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 */
39 GstRTSPLowerTrans protocols;
41 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 */
61 gboolean time_provider;
62 GstNetTimeProvider *nettime;
67 GstState target_state;
69 /* RTP session manager */
72 /* the range of media */
73 GstRTSPTimeRange range; /* protected by lock */
74 GstClockTime range_start;
75 GstClockTime range_stop;
78 #define DEFAULT_SHARED FALSE
79 #define DEFAULT_REUSABLE FALSE
80 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_TCP
81 //#define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP_MCAST
82 #define DEFAULT_EOS_SHUTDOWN FALSE
83 #define DEFAULT_BUFFER_SIZE 0x80000
84 #define DEFAULT_TIME_PROVIDER FALSE
86 /* define to dump received RTCP packets */
105 SIGNAL_REMOVED_STREAM,
112 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
113 #define GST_CAT_DEFAULT rtsp_media_debug
115 static void gst_rtsp_media_get_property (GObject * object, guint propid,
116 GValue * value, GParamSpec * pspec);
117 static void gst_rtsp_media_set_property (GObject * object, guint propid,
118 const GValue * value, GParamSpec * pspec);
119 static void gst_rtsp_media_finalize (GObject * obj);
121 static gpointer do_loop (GstRTSPMediaClass * klass);
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);
130 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
132 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
135 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
137 GObjectClass *gobject_class;
139 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
141 gobject_class = G_OBJECT_CLASS (klass);
143 gobject_class->get_property = gst_rtsp_media_get_property;
144 gobject_class->set_property = gst_rtsp_media_set_property;
145 gobject_class->finalize = gst_rtsp_media_finalize;
147 g_object_class_install_property (gobject_class, PROP_SHARED,
148 g_param_spec_boolean ("shared", "Shared",
149 "If this media pipeline can be shared", DEFAULT_SHARED,
150 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
152 g_object_class_install_property (gobject_class, PROP_REUSABLE,
153 g_param_spec_boolean ("reusable", "Reusable",
154 "If this media pipeline can be reused after an unprepare",
155 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
157 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
158 g_param_spec_flags ("protocols", "Protocols",
159 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
160 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
162 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
163 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
164 "Send an EOS event to the pipeline before unpreparing",
165 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
167 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
168 g_param_spec_uint ("buffer-size", "Buffer Size",
169 "The kernel UDP buffer size to use", 0, G_MAXUINT,
170 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
172 g_object_class_install_property (gobject_class, PROP_ELEMENT,
173 g_param_spec_object ("element", "The Element",
174 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
175 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
177 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
178 g_param_spec_boolean ("time-provider", "Time Provider",
179 "Use a NetTimeProvider for clients",
180 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
182 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
183 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
184 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
185 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
187 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
188 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
189 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
190 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
191 GST_TYPE_RTSP_STREAM);
193 gst_rtsp_media_signals[SIGNAL_PREPARED] =
194 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
195 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
196 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
198 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
199 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
200 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
201 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
203 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
204 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
205 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
206 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
208 klass->context = g_main_context_new ();
209 klass->loop = g_main_loop_new (klass->context, TRUE);
211 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
213 klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
215 klass->handle_message = default_handle_message;
216 klass->unprepare = default_unprepare;
217 klass->convert_range = default_convert_range;
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->auth);
263 g_object_unref (priv->pool);
264 g_mutex_clear (&priv->lock);
265 g_cond_clear (&priv->cond);
266 g_rec_mutex_clear (&priv->state_lock);
268 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
272 gst_rtsp_media_get_property (GObject * object, guint propid,
273 GValue * value, GParamSpec * pspec)
275 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
279 g_value_set_object (value, media->priv->element);
282 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
285 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
288 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
290 case PROP_EOS_SHUTDOWN:
291 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
293 case PROP_BUFFER_SIZE:
294 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
296 case PROP_TIME_PROVIDER:
297 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
300 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
305 gst_rtsp_media_set_property (GObject * object, guint propid,
306 const GValue * value, GParamSpec * pspec)
308 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
312 media->priv->element = g_value_get_object (value);
313 gst_object_ref_sink (media->priv->element);
316 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
319 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
322 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
324 case PROP_EOS_SHUTDOWN:
325 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
327 case PROP_BUFFER_SIZE:
328 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
330 case PROP_TIME_PROVIDER:
331 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
334 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
339 do_loop (GstRTSPMediaClass * klass)
341 GST_INFO ("enter mainloop");
342 g_main_loop_run (klass->loop);
343 GST_INFO ("exit mainloop");
348 /* must be called with state lock */
350 collect_media_stats (GstRTSPMedia * media)
352 GstRTSPMediaPrivate *priv = media->priv;
354 gint64 position, stop;
356 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
357 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
360 priv->range.unit = GST_RTSP_RANGE_NPT;
362 GST_INFO ("collect media stats");
365 priv->range.min.type = GST_RTSP_TIME_NOW;
366 priv->range.min.seconds = -1;
367 priv->range_start = -1;
368 priv->range.max.type = GST_RTSP_TIME_END;
369 priv->range.max.seconds = -1;
370 priv->range_stop = -1;
372 /* get the position */
373 if (!gst_element_query_position (priv->pipeline, GST_FORMAT_TIME,
375 GST_INFO ("position query failed");
379 /* get the current segment stop */
380 query = gst_query_new_segment (GST_FORMAT_TIME);
381 if (gst_element_query (priv->pipeline, query)) {
382 gst_query_parse_segment (query, NULL, NULL, NULL, &stop);
384 GST_INFO ("segment query failed");
387 gst_query_unref (query);
389 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
390 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
392 if (position == -1) {
393 priv->range.min.type = GST_RTSP_TIME_NOW;
394 priv->range.min.seconds = -1;
395 priv->range_start = -1;
397 priv->range.min.type = GST_RTSP_TIME_SECONDS;
398 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
399 priv->range_start = position;
402 priv->range.max.type = GST_RTSP_TIME_END;
403 priv->range.max.seconds = -1;
404 priv->range_stop = -1;
406 priv->range.max.type = GST_RTSP_TIME_SECONDS;
407 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
408 priv->range_stop = stop;
414 * gst_rtsp_media_new:
415 * @element: (transfer full): a #GstElement
417 * Create a new #GstRTSPMedia instance. @element is the bin element that
418 * provides the different streams. The #GstRTSPMedia object contains the
419 * element to produce RTP data for one or more related (audio/video/..)
422 * Ownership is taken of @element.
424 * Returns: a new #GstRTSPMedia object.
427 gst_rtsp_media_new (GstElement * element)
429 GstRTSPMedia *result;
431 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
433 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
439 * gst_rtsp_media_take_element:
440 * @media: a #GstRTSPMedia
441 * @pipeline: (transfer full): a #GstPipeline
443 * Set @pipeline as the #GstPipeline for @media. Ownership is
444 * taken of @pipeline.
447 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
449 GstRTSPMediaPrivate *priv;
451 GstNetTimeProvider *nettime;
453 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
454 g_return_if_fail (GST_IS_PIPELINE (pipeline));
458 g_mutex_lock (&priv->lock);
459 old = priv->pipeline;
460 priv->pipeline = GST_ELEMENT_CAST (pipeline);
461 nettime = priv->nettime;
462 priv->nettime = NULL;
463 g_mutex_unlock (&priv->lock);
466 gst_object_unref (old);
469 gst_object_unref (nettime);
471 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
475 * gst_rtsp_media_set_shared:
476 * @media: a #GstRTSPMedia
477 * @shared: the new value
479 * Set or unset if the pipeline for @media can be shared will multiple clients.
480 * When @shared is %TRUE, client requests for this media will share the media
484 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
486 GstRTSPMediaPrivate *priv;
488 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
492 g_mutex_lock (&priv->lock);
493 priv->shared = shared;
494 g_mutex_unlock (&priv->lock);
498 * gst_rtsp_media_is_shared:
499 * @media: a #GstRTSPMedia
501 * Check if the pipeline for @media can be shared between multiple clients.
503 * Returns: %TRUE if the media can be shared between clients.
506 gst_rtsp_media_is_shared (GstRTSPMedia * media)
508 GstRTSPMediaPrivate *priv;
511 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
515 g_mutex_lock (&priv->lock);
517 g_mutex_unlock (&priv->lock);
523 * gst_rtsp_media_set_reusable:
524 * @media: a #GstRTSPMedia
525 * @reusable: the new value
527 * Set or unset if the pipeline for @media can be reused after the pipeline has
531 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
533 GstRTSPMediaPrivate *priv;
535 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
539 g_mutex_lock (&priv->lock);
540 priv->reusable = reusable;
541 g_mutex_unlock (&priv->lock);
545 * gst_rtsp_media_is_reusable:
546 * @media: a #GstRTSPMedia
548 * Check if the pipeline for @media can be reused after an unprepare.
550 * Returns: %TRUE if the media can be reused
553 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
555 GstRTSPMediaPrivate *priv;
558 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
562 g_mutex_lock (&priv->lock);
563 res = priv->reusable;
564 g_mutex_unlock (&priv->lock);
570 * gst_rtsp_media_set_protocols:
571 * @media: a #GstRTSPMedia
572 * @protocols: the new flags
574 * Configure the allowed lower transport for @media.
577 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
579 GstRTSPMediaPrivate *priv;
581 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
585 g_mutex_lock (&priv->lock);
586 priv->protocols = protocols;
587 g_mutex_unlock (&priv->lock);
591 * gst_rtsp_media_get_protocols:
592 * @media: a #GstRTSPMedia
594 * Get the allowed protocols of @media.
596 * Returns: a #GstRTSPLowerTrans
599 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
601 GstRTSPMediaPrivate *priv;
602 GstRTSPLowerTrans res;
604 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
605 GST_RTSP_LOWER_TRANS_UNKNOWN);
609 g_mutex_lock (&priv->lock);
610 res = priv->protocols;
611 g_mutex_unlock (&priv->lock);
617 * gst_rtsp_media_set_eos_shutdown:
618 * @media: a #GstRTSPMedia
619 * @eos_shutdown: the new value
621 * Set or unset if an EOS event will be sent to the pipeline for @media before
625 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
627 GstRTSPMediaPrivate *priv;
629 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
633 g_mutex_lock (&priv->lock);
634 priv->eos_shutdown = eos_shutdown;
635 g_mutex_unlock (&priv->lock);
639 * gst_rtsp_media_is_eos_shutdown:
640 * @media: a #GstRTSPMedia
642 * Check if the pipeline for @media will send an EOS down the pipeline before
645 * Returns: %TRUE if the media will send EOS before unpreparing.
648 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
650 GstRTSPMediaPrivate *priv;
653 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
657 g_mutex_lock (&priv->lock);
658 res = priv->eos_shutdown;
659 g_mutex_unlock (&priv->lock);
665 * gst_rtsp_media_set_buffer_size:
666 * @media: a #GstRTSPMedia
667 * @size: the new value
669 * Set the kernel UDP buffer size.
672 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
674 GstRTSPMediaPrivate *priv;
676 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
678 GST_LOG_OBJECT (media, "set buffer size %u", size);
682 g_mutex_lock (&priv->lock);
683 priv->buffer_size = size;
684 g_mutex_unlock (&priv->lock);
688 * gst_rtsp_media_get_buffer_size:
689 * @media: a #GstRTSPMedia
691 * Get the kernel UDP buffer size.
693 * Returns: the kernel UDP buffer size.
696 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
698 GstRTSPMediaPrivate *priv;
701 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
705 g_mutex_unlock (&priv->lock);
706 res = priv->buffer_size;
707 g_mutex_unlock (&priv->lock);
713 * gst_rtsp_media_use_time_provider:
714 * @media: a #GstRTSPMedia
716 * Set @media to provide a GstNetTimeProvider.
719 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
721 GstRTSPMediaPrivate *priv;
723 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
727 g_mutex_lock (&priv->lock);
728 priv->time_provider = time_provider;
729 g_mutex_unlock (&priv->lock);
733 * gst_rtsp_media_is_time_provider:
734 * @media: a #GstRTSPMedia
736 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
738 * Use gst_rtsp_media_get_time_provider() to get the network clock.
740 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
743 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
745 GstRTSPMediaPrivate *priv;
748 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
752 g_mutex_unlock (&priv->lock);
753 res = priv->time_provider;
754 g_mutex_unlock (&priv->lock);
760 * gst_rtsp_media_set_auth:
761 * @media: a #GstRTSPMedia
762 * @auth: a #GstRTSPAuth
764 * configure @auth to be used as the authentication manager of @media.
767 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
769 GstRTSPMediaPrivate *priv;
772 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
776 GST_LOG_OBJECT (media, "set auth %p", auth);
778 g_mutex_lock (&priv->lock);
779 if ((old = priv->auth) != auth)
780 priv->auth = auth ? g_object_ref (auth) : NULL;
783 g_mutex_unlock (&priv->lock);
786 g_object_unref (old);
790 * gst_rtsp_media_get_auth:
791 * @media: a #GstRTSPMedia
793 * Get the #GstRTSPAuth used as the authentication manager of @media.
795 * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
799 gst_rtsp_media_get_auth (GstRTSPMedia * media)
801 GstRTSPMediaPrivate *priv;
804 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
808 g_mutex_lock (&priv->lock);
809 if ((result = priv->auth))
810 g_object_ref (result);
811 g_mutex_unlock (&priv->lock);
817 * gst_rtsp_media_set_address_pool:
818 * @media: a #GstRTSPMedia
819 * @pool: a #GstRTSPAddressPool
821 * configure @pool to be used as the address pool of @media.
824 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
825 GstRTSPAddressPool * pool)
827 GstRTSPMediaPrivate *priv;
828 GstRTSPAddressPool *old;
830 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
834 GST_LOG_OBJECT (media, "set address pool %p", pool);
836 g_mutex_lock (&priv->lock);
837 if ((old = priv->pool) != pool)
838 priv->pool = pool ? g_object_ref (pool) : NULL;
841 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
843 g_mutex_unlock (&priv->lock);
846 g_object_unref (old);
850 * gst_rtsp_media_get_address_pool:
851 * @media: a #GstRTSPMedia
853 * Get the #GstRTSPAddressPool used as the address pool of @media.
855 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
859 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
861 GstRTSPMediaPrivate *priv;
862 GstRTSPAddressPool *result;
864 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
868 g_mutex_lock (&priv->lock);
869 if ((result = priv->pool))
870 g_object_ref (result);
871 g_mutex_unlock (&priv->lock);
877 * gst_rtsp_media_collect_streams:
878 * @media: a #GstRTSPMedia
880 * Find all payloader elements, they should be named pay%d in the
881 * element of @media, and create #GstRTSPStreams for them.
883 * Collect all dynamic elements, named dynpay%d, and add them to
884 * the list of dynamic elements.
887 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
889 GstRTSPMediaPrivate *priv;
890 GstElement *element, *elem;
895 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
898 element = priv->element;
901 for (i = 0; have_elem; i++) {
906 name = g_strdup_printf ("pay%d", i);
907 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
908 GST_INFO ("found stream %d with payloader %p", i, elem);
910 /* take the pad of the payloader */
911 pad = gst_element_get_static_pad (elem, "src");
912 /* create the stream */
913 gst_rtsp_media_create_stream (media, elem, pad);
914 gst_object_unref (pad);
915 gst_object_unref (elem);
921 name = g_strdup_printf ("dynpay%d", i);
922 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
923 /* a stream that will dynamically create pads to provide RTP packets */
925 GST_INFO ("found dynamic element %d, %p", i, elem);
927 g_mutex_lock (&priv->lock);
928 priv->dynamic = g_list_prepend (priv->dynamic, elem);
929 g_mutex_unlock (&priv->lock);
938 * gst_rtsp_media_create_stream:
939 * @media: a #GstRTSPMedia
940 * @payloader: a #GstElement
941 * @srcpad: a source #GstPad
943 * Create a new stream in @media that provides RTP data on @srcpad.
944 * @srcpad should be a pad of an element inside @media->element.
946 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
950 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
953 GstRTSPMediaPrivate *priv;
954 GstRTSPStream *stream;
959 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
960 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
961 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
962 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
966 g_mutex_lock (&priv->lock);
967 idx = priv->streams->len;
969 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
971 name = g_strdup_printf ("src_%u", idx);
972 srcpad = gst_ghost_pad_new (name, pad);
973 gst_pad_set_active (srcpad, TRUE);
974 gst_element_add_pad (priv->element, srcpad);
977 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
979 gst_rtsp_stream_set_address_pool (stream, priv->pool);
981 g_ptr_array_add (priv->streams, stream);
982 g_mutex_unlock (&priv->lock);
984 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
991 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
993 GstRTSPMediaPrivate *priv;
998 g_mutex_lock (&priv->lock);
999 /* remove the ghostpad */
1000 srcpad = gst_rtsp_stream_get_srcpad (stream);
1001 gst_element_remove_pad (priv->element, srcpad);
1002 gst_object_unref (srcpad);
1003 /* now remove the stream */
1004 g_object_ref (stream);
1005 g_ptr_array_remove (priv->streams, stream);
1006 g_mutex_unlock (&priv->lock);
1008 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1011 g_object_unref (stream);
1015 * gst_rtsp_media_n_streams:
1016 * @media: a #GstRTSPMedia
1018 * Get the number of streams in this media.
1020 * Returns: The number of streams.
1023 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1025 GstRTSPMediaPrivate *priv;
1028 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1032 g_mutex_lock (&priv->lock);
1033 res = priv->streams->len;
1034 g_mutex_unlock (&priv->lock);
1040 * gst_rtsp_media_get_stream:
1041 * @media: a #GstRTSPMedia
1042 * @idx: the stream index
1044 * Retrieve the stream with index @idx from @media.
1046 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1047 * that index did not exist.
1050 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1052 GstRTSPMediaPrivate *priv;
1055 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1059 g_mutex_lock (&priv->lock);
1060 if (idx < priv->streams->len)
1061 res = g_ptr_array_index (priv->streams, idx);
1064 g_mutex_unlock (&priv->lock);
1070 * gst_rtsp_media_get_range_string:
1071 * @media: a #GstRTSPMedia
1072 * @play: for the PLAY request
1073 * @unit: the unit to use for the string
1075 * Get the current range as a string. @media must be prepared with
1076 * gst_rtsp_media_prepare ().
1078 * Returns: The range as a string, g_free() after usage.
1081 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1082 GstRTSPRangeUnit unit)
1084 GstRTSPMediaClass *klass;
1085 GstRTSPMediaPrivate *priv;
1087 GstRTSPTimeRange range;
1089 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1090 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1091 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1095 g_rec_mutex_lock (&priv->state_lock);
1096 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1099 g_mutex_lock (&priv->lock);
1101 range = priv->range;
1103 if (!play && priv->n_active > 0) {
1104 range.min.type = GST_RTSP_TIME_NOW;
1105 range.min.seconds = -1;
1107 g_mutex_unlock (&priv->lock);
1108 g_rec_mutex_unlock (&priv->state_lock);
1110 if (!klass->convert_range (media, &range, unit)) {
1111 goto conversion_failed;
1114 result = gst_rtsp_range_to_string (&range);
1121 GST_WARNING ("media %p was not prepared", media);
1122 g_rec_mutex_unlock (&priv->state_lock);
1127 GST_WARNING ("range conversion to unit %d failed", unit);
1128 g_rec_mutex_unlock (&priv->state_lock);
1134 * gst_rtsp_media_seek:
1135 * @media: a #GstRTSPMedia
1136 * @range: a #GstRTSPTimeRange
1138 * Seek the pipeline of @media to @range. @media must be prepared with
1139 * gst_rtsp_media_prepare().
1141 * Returns: %TRUE on success.
1144 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1146 GstRTSPMediaClass *klass;
1147 GstRTSPMediaPrivate *priv;
1150 GstClockTime start, stop;
1151 GstSeekType start_type, stop_type;
1153 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1155 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1156 g_return_val_if_fail (range != NULL, FALSE);
1157 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1161 g_rec_mutex_lock (&priv->state_lock);
1162 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1165 if (!priv->seekable)
1168 /* depends on the current playing state of the pipeline. We might need to
1169 * queue this until we get EOS. */
1170 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
1172 start_type = stop_type = GST_SEEK_TYPE_NONE;
1174 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1176 gst_rtsp_range_get_times (range, &start, &stop);
1178 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1179 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1180 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1181 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1183 if (priv->range_start == start)
1184 start = GST_CLOCK_TIME_NONE;
1185 else if (start != GST_CLOCK_TIME_NONE)
1186 start_type = GST_SEEK_TYPE_SET;
1188 if (priv->range_stop == stop)
1189 stop = GST_CLOCK_TIME_NONE;
1190 else if (stop != GST_CLOCK_TIME_NONE)
1191 stop_type = GST_SEEK_TYPE_SET;
1193 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1194 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1195 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1197 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1198 flags, start_type, start, stop_type, stop);
1200 /* and block for the seek to complete */
1201 GST_INFO ("done seeking %d", res);
1202 gst_element_get_state (priv->pipeline, NULL, NULL, -1);
1203 GST_INFO ("prerolled again");
1205 collect_media_stats (media);
1207 GST_INFO ("no seek needed");
1210 g_rec_mutex_unlock (&priv->state_lock);
1217 g_rec_mutex_unlock (&priv->state_lock);
1218 GST_INFO ("media %p is not prepared", media);
1223 g_rec_mutex_unlock (&priv->state_lock);
1224 GST_INFO ("pipeline is not seekable");
1229 g_rec_mutex_unlock (&priv->state_lock);
1230 GST_WARNING ("conversion to npt not supported");
1236 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1238 GstRTSPMediaPrivate *priv = media->priv;
1240 g_mutex_lock (&priv->lock);
1241 priv->status = status;
1242 GST_DEBUG ("setting new status to %d", status);
1243 g_cond_broadcast (&priv->cond);
1244 g_mutex_unlock (&priv->lock);
1248 * gst_rtsp_media_get_status:
1249 * @media: a #GstRTSPMedia
1251 * Get the status of @media. When @media is busy preparing, this function waits
1252 * until @media is prepared or in error.
1254 * Returns: the status of @media.
1257 gst_rtsp_media_get_status (GstRTSPMedia * media)
1259 GstRTSPMediaPrivate *priv = media->priv;
1260 GstRTSPMediaStatus result;
1263 g_mutex_lock (&priv->lock);
1264 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1265 /* while we are preparing, wait */
1266 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1267 GST_DEBUG ("waiting for status change");
1268 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1269 GST_DEBUG ("timeout, assuming error status");
1270 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1273 /* could be success or error */
1274 result = priv->status;
1275 GST_DEBUG ("got status %d", result);
1276 g_mutex_unlock (&priv->lock);
1281 /* called with state-lock */
1283 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1285 GstRTSPMediaPrivate *priv = media->priv;
1286 GstMessageType type;
1288 type = GST_MESSAGE_TYPE (message);
1291 case GST_MESSAGE_STATE_CHANGED:
1293 case GST_MESSAGE_BUFFERING:
1297 gst_message_parse_buffering (message, &percent);
1299 /* no state management needed for live pipelines */
1303 if (percent == 100) {
1304 /* a 100% message means buffering is done */
1305 priv->buffering = FALSE;
1306 /* if the desired state is playing, go back */
1307 if (priv->target_state == GST_STATE_PLAYING) {
1308 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1309 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1311 GST_INFO ("Buffering done");
1314 /* buffering busy */
1315 if (priv->buffering == FALSE) {
1316 if (priv->target_state == GST_STATE_PLAYING) {
1317 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1318 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1319 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1321 GST_INFO ("Buffering ...");
1324 priv->buffering = TRUE;
1328 case GST_MESSAGE_LATENCY:
1330 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1333 case GST_MESSAGE_ERROR:
1338 gst_message_parse_error (message, &gerror, &debug);
1339 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1340 g_error_free (gerror);
1343 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1346 case GST_MESSAGE_WARNING:
1351 gst_message_parse_warning (message, &gerror, &debug);
1352 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1353 g_error_free (gerror);
1357 case GST_MESSAGE_ELEMENT:
1359 case GST_MESSAGE_STREAM_STATUS:
1361 case GST_MESSAGE_ASYNC_DONE:
1363 /* when we are dynamically adding pads, the addition of the udpsrc will
1364 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1365 * wait for the final ASYNC_DONE after everything prerolled */
1366 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1368 GST_INFO ("%p: got ASYNC_DONE", media);
1369 collect_media_stats (media);
1371 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1372 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1375 case GST_MESSAGE_EOS:
1376 GST_INFO ("%p: got EOS", media);
1378 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1379 GST_DEBUG ("shutting down after EOS");
1380 finish_unprepare (media);
1384 GST_INFO ("%p: got message type %d (%s)", media, type,
1385 gst_message_type_get_name (type));
1392 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1394 GstRTSPMediaPrivate *priv = media->priv;
1395 GstRTSPMediaClass *klass;
1398 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1400 g_rec_mutex_lock (&priv->state_lock);
1401 if (klass->handle_message)
1402 ret = klass->handle_message (media, message);
1405 g_rec_mutex_unlock (&priv->state_lock);
1411 watch_destroyed (GstRTSPMedia * media)
1413 GST_DEBUG_OBJECT (media, "source destroyed");
1414 g_object_unref (media);
1417 /* called from streaming threads */
1419 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1421 GstRTSPMediaPrivate *priv = media->priv;
1422 GstRTSPStream *stream;
1424 /* FIXME, element is likely not a payloader, find the payloader here */
1425 stream = gst_rtsp_media_create_stream (media, element, pad);
1427 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1429 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1431 g_rec_mutex_lock (&priv->state_lock);
1432 /* we will be adding elements below that will cause ASYNC_DONE to be
1433 * posted in the bus. We want to ignore those messages until the
1434 * pipeline really prerolled. */
1435 priv->adding = TRUE;
1437 /* join the element in the PAUSED state because this callback is
1438 * called from the streaming thread and it is PAUSED */
1439 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1440 priv->rtpbin, GST_STATE_PAUSED);
1442 priv->adding = FALSE;
1443 g_rec_mutex_unlock (&priv->state_lock);
1447 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1449 GstRTSPMediaPrivate *priv = media->priv;
1450 GstRTSPStream *stream;
1452 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1456 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1458 g_rec_mutex_lock (&priv->state_lock);
1459 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1460 g_rec_mutex_unlock (&priv->state_lock);
1462 gst_rtsp_media_remove_stream (media, stream);
1466 remove_fakesink (GstRTSPMediaPrivate * priv)
1468 GstElement *fakesink;
1470 g_mutex_lock (&priv->lock);
1471 if ((fakesink = priv->fakesink))
1472 gst_object_ref (fakesink);
1473 priv->fakesink = NULL;
1474 g_mutex_unlock (&priv->lock);
1477 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1478 gst_element_set_state (fakesink, GST_STATE_NULL);
1479 gst_object_unref (fakesink);
1480 GST_INFO ("removed fakesink");
1485 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1487 GstRTSPMediaPrivate *priv = media->priv;
1489 GST_INFO ("no more pads");
1490 remove_fakesink (priv);
1493 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1495 struct _DynPaySignalHandlers
1497 gulong pad_added_handler;
1498 gulong pad_removed_handler;
1499 gulong no_more_pads_handler;
1503 * gst_rtsp_media_prepare:
1504 * @media: a #GstRTSPMedia
1506 * Prepare @media for streaming. This function will create the objects
1507 * to manage the streaming. A pipeline must have been set on @media with
1508 * gst_rtsp_media_take_pipeline().
1510 * It will preroll the pipeline and collect vital information about the streams
1511 * such as the duration.
1513 * Returns: %TRUE on success.
1516 gst_rtsp_media_prepare (GstRTSPMedia * media)
1518 GstRTSPMediaPrivate *priv;
1519 GstStateChangeReturn ret;
1520 GstRTSPMediaStatus status;
1522 GstRTSPMediaClass *klass;
1526 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1530 g_rec_mutex_lock (&priv->state_lock);
1531 priv->prepare_count++;
1533 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1536 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1539 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1540 goto not_unprepared;
1542 if (!priv->reusable && priv->reused)
1545 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1546 if (priv->rtpbin == NULL)
1549 GST_INFO ("preparing media %p", media);
1551 /* reset some variables */
1552 priv->is_live = FALSE;
1553 priv->seekable = FALSE;
1554 priv->buffering = FALSE;
1555 /* we're preparing now */
1556 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1558 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1560 /* add the pipeline bus to our custom mainloop */
1561 priv->source = gst_bus_create_watch (bus);
1562 gst_object_unref (bus);
1564 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1565 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1567 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1568 priv->id = g_source_attach (priv->source, klass->context);
1570 /* add stuff to the bin */
1571 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1573 /* link streams we already have, other streams might appear when we have
1574 * dynamic elements */
1575 for (i = 0; i < priv->streams->len; i++) {
1576 GstRTSPStream *stream;
1578 stream = g_ptr_array_index (priv->streams, i);
1580 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1581 priv->rtpbin, GST_STATE_NULL);
1584 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1585 GstElement *elem = walk->data;
1586 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
1588 GST_INFO ("adding callbacks for dynamic element %p", elem);
1590 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
1591 (GCallback) pad_added_cb, media);
1592 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
1593 (GCallback) pad_removed_cb, media);
1594 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
1595 (GCallback) no_more_pads_cb, media);
1597 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
1599 /* we add a fakesink here in order to make the state change async. We remove
1600 * the fakesink again in the no-more-pads callback. */
1601 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1602 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1605 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1606 /* first go to PAUSED */
1607 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1608 priv->target_state = GST_STATE_PAUSED;
1611 case GST_STATE_CHANGE_SUCCESS:
1612 GST_INFO ("SUCCESS state change for media %p", media);
1613 priv->seekable = TRUE;
1615 case GST_STATE_CHANGE_ASYNC:
1616 GST_INFO ("ASYNC state change for media %p", media);
1617 priv->seekable = TRUE;
1619 case GST_STATE_CHANGE_NO_PREROLL:
1620 /* we need to go to PLAYING */
1621 GST_INFO ("NO_PREROLL state change: live media %p", media);
1622 /* FIXME we disable seeking for live streams for now. We should perform a
1623 * seeking query in preroll instead */
1624 priv->seekable = FALSE;
1625 priv->is_live = TRUE;
1626 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1627 if (ret == GST_STATE_CHANGE_FAILURE)
1630 case GST_STATE_CHANGE_FAILURE:
1634 g_rec_mutex_unlock (&priv->state_lock);
1636 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1637 * able to do this async so that we don't block the server thread. */
1638 status = gst_rtsp_media_get_status (media);
1639 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1642 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1644 GST_INFO ("object %p is prerolled", media);
1651 GST_LOG ("media %p was prepared", media);
1652 g_rec_mutex_unlock (&priv->state_lock);
1658 GST_WARNING ("media %p was not unprepared", media);
1659 priv->prepare_count--;
1660 g_rec_mutex_unlock (&priv->state_lock);
1665 priv->prepare_count--;
1666 g_rec_mutex_unlock (&priv->state_lock);
1667 GST_WARNING ("can not reuse media %p", media);
1672 priv->prepare_count--;
1673 g_rec_mutex_unlock (&priv->state_lock);
1674 GST_WARNING ("no rtpbin element");
1675 g_warning ("failed to create element 'rtpbin', check your installation");
1680 GST_WARNING ("failed to preroll pipeline");
1681 gst_rtsp_media_unprepare (media);
1682 g_rec_mutex_unlock (&priv->state_lock);
1687 /* must be called with state-lock */
1689 finish_unprepare (GstRTSPMedia * media)
1691 GstRTSPMediaPrivate *priv = media->priv;
1695 GST_DEBUG ("shutting down");
1697 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1698 remove_fakesink (priv);
1700 for (i = 0; i < priv->streams->len; i++) {
1701 GstRTSPStream *stream;
1703 GST_INFO ("Removing elements of stream %d from pipeline", i);
1705 stream = g_ptr_array_index (priv->streams, i);
1707 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1710 /* remove the pad signal handlers */
1711 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1712 GstElement *elem = walk->data;
1713 DynPaySignalHandlers *handlers;
1716 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
1717 g_assert (handlers != NULL);
1719 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
1720 g_signal_handler_disconnect (G_OBJECT (elem),
1721 handlers->pad_removed_handler);
1722 g_signal_handler_disconnect (G_OBJECT (elem),
1723 handlers->no_more_pads_handler);
1725 g_slice_free (DynPaySignalHandlers, handlers);
1728 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1729 priv->rtpbin = NULL;
1732 gst_object_unref (priv->nettime);
1733 priv->nettime = NULL;
1735 priv->reused = TRUE;
1736 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1738 /* when the media is not reusable, this will effectively unref the media and
1740 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1742 /* the source has the last ref to the media */
1744 GST_DEBUG ("destroy source");
1745 g_source_destroy (priv->source);
1746 g_source_unref (priv->source);
1750 /* called with state-lock */
1752 default_unprepare (GstRTSPMedia * media)
1754 GstRTSPMediaPrivate *priv = media->priv;
1756 if (priv->eos_shutdown) {
1757 GST_DEBUG ("sending EOS for shutdown");
1758 /* ref so that we don't disappear */
1759 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
1760 /* we need to go to playing again for the EOS to propagate, normally in this
1761 * state, nothing is receiving data from us anymore so this is ok. */
1762 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1763 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
1765 finish_unprepare (media);
1771 * gst_rtsp_media_unprepare:
1772 * @media: a #GstRTSPMedia
1774 * Unprepare @media. After this call, the media should be prepared again before
1775 * it can be used again. If the media is set to be non-reusable, a new instance
1778 * Returns: %TRUE on success.
1781 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1783 GstRTSPMediaPrivate *priv;
1786 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1790 g_rec_mutex_lock (&priv->state_lock);
1791 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1792 goto was_unprepared;
1794 priv->prepare_count--;
1795 if (priv->prepare_count > 0)
1798 GST_INFO ("unprepare media %p", media);
1799 priv->target_state = GST_STATE_NULL;
1802 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
1803 GstRTSPMediaClass *klass;
1805 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1806 if (klass->unprepare)
1807 success = klass->unprepare (media);
1809 finish_unprepare (media);
1811 g_rec_mutex_unlock (&priv->state_lock);
1817 g_rec_mutex_unlock (&priv->state_lock);
1818 GST_INFO ("media %p was already unprepared", media);
1823 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
1824 g_rec_mutex_unlock (&priv->state_lock);
1829 /* should be called with state-lock */
1831 get_clock_unlocked (GstRTSPMedia * media)
1833 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
1834 GST_DEBUG_OBJECT (media, "media was not prepared");
1837 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
1841 * gst_rtsp_media_get_clock:
1842 * @media: a #GstRTSPMedia
1844 * Get the clock that is used by the pipeline in @media.
1846 * @media must be prepared before this method returns a valid clock object.
1848 * Returns: the #GstClock used by @media. unref after usage.
1851 gst_rtsp_media_get_clock (GstRTSPMedia * media)
1854 GstRTSPMediaPrivate *priv;
1856 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1860 g_rec_mutex_lock (&priv->state_lock);
1861 clock = get_clock_unlocked (media);
1862 g_rec_mutex_unlock (&priv->state_lock);
1868 * gst_rtsp_media_get_base_time:
1869 * @media: a #GstRTSPMedia
1871 * Get the base_time that is used by the pipeline in @media.
1873 * @media must be prepared before this method returns a valid base_time.
1875 * Returns: the base_time used by @media.
1878 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
1880 GstClockTime result;
1881 GstRTSPMediaPrivate *priv;
1883 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
1887 g_rec_mutex_lock (&priv->state_lock);
1888 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1891 result = gst_element_get_base_time (media->priv->pipeline);
1892 g_rec_mutex_unlock (&priv->state_lock);
1899 g_rec_mutex_unlock (&priv->state_lock);
1900 GST_DEBUG_OBJECT (media, "media was not prepared");
1901 return GST_CLOCK_TIME_NONE;
1906 * gst_rtsp_media_get_time_provider:
1907 * @media: a #GstRTSPMedia
1908 * @address: an address or NULL
1909 * @port: a port or 0
1911 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
1912 * will listen on @address and @port for client time requests.
1914 * Returns: the #GstNetTimeProvider of @media.
1916 GstNetTimeProvider *
1917 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
1920 GstRTSPMediaPrivate *priv;
1921 GstNetTimeProvider *provider = NULL;
1923 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1927 g_rec_mutex_lock (&priv->state_lock);
1928 if (priv->time_provider) {
1929 if ((provider = priv->nettime) == NULL) {
1932 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
1933 provider = gst_net_time_provider_new (clock, address, port);
1934 gst_object_unref (clock);
1936 priv->nettime = provider;
1940 g_rec_mutex_unlock (&priv->state_lock);
1943 gst_object_ref (provider);
1949 * gst_rtsp_media_set_state:
1950 * @media: a #GstRTSPMedia
1951 * @state: the target state of the media
1952 * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
1954 * Set the state of @media to @state and for the transports in @transports.
1956 * @media must be prepared with gst_rtsp_media_prepare();
1958 * Returns: %TRUE on success.
1961 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1962 GPtrArray * transports)
1964 GstRTSPMediaPrivate *priv;
1966 gboolean activate, deactivate, do_state;
1969 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1970 g_return_val_if_fail (transports != NULL, FALSE);
1974 g_rec_mutex_lock (&priv->state_lock);
1975 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1978 /* NULL and READY are the same */
1979 if (state == GST_STATE_READY)
1980 state = GST_STATE_NULL;
1982 activate = deactivate = FALSE;
1984 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1988 case GST_STATE_NULL:
1989 case GST_STATE_PAUSED:
1990 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
1991 if (priv->target_state == GST_STATE_PLAYING)
1994 case GST_STATE_PLAYING:
1995 /* we're going to PLAYING, activate */
2001 old_active = priv->n_active;
2003 for (i = 0; i < transports->len; i++) {
2004 GstRTSPStreamTransport *trans;
2006 /* we need a non-NULL entry in the array */
2007 trans = g_ptr_array_index (transports, i);
2012 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2014 } else if (deactivate) {
2015 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2020 /* we just activated the first media, do the playing state change */
2021 if (old_active == 0 && activate)
2023 /* if we have no more active media, do the downward state changes */
2024 else if (priv->n_active == 0)
2029 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2032 if (priv->target_state != state) {
2034 if (state == GST_STATE_NULL) {
2035 gst_rtsp_media_unprepare (media);
2037 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2039 priv->target_state = state;
2040 /* when we are buffering, don't update the state yet, this will be done
2041 * when buffering finishes */
2042 if (priv->buffering) {
2043 GST_INFO ("Buffering busy, delay state change");
2045 gst_element_set_state (priv->pipeline, state);
2049 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2053 /* remember where we are */
2054 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2055 old_active != priv->n_active))
2056 collect_media_stats (media);
2058 g_rec_mutex_unlock (&priv->state_lock);
2065 GST_WARNING ("media %p was not prepared", media);
2066 g_rec_mutex_unlock (&priv->state_lock);
2071 /* called with state-lock */
2073 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2074 GstRTSPRangeUnit unit)
2076 return gst_rtsp_range_convert_units (range, unit);