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_generic, 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_generic, 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_generic, 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_generic, 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: (nullable) (transfer none): the #GstRTSPStream at index
1327 * @idx or %NULL when a stream with 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: (nullable) (transfer none): the #GstRTSPStream with
1357 * control uri @control or %NULL when a stream with that control did
1361 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1363 GstRTSPMediaPrivate *priv;
1367 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1368 g_return_val_if_fail (control != NULL, NULL);
1374 g_mutex_lock (&priv->lock);
1375 for (i = 0; i < priv->streams->len; i++) {
1376 GstRTSPStream *test;
1378 test = g_ptr_array_index (priv->streams, i);
1379 if (gst_rtsp_stream_has_control (test, control)) {
1384 g_mutex_unlock (&priv->lock);
1389 /* called with state-lock */
1391 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1392 GstRTSPRangeUnit unit)
1394 return gst_rtsp_range_convert_units (range, unit);
1398 * gst_rtsp_media_get_range_string:
1399 * @media: a #GstRTSPMedia
1400 * @play: for the PLAY request
1401 * @unit: the unit to use for the string
1403 * Get the current range as a string. @media must be prepared with
1404 * gst_rtsp_media_prepare ().
1406 * Returns: (transfer full): The range as a string, g_free() after usage.
1409 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1410 GstRTSPRangeUnit unit)
1412 GstRTSPMediaClass *klass;
1413 GstRTSPMediaPrivate *priv;
1415 GstRTSPTimeRange range;
1417 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1418 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1419 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1423 g_rec_mutex_lock (&priv->state_lock);
1424 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1425 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1428 g_mutex_lock (&priv->lock);
1430 /* Update the range value with current position/duration */
1431 collect_media_stats (media);
1434 range = priv->range;
1436 if (!play && priv->n_active > 0) {
1437 range.min.type = GST_RTSP_TIME_NOW;
1438 range.min.seconds = -1;
1440 g_mutex_unlock (&priv->lock);
1441 g_rec_mutex_unlock (&priv->state_lock);
1443 if (!klass->convert_range (media, &range, unit))
1444 goto conversion_failed;
1446 result = gst_rtsp_range_to_string (&range);
1453 GST_WARNING ("media %p was not prepared", media);
1454 g_rec_mutex_unlock (&priv->state_lock);
1459 GST_WARNING ("range conversion to unit %d failed", unit);
1465 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1467 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1471 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1473 GstRTSPMediaPrivate *priv = media->priv;
1475 GST_DEBUG ("media %p set blocked %d", media, blocked);
1476 priv->blocked = blocked;
1477 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1481 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1483 GstRTSPMediaPrivate *priv = media->priv;
1485 g_mutex_lock (&priv->lock);
1486 priv->status = status;
1487 GST_DEBUG ("setting new status to %d", status);
1488 g_cond_broadcast (&priv->cond);
1489 g_mutex_unlock (&priv->lock);
1493 * gst_rtsp_media_get_status:
1494 * @media: a #GstRTSPMedia
1496 * Get the status of @media. When @media is busy preparing, this function waits
1497 * until @media is prepared or in error.
1499 * Returns: the status of @media.
1502 gst_rtsp_media_get_status (GstRTSPMedia * media)
1504 GstRTSPMediaPrivate *priv = media->priv;
1505 GstRTSPMediaStatus result;
1508 g_mutex_lock (&priv->lock);
1509 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1510 /* while we are preparing, wait */
1511 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1512 GST_DEBUG ("waiting for status change");
1513 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1514 GST_DEBUG ("timeout, assuming error status");
1515 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1518 /* could be success or error */
1519 result = priv->status;
1520 GST_DEBUG ("got status %d", result);
1521 g_mutex_unlock (&priv->lock);
1527 * gst_rtsp_media_seek:
1528 * @media: a #GstRTSPMedia
1529 * @range: (transfer none): a #GstRTSPTimeRange
1531 * Seek the pipeline of @media to @range. @media must be prepared with
1532 * gst_rtsp_media_prepare().
1534 * Returns: %TRUE on success.
1537 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1539 GstRTSPMediaClass *klass;
1540 GstRTSPMediaPrivate *priv;
1542 GstClockTime start, stop;
1543 GstSeekType start_type, stop_type;
1546 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1548 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1549 g_return_val_if_fail (range != NULL, FALSE);
1550 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1554 g_rec_mutex_lock (&priv->state_lock);
1555 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1558 /* Update the seekable state of the pipeline in case it changed */
1559 query = gst_query_new_seeking (GST_FORMAT_TIME);
1560 if (gst_element_query (priv->pipeline, query)) {
1565 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1566 priv->seekable = seekable;
1568 gst_query_unref (query);
1570 if (!priv->seekable)
1573 start_type = stop_type = GST_SEEK_TYPE_NONE;
1575 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1577 gst_rtsp_range_get_times (range, &start, &stop);
1579 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1580 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1581 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1582 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1584 if (start != GST_CLOCK_TIME_NONE)
1585 start_type = GST_SEEK_TYPE_SET;
1587 if (priv->range_stop == stop)
1588 stop = GST_CLOCK_TIME_NONE;
1589 else if (stop != GST_CLOCK_TIME_NONE)
1590 stop_type = GST_SEEK_TYPE_SET;
1592 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1595 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1596 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1598 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
1600 media_streams_set_blocked (media, TRUE);
1602 /* depends on the current playing state of the pipeline. We might need to
1603 * queue this until we get EOS. */
1604 flags = GST_SEEK_FLAG_FLUSH;
1606 /* if range start was not supplied we must continue from current position.
1607 * but since we're doing a flushing seek, let us query the current position
1608 * so we end up at exactly the same position after the seek. */
1609 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
1611 gboolean ret = FALSE;
1613 if (klass->query_position)
1614 ret = klass->query_position (media, &position);
1617 GST_WARNING ("position query failed");
1619 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
1620 GST_TIME_ARGS (position));
1622 start_type = GST_SEEK_TYPE_SET;
1623 flags |= GST_SEEK_FLAG_ACCURATE;
1626 /* only set keyframe flag when modifying start */
1627 if (start_type != GST_SEEK_TYPE_NONE)
1628 flags |= GST_SEEK_FLAG_KEY_UNIT;
1631 /* FIXME, we only do forwards playback, no trick modes yet */
1632 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1633 flags, start_type, start, stop_type, stop);
1635 /* and block for the seek to complete */
1636 GST_INFO ("done seeking %d", res);
1637 g_rec_mutex_unlock (&priv->state_lock);
1639 /* wait until pipeline is prerolled again, this will also collect stats */
1640 if (!wait_preroll (media))
1641 goto preroll_failed;
1643 g_rec_mutex_lock (&priv->state_lock);
1644 GST_INFO ("prerolled again");
1646 GST_INFO ("no seek needed");
1649 g_rec_mutex_unlock (&priv->state_lock);
1656 g_rec_mutex_unlock (&priv->state_lock);
1657 GST_INFO ("media %p is not prepared", media);
1662 g_rec_mutex_unlock (&priv->state_lock);
1663 GST_INFO ("pipeline is not seekable");
1668 g_rec_mutex_unlock (&priv->state_lock);
1669 GST_WARNING ("conversion to npt not supported");
1674 GST_WARNING ("failed to preroll after seek");
1680 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
1682 *blocked &= gst_rtsp_stream_is_blocking (stream);
1686 media_streams_blocking (GstRTSPMedia * media)
1688 gboolean blocking = TRUE;
1690 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
1696 static GstStateChangeReturn
1697 set_state (GstRTSPMedia * media, GstState state)
1699 GstRTSPMediaPrivate *priv = media->priv;
1700 GstStateChangeReturn ret;
1702 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
1704 ret = gst_element_set_state (priv->pipeline, state);
1709 static GstStateChangeReturn
1710 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
1712 GstRTSPMediaPrivate *priv = media->priv;
1713 GstStateChangeReturn ret;
1715 GST_INFO ("set target state to %s for media %p",
1716 gst_element_state_get_name (state), media);
1717 priv->target_state = state;
1719 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
1720 priv->target_state, NULL);
1723 ret = set_state (media, state);
1725 ret = GST_STATE_CHANGE_SUCCESS;
1730 /* called with state-lock */
1732 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1734 GstRTSPMediaPrivate *priv = media->priv;
1735 GstMessageType type;
1737 type = GST_MESSAGE_TYPE (message);
1740 case GST_MESSAGE_STATE_CHANGED:
1742 case GST_MESSAGE_BUFFERING:
1746 gst_message_parse_buffering (message, &percent);
1748 /* no state management needed for live pipelines */
1752 if (percent == 100) {
1753 /* a 100% message means buffering is done */
1754 priv->buffering = FALSE;
1755 /* if the desired state is playing, go back */
1756 if (priv->target_state == GST_STATE_PLAYING) {
1757 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1758 set_state (media, GST_STATE_PLAYING);
1760 GST_INFO ("Buffering done");
1763 /* buffering busy */
1764 if (priv->buffering == FALSE) {
1765 if (priv->target_state == GST_STATE_PLAYING) {
1766 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1767 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1768 set_state (media, GST_STATE_PAUSED);
1770 GST_INFO ("Buffering ...");
1773 priv->buffering = TRUE;
1777 case GST_MESSAGE_LATENCY:
1779 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1782 case GST_MESSAGE_ERROR:
1787 gst_message_parse_error (message, &gerror, &debug);
1788 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1789 g_error_free (gerror);
1792 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1795 case GST_MESSAGE_WARNING:
1800 gst_message_parse_warning (message, &gerror, &debug);
1801 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1802 g_error_free (gerror);
1806 case GST_MESSAGE_ELEMENT:
1808 const GstStructure *s;
1810 s = gst_message_get_structure (message);
1811 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
1812 GST_DEBUG ("media received blocking message");
1813 if (priv->blocked && media_streams_blocking (media)) {
1814 GST_DEBUG ("media is blocking");
1815 collect_media_stats (media);
1817 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1818 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1823 case GST_MESSAGE_STREAM_STATUS:
1825 case GST_MESSAGE_ASYNC_DONE:
1827 /* when we are dynamically adding pads, the addition of the udpsrc will
1828 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1829 * wait for the final ASYNC_DONE after everything prerolled */
1830 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1832 GST_INFO ("%p: got ASYNC_DONE", media);
1833 collect_media_stats (media);
1835 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1836 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1839 case GST_MESSAGE_EOS:
1840 GST_INFO ("%p: got EOS", media);
1842 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1843 GST_DEBUG ("shutting down after EOS");
1844 finish_unprepare (media);
1848 GST_INFO ("%p: got message type %d (%s)", media, type,
1849 gst_message_type_get_name (type));
1856 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1858 GstRTSPMediaPrivate *priv = media->priv;
1859 GstRTSPMediaClass *klass;
1862 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1864 g_rec_mutex_lock (&priv->state_lock);
1865 if (klass->handle_message)
1866 ret = klass->handle_message (media, message);
1869 g_rec_mutex_unlock (&priv->state_lock);
1875 watch_destroyed (GstRTSPMedia * media)
1877 GST_DEBUG_OBJECT (media, "source destroyed");
1878 g_object_unref (media);
1882 find_payload_element (GstElement * payloader)
1884 GstElement *pay = NULL;
1886 if (GST_IS_BIN (payloader)) {
1888 GValue item = { 0 };
1890 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
1891 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
1892 GstElement *element = (GstElement *) g_value_get_object (&item);
1893 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
1897 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
1901 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
1902 pay = gst_object_ref (element);
1903 g_value_unset (&item);
1906 g_value_unset (&item);
1908 gst_iterator_free (iter);
1910 pay = g_object_ref (payloader);
1916 /* called from streaming threads */
1918 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1920 GstRTSPMediaPrivate *priv = media->priv;
1921 GstRTSPStream *stream;
1924 /* find the real payload element */
1925 pay = find_payload_element (element);
1926 stream = gst_rtsp_media_create_stream (media, pay, pad);
1927 gst_object_unref (pay);
1929 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1931 g_rec_mutex_lock (&priv->state_lock);
1932 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
1935 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1937 /* we will be adding elements below that will cause ASYNC_DONE to be
1938 * posted in the bus. We want to ignore those messages until the
1939 * pipeline really prerolled. */
1940 priv->adding = TRUE;
1942 /* join the element in the PAUSED state because this callback is
1943 * called from the streaming thread and it is PAUSED */
1944 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1945 priv->rtpbin, GST_STATE_PAUSED)) {
1946 GST_WARNING ("failed to join bin element");
1949 priv->adding = FALSE;
1950 g_rec_mutex_unlock (&priv->state_lock);
1957 gst_rtsp_media_remove_stream (media, stream);
1958 g_rec_mutex_unlock (&priv->state_lock);
1959 GST_INFO ("ignore pad because we are not preparing");
1965 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1967 GstRTSPMediaPrivate *priv = media->priv;
1968 GstRTSPStream *stream;
1970 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1974 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1976 g_rec_mutex_lock (&priv->state_lock);
1977 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1978 g_rec_mutex_unlock (&priv->state_lock);
1980 gst_rtsp_media_remove_stream (media, stream);
1984 remove_fakesink (GstRTSPMediaPrivate * priv)
1986 GstElement *fakesink;
1988 g_mutex_lock (&priv->lock);
1989 if ((fakesink = priv->fakesink))
1990 gst_object_ref (fakesink);
1991 priv->fakesink = NULL;
1992 g_mutex_unlock (&priv->lock);
1995 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1996 gst_element_set_state (fakesink, GST_STATE_NULL);
1997 gst_object_unref (fakesink);
1998 GST_INFO ("removed fakesink");
2003 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2005 GstRTSPMediaPrivate *priv = media->priv;
2007 GST_INFO ("no more pads");
2008 remove_fakesink (priv);
2011 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2013 struct _DynPaySignalHandlers
2015 gulong pad_added_handler;
2016 gulong pad_removed_handler;
2017 gulong no_more_pads_handler;
2021 start_preroll (GstRTSPMedia * media)
2023 GstRTSPMediaPrivate *priv = media->priv;
2024 GstStateChangeReturn ret;
2026 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2027 /* first go to PAUSED */
2028 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2031 case GST_STATE_CHANGE_SUCCESS:
2032 GST_INFO ("SUCCESS state change for media %p", media);
2033 priv->seekable = TRUE;
2035 case GST_STATE_CHANGE_ASYNC:
2036 GST_INFO ("ASYNC state change for media %p", media);
2037 priv->seekable = TRUE;
2039 case GST_STATE_CHANGE_NO_PREROLL:
2040 /* we need to go to PLAYING */
2041 GST_INFO ("NO_PREROLL state change: live media %p", media);
2042 /* FIXME we disable seeking for live streams for now. We should perform a
2043 * seeking query in preroll instead */
2044 priv->seekable = FALSE;
2045 priv->is_live = TRUE;
2046 /* start blocked to make sure nothing goes to the sink */
2047 media_streams_set_blocked (media, TRUE);
2048 ret = set_state (media, GST_STATE_PLAYING);
2049 if (ret == GST_STATE_CHANGE_FAILURE)
2052 case GST_STATE_CHANGE_FAILURE:
2060 GST_WARNING ("failed to preroll pipeline");
2066 wait_preroll (GstRTSPMedia * media)
2068 GstRTSPMediaStatus status;
2070 GST_DEBUG ("wait to preroll pipeline");
2072 /* wait until pipeline is prerolled */
2073 status = gst_rtsp_media_get_status (media);
2074 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2075 goto preroll_failed;
2081 GST_WARNING ("failed to preroll pipeline");
2087 start_prepare (GstRTSPMedia * media)
2089 GstRTSPMediaPrivate *priv = media->priv;
2093 /* link streams we already have, other streams might appear when we have
2094 * dynamic elements */
2095 for (i = 0; i < priv->streams->len; i++) {
2096 GstRTSPStream *stream;
2098 stream = g_ptr_array_index (priv->streams, i);
2100 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2101 priv->rtpbin, GST_STATE_NULL)) {
2102 goto join_bin_failed;
2106 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2107 GstElement *elem = walk->data;
2108 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2110 GST_INFO ("adding callbacks for dynamic element %p", elem);
2112 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2113 (GCallback) pad_added_cb, media);
2114 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2115 (GCallback) pad_removed_cb, media);
2116 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2117 (GCallback) no_more_pads_cb, media);
2119 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2121 /* we add a fakesink here in order to make the state change async. We remove
2122 * the fakesink again in the no-more-pads callback. */
2123 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2124 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2127 if (!start_preroll (media))
2128 goto preroll_failed;
2134 GST_WARNING ("failed to join bin element");
2135 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2140 GST_WARNING ("failed to preroll pipeline");
2141 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2147 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2149 GstRTSPMediaPrivate *priv;
2150 GstRTSPMediaClass *klass;
2152 GMainContext *context;
2157 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2159 if (!klass->create_rtpbin)
2160 goto no_create_rtpbin;
2162 priv->rtpbin = klass->create_rtpbin (media);
2163 if (priv->rtpbin != NULL) {
2164 gboolean success = TRUE;
2166 if (klass->setup_rtpbin)
2167 success = klass->setup_rtpbin (media, priv->rtpbin);
2169 if (success == FALSE) {
2170 gst_object_unref (priv->rtpbin);
2171 priv->rtpbin = NULL;
2174 if (priv->rtpbin == NULL)
2177 priv->thread = thread;
2178 context = (thread != NULL) ? (thread->context) : NULL;
2180 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2182 /* add the pipeline bus to our custom mainloop */
2183 priv->source = gst_bus_create_watch (bus);
2184 gst_object_unref (bus);
2186 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2187 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2189 priv->id = g_source_attach (priv->source, context);
2191 /* add stuff to the bin */
2192 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2194 /* do remainder in context */
2195 source = g_idle_source_new ();
2196 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
2197 g_source_attach (source, context);
2198 g_source_unref (source);
2205 GST_ERROR ("no create_rtpbin function");
2206 g_critical ("no create_rtpbin vmethod function set");
2211 GST_WARNING ("no rtpbin element");
2212 g_warning ("failed to create element 'rtpbin', check your installation");
2218 * gst_rtsp_media_prepare:
2219 * @media: a #GstRTSPMedia
2220 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2221 * bus handler or %NULL
2223 * Prepare @media for streaming. This function will create the objects
2224 * to manage the streaming. A pipeline must have been set on @media with
2225 * gst_rtsp_media_take_pipeline().
2227 * It will preroll the pipeline and collect vital information about the streams
2228 * such as the duration.
2230 * Returns: %TRUE on success.
2233 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2235 GstRTSPMediaPrivate *priv;
2236 GstRTSPMediaClass *klass;
2238 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2242 g_rec_mutex_lock (&priv->state_lock);
2243 priv->prepare_count++;
2245 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2246 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2249 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2252 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2253 goto not_unprepared;
2255 if (!priv->reusable && priv->reused)
2258 GST_INFO ("preparing media %p", media);
2260 /* reset some variables */
2261 priv->is_live = FALSE;
2262 priv->seekable = FALSE;
2263 priv->buffering = FALSE;
2265 /* we're preparing now */
2266 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2268 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2269 if (klass->prepare) {
2270 if (!klass->prepare (media, thread))
2271 goto prepare_failed;
2275 g_rec_mutex_unlock (&priv->state_lock);
2277 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2278 * able to do this async so that we don't block the server thread. */
2279 if (!wait_preroll (media))
2280 goto preroll_failed;
2282 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2284 GST_INFO ("object %p is prerolled", media);
2291 /* we are not going to use the giving thread, so stop it. */
2293 gst_rtsp_thread_stop (thread);
2298 GST_LOG ("media %p was prepared", media);
2299 /* we are not going to use the giving thread, so stop it. */
2301 gst_rtsp_thread_stop (thread);
2302 g_rec_mutex_unlock (&priv->state_lock);
2308 /* we are not going to use the giving thread, so stop it. */
2310 gst_rtsp_thread_stop (thread);
2311 GST_WARNING ("media %p was not unprepared", media);
2312 priv->prepare_count--;
2313 g_rec_mutex_unlock (&priv->state_lock);
2318 /* we are not going to use the giving thread, so stop it. */
2320 gst_rtsp_thread_stop (thread);
2321 priv->prepare_count--;
2322 g_rec_mutex_unlock (&priv->state_lock);
2323 GST_WARNING ("can not reuse media %p", media);
2328 /* we are not going to use the giving thread, so stop it. */
2330 gst_rtsp_thread_stop (thread);
2331 priv->prepare_count--;
2332 g_rec_mutex_unlock (&priv->state_lock);
2333 GST_ERROR ("failed to prepare media");
2338 GST_WARNING ("failed to preroll pipeline");
2339 gst_rtsp_media_unprepare (media);
2344 /* must be called with state-lock */
2346 finish_unprepare (GstRTSPMedia * media)
2348 GstRTSPMediaPrivate *priv = media->priv;
2352 GST_DEBUG ("shutting down");
2354 /* release the lock on shutdown, otherwise pad_added_cb might try to
2355 * acquire the lock and then we deadlock */
2356 g_rec_mutex_unlock (&priv->state_lock);
2357 set_state (media, GST_STATE_NULL);
2358 g_rec_mutex_lock (&priv->state_lock);
2359 remove_fakesink (priv);
2361 for (i = 0; i < priv->streams->len; i++) {
2362 GstRTSPStream *stream;
2364 GST_INFO ("Removing elements of stream %d from pipeline", i);
2366 stream = g_ptr_array_index (priv->streams, i);
2368 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2371 /* remove the pad signal handlers */
2372 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2373 GstElement *elem = walk->data;
2374 DynPaySignalHandlers *handlers;
2377 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2378 g_assert (handlers != NULL);
2380 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2381 g_signal_handler_disconnect (G_OBJECT (elem),
2382 handlers->pad_removed_handler);
2383 g_signal_handler_disconnect (G_OBJECT (elem),
2384 handlers->no_more_pads_handler);
2386 g_slice_free (DynPaySignalHandlers, handlers);
2389 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2390 priv->rtpbin = NULL;
2393 gst_object_unref (priv->nettime);
2394 priv->nettime = NULL;
2396 priv->reused = TRUE;
2397 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2399 /* when the media is not reusable, this will effectively unref the media and
2401 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2403 /* the source has the last ref to the media */
2405 GST_DEBUG ("destroy source");
2406 g_source_destroy (priv->source);
2407 g_source_unref (priv->source);
2410 GST_DEBUG ("stop thread");
2411 gst_rtsp_thread_stop (priv->thread);
2415 /* called with state-lock */
2417 default_unprepare (GstRTSPMedia * media)
2419 GstRTSPMediaPrivate *priv = media->priv;
2421 if (priv->eos_shutdown) {
2422 GST_DEBUG ("sending EOS for shutdown");
2423 /* ref so that we don't disappear */
2424 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2425 /* we need to go to playing again for the EOS to propagate, normally in this
2426 * state, nothing is receiving data from us anymore so this is ok. */
2427 set_state (media, GST_STATE_PLAYING);
2429 finish_unprepare (media);
2435 * gst_rtsp_media_unprepare:
2436 * @media: a #GstRTSPMedia
2438 * Unprepare @media. After this call, the media should be prepared again before
2439 * it can be used again. If the media is set to be non-reusable, a new instance
2442 * Returns: %TRUE on success.
2445 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2447 GstRTSPMediaPrivate *priv;
2450 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2454 g_rec_mutex_lock (&priv->state_lock);
2455 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2456 goto was_unprepared;
2458 priv->prepare_count--;
2459 if (priv->prepare_count > 0)
2462 GST_INFO ("unprepare media %p", media);
2464 media_streams_set_blocked (media, FALSE);
2465 set_target_state (media, GST_STATE_NULL, FALSE);
2468 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2470 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2471 GstRTSPMediaClass *klass;
2473 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2474 if (klass->unprepare)
2475 success = klass->unprepare (media);
2477 finish_unprepare (media);
2479 g_rec_mutex_unlock (&priv->state_lock);
2485 g_rec_mutex_unlock (&priv->state_lock);
2486 GST_INFO ("media %p was already unprepared", media);
2491 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2492 g_rec_mutex_unlock (&priv->state_lock);
2497 /* should be called with state-lock */
2499 get_clock_unlocked (GstRTSPMedia * media)
2501 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2502 GST_DEBUG_OBJECT (media, "media was not prepared");
2505 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2509 * gst_rtsp_media_get_clock:
2510 * @media: a #GstRTSPMedia
2512 * Get the clock that is used by the pipeline in @media.
2514 * @media must be prepared before this method returns a valid clock object.
2516 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2519 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2522 GstRTSPMediaPrivate *priv;
2524 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2528 g_rec_mutex_lock (&priv->state_lock);
2529 clock = get_clock_unlocked (media);
2530 g_rec_mutex_unlock (&priv->state_lock);
2536 * gst_rtsp_media_get_base_time:
2537 * @media: a #GstRTSPMedia
2539 * Get the base_time that is used by the pipeline in @media.
2541 * @media must be prepared before this method returns a valid base_time.
2543 * Returns: the base_time used by @media.
2546 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2548 GstClockTime result;
2549 GstRTSPMediaPrivate *priv;
2551 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2555 g_rec_mutex_lock (&priv->state_lock);
2556 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2559 result = gst_element_get_base_time (media->priv->pipeline);
2560 g_rec_mutex_unlock (&priv->state_lock);
2567 g_rec_mutex_unlock (&priv->state_lock);
2568 GST_DEBUG_OBJECT (media, "media was not prepared");
2569 return GST_CLOCK_TIME_NONE;
2574 * gst_rtsp_media_get_time_provider:
2575 * @media: a #GstRTSPMedia
2576 * @address: (allow-none): an address or %NULL
2577 * @port: a port or 0
2579 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2580 * will listen on @address and @port for client time requests.
2582 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2584 GstNetTimeProvider *
2585 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2588 GstRTSPMediaPrivate *priv;
2589 GstNetTimeProvider *provider = NULL;
2591 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2595 g_rec_mutex_lock (&priv->state_lock);
2596 if (priv->time_provider) {
2597 if ((provider = priv->nettime) == NULL) {
2600 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2601 provider = gst_net_time_provider_new (clock, address, port);
2602 gst_object_unref (clock);
2604 priv->nettime = provider;
2608 g_rec_mutex_unlock (&priv->state_lock);
2611 gst_object_ref (provider);
2617 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
2619 return gst_rtsp_sdp_from_media (sdp, info, media);
2623 * gst_rtsp_media_setup_sdp:
2624 * @media: a #GstRTSPMedia
2625 * @sdp: (transfer none): a #GstSDPMessage
2626 * @info: (transfer none): a #GstSDPInfo
2628 * Add @media specific info to @sdp. @info is used to configure the connection
2629 * information in the SDP.
2631 * Returns: TRUE on success.
2634 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
2637 GstRTSPMediaPrivate *priv;
2638 GstRTSPMediaClass *klass;
2641 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2642 g_return_val_if_fail (sdp != NULL, FALSE);
2643 g_return_val_if_fail (info != NULL, FALSE);
2647 g_rec_mutex_lock (&priv->state_lock);
2649 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2651 if (!klass->setup_sdp)
2654 res = klass->setup_sdp (media, sdp, info);
2656 g_rec_mutex_unlock (&priv->state_lock);
2663 g_rec_mutex_unlock (&priv->state_lock);
2664 GST_ERROR ("no setup_sdp function");
2665 g_critical ("no setup_sdp vmethod function set");
2670 /* call with state_lock */
2672 default_suspend (GstRTSPMedia * media)
2674 GstRTSPMediaPrivate *priv = media->priv;
2675 GstStateChangeReturn ret;
2677 switch (priv->suspend_mode) {
2678 case GST_RTSP_SUSPEND_MODE_NONE:
2679 GST_DEBUG ("media %p no suspend", media);
2681 case GST_RTSP_SUSPEND_MODE_PAUSE:
2682 GST_DEBUG ("media %p suspend to PAUSED", media);
2683 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2684 if (ret == GST_STATE_CHANGE_FAILURE)
2687 case GST_RTSP_SUSPEND_MODE_RESET:
2688 GST_DEBUG ("media %p suspend to NULL", media);
2689 ret = set_target_state (media, GST_STATE_NULL, TRUE);
2690 if (ret == GST_STATE_CHANGE_FAILURE)
2697 /* let the streams do the state changes freely, if any */
2698 media_streams_set_blocked (media, FALSE);
2705 GST_WARNING ("failed changing pipeline's state for media %p", media);
2711 * gst_rtsp_media_suspend:
2712 * @media: a #GstRTSPMedia
2714 * Suspend @media. The state of the pipeline managed by @media is set to
2715 * GST_STATE_NULL but all streams are kept. @media can be prepared again
2716 * with gst_rtsp_media_unsuspend()
2718 * @media must be prepared with gst_rtsp_media_prepare();
2720 * Returns: %TRUE on success.
2723 gst_rtsp_media_suspend (GstRTSPMedia * media)
2725 GstRTSPMediaPrivate *priv = media->priv;
2726 GstRTSPMediaClass *klass;
2728 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2730 GST_FIXME ("suspend for dynamic pipelines needs fixing");
2732 g_rec_mutex_lock (&priv->state_lock);
2733 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2736 /* don't attempt to suspend when something is busy */
2737 if (priv->n_active > 0)
2740 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2741 if (klass->suspend) {
2742 if (!klass->suspend (media))
2743 goto suspend_failed;
2746 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
2748 g_rec_mutex_unlock (&priv->state_lock);
2755 g_rec_mutex_unlock (&priv->state_lock);
2756 GST_WARNING ("media %p was not prepared", media);
2761 g_rec_mutex_unlock (&priv->state_lock);
2762 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2763 GST_WARNING ("failed to suspend media %p", media);
2768 /* call with state_lock */
2770 default_unsuspend (GstRTSPMedia * media)
2772 GstRTSPMediaPrivate *priv = media->priv;
2774 switch (priv->suspend_mode) {
2775 case GST_RTSP_SUSPEND_MODE_NONE:
2776 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2778 case GST_RTSP_SUSPEND_MODE_PAUSE:
2779 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2781 case GST_RTSP_SUSPEND_MODE_RESET:
2783 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2784 if (!start_preroll (media))
2786 g_rec_mutex_unlock (&priv->state_lock);
2788 if (!wait_preroll (media))
2789 goto preroll_failed;
2791 g_rec_mutex_lock (&priv->state_lock);
2802 GST_WARNING ("failed to preroll pipeline");
2807 GST_WARNING ("failed to preroll pipeline");
2813 * gst_rtsp_media_unsuspend:
2814 * @media: a #GstRTSPMedia
2816 * Unsuspend @media if it was in a suspended state. This method does nothing
2817 * when the media was not in the suspended state.
2819 * Returns: %TRUE on success.
2822 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
2824 GstRTSPMediaPrivate *priv = media->priv;
2825 GstRTSPMediaClass *klass;
2827 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2829 g_rec_mutex_lock (&priv->state_lock);
2830 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2833 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2834 if (klass->unsuspend) {
2835 if (!klass->unsuspend (media))
2836 goto unsuspend_failed;
2840 g_rec_mutex_unlock (&priv->state_lock);
2847 g_rec_mutex_unlock (&priv->state_lock);
2848 GST_WARNING ("failed to unsuspend media %p", media);
2849 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2854 /* must be called with state-lock */
2856 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
2858 GstRTSPMediaPrivate *priv = media->priv;
2860 if (state == GST_STATE_NULL) {
2861 gst_rtsp_media_unprepare (media);
2863 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
2864 set_target_state (media, state, FALSE);
2865 /* when we are buffering, don't update the state yet, this will be done
2866 * when buffering finishes */
2867 if (priv->buffering) {
2868 GST_INFO ("Buffering busy, delay state change");
2870 if (state == GST_STATE_PLAYING)
2871 /* make sure pads are not blocking anymore when going to PLAYING */
2872 media_streams_set_blocked (media, FALSE);
2874 set_state (media, state);
2876 /* and suspend after pause */
2877 if (state == GST_STATE_PAUSED)
2878 gst_rtsp_media_suspend (media);
2884 * gst_rtsp_media_set_pipeline_state:
2885 * @media: a #GstRTSPMedia
2886 * @state: the target state of the pipeline
2888 * Set the state of the pipeline managed by @media to @state
2891 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
2893 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2895 g_rec_mutex_lock (&media->priv->state_lock);
2896 media_set_pipeline_state_locked (media, state);
2897 g_rec_mutex_unlock (&media->priv->state_lock);
2901 * gst_rtsp_media_set_state:
2902 * @media: a #GstRTSPMedia
2903 * @state: the target state of the media
2904 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
2905 * a #GPtrArray of #GstRTSPStreamTransport pointers
2907 * Set the state of @media to @state and for the transports in @transports.
2909 * @media must be prepared with gst_rtsp_media_prepare();
2911 * Returns: %TRUE on success.
2914 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2915 GPtrArray * transports)
2917 GstRTSPMediaPrivate *priv;
2919 gboolean activate, deactivate, do_state;
2922 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2923 g_return_val_if_fail (transports != NULL, FALSE);
2927 g_rec_mutex_lock (&priv->state_lock);
2928 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
2930 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2931 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2934 /* NULL and READY are the same */
2935 if (state == GST_STATE_READY)
2936 state = GST_STATE_NULL;
2938 activate = deactivate = FALSE;
2940 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
2944 case GST_STATE_NULL:
2945 case GST_STATE_PAUSED:
2946 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
2947 if (priv->target_state == GST_STATE_PLAYING)
2950 case GST_STATE_PLAYING:
2951 /* we're going to PLAYING, activate */
2957 old_active = priv->n_active;
2959 for (i = 0; i < transports->len; i++) {
2960 GstRTSPStreamTransport *trans;
2962 /* we need a non-NULL entry in the array */
2963 trans = g_ptr_array_index (transports, i);
2968 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2970 } else if (deactivate) {
2971 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2976 /* we just activated the first media, do the playing state change */
2977 if (old_active == 0 && activate)
2979 /* if we have no more active media, do the downward state changes */
2980 else if (priv->n_active == 0)
2985 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2988 if (priv->target_state != state) {
2990 media_set_pipeline_state_locked (media, state);
2992 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2996 /* remember where we are */
2997 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2998 old_active != priv->n_active))
2999 collect_media_stats (media);
3001 g_rec_mutex_unlock (&priv->state_lock);
3008 GST_WARNING ("media %p was not prepared", media);
3009 g_rec_mutex_unlock (&priv->state_lock);
3014 GST_WARNING ("media %p in error status while changing to state %d",
3016 if (state == GST_STATE_NULL) {
3017 for (i = 0; i < transports->len; i++) {
3018 GstRTSPStreamTransport *trans;
3020 /* we need a non-NULL entry in the array */
3021 trans = g_ptr_array_index (transports, i);
3025 gst_rtsp_stream_transport_set_active (trans, FALSE);
3029 g_rec_mutex_unlock (&priv->state_lock);