2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
23 #include <gst/app/gstappsrc.h>
24 #include <gst/app/gstappsink.h>
26 #include "rtsp-media.h"
28 #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
29 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
31 struct _GstRTSPMediaPrivate
38 GstRTSPLowerTrans protocols;
40 gboolean eos_shutdown;
43 GstRTSPAddressPool *pool;
49 GstRTSPMediaStatus status;
53 /* the pipeline for the media */
62 GstState target_state;
64 /* RTP session manager */
67 /* the range of media */
68 GstRTSPTimeRange range;
69 GstClockTime range_start;
70 GstClockTime range_stop;
73 #define DEFAULT_SHARED FALSE
74 #define DEFAULT_REUSABLE FALSE
75 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
76 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
77 #define DEFAULT_EOS_SHUTDOWN FALSE
78 #define DEFAULT_BUFFER_SIZE 0x80000
80 /* define to dump received RTCP packets */
103 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
104 #define GST_CAT_DEFAULT rtsp_media_debug
106 static void gst_rtsp_media_get_property (GObject * object, guint propid,
107 GValue * value, GParamSpec * pspec);
108 static void gst_rtsp_media_set_property (GObject * object, guint propid,
109 const GValue * value, GParamSpec * pspec);
110 static void gst_rtsp_media_finalize (GObject * obj);
112 static gpointer do_loop (GstRTSPMediaClass * klass);
113 static gboolean default_handle_message (GstRTSPMedia * media,
114 GstMessage * message);
115 static void finish_unprepare (GstRTSPMedia * media);
116 static gboolean default_unprepare (GstRTSPMedia * media);
118 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
120 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
123 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
125 GObjectClass *gobject_class;
127 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
129 gobject_class = G_OBJECT_CLASS (klass);
131 gobject_class->get_property = gst_rtsp_media_get_property;
132 gobject_class->set_property = gst_rtsp_media_set_property;
133 gobject_class->finalize = gst_rtsp_media_finalize;
135 g_object_class_install_property (gobject_class, PROP_SHARED,
136 g_param_spec_boolean ("shared", "Shared",
137 "If this media pipeline can be shared", DEFAULT_SHARED,
138 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
140 g_object_class_install_property (gobject_class, PROP_REUSABLE,
141 g_param_spec_boolean ("reusable", "Reusable",
142 "If this media pipeline can be reused after an unprepare",
143 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
145 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
146 g_param_spec_flags ("protocols", "Protocols",
147 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
148 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
150 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
151 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
152 "Send an EOS event to the pipeline before unpreparing",
153 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
155 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
156 g_param_spec_uint ("buffer-size", "Buffer Size",
157 "The kernel UDP buffer size to use", 0, G_MAXUINT,
158 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
160 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
161 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
162 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
163 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
165 gst_rtsp_media_signals[SIGNAL_PREPARED] =
166 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
167 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
168 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
170 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
171 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
172 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
173 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
175 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
176 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
177 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
178 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
180 klass->context = g_main_context_new ();
181 klass->loop = g_main_loop_new (klass->context, TRUE);
183 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
185 klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
187 klass->handle_message = default_handle_message;
188 klass->unprepare = default_unprepare;
192 gst_rtsp_media_init (GstRTSPMedia * media)
194 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
198 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
199 g_mutex_init (&priv->lock);
200 g_cond_init (&priv->cond);
201 g_rec_mutex_init (&priv->state_lock);
203 priv->shared = DEFAULT_SHARED;
204 priv->reusable = DEFAULT_REUSABLE;
205 priv->protocols = DEFAULT_PROTOCOLS;
206 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
207 priv->buffer_size = DEFAULT_BUFFER_SIZE;
211 gst_rtsp_media_finalize (GObject * obj)
213 GstRTSPMediaPrivate *priv;
216 media = GST_RTSP_MEDIA (obj);
219 GST_INFO ("finalize media %p", media);
221 gst_rtsp_media_unprepare (media);
223 g_ptr_array_unref (priv->streams);
225 g_list_free_full (priv->dynamic, gst_object_unref);
228 gst_object_unref (priv->pipeline);
229 gst_object_unref (priv->element);
231 g_object_unref (priv->auth);
233 g_object_unref (priv->pool);
234 g_mutex_clear (&priv->lock);
235 g_cond_clear (&priv->cond);
236 g_rec_mutex_clear (&priv->state_lock);
238 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
242 gst_rtsp_media_get_property (GObject * object, guint propid,
243 GValue * value, GParamSpec * pspec)
245 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
249 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
252 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
255 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
257 case PROP_EOS_SHUTDOWN:
258 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
260 case PROP_BUFFER_SIZE:
261 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
264 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
269 gst_rtsp_media_set_property (GObject * object, guint propid,
270 const GValue * value, GParamSpec * pspec)
272 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
276 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
279 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
282 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
284 case PROP_EOS_SHUTDOWN:
285 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
287 case PROP_BUFFER_SIZE:
288 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
291 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
296 do_loop (GstRTSPMediaClass * klass)
298 GST_INFO ("enter mainloop");
299 g_main_loop_run (klass->loop);
300 GST_INFO ("exit mainloop");
305 /* must be called with state lock */
307 collect_media_stats (GstRTSPMedia * media)
309 GstRTSPMediaPrivate *priv = media->priv;
310 gint64 position, duration;
312 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
313 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
316 priv->range.unit = GST_RTSP_RANGE_NPT;
318 GST_INFO ("collect media stats");
321 priv->range.min.type = GST_RTSP_TIME_NOW;
322 priv->range.min.seconds = -1;
323 priv->range_start = -1;
324 priv->range.max.type = GST_RTSP_TIME_END;
325 priv->range.max.seconds = -1;
326 priv->range_stop = -1;
328 /* get the position */
329 if (!gst_element_query_position (priv->pipeline, GST_FORMAT_TIME,
331 GST_INFO ("position query failed");
335 /* get the duration */
336 if (!gst_element_query_duration (priv->pipeline, GST_FORMAT_TIME,
338 GST_INFO ("duration query failed");
342 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
343 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
345 if (position == -1) {
346 priv->range.min.type = GST_RTSP_TIME_NOW;
347 priv->range.min.seconds = -1;
348 priv->range_start = -1;
350 priv->range.min.type = GST_RTSP_TIME_SECONDS;
351 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
352 priv->range_start = position;
354 if (duration == -1) {
355 priv->range.max.type = GST_RTSP_TIME_END;
356 priv->range.max.seconds = -1;
357 priv->range_stop = -1;
359 priv->range.max.type = GST_RTSP_TIME_SECONDS;
360 priv->range.max.seconds = ((gdouble) duration) / GST_SECOND;
361 priv->range_stop = duration;
367 * gst_rtsp_media_new:
368 * @element: (transfer full): a #GstElement
370 * Create a new #GstRTSPMedia instance. @element is the bin element that
371 * provides the different streams. The #GstRTSPMedia object contains the
372 * element to produce RTP data for one or more related (audio/video/..)
375 * Ownership is taken of @element.
377 * Returns: a new #GstRTSPMedia object.
380 gst_rtsp_media_new (GstElement * element)
382 GstRTSPMedia *result;
384 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
386 result = g_object_new (GST_TYPE_RTSP_MEDIA, NULL);
387 result->priv->element = element;
393 * gst_rtsp_media_take_element:
394 * @media: a #GstRTSPMedia
395 * @pipeline: (transfer full): a #GstPipeline
397 * Set @pipeline as the #GstPipeline for @media. Ownership is
398 * taken of @pipeline.
401 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
403 GstRTSPMediaPrivate *priv;
406 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
407 g_return_if_fail (GST_IS_PIPELINE (pipeline));
411 g_mutex_lock (&priv->lock);
412 old = priv->pipeline;
413 priv->pipeline = GST_ELEMENT_CAST (pipeline);
414 g_mutex_unlock (&priv->lock);
417 gst_object_unref (old);
419 gst_object_ref (priv->element);
420 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
424 * gst_rtsp_media_set_shared:
425 * @media: a #GstRTSPMedia
426 * @shared: the new value
428 * Set or unset if the pipeline for @media can be shared will multiple clients.
429 * When @shared is %TRUE, client requests for this media will share the media
433 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
435 GstRTSPMediaPrivate *priv;
437 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
441 g_mutex_lock (&priv->lock);
442 priv->shared = shared;
443 g_mutex_unlock (&priv->lock);
447 * gst_rtsp_media_is_shared:
448 * @media: a #GstRTSPMedia
450 * Check if the pipeline for @media can be shared between multiple clients.
452 * Returns: %TRUE if the media can be shared between clients.
455 gst_rtsp_media_is_shared (GstRTSPMedia * media)
457 GstRTSPMediaPrivate *priv;
460 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
464 g_mutex_lock (&priv->lock);
466 g_mutex_unlock (&priv->lock);
472 * gst_rtsp_media_set_reusable:
473 * @media: a #GstRTSPMedia
474 * @reusable: the new value
476 * Set or unset if the pipeline for @media can be reused after the pipeline has
480 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
482 GstRTSPMediaPrivate *priv;
484 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
488 g_mutex_lock (&priv->lock);
489 priv->reusable = reusable;
490 g_mutex_unlock (&priv->lock);
494 * gst_rtsp_media_is_reusable:
495 * @media: a #GstRTSPMedia
497 * Check if the pipeline for @media can be reused after an unprepare.
499 * Returns: %TRUE if the media can be reused
502 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
504 GstRTSPMediaPrivate *priv;
507 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
511 g_mutex_lock (&priv->lock);
512 res = priv->reusable;
513 g_mutex_unlock (&priv->lock);
519 * gst_rtsp_media_set_protocols:
520 * @media: a #GstRTSPMedia
521 * @protocols: the new flags
523 * Configure the allowed lower transport for @media.
526 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
528 GstRTSPMediaPrivate *priv;
530 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
534 g_mutex_lock (&priv->lock);
535 priv->protocols = protocols;
536 g_mutex_unlock (&priv->lock);
540 * gst_rtsp_media_get_protocols:
541 * @media: a #GstRTSPMedia
543 * Get the allowed protocols of @media.
545 * Returns: a #GstRTSPLowerTrans
548 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
550 GstRTSPMediaPrivate *priv;
551 GstRTSPLowerTrans res;
553 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
554 GST_RTSP_LOWER_TRANS_UNKNOWN);
558 g_mutex_lock (&priv->lock);
559 res = priv->protocols;
560 g_mutex_unlock (&priv->lock);
566 * gst_rtsp_media_set_eos_shutdown:
567 * @media: a #GstRTSPMedia
568 * @eos_shutdown: the new value
570 * Set or unset if an EOS event will be sent to the pipeline for @media before
574 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
576 GstRTSPMediaPrivate *priv;
578 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
582 g_mutex_lock (&priv->lock);
583 priv->eos_shutdown = eos_shutdown;
584 g_mutex_unlock (&priv->lock);
588 * gst_rtsp_media_is_eos_shutdown:
589 * @media: a #GstRTSPMedia
591 * Check if the pipeline for @media will send an EOS down the pipeline before
594 * Returns: %TRUE if the media will send EOS before unpreparing.
597 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
599 GstRTSPMediaPrivate *priv;
602 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
606 g_mutex_lock (&priv->lock);
607 res = priv->eos_shutdown;
608 g_mutex_unlock (&priv->lock);
614 * gst_rtsp_media_set_buffer_size:
615 * @media: a #GstRTSPMedia
616 * @size: the new value
618 * Set the kernel UDP buffer size.
621 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
623 GstRTSPMediaPrivate *priv;
625 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
627 GST_LOG_OBJECT (media, "set buffer size %u", size);
631 g_mutex_lock (&priv->lock);
632 priv->buffer_size = size;
633 g_mutex_unlock (&priv->lock);
637 * gst_rtsp_media_get_buffer_size:
638 * @media: a #GstRTSPMedia
640 * Get the kernel UDP buffer size.
642 * Returns: the kernel UDP buffer size.
645 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
647 GstRTSPMediaPrivate *priv;
650 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
654 g_mutex_unlock (&priv->lock);
655 res = priv->buffer_size;
656 g_mutex_unlock (&priv->lock);
662 * gst_rtsp_media_set_auth:
663 * @media: a #GstRTSPMedia
664 * @auth: a #GstRTSPAuth
666 * configure @auth to be used as the authentication manager of @media.
669 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
671 GstRTSPMediaPrivate *priv;
674 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
678 GST_LOG_OBJECT (media, "set auth %p", auth);
680 g_mutex_lock (&priv->lock);
681 if ((old = priv->auth) != auth)
682 priv->auth = auth ? g_object_ref (auth) : NULL;
685 g_mutex_unlock (&priv->lock);
688 g_object_unref (old);
692 * gst_rtsp_media_get_auth:
693 * @media: a #GstRTSPMedia
695 * Get the #GstRTSPAuth used as the authentication manager of @media.
697 * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
701 gst_rtsp_media_get_auth (GstRTSPMedia * media)
703 GstRTSPMediaPrivate *priv;
706 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
710 g_mutex_lock (&priv->lock);
711 if ((result = priv->auth))
712 g_object_ref (result);
713 g_mutex_unlock (&priv->lock);
719 * gst_rtsp_media_set_address_pool:
720 * @media: a #GstRTSPMedia
721 * @pool: a #GstRTSPAddressPool
723 * configure @pool to be used as the address pool of @media.
726 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
727 GstRTSPAddressPool * pool)
729 GstRTSPMediaPrivate *priv;
730 GstRTSPAddressPool *old;
732 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
736 GST_LOG_OBJECT (media, "set address pool %p", pool);
738 g_mutex_lock (&priv->lock);
739 if ((old = priv->pool) != pool)
740 priv->pool = pool ? g_object_ref (pool) : NULL;
743 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
745 g_mutex_unlock (&priv->lock);
748 g_object_unref (old);
752 * gst_rtsp_media_get_address_pool:
753 * @media: a #GstRTSPMedia
755 * Get the #GstRTSPAddressPool used as the address pool of @media.
757 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
761 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
763 GstRTSPMediaPrivate *priv;
764 GstRTSPAddressPool *result;
766 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
770 g_mutex_lock (&priv->lock);
771 if ((result = priv->pool))
772 g_object_ref (result);
773 g_mutex_unlock (&priv->lock);
779 * gst_rtsp_media_collect_streams:
780 * @media: a #GstRTSPMedia
782 * Find all payloader elements, they should be named pay%d in the
783 * element of @media, and create #GstRTSPStreams for them.
785 * Collect all dynamic elements, named dynpay%d, and add them to
786 * the list of dynamic elements.
789 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
791 GstRTSPMediaPrivate *priv;
792 GstElement *element, *elem;
797 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
800 element = priv->element;
803 for (i = 0; have_elem; i++) {
808 name = g_strdup_printf ("pay%d", i);
809 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
810 GST_INFO ("found stream %d with payloader %p", i, elem);
812 /* take the pad of the payloader */
813 pad = gst_element_get_static_pad (elem, "src");
814 /* create the stream */
815 gst_rtsp_media_create_stream (media, elem, pad);
816 gst_object_unref (pad);
817 gst_object_unref (elem);
823 name = g_strdup_printf ("dynpay%d", i);
824 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
825 /* a stream that will dynamically create pads to provide RTP packets */
827 GST_INFO ("found dynamic element %d, %p", i, elem);
829 g_mutex_lock (&priv->lock);
830 priv->dynamic = g_list_prepend (priv->dynamic, elem);
831 g_mutex_unlock (&priv->lock);
840 * gst_rtsp_media_create_stream:
841 * @media: a #GstRTSPMedia
842 * @payloader: a #GstElement
843 * @srcpad: a source #GstPad
845 * Create a new stream in @media that provides RTP data on @srcpad.
846 * @srcpad should be a pad of an element inside @media->element.
848 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
852 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
855 GstRTSPMediaPrivate *priv;
856 GstRTSPStream *stream;
861 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
862 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
863 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
864 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
868 g_mutex_lock (&priv->lock);
869 idx = priv->streams->len;
871 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
873 name = g_strdup_printf ("src_%u", idx);
874 srcpad = gst_ghost_pad_new (name, pad);
875 gst_pad_set_active (srcpad, TRUE);
876 gst_element_add_pad (priv->element, srcpad);
879 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
881 gst_rtsp_stream_set_address_pool (stream, priv->pool);
883 g_ptr_array_add (priv->streams, stream);
884 g_mutex_unlock (&priv->lock);
886 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
893 * gst_rtsp_media_n_streams:
894 * @media: a #GstRTSPMedia
896 * Get the number of streams in this media.
898 * Returns: The number of streams.
901 gst_rtsp_media_n_streams (GstRTSPMedia * media)
903 GstRTSPMediaPrivate *priv;
906 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
910 g_mutex_lock (&priv->lock);
911 res = priv->streams->len;
912 g_mutex_unlock (&priv->lock);
918 * gst_rtsp_media_get_stream:
919 * @media: a #GstRTSPMedia
920 * @idx: the stream index
922 * Retrieve the stream with index @idx from @media.
924 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
925 * that index did not exist.
928 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
930 GstRTSPMediaPrivate *priv;
933 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
937 g_mutex_lock (&priv->lock);
938 if (idx < priv->streams->len)
939 res = g_ptr_array_index (priv->streams, idx);
942 g_mutex_unlock (&priv->lock);
948 * gst_rtsp_media_get_range_string:
949 * @media: a #GstRTSPMedia
950 * @play: for the PLAY request
952 * Get the current range as a string. @media must be prepared with
953 * gst_rtsp_media_prepare ().
955 * Returns: The range as a string, g_free() after usage.
958 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play)
960 GstRTSPMediaPrivate *priv;
962 GstRTSPTimeRange range;
964 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
968 g_rec_mutex_lock (&priv->state_lock);
969 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
972 g_mutex_lock (&priv->lock);
976 if (!play && priv->n_active > 0) {
977 range.min.type = GST_RTSP_TIME_NOW;
978 range.min.seconds = -1;
980 g_mutex_unlock (&priv->lock);
981 g_rec_mutex_unlock (&priv->state_lock);
983 result = gst_rtsp_range_to_string (&range);
990 GST_WARNING ("media %p was not prepared", media);
991 g_rec_mutex_unlock (&priv->state_lock);
997 * gst_rtsp_media_seek:
998 * @media: a #GstRTSPMedia
999 * @range: a #GstRTSPTimeRange
1001 * Seek the pipeline of @media to @range. @media must be prepared with
1002 * gst_rtsp_media_prepare().
1004 * Returns: %TRUE on success.
1007 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1009 GstRTSPMediaPrivate *priv;
1012 GstClockTime start, stop;
1013 GstSeekType start_type, stop_type;
1015 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1016 g_return_val_if_fail (range != NULL, FALSE);
1020 g_rec_mutex_lock (&priv->state_lock);
1021 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1024 if (!priv->seekable)
1027 /* depends on the current playing state of the pipeline. We might need to
1028 * queue this until we get EOS. */
1029 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
1031 start_type = stop_type = GST_SEEK_TYPE_NONE;
1033 if (!gst_rtsp_range_get_times (range, &start, &stop))
1036 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1037 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1038 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1039 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1041 if (priv->range_start == start)
1042 start = GST_CLOCK_TIME_NONE;
1043 else if (start != GST_CLOCK_TIME_NONE)
1044 start_type = GST_SEEK_TYPE_SET;
1046 if (priv->range_stop == stop)
1047 stop = GST_CLOCK_TIME_NONE;
1048 else if (stop != GST_CLOCK_TIME_NONE)
1049 stop_type = GST_SEEK_TYPE_SET;
1051 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1052 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1053 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1055 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1056 flags, start_type, start, stop_type, stop);
1058 /* and block for the seek to complete */
1059 GST_INFO ("done seeking %d", res);
1060 gst_element_get_state (priv->pipeline, NULL, NULL, -1);
1061 GST_INFO ("prerolled again");
1063 collect_media_stats (media);
1065 GST_INFO ("no seek needed");
1068 g_rec_mutex_unlock (&priv->state_lock);
1075 g_rec_mutex_unlock (&priv->state_lock);
1076 GST_INFO ("media %p is not prepared", media);
1081 g_rec_mutex_unlock (&priv->state_lock);
1082 GST_INFO ("pipeline is not seekable");
1087 g_rec_mutex_unlock (&priv->state_lock);
1088 GST_WARNING ("seek unit %d not supported", range->unit);
1094 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1096 GstRTSPMediaPrivate *priv = media->priv;
1098 g_mutex_lock (&priv->lock);
1099 priv->status = status;
1100 GST_DEBUG ("setting new status to %d", status);
1101 g_cond_broadcast (&priv->cond);
1102 g_mutex_unlock (&priv->lock);
1106 * gst_rtsp_media_get_status:
1107 * @media: a #GstRTSPMedia
1109 * Get the status of @media. When @media is busy preparing, this function waits
1110 * until @media is prepared or in error.
1112 * Returns: the status of @media.
1115 gst_rtsp_media_get_status (GstRTSPMedia * media)
1117 GstRTSPMediaPrivate *priv = media->priv;
1118 GstRTSPMediaStatus result;
1121 g_mutex_lock (&priv->lock);
1122 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1123 /* while we are preparing, wait */
1124 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1125 GST_DEBUG ("waiting for status change");
1126 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1127 GST_DEBUG ("timeout, assuming error status");
1128 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1131 /* could be success or error */
1132 result = priv->status;
1133 GST_DEBUG ("got status %d", result);
1134 g_mutex_unlock (&priv->lock);
1139 /* called with state-lock */
1141 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1143 GstRTSPMediaPrivate *priv = media->priv;
1144 GstMessageType type;
1146 type = GST_MESSAGE_TYPE (message);
1149 case GST_MESSAGE_STATE_CHANGED:
1151 case GST_MESSAGE_BUFFERING:
1155 gst_message_parse_buffering (message, &percent);
1157 /* no state management needed for live pipelines */
1161 if (percent == 100) {
1162 /* a 100% message means buffering is done */
1163 priv->buffering = FALSE;
1164 /* if the desired state is playing, go back */
1165 if (priv->target_state == GST_STATE_PLAYING) {
1166 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1167 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1169 GST_INFO ("Buffering done");
1172 /* buffering busy */
1173 if (priv->buffering == FALSE) {
1174 if (priv->target_state == GST_STATE_PLAYING) {
1175 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1176 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1177 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1179 GST_INFO ("Buffering ...");
1182 priv->buffering = TRUE;
1186 case GST_MESSAGE_LATENCY:
1188 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1191 case GST_MESSAGE_ERROR:
1196 gst_message_parse_error (message, &gerror, &debug);
1197 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1198 g_error_free (gerror);
1201 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1204 case GST_MESSAGE_WARNING:
1209 gst_message_parse_warning (message, &gerror, &debug);
1210 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1211 g_error_free (gerror);
1215 case GST_MESSAGE_ELEMENT:
1217 case GST_MESSAGE_STREAM_STATUS:
1219 case GST_MESSAGE_ASYNC_DONE:
1220 if (!priv->adding) {
1221 /* when we are dynamically adding pads, the addition of the udpsrc will
1222 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1223 * wait for the final ASYNC_DONE after everything prerolled */
1224 GST_INFO ("%p: got ASYNC_DONE", media);
1225 collect_media_stats (media);
1227 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1228 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1230 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1233 case GST_MESSAGE_EOS:
1234 GST_INFO ("%p: got EOS", media);
1236 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1237 GST_DEBUG ("shutting down after EOS");
1238 finish_unprepare (media);
1242 GST_INFO ("%p: got message type %d (%s)", media, type,
1243 gst_message_type_get_name (type));
1250 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1252 GstRTSPMediaPrivate *priv = media->priv;
1253 GstRTSPMediaClass *klass;
1256 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1258 g_rec_mutex_lock (&priv->state_lock);
1259 if (klass->handle_message)
1260 ret = klass->handle_message (media, message);
1263 g_rec_mutex_unlock (&priv->state_lock);
1269 watch_destroyed (GstRTSPMedia * media)
1271 GST_DEBUG_OBJECT (media, "source destroyed");
1272 g_object_unref (media);
1275 /* called from streaming threads */
1277 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1279 GstRTSPMediaPrivate *priv = media->priv;
1280 GstRTSPStream *stream;
1282 /* FIXME, element is likely not a payloader, find the payloader here */
1283 stream = gst_rtsp_media_create_stream (media, element, pad);
1285 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1287 g_rec_mutex_lock (&priv->state_lock);
1288 /* we will be adding elements below that will cause ASYNC_DONE to be
1289 * posted in the bus. We want to ignore those messages until the
1290 * pipeline really prerolled. */
1291 priv->adding = TRUE;
1293 /* join the element in the PAUSED state because this callback is
1294 * called from the streaming thread and it is PAUSED */
1295 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1296 priv->rtpbin, GST_STATE_PAUSED);
1298 priv->adding = FALSE;
1299 g_rec_mutex_unlock (&priv->state_lock);
1303 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1305 GstRTSPMediaPrivate *priv = media->priv;
1306 GstElement *fakesink;
1308 g_mutex_lock (&priv->lock);
1309 GST_INFO ("no more pads");
1310 if ((fakesink = priv->fakesink)) {
1311 gst_object_ref (fakesink);
1312 priv->fakesink = NULL;
1313 g_mutex_unlock (&priv->lock);
1315 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1316 gst_element_set_state (fakesink, GST_STATE_NULL);
1317 gst_object_unref (fakesink);
1318 GST_INFO ("removed fakesink");
1323 * gst_rtsp_media_prepare:
1324 * @media: a #GstRTSPMedia
1326 * Prepare @media for streaming. This function will create the pipeline and
1327 * other objects to manage the streaming.
1329 * It will preroll the pipeline and collect vital information about the streams
1330 * such as the duration.
1332 * Returns: %TRUE on success.
1335 gst_rtsp_media_prepare (GstRTSPMedia * media)
1337 GstRTSPMediaPrivate *priv;
1338 GstStateChangeReturn ret;
1339 GstRTSPMediaStatus status;
1341 GstRTSPMediaClass *klass;
1345 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1349 g_rec_mutex_lock (&priv->state_lock);
1350 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1353 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1356 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1357 goto not_unprepared;
1359 if (!priv->reusable && priv->reused)
1362 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1363 if (priv->rtpbin == NULL)
1366 GST_INFO ("preparing media %p", media);
1368 /* reset some variables */
1369 priv->is_live = FALSE;
1370 priv->seekable = FALSE;
1371 priv->buffering = FALSE;
1372 /* we're preparing now */
1373 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1375 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1377 /* add the pipeline bus to our custom mainloop */
1378 priv->source = gst_bus_create_watch (bus);
1379 gst_object_unref (bus);
1381 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1382 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1384 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1385 priv->id = g_source_attach (priv->source, klass->context);
1387 /* add stuff to the bin */
1388 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1390 /* link streams we already have, other streams might appear when we have
1391 * dynamic elements */
1392 for (i = 0; i < priv->streams->len; i++) {
1393 GstRTSPStream *stream;
1395 stream = g_ptr_array_index (priv->streams, i);
1397 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1398 priv->rtpbin, GST_STATE_NULL);
1401 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1402 GstElement *elem = walk->data;
1404 GST_INFO ("adding callbacks for dynamic element %p", elem);
1406 g_signal_connect (elem, "pad-added", (GCallback) pad_added_cb, media);
1407 g_signal_connect (elem, "no-more-pads", (GCallback) no_more_pads_cb, media);
1409 /* we add a fakesink here in order to make the state change async. We remove
1410 * the fakesink again in the no-more-pads callback. */
1411 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1412 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1415 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1416 /* first go to PAUSED */
1417 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1418 priv->target_state = GST_STATE_PAUSED;
1421 case GST_STATE_CHANGE_SUCCESS:
1422 GST_INFO ("SUCCESS state change for media %p", media);
1423 priv->seekable = TRUE;
1425 case GST_STATE_CHANGE_ASYNC:
1426 GST_INFO ("ASYNC state change for media %p", media);
1427 priv->seekable = TRUE;
1429 case GST_STATE_CHANGE_NO_PREROLL:
1430 /* we need to go to PLAYING */
1431 GST_INFO ("NO_PREROLL state change: live media %p", media);
1432 /* FIXME we disable seeking for live streams for now. We should perform a
1433 * seeking query in preroll instead */
1434 priv->seekable = FALSE;
1435 priv->is_live = TRUE;
1436 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1437 if (ret == GST_STATE_CHANGE_FAILURE)
1440 case GST_STATE_CHANGE_FAILURE:
1444 g_rec_mutex_unlock (&priv->state_lock);
1446 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1447 * able to do this async so that we don't block the server thread. */
1448 status = gst_rtsp_media_get_status (media);
1449 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1452 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1454 GST_INFO ("object %p is prerolled", media);
1461 GST_LOG ("media %p was prepared", media);
1462 g_rec_mutex_unlock (&priv->state_lock);
1468 GST_WARNING ("media %p was not unprepared", media);
1469 g_rec_mutex_unlock (&priv->state_lock);
1474 g_rec_mutex_unlock (&priv->state_lock);
1475 GST_WARNING ("can not reuse media %p", media);
1480 g_rec_mutex_unlock (&priv->state_lock);
1481 GST_WARNING ("no rtpbin element");
1482 g_warning ("failed to create element 'rtpbin', check your installation");
1487 GST_WARNING ("failed to preroll pipeline");
1488 gst_rtsp_media_unprepare (media);
1489 g_rec_mutex_unlock (&priv->state_lock);
1494 /* must be called with state-lock */
1496 finish_unprepare (GstRTSPMedia * media)
1498 GstRTSPMediaPrivate *priv = media->priv;
1501 GST_DEBUG ("shutting down");
1503 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1505 for (i = 0; i < priv->streams->len; i++) {
1506 GstRTSPStream *stream;
1508 GST_INFO ("Removing elements of stream %d from pipeline", i);
1510 stream = g_ptr_array_index (priv->streams, i);
1512 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1514 g_ptr_array_set_size (priv->streams, 0);
1516 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1517 priv->rtpbin = NULL;
1519 gst_object_unref (priv->pipeline);
1520 priv->pipeline = NULL;
1522 priv->reused = TRUE;
1523 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1525 /* when the media is not reusable, this will effectively unref the media and
1527 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1529 /* the source has the last ref to the media */
1531 GST_DEBUG ("destroy source");
1532 g_source_destroy (priv->source);
1533 g_source_unref (priv->source);
1537 /* called with state-lock */
1539 default_unprepare (GstRTSPMedia * media)
1541 GstRTSPMediaPrivate *priv = media->priv;
1543 if (priv->eos_shutdown) {
1544 GST_DEBUG ("sending EOS for shutdown");
1545 /* ref so that we don't disappear */
1546 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
1547 /* we need to go to playing again for the EOS to propagate, normally in this
1548 * state, nothing is receiving data from us anymore so this is ok. */
1549 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1550 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
1552 finish_unprepare (media);
1558 * gst_rtsp_media_unprepare:
1559 * @media: a #GstRTSPMedia
1561 * Unprepare @media. After this call, the media should be prepared again before
1562 * it can be used again. If the media is set to be non-reusable, a new instance
1565 * Returns: %TRUE on success.
1568 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1570 GstRTSPMediaPrivate *priv;
1573 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1577 g_rec_mutex_lock (&priv->state_lock);
1578 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1579 goto was_unprepared;
1581 GST_INFO ("unprepare media %p", media);
1582 priv->target_state = GST_STATE_NULL;
1585 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
1586 GstRTSPMediaClass *klass;
1588 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1589 if (klass->unprepare)
1590 success = klass->unprepare (media);
1592 finish_unprepare (media);
1594 g_rec_mutex_unlock (&priv->state_lock);
1600 g_rec_mutex_unlock (&priv->state_lock);
1601 GST_INFO ("media %p was already unprepared", media);
1607 * gst_rtsp_media_set_state:
1608 * @media: a #GstRTSPMedia
1609 * @state: the target state of the media
1610 * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
1612 * Set the state of @media to @state and for the transports in @transports.
1614 * @media must be prepared with gst_rtsp_media_prepare();
1616 * Returns: %TRUE on success.
1619 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1620 GPtrArray * transports)
1622 GstRTSPMediaPrivate *priv;
1624 gboolean activate, deactivate, do_state;
1627 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1628 g_return_val_if_fail (transports != NULL, FALSE);
1632 g_rec_mutex_lock (&priv->state_lock);
1633 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1636 /* NULL and READY are the same */
1637 if (state == GST_STATE_READY)
1638 state = GST_STATE_NULL;
1640 activate = deactivate = FALSE;
1642 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1646 case GST_STATE_NULL:
1647 case GST_STATE_PAUSED:
1648 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
1649 if (priv->target_state == GST_STATE_PLAYING)
1652 case GST_STATE_PLAYING:
1653 /* we're going to PLAYING, activate */
1659 old_active = priv->n_active;
1661 for (i = 0; i < transports->len; i++) {
1662 GstRTSPStreamTransport *trans;
1664 /* we need a non-NULL entry in the array */
1665 trans = g_ptr_array_index (transports, i);
1670 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
1672 } else if (deactivate) {
1673 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
1678 /* we just activated the first media, do the playing state change */
1679 if (old_active == 0 && activate)
1681 /* if we have no more active media, do the downward state changes */
1682 else if (priv->n_active == 0)
1687 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
1690 if (priv->target_state != state) {
1692 if (state == GST_STATE_NULL) {
1693 gst_rtsp_media_unprepare (media);
1695 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
1697 priv->target_state = state;
1698 gst_element_set_state (priv->pipeline, state);
1701 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
1705 /* remember where we are */
1706 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
1707 old_active != priv->n_active))
1708 collect_media_stats (media);
1710 g_rec_mutex_unlock (&priv->state_lock);
1717 GST_WARNING ("media %p was not prepared", media);
1718 g_rec_mutex_unlock (&priv->state_lock);