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, target_state),
301 NULL, 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);
459 } DoQueryPositionData;
462 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
466 if (gst_rtsp_stream_query_position (stream, &tmp)) {
467 data->position = MAX (data->position, tmp);
473 default_query_position (GstRTSPMedia * media, gint64 * position)
475 GstRTSPMediaPrivate *priv;
476 DoQueryPositionData data;
483 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
485 *position = data.position;
497 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
501 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
502 data->stop = MAX (data->stop, tmp);
508 default_query_stop (GstRTSPMedia * media, gint64 * stop)
510 GstRTSPMediaPrivate *priv;
511 DoQueryStopData data;
518 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
526 default_create_rtpbin (GstRTSPMedia * media)
530 rtpbin = gst_element_factory_make ("rtpbin", NULL);
535 /* must be called with state lock */
537 collect_media_stats (GstRTSPMedia * media)
539 GstRTSPMediaPrivate *priv = media->priv;
540 gint64 position = 0, stop = -1;
542 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
543 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
546 priv->range.unit = GST_RTSP_RANGE_NPT;
548 GST_INFO ("collect media stats");
551 priv->range.min.type = GST_RTSP_TIME_NOW;
552 priv->range.min.seconds = -1;
553 priv->range_start = -1;
554 priv->range.max.type = GST_RTSP_TIME_END;
555 priv->range.max.seconds = -1;
556 priv->range_stop = -1;
558 GstRTSPMediaClass *klass;
561 klass = GST_RTSP_MEDIA_GET_CLASS (media);
563 /* get the position */
565 if (klass->query_position)
566 ret = klass->query_position (media, &position);
569 GST_INFO ("position query failed");
573 /* get the current segment stop */
575 if (klass->query_stop)
576 ret = klass->query_stop (media, &stop);
579 GST_INFO ("stop query failed");
583 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
584 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
586 if (position == -1) {
587 priv->range.min.type = GST_RTSP_TIME_NOW;
588 priv->range.min.seconds = -1;
589 priv->range_start = -1;
591 priv->range.min.type = GST_RTSP_TIME_SECONDS;
592 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
593 priv->range_start = position;
596 priv->range.max.type = GST_RTSP_TIME_END;
597 priv->range.max.seconds = -1;
598 priv->range_stop = -1;
600 priv->range.max.type = GST_RTSP_TIME_SECONDS;
601 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
602 priv->range_stop = stop;
608 * gst_rtsp_media_new:
609 * @element: (transfer full): a #GstElement
611 * Create a new #GstRTSPMedia instance. @element is the bin element that
612 * provides the different streams. The #GstRTSPMedia object contains the
613 * element to produce RTP data for one or more related (audio/video/..)
616 * Ownership is taken of @element.
618 * Returns: (transfer full): a new #GstRTSPMedia object.
621 gst_rtsp_media_new (GstElement * element)
623 GstRTSPMedia *result;
625 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
627 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
633 * gst_rtsp_media_get_element:
634 * @media: a #GstRTSPMedia
636 * Get the element that was used when constructing @media.
638 * Returns: (transfer full): a #GstElement. Unref after usage.
641 gst_rtsp_media_get_element (GstRTSPMedia * media)
643 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
645 return gst_object_ref (media->priv->element);
649 * gst_rtsp_media_take_pipeline:
650 * @media: a #GstRTSPMedia
651 * @pipeline: (transfer full): a #GstPipeline
653 * Set @pipeline as the #GstPipeline for @media. Ownership is
654 * taken of @pipeline.
657 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
659 GstRTSPMediaPrivate *priv;
661 GstNetTimeProvider *nettime;
663 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
664 g_return_if_fail (GST_IS_PIPELINE (pipeline));
668 g_mutex_lock (&priv->lock);
669 old = priv->pipeline;
670 priv->pipeline = GST_ELEMENT_CAST (pipeline);
671 nettime = priv->nettime;
672 priv->nettime = NULL;
673 g_mutex_unlock (&priv->lock);
676 gst_object_unref (old);
679 gst_object_unref (nettime);
681 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
685 * gst_rtsp_media_set_permissions:
686 * @media: a #GstRTSPMedia
687 * @permissions: (transfer none): a #GstRTSPPermissions
689 * Set @permissions on @media.
692 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
693 GstRTSPPermissions * permissions)
695 GstRTSPMediaPrivate *priv;
697 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
701 g_mutex_lock (&priv->lock);
702 if (priv->permissions)
703 gst_rtsp_permissions_unref (priv->permissions);
704 if ((priv->permissions = permissions))
705 gst_rtsp_permissions_ref (permissions);
706 g_mutex_unlock (&priv->lock);
710 * gst_rtsp_media_get_permissions:
711 * @media: a #GstRTSPMedia
713 * Get the permissions object from @media.
715 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
718 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
720 GstRTSPMediaPrivate *priv;
721 GstRTSPPermissions *result;
723 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
727 g_mutex_lock (&priv->lock);
728 if ((result = priv->permissions))
729 gst_rtsp_permissions_ref (result);
730 g_mutex_unlock (&priv->lock);
736 * gst_rtsp_media_set_suspend_mode:
737 * @media: a #GstRTSPMedia
738 * @mode: the new #GstRTSPSuspendMode
740 * Control how @ media will be suspended after the SDP has been generated and
741 * after a PAUSE request has been performed.
743 * Media must be unprepared when setting the suspend mode.
746 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
748 GstRTSPMediaPrivate *priv;
750 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
754 g_rec_mutex_lock (&priv->state_lock);
755 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
757 priv->suspend_mode = mode;
758 g_rec_mutex_unlock (&priv->state_lock);
765 GST_WARNING ("media %p was prepared", media);
766 g_rec_mutex_unlock (&priv->state_lock);
771 * gst_rtsp_media_get_suspend_mode:
772 * @media: a #GstRTSPMedia
774 * Get how @media will be suspended.
776 * Returns: #GstRTSPSuspendMode.
779 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
781 GstRTSPMediaPrivate *priv;
782 GstRTSPSuspendMode res;
784 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
788 g_rec_mutex_lock (&priv->state_lock);
789 res = priv->suspend_mode;
790 g_rec_mutex_unlock (&priv->state_lock);
796 * gst_rtsp_media_set_shared:
797 * @media: a #GstRTSPMedia
798 * @shared: the new value
800 * Set or unset if the pipeline for @media can be shared will multiple clients.
801 * When @shared is %TRUE, client requests for this media will share the media
805 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
807 GstRTSPMediaPrivate *priv;
809 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
813 g_mutex_lock (&priv->lock);
814 priv->shared = shared;
815 g_mutex_unlock (&priv->lock);
819 * gst_rtsp_media_is_shared:
820 * @media: a #GstRTSPMedia
822 * Check if the pipeline for @media can be shared between multiple clients.
824 * Returns: %TRUE if the media can be shared between clients.
827 gst_rtsp_media_is_shared (GstRTSPMedia * media)
829 GstRTSPMediaPrivate *priv;
832 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
836 g_mutex_lock (&priv->lock);
838 g_mutex_unlock (&priv->lock);
844 * gst_rtsp_media_set_reusable:
845 * @media: a #GstRTSPMedia
846 * @reusable: the new value
848 * Set or unset if the pipeline for @media can be reused after the pipeline has
852 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
854 GstRTSPMediaPrivate *priv;
856 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
860 g_mutex_lock (&priv->lock);
861 priv->reusable = reusable;
862 g_mutex_unlock (&priv->lock);
866 * gst_rtsp_media_is_reusable:
867 * @media: a #GstRTSPMedia
869 * Check if the pipeline for @media can be reused after an unprepare.
871 * Returns: %TRUE if the media can be reused
874 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
876 GstRTSPMediaPrivate *priv;
879 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
883 g_mutex_lock (&priv->lock);
884 res = priv->reusable;
885 g_mutex_unlock (&priv->lock);
891 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
893 gst_rtsp_stream_set_profiles (stream, *profiles);
897 * gst_rtsp_media_set_profiles:
898 * @media: a #GstRTSPMedia
899 * @profiles: the new flags
901 * Configure the allowed lower transport for @media.
904 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
906 GstRTSPMediaPrivate *priv;
908 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
912 g_mutex_lock (&priv->lock);
913 priv->profiles = profiles;
914 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
915 g_mutex_unlock (&priv->lock);
919 * gst_rtsp_media_get_profiles:
920 * @media: a #GstRTSPMedia
922 * Get the allowed profiles of @media.
924 * Returns: a #GstRTSPProfile
927 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
929 GstRTSPMediaPrivate *priv;
932 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
936 g_mutex_lock (&priv->lock);
937 res = priv->profiles;
938 g_mutex_unlock (&priv->lock);
944 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
946 gst_rtsp_stream_set_protocols (stream, *protocols);
950 * gst_rtsp_media_set_protocols:
951 * @media: a #GstRTSPMedia
952 * @protocols: the new flags
954 * Configure the allowed lower transport for @media.
957 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
959 GstRTSPMediaPrivate *priv;
961 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
965 g_mutex_lock (&priv->lock);
966 priv->protocols = protocols;
967 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
968 g_mutex_unlock (&priv->lock);
972 * gst_rtsp_media_get_protocols:
973 * @media: a #GstRTSPMedia
975 * Get the allowed protocols of @media.
977 * Returns: a #GstRTSPLowerTrans
980 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
982 GstRTSPMediaPrivate *priv;
983 GstRTSPLowerTrans res;
985 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
986 GST_RTSP_LOWER_TRANS_UNKNOWN);
990 g_mutex_lock (&priv->lock);
991 res = priv->protocols;
992 g_mutex_unlock (&priv->lock);
998 * gst_rtsp_media_set_eos_shutdown:
999 * @media: a #GstRTSPMedia
1000 * @eos_shutdown: the new value
1002 * Set or unset if an EOS event will be sent to the pipeline for @media before
1006 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1008 GstRTSPMediaPrivate *priv;
1010 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1014 g_mutex_lock (&priv->lock);
1015 priv->eos_shutdown = eos_shutdown;
1016 g_mutex_unlock (&priv->lock);
1020 * gst_rtsp_media_is_eos_shutdown:
1021 * @media: a #GstRTSPMedia
1023 * Check if the pipeline for @media will send an EOS down the pipeline before
1026 * Returns: %TRUE if the media will send EOS before unpreparing.
1029 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1031 GstRTSPMediaPrivate *priv;
1034 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1038 g_mutex_lock (&priv->lock);
1039 res = priv->eos_shutdown;
1040 g_mutex_unlock (&priv->lock);
1046 * gst_rtsp_media_set_buffer_size:
1047 * @media: a #GstRTSPMedia
1048 * @size: the new value
1050 * Set the kernel UDP buffer size.
1053 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1055 GstRTSPMediaPrivate *priv;
1057 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1059 GST_LOG_OBJECT (media, "set buffer size %u", size);
1063 g_mutex_lock (&priv->lock);
1064 priv->buffer_size = size;
1065 g_mutex_unlock (&priv->lock);
1069 * gst_rtsp_media_get_buffer_size:
1070 * @media: a #GstRTSPMedia
1072 * Get the kernel UDP buffer size.
1074 * Returns: the kernel UDP buffer size.
1077 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1079 GstRTSPMediaPrivate *priv;
1082 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1086 g_mutex_unlock (&priv->lock);
1087 res = priv->buffer_size;
1088 g_mutex_unlock (&priv->lock);
1094 * gst_rtsp_media_use_time_provider:
1095 * @media: a #GstRTSPMedia
1096 * @time_provider: if a #GstNetTimeProvider should be used
1098 * Set @media to provide a #GstNetTimeProvider.
1101 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1103 GstRTSPMediaPrivate *priv;
1105 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1109 g_mutex_lock (&priv->lock);
1110 priv->time_provider = time_provider;
1111 g_mutex_unlock (&priv->lock);
1115 * gst_rtsp_media_is_time_provider:
1116 * @media: a #GstRTSPMedia
1118 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1120 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1122 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1125 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1127 GstRTSPMediaPrivate *priv;
1130 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1134 g_mutex_unlock (&priv->lock);
1135 res = priv->time_provider;
1136 g_mutex_unlock (&priv->lock);
1142 * gst_rtsp_media_set_address_pool:
1143 * @media: a #GstRTSPMedia
1144 * @pool: (transfer none): a #GstRTSPAddressPool
1146 * configure @pool to be used as the address pool of @media.
1149 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1150 GstRTSPAddressPool * pool)
1152 GstRTSPMediaPrivate *priv;
1153 GstRTSPAddressPool *old;
1155 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1159 GST_LOG_OBJECT (media, "set address pool %p", pool);
1161 g_mutex_lock (&priv->lock);
1162 if ((old = priv->pool) != pool)
1163 priv->pool = pool ? g_object_ref (pool) : NULL;
1166 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1168 g_mutex_unlock (&priv->lock);
1171 g_object_unref (old);
1175 * gst_rtsp_media_get_address_pool:
1176 * @media: a #GstRTSPMedia
1178 * Get the #GstRTSPAddressPool used as the address pool of @media.
1180 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1183 GstRTSPAddressPool *
1184 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1186 GstRTSPMediaPrivate *priv;
1187 GstRTSPAddressPool *result;
1189 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1193 g_mutex_lock (&priv->lock);
1194 if ((result = priv->pool))
1195 g_object_ref (result);
1196 g_mutex_unlock (&priv->lock);
1202 * gst_rtsp_media_collect_streams:
1203 * @media: a #GstRTSPMedia
1205 * Find all payloader elements, they should be named pay\%d in the
1206 * element of @media, and create #GstRTSPStreams for them.
1208 * Collect all dynamic elements, named dynpay\%d, and add them to
1209 * the list of dynamic elements.
1212 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1214 GstRTSPMediaPrivate *priv;
1215 GstElement *element, *elem;
1220 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1223 element = priv->element;
1226 for (i = 0; have_elem; i++) {
1231 name = g_strdup_printf ("pay%d", i);
1232 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1233 GST_INFO ("found stream %d with payloader %p", i, elem);
1235 /* take the pad of the payloader */
1236 pad = gst_element_get_static_pad (elem, "src");
1237 /* create the stream */
1238 gst_rtsp_media_create_stream (media, elem, pad);
1239 gst_object_unref (pad);
1240 gst_object_unref (elem);
1246 name = g_strdup_printf ("dynpay%d", i);
1247 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1248 /* a stream that will dynamically create pads to provide RTP packets */
1249 GST_INFO ("found dynamic element %d, %p", i, elem);
1251 g_mutex_lock (&priv->lock);
1252 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1253 g_mutex_unlock (&priv->lock);
1262 * gst_rtsp_media_create_stream:
1263 * @media: a #GstRTSPMedia
1264 * @payloader: a #GstElement
1265 * @srcpad: a source #GstPad
1267 * Create a new stream in @media that provides RTP data on @srcpad.
1268 * @srcpad should be a pad of an element inside @media->element.
1270 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1274 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1277 GstRTSPMediaPrivate *priv;
1278 GstRTSPStream *stream;
1283 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1284 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1285 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1286 g_return_val_if_fail (GST_PAD_IS_SRC (pad), NULL);
1290 g_mutex_lock (&priv->lock);
1291 idx = priv->streams->len;
1293 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1295 name = g_strdup_printf ("src_%u", idx);
1296 srcpad = gst_ghost_pad_new (name, pad);
1297 gst_pad_set_active (srcpad, TRUE);
1298 gst_element_add_pad (priv->element, srcpad);
1301 stream = gst_rtsp_stream_new (idx, payloader, srcpad);
1303 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1304 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1305 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1307 g_ptr_array_add (priv->streams, stream);
1308 g_mutex_unlock (&priv->lock);
1310 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1317 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1319 GstRTSPMediaPrivate *priv;
1324 g_mutex_lock (&priv->lock);
1325 /* remove the ghostpad */
1326 srcpad = gst_rtsp_stream_get_srcpad (stream);
1327 gst_element_remove_pad (priv->element, srcpad);
1328 gst_object_unref (srcpad);
1329 /* now remove the stream */
1330 g_object_ref (stream);
1331 g_ptr_array_remove (priv->streams, stream);
1332 g_mutex_unlock (&priv->lock);
1334 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1337 g_object_unref (stream);
1341 * gst_rtsp_media_n_streams:
1342 * @media: a #GstRTSPMedia
1344 * Get the number of streams in this media.
1346 * Returns: The number of streams.
1349 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1351 GstRTSPMediaPrivate *priv;
1354 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1358 g_mutex_lock (&priv->lock);
1359 res = priv->streams->len;
1360 g_mutex_unlock (&priv->lock);
1366 * gst_rtsp_media_get_stream:
1367 * @media: a #GstRTSPMedia
1368 * @idx: the stream index
1370 * Retrieve the stream with index @idx from @media.
1372 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1373 * @idx or %NULL when a stream with that index did not exist.
1376 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1378 GstRTSPMediaPrivate *priv;
1381 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1385 g_mutex_lock (&priv->lock);
1386 if (idx < priv->streams->len)
1387 res = g_ptr_array_index (priv->streams, idx);
1390 g_mutex_unlock (&priv->lock);
1396 * gst_rtsp_media_find_stream:
1397 * @media: a #GstRTSPMedia
1398 * @control: the control of the stream
1400 * Find a stream in @media with @control as the control uri.
1402 * Returns: (nullable) (transfer none): the #GstRTSPStream with
1403 * control uri @control or %NULL when a stream with that control did
1407 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
1409 GstRTSPMediaPrivate *priv;
1413 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1414 g_return_val_if_fail (control != NULL, NULL);
1420 g_mutex_lock (&priv->lock);
1421 for (i = 0; i < priv->streams->len; i++) {
1422 GstRTSPStream *test;
1424 test = g_ptr_array_index (priv->streams, i);
1425 if (gst_rtsp_stream_has_control (test, control)) {
1430 g_mutex_unlock (&priv->lock);
1435 /* called with state-lock */
1437 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
1438 GstRTSPRangeUnit unit)
1440 return gst_rtsp_range_convert_units (range, unit);
1444 * gst_rtsp_media_get_range_string:
1445 * @media: a #GstRTSPMedia
1446 * @play: for the PLAY request
1447 * @unit: the unit to use for the string
1449 * Get the current range as a string. @media must be prepared with
1450 * gst_rtsp_media_prepare ().
1452 * Returns: (transfer full): The range as a string, g_free() after usage.
1455 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
1456 GstRTSPRangeUnit unit)
1458 GstRTSPMediaClass *klass;
1459 GstRTSPMediaPrivate *priv;
1461 GstRTSPTimeRange range;
1463 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1464 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1465 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1469 g_rec_mutex_lock (&priv->state_lock);
1470 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
1471 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
1474 g_mutex_lock (&priv->lock);
1476 /* Update the range value with current position/duration */
1477 collect_media_stats (media);
1480 range = priv->range;
1482 if (!play && priv->n_active > 0) {
1483 range.min.type = GST_RTSP_TIME_NOW;
1484 range.min.seconds = -1;
1486 g_mutex_unlock (&priv->lock);
1487 g_rec_mutex_unlock (&priv->state_lock);
1489 if (!klass->convert_range (media, &range, unit))
1490 goto conversion_failed;
1492 result = gst_rtsp_range_to_string (&range);
1499 GST_WARNING ("media %p was not prepared", media);
1500 g_rec_mutex_unlock (&priv->state_lock);
1505 GST_WARNING ("range conversion to unit %d failed", unit);
1511 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
1513 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
1517 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
1519 GstRTSPMediaPrivate *priv = media->priv;
1521 GST_DEBUG ("media %p set blocked %d", media, blocked);
1522 priv->blocked = blocked;
1523 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
1527 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
1529 GstRTSPMediaPrivate *priv = media->priv;
1531 g_mutex_lock (&priv->lock);
1532 priv->status = status;
1533 GST_DEBUG ("setting new status to %d", status);
1534 g_cond_broadcast (&priv->cond);
1535 g_mutex_unlock (&priv->lock);
1539 * gst_rtsp_media_get_status:
1540 * @media: a #GstRTSPMedia
1542 * Get the status of @media. When @media is busy preparing, this function waits
1543 * until @media is prepared or in error.
1545 * Returns: the status of @media.
1548 gst_rtsp_media_get_status (GstRTSPMedia * media)
1550 GstRTSPMediaPrivate *priv = media->priv;
1551 GstRTSPMediaStatus result;
1554 g_mutex_lock (&priv->lock);
1555 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
1556 /* while we are preparing, wait */
1557 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
1558 GST_DEBUG ("waiting for status change");
1559 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
1560 GST_DEBUG ("timeout, assuming error status");
1561 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
1564 /* could be success or error */
1565 result = priv->status;
1566 GST_DEBUG ("got status %d", result);
1567 g_mutex_unlock (&priv->lock);
1573 * gst_rtsp_media_seek:
1574 * @media: a #GstRTSPMedia
1575 * @range: (transfer none): a #GstRTSPTimeRange
1577 * Seek the pipeline of @media to @range. @media must be prepared with
1578 * gst_rtsp_media_prepare().
1580 * Returns: %TRUE on success.
1583 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
1585 GstRTSPMediaClass *klass;
1586 GstRTSPMediaPrivate *priv;
1588 GstClockTime start, stop;
1589 GstSeekType start_type, stop_type;
1592 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1594 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1595 g_return_val_if_fail (range != NULL, FALSE);
1596 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
1600 g_rec_mutex_lock (&priv->state_lock);
1601 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
1604 /* Update the seekable state of the pipeline in case it changed */
1605 query = gst_query_new_seeking (GST_FORMAT_TIME);
1606 if (gst_element_query (priv->pipeline, query)) {
1611 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
1612 priv->seekable = seekable;
1614 gst_query_unref (query);
1616 if (!priv->seekable)
1619 start_type = stop_type = GST_SEEK_TYPE_NONE;
1621 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
1623 gst_rtsp_range_get_times (range, &start, &stop);
1625 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1626 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1627 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1628 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
1630 if (start != GST_CLOCK_TIME_NONE)
1631 start_type = GST_SEEK_TYPE_SET;
1633 if (priv->range_stop == stop)
1634 stop = GST_CLOCK_TIME_NONE;
1635 else if (stop != GST_CLOCK_TIME_NONE)
1636 stop_type = GST_SEEK_TYPE_SET;
1638 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
1641 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
1642 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
1644 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
1646 media_streams_set_blocked (media, TRUE);
1648 /* depends on the current playing state of the pipeline. We might need to
1649 * queue this until we get EOS. */
1650 flags = GST_SEEK_FLAG_FLUSH;
1652 /* if range start was not supplied we must continue from current position.
1653 * but since we're doing a flushing seek, let us query the current position
1654 * so we end up at exactly the same position after the seek. */
1655 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
1657 gboolean ret = FALSE;
1659 if (klass->query_position)
1660 ret = klass->query_position (media, &position);
1663 GST_WARNING ("position query failed");
1665 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
1666 GST_TIME_ARGS (position));
1668 start_type = GST_SEEK_TYPE_SET;
1669 flags |= GST_SEEK_FLAG_ACCURATE;
1672 /* only set keyframe flag when modifying start */
1673 if (start_type != GST_SEEK_TYPE_NONE)
1674 flags |= GST_SEEK_FLAG_KEY_UNIT;
1677 /* FIXME, we only do forwards playback, no trick modes yet */
1678 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
1679 flags, start_type, start, stop_type, stop);
1681 /* and block for the seek to complete */
1682 GST_INFO ("done seeking %d", res);
1683 g_rec_mutex_unlock (&priv->state_lock);
1685 /* wait until pipeline is prerolled again, this will also collect stats */
1686 if (!wait_preroll (media))
1687 goto preroll_failed;
1689 g_rec_mutex_lock (&priv->state_lock);
1690 GST_INFO ("prerolled again");
1692 GST_INFO ("no seek needed");
1695 g_rec_mutex_unlock (&priv->state_lock);
1702 g_rec_mutex_unlock (&priv->state_lock);
1703 GST_INFO ("media %p is not prepared", media);
1708 g_rec_mutex_unlock (&priv->state_lock);
1709 GST_INFO ("pipeline is not seekable");
1714 g_rec_mutex_unlock (&priv->state_lock);
1715 GST_WARNING ("conversion to npt not supported");
1720 GST_WARNING ("failed to preroll after seek");
1726 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
1728 *blocked &= gst_rtsp_stream_is_blocking (stream);
1732 media_streams_blocking (GstRTSPMedia * media)
1734 gboolean blocking = TRUE;
1736 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
1742 static GstStateChangeReturn
1743 set_state (GstRTSPMedia * media, GstState state)
1745 GstRTSPMediaPrivate *priv = media->priv;
1746 GstStateChangeReturn ret;
1748 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
1750 ret = gst_element_set_state (priv->pipeline, state);
1755 static GstStateChangeReturn
1756 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
1758 GstRTSPMediaPrivate *priv = media->priv;
1759 GstStateChangeReturn ret;
1761 GST_INFO ("set target state to %s for media %p",
1762 gst_element_state_get_name (state), media);
1763 priv->target_state = state;
1765 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
1766 priv->target_state, NULL);
1769 ret = set_state (media, state);
1771 ret = GST_STATE_CHANGE_SUCCESS;
1776 /* called with state-lock */
1778 default_handle_message (GstRTSPMedia * media, GstMessage * message)
1780 GstRTSPMediaPrivate *priv = media->priv;
1781 GstMessageType type;
1783 type = GST_MESSAGE_TYPE (message);
1786 case GST_MESSAGE_STATE_CHANGED:
1788 case GST_MESSAGE_BUFFERING:
1792 gst_message_parse_buffering (message, &percent);
1794 /* no state management needed for live pipelines */
1798 if (percent == 100) {
1799 /* a 100% message means buffering is done */
1800 priv->buffering = FALSE;
1801 /* if the desired state is playing, go back */
1802 if (priv->target_state == GST_STATE_PLAYING) {
1803 GST_INFO ("Buffering done, setting pipeline to PLAYING");
1804 set_state (media, GST_STATE_PLAYING);
1806 GST_INFO ("Buffering done");
1809 /* buffering busy */
1810 if (priv->buffering == FALSE) {
1811 if (priv->target_state == GST_STATE_PLAYING) {
1812 /* we were not buffering but PLAYING, PAUSE the pipeline. */
1813 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
1814 set_state (media, GST_STATE_PAUSED);
1816 GST_INFO ("Buffering ...");
1819 priv->buffering = TRUE;
1823 case GST_MESSAGE_LATENCY:
1825 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
1828 case GST_MESSAGE_ERROR:
1833 gst_message_parse_error (message, &gerror, &debug);
1834 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
1835 g_error_free (gerror);
1838 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
1841 case GST_MESSAGE_WARNING:
1846 gst_message_parse_warning (message, &gerror, &debug);
1847 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
1848 g_error_free (gerror);
1852 case GST_MESSAGE_ELEMENT:
1854 const GstStructure *s;
1856 s = gst_message_get_structure (message);
1857 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
1858 GST_DEBUG ("media received blocking message");
1859 if (priv->blocked && media_streams_blocking (media)) {
1860 GST_DEBUG ("media is blocking");
1861 collect_media_stats (media);
1863 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1864 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1869 case GST_MESSAGE_STREAM_STATUS:
1871 case GST_MESSAGE_ASYNC_DONE:
1873 /* when we are dynamically adding pads, the addition of the udpsrc will
1874 * temporarily produce ASYNC_DONE messages. We have to ignore them and
1875 * wait for the final ASYNC_DONE after everything prerolled */
1876 GST_INFO ("%p: ignoring ASYNC_DONE", media);
1878 GST_INFO ("%p: got ASYNC_DONE", media);
1879 collect_media_stats (media);
1881 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
1882 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
1885 case GST_MESSAGE_EOS:
1886 GST_INFO ("%p: got EOS", media);
1888 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
1889 GST_DEBUG ("shutting down after EOS");
1890 finish_unprepare (media);
1894 GST_INFO ("%p: got message type %d (%s)", media, type,
1895 gst_message_type_get_name (type));
1902 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
1904 GstRTSPMediaPrivate *priv = media->priv;
1905 GstRTSPMediaClass *klass;
1908 klass = GST_RTSP_MEDIA_GET_CLASS (media);
1910 g_rec_mutex_lock (&priv->state_lock);
1911 if (klass->handle_message)
1912 ret = klass->handle_message (media, message);
1915 g_rec_mutex_unlock (&priv->state_lock);
1921 watch_destroyed (GstRTSPMedia * media)
1923 GST_DEBUG_OBJECT (media, "source destroyed");
1924 g_object_unref (media);
1928 find_payload_element (GstElement * payloader)
1930 GstElement *pay = NULL;
1932 if (GST_IS_BIN (payloader)) {
1934 GValue item = { 0 };
1936 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
1937 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
1938 GstElement *element = (GstElement *) g_value_get_object (&item);
1939 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
1943 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
1947 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
1948 pay = gst_object_ref (element);
1949 g_value_unset (&item);
1952 g_value_unset (&item);
1954 gst_iterator_free (iter);
1956 pay = g_object_ref (payloader);
1962 /* called from streaming threads */
1964 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
1966 GstRTSPMediaPrivate *priv = media->priv;
1967 GstRTSPStream *stream;
1970 /* find the real payload element */
1971 pay = find_payload_element (element);
1972 stream = gst_rtsp_media_create_stream (media, pay, pad);
1973 gst_object_unref (pay);
1975 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
1977 g_rec_mutex_lock (&priv->state_lock);
1978 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
1981 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
1983 /* we will be adding elements below that will cause ASYNC_DONE to be
1984 * posted in the bus. We want to ignore those messages until the
1985 * pipeline really prerolled. */
1986 priv->adding = TRUE;
1988 /* join the element in the PAUSED state because this callback is
1989 * called from the streaming thread and it is PAUSED */
1990 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
1991 priv->rtpbin, GST_STATE_PAUSED)) {
1992 GST_WARNING ("failed to join bin element");
1995 priv->adding = FALSE;
1996 g_rec_mutex_unlock (&priv->state_lock);
2003 gst_rtsp_media_remove_stream (media, stream);
2004 g_rec_mutex_unlock (&priv->state_lock);
2005 GST_INFO ("ignore pad because we are not preparing");
2011 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2013 GstRTSPMediaPrivate *priv = media->priv;
2014 GstRTSPStream *stream;
2016 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2020 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2022 g_rec_mutex_lock (&priv->state_lock);
2023 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2024 g_rec_mutex_unlock (&priv->state_lock);
2026 gst_rtsp_media_remove_stream (media, stream);
2030 remove_fakesink (GstRTSPMediaPrivate * priv)
2032 GstElement *fakesink;
2034 g_mutex_lock (&priv->lock);
2035 if ((fakesink = priv->fakesink))
2036 gst_object_ref (fakesink);
2037 priv->fakesink = NULL;
2038 g_mutex_unlock (&priv->lock);
2041 gst_bin_remove (GST_BIN (priv->pipeline), fakesink);
2042 gst_element_set_state (fakesink, GST_STATE_NULL);
2043 gst_object_unref (fakesink);
2044 GST_INFO ("removed fakesink");
2049 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2051 GstRTSPMediaPrivate *priv = media->priv;
2053 GST_INFO ("no more pads");
2054 remove_fakesink (priv);
2057 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2059 struct _DynPaySignalHandlers
2061 gulong pad_added_handler;
2062 gulong pad_removed_handler;
2063 gulong no_more_pads_handler;
2067 start_preroll (GstRTSPMedia * media)
2069 GstRTSPMediaPrivate *priv = media->priv;
2070 GstStateChangeReturn ret;
2072 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2073 /* first go to PAUSED */
2074 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2077 case GST_STATE_CHANGE_SUCCESS:
2078 GST_INFO ("SUCCESS state change for media %p", media);
2079 priv->seekable = TRUE;
2081 case GST_STATE_CHANGE_ASYNC:
2082 GST_INFO ("ASYNC state change for media %p", media);
2083 priv->seekable = TRUE;
2085 case GST_STATE_CHANGE_NO_PREROLL:
2086 /* we need to go to PLAYING */
2087 GST_INFO ("NO_PREROLL state change: live media %p", media);
2088 /* FIXME we disable seeking for live streams for now. We should perform a
2089 * seeking query in preroll instead */
2090 priv->seekable = FALSE;
2091 priv->is_live = TRUE;
2092 /* start blocked to make sure nothing goes to the sink */
2093 media_streams_set_blocked (media, TRUE);
2094 ret = set_state (media, GST_STATE_PLAYING);
2095 if (ret == GST_STATE_CHANGE_FAILURE)
2098 case GST_STATE_CHANGE_FAILURE:
2106 GST_WARNING ("failed to preroll pipeline");
2112 wait_preroll (GstRTSPMedia * media)
2114 GstRTSPMediaStatus status;
2116 GST_DEBUG ("wait to preroll pipeline");
2118 /* wait until pipeline is prerolled */
2119 status = gst_rtsp_media_get_status (media);
2120 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2121 goto preroll_failed;
2127 GST_WARNING ("failed to preroll pipeline");
2133 start_prepare (GstRTSPMedia * media)
2135 GstRTSPMediaPrivate *priv = media->priv;
2139 /* link streams we already have, other streams might appear when we have
2140 * dynamic elements */
2141 for (i = 0; i < priv->streams->len; i++) {
2142 GstRTSPStream *stream;
2144 stream = g_ptr_array_index (priv->streams, i);
2146 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2147 priv->rtpbin, GST_STATE_NULL)) {
2148 goto join_bin_failed;
2152 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2153 GstElement *elem = walk->data;
2154 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2156 GST_INFO ("adding callbacks for dynamic element %p", elem);
2158 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2159 (GCallback) pad_added_cb, media);
2160 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2161 (GCallback) pad_removed_cb, media);
2162 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2163 (GCallback) no_more_pads_cb, media);
2165 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2167 /* we add a fakesink here in order to make the state change async. We remove
2168 * the fakesink again in the no-more-pads callback. */
2169 priv->fakesink = gst_element_factory_make ("fakesink", "fakesink");
2170 gst_bin_add (GST_BIN (priv->pipeline), priv->fakesink);
2173 if (!start_preroll (media))
2174 goto preroll_failed;
2180 GST_WARNING ("failed to join bin element");
2181 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2186 GST_WARNING ("failed to preroll pipeline");
2187 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2193 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2195 GstRTSPMediaPrivate *priv;
2196 GstRTSPMediaClass *klass;
2198 GMainContext *context;
2203 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2205 if (!klass->create_rtpbin)
2206 goto no_create_rtpbin;
2208 priv->rtpbin = klass->create_rtpbin (media);
2209 if (priv->rtpbin != NULL) {
2210 gboolean success = TRUE;
2212 if (klass->setup_rtpbin)
2213 success = klass->setup_rtpbin (media, priv->rtpbin);
2215 if (success == FALSE) {
2216 gst_object_unref (priv->rtpbin);
2217 priv->rtpbin = NULL;
2220 if (priv->rtpbin == NULL)
2223 priv->thread = thread;
2224 context = (thread != NULL) ? (thread->context) : NULL;
2226 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2228 /* add the pipeline bus to our custom mainloop */
2229 priv->source = gst_bus_create_watch (bus);
2230 gst_object_unref (bus);
2232 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2233 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2235 priv->id = g_source_attach (priv->source, context);
2237 /* add stuff to the bin */
2238 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2240 /* do remainder in context */
2241 source = g_idle_source_new ();
2242 g_source_set_callback (source, (GSourceFunc) start_prepare, media, NULL);
2243 g_source_attach (source, context);
2244 g_source_unref (source);
2251 GST_ERROR ("no create_rtpbin function");
2252 g_critical ("no create_rtpbin vmethod function set");
2257 GST_WARNING ("no rtpbin element");
2258 g_warning ("failed to create element 'rtpbin', check your installation");
2264 * gst_rtsp_media_prepare:
2265 * @media: a #GstRTSPMedia
2266 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2267 * bus handler or %NULL
2269 * Prepare @media for streaming. This function will create the objects
2270 * to manage the streaming. A pipeline must have been set on @media with
2271 * gst_rtsp_media_take_pipeline().
2273 * It will preroll the pipeline and collect vital information about the streams
2274 * such as the duration.
2276 * Returns: %TRUE on success.
2279 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2281 GstRTSPMediaPrivate *priv;
2282 GstRTSPMediaClass *klass;
2284 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2288 g_rec_mutex_lock (&priv->state_lock);
2289 priv->prepare_count++;
2291 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
2292 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
2295 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2298 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
2299 goto not_unprepared;
2301 if (!priv->reusable && priv->reused)
2304 GST_INFO ("preparing media %p", media);
2306 /* reset some variables */
2307 priv->is_live = FALSE;
2308 priv->seekable = FALSE;
2309 priv->buffering = FALSE;
2311 /* we're preparing now */
2312 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2314 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2315 if (klass->prepare) {
2316 if (!klass->prepare (media, thread))
2317 goto prepare_failed;
2321 g_rec_mutex_unlock (&priv->state_lock);
2323 /* now wait for all pads to be prerolled, FIXME, we should somehow be
2324 * able to do this async so that we don't block the server thread. */
2325 if (!wait_preroll (media))
2326 goto preroll_failed;
2328 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
2330 GST_INFO ("object %p is prerolled", media);
2337 /* we are not going to use the giving thread, so stop it. */
2339 gst_rtsp_thread_stop (thread);
2344 GST_LOG ("media %p was prepared", media);
2345 /* we are not going to use the giving thread, so stop it. */
2347 gst_rtsp_thread_stop (thread);
2348 g_rec_mutex_unlock (&priv->state_lock);
2354 /* we are not going to use the giving thread, so stop it. */
2356 gst_rtsp_thread_stop (thread);
2357 GST_WARNING ("media %p was not unprepared", media);
2358 priv->prepare_count--;
2359 g_rec_mutex_unlock (&priv->state_lock);
2364 /* we are not going to use the giving thread, so stop it. */
2366 gst_rtsp_thread_stop (thread);
2367 priv->prepare_count--;
2368 g_rec_mutex_unlock (&priv->state_lock);
2369 GST_WARNING ("can not reuse media %p", media);
2374 /* we are not going to use the giving thread, so stop it. */
2376 gst_rtsp_thread_stop (thread);
2377 priv->prepare_count--;
2378 g_rec_mutex_unlock (&priv->state_lock);
2379 GST_ERROR ("failed to prepare media");
2384 GST_WARNING ("failed to preroll pipeline");
2385 gst_rtsp_media_unprepare (media);
2390 /* must be called with state-lock */
2392 finish_unprepare (GstRTSPMedia * media)
2394 GstRTSPMediaPrivate *priv = media->priv;
2398 GST_DEBUG ("shutting down");
2400 /* release the lock on shutdown, otherwise pad_added_cb might try to
2401 * acquire the lock and then we deadlock */
2402 g_rec_mutex_unlock (&priv->state_lock);
2403 set_state (media, GST_STATE_NULL);
2404 g_rec_mutex_lock (&priv->state_lock);
2405 remove_fakesink (priv);
2407 for (i = 0; i < priv->streams->len; i++) {
2408 GstRTSPStream *stream;
2410 GST_INFO ("Removing elements of stream %d from pipeline", i);
2412 stream = g_ptr_array_index (priv->streams, i);
2414 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2417 /* remove the pad signal handlers */
2418 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2419 GstElement *elem = walk->data;
2420 DynPaySignalHandlers *handlers;
2423 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
2424 g_assert (handlers != NULL);
2426 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
2427 g_signal_handler_disconnect (G_OBJECT (elem),
2428 handlers->pad_removed_handler);
2429 g_signal_handler_disconnect (G_OBJECT (elem),
2430 handlers->no_more_pads_handler);
2432 g_slice_free (DynPaySignalHandlers, handlers);
2435 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
2436 priv->rtpbin = NULL;
2439 gst_object_unref (priv->nettime);
2440 priv->nettime = NULL;
2442 priv->reused = TRUE;
2443 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
2445 /* when the media is not reusable, this will effectively unref the media and
2447 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
2449 /* the source has the last ref to the media */
2451 GST_DEBUG ("destroy source");
2452 g_source_destroy (priv->source);
2453 g_source_unref (priv->source);
2456 GST_DEBUG ("stop thread");
2457 gst_rtsp_thread_stop (priv->thread);
2461 /* called with state-lock */
2463 default_unprepare (GstRTSPMedia * media)
2465 GstRTSPMediaPrivate *priv = media->priv;
2467 if (priv->eos_shutdown) {
2468 GST_DEBUG ("sending EOS for shutdown");
2469 /* ref so that we don't disappear */
2470 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
2471 /* we need to go to playing again for the EOS to propagate, normally in this
2472 * state, nothing is receiving data from us anymore so this is ok. */
2473 set_state (media, GST_STATE_PLAYING);
2475 finish_unprepare (media);
2481 * gst_rtsp_media_unprepare:
2482 * @media: a #GstRTSPMedia
2484 * Unprepare @media. After this call, the media should be prepared again before
2485 * it can be used again. If the media is set to be non-reusable, a new instance
2488 * Returns: %TRUE on success.
2491 gst_rtsp_media_unprepare (GstRTSPMedia * media)
2493 GstRTSPMediaPrivate *priv;
2496 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2500 g_rec_mutex_lock (&priv->state_lock);
2501 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
2502 goto was_unprepared;
2504 priv->prepare_count--;
2505 if (priv->prepare_count > 0)
2508 GST_INFO ("unprepare media %p", media);
2510 media_streams_set_blocked (media, FALSE);
2511 set_target_state (media, GST_STATE_NULL, FALSE);
2514 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
2516 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
2517 GstRTSPMediaClass *klass;
2519 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2520 if (klass->unprepare)
2521 success = klass->unprepare (media);
2523 finish_unprepare (media);
2525 g_rec_mutex_unlock (&priv->state_lock);
2531 g_rec_mutex_unlock (&priv->state_lock);
2532 GST_INFO ("media %p was already unprepared", media);
2537 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
2538 g_rec_mutex_unlock (&priv->state_lock);
2543 /* should be called with state-lock */
2545 get_clock_unlocked (GstRTSPMedia * media)
2547 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
2548 GST_DEBUG_OBJECT (media, "media was not prepared");
2551 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
2555 * gst_rtsp_media_get_clock:
2556 * @media: a #GstRTSPMedia
2558 * Get the clock that is used by the pipeline in @media.
2560 * @media must be prepared before this method returns a valid clock object.
2562 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
2565 gst_rtsp_media_get_clock (GstRTSPMedia * media)
2568 GstRTSPMediaPrivate *priv;
2570 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2574 g_rec_mutex_lock (&priv->state_lock);
2575 clock = get_clock_unlocked (media);
2576 g_rec_mutex_unlock (&priv->state_lock);
2582 * gst_rtsp_media_get_base_time:
2583 * @media: a #GstRTSPMedia
2585 * Get the base_time that is used by the pipeline in @media.
2587 * @media must be prepared before this method returns a valid base_time.
2589 * Returns: the base_time used by @media.
2592 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
2594 GstClockTime result;
2595 GstRTSPMediaPrivate *priv;
2597 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
2601 g_rec_mutex_lock (&priv->state_lock);
2602 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2605 result = gst_element_get_base_time (media->priv->pipeline);
2606 g_rec_mutex_unlock (&priv->state_lock);
2613 g_rec_mutex_unlock (&priv->state_lock);
2614 GST_DEBUG_OBJECT (media, "media was not prepared");
2615 return GST_CLOCK_TIME_NONE;
2620 * gst_rtsp_media_get_time_provider:
2621 * @media: a #GstRTSPMedia
2622 * @address: (allow-none): an address or %NULL
2623 * @port: a port or 0
2625 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
2626 * will listen on @address and @port for client time requests.
2628 * Returns: (transfer full): the #GstNetTimeProvider of @media.
2630 GstNetTimeProvider *
2631 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
2634 GstRTSPMediaPrivate *priv;
2635 GstNetTimeProvider *provider = NULL;
2637 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2641 g_rec_mutex_lock (&priv->state_lock);
2642 if (priv->time_provider) {
2643 if ((provider = priv->nettime) == NULL) {
2646 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
2647 provider = gst_net_time_provider_new (clock, address, port);
2648 gst_object_unref (clock);
2650 priv->nettime = provider;
2654 g_rec_mutex_unlock (&priv->state_lock);
2657 gst_object_ref (provider);
2663 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
2665 return gst_rtsp_sdp_from_media (sdp, info, media);
2669 * gst_rtsp_media_setup_sdp:
2670 * @media: a #GstRTSPMedia
2671 * @sdp: (transfer none): a #GstSDPMessage
2672 * @info: (transfer none): a #GstSDPInfo
2674 * Add @media specific info to @sdp. @info is used to configure the connection
2675 * information in the SDP.
2677 * Returns: TRUE on success.
2680 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
2683 GstRTSPMediaPrivate *priv;
2684 GstRTSPMediaClass *klass;
2687 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2688 g_return_val_if_fail (sdp != NULL, FALSE);
2689 g_return_val_if_fail (info != NULL, FALSE);
2693 g_rec_mutex_lock (&priv->state_lock);
2695 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2697 if (!klass->setup_sdp)
2700 res = klass->setup_sdp (media, sdp, info);
2702 g_rec_mutex_unlock (&priv->state_lock);
2709 g_rec_mutex_unlock (&priv->state_lock);
2710 GST_ERROR ("no setup_sdp function");
2711 g_critical ("no setup_sdp vmethod function set");
2717 do_set_seqnum (GstRTSPStream * stream)
2720 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
2721 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
2724 /* call with state_lock */
2726 default_suspend (GstRTSPMedia * media)
2728 GstRTSPMediaPrivate *priv = media->priv;
2729 GstStateChangeReturn ret;
2731 switch (priv->suspend_mode) {
2732 case GST_RTSP_SUSPEND_MODE_NONE:
2733 GST_DEBUG ("media %p no suspend", media);
2735 case GST_RTSP_SUSPEND_MODE_PAUSE:
2736 GST_DEBUG ("media %p suspend to PAUSED", media);
2737 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2738 if (ret == GST_STATE_CHANGE_FAILURE)
2741 case GST_RTSP_SUSPEND_MODE_RESET:
2742 GST_DEBUG ("media %p suspend to NULL", media);
2743 ret = set_target_state (media, GST_STATE_NULL, TRUE);
2744 if (ret == GST_STATE_CHANGE_FAILURE)
2746 /* Because payloader needs to set the sequence number as
2747 * monotonic, we need to preserve the sequence number
2748 * after pause. (otherwise going from pause to play, which
2749 * is actually from NULL to PLAY will create a new sequence
2751 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
2757 /* let the streams do the state changes freely, if any */
2758 media_streams_set_blocked (media, FALSE);
2765 GST_WARNING ("failed changing pipeline's state for media %p", media);
2771 * gst_rtsp_media_suspend:
2772 * @media: a #GstRTSPMedia
2774 * Suspend @media. The state of the pipeline managed by @media is set to
2775 * GST_STATE_NULL but all streams are kept. @media can be prepared again
2776 * with gst_rtsp_media_unsuspend()
2778 * @media must be prepared with gst_rtsp_media_prepare();
2780 * Returns: %TRUE on success.
2783 gst_rtsp_media_suspend (GstRTSPMedia * media)
2785 GstRTSPMediaPrivate *priv = media->priv;
2786 GstRTSPMediaClass *klass;
2788 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2790 GST_FIXME ("suspend for dynamic pipelines needs fixing");
2792 g_rec_mutex_lock (&priv->state_lock);
2793 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2796 /* don't attempt to suspend when something is busy */
2797 if (priv->n_active > 0)
2800 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2801 if (klass->suspend) {
2802 if (!klass->suspend (media))
2803 goto suspend_failed;
2806 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
2808 g_rec_mutex_unlock (&priv->state_lock);
2815 g_rec_mutex_unlock (&priv->state_lock);
2816 GST_WARNING ("media %p was not prepared", media);
2821 g_rec_mutex_unlock (&priv->state_lock);
2822 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2823 GST_WARNING ("failed to suspend media %p", media);
2828 /* call with state_lock */
2830 default_unsuspend (GstRTSPMedia * media)
2832 GstRTSPMediaPrivate *priv = media->priv;
2834 switch (priv->suspend_mode) {
2835 case GST_RTSP_SUSPEND_MODE_NONE:
2836 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2838 case GST_RTSP_SUSPEND_MODE_PAUSE:
2839 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2841 case GST_RTSP_SUSPEND_MODE_RESET:
2843 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2844 if (!start_preroll (media))
2846 g_rec_mutex_unlock (&priv->state_lock);
2848 if (!wait_preroll (media))
2849 goto preroll_failed;
2851 g_rec_mutex_lock (&priv->state_lock);
2862 GST_WARNING ("failed to preroll pipeline");
2867 GST_WARNING ("failed to preroll pipeline");
2873 * gst_rtsp_media_unsuspend:
2874 * @media: a #GstRTSPMedia
2876 * Unsuspend @media if it was in a suspended state. This method does nothing
2877 * when the media was not in the suspended state.
2879 * Returns: %TRUE on success.
2882 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
2884 GstRTSPMediaPrivate *priv = media->priv;
2885 GstRTSPMediaClass *klass;
2887 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2889 g_rec_mutex_lock (&priv->state_lock);
2890 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2893 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2894 if (klass->unsuspend) {
2895 if (!klass->unsuspend (media))
2896 goto unsuspend_failed;
2900 g_rec_mutex_unlock (&priv->state_lock);
2907 g_rec_mutex_unlock (&priv->state_lock);
2908 GST_WARNING ("failed to unsuspend media %p", media);
2909 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2914 /* must be called with state-lock */
2916 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
2918 GstRTSPMediaPrivate *priv = media->priv;
2920 if (state == GST_STATE_NULL) {
2921 gst_rtsp_media_unprepare (media);
2923 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
2924 set_target_state (media, state, FALSE);
2925 /* when we are buffering, don't update the state yet, this will be done
2926 * when buffering finishes */
2927 if (priv->buffering) {
2928 GST_INFO ("Buffering busy, delay state change");
2930 if (state == GST_STATE_PLAYING)
2931 /* make sure pads are not blocking anymore when going to PLAYING */
2932 media_streams_set_blocked (media, FALSE);
2934 set_state (media, state);
2936 /* and suspend after pause */
2937 if (state == GST_STATE_PAUSED)
2938 gst_rtsp_media_suspend (media);
2944 * gst_rtsp_media_set_pipeline_state:
2945 * @media: a #GstRTSPMedia
2946 * @state: the target state of the pipeline
2948 * Set the state of the pipeline managed by @media to @state
2951 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
2953 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2955 g_rec_mutex_lock (&media->priv->state_lock);
2956 media_set_pipeline_state_locked (media, state);
2957 g_rec_mutex_unlock (&media->priv->state_lock);
2961 * gst_rtsp_media_set_state:
2962 * @media: a #GstRTSPMedia
2963 * @state: the target state of the media
2964 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
2965 * a #GPtrArray of #GstRTSPStreamTransport pointers
2967 * Set the state of @media to @state and for the transports in @transports.
2969 * @media must be prepared with gst_rtsp_media_prepare();
2971 * Returns: %TRUE on success.
2974 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
2975 GPtrArray * transports)
2977 GstRTSPMediaPrivate *priv;
2979 gboolean activate, deactivate, do_state;
2982 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2983 g_return_val_if_fail (transports != NULL, FALSE);
2987 g_rec_mutex_lock (&priv->state_lock);
2988 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
2990 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2991 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2994 /* NULL and READY are the same */
2995 if (state == GST_STATE_READY)
2996 state = GST_STATE_NULL;
2998 activate = deactivate = FALSE;
3000 GST_INFO ("going to state %s media %p", gst_element_state_get_name (state),
3004 case GST_STATE_NULL:
3005 case GST_STATE_PAUSED:
3006 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
3007 if (priv->target_state == GST_STATE_PLAYING)
3010 case GST_STATE_PLAYING:
3011 /* we're going to PLAYING, activate */
3017 old_active = priv->n_active;
3019 for (i = 0; i < transports->len; i++) {
3020 GstRTSPStreamTransport *trans;
3022 /* we need a non-NULL entry in the array */
3023 trans = g_ptr_array_index (transports, i);
3028 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
3030 } else if (deactivate) {
3031 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
3036 /* we just activated the first media, do the playing state change */
3037 if (old_active == 0 && activate)
3039 /* if we have no more active media, do the downward state changes */
3040 else if (priv->n_active == 0)
3045 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
3048 if (priv->target_state != state) {
3050 media_set_pipeline_state_locked (media, state);
3052 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
3056 /* remember where we are */
3057 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
3058 old_active != priv->n_active))
3059 collect_media_stats (media);
3061 g_rec_mutex_unlock (&priv->state_lock);
3068 GST_WARNING ("media %p was not prepared", media);
3069 g_rec_mutex_unlock (&priv->state_lock);
3074 GST_WARNING ("media %p in error status while changing to state %d",
3076 if (state == GST_STATE_NULL) {
3077 for (i = 0; i < transports->len; i++) {
3078 GstRTSPStreamTransport *trans;
3080 /* we need a non-NULL entry in the array */
3081 trans = g_ptr_array_index (transports, i);
3085 gst_rtsp_stream_transport_set_active (trans, FALSE);
3089 g_rec_mutex_unlock (&priv->state_lock);