2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: The media pipeline
24 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
25 * #GstRTSPSessionMedia
27 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
28 * streaming to the clients. The actual data transfer is done by the
29 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
31 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
32 * client does a DESCRIBE or SETUP of a resource.
34 * A media is created with gst_rtsp_media_new() that takes the element that will
35 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
36 * object needs to be made with the gst_rtsp_media_create_stream() which takes
37 * the payloader element and the source pad that produces the RTP stream.
39 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
40 * prepare method will add rtpbin and sinks and sources to send and receive RTP
41 * and RTCP packets from the clients. Each stream srcpad is connected to an
42 * input into the internal rtpbin.
44 * It is also possible to dynamically create #GstRTSPStream objects during the
45 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
48 * After the media is prepared, it is ready for streaming. It will usually be
49 * managed in a session with gst_rtsp_session_manage_media(). See
50 * #GstRTSPSession and #GstRTSPSessionMedia.
52 * The state of the media can be controlled with gst_rtsp_media_set_state ().
53 * Seeking can be done with gst_rtsp_media_seek(), or gst_rtsp_media_seek_full()
54 * or gst_rtsp_media_seek_trickmode() for finer control of the seek.
56 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
57 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
60 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
61 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
62 * can be prepared again after an unprepare.
64 * Last reviewed on 2013-07-11 (1.0.0)
74 #include <gst/app/gstappsrc.h>
75 #include <gst/app/gstappsink.h>
77 #include <gst/sdp/gstmikey.h>
78 #include <gst/rtp/gstrtppayloads.h>
80 #define AES_128_KEY_LEN 16
81 #define AES_256_KEY_LEN 32
83 #define HMAC_32_KEY_LEN 4
84 #define HMAC_80_KEY_LEN 10
86 #include "rtsp-media.h"
88 struct _GstRTSPMediaPrivate
93 /* protected by lock */
94 GstRTSPPermissions *permissions;
96 gboolean suspend_mode;
98 GstRTSPProfile profiles;
99 GstRTSPLowerTrans protocols;
101 gboolean eos_shutdown;
103 GstRTSPAddressPool *pool;
104 gchar *multicast_iface;
106 gboolean bind_mcast_address;
108 GstRTSPTransportMode transport_mode;
109 gboolean stop_on_disconnect;
112 GRecMutex state_lock; /* locking order: state lock, lock */
113 GPtrArray *streams; /* protected by lock */
114 GList *dynamic; /* protected by lock */
115 GstRTSPMediaStatus status; /* protected by lock */
119 gboolean finishing_unprepare;
121 /* the pipeline for the media */
122 GstElement *pipeline;
125 GstRTSPThread *thread;
126 GList *pending_pipeline_elements;
128 gboolean time_provider;
129 GstNetTimeProvider *nettime;
132 GstClockTimeDiff seekable;
134 GstState target_state;
136 /* RTP session manager */
139 /* the range of media */
140 GstRTSPTimeRange range; /* protected by lock */
141 GstClockTime range_start;
142 GstClockTime range_stop;
144 GList *payloads; /* protected by lock */
145 GstClockTime rtx_time; /* protected by lock */
146 gboolean do_retransmission; /* protected by lock */
147 guint latency; /* protected by lock */
148 GstClock *clock; /* protected by lock */
149 gboolean do_rate_control; /* protected by lock */
150 GstRTSPPublishClockMode publish_clock_mode;
152 /* Dynamic element handling */
153 guint nb_dynamic_elements;
154 guint no_more_pads_pending;
157 #define DEFAULT_SHARED FALSE
158 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
159 #define DEFAULT_REUSABLE FALSE
160 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
161 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
162 GST_RTSP_LOWER_TRANS_TCP
163 #define DEFAULT_EOS_SHUTDOWN FALSE
164 #define DEFAULT_BUFFER_SIZE 0x80000
165 #define DEFAULT_TIME_PROVIDER FALSE
166 #define DEFAULT_LATENCY 200
167 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
168 #define DEFAULT_STOP_ON_DISCONNECT TRUE
169 #define DEFAULT_MAX_MCAST_TTL 255
170 #define DEFAULT_BIND_MCAST_ADDRESS FALSE
171 #define DEFAULT_DO_RATE_CONTROL TRUE
173 #define DEFAULT_DO_RETRANSMISSION FALSE
175 /* define to dump received RTCP packets */
192 PROP_STOP_ON_DISCONNECT,
195 PROP_BIND_MCAST_ADDRESS,
202 SIGNAL_REMOVED_STREAM,
210 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
211 #define GST_CAT_DEFAULT rtsp_media_debug
213 static void gst_rtsp_media_get_property (GObject * object, guint propid,
214 GValue * value, GParamSpec * pspec);
215 static void gst_rtsp_media_set_property (GObject * object, guint propid,
216 const GValue * value, GParamSpec * pspec);
217 static void gst_rtsp_media_finalize (GObject * obj);
219 static gboolean default_handle_message (GstRTSPMedia * media,
220 GstMessage * message);
221 static void finish_unprepare (GstRTSPMedia * media);
222 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
223 static gboolean default_unprepare (GstRTSPMedia * media);
224 static gboolean default_suspend (GstRTSPMedia * media);
225 static gboolean default_unsuspend (GstRTSPMedia * media);
226 static gboolean default_convert_range (GstRTSPMedia * media,
227 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
228 static gboolean default_query_position (GstRTSPMedia * media,
230 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
231 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
232 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
234 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
236 static gboolean wait_preroll (GstRTSPMedia * media);
238 static GstElement *find_payload_element (GstElement * payloader);
240 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
242 static gboolean check_complete (GstRTSPMedia * media);
244 #define C_ENUM(v) ((gint) v)
247 gst_rtsp_suspend_mode_get_type (void)
250 static const GEnumValue values[] = {
251 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
252 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
254 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
259 if (g_once_init_enter (&id)) {
260 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
261 g_once_init_leave (&id, tmp);
266 #define C_FLAGS(v) ((guint) v)
269 gst_rtsp_transport_mode_get_type (void)
272 static const GFlagsValue values[] = {
273 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
275 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
280 if (g_once_init_enter (&id)) {
281 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
282 g_once_init_leave (&id, tmp);
288 gst_rtsp_publish_clock_mode_get_type (void)
291 static const GEnumValue values[] = {
292 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
293 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
294 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
295 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
297 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
298 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
303 if (g_once_init_enter (&id)) {
304 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
305 g_once_init_leave (&id, tmp);
310 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
313 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
315 GObjectClass *gobject_class;
317 gobject_class = G_OBJECT_CLASS (klass);
319 gobject_class->get_property = gst_rtsp_media_get_property;
320 gobject_class->set_property = gst_rtsp_media_set_property;
321 gobject_class->finalize = gst_rtsp_media_finalize;
323 g_object_class_install_property (gobject_class, PROP_SHARED,
324 g_param_spec_boolean ("shared", "Shared",
325 "If this media pipeline can be shared", DEFAULT_SHARED,
326 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
328 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
329 g_param_spec_enum ("suspend-mode", "Suspend Mode",
330 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
331 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_REUSABLE,
334 g_param_spec_boolean ("reusable", "Reusable",
335 "If this media pipeline can be reused after an unprepare",
336 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_PROFILES,
339 g_param_spec_flags ("profiles", "Profiles",
340 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
341 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
344 g_param_spec_flags ("protocols", "Protocols",
345 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
346 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
348 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
349 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
350 "Send an EOS event to the pipeline before unpreparing",
351 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
353 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
354 g_param_spec_uint ("buffer-size", "Buffer Size",
355 "The kernel UDP buffer size to use", 0, G_MAXUINT,
356 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
358 g_object_class_install_property (gobject_class, PROP_ELEMENT,
359 g_param_spec_object ("element", "The Element",
360 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
361 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
363 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
364 g_param_spec_boolean ("time-provider", "Time Provider",
365 "Use a NetTimeProvider for clients",
366 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
368 g_object_class_install_property (gobject_class, PROP_LATENCY,
369 g_param_spec_uint ("latency", "Latency",
370 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
371 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
373 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
374 g_param_spec_flags ("transport-mode", "Transport Mode",
375 "If this media pipeline can be used for PLAY or RECORD",
376 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
377 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
379 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
380 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
381 "If this media pipeline should be stopped "
382 "when a client disconnects without TEARDOWN",
383 DEFAULT_STOP_ON_DISCONNECT,
384 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
386 g_object_class_install_property (gobject_class, PROP_CLOCK,
387 g_param_spec_object ("clock", "Clock",
388 "Clock to be used by the media pipeline",
389 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
391 g_object_class_install_property (gobject_class, PROP_MAX_MCAST_TTL,
392 g_param_spec_uint ("max-mcast-ttl", "Maximum multicast ttl",
393 "The maximum time-to-live value of outgoing multicast packets", 1,
394 255, DEFAULT_MAX_MCAST_TTL,
395 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
397 g_object_class_install_property (gobject_class, PROP_BIND_MCAST_ADDRESS,
398 g_param_spec_boolean ("bind-mcast-address", "Bind mcast address",
399 "Whether the multicast sockets should be bound to multicast addresses "
401 DEFAULT_BIND_MCAST_ADDRESS,
402 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
404 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
405 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
406 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
407 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
409 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
410 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
411 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
412 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
413 GST_TYPE_RTSP_STREAM);
415 gst_rtsp_media_signals[SIGNAL_PREPARED] =
416 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
417 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
418 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
420 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
421 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
422 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
423 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
425 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
426 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
427 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
428 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
430 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
431 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
432 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
433 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
435 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
437 klass->handle_message = default_handle_message;
438 klass->prepare = default_prepare;
439 klass->unprepare = default_unprepare;
440 klass->suspend = default_suspend;
441 klass->unsuspend = default_unsuspend;
442 klass->convert_range = default_convert_range;
443 klass->query_position = default_query_position;
444 klass->query_stop = default_query_stop;
445 klass->create_rtpbin = default_create_rtpbin;
446 klass->setup_sdp = default_setup_sdp;
447 klass->handle_sdp = default_handle_sdp;
451 gst_rtsp_media_init (GstRTSPMedia * media)
453 GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_private (media);
457 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
458 g_mutex_init (&priv->lock);
459 g_cond_init (&priv->cond);
460 g_rec_mutex_init (&priv->state_lock);
462 priv->shared = DEFAULT_SHARED;
463 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
464 priv->reusable = DEFAULT_REUSABLE;
465 priv->profiles = DEFAULT_PROFILES;
466 priv->protocols = DEFAULT_PROTOCOLS;
467 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
468 priv->buffer_size = DEFAULT_BUFFER_SIZE;
469 priv->time_provider = DEFAULT_TIME_PROVIDER;
470 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
471 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
472 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
473 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
474 priv->max_mcast_ttl = DEFAULT_MAX_MCAST_TTL;
475 priv->bind_mcast_address = DEFAULT_BIND_MCAST_ADDRESS;
476 priv->do_rate_control = DEFAULT_DO_RATE_CONTROL;
480 gst_rtsp_media_finalize (GObject * obj)
482 GstRTSPMediaPrivate *priv;
485 media = GST_RTSP_MEDIA (obj);
488 GST_INFO ("finalize media %p", media);
490 if (priv->permissions)
491 gst_rtsp_permissions_unref (priv->permissions);
493 g_ptr_array_unref (priv->streams);
495 g_list_free_full (priv->dynamic, gst_object_unref);
496 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
499 gst_object_unref (priv->pipeline);
501 gst_object_unref (priv->nettime);
502 gst_object_unref (priv->element);
504 g_object_unref (priv->pool);
506 g_list_free (priv->payloads);
508 gst_object_unref (priv->clock);
509 g_free (priv->multicast_iface);
510 g_mutex_clear (&priv->lock);
511 g_cond_clear (&priv->cond);
512 g_rec_mutex_clear (&priv->state_lock);
514 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
518 gst_rtsp_media_get_property (GObject * object, guint propid,
519 GValue * value, GParamSpec * pspec)
521 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
525 g_value_set_object (value, media->priv->element);
528 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
530 case PROP_SUSPEND_MODE:
531 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
534 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
537 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
540 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
542 case PROP_EOS_SHUTDOWN:
543 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
545 case PROP_BUFFER_SIZE:
546 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
548 case PROP_TIME_PROVIDER:
549 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
552 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
554 case PROP_TRANSPORT_MODE:
555 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
557 case PROP_STOP_ON_DISCONNECT:
558 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
561 g_value_take_object (value, gst_rtsp_media_get_clock (media));
563 case PROP_MAX_MCAST_TTL:
564 g_value_set_uint (value, gst_rtsp_media_get_max_mcast_ttl (media));
566 case PROP_BIND_MCAST_ADDRESS:
567 g_value_set_boolean (value, gst_rtsp_media_is_bind_mcast_address (media));
570 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
575 gst_rtsp_media_set_property (GObject * object, guint propid,
576 const GValue * value, GParamSpec * pspec)
578 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
582 media->priv->element = g_value_get_object (value);
583 gst_object_ref_sink (media->priv->element);
586 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
588 case PROP_SUSPEND_MODE:
589 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
592 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
595 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
598 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
600 case PROP_EOS_SHUTDOWN:
601 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
603 case PROP_BUFFER_SIZE:
604 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
606 case PROP_TIME_PROVIDER:
607 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
610 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
612 case PROP_TRANSPORT_MODE:
613 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
615 case PROP_STOP_ON_DISCONNECT:
616 gst_rtsp_media_set_stop_on_disconnect (media,
617 g_value_get_boolean (value));
620 gst_rtsp_media_set_clock (media, g_value_get_object (value));
622 case PROP_MAX_MCAST_TTL:
623 gst_rtsp_media_set_max_mcast_ttl (media, g_value_get_uint (value));
625 case PROP_BIND_MCAST_ADDRESS:
626 gst_rtsp_media_set_bind_mcast_address (media,
627 g_value_get_boolean (value));
630 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
637 gboolean complete_streams_only;
639 } DoQueryPositionData;
642 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
646 if (!gst_rtsp_stream_is_sender (stream))
649 if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
650 GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
654 if (gst_rtsp_stream_query_position (stream, &tmp)) {
655 data->position = MIN (data->position, tmp);
659 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
660 GST_TIME_ARGS (data->position));
664 default_query_position (GstRTSPMedia * media, gint64 * position)
666 GstRTSPMediaPrivate *priv;
667 DoQueryPositionData data;
671 data.position = G_MAXINT64;
674 /* if the media is complete, i.e. one or more streams have been configured
675 * with sinks, then we want to query the position on those streams only.
676 * a query on an incmplete stream may return a position that originates from
677 * an earlier preroll */
678 if (check_complete (media))
679 data.complete_streams_only = TRUE;
681 data.complete_streams_only = FALSE;
683 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
686 *position = GST_CLOCK_TIME_NONE;
688 *position = data.position;
700 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
704 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
705 data->stop = MAX (data->stop, tmp);
711 default_query_stop (GstRTSPMedia * media, gint64 * stop)
713 GstRTSPMediaPrivate *priv;
714 DoQueryStopData data;
721 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
729 default_create_rtpbin (GstRTSPMedia * media)
733 rtpbin = gst_element_factory_make ("rtpbin", NULL);
738 /* must be called with state lock */
740 check_seekable (GstRTSPMedia * media)
743 GstRTSPMediaPrivate *priv = media->priv;
745 /* Update the seekable state of the pipeline in case it changed */
746 if (gst_rtsp_media_is_receive_only (media)) {
747 /* TODO: Seeking for "receive-only"? */
750 guint i, n = priv->streams->len;
752 for (i = 0; i < n; i++) {
753 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
755 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
756 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
763 query = gst_query_new_seeking (GST_FORMAT_TIME);
764 if (gst_element_query (priv->pipeline, query)) {
769 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
770 priv->seekable = seekable ? G_MAXINT64 : 0;
771 } else if (priv->streams->len) {
772 gboolean seekable = TRUE;
773 guint i, n = priv->streams->len;
775 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
776 for (i = 0; i < n; i++) {
777 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
778 seekable &= gst_rtsp_stream_seekable (stream);
780 priv->seekable = seekable ? G_MAXINT64 : -1;
783 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
785 gst_query_unref (query);
788 /* must be called with state lock */
790 check_complete (GstRTSPMedia * media)
792 GstRTSPMediaPrivate *priv = media->priv;
794 guint i, n = priv->streams->len;
796 for (i = 0; i < n; i++) {
797 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
799 if (gst_rtsp_stream_is_complete (stream))
806 /* must be called with state lock */
808 collect_media_stats (GstRTSPMedia * media)
810 GstRTSPMediaPrivate *priv = media->priv;
811 gint64 position = 0, stop = -1;
813 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
814 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
817 priv->range.unit = GST_RTSP_RANGE_NPT;
819 GST_INFO ("collect media stats");
822 priv->range.min.type = GST_RTSP_TIME_NOW;
823 priv->range.min.seconds = -1;
824 priv->range_start = -1;
825 priv->range.max.type = GST_RTSP_TIME_END;
826 priv->range.max.seconds = -1;
827 priv->range_stop = -1;
829 GstRTSPMediaClass *klass;
832 klass = GST_RTSP_MEDIA_GET_CLASS (media);
834 /* get the position */
836 if (klass->query_position)
837 ret = klass->query_position (media, &position);
840 GST_INFO ("position query failed");
844 /* get the current segment stop */
846 if (klass->query_stop)
847 ret = klass->query_stop (media, &stop);
850 GST_INFO ("stop query failed");
854 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
855 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
857 if (position == -1) {
858 priv->range.min.type = GST_RTSP_TIME_NOW;
859 priv->range.min.seconds = -1;
860 priv->range_start = -1;
862 priv->range.min.type = GST_RTSP_TIME_SECONDS;
863 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
864 priv->range_start = position;
867 priv->range.max.type = GST_RTSP_TIME_END;
868 priv->range.max.seconds = -1;
869 priv->range_stop = -1;
871 priv->range.max.type = GST_RTSP_TIME_SECONDS;
872 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
873 priv->range_stop = stop;
876 check_seekable (media);
881 * gst_rtsp_media_new:
882 * @element: (transfer full): a #GstElement
884 * Create a new #GstRTSPMedia instance. @element is the bin element that
885 * provides the different streams. The #GstRTSPMedia object contains the
886 * element to produce RTP data for one or more related (audio/video/..)
889 * Ownership is taken of @element.
891 * Returns: (transfer full): a new #GstRTSPMedia object.
894 gst_rtsp_media_new (GstElement * element)
896 GstRTSPMedia *result;
898 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
900 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
906 * gst_rtsp_media_get_element:
907 * @media: a #GstRTSPMedia
909 * Get the element that was used when constructing @media.
911 * Returns: (transfer full): a #GstElement. Unref after usage.
914 gst_rtsp_media_get_element (GstRTSPMedia * media)
916 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
918 return gst_object_ref (media->priv->element);
922 * gst_rtsp_media_take_pipeline:
923 * @media: a #GstRTSPMedia
924 * @pipeline: (transfer full): a #GstPipeline
926 * Set @pipeline as the #GstPipeline for @media. Ownership is
927 * taken of @pipeline.
930 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
932 GstRTSPMediaPrivate *priv;
934 GstNetTimeProvider *nettime;
937 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
938 g_return_if_fail (GST_IS_PIPELINE (pipeline));
942 g_mutex_lock (&priv->lock);
943 old = priv->pipeline;
944 priv->pipeline = GST_ELEMENT_CAST (pipeline);
945 nettime = priv->nettime;
946 priv->nettime = NULL;
947 g_mutex_unlock (&priv->lock);
950 gst_object_unref (old);
953 gst_object_unref (nettime);
955 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
957 for (l = priv->pending_pipeline_elements; l; l = l->next) {
958 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
960 g_list_free (priv->pending_pipeline_elements);
961 priv->pending_pipeline_elements = NULL;
965 * gst_rtsp_media_set_permissions:
966 * @media: a #GstRTSPMedia
967 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
969 * Set @permissions on @media.
972 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
973 GstRTSPPermissions * permissions)
975 GstRTSPMediaPrivate *priv;
977 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
981 g_mutex_lock (&priv->lock);
982 if (priv->permissions)
983 gst_rtsp_permissions_unref (priv->permissions);
984 if ((priv->permissions = permissions))
985 gst_rtsp_permissions_ref (permissions);
986 g_mutex_unlock (&priv->lock);
990 * gst_rtsp_media_get_permissions:
991 * @media: a #GstRTSPMedia
993 * Get the permissions object from @media.
995 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
998 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
1000 GstRTSPMediaPrivate *priv;
1001 GstRTSPPermissions *result;
1003 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1007 g_mutex_lock (&priv->lock);
1008 if ((result = priv->permissions))
1009 gst_rtsp_permissions_ref (result);
1010 g_mutex_unlock (&priv->lock);
1016 * gst_rtsp_media_set_suspend_mode:
1017 * @media: a #GstRTSPMedia
1018 * @mode: the new #GstRTSPSuspendMode
1020 * Control how @ media will be suspended after the SDP has been generated and
1021 * after a PAUSE request has been performed.
1023 * Media must be unprepared when setting the suspend mode.
1026 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
1028 GstRTSPMediaPrivate *priv;
1030 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1034 g_rec_mutex_lock (&priv->state_lock);
1035 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1037 priv->suspend_mode = mode;
1038 g_rec_mutex_unlock (&priv->state_lock);
1045 GST_WARNING ("media %p was prepared", media);
1046 g_rec_mutex_unlock (&priv->state_lock);
1051 * gst_rtsp_media_get_suspend_mode:
1052 * @media: a #GstRTSPMedia
1054 * Get how @media will be suspended.
1056 * Returns: #GstRTSPSuspendMode.
1059 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1061 GstRTSPMediaPrivate *priv;
1062 GstRTSPSuspendMode res;
1064 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1068 g_rec_mutex_lock (&priv->state_lock);
1069 res = priv->suspend_mode;
1070 g_rec_mutex_unlock (&priv->state_lock);
1076 * gst_rtsp_media_set_shared:
1077 * @media: a #GstRTSPMedia
1078 * @shared: the new value
1080 * Set or unset if the pipeline for @media can be shared will multiple clients.
1081 * When @shared is %TRUE, client requests for this media will share the media
1085 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1087 GstRTSPMediaPrivate *priv;
1089 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1093 g_mutex_lock (&priv->lock);
1094 priv->shared = shared;
1095 g_mutex_unlock (&priv->lock);
1099 * gst_rtsp_media_is_shared:
1100 * @media: a #GstRTSPMedia
1102 * Check if the pipeline for @media can be shared between multiple clients.
1104 * Returns: %TRUE if the media can be shared between clients.
1107 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1109 GstRTSPMediaPrivate *priv;
1112 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1116 g_mutex_lock (&priv->lock);
1118 g_mutex_unlock (&priv->lock);
1124 * gst_rtsp_media_set_reusable:
1125 * @media: a #GstRTSPMedia
1126 * @reusable: the new value
1128 * Set or unset if the pipeline for @media can be reused after the pipeline has
1132 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1134 GstRTSPMediaPrivate *priv;
1136 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1140 g_mutex_lock (&priv->lock);
1141 priv->reusable = reusable;
1142 g_mutex_unlock (&priv->lock);
1146 * gst_rtsp_media_is_reusable:
1147 * @media: a #GstRTSPMedia
1149 * Check if the pipeline for @media can be reused after an unprepare.
1151 * Returns: %TRUE if the media can be reused
1154 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1156 GstRTSPMediaPrivate *priv;
1159 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1163 g_mutex_lock (&priv->lock);
1164 res = priv->reusable;
1165 g_mutex_unlock (&priv->lock);
1171 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1173 gst_rtsp_stream_set_profiles (stream, *profiles);
1177 * gst_rtsp_media_set_profiles:
1178 * @media: a #GstRTSPMedia
1179 * @profiles: the new flags
1181 * Configure the allowed lower transport for @media.
1184 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1186 GstRTSPMediaPrivate *priv;
1188 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1192 g_mutex_lock (&priv->lock);
1193 priv->profiles = profiles;
1194 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1195 g_mutex_unlock (&priv->lock);
1199 * gst_rtsp_media_get_profiles:
1200 * @media: a #GstRTSPMedia
1202 * Get the allowed profiles of @media.
1204 * Returns: a #GstRTSPProfile
1207 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1209 GstRTSPMediaPrivate *priv;
1212 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1216 g_mutex_lock (&priv->lock);
1217 res = priv->profiles;
1218 g_mutex_unlock (&priv->lock);
1224 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1226 gst_rtsp_stream_set_protocols (stream, *protocols);
1230 * gst_rtsp_media_set_protocols:
1231 * @media: a #GstRTSPMedia
1232 * @protocols: the new flags
1234 * Configure the allowed lower transport for @media.
1237 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1239 GstRTSPMediaPrivate *priv;
1241 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1245 g_mutex_lock (&priv->lock);
1246 priv->protocols = protocols;
1247 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1248 g_mutex_unlock (&priv->lock);
1252 * gst_rtsp_media_get_protocols:
1253 * @media: a #GstRTSPMedia
1255 * Get the allowed protocols of @media.
1257 * Returns: a #GstRTSPLowerTrans
1260 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1262 GstRTSPMediaPrivate *priv;
1263 GstRTSPLowerTrans res;
1265 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1266 GST_RTSP_LOWER_TRANS_UNKNOWN);
1270 g_mutex_lock (&priv->lock);
1271 res = priv->protocols;
1272 g_mutex_unlock (&priv->lock);
1278 * gst_rtsp_media_set_eos_shutdown:
1279 * @media: a #GstRTSPMedia
1280 * @eos_shutdown: the new value
1282 * Set or unset if an EOS event will be sent to the pipeline for @media before
1286 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1288 GstRTSPMediaPrivate *priv;
1290 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1294 g_mutex_lock (&priv->lock);
1295 priv->eos_shutdown = eos_shutdown;
1296 g_mutex_unlock (&priv->lock);
1300 * gst_rtsp_media_is_eos_shutdown:
1301 * @media: a #GstRTSPMedia
1303 * Check if the pipeline for @media will send an EOS down the pipeline before
1306 * Returns: %TRUE if the media will send EOS before unpreparing.
1309 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1311 GstRTSPMediaPrivate *priv;
1314 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1318 g_mutex_lock (&priv->lock);
1319 res = priv->eos_shutdown;
1320 g_mutex_unlock (&priv->lock);
1326 * gst_rtsp_media_set_buffer_size:
1327 * @media: a #GstRTSPMedia
1328 * @size: the new value
1330 * Set the kernel UDP buffer size.
1333 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1335 GstRTSPMediaPrivate *priv;
1338 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1340 GST_LOG_OBJECT (media, "set buffer size %u", size);
1344 g_mutex_lock (&priv->lock);
1345 priv->buffer_size = size;
1347 for (i = 0; i < priv->streams->len; i++) {
1348 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1349 gst_rtsp_stream_set_buffer_size (stream, size);
1351 g_mutex_unlock (&priv->lock);
1355 * gst_rtsp_media_get_buffer_size:
1356 * @media: a #GstRTSPMedia
1358 * Get the kernel UDP buffer size.
1360 * Returns: the kernel UDP buffer size.
1363 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1365 GstRTSPMediaPrivate *priv;
1368 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1372 g_mutex_lock (&priv->lock);
1373 res = priv->buffer_size;
1374 g_mutex_unlock (&priv->lock);
1380 * gst_rtsp_media_set_stop_on_disconnect:
1381 * @media: a #GstRTSPMedia
1382 * @stop_on_disconnect: the new value
1384 * Set or unset if the pipeline for @media should be stopped when a
1385 * client disconnects without sending TEARDOWN.
1388 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1389 gboolean stop_on_disconnect)
1391 GstRTSPMediaPrivate *priv;
1393 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1397 g_mutex_lock (&priv->lock);
1398 priv->stop_on_disconnect = stop_on_disconnect;
1399 g_mutex_unlock (&priv->lock);
1403 * gst_rtsp_media_is_stop_on_disconnect:
1404 * @media: a #GstRTSPMedia
1406 * Check if the pipeline for @media will be stopped when a client disconnects
1407 * without sending TEARDOWN.
1409 * Returns: %TRUE if the media will be stopped when a client disconnects
1410 * without sending TEARDOWN.
1413 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1415 GstRTSPMediaPrivate *priv;
1418 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1422 g_mutex_lock (&priv->lock);
1423 res = priv->stop_on_disconnect;
1424 g_mutex_unlock (&priv->lock);
1430 * gst_rtsp_media_set_retransmission_time:
1431 * @media: a #GstRTSPMedia
1432 * @time: the new value
1434 * Set the amount of time to store retransmission packets.
1437 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1439 GstRTSPMediaPrivate *priv;
1442 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1444 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1448 g_mutex_lock (&priv->lock);
1449 priv->rtx_time = time;
1450 for (i = 0; i < priv->streams->len; i++) {
1451 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1453 gst_rtsp_stream_set_retransmission_time (stream, time);
1455 g_mutex_unlock (&priv->lock);
1459 * gst_rtsp_media_get_retransmission_time:
1460 * @media: a #GstRTSPMedia
1462 * Get the amount of time to store retransmission data.
1464 * Returns: the amount of time to store retransmission data.
1467 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1469 GstRTSPMediaPrivate *priv;
1472 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1476 g_mutex_lock (&priv->lock);
1477 res = priv->rtx_time;
1478 g_mutex_unlock (&priv->lock);
1484 * gst_rtsp_media_set_do_retransmission:
1486 * Set whether retransmission requests will be sent
1491 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
1492 gboolean do_retransmission)
1494 GstRTSPMediaPrivate *priv;
1496 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1500 g_mutex_lock (&priv->lock);
1501 priv->do_retransmission = do_retransmission;
1504 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1505 g_mutex_unlock (&priv->lock);
1509 * gst_rtsp_media_get_do_retransmission:
1511 * Returns: Whether retransmission requests will be sent
1516 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1518 GstRTSPMediaPrivate *priv;
1521 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1525 g_mutex_lock (&priv->lock);
1526 res = priv->do_retransmission;
1527 g_mutex_unlock (&priv->lock);
1533 * gst_rtsp_media_set_latency:
1534 * @media: a #GstRTSPMedia
1535 * @latency: latency in milliseconds
1537 * Configure the latency used for receiving media.
1540 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1542 GstRTSPMediaPrivate *priv;
1545 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1547 GST_LOG_OBJECT (media, "set latency %ums", latency);
1551 g_mutex_lock (&priv->lock);
1552 priv->latency = latency;
1554 g_object_set (priv->rtpbin, "latency", latency, NULL);
1556 for (i = 0; i < media->priv->streams->len; i++) {
1557 GObject *storage = NULL;
1559 g_signal_emit_by_name (G_OBJECT (media->priv->rtpbin), "get-storage",
1562 g_object_set (storage, "size-time",
1563 (media->priv->latency + 50) * GST_MSECOND, NULL);
1567 g_mutex_unlock (&priv->lock);
1571 * gst_rtsp_media_get_latency:
1572 * @media: a #GstRTSPMedia
1574 * Get the latency that is used for receiving media.
1576 * Returns: latency in milliseconds
1579 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1581 GstRTSPMediaPrivate *priv;
1584 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1588 g_mutex_lock (&priv->lock);
1589 res = priv->latency;
1590 g_mutex_unlock (&priv->lock);
1596 * gst_rtsp_media_use_time_provider:
1597 * @media: a #GstRTSPMedia
1598 * @time_provider: if a #GstNetTimeProvider should be used
1600 * Set @media to provide a #GstNetTimeProvider.
1603 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1605 GstRTSPMediaPrivate *priv;
1607 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1611 g_mutex_lock (&priv->lock);
1612 priv->time_provider = time_provider;
1613 g_mutex_unlock (&priv->lock);
1617 * gst_rtsp_media_is_time_provider:
1618 * @media: a #GstRTSPMedia
1620 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1622 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1624 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1627 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1629 GstRTSPMediaPrivate *priv;
1632 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1636 g_mutex_lock (&priv->lock);
1637 res = priv->time_provider;
1638 g_mutex_unlock (&priv->lock);
1644 * gst_rtsp_media_set_clock:
1645 * @media: a #GstRTSPMedia
1646 * @clock: (nullable): #GstClock to be used
1648 * Configure the clock used for the media.
1651 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1653 GstRTSPMediaPrivate *priv;
1655 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1656 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1658 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1662 g_mutex_lock (&priv->lock);
1664 gst_object_unref (priv->clock);
1665 priv->clock = clock ? gst_object_ref (clock) : NULL;
1666 if (priv->pipeline) {
1668 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1670 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1673 g_mutex_unlock (&priv->lock);
1677 * gst_rtsp_media_set_publish_clock_mode:
1678 * @media: a #GstRTSPMedia
1679 * @mode: the clock publish mode
1681 * Sets if and how the media clock should be published according to RFC7273.
1686 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1687 GstRTSPPublishClockMode mode)
1689 GstRTSPMediaPrivate *priv;
1693 g_mutex_lock (&priv->lock);
1694 priv->publish_clock_mode = mode;
1696 n = priv->streams->len;
1697 for (i = 0; i < n; i++) {
1698 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1700 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1702 g_mutex_unlock (&priv->lock);
1706 * gst_rtsp_media_get_publish_clock_mode:
1707 * @media: a #GstRTSPMedia
1709 * Gets if and how the media clock should be published according to RFC7273.
1711 * Returns: The GstRTSPPublishClockMode
1715 GstRTSPPublishClockMode
1716 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1718 GstRTSPMediaPrivate *priv;
1719 GstRTSPPublishClockMode ret;
1722 g_mutex_lock (&priv->lock);
1723 ret = priv->publish_clock_mode;
1724 g_mutex_unlock (&priv->lock);
1730 * gst_rtsp_media_set_address_pool:
1731 * @media: a #GstRTSPMedia
1732 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1734 * configure @pool to be used as the address pool of @media.
1737 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1738 GstRTSPAddressPool * pool)
1740 GstRTSPMediaPrivate *priv;
1741 GstRTSPAddressPool *old;
1743 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1747 GST_LOG_OBJECT (media, "set address pool %p", pool);
1749 g_mutex_lock (&priv->lock);
1750 if ((old = priv->pool) != pool)
1751 priv->pool = pool ? g_object_ref (pool) : NULL;
1754 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1756 g_mutex_unlock (&priv->lock);
1759 g_object_unref (old);
1763 * gst_rtsp_media_get_address_pool:
1764 * @media: a #GstRTSPMedia
1766 * Get the #GstRTSPAddressPool used as the address pool of @media.
1768 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1769 * g_object_unref() after usage.
1771 GstRTSPAddressPool *
1772 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1774 GstRTSPMediaPrivate *priv;
1775 GstRTSPAddressPool *result;
1777 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1781 g_mutex_lock (&priv->lock);
1782 if ((result = priv->pool))
1783 g_object_ref (result);
1784 g_mutex_unlock (&priv->lock);
1790 * gst_rtsp_media_set_multicast_iface:
1791 * @media: a #GstRTSPMedia
1792 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1794 * configure @multicast_iface to be used for @media.
1797 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1798 const gchar * multicast_iface)
1800 GstRTSPMediaPrivate *priv;
1803 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1807 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1809 g_mutex_lock (&priv->lock);
1810 if ((old = priv->multicast_iface) != multicast_iface)
1811 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1814 g_ptr_array_foreach (priv->streams,
1815 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1816 g_mutex_unlock (&priv->lock);
1823 * gst_rtsp_media_get_multicast_iface:
1824 * @media: a #GstRTSPMedia
1826 * Get the multicast interface used for @media.
1828 * Returns: (transfer full) (nullable): the multicast interface for @media.
1829 * g_free() after usage.
1832 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1834 GstRTSPMediaPrivate *priv;
1837 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1841 g_mutex_lock (&priv->lock);
1842 if ((result = priv->multicast_iface))
1843 result = g_strdup (result);
1844 g_mutex_unlock (&priv->lock);
1850 * gst_rtsp_media_set_max_mcast_ttl:
1851 * @media: a #GstRTSPMedia
1852 * @ttl: the new multicast ttl value
1854 * Set the maximum time-to-live value of outgoing multicast packets.
1856 * Returns: %TRUE if the requested ttl has been set successfully.
1861 gst_rtsp_media_set_max_mcast_ttl (GstRTSPMedia * media, guint ttl)
1863 GstRTSPMediaPrivate *priv;
1866 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1868 GST_LOG_OBJECT (media, "set max mcast ttl %u", ttl);
1872 g_mutex_lock (&priv->lock);
1874 if (ttl == 0 || ttl > DEFAULT_MAX_MCAST_TTL) {
1875 GST_WARNING_OBJECT (media, "The reqested mcast TTL value is not valid.");
1876 g_mutex_unlock (&priv->lock);
1879 priv->max_mcast_ttl = ttl;
1881 for (i = 0; i < priv->streams->len; i++) {
1882 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1883 gst_rtsp_stream_set_max_mcast_ttl (stream, ttl);
1885 g_mutex_unlock (&priv->lock);
1891 * gst_rtsp_media_get_max_mcast_ttl:
1892 * @media: a #GstRTSPMedia
1894 * Get the the maximum time-to-live value of outgoing multicast packets.
1896 * Returns: the maximum time-to-live value of outgoing multicast packets.
1901 gst_rtsp_media_get_max_mcast_ttl (GstRTSPMedia * media)
1903 GstRTSPMediaPrivate *priv;
1906 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1910 g_mutex_lock (&priv->lock);
1911 res = priv->max_mcast_ttl;
1912 g_mutex_unlock (&priv->lock);
1918 * gst_rtsp_media_set_bind_mcast_address:
1919 * @media: a #GstRTSPMedia
1920 * @bind_mcast_addr: the new value
1922 * Decide whether the multicast socket should be bound to a multicast address or
1928 gst_rtsp_media_set_bind_mcast_address (GstRTSPMedia * media,
1929 gboolean bind_mcast_addr)
1931 GstRTSPMediaPrivate *priv;
1934 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1938 g_mutex_lock (&priv->lock);
1939 priv->bind_mcast_address = bind_mcast_addr;
1940 for (i = 0; i < priv->streams->len; i++) {
1941 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1942 gst_rtsp_stream_set_bind_mcast_address (stream, bind_mcast_addr);
1944 g_mutex_unlock (&priv->lock);
1948 * gst_rtsp_media_is_bind_mcast_address:
1949 * @media: a #GstRTSPMedia
1951 * Check if multicast sockets are configured to be bound to multicast addresses.
1953 * Returns: %TRUE if multicast sockets are configured to be bound to multicast addresses.
1958 gst_rtsp_media_is_bind_mcast_address (GstRTSPMedia * media)
1960 GstRTSPMediaPrivate *priv;
1963 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1967 g_mutex_lock (&priv->lock);
1968 result = priv->bind_mcast_address;
1969 g_mutex_unlock (&priv->lock);
1975 _find_payload_types (GstRTSPMedia * media)
1978 GQueue queue = G_QUEUE_INIT;
1980 n = media->priv->streams->len;
1981 for (i = 0; i < n; i++) {
1982 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1983 guint pt = gst_rtsp_stream_get_pt (stream);
1985 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1992 _next_available_pt (GList * payloads)
1996 for (i = 96; i <= 127; i++) {
1997 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1999 return GPOINTER_TO_UINT (i);
2006 * gst_rtsp_media_collect_streams:
2007 * @media: a #GstRTSPMedia
2009 * Find all payloader elements, they should be named pay\%d in the
2010 * element of @media, and create #GstRTSPStreams for them.
2012 * Collect all dynamic elements, named dynpay\%d, and add them to
2013 * the list of dynamic elements.
2015 * Find all depayloader elements, they should be named depay\%d in the
2016 * element of @media, and create #GstRTSPStreams for them.
2019 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
2021 GstRTSPMediaPrivate *priv;
2022 GstElement *element, *elem;
2026 gboolean more_elem_remaining = TRUE;
2027 GstRTSPTransportMode mode = 0;
2029 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
2032 element = priv->element;
2035 for (i = 0; more_elem_remaining; i++) {
2038 more_elem_remaining = FALSE;
2040 name = g_strdup_printf ("pay%d", i);
2041 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2043 GST_INFO ("found stream %d with payloader %p", i, elem);
2045 /* take the pad of the payloader */
2046 pad = gst_element_get_static_pad (elem, "src");
2048 /* find the real payload element in case elem is a GstBin */
2049 pay = find_payload_element (elem);
2051 /* create the stream */
2053 GST_WARNING ("could not find real payloader, using bin");
2054 gst_rtsp_media_create_stream (media, elem, pad);
2056 gst_rtsp_media_create_stream (media, pay, pad);
2057 gst_object_unref (pay);
2060 gst_object_unref (pad);
2061 gst_object_unref (elem);
2064 more_elem_remaining = TRUE;
2065 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2069 name = g_strdup_printf ("dynpay%d", i);
2070 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2071 /* a stream that will dynamically create pads to provide RTP packets */
2072 GST_INFO ("found dynamic element %d, %p", i, elem);
2074 g_mutex_lock (&priv->lock);
2075 priv->dynamic = g_list_prepend (priv->dynamic, elem);
2076 g_mutex_unlock (&priv->lock);
2078 priv->nb_dynamic_elements++;
2081 more_elem_remaining = TRUE;
2082 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
2086 name = g_strdup_printf ("depay%d", i);
2087 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
2088 GST_INFO ("found stream %d with depayloader %p", i, elem);
2090 /* take the pad of the payloader */
2091 pad = gst_element_get_static_pad (elem, "sink");
2092 /* create the stream */
2093 gst_rtsp_media_create_stream (media, elem, pad);
2094 gst_object_unref (pad);
2095 gst_object_unref (elem);
2098 more_elem_remaining = TRUE;
2099 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
2105 if (priv->transport_mode != mode)
2106 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
2107 priv->transport_mode, mode);
2113 GstElement *appsink, *appsrc;
2114 GstRTSPStream *stream;
2117 static GstFlowReturn
2118 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
2120 AppSinkSrcData *data = user_data;
2124 sample = gst_app_sink_pull_sample (appsink);
2126 return GST_FLOW_FLUSHING;
2129 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
2130 gst_sample_unref (sample);
2134 static GstAppSinkCallbacks appsink_callbacks = {
2140 static GstPadProbeReturn
2141 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2143 AppSinkSrcData *data = user_data;
2145 if (GST_IS_EVENT (info->data)
2146 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
2147 GstClockTime min, max;
2149 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
2151 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
2152 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
2153 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
2155 } else if (GST_IS_QUERY (info->data)) {
2156 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
2157 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
2158 gst_object_unref (srcpad);
2159 return GST_PAD_PROBE_HANDLED;
2161 gst_object_unref (srcpad);
2164 return GST_PAD_PROBE_OK;
2167 static GstPadProbeReturn
2168 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2170 AppSinkSrcData *data = user_data;
2172 if (GST_IS_QUERY (info->data)) {
2173 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
2174 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
2175 gst_object_unref (sinkpad);
2176 return GST_PAD_PROBE_HANDLED;
2178 gst_object_unref (sinkpad);
2181 return GST_PAD_PROBE_OK;
2185 * gst_rtsp_media_create_stream:
2186 * @media: a #GstRTSPMedia
2187 * @payloader: a #GstElement
2190 * Create a new stream in @media that provides RTP data on @pad.
2191 * @pad should be a pad of an element inside @media->element.
2193 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2197 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2200 GstRTSPMediaPrivate *priv;
2201 GstRTSPStream *stream;
2205 AppSinkSrcData *data = NULL;
2207 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2208 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2209 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2213 g_mutex_lock (&priv->lock);
2214 idx = priv->streams->len;
2216 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
2218 if (GST_PAD_IS_SRC (pad))
2219 name = g_strdup_printf ("src_%u", idx);
2221 name = g_strdup_printf ("sink_%u", idx);
2223 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2224 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2225 GstElement *appsink, *appsrc;
2226 GstPad *sinkpad, *srcpad;
2228 appsink = gst_element_factory_make ("appsink", NULL);
2229 appsrc = gst_element_factory_make ("appsrc", NULL);
2231 if (GST_PAD_IS_SINK (pad)) {
2232 srcpad = gst_element_get_static_pad (appsrc, "src");
2234 gst_bin_add (GST_BIN (priv->element), appsrc);
2236 gst_pad_link (srcpad, pad);
2237 gst_object_unref (srcpad);
2239 streampad = gst_element_get_static_pad (appsink, "sink");
2241 priv->pending_pipeline_elements =
2242 g_list_prepend (priv->pending_pipeline_elements, appsink);
2244 sinkpad = gst_element_get_static_pad (appsink, "sink");
2246 gst_pad_link (pad, sinkpad);
2247 gst_object_unref (sinkpad);
2249 streampad = gst_element_get_static_pad (appsrc, "src");
2251 priv->pending_pipeline_elements =
2252 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2255 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2256 TRUE, "emit-signals", FALSE, NULL);
2257 g_object_set (appsink, "sync", FALSE, "async", FALSE, "emit-signals",
2258 FALSE, "buffer-list", TRUE, NULL);
2260 data = g_new0 (AppSinkSrcData, 1);
2261 data->appsink = appsink;
2262 data->appsrc = appsrc;
2264 sinkpad = gst_element_get_static_pad (appsink, "sink");
2265 gst_pad_add_probe (sinkpad,
2266 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2267 appsink_pad_probe, data, NULL);
2268 gst_object_unref (sinkpad);
2270 srcpad = gst_element_get_static_pad (appsrc, "src");
2271 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2272 appsrc_pad_probe, data, NULL);
2273 gst_object_unref (srcpad);
2275 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2277 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2280 streampad = gst_ghost_pad_new (name, pad);
2281 gst_pad_set_active (streampad, TRUE);
2282 gst_element_add_pad (priv->element, streampad);
2286 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2288 data->stream = stream;
2290 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2291 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2292 gst_rtsp_stream_set_max_mcast_ttl (stream, priv->max_mcast_ttl);
2293 gst_rtsp_stream_set_bind_mcast_address (stream, priv->bind_mcast_address);
2294 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2295 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2296 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2297 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2298 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2299 gst_rtsp_stream_set_rate_control (stream, priv->do_rate_control);
2301 g_ptr_array_add (priv->streams, stream);
2303 if (GST_PAD_IS_SRC (pad)) {
2307 g_list_free (priv->payloads);
2308 priv->payloads = _find_payload_types (media);
2310 n = priv->streams->len;
2311 for (i = 0; i < n; i++) {
2312 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2313 guint rtx_pt = _next_available_pt (priv->payloads);
2316 GST_WARNING ("Ran out of space of dynamic payload types");
2320 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2323 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2326 g_mutex_unlock (&priv->lock);
2328 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2335 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2337 GstRTSPMediaPrivate *priv;
2339 AppSinkSrcData *data;
2343 g_mutex_lock (&priv->lock);
2344 /* remove the ghostpad */
2345 srcpad = gst_rtsp_stream_get_srcpad (stream);
2346 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2348 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2349 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2350 else if (GST_OBJECT_PARENT (data->appsrc) ==
2351 GST_OBJECT_CAST (priv->element))
2352 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2353 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2354 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2355 else if (GST_OBJECT_PARENT (data->appsink) ==
2356 GST_OBJECT_CAST (priv->element))
2357 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2359 gst_element_remove_pad (priv->element, srcpad);
2361 gst_object_unref (srcpad);
2362 /* now remove the stream */
2363 g_object_ref (stream);
2364 g_ptr_array_remove (priv->streams, stream);
2365 g_mutex_unlock (&priv->lock);
2367 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2370 g_object_unref (stream);
2374 * gst_rtsp_media_n_streams:
2375 * @media: a #GstRTSPMedia
2377 * Get the number of streams in this media.
2379 * Returns: The number of streams.
2382 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2384 GstRTSPMediaPrivate *priv;
2387 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2391 g_mutex_lock (&priv->lock);
2392 res = priv->streams->len;
2393 g_mutex_unlock (&priv->lock);
2399 * gst_rtsp_media_get_stream:
2400 * @media: a #GstRTSPMedia
2401 * @idx: the stream index
2403 * Retrieve the stream with index @idx from @media.
2405 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2406 * @idx or %NULL when a stream with that index did not exist.
2409 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2411 GstRTSPMediaPrivate *priv;
2414 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2418 g_mutex_lock (&priv->lock);
2419 if (idx < priv->streams->len)
2420 res = g_ptr_array_index (priv->streams, idx);
2423 g_mutex_unlock (&priv->lock);
2429 * gst_rtsp_media_find_stream:
2430 * @media: a #GstRTSPMedia
2431 * @control: the control of the stream
2433 * Find a stream in @media with @control as the control uri.
2435 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2436 * control uri @control or %NULL when a stream with that control did
2440 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2442 GstRTSPMediaPrivate *priv;
2446 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2447 g_return_val_if_fail (control != NULL, NULL);
2453 g_mutex_lock (&priv->lock);
2454 for (i = 0; i < priv->streams->len; i++) {
2455 GstRTSPStream *test;
2457 test = g_ptr_array_index (priv->streams, i);
2458 if (gst_rtsp_stream_has_control (test, control)) {
2463 g_mutex_unlock (&priv->lock);
2468 /* called with state-lock */
2470 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2471 GstRTSPRangeUnit unit)
2473 return gst_rtsp_range_convert_units (range, unit);
2477 * gst_rtsp_media_get_range_string:
2478 * @media: a #GstRTSPMedia
2479 * @play: for the PLAY request
2480 * @unit: the unit to use for the string
2482 * Get the current range as a string. @media must be prepared with
2483 * gst_rtsp_media_prepare ().
2485 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2488 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2489 GstRTSPRangeUnit unit)
2491 GstRTSPMediaClass *klass;
2492 GstRTSPMediaPrivate *priv;
2494 GstRTSPTimeRange range;
2496 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2497 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2498 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2502 g_rec_mutex_lock (&priv->state_lock);
2503 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2504 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2507 g_mutex_lock (&priv->lock);
2509 /* Update the range value with current position/duration */
2510 collect_media_stats (media);
2513 range = priv->range;
2515 if (!play && priv->n_active > 0) {
2516 range.min.type = GST_RTSP_TIME_NOW;
2517 range.min.seconds = -1;
2519 g_mutex_unlock (&priv->lock);
2520 g_rec_mutex_unlock (&priv->state_lock);
2522 if (!klass->convert_range (media, &range, unit))
2523 goto conversion_failed;
2525 result = gst_rtsp_range_to_string (&range);
2532 GST_WARNING ("media %p was not prepared", media);
2533 g_rec_mutex_unlock (&priv->state_lock);
2538 GST_WARNING ("range conversion to unit %d failed", unit);
2544 * gst_rtsp_media_get_rates:
2545 * @media: a #GstRTSPMedia
2546 * @rate (allow-none): the rate of the current segment
2547 * @applied_rate (allow-none): the applied_rate of the current segment
2549 * Get the rate and applied_rate of the current segment.
2551 * Returns: %FALSE if looking up the rate and applied rate failed. Otherwise
2552 * %TRUE is returned and @rate and @applied_rate are set to the rate and
2553 * applied_rate of the current segment.
2557 gst_rtsp_media_get_rates (GstRTSPMedia * media, gdouble * rate,
2558 gdouble * applied_rate)
2560 GstRTSPMediaPrivate *priv;
2561 GstRTSPStream *stream;
2562 gboolean result = TRUE;
2564 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2566 if (!rate && !applied_rate) {
2567 GST_WARNING_OBJECT (media, "rate and applied_rate are both NULL");
2573 g_mutex_lock (&priv->lock);
2575 g_assert (priv->streams->len > 0);
2576 stream = g_ptr_array_index (priv->streams, 0);
2577 if (!gst_rtsp_stream_get_rates (stream, rate, applied_rate)) {
2578 GST_WARNING_OBJECT (media,
2579 "failed to obtain rate and applied_rate from first stream");
2583 g_mutex_unlock (&priv->lock);
2589 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2591 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2595 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2597 GstRTSPMediaPrivate *priv = media->priv;
2599 GST_DEBUG ("media %p set blocked %d", media, blocked);
2600 priv->blocked = blocked;
2601 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2605 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2607 gst_rtsp_stream_unblock_linked (stream);
2611 media_unblock_linked (GstRTSPMedia * media)
2613 GstRTSPMediaPrivate *priv = media->priv;
2615 GST_DEBUG ("media %p unblocking linked streams", media);
2616 /* media is not blocked any longer, as it contains active streams,
2617 * streams that are complete */
2618 priv->blocked = FALSE;
2619 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2623 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2625 GstRTSPMediaPrivate *priv = media->priv;
2627 g_mutex_lock (&priv->lock);
2628 priv->status = status;
2629 GST_DEBUG ("setting new status to %d", status);
2630 g_cond_broadcast (&priv->cond);
2631 g_mutex_unlock (&priv->lock);
2635 * gst_rtsp_media_get_status:
2636 * @media: a #GstRTSPMedia
2638 * Get the status of @media. When @media is busy preparing, this function waits
2639 * until @media is prepared or in error.
2641 * Returns: the status of @media.
2644 gst_rtsp_media_get_status (GstRTSPMedia * media)
2646 GstRTSPMediaPrivate *priv = media->priv;
2647 GstRTSPMediaStatus result;
2650 g_mutex_lock (&priv->lock);
2651 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2652 /* while we are preparing, wait */
2653 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2654 GST_DEBUG ("waiting for status change");
2655 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2656 GST_DEBUG ("timeout, assuming error status");
2657 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2660 /* could be success or error */
2661 result = priv->status;
2662 GST_DEBUG ("got status %d", result);
2663 g_mutex_unlock (&priv->lock);
2669 * gst_rtsp_media_seek_trickmode:
2670 * @media: a #GstRTSPMedia
2671 * @range: (transfer none): a #GstRTSPTimeRange
2672 * @flags: The minimal set of #GstSeekFlags to use
2673 * @rate: the rate to use in the seek
2674 * @trickmode_interval: The trickmode interval to use for KEY_UNITS trick mode
2676 * Seek the pipeline of @media to @range with the given @flags and @rate,
2677 * and @trickmode_interval.
2678 * @media must be prepared with gst_rtsp_media_prepare().
2679 * In order to perform the seek operation, the pipeline must contain all
2680 * needed transport parts (transport sinks).
2682 * Returns: %TRUE on success.
2687 gst_rtsp_media_seek_trickmode (GstRTSPMedia * media,
2688 GstRTSPTimeRange * range, GstSeekFlags flags, gdouble rate,
2689 GstClockTime trickmode_interval)
2691 GstRTSPMediaClass *klass;
2692 GstRTSPMediaPrivate *priv;
2694 GstClockTime start, stop;
2695 GstSeekType start_type, stop_type;
2696 gint64 current_position;
2697 gboolean force_seek;
2699 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2701 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2702 /* if there's a range then klass->convert_range must be set */
2703 g_return_val_if_fail (range == NULL || klass->convert_range != NULL, FALSE);
2705 GST_DEBUG ("flags=%x rate=%f", flags, rate);
2709 g_rec_mutex_lock (&priv->state_lock);
2710 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2713 /* check if the media pipeline is complete in order to perform a
2714 * seek operation on it */
2715 if (!check_complete (media))
2718 /* Update the seekable state of the pipeline in case it changed */
2719 check_seekable (media);
2721 if (priv->seekable == 0) {
2722 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2723 " not seekable streams.");
2726 } else if (priv->seekable < 0) {
2730 start_type = stop_type = GST_SEEK_TYPE_NONE;
2731 start = stop = GST_CLOCK_TIME_NONE;
2733 /* if caller provided a range convert it to NPT format
2734 * if no range provided the seek is assumed to be the same position but with
2735 * e.g. the rate changed */
2736 if (range != NULL) {
2737 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2739 gst_rtsp_range_get_times (range, &start, &stop);
2741 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2742 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2743 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2744 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2747 current_position = -1;
2748 if (klass->query_position)
2749 klass->query_position (media, ¤t_position);
2750 GST_INFO ("current media position %" GST_TIME_FORMAT,
2751 GST_TIME_ARGS (current_position));
2753 if (start != GST_CLOCK_TIME_NONE)
2754 start_type = GST_SEEK_TYPE_SET;
2756 if (stop != GST_CLOCK_TIME_NONE)
2757 stop_type = GST_SEEK_TYPE_SET;
2759 /* we force a seek if any seek flag is set, or if the the rate
2760 * is non-standard, i.e. not 1.0 */
2761 force_seek = flags != GST_SEEK_FLAG_NONE || rate != 1.0;
2763 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE || force_seek) {
2764 gboolean had_flags = flags != GST_SEEK_FLAG_NONE;
2766 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2767 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2769 /* depends on the current playing state of the pipeline. We might need to
2770 * queue this until we get EOS. */
2771 flags |= GST_SEEK_FLAG_FLUSH;
2773 /* if range start was not supplied we must continue from current position.
2774 * but since we're doing a flushing seek, let us query the current position
2775 * so we end up at exactly the same position after the seek. */
2776 if (range == NULL || range->min.type == GST_RTSP_TIME_END) {
2777 if (current_position == -1) {
2778 GST_WARNING ("current position unknown");
2780 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2781 GST_TIME_ARGS (current_position));
2782 start = current_position;
2783 start_type = GST_SEEK_TYPE_SET;
2785 flags |= GST_SEEK_FLAG_ACCURATE;
2788 /* only set keyframe flag when modifying start */
2789 if (start_type != GST_SEEK_TYPE_NONE)
2791 flags |= GST_SEEK_FLAG_KEY_UNIT;
2794 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE &&
2796 GST_DEBUG ("no position change, no flags set by caller, so not seeking");
2799 GstEvent *seek_event;
2801 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2804 GstClockTime temp_time = start;
2805 GstSeekType temp_type = start_type;
2808 start_type = stop_type;
2810 stop_type = temp_type;
2813 seek_event = gst_event_new_seek (rate, GST_FORMAT_TIME, flags, start_type,
2814 start, stop_type, stop);
2816 gst_event_set_seek_trickmode_interval (seek_event, trickmode_interval);
2818 res = gst_element_send_event (priv->pipeline, seek_event);
2820 /* and block for the seek to complete */
2821 GST_INFO ("done seeking %d", res);
2825 g_rec_mutex_unlock (&priv->state_lock);
2827 /* wait until pipeline is prerolled again, this will also collect stats */
2828 if (!wait_preroll (media))
2829 goto preroll_failed;
2831 g_rec_mutex_lock (&priv->state_lock);
2832 GST_INFO ("prerolled again");
2835 GST_INFO ("no seek needed");
2838 g_rec_mutex_unlock (&priv->state_lock);
2845 g_rec_mutex_unlock (&priv->state_lock);
2846 GST_INFO ("media %p is not prepared", media);
2851 g_rec_mutex_unlock (&priv->state_lock);
2852 GST_INFO ("pipeline is not complete");
2857 g_rec_mutex_unlock (&priv->state_lock);
2858 GST_INFO ("pipeline is not seekable");
2863 g_rec_mutex_unlock (&priv->state_lock);
2864 GST_WARNING ("conversion to npt not supported");
2869 g_rec_mutex_unlock (&priv->state_lock);
2870 GST_INFO ("seeking failed");
2871 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2876 GST_WARNING ("failed to preroll after seek");
2882 * gst_rtsp_media_seek_full:
2883 * @media: a #GstRTSPMedia
2884 * @range: (transfer none): a #GstRTSPTimeRange
2885 * @flags: The minimal set of #GstSeekFlags to use
2887 * Seek the pipeline of @media to @range with the given @flags.
2888 * @media must be prepared with gst_rtsp_media_prepare().
2890 * Returns: %TRUE on success.
2893 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2896 return gst_rtsp_media_seek_trickmode (media, range, flags, 1.0, 0);
2900 * gst_rtsp_media_seek:
2901 * @media: a #GstRTSPMedia
2902 * @range: (transfer none): a #GstRTSPTimeRange
2904 * Seek the pipeline of @media to @range. @media must be prepared with
2905 * gst_rtsp_media_prepare().
2907 * Returns: %TRUE on success.
2910 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2912 return gst_rtsp_media_seek_trickmode (media, range, GST_SEEK_FLAG_NONE,
2917 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2919 *blocked &= gst_rtsp_stream_is_blocking (stream);
2923 media_streams_blocking (GstRTSPMedia * media)
2925 gboolean blocking = TRUE;
2927 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2933 static GstStateChangeReturn
2934 set_state (GstRTSPMedia * media, GstState state)
2936 GstRTSPMediaPrivate *priv = media->priv;
2937 GstStateChangeReturn ret;
2939 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2941 ret = gst_element_set_state (priv->pipeline, state);
2946 static GstStateChangeReturn
2947 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2949 GstRTSPMediaPrivate *priv = media->priv;
2950 GstStateChangeReturn ret;
2952 GST_INFO ("set target state to %s for media %p",
2953 gst_element_state_get_name (state), media);
2954 priv->target_state = state;
2956 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2957 priv->target_state, NULL);
2960 ret = set_state (media, state);
2962 ret = GST_STATE_CHANGE_SUCCESS;
2967 /* called with state-lock */
2969 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2971 GstRTSPMediaPrivate *priv = media->priv;
2972 GstMessageType type;
2974 type = GST_MESSAGE_TYPE (message);
2977 case GST_MESSAGE_STATE_CHANGED:
2979 GstState old, new, pending;
2981 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2984 gst_message_parse_state_changed (message, &old, &new, &pending);
2986 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2987 gst_element_state_get_name (old), gst_element_state_get_name (new),
2988 gst_element_state_get_name (pending));
2989 if (priv->no_more_pads_pending == 0
2990 && gst_rtsp_media_is_receive_only (media) && old == GST_STATE_READY
2991 && new == GST_STATE_PAUSED) {
2992 GST_INFO ("%p: went to PAUSED, prepared now", media);
2993 collect_media_stats (media);
2995 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2996 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3001 case GST_MESSAGE_BUFFERING:
3005 gst_message_parse_buffering (message, &percent);
3007 /* no state management needed for live pipelines */
3011 if (percent == 100) {
3012 /* a 100% message means buffering is done */
3013 priv->buffering = FALSE;
3014 /* if the desired state is playing, go back */
3015 if (priv->target_state == GST_STATE_PLAYING) {
3016 GST_INFO ("Buffering done, setting pipeline to PLAYING");
3017 set_state (media, GST_STATE_PLAYING);
3019 GST_INFO ("Buffering done");
3022 /* buffering busy */
3023 if (priv->buffering == FALSE) {
3024 if (priv->target_state == GST_STATE_PLAYING) {
3025 /* we were not buffering but PLAYING, PAUSE the pipeline. */
3026 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
3027 set_state (media, GST_STATE_PAUSED);
3029 GST_INFO ("Buffering ...");
3032 priv->buffering = TRUE;
3036 case GST_MESSAGE_LATENCY:
3038 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
3041 case GST_MESSAGE_ERROR:
3046 gst_message_parse_error (message, &gerror, &debug);
3047 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
3048 g_error_free (gerror);
3051 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3054 case GST_MESSAGE_WARNING:
3059 gst_message_parse_warning (message, &gerror, &debug);
3060 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
3061 g_error_free (gerror);
3065 case GST_MESSAGE_ELEMENT:
3067 const GstStructure *s;
3069 s = gst_message_get_structure (message);
3070 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
3071 GST_DEBUG ("media received blocking message");
3072 if (priv->blocked && media_streams_blocking (media) &&
3073 priv->no_more_pads_pending == 0) {
3074 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
3075 collect_media_stats (media);
3077 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3078 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3083 case GST_MESSAGE_STREAM_STATUS:
3085 case GST_MESSAGE_ASYNC_DONE:
3086 if (priv->complete) {
3087 /* receive the final ASYNC_DONE, that is posted by the media pipeline
3088 * after all the transport parts have been successfully added to
3089 * the media streams. */
3090 GST_DEBUG_OBJECT (media, "got async-done");
3091 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3092 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3095 case GST_MESSAGE_EOS:
3096 GST_INFO ("%p: got EOS", media);
3098 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
3099 GST_DEBUG ("shutting down after EOS");
3100 finish_unprepare (media);
3104 GST_INFO ("%p: got message type %d (%s)", media, type,
3105 gst_message_type_get_name (type));
3112 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
3114 GstRTSPMediaPrivate *priv = media->priv;
3115 GstRTSPMediaClass *klass;
3118 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3120 g_rec_mutex_lock (&priv->state_lock);
3121 if (klass->handle_message)
3122 ret = klass->handle_message (media, message);
3125 g_rec_mutex_unlock (&priv->state_lock);
3131 watch_destroyed (GstRTSPMedia * media)
3133 GST_DEBUG_OBJECT (media, "source destroyed");
3134 g_object_unref (media);
3138 find_payload_element (GstElement * payloader)
3140 GstElement *pay = NULL;
3142 if (GST_IS_BIN (payloader)) {
3144 GValue item = { 0 };
3146 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
3147 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
3148 GstElement *element = (GstElement *) g_value_get_object (&item);
3149 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
3153 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
3157 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
3158 pay = gst_object_ref (element);
3159 g_value_unset (&item);
3162 g_value_unset (&item);
3164 gst_iterator_free (iter);
3166 pay = g_object_ref (payloader);
3172 /* called from streaming threads */
3174 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3176 GstRTSPMediaPrivate *priv = media->priv;
3177 GstRTSPStream *stream;
3180 /* find the real payload element */
3181 pay = find_payload_element (element);
3182 stream = gst_rtsp_media_create_stream (media, pay, pad);
3183 gst_object_unref (pay);
3185 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3187 g_rec_mutex_lock (&priv->state_lock);
3188 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3191 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
3193 /* join the element in the PAUSED state because this callback is
3194 * called from the streaming thread and it is PAUSED */
3195 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3196 priv->rtpbin, GST_STATE_PAUSED)) {
3197 GST_WARNING ("failed to join bin element");
3201 gst_rtsp_stream_set_blocked (stream, TRUE);
3203 g_rec_mutex_unlock (&priv->state_lock);
3210 gst_rtsp_media_remove_stream (media, stream);
3211 g_rec_mutex_unlock (&priv->state_lock);
3212 GST_INFO ("ignore pad because we are not preparing");
3218 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
3220 GstRTSPMediaPrivate *priv = media->priv;
3221 GstRTSPStream *stream;
3223 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
3227 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
3229 g_rec_mutex_lock (&priv->state_lock);
3230 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3231 g_rec_mutex_unlock (&priv->state_lock);
3233 gst_rtsp_media_remove_stream (media, stream);
3237 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
3239 GstRTSPMediaPrivate *priv = media->priv;
3241 GST_INFO_OBJECT (element, "no more pads");
3242 g_mutex_lock (&priv->lock);
3243 priv->no_more_pads_pending--;
3244 g_mutex_unlock (&priv->lock);
3247 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
3249 struct _DynPaySignalHandlers
3251 gulong pad_added_handler;
3252 gulong pad_removed_handler;
3253 gulong no_more_pads_handler;
3257 start_preroll (GstRTSPMedia * media)
3259 GstRTSPMediaPrivate *priv = media->priv;
3260 GstStateChangeReturn ret;
3262 GST_INFO ("setting pipeline to PAUSED for media %p", media);
3264 /* start blocked since it is possible that there are no sink elements yet */
3265 media_streams_set_blocked (media, TRUE);
3266 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3269 case GST_STATE_CHANGE_SUCCESS:
3270 GST_INFO ("SUCCESS state change for media %p", media);
3272 case GST_STATE_CHANGE_ASYNC:
3273 GST_INFO ("ASYNC state change for media %p", media);
3275 case GST_STATE_CHANGE_NO_PREROLL:
3276 /* we need to go to PLAYING */
3277 GST_INFO ("NO_PREROLL state change: live media %p", media);
3278 /* FIXME we disable seeking for live streams for now. We should perform a
3279 * seeking query in preroll instead */
3280 priv->seekable = -1;
3281 priv->is_live = TRUE;
3283 ret = set_state (media, GST_STATE_PLAYING);
3284 if (ret == GST_STATE_CHANGE_FAILURE)
3287 case GST_STATE_CHANGE_FAILURE:
3295 GST_WARNING ("failed to preroll pipeline");
3301 wait_preroll (GstRTSPMedia * media)
3303 GstRTSPMediaStatus status;
3305 GST_DEBUG ("wait to preroll pipeline");
3307 /* wait until pipeline is prerolled */
3308 status = gst_rtsp_media_get_status (media);
3309 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3310 goto preroll_failed;
3316 GST_WARNING ("failed to preroll pipeline");
3322 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3324 GstRTSPMediaPrivate *priv = media->priv;
3325 GstRTSPStream *stream = NULL;
3327 GstElement *res = NULL;
3329 g_mutex_lock (&priv->lock);
3330 for (i = 0; i < priv->streams->len; i++) {
3331 stream = g_ptr_array_index (priv->streams, i);
3333 if (sessid == gst_rtsp_stream_get_index (stream))
3338 g_mutex_unlock (&priv->lock);
3341 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3347 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3349 GstRTSPMediaPrivate *priv = media->priv;
3350 GstRTSPStream *stream = NULL;
3352 GstElement *res = NULL;
3354 g_mutex_lock (&priv->lock);
3355 for (i = 0; i < priv->streams->len; i++) {
3356 stream = g_ptr_array_index (priv->streams, i);
3358 if (sessid == gst_rtsp_stream_get_index (stream))
3363 g_mutex_unlock (&priv->lock);
3366 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3372 request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3374 GstRTSPMediaPrivate *priv = media->priv;
3375 GstRTSPStream *stream = NULL;
3377 GstElement *res = NULL;
3379 g_mutex_lock (&priv->lock);
3380 for (i = 0; i < priv->streams->len; i++) {
3381 stream = g_ptr_array_index (priv->streams, i);
3383 if (sessid == gst_rtsp_stream_get_index (stream))
3388 g_mutex_unlock (&priv->lock);
3391 res = gst_rtsp_stream_request_ulpfec_decoder (stream, rtpbin, sessid);
3398 new_storage_cb (GstElement * rtpbin, GObject * storage, guint sessid,
3399 GstRTSPMedia * media)
3401 g_object_set (storage, "size-time", (media->priv->latency + 50) * GST_MSECOND,
3406 start_prepare (GstRTSPMedia * media)
3408 GstRTSPMediaPrivate *priv = media->priv;
3412 g_rec_mutex_lock (&priv->state_lock);
3413 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3414 goto no_longer_preparing;
3416 g_signal_connect (priv->rtpbin, "new-storage", G_CALLBACK (new_storage_cb),
3418 g_signal_connect (priv->rtpbin, "request-fec-decoder",
3419 G_CALLBACK (request_fec_decoder), media);
3421 /* link streams we already have, other streams might appear when we have
3422 * dynamic elements */
3423 for (i = 0; i < priv->streams->len; i++) {
3424 GstRTSPStream *stream;
3426 stream = g_ptr_array_index (priv->streams, i);
3428 if (priv->rtx_time > 0) {
3429 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3430 g_signal_connect (priv->rtpbin, "request-aux-sender",
3431 (GCallback) request_aux_sender, media);
3434 if (priv->do_retransmission) {
3435 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3436 (GCallback) request_aux_receiver, media);
3439 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3440 priv->rtpbin, GST_STATE_NULL)) {
3441 goto join_bin_failed;
3446 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
3447 "do-lost", TRUE, NULL);
3449 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3450 GstElement *elem = walk->data;
3451 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3453 GST_INFO ("adding callbacks for dynamic element %p", elem);
3455 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3456 (GCallback) pad_added_cb, media);
3457 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3458 (GCallback) pad_removed_cb, media);
3459 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3460 (GCallback) no_more_pads_cb, media);
3462 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3465 if (priv->nb_dynamic_elements == 0 && gst_rtsp_media_is_receive_only (media)) {
3466 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3467 * a second ASYNC state change failing */
3468 priv->is_live = TRUE;
3469 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3470 } else if (!start_preroll (media)) {
3471 goto preroll_failed;
3474 g_rec_mutex_unlock (&priv->state_lock);
3478 no_longer_preparing:
3480 GST_INFO ("media is no longer preparing");
3481 g_rec_mutex_unlock (&priv->state_lock);
3486 GST_WARNING ("failed to join bin element");
3487 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3488 g_rec_mutex_unlock (&priv->state_lock);
3493 GST_WARNING ("failed to preroll pipeline");
3494 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3495 g_rec_mutex_unlock (&priv->state_lock);
3501 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3503 GstRTSPMediaPrivate *priv;
3504 GstRTSPMediaClass *klass;
3506 GMainContext *context;
3511 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3513 if (!klass->create_rtpbin)
3514 goto no_create_rtpbin;
3516 priv->rtpbin = klass->create_rtpbin (media);
3517 if (priv->rtpbin != NULL) {
3518 gboolean success = TRUE;
3520 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3522 if (klass->setup_rtpbin)
3523 success = klass->setup_rtpbin (media, priv->rtpbin);
3525 if (success == FALSE) {
3526 gst_object_unref (priv->rtpbin);
3527 priv->rtpbin = NULL;
3530 if (priv->rtpbin == NULL)
3533 priv->thread = thread;
3534 context = (thread != NULL) ? (thread->context) : NULL;
3536 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3538 /* add the pipeline bus to our custom mainloop */
3539 priv->source = gst_bus_create_watch (bus);
3540 gst_object_unref (bus);
3542 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3543 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3545 priv->id = g_source_attach (priv->source, context);
3547 /* add stuff to the bin */
3548 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3550 /* do remainder in context */
3551 source = g_idle_source_new ();
3552 g_source_set_callback (source, (GSourceFunc) start_prepare,
3553 g_object_ref (media), (GDestroyNotify) g_object_unref);
3554 g_source_attach (source, context);
3555 g_source_unref (source);
3562 GST_ERROR ("no create_rtpbin function");
3563 g_critical ("no create_rtpbin vmethod function set");
3568 GST_WARNING ("no rtpbin element");
3569 g_warning ("failed to create element 'rtpbin', check your installation");
3575 * gst_rtsp_media_prepare:
3576 * @media: a #GstRTSPMedia
3577 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3578 * bus handler or %NULL
3580 * Prepare @media for streaming. This function will create the objects
3581 * to manage the streaming. A pipeline must have been set on @media with
3582 * gst_rtsp_media_take_pipeline().
3584 * It will preroll the pipeline and collect vital information about the streams
3585 * such as the duration.
3587 * Returns: %TRUE on success.
3590 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3592 GstRTSPMediaPrivate *priv;
3593 GstRTSPMediaClass *klass;
3595 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3599 g_rec_mutex_lock (&priv->state_lock);
3600 priv->prepare_count++;
3602 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3603 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3606 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3609 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3610 goto not_unprepared;
3612 if (!priv->reusable && priv->reused)
3615 GST_INFO ("preparing media %p", media);
3617 /* reset some variables */
3618 priv->is_live = FALSE;
3619 priv->seekable = -1;
3620 priv->buffering = FALSE;
3621 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3623 /* we're preparing now */
3624 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3626 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3627 if (klass->prepare) {
3628 if (!klass->prepare (media, thread))
3629 goto prepare_failed;
3633 g_rec_mutex_unlock (&priv->state_lock);
3635 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3636 * able to do this async so that we don't block the server thread. */
3637 if (!wait_preroll (media))
3638 goto preroll_failed;
3640 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3642 GST_INFO ("object %p is prerolled", media);
3649 /* we are not going to use the giving thread, so stop it. */
3651 gst_rtsp_thread_stop (thread);
3656 GST_LOG ("media %p was prepared", media);
3657 /* we are not going to use the giving thread, so stop it. */
3659 gst_rtsp_thread_stop (thread);
3660 g_rec_mutex_unlock (&priv->state_lock);
3666 /* we are not going to use the giving thread, so stop it. */
3668 gst_rtsp_thread_stop (thread);
3669 GST_WARNING ("media %p was not unprepared", media);
3670 priv->prepare_count--;
3671 g_rec_mutex_unlock (&priv->state_lock);
3676 /* we are not going to use the giving thread, so stop it. */
3678 gst_rtsp_thread_stop (thread);
3679 priv->prepare_count--;
3680 g_rec_mutex_unlock (&priv->state_lock);
3681 GST_WARNING ("can not reuse media %p", media);
3686 /* we are not going to use the giving thread, so stop it. */
3688 gst_rtsp_thread_stop (thread);
3689 priv->prepare_count--;
3690 g_rec_mutex_unlock (&priv->state_lock);
3691 GST_ERROR ("failed to prepare media");
3696 GST_WARNING ("failed to preroll pipeline");
3697 gst_rtsp_media_unprepare (media);
3702 /* must be called with state-lock */
3704 finish_unprepare (GstRTSPMedia * media)
3706 GstRTSPMediaPrivate *priv = media->priv;
3710 if (priv->finishing_unprepare)
3712 priv->finishing_unprepare = TRUE;
3714 GST_DEBUG ("shutting down");
3716 /* release the lock on shutdown, otherwise pad_added_cb might try to
3717 * acquire the lock and then we deadlock */
3718 g_rec_mutex_unlock (&priv->state_lock);
3719 set_state (media, GST_STATE_NULL);
3720 g_rec_mutex_lock (&priv->state_lock);
3722 media_streams_set_blocked (media, FALSE);
3724 for (i = 0; i < priv->streams->len; i++) {
3725 GstRTSPStream *stream;
3727 GST_INFO ("Removing elements of stream %d from pipeline", i);
3729 stream = g_ptr_array_index (priv->streams, i);
3731 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3734 /* remove the pad signal handlers */
3735 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3736 GstElement *elem = walk->data;
3737 DynPaySignalHandlers *handlers;
3740 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3741 g_assert (handlers != NULL);
3743 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3744 g_signal_handler_disconnect (G_OBJECT (elem),
3745 handlers->pad_removed_handler);
3746 g_signal_handler_disconnect (G_OBJECT (elem),
3747 handlers->no_more_pads_handler);
3749 g_slice_free (DynPaySignalHandlers, handlers);
3752 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3753 priv->rtpbin = NULL;
3756 gst_object_unref (priv->nettime);
3757 priv->nettime = NULL;
3759 priv->reused = TRUE;
3760 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3762 /* when the media is not reusable, this will effectively unref the media and
3764 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3766 /* the source has the last ref to the media */
3768 GST_DEBUG ("destroy source");
3769 g_source_destroy (priv->source);
3770 g_source_unref (priv->source);
3773 GST_DEBUG ("stop thread");
3774 gst_rtsp_thread_stop (priv->thread);
3777 priv->finishing_unprepare = FALSE;
3780 /* called with state-lock */
3782 default_unprepare (GstRTSPMedia * media)
3784 GstRTSPMediaPrivate *priv = media->priv;
3786 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3788 if (priv->eos_shutdown) {
3789 GST_DEBUG ("sending EOS for shutdown");
3790 /* ref so that we don't disappear */
3791 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3792 /* we need to go to playing again for the EOS to propagate, normally in this
3793 * state, nothing is receiving data from us anymore so this is ok. */
3794 set_state (media, GST_STATE_PLAYING);
3796 finish_unprepare (media);
3802 * gst_rtsp_media_unprepare:
3803 * @media: a #GstRTSPMedia
3805 * Unprepare @media. After this call, the media should be prepared again before
3806 * it can be used again. If the media is set to be non-reusable, a new instance
3809 * Returns: %TRUE on success.
3812 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3814 GstRTSPMediaPrivate *priv;
3817 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3821 g_rec_mutex_lock (&priv->state_lock);
3822 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3823 goto was_unprepared;
3825 priv->prepare_count--;
3826 if (priv->prepare_count > 0)
3829 GST_INFO ("unprepare media %p", media);
3830 set_target_state (media, GST_STATE_NULL, FALSE);
3833 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3834 GstRTSPMediaClass *klass;
3836 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3837 if (klass->unprepare)
3838 success = klass->unprepare (media);
3840 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3841 finish_unprepare (media);
3843 g_rec_mutex_unlock (&priv->state_lock);
3849 g_rec_mutex_unlock (&priv->state_lock);
3850 GST_INFO ("media %p was already unprepared", media);
3855 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3856 g_rec_mutex_unlock (&priv->state_lock);
3861 /* should be called with state-lock */
3863 get_clock_unlocked (GstRTSPMedia * media)
3865 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3866 GST_DEBUG_OBJECT (media, "media was not prepared");
3869 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3873 * gst_rtsp_media_get_clock:
3874 * @media: a #GstRTSPMedia
3876 * Get the clock that is used by the pipeline in @media.
3878 * @media must be prepared before this method returns a valid clock object.
3880 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3883 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3886 GstRTSPMediaPrivate *priv;
3888 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3892 g_rec_mutex_lock (&priv->state_lock);
3893 clock = get_clock_unlocked (media);
3894 g_rec_mutex_unlock (&priv->state_lock);
3900 * gst_rtsp_media_get_base_time:
3901 * @media: a #GstRTSPMedia
3903 * Get the base_time that is used by the pipeline in @media.
3905 * @media must be prepared before this method returns a valid base_time.
3907 * Returns: the base_time used by @media.
3910 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3912 GstClockTime result;
3913 GstRTSPMediaPrivate *priv;
3915 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3919 g_rec_mutex_lock (&priv->state_lock);
3920 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3923 result = gst_element_get_base_time (media->priv->pipeline);
3924 g_rec_mutex_unlock (&priv->state_lock);
3931 g_rec_mutex_unlock (&priv->state_lock);
3932 GST_DEBUG_OBJECT (media, "media was not prepared");
3933 return GST_CLOCK_TIME_NONE;
3938 * gst_rtsp_media_get_time_provider:
3939 * @media: a #GstRTSPMedia
3940 * @address: (allow-none): an address or %NULL
3941 * @port: a port or 0
3943 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3944 * will listen on @address and @port for client time requests.
3946 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3948 GstNetTimeProvider *
3949 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3952 GstRTSPMediaPrivate *priv;
3953 GstNetTimeProvider *provider = NULL;
3955 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3959 g_rec_mutex_lock (&priv->state_lock);
3960 if (priv->time_provider) {
3961 if ((provider = priv->nettime) == NULL) {
3964 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3965 provider = gst_net_time_provider_new (clock, address, port);
3966 gst_object_unref (clock);
3968 priv->nettime = provider;
3972 g_rec_mutex_unlock (&priv->state_lock);
3975 gst_object_ref (provider);
3981 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3983 return gst_rtsp_sdp_from_media (sdp, info, media);
3987 * gst_rtsp_media_setup_sdp:
3988 * @media: a #GstRTSPMedia
3989 * @sdp: (transfer none): a #GstSDPMessage
3990 * @info: (transfer none): a #GstSDPInfo
3992 * Add @media specific info to @sdp. @info is used to configure the connection
3993 * information in the SDP.
3995 * Returns: TRUE on success.
3998 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
4001 GstRTSPMediaPrivate *priv;
4002 GstRTSPMediaClass *klass;
4005 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4006 g_return_val_if_fail (sdp != NULL, FALSE);
4007 g_return_val_if_fail (info != NULL, FALSE);
4011 g_rec_mutex_lock (&priv->state_lock);
4013 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4015 if (!klass->setup_sdp)
4018 res = klass->setup_sdp (media, sdp, info);
4020 g_rec_mutex_unlock (&priv->state_lock);
4027 g_rec_mutex_unlock (&priv->state_lock);
4028 GST_ERROR ("no setup_sdp function");
4029 g_critical ("no setup_sdp vmethod function set");
4035 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4037 GstRTSPMediaPrivate *priv = media->priv;
4040 medias_len = gst_sdp_message_medias_len (sdp);
4041 if (medias_len != priv->streams->len) {
4042 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
4043 priv->streams->len, medias_len);
4047 for (i = 0; i < medias_len; i++) {
4049 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
4050 GstRTSPStream *stream;
4051 gint j, formats_len;
4052 const gchar *control;
4053 GstRTSPProfile profile, profiles;
4055 stream = g_ptr_array_index (priv->streams, i);
4057 /* TODO: Should we do something with the other SDP information? */
4060 proto = gst_sdp_media_get_proto (sdp_media);
4061 if (proto == NULL) {
4062 GST_ERROR ("%p: SDP media %d has no proto", media, i);
4066 if (g_str_equal (proto, "RTP/AVP")) {
4067 profile = GST_RTSP_PROFILE_AVP;
4068 } else if (g_str_equal (proto, "RTP/SAVP")) {
4069 profile = GST_RTSP_PROFILE_SAVP;
4070 } else if (g_str_equal (proto, "RTP/AVPF")) {
4071 profile = GST_RTSP_PROFILE_AVPF;
4072 } else if (g_str_equal (proto, "RTP/SAVPF")) {
4073 profile = GST_RTSP_PROFILE_SAVPF;
4075 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4079 profiles = gst_rtsp_stream_get_profiles (stream);
4080 if ((profiles & profile) == 0) {
4081 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
4085 formats_len = gst_sdp_media_formats_len (sdp_media);
4086 for (j = 0; j < formats_len; j++) {
4091 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
4093 GST_DEBUG (" looking at %d pt: %d", j, pt);
4096 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
4098 GST_WARNING (" skipping pt %d without caps", pt);
4102 /* do some tweaks */
4103 GST_DEBUG ("mapping sdp session level attributes to caps");
4104 gst_sdp_message_attributes_to_caps (sdp, caps);
4105 GST_DEBUG ("mapping sdp media level attributes to caps");
4106 gst_sdp_media_attributes_to_caps (sdp_media, caps);
4108 s = gst_caps_get_structure (caps, 0);
4109 gst_structure_set_name (s, "application/x-rtp");
4111 if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
4112 gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
4114 gst_rtsp_stream_set_pt_map (stream, pt, caps);
4115 gst_caps_unref (caps);
4118 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
4120 gst_rtsp_stream_set_control (stream, control);
4128 * gst_rtsp_media_handle_sdp:
4129 * @media: a #GstRTSPMedia
4130 * @sdp: (transfer none): a #GstSDPMessage
4132 * Configure an SDP on @media for receiving streams
4134 * Returns: TRUE on success.
4137 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
4139 GstRTSPMediaPrivate *priv;
4140 GstRTSPMediaClass *klass;
4143 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4144 g_return_val_if_fail (sdp != NULL, FALSE);
4148 g_rec_mutex_lock (&priv->state_lock);
4150 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4152 if (!klass->handle_sdp)
4155 res = klass->handle_sdp (media, sdp);
4157 g_rec_mutex_unlock (&priv->state_lock);
4164 g_rec_mutex_unlock (&priv->state_lock);
4165 GST_ERROR ("no handle_sdp function");
4166 g_critical ("no handle_sdp vmethod function set");
4172 do_set_seqnum (GstRTSPStream * stream)
4176 if (gst_rtsp_stream_is_sender (stream)) {
4177 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
4178 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
4182 /* call with state_lock */
4184 default_suspend (GstRTSPMedia * media)
4186 GstRTSPMediaPrivate *priv = media->priv;
4187 GstStateChangeReturn ret;
4189 switch (priv->suspend_mode) {
4190 case GST_RTSP_SUSPEND_MODE_NONE:
4191 GST_DEBUG ("media %p no suspend", media);
4193 case GST_RTSP_SUSPEND_MODE_PAUSE:
4194 GST_DEBUG ("media %p suspend to PAUSED", media);
4195 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
4196 if (ret == GST_STATE_CHANGE_FAILURE)
4199 case GST_RTSP_SUSPEND_MODE_RESET:
4200 GST_DEBUG ("media %p suspend to NULL", media);
4201 ret = set_target_state (media, GST_STATE_NULL, TRUE);
4202 if (ret == GST_STATE_CHANGE_FAILURE)
4204 /* Because payloader needs to set the sequence number as
4205 * monotonic, we need to preserve the sequence number
4206 * after pause. (otherwise going from pause to play, which
4207 * is actually from NULL to PLAY will create a new sequence
4209 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
4220 GST_WARNING ("failed changing pipeline's state for media %p", media);
4226 * gst_rtsp_media_suspend:
4227 * @media: a #GstRTSPMedia
4229 * Suspend @media. The state of the pipeline managed by @media is set to
4230 * GST_STATE_NULL but all streams are kept. @media can be prepared again
4231 * with gst_rtsp_media_unsuspend()
4233 * @media must be prepared with gst_rtsp_media_prepare();
4235 * Returns: %TRUE on success.
4238 gst_rtsp_media_suspend (GstRTSPMedia * media)
4240 GstRTSPMediaPrivate *priv = media->priv;
4241 GstRTSPMediaClass *klass;
4243 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4245 GST_FIXME ("suspend for dynamic pipelines needs fixing");
4247 g_rec_mutex_lock (&priv->state_lock);
4248 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
4251 /* don't attempt to suspend when something is busy */
4252 if (priv->n_active > 0)
4255 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4256 if (klass->suspend) {
4257 if (!klass->suspend (media))
4258 goto suspend_failed;
4261 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
4263 g_rec_mutex_unlock (&priv->state_lock);
4270 g_rec_mutex_unlock (&priv->state_lock);
4271 GST_WARNING ("media %p was not prepared", media);
4276 g_rec_mutex_unlock (&priv->state_lock);
4277 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4278 GST_WARNING ("failed to suspend media %p", media);
4283 /* call with state_lock */
4285 default_unsuspend (GstRTSPMedia * media)
4287 GstRTSPMediaPrivate *priv = media->priv;
4288 gboolean preroll_ok;
4290 switch (priv->suspend_mode) {
4291 case GST_RTSP_SUSPEND_MODE_NONE:
4292 if (gst_rtsp_media_is_receive_only (media))
4294 if (media_streams_blocking (media)) {
4295 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4296 /* at this point the media pipeline has been updated and contain all
4297 * specific transport parts: all active streams contain at least one sink
4298 * element and it's safe to unblock any blocked streams that are active */
4299 media_unblock_linked (media);
4301 /* streams are not blocked and media is suspended from PAUSED */
4302 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4304 g_rec_mutex_unlock (&priv->state_lock);
4305 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
4306 g_rec_mutex_lock (&priv->state_lock);
4307 goto preroll_failed;
4309 g_rec_mutex_lock (&priv->state_lock);
4311 case GST_RTSP_SUSPEND_MODE_PAUSE:
4312 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4314 case GST_RTSP_SUSPEND_MODE_RESET:
4316 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4317 /* at this point the media pipeline has been updated and contain all
4318 * specific transport parts: all active streams contain at least one sink
4319 * element and it's safe to unblock any blocked streams that are active */
4320 media_unblock_linked (media);
4321 if (!start_preroll (media))
4324 g_rec_mutex_unlock (&priv->state_lock);
4325 preroll_ok = wait_preroll (media);
4326 g_rec_mutex_lock (&priv->state_lock);
4329 goto preroll_failed;
4340 GST_WARNING ("failed to preroll pipeline");
4345 GST_WARNING ("failed to preroll pipeline");
4351 * gst_rtsp_media_unsuspend:
4352 * @media: a #GstRTSPMedia
4354 * Unsuspend @media if it was in a suspended state. This method does nothing
4355 * when the media was not in the suspended state.
4357 * Returns: %TRUE on success.
4360 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4362 GstRTSPMediaPrivate *priv = media->priv;
4363 GstRTSPMediaClass *klass;
4365 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4367 g_rec_mutex_lock (&priv->state_lock);
4368 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4371 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4372 if (klass->unsuspend) {
4373 if (!klass->unsuspend (media))
4374 goto unsuspend_failed;
4378 g_rec_mutex_unlock (&priv->state_lock);
4385 g_rec_mutex_unlock (&priv->state_lock);
4386 GST_WARNING ("failed to unsuspend media %p", media);
4387 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4392 /* must be called with state-lock */
4394 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4396 GstRTSPMediaPrivate *priv = media->priv;
4398 if (state == GST_STATE_NULL) {
4399 gst_rtsp_media_unprepare (media);
4401 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4402 set_target_state (media, state, FALSE);
4403 /* when we are buffering, don't update the state yet, this will be done
4404 * when buffering finishes */
4405 if (priv->buffering) {
4406 GST_INFO ("Buffering busy, delay state change");
4408 if (state == GST_STATE_PLAYING)
4409 /* make sure pads are not blocking anymore when going to PLAYING */
4410 media_unblock_linked (media);
4412 set_state (media, state);
4414 /* and suspend after pause */
4415 if (state == GST_STATE_PAUSED)
4416 gst_rtsp_media_suspend (media);
4422 * gst_rtsp_media_set_pipeline_state:
4423 * @media: a #GstRTSPMedia
4424 * @state: the target state of the pipeline
4426 * Set the state of the pipeline managed by @media to @state
4429 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4431 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4433 g_rec_mutex_lock (&media->priv->state_lock);
4434 media_set_pipeline_state_locked (media, state);
4435 g_rec_mutex_unlock (&media->priv->state_lock);
4439 * gst_rtsp_media_set_state:
4440 * @media: a #GstRTSPMedia
4441 * @state: the target state of the media
4442 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4443 * a #GPtrArray of #GstRTSPStreamTransport pointers
4445 * Set the state of @media to @state and for the transports in @transports.
4447 * @media must be prepared with gst_rtsp_media_prepare();
4449 * Returns: %TRUE on success.
4452 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4453 GPtrArray * transports)
4455 GstRTSPMediaPrivate *priv;
4457 gboolean activate, deactivate, do_state;
4460 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4461 g_return_val_if_fail (transports != NULL, FALSE);
4465 g_rec_mutex_lock (&priv->state_lock);
4467 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING
4468 && gst_rtsp_media_is_shared (media)) {
4469 g_rec_mutex_unlock (&priv->state_lock);
4470 gst_rtsp_media_get_status (media);
4471 g_rec_mutex_lock (&priv->state_lock);
4473 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4475 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4476 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4479 /* NULL and READY are the same */
4480 if (state == GST_STATE_READY)
4481 state = GST_STATE_NULL;
4483 activate = deactivate = FALSE;
4485 GST_INFO ("going to state %s media %p, target state %s",
4486 gst_element_state_get_name (state), media,
4487 gst_element_state_get_name (priv->target_state));
4490 case GST_STATE_NULL:
4491 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4492 if (priv->target_state >= GST_STATE_PAUSED)
4495 case GST_STATE_PAUSED:
4496 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4497 if (priv->target_state == GST_STATE_PLAYING)
4500 case GST_STATE_PLAYING:
4501 /* we're going to PLAYING, activate */
4507 old_active = priv->n_active;
4509 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4510 activate, deactivate);
4511 for (i = 0; i < transports->len; i++) {
4512 GstRTSPStreamTransport *trans;
4514 /* we need a non-NULL entry in the array */
4515 trans = g_ptr_array_index (transports, i);
4520 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4522 } else if (deactivate) {
4523 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4528 /* we just activated the first media, do the playing state change */
4529 if (old_active == 0 && activate)
4531 /* if we have no more active media and prepare count is not indicate
4532 * that there are new session/sessions ongoing,
4533 * do the downward state changes */
4534 else if (priv->n_active == 0 && priv->prepare_count <= 1)
4539 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4542 if (priv->target_state != state) {
4544 media_set_pipeline_state_locked (media, state);
4545 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4550 /* remember where we are */
4551 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4552 old_active != priv->n_active))
4553 collect_media_stats (media);
4555 g_rec_mutex_unlock (&priv->state_lock);
4562 GST_WARNING ("media %p was not prepared", media);
4563 g_rec_mutex_unlock (&priv->state_lock);
4568 GST_WARNING ("media %p in error status while changing to state %d",
4570 if (state == GST_STATE_NULL) {
4571 for (i = 0; i < transports->len; i++) {
4572 GstRTSPStreamTransport *trans;
4574 /* we need a non-NULL entry in the array */
4575 trans = g_ptr_array_index (transports, i);
4579 gst_rtsp_stream_transport_set_active (trans, FALSE);
4583 g_rec_mutex_unlock (&priv->state_lock);
4589 * gst_rtsp_media_set_transport_mode:
4590 * @media: a #GstRTSPMedia
4591 * @mode: the new value
4593 * Sets if the media pipeline can work in PLAY or RECORD mode
4596 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4597 GstRTSPTransportMode mode)
4599 GstRTSPMediaPrivate *priv;
4601 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4605 g_mutex_lock (&priv->lock);
4606 priv->transport_mode = mode;
4607 g_mutex_unlock (&priv->lock);
4611 * gst_rtsp_media_get_transport_mode:
4612 * @media: a #GstRTSPMedia
4614 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4616 * Returns: The transport mode.
4618 GstRTSPTransportMode
4619 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4621 GstRTSPMediaPrivate *priv;
4622 GstRTSPTransportMode res;
4624 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4628 g_mutex_lock (&priv->lock);
4629 res = priv->transport_mode;
4630 g_mutex_unlock (&priv->lock);
4636 * gst_rtsp_media_seekable:
4637 * @media: a #GstRTSPMedia
4639 * Check if the pipeline for @media seek and up to what point in time,
4642 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4643 * and > 0 to indicate the longest duration between any two random access points.
4644 * %G_MAXINT64 means any value is possible.
4649 gst_rtsp_media_seekable (GstRTSPMedia * media)
4651 GstRTSPMediaPrivate *priv;
4652 GstClockTimeDiff res;
4654 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4658 /* Currently we are not able to seek on live streams,
4659 * and no stream is seekable only to the beginning */
4660 g_mutex_lock (&priv->lock);
4661 res = priv->seekable;
4662 g_mutex_unlock (&priv->lock);
4668 * gst_rtsp_media_complete_pipeline:
4669 * @media: a #GstRTSPMedia
4670 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4672 * Add a receiver and sender parts to the pipeline based on the transport from
4675 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4680 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4682 GstRTSPMediaPrivate *priv;
4685 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4686 g_return_val_if_fail (transports, FALSE);
4688 GST_DEBUG_OBJECT (media, "complete pipeline");
4692 g_mutex_lock (&priv->lock);
4693 for (i = 0; i < priv->streams->len; i++) {
4694 GstRTSPStreamTransport *transport;
4695 GstRTSPStream *stream;
4696 const GstRTSPTransport *rtsp_transport;
4698 transport = g_ptr_array_index (transports, i);
4702 stream = gst_rtsp_stream_transport_get_stream (transport);
4706 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4708 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4709 g_mutex_unlock (&priv->lock);
4714 priv->complete = TRUE;
4715 g_mutex_unlock (&priv->lock);
4721 * gst_rtsp_media_is_receive_only:
4723 * Returns: %TRUE if @media is receive-only, %FALSE otherwise.
4727 gst_rtsp_media_is_receive_only (GstRTSPMedia * media)
4729 GstRTSPMediaPrivate *priv = media->priv;
4730 gboolean receive_only = TRUE;
4733 for (i = 0; i < priv->streams->len; i++) {
4734 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
4735 if (gst_rtsp_stream_is_sender (stream) ||
4736 !gst_rtsp_stream_is_receiver (stream)) {
4737 receive_only = FALSE;
4742 return receive_only;
4746 * gst_rtsp_media_set_rate_control:
4748 * Define whether @media will follow the Rate-Control=no behaviour as specified
4749 * in the ONVIF replay spec.
4754 gst_rtsp_media_set_rate_control (GstRTSPMedia * media, gboolean enabled)
4756 GstRTSPMediaPrivate *priv;
4759 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4761 GST_LOG_OBJECT (media, "%s rate control", enabled ? "Enabling" : "Disabling");
4765 g_mutex_lock (&priv->lock);
4766 priv->do_rate_control = enabled;
4767 for (i = 0; i < priv->streams->len; i++) {
4768 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
4770 gst_rtsp_stream_set_rate_control (stream, enabled);
4773 g_mutex_unlock (&priv->lock);
4777 * gst_rtsp_media_get_rate_control:
4779 * Returns: whether @media will follow the Rate-Control=no behaviour as specified
4780 * in the ONVIF replay spec.
4785 gst_rtsp_media_get_rate_control (GstRTSPMedia * media)
4787 GstRTSPMediaPrivate *priv;
4790 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4794 g_mutex_lock (&priv->lock);
4795 res = priv->do_rate_control;
4796 g_mutex_unlock (&priv->lock);