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-funnel.h"
27 #include "rtsp-media.h"
29 #define DEFAULT_SHARED FALSE
30 #define DEFAULT_REUSABLE FALSE
31 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
32 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
33 #define DEFAULT_EOS_SHUTDOWN FALSE
34 #define DEFAULT_BUFFER_SIZE 0x80000
35 #define DEFAULT_MULTICAST_GROUP "224.2.0.1"
37 /* define to dump received RTCP packets */
60 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
61 #define GST_CAT_DEFAULT rtsp_media_debug
63 static GQuark ssrc_stream_map_key;
65 static void gst_rtsp_media_get_property (GObject * object, guint propid,
66 GValue * value, GParamSpec * pspec);
67 static void gst_rtsp_media_set_property (GObject * object, guint propid,
68 const GValue * value, GParamSpec * pspec);
69 static void gst_rtsp_media_finalize (GObject * obj);
71 static gpointer do_loop (GstRTSPMediaClass * klass);
72 static gboolean default_handle_message (GstRTSPMedia * media,
73 GstMessage * message);
74 static gboolean default_unprepare (GstRTSPMedia * media);
75 static void unlock_streams (GstRTSPMedia * media);
77 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
79 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
82 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
84 GObjectClass *gobject_class;
87 gobject_class = G_OBJECT_CLASS (klass);
89 gobject_class->get_property = gst_rtsp_media_get_property;
90 gobject_class->set_property = gst_rtsp_media_set_property;
91 gobject_class->finalize = gst_rtsp_media_finalize;
93 g_object_class_install_property (gobject_class, PROP_SHARED,
94 g_param_spec_boolean ("shared", "Shared",
95 "If this media pipeline can be shared", DEFAULT_SHARED,
96 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
98 g_object_class_install_property (gobject_class, PROP_REUSABLE,
99 g_param_spec_boolean ("reusable", "Reusable",
100 "If this media pipeline can be reused after an unprepare",
101 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
103 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
104 g_param_spec_flags ("protocols", "Protocols",
105 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
106 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
108 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
109 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
110 "Send an EOS event to the pipeline before unpreparing",
111 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
113 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
114 g_param_spec_uint ("buffer-size", "Buffer Size",
115 "The kernel UDP buffer size to use", 0, G_MAXUINT,
116 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
118 g_object_class_install_property (gobject_class, PROP_MULTICAST_GROUP,
119 g_param_spec_string ("multicast-group", "Multicast Group",
120 "The Multicast group to send media to",
121 DEFAULT_MULTICAST_GROUP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
123 gst_rtsp_media_signals[SIGNAL_PREPARED] =
124 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
125 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
126 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
128 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
129 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
130 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
131 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
133 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
134 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
135 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
136 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
138 klass->context = g_main_context_new ();
139 klass->loop = g_main_loop_new (klass->context, TRUE);
141 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
143 klass->thread = g_thread_create ((GThreadFunc) do_loop, klass, TRUE, &error);
145 g_critical ("could not start bus thread: %s", error->message);
147 klass->handle_message = default_handle_message;
148 klass->unprepare = default_unprepare;
150 ssrc_stream_map_key = g_quark_from_static_string ("GstRTSPServer.stream");
152 gst_element_register (NULL, "rtspfunnel", GST_RANK_NONE, RTSP_TYPE_FUNNEL);
157 gst_rtsp_media_init (GstRTSPMedia * media)
159 media->streams = g_array_new (FALSE, TRUE, sizeof (GstRTSPMediaStream *));
160 media->lock = g_mutex_new ();
161 media->cond = g_cond_new ();
163 media->shared = DEFAULT_SHARED;
164 media->reusable = DEFAULT_REUSABLE;
165 media->protocols = DEFAULT_PROTOCOLS;
166 media->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
167 media->buffer_size = DEFAULT_BUFFER_SIZE;
168 media->multicast_group = g_strdup (DEFAULT_MULTICAST_GROUP);
172 gst_rtsp_media_trans_cleanup (GstRTSPMediaTrans * trans)
174 if (trans->transport) {
175 gst_rtsp_transport_free (trans->transport);
176 trans->transport = NULL;
178 if (trans->rtpsource) {
179 g_object_set_qdata (trans->rtpsource, ssrc_stream_map_key, NULL);
180 trans->rtpsource = NULL;
185 gst_rtsp_media_stream_free (GstRTSPMediaStream * stream)
188 g_object_unref (stream->session);
191 gst_caps_unref (stream->caps);
193 if (stream->send_rtp_sink)
194 gst_object_unref (stream->send_rtp_sink);
195 if (stream->send_rtp_src)
196 gst_object_unref (stream->send_rtp_src);
197 if (stream->send_rtcp_src)
198 gst_object_unref (stream->send_rtcp_src);
199 if (stream->recv_rtcp_sink)
200 gst_object_unref (stream->recv_rtcp_sink);
201 if (stream->recv_rtp_sink)
202 gst_object_unref (stream->recv_rtp_sink);
204 g_list_free (stream->transports);
210 gst_rtsp_media_finalize (GObject * obj)
215 media = GST_RTSP_MEDIA (obj);
217 GST_INFO ("finalize media %p", media);
219 if (media->pipeline) {
220 unlock_streams (media);
221 gst_element_set_state (media->pipeline, GST_STATE_NULL);
222 gst_object_unref (media->pipeline);
225 for (i = 0; i < media->streams->len; i++) {
226 GstRTSPMediaStream *stream;
228 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
230 gst_rtsp_media_stream_free (stream);
232 g_array_free (media->streams, TRUE);
234 g_list_foreach (media->dynamic, (GFunc) gst_object_unref, NULL);
235 g_list_free (media->dynamic);
238 g_source_destroy (media->source);
239 g_source_unref (media->source);
241 g_free (media->multicast_group);
242 g_mutex_free (media->lock);
243 g_cond_free (media->cond);
245 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
249 gst_rtsp_media_get_property (GObject * object, guint propid,
250 GValue * value, GParamSpec * pspec)
252 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
256 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
259 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
262 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
264 case PROP_EOS_SHUTDOWN:
265 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
267 case PROP_BUFFER_SIZE:
268 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
270 case PROP_MULTICAST_GROUP:
271 g_value_take_string (value, gst_rtsp_media_get_multicast_group (media));
274 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
279 gst_rtsp_media_set_property (GObject * object, guint propid,
280 const GValue * value, GParamSpec * pspec)
282 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
286 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
289 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
292 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
294 case PROP_EOS_SHUTDOWN:
295 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
297 case PROP_BUFFER_SIZE:
298 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
300 case PROP_MULTICAST_GROUP:
301 gst_rtsp_media_set_multicast_group (media, g_value_get_string (value));
304 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
309 do_loop (GstRTSPMediaClass * klass)
311 GST_INFO ("enter mainloop");
312 g_main_loop_run (klass->loop);
313 GST_INFO ("exit mainloop");
319 collect_media_stats (GstRTSPMedia * media)
322 gint64 position, duration;
324 media->range.unit = GST_RTSP_RANGE_NPT;
326 if (media->is_live) {
327 media->range.min.type = GST_RTSP_TIME_NOW;
328 media->range.min.seconds = -1;
329 media->range.max.type = GST_RTSP_TIME_END;
330 media->range.max.seconds = -1;
332 /* get the position */
333 format = GST_FORMAT_TIME;
334 if (!gst_element_query_position (media->pipeline, &format, &position)) {
335 GST_INFO ("position query failed");
339 /* get the duration */
340 format = GST_FORMAT_TIME;
341 if (!gst_element_query_duration (media->pipeline, &format, &duration)) {
342 GST_INFO ("duration query failed");
346 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
347 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
349 if (position == -1) {
350 media->range.min.type = GST_RTSP_TIME_NOW;
351 media->range.min.seconds = -1;
353 media->range.min.type = GST_RTSP_TIME_SECONDS;
354 media->range.min.seconds = ((gdouble) position) / GST_SECOND;
356 if (duration == -1) {
357 media->range.max.type = GST_RTSP_TIME_END;
358 media->range.max.seconds = -1;
360 media->range.max.type = GST_RTSP_TIME_SECONDS;
361 media->range.max.seconds = ((gdouble) duration) / GST_SECOND;
367 * gst_rtsp_media_new:
369 * Create a new #GstRTSPMedia instance. The #GstRTSPMedia object contains the
370 * element to produde RTP data for one or more related (audio/video/..)
373 * Returns: a new #GstRTSPMedia object.
376 gst_rtsp_media_new (void)
378 GstRTSPMedia *result;
380 result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
386 * gst_rtsp_media_set_shared:
387 * @media: a #GstRTSPMedia
388 * @shared: the new value
390 * Set or unset if the pipeline for @media can be shared will multiple clients.
391 * When @shared is %TRUE, client requests for this media will share the media
395 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
397 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
399 media->shared = shared;
403 * gst_rtsp_media_is_shared:
404 * @media: a #GstRTSPMedia
406 * Check if the pipeline for @media can be shared between multiple clients.
408 * Returns: %TRUE if the media can be shared between clients.
411 gst_rtsp_media_is_shared (GstRTSPMedia * media)
413 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
415 return media->shared;
419 * gst_rtsp_media_set_reusable:
420 * @media: a #GstRTSPMedia
421 * @reusable: the new value
423 * Set or unset if the pipeline for @media can be reused after the pipeline has
427 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
429 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
431 media->reusable = reusable;
435 * gst_rtsp_media_is_reusable:
436 * @media: a #GstRTSPMedia
438 * Check if the pipeline for @media can be reused after an unprepare.
440 * Returns: %TRUE if the media can be reused
443 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
445 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
447 return media->reusable;
451 * gst_rtsp_media_set_protocols:
452 * @media: a #GstRTSPMedia
453 * @protocols: the new flags
455 * Configure the allowed lower transport for @media.
458 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
460 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
462 media->protocols = protocols;
466 * gst_rtsp_media_get_protocols:
467 * @media: a #GstRTSPMedia
469 * Get the allowed protocols of @media.
471 * Returns: a #GstRTSPLowerTrans
474 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
476 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
477 GST_RTSP_LOWER_TRANS_UNKNOWN);
479 return media->protocols;
483 * gst_rtsp_media_set_eos_shutdown:
484 * @media: a #GstRTSPMedia
485 * @eos_shutdown: the new value
487 * Set or unset if an EOS event will be sent to the pipeline for @media before
491 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
493 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
495 media->eos_shutdown = eos_shutdown;
499 * gst_rtsp_media_is_eos_shutdown:
500 * @media: a #GstRTSPMedia
502 * Check if the pipeline for @media will send an EOS down the pipeline before
505 * Returns: %TRUE if the media will send EOS before unpreparing.
508 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
510 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
512 return media->eos_shutdown;
516 * gst_rtsp_media_set_buffer_size:
517 * @media: a #GstRTSPMedia
518 * @size: the new value
520 * Set the kernel UDP buffer size.
523 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
525 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
527 media->buffer_size = size;
531 * gst_rtsp_media_get_buffer_size:
532 * @media: a #GstRTSPMedia
534 * Get the kernel UDP buffer size.
536 * Returns: the kernel UDP buffer size.
539 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
541 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
543 return media->buffer_size;
547 * gst_rtsp_media_set_multicast_group:
548 * @media: a #GstRTSPMedia
549 * @mc: the new multicast group
551 * Set the multicast group that media from @media will be streamed to.
554 gst_rtsp_media_set_multicast_group (GstRTSPMedia * media, const gchar * mc)
556 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
558 g_mutex_lock (media->lock);
559 g_free (media->multicast_group);
560 media->multicast_group = g_strdup (mc);
561 g_mutex_unlock (media->lock);
565 * gst_rtsp_media_get_multicast_group:
566 * @media: a #GstRTSPMedia
568 * Get the multicast group that media from @media will be streamed to.
570 * Returns: the multicast group
573 gst_rtsp_media_get_multicast_group (GstRTSPMedia * media)
577 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
579 g_mutex_lock (media->lock);
580 result = g_strdup (media->multicast_group);
581 g_mutex_unlock (media->lock);
587 * gst_rtsp_media_set_auth:
588 * @media: a #GstRTSPMedia
589 * @auth: a #GstRTSPAuth
591 * configure @auth to be used as the authentication manager of @media.
594 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
598 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
607 g_object_unref (old);
612 * gst_rtsp_media_get_auth:
613 * @media: a #GstRTSPMedia
615 * Get the #GstRTSPAuth used as the authentication manager of @media.
617 * Returns: the #GstRTSPAuth of @media. g_object_unref() after
621 gst_rtsp_media_get_auth (GstRTSPMedia * media)
625 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
627 if ((result = media->auth))
628 g_object_ref (result);
635 * gst_rtsp_media_n_streams:
636 * @media: a #GstRTSPMedia
638 * Get the number of streams in this media.
640 * Returns: The number of streams.
643 gst_rtsp_media_n_streams (GstRTSPMedia * media)
645 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
647 return media->streams->len;
651 * gst_rtsp_media_get_stream:
652 * @media: a #GstRTSPMedia
653 * @idx: the stream index
655 * Retrieve the stream with index @idx from @media.
657 * Returns: the #GstRTSPMediaStream at index @idx or %NULL when a stream with
658 * that index did not exist.
661 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
663 GstRTSPMediaStream *res;
665 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
667 if (idx < media->streams->len)
668 res = g_array_index (media->streams, GstRTSPMediaStream *, idx);
676 * gst_rtsp_media_get_range_string:
677 * @media: a #GstRTSPMedia
678 * @play: for the PLAY request
680 * Get the current range as a string.
682 * Returns: The range as a string, g_free() after usage.
685 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
688 GstRTSPTimeRange range;
691 range = media->range;
693 if (!play && media->active > 0) {
694 range.min.type = GST_RTSP_TIME_NOW;
695 range.min.seconds = -1;
698 result = gst_rtsp_range_to_string (&range);
704 * gst_rtsp_media_seek:
705 * @media: a #GstRTSPMedia
706 * @range: a #GstRTSPTimeRange
708 * Seek the pipeline to @range.
710 * Returns: %TRUE on success.
713 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
718 GstSeekType start_type, stop_type;
720 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
721 g_return_val_if_fail (range != NULL, FALSE);
723 if (range->unit != GST_RTSP_RANGE_NPT)
726 /* depends on the current playing state of the pipeline. We might need to
727 * queue this until we get EOS. */
728 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
730 start_type = stop_type = GST_SEEK_TYPE_NONE;
732 switch (range->min.type) {
733 case GST_RTSP_TIME_NOW:
736 case GST_RTSP_TIME_SECONDS:
737 /* only seek when something changed */
738 if (media->range.min.seconds == range->min.seconds) {
741 start = range->min.seconds * GST_SECOND;
742 start_type = GST_SEEK_TYPE_SET;
745 case GST_RTSP_TIME_END:
749 switch (range->max.type) {
750 case GST_RTSP_TIME_SECONDS:
751 /* only seek when something changed */
752 if (media->range.max.seconds == range->max.seconds) {
755 stop = range->max.seconds * GST_SECOND;
756 stop_type = GST_SEEK_TYPE_SET;
759 case GST_RTSP_TIME_END:
761 stop_type = GST_SEEK_TYPE_SET;
763 case GST_RTSP_TIME_NOW:
768 if (start != -1 || stop != -1) {
769 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
770 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
772 res = gst_element_seek (media->pipeline, 1.0, GST_FORMAT_TIME,
773 flags, start_type, start, stop_type, stop);
775 /* and block for the seek to complete */
776 GST_INFO ("done seeking %d", res);
777 gst_element_get_state (media->pipeline, NULL, NULL, -1);
778 GST_INFO ("prerolled again");
780 collect_media_stats (media);
782 GST_INFO ("no seek needed");
791 GST_WARNING ("seek unit %d not supported", range->unit);
796 GST_WARNING ("weird range type %d not supported", range->min.type);
802 * gst_rtsp_media_stream_rtp:
803 * @stream: a #GstRTSPMediaStream
804 * @buffer: a #GstBuffer
806 * Handle an RTP buffer for the stream. This method is usually called when a
807 * message has been received from a client using the TCP transport.
809 * This function takes ownership of @buffer.
811 * Returns: a GstFlowReturn.
814 gst_rtsp_media_stream_rtp (GstRTSPMediaStream * stream, GstBuffer * buffer)
818 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[0]), buffer);
824 * gst_rtsp_media_stream_rtcp:
825 * @stream: a #GstRTSPMediaStream
826 * @buffer: a #GstBuffer
828 * Handle an RTCP buffer for the stream. This method is usually called when a
829 * message has been received from a client using the TCP transport.
831 * This function takes ownership of @buffer.
833 * Returns: a GstFlowReturn.
836 gst_rtsp_media_stream_rtcp (GstRTSPMediaStream * stream, GstBuffer * buffer)
840 ret = gst_app_src_push_buffer (GST_APP_SRC_CAST (stream->appsrc[1]), buffer);
845 /* Allocate the udp ports and sockets */
847 alloc_udp_ports (GstRTSPMedia * media, GstRTSPMediaStream * stream)
849 GstStateChangeReturn ret;
850 GstElement *udpsrc0, *udpsrc1;
851 GstElement *udpsink0, *udpsink1;
852 gint tmp_rtp, tmp_rtcp;
854 gint rtpport, rtcpport, sockfd;
863 /* Start with random port */
867 host = "udp://[::0]";
869 host = "udp://0.0.0.0";
871 /* try to allocate 2 UDP ports, the RTP port should be an even
872 * number and the RTCP port should be the next (uneven) port */
874 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
876 goto no_udp_protocol;
877 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, NULL);
879 ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
880 if (ret == GST_STATE_CHANGE_FAILURE) {
886 gst_element_set_state (udpsrc0, GST_STATE_NULL);
887 gst_object_unref (udpsrc0);
891 goto no_udp_protocol;
894 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
896 /* check if port is even */
897 if ((tmp_rtp & 1) != 0) {
898 /* port not even, close and allocate another */
902 gst_element_set_state (udpsrc0, GST_STATE_NULL);
903 gst_object_unref (udpsrc0);
909 /* allocate port+1 for RTCP now */
910 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL);
912 goto no_udp_rtcp_protocol;
915 tmp_rtcp = tmp_rtp + 1;
916 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
918 ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
919 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
920 if (ret == GST_STATE_CHANGE_FAILURE) {
925 gst_element_set_state (udpsrc0, GST_STATE_NULL);
926 gst_object_unref (udpsrc0);
928 gst_element_set_state (udpsrc1, GST_STATE_NULL);
929 gst_object_unref (udpsrc1);
935 /* all fine, do port check */
936 g_object_get (G_OBJECT (udpsrc0), "port", &rtpport, NULL);
937 g_object_get (G_OBJECT (udpsrc1), "port", &rtcpport, NULL);
939 /* this should not happen... */
940 if (rtpport != tmp_rtp || rtcpport != tmp_rtcp)
943 udpsink0 = gst_element_factory_make ("multiudpsink", NULL);
945 goto no_udp_protocol;
947 g_object_get (G_OBJECT (udpsrc0), "sock", &sockfd, NULL);
948 g_object_set (G_OBJECT (udpsink0), "sockfd", sockfd, NULL);
949 g_object_set (G_OBJECT (udpsink0), "closefd", FALSE, NULL);
951 udpsink1 = gst_element_factory_make ("multiudpsink", NULL);
953 goto no_udp_protocol;
955 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
956 "send-duplicates")) {
957 g_object_set (G_OBJECT (udpsink0), "send-duplicates", FALSE, NULL);
958 g_object_set (G_OBJECT (udpsink1), "send-duplicates", FALSE, NULL);
961 ("old multiudpsink version found without send-duplicates property");
964 if (g_object_class_find_property (G_OBJECT_GET_CLASS (udpsink0),
966 g_object_set (G_OBJECT (udpsink0), "buffer-size", media->buffer_size, NULL);
968 GST_WARNING ("multiudpsink version found without buffer-size property");
971 g_object_get (G_OBJECT (udpsrc1), "sock", &sockfd, NULL);
972 g_object_set (G_OBJECT (udpsink1), "sockfd", sockfd, NULL);
973 g_object_set (G_OBJECT (udpsink1), "closefd", FALSE, NULL);
974 g_object_set (G_OBJECT (udpsink1), "sync", FALSE, NULL);
975 g_object_set (G_OBJECT (udpsink1), "async", FALSE, NULL);
977 g_object_set (G_OBJECT (udpsink0), "auto-multicast", FALSE, NULL);
978 g_object_set (G_OBJECT (udpsink0), "loop", FALSE, NULL);
979 g_object_set (G_OBJECT (udpsink1), "auto-multicast", FALSE, NULL);
980 g_object_set (G_OBJECT (udpsink1), "loop", FALSE, NULL);
982 /* we keep these elements, we configure all in configure_transport when the
983 * server told us to really use the UDP ports. */
984 stream->udpsrc[0] = udpsrc0;
985 stream->udpsrc[1] = udpsrc1;
986 stream->udpsink[0] = udpsink0;
987 stream->udpsink[1] = udpsink1;
988 stream->server_port.min = rtpport;
989 stream->server_port.max = rtcpport;
1002 no_udp_rtcp_protocol:
1013 gst_element_set_state (udpsrc0, GST_STATE_NULL);
1014 gst_object_unref (udpsrc0);
1017 gst_element_set_state (udpsrc1, GST_STATE_NULL);
1018 gst_object_unref (udpsrc1);
1021 gst_element_set_state (udpsink0, GST_STATE_NULL);
1022 gst_object_unref (udpsink0);
1025 gst_element_set_state (udpsink1, GST_STATE_NULL);
1026 gst_object_unref (udpsink1);
1032 /* executed from streaming thread */
1034 caps_notify (GstPad * pad, GParamSpec * unused, GstRTSPMediaStream * stream)
1037 GstCaps *newcaps, *oldcaps;
1039 if ((newcaps = GST_PAD_CAPS (pad)))
1040 gst_caps_ref (newcaps);
1042 oldcaps = stream->caps;
1043 stream->caps = newcaps;
1046 gst_caps_unref (oldcaps);
1048 capsstr = gst_caps_to_string (newcaps);
1049 GST_INFO ("stream %p received caps %p, %s", stream, newcaps, capsstr);
1054 dump_structure (const GstStructure * s)
1058 sstr = gst_structure_to_string (s);
1059 GST_INFO ("structure: %s", sstr);
1063 static GstRTSPMediaTrans *
1064 find_transport (GstRTSPMediaStream * stream, const gchar * rtcp_from)
1067 GstRTSPMediaTrans *result = NULL;
1072 if (rtcp_from == NULL)
1075 tmp = g_strrstr (rtcp_from, ":");
1079 port = atoi (tmp + 1);
1080 dest = g_strndup (rtcp_from, tmp - rtcp_from);
1082 GST_INFO ("finding %s:%d", dest, port);
1084 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1085 GstRTSPMediaTrans *trans = walk->data;
1088 min = trans->transport->client_port.min;
1089 max = trans->transport->client_port.max;
1091 if ((strcmp (trans->transport->destination, dest) == 0) && (min == port
1103 on_new_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1105 GstStructure *stats;
1106 GstRTSPMediaTrans *trans;
1108 GST_INFO ("%p: new source %p", stream, source);
1110 /* see if we have a stream to match with the origin of the RTCP packet */
1111 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1112 if (trans == NULL) {
1113 g_object_get (source, "stats", &stats, NULL);
1115 const gchar *rtcp_from;
1117 dump_structure (stats);
1119 rtcp_from = gst_structure_get_string (stats, "rtcp-from");
1120 if ((trans = find_transport (stream, rtcp_from))) {
1121 GST_INFO ("%p: found transport %p for source %p", stream, trans,
1124 /* keep ref to the source */
1125 trans->rtpsource = source;
1127 g_object_set_qdata (source, ssrc_stream_map_key, trans);
1129 gst_structure_free (stats);
1132 GST_INFO ("%p: source %p for transport %p", stream, source, trans);
1137 on_ssrc_sdes (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1139 GST_INFO ("%p: new SDES %p", stream, source);
1143 on_ssrc_active (GObject * session, GObject * source,
1144 GstRTSPMediaStream * stream)
1146 GstRTSPMediaTrans *trans;
1148 trans = g_object_get_qdata (source, ssrc_stream_map_key);
1150 GST_INFO ("%p: source %p in transport %p is active", stream, source, trans);
1152 if (trans && trans->keep_alive)
1153 trans->keep_alive (trans->ka_user_data);
1157 GstStructure *stats;
1158 g_object_get (source, "stats", &stats, NULL);
1160 dump_structure (stats);
1161 gst_structure_free (stats);
1168 on_bye_ssrc (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1170 GST_INFO ("%p: source %p bye", stream, source);
1174 on_bye_timeout (GObject * session, GObject * source,
1175 GstRTSPMediaStream * stream)
1177 GstRTSPMediaTrans *trans;
1179 GST_INFO ("%p: source %p bye timeout", stream, source);
1181 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1182 trans->rtpsource = NULL;
1183 trans->timeout = TRUE;
1188 on_timeout (GObject * session, GObject * source, GstRTSPMediaStream * stream)
1190 GstRTSPMediaTrans *trans;
1192 GST_INFO ("%p: source %p timeout", stream, source);
1194 if ((trans = g_object_get_qdata (source, ssrc_stream_map_key))) {
1195 trans->rtpsource = NULL;
1196 trans->timeout = TRUE;
1200 static GstFlowReturn
1201 handle_new_buffer (GstAppSink * sink, gpointer user_data)
1205 GstRTSPMediaStream *stream;
1207 buffer = gst_app_sink_pull_buffer (sink);
1211 stream = (GstRTSPMediaStream *) user_data;
1213 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1214 GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
1216 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
1218 tr->send_rtp (buffer, tr->transport->interleaved.min, tr->user_data);
1221 tr->send_rtcp (buffer, tr->transport->interleaved.max, tr->user_data);
1224 gst_buffer_unref (buffer);
1229 static GstFlowReturn
1230 handle_new_buffer_list (GstAppSink * sink, gpointer user_data)
1233 GstBufferList *blist;
1234 GstRTSPMediaStream *stream;
1236 blist = gst_app_sink_pull_buffer_list (sink);
1240 stream = (GstRTSPMediaStream *) user_data;
1242 for (walk = stream->transports; walk; walk = g_list_next (walk)) {
1243 GstRTSPMediaTrans *tr = (GstRTSPMediaTrans *) walk->data;
1245 if (GST_ELEMENT_CAST (sink) == stream->appsink[0]) {
1246 if (tr->send_rtp_list)
1247 tr->send_rtp_list (blist, tr->transport->interleaved.min,
1250 if (tr->send_rtcp_list)
1251 tr->send_rtcp_list (blist, tr->transport->interleaved.max,
1255 gst_buffer_list_unref (blist);
1260 static GstAppSinkCallbacks sink_cb = {
1261 NULL, /* not interested in EOS */
1262 NULL, /* not interested in preroll buffers */
1264 handle_new_buffer_list
1267 /* prepare the pipeline objects to handle @stream in @media */
1269 setup_stream (GstRTSPMediaStream * stream, guint idx, GstRTSPMedia * media)
1272 GstPad *pad, *teepad, *selpad;
1273 GstPadLinkReturn ret;
1276 /* allocate udp ports, we will have 4 of them, 2 for receiving RTP/RTCP and 2
1277 * for sending RTP/RTCP. The sender and receiver ports are shared between the
1279 if (!alloc_udp_ports (media, stream))
1282 /* add the ports to the pipeline */
1283 for (i = 0; i < 2; i++) {
1284 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsink[i]);
1285 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->udpsrc[i]);
1288 /* create elements for the TCP transfer */
1289 for (i = 0; i < 2; i++) {
1290 stream->appsrc[i] = gst_element_factory_make ("appsrc", NULL);
1291 stream->appsink[i] = gst_element_factory_make ("appsink", NULL);
1292 g_object_set (stream->appsink[i], "async", FALSE, "sync", FALSE, NULL);
1293 g_object_set (stream->appsink[i], "emit-signals", FALSE, NULL);
1294 g_object_set (stream->appsink[i], "preroll-queue-len", 1, NULL);
1295 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsink[i]);
1296 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->appsrc[i]);
1297 gst_app_sink_set_callbacks (GST_APP_SINK_CAST (stream->appsink[i]),
1298 &sink_cb, stream, NULL);
1301 /* hook up the stream to the RTP session elements. */
1302 name = g_strdup_printf ("send_rtp_sink_%d", idx);
1303 stream->send_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1305 name = g_strdup_printf ("send_rtp_src_%d", idx);
1306 stream->send_rtp_src = gst_element_get_static_pad (media->rtpbin, name);
1308 name = g_strdup_printf ("send_rtcp_src_%d", idx);
1309 stream->send_rtcp_src = gst_element_get_request_pad (media->rtpbin, name);
1311 name = g_strdup_printf ("recv_rtcp_sink_%d", idx);
1312 stream->recv_rtcp_sink = gst_element_get_request_pad (media->rtpbin, name);
1314 name = g_strdup_printf ("recv_rtp_sink_%d", idx);
1315 stream->recv_rtp_sink = gst_element_get_request_pad (media->rtpbin, name);
1318 /* get the session */
1319 g_signal_emit_by_name (media->rtpbin, "get-internal-session", idx,
1322 g_signal_connect (stream->session, "on-new-ssrc", (GCallback) on_new_ssrc,
1324 g_signal_connect (stream->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes,
1326 g_signal_connect (stream->session, "on-ssrc-active",
1327 (GCallback) on_ssrc_active, stream);
1328 g_signal_connect (stream->session, "on-bye-ssrc", (GCallback) on_bye_ssrc,
1330 g_signal_connect (stream->session, "on-bye-timeout",
1331 (GCallback) on_bye_timeout, stream);
1332 g_signal_connect (stream->session, "on-timeout", (GCallback) on_timeout,
1335 /* link the RTP pad to the session manager */
1336 ret = gst_pad_link (stream->srcpad, stream->send_rtp_sink);
1337 if (ret != GST_PAD_LINK_OK)
1340 /* make tee for RTP and link to stream */
1341 stream->tee[0] = gst_element_factory_make ("tee", NULL);
1342 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[0]);
1344 pad = gst_element_get_static_pad (stream->tee[0], "sink");
1345 gst_pad_link (stream->send_rtp_src, pad);
1346 gst_object_unref (pad);
1348 /* link RTP sink, we're pretty sure this will work. */
1349 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1350 pad = gst_element_get_static_pad (stream->udpsink[0], "sink");
1351 gst_pad_link (teepad, pad);
1352 gst_object_unref (pad);
1353 gst_object_unref (teepad);
1355 teepad = gst_element_get_request_pad (stream->tee[0], "src%d");
1356 pad = gst_element_get_static_pad (stream->appsink[0], "sink");
1357 gst_pad_link (teepad, pad);
1358 gst_object_unref (pad);
1359 gst_object_unref (teepad);
1361 /* make tee for RTCP */
1362 stream->tee[1] = gst_element_factory_make ("tee", NULL);
1363 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->tee[1]);
1365 pad = gst_element_get_static_pad (stream->tee[1], "sink");
1366 gst_pad_link (stream->send_rtcp_src, pad);
1367 gst_object_unref (pad);
1369 /* link RTCP elements */
1370 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1371 pad = gst_element_get_static_pad (stream->udpsink[1], "sink");
1372 gst_pad_link (teepad, pad);
1373 gst_object_unref (pad);
1374 gst_object_unref (teepad);
1376 teepad = gst_element_get_request_pad (stream->tee[1], "src%d");
1377 pad = gst_element_get_static_pad (stream->appsink[1], "sink");
1378 gst_pad_link (teepad, pad);
1379 gst_object_unref (pad);
1380 gst_object_unref (teepad);
1382 /* make selector for the RTP receivers */
1383 stream->selector[0] = gst_element_factory_make ("rtspfunnel", NULL);
1384 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[0]);
1386 pad = gst_element_get_static_pad (stream->selector[0], "src");
1387 gst_pad_link (pad, stream->recv_rtp_sink);
1388 gst_object_unref (pad);
1390 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1391 pad = gst_element_get_static_pad (stream->udpsrc[0], "src");
1392 gst_pad_link (pad, selpad);
1393 gst_object_unref (pad);
1394 gst_object_unref (selpad);
1396 selpad = gst_element_get_request_pad (stream->selector[0], "sink%d");
1397 pad = gst_element_get_static_pad (stream->appsrc[0], "src");
1398 gst_pad_link (pad, selpad);
1399 gst_object_unref (pad);
1400 gst_object_unref (selpad);
1402 /* make selector for the RTCP receivers */
1403 stream->selector[1] = gst_element_factory_make ("rtspfunnel", NULL);
1404 gst_bin_add (GST_BIN_CAST (media->pipeline), stream->selector[1]);
1406 pad = gst_element_get_static_pad (stream->selector[1], "src");
1407 gst_pad_link (pad, stream->recv_rtcp_sink);
1408 gst_object_unref (pad);
1410 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1411 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
1412 gst_pad_link (pad, selpad);
1413 gst_object_unref (pad);
1414 gst_object_unref (selpad);
1416 selpad = gst_element_get_request_pad (stream->selector[1], "sink%d");
1417 pad = gst_element_get_static_pad (stream->appsrc[1], "src");
1418 gst_pad_link (pad, selpad);
1419 gst_object_unref (pad);
1420 gst_object_unref (selpad);
1422 /* we set and keep these to playing so that they don't cause NO_PREROLL return
1424 gst_element_set_state (stream->udpsrc[0], GST_STATE_PLAYING);
1425 gst_element_set_state (stream->udpsrc[1], GST_STATE_PLAYING);
1426 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
1427 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
1429 /* be notified of caps changes */
1430 stream->caps_sig = g_signal_connect (stream->send_rtp_sink, "notify::caps",
1431 (GCallback) caps_notify, stream);
1433 stream->prepared = TRUE;
1440 GST_WARNING ("failed to link stream %d", idx);
1446 unlock_streams (GstRTSPMedia * media)
1450 /* unlock the udp src elements */
1451 n_streams = gst_rtsp_media_n_streams (media);
1452 for (i = 0; i < n_streams; i++) {
1453 GstRTSPMediaStream *stream;
1455 stream = gst_rtsp_media_get_stream (media, i);
1457 gst_element_set_locked_state (stream->udpsrc[0], FALSE);
1458 gst_element_set_locked_state (stream->udpsrc[1], FALSE);
1463 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1465 g_mutex_lock (media->lock);
1466 /* never overwrite the error status */
1467 if (media->status != GST_RTSP_MEDIA_STATUS_ERROR)
1468 media->status = status;
1469 GST_DEBUG ("setting new status to %d", status);
1470 g_cond_broadcast (media->cond);
1471 g_mutex_unlock (media->lock);
1474 static GstRTSPMediaStatus
1475 gst_rtsp_media_get_status (GstRTSPMedia * media)
1477 GstRTSPMediaStatus result;
1480 g_mutex_lock (media->lock);
1481 g_get_current_time (&timeout);
1482 g_time_val_add (&timeout, 20 * G_USEC_PER_SEC);
1483 /* while we are preparing, wait */
1484 while (media->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1485 GST_DEBUG ("waiting for status change");
1486 if (!g_cond_timed_wait (media->cond, media->lock, &timeout)) {
1487 GST_DEBUG ("timeout, assuming error status");
1488 media->status = GST_RTSP_MEDIA_STATUS_ERROR;
1491 /* could be success or error */
1492 result = media->status;
1493 GST_DEBUG ("got status %d", result);
1494 g_mutex_unlock (media->lock);
1500 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1502 GstMessageType type;
1504 type = GST_MESSAGE_TYPE (message);
1507 case GST_MESSAGE_STATE_CHANGED:
1509 case GST_MESSAGE_BUFFERING:
1513 gst_message_parse_buffering (message, &percent);
1515 /* no state management needed for live pipelines */
1519 if (percent == 100) {
1520 /* a 100% message means buffering is done */
1521 media->buffering = FALSE;
1522 /* if the desired state is playing, go back */
1523 if (media->target_state == GST_STATE_PLAYING) {
1524 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1525 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1527 GST_INFO ("Buffering done");
1530 /* buffering busy */
1531 if (media->buffering == FALSE) {
1532 if (media->target_state == GST_STATE_PLAYING) {
1533 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1534 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1535 gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1537 GST_INFO ("Buffering ...");
1540 media->buffering = TRUE;
1544 case GST_MESSAGE_LATENCY:
1546 gst_bin_recalculate_latency (GST_BIN_CAST (media->pipeline));
1549 case GST_MESSAGE_ERROR:
1554 gst_message_parse_error (message, &gerror, &debug);
1555 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1556 g_error_free (gerror);
1559 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1562 case GST_MESSAGE_WARNING:
1567 gst_message_parse_warning (message, &gerror, &debug);
1568 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1569 g_error_free (gerror);
1573 case GST_MESSAGE_ELEMENT:
1575 case GST_MESSAGE_STREAM_STATUS:
1577 case GST_MESSAGE_ASYNC_DONE:
1578 if (!media->adding) {
1579 /* when we are dynamically adding pads, the addition of the udpsrc will
1580 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1581 * wait for the final ASYNC_DONE after everything prerolled */
1582 GST_INFO ("%p: got ASYNC_DONE", media);
1583 collect_media_stats (media);
1585 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1587 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1590 case GST_MESSAGE_EOS:
1591 GST_INFO ("%p: got EOS", media);
1592 if (media->eos_pending) {
1593 GST_DEBUG ("shutting down after EOS");
1594 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1595 media->eos_pending = FALSE;
1596 g_object_unref (media);
1600 GST_INFO ("%p: got message type %s", media,
1601 gst_message_type_get_name (type));
1608 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1610 GstRTSPMediaClass *klass;
1613 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1615 if (klass->handle_message)
1616 ret = klass->handle_message (media, message);
1623 /* called from streaming threads */
1625 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1627 GstRTSPMediaStream *stream;
1631 i = media->streams->len + 1;
1633 GST_INFO ("pad added %s:%s, stream %d", GST_DEBUG_PAD_NAME (pad), i);
1635 stream = g_new0 (GstRTSPMediaStream, 1);
1636 stream->payloader = element;
1638 name = g_strdup_printf ("dynpay%d", i);
1640 media->adding = TRUE;
1642 /* ghost the pad of the payloader to the element */
1643 stream->srcpad = gst_ghost_pad_new (name, pad);
1644 gst_pad_set_active (stream->srcpad, TRUE);
1645 gst_element_add_pad (media->element, stream->srcpad);
1648 /* add stream now */
1649 g_array_append_val (media->streams, stream);
1651 setup_stream (stream, i, media);
1653 for (i = 0; i < 2; i++) {
1654 gst_element_set_state (stream->udpsink[i], GST_STATE_PAUSED);
1655 gst_element_set_state (stream->appsink[i], GST_STATE_PAUSED);
1656 gst_element_set_state (stream->tee[i], GST_STATE_PAUSED);
1657 gst_element_set_state (stream->selector[i], GST_STATE_PAUSED);
1658 gst_element_set_state (stream->appsrc[i], GST_STATE_PAUSED);
1660 media->adding = FALSE;
1664 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1666 GST_INFO ("no more pads");
1667 if (media->fakesink) {
1668 gst_object_ref (media->fakesink);
1669 gst_bin_remove (GST_BIN (media->pipeline), media->fakesink);
1670 gst_element_set_state (media->fakesink, GST_STATE_NULL);
1671 gst_object_unref (media->fakesink);
1672 media->fakesink = NULL;
1673 GST_INFO ("removed fakesink");
1678 * gst_rtsp_media_prepare:
1679 * @media: a #GstRTSPMedia
1681 * Prepare @media for streaming. This function will create the pipeline and
1682 * other objects to manage the streaming.
1684 * It will preroll the pipeline and collect vital information about the streams
1685 * such as the duration.
1687 * Returns: %TRUE on success.
1690 gst_rtsp_media_prepare (GstRTSPMedia * media)
1692 GstStateChangeReturn ret;
1693 GstRTSPMediaStatus status;
1695 GstRTSPMediaClass *klass;
1699 if (media->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1702 if (!media->reusable && media->reused)
1705 GST_INFO ("preparing media %p", media);
1707 /* reset some variables */
1708 media->is_live = FALSE;
1709 media->buffering = FALSE;
1710 /* we're preparing now */
1711 media->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1713 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (media->pipeline));
1715 /* add the pipeline bus to our custom mainloop */
1716 media->source = gst_bus_create_watch (bus);
1717 gst_object_unref (bus);
1719 g_source_set_callback (media->source, (GSourceFunc) bus_message, media, NULL);
1721 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1722 media->id = g_source_attach (media->source, klass->context);
1724 media->rtpbin = gst_element_factory_make ("gstrtpbin", NULL);
1726 /* add stuff to the bin */
1727 gst_bin_add (GST_BIN (media->pipeline), media->rtpbin);
1729 /* link streams we already have, other streams might appear when we have
1730 * dynamic elements */
1731 n_streams = gst_rtsp_media_n_streams (media);
1732 for (i = 0; i < n_streams; i++) {
1733 GstRTSPMediaStream *stream;
1735 stream = gst_rtsp_media_get_stream (media, i);
1737 setup_stream (stream, i, media);
1740 for (walk = media->dynamic; walk; walk = g_list_next (walk)) {
1741 GstElement *elem = walk->data;
1743 GST_INFO ("adding callbacks for dynamic element %p", elem);
1745 g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
1746 g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
1748 /* we add a fakesink here in order to make the state change async. We remove
1749 * the fakesink again in the no-more-pads callback. */
1750 media->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1751 gst_bin_add (GST_BIN (media->pipeline), media->fakesink);
1754 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1755 /* first go to PAUSED */
1756 ret = gst_element_set_state (media->pipeline, GST_STATE_PAUSED);
1757 media->target_state = GST_STATE_PAUSED;
1760 case GST_STATE_CHANGE_SUCCESS:
1761 GST_INFO ("SUCCESS state change for media %p", media);
1763 case GST_STATE_CHANGE_ASYNC:
1764 GST_INFO ("ASYNC state change for media %p", media);
1766 case GST_STATE_CHANGE_NO_PREROLL:
1767 /* we need to go to PLAYING */
1768 GST_INFO ("NO_PREROLL state change: live media %p", media);
1769 media->is_live = TRUE;
1770 ret = gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1771 if (ret == GST_STATE_CHANGE_FAILURE)
1774 case GST_STATE_CHANGE_FAILURE:
1778 /* now wait for all pads to be prerolled */
1779 status = gst_rtsp_media_get_status (media);
1780 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1783 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1785 GST_INFO ("object %p is prerolled", media);
1797 GST_WARNING ("can not reuse media %p", media);
1802 GST_WARNING ("failed to preroll pipeline");
1803 unlock_streams (media);
1804 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1805 gst_rtsp_media_unprepare (media);
1811 * gst_rtsp_media_unprepare:
1812 * @media: a #GstRTSPMedia
1814 * Unprepare @media. After this call, the media should be prepared again before
1815 * it can be used again. If the media is set to be non-reusable, a new instance
1818 * Returns: %TRUE on success.
1821 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1823 GstRTSPMediaClass *klass;
1826 if (media->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1829 GST_INFO ("unprepare media %p", media);
1830 media->target_state = GST_STATE_NULL;
1832 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1833 if (klass->unprepare)
1834 success = klass->unprepare (media);
1838 media->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1839 media->reused = TRUE;
1841 /* when the media is not reusable, this will effectively unref the media and
1843 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1849 default_unprepare (GstRTSPMedia * media)
1851 if (media->eos_shutdown) {
1852 GST_DEBUG ("sending EOS for shutdown");
1853 /* ref so that we don't disappear */
1854 g_object_ref (media);
1855 media->eos_pending = TRUE;
1856 gst_element_send_event (media->pipeline, gst_event_new_eos ());
1857 /* we need to go to playing again for the EOS to propagate, normally in this
1858 * state, nothing is receiving data from us anymore so this is ok. */
1859 gst_element_set_state (media->pipeline, GST_STATE_PLAYING);
1861 GST_DEBUG ("shutting down");
1862 gst_element_set_state (media->pipeline, GST_STATE_NULL);
1868 add_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1869 gchar * dest, gint min, gint max)
1871 GST_INFO ("adding %s:%d-%d", dest, min, max);
1872 g_signal_emit_by_name (stream->udpsink[0], "add", dest, min, NULL);
1873 g_signal_emit_by_name (stream->udpsink[1], "add", dest, max, NULL);
1877 remove_udp_destination (GstRTSPMedia * media, GstRTSPMediaStream * stream,
1878 gchar * dest, gint min, gint max)
1880 GST_INFO ("removing %s:%d-%d", dest, min, max);
1881 g_signal_emit_by_name (stream->udpsink[0], "remove", dest, min, NULL);
1882 g_signal_emit_by_name (stream->udpsink[1], "remove", dest, max, NULL);
1886 * gst_rtsp_media_set_state:
1887 * @media: a #GstRTSPMedia
1888 * @state: the target state of the media
1889 * @transports: a #GArray of #GstRTSPMediaTrans pointers
1891 * Set the state of @media to @state and for the transports in @transports.
1893 * Returns: %TRUE on success.
1896 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1897 GArray * transports)
1900 gboolean add, remove, do_state;
1903 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1904 g_return_val_if_fail (transports != NULL, FALSE);
1906 /* NULL and READY are the same */
1907 if (state == GST_STATE_READY)
1908 state = GST_STATE_NULL;
1910 add = remove = FALSE;
1912 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1916 case GST_STATE_NULL:
1917 /* unlock the streams so that they follow the state changes from now on */
1918 unlock_streams (media);
1920 case GST_STATE_PAUSED:
1921 /* we're going from PLAYING to PAUSED, READY or NULL, remove */
1922 if (media->target_state == GST_STATE_PLAYING)
1925 case GST_STATE_PLAYING:
1926 /* we're going to PLAYING, add */
1932 old_active = media->active;
1934 for (i = 0; i < transports->len; i++) {
1935 GstRTSPMediaTrans *tr;
1936 GstRTSPMediaStream *stream;
1937 GstRTSPTransport *trans;
1939 /* we need a non-NULL entry in the array */
1940 tr = g_array_index (transports, GstRTSPMediaTrans *, i);
1944 /* we need a transport */
1945 if (!(trans = tr->transport))
1948 /* get the stream and add the destinations */
1949 stream = gst_rtsp_media_get_stream (media, tr->idx);
1950 switch (trans->lower_transport) {
1951 case GST_RTSP_LOWER_TRANS_UDP:
1952 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
1957 dest = trans->destination;
1958 if (trans->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
1959 min = trans->port.min;
1960 max = trans->port.max;
1962 min = trans->client_port.min;
1963 max = trans->client_port.max;
1966 if (add && !tr->active) {
1967 add_udp_destination (media, stream, dest, min, max);
1968 stream->transports = g_list_prepend (stream->transports, tr);
1971 } else if (remove && tr->active) {
1972 remove_udp_destination (media, stream, dest, min, max);
1973 stream->transports = g_list_remove (stream->transports, tr);
1979 case GST_RTSP_LOWER_TRANS_TCP:
1980 if (add && !tr->active) {
1981 GST_INFO ("adding TCP %s", trans->destination);
1982 stream->transports = g_list_prepend (stream->transports, tr);
1985 } else if (remove && tr->active) {
1986 GST_INFO ("removing TCP %s", trans->destination);
1987 stream->transports = g_list_remove (stream->transports, tr);
1993 GST_INFO ("Unknown transport %d", trans->lower_transport);
1998 /* we just added the first media, do the playing state change */
1999 if (old_active == 0 && add)
2001 /* if we have no more active media, do the downward state changes */
2002 else if (media->active == 0)
2007 GST_INFO ("state %d active %d media %p do_state %d", state, media->active,
2010 if (media->target_state != state) {
2012 if (state == GST_STATE_NULL) {
2013 gst_rtsp_media_unprepare (media);
2015 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2017 media->target_state = state;
2018 gst_element_set_state (media->pipeline, state);
2021 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2025 /* remember where we are */
2026 if (state == GST_STATE_PAUSED || old_active != media->active)
2027 collect_media_stats (media);
2033 * gst_rtsp_media_remove_elements:
2034 * @media: a #GstRTSPMedia
2036 * Remove all elements and the pipeline controlled by @media.
2039 gst_rtsp_media_remove_elements (GstRTSPMedia * media)
2043 unlock_streams (media);
2045 for (i = 0; i < media->streams->len; i++) {
2046 GstRTSPMediaStream *stream;
2048 GST_INFO ("Removing elements of stream %d from pipeline", i);
2050 stream = g_array_index (media->streams, GstRTSPMediaStream *, i);
2052 gst_pad_unlink (stream->srcpad, stream->send_rtp_sink);
2054 g_signal_handler_disconnect (stream->send_rtp_sink, stream->caps_sig);
2056 for (j = 0; j < 2; j++) {
2057 gst_element_set_state (stream->udpsrc[j], GST_STATE_NULL);
2058 gst_element_set_state (stream->udpsink[j], GST_STATE_NULL);
2059 gst_element_set_state (stream->appsrc[j], GST_STATE_NULL);
2060 gst_element_set_state (stream->appsink[j], GST_STATE_NULL);
2061 gst_element_set_state (stream->tee[j], GST_STATE_NULL);
2062 gst_element_set_state (stream->selector[j], GST_STATE_NULL);
2064 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsrc[j]);
2065 gst_bin_remove (GST_BIN (media->pipeline), stream->udpsink[j]);
2066 gst_bin_remove (GST_BIN (media->pipeline), stream->appsrc[j]);
2067 gst_bin_remove (GST_BIN (media->pipeline), stream->appsink[j]);
2068 gst_bin_remove (GST_BIN (media->pipeline), stream->tee[j]);
2069 gst_bin_remove (GST_BIN (media->pipeline), stream->selector[j]);
2072 gst_caps_unref (stream->caps);
2073 stream->caps = NULL;
2074 gst_rtsp_media_stream_free (stream);
2076 g_array_remove_range (media->streams, 0, media->streams->len);
2078 gst_element_set_state (media->rtpbin, GST_STATE_NULL);
2079 gst_bin_remove (GST_BIN (media->pipeline), media->rtpbin);
2081 gst_object_unref (media->pipeline);
2082 media->pipeline = NULL;