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,
164 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
165 #define GST_CAT_DEFAULT rtsp_media_debug
167 static void gst_rtsp_media_get_property (GObject * object, guint propid,
168 GValue * value, GParamSpec * pspec);
169 static void gst_rtsp_media_set_property (GObject * object, guint propid,
170 const GValue * value, GParamSpec * pspec);
171 static void gst_rtsp_media_finalize (GObject * obj);
173 static gboolean default_handle_message (GstRTSPMedia * media,
174 GstMessage * message);
175 static void finish_unprepare (GstRTSPMedia * media);
176 static gboolean default_unprepare (GstRTSPMedia * media);
177 static gboolean default_convert_range (GstRTSPMedia * media,
178 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
179 static gboolean default_query_position (GstRTSPMedia * media,
181 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
182 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
183 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
186 static gboolean wait_preroll (GstRTSPMedia * media);
188 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
190 #define C_ENUM(v) ((gint) v)
192 #define GST_TYPE_RTSP_SUSPEND_MODE (gst_rtsp_suspend_mode_get_type())
194 gst_rtsp_suspend_mode_get_type (void)
197 static const GEnumValue values[] = {
198 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
199 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
201 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
206 if (g_once_init_enter (&id)) {
207 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
208 g_once_init_leave (&id, tmp);
213 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
216 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
218 GObjectClass *gobject_class;
220 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
222 gobject_class = G_OBJECT_CLASS (klass);
224 gobject_class->get_property = gst_rtsp_media_get_property;
225 gobject_class->set_property = gst_rtsp_media_set_property;
226 gobject_class->finalize = gst_rtsp_media_finalize;
228 g_object_class_install_property (gobject_class, PROP_SHARED,
229 g_param_spec_boolean ("shared", "Shared",
230 "If this media pipeline can be shared", DEFAULT_SHARED,
231 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
233 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
234 g_param_spec_enum ("suspend-mode", "Suspend Mode",
235 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
236 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
238 g_object_class_install_property (gobject_class, PROP_REUSABLE,
239 g_param_spec_boolean ("reusable", "Reusable",
240 "If this media pipeline can be reused after an unprepare",
241 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
243 g_object_class_install_property (gobject_class, PROP_PROFILES,
244 g_param_spec_flags ("profiles", "Profiles",
245 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
246 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
248 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
249 g_param_spec_flags ("protocols", "Protocols",
250 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
251 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
253 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
254 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
255 "Send an EOS event to the pipeline before unpreparing",
256 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
258 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
259 g_param_spec_uint ("buffer-size", "Buffer Size",
260 "The kernel UDP buffer size to use", 0, G_MAXUINT,
261 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
263 g_object_class_install_property (gobject_class, PROP_ELEMENT,
264 g_param_spec_object ("element", "The Element",
265 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
266 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
268 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
269 g_param_spec_boolean ("time-provider", "Time Provider",
270 "Use a NetTimeProvider for clients",
271 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
273 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
274 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
275 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
276 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
278 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
279 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
280 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
281 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
282 GST_TYPE_RTSP_STREAM);
284 gst_rtsp_media_signals[SIGNAL_PREPARED] =
285 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
286 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
287 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
289 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
290 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
291 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
292 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
294 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
295 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
296 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
297 g_cclosure_marshal_VOID__INT, G_TYPE_NONE, 1, G_TYPE_INT);
299 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
301 klass->handle_message = default_handle_message;
302 klass->unprepare = default_unprepare;
303 klass->convert_range = default_convert_range;
304 klass->query_position = default_query_position;
305 klass->query_stop = default_query_stop;
306 klass->create_rtpbin = default_create_rtpbin;
307 klass->setup_sdp = default_setup_sdp;
311 gst_rtsp_media_init (GstRTSPMedia * media)
313 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
317 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
318 g_mutex_init (&priv->lock);
319 g_cond_init (&priv->cond);
320 g_rec_mutex_init (&priv->state_lock);
322 priv->shared = DEFAULT_SHARED;
323 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
324 priv->reusable = DEFAULT_REUSABLE;
325 priv->profiles = DEFAULT_PROFILES;
326 priv->protocols = DEFAULT_PROTOCOLS;
327 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
328 priv->buffer_size = DEFAULT_BUFFER_SIZE;
329 priv->time_provider = DEFAULT_TIME_PROVIDER;
333 gst_rtsp_media_finalize (GObject * obj)
335 GstRTSPMediaPrivate *priv;
338 media = GST_RTSP_MEDIA (obj);
341 GST_INFO ("finalize media %p", media);
343 if (priv->permissions)
344 gst_rtsp_permissions_unref (priv->permissions);
346 g_ptr_array_unref (priv->streams);
348 g_list_free_full (priv->dynamic, gst_object_unref);
351 gst_object_unref (priv->pipeline);
353 gst_object_unref (priv->nettime);
354 gst_object_unref (priv->element);
356 g_object_unref (priv->pool);
357 g_mutex_clear (&priv->lock);
358 g_cond_clear (&priv->cond);
359 g_rec_mutex_clear (&priv->state_lock);
361 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
365 gst_rtsp_media_get_property (GObject * object, guint propid,
366 GValue * value, GParamSpec * pspec)
368 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
372 g_value_set_object (value, media->priv->element);
375 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
377 case PROP_SUSPEND_MODE:
378 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
381 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
384 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
387 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
389 case PROP_EOS_SHUTDOWN:
390 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
392 case PROP_BUFFER_SIZE:
393 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
395 case PROP_TIME_PROVIDER:
396 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
399 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
404 gst_rtsp_media_set_property (GObject * object, guint propid,
405 const GValue * value, GParamSpec * pspec)
407 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
411 media->priv->element = g_value_get_object (value);
412 gst_object_ref_sink (media->priv->element);
415 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
417 case PROP_SUSPEND_MODE:
418 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
421 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
424 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
427 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
429 case PROP_EOS_SHUTDOWN:
430 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
432 case PROP_BUFFER_SIZE:
433 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
435 case PROP_TIME_PROVIDER:
436 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
439 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
444 default_query_position (GstRTSPMedia * media, gint64 * position)
446 return gst_element_query_position (media->priv->pipeline, GST_FORMAT_TIME,
451 default_query_stop (GstRTSPMedia * media, gint64 * stop)
456 query = gst_query_new_segment (GST_FORMAT_TIME);
457 if ((res = gst_element_query (media->priv->pipeline, query))) {
459 gst_query_parse_segment (query, NULL, &format, NULL, stop);
460 if (format != GST_FORMAT_TIME)
463 gst_query_unref (query);
468 default_create_rtpbin (GstRTSPMedia * media)
472 rtpbin = gst_element_factory_make ("rtpbin", NULL);
477 /* must be called with state lock */
479 collect_media_stats (GstRTSPMedia * media)
481 GstRTSPMediaPrivate *priv = media->priv;
482 gint64 position, stop;
484 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
485 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
488 priv->range.unit = GST_RTSP_RANGE_NPT;
490 GST_INFO ("collect media stats");
493 priv->range.min.type = GST_RTSP_TIME_NOW;
494 priv->range.min.seconds = -1;
495 priv->range_start = -1;
496 priv->range.max.type = GST_RTSP_TIME_END;
497 priv->range.max.seconds = -1;
498 priv->range_stop = -1;
500 GstRTSPMediaClass *klass;
503 klass = GST_RTSP_MEDIA_GET_CLASS (media);
505 /* get the position */
507 if (klass->query_position)
508 ret = klass->query_position (media, &position);
511 GST_INFO ("position query failed");
515 /* get the current segment stop */
517 if (klass->query_stop)
518 ret = klass->query_stop (media, &stop);
521 GST_INFO ("stop query failed");
525 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
526 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
528 if (position == -1) {
529 priv->range.min.type = GST_RTSP_TIME_NOW;
530 priv->range.min.seconds = -1;
531 priv->range_start = -1;
533 priv->range.min.type = GST_RTSP_TIME_SECONDS;
534 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
535 priv->range_start = position;
538 priv->range.max.type = GST_RTSP_TIME_END;
539 priv->range.max.seconds = -1;
540 priv->range_stop = -1;
542 priv->range.max.type = GST_RTSP_TIME_SECONDS;
543 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
544 priv->range_stop = stop;
550 * gst_rtsp_media_new:
551 * @element: (transfer full): a #GstElement
553 * Create a new #GstRTSPMedia instance. @element is the bin element that
554 * provides the different streams. The #GstRTSPMedia object contains the
555 * element to produce RTP data for one or more related (audio/video/..)
558 * Ownership is taken of @element.
560 * Returns: a new #GstRTSPMedia object.
563 gst_rtsp_media_new (GstElement * element)
565 GstRTSPMedia *result;
567 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
569 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
575 * gst_rtsp_media_get_element:
576 * @media: a #GstRTSPMedia
578 * Get the element that was used when constructing @media.
580 * Returns: (transfer full): a #GstElement. Unref after usage.
583 gst_rtsp_media_get_element (GstRTSPMedia * media)
585 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
587 return gst_object_ref (media->priv->element);
591 * gst_rtsp_media_take_pipeline:
592 * @media: a #GstRTSPMedia
593 * @pipeline: (transfer full): a #GstPipeline
595 * Set @pipeline as the #GstPipeline for @media. Ownership is
596 * taken of @pipeline.
599 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
601 GstRTSPMediaPrivate *priv;
603 GstNetTimeProvider *nettime;
605 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
606 g_return_if_fail (GST_IS_PIPELINE (pipeline));
610 g_mutex_lock (&priv->lock);
611 old = priv->pipeline;
612 priv->pipeline = GST_ELEMENT_CAST (pipeline);
613 nettime = priv->nettime;
614 priv->nettime = NULL;
615 g_mutex_unlock (&priv->lock);
618 gst_object_unref (old);
621 gst_object_unref (nettime);
623 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
627 * gst_rtsp_media_set_permissions:
628 * @media: a #GstRTSPMedia
629 * @permissions: a #GstRTSPPermissions
631 * Set @permissions on @media.
634 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
635 GstRTSPPermissions * permissions)
637 GstRTSPMediaPrivate *priv;
639 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
643 g_mutex_lock (&priv->lock);
644 if (priv->permissions)
645 gst_rtsp_permissions_unref (priv->permissions);
646 if ((priv->permissions = permissions))
647 gst_rtsp_permissions_ref (permissions);
648 g_mutex_unlock (&priv->lock);
652 * gst_rtsp_media_get_permissions:
653 * @media: a #GstRTSPMedia
655 * Get the permissions object from @media.
657 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
660 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
662 GstRTSPMediaPrivate *priv;
663 GstRTSPPermissions *result;
665 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
669 g_mutex_lock (&priv->lock);
670 if ((result = priv->permissions))
671 gst_rtsp_permissions_ref (result);
672 g_mutex_unlock (&priv->lock);
678 * gst_rtsp_media_set_suspend_mode:
679 * @media: a #GstRTSPMedia
680 * @mode: the new #GstRTSPSuspendMode
682 * Control how @ media will be suspended after the SDP has been generated and
683 * after a PAUSE request has been performed.
685 * Media must be unprepared when setting the suspend mode.
688 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
690 GstRTSPMediaPrivate *priv;
692 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
696 g_rec_mutex_lock (&priv->state_lock);
697 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
699 priv->suspend_mode = mode;
700 g_rec_mutex_unlock (&priv->state_lock);
707 GST_WARNING ("media %p was prepared", media);
708 g_rec_mutex_unlock (&priv->state_lock);
713 * gst_rtsp_media_get_suspend_mode:
714 * @media: a #GstRTSPMedia
716 * Get how @media will be suspended.
718 * Returns: #GstRTSPSuspendMode.
721 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
723 GstRTSPMediaPrivate *priv;
724 GstRTSPSuspendMode res;
726 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
730 g_rec_mutex_lock (&priv->state_lock);
731 res = priv->suspend_mode;
732 g_rec_mutex_unlock (&priv->state_lock);
738 * gst_rtsp_media_set_shared:
739 * @media: a #GstRTSPMedia
740 * @shared: the new value
742 * Set or unset if the pipeline for @media can be shared will multiple clients.
743 * When @shared is %TRUE, client requests for this media will share the media
747 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
749 GstRTSPMediaPrivate *priv;
751 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
755 g_mutex_lock (&priv->lock);
756 priv->shared = shared;
757 g_mutex_unlock (&priv->lock);
761 * gst_rtsp_media_is_shared:
762 * @media: a #GstRTSPMedia
764 * Check if the pipeline for @media can be shared between multiple clients.
766 * Returns: %TRUE if the media can be shared between clients.
769 gst_rtsp_media_is_shared (GstRTSPMedia * media)
771 GstRTSPMediaPrivate *priv;
774 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
778 g_mutex_lock (&priv->lock);
780 g_mutex_unlock (&priv->lock);
786 * gst_rtsp_media_set_reusable:
787 * @media: a #GstRTSPMedia
788 * @reusable: the new value
790 * Set or unset if the pipeline for @media can be reused after the pipeline has
794 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
796 GstRTSPMediaPrivate *priv;
798 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
802 g_mutex_lock (&priv->lock);
803 priv->reusable = reusable;
804 g_mutex_unlock (&priv->lock);
808 * gst_rtsp_media_is_reusable:
809 * @media: a #GstRTSPMedia
811 * Check if the pipeline for @media can be reused after an unprepare.
813 * Returns: %TRUE if the media can be reused
816 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
818 GstRTSPMediaPrivate *priv;
821 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
825 g_mutex_lock (&priv->lock);
826 res = priv->reusable;
827 g_mutex_unlock (&priv->lock);
833 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
835 gst_rtsp_stream_set_profiles (stream, *profiles);
839 * gst_rtsp_media_set_profiles:
840 * @media: a #GstRTSPMedia
841 * @profiles: the new flags
843 * Configure the allowed lower transport for @media.
846 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
848 GstRTSPMediaPrivate *priv;
850 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
854 g_mutex_lock (&priv->lock);
855 priv->profiles = profiles;
856 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
857 g_mutex_unlock (&priv->lock);
861 * gst_rtsp_media_get_profiles:
862 * @media: a #GstRTSPMedia
864 * Get the allowed profiles of @media.
866 * Returns: a #GstRTSPProfile
869 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
871 GstRTSPMediaPrivate *priv;
874 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
878 g_mutex_lock (&priv->lock);
879 res = priv->profiles;
880 g_mutex_unlock (&priv->lock);
886 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
888 gst_rtsp_stream_set_protocols (stream, *protocols);
892 * gst_rtsp_media_set_protocols:
893 * @media: a #GstRTSPMedia
894 * @protocols: the new flags
896 * Configure the allowed lower transport for @media.
899 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
901 GstRTSPMediaPrivate *priv;
903 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
907 g_mutex_lock (&priv->lock);
908 priv->protocols = protocols;
909 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
910 g_mutex_unlock (&priv->lock);
914 * gst_rtsp_media_get_protocols:
915 * @media: a #GstRTSPMedia
917 * Get the allowed protocols of @media.
919 * Returns: a #GstRTSPLowerTrans
922 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
924 GstRTSPMediaPrivate *priv;
925 GstRTSPLowerTrans res;
927 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
928 GST_RTSP_LOWER_TRANS_UNKNOWN);
932 g_mutex_lock (&priv->lock);
933 res = priv->protocols;
934 g_mutex_unlock (&priv->lock);
940 * gst_rtsp_media_set_eos_shutdown:
941 * @media: a #GstRTSPMedia
942 * @eos_shutdown: the new value
944 * Set or unset if an EOS event will be sent to the pipeline for @media before
948 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
950 GstRTSPMediaPrivate *priv;
952 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
956 g_mutex_lock (&priv->lock);
957 priv->eos_shutdown = eos_shutdown;
958 g_mutex_unlock (&priv->lock);
962 * gst_rtsp_media_is_eos_shutdown:
963 * @media: a #GstRTSPMedia
965 * Check if the pipeline for @media will send an EOS down the pipeline before
968 * Returns: %TRUE if the media will send EOS before unpreparing.
971 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
973 GstRTSPMediaPrivate *priv;
976 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
980 g_mutex_lock (&priv->lock);
981 res = priv->eos_shutdown;
982 g_mutex_unlock (&priv->lock);
988 * gst_rtsp_media_set_buffer_size:
989 * @media: a #GstRTSPMedia
990 * @size: the new value
992 * Set the kernel UDP buffer size.
995 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
997 GstRTSPMediaPrivate *priv;
999 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1001 GST_LOG_OBJECT (media, "set buffer size %u", size);
1005 g_mutex_lock (&priv->lock);
1006 priv->buffer_size = size;
1007 g_mutex_unlock (&priv->lock);
1011 * gst_rtsp_media_get_buffer_size:
1012 * @media: a #GstRTSPMedia
1014 * Get the kernel UDP buffer size.
1016 * Returns: the kernel UDP buffer size.
1019 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1021 GstRTSPMediaPrivate *priv;
1024 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1028 g_mutex_unlock (&priv->lock);
1029 res = priv->buffer_size;
1030 g_mutex_unlock (&priv->lock);
1036 * gst_rtsp_media_use_time_provider:
1037 * @media: a #GstRTSPMedia
1038 * @time_provider: if a #GstNetTimeProvider should be used
1040 * Set @media to provide a #GstNetTimeProvider.
1043 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1045 GstRTSPMediaPrivate *priv;
1047 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1051 g_mutex_lock (&priv->lock);
1052 priv->time_provider = time_provider;
1053 g_mutex_unlock (&priv->lock);
1057 * gst_rtsp_media_is_time_provider:
1058 * @media: a #GstRTSPMedia
1060 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1062 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1064 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1067 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1069 GstRTSPMediaPrivate *priv;
1072 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1076 g_mutex_unlock (&priv->lock);
1077 res = priv->time_provider;
1078 g_mutex_unlock (&priv->lock);
1084 * gst_rtsp_media_set_address_pool:
1085 * @media: a #GstRTSPMedia
1086 * @pool: a #GstRTSPAddressPool
1088 * configure @pool to be used as the address pool of @media.
1091 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1092 GstRTSPAddressPool * pool)
1094 GstRTSPMediaPrivate *priv;
1095 GstRTSPAddressPool *old;
1097 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1101 GST_LOG_OBJECT (media, "set address pool %p", pool);
1103 g_mutex_lock (&priv->lock);
1104 if ((old = priv->pool) != pool)
1105 priv->pool = pool ? g_object_ref (pool) : NULL;
1108 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1110 g_mutex_unlock (&priv->lock);
1113 g_object_unref (old);
1117 * gst_rtsp_media_get_address_pool:
1118 * @media: a #GstRTSPMedia
1120 * Get the #GstRTSPAddressPool used as the address pool of @media.
1122 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1125 GstRTSPAddressPool *
1126 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1128 GstRTSPMediaPrivate *priv;
1129 GstRTSPAddressPool *result;
1131 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1135 g_mutex_lock (&priv->lock);
1136 if ((result = priv->pool))
1137 g_object_ref (result);
1138 g_mutex_unlock (&priv->lock);
1144 * gst_rtsp_media_collect_streams:
1145 * @media: a #GstRTSPMedia
1147 * Find all payloader elements, they should be named pay\%d in the
1148 * element of @media, and create #GstRTSPStreams for them.
1150 * Collect all dynamic elements, named dynpay\%d, and add them to
1151 * the list of dynamic elements.
1154 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1156 GstRTSPMediaPrivate *priv;
1157 GstElement *element, *elem;
1162 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1165 element = priv->element;
1168 for (i = 0; have_elem; i++) {
1173 name = g_strdup_printf ("pay%d", i);
1174 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1175 GST_INFO ("found stream %d with payloader %p", i, elem);
1177 /* take the pad of the payloader */
1178 pad = gst_element_get_static_pad (elem, "src");
1179 /* create the stream */
1180 gst_rtsp_media_create_stream (media, elem, pad);
1181 gst_object_unref (pad);
1182 gst_object_unref (elem);
1188 name = g_strdup_printf ("dynpay%d", i);
1189 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1190 /* a stream that will dynamically create pads to provide RTP packets */
1192 GST_INFO ("found dynamic element %d, %p", i, elem);
1194 g_mutex_lock (&priv->lock);
1195 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1196 g_mutex_unlock (&priv->lock);
1205 * gst_rtsp_media_create_stream:
1206 * @media: a #GstRTSPMedia
1207 * @payloader: a #GstElement
1208 * @srcpad: a source #GstPad
1210 * Create a new stream in @media that provides RTP data on @srcpad.
1211 * @srcpad should be a pad of an element inside @media->element.
1213 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1217 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1220 GstRTSPMediaPrivate *priv;
1221 GstRTSPStream *stream;
1226 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1227 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1228 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1229 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
1233 g_mutex_lock (&priv->lock);
1234 idx = priv->streams->len;
1236 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1238 name = g_strdup_printf ("src_%u", idx);
1239 srcpad = gst_ghost_pad_new (name, pad);
1240 gst_pad_set_active (srcpad, TRUE);
1241 gst_element_add_pad (priv->element, srcpad);
1244 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
1246 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1247 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1249 g_ptr_array_add (priv->streams, stream);
1250 g_mutex_unlock (&priv->lock);
1252 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1259 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1261 GstRTSPMediaPrivate *priv;
1266 g_mutex_lock (&priv->lock);
1267 /* remove the ghostpad */
1268 srcpad = gst_rtsp_stream_get_srcpad (stream);
1269 gst_element_remove_pad (priv->element, srcpad);
1270 gst_object_unref (srcpad);
1271 /* now remove the stream */
1272 g_object_ref (stream);
1273 g_ptr_array_remove (priv->streams, stream);
1274 g_mutex_unlock (&priv->lock);
1276 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1279 g_object_unref (stream);
1283 * gst_rtsp_media_n_streams:
1284 * @media: a #GstRTSPMedia
1286 * Get the number of streams in this media.
1288 * Returns: The number of streams.
1291 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1293 GstRTSPMediaPrivate *priv;
1296 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1300 g_mutex_lock (&priv->lock);
1301 res = priv->streams->len;
1302 g_mutex_unlock (&priv->lock);
1308 * gst_rtsp_media_get_stream:
1309 * @media: a #GstRTSPMedia
1310 * @idx: the stream index
1312 * Retrieve the stream with index @idx from @media.
1314 * Returns: (transfer none): the #GstRTSPStream at index @idx or %NULL when a stream with
1315 * that index did not exist.
1318 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1320 GstRTSPMediaPrivate *priv;
1323 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1327 g_mutex_lock (&priv->lock);
1328 if (idx < priv->streams->len)
1329 res = g_ptr_array_index (priv->streams, idx);
1332 g_mutex_unlock (&priv->lock);
1338 * gst_rtsp_media_find_stream:
1339 * @media: a #GstRTSPMedia
1340 * @control: the control of the stream
1342 * Find a stream in @media with @control as the control uri.
1344 * Returns: (transfer none): the #GstRTSPStream with control uri @control
1345 * or %NULL when a stream with that control did not exist.
1348 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1350 GstRTSPMediaPrivate *priv;
1354 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1355 g_return_val_if_fail (control != NULL, NULL);
1361 g_mutex_lock (&priv->lock);
1362 for (i = 0; i < priv->streams->len; i++) {
1363 GstRTSPStream *test;
1365 test = g_ptr_array_index (priv->streams, i);
1366 if (gst_rtsp_stream_has_control (test, control)) {
1371 g_mutex_unlock (&priv->lock);
1376 /* called with state-lock */
1378 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1379 GstRTSPRangeUnit unit)
1381 return gst_rtsp_range_convert_units (range, unit);
1385 * gst_rtsp_media_get_range_string:
1386 * @media: a #GstRTSPMedia
1387 * @play: for the PLAY request
1388 * @unit: the unit to use for the string
1390 * Get the current range as a string. @media must be prepared with
1391 * gst_rtsp_media_prepare ().
1393 * Returns: The range as a string, g_free() after usage.
1396 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1397 GstRTSPRangeUnit unit)
1399 GstRTSPMediaClass *klass;
1400 GstRTSPMediaPrivate *priv;
1402 GstRTSPTimeRange range;
1404 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1405 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1406 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1410 g_rec_mutex_lock (&priv->state_lock);
1411 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1412 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1415 g_mutex_lock (&priv->lock);
1417 /* Update the range value with current position/duration */
1418 collect_media_stats (media);
1421 range = priv->range;
1423 if (!play && priv->n_active > 0) {
1424 range.min.type = GST_RTSP_TIME_NOW;
1425 range.min.seconds = -1;
1427 g_mutex_unlock (&priv->lock);
1428 g_rec_mutex_unlock (&priv->state_lock);
1430 if (!klass->convert_range (media, &range, unit))
1431 goto conversion_failed;
1433 result = gst_rtsp_range_to_string (&range);
1440 GST_WARNING ("media %p was not prepared", media);
1441 g_rec_mutex_unlock (&priv->state_lock);
1446 GST_WARNING ("range conversion to unit %d failed", unit);
1452 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1454 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1458 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1460 GstRTSPMediaPrivate *priv = media->priv;
1462 GST_DEBUG ("media %p set blocked %d", media, blocked);
1463 priv->blocked = blocked;
1464 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1468 * gst_rtsp_media_seek:
1469 * @media: a #GstRTSPMedia
1470 * @range: a #GstRTSPTimeRange
1472 * Seek the pipeline of @media to @range. @media must be prepared with
1473 * gst_rtsp_media_prepare().
1475 * Returns: %TRUE on success.
1478 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1480 GstRTSPMediaClass *klass;
1481 GstRTSPMediaPrivate *priv;
1484 GstClockTime start, stop;
1485 GstSeekType start_type, stop_type;
1488 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1490 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1491 g_return_val_if_fail (range != NULL, FALSE);
1492 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1496 g_rec_mutex_lock (&priv->state_lock);
1497 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1500 /* Update the seekable state of the pipeline in case it changed */
1501 query = gst_query_new_seeking (GST_FORMAT_TIME);
1502 if (gst_element_query (priv->pipeline, query)) {
1507 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1508 priv->seekable = seekable;
1510 gst_query_unref (query);
1512 if (!priv->seekable)
1515 /* depends on the current playing state of the pipeline. We might need to
1516 * queue this until we get EOS. */
1517 flags = GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT;
1519 start_type = stop_type = GST_SEEK_TYPE_NONE;
1521 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1523 gst_rtsp_range_get_times (range, &start, &stop);
1525 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1526 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1527 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1528 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1530 if (priv->range_start == start)
1531 start = GST_CLOCK_TIME_NONE;
1532 else if (start != GST_CLOCK_TIME_NONE)
1533 start_type = GST_SEEK_TYPE_SET;
1535 if (priv->range_stop == stop)
1536 stop = GST_CLOCK_TIME_NONE;
1537 else if (stop != GST_CLOCK_TIME_NONE)
1538 stop_type = GST_SEEK_TYPE_SET;
1540 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1541 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1542 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1544 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
1546 media_streams_set_blocked (media, TRUE);
1548 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1549 flags, start_type, start, stop_type, stop);
1551 /* and block for the seek to complete */
1552 GST_INFO ("done seeking %d", res);
1553 g_rec_mutex_unlock (&priv->state_lock);
1555 /* wait until pipeline is prerolled again, this will also collect stats */
1556 if (!wait_preroll (media))
1557 goto preroll_failed;
1559 g_rec_mutex_lock (&priv->state_lock);
1560 GST_INFO ("prerolled again");
1562 GST_INFO ("no seek needed");
1565 g_rec_mutex_unlock (&priv->state_lock);
1572 g_rec_mutex_unlock (&priv->state_lock);
1573 GST_INFO ("media %p is not prepared", media);
1578 g_rec_mutex_unlock (&priv->state_lock);
1579 GST_INFO ("pipeline is not seekable");
1584 g_rec_mutex_unlock (&priv->state_lock);
1585 GST_WARNING ("conversion to npt not supported");
1590 GST_WARNING ("failed to preroll after seek");
1596 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1598 GstRTSPMediaPrivate *priv = media->priv;
1600 g_mutex_lock (&priv->lock);
1601 priv->status = status;
1602 GST_DEBUG ("setting new status to %d", status);
1603 g_cond_broadcast (&priv->cond);
1604 g_mutex_unlock (&priv->lock);
1608 * gst_rtsp_media_get_status:
1609 * @media: a #GstRTSPMedia
1611 * Get the status of @media. When @media is busy preparing, this function waits
1612 * until @media is prepared or in error.
1614 * Returns: the status of @media.
1617 gst_rtsp_media_get_status (GstRTSPMedia * media)
1619 GstRTSPMediaPrivate *priv = media->priv;
1620 GstRTSPMediaStatus result;
1623 g_mutex_lock (&priv->lock);
1624 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1625 /* while we are preparing, wait */
1626 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1627 GST_DEBUG ("waiting for status change");
1628 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1629 GST_DEBUG ("timeout, assuming error status");
1630 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1633 /* could be success or error */
1634 result = priv->status;
1635 GST_DEBUG ("got status %d", result);
1636 g_mutex_unlock (&priv->lock);
1642 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
1644 *blocked &= gst_rtsp_stream_is_blocking (stream);
1648 media_streams_blocking (GstRTSPMedia * media)
1650 gboolean blocking = TRUE;
1652 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
1658 /* called with state-lock */
1660 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1662 GstRTSPMediaPrivate *priv = media->priv;
1663 GstMessageType type;
1665 type = GST_MESSAGE_TYPE (message);
1668 case GST_MESSAGE_STATE_CHANGED:
1670 case GST_MESSAGE_BUFFERING:
1674 gst_message_parse_buffering (message, &percent);
1676 /* no state management needed for live pipelines */
1680 if (percent == 100) {
1681 /* a 100% message means buffering is done */
1682 priv->buffering = FALSE;
1683 /* if the desired state is playing, go back */
1684 if (priv->target_state == GST_STATE_PLAYING) {
1685 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1686 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1688 GST_INFO ("Buffering done");
1691 /* buffering busy */
1692 if (priv->buffering == FALSE) {
1693 if (priv->target_state == GST_STATE_PLAYING) {
1694 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1695 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1696 gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1698 GST_INFO ("Buffering ...");
1701 priv->buffering = TRUE;
1705 case GST_MESSAGE_LATENCY:
1707 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1710 case GST_MESSAGE_ERROR:
1715 gst_message_parse_error (message, &gerror, &debug);
1716 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1717 g_error_free (gerror);
1720 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1723 case GST_MESSAGE_WARNING:
1728 gst_message_parse_warning (message, &gerror, &debug);
1729 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1730 g_error_free (gerror);
1734 case GST_MESSAGE_ELEMENT:
1736 const GstStructure *s;
1738 s = gst_message_get_structure (message);
1739 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
1740 GST_DEBUG ("media received blocking message");
1741 if (priv->blocked && media_streams_blocking (media)) {
1742 GST_DEBUG ("media is blocking");
1743 collect_media_stats (media);
1745 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1746 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1751 case GST_MESSAGE_STREAM_STATUS:
1753 case GST_MESSAGE_ASYNC_DONE:
1755 /* when we are dynamically adding pads, the addition of the udpsrc will
1756 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1757 * wait for the final ASYNC_DONE after everything prerolled */
1758 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1760 GST_INFO ("%p: got ASYNC_DONE", media);
1761 collect_media_stats (media);
1763 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1764 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1767 case GST_MESSAGE_EOS:
1768 GST_INFO ("%p: got EOS", media);
1770 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1771 GST_DEBUG ("shutting down after EOS");
1772 finish_unprepare (media);
1776 GST_INFO ("%p: got message type %d (%s)", media, type,
1777 gst_message_type_get_name (type));
1784 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1786 GstRTSPMediaPrivate *priv = media->priv;
1787 GstRTSPMediaClass *klass;
1790 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1792 g_rec_mutex_lock (&priv->state_lock);
1793 if (klass->handle_message)
1794 ret = klass->handle_message (media, message);
1797 g_rec_mutex_unlock (&priv->state_lock);
1803 watch_destroyed (GstRTSPMedia * media)
1805 GST_DEBUG_OBJECT (media, "source destroyed");
1806 g_object_unref (media);
1810 find_payload_element (GstElement * payloader)
1812 GstElement *pay = NULL;
1814 if (GST_IS_BIN (payloader)) {
1816 GValue item = { 0 };
1818 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
1819 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
1820 GstElement *element = (GstElement *) g_value_get_object (&item);
1821 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
1825 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
1829 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
1830 pay = gst_object_ref (element);
1831 g_value_unset (&item);
1834 g_value_unset (&item);
1836 gst_iterator_free (iter);
1838 pay = g_object_ref (payloader);
1844 /* called from streaming threads */
1846 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1848 GstRTSPMediaPrivate *priv = media->priv;
1849 GstRTSPStream *stream;
1852 /* find the real payload element */
1853 pay = find_payload_element (element);
1854 stream = gst_rtsp_media_create_stream (media, pay, pad);
1855 gst_object_unref (pay);
1857 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1859 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1861 g_rec_mutex_lock (&priv->state_lock);
1862 /* we will be adding elements below that will cause ASYNC_DONE to be
1863 * posted in the bus. We want to ignore those messages until the
1864 * pipeline really prerolled. */
1865 priv->adding = TRUE;
1867 /* join the element in the PAUSED state because this callback is
1868 * called from the streaming thread and it is PAUSED */
1869 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1870 priv->rtpbin, GST_STATE_PAUSED);
1872 priv->adding = FALSE;
1873 g_rec_mutex_unlock (&priv->state_lock);
1877 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1879 GstRTSPMediaPrivate *priv = media->priv;
1880 GstRTSPStream *stream;
1882 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
1886 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1888 g_rec_mutex_lock (&priv->state_lock);
1889 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
1890 g_rec_mutex_unlock (&priv->state_lock);
1892 gst_rtsp_media_remove_stream (media, stream);
1896 remove_fakesink (GstRTSPMediaPrivate * priv)
1898 GstElement *fakesink;
1900 g_mutex_lock (&priv->lock);
1901 if ((fakesink = priv->fakesink))
1902 gst_object_ref (fakesink);
1903 priv->fakesink = NULL;
1904 g_mutex_unlock (&priv->lock);
1907 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
1908 gst_element_set_state (fakesink, GST_STATE_NULL);
1909 gst_object_unref (fakesink);
1910 GST_INFO ("removed fakesink");
1915 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
1917 GstRTSPMediaPrivate *priv = media->priv;
1919 GST_INFO ("no more pads");
1920 remove_fakesink (priv);
1923 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
1925 struct _DynPaySignalHandlers
1927 gulong pad_added_handler;
1928 gulong pad_removed_handler;
1929 gulong no_more_pads_handler;
1933 start_preroll (GstRTSPMedia * media)
1935 GstRTSPMediaPrivate *priv = media->priv;
1936 GstStateChangeReturn ret;
1938 GST_INFO ("setting pipeline to PAUSED for media %p", media);
1939 /* first go to PAUSED */
1940 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
1941 priv->target_state = GST_STATE_PAUSED;
1944 case GST_STATE_CHANGE_SUCCESS:
1945 GST_INFO ("SUCCESS state change for media %p", media);
1946 priv->seekable = TRUE;
1948 case GST_STATE_CHANGE_ASYNC:
1949 GST_INFO ("ASYNC state change for media %p", media);
1950 priv->seekable = TRUE;
1952 case GST_STATE_CHANGE_NO_PREROLL:
1953 /* we need to go to PLAYING */
1954 GST_INFO ("NO_PREROLL state change: live media %p", media);
1955 /* FIXME we disable seeking for live streams for now. We should perform a
1956 * seeking query in preroll instead */
1957 priv->seekable = FALSE;
1958 priv->is_live = TRUE;
1959 /* start blocked to make sure nothing goes to the sink */
1960 media_streams_set_blocked (media, TRUE);
1961 ret = gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
1962 if (ret == GST_STATE_CHANGE_FAILURE)
1965 case GST_STATE_CHANGE_FAILURE:
1973 GST_WARNING ("failed to preroll pipeline");
1979 wait_preroll (GstRTSPMedia * media)
1981 GstRTSPMediaStatus status;
1983 GST_DEBUG ("wait to preroll pipeline");
1985 /* wait until pipeline is prerolled */
1986 status = gst_rtsp_media_get_status (media);
1987 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
1988 goto preroll_failed;
1994 GST_WARNING ("failed to preroll pipeline");
2000 start_prepare (GstRTSPMedia * media)
2002 GstRTSPMediaPrivate *priv = media->priv;
2006 /* link streams we already have, other streams might appear when we have
2007 * dynamic elements */
2008 for (i = 0; i < priv->streams->len; i++) {
2009 GstRTSPStream *stream;
2011 stream = g_ptr_array_index (priv->streams, i);
2013 gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2014 priv->rtpbin, GST_STATE_NULL);
2017 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2018 GstElement *elem = walk->data;
2019 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2021 GST_INFO ("adding callbacks for dynamic element %p", elem);
2023 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2024 (GCallback) pad_added_cb, media);
2025 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2026 (GCallback) pad_removed_cb, media);
2027 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2028 (GCallback) no_more_pads_cb, media);
2030 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2032 /* we add a fakesink here in order to make the state change async. We remove
2033 * the fakesink again in the no-more-pads callback. */
2034 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2035 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2038 if (!start_preroll (media))
2039 goto preroll_failed;
2045 GST_WARNING ("failed to preroll pipeline");
2046 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2052 * gst_rtsp_media_prepare:
2053 * @media: a #GstRTSPMedia
2054 * @thread: a #GstRTSPThread to run the bus handler or %NULL
2056 * Prepare @media for streaming. This function will create the objects
2057 * to manage the streaming. A pipeline must have been set on @media with
2058 * gst_rtsp_media_take_pipeline().
2060 * It will preroll the pipeline and collect vital information about the streams
2061 * such as the duration.
2063 * Returns: %TRUE on success.
2066 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2068 GstRTSPMediaPrivate *priv;
2071 GstRTSPMediaClass *klass;
2073 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2074 g_return_val_if_fail (GST_IS_RTSP_THREAD (thread), FALSE);
2078 g_rec_mutex_lock (&priv->state_lock);
2079 priv->prepare_count++;
2081 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2082 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2085 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2088 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2089 goto not_unprepared;
2091 if (!priv->reusable && priv->reused)
2094 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2096 if (!klass->create_rtpbin)
2097 goto no_create_rtpbin;
2099 priv->rtpbin = klass->create_rtpbin (media);
2100 if (priv->rtpbin != NULL) {
2101 gboolean success = TRUE;
2103 if (klass->setup_rtpbin)
2104 success = klass->setup_rtpbin (media, priv->rtpbin);
2106 if (success == FALSE) {
2107 gst_object_unref (priv->rtpbin);
2108 priv->rtpbin = NULL;
2111 if (priv->rtpbin == NULL)
2114 GST_INFO ("preparing media %p", media);
2116 /* reset some variables */
2117 priv->is_live = FALSE;
2118 priv->seekable = FALSE;
2119 priv->buffering = FALSE;
2120 priv->thread = thread;
2121 /* we're preparing now */
2122 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
2124 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2126 /* add the pipeline bus to our custom mainloop */
2127 priv->source = gst_bus_create_watch (bus);
2128 gst_object_unref (bus);
2130 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2131 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2133 priv->id = g_source_attach (priv->source, thread->context);
2135 /* add stuff to the bin */
2136 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2138 /* do remainder in context */
2139 source = g_idle_source_new ();
2140 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
2141 g_source_attach (source, thread->context);
2142 g_source_unref (source);
2145 g_rec_mutex_unlock (&priv->state_lock);
2147 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2148 * able to do this async so that we don't block the server thread. */
2149 if (!wait_preroll (media))
2150 goto preroll_failed;
2152 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2154 GST_INFO ("object %p is prerolled", media);
2161 GST_LOG ("media %p was prepared", media);
2162 g_rec_mutex_unlock (&priv->state_lock);
2168 GST_WARNING ("media %p was not unprepared", media);
2169 priv->prepare_count--;
2170 g_rec_mutex_unlock (&priv->state_lock);
2175 priv->prepare_count--;
2176 g_rec_mutex_unlock (&priv->state_lock);
2177 GST_WARNING ("can not reuse media %p", media);
2182 priv->prepare_count--;
2183 g_rec_mutex_unlock (&priv->state_lock);
2184 GST_ERROR ("no create_rtpbin function");
2185 g_critical ("no create_rtpbin vmethod function set");
2190 priv->prepare_count--;
2191 g_rec_mutex_unlock (&priv->state_lock);
2192 GST_WARNING ("no rtpbin element");
2193 g_warning ("failed to create element 'rtpbin', check your installation");
2198 GST_WARNING ("failed to preroll pipeline");
2199 gst_rtsp_media_unprepare (media);
2204 /* must be called with state-lock */
2206 finish_unprepare (GstRTSPMedia * media)
2208 GstRTSPMediaPrivate *priv = media->priv;
2212 GST_DEBUG ("shutting down");
2214 gst_element_set_state (priv->pipeline, GST_STATE_NULL);
2215 remove_fakesink (priv);
2217 for (i = 0; i < priv->streams->len; i++) {
2218 GstRTSPStream *stream;
2220 GST_INFO ("Removing elements of stream %d from pipeline", i);
2222 stream = g_ptr_array_index (priv->streams, i);
2224 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2227 /* remove the pad signal handlers */
2228 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2229 GstElement *elem = walk->data;
2230 DynPaySignalHandlers *handlers;
2233 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2234 g_assert (handlers != NULL);
2236 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2237 g_signal_handler_disconnect (G_OBJECT (elem),
2238 handlers->pad_removed_handler);
2239 g_signal_handler_disconnect (G_OBJECT (elem),
2240 handlers->no_more_pads_handler);
2242 g_slice_free (DynPaySignalHandlers, handlers);
2245 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2246 priv->rtpbin = NULL;
2249 gst_object_unref (priv->nettime);
2250 priv->nettime = NULL;
2252 priv->reused = TRUE;
2253 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARED;
2255 /* when the media is not reusable, this will effectively unref the media and
2257 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2259 /* the source has the last ref to the media */
2261 GST_DEBUG ("destroy source");
2262 g_source_destroy (priv->source);
2263 g_source_unref (priv->source);
2266 GST_DEBUG ("stop thread");
2267 gst_rtsp_thread_stop (priv->thread);
2271 /* called with state-lock */
2273 default_unprepare (GstRTSPMedia * media)
2275 GstRTSPMediaPrivate *priv = media->priv;
2277 if (priv->eos_shutdown) {
2278 GST_DEBUG ("sending EOS for shutdown");
2279 /* ref so that we don't disappear */
2280 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2281 /* we need to go to playing again for the EOS to propagate, normally in this
2282 * state, nothing is receiving data from us anymore so this is ok. */
2283 gst_element_set_state (priv->pipeline, GST_STATE_PLAYING);
2284 priv->status = GST_RTSP_MEDIA_STATUS_UNPREPARING;
2286 finish_unprepare (media);
2292 * gst_rtsp_media_unprepare:
2293 * @media: a #GstRTSPMedia
2295 * Unprepare @media. After this call, the media should be prepared again before
2296 * it can be used again. If the media is set to be non-reusable, a new instance
2299 * Returns: %TRUE on success.
2302 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2304 GstRTSPMediaPrivate *priv;
2307 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2311 g_rec_mutex_lock (&priv->state_lock);
2312 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2313 goto was_unprepared;
2315 priv->prepare_count--;
2316 if (priv->prepare_count > 0)
2319 GST_INFO ("unprepare media %p", media);
2320 priv->target_state = GST_STATE_NULL;
2323 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2324 GstRTSPMediaClass *klass;
2326 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2327 if (klass->unprepare)
2328 success = klass->unprepare (media);
2330 finish_unprepare (media);
2332 g_rec_mutex_unlock (&priv->state_lock);
2338 g_rec_mutex_unlock (&priv->state_lock);
2339 GST_INFO ("media %p was already unprepared", media);
2344 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2345 g_rec_mutex_unlock (&priv->state_lock);
2350 /* should be called with state-lock */
2352 get_clock_unlocked (GstRTSPMedia * media)
2354 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2355 GST_DEBUG_OBJECT (media, "media was not prepared");
2358 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2362 * gst_rtsp_media_get_clock:
2363 * @media: a #GstRTSPMedia
2365 * Get the clock that is used by the pipeline in @media.
2367 * @media must be prepared before this method returns a valid clock object.
2369 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2372 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2375 GstRTSPMediaPrivate *priv;
2377 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2381 g_rec_mutex_lock (&priv->state_lock);
2382 clock = get_clock_unlocked (media);
2383 g_rec_mutex_unlock (&priv->state_lock);
2389 * gst_rtsp_media_get_base_time:
2390 * @media: a #GstRTSPMedia
2392 * Get the base_time that is used by the pipeline in @media.
2394 * @media must be prepared before this method returns a valid base_time.
2396 * Returns: the base_time used by @media.
2399 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2401 GstClockTime result;
2402 GstRTSPMediaPrivate *priv;
2404 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2408 g_rec_mutex_lock (&priv->state_lock);
2409 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2412 result = gst_element_get_base_time (media->priv->pipeline);
2413 g_rec_mutex_unlock (&priv->state_lock);
2420 g_rec_mutex_unlock (&priv->state_lock);
2421 GST_DEBUG_OBJECT (media, "media was not prepared");
2422 return GST_CLOCK_TIME_NONE;
2427 * gst_rtsp_media_get_time_provider:
2428 * @media: a #GstRTSPMedia
2429 * @address: an address or %NULL
2430 * @port: a port or 0
2432 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2433 * will listen on @address and @port for client time requests.
2435 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2437 GstNetTimeProvider *
2438 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2441 GstRTSPMediaPrivate *priv;
2442 GstNetTimeProvider *provider = NULL;
2444 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2448 g_rec_mutex_lock (&priv->state_lock);
2449 if (priv->time_provider) {
2450 if ((provider = priv->nettime) == NULL) {
2453 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2454 provider = gst_net_time_provider_new (clock, address, port);
2455 gst_object_unref (clock);
2457 priv->nettime = provider;
2461 g_rec_mutex_unlock (&priv->state_lock);
2464 gst_object_ref (provider);
2470 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
2472 return gst_rtsp_sdp_from_media (sdp, info, media);
2476 * gst_rtsp_media_setup_sdp:
2477 * @sdp: a #GstSDPMessage
2479 * @media: a #GstRTSPMedia
2481 * Add @media specific info to @sdp. @info is used to configure the connection
2482 * information in the SDP.
2484 * Returns: TRUE on success.
2487 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
2490 GstRTSPMediaPrivate *priv;
2491 GstRTSPMediaClass *klass;
2494 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2495 g_return_val_if_fail (sdp != NULL, FALSE);
2496 g_return_val_if_fail (info != NULL, FALSE);
2500 g_rec_mutex_lock (&priv->state_lock);
2502 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2504 if (!klass->setup_sdp)
2507 res = klass->setup_sdp (media, sdp, info);
2509 g_rec_mutex_unlock (&priv->state_lock);
2516 g_rec_mutex_unlock (&priv->state_lock);
2517 GST_ERROR ("no setup_sdp function");
2518 g_critical ("no setup_sdp vmethod function set");
2524 * gst_rtsp_media_suspend:
2525 * @media: a #GstRTSPMedia
2527 * Suspend @media. The state of the pipeline managed by @media is set to
2528 * GST_STATE_NULL but all streams are kept. @media can be prepared again
2529 * with gst_rtsp_media_undo_reset()
2531 * @media must be prepared with gst_rtsp_media_prepare();
2533 * Returns: %TRUE on success.
2536 gst_rtsp_media_suspend (GstRTSPMedia * media)
2538 GstRTSPMediaPrivate *priv = media->priv;
2539 GstStateChangeReturn ret;
2541 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2543 GST_FIXME ("suspend for dynamic pipelines needs fixing");
2545 g_rec_mutex_lock (&priv->state_lock);
2546 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2549 /* don't attempt to suspend when something is busy */
2550 if (priv->n_active > 0)
2553 switch (priv->suspend_mode) {
2554 case GST_RTSP_SUSPEND_MODE_NONE:
2555 GST_DEBUG ("media %p no suspend", media);
2557 case GST_RTSP_SUSPEND_MODE_PAUSE:
2558 GST_DEBUG ("media %p suspend to PAUSED", media);
2559 priv->target_state = GST_STATE_PAUSED;
2560 ret = gst_element_set_state (priv->pipeline, GST_STATE_PAUSED);
2561 if (ret == GST_STATE_CHANGE_FAILURE)
2564 case GST_RTSP_SUSPEND_MODE_RESET:
2565 GST_DEBUG ("media %p suspend to NULL", media);
2566 priv->target_state = GST_STATE_NULL;
2567 ret = gst_element_set_state (priv->pipeline, GST_STATE_NULL);
2568 if (ret == GST_STATE_CHANGE_FAILURE)
2574 /* let the streams do the state changes freely, if any */
2575 media_streams_set_blocked (media, FALSE);
2576 priv->status = GST_RTSP_MEDIA_STATUS_SUSPENDED;
2578 g_rec_mutex_unlock (&priv->state_lock);
2585 g_rec_mutex_unlock (&priv->state_lock);
2586 GST_WARNING ("media %p was not prepared", media);
2591 g_rec_mutex_unlock (&priv->state_lock);
2592 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2593 GST_WARNING ("failed changing pipeline's state for media %p", media);
2599 * gst_rtsp_media_unsuspend:
2600 * @media: a #GstRTSPMedia
2602 * Unsuspend @media if it was in a suspended state. This method does nothing
2603 * when the media was not in the suspended state.
2605 * Returns: %TRUE on success.
2608 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
2610 GstRTSPMediaPrivate *priv = media->priv;
2612 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2614 g_rec_mutex_lock (&priv->state_lock);
2615 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2618 switch (priv->suspend_mode) {
2619 case GST_RTSP_SUSPEND_MODE_NONE:
2620 priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
2622 case GST_RTSP_SUSPEND_MODE_PAUSE:
2623 priv->status = GST_RTSP_MEDIA_STATUS_PREPARED;
2625 case GST_RTSP_SUSPEND_MODE_RESET:
2627 priv->status = GST_RTSP_MEDIA_STATUS_PREPARING;
2628 if (!start_preroll (media))
2630 g_rec_mutex_unlock (&priv->state_lock);
2632 if (!wait_preroll (media))
2633 goto preroll_failed;
2635 g_rec_mutex_lock (&priv->state_lock);
2641 g_rec_mutex_unlock (&priv->state_lock);
2648 g_rec_mutex_unlock (&priv->state_lock);
2649 GST_WARNING ("failed to preroll pipeline");
2650 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2655 GST_WARNING ("failed to preroll pipeline");
2660 /* must be called with state-lock */
2662 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
2664 GstRTSPMediaPrivate *priv = media->priv;
2666 if (state == GST_STATE_NULL) {
2667 gst_rtsp_media_unprepare (media);
2669 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
2670 priv->target_state = state;
2671 /* when we are buffering, don't update the state yet, this will be done
2672 * when buffering finishes */
2673 if (priv->buffering) {
2674 GST_INFO ("Buffering busy, delay state change");
2676 if (state == GST_STATE_PLAYING)
2677 /* make sure pads are not blocking anymore when going to PLAYING */
2678 media_streams_set_blocked (media, FALSE);
2680 gst_element_set_state (priv->pipeline, state);
2682 /* and suspend after pause */
2683 if (state == GST_STATE_PAUSED)
2684 gst_rtsp_media_suspend (media);
2690 * gst_rtsp_media_set_pipeline_state:
2691 * @media: a #GstRTSPMedia
2692 * @state: the target state of the pipeline
2694 * Set the state of the pipeline managed by @media to @state
2697 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
2699 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2701 g_rec_mutex_lock (&media->priv->state_lock);
2702 media_set_pipeline_state_locked (media, state);
2703 g_rec_mutex_unlock (&media->priv->state_lock);
2707 * gst_rtsp_media_set_state:
2708 * @media: a #GstRTSPMedia
2709 * @state: the target state of the media
2710 * @transports: (element-type GstRtspServer.RTSPStreamTransport): a #GPtrArray
2711 * of #GstRTSPStreamTransport pointers
2713 * Set the state of @media to @state and for the transports in @transports.
2715 * @media must be prepared with gst_rtsp_media_prepare();
2717 * Returns: %TRUE on success.
2720 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2721 GPtrArray * transports)
2723 GstRTSPMediaPrivate *priv;
2725 gboolean activate, deactivate, do_state;
2728 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2729 g_return_val_if_fail (transports != NULL, FALSE);
2733 g_rec_mutex_lock (&priv->state_lock);
2734 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
2736 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2737 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2740 /* NULL and READY are the same */
2741 if (state == GST_STATE_READY)
2742 state = GST_STATE_NULL;
2744 activate = deactivate = FALSE;
2746 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
2750 case GST_STATE_NULL:
2751 case GST_STATE_PAUSED:
2752 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
2753 if (priv->target_state == GST_STATE_PLAYING)
2756 case GST_STATE_PLAYING:
2757 /* we're going to PLAYING, activate */
2763 old_active = priv->n_active;
2765 for (i = 0; i < transports->len; i++) {
2766 GstRTSPStreamTransport *trans;
2768 /* we need a non-NULL entry in the array */
2769 trans = g_ptr_array_index (transports, i);
2774 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
2776 } else if (deactivate) {
2777 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
2782 /* we just activated the first media, do the playing state change */
2783 if (old_active == 0 && activate)
2785 /* if we have no more active media, do the downward state changes */
2786 else if (priv->n_active == 0)
2791 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
2794 if (priv->target_state != state) {
2796 media_set_pipeline_state_locked (media, state);
2798 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
2802 /* remember where we are */
2803 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
2804 old_active != priv->n_active))
2805 collect_media_stats (media);
2807 g_rec_mutex_unlock (&priv->state_lock);
2814 GST_WARNING ("media %p was not prepared", media);
2815 g_rec_mutex_unlock (&priv->state_lock);
2820 GST_WARNING ("media %p in error status while changing to state %d",
2822 if (state == GST_STATE_NULL) {
2823 for (i = 0; i < transports->len; i++) {
2824 GstRTSPStreamTransport *trans;
2826 /* we need a non-NULL entry in the array */
2827 trans = g_ptr_array_index (transports, i);
2831 gst_rtsp_stream_transport_set_active (trans, FALSE);
2835 g_rec_mutex_unlock (&priv->state_lock);