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
36 /* protected by lock */
37 GstRTSPPermissions *permissions;
40 GstRTSPLowerTrans protocols;
42 gboolean eos_shutdown;
44 GstRTSPAddressPool *pool;
47 GRecMutex state_lock; /* locking order: state lock, lock */
48 GPtrArray *streams; /* protected by lock */
49 GList *dynamic; /* protected by lock */
50 GstRTSPMediaStatus status; /* protected by lock */
55 /* the pipeline for the media */
57 GstElement *fakesink; /* protected by lock */
60 GstRTSPThread *thread;
62 gboolean time_provider;
63 GstNetTimeProvider *nettime;
68 GstState target_state;
70 /* RTP session manager */
73 /* the range of media */
74 GstRTSPTimeRange range; /* protected by lock */
75 GstClockTime range_start;
76 GstClockTime range_stop;
79 #define DEFAULT_SHARED FALSE
80 #define DEFAULT_REUSABLE FALSE
81 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
82 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
83 #define DEFAULT_EOS_SHUTDOWN FALSE
84 #define DEFAULT_BUFFER_SIZE 0x80000
85 #define DEFAULT_TIME_PROVIDER FALSE
87 /* define to dump received RTCP packets */
106 SIGNAL_REMOVED_STREAM,
113 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
114 #define GST_CAT_DEFAULT rtsp_media_debug
116 static void gst_rtsp_media_get_property (GObject * object, guint propid,
117 GValue * value, GParamSpec * pspec);
118 static void gst_rtsp_media_set_property (GObject * object, guint propid,
119 const GValue * value, GParamSpec * pspec);
120 static void gst_rtsp_media_finalize (GObject * obj);
122 static gboolean default_handle_message (GstRTSPMedia * media,
123 GstMessage * message);
124 static void finish_unprepare (GstRTSPMedia * media);
125 static gboolean default_unprepare (GstRTSPMedia * media);
127 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
128 GstRTSPRangeUnit unit);
129 static gboolean default_query_position (GstRTSPMedia * media,
131 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
133 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
135 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
138 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
140 GObjectClass *gobject_class;
142 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
144 gobject_class = G_OBJECT_CLASS (klass);
146 gobject_class->get_property = gst_rtsp_media_get_property;
147 gobject_class->set_property = gst_rtsp_media_set_property;
148 gobject_class->finalize = gst_rtsp_media_finalize;
150 g_object_class_install_property (gobject_class, PROP_SHARED,
151 g_param_spec_boolean ("shared", "Shared",
152 "If this media pipeline can be shared", DEFAULT_SHARED,
153 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
155 g_object_class_install_property (gobject_class, PROP_REUSABLE,
156 g_param_spec_boolean ("reusable", "Reusable",
157 "If this media pipeline can be reused after an unprepare",
158 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
160 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
161 g_param_spec_flags ("protocols", "Protocols",
162 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
163 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
165 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
166 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
167 "Send an EOS event to the pipeline before unpreparing",
168 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
170 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
171 g_param_spec_uint ("buffer-size", "Buffer Size",
172 "The kernel UDP buffer size to use", 0, G_MAXUINT,
173 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
175 g_object_class_install_property (gobject_class, PROP_ELEMENT,
176 g_param_spec_object ("element", "The Element",
177 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
178 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
180 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
181 g_param_spec_boolean ("time-provider", "Time Provider",
182 "Use a NetTimeProvider for clients",
183 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
185 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
186 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
187 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
188 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
190 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
191 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
192 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
193 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
194 GST_TYPE_RTSP_STREAM);
196 gst_rtsp_media_signals[SIGNAL_PREPARED] =
197 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
198 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
199 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
201 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
202 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
203 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
204 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
206 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
207 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
208 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
209 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
211 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
213 klass->handle_message = default_handle_message;
214 klass->unprepare = default_unprepare;
215 klass->convert_range = default_convert_range;
216 klass->query_position = default_query_position;
217 klass->query_stop = default_query_stop;
221 gst_rtsp_media_init (GstRTSPMedia * media)
223 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
227 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
228 g_mutex_init (&priv->lock);
229 g_cond_init (&priv->cond);
230 g_rec_mutex_init (&priv->state_lock);
232 priv->shared = DEFAULT_SHARED;
233 priv->reusable = DEFAULT_REUSABLE;
234 priv->protocols = DEFAULT_PROTOCOLS;
235 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
236 priv->buffer_size = DEFAULT_BUFFER_SIZE;
237 priv->time_provider = DEFAULT_TIME_PROVIDER;
241 gst_rtsp_media_finalize (GObject * obj)
243 GstRTSPMediaPrivate *priv;
246 media = GST_RTSP_MEDIA (obj);
249 GST_INFO ("finalize media %p", media);
251 g_ptr_array_unref (priv->streams);
253 g_list_free_full (priv->dynamic, gst_object_unref);
256 gst_object_unref (priv->pipeline);
258 gst_object_unref (priv->nettime);
259 gst_object_unref (priv->element);
261 g_object_unref (priv->pool);
262 g_mutex_clear (&priv->lock);
263 g_cond_clear (&priv->cond);
264 g_rec_mutex_clear (&priv->state_lock);
266 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
270 gst_rtsp_media_get_property (GObject * object, guint propid,
271 GValue * value, GParamSpec * pspec)
273 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
277 g_value_set_object (value, media->priv->element);
280 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
283 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
286 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
288 case PROP_EOS_SHUTDOWN:
289 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
291 case PROP_BUFFER_SIZE:
292 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
294 case PROP_TIME_PROVIDER:
295 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
298 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
303 gst_rtsp_media_set_property (GObject * object, guint propid,
304 const GValue * value, GParamSpec * pspec)
306 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
310 media->priv->element = g_value_get_object (value);
311 gst_object_ref_sink (media->priv->element);
314 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
317 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
320 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
322 case PROP_EOS_SHUTDOWN:
323 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
325 case PROP_BUFFER_SIZE:
326 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
328 case PROP_TIME_PROVIDER:
329 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
332 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
336 /* must be called with state lock */
338 collect_media_stats (GstRTSPMedia * media)
340 GstRTSPMediaPrivate *priv = media->priv;
341 gint64 position, stop;
343 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
344 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
347 priv->range.unit = GST_RTSP_RANGE_NPT;
349 GST_INFO ("collect media stats");
352 priv->range.min.type = GST_RTSP_TIME_NOW;
353 priv->range.min.seconds = -1;
354 priv->range_start = -1;
355 priv->range.max.type = GST_RTSP_TIME_END;
356 priv->range.max.seconds = -1;
357 priv->range_stop = -1;
359 GstRTSPMediaClass *klass;
362 klass = GST_RTSP_MEDIA_GET_CLASS (media);
364 /* get the position */
366 if (klass->query_position)
367 ret = klass->query_position (media, &position);
370 GST_INFO ("position query failed");
374 /* get the current segment stop */
376 if (klass->query_stop)
377 ret = klass->query_stop (media, &stop);
380 GST_INFO ("stop query failed");
384 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
385 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
387 if (position == -1) {
388 priv->range.min.type = GST_RTSP_TIME_NOW;
389 priv->range.min.seconds = -1;
390 priv->range_start = -1;
392 priv->range.min.type = GST_RTSP_TIME_SECONDS;
393 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
394 priv->range_start = position;
397 priv->range.max.type = GST_RTSP_TIME_END;
398 priv->range.max.seconds = -1;
399 priv->range_stop = -1;
401 priv->range.max.type = GST_RTSP_TIME_SECONDS;
402 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
403 priv->range_stop = stop;
409 * gst_rtsp_media_new:
410 * @element: (transfer full): a #GstElement
412 * Create a new #GstRTSPMedia instance. @element is the bin element that
413 * provides the different streams. The #GstRTSPMedia object contains the
414 * element to produce RTP data for one or more related (audio/video/..)
417 * Ownership is taken of @element.
419 * Returns: a new #GstRTSPMedia object.
422 gst_rtsp_media_new (GstElement * element)
424 GstRTSPMedia *result;
426 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
428 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
434 * gst_rtsp_media_get_element:
435 * @media: a #GstRTSPMedia
437 * Get the element that was used when constructing @media.
439 * Returns: a #GstElement. Unref after usage.
442 gst_rtsp_media_get_element (GstRTSPMedia * media)
444 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
446 return gst_object_ref (media->priv->element);
450 * gst_rtsp_media_take_pipeline:
451 * @media: a #GstRTSPMedia
452 * @pipeline: (transfer full): a #GstPipeline
454 * Set @pipeline as the #GstPipeline for @media. Ownership is
455 * taken of @pipeline.
458 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
460 GstRTSPMediaPrivate *priv;
462 GstNetTimeProvider *nettime;
464 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
465 g_return_if_fail (GST_IS_PIPELINE (pipeline));
469 g_mutex_lock (&priv->lock);
470 old = priv->pipeline;
471 priv->pipeline = GST_ELEMENT_CAST (pipeline);
472 nettime = priv->nettime;
473 priv->nettime = NULL;
474 g_mutex_unlock (&priv->lock);
477 gst_object_unref (old);
480 gst_object_unref (nettime);
482 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
486 * gst_rtsp_media_set_permissions:
487 * @media: a #GstRTSPMedia
488 * @permissions: a #GstRTSPPermissions
490 * Set @permissions on @media.
493 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
494 GstRTSPPermissions * permissions)
496 GstRTSPMediaPrivate *priv;
498 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
502 g_mutex_lock (&priv->lock);
503 if (priv->permissions)
504 gst_rtsp_permissions_unref (priv->permissions);
505 if ((priv->permissions = permissions))
506 gst_rtsp_permissions_ref (permissions);
507 g_mutex_unlock (&priv->lock);
511 * gst_rtsp_media_get_permissions:
512 * @media: a #GstRTSPMedia
514 * Get the permissions object from @media.
516 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
519 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
521 GstRTSPMediaPrivate *priv;
522 GstRTSPPermissions *result;
524 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
528 g_mutex_lock (&priv->lock);
529 if ((result = priv->permissions))
530 gst_rtsp_permissions_ref (result);
531 g_mutex_unlock (&priv->lock);
537 * gst_rtsp_media_set_shared:
538 * @media: a #GstRTSPMedia
539 * @shared: the new value
541 * Set or unset if the pipeline for @media can be shared will multiple clients.
542 * When @shared is %TRUE, client requests for this media will share the media
546 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
548 GstRTSPMediaPrivate *priv;
550 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
554 g_mutex_lock (&priv->lock);
555 priv->shared = shared;
556 g_mutex_unlock (&priv->lock);
560 * gst_rtsp_media_is_shared:
561 * @media: a #GstRTSPMedia
563 * Check if the pipeline for @media can be shared between multiple clients.
565 * Returns: %TRUE if the media can be shared between clients.
568 gst_rtsp_media_is_shared (GstRTSPMedia * media)
570 GstRTSPMediaPrivate *priv;
573 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
577 g_mutex_lock (&priv->lock);
579 g_mutex_unlock (&priv->lock);
585 * gst_rtsp_media_set_reusable:
586 * @media: a #GstRTSPMedia
587 * @reusable: the new value
589 * Set or unset if the pipeline for @media can be reused after the pipeline has
593 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
595 GstRTSPMediaPrivate *priv;
597 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
601 g_mutex_lock (&priv->lock);
602 priv->reusable = reusable;
603 g_mutex_unlock (&priv->lock);
607 * gst_rtsp_media_is_reusable:
608 * @media: a #GstRTSPMedia
610 * Check if the pipeline for @media can be reused after an unprepare.
612 * Returns: %TRUE if the media can be reused
615 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
617 GstRTSPMediaPrivate *priv;
620 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
624 g_mutex_lock (&priv->lock);
625 res = priv->reusable;
626 g_mutex_unlock (&priv->lock);
632 * gst_rtsp_media_set_protocols:
633 * @media: a #GstRTSPMedia
634 * @protocols: the new flags
636 * Configure the allowed lower transport for @media.
639 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
641 GstRTSPMediaPrivate *priv;
643 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
647 g_mutex_lock (&priv->lock);
648 priv->protocols = protocols;
649 g_mutex_unlock (&priv->lock);
653 * gst_rtsp_media_get_protocols:
654 * @media: a #GstRTSPMedia
656 * Get the allowed protocols of @media.
658 * Returns: a #GstRTSPLowerTrans
661 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
663 GstRTSPMediaPrivate *priv;
664 GstRTSPLowerTrans res;
666 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
667 GST_RTSP_LOWER_TRANS_UNKNOWN);
671 g_mutex_lock (&priv->lock);
672 res = priv->protocols;
673 g_mutex_unlock (&priv->lock);
679 * gst_rtsp_media_set_eos_shutdown:
680 * @media: a #GstRTSPMedia
681 * @eos_shutdown: the new value
683 * Set or unset if an EOS event will be sent to the pipeline for @media before
687 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
689 GstRTSPMediaPrivate *priv;
691 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
695 g_mutex_lock (&priv->lock);
696 priv->eos_shutdown = eos_shutdown;
697 g_mutex_unlock (&priv->lock);
701 * gst_rtsp_media_is_eos_shutdown:
702 * @media: a #GstRTSPMedia
704 * Check if the pipeline for @media will send an EOS down the pipeline before
707 * Returns: %TRUE if the media will send EOS before unpreparing.
710 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
712 GstRTSPMediaPrivate *priv;
715 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
719 g_mutex_lock (&priv->lock);
720 res = priv->eos_shutdown;
721 g_mutex_unlock (&priv->lock);
727 * gst_rtsp_media_set_buffer_size:
728 * @media: a #GstRTSPMedia
729 * @size: the new value
731 * Set the kernel UDP buffer size.
734 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
736 GstRTSPMediaPrivate *priv;
738 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
740 GST_LOG_OBJECT (media, "set buffer size %u", size);
744 g_mutex_lock (&priv->lock);
745 priv->buffer_size = size;
746 g_mutex_unlock (&priv->lock);
750 * gst_rtsp_media_get_buffer_size:
751 * @media: a #GstRTSPMedia
753 * Get the kernel UDP buffer size.
755 * Returns: the kernel UDP buffer size.
758 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
760 GstRTSPMediaPrivate *priv;
763 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
767 g_mutex_unlock (&priv->lock);
768 res = priv->buffer_size;
769 g_mutex_unlock (&priv->lock);
775 * gst_rtsp_media_use_time_provider:
776 * @media: a #GstRTSPMedia
777 * @time_provider: if a #GstNetTimeProvider should be used
779 * Set @media to provide a #GstNetTimeProvider.
782 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
784 GstRTSPMediaPrivate *priv;
786 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
790 g_mutex_lock (&priv->lock);
791 priv->time_provider = time_provider;
792 g_mutex_unlock (&priv->lock);
796 * gst_rtsp_media_is_time_provider:
797 * @media: a #GstRTSPMedia
799 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
801 * Use gst_rtsp_media_get_time_provider() to get the network clock.
803 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
806 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
808 GstRTSPMediaPrivate *priv;
811 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
815 g_mutex_unlock (&priv->lock);
816 res = priv->time_provider;
817 g_mutex_unlock (&priv->lock);
823 * gst_rtsp_media_set_address_pool:
824 * @media: a #GstRTSPMedia
825 * @pool: a #GstRTSPAddressPool
827 * configure @pool to be used as the address pool of @media.
830 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
831 GstRTSPAddressPool * pool)
833 GstRTSPMediaPrivate *priv;
834 GstRTSPAddressPool *old;
836 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
840 GST_LOG_OBJECT (media, "set address pool %p", pool);
842 g_mutex_lock (&priv->lock);
843 if ((old = priv->pool) != pool)
844 priv->pool = pool ? g_object_ref (pool) : NULL;
847 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
849 g_mutex_unlock (&priv->lock);
852 g_object_unref (old);
856 * gst_rtsp_media_get_address_pool:
857 * @media: a #GstRTSPMedia
859 * Get the #GstRTSPAddressPool used as the address pool of @media.
861 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
865 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
867 GstRTSPMediaPrivate *priv;
868 GstRTSPAddressPool *result;
870 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
874 g_mutex_lock (&priv->lock);
875 if ((result = priv->pool))
876 g_object_ref (result);
877 g_mutex_unlock (&priv->lock);
883 * gst_rtsp_media_collect_streams:
884 * @media: a #GstRTSPMedia
886 * Find all payloader elements, they should be named pay%d in the
887 * element of @media, and create #GstRTSPStreams for them.
889 * Collect all dynamic elements, named dynpay%d, and add them to
890 * the list of dynamic elements.
893 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
895 GstRTSPMediaPrivate *priv;
896 GstElement *element, *elem;
901 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
904 element = priv->element;
907 for (i = 0; have_elem; i++) {
912 name = g_strdup_printf ("pay%d", i);
913 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
914 GST_INFO ("found stream %d with payloader %p", i, elem);
916 /* take the pad of the payloader */
917 pad = gst_element_get_static_pad (elem, "src");
918 /* create the stream */
919 gst_rtsp_media_create_stream (media, elem, pad);
920 gst_object_unref (pad);
921 gst_object_unref (elem);
927 name = g_strdup_printf ("dynpay%d", i);
928 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
929 /* a stream that will dynamically create pads to provide RTP packets */
931 GST_INFO ("found dynamic element %d, %p", i, elem);
933 g_mutex_lock (&priv->lock);
934 priv->dynamic = g_list_prepend (priv->dynamic, elem);
935 g_mutex_unlock (&priv->lock);
944 * gst_rtsp_media_create_stream:
945 * @media: a #GstRTSPMedia
946 * @payloader: a #GstElement
947 * @srcpad: a source #GstPad
949 * Create a new stream in @media that provides RTP data on @srcpad.
950 * @srcpad should be a pad of an element inside @media->element.
952 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
956 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
959 GstRTSPMediaPrivate *priv;
960 GstRTSPStream *stream;
965 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
966 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
967 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
968 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
972 g_mutex_lock (&priv->lock);
973 idx = priv->streams->len;
975 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
977 name = g_strdup_printf ("src_%u", idx);
978 srcpad = gst_ghost_pad_new (name, pad);
979 gst_pad_set_active (srcpad, TRUE);
980 gst_element_add_pad (priv->element, srcpad);
983 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
985 gst_rtsp_stream_set_address_pool (stream, priv->pool);
987 g_ptr_array_add (priv->streams, stream);
988 g_mutex_unlock (&priv->lock);
990 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
997 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
999 GstRTSPMediaPrivate *priv;
1004 g_mutex_lock (&priv->lock);
1005 /* remove the ghostpad */
1006 srcpad = gst_rtsp_stream_get_srcpad (stream);
1007 gst_element_remove_pad (priv->element, srcpad);
1008 gst_object_unref (srcpad);
1009 /* now remove the stream */
1010 g_object_ref (stream);
1011 g_ptr_array_remove (priv->streams, stream);
1012 g_mutex_unlock (&priv->lock);
1014 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1017 g_object_unref (stream);
1021 * gst_rtsp_media_n_streams:
1022 * @media: a #GstRTSPMedia
1024 * Get the number of streams in this media.
1026 * Returns: The number of streams.
1029 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1031 GstRTSPMediaPrivate *priv;
1034 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1038 g_mutex_lock (&priv->lock);
1039 res = priv->streams->len;
1040 g_mutex_unlock (&priv->lock);
1046 * gst_rtsp_media_get_stream:
1047 * @media: a #GstRTSPMedia
1048 * @idx: the stream index
1050 * Retrieve the stream with index @idx from @media.
1052 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1053 * that index did not exist.
1056 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1058 GstRTSPMediaPrivate *priv;
1061 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1065 g_mutex_lock (&priv->lock);
1066 if (idx < priv->streams->len)
1067 res = g_ptr_array_index (priv->streams, idx);
1070 g_mutex_unlock (&priv->lock);
1076 * gst_rtsp_media_find_stream:
1077 * @media: a #GstRTSPMedia
1078 * @control: the control of the stream
1080 * Find a stream in @media with @control as the control uri.
1082 * Returns: (transfer none): the #GstRTSPStream with control uri @control
1083 * or %NULL when a stream with that control did not exist.
1086 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1088 GstRTSPMediaPrivate *priv;
1092 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1093 g_return_val_if_fail (control != NULL, NULL);
1099 g_mutex_lock (&priv->lock);
1100 for (i = 0; i < priv->streams->len; i++) {
1101 GstRTSPStream *test;
1103 test = g_ptr_array_index (priv->streams, i);
1104 if (gst_rtsp_stream_has_control (test, control)) {
1109 g_mutex_unlock (&priv->lock);
1115 * gst_rtsp_media_get_range_string:
1116 * @media: a #GstRTSPMedia
1117 * @play: for the PLAY request
1118 * @unit: the unit to use for the string
1120 * Get the current range as a string. @media must be prepared with
1121 * gst_rtsp_media_prepare ().
1123 * Returns: The range as a string, g_free() after usage.
1126 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1127 GstRTSPRangeUnit unit)
1129 GstRTSPMediaClass *klass;
1130 GstRTSPMediaPrivate *priv;
1132 GstRTSPTimeRange range;
1134 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1135 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1136 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1140 g_rec_mutex_lock (&priv->state_lock);
1141 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1144 g_mutex_lock (&priv->lock);
1146 range = priv->range;
1148 if (!play && priv->n_active > 0) {
1149 range.min.type = GST_RTSP_TIME_NOW;
1150 range.min.seconds = -1;
1152 g_mutex_unlock (&priv->lock);
1153 g_rec_mutex_unlock (&priv->state_lock);
1155 if (!klass->convert_range (media, &range, unit))
1156 goto conversion_failed;
1158 result = gst_rtsp_range_to_string (&range);
1165 GST_WARNING ("media %p was not prepared", media);
1166 g_rec_mutex_unlock (&priv->state_lock);
1171 GST_WARNING ("range conversion to unit %d failed", unit);
1177 * gst_rtsp_media_seek:
1178 * @media: a #GstRTSPMedia
1179 * @range: a #GstRTSPTimeRange
1181 * Seek the pipeline of @media to @range. @media must be prepared with
1182 * gst_rtsp_media_prepare().
1184 * Returns: %TRUE on success.
1187 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1189 GstRTSPMediaClass *klass;
1190 GstRTSPMediaPrivate *priv;
1193 GstClockTime start, stop;
1194 GstSeekType start_type, stop_type;
1196 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1198 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1199 g_return_val_if_fail (range != NULL, FALSE);
1200 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1204 g_rec_mutex_lock (&priv->state_lock);
1205 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1208 if (!priv->seekable)
1211 /* depends on the current playing state of the pipeline. We might need to
1212 * queue this until we get EOS. */
1213 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
1215 start_type = stop_type = GST_SEEK_TYPE_NONE;
1217 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1219 gst_rtsp_range_get_times (range, &start, &stop);
1221 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1222 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1223 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1224 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1226 if (priv->range_start == start)
1227 start = GST_CLOCK_TIME_NONE;
1228 else if (start != GST_CLOCK_TIME_NONE)
1229 start_type = GST_SEEK_TYPE_SET;
1231 if (priv->range_stop == stop)
1232 stop = GST_CLOCK_TIME_NONE;
1233 else if (stop != GST_CLOCK_TIME_NONE)
1234 stop_type = GST_SEEK_TYPE_SET;
1236 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1237 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1238 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1240 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1241 flags, start_type, start, stop_type, stop);
1243 /* and block for the seek to complete */
1244 GST_INFO ("done seeking %d", res);
1245 gst_element_get_state (priv->pipeline, NULL, NULL, -1);
1246 GST_INFO ("prerolled again");
1248 collect_media_stats (media);
1250 GST_INFO ("no seek needed");
1253 g_rec_mutex_unlock (&priv->state_lock);
1260 g_rec_mutex_unlock (&priv->state_lock);
1261 GST_INFO ("media %p is not prepared", media);
1266 g_rec_mutex_unlock (&priv->state_lock);
1267 GST_INFO ("pipeline is not seekable");
1272 g_rec_mutex_unlock (&priv->state_lock);
1273 GST_WARNING ("conversion to npt not supported");
1279 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1281 GstRTSPMediaPrivate *priv = media->priv;
1283 g_mutex_lock (&priv->lock);
1284 priv->status = status;
1285 GST_DEBUG ("setting new status to %d", status);
1286 g_cond_broadcast (&priv->cond);
1287 g_mutex_unlock (&priv->lock);
1291 * gst_rtsp_media_get_status:
1292 * @media: a #GstRTSPMedia
1294 * Get the status of @media. When @media is busy preparing, this function waits
1295 * until @media is prepared or in error.
1297 * Returns: the status of @media.
1300 gst_rtsp_media_get_status (GstRTSPMedia * media)
1302 GstRTSPMediaPrivate *priv = media->priv;
1303 GstRTSPMediaStatus result;
1306 g_mutex_lock (&priv->lock);
1307 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1308 /* while we are preparing, wait */
1309 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1310 GST_DEBUG ("waiting for status change");
1311 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1312 GST_DEBUG ("timeout, assuming error status");
1313 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1316 /* could be success or error */
1317 result = priv->status;
1318 GST_DEBUG ("got status %d", result);
1319 g_mutex_unlock (&priv->lock);
1324 /* called with state-lock */
1326 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1328 GstRTSPMediaPrivate *priv = media->priv;
1329 GstMessageType type;
1331 type = GST_MESSAGE_TYPE (message);
1334 case GST_MESSAGE_STATE_CHANGED:
1336 case GST_MESSAGE_BUFFERING:
1340 gst_message_parse_buffering (message, &percent);
1342 /* no state management needed for live pipelines */
1346 if (percent == 100) {
1347 /* a 100% message means buffering is done */
1348 priv->buffering = FALSE;
1349 /* if the desired state is playing, go back */
1350 if (priv->target_state == GST_STATE_PLAYING) {
1351 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1352 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1354 GST_INFO ("Buffering done");
1357 /* buffering busy */
1358 if (priv->buffering == FALSE) {
1359 if (priv->target_state == GST_STATE_PLAYING) {
1360 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1361 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1362 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1364 GST_INFO ("Buffering ...");
1367 priv->buffering = TRUE;
1371 case GST_MESSAGE_LATENCY:
1373 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1376 case GST_MESSAGE_ERROR:
1381 gst_message_parse_error (message, &gerror, &debug);
1382 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1383 g_error_free (gerror);
1386 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1389 case GST_MESSAGE_WARNING:
1394 gst_message_parse_warning (message, &gerror, &debug);
1395 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1396 g_error_free (gerror);
1400 case GST_MESSAGE_ELEMENT:
1402 case GST_MESSAGE_STREAM_STATUS:
1404 case GST_MESSAGE_ASYNC_DONE:
1406 /* when we are dynamically adding pads, the addition of the udpsrc will
1407 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1408 * wait for the final ASYNC_DONE after everything prerolled */
1409 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1411 GST_INFO ("%p: got ASYNC_DONE", media);
1412 collect_media_stats (media);
1414 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1415 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1418 case GST_MESSAGE_EOS:
1419 GST_INFO ("%p: got EOS", media);
1421 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1422 GST_DEBUG ("shutting down after EOS");
1423 finish_unprepare (media);
1427 GST_INFO ("%p: got message type %d (%s)", media, type,
1428 gst_message_type_get_name (type));
1435 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1437 GstRTSPMediaPrivate *priv = media->priv;
1438 GstRTSPMediaClass *klass;
1441 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1443 g_rec_mutex_lock (&priv->state_lock);
1444 if (klass->handle_message)
1445 ret = klass->handle_message (media, message);
1448 g_rec_mutex_unlock (&priv->state_lock);
1454 watch_destroyed (GstRTSPMedia * media)
1456 GST_DEBUG_OBJECT (media, "source destroyed");
1457 g_object_unref (media);
1460 /* called from streaming threads */
1462 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1464 GstRTSPMediaPrivate *priv = media->priv;
1465 GstRTSPStream *stream;
1467 /* FIXME, element is likely not a payloader, find the payloader here */
1468 stream = gst_rtsp_media_create_stream (media, element, pad);
1470 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1472 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1474 g_rec_mutex_lock (&priv->state_lock);
1475 /* we will be adding elements below that will cause ASYNC_DONE to be
1476 * posted in the bus. We want to ignore those messages until the
1477 * pipeline really prerolled. */
1478 priv->adding = TRUE;
1480 /* join the element in the PAUSED state because this callback is
1481 * called from the streaming thread and it is PAUSED */
1482 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1483 priv->rtpbin, GST_STATE_PAUSED);
1485 priv->adding = FALSE;
1486 g_rec_mutex_unlock (&priv->state_lock);
1490 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1492 GstRTSPMediaPrivate *priv = media->priv;
1493 GstRTSPStream *stream;
1495 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1499 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1501 g_rec_mutex_lock (&priv->state_lock);
1502 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1503 g_rec_mutex_unlock (&priv->state_lock);
1505 gst_rtsp_media_remove_stream (media, stream);
1509 remove_fakesink (GstRTSPMediaPrivate * priv)
1511 GstElement *fakesink;
1513 g_mutex_lock (&priv->lock);
1514 if ((fakesink = priv->fakesink))
1515 gst_object_ref (fakesink);
1516 priv->fakesink = NULL;
1517 g_mutex_unlock (&priv->lock);
1520 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1521 gst_element_set_state (fakesink, GST_STATE_NULL);
1522 gst_object_unref (fakesink);
1523 GST_INFO ("removed fakesink");
1528 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1530 GstRTSPMediaPrivate *priv = media->priv;
1532 GST_INFO ("no more pads");
1533 remove_fakesink (priv);
1536 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1538 struct _DynPaySignalHandlers
1540 gulong pad_added_handler;
1541 gulong pad_removed_handler;
1542 gulong no_more_pads_handler;
1546 start_prepare (GstRTSPMedia * media)
1548 GstRTSPMediaPrivate *priv = media->priv;
1549 GstStateChangeReturn ret;
1553 /* link streams we already have, other streams might appear when we have
1554 * dynamic elements */
1555 for (i = 0; i < priv->streams->len; i++) {
1556 GstRTSPStream *stream;
1558 stream = g_ptr_array_index (priv->streams, i);
1560 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1561 priv->rtpbin, GST_STATE_NULL);
1564 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1565 GstElement *elem = walk->data;
1566 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
1568 GST_INFO ("adding callbacks for dynamic element %p", elem);
1570 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
1571 (GCallback) pad_added_cb, media);
1572 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
1573 (GCallback) pad_removed_cb, media);
1574 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
1575 (GCallback) no_more_pads_cb, media);
1577 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
1579 /* we add a fakesink here in order to make the state change async. We remove
1580 * the fakesink again in the no-more-pads callback. */
1581 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1582 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1585 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1586 /* first go to PAUSED */
1587 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1588 priv->target_state = GST_STATE_PAUSED;
1591 case GST_STATE_CHANGE_SUCCESS:
1592 GST_INFO ("SUCCESS state change for media %p", media);
1593 priv->seekable = TRUE;
1595 case GST_STATE_CHANGE_ASYNC:
1596 GST_INFO ("ASYNC state change for media %p", media);
1597 priv->seekable = TRUE;
1599 case GST_STATE_CHANGE_NO_PREROLL:
1600 /* we need to go to PLAYING */
1601 GST_INFO ("NO_PREROLL state change: live media %p", media);
1602 /* FIXME we disable seeking for live streams for now. We should perform a
1603 * seeking query in preroll instead */
1604 priv->seekable = FALSE;
1605 priv->is_live = TRUE;
1606 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1607 if (ret == GST_STATE_CHANGE_FAILURE)
1610 case GST_STATE_CHANGE_FAILURE:
1618 GST_WARNING ("failed to preroll pipeline");
1619 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1625 * gst_rtsp_media_prepare:
1626 * @media: a #GstRTSPMedia
1627 * @thread: a #GstRTSPThread to run the bus handler or %NULL
1629 * Prepare @media for streaming. This function will create the objects
1630 * to manage the streaming. A pipeline must have been set on @media with
1631 * gst_rtsp_media_take_pipeline().
1633 * It will preroll the pipeline and collect vital information about the streams
1634 * such as the duration.
1636 * Returns: %TRUE on success.
1639 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
1641 GstRTSPMediaPrivate *priv;
1642 GstRTSPMediaStatus status;
1646 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1647 g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
1651 g_rec_mutex_lock (&priv->state_lock);
1652 priv->prepare_count++;
1654 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1657 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1660 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1661 goto not_unprepared;
1663 if (!priv->reusable && priv->reused)
1666 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1667 if (priv->rtpbin == NULL)
1670 GST_INFO ("preparing media %p", media);
1672 /* reset some variables */
1673 priv->is_live = FALSE;
1674 priv->seekable = FALSE;
1675 priv->buffering = FALSE;
1676 priv->thread = thread;
1677 /* we're preparing now */
1678 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1680 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1682 /* add the pipeline bus to our custom mainloop */
1683 priv->source = gst_bus_create_watch (bus);
1684 gst_object_unref (bus);
1686 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1687 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1689 priv->id = g_source_attach (priv->source, thread->context);
1691 /* add stuff to the bin */
1692 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1694 /* do remainder in context */
1695 source = g_idle_source_new ();
1696 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
1697 g_source_attach (source, thread->context);
1698 g_source_unref (source);
1701 g_rec_mutex_unlock (&priv->state_lock);
1703 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1704 * able to do this async so that we don't block the server thread. */
1705 status = gst_rtsp_media_get_status (media);
1706 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1709 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1711 GST_INFO ("object %p is prerolled", media);
1718 GST_LOG ("media %p was prepared", media);
1719 g_rec_mutex_unlock (&priv->state_lock);
1725 GST_WARNING ("media %p was not unprepared", media);
1726 priv->prepare_count--;
1727 g_rec_mutex_unlock (&priv->state_lock);
1732 priv->prepare_count--;
1733 g_rec_mutex_unlock (&priv->state_lock);
1734 GST_WARNING ("can not reuse media %p", media);
1739 priv->prepare_count--;
1740 g_rec_mutex_unlock (&priv->state_lock);
1741 GST_WARNING ("no rtpbin element");
1742 g_warning ("failed to create element 'rtpbin', check your installation");
1747 GST_WARNING ("failed to preroll pipeline");
1748 gst_rtsp_media_unprepare (media);
1753 /* must be called with state-lock */
1755 finish_unprepare (GstRTSPMedia * media)
1757 GstRTSPMediaPrivate *priv = media->priv;
1761 GST_DEBUG ("shutting down");
1763 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1764 remove_fakesink (priv);
1766 for (i = 0; i < priv->streams->len; i++) {
1767 GstRTSPStream *stream;
1769 GST_INFO ("Removing elements of stream %d from pipeline", i);
1771 stream = g_ptr_array_index (priv->streams, i);
1773 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1776 /* remove the pad signal handlers */
1777 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1778 GstElement *elem = walk->data;
1779 DynPaySignalHandlers *handlers;
1782 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
1783 g_assert (handlers != NULL);
1785 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
1786 g_signal_handler_disconnect (G_OBJECT (elem),
1787 handlers->pad_removed_handler);
1788 g_signal_handler_disconnect (G_OBJECT (elem),
1789 handlers->no_more_pads_handler);
1791 g_slice_free (DynPaySignalHandlers, handlers);
1794 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1795 priv->rtpbin = NULL;
1798 gst_object_unref (priv->nettime);
1799 priv->nettime = NULL;
1801 priv->reused = TRUE;
1802 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1804 /* when the media is not reusable, this will effectively unref the media and
1806 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1808 /* the source has the last ref to the media */
1810 GST_DEBUG ("destroy source");
1811 g_source_destroy (priv->source);
1812 g_source_unref (priv->source);
1815 GST_DEBUG ("stop thread");
1816 gst_rtsp_thread_stop (priv->thread);
1820 /* called with state-lock */
1822 default_unprepare (GstRTSPMedia * media)
1824 GstRTSPMediaPrivate *priv = media->priv;
1826 if (priv->eos_shutdown) {
1827 GST_DEBUG ("sending EOS for shutdown");
1828 /* ref so that we don't disappear */
1829 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
1830 /* we need to go to playing again for the EOS to propagate, normally in this
1831 * state, nothing is receiving data from us anymore so this is ok. */
1832 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1833 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
1835 finish_unprepare (media);
1841 * gst_rtsp_media_unprepare:
1842 * @media: a #GstRTSPMedia
1844 * Unprepare @media. After this call, the media should be prepared again before
1845 * it can be used again. If the media is set to be non-reusable, a new instance
1848 * Returns: %TRUE on success.
1851 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1853 GstRTSPMediaPrivate *priv;
1856 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1860 g_rec_mutex_lock (&priv->state_lock);
1861 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1862 goto was_unprepared;
1864 priv->prepare_count--;
1865 if (priv->prepare_count > 0)
1868 GST_INFO ("unprepare media %p", media);
1869 priv->target_state = GST_STATE_NULL;
1872 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
1873 GstRTSPMediaClass *klass;
1875 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1876 if (klass->unprepare)
1877 success = klass->unprepare (media);
1879 finish_unprepare (media);
1881 g_rec_mutex_unlock (&priv->state_lock);
1887 g_rec_mutex_unlock (&priv->state_lock);
1888 GST_INFO ("media %p was already unprepared", media);
1893 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
1894 g_rec_mutex_unlock (&priv->state_lock);
1899 /* should be called with state-lock */
1901 get_clock_unlocked (GstRTSPMedia * media)
1903 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
1904 GST_DEBUG_OBJECT (media, "media was not prepared");
1907 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
1911 * gst_rtsp_media_get_clock:
1912 * @media: a #GstRTSPMedia
1914 * Get the clock that is used by the pipeline in @media.
1916 * @media must be prepared before this method returns a valid clock object.
1918 * Returns: the #GstClock used by @media. unref after usage.
1921 gst_rtsp_media_get_clock (GstRTSPMedia * media)
1924 GstRTSPMediaPrivate *priv;
1926 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1930 g_rec_mutex_lock (&priv->state_lock);
1931 clock = get_clock_unlocked (media);
1932 g_rec_mutex_unlock (&priv->state_lock);
1938 * gst_rtsp_media_get_base_time:
1939 * @media: a #GstRTSPMedia
1941 * Get the base_time that is used by the pipeline in @media.
1943 * @media must be prepared before this method returns a valid base_time.
1945 * Returns: the base_time used by @media.
1948 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
1950 GstClockTime result;
1951 GstRTSPMediaPrivate *priv;
1953 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
1957 g_rec_mutex_lock (&priv->state_lock);
1958 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1961 result = gst_element_get_base_time (media->priv->pipeline);
1962 g_rec_mutex_unlock (&priv->state_lock);
1969 g_rec_mutex_unlock (&priv->state_lock);
1970 GST_DEBUG_OBJECT (media, "media was not prepared");
1971 return GST_CLOCK_TIME_NONE;
1976 * gst_rtsp_media_get_time_provider:
1977 * @media: a #GstRTSPMedia
1978 * @address: an address or NULL
1979 * @port: a port or 0
1981 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
1982 * will listen on @address and @port for client time requests.
1984 * Returns: the #GstNetTimeProvider of @media.
1986 GstNetTimeProvider *
1987 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
1990 GstRTSPMediaPrivate *priv;
1991 GstNetTimeProvider *provider = NULL;
1993 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1997 g_rec_mutex_lock (&priv->state_lock);
1998 if (priv->time_provider) {
1999 if ((provider = priv->nettime) == NULL) {
2002 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2003 provider = gst_net_time_provider_new (clock, address, port);
2004 gst_object_unref (clock);
2006 priv->nettime = provider;
2010 g_rec_mutex_unlock (&priv->state_lock);
2013 gst_object_ref (provider);
2019 * gst_rtsp_media_set_state:
2020 * @media: a #GstRTSPMedia
2021 * @state: the target state of the media
2022 * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
2024 * Set the state of @media to @state and for the transports in @transports.
2026 * @media must be prepared with gst_rtsp_media_prepare();
2028 * Returns: %TRUE on success.
2031 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2032 GPtrArray * transports)
2034 GstRTSPMediaPrivate *priv;
2036 gboolean activate, deactivate, do_state;
2039 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2040 g_return_val_if_fail (transports != NULL, FALSE);
2044 g_rec_mutex_lock (&priv->state_lock);
2045 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2048 /* NULL and READY are the same */
2049 if (state == GST_STATE_READY)
2050 state = GST_STATE_NULL;
2052 activate = deactivate = FALSE;
2054 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
2058 case GST_STATE_NULL:
2059 case GST_STATE_PAUSED:
2060 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
2061 if (priv->target_state == GST_STATE_PLAYING)
2064 case GST_STATE_PLAYING:
2065 /* we're going to PLAYING, activate */
2071 old_active = priv->n_active;
2073 for (i = 0; i < transports->len; i++) {
2074 GstRTSPStreamTransport *trans;
2076 /* we need a non-NULL entry in the array */
2077 trans = g_ptr_array_index (transports, i);
2082 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2084 } else if (deactivate) {
2085 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2090 /* we just activated the first media, do the playing state change */
2091 if (old_active == 0 && activate)
2093 /* if we have no more active media, do the downward state changes */
2094 else if (priv->n_active == 0)
2099 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2102 if (priv->target_state != state) {
2104 if (state == GST_STATE_NULL) {
2105 gst_rtsp_media_unprepare (media);
2107 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2109 priv->target_state = state;
2110 /* when we are buffering, don't update the state yet, this will be done
2111 * when buffering finishes */
2112 if (priv->buffering) {
2113 GST_INFO ("Buffering busy, delay state change");
2115 gst_element_set_state (priv->pipeline, state);
2119 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2123 /* remember where we are */
2124 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2125 old_active != priv->n_active))
2126 collect_media_stats (media);
2128 g_rec_mutex_unlock (&priv->state_lock);
2135 GST_WARNING ("media %p was not prepared", media);
2136 g_rec_mutex_unlock (&priv->state_lock);
2141 /* called with state-lock */
2143 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2144 GstRTSPRangeUnit unit)
2146 return gst_rtsp_range_convert_units (range, unit);
2150 default_query_position (GstRTSPMedia * media, gint64 * position)
2152 return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
2157 default_query_stop (GstRTSPMedia * media, gint64 * stop)
2162 query = gst_query_new_segment (GST_FORMAT_TIME);
2163 if ((res = gst_element_query (media->priv->pipeline, query))) {
2165 gst_query_parse_segment (query, NULL, &format, NULL, stop);
2166 if (format != GST_FORMAT_TIME)
2169 gst_query_unref (query);