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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
23 #include <gst/app/gstappsrc.h>
24 #include <gst/app/gstappsink.h>
26 #include "rtsp-media.h"
28 #define DEFAULT_SHARED FALSE
29 #define DEFAULT_REUSABLE FALSE
30 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
31 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
33 /* define to dump received RTCP packets */
51 GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
52 #define GST_CAT_DEFAULT rtsp_media_debug
54 static GQuark ssrc_stream_map_key;
56 static void gst_rtsp_media_get_property (GObject * object, guint propid,
57 GValue * value, GParamSpec * pspec);
58 static void gst_rtsp_media_set_property (GObject * object, guint propid,
59 const GValue * value, GParamSpec * pspec);
60 static void gst_rtsp_media_finalize (GObject * obj);
62 static gpointer do_loop (GstRTSPMediaClass * klass);
63 static gboolean default_handle_message (GstRTSPMedia * media,
64 GstMessage * message);
65 static gboolean default_unprepare (GstRTSPMedia * media);
66 static void unlock_streams (GstRTSPMedia * media);
68 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
70 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
73 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
75 GObjectClass *gobject_class;
78 gobject_class = G_OBJECT_CLASS (klass);
80 gobject_class->get_property = gst_rtsp_media_get_property;
81 gobject_class->set_property = gst_rtsp_media_set_property;
82 gobject_class->finalize = gst_rtsp_media_finalize;
84 g_object_class_install_property (gobject_class, PROP_SHARED,
85 g_param_spec_boolean ("shared", "Shared",
86 "If this media pipeline can be shared", DEFAULT_SHARED,
87 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
89 g_object_class_install_property (gobject_class, PROP_REUSABLE,
90 g_param_spec_boolean ("reusable", "Reusable",
91 "If this media pipeline can be reused after an unprepare",
92 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
94 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
95 g_param_spec_flags ("protocols", "Protocols",
96 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
97 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
99 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
100 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
101 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
102 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
104 klass->context = g_main_context_new ();
105 klass->loop = g_main_loop_new (klass->context, TRUE);
107 klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
109 g_critical ("could not start bus thread: %s", error->message);
111 klass->handle_message = default_handle_message;
112 klass->unprepare = default_unprepare;
114 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
118 gst_rtsp_media_init (GstRTSPMedia * media)
120 media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
121 media->lock = g_mutex_new ();
122 media->cond = g_cond_new ();
124 media->shared = DEFAULT_SHARED;
125 media->reusable = DEFAULT_REUSABLE;
126 media->protocols = DEFAULT_PROTOCOLS;
130 gst_rtsp_media_stream_free (GstRTSPMediaStream * stream)
133 g_object_unref (stream->session);
136 gst_caps_unref (stream->caps);
138 if (stream->send_rtp_sink)
139 gst_object_unref (stream->send_rtp_sink);
140 if (stream->send_rtp_src)
141 gst_object_unref (stream->send_rtp_src);
142 if (stream->send_rtcp_src)
143 gst_object_unref (stream->send_rtcp_src);
144 if (stream->recv_rtcp_sink)
145 gst_object_unref (stream->recv_rtcp_sink);
146 if (stream->recv_rtp_sink)
147 gst_object_unref (stream->recv_rtp_sink);
149 g_list_free (stream->transports);
155 gst_rtsp_media_finalize (GObject * obj)
160 media = GST_RTSP_MEDIA (obj);
162 GST_INFO ("finalize media %p", media);
164 if (media->pipeline) {
165 unlock_streams (media);
166 gst_element_set_state (media->pipeline, GST_STATE_NULL);
167 gst_object_unref (media->pipeline);
170 for (i = 0; i < media->streams->len; i++) {
171 GstRTSPMediaStream *stream;
173 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
175 gst_rtsp_media_stream_free (stream);
177 g_array_free (media->streams, TRUE);
179 g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL);
180 g_list_free (media->dynamic);
183 g_source_destroy (media->source);
184 g_source_unref (media->source);
186 g_mutex_free (media->lock);
187 g_cond_free (media->cond);
189 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
193 gst_rtsp_media_get_property (GObject * object, guint propid,
194 GValue * value, GParamSpec * pspec)
196 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
200 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
203 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
206 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
209 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
214 gst_rtsp_media_set_property (GObject * object, guint propid,
215 const GValue * value, GParamSpec * pspec)
217 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
221 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
224 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
227 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
230 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
235 do_loop (GstRTSPMediaClass * klass)
237 GST_INFO ("enter mainloop");
238 g_main_loop_run (klass->loop);
239 GST_INFO ("exit mainloop");
245 collect_media_stats (GstRTSPMedia * media)
248 gint64 position, duration;
250 media->range.unit = GST_RTSP_RANGE_NPT;
252 if (media->is_live) {
253 media->range.min.type = GST_RTSP_TIME_NOW;
254 media->range.min.seconds = -1;
255 media->range.max.type = GST_RTSP_TIME_END;
256 media->range.max.seconds = -1;
258 /* get the position */
259 format = GST_FORMAT_TIME;
260 if (!gst_element_query_position (media->pipeline, &format, &position)) {
261 GST_INFO ("position query failed");
265 /* get the duration */
266 format = GST_FORMAT_TIME;
267 if (!gst_element_query_duration (media->pipeline, &format, &duration)) {
268 GST_INFO ("duration query failed");
272 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
273 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
275 if (position == -1) {
276 media->range.min.type = GST_RTSP_TIME_NOW;
277 media->range.min.seconds = -1;
279 media->range.min.type = GST_RTSP_TIME_SECONDS;
280 media->range.min.seconds = ((gdouble) position) / GST_SECOND;
282 if (duration == -1) {
283 media->range.max.type = GST_RTSP_TIME_END;
284 media->range.max.seconds = -1;
286 media->range.max.type = GST_RTSP_TIME_SECONDS;
287 media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
293 * gst_rtsp_media_new:
295 * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
296 * element to produde RTP data for one or more related (audio/video/..)
299 * Returns: a new #GstRTSPMedia object.
302 gst_rtsp_media_new (void)
304 GstRTSPMedia *result;
306 result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
312 * gst_rtsp_media_set_shared:
313 * @media: a #GstRTSPMedia
314 * @shared: the new value
316 * Set or unset if the pipeline for @media can be shared will multiple clients.
317 * When @shared is %TRUE, client requests for this media will share the media
321 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
323 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
325 media->shared = shared;
329 * gst_rtsp_media_is_shared:
330 * @media: a #GstRTSPMedia
332 * Check if the pipeline for @media can be shared between multiple clients.
334 * Returns: %TRUE if the media can be shared between clients.
337 gst_rtsp_media_is_shared (GstRTSPMedia * media)
339 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
341 return media->shared;
345 * gst_rtsp_media_set_reusable:
346 * @media: a #GstRTSPMedia
347 * @reusable: the new value
349 * Set or unset if the pipeline for @media can be reused after the pipeline has
353 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
355 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
357 media->reusable = reusable;
361 * gst_rtsp_media_is_reusable:
362 * @media: a #GstRTSPMedia
364 * Check if the pipeline for @media can be reused after an unprepare.
366 * Returns: %TRUE if the media can be reused
369 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
371 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
373 return media->reusable;
377 * gst_rtsp_media_set_protocols:
378 * @media: a #GstRTSPMedia
379 * @protocols: the new flags
381 * Configure the allowed lower transport for @media.
384 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
386 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
388 media->protocols = protocols;
392 * gst_rtsp_media_get_protocols:
393 * @media: a #GstRTSPMedia
395 * Get the allowed protocols of @media.
397 * Returns: a #GstRTSPLowerTrans
400 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
402 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_LOWER_TRANS_UNKNOWN);
404 return media->protocols;
408 * gst_rtsp_media_n_streams:
409 * @media: a #GstRTSPMedia
411 * Get the number of streams in this media.
413 * Returns: The number of streams.
416 gst_rtsp_media_n_streams (GstRTSPMedia * media)
418 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
420 return media->streams->len;
424 * gst_rtsp_media_get_stream:
425 * @media: a #GstRTSPMedia
426 * @idx: the stream index
428 * Retrieve the stream with index @idx from @media.
430 * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
431 * that index did not exist.
434 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
436 GstRTSPMediaStream *res;
438 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
440 if (idx < media->streams->len)
441 res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
449 * gst_rtsp_media_seek:
450 * @stream: a #GstRTSPMediaStream
451 * @range: a #GstRTSPTimeRange
453 * Seek the pipeline to @range.
455 * Returns: %TRUE on success.
458 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
463 GstSeekType start_type, stop_type;
465 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
466 g_return_val_if_fail (range != NULL, FALSE);
468 if (range->unit != GST_RTSP_RANGE_NPT)
471 /* depends on the current playing state of the pipeline. We might need to
472 * queue this until we get EOS. */
473 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
475 start_type = stop_type = GST_SEEK_TYPE_NONE;
477 switch (range->min.type) {
478 case GST_RTSP_TIME_NOW:
481 case GST_RTSP_TIME_SECONDS:
482 /* only seek when something changed */
483 if (media->range.min.seconds == range->min.seconds) {
486 start = range->min.seconds * GST_SECOND;
487 start_type = GST_SEEK_TYPE_SET;
490 case GST_RTSP_TIME_END:
494 switch (range->max.type) {
495 case GST_RTSP_TIME_SECONDS:
496 /* only seek when something changed */
497 if (media->range.max.seconds == range->max.seconds) {
500 stop = range->max.seconds * GST_SECOND;
501 stop_type = GST_SEEK_TYPE_SET;
504 case GST_RTSP_TIME_END:
506 stop_type = GST_SEEK_TYPE_SET;
508 case GST_RTSP_TIME_NOW:
513 if (start != -1 || stop != -1) {
514 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
515 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
517 res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
518 flags, start_type, start, stop_type, stop);
520 /* and block for the seek to complete */
521 GST_INFO ("done seeking %d", res);
522 gst_element_get_state (media->pipeline, NULL, NULL, -1);
523 GST_INFO ("prerolled again");
525 collect_media_stats (media);
527 GST_INFO ("no seek needed");
536 GST_WARNING ("seek unit %d not supported", range->unit);
541 GST_WARNING ("weird range type %d not supported", range->min.type);
547 * gst_rtsp_media_stream_rtp:
548 * @stream: a #GstRTSPMediaStream
549 * @buffer: a #GstBuffer
551 * Handle an RTP buffer for the stream. This method is usually called when a
552 * message has been received from a client using the TCP transport.
554 * This function takes ownership of @buffer.
556 * Returns: a GstFlowReturn.
559 gst_rtsp_media_stream_rtp (GstRTSPMediaStream * stream, GstBuffer * buffer)
563 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
569 * gst_rtsp_media_stream_rtcp:
570 * @stream: a #GstRTSPMediaStream
571 * @buffer: a #GstBuffer
573 * Handle an RTCP buffer for the stream. This method is usually called when a
574 * message has been received from a client using the TCP transport.
576 * This function takes ownership of @buffer.
578 * Returns: a GstFlowReturn.
581 gst_rtsp_media_stream_rtcp (GstRTSPMediaStream * stream, GstBuffer * buffer)
585 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
590 /* Allocate the udp ports and sockets */
592 alloc_udp_ports (GstRTSPMedia * media, GstRTSPMediaStream * stream)
594 GstStateChangeReturn ret;
595 GstElement *udpsrc0, *udpsrc1;
596 GstElement *udpsink0, *udpsink1;
597 gint tmp_rtp, tmp_rtcp;
599 gint rtpport, rtcpport, sockfd;
608 /* Start with random port */
612 host = "udp://[::0]";
614 host = "udp://0.0.0.0";
616 /* try to allocate 2 UDP ports, the RTP port should be an even
617 * number and the RTCP port should be the next (uneven) port */
619 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
621 goto no_udp_protocol;
622 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
624 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
625 if (ret == GST_STATE_CHANGE_FAILURE) {
631 gst_element_set_state (udpsrc0, GST_STATE_NULL);
632 gst_object_unref (udpsrc0);
636 goto no_udp_protocol;
639 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
641 /* check if port is even */
642 if ((tmp_rtp & 1) != 0) {
643 /* port not even, close and allocate another */
647 gst_element_set_state (udpsrc0, GST_STATE_NULL);
648 gst_object_unref (udpsrc0);
654 /* allocate port+1 for RTCP now */
655 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
657 goto no_udp_rtcp_protocol;
660 tmp_rtcp = tmp_rtp + 1;
661 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
663 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
664 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
665 if (ret == GST_STATE_CHANGE_FAILURE) {
670 gst_element_set_state (udpsrc0, GST_STATE_NULL);
671 gst_object_unref (udpsrc0);
673 gst_element_set_state (udpsrc1, GST_STATE_NULL);
674 gst_object_unref (udpsrc1);
680 /* all fine, do port check */
681 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
682 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
684 /* this should not happen... */
685 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
688 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
690 goto no_udp_protocol;
692 g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
693 g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
694 g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);
696 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
698 goto no_udp_protocol;
700 g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
701 g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
702 g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
703 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
704 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
706 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
707 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
708 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
709 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
711 /* we keep these elements, we configure all in configure_transport when the
712 * server told us to really use the UDP ports. */
713 stream->udpsrc[0] = udpsrc0;
714 stream->udpsrc[1] = udpsrc1;
715 stream->udpsink[0] = udpsink0;
716 stream->udpsink[1] = udpsink1;
717 stream->server_port.min = rtpport;
718 stream->server_port.max = rtcpport;
731 no_udp_rtcp_protocol:
742 gst_element_set_state (udpsrc0, GST_STATE_NULL);
743 gst_object_unref (udpsrc0);
746 gst_element_set_state (udpsrc1, GST_STATE_NULL);
747 gst_object_unref (udpsrc1);
750 gst_element_set_state (udpsink0, GST_STATE_NULL);
751 gst_object_unref (udpsink0);
754 gst_element_set_state (udpsink1, GST_STATE_NULL);
755 gst_object_unref (udpsink1);
762 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
765 GstCaps *newcaps, *oldcaps;
767 if ((newcaps = GST_PAD_CAPS (pad)))
768 gst_caps_ref (newcaps);
770 oldcaps = stream->caps;
771 stream->caps = newcaps;
774 gst_caps_unref (oldcaps);
776 capsstr = gst_caps_to_string (newcaps);
777 GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr);
782 dump_structure (const GstStructure * s)
786 sstr = gst_structure_to_string (s);
787 GST_INFO ("structure: %s", sstr);
791 static GstRTSPMediaTrans *
792 find_transport (GstRTSPMediaStream * stream, const gchar * rtcp_from)
795 GstRTSPMediaTrans *result = NULL;
800 if (rtcp_from == NULL)
803 tmp = g_strrstr (rtcp_from, ":");
807 port = atoi (tmp + 1);
808 dest = g_strndup (rtcp_from, tmp - rtcp_from);
810 GST_INFO ("finding %s:%d", dest, port);
812 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
813 GstRTSPMediaTrans *trans = walk->data;
816 min = trans->transport->client_port.min;
817 max = trans->transport->client_port.max;
819 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
831 on_new_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
834 GstRTSPMediaTrans *trans;
836 GST_INFO ("%p: new source %p", stream, source);
838 /* see if we have a stream to match with the origin of the RTCP packet */
839 trans = g_object_get_qdata (source, ssrc_stream_map_key);
841 g_object_get (source, "stats", &stats, NULL);
843 const gchar *rtcp_from;
845 dump_structure (stats);
847 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
848 if ((trans = find_transport (stream, rtcp_from))) {
849 GST_INFO ("%p: found transport %p for source %p", stream, trans,
852 /* keep ref to the source */
853 trans->rtpsource = source;
855 g_object_set_qdata (source, ssrc_stream_map_key, trans);
857 gst_structure_free (stats);
860 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
865 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPMediaStream * stream)
867 GST_INFO ("%p: new SDES %p", stream, source);
871 on_ssrc_active (GObject * session, GObject * source,
872 GstRTSPMediaStream * stream)
874 GstRTSPMediaTrans *trans;
876 trans = g_object_get_qdata (source, ssrc_stream_map_key);
878 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
880 if (trans && trans->keep_alive)
881 trans->keep_alive (trans->ka_user_data);
886 g_object_get (source, "stats", &stats, NULL);
888 dump_structure (stats);
889 gst_structure_free (stats);
896 on_bye_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
898 GST_INFO ("%p: source %p bye", stream, source);
902 on_bye_timeout (GObject * session, GObject * source,
903 GstRTSPMediaStream * stream)
905 GstRTSPMediaTrans *trans;
907 GST_INFO ("%p: source %p bye timeout", stream, source);
909 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
910 trans->rtpsource = NULL;
911 trans->timeout = TRUE;
916 on_timeout (GObject * session, GObject * source, GstRTSPMediaStream * stream)
918 GstRTSPMediaTrans *trans;
920 GST_INFO ("%p: source %p timeout", stream, source);
922 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
923 trans->rtpsource = NULL;
924 trans->timeout = TRUE;
929 handle_new_buffer (GstAppSink * sink, gpointer user_data)
933 GstRTSPMediaStream *stream;
935 buffer = gst_app_sink_pull_buffer (sink);
939 stream = (GstRTSPMediaStream *) user_data;
941 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
942 GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
944 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
946 tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
949 tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
952 gst_buffer_unref (buffer);
957 static GstAppSinkCallbacks sink_cb = {
958 NULL, /* not interested in EOS */
959 NULL, /* not interested in preroll buffers */
963 /* prepare the pipeline objects to handle @stream in @media */
965 setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
968 GstPad *pad, *teepad, *selpad;
969 GstPadLinkReturn ret;
972 /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2
973 * for sending RTP/RTCP. The sender and receiver ports are shared between the
975 if (!alloc_udp_ports (media, stream))
978 /* add the ports to the pipeline */
979 for (i = 0; i < 2; i++) {
980 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]);
981 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]);
984 /* create elements for the TCP transfer */
985 for (i = 0; i < 2; i++) {
986 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
987 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
988 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
989 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
990 g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL);
991 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
992 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
993 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
994 &sink_cb, stream, NULL);
997 /* hook up the stream to the RTP session elements. */
998 name = g_strdup_printf ("send_rtp_sink_%d", idx);
999 stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1001 name = g_strdup_printf ("send_rtp_src_%d", idx);
1002 stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
1004 name = g_strdup_printf ("send_rtcp_src_%d", idx);
1005 stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
1007 name = g_strdup_printf ("recv_rtcp_sink_%d", idx);
1008 stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
1010 name = g_strdup_printf ("recv_rtp_sink_%d", idx);
1011 stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1014 /* get the session */
1015 g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
1018 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1020 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1022 g_signal_connect (stream->session, "on-ssrc-active",
1023 (GCallback) on_ssrc_active, stream);
1024 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1026 g_signal_connect (stream->session, "on-bye-timeout",
1027 (GCallback) on_bye_timeout, stream);
1028 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
1031 /* link the RTP pad to the session manager */
1032 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
1033 if (ret != GST_PAD_LINK_OK)
1036 /* make tee for RTP and link to stream */
1037 stream->tee[0] = gst_element_factory_make ("tee", NULL);
1038 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]);
1040 pad = gst_element_get_static_pad (stream->tee[0], "sink");
1041 gst_pad_link (stream->send_rtp_src, pad);
1042 gst_object_unref (pad);
1044 /* link RTP sink, we're pretty sure this will work. */
1045 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1046 pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
1047 gst_pad_link (teepad, pad);
1048 gst_object_unref (pad);
1049 gst_object_unref (teepad);
1051 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1052 pad = gst_element_get_static_pad (stream->appsink[0], "sink");
1053 gst_pad_link (teepad, pad);
1054 gst_object_unref (pad);
1055 gst_object_unref (teepad);
1057 /* make tee for RTCP */
1058 stream->tee[1] = gst_element_factory_make ("tee", NULL);
1059 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]);
1061 pad = gst_element_get_static_pad (stream->tee[1], "sink");
1062 gst_pad_link (stream->send_rtcp_src, pad);
1063 gst_object_unref (pad);
1065 /* link RTCP elements */
1066 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1067 pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
1068 gst_pad_link (teepad, pad);
1069 gst_object_unref (pad);
1070 gst_object_unref (teepad);
1072 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1073 pad = gst_element_get_static_pad (stream->appsink[1], "sink");
1074 gst_pad_link (teepad, pad);
1075 gst_object_unref (pad);
1076 gst_object_unref (teepad);
1078 /* make selector for the RTP receivers */
1079 stream->selector[0] = gst_element_factory_make ("input-selector", NULL);
1080 g_object_set (stream->selector[0], "select-all", TRUE, NULL);
1081 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
1083 pad = gst_element_get_static_pad (stream->selector[0], "src");
1084 gst_pad_link (pad, stream->recv_rtp_sink);
1085 gst_object_unref (pad);
1087 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1088 pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
1089 gst_pad_link (pad, selpad);
1090 gst_object_unref (pad);
1091 gst_object_unref (selpad);
1093 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1094 pad = gst_element_get_static_pad (stream->appsrc[0], "src");
1095 gst_pad_link (pad, selpad);
1096 gst_object_unref (pad);
1097 gst_object_unref (selpad);
1099 /* make selector for the RTCP receivers */
1100 stream->selector[1] = gst_element_factory_make ("input-selector", NULL);
1101 g_object_set (stream->selector[1], "select-all", TRUE, NULL);
1102 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
1104 pad = gst_element_get_static_pad (stream->selector[1], "src");
1105 gst_pad_link (pad, stream->recv_rtcp_sink);
1106 gst_object_unref (pad);
1108 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1109 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
1110 gst_pad_link (pad, selpad);
1111 gst_object_unref (pad);
1112 gst_object_unref (selpad);
1114 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1115 pad = gst_element_get_static_pad (stream->appsrc[1], "src");
1116 gst_pad_link (pad, selpad);
1117 gst_object_unref (pad);
1118 gst_object_unref (selpad);
1120 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1122 gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
1123 gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
1124 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
1125 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
1127 /* be notified of caps changes */
1128 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
1129 (GCallback) caps_notify, stream);
1131 stream->prepared = TRUE;
1138 GST_WARNING ("failed to link stream %d", idx);
1144 unlock_streams (GstRTSPMedia * media)
1148 /* unlock the udp src elements */
1149 n_streams = gst_rtsp_media_n_streams (media);
1150 for (i = 0; i < n_streams; i++) {
1151 GstRTSPMediaStream *stream;
1153 stream = gst_rtsp_media_get_stream (media, i);
1155 gst_element_set_locked_state (stream->udpsrc[0], FALSE);
1156 gst_element_set_locked_state (stream->udpsrc[1], FALSE);
1161 gst_rtsp_media_set_status (GstRTSPMedia *media, GstRTSPMediaStatus status)
1163 g_mutex_lock (media->lock);
1164 /* never overwrite the error status */
1165 if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
1166 media->status = status;
1167 GST_DEBUG ("setting new status to %d", status);
1168 g_cond_broadcast (media->cond);
1169 g_mutex_unlock (media->lock);
1172 static GstRTSPMediaStatus
1173 gst_rtsp_media_get_status (GstRTSPMedia *media)
1175 GstRTSPMediaStatus result;
1178 g_mutex_lock (media->lock);
1179 g_get_current_time (&timeout);
1180 g_time_val_add (&timeout, 20 * G_USEC_PER_SEC);
1181 /* while we are preparing, wait */
1182 while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1183 GST_DEBUG ("waiting for status change");
1184 if (!g_cond_timed_wait (media->cond, media->lock, &timeout)) {
1185 GST_DEBUG ("timeout, assuming error status");
1186 media->status = GST_RTSP_MEDIA_STATUS_ERROR;
1189 /* could be success or error */
1190 result = media->status;
1191 GST_DEBUG ("got status %d", result);
1192 g_mutex_unlock (media->lock);
1198 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1200 GstMessageType type;
1202 type = GST_MESSAGE_TYPE (message);
1205 case GST_MESSAGE_STATE_CHANGED:
1207 case GST_MESSAGE_BUFFERING:
1211 gst_message_parse_buffering (message, &percent);
1213 /* no state management needed for live pipelines */
1217 if (percent == 100) {
1218 /* a 100% message means buffering is done */
1219 media->buffering = FALSE;
1220 /* if the desired state is playing, go back */
1221 if (media->target_state == GST_STATE_PLAYING) {
1222 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1223 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1225 GST_INFO ("Buffering done");
1228 /* buffering busy */
1229 if (media->buffering == FALSE) {
1230 if (media->target_state == GST_STATE_PLAYING) {
1231 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1232 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1233 gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1235 GST_INFO ("Buffering ...");
1238 media->buffering = TRUE;
1242 case GST_MESSAGE_LATENCY:
1244 gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
1247 case GST_MESSAGE_ERROR:
1252 gst_message_parse_error (message, &gerror, &debug);
1253 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1254 g_error_free (gerror);
1257 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1260 case GST_MESSAGE_WARNING:
1265 gst_message_parse_warning (message, &gerror, &debug);
1266 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1267 g_error_free (gerror);
1271 case GST_MESSAGE_ELEMENT:
1273 case GST_MESSAGE_STREAM_STATUS:
1275 case GST_MESSAGE_ASYNC_DONE:
1276 GST_INFO ("%p: got ASYNC_DONE", media);
1277 collect_media_stats (media);
1279 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1282 GST_INFO ("%p: got message type %s", media,
1283 gst_message_type_get_name (type));
1290 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1292 GstRTSPMediaClass *klass;
1295 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1297 if (klass->handle_message)
1298 ret = klass->handle_message (media, message);
1306 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1308 GstRTSPMediaStream *stream;
1312 i = media->streams->len + 1;
1314 GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
1316 stream = g_new0 (GstRTSPMediaStream, 1);
1317 stream->payloader = element;
1319 name = g_strdup_printf ("dynpay%d", i);
1321 /* ghost the pad of the payloader to the element */
1322 stream->srcpad = gst_ghost_pad_new (name, pad);
1323 gst_pad_set_active (stream->srcpad, TRUE);
1324 gst_element_add_pad (media->element, stream->srcpad);
1327 /* add stream now */
1328 g_array_append_val (media->streams, stream);
1330 setup_stream (stream, i, media);
1332 for (i = 0; i < 2; i++) {
1333 gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
1334 gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
1335 gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
1336 gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
1337 gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
1342 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1344 GST_INFO ("no more pads");
1345 if (media->fakesink) {
1346 gst_object_ref (media->fakesink);
1347 gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
1348 gst_element_set_state (media->fakesink, GST_STATE_NULL);
1349 gst_object_unref (media->fakesink);
1350 media->fakesink = NULL;
1351 GST_INFO ("removed fakesink");
1356 * gst_rtsp_media_prepare:
1357 * @obj: a #GstRTSPMedia
1359 * Prepare @media for streaming. This function will create the pipeline and
1360 * other objects to manage the streaming.
1362 * It will preroll the pipeline and collect vital information about the streams
1363 * such as the duration.
1365 * Returns: %TRUE on success.
1368 gst_rtsp_media_prepare (GstRTSPMedia * media)
1370 GstStateChangeReturn ret;
1371 GstRTSPMediaStatus status;
1373 GstRTSPMediaClass *klass;
1377 if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1380 if (!media->reusable && media->reused)
1383 GST_INFO ("preparing media %p", media);
1385 /* reset some variables */
1386 media->is_live = FALSE;
1387 media->buffering = FALSE;
1388 /* we're preparing now */
1389 media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1391 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
1393 /* add the pipeline bus to our custom mainloop */
1394 media->source = gst_bus_create_watch (bus);
1395 gst_object_unref (bus);
1397 g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
1399 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1400 media->id = g_source_attach (media->source, klass->context);
1402 media->rtpbin = gst_element_factory_make ("gstrtpbin", NULL);
1404 /* add stuff to the bin */
1405 gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
1407 /* link streams we already have, other streams might appear when we have
1408 * dynamic elements */
1409 n_streams = gst_rtsp_media_n_streams (media);
1410 for (i = 0; i < n_streams; i++) {
1411 GstRTSPMediaStream *stream;
1413 stream = gst_rtsp_media_get_stream (media, i);
1415 setup_stream (stream, i, media);
1418 for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
1419 GstElement *elem = walk->data;
1421 g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
1422 g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
1424 /* we add a fakesink here in order to make the state change async. We remove
1425 * the fakesink again in the no-more-pads callback. */
1426 media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1427 gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
1430 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1431 /* first go to PAUSED */
1432 ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1433 media->target_state = GST_STATE_PAUSED;
1436 case GST_STATE_CHANGE_SUCCESS:
1437 GST_INFO ("SUCCESS state change for media %p", media);
1439 case GST_STATE_CHANGE_ASYNC:
1440 GST_INFO ("ASYNC state change for media %p", media);
1442 case GST_STATE_CHANGE_NO_PREROLL:
1443 /* we need to go to PLAYING */
1444 GST_INFO ("NO_PREROLL state change: live media %p", media);
1445 media->is_live = TRUE;
1446 ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1447 if (ret == GST_STATE_CHANGE_FAILURE)
1450 case GST_STATE_CHANGE_FAILURE:
1454 /* now wait for all pads to be prerolled */
1455 status = gst_rtsp_media_get_status (media);
1456 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1459 GST_INFO ("object %p is prerolled", media);
1471 GST_WARNING ("can not reuse media %p", media);
1476 GST_WARNING ("failed to preroll pipeline");
1477 unlock_streams (media);
1478 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1479 gst_rtsp_media_unprepare (media);
1485 * gst_rtsp_media_unprepare:
1486 * @obj: a #GstRTSPMedia
1488 * Unprepare @media. After this call, the media should be prepared again before
1489 * it can be used again. If the media is set to be non-reusable, a new instance
1492 * Returns: %TRUE on success.
1495 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1497 GstRTSPMediaClass *klass;
1500 if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1503 GST_INFO ("unprepare media %p", media);
1504 media->target_state = GST_STATE_NULL;
1506 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1507 if (klass->unprepare)
1508 success = klass->unprepare (media);
1512 media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1513 media->reused = TRUE;
1515 /* when the media is not reusable, this will effectively unref the media and
1517 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1523 default_unprepare (GstRTSPMedia * media)
1525 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1531 * gst_rtsp_media_set_state:
1532 * @media: a #GstRTSPMedia
1533 * @state: the target state of the media
1534 * @transports: a GArray of #GstRTSPMediaTrans pointers
1536 * Set the state of @media to @state and for the transports in @transports.
1538 * Returns: %TRUE on success.
1541 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1542 GArray * transports)
1545 GstStateChangeReturn ret;
1546 gboolean add, remove, do_state;
1549 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1550 g_return_val_if_fail (transports != NULL, FALSE);
1552 /* NULL and READY are the same */
1553 if (state == GST_STATE_READY)
1554 state = GST_STATE_NULL;
1556 add = remove = FALSE;
1558 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1562 case GST_STATE_NULL:
1563 /* unlock the streams so that they follow the state changes from now on */
1564 unlock_streams (media);
1566 case GST_STATE_PAUSED:
1567 /* we're going from PLAYING to PAUSED, READY or NULL, remove */
1568 if (media->target_state == GST_STATE_PLAYING)
1571 case GST_STATE_PLAYING:
1572 /* we're going to PLAYING, add */
1578 old_active = media->active;
1580 for (i = 0; i < transports->len; i++) {
1581 GstRTSPMediaTrans *tr;
1582 GstRTSPMediaStream *stream;
1583 GstRTSPTransport *trans;
1585 /* we need a non-NULL entry in the array */
1586 tr = g_array_index (transports, GstRTSPMediaTrans *, i);
1590 /* we need a transport */
1591 if (!(trans = tr->transport))
1594 /* get the stream and add the destinations */
1595 stream = gst_rtsp_media_get_stream (media, tr->idx);
1596 switch (trans->lower_transport) {
1597 case GST_RTSP_LOWER_TRANS_UDP:
1598 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1603 dest = trans->destination;
1604 if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1605 min = trans->port.min;
1606 max = trans->port.max;
1608 min = trans->client_port.min;
1609 max = trans->client_port.max;
1612 if (add && !tr->active) {
1613 GST_INFO ("adding %s:%d-%d", dest, min, max);
1614 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1615 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1616 stream->transports = g_list_prepend (stream->transports, tr);
1619 } else if (remove && tr->active) {
1620 GST_INFO ("removing %s:%d-%d", dest, min, max);
1621 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1622 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1623 stream->transports = g_list_remove (stream->transports, tr);
1629 case GST_RTSP_LOWER_TRANS_TCP:
1630 if (add && !tr->active) {
1631 GST_INFO ("adding TCP %s", trans->destination);
1632 stream->transports = g_list_prepend (stream->transports, tr);
1635 } else if (remove && tr->active) {
1636 GST_INFO ("removing TCP %s", trans->destination);
1637 stream->transports = g_list_remove (stream->transports, tr);
1643 GST_INFO ("Unknown transport %d", trans->lower_transport);
1648 /* we just added the first media, do the playing state change */
1649 if (old_active == 0 && add)
1651 /* if we have no more active media, do the downward state changes */
1652 else if (media->active == 0)
1657 GST_INFO ("active %d media %p", media->active, media);
1659 if (do_state && media->target_state != state) {
1660 if (state == GST_STATE_NULL) {
1661 gst_rtsp_media_unprepare (media);
1663 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
1664 media->target_state = state;
1665 ret = gst_element_set_state (media->pipeline, state);
1669 /* remember where we are */
1670 if (state == GST_STATE_PAUSED)
1671 collect_media_stats (media);
1677 * gst_rtsp_media_remove_elements:
1678 * @media: a #GstRTSPMedia
1680 * Remove all elements and the pipeline controlled by @media.
1683 gst_rtsp_media_remove_elements (GstRTSPMedia * media)
1687 unlock_streams (media);
1689 for (i = 0; i < media->streams->len; i++) {
1690 GstRTSPMediaStream *stream;
1692 GST_INFO ("Removing elements of stream %d from pipeline", i);
1694 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
1696 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
1698 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
1700 for (j = 0; j < 2; j++) {
1701 gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL);
1702 gst_element_set_state (stream->udpsink[j], GST_STATE_NULL);
1703 gst_element_set_state (stream->appsrc[j], GST_STATE_NULL);
1704 gst_element_set_state (stream->appsink[j], GST_STATE_NULL);
1705 gst_element_set_state (stream->tee[j], GST_STATE_NULL);
1706 gst_element_set_state (stream->selector[j], GST_STATE_NULL);
1708 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]);
1709 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]);
1710 gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]);
1711 gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]);
1712 gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]);
1713 gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]);
1716 gst_caps_unref (stream->caps);
1717 stream->caps = NULL;
1718 gst_rtsp_media_stream_free (stream);
1720 g_array_remove_range (media->streams, 0, media->streams->len);
1722 gst_element_set_state (media->rtpbin, GST_STATE_NULL);
1723 gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
1725 gst_object_unref (media->pipeline);
1726 media->pipeline = NULL;