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;
83 gboolean suspend_mode;
85 GstRTSPProfile profiles;
86 GstRTSPLowerTrans protocols;
88 gboolean eos_shutdown;
90 GstRTSPAddressPool *pool;
94 GRecMutex state_lock; /* locking order: state lock, lock */
95 GPtrArray *streams; /* protected by lock */
96 GList *dynamic; /* protected by lock */
97 GstRTSPMediaStatus status; /* protected by lock */
102 /* the pipeline for the media */
103 GstElement *pipeline;
104 GstElement *fakesink; /* protected by lock */
107 GstRTSPThread *thread;
109 gboolean time_provider;
110 GstNetTimeProvider *nettime;
115 GstState target_state;
117 /* RTP session manager */
120 /* the range of media */
121 GstRTSPTimeRange range; /* protected by lock */
122 GstClockTime range_start;
123 GstClockTime range_stop;
126 #define DEFAULT_SHARED FALSE
127 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
128 #define DEFAULT_REUSABLE FALSE
129 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
130 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
131 GST_RTSP_LOWER_TRANS_TCP
132 #define DEFAULT_EOS_SHUTDOWN FALSE
133 #define DEFAULT_BUFFER_SIZE 0x80000
134 #define DEFAULT_TIME_PROVIDER FALSE
136 /* define to dump received RTCP packets */
157 SIGNAL_REMOVED_STREAM,
165 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
166 #define GST_CAT_DEFAULT rtsp_media_debug
168 static void gst_rtsp_media_get_property (GObject * object, guint propid,
169 GValue * value, GParamSpec * pspec);
170 static void gst_rtsp_media_set_property (GObject * object, guint propid,
171 const GValue * value, GParamSpec * pspec);
172 static void gst_rtsp_media_finalize (GObject * obj);
174 static gboolean default_handle_message (GstRTSPMedia * media,
175 GstMessage * message);
176 static void finish_unprepare (GstRTSPMedia * media);
177 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
178 static gboolean default_unprepare (GstRTSPMedia * media);
179 static gboolean default_suspend (GstRTSPMedia * media);
180 static gboolean default_unsuspend (GstRTSPMedia * media);
181 static gboolean default_convert_range (GstRTSPMedia * media,
182 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
183 static gboolean default_query_position (GstRTSPMedia * media,
185 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
186 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
187 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
190 static gboolean wait_preroll (GstRTSPMedia * media);
192 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
194 #define C_ENUM(v) ((gint) v)
196 #define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
198 gst_rtsp_suspend_mode_get_type (void)
201 static const GEnumValue values[] = {
202 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
203 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
205 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
210 if (g_once_init_enter (&id)) {
211 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
212 g_once_init_leave (&id, tmp);
217 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
220 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
222 GObjectClass *gobject_class;
224 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
226 gobject_class = G_OBJECT_CLASS (klass);
228 gobject_class->get_property = gst_rtsp_media_get_property;
229 gobject_class->set_property = gst_rtsp_media_set_property;
230 gobject_class->finalize = gst_rtsp_media_finalize;
232 g_object_class_install_property (gobject_class, PROP_SHARED,
233 g_param_spec_boolean ("shared", "Shared",
234 "If this media pipeline can be shared", DEFAULT_SHARED,
235 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
237 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
238 g_param_spec_enum ("suspend-mode", "Suspend Mode",
239 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
240 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
242 g_object_class_install_property (gobject_class, PROP_REUSABLE,
243 g_param_spec_boolean ("reusable", "Reusable",
244 "If this media pipeline can be reused after an unprepare",
245 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
247 g_object_class_install_property (gobject_class, PROP_PROFILES,
248 g_param_spec_flags ("profiles", "Profiles",
249 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
250 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
252 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
253 g_param_spec_flags ("protocols", "Protocols",
254 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
255 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
257 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
258 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
259 "Send an EOS event to the pipeline before unpreparing",
260 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
262 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
263 g_param_spec_uint ("buffer-size", "Buffer Size",
264 "The kernel UDP buffer size to use", 0, G_MAXUINT,
265 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
267 g_object_class_install_property (gobject_class, PROP_ELEMENT,
268 g_param_spec_object ("element", "The Element",
269 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
270 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
272 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
273 g_param_spec_boolean ("time-provider", "Time Provider",
274 "Use a NetTimeProvider for clients",
275 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
277 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
278 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
279 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
280 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
282 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
283 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
284 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
285 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
286 GST_TYPE_RTSP_STREAM);
288 gst_rtsp_media_signals[SIGNAL_PREPARED] =
289 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
290 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
291 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
293 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
294 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
295 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
296 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
298 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
299 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
300 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL,
301 NULL, g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
303 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
304 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
305 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
306 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
308 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
310 klass->handle_message = default_handle_message;
311 klass->prepare = default_prepare;
312 klass->unprepare = default_unprepare;
313 klass->suspend = default_suspend;
314 klass->unsuspend = default_unsuspend;
315 klass->convert_range = default_convert_range;
316 klass->query_position = default_query_position;
317 klass->query_stop = default_query_stop;
318 klass->create_rtpbin = default_create_rtpbin;
319 klass->setup_sdp = default_setup_sdp;
323 gst_rtsp_media_init (GstRTSPMedia * media)
325 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
329 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
330 g_mutex_init (&priv->lock);
331 g_cond_init (&priv->cond);
332 g_rec_mutex_init (&priv->state_lock);
334 priv->shared = DEFAULT_SHARED;
335 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
336 priv->reusable = DEFAULT_REUSABLE;
337 priv->profiles = DEFAULT_PROFILES;
338 priv->protocols = DEFAULT_PROTOCOLS;
339 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
340 priv->buffer_size = DEFAULT_BUFFER_SIZE;
341 priv->time_provider = DEFAULT_TIME_PROVIDER;
345 gst_rtsp_media_finalize (GObject * obj)
347 GstRTSPMediaPrivate *priv;
350 media = GST_RTSP_MEDIA (obj);
353 GST_INFO ("finalize media %p", media);
355 if (priv->permissions)
356 gst_rtsp_permissions_unref (priv->permissions);
358 g_ptr_array_unref (priv->streams);
360 g_list_free_full (priv->dynamic, gst_object_unref);
363 gst_object_unref (priv->pipeline);
365 gst_object_unref (priv->nettime);
366 gst_object_unref (priv->element);
368 g_object_unref (priv->pool);
369 g_mutex_clear (&priv->lock);
370 g_cond_clear (&priv->cond);
371 g_rec_mutex_clear (&priv->state_lock);
373 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
377 gst_rtsp_media_get_property (GObject * object, guint propid,
378 GValue * value, GParamSpec * pspec)
380 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
384 g_value_set_object (value, media->priv->element);
387 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
389 case PROP_SUSPEND_MODE:
390 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
393 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
396 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
399 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
401 case PROP_EOS_SHUTDOWN:
402 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
404 case PROP_BUFFER_SIZE:
405 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
407 case PROP_TIME_PROVIDER:
408 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
411 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
416 gst_rtsp_media_set_property (GObject * object, guint propid,
417 const GValue * value, GParamSpec * pspec)
419 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
423 media->priv->element = g_value_get_object (value);
424 gst_object_ref_sink (media->priv->element);
427 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
429 case PROP_SUSPEND_MODE:
430 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
433 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
436 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
439 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
441 case PROP_EOS_SHUTDOWN:
442 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
444 case PROP_BUFFER_SIZE:
445 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
447 case PROP_TIME_PROVIDER:
448 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
451 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
456 default_query_position (GstRTSPMedia * media, gint64 * position)
458 return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
463 default_query_stop (GstRTSPMedia * media, gint64 * stop)
468 query = gst_query_new_segment (GST_FORMAT_TIME);
469 if ((res = gst_element_query (media->priv->pipeline, query))) {
471 gst_query_parse_segment (query, NULL, &format, NULL, stop);
472 if (format != GST_FORMAT_TIME)
475 gst_query_unref (query);
480 default_create_rtpbin (GstRTSPMedia * media)
484 rtpbin = gst_element_factory_make ("rtpbin", NULL);
489 /* must be called with state lock */
491 collect_media_stats (GstRTSPMedia * media)
493 GstRTSPMediaPrivate *priv = media->priv;
494 gint64 position, stop;
496 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
497 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
500 priv->range.unit = GST_RTSP_RANGE_NPT;
502 GST_INFO ("collect media stats");
505 priv->range.min.type = GST_RTSP_TIME_NOW;
506 priv->range.min.seconds = -1;
507 priv->range_start = -1;
508 priv->range.max.type = GST_RTSP_TIME_END;
509 priv->range.max.seconds = -1;
510 priv->range_stop = -1;
512 GstRTSPMediaClass *klass;
515 klass = GST_RTSP_MEDIA_GET_CLASS (media);
517 /* get the position */
519 if (klass->query_position)
520 ret = klass->query_position (media, &position);
523 GST_INFO ("position query failed");
527 /* get the current segment stop */
529 if (klass->query_stop)
530 ret = klass->query_stop (media, &stop);
533 GST_INFO ("stop query failed");
537 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
538 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
540 if (position == -1) {
541 priv->range.min.type = GST_RTSP_TIME_NOW;
542 priv->range.min.seconds = -1;
543 priv->range_start = -1;
545 priv->range.min.type = GST_RTSP_TIME_SECONDS;
546 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
547 priv->range_start = position;
550 priv->range.max.type = GST_RTSP_TIME_END;
551 priv->range.max.seconds = -1;
552 priv->range_stop = -1;
554 priv->range.max.type = GST_RTSP_TIME_SECONDS;
555 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
556 priv->range_stop = stop;
562 * gst_rtsp_media_new:
563 * @element: (transfer full): a #GstElement
565 * Create a new #GstRTSPMedia instance. @element is the bin element that
566 * provides the different streams. The #GstRTSPMedia object contains the
567 * element to produce RTP data for one or more related (audio/video/..)
570 * Ownership is taken of @element.
572 * Returns: (transfer full): a new #GstRTSPMedia object.
575 gst_rtsp_media_new (GstElement * element)
577 GstRTSPMedia *result;
579 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
581 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
587 * gst_rtsp_media_get_element:
588 * @media: a #GstRTSPMedia
590 * Get the element that was used when constructing @media.
592 * Returns: (transfer full): a #GstElement. Unref after usage.
595 gst_rtsp_media_get_element (GstRTSPMedia * media)
597 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
599 return gst_object_ref (media->priv->element);
603 * gst_rtsp_media_take_pipeline:
604 * @media: a #GstRTSPMedia
605 * @pipeline: (transfer full): a #GstPipeline
607 * Set @pipeline as the #GstPipeline for @media. Ownership is
608 * taken of @pipeline.
611 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
613 GstRTSPMediaPrivate *priv;
615 GstNetTimeProvider *nettime;
617 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
618 g_return_if_fail (GST_IS_PIPELINE (pipeline));
622 g_mutex_lock (&priv->lock);
623 old = priv->pipeline;
624 priv->pipeline = GST_ELEMENT_CAST (pipeline);
625 nettime = priv->nettime;
626 priv->nettime = NULL;
627 g_mutex_unlock (&priv->lock);
630 gst_object_unref (old);
633 gst_object_unref (nettime);
635 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
639 * gst_rtsp_media_set_permissions:
640 * @media: a #GstRTSPMedia
641 * @permissions: (transfer none): a #GstRTSPPermissions
643 * Set @permissions on @media.
646 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
647 GstRTSPPermissions * permissions)
649 GstRTSPMediaPrivate *priv;
651 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
655 g_mutex_lock (&priv->lock);
656 if (priv->permissions)
657 gst_rtsp_permissions_unref (priv->permissions);
658 if ((priv->permissions = permissions))
659 gst_rtsp_permissions_ref (permissions);
660 g_mutex_unlock (&priv->lock);
664 * gst_rtsp_media_get_permissions:
665 * @media: a #GstRTSPMedia
667 * Get the permissions object from @media.
669 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
672 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
674 GstRTSPMediaPrivate *priv;
675 GstRTSPPermissions *result;
677 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
681 g_mutex_lock (&priv->lock);
682 if ((result = priv->permissions))
683 gst_rtsp_permissions_ref (result);
684 g_mutex_unlock (&priv->lock);
690 * gst_rtsp_media_set_suspend_mode:
691 * @media: a #GstRTSPMedia
692 * @mode: the new #GstRTSPSuspendMode
694 * Control how @ media will be suspended after the SDP has been generated and
695 * after a PAUSE request has been performed.
697 * Media must be unprepared when setting the suspend mode.
700 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
702 GstRTSPMediaPrivate *priv;
704 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
708 g_rec_mutex_lock (&priv->state_lock);
709 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
711 priv->suspend_mode = mode;
712 g_rec_mutex_unlock (&priv->state_lock);
719 GST_WARNING ("media %p was prepared", media);
720 g_rec_mutex_unlock (&priv->state_lock);
725 * gst_rtsp_media_get_suspend_mode:
726 * @media: a #GstRTSPMedia
728 * Get how @media will be suspended.
730 * Returns: #GstRTSPSuspendMode.
733 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
735 GstRTSPMediaPrivate *priv;
736 GstRTSPSuspendMode res;
738 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
742 g_rec_mutex_lock (&priv->state_lock);
743 res = priv->suspend_mode;
744 g_rec_mutex_unlock (&priv->state_lock);
750 * gst_rtsp_media_set_shared:
751 * @media: a #GstRTSPMedia
752 * @shared: the new value
754 * Set or unset if the pipeline for @media can be shared will multiple clients.
755 * When @shared is %TRUE, client requests for this media will share the media
759 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
761 GstRTSPMediaPrivate *priv;
763 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
767 g_mutex_lock (&priv->lock);
768 priv->shared = shared;
769 g_mutex_unlock (&priv->lock);
773 * gst_rtsp_media_is_shared:
774 * @media: a #GstRTSPMedia
776 * Check if the pipeline for @media can be shared between multiple clients.
778 * Returns: %TRUE if the media can be shared between clients.
781 gst_rtsp_media_is_shared (GstRTSPMedia * media)
783 GstRTSPMediaPrivate *priv;
786 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
790 g_mutex_lock (&priv->lock);
792 g_mutex_unlock (&priv->lock);
798 * gst_rtsp_media_set_reusable:
799 * @media: a #GstRTSPMedia
800 * @reusable: the new value
802 * Set or unset if the pipeline for @media can be reused after the pipeline has
806 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
808 GstRTSPMediaPrivate *priv;
810 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
814 g_mutex_lock (&priv->lock);
815 priv->reusable = reusable;
816 g_mutex_unlock (&priv->lock);
820 * gst_rtsp_media_is_reusable:
821 * @media: a #GstRTSPMedia
823 * Check if the pipeline for @media can be reused after an unprepare.
825 * Returns: %TRUE if the media can be reused
828 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
830 GstRTSPMediaPrivate *priv;
833 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
837 g_mutex_lock (&priv->lock);
838 res = priv->reusable;
839 g_mutex_unlock (&priv->lock);
845 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
847 gst_rtsp_stream_set_profiles (stream, *profiles);
851 * gst_rtsp_media_set_profiles:
852 * @media: a #GstRTSPMedia
853 * @profiles: the new flags
855 * Configure the allowed lower transport for @media.
858 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
860 GstRTSPMediaPrivate *priv;
862 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
866 g_mutex_lock (&priv->lock);
867 priv->profiles = profiles;
868 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
869 g_mutex_unlock (&priv->lock);
873 * gst_rtsp_media_get_profiles:
874 * @media: a #GstRTSPMedia
876 * Get the allowed profiles of @media.
878 * Returns: a #GstRTSPProfile
881 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
883 GstRTSPMediaPrivate *priv;
886 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
890 g_mutex_lock (&priv->lock);
891 res = priv->profiles;
892 g_mutex_unlock (&priv->lock);
898 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
900 gst_rtsp_stream_set_protocols (stream, *protocols);
904 * gst_rtsp_media_set_protocols:
905 * @media: a #GstRTSPMedia
906 * @protocols: the new flags
908 * Configure the allowed lower transport for @media.
911 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
913 GstRTSPMediaPrivate *priv;
915 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
919 g_mutex_lock (&priv->lock);
920 priv->protocols = protocols;
921 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
922 g_mutex_unlock (&priv->lock);
926 * gst_rtsp_media_get_protocols:
927 * @media: a #GstRTSPMedia
929 * Get the allowed protocols of @media.
931 * Returns: a #GstRTSPLowerTrans
934 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
936 GstRTSPMediaPrivate *priv;
937 GstRTSPLowerTrans res;
939 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
940 GST_RTSP_LOWER_TRANS_UNKNOWN);
944 g_mutex_lock (&priv->lock);
945 res = priv->protocols;
946 g_mutex_unlock (&priv->lock);
952 * gst_rtsp_media_set_eos_shutdown:
953 * @media: a #GstRTSPMedia
954 * @eos_shutdown: the new value
956 * Set or unset if an EOS event will be sent to the pipeline for @media before
960 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
962 GstRTSPMediaPrivate *priv;
964 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
968 g_mutex_lock (&priv->lock);
969 priv->eos_shutdown = eos_shutdown;
970 g_mutex_unlock (&priv->lock);
974 * gst_rtsp_media_is_eos_shutdown:
975 * @media: a #GstRTSPMedia
977 * Check if the pipeline for @media will send an EOS down the pipeline before
980 * Returns: %TRUE if the media will send EOS before unpreparing.
983 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
985 GstRTSPMediaPrivate *priv;
988 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
992 g_mutex_lock (&priv->lock);
993 res = priv->eos_shutdown;
994 g_mutex_unlock (&priv->lock);
1000 * gst_rtsp_media_set_buffer_size:
1001 * @media: a #GstRTSPMedia
1002 * @size: the new value
1004 * Set the kernel UDP buffer size.
1007 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1009 GstRTSPMediaPrivate *priv;
1011 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1013 GST_LOG_OBJECT (media, "set buffer size %u", size);
1017 g_mutex_lock (&priv->lock);
1018 priv->buffer_size = size;
1019 g_mutex_unlock (&priv->lock);
1023 * gst_rtsp_media_get_buffer_size:
1024 * @media: a #GstRTSPMedia
1026 * Get the kernel UDP buffer size.
1028 * Returns: the kernel UDP buffer size.
1031 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1033 GstRTSPMediaPrivate *priv;
1036 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1040 g_mutex_unlock (&priv->lock);
1041 res = priv->buffer_size;
1042 g_mutex_unlock (&priv->lock);
1048 * gst_rtsp_media_use_time_provider:
1049 * @media: a #GstRTSPMedia
1050 * @time_provider: if a #GstNetTimeProvider should be used
1052 * Set @media to provide a #GstNetTimeProvider.
1055 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1057 GstRTSPMediaPrivate *priv;
1059 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1063 g_mutex_lock (&priv->lock);
1064 priv->time_provider = time_provider;
1065 g_mutex_unlock (&priv->lock);
1069 * gst_rtsp_media_is_time_provider:
1070 * @media: a #GstRTSPMedia
1072 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1074 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1076 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1079 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1081 GstRTSPMediaPrivate *priv;
1084 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1088 g_mutex_unlock (&priv->lock);
1089 res = priv->time_provider;
1090 g_mutex_unlock (&priv->lock);
1096 * gst_rtsp_media_set_address_pool:
1097 * @media: a #GstRTSPMedia
1098 * @pool: (transfer none): a #GstRTSPAddressPool
1100 * configure @pool to be used as the address pool of @media.
1103 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1104 GstRTSPAddressPool * pool)
1106 GstRTSPMediaPrivate *priv;
1107 GstRTSPAddressPool *old;
1109 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1113 GST_LOG_OBJECT (media, "set address pool %p", pool);
1115 g_mutex_lock (&priv->lock);
1116 if ((old = priv->pool) != pool)
1117 priv->pool = pool ? g_object_ref (pool) : NULL;
1120 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1122 g_mutex_unlock (&priv->lock);
1125 g_object_unref (old);
1129 * gst_rtsp_media_get_address_pool:
1130 * @media: a #GstRTSPMedia
1132 * Get the #GstRTSPAddressPool used as the address pool of @media.
1134 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1137 GstRTSPAddressPool *
1138 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1140 GstRTSPMediaPrivate *priv;
1141 GstRTSPAddressPool *result;
1143 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1147 g_mutex_lock (&priv->lock);
1148 if ((result = priv->pool))
1149 g_object_ref (result);
1150 g_mutex_unlock (&priv->lock);
1156 * gst_rtsp_media_collect_streams:
1157 * @media: a #GstRTSPMedia
1159 * Find all payloader elements, they should be named pay\%d in the
1160 * element of @media, and create #GstRTSPStreams for them.
1162 * Collect all dynamic elements, named dynpay\%d, and add them to
1163 * the list of dynamic elements.
1166 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1168 GstRTSPMediaPrivate *priv;
1169 GstElement *element, *elem;
1174 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1177 element = priv->element;
1180 for (i = 0; have_elem; i++) {
1185 name = g_strdup_printf ("pay%d", i);
1186 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1187 GST_INFO ("found stream %d with payloader %p", i, elem);
1189 /* take the pad of the payloader */
1190 pad = gst_element_get_static_pad (elem, "src");
1191 /* create the stream */
1192 gst_rtsp_media_create_stream (media, elem, pad);
1193 gst_object_unref (pad);
1194 gst_object_unref (elem);
1200 name = g_strdup_printf ("dynpay%d", i);
1201 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1202 /* a stream that will dynamically create pads to provide RTP packets */
1203 GST_INFO ("found dynamic element %d, %p", i, elem);
1205 g_mutex_lock (&priv->lock);
1206 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1207 g_mutex_unlock (&priv->lock);
1216 * gst_rtsp_media_create_stream:
1217 * @media: a #GstRTSPMedia
1218 * @payloader: a #GstElement
1219 * @srcpad: a source #GstPad
1221 * Create a new stream in @media that provides RTP data on @srcpad.
1222 * @srcpad should be a pad of an element inside @media->element.
1224 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1228 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1231 GstRTSPMediaPrivate *priv;
1232 GstRTSPStream *stream;
1237 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1238 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1239 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1240 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
1244 g_mutex_lock (&priv->lock);
1245 idx = priv->streams->len;
1247 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1249 name = g_strdup_printf ("src_%u", idx);
1250 srcpad = gst_ghost_pad_new (name, pad);
1251 gst_pad_set_active (srcpad, TRUE);
1252 gst_element_add_pad (priv->element, srcpad);
1255 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
1257 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1258 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1259 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1261 g_ptr_array_add (priv->streams, stream);
1262 g_mutex_unlock (&priv->lock);
1264 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1271 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1273 GstRTSPMediaPrivate *priv;
1278 g_mutex_lock (&priv->lock);
1279 /* remove the ghostpad */
1280 srcpad = gst_rtsp_stream_get_srcpad (stream);
1281 gst_element_remove_pad (priv->element, srcpad);
1282 gst_object_unref (srcpad);
1283 /* now remove the stream */
1284 g_object_ref (stream);
1285 g_ptr_array_remove (priv->streams, stream);
1286 g_mutex_unlock (&priv->lock);
1288 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1291 g_object_unref (stream);
1295 * gst_rtsp_media_n_streams:
1296 * @media: a #GstRTSPMedia
1298 * Get the number of streams in this media.
1300 * Returns: The number of streams.
1303 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1305 GstRTSPMediaPrivate *priv;
1308 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1312 g_mutex_lock (&priv->lock);
1313 res = priv->streams->len;
1314 g_mutex_unlock (&priv->lock);
1320 * gst_rtsp_media_get_stream:
1321 * @media: a #GstRTSPMedia
1322 * @idx: the stream index
1324 * Retrieve the stream with index @idx from @media.
1326 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1327 * that index did not exist.
1330 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1332 GstRTSPMediaPrivate *priv;
1335 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1339 g_mutex_lock (&priv->lock);
1340 if (idx < priv->streams->len)
1341 res = g_ptr_array_index (priv->streams, idx);
1344 g_mutex_unlock (&priv->lock);
1350 * gst_rtsp_media_find_stream:
1351 * @media: a #GstRTSPMedia
1352 * @control: the control of the stream
1354 * Find a stream in @media with @control as the control uri.
1356 * Returns: (transfer none): the #GstRTSPStream with control uri @control
1357 * or %NULL when a stream with that control did not exist.
1360 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1362 GstRTSPMediaPrivate *priv;
1366 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1367 g_return_val_if_fail (control != NULL, NULL);
1373 g_mutex_lock (&priv->lock);
1374 for (i = 0; i < priv->streams->len; i++) {
1375 GstRTSPStream *test;
1377 test = g_ptr_array_index (priv->streams, i);
1378 if (gst_rtsp_stream_has_control (test, control)) {
1383 g_mutex_unlock (&priv->lock);
1388 /* called with state-lock */
1390 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1391 GstRTSPRangeUnit unit)
1393 return gst_rtsp_range_convert_units (range, unit);
1397 * gst_rtsp_media_get_range_string:
1398 * @media: a #GstRTSPMedia
1399 * @play: for the PLAY request
1400 * @unit: the unit to use for the string
1402 * Get the current range as a string. @media must be prepared with
1403 * gst_rtsp_media_prepare ().
1405 * Returns: (transfer full): The range as a string, g_free() after usage.
1408 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1409 GstRTSPRangeUnit unit)
1411 GstRTSPMediaClass *klass;
1412 GstRTSPMediaPrivate *priv;
1414 GstRTSPTimeRange range;
1416 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1417 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1418 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1422 g_rec_mutex_lock (&priv->state_lock);
1423 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1424 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1427 g_mutex_lock (&priv->lock);
1429 /* Update the range value with current position/duration */
1430 collect_media_stats (media);
1433 range = priv->range;
1435 if (!play && priv->n_active > 0) {
1436 range.min.type = GST_RTSP_TIME_NOW;
1437 range.min.seconds = -1;
1439 g_mutex_unlock (&priv->lock);
1440 g_rec_mutex_unlock (&priv->state_lock);
1442 if (!klass->convert_range (media, &range, unit))
1443 goto conversion_failed;
1445 result = gst_rtsp_range_to_string (&range);
1452 GST_WARNING ("media %p was not prepared", media);
1453 g_rec_mutex_unlock (&priv->state_lock);
1458 GST_WARNING ("range conversion to unit %d failed", unit);
1464 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1466 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1470 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1472 GstRTSPMediaPrivate *priv = media->priv;
1474 GST_DEBUG ("media %p set blocked %d", media, blocked);
1475 priv->blocked = blocked;
1476 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1480 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1482 GstRTSPMediaPrivate *priv = media->priv;
1484 g_mutex_lock (&priv->lock);
1485 priv->status = status;
1486 GST_DEBUG ("setting new status to %d", status);
1487 g_cond_broadcast (&priv->cond);
1488 g_mutex_unlock (&priv->lock);
1492 * gst_rtsp_media_get_status:
1493 * @media: a #GstRTSPMedia
1495 * Get the status of @media. When @media is busy preparing, this function waits
1496 * until @media is prepared or in error.
1498 * Returns: the status of @media.
1501 gst_rtsp_media_get_status (GstRTSPMedia * media)
1503 GstRTSPMediaPrivate *priv = media->priv;
1504 GstRTSPMediaStatus result;
1507 g_mutex_lock (&priv->lock);
1508 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1509 /* while we are preparing, wait */
1510 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1511 GST_DEBUG ("waiting for status change");
1512 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1513 GST_DEBUG ("timeout, assuming error status");
1514 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1517 /* could be success or error */
1518 result = priv->status;
1519 GST_DEBUG ("got status %d", result);
1520 g_mutex_unlock (&priv->lock);
1526 * gst_rtsp_media_seek:
1527 * @media: a #GstRTSPMedia
1528 * @range: (transfer none): a #GstRTSPTimeRange
1530 * Seek the pipeline of @media to @range. @media must be prepared with
1531 * gst_rtsp_media_prepare().
1533 * Returns: %TRUE on success.
1536 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1538 GstRTSPMediaClass *klass;
1539 GstRTSPMediaPrivate *priv;
1541 GstClockTime start, stop;
1542 GstSeekType start_type, stop_type;
1545 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1547 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1548 g_return_val_if_fail (range != NULL, FALSE);
1549 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1553 g_rec_mutex_lock (&priv->state_lock);
1554 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1557 /* Update the seekable state of the pipeline in case it changed */
1558 query = gst_query_new_seeking (GST_FORMAT_TIME);
1559 if (gst_element_query (priv->pipeline, query)) {
1564 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1565 priv->seekable = seekable;
1567 gst_query_unref (query);
1569 if (!priv->seekable)
1572 start_type = stop_type = GST_SEEK_TYPE_NONE;
1574 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1576 gst_rtsp_range_get_times (range, &start, &stop);
1578 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1579 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1580 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1581 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1583 if (start != GST_CLOCK_TIME_NONE)
1584 start_type = GST_SEEK_TYPE_SET;
1586 if (priv->range_stop == stop)
1587 stop = GST_CLOCK_TIME_NONE;
1588 else if (stop != GST_CLOCK_TIME_NONE)
1589 stop_type = GST_SEEK_TYPE_SET;
1591 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1594 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1595 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1597 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
1599 media_streams_set_blocked (media, TRUE);
1601 /* depends on the current playing state of the pipeline. We might need to
1602 * queue this until we get EOS. */
1603 flags = GST_SEEK_FLAG_FLUSH;
1605 /* if range start was not supplied we must continue from current position.
1606 * but since we're doing a flushing seek, let us query the current position
1607 * so we end up at exactly the same position after the seek. */
1608 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
1610 gboolean ret = FALSE;
1612 if (klass->query_position)
1613 ret = klass->query_position (media, &position);
1616 GST_WARNING ("position query failed");
1618 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
1619 GST_TIME_ARGS (position));
1621 start_type = GST_SEEK_TYPE_SET;
1622 flags |= GST_SEEK_FLAG_ACCURATE;
1625 /* only set keyframe flag when modifying start */
1626 if (start_type != GST_SEEK_TYPE_NONE)
1627 flags |= GST_SEEK_FLAG_KEY_UNIT;
1630 /* FIXME, we only do forwards playback, no trick modes yet */
1631 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1632 flags, start_type, start, stop_type, stop);
1634 /* and block for the seek to complete */
1635 GST_INFO ("done seeking %d", res);
1636 g_rec_mutex_unlock (&priv->state_lock);
1638 /* wait until pipeline is prerolled again, this will also collect stats */
1639 if (!wait_preroll (media))
1640 goto preroll_failed;
1642 g_rec_mutex_lock (&priv->state_lock);
1643 GST_INFO ("prerolled again");
1645 GST_INFO ("no seek needed");
1648 g_rec_mutex_unlock (&priv->state_lock);
1655 g_rec_mutex_unlock (&priv->state_lock);
1656 GST_INFO ("media %p is not prepared", media);
1661 g_rec_mutex_unlock (&priv->state_lock);
1662 GST_INFO ("pipeline is not seekable");
1667 g_rec_mutex_unlock (&priv->state_lock);
1668 GST_WARNING ("conversion to npt not supported");
1673 GST_WARNING ("failed to preroll after seek");
1679 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
1681 *blocked &= gst_rtsp_stream_is_blocking (stream);
1685 media_streams_blocking (GstRTSPMedia * media)
1687 gboolean blocking = TRUE;
1689 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
1695 static GstStateChangeReturn
1696 set_state (GstRTSPMedia * media, GstState state)
1698 GstRTSPMediaPrivate *priv = media->priv;
1699 GstStateChangeReturn ret;
1701 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
1703 ret = gst_element_set_state (priv->pipeline, state);
1708 static GstStateChangeReturn
1709 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
1711 GstRTSPMediaPrivate *priv = media->priv;
1712 GstStateChangeReturn ret;
1714 GST_INFO ("set target state to %s for media %p",
1715 gst_element_state_get_name (state), media);
1716 priv->target_state = state;
1718 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
1719 priv->target_state, NULL);
1722 ret = set_state (media, state);
1724 ret = GST_STATE_CHANGE_SUCCESS;
1729 /* called with state-lock */
1731 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1733 GstRTSPMediaPrivate *priv = media->priv;
1734 GstMessageType type;
1736 type = GST_MESSAGE_TYPE (message);
1739 case GST_MESSAGE_STATE_CHANGED:
1741 case GST_MESSAGE_BUFFERING:
1745 gst_message_parse_buffering (message, &percent);
1747 /* no state management needed for live pipelines */
1751 if (percent == 100) {
1752 /* a 100% message means buffering is done */
1753 priv->buffering = FALSE;
1754 /* if the desired state is playing, go back */
1755 if (priv->target_state == GST_STATE_PLAYING) {
1756 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1757 set_state (media, GST_STATE_PLAYING);
1759 GST_INFO ("Buffering done");
1762 /* buffering busy */
1763 if (priv->buffering == FALSE) {
1764 if (priv->target_state == GST_STATE_PLAYING) {
1765 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1766 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1767 set_state (media, GST_STATE_PAUSED);
1769 GST_INFO ("Buffering ...");
1772 priv->buffering = TRUE;
1776 case GST_MESSAGE_LATENCY:
1778 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1781 case GST_MESSAGE_ERROR:
1786 gst_message_parse_error (message, &gerror, &debug);
1787 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1788 g_error_free (gerror);
1791 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1794 case GST_MESSAGE_WARNING:
1799 gst_message_parse_warning (message, &gerror, &debug);
1800 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1801 g_error_free (gerror);
1805 case GST_MESSAGE_ELEMENT:
1807 const GstStructure *s;
1809 s = gst_message_get_structure (message);
1810 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
1811 GST_DEBUG ("media received blocking message");
1812 if (priv->blocked && media_streams_blocking (media)) {
1813 GST_DEBUG ("media is blocking");
1814 collect_media_stats (media);
1816 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1817 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1822 case GST_MESSAGE_STREAM_STATUS:
1824 case GST_MESSAGE_ASYNC_DONE:
1826 /* when we are dynamically adding pads, the addition of the udpsrc will
1827 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1828 * wait for the final ASYNC_DONE after everything prerolled */
1829 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1831 GST_INFO ("%p: got ASYNC_DONE", media);
1832 collect_media_stats (media);
1834 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1835 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1838 case GST_MESSAGE_EOS:
1839 GST_INFO ("%p: got EOS", media);
1841 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1842 GST_DEBUG ("shutting down after EOS");
1843 finish_unprepare (media);
1847 GST_INFO ("%p: got message type %d (%s)", media, type,
1848 gst_message_type_get_name (type));
1855 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1857 GstRTSPMediaPrivate *priv = media->priv;
1858 GstRTSPMediaClass *klass;
1861 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1863 g_rec_mutex_lock (&priv->state_lock);
1864 if (klass->handle_message)
1865 ret = klass->handle_message (media, message);
1868 g_rec_mutex_unlock (&priv->state_lock);
1874 watch_destroyed (GstRTSPMedia * media)
1876 GST_DEBUG_OBJECT (media, "source destroyed");
1877 g_object_unref (media);
1881 find_payload_element (GstElement * payloader)
1883 GstElement *pay = NULL;
1885 if (GST_IS_BIN (payloader)) {
1887 GValue item = { 0 };
1889 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
1890 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
1891 GstElement *element = (GstElement *) g_value_get_object (&item);
1892 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
1896 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
1900 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
1901 pay = gst_object_ref (element);
1902 g_value_unset (&item);
1905 g_value_unset (&item);
1907 gst_iterator_free (iter);
1909 pay = g_object_ref (payloader);
1915 /* called from streaming threads */
1917 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1919 GstRTSPMediaPrivate *priv = media->priv;
1920 GstRTSPStream *stream;
1923 /* find the real payload element */
1924 pay = find_payload_element (element);
1925 stream = gst_rtsp_media_create_stream (media, pay, pad);
1926 gst_object_unref (pay);
1928 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1930 g_rec_mutex_lock (&priv->state_lock);
1931 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
1934 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1936 /* we will be adding elements below that will cause ASYNC_DONE to be
1937 * posted in the bus. We want to ignore those messages until the
1938 * pipeline really prerolled. */
1939 priv->adding = TRUE;
1941 /* join the element in the PAUSED state because this callback is
1942 * called from the streaming thread and it is PAUSED */
1943 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1944 priv->rtpbin, GST_STATE_PAUSED)) {
1945 GST_WARNING ("failed to join bin element");
1948 priv->adding = FALSE;
1949 g_rec_mutex_unlock (&priv->state_lock);
1956 gst_rtsp_media_remove_stream (media, stream);
1957 g_rec_mutex_unlock (&priv->state_lock);
1958 GST_INFO ("ignore pad because we are not preparing");
1964 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1966 GstRTSPMediaPrivate *priv = media->priv;
1967 GstRTSPStream *stream;
1969 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1973 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1975 g_rec_mutex_lock (&priv->state_lock);
1976 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1977 g_rec_mutex_unlock (&priv->state_lock);
1979 gst_rtsp_media_remove_stream (media, stream);
1983 remove_fakesink (GstRTSPMediaPrivate * priv)
1985 GstElement *fakesink;
1987 g_mutex_lock (&priv->lock);
1988 if ((fakesink = priv->fakesink))
1989 gst_object_ref (fakesink);
1990 priv->fakesink = NULL;
1991 g_mutex_unlock (&priv->lock);
1994 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1995 gst_element_set_state (fakesink, GST_STATE_NULL);
1996 gst_object_unref (fakesink);
1997 GST_INFO ("removed fakesink");
2002 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2004 GstRTSPMediaPrivate *priv = media->priv;
2006 GST_INFO ("no more pads");
2007 remove_fakesink (priv);
2010 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2012 struct _DynPaySignalHandlers
2014 gulong pad_added_handler;
2015 gulong pad_removed_handler;
2016 gulong no_more_pads_handler;
2020 start_preroll (GstRTSPMedia * media)
2022 GstRTSPMediaPrivate *priv = media->priv;
2023 GstStateChangeReturn ret;
2025 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2026 /* first go to PAUSED */
2027 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2030 case GST_STATE_CHANGE_SUCCESS:
2031 GST_INFO ("SUCCESS state change for media %p", media);
2032 priv->seekable = TRUE;
2034 case GST_STATE_CHANGE_ASYNC:
2035 GST_INFO ("ASYNC state change for media %p", media);
2036 priv->seekable = TRUE;
2038 case GST_STATE_CHANGE_NO_PREROLL:
2039 /* we need to go to PLAYING */
2040 GST_INFO ("NO_PREROLL state change: live media %p", media);
2041 /* FIXME we disable seeking for live streams for now. We should perform a
2042 * seeking query in preroll instead */
2043 priv->seekable = FALSE;
2044 priv->is_live = TRUE;
2045 /* start blocked to make sure nothing goes to the sink */
2046 media_streams_set_blocked (media, TRUE);
2047 ret = set_state (media, GST_STATE_PLAYING);
2048 if (ret == GST_STATE_CHANGE_FAILURE)
2051 case GST_STATE_CHANGE_FAILURE:
2059 GST_WARNING ("failed to preroll pipeline");
2065 wait_preroll (GstRTSPMedia * media)
2067 GstRTSPMediaStatus status;
2069 GST_DEBUG ("wait to preroll pipeline");
2071 /* wait until pipeline is prerolled */
2072 status = gst_rtsp_media_get_status (media);
2073 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2074 goto preroll_failed;
2080 GST_WARNING ("failed to preroll pipeline");
2086 start_prepare (GstRTSPMedia * media)
2088 GstRTSPMediaPrivate *priv = media->priv;
2092 /* link streams we already have, other streams might appear when we have
2093 * dynamic elements */
2094 for (i = 0; i < priv->streams->len; i++) {
2095 GstRTSPStream *stream;
2097 stream = g_ptr_array_index (priv->streams, i);
2099 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2100 priv->rtpbin, GST_STATE_NULL)) {
2101 goto join_bin_failed;
2105 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2106 GstElement *elem = walk->data;
2107 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2109 GST_INFO ("adding callbacks for dynamic element %p", elem);
2111 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2112 (GCallback) pad_added_cb, media);
2113 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2114 (GCallback) pad_removed_cb, media);
2115 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2116 (GCallback) no_more_pads_cb, media);
2118 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2120 /* we add a fakesink here in order to make the state change async. We remove
2121 * the fakesink again in the no-more-pads callback. */
2122 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2123 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2126 if (!start_preroll (media))
2127 goto preroll_failed;
2133 GST_WARNING ("failed to join bin element");
2134 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2139 GST_WARNING ("failed to preroll pipeline");
2140 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2146 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2148 GstRTSPMediaPrivate *priv;
2149 GstRTSPMediaClass *klass;
2151 GMainContext *context;
2156 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2158 if (!klass->create_rtpbin)
2159 goto no_create_rtpbin;
2161 priv->rtpbin = klass->create_rtpbin (media);
2162 if (priv->rtpbin != NULL) {
2163 gboolean success = TRUE;
2165 if (klass->setup_rtpbin)
2166 success = klass->setup_rtpbin (media, priv->rtpbin);
2168 if (success == FALSE) {
2169 gst_object_unref (priv->rtpbin);
2170 priv->rtpbin = NULL;
2173 if (priv->rtpbin == NULL)
2176 priv->thread = thread;
2177 context = (thread != NULL) ? (thread->context) : NULL;
2179 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2181 /* add the pipeline bus to our custom mainloop */
2182 priv->source = gst_bus_create_watch (bus);
2183 gst_object_unref (bus);
2185 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2186 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2188 priv->id = g_source_attach (priv->source, context);
2190 /* add stuff to the bin */
2191 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2193 /* do remainder in context */
2194 source = g_idle_source_new ();
2195 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
2196 g_source_attach (source, context);
2197 g_source_unref (source);
2204 GST_ERROR ("no create_rtpbin function");
2205 g_critical ("no create_rtpbin vmethod function set");
2210 GST_WARNING ("no rtpbin element");
2211 g_warning ("failed to create element 'rtpbin', check your installation");
2217 * gst_rtsp_media_prepare:
2218 * @media: a #GstRTSPMedia
2219 * @thread: (transfer full): a #GstRTSPThread to run the bus handler or %NULL
2221 * Prepare @media for streaming. This function will create the objects
2222 * to manage the streaming. A pipeline must have been set on @media with
2223 * gst_rtsp_media_take_pipeline().
2225 * It will preroll the pipeline and collect vital information about the streams
2226 * such as the duration.
2228 * Returns: %TRUE on success.
2231 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2233 GstRTSPMediaPrivate *priv;
2234 GstRTSPMediaClass *klass;
2236 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2240 g_rec_mutex_lock (&priv->state_lock);
2241 priv->prepare_count++;
2243 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2244 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2247 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2250 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2251 goto not_unprepared;
2253 if (!priv->reusable && priv->reused)
2256 GST_INFO ("preparing media %p", media);
2258 /* reset some variables */
2259 priv->is_live = FALSE;
2260 priv->seekable = FALSE;
2261 priv->buffering = FALSE;
2263 /* we're preparing now */
2264 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2266 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2267 if (klass->prepare) {
2268 if (!klass->prepare (media, thread))
2269 goto prepare_failed;
2273 g_rec_mutex_unlock (&priv->state_lock);
2275 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2276 * able to do this async so that we don't block the server thread. */
2277 if (!wait_preroll (media))
2278 goto preroll_failed;
2280 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2282 GST_INFO ("object %p is prerolled", media);
2289 /* we are not going to use the giving thread, so stop it. */
2291 gst_rtsp_thread_stop (thread);
2296 GST_LOG ("media %p was prepared", media);
2297 /* we are not going to use the giving thread, so stop it. */
2299 gst_rtsp_thread_stop (thread);
2300 g_rec_mutex_unlock (&priv->state_lock);
2306 /* we are not going to use the giving thread, so stop it. */
2308 gst_rtsp_thread_stop (thread);
2309 GST_WARNING ("media %p was not unprepared", media);
2310 priv->prepare_count--;
2311 g_rec_mutex_unlock (&priv->state_lock);
2316 /* we are not going to use the giving thread, so stop it. */
2318 gst_rtsp_thread_stop (thread);
2319 priv->prepare_count--;
2320 g_rec_mutex_unlock (&priv->state_lock);
2321 GST_WARNING ("can not reuse media %p", media);
2326 /* we are not going to use the giving thread, so stop it. */
2328 gst_rtsp_thread_stop (thread);
2329 priv->prepare_count--;
2330 g_rec_mutex_unlock (&priv->state_lock);
2331 GST_ERROR ("failed to prepare media");
2336 GST_WARNING ("failed to preroll pipeline");
2337 gst_rtsp_media_unprepare (media);
2342 /* must be called with state-lock */
2344 finish_unprepare (GstRTSPMedia * media)
2346 GstRTSPMediaPrivate *priv = media->priv;
2350 GST_DEBUG ("shutting down");
2352 /* release the lock on shutdown, otherwise pad_added_cb might try to
2353 * acquire the lock and then we deadlock */
2354 g_rec_mutex_unlock (&priv->state_lock);
2355 set_state (media, GST_STATE_NULL);
2356 g_rec_mutex_lock (&priv->state_lock);
2357 remove_fakesink (priv);
2359 for (i = 0; i < priv->streams->len; i++) {
2360 GstRTSPStream *stream;
2362 GST_INFO ("Removing elements of stream %d from pipeline", i);
2364 stream = g_ptr_array_index (priv->streams, i);
2366 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2369 /* remove the pad signal handlers */
2370 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2371 GstElement *elem = walk->data;
2372 DynPaySignalHandlers *handlers;
2375 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2376 g_assert (handlers != NULL);
2378 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2379 g_signal_handler_disconnect (G_OBJECT (elem),
2380 handlers->pad_removed_handler);
2381 g_signal_handler_disconnect (G_OBJECT (elem),
2382 handlers->no_more_pads_handler);
2384 g_slice_free (DynPaySignalHandlers, handlers);
2387 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2388 priv->rtpbin = NULL;
2391 gst_object_unref (priv->nettime);
2392 priv->nettime = NULL;
2394 priv->reused = TRUE;
2395 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2397 /* when the media is not reusable, this will effectively unref the media and
2399 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2401 /* the source has the last ref to the media */
2403 GST_DEBUG ("destroy source");
2404 g_source_destroy (priv->source);
2405 g_source_unref (priv->source);
2408 GST_DEBUG ("stop thread");
2409 gst_rtsp_thread_stop (priv->thread);
2413 /* called with state-lock */
2415 default_unprepare (GstRTSPMedia * media)
2417 GstRTSPMediaPrivate *priv = media->priv;
2419 if (priv->eos_shutdown) {
2420 GST_DEBUG ("sending EOS for shutdown");
2421 /* ref so that we don't disappear */
2422 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2423 /* we need to go to playing again for the EOS to propagate, normally in this
2424 * state, nothing is receiving data from us anymore so this is ok. */
2425 set_state (media, GST_STATE_PLAYING);
2427 finish_unprepare (media);
2433 * gst_rtsp_media_unprepare:
2434 * @media: a #GstRTSPMedia
2436 * Unprepare @media. After this call, the media should be prepared again before
2437 * it can be used again. If the media is set to be non-reusable, a new instance
2440 * Returns: %TRUE on success.
2443 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2445 GstRTSPMediaPrivate *priv;
2448 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2452 g_rec_mutex_lock (&priv->state_lock);
2453 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2454 goto was_unprepared;
2456 priv->prepare_count--;
2457 if (priv->prepare_count > 0)
2460 GST_INFO ("unprepare media %p", media);
2462 media_streams_set_blocked (media, FALSE);
2463 set_target_state (media, GST_STATE_NULL, FALSE);
2466 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2468 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2469 GstRTSPMediaClass *klass;
2471 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2472 if (klass->unprepare)
2473 success = klass->unprepare (media);
2475 finish_unprepare (media);
2477 g_rec_mutex_unlock (&priv->state_lock);
2483 g_rec_mutex_unlock (&priv->state_lock);
2484 GST_INFO ("media %p was already unprepared", media);
2489 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2490 g_rec_mutex_unlock (&priv->state_lock);
2495 /* should be called with state-lock */
2497 get_clock_unlocked (GstRTSPMedia * media)
2499 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2500 GST_DEBUG_OBJECT (media, "media was not prepared");
2503 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2507 * gst_rtsp_media_get_clock:
2508 * @media: a #GstRTSPMedia
2510 * Get the clock that is used by the pipeline in @media.
2512 * @media must be prepared before this method returns a valid clock object.
2514 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2517 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2520 GstRTSPMediaPrivate *priv;
2522 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2526 g_rec_mutex_lock (&priv->state_lock);
2527 clock = get_clock_unlocked (media);
2528 g_rec_mutex_unlock (&priv->state_lock);
2534 * gst_rtsp_media_get_base_time:
2535 * @media: a #GstRTSPMedia
2537 * Get the base_time that is used by the pipeline in @media.
2539 * @media must be prepared before this method returns a valid base_time.
2541 * Returns: the base_time used by @media.
2544 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2546 GstClockTime result;
2547 GstRTSPMediaPrivate *priv;
2549 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2553 g_rec_mutex_lock (&priv->state_lock);
2554 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2557 result = gst_element_get_base_time (media->priv->pipeline);
2558 g_rec_mutex_unlock (&priv->state_lock);
2565 g_rec_mutex_unlock (&priv->state_lock);
2566 GST_DEBUG_OBJECT (media, "media was not prepared");
2567 return GST_CLOCK_TIME_NONE;
2572 * gst_rtsp_media_get_time_provider:
2573 * @media: a #GstRTSPMedia
2574 * @address: an address or %NULL
2575 * @port: a port or 0
2577 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2578 * will listen on @address and @port for client time requests.
2580 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2582 GstNetTimeProvider *
2583 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2586 GstRTSPMediaPrivate *priv;
2587 GstNetTimeProvider *provider = NULL;
2589 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2593 g_rec_mutex_lock (&priv->state_lock);
2594 if (priv->time_provider) {
2595 if ((provider = priv->nettime) == NULL) {
2598 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2599 provider = gst_net_time_provider_new (clock, address, port);
2600 gst_object_unref (clock);
2602 priv->nettime = provider;
2606 g_rec_mutex_unlock (&priv->state_lock);
2609 gst_object_ref (provider);
2615 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
2617 return gst_rtsp_sdp_from_media (sdp, info, media);
2621 * gst_rtsp_media_setup_sdp:
2622 * @media: a #GstRTSPMedia
2623 * @sdp: (transfer none): a #GstSDPMessage
2624 * @info: (transfer none): a #GstSDPInfo
2626 * Add @media specific info to @sdp. @info is used to configure the connection
2627 * information in the SDP.
2629 * Returns: TRUE on success.
2632 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
2635 GstRTSPMediaPrivate *priv;
2636 GstRTSPMediaClass *klass;
2639 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2640 g_return_val_if_fail (sdp != NULL, FALSE);
2641 g_return_val_if_fail (info != NULL, FALSE);
2645 g_rec_mutex_lock (&priv->state_lock);
2647 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2649 if (!klass->setup_sdp)
2652 res = klass->setup_sdp (media, sdp, info);
2654 g_rec_mutex_unlock (&priv->state_lock);
2661 g_rec_mutex_unlock (&priv->state_lock);
2662 GST_ERROR ("no setup_sdp function");
2663 g_critical ("no setup_sdp vmethod function set");
2668 /* call with state_lock */
2670 default_suspend (GstRTSPMedia * media)
2672 GstRTSPMediaPrivate *priv = media->priv;
2673 GstStateChangeReturn ret;
2675 switch (priv->suspend_mode) {
2676 case GST_RTSP_SUSPEND_MODE_NONE:
2677 GST_DEBUG ("media %p no suspend", media);
2679 case GST_RTSP_SUSPEND_MODE_PAUSE:
2680 GST_DEBUG ("media %p suspend to PAUSED", media);
2681 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2682 if (ret == GST_STATE_CHANGE_FAILURE)
2685 case GST_RTSP_SUSPEND_MODE_RESET:
2686 GST_DEBUG ("media %p suspend to NULL", media);
2687 ret = set_target_state (media, GST_STATE_NULL, TRUE);
2688 if (ret == GST_STATE_CHANGE_FAILURE)
2695 /* let the streams do the state changes freely, if any */
2696 media_streams_set_blocked (media, FALSE);
2703 GST_WARNING ("failed changing pipeline's state for media %p", media);
2709 * gst_rtsp_media_suspend:
2710 * @media: a #GstRTSPMedia
2712 * Suspend @media. The state of the pipeline managed by @media is set to
2713 * GST_STATE_NULL but all streams are kept. @media can be prepared again
2714 * with gst_rtsp_media_unsuspend()
2716 * @media must be prepared with gst_rtsp_media_prepare();
2718 * Returns: %TRUE on success.
2721 gst_rtsp_media_suspend (GstRTSPMedia * media)
2723 GstRTSPMediaPrivate *priv = media->priv;
2724 GstRTSPMediaClass *klass;
2726 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2728 GST_FIXME ("suspend for dynamic pipelines needs fixing");
2730 g_rec_mutex_lock (&priv->state_lock);
2731 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2734 /* don't attempt to suspend when something is busy */
2735 if (priv->n_active > 0)
2738 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2739 if (klass->suspend) {
2740 if (!klass->suspend (media))
2741 goto suspend_failed;
2744 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
2746 g_rec_mutex_unlock (&priv->state_lock);
2753 g_rec_mutex_unlock (&priv->state_lock);
2754 GST_WARNING ("media %p was not prepared", media);
2759 g_rec_mutex_unlock (&priv->state_lock);
2760 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2761 GST_WARNING ("failed to suspend media %p", media);
2766 /* call with state_lock */
2768 default_unsuspend (GstRTSPMedia * media)
2770 GstRTSPMediaPrivate *priv = media->priv;
2772 switch (priv->suspend_mode) {
2773 case GST_RTSP_SUSPEND_MODE_NONE:
2774 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2776 case GST_RTSP_SUSPEND_MODE_PAUSE:
2777 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2779 case GST_RTSP_SUSPEND_MODE_RESET:
2781 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2782 if (!start_preroll (media))
2784 g_rec_mutex_unlock (&priv->state_lock);
2786 if (!wait_preroll (media))
2787 goto preroll_failed;
2789 g_rec_mutex_lock (&priv->state_lock);
2800 GST_WARNING ("failed to preroll pipeline");
2805 GST_WARNING ("failed to preroll pipeline");
2811 * gst_rtsp_media_unsuspend:
2812 * @media: a #GstRTSPMedia
2814 * Unsuspend @media if it was in a suspended state. This method does nothing
2815 * when the media was not in the suspended state.
2817 * Returns: %TRUE on success.
2820 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
2822 GstRTSPMediaPrivate *priv = media->priv;
2823 GstRTSPMediaClass *klass;
2825 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2827 g_rec_mutex_lock (&priv->state_lock);
2828 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2831 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2832 if (klass->unsuspend) {
2833 if (!klass->unsuspend (media))
2834 goto unsuspend_failed;
2838 g_rec_mutex_unlock (&priv->state_lock);
2845 g_rec_mutex_unlock (&priv->state_lock);
2846 GST_WARNING ("failed to unsuspend media %p", media);
2847 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2852 /* must be called with state-lock */
2854 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
2856 GstRTSPMediaPrivate *priv = media->priv;
2858 if (state == GST_STATE_NULL) {
2859 gst_rtsp_media_unprepare (media);
2861 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
2862 set_target_state (media, state, FALSE);
2863 /* when we are buffering, don't update the state yet, this will be done
2864 * when buffering finishes */
2865 if (priv->buffering) {
2866 GST_INFO ("Buffering busy, delay state change");
2868 if (state == GST_STATE_PLAYING)
2869 /* make sure pads are not blocking anymore when going to PLAYING */
2870 media_streams_set_blocked (media, FALSE);
2872 set_state (media, state);
2874 /* and suspend after pause */
2875 if (state == GST_STATE_PAUSED)
2876 gst_rtsp_media_suspend (media);
2882 * gst_rtsp_media_set_pipeline_state:
2883 * @media: a #GstRTSPMedia
2884 * @state: the target state of the pipeline
2886 * Set the state of the pipeline managed by @media to @state
2889 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
2891 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2893 g_rec_mutex_lock (&media->priv->state_lock);
2894 media_set_pipeline_state_locked (media, state);
2895 g_rec_mutex_unlock (&media->priv->state_lock);
2899 * gst_rtsp_media_set_state:
2900 * @media: a #GstRTSPMedia
2901 * @state: the target state of the media
2902 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
2903 * a #GPtrArray of #GstRTSPStreamTransport pointers
2905 * Set the state of @media to @state and for the transports in @transports.
2907 * @media must be prepared with gst_rtsp_media_prepare();
2909 * Returns: %TRUE on success.
2912 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2913 GPtrArray * transports)
2915 GstRTSPMediaPrivate *priv;
2917 gboolean activate, deactivate, do_state;
2920 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2921 g_return_val_if_fail (transports != NULL, FALSE);
2925 g_rec_mutex_lock (&priv->state_lock);
2926 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
2928 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2929 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2932 /* NULL and READY are the same */
2933 if (state == GST_STATE_READY)
2934 state = GST_STATE_NULL;
2936 activate = deactivate = FALSE;
2938 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
2942 case GST_STATE_NULL:
2943 case GST_STATE_PAUSED:
2944 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
2945 if (priv->target_state == GST_STATE_PLAYING)
2948 case GST_STATE_PLAYING:
2949 /* we're going to PLAYING, activate */
2955 old_active = priv->n_active;
2957 for (i = 0; i < transports->len; i++) {
2958 GstRTSPStreamTransport *trans;
2960 /* we need a non-NULL entry in the array */
2961 trans = g_ptr_array_index (transports, i);
2966 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2968 } else if (deactivate) {
2969 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2974 /* we just activated the first media, do the playing state change */
2975 if (old_active == 0 && activate)
2977 /* if we have no more active media, do the downward state changes */
2978 else if (priv->n_active == 0)
2983 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2986 if (priv->target_state != state) {
2988 media_set_pipeline_state_locked (media, state);
2990 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2994 /* remember where we are */
2995 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2996 old_active != priv->n_active))
2997 collect_media_stats (media);
2999 g_rec_mutex_unlock (&priv->state_lock);
3006 GST_WARNING ("media %p was not prepared", media);
3007 g_rec_mutex_unlock (&priv->state_lock);
3012 GST_WARNING ("media %p in error status while changing to state %d",
3014 if (state == GST_STATE_NULL) {
3015 for (i = 0; i < transports->len; i++) {
3016 GstRTSPStreamTransport *trans;
3018 /* we need a non-NULL entry in the array */
3019 trans = g_ptr_array_index (transports, i);
3023 gst_rtsp_stream_transport_set_active (trans, FALSE);
3027 g_rec_mutex_unlock (&priv->state_lock);