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.
21 * @short_description: The media pipeline
22 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
23 * #GstRTSPSessionMedia
25 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
26 * streaming to the clients. The actual data transfer is done by the
27 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
29 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
30 * client does a DESCRIBE or SETUP of a resource.
32 * A media is created with gst_rtsp_media_new() that takes the element that will
33 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
34 * object needs to be made with the gst_rtsp_media_create_stream() which takes
35 * the payloader element and the source pad that produces the RTP stream.
37 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
38 * prepare method will add rtpbin and sinks and sources to send and receive RTP
39 * and RTCP packets from the clients. Each stream srcpad is connected to an
40 * input into the internal rtpbin.
42 * It is also possible to dynamically create #GstRTSPStream objects during the
43 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
46 * After the media is prepared, it is ready for streaming. It will usually be
47 * managed in a session with gst_rtsp_session_manage_media(). See
48 * #GstRTSPSession and #GstRTSPSessionMedia.
50 * The state of the media can be controlled with gst_rtsp_media_set_state ().
51 * Seeking can be done with gst_rtsp_media_seek().
53 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
54 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
57 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
58 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
59 * can be prepared again after an unprepare.
61 * Last reviewed on 2013-07-11 (1.0.0)
67 #include <gst/app/gstappsrc.h>
68 #include <gst/app/gstappsink.h>
70 #include "rtsp-media.h"
72 #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
73 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
75 struct _GstRTSPMediaPrivate
80 /* protected by lock */
81 GstRTSPPermissions *permissions;
84 GstRTSPLowerTrans protocols;
86 gboolean eos_shutdown;
88 GstRTSPAddressPool *pool;
91 GRecMutex state_lock; /* locking order: state lock, lock */
92 GPtrArray *streams; /* protected by lock */
93 GList *dynamic; /* protected by lock */
94 GstRTSPMediaStatus status; /* protected by lock */
99 /* the pipeline for the media */
100 GstElement *pipeline;
101 GstElement *fakesink; /* protected by lock */
104 GstRTSPThread *thread;
106 gboolean time_provider;
107 GstNetTimeProvider *nettime;
112 GstState target_state;
114 /* RTP session manager */
117 /* the range of media */
118 GstRTSPTimeRange range; /* protected by lock */
119 GstClockTime range_start;
120 GstClockTime range_stop;
123 #define DEFAULT_SHARED FALSE
124 #define DEFAULT_REUSABLE FALSE
125 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
126 GST_RTSP_LOWER_TRANS_TCP
127 #define DEFAULT_EOS_SHUTDOWN FALSE
128 #define DEFAULT_BUFFER_SIZE 0x80000
129 #define DEFAULT_TIME_PROVIDER FALSE
131 /* define to dump received RTCP packets */
150 SIGNAL_REMOVED_STREAM,
157 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
158 #define GST_CAT_DEFAULT rtsp_media_debug
160 static void gst_rtsp_media_get_property (GObject * object, guint propid,
161 GValue * value, GParamSpec * pspec);
162 static void gst_rtsp_media_set_property (GObject * object, guint propid,
163 const GValue * value, GParamSpec * pspec);
164 static void gst_rtsp_media_finalize (GObject * obj);
166 static gboolean default_handle_message (GstRTSPMedia * media,
167 GstMessage * message);
168 static void finish_unprepare (GstRTSPMedia * media);
169 static gboolean default_unprepare (GstRTSPMedia * media);
170 static gboolean default_convert_range (GstRTSPMedia * media,
171 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
172 static gboolean default_query_position (GstRTSPMedia * media,
174 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
176 static gboolean wait_preroll (GstRTSPMedia * media);
178 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
180 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
183 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
185 GObjectClass *gobject_class;
187 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
189 gobject_class = G_OBJECT_CLASS (klass);
191 gobject_class->get_property = gst_rtsp_media_get_property;
192 gobject_class->set_property = gst_rtsp_media_set_property;
193 gobject_class->finalize = gst_rtsp_media_finalize;
195 g_object_class_install_property (gobject_class, PROP_SHARED,
196 g_param_spec_boolean ("shared", "Shared",
197 "If this media pipeline can be shared", DEFAULT_SHARED,
198 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
200 g_object_class_install_property (gobject_class, PROP_REUSABLE,
201 g_param_spec_boolean ("reusable", "Reusable",
202 "If this media pipeline can be reused after an unprepare",
203 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
205 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
206 g_param_spec_flags ("protocols", "Protocols",
207 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
208 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
210 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
211 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
212 "Send an EOS event to the pipeline before unpreparing",
213 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
215 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
216 g_param_spec_uint ("buffer-size", "Buffer Size",
217 "The kernel UDP buffer size to use", 0, G_MAXUINT,
218 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
220 g_object_class_install_property (gobject_class, PROP_ELEMENT,
221 g_param_spec_object ("element", "The Element",
222 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
223 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
225 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
226 g_param_spec_boolean ("time-provider", "Time Provider",
227 "Use a NetTimeProvider for clients",
228 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
230 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
231 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
232 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
233 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
235 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
236 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
237 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
238 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
239 GST_TYPE_RTSP_STREAM);
241 gst_rtsp_media_signals[SIGNAL_PREPARED] =
242 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
243 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
244 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
246 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
247 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
248 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
249 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
251 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
252 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
253 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
254 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
256 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
258 klass->handle_message = default_handle_message;
259 klass->unprepare = default_unprepare;
260 klass->convert_range = default_convert_range;
261 klass->query_position = default_query_position;
262 klass->query_stop = default_query_stop;
266 gst_rtsp_media_init (GstRTSPMedia * media)
268 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
272 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
273 g_mutex_init (&priv->lock);
274 g_cond_init (&priv->cond);
275 g_rec_mutex_init (&priv->state_lock);
277 priv->shared = DEFAULT_SHARED;
278 priv->reusable = DEFAULT_REUSABLE;
279 priv->protocols = DEFAULT_PROTOCOLS;
280 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
281 priv->buffer_size = DEFAULT_BUFFER_SIZE;
282 priv->time_provider = DEFAULT_TIME_PROVIDER;
286 gst_rtsp_media_finalize (GObject * obj)
288 GstRTSPMediaPrivate *priv;
291 media = GST_RTSP_MEDIA (obj);
294 GST_INFO ("finalize media %p", media);
296 if (priv->permissions)
297 gst_rtsp_permissions_unref (priv->permissions);
299 g_ptr_array_unref (priv->streams);
301 g_list_free_full (priv->dynamic, gst_object_unref);
304 gst_object_unref (priv->pipeline);
306 gst_object_unref (priv->nettime);
307 gst_object_unref (priv->element);
309 g_object_unref (priv->pool);
310 g_mutex_clear (&priv->lock);
311 g_cond_clear (&priv->cond);
312 g_rec_mutex_clear (&priv->state_lock);
314 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
318 gst_rtsp_media_get_property (GObject * object, guint propid,
319 GValue * value, GParamSpec * pspec)
321 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
325 g_value_set_object (value, media->priv->element);
328 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
331 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
334 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
336 case PROP_EOS_SHUTDOWN:
337 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
339 case PROP_BUFFER_SIZE:
340 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
342 case PROP_TIME_PROVIDER:
343 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
346 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
351 gst_rtsp_media_set_property (GObject * object, guint propid,
352 const GValue * value, GParamSpec * pspec)
354 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
358 media->priv->element = g_value_get_object (value);
359 gst_object_ref_sink (media->priv->element);
362 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
365 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
368 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
370 case PROP_EOS_SHUTDOWN:
371 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
373 case PROP_BUFFER_SIZE:
374 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
376 case PROP_TIME_PROVIDER:
377 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
380 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
385 default_query_position (GstRTSPMedia * media, gint64 * position)
387 return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
392 default_query_stop (GstRTSPMedia * media, gint64 * stop)
397 query = gst_query_new_segment (GST_FORMAT_TIME);
398 if ((res = gst_element_query (media->priv->pipeline, query))) {
400 gst_query_parse_segment (query, NULL, &format, NULL, stop);
401 if (format != GST_FORMAT_TIME)
404 gst_query_unref (query);
408 /* must be called with state lock */
410 collect_media_stats (GstRTSPMedia * media)
412 GstRTSPMediaPrivate *priv = media->priv;
413 gint64 position, stop;
415 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
416 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
419 priv->range.unit = GST_RTSP_RANGE_NPT;
421 GST_INFO ("collect media stats");
424 priv->range.min.type = GST_RTSP_TIME_NOW;
425 priv->range.min.seconds = -1;
426 priv->range_start = -1;
427 priv->range.max.type = GST_RTSP_TIME_END;
428 priv->range.max.seconds = -1;
429 priv->range_stop = -1;
431 GstRTSPMediaClass *klass;
434 klass = GST_RTSP_MEDIA_GET_CLASS (media);
436 /* get the position */
438 if (klass->query_position)
439 ret = klass->query_position (media, &position);
442 GST_INFO ("position query failed");
446 /* get the current segment stop */
448 if (klass->query_stop)
449 ret = klass->query_stop (media, &stop);
452 GST_INFO ("stop query failed");
456 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
457 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
459 if (position == -1) {
460 priv->range.min.type = GST_RTSP_TIME_NOW;
461 priv->range.min.seconds = -1;
462 priv->range_start = -1;
464 priv->range.min.type = GST_RTSP_TIME_SECONDS;
465 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
466 priv->range_start = position;
469 priv->range.max.type = GST_RTSP_TIME_END;
470 priv->range.max.seconds = -1;
471 priv->range_stop = -1;
473 priv->range.max.type = GST_RTSP_TIME_SECONDS;
474 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
475 priv->range_stop = stop;
481 * gst_rtsp_media_new:
482 * @element: (transfer full): a #GstElement
484 * Create a new #GstRTSPMedia instance. @element is the bin element that
485 * provides the different streams. The #GstRTSPMedia object contains the
486 * element to produce RTP data for one or more related (audio/video/..)
489 * Ownership is taken of @element.
491 * Returns: a new #GstRTSPMedia object.
494 gst_rtsp_media_new (GstElement * element)
496 GstRTSPMedia *result;
498 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
500 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
506 * gst_rtsp_media_get_element:
507 * @media: a #GstRTSPMedia
509 * Get the element that was used when constructing @media.
511 * Returns: (transfer full): a #GstElement. Unref after usage.
514 gst_rtsp_media_get_element (GstRTSPMedia * media)
516 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
518 return gst_object_ref (media->priv->element);
522 * gst_rtsp_media_take_pipeline:
523 * @media: a #GstRTSPMedia
524 * @pipeline: (transfer full): a #GstPipeline
526 * Set @pipeline as the #GstPipeline for @media. Ownership is
527 * taken of @pipeline.
530 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
532 GstRTSPMediaPrivate *priv;
534 GstNetTimeProvider *nettime;
536 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
537 g_return_if_fail (GST_IS_PIPELINE (pipeline));
541 g_mutex_lock (&priv->lock);
542 old = priv->pipeline;
543 priv->pipeline = GST_ELEMENT_CAST (pipeline);
544 nettime = priv->nettime;
545 priv->nettime = NULL;
546 g_mutex_unlock (&priv->lock);
549 gst_object_unref (old);
552 gst_object_unref (nettime);
554 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
558 * gst_rtsp_media_set_permissions:
559 * @media: a #GstRTSPMedia
560 * @permissions: a #GstRTSPPermissions
562 * Set @permissions on @media.
565 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
566 GstRTSPPermissions * permissions)
568 GstRTSPMediaPrivate *priv;
570 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
574 g_mutex_lock (&priv->lock);
575 if (priv->permissions)
576 gst_rtsp_permissions_unref (priv->permissions);
577 if ((priv->permissions = permissions))
578 gst_rtsp_permissions_ref (permissions);
579 g_mutex_unlock (&priv->lock);
583 * gst_rtsp_media_get_permissions:
584 * @media: a #GstRTSPMedia
586 * Get the permissions object from @media.
588 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
591 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
593 GstRTSPMediaPrivate *priv;
594 GstRTSPPermissions *result;
596 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
600 g_mutex_lock (&priv->lock);
601 if ((result = priv->permissions))
602 gst_rtsp_permissions_ref (result);
603 g_mutex_unlock (&priv->lock);
609 * gst_rtsp_media_set_shared:
610 * @media: a #GstRTSPMedia
611 * @shared: the new value
613 * Set or unset if the pipeline for @media can be shared will multiple clients.
614 * When @shared is %TRUE, client requests for this media will share the media
618 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
620 GstRTSPMediaPrivate *priv;
622 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
626 g_mutex_lock (&priv->lock);
627 priv->shared = shared;
628 g_mutex_unlock (&priv->lock);
632 * gst_rtsp_media_is_shared:
633 * @media: a #GstRTSPMedia
635 * Check if the pipeline for @media can be shared between multiple clients.
637 * Returns: %TRUE if the media can be shared between clients.
640 gst_rtsp_media_is_shared (GstRTSPMedia * media)
642 GstRTSPMediaPrivate *priv;
645 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
649 g_mutex_lock (&priv->lock);
651 g_mutex_unlock (&priv->lock);
657 * gst_rtsp_media_set_reusable:
658 * @media: a #GstRTSPMedia
659 * @reusable: the new value
661 * Set or unset if the pipeline for @media can be reused after the pipeline has
665 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
667 GstRTSPMediaPrivate *priv;
669 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
673 g_mutex_lock (&priv->lock);
674 priv->reusable = reusable;
675 g_mutex_unlock (&priv->lock);
679 * gst_rtsp_media_is_reusable:
680 * @media: a #GstRTSPMedia
682 * Check if the pipeline for @media can be reused after an unprepare.
684 * Returns: %TRUE if the media can be reused
687 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
689 GstRTSPMediaPrivate *priv;
692 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
696 g_mutex_lock (&priv->lock);
697 res = priv->reusable;
698 g_mutex_unlock (&priv->lock);
704 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
706 gst_rtsp_stream_set_protocols (stream, *protocols);
710 * gst_rtsp_media_set_protocols:
711 * @media: a #GstRTSPMedia
712 * @protocols: the new flags
714 * Configure the allowed lower transport for @media.
717 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
719 GstRTSPMediaPrivate *priv;
721 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
725 g_mutex_lock (&priv->lock);
726 priv->protocols = protocols;
727 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
728 g_mutex_unlock (&priv->lock);
732 * gst_rtsp_media_get_protocols:
733 * @media: a #GstRTSPMedia
735 * Get the allowed protocols of @media.
737 * Returns: a #GstRTSPLowerTrans
740 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
742 GstRTSPMediaPrivate *priv;
743 GstRTSPLowerTrans res;
745 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
746 GST_RTSP_LOWER_TRANS_UNKNOWN);
750 g_mutex_lock (&priv->lock);
751 res = priv->protocols;
752 g_mutex_unlock (&priv->lock);
758 * gst_rtsp_media_set_eos_shutdown:
759 * @media: a #GstRTSPMedia
760 * @eos_shutdown: the new value
762 * Set or unset if an EOS event will be sent to the pipeline for @media before
766 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
768 GstRTSPMediaPrivate *priv;
770 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
774 g_mutex_lock (&priv->lock);
775 priv->eos_shutdown = eos_shutdown;
776 g_mutex_unlock (&priv->lock);
780 * gst_rtsp_media_is_eos_shutdown:
781 * @media: a #GstRTSPMedia
783 * Check if the pipeline for @media will send an EOS down the pipeline before
786 * Returns: %TRUE if the media will send EOS before unpreparing.
789 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
791 GstRTSPMediaPrivate *priv;
794 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
798 g_mutex_lock (&priv->lock);
799 res = priv->eos_shutdown;
800 g_mutex_unlock (&priv->lock);
806 * gst_rtsp_media_set_buffer_size:
807 * @media: a #GstRTSPMedia
808 * @size: the new value
810 * Set the kernel UDP buffer size.
813 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
815 GstRTSPMediaPrivate *priv;
817 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
819 GST_LOG_OBJECT (media, "set buffer size %u", size);
823 g_mutex_lock (&priv->lock);
824 priv->buffer_size = size;
825 g_mutex_unlock (&priv->lock);
829 * gst_rtsp_media_get_buffer_size:
830 * @media: a #GstRTSPMedia
832 * Get the kernel UDP buffer size.
834 * Returns: the kernel UDP buffer size.
837 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
839 GstRTSPMediaPrivate *priv;
842 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
846 g_mutex_unlock (&priv->lock);
847 res = priv->buffer_size;
848 g_mutex_unlock (&priv->lock);
854 * gst_rtsp_media_use_time_provider:
855 * @media: a #GstRTSPMedia
856 * @time_provider: if a #GstNetTimeProvider should be used
858 * Set @media to provide a #GstNetTimeProvider.
861 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
863 GstRTSPMediaPrivate *priv;
865 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
869 g_mutex_lock (&priv->lock);
870 priv->time_provider = time_provider;
871 g_mutex_unlock (&priv->lock);
875 * gst_rtsp_media_is_time_provider:
876 * @media: a #GstRTSPMedia
878 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
880 * Use gst_rtsp_media_get_time_provider() to get the network clock.
882 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
885 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
887 GstRTSPMediaPrivate *priv;
890 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
894 g_mutex_unlock (&priv->lock);
895 res = priv->time_provider;
896 g_mutex_unlock (&priv->lock);
902 * gst_rtsp_media_set_address_pool:
903 * @media: a #GstRTSPMedia
904 * @pool: a #GstRTSPAddressPool
906 * configure @pool to be used as the address pool of @media.
909 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
910 GstRTSPAddressPool * pool)
912 GstRTSPMediaPrivate *priv;
913 GstRTSPAddressPool *old;
915 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
919 GST_LOG_OBJECT (media, "set address pool %p", pool);
921 g_mutex_lock (&priv->lock);
922 if ((old = priv->pool) != pool)
923 priv->pool = pool ? g_object_ref (pool) : NULL;
926 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
928 g_mutex_unlock (&priv->lock);
931 g_object_unref (old);
935 * gst_rtsp_media_get_address_pool:
936 * @media: a #GstRTSPMedia
938 * Get the #GstRTSPAddressPool used as the address pool of @media.
940 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
944 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
946 GstRTSPMediaPrivate *priv;
947 GstRTSPAddressPool *result;
949 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
953 g_mutex_lock (&priv->lock);
954 if ((result = priv->pool))
955 g_object_ref (result);
956 g_mutex_unlock (&priv->lock);
962 * gst_rtsp_media_collect_streams:
963 * @media: a #GstRTSPMedia
965 * Find all payloader elements, they should be named pay\%d in the
966 * element of @media, and create #GstRTSPStreams for them.
968 * Collect all dynamic elements, named dynpay\%d, and add them to
969 * the list of dynamic elements.
972 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
974 GstRTSPMediaPrivate *priv;
975 GstElement *element, *elem;
980 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
983 element = priv->element;
986 for (i = 0; have_elem; i++) {
991 name = g_strdup_printf ("pay%d", i);
992 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
993 GST_INFO ("found stream %d with payloader %p", i, elem);
995 /* take the pad of the payloader */
996 pad = gst_element_get_static_pad (elem, "src");
997 /* create the stream */
998 gst_rtsp_media_create_stream (media, elem, pad);
999 gst_object_unref (pad);
1000 gst_object_unref (elem);
1006 name = g_strdup_printf ("dynpay%d", i);
1007 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1008 /* a stream that will dynamically create pads to provide RTP packets */
1010 GST_INFO ("found dynamic element %d, %p", i, elem);
1012 g_mutex_lock (&priv->lock);
1013 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1014 g_mutex_unlock (&priv->lock);
1023 * gst_rtsp_media_create_stream:
1024 * @media: a #GstRTSPMedia
1025 * @payloader: a #GstElement
1026 * @srcpad: a source #GstPad
1028 * Create a new stream in @media that provides RTP data on @srcpad.
1029 * @srcpad should be a pad of an element inside @media->element.
1031 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1035 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1038 GstRTSPMediaPrivate *priv;
1039 GstRTSPStream *stream;
1044 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1045 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1046 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1047 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
1051 g_mutex_lock (&priv->lock);
1052 idx = priv->streams->len;
1054 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1056 name = g_strdup_printf ("src_%u", idx);
1057 srcpad = gst_ghost_pad_new (name, pad);
1058 gst_pad_set_active (srcpad, TRUE);
1059 gst_element_add_pad (priv->element, srcpad);
1062 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
1064 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1065 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1067 g_ptr_array_add (priv->streams, stream);
1068 g_mutex_unlock (&priv->lock);
1070 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1077 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1079 GstRTSPMediaPrivate *priv;
1084 g_mutex_lock (&priv->lock);
1085 /* remove the ghostpad */
1086 srcpad = gst_rtsp_stream_get_srcpad (stream);
1087 gst_element_remove_pad (priv->element, srcpad);
1088 gst_object_unref (srcpad);
1089 /* now remove the stream */
1090 g_object_ref (stream);
1091 g_ptr_array_remove (priv->streams, stream);
1092 g_mutex_unlock (&priv->lock);
1094 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1097 g_object_unref (stream);
1101 * gst_rtsp_media_n_streams:
1102 * @media: a #GstRTSPMedia
1104 * Get the number of streams in this media.
1106 * Returns: The number of streams.
1109 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1111 GstRTSPMediaPrivate *priv;
1114 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1118 g_mutex_lock (&priv->lock);
1119 res = priv->streams->len;
1120 g_mutex_unlock (&priv->lock);
1126 * gst_rtsp_media_get_stream:
1127 * @media: a #GstRTSPMedia
1128 * @idx: the stream index
1130 * Retrieve the stream with index @idx from @media.
1132 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1133 * that index did not exist.
1136 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1138 GstRTSPMediaPrivate *priv;
1141 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1145 g_mutex_lock (&priv->lock);
1146 if (idx < priv->streams->len)
1147 res = g_ptr_array_index (priv->streams, idx);
1150 g_mutex_unlock (&priv->lock);
1156 * gst_rtsp_media_find_stream:
1157 * @media: a #GstRTSPMedia
1158 * @control: the control of the stream
1160 * Find a stream in @media with @control as the control uri.
1162 * Returns: (transfer none): the #GstRTSPStream with control uri @control
1163 * or %NULL when a stream with that control did not exist.
1166 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1168 GstRTSPMediaPrivate *priv;
1172 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1173 g_return_val_if_fail (control != NULL, NULL);
1179 g_mutex_lock (&priv->lock);
1180 for (i = 0; i < priv->streams->len; i++) {
1181 GstRTSPStream *test;
1183 test = g_ptr_array_index (priv->streams, i);
1184 if (gst_rtsp_stream_has_control (test, control)) {
1189 g_mutex_unlock (&priv->lock);
1194 /* called with state-lock */
1196 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1197 GstRTSPRangeUnit unit)
1199 return gst_rtsp_range_convert_units (range, unit);
1203 * gst_rtsp_media_get_range_string:
1204 * @media: a #GstRTSPMedia
1205 * @play: for the PLAY request
1206 * @unit: the unit to use for the string
1208 * Get the current range as a string. @media must be prepared with
1209 * gst_rtsp_media_prepare ().
1211 * Returns: The range as a string, g_free() after usage.
1214 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1215 GstRTSPRangeUnit unit)
1217 GstRTSPMediaClass *klass;
1218 GstRTSPMediaPrivate *priv;
1220 GstRTSPTimeRange range;
1222 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1223 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1224 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1228 g_rec_mutex_lock (&priv->state_lock);
1229 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1232 g_mutex_lock (&priv->lock);
1234 /* Update the range value with current position/duration */
1235 collect_media_stats (media);
1238 range = priv->range;
1240 if (!play && priv->n_active > 0) {
1241 range.min.type = GST_RTSP_TIME_NOW;
1242 range.min.seconds = -1;
1244 g_mutex_unlock (&priv->lock);
1245 g_rec_mutex_unlock (&priv->state_lock);
1247 if (!klass->convert_range (media, &range, unit))
1248 goto conversion_failed;
1250 result = gst_rtsp_range_to_string (&range);
1257 GST_WARNING ("media %p was not prepared", media);
1258 g_rec_mutex_unlock (&priv->state_lock);
1263 GST_WARNING ("range conversion to unit %d failed", unit);
1269 * gst_rtsp_media_seek:
1270 * @media: a #GstRTSPMedia
1271 * @range: a #GstRTSPTimeRange
1273 * Seek the pipeline of @media to @range. @media must be prepared with
1274 * gst_rtsp_media_prepare().
1276 * Returns: %TRUE on success.
1279 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1281 GstRTSPMediaClass *klass;
1282 GstRTSPMediaPrivate *priv;
1285 GstClockTime start, stop;
1286 GstSeekType start_type, stop_type;
1289 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1291 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1292 g_return_val_if_fail (range != NULL, FALSE);
1293 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1297 g_rec_mutex_lock (&priv->state_lock);
1298 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1301 /* Update the seekable state of the pipeline in case it changed */
1302 query = gst_query_new_seeking (GST_FORMAT_TIME);
1303 if (gst_element_query (priv->pipeline, query)) {
1308 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1309 priv->seekable = seekable;
1311 gst_query_unref (query);
1313 if (!priv->seekable)
1316 /* depends on the current playing state of the pipeline. We might need to
1317 * queue this until we get EOS. */
1318 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT;
1320 start_type = stop_type = GST_SEEK_TYPE_NONE;
1322 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1324 gst_rtsp_range_get_times (range, &start, &stop);
1326 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1327 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1328 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1329 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1331 if (priv->range_start == start)
1332 start = GST_CLOCK_TIME_NONE;
1333 else if (start != GST_CLOCK_TIME_NONE)
1334 start_type = GST_SEEK_TYPE_SET;
1336 if (priv->range_stop == stop)
1337 stop = GST_CLOCK_TIME_NONE;
1338 else if (stop != GST_CLOCK_TIME_NONE)
1339 stop_type = GST_SEEK_TYPE_SET;
1341 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1342 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1343 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1345 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1346 flags, start_type, start, stop_type, stop);
1348 /* and block for the seek to complete */
1349 GST_INFO ("done seeking %d", res);
1350 g_rec_mutex_unlock (&priv->state_lock);
1352 /* wait until pipeline is prerolled again, this will also collect stats */
1353 if (!wait_preroll (media))
1354 goto preroll_failed;
1356 g_rec_mutex_lock (&priv->state_lock);
1357 GST_INFO ("prerolled again");
1359 GST_INFO ("no seek needed");
1362 g_rec_mutex_unlock (&priv->state_lock);
1369 g_rec_mutex_unlock (&priv->state_lock);
1370 GST_INFO ("media %p is not prepared", media);
1375 g_rec_mutex_unlock (&priv->state_lock);
1376 GST_INFO ("pipeline is not seekable");
1381 g_rec_mutex_unlock (&priv->state_lock);
1382 GST_WARNING ("conversion to npt not supported");
1387 GST_WARNING ("failed to preroll after seek");
1393 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1395 GstRTSPMediaPrivate *priv = media->priv;
1397 g_mutex_lock (&priv->lock);
1398 priv->status = status;
1399 GST_DEBUG ("setting new status to %d", status);
1400 g_cond_broadcast (&priv->cond);
1401 g_mutex_unlock (&priv->lock);
1405 * gst_rtsp_media_get_status:
1406 * @media: a #GstRTSPMedia
1408 * Get the status of @media. When @media is busy preparing, this function waits
1409 * until @media is prepared or in error.
1411 * Returns: the status of @media.
1414 gst_rtsp_media_get_status (GstRTSPMedia * media)
1416 GstRTSPMediaPrivate *priv = media->priv;
1417 GstRTSPMediaStatus result;
1420 g_mutex_lock (&priv->lock);
1421 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1422 /* while we are preparing, wait */
1423 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1424 GST_DEBUG ("waiting for status change");
1425 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1426 GST_DEBUG ("timeout, assuming error status");
1427 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1430 /* could be success or error */
1431 result = priv->status;
1432 GST_DEBUG ("got status %d", result);
1433 g_mutex_unlock (&priv->lock);
1438 /* called with state-lock */
1440 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1442 GstRTSPMediaPrivate *priv = media->priv;
1443 GstMessageType type;
1445 type = GST_MESSAGE_TYPE (message);
1448 case GST_MESSAGE_STATE_CHANGED:
1450 case GST_MESSAGE_BUFFERING:
1454 gst_message_parse_buffering (message, &percent);
1456 /* no state management needed for live pipelines */
1460 if (percent == 100) {
1461 /* a 100% message means buffering is done */
1462 priv->buffering = FALSE;
1463 /* if the desired state is playing, go back */
1464 if (priv->target_state == GST_STATE_PLAYING) {
1465 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1466 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1468 GST_INFO ("Buffering done");
1471 /* buffering busy */
1472 if (priv->buffering == FALSE) {
1473 if (priv->target_state == GST_STATE_PLAYING) {
1474 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1475 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1476 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1478 GST_INFO ("Buffering ...");
1481 priv->buffering = TRUE;
1485 case GST_MESSAGE_LATENCY:
1487 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1490 case GST_MESSAGE_ERROR:
1495 gst_message_parse_error (message, &gerror, &debug);
1496 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1497 g_error_free (gerror);
1500 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1503 case GST_MESSAGE_WARNING:
1508 gst_message_parse_warning (message, &gerror, &debug);
1509 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1510 g_error_free (gerror);
1514 case GST_MESSAGE_ELEMENT:
1516 case GST_MESSAGE_STREAM_STATUS:
1518 case GST_MESSAGE_ASYNC_DONE:
1520 /* when we are dynamically adding pads, the addition of the udpsrc will
1521 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1522 * wait for the final ASYNC_DONE after everything prerolled */
1523 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1525 GST_INFO ("%p: got ASYNC_DONE", media);
1526 collect_media_stats (media);
1528 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1529 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1532 case GST_MESSAGE_EOS:
1533 GST_INFO ("%p: got EOS", media);
1535 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1536 GST_DEBUG ("shutting down after EOS");
1537 finish_unprepare (media);
1541 GST_INFO ("%p: got message type %d (%s)", media, type,
1542 gst_message_type_get_name (type));
1549 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1551 GstRTSPMediaPrivate *priv = media->priv;
1552 GstRTSPMediaClass *klass;
1555 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1557 g_rec_mutex_lock (&priv->state_lock);
1558 if (klass->handle_message)
1559 ret = klass->handle_message (media, message);
1562 g_rec_mutex_unlock (&priv->state_lock);
1568 watch_destroyed (GstRTSPMedia * media)
1570 GST_DEBUG_OBJECT (media, "source destroyed");
1571 g_object_unref (media);
1575 find_payload_element (GstElement * payloader)
1577 GstElement *pay = NULL;
1579 if (GST_IS_BIN (payloader)) {
1581 GValue item = { 0 };
1583 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
1584 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
1585 GstElement *element = (GstElement *) g_value_get_object (&item);
1586 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
1590 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
1594 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
1595 pay = gst_object_ref (element);
1596 g_value_unset (&item);
1599 g_value_unset (&item);
1601 gst_iterator_free (iter);
1603 pay = g_object_ref (payloader);
1609 /* called from streaming threads */
1611 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1613 GstRTSPMediaPrivate *priv = media->priv;
1614 GstRTSPStream *stream;
1617 /* find the real payload element */
1618 pay = find_payload_element (element);
1619 stream = gst_rtsp_media_create_stream (media, pay, pad);
1620 gst_object_unref (pay);
1622 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1624 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1626 g_rec_mutex_lock (&priv->state_lock);
1627 /* we will be adding elements below that will cause ASYNC_DONE to be
1628 * posted in the bus. We want to ignore those messages until the
1629 * pipeline really prerolled. */
1630 priv->adding = TRUE;
1632 /* join the element in the PAUSED state because this callback is
1633 * called from the streaming thread and it is PAUSED */
1634 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1635 priv->rtpbin, GST_STATE_PAUSED);
1637 priv->adding = FALSE;
1638 g_rec_mutex_unlock (&priv->state_lock);
1642 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1644 GstRTSPMediaPrivate *priv = media->priv;
1645 GstRTSPStream *stream;
1647 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1651 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1653 g_rec_mutex_lock (&priv->state_lock);
1654 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1655 g_rec_mutex_unlock (&priv->state_lock);
1657 gst_rtsp_media_remove_stream (media, stream);
1661 remove_fakesink (GstRTSPMediaPrivate * priv)
1663 GstElement *fakesink;
1665 g_mutex_lock (&priv->lock);
1666 if ((fakesink = priv->fakesink))
1667 gst_object_ref (fakesink);
1668 priv->fakesink = NULL;
1669 g_mutex_unlock (&priv->lock);
1672 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1673 gst_element_set_state (fakesink, GST_STATE_NULL);
1674 gst_object_unref (fakesink);
1675 GST_INFO ("removed fakesink");
1680 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1682 GstRTSPMediaPrivate *priv = media->priv;
1684 GST_INFO ("no more pads");
1685 remove_fakesink (priv);
1688 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1690 struct _DynPaySignalHandlers
1692 gulong pad_added_handler;
1693 gulong pad_removed_handler;
1694 gulong no_more_pads_handler;
1698 start_preroll (GstRTSPMedia * media)
1700 GstRTSPMediaPrivate *priv = media->priv;
1701 GstStateChangeReturn ret;
1703 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1704 /* first go to PAUSED */
1705 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1706 priv->target_state = GST_STATE_PAUSED;
1709 case GST_STATE_CHANGE_SUCCESS:
1710 GST_INFO ("SUCCESS state change for media %p", media);
1711 priv->seekable = TRUE;
1713 case GST_STATE_CHANGE_ASYNC:
1714 GST_INFO ("ASYNC state change for media %p", media);
1715 priv->seekable = TRUE;
1717 case GST_STATE_CHANGE_NO_PREROLL:
1718 /* we need to go to PLAYING */
1719 GST_INFO ("NO_PREROLL state change: live media %p", media);
1720 /* FIXME we disable seeking for live streams for now. We should perform a
1721 * seeking query in preroll instead */
1722 priv->seekable = FALSE;
1723 priv->is_live = TRUE;
1724 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1725 if (ret == GST_STATE_CHANGE_FAILURE)
1728 case GST_STATE_CHANGE_FAILURE:
1736 GST_WARNING ("failed to preroll pipeline");
1742 wait_preroll (GstRTSPMedia * media)
1744 GstRTSPMediaStatus status;
1746 GST_DEBUG ("wait to preroll pipeline");
1748 /* wait until pipeline is prerolled */
1749 status = gst_rtsp_media_get_status (media);
1750 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1751 goto preroll_failed;
1757 GST_WARNING ("failed to preroll pipeline");
1763 start_prepare (GstRTSPMedia * media)
1765 GstRTSPMediaPrivate *priv = media->priv;
1769 /* link streams we already have, other streams might appear when we have
1770 * dynamic elements */
1771 for (i = 0; i < priv->streams->len; i++) {
1772 GstRTSPStream *stream;
1774 stream = g_ptr_array_index (priv->streams, i);
1776 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1777 priv->rtpbin, GST_STATE_NULL);
1780 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1781 GstElement *elem = walk->data;
1782 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
1784 GST_INFO ("adding callbacks for dynamic element %p", elem);
1786 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
1787 (GCallback) pad_added_cb, media);
1788 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
1789 (GCallback) pad_removed_cb, media);
1790 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
1791 (GCallback) no_more_pads_cb, media);
1793 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
1795 /* we add a fakesink here in order to make the state change async. We remove
1796 * the fakesink again in the no-more-pads callback. */
1797 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
1798 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
1801 if (!start_preroll (media))
1802 goto preroll_failed;
1808 GST_WARNING ("failed to preroll pipeline");
1809 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1815 * gst_rtsp_media_prepare:
1816 * @media: a #GstRTSPMedia
1817 * @thread: a #GstRTSPThread to run the bus handler or %NULL
1819 * Prepare @media for streaming. This function will create the objects
1820 * to manage the streaming. A pipeline must have been set on @media with
1821 * gst_rtsp_media_take_pipeline().
1823 * It will preroll the pipeline and collect vital information about the streams
1824 * such as the duration.
1826 * Returns: %TRUE on success.
1829 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
1831 GstRTSPMediaPrivate *priv;
1835 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1836 g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
1840 g_rec_mutex_lock (&priv->state_lock);
1841 priv->prepare_count++;
1843 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1846 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1849 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
1850 goto not_unprepared;
1852 if (!priv->reusable && priv->reused)
1855 priv->rtpbin = gst_element_factory_make ("rtpbin", NULL);
1856 if (priv->rtpbin != NULL) {
1857 GstRTSPMediaClass *klass;
1858 gboolean success = TRUE;
1860 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1861 if (klass->setup_rtpbin)
1862 success = klass->setup_rtpbin (media, priv->rtpbin);
1864 if (success == FALSE) {
1865 gst_object_unref (priv->rtpbin);
1866 priv->rtpbin = NULL;
1869 if (priv->rtpbin == NULL)
1872 GST_INFO ("preparing media %p", media);
1874 /* reset some variables */
1875 priv->is_live = FALSE;
1876 priv->seekable = FALSE;
1877 priv->buffering = FALSE;
1878 priv->thread = thread;
1879 /* we're preparing now */
1880 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1882 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
1884 /* add the pipeline bus to our custom mainloop */
1885 priv->source = gst_bus_create_watch (bus);
1886 gst_object_unref (bus);
1888 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
1889 g_object_ref (media), (GDestroyNotify) watch_destroyed);
1891 priv->id = g_source_attach (priv->source, thread->context);
1893 /* add stuff to the bin */
1894 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
1896 /* do remainder in context */
1897 source = g_idle_source_new ();
1898 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
1899 g_source_attach (source, thread->context);
1900 g_source_unref (source);
1903 g_rec_mutex_unlock (&priv->state_lock);
1905 /* now wait for all pads to be prerolled, FIXME, we should somehow be
1906 * able to do this async so that we don't block the server thread. */
1907 if (!wait_preroll (media))
1908 goto preroll_failed;
1910 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
1912 GST_INFO ("object %p is prerolled", media);
1919 GST_LOG ("media %p was prepared", media);
1920 g_rec_mutex_unlock (&priv->state_lock);
1926 GST_WARNING ("media %p was not unprepared", media);
1927 priv->prepare_count--;
1928 g_rec_mutex_unlock (&priv->state_lock);
1933 priv->prepare_count--;
1934 g_rec_mutex_unlock (&priv->state_lock);
1935 GST_WARNING ("can not reuse media %p", media);
1940 priv->prepare_count--;
1941 g_rec_mutex_unlock (&priv->state_lock);
1942 GST_WARNING ("no rtpbin element");
1943 g_warning ("failed to create element 'rtpbin', check your installation");
1948 GST_WARNING ("failed to preroll pipeline");
1949 gst_rtsp_media_unprepare (media);
1954 /* must be called with state-lock */
1956 finish_unprepare (GstRTSPMedia * media)
1958 GstRTSPMediaPrivate *priv = media->priv;
1962 GST_DEBUG ("shutting down");
1964 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
1965 remove_fakesink (priv);
1967 for (i = 0; i < priv->streams->len; i++) {
1968 GstRTSPStream *stream;
1970 GST_INFO ("Removing elements of stream %d from pipeline", i);
1972 stream = g_ptr_array_index (priv->streams, i);
1974 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1977 /* remove the pad signal handlers */
1978 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
1979 GstElement *elem = walk->data;
1980 DynPaySignalHandlers *handlers;
1983 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
1984 g_assert (handlers != NULL);
1986 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
1987 g_signal_handler_disconnect (G_OBJECT (elem),
1988 handlers->pad_removed_handler);
1989 g_signal_handler_disconnect (G_OBJECT (elem),
1990 handlers->no_more_pads_handler);
1992 g_slice_free (DynPaySignalHandlers, handlers);
1995 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
1996 priv->rtpbin = NULL;
1999 gst_object_unref (priv->nettime);
2000 priv->nettime = NULL;
2002 priv->reused = TRUE;
2003 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
2005 /* when the media is not reusable, this will effectively unref the media and
2007 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2009 /* the source has the last ref to the media */
2011 GST_DEBUG ("destroy source");
2012 g_source_destroy (priv->source);
2013 g_source_unref (priv->source);
2016 GST_DEBUG ("stop thread");
2017 gst_rtsp_thread_stop (priv->thread);
2021 /* called with state-lock */
2023 default_unprepare (GstRTSPMedia * media)
2025 GstRTSPMediaPrivate *priv = media->priv;
2027 if (priv->eos_shutdown) {
2028 GST_DEBUG ("sending EOS for shutdown");
2029 /* ref so that we don't disappear */
2030 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2031 /* we need to go to playing again for the EOS to propagate, normally in this
2032 * state, nothing is receiving data from us anymore so this is ok. */
2033 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
2034 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
2036 finish_unprepare (media);
2042 * gst_rtsp_media_unprepare:
2043 * @media: a #GstRTSPMedia
2045 * Unprepare @media. After this call, the media should be prepared again before
2046 * it can be used again. If the media is set to be non-reusable, a new instance
2049 * Returns: %TRUE on success.
2052 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2054 GstRTSPMediaPrivate *priv;
2057 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2061 g_rec_mutex_lock (&priv->state_lock);
2062 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2063 goto was_unprepared;
2065 priv->prepare_count--;
2066 if (priv->prepare_count > 0)
2069 GST_INFO ("unprepare media %p", media);
2070 priv->target_state = GST_STATE_NULL;
2073 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2074 GstRTSPMediaClass *klass;
2076 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2077 if (klass->unprepare)
2078 success = klass->unprepare (media);
2080 finish_unprepare (media);
2082 g_rec_mutex_unlock (&priv->state_lock);
2088 g_rec_mutex_unlock (&priv->state_lock);
2089 GST_INFO ("media %p was already unprepared", media);
2094 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2095 g_rec_mutex_unlock (&priv->state_lock);
2100 /* should be called with state-lock */
2102 get_clock_unlocked (GstRTSPMedia * media)
2104 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2105 GST_DEBUG_OBJECT (media, "media was not prepared");
2108 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2112 * gst_rtsp_media_get_clock:
2113 * @media: a #GstRTSPMedia
2115 * Get the clock that is used by the pipeline in @media.
2117 * @media must be prepared before this method returns a valid clock object.
2119 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2122 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2125 GstRTSPMediaPrivate *priv;
2127 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2131 g_rec_mutex_lock (&priv->state_lock);
2132 clock = get_clock_unlocked (media);
2133 g_rec_mutex_unlock (&priv->state_lock);
2139 * gst_rtsp_media_get_base_time:
2140 * @media: a #GstRTSPMedia
2142 * Get the base_time that is used by the pipeline in @media.
2144 * @media must be prepared before this method returns a valid base_time.
2146 * Returns: the base_time used by @media.
2149 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2151 GstClockTime result;
2152 GstRTSPMediaPrivate *priv;
2154 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2158 g_rec_mutex_lock (&priv->state_lock);
2159 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2162 result = gst_element_get_base_time (media->priv->pipeline);
2163 g_rec_mutex_unlock (&priv->state_lock);
2170 g_rec_mutex_unlock (&priv->state_lock);
2171 GST_DEBUG_OBJECT (media, "media was not prepared");
2172 return GST_CLOCK_TIME_NONE;
2177 * gst_rtsp_media_get_time_provider:
2178 * @media: a #GstRTSPMedia
2179 * @address: an address or %NULL
2180 * @port: a port or 0
2182 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2183 * will listen on @address and @port for client time requests.
2185 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2187 GstNetTimeProvider *
2188 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2191 GstRTSPMediaPrivate *priv;
2192 GstNetTimeProvider *provider = NULL;
2194 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2198 g_rec_mutex_lock (&priv->state_lock);
2199 if (priv->time_provider) {
2200 if ((provider = priv->nettime) == NULL) {
2203 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2204 provider = gst_net_time_provider_new (clock, address, port);
2205 gst_object_unref (clock);
2207 priv->nettime = provider;
2211 g_rec_mutex_unlock (&priv->state_lock);
2214 gst_object_ref (provider);
2220 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
2222 GstRTSPMediaPrivate *priv = media->priv;
2224 if (state == GST_STATE_NULL) {
2225 gst_rtsp_media_unprepare (media);
2227 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
2228 priv->target_state = state;
2229 /* when we are buffering, don't update the state yet, this will be done
2230 * when buffering finishes */
2231 if (priv->buffering) {
2232 GST_INFO ("Buffering busy, delay state change");
2234 gst_element_set_state (priv->pipeline, state);
2240 * gst_rtsp_media_set_pipeline_state:
2241 * @media: a #GstRTSPMedia
2242 * @state: the target state of the pipeline
2244 * Set the state of the pipeline managed by @media to @state
2247 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
2249 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2251 g_rec_mutex_lock (&media->priv->state_lock);
2252 media_set_pipeline_state_locked (media, state);
2253 g_rec_mutex_unlock (&media->priv->state_lock);
2257 * gst_rtsp_media_set_state:
2258 * @media: a #GstRTSPMedia
2259 * @state: the target state of the media
2260 * @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
2261 * of #GstRTSPStreamTransport pointers
2263 * Set the state of @media to @state and for the transports in @transports.
2265 * @media must be prepared with gst_rtsp_media_prepare();
2267 * Returns: %TRUE on success.
2270 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2271 GPtrArray * transports)
2273 GstRTSPMediaPrivate *priv;
2275 gboolean activate, deactivate, do_state;
2278 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2279 g_return_val_if_fail (transports != NULL, FALSE);
2283 g_rec_mutex_lock (&priv->state_lock);
2284 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
2286 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2289 /* NULL and READY are the same */
2290 if (state == GST_STATE_READY)
2291 state = GST_STATE_NULL;
2293 activate = deactivate = FALSE;
2295 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
2299 case GST_STATE_NULL:
2300 case GST_STATE_PAUSED:
2301 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
2302 if (priv->target_state == GST_STATE_PLAYING)
2305 case GST_STATE_PLAYING:
2306 /* we're going to PLAYING, activate */
2312 old_active = priv->n_active;
2314 for (i = 0; i < transports->len; i++) {
2315 GstRTSPStreamTransport *trans;
2317 /* we need a non-NULL entry in the array */
2318 trans = g_ptr_array_index (transports, i);
2323 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2325 } else if (deactivate) {
2326 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2331 /* we just activated the first media, do the playing state change */
2332 if (old_active == 0 && activate)
2334 /* if we have no more active media, do the downward state changes */
2335 else if (priv->n_active == 0)
2340 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2343 if (priv->target_state != state) {
2345 media_set_pipeline_state_locked (media, state);
2347 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2351 /* remember where we are */
2352 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2353 old_active != priv->n_active))
2354 collect_media_stats (media);
2356 g_rec_mutex_unlock (&priv->state_lock);
2363 GST_WARNING ("media %p was not prepared", media);
2364 g_rec_mutex_unlock (&priv->state_lock);
2369 GST_WARNING ("media %p in error status while changing to state %d",
2371 if (state == GST_STATE_NULL) {
2372 for (i = 0; i < transports->len; i++) {
2373 GstRTSPStreamTransport *trans;
2375 /* we need a non-NULL entry in the array */
2376 trans = g_ptr_array_index (transports, i);
2380 gst_rtsp_stream_transport_set_active (trans, FALSE);
2384 g_rec_mutex_unlock (&priv->state_lock);