1114 stream->transports = g_list_remove (stream->transports, trans);
1115 trans->active = FALSE;
1120 GST_INFO ("Unknown transport %d", tr->lower_transport);
1128 * gst_rtsp_stream_add_transport:
1129 * @stream: a #GstRTSPStream
1130 * @trans: a #GstRTSPStreamTransport
1132 * Add the transport in @trans to @stream. The media of @stream will
1133 * then also be send to the values configured in @trans.
1135 * @stream must be joined to a bin.
1137 * @trans must contain a valid #GstRTSPTransport.
1139 * Returns: %TRUE if @trans was added
1142 gst_rtsp_stream_add_transport (GstRTSPStream * stream,
1143 GstRTSPStreamTransport * trans)
1147 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1148 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1149 g_return_val_if_fail (stream->is_joined, FALSE);
1150 g_return_val_if_fail (trans->transport != NULL, FALSE);
1152 g_mutex_lock (&stream->lock);
1153 res = update_transport (stream, trans, TRUE);
1154 g_mutex_unlock (&stream->lock);
1160 * gst_rtsp_stream_remove_transport:
1161 * @stream: a #GstRTSPStream
1162 * @trans: a #GstRTSPStreamTransport
1164 * Remove the transport in @trans from @stream. The media of @stream will
1165 * not be sent to the values configured in @trans.
1167 * @stream must be joined to a bin.
1169 * @trans must contain a valid #GstRTSPTransport.
1171 * Returns: %TRUE if @trans was removed
1174 gst_rtsp_stream_remove_transport (GstRTSPStream * stream,
1175 GstRTSPStreamTransport * trans)
1179 g_return_val_if_fail (GST_IS_RTSP_STREAM (stream), FALSE);
1180 g_return_val_if_fail (GST_IS_RTSP_STREAM_TRANSPORT (trans), FALSE);
1181 g_return_val_if_fail (stream->is_joined, FALSE);
1182 g_return_val_if_fail (trans->transport != NULL, FALSE);
1184 g_mutex_lock (&stream->lock);
1185 res = update_transport (stream, trans, FALSE);
1186 g_mutex_unlock (&stream->lock);