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);
126 static gboolean default_get_range_times (GstRTSPMedia * media,
127 const GstRTSPTimeRange * range, GstClockTime * min, GstClockTime * max);
129 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
131 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
134 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
136 GObjectClass *gobject_class;
138 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
140 gobject_class = G_OBJECT_CLASS (klass);
142 gobject_class->get_property = gst_rtsp_media_get_property;
143 gobject_class->set_property = gst_rtsp_media_set_property;
144 gobject_class->finalize = gst_rtsp_media_finalize;
146 g_object_class_install_property (gobject_class, PROP_SHARED,
147 g_param_spec_boolean ("shared", "Shared",
148 "If this media pipeline can be shared", DEFAULT_SHARED,
149 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
151 g_object_class_install_property (gobject_class, PROP_REUSABLE,
152 g_param_spec_boolean ("reusable", "Reusable",
153 "If this media pipeline can be reused after an unprepare",
154 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
156 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
157 g_param_spec_flags ("protocols", "Protocols",
158 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
159 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
161 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
162 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
163 "Send an EOS event to the pipeline before unpreparing",
164 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
166 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
167 g_param_spec_uint ("buffer-size", "Buffer Size",
168 "The kernel UDP buffer size to use", 0, G_MAXUINT,
169 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
171 g_object_class_install_property (gobject_class, PROP_ELEMENT,
172 g_param_spec_object ("element", "The Element",
173 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
174 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
176 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
177 g_param_spec_boolean ("time-provider", "Time Provider",
178 "Use a NetTimeProvider for clients",
179 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
181 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
182 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
183 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
184 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
186 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
187 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
188 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
189 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
190 GST_TYPE_RTSP_STREAM);
192 gst_rtsp_media_signals[SIGNAL_PREPARED] =
193 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
194 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
195 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
197 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
198 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
199 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
200 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
202 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
203 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
204 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
205 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 0, G_TYPE_INT);
207 klass->context = g_main_context_new ();
208 klass->loop = g_main_loop_new (klass->context, TRUE);
210 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
212 klass->thread = g_thread_new ("Bus Thread", (GThreadFunc) do_loop, klass);
214 klass->handle_message = default_handle_message;
215 klass->unprepare = default_unprepare;
216 klass->get_range_times = default_get_range_times;
220 gst_rtsp_media_init (GstRTSPMedia * media)
222 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
226 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
227 g_mutex_init (&priv->lock);
228 g_cond_init (&priv->cond);
229 g_rec_mutex_init (&priv->state_lock);
231 priv->shared = DEFAULT_SHARED;
232 priv->reusable = DEFAULT_REUSABLE;
233 priv->protocols = DEFAULT_PROTOCOLS;
234 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
235 priv->buffer_size = DEFAULT_BUFFER_SIZE;
236 priv->time_provider = DEFAULT_TIME_PROVIDER;
240 gst_rtsp_media_finalize (GObject * obj)
242 GstRTSPMediaPrivate *priv;
245 media = GST_RTSP_MEDIA (obj);
248 GST_INFO ("finalize media %p", media);
250 g_ptr_array_unref (priv->streams);
252 g_list_free_full (priv->dynamic, gst_object_unref);
255 gst_object_unref (priv->pipeline);
257 gst_object_unref (priv->nettime);
258 gst_object_unref (priv->element);
260 g_object_unref (priv->auth);
262 g_object_unref (priv->pool);
263 g_mutex_clear (&priv->lock);
264 g_cond_clear (&priv->cond);
265 g_rec_mutex_clear (&priv->state_lock);
267 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
271 gst_rtsp_media_get_property (GObject * object, guint propid,
272 GValue * value, GParamSpec * pspec)
274 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
278 g_value_set_object (value, media->priv->element);
281 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
284 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
287 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
289 case PROP_EOS_SHUTDOWN:
290 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
292 case PROP_BUFFER_SIZE:
293 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
295 case PROP_TIME_PROVIDER:
296 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
299 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
304 gst_rtsp_media_set_property (GObject * object, guint propid,
305 const GValue * value, GParamSpec * pspec)
307 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
311 media->priv->element = g_value_get_object (value);
314 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
317 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
320 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
322 case PROP_EOS_SHUTDOWN:
323 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
325 case PROP_BUFFER_SIZE:
326 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
328 case PROP_TIME_PROVIDER:
329 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
332 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
337 do_loop (GstRTSPMediaClass * klass)
339 GST_INFO ("enter mainloop");
340 g_main_loop_run (klass->loop);
341 GST_INFO ("exit mainloop");
346 /* must be called with state lock */
348 collect_media_stats (GstRTSPMedia * media)
350 GstRTSPMediaPrivate *priv = media->priv;
351 gint64 position, duration;
353 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
354 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
357 priv->range.unit = GST_RTSP_RANGE_NPT;
359 GST_INFO ("collect media stats");
362 priv->range.min.type = GST_RTSP_TIME_NOW;
363 priv->range.min.seconds = -1;
364 priv->range_start = -1;
365 priv->range.max.type = GST_RTSP_TIME_END;
366 priv->range.max.seconds = -1;
367 priv->range_stop = -1;
369 /* get the position */
370 if (!gst_element_query_position (priv->pipeline, GST_FORMAT_TIME,
372 GST_INFO ("position query failed");
376 /* get the duration */
377 if (!gst_element_query_duration (priv->pipeline, GST_FORMAT_TIME,
379 GST_INFO ("duration query failed");
383 GST_INFO ("stats: position %" GST_TIME_FORMAT ", duration %"
384 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (duration));
386 if (position == -1) {
387 priv->range.min.type = GST_RTSP_TIME_NOW;
388 priv->range.min.seconds = -1;
389 priv->range_start = -1;
391 priv->range.min.type = GST_RTSP_TIME_SECONDS;
392 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
393 priv->range_start = position;
395 if (duration == -1) {
396 priv->range.max.type = GST_RTSP_TIME_END;
397 priv->range.max.seconds = -1;
398 priv->range_stop = -1;
400 priv->range.max.type = GST_RTSP_TIME_SECONDS;
401 priv->range.max.seconds = ((gdouble) duration) / GST_SECOND;
402 priv->range_stop = duration;
408 * gst_rtsp_media_new:
409 * @element: (transfer full): a #GstElement
411 * Create a new #GstRTSPMedia instance. @element is the bin element that
412 * provides the different streams. The #GstRTSPMedia object contains the
413 * element to produce RTP data for one or more related (audio/video/..)
416 * Ownership is taken of @element.
418 * Returns: a new #GstRTSPMedia object.
421 gst_rtsp_media_new (GstElement * element)
423 GstRTSPMedia *result;
425 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
427 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
433 * gst_rtsp_media_take_element:
434 * @media: a #GstRTSPMedia
435 * @pipeline: (transfer full): a #GstPipeline
437 * Set @pipeline as the #GstPipeline for @media. Ownership is
438 * taken of @pipeline.
441 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
443 GstRTSPMediaPrivate *priv;
445 GstNetTimeProvider *nettime;
447 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
448 g_return_if_fail (GST_IS_PIPELINE (pipeline));
452 g_mutex_lock (&priv->lock);
453 old = priv->pipeline;
454 priv->pipeline = GST_ELEMENT_CAST (pipeline);
455 nettime = priv->nettime;
456 priv->nettime = NULL;
457 g_mutex_unlock (&priv->lock);
460 gst_object_unref (old);
463 gst_object_unref (nettime);
465 gst_object_ref (priv->element);
466 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
470 * gst_rtsp_media_set_shared:
471 * @media: a #GstRTSPMedia
472 * @shared: the new value
474 * Set or unset if the pipeline for @media can be shared will multiple clients.
475 * When @shared is %TRUE, client requests for this media will share the media
479 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
481 GstRTSPMediaPrivate *priv;
483 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
487 g_mutex_lock (&priv->lock);
488 priv->shared = shared;
489 g_mutex_unlock (&priv->lock);
493 * gst_rtsp_media_is_shared:
494 * @media: a #GstRTSPMedia
496 * Check if the pipeline for @media can be shared between multiple clients.
498 * Returns: %TRUE if the media can be shared between clients.
501 gst_rtsp_media_is_shared (GstRTSPMedia * media)
503 GstRTSPMediaPrivate *priv;
506 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
510 g_mutex_lock (&priv->lock);
512 g_mutex_unlock (&priv->lock);
518 * gst_rtsp_media_set_reusable:
519 * @media: a #GstRTSPMedia
520 * @reusable: the new value
522 * Set or unset if the pipeline for @media can be reused after the pipeline has
526 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
528 GstRTSPMediaPrivate *priv;
530 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
534 g_mutex_lock (&priv->lock);
535 priv->reusable = reusable;
536 g_mutex_unlock (&priv->lock);
540 * gst_rtsp_media_is_reusable:
541 * @media: a #GstRTSPMedia
543 * Check if the pipeline for @media can be reused after an unprepare.
545 * Returns: %TRUE if the media can be reused
548 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
550 GstRTSPMediaPrivate *priv;
553 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
557 g_mutex_lock (&priv->lock);
558 res = priv->reusable;
559 g_mutex_unlock (&priv->lock);
565 * gst_rtsp_media_set_protocols:
566 * @media: a #GstRTSPMedia
567 * @protocols: the new flags
569 * Configure the allowed lower transport for @media.
572 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
574 GstRTSPMediaPrivate *priv;
576 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
580 g_mutex_lock (&priv->lock);
581 priv->protocols = protocols;
582 g_mutex_unlock (&priv->lock);
586 * gst_rtsp_media_get_protocols:
587 * @media: a #GstRTSPMedia
589 * Get the allowed protocols of @media.
591 * Returns: a #GstRTSPLowerTrans
594 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
596 GstRTSPMediaPrivate *priv;
597 GstRTSPLowerTrans res;
599 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
600 GST_RTSP_LOWER_TRANS_UNKNOWN);
604 g_mutex_lock (&priv->lock);
605 res = priv->protocols;
606 g_mutex_unlock (&priv->lock);
612 * gst_rtsp_media_set_eos_shutdown:
613 * @media: a #GstRTSPMedia
614 * @eos_shutdown: the new value
616 * Set or unset if an EOS event will be sent to the pipeline for @media before
620 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
622 GstRTSPMediaPrivate *priv;
624 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
628 g_mutex_lock (&priv->lock);
629 priv->eos_shutdown = eos_shutdown;
630 g_mutex_unlock (&priv->lock);
634 * gst_rtsp_media_is_eos_shutdown:
635 * @media: a #GstRTSPMedia
637 * Check if the pipeline for @media will send an EOS down the pipeline before
640 * Returns: %TRUE if the media will send EOS before unpreparing.
643 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
645 GstRTSPMediaPrivate *priv;
648 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
652 g_mutex_lock (&priv->lock);
653 res = priv->eos_shutdown;
654 g_mutex_unlock (&priv->lock);
660 * gst_rtsp_media_set_buffer_size:
661 * @media: a #GstRTSPMedia
662 * @size: the new value
664 * Set the kernel UDP buffer size.
667 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
669 GstRTSPMediaPrivate *priv;
671 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
673 GST_LOG_OBJECT (media, "set buffer size %u", size);
677 g_mutex_lock (&priv->lock);
678 priv->buffer_size = size;
679 g_mutex_unlock (&priv->lock);
683 * gst_rtsp_media_get_buffer_size:
684 * @media: a #GstRTSPMedia
686 * Get the kernel UDP buffer size.
688 * Returns: the kernel UDP buffer size.
691 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
693 GstRTSPMediaPrivate *priv;
696 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
700 g_mutex_unlock (&priv->lock);
701 res = priv->buffer_size;
702 g_mutex_unlock (&priv->lock);
708 * gst_rtsp_media_use_time_provider:
709 * @media: a #GstRTSPMedia
711 * Set @media to provide a GstNetTimeProvider.
714 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
716 GstRTSPMediaPrivate *priv;
718 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
722 g_mutex_lock (&priv->lock);
723 priv->time_provider = time_provider;
724 g_mutex_unlock (&priv->lock);
728 * gst_rtsp_media_is_time_provider:
729 * @media: a #GstRTSPMedia
731 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
733 * Use gst_rtsp_media_get_time_provider() to get the network clock.
735 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
738 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
740 GstRTSPMediaPrivate *priv;
743 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
747 g_mutex_unlock (&priv->lock);
748 res = priv->time_provider;
749 g_mutex_unlock (&priv->lock);
755 * gst_rtsp_media_set_auth:
756 * @media: a #GstRTSPMedia
757 * @auth: a #GstRTSPAuth
759 * configure @auth to be used as the authentication manager of @media.
762 gst_rtsp_media_set_auth (GstRTSPMedia * media, GstRTSPAuth * auth)
764 GstRTSPMediaPrivate *priv;
767 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
771 GST_LOG_OBJECT (media, "set auth %p", auth);
773 g_mutex_lock (&priv->lock);
774 if ((old = priv->auth) != auth)
775 priv->auth = auth ? g_object_ref (auth) : NULL;
778 g_mutex_unlock (&priv->lock);
781 g_object_unref (old);
785 * gst_rtsp_media_get_auth:
786 * @media: a #GstRTSPMedia
788 * Get the #GstRTSPAuth used as the authentication manager of @media.
790 * Returns: (transfer full): the #GstRTSPAuth of @media. g_object_unref() after
794 gst_rtsp_media_get_auth (GstRTSPMedia * media)
796 GstRTSPMediaPrivate *priv;
799 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
803 g_mutex_lock (&priv->lock);
804 if ((result = priv->auth))
805 g_object_ref (result);
806 g_mutex_unlock (&priv->lock);
812 * gst_rtsp_media_set_address_pool:
813 * @media: a #GstRTSPMedia
814 * @pool: a #GstRTSPAddressPool
816 * configure @pool to be used as the address pool of @media.
819 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
820 GstRTSPAddressPool * pool)
822 GstRTSPMediaPrivate *priv;
823 GstRTSPAddressPool *old;
825 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
829 GST_LOG_OBJECT (media, "set address pool %p", pool);
831 g_mutex_lock (&priv->lock);
832 if ((old = priv->pool) != pool)
833 priv->pool = pool ? g_object_ref (pool) : NULL;
836 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
838 g_mutex_unlock (&priv->lock);
841 g_object_unref (old);
845 * gst_rtsp_media_get_address_pool:
846 * @media: a #GstRTSPMedia
848 * Get the #GstRTSPAddressPool used as the address pool of @media.
850 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
854 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
856 GstRTSPMediaPrivate *priv;
857 GstRTSPAddressPool *result;
859 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
863 g_mutex_lock (&priv->lock);
864 if ((result = priv->pool))
865 g_object_ref (result);
866 g_mutex_unlock (&priv->lock);
872 * gst_rtsp_media_collect_streams:
873 * @media: a #GstRTSPMedia
875 * Find all payloader elements, they should be named pay%d in the
876 * element of @media, and create #GstRTSPStreams for them.
878 * Collect all dynamic elements, named dynpay%d, and add them to
879 * the list of dynamic elements.
882 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
884 GstRTSPMediaPrivate *priv;
885 GstElement *element, *elem;
890 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
893 element = priv->element;
896 for (i = 0; have_elem; i++) {
901 name = g_strdup_printf ("pay%d", i);
902 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
903 GST_INFO ("found stream %d with payloader %p", i, elem);
905 /* take the pad of the payloader */
906 pad = gst_element_get_static_pad (elem, "src");
907 /* create the stream */
908 gst_rtsp_media_create_stream (media, elem, pad);
909 gst_object_unref (pad);
910 gst_object_unref (elem);
916 name = g_strdup_printf ("dynpay%d", i);
917 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
918 /* a stream that will dynamically create pads to provide RTP packets */
920 GST_INFO ("found dynamic element %d, %p", i, elem);
922 g_mutex_lock (&priv->lock);
923 priv->dynamic = g_list_prepend (priv->dynamic, elem);
924 g_mutex_unlock (&priv->lock);
933 * gst_rtsp_media_create_stream:
934 * @media: a #GstRTSPMedia
935 * @payloader: a #GstElement
936 * @srcpad: a source #GstPad
938 * Create a new stream in @media that provides RTP data on @srcpad.
939 * @srcpad should be a pad of an element inside @media->element.
941 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
945 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
948 GstRTSPMediaPrivate *priv;
949 GstRTSPStream *stream;
954 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
955 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
956 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
957 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
961 g_mutex_lock (&priv->lock);
962 idx = priv->streams->len;
964 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
966 name = g_strdup_printf ("src_%u", idx);
967 srcpad = gst_ghost_pad_new (name, pad);
968 gst_pad_set_active (srcpad, TRUE);
969 gst_element_add_pad (priv->element, srcpad);
972 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
974 gst_rtsp_stream_set_address_pool (stream, priv->pool);
976 g_ptr_array_add (priv->streams, stream);
977 g_mutex_unlock (&priv->lock);
979 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
986 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
988 GstRTSPMediaPrivate *priv;
993 g_mutex_lock (&priv->lock);
994 /* remove the ghostpad */
995 srcpad = gst_rtsp_stream_get_srcpad (stream);
996 gst_element_remove_pad (priv->element, srcpad);
997 gst_object_unref (srcpad);
998 /* now remove the stream */
999 g_object_ref (stream);
1000 g_ptr_array_remove (priv->streams, stream);
1001 g_mutex_unlock (&priv->lock);
1003 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1006 g_object_unref (stream);
1010 * gst_rtsp_media_n_streams:
1011 * @media: a #GstRTSPMedia
1013 * Get the number of streams in this media.
1015 * Returns: The number of streams.
1018 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1020 GstRTSPMediaPrivate *priv;
1023 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1027 g_mutex_lock (&priv->lock);
1028 res = priv->streams->len;
1029 g_mutex_unlock (&priv->lock);
1035 * gst_rtsp_media_get_stream:
1036 * @media: a #GstRTSPMedia
1037 * @idx: the stream index
1039 * Retrieve the stream with index @idx from @media.
1041 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1042 * that index did not exist.
1045 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1047 GstRTSPMediaPrivate *priv;
1050 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1054 g_mutex_lock (&priv->lock);
1055 if (idx < priv->streams->len)
1056 res = g_ptr_array_index (priv->streams, idx);
1059 g_mutex_unlock (&priv->lock);
1065 * gst_rtsp_media_get_range_string:
1066 * @media: a #GstRTSPMedia
1067 * @play: for the PLAY request
1068 * @unit: the unit to use for the string
1070 * Get the current range as a string. @media must be prepared with
1071 * gst_rtsp_media_prepare ().
1073 * Returns: The range as a string, g_free() after usage.
1076 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1077 GstRTSPRangeUnit unit)
1079 GstRTSPMediaPrivate *priv;
1081 GstRTSPTimeRange range;
1083 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1087 g_rec_mutex_lock (&priv->state_lock);
1088 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1091 g_mutex_lock (&priv->lock);
1093 range = priv->range;
1095 if (!play && priv->n_active > 0) {
1096 range.min.type = GST_RTSP_TIME_NOW;
1097 range.min.seconds = -1;
1099 g_mutex_unlock (&priv->lock);
1100 g_rec_mutex_unlock (&priv->state_lock);
1102 gst_rtsp_range_convert_units (&range, unit);
1104 result = gst_rtsp_range_to_string (&range);
1111 GST_WARNING ("media %p was not prepared", media);
1112 g_rec_mutex_unlock (&priv->state_lock);
1118 * gst_rtsp_media_seek:
1119 * @media: a #GstRTSPMedia
1120 * @range: a #GstRTSPTimeRange
1122 * Seek the pipeline of @media to @range. @media must be prepared with
1123 * gst_rtsp_media_prepare().
1125 * Returns: %TRUE on success.
1128 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1130 GstRTSPMediaClass *klass;
1131 GstRTSPMediaPrivate *priv;
1134 GstClockTime start, stop;
1135 GstSeekType start_type, stop_type;
1137 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1139 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1140 g_return_val_if_fail (range != NULL, FALSE);
1141 g_return_val_if_fail (klass->get_range_times != NULL, FALSE);
1145 g_rec_mutex_lock (&priv->state_lock);
1146 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1149 if (!priv->seekable)
1152 /* depends on the current playing state of the pipeline. We might need to
1153 * queue this until we get EOS. */
1154 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT;
1156 start_type = stop_type = GST_SEEK_TYPE_NONE;
1158 if (!klass->get_range_times (media, range, &start, &stop))
1161 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1162 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1163 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1164 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1166 if (priv->range_start == start)
1167 start = GST_CLOCK_TIME_NONE;
1168 else if (start != GST_CLOCK_TIME_NONE)
1169 start_type = GST_SEEK_TYPE_SET;
1171 if (priv->range_stop == stop)
1172 stop = GST_CLOCK_TIME_NONE;
1173 else if (stop != GST_CLOCK_TIME_NONE)
1174 stop_type = GST_SEEK_TYPE_SET;
1176 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1177 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1178 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1180 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1181 flags, start_type, start, stop_type, stop);
1183 /* and block for the seek to complete */
1184 GST_INFO ("done seeking %d", res);
1185 gst_element_get_state (priv->pipeline, NULL, NULL, -1);
1186 GST_INFO ("prerolled again");
1188 collect_media_stats (media);
1190 GST_INFO ("no seek needed");
1193 g_rec_mutex_unlock (&priv->state_lock);
1200 g_rec_mutex_unlock (&priv->state_lock);
1201 GST_INFO ("media %p is not prepared", media);
1206 g_rec_mutex_unlock (&priv->state_lock);
1207 GST_INFO ("pipeline is not seekable");
1212 g_rec_mutex_unlock (&priv->state_lock);
1213 GST_WARNING ("seek unit %d not supported", range->unit);
1219 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1221 GstRTSPMediaPrivate *priv = media->priv;
1223 g_mutex_lock (&priv->lock);
1224 priv->status = status;
1225 GST_DEBUG ("setting new status to %d", status);
1226 g_cond_broadcast (&priv->cond);
1227 g_mutex_unlock (&priv->lock);
1231 * gst_rtsp_media_get_status:
1232 * @media: a #GstRTSPMedia
1234 * Get the status of @media. When @media is busy preparing, this function waits
1235 * until @media is prepared or in error.
1237 * Returns: the status of @media.
1240 gst_rtsp_media_get_status (GstRTSPMedia * media)
1242 GstRTSPMediaPrivate *priv = media->priv;
1243 GstRTSPMediaStatus result;
1246 g_mutex_lock (&priv->lock);
1247 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1248 /* while we are preparing, wait */
1249 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1250 GST_DEBUG ("waiting for status change");
1251 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1252 GST_DEBUG ("timeout, assuming error status");
1253 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1256 /* could be success or error */
1257 result = priv->status;
1258 GST_DEBUG ("got status %d", result);
1259 g_mutex_unlock (&priv->lock);
1264 /* called with state-lock */
1266 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1268 GstRTSPMediaPrivate *priv = media->priv;
1269 GstMessageType type;
1271 type = GST_MESSAGE_TYPE (message);
1274 case GST_MESSAGE_STATE_CHANGED:
1276 case GST_MESSAGE_BUFFERING:
1280 gst_message_parse_buffering (message, &percent);
1282 /* no state management needed for live pipelines */
1286 if (percent == 100) {
1287 /* a 100% message means buffering is done */
1288 priv->buffering = FALSE;
1289 /* if the desired state is playing, go back */
1290 if (priv->target_state == GST_STATE_PLAYING) {
1291 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1292 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1294 GST_INFO ("Buffering done");
1297 /* buffering busy */
1298 if (priv->buffering == FALSE) {
1299 if (priv->target_state == GST_STATE_PLAYING) {
1300 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1301 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1302 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1304 GST_INFO ("Buffering ...");
1307 priv->buffering = TRUE;
1311 case GST_MESSAGE_LATENCY:
1313 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1316 case GST_MESSAGE_ERROR:
1321 gst_message_parse_error (message, &gerror, &debug);
1322 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1323 g_error_free (gerror);
1326 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1329 case GST_MESSAGE_WARNING:
1334 gst_message_parse_warning (message, &gerror, &debug);
1335 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1336 g_error_free (gerror);
1340 case GST_MESSAGE_ELEMENT:
1342 case GST_MESSAGE_STREAM_STATUS:
1344 case GST_MESSAGE_ASYNC_DONE:
1346 /* when we are dynamically adding pads, the addition of the udpsrc will
1347 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1348 * wait for the final ASYNC_DONE after everything prerolled */
1349 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1351 GST_INFO ("%p: got ASYNC_DONE", media);
1352 collect_media_stats (media);
1354 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1355 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1358 case GST_MESSAGE_EOS:
1359 GST_INFO ("%p: got EOS", media);
1361 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1362 GST_DEBUG ("shutting down after EOS");
1363 finish_unprepare (media);
1367 GST_INFO ("%p: got message type %d (%s)", media, type,
1368 gst_message_type_get_name (type));
1375 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1377 GstRTSPMediaPrivate *priv = media->priv;
1378 GstRTSPMediaClass *klass;
1381 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1383 g_rec_mutex_lock (&priv->state_lock);
1384 if (klass->handle_message)
1385 ret = klass->handle_message (media, message);
1388 g_rec_mutex_unlock (&priv->state_lock);
1394 watch_destroyed (GstRTSPMedia * media)
1396 GST_DEBUG_OBJECT (media, "source destroyed");
1397 g_object_unref (media);
1400 /* called from streaming threads */
1402 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1404 GstRTSPMediaPrivate *priv = media->priv;
1405 GstRTSPStream *stream;
1407 /* FIXME, element is likely not a payloader, find the payloader here */
1408 stream = gst_rtsp_media_create_stream (media, element, pad);
1410 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1412 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1414 g_rec_mutex_lock (&priv->state_lock);
1415 /* we will be adding elements below that will cause ASYNC_DONE to be
1416 * posted in the bus. We want to ignore those messages until the
1417 * pipeline really prerolled. */
1418 priv->adding = TRUE;
1420 /* join the element in the PAUSED state because this callback is
1421 * called from the streaming thread and it is PAUSED */
1422 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1423 priv->rtpbin, GST_STATE_PAUSED);
1425 priv->adding = FALSE;
1426 g_rec_mutex_unlock (&priv->state_lock);
1430 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1432 GstRTSPMediaPrivate *priv = media->priv;
1433 GstRTSPStream *stream;
1435 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1439 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1441 g_rec_mutex_lock (&priv->state_lock);
1442 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1443 g_rec_mutex_unlock (&priv->state_lock);
1445 gst_rtsp_media_remove_stream (media, stream);
1449 remove_fakesink (GstRTSPMediaPrivate * priv)
1451 GstElement *fakesink;
1453 g_mutex_lock (&priv->lock);
1454 if ((fakesink = priv->fakesink))
1455 gst_object_ref (fakesink);
1456 priv->fakesink = NULL;
1457 g_mutex_unlock (&priv->lock);
1460 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1461 gst_element_set_state (fakesink, GST_STATE_NULL);
1462 gst_object_unref (fakesink);
1463 GST_INFO ("removed fakesink");
1468 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1470 GstRTSPMediaPrivate *priv = media->priv;
1472 GST_INFO ("no more pads");
1473 remove_fakesink (priv);
1476 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1478 struct _DynPaySignalHandlers
1480 gulong pad_added_handler;
1481 gulong pad_removed_handler;
1482 gulong no_more_pads_handler;
1486 * gst_rtsp_media_prepare:
1487 * @media: a #GstRTSPMedia
1489 * Prepare @media for streaming. This function will create the objects
1490 * to manage the streaming. A pipeline must have been set on @media with
1491 * gst_rtsp_media_take_pipeline().
1493 * It will preroll the pipeline and collect vital information about the streams
1494 * such as the duration.
1496 * Returns: %TRUE on success.
1499 gst_rtsp_media_prepare (GstRTSPMedia * media)
1501 GstRTSPMediaPrivate *priv;
1502 GstStateChangeReturn ret;
1503 GstRTSPMediaStatus status;
1505 GstRTSPMediaClass *klass;
1509 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1513 g_rec_mutex_lock (&priv->state_lock);
1514 priv->prepare_count++;
1516 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1519 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1522 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1523 goto not_unprepared;
1525 if (!priv->reusable && priv->reused)
1528 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1529 if (priv->rtpbin == NULL)
1532 GST_INFO ("preparing media %p", media);
1534 /* reset some variables */
1535 priv->is_live = FALSE;
1536 priv->seekable = FALSE;
1537 priv->buffering = FALSE;
1538 /* we're preparing now */
1539 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1541 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1543 /* add the pipeline bus to our custom mainloop */
1544 priv->source = gst_bus_create_watch (bus);
1545 gst_object_unref (bus);
1547 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1548 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1550 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1551 priv->id = g_source_attach (priv->source, klass->context);
1553 /* add stuff to the bin */
1554 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1556 /* link streams we already have, other streams might appear when we have
1557 * dynamic elements */
1558 for (i = 0; i < priv->streams->len; i++) {
1559 GstRTSPStream *stream;
1561 stream = g_ptr_array_index (priv->streams, i);
1563 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1564 priv->rtpbin, GST_STATE_NULL);
1567 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1568 GstElement *elem = walk->data;
1569 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
1571 GST_INFO ("adding callbacks for dynamic element %p", elem);
1573 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
1574 (GCallback) pad_added_cb, media);
1575 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
1576 (GCallback) pad_removed_cb, media);
1577 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
1578 (GCallback) no_more_pads_cb, media);
1580 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
1582 /* we add a fakesink here in order to make the state change async. We remove
1583 * the fakesink again in the no-more-pads callback. */
1584 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1585 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1588 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1589 /* first go to PAUSED */
1590 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1591 priv->target_state = GST_STATE_PAUSED;
1594 case GST_STATE_CHANGE_SUCCESS:
1595 GST_INFO ("SUCCESS state change for media %p", media);
1596 priv->seekable = TRUE;
1598 case GST_STATE_CHANGE_ASYNC:
1599 GST_INFO ("ASYNC state change for media %p", media);
1600 priv->seekable = TRUE;
1602 case GST_STATE_CHANGE_NO_PREROLL:
1603 /* we need to go to PLAYING */
1604 GST_INFO ("NO_PREROLL state change: live media %p", media);
1605 /* FIXME we disable seeking for live streams for now. We should perform a
1606 * seeking query in preroll instead */
1607 priv->seekable = FALSE;
1608 priv->is_live = TRUE;
1609 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1610 if (ret == GST_STATE_CHANGE_FAILURE)
1613 case GST_STATE_CHANGE_FAILURE:
1617 g_rec_mutex_unlock (&priv->state_lock);
1619 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1620 * able to do this async so that we don't block the server thread. */
1621 status = gst_rtsp_media_get_status (media);
1622 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1625 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1627 GST_INFO ("object %p is prerolled", media);
1634 GST_LOG ("media %p was prepared", media);
1635 g_rec_mutex_unlock (&priv->state_lock);
1641 GST_WARNING ("media %p was not unprepared", media);
1642 priv->prepare_count--;
1643 g_rec_mutex_unlock (&priv->state_lock);
1648 priv->prepare_count--;
1649 g_rec_mutex_unlock (&priv->state_lock);
1650 GST_WARNING ("can not reuse media %p", media);
1655 priv->prepare_count--;
1656 g_rec_mutex_unlock (&priv->state_lock);
1657 GST_WARNING ("no rtpbin element");
1658 g_warning ("failed to create element 'rtpbin', check your installation");
1663 GST_WARNING ("failed to preroll pipeline");
1664 gst_rtsp_media_unprepare (media);
1665 g_rec_mutex_unlock (&priv->state_lock);
1670 /* must be called with state-lock */
1672 finish_unprepare (GstRTSPMedia * media)
1674 GstRTSPMediaPrivate *priv = media->priv;
1678 GST_DEBUG ("shutting down");
1680 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1681 remove_fakesink (priv);
1683 for (i = 0; i < priv->streams->len; i++) {
1684 GstRTSPStream *stream;
1686 GST_INFO ("Removing elements of stream %d from pipeline", i);
1688 stream = g_ptr_array_index (priv->streams, i);
1690 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1693 /* remove the pad signal handlers */
1694 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1695 GstElement *elem = walk->data;
1696 DynPaySignalHandlers *handlers;
1699 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
1700 g_assert (handlers != NULL);
1702 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
1703 g_signal_handler_disconnect (G_OBJECT (elem),
1704 handlers->pad_removed_handler);
1705 g_signal_handler_disconnect (G_OBJECT (elem),
1706 handlers->no_more_pads_handler);
1708 g_slice_free (DynPaySignalHandlers, handlers);
1711 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1712 priv->rtpbin = NULL;
1715 gst_object_unref (priv->nettime);
1716 priv->nettime = NULL;
1718 priv->reused = TRUE;
1719 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
1721 /* when the media is not reusable, this will effectively unref the media and
1723 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
1725 /* the source has the last ref to the media */
1727 GST_DEBUG ("destroy source");
1728 g_source_destroy (priv->source);
1729 g_source_unref (priv->source);
1733 /* called with state-lock */
1735 default_unprepare (GstRTSPMedia * media)
1737 GstRTSPMediaPrivate *priv = media->priv;
1739 if (priv->eos_shutdown) {
1740 GST_DEBUG ("sending EOS for shutdown");
1741 /* ref so that we don't disappear */
1742 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
1743 /* we need to go to playing again for the EOS to propagate, normally in this
1744 * state, nothing is receiving data from us anymore so this is ok. */
1745 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1746 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
1748 finish_unprepare (media);
1754 * gst_rtsp_media_unprepare:
1755 * @media: a #GstRTSPMedia
1757 * Unprepare @media. After this call, the media should be prepared again before
1758 * it can be used again. If the media is set to be non-reusable, a new instance
1761 * Returns: %TRUE on success.
1764 gst_rtsp_media_unprepare (GstRTSPMedia * media)
1766 GstRTSPMediaPrivate *priv;
1769 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1773 g_rec_mutex_lock (&priv->state_lock);
1774 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
1775 goto was_unprepared;
1777 priv->prepare_count--;
1778 if (priv->prepare_count > 0)
1781 GST_INFO ("unprepare media %p", media);
1782 priv->target_state = GST_STATE_NULL;
1785 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
1786 GstRTSPMediaClass *klass;
1788 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1789 if (klass->unprepare)
1790 success = klass->unprepare (media);
1792 finish_unprepare (media);
1794 g_rec_mutex_unlock (&priv->state_lock);
1800 g_rec_mutex_unlock (&priv->state_lock);
1801 GST_INFO ("media %p was already unprepared", media);
1806 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
1807 g_rec_mutex_unlock (&priv->state_lock);
1812 /* should be called with state-lock */
1814 get_clock_unlocked (GstRTSPMedia * media)
1816 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
1817 GST_DEBUG_OBJECT (media, "media was not prepared");
1820 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
1824 * gst_rtsp_media_get_clock:
1825 * @media: a #GstRTSPMedia
1827 * Get the clock that is used by the pipeline in @media.
1829 * @media must be prepared before this method returns a valid clock object.
1831 * Returns: the #GstClock used by @media. unref after usage.
1834 gst_rtsp_media_get_clock (GstRTSPMedia * media)
1837 GstRTSPMediaPrivate *priv;
1839 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1843 g_rec_mutex_lock (&priv->state_lock);
1844 clock = get_clock_unlocked (media);
1845 g_rec_mutex_unlock (&priv->state_lock);
1851 * gst_rtsp_media_get_base_time:
1852 * @media: a #GstRTSPMedia
1854 * Get the base_time that is used by the pipeline in @media.
1856 * @media must be prepared before this method returns a valid base_time.
1858 * Returns: the base_time used by @media.
1861 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
1863 GstClockTime result;
1864 GstRTSPMediaPrivate *priv;
1866 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
1870 g_rec_mutex_lock (&priv->state_lock);
1871 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1874 result = gst_element_get_base_time (media->priv->pipeline);
1875 g_rec_mutex_unlock (&priv->state_lock);
1882 g_rec_mutex_unlock (&priv->state_lock);
1883 GST_DEBUG_OBJECT (media, "media was not prepared");
1884 return GST_CLOCK_TIME_NONE;
1889 * gst_rtsp_media_get_time_provider:
1890 * @media: a #GstRTSPMedia
1891 * @address: an address or NULL
1892 * @port: a port or 0
1894 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
1895 * will listen on @address and @port for client time requests.
1897 * Returns: the #GstNetTimeProvider of @media.
1899 GstNetTimeProvider *
1900 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
1903 GstRTSPMediaPrivate *priv;
1904 GstNetTimeProvider *provider = NULL;
1906 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1910 g_rec_mutex_lock (&priv->state_lock);
1911 if (priv->time_provider) {
1912 if ((provider = priv->nettime) == NULL) {
1915 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
1916 provider = gst_net_time_provider_new (clock, address, port);
1917 gst_object_unref (clock);
1919 priv->nettime = provider;
1923 g_rec_mutex_unlock (&priv->state_lock);
1926 gst_object_ref (provider);
1932 * gst_rtsp_media_set_state:
1933 * @media: a #GstRTSPMedia
1934 * @state: the target state of the media
1935 * @transports: a #GPtrArray of #GstRTSPStreamTransport pointers
1937 * Set the state of @media to @state and for the transports in @transports.
1939 * @media must be prepared with gst_rtsp_media_prepare();
1941 * Returns: %TRUE on success.
1944 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
1945 GPtrArray * transports)
1947 GstRTSPMediaPrivate *priv;
1949 gboolean activate, deactivate, do_state;
1952 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1953 g_return_val_if_fail (transports != NULL, FALSE);
1957 g_rec_mutex_lock (&priv->state_lock);
1958 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1961 /* NULL and READY are the same */
1962 if (state == GST_STATE_READY)
1963 state = GST_STATE_NULL;
1965 activate = deactivate = FALSE;
1967 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
1971 case GST_STATE_NULL:
1972 case GST_STATE_PAUSED:
1973 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
1974 if (priv->target_state == GST_STATE_PLAYING)
1977 case GST_STATE_PLAYING:
1978 /* we're going to PLAYING, activate */
1984 old_active = priv->n_active;
1986 for (i = 0; i < transports->len; i++) {
1987 GstRTSPStreamTransport *trans;
1989 /* we need a non-NULL entry in the array */
1990 trans = g_ptr_array_index (transports, i);
1995 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
1997 } else if (deactivate) {
1998 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2003 /* we just activated the first media, do the playing state change */
2004 if (old_active == 0 && activate)
2006 /* if we have no more active media, do the downward state changes */
2007 else if (priv->n_active == 0)
2012 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2015 if (priv->target_state != state) {
2017 if (state == GST_STATE_NULL) {
2018 gst_rtsp_media_unprepare (media);
2020 GST_INFO ("state %s media %p", gst_element_state_get_name (state),
2022 priv->target_state = state;
2023 /* when we are buffering, don't update the state yet, this will be done
2024 * when buffering finishes */
2025 if (priv->buffering) {
2026 GST_INFO ("Buffering busy, delay state change");
2028 gst_element_set_state (priv->pipeline, state);
2032 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2036 /* remember where we are */
2037 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2038 old_active != priv->n_active))
2039 collect_media_stats (media);
2041 g_rec_mutex_unlock (&priv->state_lock);
2048 GST_WARNING ("media %p was not prepared", media);
2049 g_rec_mutex_unlock (&priv->state_lock);
2054 /* called with state-lock */
2056 default_get_range_times (GstRTSPMedia * media,
2057 const GstRTSPTimeRange * range, GstClockTime * min, GstClockTime * max)
2059 return gst_rtsp_range_get_times (range, min, max);