2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: The media pipeline
24 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
25 * #GstRTSPSessionMedia
27 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
28 * streaming to the clients. The actual data transfer is done by the
29 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
31 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
32 * client does a DESCRIBE or SETUP of a resource.
34 * A media is created with gst_rtsp_media_new() that takes the element that will
35 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
36 * object needs to be made with the gst_rtsp_media_create_stream() which takes
37 * the payloader element and the source pad that produces the RTP stream.
39 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
40 * prepare method will add rtpbin and sinks and sources to send and receive RTP
41 * and RTCP packets from the clients. Each stream srcpad is connected to an
42 * input into the internal rtpbin.
44 * It is also possible to dynamically create #GstRTSPStream objects during the
45 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
48 * After the media is prepared, it is ready for streaming. It will usually be
49 * managed in a session with gst_rtsp_session_manage_media(). See
50 * #GstRTSPSession and #GstRTSPSessionMedia.
52 * The state of the media can be controlled with gst_rtsp_media_set_state ().
53 * Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
54 * or gst_rtsp_media_seek_trickmode() for finer control of the seek.
56 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
57 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
60 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
61 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
62 * can be prepared again after an unprepare.
64 * Last reviewed on 2013-07-11 (1.0.0)
74 #include <gst/app/gstappsrc.h>
75 #include <gst/app/gstappsink.h>
77 #include <gst/sdp/gstmikey.h>
78 #include <gst/rtp/gstrtppayloads.h>
80 #define AES_128_KEY_LEN 16
81 #define AES_256_KEY_LEN 32
83 #define HMAC_32_KEY_LEN 4
84 #define HMAC_80_KEY_LEN 10
86 #include "rtsp-media.h"
88 struct _GstRTSPMediaPrivate
93 /* protected by lock */
94 GstRTSPPermissions *permissions;
96 gboolean suspend_mode;
98 GstRTSPProfile profiles;
99 GstRTSPLowerTrans protocols;
101 gboolean eos_shutdown;
103 GstRTSPAddressPool *pool;
104 gchar *multicast_iface;
106 gboolean bind_mcast_address;
108 GstRTSPTransportMode transport_mode;
109 gboolean stop_on_disconnect;
112 GRecMutex state_lock; /* locking order: state lock, lock */
113 GPtrArray *streams; /* protected by lock */
114 GList *dynamic; /* protected by lock */
115 GstRTSPMediaStatus status; /* protected by lock */
119 gboolean finishing_unprepare;
121 /* the pipeline for the media */
122 GstElement *pipeline;
125 GstRTSPThread *thread;
126 GList *pending_pipeline_elements;
128 gboolean time_provider;
129 GstNetTimeProvider *nettime;
132 GstClockTimeDiff seekable;
134 GstState target_state;
136 /* RTP session manager */
139 /* the range of media */
140 GstRTSPTimeRange range; /* protected by lock */
141 GstClockTime range_start;
142 GstClockTime range_stop;
144 GList *payloads; /* protected by lock */
145 GstClockTime rtx_time; /* protected by lock */
146 gboolean do_retransmission; /* protected by lock */
147 guint latency; /* protected by lock */
148 GstClock *clock; /* protected by lock */
149 gboolean do_rate_control; /* protected by lock */
150 GstRTSPPublishClockMode publish_clock_mode;
152 /* Dynamic element handling */
153 guint nb_dynamic_elements;
154 guint no_more_pads_pending;
155 gboolean expected_async_done;
158 #define DEFAULT_SHARED FALSE
159 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
160 #define DEFAULT_REUSABLE FALSE
161 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
162 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
163 GST_RTSP_LOWER_TRANS_TCP
164 #define DEFAULT_EOS_SHUTDOWN FALSE
165 #define DEFAULT_BUFFER_SIZE 0x80000
166 #define DEFAULT_TIME_PROVIDER FALSE
167 #define DEFAULT_LATENCY 200
168 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
169 #define DEFAULT_STOP_ON_DISCONNECT TRUE
170 #define DEFAULT_MAX_MCAST_TTL 255
171 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
172 #define DEFAULT_DO_RATE_CONTROL TRUE
174 #define DEFAULT_DO_RETRANSMISSION FALSE
176 /* define to dump received RTCP packets */
193 PROP_STOP_ON_DISCONNECT,
196 PROP_BIND_MCAST_ADDRESS,
203 SIGNAL_REMOVED_STREAM,
211 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
212 #define GST_CAT_DEFAULT rtsp_media_debug
214 static void gst_rtsp_media_get_property (GObject * object, guint propid,
215 GValue * value, GParamSpec * pspec);
216 static void gst_rtsp_media_set_property (GObject * object, guint propid,
217 const GValue * value, GParamSpec * pspec);
218 static void gst_rtsp_media_finalize (GObject * obj);
220 static gboolean default_handle_message (GstRTSPMedia * media,
221 GstMessage * message);
222 static void finish_unprepare (GstRTSPMedia * media);
223 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
224 static gboolean default_unprepare (GstRTSPMedia * media);
225 static gboolean default_suspend (GstRTSPMedia * media);
226 static gboolean default_unsuspend (GstRTSPMedia * media);
227 static gboolean default_convert_range (GstRTSPMedia * media,
228 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
229 static gboolean default_query_position (GstRTSPMedia * media,
231 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
232 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
233 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
235 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
237 static gboolean wait_preroll (GstRTSPMedia * media);
239 static GstElement *find_payload_element (GstElement * payloader, GstPad * pad);
241 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
243 static gboolean check_complete (GstRTSPMedia * media);
245 #define C_ENUM(v) ((gint) v)
247 #define TRICKMODE_FLAGS (GST_SEEK_FLAG_TRICKMODE | GST_SEEK_FLAG_TRICKMODE_KEY_UNITS | GST_SEEK_FLAG_TRICKMODE_FORWARD_PREDICTED)
250 gst_rtsp_suspend_mode_get_type (void)
253 static const GEnumValue values[] = {
254 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
255 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
257 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
262 if (g_once_init_enter (&id)) {
263 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
264 g_once_init_leave (&id, tmp);
269 #define C_FLAGS(v) ((guint) v)
272 gst_rtsp_transport_mode_get_type (void)
275 static const GFlagsValue values[] = {
276 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
278 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
283 if (g_once_init_enter (&id)) {
284 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
285 g_once_init_leave (&id, tmp);
291 gst_rtsp_publish_clock_mode_get_type (void)
294 static const GEnumValue values[] = {
295 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
296 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
297 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
298 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
300 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
301 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
306 if (g_once_init_enter (&id)) {
307 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
308 g_once_init_leave (&id, tmp);
313 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
316 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
318 GObjectClass *gobject_class;
320 gobject_class = G_OBJECT_CLASS (klass);
322 gobject_class->get_property = gst_rtsp_media_get_property;
323 gobject_class->set_property = gst_rtsp_media_set_property;
324 gobject_class->finalize = gst_rtsp_media_finalize;
326 g_object_class_install_property (gobject_class, PROP_SHARED,
327 g_param_spec_boolean ("shared", "Shared",
328 "If this media pipeline can be shared", DEFAULT_SHARED,
329 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
332 g_param_spec_enum ("suspend-mode", "Suspend Mode",
333 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
334 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
336 g_object_class_install_property (gobject_class, PROP_REUSABLE,
337 g_param_spec_boolean ("reusable", "Reusable",
338 "If this media pipeline can be reused after an unprepare",
339 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
341 g_object_class_install_property (gobject_class, PROP_PROFILES,
342 g_param_spec_flags ("profiles", "Profiles",
343 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
344 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
346 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
347 g_param_spec_flags ("protocols", "Protocols",
348 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
349 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
351 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
352 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
353 "Send an EOS event to the pipeline before unpreparing",
354 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
356 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
357 g_param_spec_uint ("buffer-size", "Buffer Size",
358 "The kernel UDP buffer size to use", 0, G_MAXUINT,
359 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
361 g_object_class_install_property (gobject_class, PROP_ELEMENT,
362 g_param_spec_object ("element", "The Element",
363 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
364 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
366 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
367 g_param_spec_boolean ("time-provider", "Time Provider",
368 "Use a NetTimeProvider for clients",
369 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
371 g_object_class_install_property (gobject_class, PROP_LATENCY,
372 g_param_spec_uint ("latency", "Latency",
373 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
374 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
377 g_param_spec_flags ("transport-mode", "Transport Mode",
378 "If this media pipeline can be used for PLAY or RECORD",
379 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
380 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
383 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
384 "If this media pipeline should be stopped "
385 "when a client disconnects without TEARDOWN",
386 DEFAULT_STOP_ON_DISCONNECT,
387 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
389 g_object_class_install_property (gobject_class, PROP_CLOCK,
390 g_param_spec_object ("clock", "Clock",
391 "Clock to be used by the media pipeline",
392 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
395 g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
396 "The maximum time-to-live value of outgoing multicast packets", 1,
397 255, DEFAULT_MAX_MCAST_TTL,
398 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
400 g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
401 g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
402 "Whether the multicast sockets should be bound to multicast addresses "
404 DEFAULT_BIND_MCAST_ADDRESS,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
407 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
408 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
409 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
410 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
412 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
413 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
414 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
415 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
416 GST_TYPE_RTSP_STREAM);
418 gst_rtsp_media_signals[SIGNAL_PREPARED] =
419 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
420 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
421 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
423 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
424 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
425 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
426 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
428 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
429 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
430 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
431 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
433 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
434 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
435 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
436 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
438 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
440 klass->handle_message = default_handle_message;
441 klass->prepare = default_prepare;
442 klass->unprepare = default_unprepare;
443 klass->suspend = default_suspend;
444 klass->unsuspend = default_unsuspend;
445 klass->convert_range = default_convert_range;
446 klass->query_position = default_query_position;
447 klass->query_stop = default_query_stop;
448 klass->create_rtpbin = default_create_rtpbin;
449 klass->setup_sdp = default_setup_sdp;
450 klass->handle_sdp = default_handle_sdp;
454 gst_rtsp_media_init (GstRTSPMedia * media)
456 GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_private (media);
460 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
461 g_mutex_init (&priv->lock);
462 g_cond_init (&priv->cond);
463 g_rec_mutex_init (&priv->state_lock);
465 priv->shared = DEFAULT_SHARED;
466 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
467 priv->reusable = DEFAULT_REUSABLE;
468 priv->profiles = DEFAULT_PROFILES;
469 priv->protocols = DEFAULT_PROTOCOLS;
470 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
471 priv->buffer_size = DEFAULT_BUFFER_SIZE;
472 priv->time_provider = DEFAULT_TIME_PROVIDER;
473 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
474 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
475 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
476 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
477 priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
478 priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
479 priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
480 priv->expected_async_done = FALSE;
484 gst_rtsp_media_finalize (GObject * obj)
486 GstRTSPMediaPrivate *priv;
489 media = GST_RTSP_MEDIA (obj);
492 GST_INFO ("finalize media %p", media);
494 if (priv->permissions)
495 gst_rtsp_permissions_unref (priv->permissions);
497 g_ptr_array_unref (priv->streams);
499 g_list_free_full (priv->dynamic, gst_object_unref);
500 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
503 gst_object_unref (priv->pipeline);
505 gst_object_unref (priv->nettime);
506 gst_object_unref (priv->element);
508 g_object_unref (priv->pool);
510 g_list_free (priv->payloads);
512 gst_object_unref (priv->clock);
513 g_free (priv->multicast_iface);
514 g_mutex_clear (&priv->lock);
515 g_cond_clear (&priv->cond);
516 g_rec_mutex_clear (&priv->state_lock);
518 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
522 gst_rtsp_media_get_property (GObject * object, guint propid,
523 GValue * value, GParamSpec * pspec)
525 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
529 g_value_set_object (value, media->priv->element);
532 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
534 case PROP_SUSPEND_MODE:
535 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
538 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
541 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
544 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
546 case PROP_EOS_SHUTDOWN:
547 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
549 case PROP_BUFFER_SIZE:
550 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
552 case PROP_TIME_PROVIDER:
553 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
556 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
558 case PROP_TRANSPORT_MODE:
559 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
561 case PROP_STOP_ON_DISCONNECT:
562 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
565 g_value_take_object (value, gst_rtsp_media_get_clock (media));
567 case PROP_MAX_MCAST_TTL:
568 g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
570 case PROP_BIND_MCAST_ADDRESS:
571 g_value_set_boolean (value, gst_rtsp_media_is_bind_mcast_address (media));
574 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
579 gst_rtsp_media_set_property (GObject * object, guint propid,
580 const GValue * value, GParamSpec * pspec)
582 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
586 media->priv->element = g_value_get_object (value);
587 gst_object_ref_sink (media->priv->element);
590 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
592 case PROP_SUSPEND_MODE:
593 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
596 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
599 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
602 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
604 case PROP_EOS_SHUTDOWN:
605 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
607 case PROP_BUFFER_SIZE:
608 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
610 case PROP_TIME_PROVIDER:
611 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
614 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
616 case PROP_TRANSPORT_MODE:
617 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
619 case PROP_STOP_ON_DISCONNECT:
620 gst_rtsp_media_set_stop_on_disconnect (media,
621 g_value_get_boolean (value));
624 gst_rtsp_media_set_clock (media, g_value_get_object (value));
626 case PROP_MAX_MCAST_TTL:
627 gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
629 case PROP_BIND_MCAST_ADDRESS:
630 gst_rtsp_media_set_bind_mcast_address (media,
631 g_value_get_boolean (value));
634 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
641 gboolean complete_streams_only;
643 } DoQueryPositionData;
646 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
650 if (!gst_rtsp_stream_is_sender (stream))
653 if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
654 GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
658 if (gst_rtsp_stream_query_position (stream, &tmp)) {
659 data->position = MIN (data->position, tmp);
663 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
664 GST_TIME_ARGS (data->position));
668 default_query_position (GstRTSPMedia * media, gint64 * position)
670 GstRTSPMediaPrivate *priv;
671 DoQueryPositionData data;
675 data.position = G_MAXINT64;
678 /* if the media is complete, i.e. one or more streams have been configured
679 * with sinks, then we want to query the position on those streams only.
680 * a query on an incmplete stream may return a position that originates from
681 * an earlier preroll */
682 if (check_complete (media))
683 data.complete_streams_only = TRUE;
685 data.complete_streams_only = FALSE;
687 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
690 *position = GST_CLOCK_TIME_NONE;
692 *position = data.position;
704 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
708 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
709 data->stop = MAX (data->stop, tmp);
715 default_query_stop (GstRTSPMedia * media, gint64 * stop)
717 GstRTSPMediaPrivate *priv;
718 DoQueryStopData data;
725 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
733 default_create_rtpbin (GstRTSPMedia * media)
737 rtpbin = gst_element_factory_make ("rtpbin", NULL);
742 /* must be called with state lock */
744 check_seekable (GstRTSPMedia * media)
747 GstRTSPMediaPrivate *priv = media->priv;
749 /* Update the seekable state of the pipeline in case it changed */
750 if (gst_rtsp_media_is_receive_only (media)) {
751 /* TODO: Seeking for "receive-only"? */
754 guint i, n = priv->streams->len;
756 for (i = 0; i < n; i++) {
757 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
759 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
760 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
767 query = gst_query_new_seeking (GST_FORMAT_TIME);
768 if (gst_element_query (priv->pipeline, query)) {
773 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
774 priv->seekable = seekable ? G_MAXINT64 : 0;
775 } else if (priv->streams->len) {
776 gboolean seekable = TRUE;
777 guint i, n = priv->streams->len;
779 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
780 for (i = 0; i < n; i++) {
781 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
782 seekable &= gst_rtsp_stream_seekable (stream);
784 priv->seekable = seekable ? G_MAXINT64 : -1;
787 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
789 gst_query_unref (query);
792 /* must be called with state lock */
794 check_complete (GstRTSPMedia * media)
796 GstRTSPMediaPrivate *priv = media->priv;
798 guint i, n = priv->streams->len;
800 for (i = 0; i < n; i++) {
801 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
803 if (gst_rtsp_stream_is_complete (stream))
810 /* must be called with state lock */
812 collect_media_stats (GstRTSPMedia * media)
814 GstRTSPMediaPrivate *priv = media->priv;
815 gint64 position = 0, stop = -1;
817 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
818 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
821 priv->range.unit = GST_RTSP_RANGE_NPT;
823 GST_INFO ("collect media stats");
826 priv->range.min.type = GST_RTSP_TIME_NOW;
827 priv->range.min.seconds = -1;
828 priv->range_start = -1;
829 priv->range.max.type = GST_RTSP_TIME_END;
830 priv->range.max.seconds = -1;
831 priv->range_stop = -1;
833 GstRTSPMediaClass *klass;
836 klass = GST_RTSP_MEDIA_GET_CLASS (media);
838 /* get the position */
840 if (klass->query_position)
841 ret = klass->query_position (media, &position);
844 GST_INFO ("position query failed");
848 /* get the current segment stop */
850 if (klass->query_stop)
851 ret = klass->query_stop (media, &stop);
854 GST_INFO ("stop query failed");
858 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
859 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
861 if (position == -1) {
862 priv->range.min.type = GST_RTSP_TIME_NOW;
863 priv->range.min.seconds = -1;
864 priv->range_start = -1;
866 priv->range.min.type = GST_RTSP_TIME_SECONDS;
867 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
868 priv->range_start = position;
871 priv->range.max.type = GST_RTSP_TIME_END;
872 priv->range.max.seconds = -1;
873 priv->range_stop = -1;
875 priv->range.max.type = GST_RTSP_TIME_SECONDS;
876 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
877 priv->range_stop = stop;
880 check_seekable (media);
885 * gst_rtsp_media_new:
886 * @element: (transfer full): a #GstElement
888 * Create a new #GstRTSPMedia instance. @element is the bin element that
889 * provides the different streams. The #GstRTSPMedia object contains the
890 * element to produce RTP data for one or more related (audio/video/..)
893 * Ownership is taken of @element.
895 * Returns: (transfer full): a new #GstRTSPMedia object.
898 gst_rtsp_media_new (GstElement * element)
900 GstRTSPMedia *result;
902 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
904 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
910 * gst_rtsp_media_get_element:
911 * @media: a #GstRTSPMedia
913 * Get the element that was used when constructing @media.
915 * Returns: (transfer full): a #GstElement. Unref after usage.
918 gst_rtsp_media_get_element (GstRTSPMedia * media)
920 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
922 return gst_object_ref (media->priv->element);
926 * gst_rtsp_media_take_pipeline:
927 * @media: a #GstRTSPMedia
928 * @pipeline: (transfer full): a #GstPipeline
930 * Set @pipeline as the #GstPipeline for @media. Ownership is
931 * taken of @pipeline.
934 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
936 GstRTSPMediaPrivate *priv;
938 GstNetTimeProvider *nettime;
941 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
942 g_return_if_fail (GST_IS_PIPELINE (pipeline));
946 g_mutex_lock (&priv->lock);
947 old = priv->pipeline;
948 priv->pipeline = GST_ELEMENT_CAST (pipeline);
949 nettime = priv->nettime;
950 priv->nettime = NULL;
951 g_mutex_unlock (&priv->lock);
954 gst_object_unref (old);
957 gst_object_unref (nettime);
959 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
961 for (l = priv->pending_pipeline_elements; l; l = l->next) {
962 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
964 g_list_free (priv->pending_pipeline_elements);
965 priv->pending_pipeline_elements = NULL;
969 * gst_rtsp_media_set_permissions:
970 * @media: a #GstRTSPMedia
971 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
973 * Set @permissions on @media.
976 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
977 GstRTSPPermissions * permissions)
979 GstRTSPMediaPrivate *priv;
981 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
985 g_mutex_lock (&priv->lock);
986 if (priv->permissions)
987 gst_rtsp_permissions_unref (priv->permissions);
988 if ((priv->permissions = permissions))
989 gst_rtsp_permissions_ref (permissions);
990 g_mutex_unlock (&priv->lock);
994 * gst_rtsp_media_get_permissions:
995 * @media: a #GstRTSPMedia
997 * Get the permissions object from @media.
999 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
1001 GstRTSPPermissions *
1002 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
1004 GstRTSPMediaPrivate *priv;
1005 GstRTSPPermissions *result;
1007 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1011 g_mutex_lock (&priv->lock);
1012 if ((result = priv->permissions))
1013 gst_rtsp_permissions_ref (result);
1014 g_mutex_unlock (&priv->lock);
1020 * gst_rtsp_media_set_suspend_mode:
1021 * @media: a #GstRTSPMedia
1022 * @mode: the new #GstRTSPSuspendMode
1024 * Control how @ media will be suspended after the SDP has been generated and
1025 * after a PAUSE request has been performed.
1027 * Media must be unprepared when setting the suspend mode.
1030 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
1032 GstRTSPMediaPrivate *priv;
1034 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1038 g_rec_mutex_lock (&priv->state_lock);
1039 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1041 priv->suspend_mode = mode;
1042 g_rec_mutex_unlock (&priv->state_lock);
1049 GST_WARNING ("media %p was prepared", media);
1050 g_rec_mutex_unlock (&priv->state_lock);
1055 * gst_rtsp_media_get_suspend_mode:
1056 * @media: a #GstRTSPMedia
1058 * Get how @media will be suspended.
1060 * Returns: #GstRTSPSuspendMode.
1063 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1065 GstRTSPMediaPrivate *priv;
1066 GstRTSPSuspendMode res;
1068 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1072 g_rec_mutex_lock (&priv->state_lock);
1073 res = priv->suspend_mode;
1074 g_rec_mutex_unlock (&priv->state_lock);
1080 * gst_rtsp_media_set_shared:
1081 * @media: a #GstRTSPMedia
1082 * @shared: the new value
1084 * Set or unset if the pipeline for @media can be shared will multiple clients.
1085 * When @shared is %TRUE, client requests for this media will share the media
1089 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1091 GstRTSPMediaPrivate *priv;
1093 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1097 g_mutex_lock (&priv->lock);
1098 priv->shared = shared;
1099 g_mutex_unlock (&priv->lock);
1103 * gst_rtsp_media_is_shared:
1104 * @media: a #GstRTSPMedia
1106 * Check if the pipeline for @media can be shared between multiple clients.
1108 * Returns: %TRUE if the media can be shared between clients.
1111 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1113 GstRTSPMediaPrivate *priv;
1116 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1120 g_mutex_lock (&priv->lock);
1122 g_mutex_unlock (&priv->lock);
1128 * gst_rtsp_media_set_reusable:
1129 * @media: a #GstRTSPMedia
1130 * @reusable: the new value
1132 * Set or unset if the pipeline for @media can be reused after the pipeline has
1136 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1138 GstRTSPMediaPrivate *priv;
1140 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1144 g_mutex_lock (&priv->lock);
1145 priv->reusable = reusable;
1146 g_mutex_unlock (&priv->lock);
1150 * gst_rtsp_media_is_reusable:
1151 * @media: a #GstRTSPMedia
1153 * Check if the pipeline for @media can be reused after an unprepare.
1155 * Returns: %TRUE if the media can be reused
1158 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1160 GstRTSPMediaPrivate *priv;
1163 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1167 g_mutex_lock (&priv->lock);
1168 res = priv->reusable;
1169 g_mutex_unlock (&priv->lock);
1175 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1177 gst_rtsp_stream_set_profiles (stream, *profiles);
1181 * gst_rtsp_media_set_profiles:
1182 * @media: a #GstRTSPMedia
1183 * @profiles: the new flags
1185 * Configure the allowed lower transport for @media.
1188 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1190 GstRTSPMediaPrivate *priv;
1192 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1196 g_mutex_lock (&priv->lock);
1197 priv->profiles = profiles;
1198 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1199 g_mutex_unlock (&priv->lock);
1203 * gst_rtsp_media_get_profiles:
1204 * @media: a #GstRTSPMedia
1206 * Get the allowed profiles of @media.
1208 * Returns: a #GstRTSPProfile
1211 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1213 GstRTSPMediaPrivate *priv;
1216 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1220 g_mutex_lock (&priv->lock);
1221 res = priv->profiles;
1222 g_mutex_unlock (&priv->lock);
1228 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1230 gst_rtsp_stream_set_protocols (stream, *protocols);
1234 * gst_rtsp_media_set_protocols:
1235 * @media: a #GstRTSPMedia
1236 * @protocols: the new flags
1238 * Configure the allowed lower transport for @media.
1241 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1243 GstRTSPMediaPrivate *priv;
1245 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1249 g_mutex_lock (&priv->lock);
1250 priv->protocols = protocols;
1251 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1252 g_mutex_unlock (&priv->lock);
1256 * gst_rtsp_media_get_protocols:
1257 * @media: a #GstRTSPMedia
1259 * Get the allowed protocols of @media.
1261 * Returns: a #GstRTSPLowerTrans
1264 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1266 GstRTSPMediaPrivate *priv;
1267 GstRTSPLowerTrans res;
1269 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1270 GST_RTSP_LOWER_TRANS_UNKNOWN);
1274 g_mutex_lock (&priv->lock);
1275 res = priv->protocols;
1276 g_mutex_unlock (&priv->lock);
1282 * gst_rtsp_media_set_eos_shutdown:
1283 * @media: a #GstRTSPMedia
1284 * @eos_shutdown: the new value
1286 * Set or unset if an EOS event will be sent to the pipeline for @media before
1290 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1292 GstRTSPMediaPrivate *priv;
1294 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1298 g_mutex_lock (&priv->lock);
1299 priv->eos_shutdown = eos_shutdown;
1300 g_mutex_unlock (&priv->lock);
1304 * gst_rtsp_media_is_eos_shutdown:
1305 * @media: a #GstRTSPMedia
1307 * Check if the pipeline for @media will send an EOS down the pipeline before
1310 * Returns: %TRUE if the media will send EOS before unpreparing.
1313 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1315 GstRTSPMediaPrivate *priv;
1318 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1322 g_mutex_lock (&priv->lock);
1323 res = priv->eos_shutdown;
1324 g_mutex_unlock (&priv->lock);
1330 * gst_rtsp_media_set_buffer_size:
1331 * @media: a #GstRTSPMedia
1332 * @size: the new value
1334 * Set the kernel UDP buffer size.
1337 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1339 GstRTSPMediaPrivate *priv;
1342 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1344 GST_LOG_OBJECT (media, "set buffer size %u", size);
1348 g_mutex_lock (&priv->lock);
1349 priv->buffer_size = size;
1351 for (i = 0; i < priv->streams->len; i++) {
1352 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1353 gst_rtsp_stream_set_buffer_size (stream, size);
1355 g_mutex_unlock (&priv->lock);
1359 * gst_rtsp_media_get_buffer_size:
1360 * @media: a #GstRTSPMedia
1362 * Get the kernel UDP buffer size.
1364 * Returns: the kernel UDP buffer size.
1367 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1369 GstRTSPMediaPrivate *priv;
1372 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1376 g_mutex_lock (&priv->lock);
1377 res = priv->buffer_size;
1378 g_mutex_unlock (&priv->lock);
1384 * gst_rtsp_media_set_stop_on_disconnect:
1385 * @media: a #GstRTSPMedia
1386 * @stop_on_disconnect: the new value
1388 * Set or unset if the pipeline for @media should be stopped when a
1389 * client disconnects without sending TEARDOWN.
1392 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1393 gboolean stop_on_disconnect)
1395 GstRTSPMediaPrivate *priv;
1397 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1401 g_mutex_lock (&priv->lock);
1402 priv->stop_on_disconnect = stop_on_disconnect;
1403 g_mutex_unlock (&priv->lock);
1407 * gst_rtsp_media_is_stop_on_disconnect:
1408 * @media: a #GstRTSPMedia
1410 * Check if the pipeline for @media will be stopped when a client disconnects
1411 * without sending TEARDOWN.
1413 * Returns: %TRUE if the media will be stopped when a client disconnects
1414 * without sending TEARDOWN.
1417 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1419 GstRTSPMediaPrivate *priv;
1422 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1426 g_mutex_lock (&priv->lock);
1427 res = priv->stop_on_disconnect;
1428 g_mutex_unlock (&priv->lock);
1434 * gst_rtsp_media_set_retransmission_time:
1435 * @media: a #GstRTSPMedia
1436 * @time: the new value
1438 * Set the amount of time to store retransmission packets.
1441 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1443 GstRTSPMediaPrivate *priv;
1446 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1448 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1452 g_mutex_lock (&priv->lock);
1453 priv->rtx_time = time;
1454 for (i = 0; i < priv->streams->len; i++) {
1455 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1457 gst_rtsp_stream_set_retransmission_time (stream, time);
1459 g_mutex_unlock (&priv->lock);
1463 * gst_rtsp_media_get_retransmission_time:
1464 * @media: a #GstRTSPMedia
1466 * Get the amount of time to store retransmission data.
1468 * Returns: the amount of time to store retransmission data.
1471 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1473 GstRTSPMediaPrivate *priv;
1476 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1480 g_mutex_lock (&priv->lock);
1481 res = priv->rtx_time;
1482 g_mutex_unlock (&priv->lock);
1488 * gst_rtsp_media_set_do_retransmission:
1490 * Set whether retransmission requests will be sent
1495 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
1496 gboolean do_retransmission)
1498 GstRTSPMediaPrivate *priv;
1500 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1504 g_mutex_lock (&priv->lock);
1505 priv->do_retransmission = do_retransmission;
1508 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1509 g_mutex_unlock (&priv->lock);
1513 * gst_rtsp_media_get_do_retransmission:
1515 * Returns: Whether retransmission requests will be sent
1520 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1522 GstRTSPMediaPrivate *priv;
1525 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1529 g_mutex_lock (&priv->lock);
1530 res = priv->do_retransmission;
1531 g_mutex_unlock (&priv->lock);
1537 * gst_rtsp_media_set_latency:
1538 * @media: a #GstRTSPMedia
1539 * @latency: latency in milliseconds
1541 * Configure the latency used for receiving media.
1544 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1546 GstRTSPMediaPrivate *priv;
1549 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1551 GST_LOG_OBJECT (media, "set latency %ums", latency);
1555 g_mutex_lock (&priv->lock);
1556 priv->latency = latency;
1558 g_object_set (priv->rtpbin, "latency", latency, NULL);
1560 for (i = 0; i < media->priv->streams->len; i++) {
1561 GObject *storage = NULL;
1563 g_signal_emit_by_name (G_OBJECT (media->priv->rtpbin), "get-storage",
1566 g_object_set (storage, "size-time",
1567 (media->priv->latency + 50) * GST_MSECOND, NULL);
1571 g_mutex_unlock (&priv->lock);
1575 * gst_rtsp_media_get_latency:
1576 * @media: a #GstRTSPMedia
1578 * Get the latency that is used for receiving media.
1580 * Returns: latency in milliseconds
1583 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1585 GstRTSPMediaPrivate *priv;
1588 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1592 g_mutex_lock (&priv->lock);
1593 res = priv->latency;
1594 g_mutex_unlock (&priv->lock);
1600 * gst_rtsp_media_use_time_provider:
1601 * @media: a #GstRTSPMedia
1602 * @time_provider: if a #GstNetTimeProvider should be used
1604 * Set @media to provide a #GstNetTimeProvider.
1607 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1609 GstRTSPMediaPrivate *priv;
1611 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1615 g_mutex_lock (&priv->lock);
1616 priv->time_provider = time_provider;
1617 g_mutex_unlock (&priv->lock);
1621 * gst_rtsp_media_is_time_provider:
1622 * @media: a #GstRTSPMedia
1624 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1626 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1628 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1631 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1633 GstRTSPMediaPrivate *priv;
1636 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1640 g_mutex_lock (&priv->lock);
1641 res = priv->time_provider;
1642 g_mutex_unlock (&priv->lock);
1648 * gst_rtsp_media_set_clock:
1649 * @media: a #GstRTSPMedia
1650 * @clock: (nullable): #GstClock to be used
1652 * Configure the clock used for the media.
1655 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1657 GstRTSPMediaPrivate *priv;
1659 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1660 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1662 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1666 g_mutex_lock (&priv->lock);
1668 gst_object_unref (priv->clock);
1669 priv->clock = clock ? gst_object_ref (clock) : NULL;
1670 if (priv->pipeline) {
1672 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1674 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1677 g_mutex_unlock (&priv->lock);
1681 * gst_rtsp_media_set_publish_clock_mode:
1682 * @media: a #GstRTSPMedia
1683 * @mode: the clock publish mode
1685 * Sets if and how the media clock should be published according to RFC7273.
1690 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1691 GstRTSPPublishClockMode mode)
1693 GstRTSPMediaPrivate *priv;
1697 g_mutex_lock (&priv->lock);
1698 priv->publish_clock_mode = mode;
1700 n = priv->streams->len;
1701 for (i = 0; i < n; i++) {
1702 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1704 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1706 g_mutex_unlock (&priv->lock);
1710 * gst_rtsp_media_get_publish_clock_mode:
1711 * @media: a #GstRTSPMedia
1713 * Gets if and how the media clock should be published according to RFC7273.
1715 * Returns: The GstRTSPPublishClockMode
1719 GstRTSPPublishClockMode
1720 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1722 GstRTSPMediaPrivate *priv;
1723 GstRTSPPublishClockMode ret;
1726 g_mutex_lock (&priv->lock);
1727 ret = priv->publish_clock_mode;
1728 g_mutex_unlock (&priv->lock);
1734 * gst_rtsp_media_set_address_pool:
1735 * @media: a #GstRTSPMedia
1736 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1738 * configure @pool to be used as the address pool of @media.
1741 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1742 GstRTSPAddressPool * pool)
1744 GstRTSPMediaPrivate *priv;
1745 GstRTSPAddressPool *old;
1747 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1751 GST_LOG_OBJECT (media, "set address pool %p", pool);
1753 g_mutex_lock (&priv->lock);
1754 if ((old = priv->pool) != pool)
1755 priv->pool = pool ? g_object_ref (pool) : NULL;
1758 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1760 g_mutex_unlock (&priv->lock);
1763 g_object_unref (old);
1767 * gst_rtsp_media_get_address_pool:
1768 * @media: a #GstRTSPMedia
1770 * Get the #GstRTSPAddressPool used as the address pool of @media.
1772 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1773 * g_object_unref() after usage.
1775 GstRTSPAddressPool *
1776 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1778 GstRTSPMediaPrivate *priv;
1779 GstRTSPAddressPool *result;
1781 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1785 g_mutex_lock (&priv->lock);
1786 if ((result = priv->pool))
1787 g_object_ref (result);
1788 g_mutex_unlock (&priv->lock);
1794 * gst_rtsp_media_set_multicast_iface:
1795 * @media: a #GstRTSPMedia
1796 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1798 * configure @multicast_iface to be used for @media.
1801 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1802 const gchar * multicast_iface)
1804 GstRTSPMediaPrivate *priv;
1807 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1811 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1813 g_mutex_lock (&priv->lock);
1814 if ((old = priv->multicast_iface) != multicast_iface)
1815 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1818 g_ptr_array_foreach (priv->streams,
1819 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1820 g_mutex_unlock (&priv->lock);
1827 * gst_rtsp_media_get_multicast_iface:
1828 * @media: a #GstRTSPMedia
1830 * Get the multicast interface used for @media.
1832 * Returns: (transfer full) (nullable): the multicast interface for @media.
1833 * g_free() after usage.
1836 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1838 GstRTSPMediaPrivate *priv;
1841 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1845 g_mutex_lock (&priv->lock);
1846 if ((result = priv->multicast_iface))
1847 result = g_strdup (result);
1848 g_mutex_unlock (&priv->lock);
1854 * gst_rtsp_media_set_max_mcast_ttl:
1855 * @media: a #GstRTSPMedia
1856 * @ttl: the new multicast ttl value
1858 * Set the maximum time-to-live value of outgoing multicast packets.
1860 * Returns: %TRUE if the requested ttl has been set successfully.
1865 gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
1867 GstRTSPMediaPrivate *priv;
1870 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1872 GST_LOG_OBJECT (media, "set max mcast ttl %u", ttl);
1876 g_mutex_lock (&priv->lock);
1878 if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
1879 GST_WARNING_OBJECT (media, "The reqested mcast TTL value is not valid.");
1880 g_mutex_unlock (&priv->lock);
1883 priv->max_mcast_ttl = ttl;
1885 for (i = 0; i < priv->streams->len; i++) {
1886 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1887 gst_rtsp_stream_set_max_mcast_ttl (stream, ttl);
1889 g_mutex_unlock (&priv->lock);
1895 * gst_rtsp_media_get_max_mcast_ttl:
1896 * @media: a #GstRTSPMedia
1898 * Get the the maximum time-to-live value of outgoing multicast packets.
1900 * Returns: the maximum time-to-live value of outgoing multicast packets.
1905 gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
1907 GstRTSPMediaPrivate *priv;
1910 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1914 g_mutex_lock (&priv->lock);
1915 res = priv->max_mcast_ttl;
1916 g_mutex_unlock (&priv->lock);
1922 * gst_rtsp_media_set_bind_mcast_address:
1923 * @media: a #GstRTSPMedia
1924 * @bind_mcast_addr: the new value
1926 * Decide whether the multicast socket should be bound to a multicast address or
1932 gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
1933 gboolean bind_mcast_addr)
1935 GstRTSPMediaPrivate *priv;
1938 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1942 g_mutex_lock (&priv->lock);
1943 priv->bind_mcast_address = bind_mcast_addr;
1944 for (i = 0; i < priv->streams->len; i++) {
1945 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1946 gst_rtsp_stream_set_bind_mcast_address (stream, bind_mcast_addr);
1948 g_mutex_unlock (&priv->lock);
1952 * gst_rtsp_media_is_bind_mcast_address:
1953 * @media: a #GstRTSPMedia
1955 * Check if multicast sockets are configured to be bound to multicast addresses.
1957 * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
1962 gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
1964 GstRTSPMediaPrivate *priv;
1967 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1971 g_mutex_lock (&priv->lock);
1972 result = priv->bind_mcast_address;
1973 g_mutex_unlock (&priv->lock);
1979 _find_payload_types (GstRTSPMedia * media)
1982 GQueue queue = G_QUEUE_INIT;
1984 n = media->priv->streams->len;
1985 for (i = 0; i < n; i++) {
1986 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1987 guint pt = gst_rtsp_stream_get_pt (stream);
1989 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1996 _next_available_pt (GList * payloads)
2000 for (i = 96; i <= 127; i++) {
2001 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
2003 return GPOINTER_TO_UINT (i);
2010 * gst_rtsp_media_collect_streams:
2011 * @media: a #GstRTSPMedia
2013 * Find all payloader elements, they should be named pay\%d in the
2014 * element of @media, and create #GstRTSPStreams for them.
2016 * Collect all dynamic elements, named dynpay\%d, and add them to
2017 * the list of dynamic elements.
2019 * Find all depayloader elements, they should be named depay\%d in the
2020 * element of @media, and create #GstRTSPStreams for them.
2023 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
2025 GstRTSPMediaPrivate *priv;
2026 GstElement *element, *elem;
2030 gboolean more_elem_remaining = TRUE;
2031 GstRTSPTransportMode mode = 0;
2033 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2036 element = priv->element;
2039 for (i = 0; more_elem_remaining; i++) {
2042 more_elem_remaining = FALSE;
2044 name = g_strdup_printf ("pay%d", i);
2045 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2047 GST_INFO ("found stream %d with payloader %p", i, elem);
2049 /* take the pad of the payloader */
2050 pad = gst_element_get_static_pad (elem, "src");
2052 /* find the real payload element in case elem is a GstBin */
2053 pay = find_payload_element (elem, pad);
2055 /* create the stream */
2057 GST_WARNING ("could not find real payloader, using bin");
2058 gst_rtsp_media_create_stream (media, elem, pad);
2060 gst_rtsp_media_create_stream (media, pay, pad);
2061 gst_object_unref (pay);
2064 gst_object_unref (pad);
2065 gst_object_unref (elem);
2068 more_elem_remaining = TRUE;
2069 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2073 name = g_strdup_printf ("dynpay%d", i);
2074 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2075 /* a stream that will dynamically create pads to provide RTP packets */
2076 GST_INFO ("found dynamic element %d, %p", i, elem);
2078 g_mutex_lock (&priv->lock);
2079 priv->dynamic = g_list_prepend (priv->dynamic, elem);
2080 g_mutex_unlock (&priv->lock);
2082 priv->nb_dynamic_elements++;
2085 more_elem_remaining = TRUE;
2086 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2090 name = g_strdup_printf ("depay%d", i);
2091 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2092 GST_INFO ("found stream %d with depayloader %p", i, elem);
2094 /* take the pad of the payloader */
2095 pad = gst_element_get_static_pad (elem, "sink");
2096 /* create the stream */
2097 gst_rtsp_media_create_stream (media, elem, pad);
2098 gst_object_unref (pad);
2099 gst_object_unref (elem);
2102 more_elem_remaining = TRUE;
2103 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
2109 if (priv->transport_mode != mode)
2110 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
2111 priv->transport_mode, mode);
2117 GstElement *appsink, *appsrc;
2118 GstRTSPStream *stream;
2121 static GstFlowReturn
2122 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
2124 AppSinkSrcData *data = user_data;
2128 sample = gst_app_sink_pull_sample (appsink);
2130 return GST_FLOW_FLUSHING;
2133 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
2134 gst_sample_unref (sample);
2138 static GstAppSinkCallbacks appsink_callbacks = {
2144 static GstPadProbeReturn
2145 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2147 AppSinkSrcData *data = user_data;
2149 if (GST_IS_EVENT (info->data)
2150 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
2151 GstClockTime min, max;
2153 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
2155 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
2156 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
2157 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
2159 } else if (GST_IS_QUERY (info->data)) {
2160 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
2161 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
2162 gst_object_unref (srcpad);
2163 return GST_PAD_PROBE_HANDLED;
2165 gst_object_unref (srcpad);
2168 return GST_PAD_PROBE_OK;
2171 static GstPadProbeReturn
2172 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2174 AppSinkSrcData *data = user_data;
2176 if (GST_IS_QUERY (info->data)) {
2177 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
2178 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
2179 gst_object_unref (sinkpad);
2180 return GST_PAD_PROBE_HANDLED;
2182 gst_object_unref (sinkpad);
2185 return GST_PAD_PROBE_OK;
2189 * gst_rtsp_media_create_stream:
2190 * @media: a #GstRTSPMedia
2191 * @payloader: a #GstElement
2194 * Create a new stream in @media that provides RTP data on @pad.
2195 * @pad should be a pad of an element inside @media->element.
2197 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2201 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2204 GstRTSPMediaPrivate *priv;
2205 GstRTSPStream *stream;
2209 AppSinkSrcData *data = NULL;
2211 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2212 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2213 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2217 g_mutex_lock (&priv->lock);
2218 idx = priv->streams->len;
2220 GST_DEBUG ("media %p: creating stream with index %d and payloader %"
2221 GST_PTR_FORMAT, media, idx, payloader);
2223 if (GST_PAD_IS_SRC (pad))
2224 name = g_strdup_printf ("src_%u", idx);
2226 name = g_strdup_printf ("sink_%u", idx);
2228 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2229 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2230 GstElement *appsink, *appsrc;
2231 GstPad *sinkpad, *srcpad;
2233 appsink = gst_element_factory_make ("appsink", NULL);
2234 appsrc = gst_element_factory_make ("appsrc", NULL);
2236 if (GST_PAD_IS_SINK (pad)) {
2237 srcpad = gst_element_get_static_pad (appsrc, "src");
2239 gst_bin_add (GST_BIN (priv->element), appsrc);
2241 gst_pad_link (srcpad, pad);
2242 gst_object_unref (srcpad);
2244 streampad = gst_element_get_static_pad (appsink, "sink");
2246 priv->pending_pipeline_elements =
2247 g_list_prepend (priv->pending_pipeline_elements, appsink);
2249 sinkpad = gst_element_get_static_pad (appsink, "sink");
2251 gst_pad_link (pad, sinkpad);
2252 gst_object_unref (sinkpad);
2254 streampad = gst_element_get_static_pad (appsrc, "src");
2256 priv->pending_pipeline_elements =
2257 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2260 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2261 TRUE, "emit-signals", FALSE, NULL);
2262 g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
2263 FALSE, "buffer-list", TRUE, NULL);
2265 data = g_new0 (AppSinkSrcData, 1);
2266 data->appsink = appsink;
2267 data->appsrc = appsrc;
2269 sinkpad = gst_element_get_static_pad (appsink, "sink");
2270 gst_pad_add_probe (sinkpad,
2271 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2272 appsink_pad_probe, data, NULL);
2273 gst_object_unref (sinkpad);
2275 srcpad = gst_element_get_static_pad (appsrc, "src");
2276 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2277 appsrc_pad_probe, data, NULL);
2278 gst_object_unref (srcpad);
2280 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2282 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2285 streampad = gst_ghost_pad_new (name, pad);
2286 gst_pad_set_active (streampad, TRUE);
2287 gst_element_add_pad (priv->element, streampad);
2291 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2293 data->stream = stream;
2295 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2296 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2297 gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
2298 gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
2299 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2300 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2301 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2302 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2303 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2304 gst_rtsp_stream_set_rate_control (stream, priv->do_rate_control);
2306 g_ptr_array_add (priv->streams, stream);
2308 if (GST_PAD_IS_SRC (pad)) {
2312 g_list_free (priv->payloads);
2313 priv->payloads = _find_payload_types (media);
2315 n = priv->streams->len;
2316 for (i = 0; i < n; i++) {
2317 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2318 guint rtx_pt = _next_available_pt (priv->payloads);
2321 GST_WARNING ("Ran out of space of dynamic payload types");
2325 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2328 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2331 g_mutex_unlock (&priv->lock);
2333 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2340 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2342 GstRTSPMediaPrivate *priv;
2344 AppSinkSrcData *data;
2348 g_mutex_lock (&priv->lock);
2349 /* remove the ghostpad */
2350 srcpad = gst_rtsp_stream_get_srcpad (stream);
2351 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2353 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2354 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2355 else if (GST_OBJECT_PARENT (data->appsrc) ==
2356 GST_OBJECT_CAST (priv->element))
2357 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2358 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2359 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2360 else if (GST_OBJECT_PARENT (data->appsink) ==
2361 GST_OBJECT_CAST (priv->element))
2362 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2364 gst_element_remove_pad (priv->element, srcpad);
2366 gst_object_unref (srcpad);
2367 /* now remove the stream */
2368 g_object_ref (stream);
2369 g_ptr_array_remove (priv->streams, stream);
2370 g_mutex_unlock (&priv->lock);
2372 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2375 g_object_unref (stream);
2379 * gst_rtsp_media_n_streams:
2380 * @media: a #GstRTSPMedia
2382 * Get the number of streams in this media.
2384 * Returns: The number of streams.
2387 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2389 GstRTSPMediaPrivate *priv;
2392 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2396 g_mutex_lock (&priv->lock);
2397 res = priv->streams->len;
2398 g_mutex_unlock (&priv->lock);
2404 * gst_rtsp_media_get_stream:
2405 * @media: a #GstRTSPMedia
2406 * @idx: the stream index
2408 * Retrieve the stream with index @idx from @media.
2410 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2411 * @idx or %NULL when a stream with that index did not exist.
2414 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2416 GstRTSPMediaPrivate *priv;
2419 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2423 g_mutex_lock (&priv->lock);
2424 if (idx < priv->streams->len)
2425 res = g_ptr_array_index (priv->streams, idx);
2428 g_mutex_unlock (&priv->lock);
2434 * gst_rtsp_media_find_stream:
2435 * @media: a #GstRTSPMedia
2436 * @control: the control of the stream
2438 * Find a stream in @media with @control as the control uri.
2440 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2441 * control uri @control or %NULL when a stream with that control did
2445 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2447 GstRTSPMediaPrivate *priv;
2451 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2452 g_return_val_if_fail (control != NULL, NULL);
2458 g_mutex_lock (&priv->lock);
2459 for (i = 0; i < priv->streams->len; i++) {
2460 GstRTSPStream *test;
2462 test = g_ptr_array_index (priv->streams, i);
2463 if (gst_rtsp_stream_has_control (test, control)) {
2468 g_mutex_unlock (&priv->lock);
2473 /* called with state-lock */
2475 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2476 GstRTSPRangeUnit unit)
2478 return gst_rtsp_range_convert_units (range, unit);
2482 * gst_rtsp_media_get_range_string:
2483 * @media: a #GstRTSPMedia
2484 * @play: for the PLAY request
2485 * @unit: the unit to use for the string
2487 * Get the current range as a string. @media must be prepared with
2488 * gst_rtsp_media_prepare ().
2490 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2493 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2494 GstRTSPRangeUnit unit)
2496 GstRTSPMediaClass *klass;
2497 GstRTSPMediaPrivate *priv;
2499 GstRTSPTimeRange range;
2501 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2502 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2503 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2507 g_rec_mutex_lock (&priv->state_lock);
2508 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2509 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2512 g_mutex_lock (&priv->lock);
2514 /* Update the range value with current position/duration */
2515 collect_media_stats (media);
2518 range = priv->range;
2520 if (!play && priv->n_active > 0) {
2521 range.min.type = GST_RTSP_TIME_NOW;
2522 range.min.seconds = -1;
2524 g_mutex_unlock (&priv->lock);
2525 g_rec_mutex_unlock (&priv->state_lock);
2527 if (!klass->convert_range (media, &range, unit))
2528 goto conversion_failed;
2530 result = gst_rtsp_range_to_string (&range);
2537 GST_WARNING ("media %p was not prepared", media);
2538 g_rec_mutex_unlock (&priv->state_lock);
2543 GST_WARNING ("range conversion to unit %d failed", unit);
2549 * gst_rtsp_media_get_rates:
2550 * @media: a #GstRTSPMedia
2551 * @rate (allow-none): the rate of the current segment
2552 * @applied_rate (allow-none): the applied_rate of the current segment
2554 * Get the rate and applied_rate of the current segment.
2556 * Returns: %FALSE if looking up the rate and applied rate failed. Otherwise
2557 * %TRUE is returned and @rate and @applied_rate are set to the rate and
2558 * applied_rate of the current segment.
2562 gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
2563 gdouble * applied_rate)
2565 GstRTSPMediaPrivate *priv;
2566 GstRTSPStream *stream;
2567 gdouble save_rate, save_applied_rate;
2568 gboolean result = TRUE;
2569 gboolean first_stream = TRUE;
2572 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2574 if (!rate && !applied_rate) {
2575 GST_WARNING_OBJECT (media, "rate and applied_rate are both NULL");
2581 g_mutex_lock (&priv->lock);
2583 g_assert (priv->streams->len > 0);
2584 for (i = 0; i < priv->streams->len; i++) {
2585 stream = g_ptr_array_index (priv->streams, i);
2586 if (gst_rtsp_stream_is_complete (stream)) {
2587 if (gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
2590 save_applied_rate = *applied_rate;
2591 first_stream = FALSE;
2593 if (save_rate != *rate || save_applied_rate != *applied_rate) {
2594 /* diffrent rate or applied_rate, weird */
2601 /* complete stream withot rate and applied_rate, weird */
2610 GST_WARNING_OBJECT (media,
2611 "failed to obtain consistent rate and applied_rate");
2614 g_mutex_unlock (&priv->lock);
2620 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2622 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2626 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2628 GstRTSPMediaPrivate *priv = media->priv;
2630 GST_DEBUG ("media %p set blocked %d", media, blocked);
2631 priv->blocked = blocked;
2632 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2636 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2638 gst_rtsp_stream_set_blocked (stream, FALSE);
2642 media_unblock (GstRTSPMedia * media)
2644 GstRTSPMediaPrivate *priv = media->priv;
2646 GST_DEBUG ("media %p unblocking streams", media);
2647 /* media is not blocked any longer, as it contains active streams,
2648 * streams that are complete */
2649 priv->blocked = FALSE;
2650 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2654 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2656 GstRTSPMediaPrivate *priv = media->priv;
2658 g_mutex_lock (&priv->lock);
2659 priv->status = status;
2660 GST_DEBUG ("setting new status to %d", status);
2661 g_cond_broadcast (&priv->cond);
2662 g_mutex_unlock (&priv->lock);
2666 * gst_rtsp_media_get_status:
2667 * @media: a #GstRTSPMedia
2669 * Get the status of @media. When @media is busy preparing, this function waits
2670 * until @media is prepared or in error.
2672 * Returns: the status of @media.
2675 gst_rtsp_media_get_status (GstRTSPMedia * media)
2677 GstRTSPMediaPrivate *priv = media->priv;
2678 GstRTSPMediaStatus result;
2681 g_mutex_lock (&priv->lock);
2682 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2683 /* while we are preparing, wait */
2684 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2685 GST_DEBUG ("waiting for status change");
2686 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2687 GST_DEBUG ("timeout, assuming error status");
2688 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2691 /* could be success or error */
2692 result = priv->status;
2693 GST_DEBUG ("got status %d", result);
2694 g_mutex_unlock (&priv->lock);
2700 * gst_rtsp_media_seek_trickmode:
2701 * @media: a #GstRTSPMedia
2702 * @range: (transfer none): a #GstRTSPTimeRange
2703 * @flags: The minimal set of #GstSeekFlags to use
2704 * @rate: the rate to use in the seek
2705 * @trickmode_interval: The trickmode interval to use for KEY_UNITS trick mode
2707 * Seek the pipeline of @media to @range with the given @flags and @rate,
2708 * and @trickmode_interval.
2709 * @media must be prepared with gst_rtsp_media_prepare().
2710 * In order to perform the seek operation, the pipeline must contain all
2711 * needed transport parts (transport sinks).
2713 * Returns: %TRUE on success.
2718 gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
2719 GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate,
2720 GstClockTime trickmode_interval)
2722 GstRTSPMediaClass *klass;
2723 GstRTSPMediaPrivate *priv;
2725 GstClockTime start, stop;
2726 GstSeekType start_type, stop_type;
2727 gint64 current_position;
2728 gboolean force_seek;
2730 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2732 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2733 /* if there's a range then klass->convert_range must be set */
2734 g_return_val_if_fail (range == NULL || klass->convert_range != NULL, FALSE);
2736 GST_DEBUG ("flags=%x rate=%f", flags, rate);
2740 g_rec_mutex_lock (&priv->state_lock);
2741 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2744 /* check if the media pipeline is complete in order to perform a
2745 * seek operation on it */
2746 if (!check_complete (media))
2749 /* Update the seekable state of the pipeline in case it changed */
2750 check_seekable (media);
2752 if (priv->seekable == 0) {
2753 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2754 " not seekable streams.");
2757 } else if (priv->seekable < 0) {
2761 start_type = stop_type = GST_SEEK_TYPE_NONE;
2762 start = stop = GST_CLOCK_TIME_NONE;
2764 /* if caller provided a range convert it to NPT format
2765 * if no range provided the seek is assumed to be the same position but with
2766 * e.g. the rate changed */
2767 if (range != NULL) {
2768 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2770 gst_rtsp_range_get_times (range, &start, &stop);
2772 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2773 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2774 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2775 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2778 current_position = -1;
2779 if (klass->query_position)
2780 klass->query_position (media, ¤t_position);
2781 GST_INFO ("current media position %" GST_TIME_FORMAT,
2782 GST_TIME_ARGS (current_position));
2784 if (start != GST_CLOCK_TIME_NONE)
2785 start_type = GST_SEEK_TYPE_SET;
2787 if (stop != GST_CLOCK_TIME_NONE)
2788 stop_type = GST_SEEK_TYPE_SET;
2790 /* we force a seek if any trickmode flag is set, or if the rate
2791 * is non-standard, i.e. not 1.0 */
2792 force_seek = (flags & TRICKMODE_FLAGS) || rate != 1.0;
2794 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE || force_seek) {
2795 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2796 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2798 /* depends on the current playing state of the pipeline. We might need to
2799 * queue this until we get EOS. */
2800 flags |= GST_SEEK_FLAG_FLUSH;
2802 /* if range start was not supplied we must continue from current position.
2803 * but since we're doing a flushing seek, let us query the current position
2804 * so we end up at exactly the same position after the seek. */
2805 if (range == NULL || range->min.type == GST_RTSP_TIME_END) {
2806 if (current_position == -1) {
2807 GST_WARNING ("current position unknown");
2809 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2810 GST_TIME_ARGS (current_position));
2811 start = current_position;
2812 start_type = GST_SEEK_TYPE_SET;
2816 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE &&
2818 GST_DEBUG ("no position change, no flags set by caller, so not seeking");
2821 GstEvent *seek_event;
2822 gboolean unblock = FALSE;
2824 /* Handle expected async-done before waiting on next async-done.
2826 * Since the seek further down in code will cause a preroll and
2827 * a async-done will be generated it's important to wait on async-done
2828 * if that is expected. Otherwise there is the risk that the waiting
2829 * for async-done after the seek is detecting the expected async-done
2830 * instead of the one that corresponds to the seek. Then execution
2831 * continue and act as if the pipeline is prerolled, but it's not.
2833 * During wait_preroll message GST_MESSAGE_ASYNC_DONE will come
2834 * and then the state will change from preparing to prepared */
2835 if (priv->expected_async_done) {
2836 GST_DEBUG (" expected to get async-done, waiting ");
2837 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2838 g_rec_mutex_unlock (&priv->state_lock);
2840 /* wait until pipeline is prerolled */
2841 if (!wait_preroll (media))
2842 goto preroll_failed_expected_async_done;
2844 g_rec_mutex_lock (&priv->state_lock);
2845 GST_DEBUG (" got expected async-done");
2848 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2851 GstClockTime temp_time = start;
2852 GstSeekType temp_type = start_type;
2855 start_type = stop_type;
2857 stop_type = temp_type;
2860 seek_event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type,
2861 start, stop_type, stop);
2863 gst_event_set_seek_trickmode_interval (seek_event, trickmode_interval);
2865 if (!media->priv->blocked) {
2866 /* Prevent a race condition with multiple streams,
2867 * where one stream may have time to preroll before others
2868 * have even started flushing, causing async-done to be
2871 media_streams_set_blocked (media, TRUE);
2875 res = gst_element_send_event (priv->pipeline, seek_event);
2878 media_streams_set_blocked (media, FALSE);
2880 /* and block for the seek to complete */
2881 GST_INFO ("done seeking %d", res);
2885 g_rec_mutex_unlock (&priv->state_lock);
2887 /* wait until pipeline is prerolled again, this will also collect stats */
2888 if (!wait_preroll (media))
2889 goto preroll_failed;
2891 g_rec_mutex_lock (&priv->state_lock);
2892 GST_INFO ("prerolled again");
2895 GST_INFO ("no seek needed");
2898 g_rec_mutex_unlock (&priv->state_lock);
2905 g_rec_mutex_unlock (&priv->state_lock);
2906 GST_INFO ("media %p is not prepared", media);
2911 g_rec_mutex_unlock (&priv->state_lock);
2912 GST_INFO ("pipeline is not complete");
2917 g_rec_mutex_unlock (&priv->state_lock);
2918 GST_INFO ("pipeline is not seekable");
2923 g_rec_mutex_unlock (&priv->state_lock);
2924 GST_WARNING ("conversion to npt not supported");
2929 g_rec_mutex_unlock (&priv->state_lock);
2930 GST_INFO ("seeking failed");
2931 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2936 GST_WARNING ("failed to preroll after seek");
2939 preroll_failed_expected_async_done:
2941 GST_WARNING ("failed to preroll");
2947 * gst_rtsp_media_seek_full:
2948 * @media: a #GstRTSPMedia
2949 * @range: (transfer none): a #GstRTSPTimeRange
2950 * @flags: The minimal set of #GstSeekFlags to use
2952 * Seek the pipeline of @media to @range with the given @flags.
2953 * @media must be prepared with gst_rtsp_media_prepare().
2955 * Returns: %TRUE on success.
2959 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2962 return gst_rtsp_media_seek_trickmode (media, range, flags, 1.0, 0);
2966 * gst_rtsp_media_seek:
2967 * @media: a #GstRTSPMedia
2968 * @range: (transfer none): a #GstRTSPTimeRange
2970 * Seek the pipeline of @media to @range. @media must be prepared with
2971 * gst_rtsp_media_prepare().
2973 * Returns: %TRUE on success.
2976 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2978 return gst_rtsp_media_seek_trickmode (media, range, GST_SEEK_FLAG_NONE,
2983 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2985 *blocked &= gst_rtsp_stream_is_blocking (stream);
2989 media_streams_blocking (GstRTSPMedia * media)
2991 gboolean blocking = TRUE;
2993 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2999 static GstStateChangeReturn
3000 set_state (GstRTSPMedia * media, GstState state)
3002 GstRTSPMediaPrivate *priv = media->priv;
3003 GstStateChangeReturn ret;
3005 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
3007 ret = gst_element_set_state (priv->pipeline, state);
3012 static GstStateChangeReturn
3013 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
3015 GstRTSPMediaPrivate *priv = media->priv;
3016 GstStateChangeReturn ret;
3018 GST_INFO ("set target state to %s for media %p",
3019 gst_element_state_get_name (state), media);
3020 priv->target_state = state;
3022 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
3023 priv->target_state, NULL);
3026 ret = set_state (media, state);
3028 ret = GST_STATE_CHANGE_SUCCESS;
3033 /* called with state-lock */
3035 default_handle_message (GstRTSPMedia * media, GstMessage * message)
3037 GstRTSPMediaPrivate *priv = media->priv;
3038 GstMessageType type;
3040 type = GST_MESSAGE_TYPE (message);
3043 case GST_MESSAGE_STATE_CHANGED:
3045 GstState old, new, pending;
3047 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
3050 gst_message_parse_state_changed (message, &old, &new, &pending);
3052 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
3053 gst_element_state_get_name (old), gst_element_state_get_name (new),
3054 gst_element_state_get_name (pending));
3055 if (priv->no_more_pads_pending == 0
3056 && gst_rtsp_media_is_receive_only (media) && old == GST_STATE_READY
3057 && new == GST_STATE_PAUSED) {
3058 GST_INFO ("%p: went to PAUSED, prepared now", media);
3059 collect_media_stats (media);
3061 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3062 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3067 case GST_MESSAGE_BUFFERING:
3071 gst_message_parse_buffering (message, &percent);
3073 /* no state management needed for live pipelines */
3077 if (percent == 100) {
3078 /* a 100% message means buffering is done */
3079 priv->buffering = FALSE;
3080 /* if the desired state is playing, go back */
3081 if (priv->target_state == GST_STATE_PLAYING) {
3082 GST_INFO ("Buffering done, setting pipeline to PLAYING");
3083 set_state (media, GST_STATE_PLAYING);
3085 GST_INFO ("Buffering done");
3088 /* buffering busy */
3089 if (priv->buffering == FALSE) {
3090 if (priv->target_state == GST_STATE_PLAYING) {
3091 /* we were not buffering but PLAYING, PAUSE the pipeline. */
3092 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
3093 set_state (media, GST_STATE_PAUSED);
3095 GST_INFO ("Buffering ...");
3098 priv->buffering = TRUE;
3102 case GST_MESSAGE_LATENCY:
3104 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
3107 case GST_MESSAGE_ERROR:
3112 gst_message_parse_error (message, &gerror, &debug);
3113 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
3114 g_error_free (gerror);
3117 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3120 case GST_MESSAGE_WARNING:
3125 gst_message_parse_warning (message, &gerror, &debug);
3126 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
3127 g_error_free (gerror);
3131 case GST_MESSAGE_ELEMENT:
3133 const GstStructure *s;
3135 s = gst_message_get_structure (message);
3136 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
3137 GST_DEBUG ("media received blocking message");
3138 if (priv->blocked && media_streams_blocking (media) &&
3139 priv->no_more_pads_pending == 0) {
3140 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
3141 collect_media_stats (media);
3143 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3144 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3149 case GST_MESSAGE_STREAM_STATUS:
3151 case GST_MESSAGE_ASYNC_DONE:
3152 if (priv->expected_async_done)
3153 priv->expected_async_done = FALSE;
3154 if (priv->complete) {
3155 /* receive the final ASYNC_DONE, that is posted by the media pipeline
3156 * after all the transport parts have been successfully added to
3157 * the media streams. */
3158 GST_DEBUG_OBJECT (media, "got async-done");
3159 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3160 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3163 case GST_MESSAGE_EOS:
3164 GST_INFO ("%p: got EOS", media);
3166 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
3167 GST_DEBUG ("shutting down after EOS");
3168 finish_unprepare (media);
3172 GST_INFO ("%p: got message type %d (%s)", media, type,
3173 gst_message_type_get_name (type));
3180 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
3182 GstRTSPMediaPrivate *priv = media->priv;
3183 GstRTSPMediaClass *klass;
3186 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3188 g_rec_mutex_lock (&priv->state_lock);
3189 if (klass->handle_message)
3190 ret = klass->handle_message (media, message);
3193 g_rec_mutex_unlock (&priv->state_lock);
3199 watch_destroyed (GstRTSPMedia * media)
3201 GST_DEBUG_OBJECT (media, "source destroyed");
3202 g_object_unref (media);
3206 is_payloader (GstElement * element)
3208 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
3211 klass = gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
3215 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
3223 find_payload_element (GstElement * payloader, GstPad * pad)
3225 GstElement *pay = NULL;
3227 if (GST_IS_BIN (payloader)) {
3229 GValue item = { 0 };
3230 gchar *pad_name, *payloader_name;
3231 GstElement *element;
3233 if ((element = gst_bin_get_by_name (GST_BIN (payloader), "pay"))) {
3234 if (is_payloader (element))
3236 gst_object_unref (element);
3239 pad_name = gst_object_get_name (GST_OBJECT (pad));
3240 payloader_name = g_strdup_printf ("pay_%s", pad_name);
3242 if ((element = gst_bin_get_by_name (GST_BIN (payloader), payloader_name))) {
3243 g_free (payloader_name);
3244 if (is_payloader (element))
3246 gst_object_unref (element);
3248 g_free (payloader_name);
3251 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
3252 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
3253 element = (GstElement *) g_value_get_object (&item);
3255 if (is_payloader (element)) {
3256 pay = gst_object_ref (element);
3257 g_value_unset (&item);
3260 g_value_unset (&item);
3262 gst_iterator_free (iter);
3264 pay = g_object_ref (payloader);
3270 /* called from streaming threads */
3272 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3274 GstRTSPMediaPrivate *priv = media->priv;
3275 GstRTSPStream *stream;
3278 /* find the real payload element */
3279 pay = find_payload_element (element, pad);
3280 stream = gst_rtsp_media_create_stream (media, pay, pad);
3281 gst_object_unref (pay);
3283 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3285 g_rec_mutex_lock (&priv->state_lock);
3286 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3289 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
3291 /* join the element in the PAUSED state because this callback is
3292 * called from the streaming thread and it is PAUSED */
3293 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3294 priv->rtpbin, GST_STATE_PAUSED)) {
3295 GST_WARNING ("failed to join bin element");
3299 gst_rtsp_stream_set_blocked (stream, TRUE);
3301 g_rec_mutex_unlock (&priv->state_lock);
3308 gst_rtsp_media_remove_stream (media, stream);
3309 g_rec_mutex_unlock (&priv->state_lock);
3310 GST_INFO ("ignore pad because we are not preparing");
3316 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3318 GstRTSPMediaPrivate *priv = media->priv;
3319 GstRTSPStream *stream;
3321 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
3325 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3327 g_rec_mutex_lock (&priv->state_lock);
3328 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3329 g_rec_mutex_unlock (&priv->state_lock);
3331 gst_rtsp_media_remove_stream (media, stream);
3335 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
3337 GstRTSPMediaPrivate *priv = media->priv;
3339 GST_INFO_OBJECT (element, "no more pads");
3340 g_mutex_lock (&priv->lock);
3341 priv->no_more_pads_pending--;
3342 g_mutex_unlock (&priv->lock);
3345 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
3347 struct _DynPaySignalHandlers
3349 gulong pad_added_handler;
3350 gulong pad_removed_handler;
3351 gulong no_more_pads_handler;
3355 start_preroll (GstRTSPMedia * media)
3357 GstRTSPMediaPrivate *priv = media->priv;
3358 GstStateChangeReturn ret;
3360 GST_INFO ("setting pipeline to PAUSED for media %p", media);
3362 /* start blocked since it is possible that there are no sink elements yet */
3363 media_streams_set_blocked (media, TRUE);
3364 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3367 case GST_STATE_CHANGE_SUCCESS:
3368 GST_INFO ("SUCCESS state change for media %p", media);
3370 case GST_STATE_CHANGE_ASYNC:
3371 GST_INFO ("ASYNC state change for media %p", media);
3373 case GST_STATE_CHANGE_NO_PREROLL:
3374 /* we need to go to PLAYING */
3375 GST_INFO ("NO_PREROLL state change: live media %p", media);
3376 /* FIXME we disable seeking for live streams for now. We should perform a
3377 * seeking query in preroll instead */
3378 priv->seekable = -1;
3379 priv->is_live = TRUE;
3381 ret = set_state (media, GST_STATE_PLAYING);
3382 if (ret == GST_STATE_CHANGE_FAILURE)
3385 case GST_STATE_CHANGE_FAILURE:
3393 GST_WARNING ("failed to preroll pipeline");
3399 wait_preroll (GstRTSPMedia * media)
3401 GstRTSPMediaStatus status;
3403 GST_DEBUG ("wait to preroll pipeline");
3405 /* wait until pipeline is prerolled */
3406 status = gst_rtsp_media_get_status (media);
3407 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3408 goto preroll_failed;
3414 GST_WARNING ("failed to preroll pipeline");
3420 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3422 GstRTSPMediaPrivate *priv = media->priv;
3423 GstRTSPStream *stream = NULL;
3425 GstElement *res = NULL;
3427 g_mutex_lock (&priv->lock);
3428 for (i = 0; i < priv->streams->len; i++) {
3429 stream = g_ptr_array_index (priv->streams, i);
3431 if (sessid == gst_rtsp_stream_get_index (stream))
3436 g_mutex_unlock (&priv->lock);
3439 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3445 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3447 GstRTSPMediaPrivate *priv = media->priv;
3448 GstRTSPStream *stream = NULL;
3450 GstElement *res = NULL;
3452 g_mutex_lock (&priv->lock);
3453 for (i = 0; i < priv->streams->len; i++) {
3454 stream = g_ptr_array_index (priv->streams, i);
3456 if (sessid == gst_rtsp_stream_get_index (stream))
3461 g_mutex_unlock (&priv->lock);
3464 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3470 request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3472 GstRTSPMediaPrivate *priv = media->priv;
3473 GstRTSPStream *stream = NULL;
3475 GstElement *res = NULL;
3477 g_mutex_lock (&priv->lock);
3478 for (i = 0; i < priv->streams->len; i++) {
3479 stream = g_ptr_array_index (priv->streams, i);
3481 if (sessid == gst_rtsp_stream_get_index (stream))
3486 g_mutex_unlock (&priv->lock);
3489 res = gst_rtsp_stream_request_ulpfec_decoder (stream, rtpbin, sessid);
3496 new_storage_cb (GstElement * rtpbin, GObject * storage, guint sessid,
3497 GstRTSPMedia * media)
3499 g_object_set (storage, "size-time", (media->priv->latency + 50) * GST_MSECOND,
3504 start_prepare (GstRTSPMedia * media)
3506 GstRTSPMediaPrivate *priv = media->priv;
3510 g_rec_mutex_lock (&priv->state_lock);
3511 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3512 goto no_longer_preparing;
3514 g_signal_connect (priv->rtpbin, "new-storage", G_CALLBACK (new_storage_cb),
3516 g_signal_connect (priv->rtpbin, "request-fec-decoder",
3517 G_CALLBACK (request_fec_decoder), media);
3519 /* link streams we already have, other streams might appear when we have
3520 * dynamic elements */
3521 for (i = 0; i < priv->streams->len; i++) {
3522 GstRTSPStream *stream;
3524 stream = g_ptr_array_index (priv->streams, i);
3526 if (priv->rtx_time > 0) {
3527 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3528 g_signal_connect (priv->rtpbin, "request-aux-sender",
3529 (GCallback) request_aux_sender, media);
3532 if (priv->do_retransmission) {
3533 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3534 (GCallback) request_aux_receiver, media);
3537 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3538 priv->rtpbin, GST_STATE_NULL)) {
3539 goto join_bin_failed;
3544 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
3545 "do-lost", TRUE, NULL);
3547 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3548 GstElement *elem = walk->data;
3549 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3551 GST_INFO ("adding callbacks for dynamic element %p", elem);
3553 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3554 (GCallback) pad_added_cb, media);
3555 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3556 (GCallback) pad_removed_cb, media);
3557 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3558 (GCallback) no_more_pads_cb, media);
3560 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3563 if (priv->nb_dynamic_elements == 0 && gst_rtsp_media_is_receive_only (media)) {
3564 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3565 * a second ASYNC state change failing */
3566 priv->is_live = TRUE;
3567 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3568 } else if (!start_preroll (media)) {
3569 goto preroll_failed;
3572 g_rec_mutex_unlock (&priv->state_lock);
3576 no_longer_preparing:
3578 GST_INFO ("media is no longer preparing");
3579 g_rec_mutex_unlock (&priv->state_lock);
3584 GST_WARNING ("failed to join bin element");
3585 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3586 g_rec_mutex_unlock (&priv->state_lock);
3591 GST_WARNING ("failed to preroll pipeline");
3592 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3593 g_rec_mutex_unlock (&priv->state_lock);
3599 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3601 GstRTSPMediaPrivate *priv;
3602 GstRTSPMediaClass *klass;
3604 GMainContext *context;
3609 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3611 if (!klass->create_rtpbin)
3612 goto no_create_rtpbin;
3614 priv->rtpbin = klass->create_rtpbin (media);
3615 if (priv->rtpbin != NULL) {
3616 gboolean success = TRUE;
3618 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3620 if (klass->setup_rtpbin)
3621 success = klass->setup_rtpbin (media, priv->rtpbin);
3623 if (success == FALSE) {
3624 gst_object_unref (priv->rtpbin);
3625 priv->rtpbin = NULL;
3628 if (priv->rtpbin == NULL)
3631 priv->thread = thread;
3632 context = (thread != NULL) ? (thread->context) : NULL;
3634 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3636 /* add the pipeline bus to our custom mainloop */
3637 priv->source = gst_bus_create_watch (bus);
3638 gst_object_unref (bus);
3640 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3641 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3643 priv->id = g_source_attach (priv->source, context);
3645 /* add stuff to the bin */
3646 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3648 /* do remainder in context */
3649 source = g_idle_source_new ();
3650 g_source_set_callback (source, (GSourceFunc) start_prepare,
3651 g_object_ref (media), (GDestroyNotify) g_object_unref);
3652 g_source_attach (source, context);
3653 g_source_unref (source);
3660 GST_ERROR ("no create_rtpbin function");
3661 g_critical ("no create_rtpbin vmethod function set");
3666 GST_WARNING ("no rtpbin element");
3667 g_warning ("failed to create element 'rtpbin', check your installation");
3673 * gst_rtsp_media_prepare:
3674 * @media: a #GstRTSPMedia
3675 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3676 * bus handler or %NULL
3678 * Prepare @media for streaming. This function will create the objects
3679 * to manage the streaming. A pipeline must have been set on @media with
3680 * gst_rtsp_media_take_pipeline().
3682 * It will preroll the pipeline and collect vital information about the streams
3683 * such as the duration.
3685 * Returns: %TRUE on success.
3688 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3690 GstRTSPMediaPrivate *priv;
3691 GstRTSPMediaClass *klass;
3693 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3697 g_rec_mutex_lock (&priv->state_lock);
3698 priv->prepare_count++;
3700 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3701 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3704 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3707 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3708 goto not_unprepared;
3710 if (!priv->reusable && priv->reused)
3713 GST_INFO ("preparing media %p", media);
3715 /* reset some variables */
3716 priv->is_live = FALSE;
3717 priv->seekable = -1;
3718 priv->buffering = FALSE;
3719 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3721 /* we're preparing now */
3722 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3724 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3725 if (klass->prepare) {
3726 if (!klass->prepare (media, thread))
3727 goto prepare_failed;
3731 g_rec_mutex_unlock (&priv->state_lock);
3733 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3734 * able to do this async so that we don't block the server thread. */
3735 if (!wait_preroll (media))
3736 goto preroll_failed;
3738 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3740 GST_INFO ("object %p is prerolled", media);
3747 /* we are not going to use the giving thread, so stop it. */
3749 gst_rtsp_thread_stop (thread);
3754 GST_LOG ("media %p was prepared", media);
3755 /* we are not going to use the giving thread, so stop it. */
3757 gst_rtsp_thread_stop (thread);
3758 g_rec_mutex_unlock (&priv->state_lock);
3764 /* we are not going to use the giving thread, so stop it. */
3766 gst_rtsp_thread_stop (thread);
3767 GST_WARNING ("media %p was not unprepared", media);
3768 priv->prepare_count--;
3769 g_rec_mutex_unlock (&priv->state_lock);
3774 /* we are not going to use the giving thread, so stop it. */
3776 gst_rtsp_thread_stop (thread);
3777 priv->prepare_count--;
3778 g_rec_mutex_unlock (&priv->state_lock);
3779 GST_WARNING ("can not reuse media %p", media);
3784 /* we are not going to use the giving thread, so stop it. */
3786 gst_rtsp_thread_stop (thread);
3787 priv->prepare_count--;
3788 g_rec_mutex_unlock (&priv->state_lock);
3789 GST_ERROR ("failed to prepare media");
3794 GST_WARNING ("failed to preroll pipeline");
3795 gst_rtsp_media_unprepare (media);
3800 /* must be called with state-lock */
3802 finish_unprepare (GstRTSPMedia * media)
3804 GstRTSPMediaPrivate *priv = media->priv;
3808 if (priv->finishing_unprepare)
3810 priv->finishing_unprepare = TRUE;
3812 GST_DEBUG ("shutting down");
3814 /* release the lock on shutdown, otherwise pad_added_cb might try to
3815 * acquire the lock and then we deadlock */
3816 g_rec_mutex_unlock (&priv->state_lock);
3817 set_state (media, GST_STATE_NULL);
3818 g_rec_mutex_lock (&priv->state_lock);
3820 media_streams_set_blocked (media, FALSE);
3822 for (i = 0; i < priv->streams->len; i++) {
3823 GstRTSPStream *stream;
3825 GST_INFO ("Removing elements of stream %d from pipeline", i);
3827 stream = g_ptr_array_index (priv->streams, i);
3829 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3832 /* remove the pad signal handlers */
3833 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3834 GstElement *elem = walk->data;
3835 DynPaySignalHandlers *handlers;
3838 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3839 g_assert (handlers != NULL);
3841 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3842 g_signal_handler_disconnect (G_OBJECT (elem),
3843 handlers->pad_removed_handler);
3844 g_signal_handler_disconnect (G_OBJECT (elem),
3845 handlers->no_more_pads_handler);
3847 g_slice_free (DynPaySignalHandlers, handlers);
3850 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3851 priv->rtpbin = NULL;
3854 gst_object_unref (priv->nettime);
3855 priv->nettime = NULL;
3857 priv->reused = TRUE;
3858 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3860 /* when the media is not reusable, this will effectively unref the media and
3862 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3864 /* the source has the last ref to the media */
3866 GST_DEBUG ("destroy source");
3867 g_source_destroy (priv->source);
3868 g_source_unref (priv->source);
3871 GST_DEBUG ("stop thread");
3872 gst_rtsp_thread_stop (priv->thread);
3875 priv->finishing_unprepare = FALSE;
3878 /* called with state-lock */
3880 default_unprepare (GstRTSPMedia * media)
3882 GstRTSPMediaPrivate *priv = media->priv;
3884 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3886 if (priv->eos_shutdown) {
3887 GST_DEBUG ("sending EOS for shutdown");
3888 /* ref so that we don't disappear */
3889 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3890 /* we need to go to playing again for the EOS to propagate, normally in this
3891 * state, nothing is receiving data from us anymore so this is ok. */
3892 set_state (media, GST_STATE_PLAYING);
3894 finish_unprepare (media);
3900 * gst_rtsp_media_unprepare:
3901 * @media: a #GstRTSPMedia
3903 * Unprepare @media. After this call, the media should be prepared again before
3904 * it can be used again. If the media is set to be non-reusable, a new instance
3907 * Returns: %TRUE on success.
3910 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3912 GstRTSPMediaPrivate *priv;
3915 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3919 g_rec_mutex_lock (&priv->state_lock);
3920 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3921 goto was_unprepared;
3923 priv->prepare_count--;
3924 if (priv->prepare_count > 0)
3927 GST_INFO ("unprepare media %p", media);
3928 set_target_state (media, GST_STATE_NULL, FALSE);
3931 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3932 GstRTSPMediaClass *klass;
3934 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3935 if (klass->unprepare)
3936 success = klass->unprepare (media);
3938 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3939 finish_unprepare (media);
3941 g_rec_mutex_unlock (&priv->state_lock);
3947 g_rec_mutex_unlock (&priv->state_lock);
3948 GST_INFO ("media %p was already unprepared", media);
3953 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3954 g_rec_mutex_unlock (&priv->state_lock);
3959 /* should be called with state-lock */
3961 get_clock_unlocked (GstRTSPMedia * media)
3963 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3964 GST_DEBUG_OBJECT (media, "media was not prepared");
3967 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3971 * gst_rtsp_media_get_clock:
3972 * @media: a #GstRTSPMedia
3974 * Get the clock that is used by the pipeline in @media.
3976 * @media must be prepared before this method returns a valid clock object.
3978 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3981 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3984 GstRTSPMediaPrivate *priv;
3986 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3990 g_rec_mutex_lock (&priv->state_lock);
3991 clock = get_clock_unlocked (media);
3992 g_rec_mutex_unlock (&priv->state_lock);
3998 * gst_rtsp_media_get_base_time:
3999 * @media: a #GstRTSPMedia
4001 * Get the base_time that is used by the pipeline in @media.
4003 * @media must be prepared before this method returns a valid base_time.
4005 * Returns: the base_time used by @media.
4008 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
4010 GstClockTime result;
4011 GstRTSPMediaPrivate *priv;
4013 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
4017 g_rec_mutex_lock (&priv->state_lock);
4018 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
4021 result = gst_element_get_base_time (media->priv->pipeline);
4022 g_rec_mutex_unlock (&priv->state_lock);
4029 g_rec_mutex_unlock (&priv->state_lock);
4030 GST_DEBUG_OBJECT (media, "media was not prepared");
4031 return GST_CLOCK_TIME_NONE;
4036 * gst_rtsp_media_get_time_provider:
4037 * @media: a #GstRTSPMedia
4038 * @address: (allow-none): an address or %NULL
4039 * @port: a port or 0
4041 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
4042 * will listen on @address and @port for client time requests.
4044 * Returns: (transfer full): the #GstNetTimeProvider of @media.
4046 GstNetTimeProvider *
4047 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
4050 GstRTSPMediaPrivate *priv;
4051 GstNetTimeProvider *provider = NULL;
4053 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
4057 g_rec_mutex_lock (&priv->state_lock);
4058 if (priv->time_provider) {
4059 if ((provider = priv->nettime) == NULL) {
4062 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
4063 provider = gst_net_time_provider_new (clock, address, port);
4064 gst_object_unref (clock);
4066 priv->nettime = provider;
4070 g_rec_mutex_unlock (&priv->state_lock);
4073 gst_object_ref (provider);
4079 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
4081 return gst_rtsp_sdp_from_media (sdp, info, media);
4085 * gst_rtsp_media_setup_sdp:
4086 * @media: a #GstRTSPMedia
4087 * @sdp: (transfer none): a #GstSDPMessage
4088 * @info: (transfer none): a #GstSDPInfo
4090 * Add @media specific info to @sdp. @info is used to configure the connection
4091 * information in the SDP.
4093 * Returns: TRUE on success.
4096 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
4099 GstRTSPMediaPrivate *priv;
4100 GstRTSPMediaClass *klass;
4103 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4104 g_return_val_if_fail (sdp != NULL, FALSE);
4105 g_return_val_if_fail (info != NULL, FALSE);
4109 g_rec_mutex_lock (&priv->state_lock);
4111 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4113 if (!klass->setup_sdp)
4116 res = klass->setup_sdp (media, sdp, info);
4118 g_rec_mutex_unlock (&priv->state_lock);
4125 g_rec_mutex_unlock (&priv->state_lock);
4126 GST_ERROR ("no setup_sdp function");
4127 g_critical ("no setup_sdp vmethod function set");
4133 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4135 GstRTSPMediaPrivate *priv = media->priv;
4138 medias_len = gst_sdp_message_medias_len (sdp);
4139 if (medias_len != priv->streams->len) {
4140 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
4141 priv->streams->len, medias_len);
4145 for (i = 0; i < medias_len; i++) {
4147 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
4148 GstRTSPStream *stream;
4149 gint j, formats_len;
4150 const gchar *control;
4151 GstRTSPProfile profile, profiles;
4153 stream = g_ptr_array_index (priv->streams, i);
4155 /* TODO: Should we do something with the other SDP information? */
4158 proto = gst_sdp_media_get_proto (sdp_media);
4159 if (proto == NULL) {
4160 GST_ERROR ("%p: SDP media %d has no proto", media, i);
4164 if (g_str_equal (proto, "RTP/AVP")) {
4165 profile = GST_RTSP_PROFILE_AVP;
4166 } else if (g_str_equal (proto, "RTP/SAVP")) {
4167 profile = GST_RTSP_PROFILE_SAVP;
4168 } else if (g_str_equal (proto, "RTP/AVPF")) {
4169 profile = GST_RTSP_PROFILE_AVPF;
4170 } else if (g_str_equal (proto, "RTP/SAVPF")) {
4171 profile = GST_RTSP_PROFILE_SAVPF;
4173 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4177 profiles = gst_rtsp_stream_get_profiles (stream);
4178 if ((profiles & profile) == 0) {
4179 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4183 formats_len = gst_sdp_media_formats_len (sdp_media);
4184 for (j = 0; j < formats_len; j++) {
4189 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
4191 GST_DEBUG (" looking at %d pt: %d", j, pt);
4194 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
4196 GST_WARNING (" skipping pt %d without caps", pt);
4200 /* do some tweaks */
4201 GST_DEBUG ("mapping sdp session level attributes to caps");
4202 gst_sdp_message_attributes_to_caps (sdp, caps);
4203 GST_DEBUG ("mapping sdp media level attributes to caps");
4204 gst_sdp_media_attributes_to_caps (sdp_media, caps);
4206 s = gst_caps_get_structure (caps, 0);
4207 gst_structure_set_name (s, "application/x-rtp");
4209 if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
4210 gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
4212 gst_rtsp_stream_set_pt_map (stream, pt, caps);
4213 gst_caps_unref (caps);
4216 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
4218 gst_rtsp_stream_set_control (stream, control);
4226 * gst_rtsp_media_handle_sdp:
4227 * @media: a #GstRTSPMedia
4228 * @sdp: (transfer none): a #GstSDPMessage
4230 * Configure an SDP on @media for receiving streams
4232 * Returns: TRUE on success.
4235 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4237 GstRTSPMediaPrivate *priv;
4238 GstRTSPMediaClass *klass;
4241 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4242 g_return_val_if_fail (sdp != NULL, FALSE);
4246 g_rec_mutex_lock (&priv->state_lock);
4248 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4250 if (!klass->handle_sdp)
4253 res = klass->handle_sdp (media, sdp);
4255 g_rec_mutex_unlock (&priv->state_lock);
4262 g_rec_mutex_unlock (&priv->state_lock);
4263 GST_ERROR ("no handle_sdp function");
4264 g_critical ("no handle_sdp vmethod function set");
4270 do_set_seqnum (GstRTSPStream * stream)
4274 if (gst_rtsp_stream_is_sender (stream)) {
4275 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
4276 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
4280 /* call with state_lock */
4282 default_suspend (GstRTSPMedia * media)
4284 GstRTSPMediaPrivate *priv = media->priv;
4285 GstStateChangeReturn ret;
4287 switch (priv->suspend_mode) {
4288 case GST_RTSP_SUSPEND_MODE_NONE:
4289 GST_DEBUG ("media %p no suspend", media);
4291 case GST_RTSP_SUSPEND_MODE_PAUSE:
4292 GST_DEBUG ("media %p suspend to PAUSED", media);
4293 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
4294 if (ret == GST_STATE_CHANGE_FAILURE)
4297 case GST_RTSP_SUSPEND_MODE_RESET:
4298 GST_DEBUG ("media %p suspend to NULL", media);
4299 ret = set_target_state (media, GST_STATE_NULL, TRUE);
4300 if (ret == GST_STATE_CHANGE_FAILURE)
4302 /* Because payloader needs to set the sequence number as
4303 * monotonic, we need to preserve the sequence number
4304 * after pause. (otherwise going from pause to play, which
4305 * is actually from NULL to PLAY will create a new sequence
4307 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
4318 GST_WARNING ("failed changing pipeline's state for media %p", media);
4324 * gst_rtsp_media_suspend:
4325 * @media: a #GstRTSPMedia
4327 * Suspend @media. The state of the pipeline managed by @media is set to
4328 * GST_STATE_NULL but all streams are kept. @media can be prepared again
4329 * with gst_rtsp_media_unsuspend()
4331 * @media must be prepared with gst_rtsp_media_prepare();
4333 * Returns: %TRUE on success.
4336 gst_rtsp_media_suspend (GstRTSPMedia * media)
4338 GstRTSPMediaPrivate *priv = media->priv;
4339 GstRTSPMediaClass *klass;
4341 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4343 GST_FIXME ("suspend for dynamic pipelines needs fixing");
4345 g_rec_mutex_lock (&priv->state_lock);
4346 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
4349 /* don't attempt to suspend when something is busy */
4350 if (priv->n_active > 0)
4353 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4354 if (klass->suspend) {
4355 if (!klass->suspend (media))
4356 goto suspend_failed;
4359 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
4361 g_rec_mutex_unlock (&priv->state_lock);
4368 g_rec_mutex_unlock (&priv->state_lock);
4369 GST_WARNING ("media %p was not prepared", media);
4374 g_rec_mutex_unlock (&priv->state_lock);
4375 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4376 GST_WARNING ("failed to suspend media %p", media);
4381 /* call with state_lock */
4383 default_unsuspend (GstRTSPMedia * media)
4385 GstRTSPMediaPrivate *priv = media->priv;
4386 gboolean preroll_ok;
4388 switch (priv->suspend_mode) {
4389 case GST_RTSP_SUSPEND_MODE_NONE:
4390 if (gst_rtsp_media_is_receive_only (media))
4392 if (media_streams_blocking (media)) {
4393 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4394 /* at this point the media pipeline has been updated and contain all
4395 * specific transport parts: all active streams contain at least one sink
4396 * element and it's safe to unblock all blocked streams */
4397 media_unblock (media);
4399 /* streams are not blocked and media is suspended from PAUSED */
4400 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4402 g_rec_mutex_unlock (&priv->state_lock);
4403 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
4404 g_rec_mutex_lock (&priv->state_lock);
4405 goto preroll_failed;
4407 g_rec_mutex_lock (&priv->state_lock);
4409 case GST_RTSP_SUSPEND_MODE_PAUSE:
4410 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4412 case GST_RTSP_SUSPEND_MODE_RESET:
4414 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4415 /* at this point the media pipeline has been updated and contain all
4416 * specific transport parts: all active streams contain at least one sink
4417 * element and it's safe to unblock all blocked streams */
4418 media_unblock (media);
4419 if (!start_preroll (media))
4422 g_rec_mutex_unlock (&priv->state_lock);
4423 preroll_ok = wait_preroll (media);
4424 g_rec_mutex_lock (&priv->state_lock);
4427 goto preroll_failed;
4438 GST_WARNING ("failed to preroll pipeline");
4443 GST_WARNING ("failed to preroll pipeline");
4449 * gst_rtsp_media_unsuspend:
4450 * @media: a #GstRTSPMedia
4452 * Unsuspend @media if it was in a suspended state. This method does nothing
4453 * when the media was not in the suspended state.
4455 * Returns: %TRUE on success.
4458 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4460 GstRTSPMediaPrivate *priv = media->priv;
4461 GstRTSPMediaClass *klass;
4463 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4465 g_rec_mutex_lock (&priv->state_lock);
4466 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4469 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4470 if (klass->unsuspend) {
4471 if (!klass->unsuspend (media))
4472 goto unsuspend_failed;
4476 g_rec_mutex_unlock (&priv->state_lock);
4483 g_rec_mutex_unlock (&priv->state_lock);
4484 GST_WARNING ("failed to unsuspend media %p", media);
4485 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4490 /* must be called with state-lock */
4492 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4494 GstRTSPMediaPrivate *priv = media->priv;
4495 GstStateChangeReturn set_state_ret;
4496 priv->expected_async_done = FALSE;
4498 if (state == GST_STATE_NULL) {
4499 gst_rtsp_media_unprepare (media);
4501 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4502 set_target_state (media, state, FALSE);
4503 /* when we are buffering, don't update the state yet, this will be done
4504 * when buffering finishes */
4505 if (priv->buffering) {
4506 GST_INFO ("Buffering busy, delay state change");
4508 if (state == GST_STATE_PLAYING)
4509 /* make sure pads are not blocking anymore when going to PLAYING */
4510 media_unblock (media);
4512 if (state == GST_STATE_PAUSED) {
4513 set_state_ret = set_state (media, state);
4514 if (set_state_ret == GST_STATE_CHANGE_ASYNC)
4515 priv->expected_async_done = TRUE;
4516 /* and suspend after pause */
4517 gst_rtsp_media_suspend (media);
4519 set_state (media, state);
4526 * gst_rtsp_media_set_pipeline_state:
4527 * @media: a #GstRTSPMedia
4528 * @state: the target state of the pipeline
4530 * Set the state of the pipeline managed by @media to @state
4533 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4535 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4537 g_rec_mutex_lock (&media->priv->state_lock);
4538 media_set_pipeline_state_locked (media, state);
4539 g_rec_mutex_unlock (&media->priv->state_lock);
4543 * gst_rtsp_media_set_state:
4544 * @media: a #GstRTSPMedia
4545 * @state: the target state of the media
4546 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4547 * a #GPtrArray of #GstRTSPStreamTransport pointers
4549 * Set the state of @media to @state and for the transports in @transports.
4551 * @media must be prepared with gst_rtsp_media_prepare();
4553 * Returns: %TRUE on success.
4556 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4557 GPtrArray * transports)
4559 GstRTSPMediaPrivate *priv;
4561 gboolean activate, deactivate, do_state;
4564 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4565 g_return_val_if_fail (transports != NULL, FALSE);
4569 g_rec_mutex_lock (&priv->state_lock);
4571 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING
4572 && gst_rtsp_media_is_shared (media)) {
4573 g_rec_mutex_unlock (&priv->state_lock);
4574 gst_rtsp_media_get_status (media);
4575 g_rec_mutex_lock (&priv->state_lock);
4577 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4579 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4580 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4583 /* NULL and READY are the same */
4584 if (state == GST_STATE_READY)
4585 state = GST_STATE_NULL;
4587 activate = deactivate = FALSE;
4589 GST_INFO ("going to state %s media %p, target state %s",
4590 gst_element_state_get_name (state), media,
4591 gst_element_state_get_name (priv->target_state));
4594 case GST_STATE_NULL:
4595 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4596 if (priv->target_state >= GST_STATE_PAUSED)
4599 case GST_STATE_PAUSED:
4600 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4601 if (priv->target_state == GST_STATE_PLAYING)
4604 case GST_STATE_PLAYING:
4605 /* we're going to PLAYING, activate */
4611 old_active = priv->n_active;
4613 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4614 activate, deactivate);
4615 for (i = 0; i < transports->len; i++) {
4616 GstRTSPStreamTransport *trans;
4618 /* we need a non-NULL entry in the array */
4619 trans = g_ptr_array_index (transports, i);
4624 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4626 } else if (deactivate) {
4627 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4632 /* we just activated the first media, do the playing state change */
4633 if (old_active == 0 && activate)
4635 /* if we have no more active media and prepare count is not indicate
4636 * that there are new session/sessions ongoing,
4637 * do the downward state changes */
4638 else if (priv->n_active == 0 && priv->prepare_count <= 1)
4643 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4646 if (priv->target_state != state) {
4648 media_set_pipeline_state_locked (media, state);
4649 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4654 /* remember where we are */
4655 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4656 old_active != priv->n_active))
4657 collect_media_stats (media);
4659 g_rec_mutex_unlock (&priv->state_lock);
4666 GST_WARNING ("media %p was not prepared", media);
4667 g_rec_mutex_unlock (&priv->state_lock);
4672 GST_WARNING ("media %p in error status while changing to state %d",
4674 if (state == GST_STATE_NULL) {
4675 for (i = 0; i < transports->len; i++) {
4676 GstRTSPStreamTransport *trans;
4678 /* we need a non-NULL entry in the array */
4679 trans = g_ptr_array_index (transports, i);
4683 gst_rtsp_stream_transport_set_active (trans, FALSE);
4687 g_rec_mutex_unlock (&priv->state_lock);
4693 * gst_rtsp_media_set_transport_mode:
4694 * @media: a #GstRTSPMedia
4695 * @mode: the new value
4697 * Sets if the media pipeline can work in PLAY or RECORD mode
4700 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4701 GstRTSPTransportMode mode)
4703 GstRTSPMediaPrivate *priv;
4705 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4709 g_mutex_lock (&priv->lock);
4710 priv->transport_mode = mode;
4711 g_mutex_unlock (&priv->lock);
4715 * gst_rtsp_media_get_transport_mode:
4716 * @media: a #GstRTSPMedia
4718 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4720 * Returns: The transport mode.
4722 GstRTSPTransportMode
4723 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4725 GstRTSPMediaPrivate *priv;
4726 GstRTSPTransportMode res;
4728 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4732 g_mutex_lock (&priv->lock);
4733 res = priv->transport_mode;
4734 g_mutex_unlock (&priv->lock);
4740 * gst_rtsp_media_seekable:
4741 * @media: a #GstRTSPMedia
4743 * Check if the pipeline for @media seek and up to what point in time,
4746 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4747 * and > 0 to indicate the longest duration between any two random access points.
4748 * %G_MAXINT64 means any value is possible.
4753 gst_rtsp_media_seekable (GstRTSPMedia * media)
4755 GstRTSPMediaPrivate *priv;
4756 GstClockTimeDiff res;
4758 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4762 /* Currently we are not able to seek on live streams,
4763 * and no stream is seekable only to the beginning */
4764 g_mutex_lock (&priv->lock);
4765 res = priv->seekable;
4766 g_mutex_unlock (&priv->lock);
4772 * gst_rtsp_media_complete_pipeline:
4773 * @media: a #GstRTSPMedia
4774 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4776 * Add a receiver and sender parts to the pipeline based on the transport from
4779 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4784 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4786 GstRTSPMediaPrivate *priv;
4789 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4790 g_return_val_if_fail (transports, FALSE);
4792 GST_DEBUG_OBJECT (media, "complete pipeline");
4796 g_mutex_lock (&priv->lock);
4797 for (i = 0; i < priv->streams->len; i++) {
4798 GstRTSPStreamTransport *transport;
4799 GstRTSPStream *stream;
4800 const GstRTSPTransport *rtsp_transport;
4802 transport = g_ptr_array_index (transports, i);
4806 stream = gst_rtsp_stream_transport_get_stream (transport);
4810 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4812 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4813 g_mutex_unlock (&priv->lock);
4818 priv->complete = TRUE;
4819 g_mutex_unlock (&priv->lock);
4825 * gst_rtsp_media_is_receive_only:
4827 * Returns: %TRUE if @media is receive-only, %FALSE otherwise.
4831 gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
4833 GstRTSPMediaPrivate *priv = media->priv;
4834 gboolean receive_only = TRUE;
4837 for (i = 0; i < priv->streams->len; i++) {
4838 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
4839 if (gst_rtsp_stream_is_sender (stream) ||
4840 !gst_rtsp_stream_is_receiver (stream)) {
4841 receive_only = FALSE;
4846 return receive_only;
4850 * gst_rtsp_media_has_completed_sender:
4852 * See gst_rtsp_stream_is_complete(), gst_rtsp_stream_is_sender().
4854 * Returns: whether @media has at least one complete sender stream.
4858 gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)
4860 GstRTSPMediaPrivate *priv = media->priv;
4861 gboolean sender = FALSE;
4864 g_mutex_lock (&priv->lock);
4865 for (i = 0; i < priv->streams->len; i++) {
4866 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
4867 if (gst_rtsp_stream_is_complete (stream))
4868 if (gst_rtsp_stream_is_sender (stream) ||
4869 !gst_rtsp_stream_is_receiver (stream)) {
4874 g_mutex_unlock (&priv->lock);
4880 * gst_rtsp_media_set_rate_control:
4882 * Define whether @media will follow the Rate-Control=no behaviour as specified
4883 * in the ONVIF replay spec.
4888 gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled)
4890 GstRTSPMediaPrivate *priv;
4893 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4895 GST_LOG_OBJECT (media, "%s rate control", enabled ? "Enabling" : "Disabling");
4899 g_mutex_lock (&priv->lock);
4900 priv->do_rate_control = enabled;
4901 for (i = 0; i < priv->streams->len; i++) {
4902 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
4904 gst_rtsp_stream_set_rate_control (stream, enabled);
4907 g_mutex_unlock (&priv->lock);
4911 * gst_rtsp_media_get_rate_control:
4913 * Returns: whether @media will follow the Rate-Control=no behaviour as specified
4914 * in the ONVIF replay spec.
4919 gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
4921 GstRTSPMediaPrivate *priv;
4924 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4928 g_mutex_lock (&priv->lock);
4929 res = priv->do_rate_control;
4930 g_mutex_unlock (&priv->lock);