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
32 #define DEFAULT_EOS_SHUTDOWN FALSE
34 /* define to dump received RTCP packets */
55 GST_DEBUG_CATEGORY_EXTERN (rtsp_media_debug);
56 #define GST_CAT_DEFAULT rtsp_media_debug
58 static GQuark ssrc_stream_map_key;
60 static void gst_rtsp_media_get_property (GObject * object, guint propid,
61 GValue * value, GParamSpec * pspec);
62 static void gst_rtsp_media_set_property (GObject * object, guint propid,
63 const GValue * value, GParamSpec * pspec);
64 static void gst_rtsp_media_finalize (GObject * obj);
66 static gpointer do_loop (GstRTSPMediaClass * klass);
67 static gboolean default_handle_message (GstRTSPMedia * media,
68 GstMessage * message);
69 static gboolean default_unprepare (GstRTSPMedia * media);
70 static void unlock_streams (GstRTSPMedia * media);
72 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
74 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
77 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
79 GObjectClass *gobject_class;
82 gobject_class = G_OBJECT_CLASS (klass);
84 gobject_class->get_property = gst_rtsp_media_get_property;
85 gobject_class->set_property = gst_rtsp_media_set_property;
86 gobject_class->finalize = gst_rtsp_media_finalize;
88 g_object_class_install_property (gobject_class, PROP_SHARED,
89 g_param_spec_boolean ("shared", "Shared",
90 "If this media pipeline can be shared", DEFAULT_SHARED,
91 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
93 g_object_class_install_property (gobject_class, PROP_REUSABLE,
94 g_param_spec_boolean ("reusable", "Reusable",
95 "If this media pipeline can be reused after an unprepare",
96 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
98 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
99 g_param_spec_flags ("protocols", "Protocols",
100 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
101 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
103 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
104 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
105 "Send an EOS event to the pipeline before unpreparing",
106 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
108 gst_rtsp_media_signals[SIGNAL_PREPARED] =
109 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
110 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
111 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
113 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
114 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
115 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
116 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
118 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
119 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
120 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
121 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
123 klass->context = g_main_context_new ();
124 klass->loop = g_main_loop_new (klass->context, TRUE);
126 klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
128 g_critical ("could not start bus thread: %s", error->message);
130 klass->handle_message = default_handle_message;
131 klass->unprepare = default_unprepare;
133 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
137 gst_rtsp_media_init (GstRTSPMedia * media)
139 media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
140 media->lock = g_mutex_new ();
141 media->cond = g_cond_new ();
143 media->shared = DEFAULT_SHARED;
144 media->reusable = DEFAULT_REUSABLE;
145 media->protocols = DEFAULT_PROTOCOLS;
146 media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
149 /* FIXME. this should be done in multiudpsink */
158 dest_compare (RTSPDestination * a, RTSPDestination * b)
160 if ((a->min == b->min) && (a->max == b->max)
161 && (strcmp (a->dest, b->dest) == 0))
167 static RTSPDestination *
168 create_destination (const gchar * dest, gint min, gint max)
170 RTSPDestination *res;
172 res = g_slice_new (RTSPDestination);
174 res->dest = g_strdup (dest);
182 free_destination (RTSPDestination * dest)
185 g_slice_free (RTSPDestination, dest);
189 gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans * trans)
191 if (trans->transport) {
192 gst_rtsp_transport_free (trans->transport);
193 trans->transport = NULL;
195 if (trans->rtpsource) {
196 g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
197 trans->rtpsource = NULL;
202 gst_rtsp_media_stream_free (GstRTSPMediaStream * stream)
205 g_object_unref (stream->session);
208 gst_caps_unref (stream->caps);
210 if (stream->send_rtp_sink)
211 gst_object_unref (stream->send_rtp_sink);
212 if (stream->send_rtp_src)
213 gst_object_unref (stream->send_rtp_src);
214 if (stream->send_rtcp_src)
215 gst_object_unref (stream->send_rtcp_src);
216 if (stream->recv_rtcp_sink)
217 gst_object_unref (stream->recv_rtcp_sink);
218 if (stream->recv_rtp_sink)
219 gst_object_unref (stream->recv_rtp_sink);
221 g_list_free (stream->transports);
223 g_list_foreach (stream->destinations, (GFunc) free_destination, NULL);
224 g_list_free (stream->destinations);
230 gst_rtsp_media_finalize (GObject * obj)
235 media = GST_RTSP_MEDIA (obj);
237 GST_INFO ("finalize media %p", media);
239 if (media->pipeline) {
240 unlock_streams (media);
241 gst_element_set_state (media->pipeline, GST_STATE_NULL);
242 gst_object_unref (media->pipeline);
245 for (i = 0; i < media->streams->len; i++) {
246 GstRTSPMediaStream *stream;
248 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
250 gst_rtsp_media_stream_free (stream);
252 g_array_free (media->streams, TRUE);
254 g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL);
255 g_list_free (media->dynamic);
258 g_source_destroy (media->source);
259 g_source_unref (media->source);
261 g_mutex_free (media->lock);
262 g_cond_free (media->cond);
264 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
268 gst_rtsp_media_get_property (GObject * object, guint propid,
269 GValue * value, GParamSpec * pspec)
271 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
275 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
278 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
281 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
283 case PROP_EOS_SHUTDOWN:
284 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
287 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
292 gst_rtsp_media_set_property (GObject * object, guint propid,
293 const GValue * value, GParamSpec * pspec)
295 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
299 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
302 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
305 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
307 case PROP_EOS_SHUTDOWN:
308 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
311 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
316 do_loop (GstRTSPMediaClass * klass)
318 GST_INFO ("enter mainloop");
319 g_main_loop_run (klass->loop);
320 GST_INFO ("exit mainloop");
326 collect_media_stats (GstRTSPMedia * media)
329 gint64 position, duration;
331 media->range.unit = GST_RTSP_RANGE_NPT;
333 if (media->is_live) {
334 media->range.min.type = GST_RTSP_TIME_NOW;
335 media->range.min.seconds = -1;
336 media->range.max.type = GST_RTSP_TIME_END;
337 media->range.max.seconds = -1;
339 /* get the position */
340 format = GST_FORMAT_TIME;
341 if (!gst_element_query_position (media->pipeline, &format, &position)) {
342 GST_INFO ("position query failed");
346 /* get the duration */
347 format = GST_FORMAT_TIME;
348 if (!gst_element_query_duration (media->pipeline, &format, &duration)) {
349 GST_INFO ("duration query failed");
353 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
354 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
356 if (position == -1 || media->active > 0) {
357 media->range.min.type = GST_RTSP_TIME_NOW;
358 media->range.min.seconds = -1;
360 media->range.min.type = GST_RTSP_TIME_SECONDS;
361 media->range.min.seconds = ((gdouble) position) / GST_SECOND;
363 if (duration == -1) {
364 media->range.max.type = GST_RTSP_TIME_END;
365 media->range.max.seconds = -1;
367 media->range.max.type = GST_RTSP_TIME_SECONDS;
368 media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
374 * gst_rtsp_media_new:
376 * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
377 * element to produde RTP data for one or more related (audio/video/..)
380 * Returns: a new #GstRTSPMedia object.
383 gst_rtsp_media_new (void)
385 GstRTSPMedia *result;
387 result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
393 * gst_rtsp_media_set_shared:
394 * @media: a #GstRTSPMedia
395 * @shared: the new value
397 * Set or unset if the pipeline for @media can be shared will multiple clients.
398 * When @shared is %TRUE, client requests for this media will share the media
402 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
404 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
406 media->shared = shared;
410 * gst_rtsp_media_is_shared:
411 * @media: a #GstRTSPMedia
413 * Check if the pipeline for @media can be shared between multiple clients.
415 * Returns: %TRUE if the media can be shared between clients.
418 gst_rtsp_media_is_shared (GstRTSPMedia * media)
420 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
422 return media->shared;
426 * gst_rtsp_media_set_reusable:
427 * @media: a #GstRTSPMedia
428 * @reusable: the new value
430 * Set or unset if the pipeline for @media can be reused after the pipeline has
434 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
436 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
438 media->reusable = reusable;
442 * gst_rtsp_media_is_reusable:
443 * @media: a #GstRTSPMedia
445 * Check if the pipeline for @media can be reused after an unprepare.
447 * Returns: %TRUE if the media can be reused
450 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
452 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
454 return media->reusable;
458 * gst_rtsp_media_set_protocols:
459 * @media: a #GstRTSPMedia
460 * @protocols: the new flags
462 * Configure the allowed lower transport for @media.
465 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
467 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
469 media->protocols = protocols;
473 * gst_rtsp_media_get_protocols:
474 * @media: a #GstRTSPMedia
476 * Get the allowed protocols of @media.
478 * Returns: a #GstRTSPLowerTrans
481 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
483 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
484 GST_RTSP_LOWER_TRANS_UNKNOWN);
486 return media->protocols;
490 * gst_rtsp_media_set_eos_shutdown:
491 * @media: a #GstRTSPMedia
492 * @eos_shutdown: the new value
494 * Set or unset if an EOS event will be sent to the pipeline for @media before
498 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
500 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
502 media->eos_shutdown = eos_shutdown;
506 * gst_rtsp_media_is_eos_shutdown:
507 * @media: a #GstRTSPMedia
509 * Check if the pipeline for @media will send an EOS down the pipeline before
512 * Returns: %TRUE if the media will send EOS before unpreparing.
515 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
517 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
519 return media->eos_shutdown;
523 * gst_rtsp_media_n_streams:
524 * @media: a #GstRTSPMedia
526 * Get the number of streams in this media.
528 * Returns: The number of streams.
531 gst_rtsp_media_n_streams (GstRTSPMedia * media)
533 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
535 return media->streams->len;
539 * gst_rtsp_media_get_stream:
540 * @media: a #GstRTSPMedia
541 * @idx: the stream index
543 * Retrieve the stream with index @idx from @media.
545 * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
546 * that index did not exist.
549 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
551 GstRTSPMediaStream *res;
553 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
555 if (idx < media->streams->len)
556 res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
564 * gst_rtsp_media_seek:
565 * @media: a #GstRTSPMedia
566 * @range: a #GstRTSPTimeRange
568 * Seek the pipeline to @range.
570 * Returns: %TRUE on success.
573 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
578 GstSeekType start_type, stop_type;
580 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
581 g_return_val_if_fail (range != NULL, FALSE);
583 if (range->unit != GST_RTSP_RANGE_NPT)
586 /* depends on the current playing state of the pipeline. We might need to
587 * queue this until we get EOS. */
588 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
590 start_type = stop_type = GST_SEEK_TYPE_NONE;
592 switch (range->min.type) {
593 case GST_RTSP_TIME_NOW:
596 case GST_RTSP_TIME_SECONDS:
597 /* only seek when something changed */
598 if (media->range.min.seconds == range->min.seconds) {
601 start = range->min.seconds * GST_SECOND;
602 start_type = GST_SEEK_TYPE_SET;
605 case GST_RTSP_TIME_END:
609 switch (range->max.type) {
610 case GST_RTSP_TIME_SECONDS:
611 /* only seek when something changed */
612 if (media->range.max.seconds == range->max.seconds) {
615 stop = range->max.seconds * GST_SECOND;
616 stop_type = GST_SEEK_TYPE_SET;
619 case GST_RTSP_TIME_END:
621 stop_type = GST_SEEK_TYPE_SET;
623 case GST_RTSP_TIME_NOW:
628 if (start != -1 || stop != -1) {
629 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
630 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
632 res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
633 flags, start_type, start, stop_type, stop);
635 /* and block for the seek to complete */
636 GST_INFO ("done seeking %d", res);
637 gst_element_get_state (media->pipeline, NULL, NULL, -1);
638 GST_INFO ("prerolled again");
640 collect_media_stats (media);
642 GST_INFO ("no seek needed");
651 GST_WARNING ("seek unit %d not supported", range->unit);
656 GST_WARNING ("weird range type %d not supported", range->min.type);
662 * gst_rtsp_media_stream_rtp:
663 * @stream: a #GstRTSPMediaStream
664 * @buffer: a #GstBuffer
666 * Handle an RTP buffer for the stream. This method is usually called when a
667 * message has been received from a client using the TCP transport.
669 * This function takes ownership of @buffer.
671 * Returns: a GstFlowReturn.
674 gst_rtsp_media_stream_rtp (GstRTSPMediaStream * stream, GstBuffer * buffer)
678 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
684 * gst_rtsp_media_stream_rtcp:
685 * @stream: a #GstRTSPMediaStream
686 * @buffer: a #GstBuffer
688 * Handle an RTCP buffer for the stream. This method is usually called when a
689 * message has been received from a client using the TCP transport.
691 * This function takes ownership of @buffer.
693 * Returns: a GstFlowReturn.
696 gst_rtsp_media_stream_rtcp (GstRTSPMediaStream * stream, GstBuffer * buffer)
700 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
705 /* Allocate the udp ports and sockets */
707 alloc_udp_ports (GstRTSPMedia * media, GstRTSPMediaStream * stream)
709 GstStateChangeReturn ret;
710 GstElement *udpsrc0, *udpsrc1;
711 GstElement *udpsink0, *udpsink1;
712 gint tmp_rtp, tmp_rtcp;
714 gint rtpport, rtcpport, sockfd;
723 /* Start with random port */
727 host = "udp://[::0]";
729 host = "udp://0.0.0.0";
731 /* try to allocate 2 UDP ports, the RTP port should be an even
732 * number and the RTCP port should be the next (uneven) port */
734 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
736 goto no_udp_protocol;
737 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
739 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
740 if (ret == GST_STATE_CHANGE_FAILURE) {
746 gst_element_set_state (udpsrc0, GST_STATE_NULL);
747 gst_object_unref (udpsrc0);
751 goto no_udp_protocol;
754 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
756 /* check if port is even */
757 if ((tmp_rtp & 1) != 0) {
758 /* port not even, close and allocate another */
762 gst_element_set_state (udpsrc0, GST_STATE_NULL);
763 gst_object_unref (udpsrc0);
769 /* allocate port+1 for RTCP now */
770 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
772 goto no_udp_rtcp_protocol;
775 tmp_rtcp = tmp_rtp + 1;
776 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
778 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
779 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
780 if (ret == GST_STATE_CHANGE_FAILURE) {
785 gst_element_set_state (udpsrc0, GST_STATE_NULL);
786 gst_object_unref (udpsrc0);
788 gst_element_set_state (udpsrc1, GST_STATE_NULL);
789 gst_object_unref (udpsrc1);
795 /* all fine, do port check */
796 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
797 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
799 /* this should not happen... */
800 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
803 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
805 goto no_udp_protocol;
807 g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
808 g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
809 g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);
811 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
813 goto no_udp_protocol;
815 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
816 "send-duplicates")) {
817 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
818 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
819 stream->filter_duplicates = FALSE;
821 GST_WARNING ("multiudpsink version found without send-duplicates property");
822 stream->filter_duplicates = TRUE;
825 g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
826 g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
827 g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
828 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
829 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
831 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
832 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
833 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
834 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
836 /* we keep these elements, we configure all in configure_transport when the
837 * server told us to really use the UDP ports. */
838 stream->udpsrc[0] = udpsrc0;
839 stream->udpsrc[1] = udpsrc1;
840 stream->udpsink[0] = udpsink0;
841 stream->udpsink[1] = udpsink1;
842 stream->server_port.min = rtpport;
843 stream->server_port.max = rtcpport;
856 no_udp_rtcp_protocol:
867 gst_element_set_state (udpsrc0, GST_STATE_NULL);
868 gst_object_unref (udpsrc0);
871 gst_element_set_state (udpsrc1, GST_STATE_NULL);
872 gst_object_unref (udpsrc1);
875 gst_element_set_state (udpsink0, GST_STATE_NULL);
876 gst_object_unref (udpsink0);
879 gst_element_set_state (udpsink1, GST_STATE_NULL);
880 gst_object_unref (udpsink1);
886 /* executed from streaming thread */
888 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
891 GstCaps *newcaps, *oldcaps;
893 if ((newcaps = GST_PAD_CAPS (pad)))
894 gst_caps_ref (newcaps);
896 oldcaps = stream->caps;
897 stream->caps = newcaps;
900 gst_caps_unref (oldcaps);
902 capsstr = gst_caps_to_string (newcaps);
903 GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr);
908 dump_structure (const GstStructure * s)
912 sstr = gst_structure_to_string (s);
913 GST_INFO ("structure: %s", sstr);
917 static GstRTSPMediaTrans *
918 find_transport (GstRTSPMediaStream * stream, const gchar * rtcp_from)
921 GstRTSPMediaTrans *result = NULL;
926 if (rtcp_from == NULL)
929 tmp = g_strrstr (rtcp_from, ":");
933 port = atoi (tmp + 1);
934 dest = g_strndup (rtcp_from, tmp - rtcp_from);
936 GST_INFO ("finding %s:%d", dest, port);
938 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
939 GstRTSPMediaTrans *trans = walk->data;
942 min = trans->transport->client_port.min;
943 max = trans->transport->client_port.max;
945 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
957 on_new_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
960 GstRTSPMediaTrans *trans;
962 GST_INFO ("%p: new source %p", stream, source);
964 /* see if we have a stream to match with the origin of the RTCP packet */
965 trans = g_object_get_qdata (source, ssrc_stream_map_key);
967 g_object_get (source, "stats", &stats, NULL);
969 const gchar *rtcp_from;
971 dump_structure (stats);
973 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
974 if ((trans = find_transport (stream, rtcp_from))) {
975 GST_INFO ("%p: found transport %p for source %p", stream, trans,
978 /* keep ref to the source */
979 trans->rtpsource = source;
981 g_object_set_qdata (source, ssrc_stream_map_key, trans);
983 gst_structure_free (stats);
986 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
991 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPMediaStream * stream)
993 GST_INFO ("%p: new SDES %p", stream, source);
997 on_ssrc_active (GObject * session, GObject * source,
998 GstRTSPMediaStream * stream)
1000 GstRTSPMediaTrans *trans;
1002 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1004 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1006 if (trans && trans->keep_alive)
1007 trans->keep_alive (trans->ka_user_data);
1011 GstStructure *stats;
1012 g_object_get (source, "stats", &stats, NULL);
1014 dump_structure (stats);
1015 gst_structure_free (stats);
1022 on_bye_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1024 GST_INFO ("%p: source %p bye", stream, source);
1028 on_bye_timeout (GObject * session, GObject * source,
1029 GstRTSPMediaStream * stream)
1031 GstRTSPMediaTrans *trans;
1033 GST_INFO ("%p: source %p bye timeout", stream, source);
1035 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1036 trans->rtpsource = NULL;
1037 trans->timeout = TRUE;
1042 on_timeout (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1044 GstRTSPMediaTrans *trans;
1046 GST_INFO ("%p: source %p timeout", stream, source);
1048 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1049 trans->rtpsource = NULL;
1050 trans->timeout = TRUE;
1054 static GstFlowReturn
1055 handle_new_buffer (GstAppSink * sink, gpointer user_data)
1059 GstRTSPMediaStream *stream;
1061 buffer = gst_app_sink_pull_buffer (sink);
1065 stream = (GstRTSPMediaStream *) user_data;
1067 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1068 GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
1070 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
1072 tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
1075 tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
1078 gst_buffer_unref (buffer);
1083 static GstAppSinkCallbacks sink_cb = {
1084 NULL, /* not interested in EOS */
1085 NULL, /* not interested in preroll buffers */
1089 /* prepare the pipeline objects to handle @stream in @media */
1091 setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
1094 GstPad *pad, *teepad, *selpad;
1095 GstPadLinkReturn ret;
1098 /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2
1099 * for sending RTP/RTCP. The sender and receiver ports are shared between the
1101 if (!alloc_udp_ports (media, stream))
1104 /* add the ports to the pipeline */
1105 for (i = 0; i < 2; i++) {
1106 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]);
1107 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]);
1110 /* create elements for the TCP transfer */
1111 for (i = 0; i < 2; i++) {
1112 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1113 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
1114 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1115 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
1116 g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL);
1117 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
1118 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
1119 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
1120 &sink_cb, stream, NULL);
1123 /* hook up the stream to the RTP session elements. */
1124 name = g_strdup_printf ("send_rtp_sink_%d", idx);
1125 stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1127 name = g_strdup_printf ("send_rtp_src_%d", idx);
1128 stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
1130 name = g_strdup_printf ("send_rtcp_src_%d", idx);
1131 stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
1133 name = g_strdup_printf ("recv_rtcp_sink_%d", idx);
1134 stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
1136 name = g_strdup_printf ("recv_rtp_sink_%d", idx);
1137 stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1140 /* get the session */
1141 g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
1144 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1146 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1148 g_signal_connect (stream->session, "on-ssrc-active",
1149 (GCallback) on_ssrc_active, stream);
1150 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1152 g_signal_connect (stream->session, "on-bye-timeout",
1153 (GCallback) on_bye_timeout, stream);
1154 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
1157 /* link the RTP pad to the session manager */
1158 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
1159 if (ret != GST_PAD_LINK_OK)
1162 /* make tee for RTP and link to stream */
1163 stream->tee[0] = gst_element_factory_make ("tee", NULL);
1164 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]);
1166 pad = gst_element_get_static_pad (stream->tee[0], "sink");
1167 gst_pad_link (stream->send_rtp_src, pad);
1168 gst_object_unref (pad);
1170 /* link RTP sink, we're pretty sure this will work. */
1171 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1172 pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
1173 gst_pad_link (teepad, pad);
1174 gst_object_unref (pad);
1175 gst_object_unref (teepad);
1177 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1178 pad = gst_element_get_static_pad (stream->appsink[0], "sink");
1179 gst_pad_link (teepad, pad);
1180 gst_object_unref (pad);
1181 gst_object_unref (teepad);
1183 /* make tee for RTCP */
1184 stream->tee[1] = gst_element_factory_make ("tee", NULL);
1185 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]);
1187 pad = gst_element_get_static_pad (stream->tee[1], "sink");
1188 gst_pad_link (stream->send_rtcp_src, pad);
1189 gst_object_unref (pad);
1191 /* link RTCP elements */
1192 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1193 pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
1194 gst_pad_link (teepad, pad);
1195 gst_object_unref (pad);
1196 gst_object_unref (teepad);
1198 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1199 pad = gst_element_get_static_pad (stream->appsink[1], "sink");
1200 gst_pad_link (teepad, pad);
1201 gst_object_unref (pad);
1202 gst_object_unref (teepad);
1204 /* make selector for the RTP receivers */
1205 stream->selector[0] = gst_element_factory_make ("input-selector", NULL);
1206 g_object_set (stream->selector[0], "select-all", TRUE, NULL);
1207 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
1209 pad = gst_element_get_static_pad (stream->selector[0], "src");
1210 gst_pad_link (pad, stream->recv_rtp_sink);
1211 gst_object_unref (pad);
1213 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1214 pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
1215 gst_pad_link (pad, selpad);
1216 gst_object_unref (pad);
1217 gst_object_unref (selpad);
1219 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1220 pad = gst_element_get_static_pad (stream->appsrc[0], "src");
1221 gst_pad_link (pad, selpad);
1222 gst_object_unref (pad);
1223 gst_object_unref (selpad);
1225 /* make selector for the RTCP receivers */
1226 stream->selector[1] = gst_element_factory_make ("input-selector", NULL);
1227 g_object_set (stream->selector[1], "select-all", TRUE, NULL);
1228 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
1230 pad = gst_element_get_static_pad (stream->selector[1], "src");
1231 gst_pad_link (pad, stream->recv_rtcp_sink);
1232 gst_object_unref (pad);
1234 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1235 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
1236 gst_pad_link (pad, selpad);
1237 gst_object_unref (pad);
1238 gst_object_unref (selpad);
1240 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1241 pad = gst_element_get_static_pad (stream->appsrc[1], "src");
1242 gst_pad_link (pad, selpad);
1243 gst_object_unref (pad);
1244 gst_object_unref (selpad);
1246 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1248 gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
1249 gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
1250 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
1251 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
1253 /* be notified of caps changes */
1254 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
1255 (GCallback) caps_notify, stream);
1257 stream->prepared = TRUE;
1264 GST_WARNING ("failed to link stream %d", idx);
1270 unlock_streams (GstRTSPMedia * media)
1274 /* unlock the udp src elements */
1275 n_streams = gst_rtsp_media_n_streams (media);
1276 for (i = 0; i < n_streams; i++) {
1277 GstRTSPMediaStream *stream;
1279 stream = gst_rtsp_media_get_stream (media, i);
1281 gst_element_set_locked_state (stream->udpsrc[0], FALSE);
1282 gst_element_set_locked_state (stream->udpsrc[1], FALSE);
1287 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1289 g_mutex_lock (media->lock);
1290 /* never overwrite the error status */
1291 if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
1292 media->status = status;
1293 GST_DEBUG ("setting new status to %d", status);
1294 g_cond_broadcast (media->cond);
1295 g_mutex_unlock (media->lock);
1298 static GstRTSPMediaStatus
1299 gst_rtsp_media_get_status (GstRTSPMedia * media)
1301 GstRTSPMediaStatus result;
1304 g_mutex_lock (media->lock);
1305 g_get_current_time (&timeout);
1306 g_time_val_add (&timeout, 20 * G_USEC_PER_SEC);
1307 /* while we are preparing, wait */
1308 while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1309 GST_DEBUG ("waiting for status change");
1310 if (!g_cond_timed_wait (media->cond, media->lock, &timeout)) {
1311 GST_DEBUG ("timeout, assuming error status");
1312 media->status = GST_RTSP_MEDIA_STATUS_ERROR;
1315 /* could be success or error */
1316 result = media->status;
1317 GST_DEBUG ("got status %d", result);
1318 g_mutex_unlock (media->lock);
1324 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1326 GstMessageType type;
1328 type = GST_MESSAGE_TYPE (message);
1331 case GST_MESSAGE_STATE_CHANGED:
1333 case GST_MESSAGE_BUFFERING:
1337 gst_message_parse_buffering (message, &percent);
1339 /* no state management needed for live pipelines */
1343 if (percent == 100) {
1344 /* a 100% message means buffering is done */
1345 media->buffering = FALSE;
1346 /* if the desired state is playing, go back */
1347 if (media->target_state == GST_STATE_PLAYING) {
1348 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1349 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1351 GST_INFO ("Buffering done");
1354 /* buffering busy */
1355 if (media->buffering == FALSE) {
1356 if (media->target_state == GST_STATE_PLAYING) {
1357 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1358 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1359 gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1361 GST_INFO ("Buffering ...");
1364 media->buffering = TRUE;
1368 case GST_MESSAGE_LATENCY:
1370 gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
1373 case GST_MESSAGE_ERROR:
1378 gst_message_parse_error (message, &gerror, &debug);
1379 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1380 g_error_free (gerror);
1383 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1386 case GST_MESSAGE_WARNING:
1391 gst_message_parse_warning (message, &gerror, &debug);
1392 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1393 g_error_free (gerror);
1397 case GST_MESSAGE_ELEMENT:
1399 case GST_MESSAGE_STREAM_STATUS:
1401 case GST_MESSAGE_ASYNC_DONE:
1402 if (!media->adding) {
1403 /* when we are dynamically adding pads, the addition of the udpsrc will
1404 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1405 * wait for the final ASYNC_DONE after everything prerolled */
1406 GST_INFO ("%p: got ASYNC_DONE", media);
1407 collect_media_stats (media);
1409 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1411 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1414 case GST_MESSAGE_EOS:
1415 GST_INFO ("%p: got EOS", media);
1416 if (media->eos_pending) {
1417 GST_DEBUG ("shutting down after EOS");
1418 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1419 media->eos_pending = FALSE;
1420 g_object_unref (media);
1424 GST_INFO ("%p: got message type %s", media,
1425 gst_message_type_get_name (type));
1432 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1434 GstRTSPMediaClass *klass;
1437 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1439 if (klass->handle_message)
1440 ret = klass->handle_message (media, message);
1447 /* called from streaming threads */
1449 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1451 GstRTSPMediaStream *stream;
1455 i = media->streams->len + 1;
1457 GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
1459 stream = g_new0 (GstRTSPMediaStream, 1);
1460 stream->payloader = element;
1462 name = g_strdup_printf ("dynpay%d", i);
1464 media->adding = TRUE;
1466 /* ghost the pad of the payloader to the element */
1467 stream->srcpad = gst_ghost_pad_new (name, pad);
1468 gst_pad_set_active (stream->srcpad, TRUE);
1469 gst_element_add_pad (media->element, stream->srcpad);
1472 /* add stream now */
1473 g_array_append_val (media->streams, stream);
1475 setup_stream (stream, i, media);
1477 for (i = 0; i < 2; i++) {
1478 gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
1479 gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
1480 gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
1481 gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
1482 gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
1484 media->adding = FALSE;
1488 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1490 GST_INFO ("no more pads");
1491 if (media->fakesink) {
1492 gst_object_ref (media->fakesink);
1493 gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
1494 gst_element_set_state (media->fakesink, GST_STATE_NULL);
1495 gst_object_unref (media->fakesink);
1496 media->fakesink = NULL;
1497 GST_INFO ("removed fakesink");
1502 * gst_rtsp_media_prepare:
1503 * @media: a #GstRTSPMedia
1505 * Prepare @media for streaming. This function will create the pipeline and
1506 * other objects to manage the streaming.
1508 * It will preroll the pipeline and collect vital information about the streams
1509 * such as the duration.
1511 * Returns: %TRUE on success.
1514 gst_rtsp_media_prepare (GstRTSPMedia * media)
1516 GstStateChangeReturn ret;
1517 GstRTSPMediaStatus status;
1519 GstRTSPMediaClass *klass;
1523 if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1526 if (!media->reusable && media->reused)
1529 GST_INFO ("preparing media %p", media);
1531 /* reset some variables */
1532 media->is_live = FALSE;
1533 media->buffering = FALSE;
1534 /* we're preparing now */
1535 media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1537 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
1539 /* add the pipeline bus to our custom mainloop */
1540 media->source = gst_bus_create_watch (bus);
1541 gst_object_unref (bus);
1543 g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
1545 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1546 media->id = g_source_attach (media->source, klass->context);
1548 media->rtpbin = gst_element_factory_make ("gstrtpbin", NULL);
1550 /* add stuff to the bin */
1551 gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
1553 /* link streams we already have, other streams might appear when we have
1554 * dynamic elements */
1555 n_streams = gst_rtsp_media_n_streams (media);
1556 for (i = 0; i < n_streams; i++) {
1557 GstRTSPMediaStream *stream;
1559 stream = gst_rtsp_media_get_stream (media, i);
1561 setup_stream (stream, i, media);
1564 for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
1565 GstElement *elem = walk->data;
1567 GST_INFO ("adding callbacks for dynamic element %p", elem);
1569 g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
1570 g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
1572 /* we add a fakesink here in order to make the state change async. We remove
1573 * the fakesink again in the no-more-pads callback. */
1574 media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1575 gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
1578 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1579 /* first go to PAUSED */
1580 ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1581 media->target_state = GST_STATE_PAUSED;
1584 case GST_STATE_CHANGE_SUCCESS:
1585 GST_INFO ("SUCCESS state change for media %p", media);
1587 case GST_STATE_CHANGE_ASYNC:
1588 GST_INFO ("ASYNC state change for media %p", media);
1590 case GST_STATE_CHANGE_NO_PREROLL:
1591 /* we need to go to PLAYING */
1592 GST_INFO ("NO_PREROLL state change: live media %p", media);
1593 media->is_live = TRUE;
1594 ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1595 if (ret == GST_STATE_CHANGE_FAILURE)
1598 case GST_STATE_CHANGE_FAILURE:
1602 /* now wait for all pads to be prerolled */
1603 status = gst_rtsp_media_get_status (media);
1604 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1607 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1609 GST_INFO ("object %p is prerolled", media);
1621 GST_WARNING ("can not reuse media %p", media);
1626 GST_WARNING ("failed to preroll pipeline");
1627 unlock_streams (media);
1628 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1629 gst_rtsp_media_unprepare (media);
1635 * gst_rtsp_media_unprepare:
1636 * @media: a #GstRTSPMedia
1638 * Unprepare @media. After this call, the media should be prepared again before
1639 * it can be used again. If the media is set to be non-reusable, a new instance
1642 * Returns: %TRUE on success.
1645 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1647 GstRTSPMediaClass *klass;
1650 if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1653 GST_INFO ("unprepare media %p", media);
1654 media->target_state = GST_STATE_NULL;
1656 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1657 if (klass->unprepare)
1658 success = klass->unprepare (media);
1662 media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1663 media->reused = TRUE;
1665 /* when the media is not reusable, this will effectively unref the media and
1667 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1673 default_unprepare (GstRTSPMedia * media)
1675 if (media->eos_shutdown) {
1676 GST_DEBUG ("sending EOS for shutdown");
1677 /* ref so that we don't disappear */
1678 g_object_ref (media);
1679 media->eos_pending = TRUE;
1680 gst_element_send_event (media->pipeline, gst_event_new_eos ());
1681 /* we need to go to playing again for the EOS to propagate, normally in this
1682 * state, nothing is receiving data from us anymore so this is ok. */
1683 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1685 GST_DEBUG ("shutting down");
1686 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1692 add_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1693 gchar * dest, gint min, gint max)
1695 gboolean do_add = TRUE;
1696 RTSPDestination *ndest;
1698 if (stream->filter_duplicates) {
1699 RTSPDestination fdest;
1706 /* first see if we already added this destination */
1708 g_list_find_custom (stream->destinations, &fdest,
1709 (GCompareFunc) dest_compare);
1711 ndest = (RTSPDestination *) find->data;
1713 GST_INFO ("already streaming to %s:%d-%d with %d clients", dest, min, max,
1721 GST_INFO ("adding %s:%d-%d", dest, min, max);
1722 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1723 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1725 if (stream->filter_duplicates) {
1726 ndest = create_destination (dest, min, max);
1727 stream->destinations = g_list_prepend (stream->destinations, ndest);
1733 remove_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1734 gchar * dest, gint min, gint max)
1736 gboolean do_remove = TRUE;
1737 RTSPDestination *ndest = NULL;
1740 if (stream->filter_duplicates) {
1741 RTSPDestination fdest;
1747 /* first see if we already added this destination */
1749 g_list_find_custom (stream->destinations, &fdest,
1750 (GCompareFunc) dest_compare);
1754 ndest = (RTSPDestination *) find->data;
1755 if (--ndest->count > 0) {
1757 GST_INFO ("still streaming to %s:%d-%d with %d clients", dest, min, max,
1763 GST_INFO ("removing %s:%d-%d", dest, min, max);
1764 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1765 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1767 if (stream->filter_duplicates) {
1768 stream->destinations = g_list_delete_link (stream->destinations, find);
1769 free_destination (ndest);
1775 * gst_rtsp_media_set_state:
1776 * @media: a #GstRTSPMedia
1777 * @state: the target state of the media
1778 * @transports: a #GArray of #GstRTSPMediaTrans pointers
1780 * Set the state of @media to @state and for the transports in @transports.
1782 * Returns: %TRUE on success.
1785 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1786 GArray * transports)
1789 GstStateChangeReturn ret;
1790 gboolean add, remove, do_state;
1793 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1794 g_return_val_if_fail (transports != NULL, FALSE);
1796 /* NULL and READY are the same */
1797 if (state == GST_STATE_READY)
1798 state = GST_STATE_NULL;
1800 add = remove = FALSE;
1802 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1806 case GST_STATE_NULL:
1807 /* unlock the streams so that they follow the state changes from now on */
1808 unlock_streams (media);
1810 case GST_STATE_PAUSED:
1811 /* we're going from PLAYING to PAUSED, READY or NULL, remove */
1812 if (media->target_state == GST_STATE_PLAYING)
1815 case GST_STATE_PLAYING:
1816 /* we're going to PLAYING, add */
1822 old_active = media->active;
1824 for (i = 0; i < transports->len; i++) {
1825 GstRTSPMediaTrans *tr;
1826 GstRTSPMediaStream *stream;
1827 GstRTSPTransport *trans;
1829 /* we need a non-NULL entry in the array */
1830 tr = g_array_index (transports, GstRTSPMediaTrans *, i);
1834 /* we need a transport */
1835 if (!(trans = tr->transport))
1838 /* get the stream and add the destinations */
1839 stream = gst_rtsp_media_get_stream (media, tr->idx);
1840 switch (trans->lower_transport) {
1841 case GST_RTSP_LOWER_TRANS_UDP:
1842 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1847 dest = trans->destination;
1848 if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1849 min = trans->port.min;
1850 max = trans->port.max;
1852 min = trans->client_port.min;
1853 max = trans->client_port.max;
1856 if (add && !tr->active) {
1857 add_udp_destination (media, stream, dest, min, max);
1858 stream->transports = g_list_prepend (stream->transports, tr);
1861 } else if (remove && tr->active) {
1862 remove_udp_destination (media, stream, dest, min, max);
1863 stream->transports = g_list_remove (stream->transports, tr);
1869 case GST_RTSP_LOWER_TRANS_TCP:
1870 if (add && !tr->active) {
1871 GST_INFO ("adding TCP %s", trans->destination);
1872 stream->transports = g_list_prepend (stream->transports, tr);
1875 } else if (remove && tr->active) {
1876 GST_INFO ("removing TCP %s", trans->destination);
1877 stream->transports = g_list_remove (stream->transports, tr);
1883 GST_INFO ("Unknown transport %d", trans->lower_transport);
1888 /* we just added the first media, do the playing state change */
1889 if (old_active == 0 && add)
1891 /* if we have no more active media, do the downward state changes */
1892 else if (media->active == 0)
1897 GST_INFO ("state %d active %d media %p do_state %d", state, media->active,
1900 if (media->target_state != state) {
1902 if (state == GST_STATE_NULL) {
1903 gst_rtsp_media_unprepare (media);
1905 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
1907 media->target_state = state;
1908 ret = gst_element_set_state (media->pipeline, state);
1911 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
1915 /* remember where we are */
1916 if (state == GST_STATE_PAUSED || old_active != media->active)
1917 collect_media_stats (media);
1923 * gst_rtsp_media_remove_elements:
1924 * @media: a #GstRTSPMedia
1926 * Remove all elements and the pipeline controlled by @media.
1929 gst_rtsp_media_remove_elements (GstRTSPMedia * media)
1933 unlock_streams (media);
1935 for (i = 0; i < media->streams->len; i++) {
1936 GstRTSPMediaStream *stream;
1938 GST_INFO ("Removing elements of stream %d from pipeline", i);
1940 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
1942 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
1944 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
1946 for (j = 0; j < 2; j++) {
1947 gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL);
1948 gst_element_set_state (stream->udpsink[j], GST_STATE_NULL);
1949 gst_element_set_state (stream->appsrc[j], GST_STATE_NULL);
1950 gst_element_set_state (stream->appsink[j], GST_STATE_NULL);
1951 gst_element_set_state (stream->tee[j], GST_STATE_NULL);
1952 gst_element_set_state (stream->selector[j], GST_STATE_NULL);
1954 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]);
1955 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]);
1956 gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]);
1957 gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]);
1958 gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]);
1959 gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]);
1962 gst_caps_unref (stream->caps);
1963 stream->caps = NULL;
1964 gst_rtsp_media_stream_free (stream);
1966 g_array_remove_range (media->streams, 0, media->streams->len);
1968 gst_element_set_state (media->rtpbin, GST_STATE_NULL);
1969 gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
1971 gst_object_unref (media->pipeline);
1972 media->pipeline = NULL;