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)) {
383 gst_query_parse_segment (query, NULL, &format, NULL, &stop);
384 if (format != GST_FORMAT_TIME)
387 GST_INFO ("segment query failed");
390 gst_query_unref (query);
392 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
393 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
395 if (position == -1) {
396 priv->range.min.type = GST_RTSP_TIME_NOW;
397 priv->range.min.seconds = -1;
398 priv->range_start = -1;
400 priv->range.min.type = GST_RTSP_TIME_SECONDS;
401 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
402 priv->range_start = position;
405 priv->range.max.type = GST_RTSP_TIME_END;
406 priv->range.max.seconds = -1;
407 priv->range_stop = -1;
409 priv->range.max.type = GST_RTSP_TIME_SECONDS;
410 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
411 priv->range_stop = stop;
417 * gst_rtsp_media_new:
418 * @element: (transfer full): a #GstElement
420 * Create a new #GstRTSPMedia instance. @element is the bin element that
421 * provides the different streams. The #GstRTSPMedia object contains the
422 * element to produce RTP data for one or more related (audio/video/..)
425 * Ownership is taken of @element.
427 * Returns: a new #GstRTSPMedia object.
430 gst_rtsp_media_new (GstElement * element)
432 GstRTSPMedia *result;
434 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
436 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
442 * gst_rtsp_media_take_element:
443 * @media: a #GstRTSPMedia
444 * @pipeline: (transfer full): a #GstPipeline
446 * Set @pipeline as the #GstPipeline for @media. Ownership is
447 * taken of @pipeline.
450 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
452 GstRTSPMediaPrivate *priv;
454 GstNetTimeProvider *nettime;
456 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
457 g_return_if_fail (GST_IS_PIPELINE (pipeline));
461 g_mutex_lock (&priv->lock);
462 old = priv->pipeline;
463 priv->pipeline = GST_ELEMENT_CAST (pipeline);
464 nettime = priv->nettime;
465 priv->nettime = NULL;
466 g_mutex_unlock (&priv->lock);
469 gst_object_unref (old);
472 gst_object_unref (nettime);
474 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
478 * gst_rtsp_media_set_shared:
479 * @media: a #GstRTSPMedia
480 * @shared: the new value
482 * Set or unset if the pipeline for @media can be shared will multiple clients.
483 * When @shared is %TRUE, client requests for this media will share the media
487 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
489 GstRTSPMediaPrivate *priv;
491 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
495 g_mutex_lock (&priv->lock);
496 priv->shared = shared;
497 g_mutex_unlock (&priv->lock);
501 * gst_rtsp_media_is_shared:
502 * @media: a #GstRTSPMedia
504 * Check if the pipeline for @media can be shared between multiple clients.
506 * Returns: %TRUE if the media can be shared between clients.
509 gst_rtsp_media_is_shared (GstRTSPMedia * media)
511 GstRTSPMediaPrivate *priv;
514 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
518 g_mutex_lock (&priv->lock);
520 g_mutex_unlock (&priv->lock);
526 * gst_rtsp_media_set_reusable:
527 * @media: a #GstRTSPMedia
528 * @reusable: the new value
530 * Set or unset if the pipeline for @media can be reused after the pipeline has
534 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
536 GstRTSPMediaPrivate *priv;
538 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
542 g_mutex_lock (&priv->lock);
543 priv->reusable = reusable;
544 g_mutex_unlock (&priv->lock);
548 * gst_rtsp_media_is_reusable:
549 * @media: a #GstRTSPMedia
551 * Check if the pipeline for @media can be reused after an unprepare.
553 * Returns: %TRUE if the media can be reused
556 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
558 GstRTSPMediaPrivate *priv;
561 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
565 g_mutex_lock (&priv->lock);
566 res = priv->reusable;
567 g_mutex_unlock (&priv->lock);
573 * gst_rtsp_media_set_protocols:
574 * @media: a #GstRTSPMedia
575 * @protocols: the new flags
577 * Configure the allowed lower transport for @media.
580 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
582 GstRTSPMediaPrivate *priv;
584 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
588 g_mutex_lock (&priv->lock);
589 priv->protocols = protocols;
590 g_mutex_unlock (&priv->lock);
594 * gst_rtsp_media_get_protocols:
595 * @media: a #GstRTSPMedia
597 * Get the allowed protocols of @media.
599 * Returns: a #GstRTSPLowerTrans
602 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
604 GstRTSPMediaPrivate *priv;
605 GstRTSPLowerTrans res;
607 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
608 GST_RTSP_LOWER_TRANS_UNKNOWN);
612 g_mutex_lock (&priv->lock);
613 res = priv->protocols;
614 g_mutex_unlock (&priv->lock);
620 * gst_rtsp_media_set_eos_shutdown:
621 * @media: a #GstRTSPMedia
622 * @eos_shutdown: the new value
624 * Set or unset if an EOS event will be sent to the pipeline for @media before
628 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
630 GstRTSPMediaPrivate *priv;
632 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
636 g_mutex_lock (&priv->lock);
637 priv->eos_shutdown = eos_shutdown;
638 g_mutex_unlock (&priv->lock);
642 * gst_rtsp_media_is_eos_shutdown:
643 * @media: a #GstRTSPMedia
645 * Check if the pipeline for @media will send an EOS down the pipeline before
648 * Returns: %TRUE if the media will send EOS before unpreparing.
651 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
653 GstRTSPMediaPrivate *priv;
656 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
660 g_mutex_lock (&priv->lock);
661 res = priv->eos_shutdown;
662 g_mutex_unlock (&priv->lock);
668 * gst_rtsp_media_set_buffer_size:
669 * @media: a #GstRTSPMedia
670 * @size: the new value
672 * Set the kernel UDP buffer size.
675 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
677 GstRTSPMediaPrivate *priv;
679 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
681 GST_LOG_OBJECT (media, "set buffer size %u", size);
685 g_mutex_lock (&priv->lock);
686 priv->buffer_size = size;
687 g_mutex_unlock (&priv->lock);
691 * gst_rtsp_media_get_buffer_size:
692 * @media: a #GstRTSPMedia
694 * Get the kernel UDP buffer size.
696 * Returns: the kernel UDP buffer size.
699 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
701 GstRTSPMediaPrivate *priv;
704 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
708 g_mutex_unlock (&priv->lock);
709 res = priv->buffer_size;
710 g_mutex_unlock (&priv->lock);
716 * gst_rtsp_media_use_time_provider:
717 * @media: a #GstRTSPMedia
719 * Set @media to provide a GstNetTimeProvider.
722 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
724 GstRTSPMediaPrivate *priv;
726 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
730 g_mutex_lock (&priv->lock);
731 priv->time_provider = time_provider;
732 g_mutex_unlock (&priv->lock);
736 * gst_rtsp_media_is_time_provider:
737 * @media: a #GstRTSPMedia
739 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
741 * Use gst_rtsp_media_get_time_provider() to get the network clock.
743 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
746 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
748 GstRTSPMediaPrivate *priv;
751 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
755 g_mutex_unlock (&priv->lock);
756 res = priv->time_provider;
757 g_mutex_unlock (&priv->lock);
763 * gst_rtsp_media_set_auth:
764 * @media: a #GstRTSPMedia
765 * @auth: a #GstRTSPAuth
767 * configure @auth to be used as the authentication manager of @media.
770 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
772 GstRTSPMediaPrivate *priv;
775 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
779 GST_LOG_OBJECT (media, "set auth %p", auth);
781 g_mutex_lock (&priv->lock);
782 if ((old = priv->auth) != auth)
783 priv->auth = auth ? g_object_ref (auth) : NULL;
786 g_mutex_unlock (&priv->lock);
789 g_object_unref (old);
793 * gst_rtsp_media_get_auth:
794 * @media: a #GstRTSPMedia
796 * Get the #GstRTSPAuth used as the authentication manager of @media.
798 * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
802 gst_rtsp_media_get_auth (GstRTSPMedia * media)
804 GstRTSPMediaPrivate *priv;
807 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
811 g_mutex_lock (&priv->lock);
812 if ((result = priv->auth))
813 g_object_ref (result);
814 g_mutex_unlock (&priv->lock);
820 * gst_rtsp_media_set_address_pool:
821 * @media: a #GstRTSPMedia
822 * @pool: a #GstRTSPAddressPool
824 * configure @pool to be used as the address pool of @media.
827 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
828 GstRTSPAddressPool * pool)
830 GstRTSPMediaPrivate *priv;
831 GstRTSPAddressPool *old;
833 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
837 GST_LOG_OBJECT (media, "set address pool %p", pool);
839 g_mutex_lock (&priv->lock);
840 if ((old = priv->pool) != pool)
841 priv->pool = pool ? g_object_ref (pool) : NULL;
844 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
846 g_mutex_unlock (&priv->lock);
849 g_object_unref (old);
853 * gst_rtsp_media_get_address_pool:
854 * @media: a #GstRTSPMedia
856 * Get the #GstRTSPAddressPool used as the address pool of @media.
858 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
862 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
864 GstRTSPMediaPrivate *priv;
865 GstRTSPAddressPool *result;
867 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
871 g_mutex_lock (&priv->lock);
872 if ((result = priv->pool))
873 g_object_ref (result);
874 g_mutex_unlock (&priv->lock);
880 * gst_rtsp_media_collect_streams:
881 * @media: a #GstRTSPMedia
883 * Find all payloader elements, they should be named pay%d in the
884 * element of @media, and create #GstRTSPStreams for them.
886 * Collect all dynamic elements, named dynpay%d, and add them to
887 * the list of dynamic elements.
890 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
892 GstRTSPMediaPrivate *priv;
893 GstElement *element, *elem;
898 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
901 element = priv->element;
904 for (i = 0; have_elem; i++) {
909 name = g_strdup_printf ("pay%d", i);
910 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
911 GST_INFO ("found stream %d with payloader %p", i, elem);
913 /* take the pad of the payloader */
914 pad = gst_element_get_static_pad (elem, "src");
915 /* create the stream */
916 gst_rtsp_media_create_stream (media, elem, pad);
917 gst_object_unref (pad);
918 gst_object_unref (elem);
924 name = g_strdup_printf ("dynpay%d", i);
925 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
926 /* a stream that will dynamically create pads to provide RTP packets */
928 GST_INFO ("found dynamic element %d, %p", i, elem);
930 g_mutex_lock (&priv->lock);
931 priv->dynamic = g_list_prepend (priv->dynamic, elem);
932 g_mutex_unlock (&priv->lock);
941 * gst_rtsp_media_create_stream:
942 * @media: a #GstRTSPMedia
943 * @payloader: a #GstElement
944 * @srcpad: a source #GstPad
946 * Create a new stream in @media that provides RTP data on @srcpad.
947 * @srcpad should be a pad of an element inside @media->element.
949 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
953 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
956 GstRTSPMediaPrivate *priv;
957 GstRTSPStream *stream;
962 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
963 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
964 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
965 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
969 g_mutex_lock (&priv->lock);
970 idx = priv->streams->len;
972 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
974 name = g_strdup_printf ("src_%u", idx);
975 srcpad = gst_ghost_pad_new (name, pad);
976 gst_pad_set_active (srcpad, TRUE);
977 gst_element_add_pad (priv->element, srcpad);
980 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
982 gst_rtsp_stream_set_address_pool (stream, priv->pool);
984 g_ptr_array_add (priv->streams, stream);
985 g_mutex_unlock (&priv->lock);
987 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
994 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
996 GstRTSPMediaPrivate *priv;
1001 g_mutex_lock (&priv->lock);
1002 /* remove the ghostpad */
1003 srcpad = gst_rtsp_stream_get_srcpad (stream);
1004 gst_element_remove_pad (priv->element, srcpad);
1005 gst_object_unref (srcpad);
1006 /* now remove the stream */
1007 g_object_ref (stream);
1008 g_ptr_array_remove (priv->streams, stream);
1009 g_mutex_unlock (&priv->lock);
1011 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1014 g_object_unref (stream);
1018 * gst_rtsp_media_n_streams:
1019 * @media: a #GstRTSPMedia
1021 * Get the number of streams in this media.
1023 * Returns: The number of streams.
1026 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1028 GstRTSPMediaPrivate *priv;
1031 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1035 g_mutex_lock (&priv->lock);
1036 res = priv->streams->len;
1037 g_mutex_unlock (&priv->lock);
1043 * gst_rtsp_media_get_stream:
1044 * @media: a #GstRTSPMedia
1045 * @idx: the stream index
1047 * Retrieve the stream with index @idx from @media.
1049 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1050 * that index did not exist.
1053 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1055 GstRTSPMediaPrivate *priv;
1058 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1062 g_mutex_lock (&priv->lock);
1063 if (idx < priv->streams->len)
1064 res = g_ptr_array_index (priv->streams, idx);
1067 g_mutex_unlock (&priv->lock);
1073 * gst_rtsp_media_get_range_string:
1074 * @media: a #GstRTSPMedia
1075 * @play: for the PLAY request
1076 * @unit: the unit to use for the string
1078 * Get the current range as a string. @media must be prepared with
1079 * gst_rtsp_media_prepare ().
1081 * Returns: The range as a string, g_free() after usage.
1084 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1085 GstRTSPRangeUnit unit)
1087 GstRTSPMediaClass *klass;
1088 GstRTSPMediaPrivate *priv;
1090 GstRTSPTimeRange range;
1092 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1093 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1094 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1098 g_rec_mutex_lock (&priv->state_lock);
1099 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1102 g_mutex_lock (&priv->lock);
1104 range = priv->range;
1106 if (!play && priv->n_active > 0) {
1107 range.min.type = GST_RTSP_TIME_NOW;
1108 range.min.seconds = -1;
1110 g_mutex_unlock (&priv->lock);
1111 g_rec_mutex_unlock (&priv->state_lock);
1113 if (!klass->convert_range (media, &range, unit)) {
1114 goto conversion_failed;
1117 result = gst_rtsp_range_to_string (&range);
1124 GST_WARNING ("media %p was not prepared", media);
1125 g_rec_mutex_unlock (&priv->state_lock);
1130 GST_WARNING ("range conversion to unit %d failed", unit);
1131 g_rec_mutex_unlock (&priv->state_lock);
1137 * gst_rtsp_media_seek:
1138 * @media: a #GstRTSPMedia
1139 * @range: a #GstRTSPTimeRange
1141 * Seek the pipeline of @media to @range. @media must be prepared with
1142 * gst_rtsp_media_prepare().
1144 * Returns: %TRUE on success.
1147 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1149 GstRTSPMediaClass *klass;
1150 GstRTSPMediaPrivate *priv;
1153 GstClockTime start, stop;
1154 GstSeekType start_type, stop_type;
1156 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1158 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1159 g_return_val_if_fail (range != NULL, FALSE);
1160 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1164 g_rec_mutex_lock (&priv->state_lock);
1165 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1168 if (!priv->seekable)
1171 /* depends on the current playing state of the pipeline. We might need to
1172 * queue this until we get EOS. */
1173 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
1175 start_type = stop_type = GST_SEEK_TYPE_NONE;
1177 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1179 gst_rtsp_range_get_times (range, &start, &stop);
1181 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1182 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1183 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1184 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1186 if (priv->range_start == start)
1187 start = GST_CLOCK_TIME_NONE;
1188 else if (start != GST_CLOCK_TIME_NONE)
1189 start_type = GST_SEEK_TYPE_SET;
1191 if (priv->range_stop == stop)
1192 stop = GST_CLOCK_TIME_NONE;
1193 else if (stop != GST_CLOCK_TIME_NONE)
1194 stop_type = GST_SEEK_TYPE_SET;
1196 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1197 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1198 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1200 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1201 flags, start_type, start, stop_type, stop);
1203 /* and block for the seek to complete */
1204 GST_INFO ("done seeking %d", res);
1205 gst_element_get_state (priv->pipeline, NULL, NULL, -1);
1206 GST_INFO ("prerolled again");
1208 collect_media_stats (media);
1210 GST_INFO ("no seek needed");
1213 g_rec_mutex_unlock (&priv->state_lock);
1220 g_rec_mutex_unlock (&priv->state_lock);
1221 GST_INFO ("media %p is not prepared", media);
1226 g_rec_mutex_unlock (&priv->state_lock);
1227 GST_INFO ("pipeline is not seekable");
1232 g_rec_mutex_unlock (&priv->state_lock);
1233 GST_WARNING ("conversion to npt not supported");
1239 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1241 GstRTSPMediaPrivate *priv = media->priv;
1243 g_mutex_lock (&priv->lock);
1244 priv->status = status;
1245 GST_DEBUG ("setting new status to %d", status);
1246 g_cond_broadcast (&priv->cond);
1247 g_mutex_unlock (&priv->lock);
1251 * gst_rtsp_media_get_status:
1252 * @media: a #GstRTSPMedia
1254 * Get the status of @media. When @media is busy preparing, this function waits
1255 * until @media is prepared or in error.
1257 * Returns: the status of @media.
1260 gst_rtsp_media_get_status (GstRTSPMedia * media)
1262 GstRTSPMediaPrivate *priv = media->priv;
1263 GstRTSPMediaStatus result;
1266 g_mutex_lock (&priv->lock);
1267 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1268 /* while we are preparing, wait */
1269 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1270 GST_DEBUG ("waiting for status change");
1271 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1272 GST_DEBUG ("timeout, assuming error status");
1273 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1276 /* could be success or error */
1277 result = priv->status;
1278 GST_DEBUG ("got status %d", result);
1279 g_mutex_unlock (&priv->lock);
1284 /* called with state-lock */
1286 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1288 GstRTSPMediaPrivate *priv = media->priv;
1289 GstMessageType type;
1291 type = GST_MESSAGE_TYPE (message);
1294 case GST_MESSAGE_STATE_CHANGED:
1296 case GST_MESSAGE_BUFFERING:
1300 gst_message_parse_buffering (message, &percent);
1302 /* no state management needed for live pipelines */
1306 if (percent == 100) {
1307 /* a 100% message means buffering is done */
1308 priv->buffering = FALSE;
1309 /* if the desired state is playing, go back */
1310 if (priv->target_state == GST_STATE_PLAYING) {
1311 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1312 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1314 GST_INFO ("Buffering done");
1317 /* buffering busy */
1318 if (priv->buffering == FALSE) {
1319 if (priv->target_state == GST_STATE_PLAYING) {
1320 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1321 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1322 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1324 GST_INFO ("Buffering ...");
1327 priv->buffering = TRUE;
1331 case GST_MESSAGE_LATENCY:
1333 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1336 case GST_MESSAGE_ERROR:
1341 gst_message_parse_error (message, &gerror, &debug);
1342 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1343 g_error_free (gerror);
1346 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1349 case GST_MESSAGE_WARNING:
1354 gst_message_parse_warning (message, &gerror, &debug);
1355 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1356 g_error_free (gerror);
1360 case GST_MESSAGE_ELEMENT:
1362 case GST_MESSAGE_STREAM_STATUS:
1364 case GST_MESSAGE_ASYNC_DONE:
1366 /* when we are dynamically adding pads, the addition of the udpsrc will
1367 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1368 * wait for the final ASYNC_DONE after everything prerolled */
1369 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1371 GST_INFO ("%p: got ASYNC_DONE", media);
1372 collect_media_stats (media);
1374 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1375 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1378 case GST_MESSAGE_EOS:
1379 GST_INFO ("%p: got EOS", media);
1381 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1382 GST_DEBUG ("shutting down after EOS");
1383 finish_unprepare (media);
1387 GST_INFO ("%p: got message type %d (%s)", media, type,
1388 gst_message_type_get_name (type));
1395 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1397 GstRTSPMediaPrivate *priv = media->priv;
1398 GstRTSPMediaClass *klass;
1401 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1403 g_rec_mutex_lock (&priv->state_lock);
1404 if (klass->handle_message)
1405 ret = klass->handle_message (media, message);
1408 g_rec_mutex_unlock (&priv->state_lock);
1414 watch_destroyed (GstRTSPMedia * media)
1416 GST_DEBUG_OBJECT (media, "source destroyed");
1417 g_object_unref (media);
1420 /* called from streaming threads */
1422 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1424 GstRTSPMediaPrivate *priv = media->priv;
1425 GstRTSPStream *stream;
1427 /* FIXME, element is likely not a payloader, find the payloader here */
1428 stream = gst_rtsp_media_create_stream (media, element, pad);
1430 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1432 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1434 g_rec_mutex_lock (&priv->state_lock);
1435 /* we will be adding elements below that will cause ASYNC_DONE to be
1436 * posted in the bus. We want to ignore those messages until the
1437 * pipeline really prerolled. */
1438 priv->adding = TRUE;
1440 /* join the element in the PAUSED state because this callback is
1441 * called from the streaming thread and it is PAUSED */
1442 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1443 priv->rtpbin, GST_STATE_PAUSED);
1445 priv->adding = FALSE;
1446 g_rec_mutex_unlock (&priv->state_lock);
1450 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1452 GstRTSPMediaPrivate *priv = media->priv;
1453 GstRTSPStream *stream;
1455 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1459 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1461 g_rec_mutex_lock (&priv->state_lock);
1462 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1463 g_rec_mutex_unlock (&priv->state_lock);
1465 gst_rtsp_media_remove_stream (media, stream);
1469 remove_fakesink (GstRTSPMediaPrivate * priv)
1471 GstElement *fakesink;
1473 g_mutex_lock (&priv->lock);
1474 if ((fakesink = priv->fakesink))
1475 gst_object_ref (fakesink);
1476 priv->fakesink = NULL;
1477 g_mutex_unlock (&priv->lock);
1480 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1481 gst_element_set_state (fakesink, GST_STATE_NULL);
1482 gst_object_unref (fakesink);
1483 GST_INFO ("removed fakesink");
1488 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1490 GstRTSPMediaPrivate *priv = media->priv;
1492 GST_INFO ("no more pads");
1493 remove_fakesink (priv);
1496 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1498 struct _DynPaySignalHandlers
1500 gulong pad_added_handler;
1501 gulong pad_removed_handler;
1502 gulong no_more_pads_handler;
1506 * gst_rtsp_media_prepare:
1507 * @media: a #GstRTSPMedia
1509 * Prepare @media for streaming. This function will create the objects
1510 * to manage the streaming. A pipeline must have been set on @media with
1511 * gst_rtsp_media_take_pipeline().
1513 * It will preroll the pipeline and collect vital information about the streams
1514 * such as the duration.
1516 * Returns: %TRUE on success.
1519 gst_rtsp_media_prepare (GstRTSPMedia * media)
1521 GstRTSPMediaPrivate *priv;
1522 GstStateChangeReturn ret;
1523 GstRTSPMediaStatus status;
1525 GstRTSPMediaClass *klass;
1529 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1533 g_rec_mutex_lock (&priv->state_lock);
1534 priv->prepare_count++;
1536 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1539 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1542 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1543 goto not_unprepared;
1545 if (!priv->reusable && priv->reused)
1548 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1549 if (priv->rtpbin == NULL)
1552 GST_INFO ("preparing media %p", media);
1554 /* reset some variables */
1555 priv->is_live = FALSE;
1556 priv->seekable = FALSE;
1557 priv->buffering = FALSE;
1558 /* we're preparing now */
1559 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1561 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1563 /* add the pipeline bus to our custom mainloop */
1564 priv->source = gst_bus_create_watch (bus);
1565 gst_object_unref (bus);
1567 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1568 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1570 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1571 priv->id = g_source_attach (priv->source, klass->context);
1573 /* add stuff to the bin */
1574 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1576 /* link streams we already have, other streams might appear when we have
1577 * dynamic elements */
1578 for (i = 0; i < priv->streams->len; i++) {
1579 GstRTSPStream *stream;
1581 stream = g_ptr_array_index (priv->streams, i);
1583 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1584 priv->rtpbin, GST_STATE_NULL);
1587 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1588 GstElement *elem = walk->data;
1589 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
1591 GST_INFO ("adding callbacks for dynamic element %p", elem);
1593 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
1594 (GCallback) pad_added_cb, media);
1595 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
1596 (GCallback) pad_removed_cb, media);
1597 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
1598 (GCallback) no_more_pads_cb, media);
1600 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
1602 /* we add a fakesink here in order to make the state change async. We remove
1603 * the fakesink again in the no-more-pads callback. */
1604 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1605 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1608 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1609 /* first go to PAUSED */
1610 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1611 priv->target_state = GST_STATE_PAUSED;
1614 case GST_STATE_CHANGE_SUCCESS:
1615 GST_INFO ("SUCCESS state change for media %p", media);
1616 priv->seekable = TRUE;
1618 case GST_STATE_CHANGE_ASYNC:
1619 GST_INFO ("ASYNC state change for media %p", media);
1620 priv->seekable = TRUE;
1622 case GST_STATE_CHANGE_NO_PREROLL:
1623 /* we need to go to PLAYING */
1624 GST_INFO ("NO_PREROLL state change: live media %p", media);
1625 /* FIXME we disable seeking for live streams for now. We should perform a
1626 * seeking query in preroll instead */
1627 priv->seekable = FALSE;
1628 priv->is_live = TRUE;
1629 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1630 if (ret == GST_STATE_CHANGE_FAILURE)
1633 case GST_STATE_CHANGE_FAILURE:
1637 g_rec_mutex_unlock (&priv->state_lock);
1639 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1640 * able to do this async so that we don't block the server thread. */
1641 status = gst_rtsp_media_get_status (media);
1642 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1645 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1647 GST_INFO ("object %p is prerolled", media);
1654 GST_LOG ("media %p was prepared", media);
1655 g_rec_mutex_unlock (&priv->state_lock);
1661 GST_WARNING ("media %p was not unprepared", media);
1662 priv->prepare_count--;
1663 g_rec_mutex_unlock (&priv->state_lock);
1668 priv->prepare_count--;
1669 g_rec_mutex_unlock (&priv->state_lock);
1670 GST_WARNING ("can not reuse media %p", media);
1675 priv->prepare_count--;
1676 g_rec_mutex_unlock (&priv->state_lock);
1677 GST_WARNING ("no rtpbin element");
1678 g_warning ("failed to create element 'rtpbin', check your installation");
1683 GST_WARNING ("failed to preroll pipeline");
1684 gst_rtsp_media_unprepare (media);
1685 g_rec_mutex_unlock (&priv->state_lock);
1690 /* must be called with state-lock */
1692 finish_unprepare (GstRTSPMedia * media)
1694 GstRTSPMediaPrivate *priv = media->priv;
1698 GST_DEBUG ("shutting down");
1700 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1701 remove_fakesink (priv);
1703 for (i = 0; i < priv->streams->len; i++) {
1704 GstRTSPStream *stream;
1706 GST_INFO ("Removing elements of stream %d from pipeline", i);
1708 stream = g_ptr_array_index (priv->streams, i);
1710 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1713 /* remove the pad signal handlers */
1714 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1715 GstElement *elem = walk->data;
1716 DynPaySignalHandlers *handlers;
1719 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
1720 g_assert (handlers != NULL);
1722 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
1723 g_signal_handler_disconnect (G_OBJECT (elem),
1724 handlers->pad_removed_handler);
1725 g_signal_handler_disconnect (G_OBJECT (elem),
1726 handlers->no_more_pads_handler);
1728 g_slice_free (DynPaySignalHandlers, handlers);
1731 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1732 priv->rtpbin = NULL;
1735 gst_object_unref (priv->nettime);
1736 priv->nettime = NULL;
1738 priv->reused = TRUE;
1739 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1741 /* when the media is not reusable, this will effectively unref the media and
1743 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1745 /* the source has the last ref to the media */
1747 GST_DEBUG ("destroy source");
1748 g_source_destroy (priv->source);
1749 g_source_unref (priv->source);
1753 /* called with state-lock */
1755 default_unprepare (GstRTSPMedia * media)
1757 GstRTSPMediaPrivate *priv = media->priv;
1759 if (priv->eos_shutdown) {
1760 GST_DEBUG ("sending EOS for shutdown");
1761 /* ref so that we don't disappear */
1762 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
1763 /* we need to go to playing again for the EOS to propagate, normally in this
1764 * state, nothing is receiving data from us anymore so this is ok. */
1765 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1766 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
1768 finish_unprepare (media);
1774 * gst_rtsp_media_unprepare:
1775 * @media: a #GstRTSPMedia
1777 * Unprepare @media. After this call, the media should be prepared again before
1778 * it can be used again. If the media is set to be non-reusable, a new instance
1781 * Returns: %TRUE on success.
1784 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1786 GstRTSPMediaPrivate *priv;
1789 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1793 g_rec_mutex_lock (&priv->state_lock);
1794 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1795 goto was_unprepared;
1797 priv->prepare_count--;
1798 if (priv->prepare_count > 0)
1801 GST_INFO ("unprepare media %p", media);
1802 priv->target_state = GST_STATE_NULL;
1805 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
1806 GstRTSPMediaClass *klass;
1808 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1809 if (klass->unprepare)
1810 success = klass->unprepare (media);
1812 finish_unprepare (media);
1814 g_rec_mutex_unlock (&priv->state_lock);
1820 g_rec_mutex_unlock (&priv->state_lock);
1821 GST_INFO ("media %p was already unprepared", media);
1826 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
1827 g_rec_mutex_unlock (&priv->state_lock);
1832 /* should be called with state-lock */
1834 get_clock_unlocked (GstRTSPMedia * media)
1836 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
1837 GST_DEBUG_OBJECT (media, "media was not prepared");
1840 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
1844 * gst_rtsp_media_get_clock:
1845 * @media: a #GstRTSPMedia
1847 * Get the clock that is used by the pipeline in @media.
1849 * @media must be prepared before this method returns a valid clock object.
1851 * Returns: the #GstClock used by @media. unref after usage.
1854 gst_rtsp_media_get_clock (GstRTSPMedia * media)
1857 GstRTSPMediaPrivate *priv;
1859 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1863 g_rec_mutex_lock (&priv->state_lock);
1864 clock = get_clock_unlocked (media);
1865 g_rec_mutex_unlock (&priv->state_lock);
1871 * gst_rtsp_media_get_base_time:
1872 * @media: a #GstRTSPMedia
1874 * Get the base_time that is used by the pipeline in @media.
1876 * @media must be prepared before this method returns a valid base_time.
1878 * Returns: the base_time used by @media.
1881 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
1883 GstClockTime result;
1884 GstRTSPMediaPrivate *priv;
1886 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
1890 g_rec_mutex_lock (&priv->state_lock);
1891 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1894 result = gst_element_get_base_time (media->priv->pipeline);
1895 g_rec_mutex_unlock (&priv->state_lock);
1902 g_rec_mutex_unlock (&priv->state_lock);
1903 GST_DEBUG_OBJECT (media, "media was not prepared");
1904 return GST_CLOCK_TIME_NONE;
1909 * gst_rtsp_media_get_time_provider:
1910 * @media: a #GstRTSPMedia
1911 * @address: an address or NULL
1912 * @port: a port or 0
1914 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
1915 * will listen on @address and @port for client time requests.
1917 * Returns: the #GstNetTimeProvider of @media.
1919 GstNetTimeProvider *
1920 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
1923 GstRTSPMediaPrivate *priv;
1924 GstNetTimeProvider *provider = NULL;
1926 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1930 g_rec_mutex_lock (&priv->state_lock);
1931 if (priv->time_provider) {
1932 if ((provider = priv->nettime) == NULL) {
1935 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
1936 provider = gst_net_time_provider_new (clock, address, port);
1937 gst_object_unref (clock);
1939 priv->nettime = provider;
1943 g_rec_mutex_unlock (&priv->state_lock);
1946 gst_object_ref (provider);
1952 * gst_rtsp_media_set_state:
1953 * @media: a #GstRTSPMedia
1954 * @state: the target state of the media
1955 * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
1957 * Set the state of @media to @state and for the transports in @transports.
1959 * @media must be prepared with gst_rtsp_media_prepare();
1961 * Returns: %TRUE on success.
1964 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1965 GPtrArray * transports)
1967 GstRTSPMediaPrivate *priv;
1969 gboolean activate, deactivate, do_state;
1972 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1973 g_return_val_if_fail (transports != NULL, FALSE);
1977 g_rec_mutex_lock (&priv->state_lock);
1978 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1981 /* NULL and READY are the same */
1982 if (state == GST_STATE_READY)
1983 state = GST_STATE_NULL;
1985 activate = deactivate = FALSE;
1987 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1991 case GST_STATE_NULL:
1992 case GST_STATE_PAUSED:
1993 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
1994 if (priv->target_state == GST_STATE_PLAYING)
1997 case GST_STATE_PLAYING:
1998 /* we're going to PLAYING, activate */
2004 old_active = priv->n_active;
2006 for (i = 0; i < transports->len; i++) {
2007 GstRTSPStreamTransport *trans;
2009 /* we need a non-NULL entry in the array */
2010 trans = g_ptr_array_index (transports, i);
2015 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2017 } else if (deactivate) {
2018 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2023 /* we just activated the first media, do the playing state change */
2024 if (old_active == 0 && activate)
2026 /* if we have no more active media, do the downward state changes */
2027 else if (priv->n_active == 0)
2032 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2035 if (priv->target_state != state) {
2037 if (state == GST_STATE_NULL) {
2038 gst_rtsp_media_unprepare (media);
2040 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2042 priv->target_state = state;
2043 /* when we are buffering, don't update the state yet, this will be done
2044 * when buffering finishes */
2045 if (priv->buffering) {
2046 GST_INFO ("Buffering busy, delay state change");
2048 gst_element_set_state (priv->pipeline, state);
2052 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2056 /* remember where we are */
2057 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2058 old_active != priv->n_active))
2059 collect_media_stats (media);
2061 g_rec_mutex_unlock (&priv->state_lock);
2068 GST_WARNING ("media %p was not prepared", media);
2069 g_rec_mutex_unlock (&priv->state_lock);
2074 /* called with state-lock */
2076 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2077 GstRTSPRangeUnit unit)
2079 return gst_rtsp_range_convert_units (range, unit);