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().
55 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
56 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
59 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
60 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
61 * can be prepared again after an unprepare.
63 * Last reviewed on 2013-07-11 (1.0.0)
73 #include <gst/app/gstappsrc.h>
74 #include <gst/app/gstappsink.h>
76 #include <gst/sdp/gstmikey.h>
77 #include <gst/rtp/gstrtppayloads.h>
79 #define AES_128_KEY_LEN 16
80 #define AES_256_KEY_LEN 32
82 #define HMAC_32_KEY_LEN 4
83 #define HMAC_80_KEY_LEN 10
85 #include "rtsp-media.h"
87 struct _GstRTSPMediaPrivate
92 /* protected by lock */
93 GstRTSPPermissions *permissions;
95 gboolean suspend_mode;
97 GstRTSPProfile profiles;
98 GstRTSPLowerTrans protocols;
100 gboolean eos_shutdown;
102 GstRTSPAddressPool *pool;
103 gchar *multicast_iface;
105 gboolean bind_mcast_address;
107 GstRTSPTransportMode transport_mode;
108 gboolean stop_on_disconnect;
111 GRecMutex state_lock; /* locking order: state lock, lock */
112 GPtrArray *streams; /* protected by lock */
113 GList *dynamic; /* protected by lock */
114 GstRTSPMediaStatus status; /* protected by lock */
118 gboolean finishing_unprepare;
120 /* the pipeline for the media */
121 GstElement *pipeline;
124 GstRTSPThread *thread;
125 GList *pending_pipeline_elements;
127 gboolean time_provider;
128 GstNetTimeProvider *nettime;
131 GstClockTimeDiff seekable;
133 GstState target_state;
135 /* RTP session manager */
138 /* the range of media */
139 GstRTSPTimeRange range; /* protected by lock */
140 GstClockTime range_start;
141 GstClockTime range_stop;
143 GList *payloads; /* protected by lock */
144 GstClockTime rtx_time; /* protected by lock */
145 gboolean do_retransmission; /* protected by lock */
146 guint latency; /* protected by lock */
147 GstClock *clock; /* protected by lock */
148 GstRTSPPublishClockMode publish_clock_mode;
150 /* Dynamic element handling */
151 guint nb_dynamic_elements;
152 guint no_more_pads_pending;
155 #define DEFAULT_SHARED FALSE
156 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
157 #define DEFAULT_REUSABLE FALSE
158 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
159 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
160 GST_RTSP_LOWER_TRANS_TCP
161 #define DEFAULT_EOS_SHUTDOWN FALSE
162 #define DEFAULT_BUFFER_SIZE 0x80000
163 #define DEFAULT_TIME_PROVIDER FALSE
164 #define DEFAULT_LATENCY 200
165 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
166 #define DEFAULT_STOP_ON_DISCONNECT TRUE
167 #define DEFAULT_MAX_MCAST_TTL 255
168 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
170 #define DEFAULT_DO_RETRANSMISSION FALSE
172 /* define to dump received RTCP packets */
189 PROP_STOP_ON_DISCONNECT,
192 PROP_BIND_MCAST_ADDRESS,
199 SIGNAL_REMOVED_STREAM,
209 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
210 #define GST_CAT_DEFAULT rtsp_media_debug
212 static void gst_rtsp_media_get_property (GObject * object, guint propid,
213 GValue * value, GParamSpec * pspec);
214 static void gst_rtsp_media_set_property (GObject * object, guint propid,
215 const GValue * value, GParamSpec * pspec);
216 static void gst_rtsp_media_finalize (GObject * obj);
218 static gboolean default_handle_message (GstRTSPMedia * media,
219 GstMessage * message);
220 static void finish_unprepare (GstRTSPMedia * media);
221 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
222 static gboolean default_unprepare (GstRTSPMedia * media);
223 static gboolean default_suspend (GstRTSPMedia * media);
224 static gboolean default_unsuspend (GstRTSPMedia * media);
225 static gboolean default_convert_range (GstRTSPMedia * media,
226 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
227 static gboolean default_query_position (GstRTSPMedia * media,
229 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
230 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
231 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
233 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
235 static gboolean wait_preroll (GstRTSPMedia * media);
237 static GstElement *find_payload_element (GstElement * payloader);
239 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
241 static gboolean check_complete (GstRTSPMedia * media);
242 gboolean gst_rtsp_media_has_completed_sender (GstRTSPMedia * media);
244 static gboolean gst_rtsp_media_is_receive_only (GstRTSPMedia * media);
246 #define C_ENUM(v) ((gint) v)
249 gst_rtsp_suspend_mode_get_type (void)
252 static const GEnumValue values[] = {
253 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
254 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
256 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
261 if (g_once_init_enter (&id)) {
262 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
263 g_once_init_leave (&id, tmp);
268 #define C_FLAGS(v) ((guint) v)
271 gst_rtsp_transport_mode_get_type (void)
274 static const GFlagsValue values[] = {
275 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
277 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
282 if (g_once_init_enter (&id)) {
283 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
284 g_once_init_leave (&id, tmp);
290 gst_rtsp_publish_clock_mode_get_type (void)
293 static const GEnumValue values[] = {
294 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
295 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
296 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
297 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
299 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
300 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
305 if (g_once_init_enter (&id)) {
306 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
307 g_once_init_leave (&id, tmp);
312 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
315 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
317 GObjectClass *gobject_class;
319 gobject_class = G_OBJECT_CLASS (klass);
321 gobject_class->get_property = gst_rtsp_media_get_property;
322 gobject_class->set_property = gst_rtsp_media_set_property;
323 gobject_class->finalize = gst_rtsp_media_finalize;
325 g_object_class_install_property (gobject_class, PROP_SHARED,
326 g_param_spec_boolean ("shared", "Shared",
327 "If this media pipeline can be shared", DEFAULT_SHARED,
328 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
330 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
331 g_param_spec_enum ("suspend-mode", "Suspend Mode",
332 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
333 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_REUSABLE,
336 g_param_spec_boolean ("reusable", "Reusable",
337 "If this media pipeline can be reused after an unprepare",
338 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
340 g_object_class_install_property (gobject_class, PROP_PROFILES,
341 g_param_spec_flags ("profiles", "Profiles",
342 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
343 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
346 g_param_spec_flags ("protocols", "Protocols",
347 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
348 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
350 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
351 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
352 "Send an EOS event to the pipeline before unpreparing",
353 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
356 g_param_spec_uint ("buffer-size", "Buffer Size",
357 "The kernel UDP buffer size to use", 0, G_MAXUINT,
358 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 g_object_class_install_property (gobject_class, PROP_ELEMENT,
361 g_param_spec_object ("element", "The Element",
362 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
363 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
365 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
366 g_param_spec_boolean ("time-provider", "Time Provider",
367 "Use a NetTimeProvider for clients",
368 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_LATENCY,
371 g_param_spec_uint ("latency", "Latency",
372 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
373 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
375 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
376 g_param_spec_flags ("transport-mode", "Transport Mode",
377 "If this media pipeline can be used for PLAY or RECORD",
378 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
379 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
381 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
382 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
383 "If this media pipeline should be stopped "
384 "when a client disconnects without TEARDOWN",
385 DEFAULT_STOP_ON_DISCONNECT,
386 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_CLOCK,
389 g_param_spec_object ("clock", "Clock",
390 "Clock to be used by the media pipeline",
391 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
393 g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
394 g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
395 "The maximum time-to-live value of outgoing multicast packets", 1,
396 255, DEFAULT_MAX_MCAST_TTL,
397 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
399 g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
400 g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
401 "Whether the multicast sockets should be bound to multicast addresses "
403 DEFAULT_BIND_MCAST_ADDRESS,
404 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
406 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
407 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
408 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
409 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
411 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
412 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
413 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
414 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
415 GST_TYPE_RTSP_STREAM);
417 gst_rtsp_media_signals[SIGNAL_PREPARED] =
418 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
419 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
420 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
422 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
423 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
424 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
425 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
427 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
428 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
429 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
430 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
432 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
433 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
434 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
435 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
437 gst_rtsp_media_signals[SIGNAL_PREPARING] =
438 g_signal_new ("preparing", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
439 G_STRUCT_OFFSET (GstRTSPMediaClass, preparing), NULL, NULL,
440 g_cclosure_marshal_generic, G_TYPE_NONE, 2, GST_TYPE_RTSP_STREAM,
443 gst_rtsp_media_signals[SIGNAL_UNPREPARING] =
444 g_signal_new ("unpreparing", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
445 G_STRUCT_OFFSET (GstRTSPMediaClass, unpreparing), NULL, NULL,
446 g_cclosure_marshal_generic, G_TYPE_NONE, 2, GST_TYPE_RTSP_STREAM,
449 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
451 klass->handle_message = default_handle_message;
452 klass->prepare = default_prepare;
453 klass->unprepare = default_unprepare;
454 klass->suspend = default_suspend;
455 klass->unsuspend = default_unsuspend;
456 klass->convert_range = default_convert_range;
457 klass->query_position = default_query_position;
458 klass->query_stop = default_query_stop;
459 klass->create_rtpbin = default_create_rtpbin;
460 klass->setup_sdp = default_setup_sdp;
461 klass->handle_sdp = default_handle_sdp;
465 gst_rtsp_media_init (GstRTSPMedia * media)
467 GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_private (media);
471 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
472 g_mutex_init (&priv->lock);
473 g_cond_init (&priv->cond);
474 g_rec_mutex_init (&priv->state_lock);
476 priv->shared = DEFAULT_SHARED;
477 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
478 priv->reusable = DEFAULT_REUSABLE;
479 priv->profiles = DEFAULT_PROFILES;
480 priv->protocols = DEFAULT_PROTOCOLS;
481 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
482 priv->buffer_size = DEFAULT_BUFFER_SIZE;
483 priv->time_provider = DEFAULT_TIME_PROVIDER;
484 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
485 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
486 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
487 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
488 priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
489 priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
493 gst_rtsp_media_finalize (GObject * obj)
495 GstRTSPMediaPrivate *priv;
498 media = GST_RTSP_MEDIA (obj);
501 GST_INFO ("finalize media %p", media);
503 if (priv->permissions)
504 gst_rtsp_permissions_unref (priv->permissions);
506 g_ptr_array_unref (priv->streams);
508 g_list_free_full (priv->dynamic, gst_object_unref);
509 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
512 gst_object_unref (priv->pipeline);
514 gst_object_unref (priv->nettime);
515 gst_object_unref (priv->element);
517 g_object_unref (priv->pool);
519 g_list_free (priv->payloads);
521 gst_object_unref (priv->clock);
522 g_free (priv->multicast_iface);
523 g_mutex_clear (&priv->lock);
524 g_cond_clear (&priv->cond);
525 g_rec_mutex_clear (&priv->state_lock);
527 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
531 gst_rtsp_media_get_property (GObject * object, guint propid,
532 GValue * value, GParamSpec * pspec)
534 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
538 g_value_set_object (value, media->priv->element);
541 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
543 case PROP_SUSPEND_MODE:
544 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
547 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
550 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
553 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
555 case PROP_EOS_SHUTDOWN:
556 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
558 case PROP_BUFFER_SIZE:
559 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
561 case PROP_TIME_PROVIDER:
562 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
565 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
567 case PROP_TRANSPORT_MODE:
568 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
570 case PROP_STOP_ON_DISCONNECT:
571 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
574 g_value_take_object (value, gst_rtsp_media_get_clock (media));
576 case PROP_MAX_MCAST_TTL:
577 g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
579 case PROP_BIND_MCAST_ADDRESS:
580 g_value_set_boolean (value, gst_rtsp_media_is_bind_mcast_address (media));
583 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
588 gst_rtsp_media_set_property (GObject * object, guint propid,
589 const GValue * value, GParamSpec * pspec)
591 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
595 media->priv->element = g_value_get_object (value);
596 gst_object_ref_sink (media->priv->element);
599 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
601 case PROP_SUSPEND_MODE:
602 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
605 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
608 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
611 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
613 case PROP_EOS_SHUTDOWN:
614 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
616 case PROP_BUFFER_SIZE:
617 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
619 case PROP_TIME_PROVIDER:
620 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
623 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
625 case PROP_TRANSPORT_MODE:
626 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
628 case PROP_STOP_ON_DISCONNECT:
629 gst_rtsp_media_set_stop_on_disconnect (media,
630 g_value_get_boolean (value));
633 gst_rtsp_media_set_clock (media, g_value_get_object (value));
635 case PROP_MAX_MCAST_TTL:
636 gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
638 case PROP_BIND_MCAST_ADDRESS:
639 gst_rtsp_media_set_bind_mcast_address (media,
640 g_value_get_boolean (value));
643 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
650 gboolean complete_streams_only;
652 } DoQueryPositionData;
655 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
659 if (!gst_rtsp_stream_is_sender (stream))
662 if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
663 GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
667 if (gst_rtsp_stream_query_position (stream, &tmp)) {
668 data->position = MIN (data->position, tmp);
672 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
673 GST_TIME_ARGS (data->position));
677 default_query_position (GstRTSPMedia * media, gint64 * position)
679 GstRTSPMediaPrivate *priv;
680 DoQueryPositionData data;
684 data.position = G_MAXINT64;
687 /* if the media is complete, i.e. one or more streams have been configured
688 * with sinks, then we want to query the position on those streams only.
689 * a query on an incmplete stream may return a position that originates from
690 * an earlier preroll */
691 if (check_complete (media))
692 data.complete_streams_only = TRUE;
694 data.complete_streams_only = FALSE;
696 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
699 *position = GST_CLOCK_TIME_NONE;
701 *position = data.position;
713 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
717 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
718 data->stop = MAX (data->stop, tmp);
724 default_query_stop (GstRTSPMedia * media, gint64 * stop)
726 GstRTSPMediaPrivate *priv;
727 DoQueryStopData data;
734 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
742 default_create_rtpbin (GstRTSPMedia * media)
746 rtpbin = gst_element_factory_make ("rtpbin", NULL);
751 /* Must be called with priv->lock */
753 is_receive_only (GstRTSPMedia * media)
755 GstRTSPMediaPrivate *priv = media->priv;
756 gboolean receive_only = TRUE;
759 for (i = 0; i < priv->streams->len; i++) {
760 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
761 if (gst_rtsp_stream_is_sender (stream) ||
762 !gst_rtsp_stream_is_receiver (stream)) {
763 receive_only = FALSE;
771 /* must be called with state lock */
773 check_seekable (GstRTSPMedia * media)
776 GstRTSPMediaPrivate *priv = media->priv;
778 g_mutex_lock (&priv->lock);
779 /* Update the seekable state of the pipeline in case it changed */
780 if (is_receive_only (media)) {
781 /* TODO: Seeking for "receive-only"? */
784 guint i, n = priv->streams->len;
786 for (i = 0; i < n; i++) {
787 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
789 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
790 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
792 g_mutex_unlock (&priv->lock);
798 query = gst_query_new_seeking (GST_FORMAT_TIME);
799 if (gst_element_query (priv->pipeline, query)) {
804 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
805 priv->seekable = seekable ? G_MAXINT64 : 0;
806 } else if (priv->streams->len) {
807 gboolean seekable = TRUE;
808 guint i, n = priv->streams->len;
810 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
811 for (i = 0; i < n; i++) {
812 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
813 seekable &= gst_rtsp_stream_seekable (stream);
815 priv->seekable = seekable ? G_MAXINT64 : -1;
818 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
819 g_mutex_unlock (&priv->lock);
820 gst_query_unref (query);
823 /* must be called with state lock */
825 check_complete (GstRTSPMedia * media)
827 GstRTSPMediaPrivate *priv = media->priv;
829 guint i, n = priv->streams->len;
831 for (i = 0; i < n; i++) {
832 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
834 if (gst_rtsp_stream_is_complete (stream))
841 /* must be called with state lock and private lock */
843 collect_media_stats (GstRTSPMedia * media)
845 GstRTSPMediaPrivate *priv = media->priv;
846 gint64 position = 0, stop = -1;
848 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
849 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING) {
853 priv->range.unit = GST_RTSP_RANGE_NPT;
855 GST_INFO ("collect media stats");
858 priv->range.min.type = GST_RTSP_TIME_NOW;
859 priv->range.min.seconds = -1;
860 priv->range_start = -1;
861 priv->range.max.type = GST_RTSP_TIME_END;
862 priv->range.max.seconds = -1;
863 priv->range_stop = -1;
865 GstRTSPMediaClass *klass;
868 klass = GST_RTSP_MEDIA_GET_CLASS (media);
870 /* get the position */
872 if (klass->query_position)
873 ret = klass->query_position (media, &position);
876 GST_INFO ("position query failed");
880 /* get the current segment stop */
882 if (klass->query_stop)
883 ret = klass->query_stop (media, &stop);
886 GST_INFO ("stop query failed");
890 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
891 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
893 if (position == -1) {
894 priv->range.min.type = GST_RTSP_TIME_NOW;
895 priv->range.min.seconds = -1;
896 priv->range_start = -1;
898 priv->range.min.type = GST_RTSP_TIME_SECONDS;
899 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
900 priv->range_start = position;
903 priv->range.max.type = GST_RTSP_TIME_END;
904 priv->range.max.seconds = -1;
905 priv->range_stop = -1;
907 priv->range.max.type = GST_RTSP_TIME_SECONDS;
908 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
909 priv->range_stop = stop;
911 g_mutex_unlock (&priv->lock);
912 check_seekable (media);
913 g_mutex_lock (&priv->lock);
918 * gst_rtsp_media_new:
919 * @element: (transfer full): a #GstElement
921 * Create a new #GstRTSPMedia instance. @element is the bin element that
922 * provides the different streams. The #GstRTSPMedia object contains the
923 * element to produce RTP data for one or more related (audio/video/..)
926 * Ownership is taken of @element.
928 * Returns: (transfer full): a new #GstRTSPMedia object.
931 gst_rtsp_media_new (GstElement * element)
933 GstRTSPMedia *result;
935 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
937 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
943 * gst_rtsp_media_get_element:
944 * @media: a #GstRTSPMedia
946 * Get the element that was used when constructing @media.
948 * Returns: (transfer full): a #GstElement. Unref after usage.
951 gst_rtsp_media_get_element (GstRTSPMedia * media)
953 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
955 return gst_object_ref (media->priv->element);
959 * gst_rtsp_media_take_pipeline:
960 * @media: a #GstRTSPMedia
961 * @pipeline: (transfer full): a #GstPipeline
963 * Set @pipeline as the #GstPipeline for @media. Ownership is
964 * taken of @pipeline.
967 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
969 GstRTSPMediaPrivate *priv;
971 GstNetTimeProvider *nettime;
974 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
975 g_return_if_fail (GST_IS_PIPELINE (pipeline));
979 g_mutex_lock (&priv->lock);
980 old = priv->pipeline;
981 priv->pipeline = GST_ELEMENT_CAST (pipeline);
982 nettime = priv->nettime;
983 priv->nettime = NULL;
984 g_mutex_unlock (&priv->lock);
987 gst_object_unref (old);
990 gst_object_unref (nettime);
992 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
994 for (l = priv->pending_pipeline_elements; l; l = l->next) {
995 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
997 g_list_free (priv->pending_pipeline_elements);
998 priv->pending_pipeline_elements = NULL;
1002 * gst_rtsp_media_set_permissions:
1003 * @media: a #GstRTSPMedia
1004 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
1006 * Set @permissions on @media.
1009 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
1010 GstRTSPPermissions * permissions)
1012 GstRTSPMediaPrivate *priv;
1014 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1018 g_mutex_lock (&priv->lock);
1019 if (priv->permissions)
1020 gst_rtsp_permissions_unref (priv->permissions);
1021 if ((priv->permissions = permissions))
1022 gst_rtsp_permissions_ref (permissions);
1023 g_mutex_unlock (&priv->lock);
1027 * gst_rtsp_media_get_permissions:
1028 * @media: a #GstRTSPMedia
1030 * Get the permissions object from @media.
1032 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
1034 GstRTSPPermissions *
1035 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
1037 GstRTSPMediaPrivate *priv;
1038 GstRTSPPermissions *result;
1040 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1044 g_mutex_lock (&priv->lock);
1045 if ((result = priv->permissions))
1046 gst_rtsp_permissions_ref (result);
1047 g_mutex_unlock (&priv->lock);
1053 * gst_rtsp_media_set_suspend_mode:
1054 * @media: a #GstRTSPMedia
1055 * @mode: the new #GstRTSPSuspendMode
1057 * Control how @ media will be suspended after the SDP has been generated and
1058 * after a PAUSE request has been performed.
1060 * Media must be unprepared when setting the suspend mode.
1063 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
1065 GstRTSPMediaPrivate *priv;
1067 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1071 g_rec_mutex_lock (&priv->state_lock);
1072 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1074 priv->suspend_mode = mode;
1075 g_rec_mutex_unlock (&priv->state_lock);
1082 GST_WARNING ("media %p was prepared", media);
1083 g_rec_mutex_unlock (&priv->state_lock);
1088 * gst_rtsp_media_get_suspend_mode:
1089 * @media: a #GstRTSPMedia
1091 * Get how @media will be suspended.
1093 * Returns: #GstRTSPSuspendMode.
1096 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1098 GstRTSPMediaPrivate *priv;
1099 GstRTSPSuspendMode res;
1101 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1105 g_rec_mutex_lock (&priv->state_lock);
1106 res = priv->suspend_mode;
1107 g_rec_mutex_unlock (&priv->state_lock);
1113 * gst_rtsp_media_set_shared:
1114 * @media: a #GstRTSPMedia
1115 * @shared: the new value
1117 * Set or unset if the pipeline for @media can be shared will multiple clients.
1118 * When @shared is %TRUE, client requests for this media will share the media
1122 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1124 GstRTSPMediaPrivate *priv;
1126 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1130 g_mutex_lock (&priv->lock);
1131 priv->shared = shared;
1132 g_mutex_unlock (&priv->lock);
1136 * gst_rtsp_media_is_shared:
1137 * @media: a #GstRTSPMedia
1139 * Check if the pipeline for @media can be shared between multiple clients.
1141 * Returns: %TRUE if the media can be shared between clients.
1144 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1146 GstRTSPMediaPrivate *priv;
1149 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1153 g_mutex_lock (&priv->lock);
1155 g_mutex_unlock (&priv->lock);
1161 * gst_rtsp_media_set_reusable:
1162 * @media: a #GstRTSPMedia
1163 * @reusable: the new value
1165 * Set or unset if the pipeline for @media can be reused after the pipeline has
1169 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1171 GstRTSPMediaPrivate *priv;
1173 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1177 g_mutex_lock (&priv->lock);
1178 priv->reusable = reusable;
1179 g_mutex_unlock (&priv->lock);
1183 * gst_rtsp_media_is_reusable:
1184 * @media: a #GstRTSPMedia
1186 * Check if the pipeline for @media can be reused after an unprepare.
1188 * Returns: %TRUE if the media can be reused
1191 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1193 GstRTSPMediaPrivate *priv;
1196 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1200 g_mutex_lock (&priv->lock);
1201 res = priv->reusable;
1202 g_mutex_unlock (&priv->lock);
1208 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1210 gst_rtsp_stream_set_profiles (stream, *profiles);
1214 * gst_rtsp_media_set_profiles:
1215 * @media: a #GstRTSPMedia
1216 * @profiles: the new flags
1218 * Configure the allowed lower transport for @media.
1221 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1223 GstRTSPMediaPrivate *priv;
1225 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1229 g_mutex_lock (&priv->lock);
1230 priv->profiles = profiles;
1231 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1232 g_mutex_unlock (&priv->lock);
1236 * gst_rtsp_media_get_profiles:
1237 * @media: a #GstRTSPMedia
1239 * Get the allowed profiles of @media.
1241 * Returns: a #GstRTSPProfile
1244 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1246 GstRTSPMediaPrivate *priv;
1249 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1253 g_mutex_lock (&priv->lock);
1254 res = priv->profiles;
1255 g_mutex_unlock (&priv->lock);
1261 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1263 gst_rtsp_stream_set_protocols (stream, *protocols);
1267 * gst_rtsp_media_set_protocols:
1268 * @media: a #GstRTSPMedia
1269 * @protocols: the new flags
1271 * Configure the allowed lower transport for @media.
1274 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1276 GstRTSPMediaPrivate *priv;
1278 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1282 g_mutex_lock (&priv->lock);
1283 priv->protocols = protocols;
1284 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1285 g_mutex_unlock (&priv->lock);
1289 * gst_rtsp_media_get_protocols:
1290 * @media: a #GstRTSPMedia
1292 * Get the allowed protocols of @media.
1294 * Returns: a #GstRTSPLowerTrans
1297 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1299 GstRTSPMediaPrivate *priv;
1300 GstRTSPLowerTrans res;
1302 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1303 GST_RTSP_LOWER_TRANS_UNKNOWN);
1307 g_mutex_lock (&priv->lock);
1308 res = priv->protocols;
1309 g_mutex_unlock (&priv->lock);
1315 * gst_rtsp_media_set_eos_shutdown:
1316 * @media: a #GstRTSPMedia
1317 * @eos_shutdown: the new value
1319 * Set or unset if an EOS event will be sent to the pipeline for @media before
1323 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1325 GstRTSPMediaPrivate *priv;
1327 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1331 g_mutex_lock (&priv->lock);
1332 priv->eos_shutdown = eos_shutdown;
1333 g_mutex_unlock (&priv->lock);
1337 * gst_rtsp_media_is_eos_shutdown:
1338 * @media: a #GstRTSPMedia
1340 * Check if the pipeline for @media will send an EOS down the pipeline before
1343 * Returns: %TRUE if the media will send EOS before unpreparing.
1346 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1348 GstRTSPMediaPrivate *priv;
1351 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1355 g_mutex_lock (&priv->lock);
1356 res = priv->eos_shutdown;
1357 g_mutex_unlock (&priv->lock);
1363 * gst_rtsp_media_set_buffer_size:
1364 * @media: a #GstRTSPMedia
1365 * @size: the new value
1367 * Set the kernel UDP buffer size.
1370 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1372 GstRTSPMediaPrivate *priv;
1375 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1377 GST_LOG_OBJECT (media, "set buffer size %u", size);
1381 g_mutex_lock (&priv->lock);
1382 priv->buffer_size = size;
1384 for (i = 0; i < priv->streams->len; i++) {
1385 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1386 gst_rtsp_stream_set_buffer_size (stream, size);
1388 g_mutex_unlock (&priv->lock);
1392 * gst_rtsp_media_get_buffer_size:
1393 * @media: a #GstRTSPMedia
1395 * Get the kernel UDP buffer size.
1397 * Returns: the kernel UDP buffer size.
1400 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1402 GstRTSPMediaPrivate *priv;
1405 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1409 g_mutex_lock (&priv->lock);
1410 res = priv->buffer_size;
1411 g_mutex_unlock (&priv->lock);
1417 * gst_rtsp_media_set_stop_on_disconnect:
1418 * @media: a #GstRTSPMedia
1419 * @stop_on_disconnect: the new value
1421 * Set or unset if the pipeline for @media should be stopped when a
1422 * client disconnects without sending TEARDOWN.
1425 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1426 gboolean stop_on_disconnect)
1428 GstRTSPMediaPrivate *priv;
1430 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1434 g_mutex_lock (&priv->lock);
1435 priv->stop_on_disconnect = stop_on_disconnect;
1436 g_mutex_unlock (&priv->lock);
1440 * gst_rtsp_media_is_stop_on_disconnect:
1441 * @media: a #GstRTSPMedia
1443 * Check if the pipeline for @media will be stopped when a client disconnects
1444 * without sending TEARDOWN.
1446 * Returns: %TRUE if the media will be stopped when a client disconnects
1447 * without sending TEARDOWN.
1450 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1452 GstRTSPMediaPrivate *priv;
1455 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1459 g_mutex_lock (&priv->lock);
1460 res = priv->stop_on_disconnect;
1461 g_mutex_unlock (&priv->lock);
1467 * gst_rtsp_media_set_retransmission_time:
1468 * @media: a #GstRTSPMedia
1469 * @time: the new value
1471 * Set the amount of time to store retransmission packets.
1474 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1476 GstRTSPMediaPrivate *priv;
1479 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1481 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1485 g_mutex_lock (&priv->lock);
1486 priv->rtx_time = time;
1487 for (i = 0; i < priv->streams->len; i++) {
1488 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1490 gst_rtsp_stream_set_retransmission_time (stream, time);
1492 g_mutex_unlock (&priv->lock);
1496 * gst_rtsp_media_get_retransmission_time:
1497 * @media: a #GstRTSPMedia
1499 * Get the amount of time to store retransmission data.
1501 * Returns: the amount of time to store retransmission data.
1504 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1506 GstRTSPMediaPrivate *priv;
1509 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1513 g_mutex_lock (&priv->lock);
1514 res = priv->rtx_time;
1515 g_mutex_unlock (&priv->lock);
1521 * gst_rtsp_media_set_do_retransmission:
1523 * Set whether retransmission requests will be sent
1528 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
1529 gboolean do_retransmission)
1531 GstRTSPMediaPrivate *priv;
1533 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1537 g_mutex_lock (&priv->lock);
1538 priv->do_retransmission = do_retransmission;
1541 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1542 g_mutex_unlock (&priv->lock);
1546 * gst_rtsp_media_get_do_retransmission:
1548 * Returns: Whether retransmission requests will be sent
1553 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1555 GstRTSPMediaPrivate *priv;
1558 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1562 g_mutex_lock (&priv->lock);
1563 res = priv->do_retransmission;
1564 g_mutex_unlock (&priv->lock);
1570 * gst_rtsp_media_set_latency:
1571 * @media: a #GstRTSPMedia
1572 * @latency: latency in milliseconds
1574 * Configure the latency used for receiving media.
1577 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1579 GstRTSPMediaPrivate *priv;
1582 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1584 GST_LOG_OBJECT (media, "set latency %ums", latency);
1588 g_mutex_lock (&priv->lock);
1589 priv->latency = latency;
1591 g_object_set (priv->rtpbin, "latency", latency, NULL);
1593 for (i = 0; i < media->priv->streams->len; i++) {
1594 GObject *storage = NULL;
1596 g_signal_emit_by_name (G_OBJECT (media->priv->rtpbin), "get-storage",
1599 g_object_set (storage, "size-time",
1600 (media->priv->latency + 50) * GST_MSECOND, NULL);
1604 g_mutex_unlock (&priv->lock);
1608 * gst_rtsp_media_get_latency:
1609 * @media: a #GstRTSPMedia
1611 * Get the latency that is used for receiving media.
1613 * Returns: latency in milliseconds
1616 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1618 GstRTSPMediaPrivate *priv;
1621 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1625 g_mutex_lock (&priv->lock);
1626 res = priv->latency;
1627 g_mutex_unlock (&priv->lock);
1633 * gst_rtsp_media_use_time_provider:
1634 * @media: a #GstRTSPMedia
1635 * @time_provider: if a #GstNetTimeProvider should be used
1637 * Set @media to provide a #GstNetTimeProvider.
1640 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1642 GstRTSPMediaPrivate *priv;
1644 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1648 g_mutex_lock (&priv->lock);
1649 priv->time_provider = time_provider;
1650 g_mutex_unlock (&priv->lock);
1654 * gst_rtsp_media_is_time_provider:
1655 * @media: a #GstRTSPMedia
1657 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1659 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1661 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1664 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1666 GstRTSPMediaPrivate *priv;
1669 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1673 g_mutex_lock (&priv->lock);
1674 res = priv->time_provider;
1675 g_mutex_unlock (&priv->lock);
1681 * gst_rtsp_media_set_clock:
1682 * @media: a #GstRTSPMedia
1683 * @clock: (nullable): #GstClock to be used
1685 * Configure the clock used for the media.
1688 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1690 GstRTSPMediaPrivate *priv;
1692 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1693 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1695 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1699 g_mutex_lock (&priv->lock);
1701 gst_object_unref (priv->clock);
1702 priv->clock = clock ? gst_object_ref (clock) : NULL;
1703 if (priv->pipeline) {
1705 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1707 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1710 g_mutex_unlock (&priv->lock);
1714 * gst_rtsp_media_set_publish_clock_mode:
1715 * @media: a #GstRTSPMedia
1716 * @mode: the clock publish mode
1718 * Sets if and how the media clock should be published according to RFC7273.
1723 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1724 GstRTSPPublishClockMode mode)
1726 GstRTSPMediaPrivate *priv;
1730 g_mutex_lock (&priv->lock);
1731 priv->publish_clock_mode = mode;
1733 n = priv->streams->len;
1734 for (i = 0; i < n; i++) {
1735 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1737 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1739 g_mutex_unlock (&priv->lock);
1743 * gst_rtsp_media_get_publish_clock_mode:
1744 * @media: a #GstRTSPMedia
1746 * Gets if and how the media clock should be published according to RFC7273.
1748 * Returns: The GstRTSPPublishClockMode
1752 GstRTSPPublishClockMode
1753 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1755 GstRTSPMediaPrivate *priv;
1756 GstRTSPPublishClockMode ret;
1759 g_mutex_lock (&priv->lock);
1760 ret = priv->publish_clock_mode;
1761 g_mutex_unlock (&priv->lock);
1767 * gst_rtsp_media_set_address_pool:
1768 * @media: a #GstRTSPMedia
1769 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1771 * configure @pool to be used as the address pool of @media.
1774 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1775 GstRTSPAddressPool * pool)
1777 GstRTSPMediaPrivate *priv;
1778 GstRTSPAddressPool *old;
1780 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1784 GST_LOG_OBJECT (media, "set address pool %p", pool);
1786 g_mutex_lock (&priv->lock);
1787 if ((old = priv->pool) != pool)
1788 priv->pool = pool ? g_object_ref (pool) : NULL;
1791 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1793 g_mutex_unlock (&priv->lock);
1796 g_object_unref (old);
1800 * gst_rtsp_media_get_address_pool:
1801 * @media: a #GstRTSPMedia
1803 * Get the #GstRTSPAddressPool used as the address pool of @media.
1805 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1806 * g_object_unref() after usage.
1808 GstRTSPAddressPool *
1809 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1811 GstRTSPMediaPrivate *priv;
1812 GstRTSPAddressPool *result;
1814 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1818 g_mutex_lock (&priv->lock);
1819 if ((result = priv->pool))
1820 g_object_ref (result);
1821 g_mutex_unlock (&priv->lock);
1827 * gst_rtsp_media_set_multicast_iface:
1828 * @media: a #GstRTSPMedia
1829 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1831 * configure @multicast_iface to be used for @media.
1834 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1835 const gchar * multicast_iface)
1837 GstRTSPMediaPrivate *priv;
1840 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1844 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1846 g_mutex_lock (&priv->lock);
1847 if ((old = priv->multicast_iface) != multicast_iface)
1848 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1851 g_ptr_array_foreach (priv->streams,
1852 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1853 g_mutex_unlock (&priv->lock);
1860 * gst_rtsp_media_get_multicast_iface:
1861 * @media: a #GstRTSPMedia
1863 * Get the multicast interface used for @media.
1865 * Returns: (transfer full) (nullable): the multicast interface for @media.
1866 * g_free() after usage.
1869 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1871 GstRTSPMediaPrivate *priv;
1874 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1878 g_mutex_lock (&priv->lock);
1879 if ((result = priv->multicast_iface))
1880 result = g_strdup (result);
1881 g_mutex_unlock (&priv->lock);
1887 * gst_rtsp_media_set_max_mcast_ttl:
1888 * @media: a #GstRTSPMedia
1889 * @ttl: the new multicast ttl value
1891 * Set the maximum time-to-live value of outgoing multicast packets.
1893 * Returns: %TRUE if the requested ttl has been set successfully.
1898 gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
1900 GstRTSPMediaPrivate *priv;
1903 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1905 GST_LOG_OBJECT (media, "set max mcast ttl %u", ttl);
1909 g_mutex_lock (&priv->lock);
1911 if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
1912 GST_WARNING_OBJECT (media, "The reqested mcast TTL value is not valid.");
1913 g_mutex_unlock (&priv->lock);
1916 priv->max_mcast_ttl = ttl;
1918 for (i = 0; i < priv->streams->len; i++) {
1919 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1920 gst_rtsp_stream_set_max_mcast_ttl (stream, ttl);
1922 g_mutex_unlock (&priv->lock);
1928 * gst_rtsp_media_get_max_mcast_ttl:
1929 * @media: a #GstRTSPMedia
1931 * Get the the maximum time-to-live value of outgoing multicast packets.
1933 * Returns: the maximum time-to-live value of outgoing multicast packets.
1938 gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
1940 GstRTSPMediaPrivate *priv;
1943 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1947 g_mutex_lock (&priv->lock);
1948 res = priv->max_mcast_ttl;
1949 g_mutex_unlock (&priv->lock);
1955 * gst_rtsp_media_set_bind_mcast_address:
1956 * @media: a #GstRTSPMedia
1957 * @bind_mcast_addr: the new value
1959 * Decide whether the multicast socket should be bound to a multicast address or
1965 gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
1966 gboolean bind_mcast_addr)
1968 GstRTSPMediaPrivate *priv;
1971 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1975 g_mutex_lock (&priv->lock);
1976 priv->bind_mcast_address = bind_mcast_addr;
1977 for (i = 0; i < priv->streams->len; i++) {
1978 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1979 gst_rtsp_stream_set_bind_mcast_address (stream, bind_mcast_addr);
1981 g_mutex_unlock (&priv->lock);
1985 * gst_rtsp_media_is_bind_mcast_address:
1986 * @media: a #GstRTSPMedia
1988 * Check if multicast sockets are configured to be bound to multicast addresses.
1990 * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
1995 gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
1997 GstRTSPMediaPrivate *priv;
2000 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2004 g_mutex_lock (&priv->lock);
2005 result = priv->bind_mcast_address;
2006 g_mutex_unlock (&priv->lock);
2012 _find_payload_types (GstRTSPMedia * media)
2015 GQueue queue = G_QUEUE_INIT;
2017 n = media->priv->streams->len;
2018 for (i = 0; i < n; i++) {
2019 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
2020 guint pt = gst_rtsp_stream_get_pt (stream);
2022 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
2029 _next_available_pt (GList * payloads)
2033 for (i = 96; i <= 127; i++) {
2034 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
2036 return GPOINTER_TO_UINT (i);
2043 * gst_rtsp_media_collect_streams:
2044 * @media: a #GstRTSPMedia
2046 * Find all payloader elements, they should be named pay\%d in the
2047 * element of @media, and create #GstRTSPStreams for them.
2049 * Collect all dynamic elements, named dynpay\%d, and add them to
2050 * the list of dynamic elements.
2052 * Find all depayloader elements, they should be named depay\%d in the
2053 * element of @media, and create #GstRTSPStreams for them.
2056 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
2058 GstRTSPMediaPrivate *priv;
2059 GstElement *element, *elem;
2063 gboolean more_elem_remaining = TRUE;
2064 GstRTSPTransportMode mode = 0;
2066 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2069 element = priv->element;
2072 for (i = 0; more_elem_remaining; i++) {
2075 more_elem_remaining = FALSE;
2077 name = g_strdup_printf ("pay%d", i);
2078 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2080 GST_INFO ("found stream %d with payloader %p", i, elem);
2082 /* take the pad of the payloader */
2083 pad = gst_element_get_static_pad (elem, "src");
2085 /* find the real payload element in case elem is a GstBin */
2086 pay = find_payload_element (elem);
2088 /* create the stream */
2090 GST_WARNING ("could not find real payloader, using bin");
2091 gst_rtsp_media_create_stream (media, elem, pad);
2093 gst_rtsp_media_create_stream (media, pay, pad);
2094 gst_object_unref (pay);
2097 gst_object_unref (pad);
2098 gst_object_unref (elem);
2101 more_elem_remaining = TRUE;
2102 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2106 name = g_strdup_printf ("dynpay%d", i);
2107 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2108 /* a stream that will dynamically create pads to provide RTP packets */
2109 GST_INFO ("found dynamic element %d, %p", i, elem);
2111 g_mutex_lock (&priv->lock);
2112 priv->dynamic = g_list_prepend (priv->dynamic, elem);
2113 g_mutex_unlock (&priv->lock);
2115 priv->nb_dynamic_elements++;
2118 more_elem_remaining = TRUE;
2119 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2123 name = g_strdup_printf ("depay%d", i);
2124 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2125 GST_INFO ("found stream %d with depayloader %p", i, elem);
2127 /* take the pad of the payloader */
2128 pad = gst_element_get_static_pad (elem, "sink");
2129 /* create the stream */
2130 gst_rtsp_media_create_stream (media, elem, pad);
2131 gst_object_unref (pad);
2132 gst_object_unref (elem);
2135 more_elem_remaining = TRUE;
2136 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
2142 if (priv->transport_mode != mode)
2143 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
2144 priv->transport_mode, mode);
2150 GstElement *appsink, *appsrc;
2151 GstRTSPStream *stream;
2154 static GstFlowReturn
2155 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
2157 AppSinkSrcData *data = user_data;
2161 sample = gst_app_sink_pull_sample (appsink);
2163 return GST_FLOW_FLUSHING;
2166 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
2167 gst_sample_unref (sample);
2171 static GstAppSinkCallbacks appsink_callbacks = {
2177 static GstPadProbeReturn
2178 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2180 AppSinkSrcData *data = user_data;
2182 if (GST_IS_EVENT (info->data)
2183 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
2184 GstClockTime min, max;
2186 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
2188 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
2189 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
2190 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
2192 } else if (GST_IS_QUERY (info->data)) {
2193 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
2194 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
2195 gst_object_unref (srcpad);
2196 return GST_PAD_PROBE_HANDLED;
2198 gst_object_unref (srcpad);
2201 return GST_PAD_PROBE_OK;
2204 static GstPadProbeReturn
2205 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2207 AppSinkSrcData *data = user_data;
2209 if (GST_IS_QUERY (info->data)) {
2210 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
2211 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
2212 gst_object_unref (sinkpad);
2213 return GST_PAD_PROBE_HANDLED;
2215 gst_object_unref (sinkpad);
2218 return GST_PAD_PROBE_OK;
2222 * gst_rtsp_media_create_stream:
2223 * @media: a #GstRTSPMedia
2224 * @payloader: a #GstElement
2227 * Create a new stream in @media that provides RTP data on @pad.
2228 * @pad should be a pad of an element inside @media->element.
2230 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2234 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2237 GstRTSPMediaPrivate *priv;
2238 GstRTSPStream *stream;
2242 AppSinkSrcData *data = NULL;
2244 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2245 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2246 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2250 g_mutex_lock (&priv->lock);
2251 idx = priv->streams->len;
2253 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
2255 if (GST_PAD_IS_SRC (pad))
2256 name = g_strdup_printf ("src_%u", idx);
2258 name = g_strdup_printf ("sink_%u", idx);
2260 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2261 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2262 GstElement *appsink, *appsrc;
2263 GstPad *sinkpad, *srcpad;
2265 appsink = gst_element_factory_make ("appsink", NULL);
2266 appsrc = gst_element_factory_make ("appsrc", NULL);
2268 if (GST_PAD_IS_SINK (pad)) {
2269 srcpad = gst_element_get_static_pad (appsrc, "src");
2271 gst_bin_add (GST_BIN (priv->element), appsrc);
2273 gst_pad_link (srcpad, pad);
2274 gst_object_unref (srcpad);
2276 streampad = gst_element_get_static_pad (appsink, "sink");
2278 priv->pending_pipeline_elements =
2279 g_list_prepend (priv->pending_pipeline_elements, appsink);
2281 sinkpad = gst_element_get_static_pad (appsink, "sink");
2283 gst_pad_link (pad, sinkpad);
2284 gst_object_unref (sinkpad);
2286 streampad = gst_element_get_static_pad (appsrc, "src");
2288 priv->pending_pipeline_elements =
2289 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2292 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2293 TRUE, "emit-signals", FALSE, NULL);
2294 g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
2295 FALSE, "buffer-list", TRUE, NULL);
2297 data = g_new0 (AppSinkSrcData, 1);
2298 data->appsink = appsink;
2299 data->appsrc = appsrc;
2301 sinkpad = gst_element_get_static_pad (appsink, "sink");
2302 gst_pad_add_probe (sinkpad,
2303 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2304 appsink_pad_probe, data, NULL);
2305 gst_object_unref (sinkpad);
2307 srcpad = gst_element_get_static_pad (appsrc, "src");
2308 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2309 appsrc_pad_probe, data, NULL);
2310 gst_object_unref (srcpad);
2312 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2314 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2317 streampad = gst_ghost_pad_new (name, pad);
2318 gst_pad_set_active (streampad, TRUE);
2319 gst_element_add_pad (priv->element, streampad);
2323 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2325 data->stream = stream;
2327 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2328 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2329 gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
2330 gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
2331 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2332 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2333 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2334 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2335 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2337 g_ptr_array_add (priv->streams, stream);
2339 if (GST_PAD_IS_SRC (pad)) {
2343 g_list_free (priv->payloads);
2344 priv->payloads = _find_payload_types (media);
2346 n = priv->streams->len;
2347 for (i = 0; i < n; i++) {
2348 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2349 guint rtx_pt = _next_available_pt (priv->payloads);
2352 GST_WARNING ("Ran out of space of dynamic payload types");
2356 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2359 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2362 g_mutex_unlock (&priv->lock);
2364 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2371 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2373 GstRTSPMediaPrivate *priv;
2375 AppSinkSrcData *data;
2379 g_mutex_lock (&priv->lock);
2380 /* remove the ghostpad */
2381 srcpad = gst_rtsp_stream_get_srcpad (stream);
2382 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2384 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2385 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2386 else if (GST_OBJECT_PARENT (data->appsrc) ==
2387 GST_OBJECT_CAST (priv->element))
2388 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2389 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2390 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2391 else if (GST_OBJECT_PARENT (data->appsink) ==
2392 GST_OBJECT_CAST (priv->element))
2393 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2395 gst_element_remove_pad (priv->element, srcpad);
2397 gst_object_unref (srcpad);
2398 /* now remove the stream */
2399 g_object_ref (stream);
2400 g_ptr_array_remove (priv->streams, stream);
2401 g_mutex_unlock (&priv->lock);
2403 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2406 g_object_unref (stream);
2410 * gst_rtsp_media_n_streams:
2411 * @media: a #GstRTSPMedia
2413 * Get the number of streams in this media.
2415 * Returns: The number of streams.
2418 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2420 GstRTSPMediaPrivate *priv;
2423 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2427 g_mutex_lock (&priv->lock);
2428 res = priv->streams->len;
2429 g_mutex_unlock (&priv->lock);
2435 * gst_rtsp_media_get_stream:
2436 * @media: a #GstRTSPMedia
2437 * @idx: the stream index
2439 * Retrieve the stream with index @idx from @media.
2441 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2442 * @idx or %NULL when a stream with that index did not exist.
2445 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2447 GstRTSPMediaPrivate *priv;
2450 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2454 g_mutex_lock (&priv->lock);
2455 if (idx < priv->streams->len)
2456 res = g_ptr_array_index (priv->streams, idx);
2459 g_mutex_unlock (&priv->lock);
2465 * gst_rtsp_media_find_stream:
2466 * @media: a #GstRTSPMedia
2467 * @control: the control of the stream
2469 * Find a stream in @media with @control as the control uri.
2471 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2472 * control uri @control or %NULL when a stream with that control did
2476 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2478 GstRTSPMediaPrivate *priv;
2482 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2483 g_return_val_if_fail (control != NULL, NULL);
2489 g_mutex_lock (&priv->lock);
2490 for (i = 0; i < priv->streams->len; i++) {
2491 GstRTSPStream *test;
2493 test = g_ptr_array_index (priv->streams, i);
2494 if (gst_rtsp_stream_has_control (test, control)) {
2499 g_mutex_unlock (&priv->lock);
2504 /* called with state-lock */
2506 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2507 GstRTSPRangeUnit unit)
2509 return gst_rtsp_range_convert_units (range, unit);
2513 * gst_rtsp_media_get_range_string:
2514 * @media: a #GstRTSPMedia
2515 * @play: for the PLAY request
2516 * @unit: the unit to use for the string
2518 * Get the current range as a string. @media must be prepared with
2519 * gst_rtsp_media_prepare ().
2521 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2524 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2525 GstRTSPRangeUnit unit)
2527 GstRTSPMediaClass *klass;
2528 GstRTSPMediaPrivate *priv;
2530 GstRTSPTimeRange range;
2532 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2533 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2534 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2538 g_rec_mutex_lock (&priv->state_lock);
2539 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2540 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2543 /* Update the range value with current position/duration */
2544 g_mutex_lock (&priv->lock);
2545 collect_media_stats (media);
2548 range = priv->range;
2550 if (!play && priv->n_active > 0) {
2551 range.min.type = GST_RTSP_TIME_NOW;
2552 range.min.seconds = -1;
2554 g_mutex_unlock (&priv->lock);
2555 g_rec_mutex_unlock (&priv->state_lock);
2557 if (!klass->convert_range (media, &range, unit))
2558 goto conversion_failed;
2560 result = gst_rtsp_range_to_string (&range);
2567 GST_WARNING ("media %p was not prepared", media);
2568 g_rec_mutex_unlock (&priv->state_lock);
2573 GST_WARNING ("range conversion to unit %d failed", unit);
2579 gst_rtsp_media_has_completed_sender (GstRTSPMedia * media)
2581 GstRTSPMediaPrivate *priv = media->priv;
2582 gboolean sender = FALSE;
2585 g_mutex_lock (&priv->lock);
2586 for (i = 0; i < priv->streams->len; i++) {
2587 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2588 if (gst_rtsp_stream_is_complete (stream) &&
2589 gst_rtsp_stream_is_sender (stream)) {
2594 g_mutex_unlock (&priv->lock);
2600 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2602 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2606 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2608 GstRTSPMediaPrivate *priv = media->priv;
2610 GST_DEBUG ("media %p set blocked %d", media, blocked);
2611 priv->blocked = blocked;
2612 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2616 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2618 gst_rtsp_stream_unblock_linked (stream);
2622 media_unblock_linked (GstRTSPMedia * media)
2624 GstRTSPMediaPrivate *priv = media->priv;
2626 GST_DEBUG ("media %p unblocking linked streams", media);
2627 /* media is not blocked any longer, as it contains active streams,
2628 * streams that are complete */
2629 priv->blocked = FALSE;
2630 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2634 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2636 GstRTSPMediaPrivate *priv = media->priv;
2638 g_mutex_lock (&priv->lock);
2639 priv->status = status;
2640 GST_DEBUG ("setting new status to %d", status);
2641 g_cond_broadcast (&priv->cond);
2642 g_mutex_unlock (&priv->lock);
2646 * gst_rtsp_media_get_status:
2647 * @media: a #GstRTSPMedia
2649 * Get the status of @media. When @media is busy preparing, this function waits
2650 * until @media is prepared or in error.
2652 * Returns: the status of @media.
2655 gst_rtsp_media_get_status (GstRTSPMedia * media)
2657 GstRTSPMediaPrivate *priv = media->priv;
2658 GstRTSPMediaStatus result;
2661 g_mutex_lock (&priv->lock);
2662 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2663 /* while we are preparing, wait */
2664 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2665 GST_DEBUG ("waiting for status change");
2666 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2667 GST_DEBUG ("timeout, assuming error status");
2668 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2671 /* could be success or error */
2672 result = priv->status;
2673 GST_DEBUG ("got status %d", result);
2674 g_mutex_unlock (&priv->lock);
2680 * gst_rtsp_media_seek_full:
2681 * @media: a #GstRTSPMedia
2682 * @range: (transfer none): a #GstRTSPTimeRange
2683 * @flags: The minimal set of #GstSeekFlags to use
2685 * Seek the pipeline of @media to @range. @media must be prepared with
2686 * gst_rtsp_media_prepare(). In order to perform the seek operation,
2687 * the pipeline must contain all needed transport parts (transport sinks).
2689 * Returns: %TRUE on success.
2694 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2697 GstRTSPMediaClass *klass;
2698 GstRTSPMediaPrivate *priv;
2700 GstClockTime start, stop;
2701 GstSeekType start_type, stop_type;
2702 gint64 current_position;
2704 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2706 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2707 g_return_val_if_fail (range != NULL, FALSE);
2708 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2712 g_rec_mutex_lock (&priv->state_lock);
2713 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2716 /* check if the media pipeline is complete in order to perform a
2717 * seek operation on it */
2718 if (!check_complete (media))
2721 /* Update the seekable state of the pipeline in case it changed */
2722 check_seekable (media);
2724 if (priv->seekable == 0) {
2725 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2726 " not seekable streams.");
2729 } else if (priv->seekable < 0) {
2733 start_type = stop_type = GST_SEEK_TYPE_NONE;
2735 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2737 gst_rtsp_range_get_times (range, &start, &stop);
2739 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2740 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2741 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2742 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2744 current_position = -1;
2745 if (klass->query_position)
2746 klass->query_position (media, ¤t_position);
2747 GST_INFO ("current media position %" GST_TIME_FORMAT,
2748 GST_TIME_ARGS (current_position));
2750 if (start != GST_CLOCK_TIME_NONE)
2751 start_type = GST_SEEK_TYPE_SET;
2753 if (priv->range_stop == stop)
2754 stop = GST_CLOCK_TIME_NONE;
2755 else if (stop != GST_CLOCK_TIME_NONE)
2756 stop_type = GST_SEEK_TYPE_SET;
2758 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2759 gboolean had_flags = flags != 0;
2761 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2762 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2764 /* depends on the current playing state of the pipeline. We might need to
2765 * queue this until we get EOS. */
2767 flags |= GST_SEEK_FLAG_FLUSH;
2769 flags = GST_SEEK_FLAG_FLUSH;
2772 /* if range start was not supplied we must continue from current position.
2773 * but since we're doing a flushing seek, let us query the current position
2774 * so we end up at exactly the same position after the seek. */
2775 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2776 if (current_position == -1) {
2777 GST_WARNING ("current position unknown");
2779 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2780 GST_TIME_ARGS (current_position));
2781 start = current_position;
2782 start_type = GST_SEEK_TYPE_SET;
2784 flags |= GST_SEEK_FLAG_ACCURATE;
2787 /* only set keyframe flag when modifying start */
2788 if (start_type != GST_SEEK_TYPE_NONE)
2790 flags |= GST_SEEK_FLAG_KEY_UNIT;
2793 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2794 GST_DEBUG ("not seeking because no position change");
2797 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2799 /* FIXME, we only do forwards playback, no trick modes yet */
2800 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2801 flags, start_type, start, stop_type, stop);
2803 /* and block for the seek to complete */
2804 GST_INFO ("done seeking %d", res);
2808 g_rec_mutex_unlock (&priv->state_lock);
2810 /* wait until pipeline is prerolled again, this will also collect stats */
2811 if (!wait_preroll (media))
2812 goto preroll_failed;
2814 g_rec_mutex_lock (&priv->state_lock);
2815 GST_INFO ("prerolled again");
2818 GST_INFO ("no seek needed");
2821 g_rec_mutex_unlock (&priv->state_lock);
2828 g_rec_mutex_unlock (&priv->state_lock);
2829 GST_INFO ("media %p is not prepared", media);
2834 g_rec_mutex_unlock (&priv->state_lock);
2835 GST_INFO ("pipeline is not complete");
2840 g_rec_mutex_unlock (&priv->state_lock);
2841 GST_INFO ("pipeline is not seekable");
2846 g_rec_mutex_unlock (&priv->state_lock);
2847 GST_WARNING ("conversion to npt not supported");
2852 g_rec_mutex_unlock (&priv->state_lock);
2853 GST_INFO ("seeking failed");
2854 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2859 GST_WARNING ("failed to preroll after seek");
2866 * gst_rtsp_media_seek:
2867 * @media: a #GstRTSPMedia
2868 * @range: (transfer none): a #GstRTSPTimeRange
2870 * Seek the pipeline of @media to @range. @media must be prepared with
2871 * gst_rtsp_media_prepare().
2873 * Returns: %TRUE on success.
2876 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2878 return gst_rtsp_media_seek_full (media, range, 0);
2883 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2885 *blocked &= gst_rtsp_stream_is_blocking (stream);
2889 media_streams_blocking (GstRTSPMedia * media)
2891 gboolean blocking = TRUE;
2893 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2899 static GstStateChangeReturn
2900 set_state (GstRTSPMedia * media, GstState state)
2902 GstRTSPMediaPrivate *priv = media->priv;
2903 GstStateChangeReturn ret;
2905 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2907 ret = gst_element_set_state (priv->pipeline, state);
2910 gchar *filename = NULL;
2911 filename = g_strdup_printf ("media_%s", gst_element_state_get_name (state));
2912 GST_DEBUG_BIN_TO_DOT_FILE (GST_BIN (priv->pipeline),
2913 GST_DEBUG_GRAPH_SHOW_ALL, filename);
2921 GstStateChangeReturn
2922 gst_rtsp_media_set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2924 GstRTSPMediaPrivate *priv = media->priv;
2925 GstStateChangeReturn ret;
2927 GST_INFO ("set target state to %s for media %p",
2928 gst_element_state_get_name (state), media);
2929 priv->target_state = state;
2931 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2932 priv->target_state, NULL);
2935 ret = set_state (media, state);
2937 ret = GST_STATE_CHANGE_SUCCESS;
2942 /* called with state-lock */
2944 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2946 GstRTSPMediaPrivate *priv = media->priv;
2947 GstMessageType type;
2949 type = GST_MESSAGE_TYPE (message);
2952 case GST_MESSAGE_STATE_CHANGED:
2954 GstState old, new, pending;
2956 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2959 gst_message_parse_state_changed (message, &old, &new, &pending);
2961 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2962 gst_element_state_get_name (old), gst_element_state_get_name (new),
2963 gst_element_state_get_name (pending));
2964 if (priv->no_more_pads_pending == 0
2965 && gst_rtsp_media_is_receive_only (media) && old == GST_STATE_READY
2966 && new == GST_STATE_PAUSED) {
2967 GST_INFO ("%p: went to PAUSED, prepared now", media);
2968 g_mutex_lock (&priv->lock);
2969 collect_media_stats (media);
2970 g_mutex_unlock (&priv->lock);
2972 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2973 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2978 case GST_MESSAGE_BUFFERING:
2982 gst_message_parse_buffering (message, &percent);
2984 /* no state management needed for live pipelines */
2988 if (percent == 100) {
2989 /* a 100% message means buffering is done */
2990 priv->buffering = FALSE;
2991 /* if the desired state is playing, go back */
2992 if (priv->target_state == GST_STATE_PLAYING) {
2993 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2994 set_state (media, GST_STATE_PLAYING);
2996 GST_INFO ("Buffering done");
2999 /* buffering busy */
3000 if (priv->buffering == FALSE) {
3001 if (priv->target_state == GST_STATE_PLAYING) {
3002 /* we were not buffering but PLAYING, PAUSE the pipeline. */
3003 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
3004 set_state (media, GST_STATE_PAUSED);
3006 GST_INFO ("Buffering ...");
3009 priv->buffering = TRUE;
3013 case GST_MESSAGE_LATENCY:
3015 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
3018 case GST_MESSAGE_ERROR:
3023 gst_message_parse_error (message, &gerror, &debug);
3024 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
3025 g_error_free (gerror);
3028 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3031 case GST_MESSAGE_WARNING:
3036 gst_message_parse_warning (message, &gerror, &debug);
3037 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
3038 g_error_free (gerror);
3042 case GST_MESSAGE_ELEMENT:
3044 const GstStructure *s;
3046 s = gst_message_get_structure (message);
3047 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
3048 GST_DEBUG ("media received blocking message");
3049 if (priv->blocked && media_streams_blocking (media) &&
3050 priv->no_more_pads_pending == 0) {
3051 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
3052 g_mutex_lock (&priv->lock);
3053 collect_media_stats (media);
3054 g_mutex_unlock (&priv->lock);
3056 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3057 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3062 case GST_MESSAGE_STREAM_STATUS:
3064 case GST_MESSAGE_ASYNC_DONE:
3065 if (priv->complete) {
3066 /* receive the final ASYNC_DONE, that is posted by the media pipeline
3067 * after all the transport parts have been successfully added to
3068 * the media streams. */
3069 GST_DEBUG_OBJECT (media, "got async-done");
3070 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3071 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3074 case GST_MESSAGE_EOS:
3075 GST_INFO ("%p: got EOS", media);
3077 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
3078 GST_DEBUG ("shutting down after EOS");
3079 finish_unprepare (media);
3083 GST_INFO ("%p: got message type %d (%s)", media, type,
3084 gst_message_type_get_name (type));
3091 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
3093 GstRTSPMediaPrivate *priv = media->priv;
3094 GstRTSPMediaClass *klass;
3097 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3099 g_rec_mutex_lock (&priv->state_lock);
3100 if (klass->handle_message)
3101 ret = klass->handle_message (media, message);
3104 g_rec_mutex_unlock (&priv->state_lock);
3110 watch_destroyed (GstRTSPMedia * media)
3112 GST_DEBUG_OBJECT (media, "source destroyed");
3113 g_object_unref (media);
3117 find_payload_element (GstElement * payloader)
3119 GstElement *pay = NULL;
3121 if (GST_IS_BIN (payloader)) {
3123 GValue item = { 0 };
3125 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
3126 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
3127 GstElement *element = (GstElement *) g_value_get_object (&item);
3128 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
3132 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
3136 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
3137 pay = gst_object_ref (element);
3138 g_value_unset (&item);
3141 g_value_unset (&item);
3143 gst_iterator_free (iter);
3145 pay = g_object_ref (payloader);
3151 /* called from streaming threads */
3153 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3155 GstRTSPMediaPrivate *priv = media->priv;
3156 GstRTSPStream *stream;
3159 /* find the real payload element */
3160 pay = find_payload_element (element);
3161 stream = gst_rtsp_media_create_stream (media, pay, pad);
3162 gst_object_unref (pay);
3164 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3166 g_rec_mutex_lock (&priv->state_lock);
3167 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3170 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
3172 /* join the element in the PAUSED state because this callback is
3173 * called from the streaming thread and it is PAUSED */
3174 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3175 priv->rtpbin, GST_STATE_PAUSED)) {
3176 GST_WARNING ("failed to join bin element");
3180 gst_rtsp_stream_set_blocked (stream, TRUE);
3182 g_rec_mutex_unlock (&priv->state_lock);
3189 gst_rtsp_media_remove_stream (media, stream);
3190 g_rec_mutex_unlock (&priv->state_lock);
3191 GST_INFO ("ignore pad because we are not preparing");
3197 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3199 GstRTSPMediaPrivate *priv = media->priv;
3200 GstRTSPStream *stream;
3202 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
3206 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3208 g_rec_mutex_lock (&priv->state_lock);
3209 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3210 g_rec_mutex_unlock (&priv->state_lock);
3212 gst_rtsp_media_remove_stream (media, stream);
3216 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
3218 GstRTSPMediaPrivate *priv = media->priv;
3220 GST_INFO_OBJECT (element, "no more pads");
3221 g_mutex_lock (&priv->lock);
3222 priv->no_more_pads_pending--;
3223 g_mutex_unlock (&priv->lock);
3226 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
3228 struct _DynPaySignalHandlers
3230 gulong pad_added_handler;
3231 gulong pad_removed_handler;
3232 gulong no_more_pads_handler;
3236 start_preroll (GstRTSPMedia * media)
3238 GstRTSPMediaPrivate *priv = media->priv;
3239 GstStateChangeReturn ret;
3241 GST_INFO ("setting pipeline to PAUSED for media %p", media);
3243 /* start blocked since it is possible that there are no sink elements yet */
3244 media_streams_set_blocked (media, TRUE);
3245 ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
3248 case GST_STATE_CHANGE_SUCCESS:
3249 GST_INFO ("SUCCESS state change for media %p", media);
3251 case GST_STATE_CHANGE_ASYNC:
3252 GST_INFO ("ASYNC state change for media %p", media);
3254 case GST_STATE_CHANGE_NO_PREROLL:
3255 /* we need to go to PLAYING */
3256 GST_INFO ("NO_PREROLL state change: live media %p", media);
3257 /* FIXME we disable seeking for live streams for now. We should perform a
3258 * seeking query in preroll instead */
3259 priv->seekable = -1;
3260 priv->is_live = TRUE;
3262 ret = set_state (media, GST_STATE_PLAYING);
3263 if (ret == GST_STATE_CHANGE_FAILURE)
3266 case GST_STATE_CHANGE_FAILURE:
3274 GST_WARNING ("failed to preroll pipeline");
3280 wait_preroll (GstRTSPMedia * media)
3282 GstRTSPMediaStatus status;
3284 GST_DEBUG ("wait to preroll pipeline");
3286 /* wait until pipeline is prerolled */
3287 status = gst_rtsp_media_get_status (media);
3288 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3289 goto preroll_failed;
3295 GST_WARNING ("failed to preroll pipeline");
3301 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3303 GstRTSPMediaPrivate *priv = media->priv;
3304 GstRTSPStream *stream = NULL;
3306 GstElement *res = NULL;
3308 g_mutex_lock (&priv->lock);
3309 for (i = 0; i < priv->streams->len; i++) {
3310 stream = g_ptr_array_index (priv->streams, i);
3312 if (sessid == gst_rtsp_stream_get_index (stream))
3317 g_mutex_unlock (&priv->lock);
3320 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3326 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3328 GstRTSPMediaPrivate *priv = media->priv;
3329 GstRTSPStream *stream = NULL;
3331 GstElement *res = NULL;
3333 g_mutex_lock (&priv->lock);
3334 for (i = 0; i < priv->streams->len; i++) {
3335 stream = g_ptr_array_index (priv->streams, i);
3337 if (sessid == gst_rtsp_stream_get_index (stream))
3342 g_mutex_unlock (&priv->lock);
3345 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3351 request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3353 GstRTSPMediaPrivate *priv = media->priv;
3354 GstRTSPStream *stream = NULL;
3356 GstElement *res = NULL;
3358 g_mutex_lock (&priv->lock);
3359 for (i = 0; i < priv->streams->len; i++) {
3360 stream = g_ptr_array_index (priv->streams, i);
3362 if (sessid == gst_rtsp_stream_get_index (stream))
3367 g_mutex_unlock (&priv->lock);
3370 res = gst_rtsp_stream_request_ulpfec_decoder (stream, rtpbin, sessid);
3377 new_storage_cb (GstElement * rtpbin, GObject * storage, guint sessid,
3378 GstRTSPMedia * media)
3380 g_object_set (storage, "size-time", (media->priv->latency + 50) * GST_MSECOND,
3385 start_prepare (GstRTSPMedia * media)
3387 GstRTSPMediaPrivate *priv = media->priv;
3391 g_rec_mutex_lock (&priv->state_lock);
3392 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3393 goto no_longer_preparing;
3395 g_signal_connect (priv->rtpbin, "new-storage", G_CALLBACK (new_storage_cb),
3397 g_signal_connect (priv->rtpbin, "request-fec-decoder",
3398 G_CALLBACK (request_fec_decoder), media);
3400 /* link streams we already have, other streams might appear when we have
3401 * dynamic elements */
3402 for (i = 0; i < priv->streams->len; i++) {
3403 GstRTSPStream *stream;
3405 stream = g_ptr_array_index (priv->streams, i);
3407 if (priv->rtx_time > 0) {
3408 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3409 g_signal_connect (priv->rtpbin, "request-aux-sender",
3410 (GCallback) request_aux_sender, media);
3413 if (priv->do_retransmission) {
3414 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3415 (GCallback) request_aux_receiver, media);
3418 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3419 priv->rtpbin, GST_STATE_NULL)) {
3420 goto join_bin_failed;
3423 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARING], 0, stream,
3428 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
3429 "do-lost", TRUE, NULL);
3431 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3432 GstElement *elem = walk->data;
3433 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3435 GST_INFO ("adding callbacks for dynamic element %p", elem);
3437 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3438 (GCallback) pad_added_cb, media);
3439 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3440 (GCallback) pad_removed_cb, media);
3441 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3442 (GCallback) no_more_pads_cb, media);
3444 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3447 if (priv->nb_dynamic_elements == 0 && gst_rtsp_media_is_receive_only (media)) {
3448 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3449 * a second ASYNC state change failing */
3450 priv->is_live = TRUE;
3451 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3452 } else if (!start_preroll (media)) {
3453 goto preroll_failed;
3456 g_rec_mutex_unlock (&priv->state_lock);
3460 no_longer_preparing:
3462 GST_INFO ("media is no longer preparing");
3463 g_rec_mutex_unlock (&priv->state_lock);
3468 GST_WARNING ("failed to join bin element");
3469 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3470 g_rec_mutex_unlock (&priv->state_lock);
3475 GST_WARNING ("failed to preroll pipeline");
3476 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3477 g_rec_mutex_unlock (&priv->state_lock);
3483 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3485 GstRTSPMediaPrivate *priv;
3486 GstRTSPMediaClass *klass;
3488 GMainContext *context;
3493 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3495 if (!klass->create_rtpbin)
3496 goto no_create_rtpbin;
3498 priv->rtpbin = klass->create_rtpbin (media);
3499 if (priv->rtpbin != NULL) {
3500 gboolean success = TRUE;
3502 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3504 if (klass->setup_rtpbin)
3505 success = klass->setup_rtpbin (media, priv->rtpbin);
3507 if (success == FALSE) {
3508 gst_object_unref (priv->rtpbin);
3509 priv->rtpbin = NULL;
3512 if (priv->rtpbin == NULL)
3515 priv->thread = thread;
3516 context = (thread != NULL) ? (thread->context) : NULL;
3518 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3520 /* add the pipeline bus to our custom mainloop */
3521 priv->source = gst_bus_create_watch (bus);
3522 gst_object_unref (bus);
3524 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3525 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3527 priv->id = g_source_attach (priv->source, context);
3529 /* add stuff to the bin */
3530 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3532 /* do remainder in context */
3533 source = g_idle_source_new ();
3534 g_source_set_callback (source, (GSourceFunc) start_prepare,
3535 g_object_ref (media), (GDestroyNotify) g_object_unref);
3536 g_source_attach (source, context);
3537 g_source_unref (source);
3544 GST_ERROR ("no create_rtpbin function");
3545 g_critical ("no create_rtpbin vmethod function set");
3550 GST_WARNING ("no rtpbin element");
3551 g_warning ("failed to create element 'rtpbin', check your installation");
3557 * gst_rtsp_media_prepare:
3558 * @media: a #GstRTSPMedia
3559 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3560 * bus handler or %NULL
3562 * Prepare @media for streaming. This function will create the objects
3563 * to manage the streaming. A pipeline must have been set on @media with
3564 * gst_rtsp_media_take_pipeline().
3566 * It will preroll the pipeline and collect vital information about the streams
3567 * such as the duration.
3569 * Returns: %TRUE on success.
3572 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3574 GstRTSPMediaPrivate *priv;
3575 GstRTSPMediaClass *klass;
3577 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3581 g_rec_mutex_lock (&priv->state_lock);
3582 priv->prepare_count++;
3584 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3585 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3588 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3591 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3592 goto not_unprepared;
3594 if (!priv->reusable && priv->reused)
3597 GST_INFO ("preparing media %p", media);
3599 /* reset some variables */
3600 priv->is_live = FALSE;
3601 priv->seekable = -1;
3602 priv->buffering = FALSE;
3603 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3605 /* we're preparing now */
3606 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3608 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3609 if (klass->prepare) {
3610 if (!klass->prepare (media, thread))
3611 goto prepare_failed;
3615 g_rec_mutex_unlock (&priv->state_lock);
3617 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3618 * able to do this async so that we don't block the server thread. */
3619 if (!wait_preroll (media))
3620 goto preroll_failed;
3622 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3624 GST_INFO ("object %p is prerolled", media);
3631 /* we are not going to use the giving thread, so stop it. */
3633 gst_rtsp_thread_stop (thread);
3638 GST_LOG ("media %p was prepared", media);
3639 /* we are not going to use the giving thread, so stop it. */
3641 gst_rtsp_thread_stop (thread);
3642 g_rec_mutex_unlock (&priv->state_lock);
3648 /* we are not going to use the giving thread, so stop it. */
3650 gst_rtsp_thread_stop (thread);
3651 GST_WARNING ("media %p was not unprepared", media);
3652 priv->prepare_count--;
3653 g_rec_mutex_unlock (&priv->state_lock);
3658 /* we are not going to use the giving thread, so stop it. */
3660 gst_rtsp_thread_stop (thread);
3661 priv->prepare_count--;
3662 g_rec_mutex_unlock (&priv->state_lock);
3663 GST_WARNING ("can not reuse media %p", media);
3668 /* we are not going to use the giving thread, so stop it. */
3670 gst_rtsp_thread_stop (thread);
3671 priv->prepare_count--;
3672 g_rec_mutex_unlock (&priv->state_lock);
3673 GST_ERROR ("failed to prepare media");
3678 GST_WARNING ("failed to preroll pipeline");
3679 gst_rtsp_media_unprepare (media);
3684 /* must be called with state-lock */
3686 finish_unprepare (GstRTSPMedia * media)
3688 GstRTSPMediaPrivate *priv = media->priv;
3692 if (priv->finishing_unprepare)
3694 priv->finishing_unprepare = TRUE;
3696 GST_DEBUG ("shutting down");
3698 /* release the lock on shutdown, otherwise pad_added_cb might try to
3699 * acquire the lock and then we deadlock */
3700 g_rec_mutex_unlock (&priv->state_lock);
3701 set_state (media, GST_STATE_NULL);
3702 g_rec_mutex_lock (&priv->state_lock);
3704 media_streams_set_blocked (media, FALSE);
3706 for (i = 0; i < priv->streams->len; i++) {
3707 GstRTSPStream *stream;
3709 GST_INFO ("Removing elements of stream %d from pipeline", i);
3711 stream = g_ptr_array_index (priv->streams, i);
3713 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3715 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARING], 0, stream,
3719 /* remove the pad signal handlers */
3720 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3721 GstElement *elem = walk->data;
3722 DynPaySignalHandlers *handlers;
3725 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3726 g_assert (handlers != NULL);
3728 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3729 g_signal_handler_disconnect (G_OBJECT (elem),
3730 handlers->pad_removed_handler);
3731 g_signal_handler_disconnect (G_OBJECT (elem),
3732 handlers->no_more_pads_handler);
3734 g_slice_free (DynPaySignalHandlers, handlers);
3737 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3738 priv->rtpbin = NULL;
3741 gst_object_unref (priv->nettime);
3742 priv->nettime = NULL;
3744 priv->reused = TRUE;
3745 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3747 /* when the media is not reusable, this will effectively unref the media and
3749 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3751 /* the source has the last ref to the media */
3753 GST_DEBUG ("destroy source");
3754 g_source_destroy (priv->source);
3755 g_source_unref (priv->source);
3758 GST_DEBUG ("stop thread");
3759 gst_rtsp_thread_stop (priv->thread);
3762 priv->finishing_unprepare = FALSE;
3765 /* called with state-lock */
3767 default_unprepare (GstRTSPMedia * media)
3769 GstRTSPMediaPrivate *priv = media->priv;
3771 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3773 if (priv->eos_shutdown) {
3774 GST_DEBUG ("sending EOS for shutdown");
3775 /* ref so that we don't disappear */
3776 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3777 /* we need to go to playing again for the EOS to propagate, normally in this
3778 * state, nothing is receiving data from us anymore so this is ok. */
3779 set_state (media, GST_STATE_PLAYING);
3781 finish_unprepare (media);
3787 * gst_rtsp_media_unprepare:
3788 * @media: a #GstRTSPMedia
3790 * Unprepare @media. After this call, the media should be prepared again before
3791 * it can be used again. If the media is set to be non-reusable, a new instance
3794 * Returns: %TRUE on success.
3797 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3799 GstRTSPMediaPrivate *priv;
3802 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3806 g_rec_mutex_lock (&priv->state_lock);
3807 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3808 goto was_unprepared;
3810 priv->prepare_count--;
3811 if (priv->prepare_count > 0)
3814 GST_INFO ("unprepare media %p", media);
3815 gst_rtsp_media_set_target_state (media, GST_STATE_NULL, FALSE);
3818 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3819 GstRTSPMediaClass *klass;
3821 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3822 if (klass->unprepare)
3823 success = klass->unprepare (media);
3825 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3826 finish_unprepare (media);
3828 g_rec_mutex_unlock (&priv->state_lock);
3834 g_rec_mutex_unlock (&priv->state_lock);
3835 GST_INFO ("media %p was already unprepared", media);
3840 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3841 g_rec_mutex_unlock (&priv->state_lock);
3846 /* should be called with state-lock */
3848 get_clock_unlocked (GstRTSPMedia * media)
3850 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3851 GST_DEBUG_OBJECT (media, "media was not prepared");
3854 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3858 * gst_rtsp_media_get_clock:
3859 * @media: a #GstRTSPMedia
3861 * Get the clock that is used by the pipeline in @media.
3863 * @media must be prepared before this method returns a valid clock object.
3865 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3868 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3871 GstRTSPMediaPrivate *priv;
3873 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3877 g_rec_mutex_lock (&priv->state_lock);
3878 clock = get_clock_unlocked (media);
3879 g_rec_mutex_unlock (&priv->state_lock);
3885 * gst_rtsp_media_get_base_time:
3886 * @media: a #GstRTSPMedia
3888 * Get the base_time that is used by the pipeline in @media.
3890 * @media must be prepared before this method returns a valid base_time.
3892 * Returns: the base_time used by @media.
3895 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3897 GstClockTime result;
3898 GstRTSPMediaPrivate *priv;
3900 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3904 g_rec_mutex_lock (&priv->state_lock);
3905 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3908 result = gst_element_get_base_time (media->priv->pipeline);
3909 g_rec_mutex_unlock (&priv->state_lock);
3916 g_rec_mutex_unlock (&priv->state_lock);
3917 GST_DEBUG_OBJECT (media, "media was not prepared");
3918 return GST_CLOCK_TIME_NONE;
3923 * gst_rtsp_media_get_time_provider:
3924 * @media: a #GstRTSPMedia
3925 * @address: (allow-none): an address or %NULL
3926 * @port: a port or 0
3928 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3929 * will listen on @address and @port for client time requests.
3931 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3933 GstNetTimeProvider *
3934 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3937 GstRTSPMediaPrivate *priv;
3938 GstNetTimeProvider *provider = NULL;
3940 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3944 g_rec_mutex_lock (&priv->state_lock);
3945 if (priv->time_provider) {
3946 if ((provider = priv->nettime) == NULL) {
3949 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3950 provider = gst_net_time_provider_new (clock, address, port);
3951 gst_object_unref (clock);
3953 priv->nettime = provider;
3957 g_rec_mutex_unlock (&priv->state_lock);
3960 gst_object_ref (provider);
3966 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3968 return gst_rtsp_sdp_from_media (sdp, info, media);
3972 * gst_rtsp_media_setup_sdp:
3973 * @media: a #GstRTSPMedia
3974 * @sdp: (transfer none): a #GstSDPMessage
3975 * @info: (transfer none): a #GstSDPInfo
3977 * Add @media specific info to @sdp. @info is used to configure the connection
3978 * information in the SDP.
3980 * Returns: TRUE on success.
3983 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3986 GstRTSPMediaPrivate *priv;
3987 GstRTSPMediaClass *klass;
3990 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3991 g_return_val_if_fail (sdp != NULL, FALSE);
3992 g_return_val_if_fail (info != NULL, FALSE);
3996 g_rec_mutex_lock (&priv->state_lock);
3998 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4000 if (!klass->setup_sdp)
4003 res = klass->setup_sdp (media, sdp, info);
4005 g_rec_mutex_unlock (&priv->state_lock);
4012 g_rec_mutex_unlock (&priv->state_lock);
4013 GST_ERROR ("no setup_sdp function");
4014 g_critical ("no setup_sdp vmethod function set");
4020 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4022 GstRTSPMediaPrivate *priv = media->priv;
4025 medias_len = gst_sdp_message_medias_len (sdp);
4026 if (medias_len != priv->streams->len) {
4027 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
4028 priv->streams->len, medias_len);
4032 for (i = 0; i < medias_len; i++) {
4034 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
4035 GstRTSPStream *stream;
4036 gint j, formats_len;
4037 const gchar *control;
4038 GstRTSPProfile profile, profiles;
4040 stream = g_ptr_array_index (priv->streams, i);
4042 /* TODO: Should we do something with the other SDP information? */
4045 proto = gst_sdp_media_get_proto (sdp_media);
4046 if (proto == NULL) {
4047 GST_ERROR ("%p: SDP media %d has no proto", media, i);
4051 if (g_str_equal (proto, "RTP/AVP")) {
4052 profile = GST_RTSP_PROFILE_AVP;
4053 } else if (g_str_equal (proto, "RTP/SAVP")) {
4054 profile = GST_RTSP_PROFILE_SAVP;
4055 } else if (g_str_equal (proto, "RTP/AVPF")) {
4056 profile = GST_RTSP_PROFILE_AVPF;
4057 } else if (g_str_equal (proto, "RTP/SAVPF")) {
4058 profile = GST_RTSP_PROFILE_SAVPF;
4060 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4064 profiles = gst_rtsp_stream_get_profiles (stream);
4065 if ((profiles & profile) == 0) {
4066 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4070 formats_len = gst_sdp_media_formats_len (sdp_media);
4071 for (j = 0; j < formats_len; j++) {
4076 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
4078 GST_DEBUG (" looking at %d pt: %d", j, pt);
4081 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
4083 GST_WARNING (" skipping pt %d without caps", pt);
4087 /* do some tweaks */
4088 GST_DEBUG ("mapping sdp session level attributes to caps");
4089 gst_sdp_message_attributes_to_caps (sdp, caps);
4090 GST_DEBUG ("mapping sdp media level attributes to caps");
4091 gst_sdp_media_attributes_to_caps (sdp_media, caps);
4093 s = gst_caps_get_structure (caps, 0);
4094 gst_structure_set_name (s, "application/x-rtp");
4096 if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
4097 gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
4099 gst_rtsp_stream_set_pt_map (stream, pt, caps);
4100 gst_caps_unref (caps);
4103 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
4105 gst_rtsp_stream_set_control (stream, control);
4113 * gst_rtsp_media_handle_sdp:
4114 * @media: a #GstRTSPMedia
4115 * @sdp: (transfer none): a #GstSDPMessage
4117 * Configure an SDP on @media for receiving streams
4119 * Returns: TRUE on success.
4122 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4124 GstRTSPMediaPrivate *priv;
4125 GstRTSPMediaClass *klass;
4128 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4129 g_return_val_if_fail (sdp != NULL, FALSE);
4133 g_rec_mutex_lock (&priv->state_lock);
4135 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4137 if (!klass->handle_sdp)
4140 res = klass->handle_sdp (media, sdp);
4142 g_rec_mutex_unlock (&priv->state_lock);
4149 g_rec_mutex_unlock (&priv->state_lock);
4150 GST_ERROR ("no handle_sdp function");
4151 g_critical ("no handle_sdp vmethod function set");
4157 do_set_seqnum (GstRTSPStream * stream)
4161 if (gst_rtsp_stream_is_sender (stream)) {
4162 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
4163 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
4167 /* call with state_lock */
4169 default_suspend (GstRTSPMedia * media)
4171 GstRTSPMediaPrivate *priv = media->priv;
4172 GstStateChangeReturn ret;
4174 switch (priv->suspend_mode) {
4175 case GST_RTSP_SUSPEND_MODE_NONE:
4176 GST_DEBUG ("media %p no suspend", media);
4178 case GST_RTSP_SUSPEND_MODE_PAUSE:
4179 GST_DEBUG ("media %p suspend to PAUSED", media);
4180 ret = gst_rtsp_media_set_target_state (media, GST_STATE_PAUSED, TRUE);
4181 if (ret == GST_STATE_CHANGE_FAILURE)
4184 case GST_RTSP_SUSPEND_MODE_RESET:
4185 GST_DEBUG ("media %p suspend to NULL", media);
4186 ret = gst_rtsp_media_set_target_state (media, GST_STATE_NULL, TRUE);
4187 if (ret == GST_STATE_CHANGE_FAILURE)
4189 /* Because payloader needs to set the sequence number as
4190 * monotonic, we need to preserve the sequence number
4191 * after pause. (otherwise going from pause to play, which
4192 * is actually from NULL to PLAY will create a new sequence
4194 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
4205 GST_WARNING ("failed changing pipeline's state for media %p", media);
4211 * gst_rtsp_media_suspend:
4212 * @media: a #GstRTSPMedia
4214 * Suspend @media. The state of the pipeline managed by @media is set to
4215 * GST_STATE_NULL but all streams are kept. @media can be prepared again
4216 * with gst_rtsp_media_unsuspend()
4218 * @media must be prepared with gst_rtsp_media_prepare();
4220 * Returns: %TRUE on success.
4223 gst_rtsp_media_suspend (GstRTSPMedia * media)
4225 GstRTSPMediaPrivate *priv = media->priv;
4226 GstRTSPMediaClass *klass;
4228 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4230 GST_FIXME ("suspend for dynamic pipelines needs fixing");
4232 g_rec_mutex_lock (&priv->state_lock);
4233 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
4236 /* don't attempt to suspend when something is busy */
4237 if (priv->n_active > 0)
4240 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4241 if (klass->suspend) {
4242 if (!klass->suspend (media))
4243 goto suspend_failed;
4246 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
4248 g_rec_mutex_unlock (&priv->state_lock);
4255 g_rec_mutex_unlock (&priv->state_lock);
4256 GST_WARNING ("media %p was not prepared", media);
4261 g_rec_mutex_unlock (&priv->state_lock);
4262 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4263 GST_WARNING ("failed to suspend media %p", media);
4268 /* call with state_lock */
4270 default_unsuspend (GstRTSPMedia * media)
4272 GstRTSPMediaPrivate *priv = media->priv;
4273 gboolean preroll_ok;
4275 switch (priv->suspend_mode) {
4276 case GST_RTSP_SUSPEND_MODE_NONE:
4277 if (gst_rtsp_media_is_receive_only (media))
4279 if (media_streams_blocking (media)) {
4280 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4281 /* at this point the media pipeline has been updated and contain all
4282 * specific transport parts: all active streams contain at least one sink
4283 * element and it's safe to unblock any blocked streams that are active */
4284 media_unblock_linked (media);
4286 /* streams are not blocked and media is suspended from PAUSED */
4287 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4289 g_rec_mutex_unlock (&priv->state_lock);
4290 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
4291 g_rec_mutex_lock (&priv->state_lock);
4292 goto preroll_failed;
4294 g_rec_mutex_lock (&priv->state_lock);
4296 case GST_RTSP_SUSPEND_MODE_PAUSE:
4297 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4299 case GST_RTSP_SUSPEND_MODE_RESET:
4301 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4302 /* at this point the media pipeline has been updated and contain all
4303 * specific transport parts: all active streams contain at least one sink
4304 * element and it's safe to unblock any blocked streams that are active */
4305 media_unblock_linked (media);
4306 if (!start_preroll (media))
4309 g_rec_mutex_unlock (&priv->state_lock);
4310 preroll_ok = wait_preroll (media);
4311 g_rec_mutex_lock (&priv->state_lock);
4314 goto preroll_failed;
4325 GST_WARNING ("failed to preroll pipeline");
4330 GST_WARNING ("failed to preroll pipeline");
4336 * gst_rtsp_media_unsuspend:
4337 * @media: a #GstRTSPMedia
4339 * Unsuspend @media if it was in a suspended state. This method does nothing
4340 * when the media was not in the suspended state.
4342 * Returns: %TRUE on success.
4345 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4347 GstRTSPMediaPrivate *priv = media->priv;
4348 GstRTSPMediaClass *klass;
4350 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4352 g_rec_mutex_lock (&priv->state_lock);
4353 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4356 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4357 if (klass->unsuspend) {
4358 if (!klass->unsuspend (media))
4359 goto unsuspend_failed;
4363 g_rec_mutex_unlock (&priv->state_lock);
4370 g_rec_mutex_unlock (&priv->state_lock);
4371 GST_WARNING ("failed to unsuspend media %p", media);
4372 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4377 /* must be called with state-lock */
4379 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4381 GstRTSPMediaPrivate *priv = media->priv;
4383 if (state == GST_STATE_NULL) {
4384 gst_rtsp_media_unprepare (media);
4386 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4387 gst_rtsp_media_set_target_state (media, state, FALSE);
4388 /* when we are buffering, don't update the state yet, this will be done
4389 * when buffering finishes */
4390 if (priv->buffering) {
4391 GST_INFO ("Buffering busy, delay state change");
4393 if (state == GST_STATE_PLAYING)
4394 /* make sure pads are not blocking anymore when going to PLAYING */
4395 media_unblock_linked (media);
4397 set_state (media, state);
4399 /* and suspend after pause */
4400 if (state == GST_STATE_PAUSED)
4401 gst_rtsp_media_suspend (media);
4407 * gst_rtsp_media_set_pipeline_state:
4408 * @media: a #GstRTSPMedia
4409 * @state: the target state of the pipeline
4411 * Set the state of the pipeline managed by @media to @state
4414 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4416 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4418 g_rec_mutex_lock (&media->priv->state_lock);
4419 media_set_pipeline_state_locked (media, state);
4420 g_rec_mutex_unlock (&media->priv->state_lock);
4424 * gst_rtsp_media_set_state:
4425 * @media: a #GstRTSPMedia
4426 * @state: the target state of the media
4427 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4428 * a #GPtrArray of #GstRTSPStreamTransport pointers
4430 * Set the state of @media to @state and for the transports in @transports.
4432 * @media must be prepared with gst_rtsp_media_prepare();
4434 * Returns: %TRUE on success.
4437 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4438 GPtrArray * transports)
4440 GstRTSPMediaPrivate *priv;
4442 gboolean activate, deactivate, do_state;
4445 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4446 g_return_val_if_fail (transports != NULL, FALSE);
4450 g_rec_mutex_lock (&priv->state_lock);
4452 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING
4453 && gst_rtsp_media_is_shared (media)) {
4454 g_rec_mutex_unlock (&priv->state_lock);
4455 gst_rtsp_media_get_status (media);
4456 g_rec_mutex_lock (&priv->state_lock);
4458 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4460 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4461 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4464 /* NULL and READY are the same */
4465 if (state == GST_STATE_READY)
4466 state = GST_STATE_NULL;
4468 activate = deactivate = FALSE;
4470 GST_INFO ("going to state %s media %p, target state %s",
4471 gst_element_state_get_name (state), media,
4472 gst_element_state_get_name (priv->target_state));
4475 case GST_STATE_NULL:
4476 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4477 if (priv->target_state >= GST_STATE_PAUSED)
4480 case GST_STATE_PAUSED:
4481 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4482 if (priv->target_state == GST_STATE_PLAYING)
4485 case GST_STATE_PLAYING:
4486 /* we're going to PLAYING, activate */
4492 old_active = priv->n_active;
4494 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4495 activate, deactivate);
4496 for (i = 0; i < transports->len; i++) {
4497 GstRTSPStreamTransport *trans;
4499 /* we need a non-NULL entry in the array */
4500 trans = g_ptr_array_index (transports, i);
4505 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4507 } else if (deactivate) {
4508 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4513 /* we just activated the first media, do the playing state change */
4514 if (old_active == 0 && activate)
4516 /* if we have no more active media and prepare count is not indicate
4517 * that there are new session/sessions ongoing,
4518 * do the downward state changes */
4519 else if (priv->n_active == 0 && priv->prepare_count <= 1)
4524 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4527 if (priv->target_state != state) {
4529 media_set_pipeline_state_locked (media, state);
4530 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4535 /* remember where we are */
4536 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4537 old_active != priv->n_active)) {
4538 g_mutex_lock (&priv->lock);
4539 collect_media_stats (media);
4540 g_mutex_unlock (&priv->lock);
4542 g_rec_mutex_unlock (&priv->state_lock);
4549 GST_WARNING ("media %p was not prepared", media);
4550 g_rec_mutex_unlock (&priv->state_lock);
4555 GST_WARNING ("media %p in error status while changing to state %d",
4557 if (state == GST_STATE_NULL) {
4558 for (i = 0; i < transports->len; i++) {
4559 GstRTSPStreamTransport *trans;
4561 /* we need a non-NULL entry in the array */
4562 trans = g_ptr_array_index (transports, i);
4566 gst_rtsp_stream_transport_set_active (trans, FALSE);
4570 g_rec_mutex_unlock (&priv->state_lock);
4576 * gst_rtsp_media_set_transport_mode:
4577 * @media: a #GstRTSPMedia
4578 * @mode: the new value
4580 * Sets if the media pipeline can work in PLAY or RECORD mode
4583 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4584 GstRTSPTransportMode mode)
4586 GstRTSPMediaPrivate *priv;
4588 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4592 g_mutex_lock (&priv->lock);
4593 priv->transport_mode = mode;
4594 g_mutex_unlock (&priv->lock);
4598 * gst_rtsp_media_get_transport_mode:
4599 * @media: a #GstRTSPMedia
4601 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4603 * Returns: The transport mode.
4605 GstRTSPTransportMode
4606 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4608 GstRTSPMediaPrivate *priv;
4609 GstRTSPTransportMode res;
4611 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4615 g_mutex_lock (&priv->lock);
4616 res = priv->transport_mode;
4617 g_mutex_unlock (&priv->lock);
4623 * gst_rtsp_media_seekable:
4624 * @media: a #GstRTSPMedia
4626 * Check if the pipeline for @media seek and up to what point in time,
4629 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4630 * and > 0 to indicate the longest duration between any two random access points.
4631 * %G_MAXINT64 means any value is possible.
4636 gst_rtsp_media_seekable (GstRTSPMedia * media)
4638 GstRTSPMediaPrivate *priv;
4639 GstClockTimeDiff res;
4641 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4645 /* Currently we are not able to seek on live streams,
4646 * and no stream is seekable only to the beginning */
4647 g_mutex_lock (&priv->lock);
4648 res = priv->seekable;
4649 g_mutex_unlock (&priv->lock);
4655 * gst_rtsp_media_complete_pipeline:
4656 * @media: a #GstRTSPMedia
4657 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4659 * Add a receiver and sender parts to the pipeline based on the transport from
4662 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4667 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4669 GstRTSPMediaPrivate *priv;
4672 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4673 g_return_val_if_fail (transports, FALSE);
4675 GST_DEBUG_OBJECT (media, "complete pipeline");
4679 g_mutex_lock (&priv->lock);
4680 for (i = 0; i < priv->streams->len; i++) {
4681 GstRTSPStreamTransport *transport;
4682 GstRTSPStream *stream;
4683 const GstRTSPTransport *rtsp_transport;
4685 transport = g_ptr_array_index (transports, i);
4689 stream = gst_rtsp_stream_transport_get_stream (transport);
4693 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4695 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4696 g_mutex_unlock (&priv->lock);
4701 priv->complete = TRUE;
4702 g_mutex_unlock (&priv->lock);
4708 gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
4710 GstRTSPMediaPrivate *priv = media->priv;
4711 gboolean receive_only;
4713 g_mutex_lock (&priv->lock);
4714 receive_only = is_receive_only (media);
4715 g_mutex_unlock (&priv->lock);
4717 return receive_only;
4721 gst_rtsp_media_get_pipeline (GstRTSPMedia * media)
4723 GstRTSPMediaPrivate *priv;
4725 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
4729 g_mutex_lock (&priv->lock);
4730 g_object_ref (priv->pipeline);
4731 g_mutex_unlock (&priv->lock);
4733 return priv->pipeline;
4738 gst_rtsp_media_get_rtpbin (GstRTSPMedia * media)
4740 GstRTSPMediaPrivate *priv;
4742 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
4746 g_mutex_lock (&priv->lock);
4747 g_object_ref (priv->rtpbin);
4748 g_mutex_unlock (&priv->lock);
4750 return priv->rtpbin;