2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: The media pipeline
24 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
25 * #GstRTSPSessionMedia
27 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
28 * streaming to the clients. The actual data transfer is done by the
29 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
31 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
32 * client does a DESCRIBE or SETUP of a resource.
34 * A media is created with gst_rtsp_media_new() that takes the element that will
35 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
36 * object needs to be made with the gst_rtsp_media_create_stream() which takes
37 * the payloader element and the source pad that produces the RTP stream.
39 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
40 * prepare method will add rtpbin and sinks and sources to send and receive RTP
41 * and RTCP packets from the clients. Each stream srcpad is connected to an
42 * input into the internal rtpbin.
44 * It is also possible to dynamically create #GstRTSPStream objects during the
45 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
48 * After the media is prepared, it is ready for streaming. It will usually be
49 * managed in a session with gst_rtsp_session_manage_media(). See
50 * #GstRTSPSession and #GstRTSPSessionMedia.
52 * The state of the media can be controlled with gst_rtsp_media_set_state ().
53 * Seeking can be done with gst_rtsp_media_seek().
55 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
56 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
59 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
60 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
61 * can be prepared again after an unprepare.
63 * Last reviewed on 2013-07-11 (1.0.0)
70 #include <gst/app/gstappsrc.h>
71 #include <gst/app/gstappsink.h>
73 #include <gst/sdp/gstmikey.h>
74 #include <gst/rtp/gstrtppayloads.h>
76 #define AES_128_KEY_LEN 16
77 #define AES_256_KEY_LEN 32
79 #define HMAC_32_KEY_LEN 4
80 #define HMAC_80_KEY_LEN 10
82 #include "rtsp-media.h"
84 #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
85 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
87 struct _GstRTSPMediaPrivate
92 /* protected by lock */
93 GstRTSPPermissions *permissions;
95 gboolean suspend_mode;
97 GstRTSPProfile profiles;
98 GstRTSPLowerTrans protocols;
100 gboolean eos_shutdown;
102 GstRTSPAddressPool *pool;
103 gchar *multicast_iface;
105 GstRTSPTransportMode transport_mode;
106 gboolean stop_on_disconnect;
109 GRecMutex state_lock; /* locking order: state lock, lock */
110 GPtrArray *streams; /* protected by lock */
111 GList *dynamic; /* protected by lock */
112 GstRTSPMediaStatus status; /* protected by lock */
117 /* the pipeline for the media */
118 GstElement *pipeline;
121 GstRTSPThread *thread;
122 GList *pending_pipeline_elements;
124 gboolean time_provider;
125 GstNetTimeProvider *nettime;
128 GstClockTimeDiff seekable;
130 GstState target_state;
132 /* RTP session manager */
135 /* the range of media */
136 GstRTSPTimeRange range; /* protected by lock */
137 GstClockTime range_start;
138 GstClockTime range_stop;
140 GList *payloads; /* protected by lock */
141 GstClockTime rtx_time; /* protected by lock */
142 gboolean do_retransmission; /* protected by lock */
143 guint latency; /* protected by lock */
144 GstClock *clock; /* protected by lock */
145 GstRTSPPublishClockMode publish_clock_mode;
147 /* Dynamic element handling */
148 guint nb_dynamic_elements;
149 guint no_more_pads_pending;
152 #define DEFAULT_SHARED FALSE
153 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
154 #define DEFAULT_REUSABLE FALSE
155 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
156 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
157 GST_RTSP_LOWER_TRANS_TCP
158 #define DEFAULT_EOS_SHUTDOWN FALSE
159 #define DEFAULT_BUFFER_SIZE 0x80000
160 #define DEFAULT_TIME_PROVIDER FALSE
161 #define DEFAULT_LATENCY 200
162 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
163 #define DEFAULT_STOP_ON_DISCONNECT TRUE
165 #define DEFAULT_DO_RETRANSMISSION FALSE
167 /* define to dump received RTCP packets */
184 PROP_STOP_ON_DISCONNECT,
192 SIGNAL_REMOVED_STREAM,
200 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
201 #define GST_CAT_DEFAULT rtsp_media_debug
203 static void gst_rtsp_media_get_property (GObject * object, guint propid,
204 GValue * value, GParamSpec * pspec);
205 static void gst_rtsp_media_set_property (GObject * object, guint propid,
206 const GValue * value, GParamSpec * pspec);
207 static void gst_rtsp_media_finalize (GObject * obj);
209 static gboolean default_handle_message (GstRTSPMedia * media,
210 GstMessage * message);
211 static void finish_unprepare (GstRTSPMedia * media);
212 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
213 static gboolean default_unprepare (GstRTSPMedia * media);
214 static gboolean default_suspend (GstRTSPMedia * media);
215 static gboolean default_unsuspend (GstRTSPMedia * media);
216 static gboolean default_convert_range (GstRTSPMedia * media,
217 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
218 static gboolean default_query_position (GstRTSPMedia * media,
220 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
221 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
222 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
224 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
226 static gboolean wait_preroll (GstRTSPMedia * media);
228 static GstElement *find_payload_element (GstElement * payloader);
230 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
231 static gboolean check_complete (GstRTSPMedia * media);
233 #define C_ENUM(v) ((gint) v)
236 gst_rtsp_suspend_mode_get_type (void)
239 static const GEnumValue values[] = {
240 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
241 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
243 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
248 if (g_once_init_enter (&id)) {
249 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
250 g_once_init_leave (&id, tmp);
255 #define C_FLAGS(v) ((guint) v)
258 gst_rtsp_transport_mode_get_type (void)
261 static const GFlagsValue values[] = {
262 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
264 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
269 if (g_once_init_enter (&id)) {
270 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
271 g_once_init_leave (&id, tmp);
277 gst_rtsp_publish_clock_mode_get_type (void)
280 static const GEnumValue values[] = {
281 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
282 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
283 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
284 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
286 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
287 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
292 if (g_once_init_enter (&id)) {
293 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
294 g_once_init_leave (&id, tmp);
299 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
302 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
304 GObjectClass *gobject_class;
306 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
308 gobject_class = G_OBJECT_CLASS (klass);
310 gobject_class->get_property = gst_rtsp_media_get_property;
311 gobject_class->set_property = gst_rtsp_media_set_property;
312 gobject_class->finalize = gst_rtsp_media_finalize;
314 g_object_class_install_property (gobject_class, PROP_SHARED,
315 g_param_spec_boolean ("shared", "Shared",
316 "If this media pipeline can be shared", DEFAULT_SHARED,
317 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
319 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
320 g_param_spec_enum ("suspend-mode", "Suspend Mode",
321 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
322 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
324 g_object_class_install_property (gobject_class, PROP_REUSABLE,
325 g_param_spec_boolean ("reusable", "Reusable",
326 "If this media pipeline can be reused after an unprepare",
327 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
329 g_object_class_install_property (gobject_class, PROP_PROFILES,
330 g_param_spec_flags ("profiles", "Profiles",
331 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
332 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
334 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
335 g_param_spec_flags ("protocols", "Protocols",
336 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
337 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
340 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
341 "Send an EOS event to the pipeline before unpreparing",
342 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
344 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
345 g_param_spec_uint ("buffer-size", "Buffer Size",
346 "The kernel UDP buffer size to use", 0, G_MAXUINT,
347 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 g_object_class_install_property (gobject_class, PROP_ELEMENT,
350 g_param_spec_object ("element", "The Element",
351 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
352 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
354 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
355 g_param_spec_boolean ("time-provider", "Time Provider",
356 "Use a NetTimeProvider for clients",
357 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
359 g_object_class_install_property (gobject_class, PROP_LATENCY,
360 g_param_spec_uint ("latency", "Latency",
361 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
362 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
364 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
365 g_param_spec_flags ("transport-mode", "Transport Mode",
366 "If this media pipeline can be used for PLAY or RECORD",
367 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
368 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
371 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
372 "If this media pipeline should be stopped "
373 "when a client disconnects without TEARDOWN",
374 DEFAULT_STOP_ON_DISCONNECT,
375 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
377 g_object_class_install_property (gobject_class, PROP_CLOCK,
378 g_param_spec_object ("clock", "Clock",
379 "Clock to be used by the media pipeline",
380 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
383 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
384 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
385 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
387 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
388 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
389 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
390 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
391 GST_TYPE_RTSP_STREAM);
393 gst_rtsp_media_signals[SIGNAL_PREPARED] =
394 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
395 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
396 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
398 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
399 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
400 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
401 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
403 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
404 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
405 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
406 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
408 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
409 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
410 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
411 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
413 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
415 klass->handle_message = default_handle_message;
416 klass->prepare = default_prepare;
417 klass->unprepare = default_unprepare;
418 klass->suspend = default_suspend;
419 klass->unsuspend = default_unsuspend;
420 klass->convert_range = default_convert_range;
421 klass->query_position = default_query_position;
422 klass->query_stop = default_query_stop;
423 klass->create_rtpbin = default_create_rtpbin;
424 klass->setup_sdp = default_setup_sdp;
425 klass->handle_sdp = default_handle_sdp;
429 gst_rtsp_media_init (GstRTSPMedia * media)
431 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
435 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
436 g_mutex_init (&priv->lock);
437 g_cond_init (&priv->cond);
438 g_rec_mutex_init (&priv->state_lock);
440 priv->shared = DEFAULT_SHARED;
441 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
442 priv->reusable = DEFAULT_REUSABLE;
443 priv->profiles = DEFAULT_PROFILES;
444 priv->protocols = DEFAULT_PROTOCOLS;
445 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
446 priv->buffer_size = DEFAULT_BUFFER_SIZE;
447 priv->time_provider = DEFAULT_TIME_PROVIDER;
448 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
449 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
450 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
451 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
455 gst_rtsp_media_finalize (GObject * obj)
457 GstRTSPMediaPrivate *priv;
460 media = GST_RTSP_MEDIA (obj);
463 GST_INFO ("finalize media %p", media);
465 if (priv->permissions)
466 gst_rtsp_permissions_unref (priv->permissions);
468 g_ptr_array_unref (priv->streams);
470 g_list_free_full (priv->dynamic, gst_object_unref);
471 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
474 gst_object_unref (priv->pipeline);
476 gst_object_unref (priv->nettime);
477 gst_object_unref (priv->element);
479 g_object_unref (priv->pool);
481 g_list_free (priv->payloads);
482 g_free (priv->multicast_iface);
483 g_mutex_clear (&priv->lock);
484 g_cond_clear (&priv->cond);
485 g_rec_mutex_clear (&priv->state_lock);
487 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
491 gst_rtsp_media_get_property (GObject * object, guint propid,
492 GValue * value, GParamSpec * pspec)
494 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
498 g_value_set_object (value, media->priv->element);
501 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
503 case PROP_SUSPEND_MODE:
504 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
507 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
510 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
513 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
515 case PROP_EOS_SHUTDOWN:
516 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
518 case PROP_BUFFER_SIZE:
519 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
521 case PROP_TIME_PROVIDER:
522 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
525 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
527 case PROP_TRANSPORT_MODE:
528 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
530 case PROP_STOP_ON_DISCONNECT:
531 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
534 g_value_take_object (value, gst_rtsp_media_get_clock (media));
537 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
542 gst_rtsp_media_set_property (GObject * object, guint propid,
543 const GValue * value, GParamSpec * pspec)
545 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
549 media->priv->element = g_value_get_object (value);
550 gst_object_ref_sink (media->priv->element);
553 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
555 case PROP_SUSPEND_MODE:
556 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
559 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
562 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
565 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
567 case PROP_EOS_SHUTDOWN:
568 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
570 case PROP_BUFFER_SIZE:
571 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
573 case PROP_TIME_PROVIDER:
574 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
577 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
579 case PROP_TRANSPORT_MODE:
580 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
582 case PROP_STOP_ON_DISCONNECT:
583 gst_rtsp_media_set_stop_on_disconnect (media,
584 g_value_get_boolean (value));
587 gst_rtsp_media_set_clock (media, g_value_get_object (value));
590 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
597 gboolean complete_streams_only;
599 } DoQueryPositionData;
602 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
606 if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream))
608 GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
612 if (gst_rtsp_stream_query_position (stream, &tmp)) {
613 data->position = MIN (data->position, tmp);
617 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
618 GST_TIME_ARGS (data->position));
622 default_query_position (GstRTSPMedia * media, gint64 * position)
624 GstRTSPMediaPrivate *priv;
625 DoQueryPositionData data;
629 data.position = G_MAXINT64;
632 /* if the media is complete, i.e. one or more streams have been configured
633 * with sinks, then we want to query the position on those streams only.
634 * a query on an incmplete stream may return a position that originates from
635 * an earlier preroll */
636 if (check_complete (media))
637 data.complete_streams_only = TRUE;
639 data.complete_streams_only = FALSE;
641 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
644 *position = GST_CLOCK_TIME_NONE;
646 *position = data.position;
658 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
662 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
663 data->stop = MAX (data->stop, tmp);
669 default_query_stop (GstRTSPMedia * media, gint64 * stop)
671 GstRTSPMediaPrivate *priv;
672 DoQueryStopData data;
679 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
687 default_create_rtpbin (GstRTSPMedia * media)
691 rtpbin = gst_element_factory_make ("rtpbin", NULL);
697 is_receive_only (GstRTSPMedia * media)
699 GstRTSPMediaPrivate *priv = media->priv;
700 gboolean recive_only = TRUE;
703 for (i = 0; i < priv->streams->len; i++) {
704 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
705 if (gst_rtsp_stream_is_sender (stream) ||
706 !gst_rtsp_stream_is_receiver (stream)) {
715 /* must be called with state lock */
717 check_seekable (GstRTSPMedia * media)
720 GstRTSPMediaPrivate *priv = media->priv;
722 /* Update the seekable state of the pipeline in case it changed */
723 if (is_receive_only (media)) {
724 /* TODO: Seeking for "receive-only"? */
727 guint i, n = priv->streams->len;
729 for (i = 0; i < n; i++) {
730 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
732 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
733 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
740 query = gst_query_new_seeking (GST_FORMAT_TIME);
741 if (gst_element_query (priv->pipeline, query)) {
746 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
747 priv->seekable = seekable ? G_MAXINT64 : 0;
748 } else if (priv->streams->len) {
749 gboolean seekable = TRUE;
750 guint i, n = priv->streams->len;
752 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
753 for (i = 0; i < n; i++) {
754 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
755 seekable &= gst_rtsp_stream_seekable (stream);
757 priv->seekable = seekable ? G_MAXINT64 : -1;
760 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
762 gst_query_unref (query);
765 /* must be called with state lock */
767 check_complete (GstRTSPMedia * media)
769 GstRTSPMediaPrivate *priv = media->priv;
771 guint i, n = priv->streams->len;
773 for (i = 0; i < n; i++) {
774 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
776 if (gst_rtsp_stream_is_complete (stream))
783 /* must be called with state lock */
785 collect_media_stats (GstRTSPMedia * media)
787 GstRTSPMediaPrivate *priv = media->priv;
788 gint64 position = 0, stop = -1;
790 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
791 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
794 priv->range.unit = GST_RTSP_RANGE_NPT;
796 GST_INFO ("collect media stats");
799 priv->range.min.type = GST_RTSP_TIME_NOW;
800 priv->range.min.seconds = -1;
801 priv->range_start = -1;
802 priv->range.max.type = GST_RTSP_TIME_END;
803 priv->range.max.seconds = -1;
804 priv->range_stop = -1;
806 GstRTSPMediaClass *klass;
809 klass = GST_RTSP_MEDIA_GET_CLASS (media);
811 /* get the position */
813 if (klass->query_position)
814 ret = klass->query_position (media, &position);
817 GST_INFO ("position query failed");
821 /* get the current segment stop */
823 if (klass->query_stop)
824 ret = klass->query_stop (media, &stop);
827 GST_INFO ("stop query failed");
831 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
832 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
834 if (position == -1) {
835 priv->range.min.type = GST_RTSP_TIME_NOW;
836 priv->range.min.seconds = -1;
837 priv->range_start = -1;
839 priv->range.min.type = GST_RTSP_TIME_SECONDS;
840 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
841 priv->range_start = position;
844 priv->range.max.type = GST_RTSP_TIME_END;
845 priv->range.max.seconds = -1;
846 priv->range_stop = -1;
848 priv->range.max.type = GST_RTSP_TIME_SECONDS;
849 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
850 priv->range_stop = stop;
853 check_seekable (media);
858 * gst_rtsp_media_new:
859 * @element: (transfer full): a #GstElement
861 * Create a new #GstRTSPMedia instance. @element is the bin element that
862 * provides the different streams. The #GstRTSPMedia object contains the
863 * element to produce RTP data for one or more related (audio/video/..)
866 * Ownership is taken of @element.
868 * Returns: (transfer full): a new #GstRTSPMedia object.
871 gst_rtsp_media_new (GstElement * element)
873 GstRTSPMedia *result;
875 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
877 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
883 * gst_rtsp_media_get_element:
884 * @media: a #GstRTSPMedia
886 * Get the element that was used when constructing @media.
888 * Returns: (transfer full): a #GstElement. Unref after usage.
891 gst_rtsp_media_get_element (GstRTSPMedia * media)
893 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
895 return gst_object_ref (media->priv->element);
899 * gst_rtsp_media_take_pipeline:
900 * @media: a #GstRTSPMedia
901 * @pipeline: (transfer full): a #GstPipeline
903 * Set @pipeline as the #GstPipeline for @media. Ownership is
904 * taken of @pipeline.
907 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
909 GstRTSPMediaPrivate *priv;
911 GstNetTimeProvider *nettime;
914 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
915 g_return_if_fail (GST_IS_PIPELINE (pipeline));
919 g_mutex_lock (&priv->lock);
920 old = priv->pipeline;
921 priv->pipeline = GST_ELEMENT_CAST (pipeline);
922 nettime = priv->nettime;
923 priv->nettime = NULL;
924 g_mutex_unlock (&priv->lock);
927 gst_object_unref (old);
930 gst_object_unref (nettime);
932 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
934 for (l = priv->pending_pipeline_elements; l; l = l->next) {
935 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
937 g_list_free (priv->pending_pipeline_elements);
938 priv->pending_pipeline_elements = NULL;
942 * gst_rtsp_media_set_permissions:
943 * @media: a #GstRTSPMedia
944 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
946 * Set @permissions on @media.
949 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
950 GstRTSPPermissions * permissions)
952 GstRTSPMediaPrivate *priv;
954 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
958 g_mutex_lock (&priv->lock);
959 if (priv->permissions)
960 gst_rtsp_permissions_unref (priv->permissions);
961 if ((priv->permissions = permissions))
962 gst_rtsp_permissions_ref (permissions);
963 g_mutex_unlock (&priv->lock);
967 * gst_rtsp_media_get_permissions:
968 * @media: a #GstRTSPMedia
970 * Get the permissions object from @media.
972 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
975 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
977 GstRTSPMediaPrivate *priv;
978 GstRTSPPermissions *result;
980 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
984 g_mutex_lock (&priv->lock);
985 if ((result = priv->permissions))
986 gst_rtsp_permissions_ref (result);
987 g_mutex_unlock (&priv->lock);
993 * gst_rtsp_media_set_suspend_mode:
994 * @media: a #GstRTSPMedia
995 * @mode: the new #GstRTSPSuspendMode
997 * Control how @ media will be suspended after the SDP has been generated and
998 * after a PAUSE request has been performed.
1000 * Media must be unprepared when setting the suspend mode.
1003 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
1005 GstRTSPMediaPrivate *priv;
1007 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1011 g_rec_mutex_lock (&priv->state_lock);
1012 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1014 priv->suspend_mode = mode;
1015 g_rec_mutex_unlock (&priv->state_lock);
1022 GST_WARNING ("media %p was prepared", media);
1023 g_rec_mutex_unlock (&priv->state_lock);
1028 * gst_rtsp_media_get_suspend_mode:
1029 * @media: a #GstRTSPMedia
1031 * Get how @media will be suspended.
1033 * Returns: #GstRTSPSuspendMode.
1036 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1038 GstRTSPMediaPrivate *priv;
1039 GstRTSPSuspendMode res;
1041 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1045 g_rec_mutex_lock (&priv->state_lock);
1046 res = priv->suspend_mode;
1047 g_rec_mutex_unlock (&priv->state_lock);
1053 * gst_rtsp_media_set_shared:
1054 * @media: a #GstRTSPMedia
1055 * @shared: the new value
1057 * Set or unset if the pipeline for @media can be shared will multiple clients.
1058 * When @shared is %TRUE, client requests for this media will share the media
1062 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1064 GstRTSPMediaPrivate *priv;
1066 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1070 g_mutex_lock (&priv->lock);
1071 priv->shared = shared;
1072 g_mutex_unlock (&priv->lock);
1076 * gst_rtsp_media_is_shared:
1077 * @media: a #GstRTSPMedia
1079 * Check if the pipeline for @media can be shared between multiple clients.
1081 * Returns: %TRUE if the media can be shared between clients.
1084 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1086 GstRTSPMediaPrivate *priv;
1089 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1093 g_mutex_lock (&priv->lock);
1095 g_mutex_unlock (&priv->lock);
1101 * gst_rtsp_media_set_reusable:
1102 * @media: a #GstRTSPMedia
1103 * @reusable: the new value
1105 * Set or unset if the pipeline for @media can be reused after the pipeline has
1109 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1111 GstRTSPMediaPrivate *priv;
1113 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1117 g_mutex_lock (&priv->lock);
1118 priv->reusable = reusable;
1119 g_mutex_unlock (&priv->lock);
1123 * gst_rtsp_media_is_reusable:
1124 * @media: a #GstRTSPMedia
1126 * Check if the pipeline for @media can be reused after an unprepare.
1128 * Returns: %TRUE if the media can be reused
1131 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1133 GstRTSPMediaPrivate *priv;
1136 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1140 g_mutex_lock (&priv->lock);
1141 res = priv->reusable;
1142 g_mutex_unlock (&priv->lock);
1148 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1150 gst_rtsp_stream_set_profiles (stream, *profiles);
1154 * gst_rtsp_media_set_profiles:
1155 * @media: a #GstRTSPMedia
1156 * @profiles: the new flags
1158 * Configure the allowed lower transport for @media.
1161 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1163 GstRTSPMediaPrivate *priv;
1165 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1169 g_mutex_lock (&priv->lock);
1170 priv->profiles = profiles;
1171 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1172 g_mutex_unlock (&priv->lock);
1176 * gst_rtsp_media_get_profiles:
1177 * @media: a #GstRTSPMedia
1179 * Get the allowed profiles of @media.
1181 * Returns: a #GstRTSPProfile
1184 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1186 GstRTSPMediaPrivate *priv;
1189 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1193 g_mutex_lock (&priv->lock);
1194 res = priv->profiles;
1195 g_mutex_unlock (&priv->lock);
1201 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1203 gst_rtsp_stream_set_protocols (stream, *protocols);
1207 * gst_rtsp_media_set_protocols:
1208 * @media: a #GstRTSPMedia
1209 * @protocols: the new flags
1211 * Configure the allowed lower transport for @media.
1214 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1216 GstRTSPMediaPrivate *priv;
1218 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1222 g_mutex_lock (&priv->lock);
1223 priv->protocols = protocols;
1224 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1225 g_mutex_unlock (&priv->lock);
1229 * gst_rtsp_media_get_protocols:
1230 * @media: a #GstRTSPMedia
1232 * Get the allowed protocols of @media.
1234 * Returns: a #GstRTSPLowerTrans
1237 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1239 GstRTSPMediaPrivate *priv;
1240 GstRTSPLowerTrans res;
1242 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1243 GST_RTSP_LOWER_TRANS_UNKNOWN);
1247 g_mutex_lock (&priv->lock);
1248 res = priv->protocols;
1249 g_mutex_unlock (&priv->lock);
1255 * gst_rtsp_media_set_eos_shutdown:
1256 * @media: a #GstRTSPMedia
1257 * @eos_shutdown: the new value
1259 * Set or unset if an EOS event will be sent to the pipeline for @media before
1263 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1265 GstRTSPMediaPrivate *priv;
1267 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1271 g_mutex_lock (&priv->lock);
1272 priv->eos_shutdown = eos_shutdown;
1273 g_mutex_unlock (&priv->lock);
1277 * gst_rtsp_media_is_eos_shutdown:
1278 * @media: a #GstRTSPMedia
1280 * Check if the pipeline for @media will send an EOS down the pipeline before
1283 * Returns: %TRUE if the media will send EOS before unpreparing.
1286 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1288 GstRTSPMediaPrivate *priv;
1291 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1295 g_mutex_lock (&priv->lock);
1296 res = priv->eos_shutdown;
1297 g_mutex_unlock (&priv->lock);
1303 * gst_rtsp_media_set_buffer_size:
1304 * @media: a #GstRTSPMedia
1305 * @size: the new value
1307 * Set the kernel UDP buffer size.
1310 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1312 GstRTSPMediaPrivate *priv;
1315 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1317 GST_LOG_OBJECT (media, "set buffer size %u", size);
1321 g_mutex_lock (&priv->lock);
1322 priv->buffer_size = size;
1324 for (i = 0; i < priv->streams->len; i++) {
1325 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1326 gst_rtsp_stream_set_buffer_size (stream, size);
1328 g_mutex_unlock (&priv->lock);
1332 * gst_rtsp_media_get_buffer_size:
1333 * @media: a #GstRTSPMedia
1335 * Get the kernel UDP buffer size.
1337 * Returns: the kernel UDP buffer size.
1340 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1342 GstRTSPMediaPrivate *priv;
1345 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1349 g_mutex_lock (&priv->lock);
1350 res = priv->buffer_size;
1351 g_mutex_unlock (&priv->lock);
1357 * gst_rtsp_media_set_stop_on_disconnect:
1358 * @media: a #GstRTSPMedia
1359 * @stop_on_disconnect: the new value
1361 * Set or unset if the pipeline for @media should be stopped when a
1362 * client disconnects without sending TEARDOWN.
1365 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1366 gboolean stop_on_disconnect)
1368 GstRTSPMediaPrivate *priv;
1370 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1374 g_mutex_lock (&priv->lock);
1375 priv->stop_on_disconnect = stop_on_disconnect;
1376 g_mutex_unlock (&priv->lock);
1380 * gst_rtsp_media_is_stop_on_disconnect:
1381 * @media: a #GstRTSPMedia
1383 * Check if the pipeline for @media will be stopped when a client disconnects
1384 * without sending TEARDOWN.
1386 * Returns: %TRUE if the media will be stopped when a client disconnects
1387 * without sending TEARDOWN.
1390 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1392 GstRTSPMediaPrivate *priv;
1395 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1399 g_mutex_lock (&priv->lock);
1400 res = priv->stop_on_disconnect;
1401 g_mutex_unlock (&priv->lock);
1407 * gst_rtsp_media_set_retransmission_time:
1408 * @media: a #GstRTSPMedia
1409 * @time: the new value
1411 * Set the amount of time to store retransmission packets.
1414 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1416 GstRTSPMediaPrivate *priv;
1419 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1421 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1425 g_mutex_lock (&priv->lock);
1426 priv->rtx_time = time;
1427 for (i = 0; i < priv->streams->len; i++) {
1428 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1430 gst_rtsp_stream_set_retransmission_time (stream, time);
1432 g_mutex_unlock (&priv->lock);
1436 * gst_rtsp_media_get_retransmission_time:
1437 * @media: a #GstRTSPMedia
1439 * Get the amount of time to store retransmission data.
1441 * Returns: the amount of time to store retransmission data.
1444 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1446 GstRTSPMediaPrivate *priv;
1449 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1453 g_mutex_lock (&priv->lock);
1454 res = priv->rtx_time;
1455 g_mutex_unlock (&priv->lock);
1461 * gst_rtsp_media_set_do_retransmission:
1463 * Set whether retransmission requests will be sent
1468 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media, gboolean do_retransmission)
1470 GstRTSPMediaPrivate *priv;
1472 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1476 g_mutex_lock (&priv->lock);
1477 priv->do_retransmission = do_retransmission;
1480 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1481 g_mutex_unlock (&priv->lock);
1485 * gst_rtsp_media_get_do_retransmission:
1487 * Returns: Whether retransmission requests will be sent
1492 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1494 GstRTSPMediaPrivate *priv;
1497 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1501 g_mutex_lock (&priv->lock);
1502 res = priv->do_retransmission;
1503 g_mutex_unlock (&priv->lock);
1509 * gst_rtsp_media_set_latency:
1510 * @media: a #GstRTSPMedia
1511 * @latency: latency in milliseconds
1513 * Configure the latency used for receiving media.
1516 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1518 GstRTSPMediaPrivate *priv;
1520 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1522 GST_LOG_OBJECT (media, "set latency %ums", latency);
1526 g_mutex_lock (&priv->lock);
1527 priv->latency = latency;
1529 g_object_set (priv->rtpbin, "latency", latency, NULL);
1530 g_mutex_unlock (&priv->lock);
1534 * gst_rtsp_media_get_latency:
1535 * @media: a #GstRTSPMedia
1537 * Get the latency that is used for receiving media.
1539 * Returns: latency in milliseconds
1542 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1544 GstRTSPMediaPrivate *priv;
1547 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1551 g_mutex_lock (&priv->lock);
1552 res = priv->latency;
1553 g_mutex_unlock (&priv->lock);
1559 * gst_rtsp_media_use_time_provider:
1560 * @media: a #GstRTSPMedia
1561 * @time_provider: if a #GstNetTimeProvider should be used
1563 * Set @media to provide a #GstNetTimeProvider.
1566 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1568 GstRTSPMediaPrivate *priv;
1570 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1574 g_mutex_lock (&priv->lock);
1575 priv->time_provider = time_provider;
1576 g_mutex_unlock (&priv->lock);
1580 * gst_rtsp_media_is_time_provider:
1581 * @media: a #GstRTSPMedia
1583 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1585 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1587 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1590 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1592 GstRTSPMediaPrivate *priv;
1595 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1599 g_mutex_lock (&priv->lock);
1600 res = priv->time_provider;
1601 g_mutex_unlock (&priv->lock);
1607 * gst_rtsp_media_set_clock:
1608 * @media: a #GstRTSPMedia
1609 * @clock: (nullable): #GstClock to be used
1611 * Configure the clock used for the media.
1614 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1616 GstRTSPMediaPrivate *priv;
1618 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1619 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1621 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1625 g_mutex_lock (&priv->lock);
1627 gst_object_unref (priv->clock);
1628 priv->clock = clock ? gst_object_ref (clock) : NULL;
1629 if (priv->pipeline) {
1631 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1633 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1636 g_mutex_unlock (&priv->lock);
1640 * gst_rtsp_media_set_publish_clock_mode:
1641 * @media: a #GstRTSPMedia
1642 * @mode: the clock publish mode
1644 * Sets if and how the media clock should be published according to RFC7273.
1649 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1650 GstRTSPPublishClockMode mode)
1652 GstRTSPMediaPrivate *priv;
1656 g_mutex_lock (&priv->lock);
1657 priv->publish_clock_mode = mode;
1659 n = priv->streams->len;
1660 for (i = 0; i < n; i++) {
1661 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1663 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1665 g_mutex_unlock (&priv->lock);
1669 * gst_rtsp_media_get_publish_clock_mode:
1670 * @media: a #GstRTSPMedia
1672 * Gets if and how the media clock should be published according to RFC7273.
1674 * Returns: The GstRTSPPublishClockMode
1678 GstRTSPPublishClockMode
1679 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1681 GstRTSPMediaPrivate *priv;
1682 GstRTSPPublishClockMode ret;
1685 g_mutex_lock (&priv->lock);
1686 ret = priv->publish_clock_mode;
1687 g_mutex_unlock (&priv->lock);
1693 * gst_rtsp_media_set_address_pool:
1694 * @media: a #GstRTSPMedia
1695 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1697 * configure @pool to be used as the address pool of @media.
1700 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1701 GstRTSPAddressPool * pool)
1703 GstRTSPMediaPrivate *priv;
1704 GstRTSPAddressPool *old;
1706 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1710 GST_LOG_OBJECT (media, "set address pool %p", pool);
1712 g_mutex_lock (&priv->lock);
1713 if ((old = priv->pool) != pool)
1714 priv->pool = pool ? g_object_ref (pool) : NULL;
1717 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1719 g_mutex_unlock (&priv->lock);
1722 g_object_unref (old);
1726 * gst_rtsp_media_get_address_pool:
1727 * @media: a #GstRTSPMedia
1729 * Get the #GstRTSPAddressPool used as the address pool of @media.
1731 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1732 * g_object_unref() after usage.
1734 GstRTSPAddressPool *
1735 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1737 GstRTSPMediaPrivate *priv;
1738 GstRTSPAddressPool *result;
1740 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1744 g_mutex_lock (&priv->lock);
1745 if ((result = priv->pool))
1746 g_object_ref (result);
1747 g_mutex_unlock (&priv->lock);
1753 * gst_rtsp_media_set_multicast_iface:
1754 * @media: a #GstRTSPMedia
1755 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1757 * configure @multicast_iface to be used for @media.
1760 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1761 const gchar * multicast_iface)
1763 GstRTSPMediaPrivate *priv;
1766 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1770 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1772 g_mutex_lock (&priv->lock);
1773 if ((old = priv->multicast_iface) != multicast_iface)
1774 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1777 g_ptr_array_foreach (priv->streams,
1778 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1779 g_mutex_unlock (&priv->lock);
1786 * gst_rtsp_media_get_multicast_iface:
1787 * @media: a #GstRTSPMedia
1789 * Get the multicast interface used for @media.
1791 * Returns: (transfer full) (nullable): the multicast interface for @media.
1792 * g_free() after usage.
1795 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1797 GstRTSPMediaPrivate *priv;
1800 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1804 g_mutex_lock (&priv->lock);
1805 if ((result = priv->multicast_iface))
1806 result = g_strdup (result);
1807 g_mutex_unlock (&priv->lock);
1813 _find_payload_types (GstRTSPMedia * media)
1816 GQueue queue = G_QUEUE_INIT;
1818 n = media->priv->streams->len;
1819 for (i = 0; i < n; i++) {
1820 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1821 guint pt = gst_rtsp_stream_get_pt (stream);
1823 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1830 _next_available_pt (GList * payloads)
1834 for (i = 96; i <= 127; i++) {
1835 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1837 return GPOINTER_TO_UINT (i);
1844 * gst_rtsp_media_collect_streams:
1845 * @media: a #GstRTSPMedia
1847 * Find all payloader elements, they should be named pay\%d in the
1848 * element of @media, and create #GstRTSPStreams for them.
1850 * Collect all dynamic elements, named dynpay\%d, and add them to
1851 * the list of dynamic elements.
1853 * Find all depayloader elements, they should be named depay\%d in the
1854 * element of @media, and create #GstRTSPStreams for them.
1857 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1859 GstRTSPMediaPrivate *priv;
1860 GstElement *element, *elem;
1864 gboolean more_elem_remaining = TRUE;
1865 GstRTSPTransportMode mode = 0;
1867 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1870 element = priv->element;
1873 for (i = 0; more_elem_remaining; i++) {
1876 more_elem_remaining = FALSE;
1878 name = g_strdup_printf ("pay%d", i);
1879 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1881 GST_INFO ("found stream %d with payloader %p", i, elem);
1883 /* take the pad of the payloader */
1884 pad = gst_element_get_static_pad (elem, "src");
1886 /* find the real payload element in case elem is a GstBin */
1887 pay = find_payload_element (elem);
1889 /* create the stream */
1891 GST_WARNING ("could not find real payloader, using bin");
1892 gst_rtsp_media_create_stream (media, elem, pad);
1894 gst_rtsp_media_create_stream (media, pay, pad);
1895 gst_object_unref (pay);
1898 gst_object_unref (pad);
1899 gst_object_unref (elem);
1902 more_elem_remaining = TRUE;
1903 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1907 name = g_strdup_printf ("dynpay%d", i);
1908 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1909 /* a stream that will dynamically create pads to provide RTP packets */
1910 GST_INFO ("found dynamic element %d, %p", i, elem);
1912 g_mutex_lock (&priv->lock);
1913 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1914 g_mutex_unlock (&priv->lock);
1916 priv->nb_dynamic_elements++;
1919 more_elem_remaining = TRUE;
1920 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1924 name = g_strdup_printf ("depay%d", i);
1925 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1926 GST_INFO ("found stream %d with depayloader %p", i, elem);
1928 /* take the pad of the payloader */
1929 pad = gst_element_get_static_pad (elem, "sink");
1930 /* create the stream */
1931 gst_rtsp_media_create_stream (media, elem, pad);
1932 gst_object_unref (pad);
1933 gst_object_unref (elem);
1936 more_elem_remaining = TRUE;
1937 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1943 if (priv->transport_mode != mode)
1944 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1945 priv->transport_mode, mode);
1951 GstElement *appsink, *appsrc;
1952 GstRTSPStream *stream;
1955 static GstFlowReturn
1956 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
1958 AppSinkSrcData *data = user_data;
1962 sample = gst_app_sink_pull_sample (appsink);
1964 return GST_FLOW_FLUSHING;
1967 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
1968 gst_sample_unref (sample);
1972 static GstAppSinkCallbacks appsink_callbacks = {
1978 static GstPadProbeReturn
1979 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
1981 AppSinkSrcData *data = user_data;
1983 if (GST_IS_EVENT (info->data)
1984 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
1985 GstClockTime min, max;
1987 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
1989 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
1990 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
1991 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
1993 } else if (GST_IS_QUERY (info->data)) {
1994 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
1995 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
1996 gst_object_unref (srcpad);
1997 return GST_PAD_PROBE_HANDLED;
1999 gst_object_unref (srcpad);
2002 return GST_PAD_PROBE_OK;
2005 static GstPadProbeReturn
2006 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2008 AppSinkSrcData *data = user_data;
2010 if (GST_IS_QUERY (info->data)) {
2011 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
2012 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
2013 gst_object_unref (sinkpad);
2014 return GST_PAD_PROBE_HANDLED;
2016 gst_object_unref (sinkpad);
2019 return GST_PAD_PROBE_OK;
2023 * gst_rtsp_media_create_stream:
2024 * @media: a #GstRTSPMedia
2025 * @payloader: a #GstElement
2028 * Create a new stream in @media that provides RTP data on @pad.
2029 * @pad should be a pad of an element inside @media->element.
2031 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2035 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2038 GstRTSPMediaPrivate *priv;
2039 GstRTSPStream *stream;
2043 AppSinkSrcData *data = NULL;
2045 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2046 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2047 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2051 g_mutex_lock (&priv->lock);
2052 idx = priv->streams->len;
2054 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
2056 if (GST_PAD_IS_SRC (pad))
2057 name = g_strdup_printf ("src_%u", idx);
2059 name = g_strdup_printf ("sink_%u", idx);
2061 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2062 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2063 GstElement *appsink, *appsrc;
2064 GstPad *sinkpad, *srcpad;
2066 appsink = gst_element_factory_make ("appsink", NULL);
2067 appsrc = gst_element_factory_make ("appsrc", NULL);
2069 if (GST_PAD_IS_SINK (pad)) {
2070 srcpad = gst_element_get_static_pad (appsrc, "src");
2072 gst_bin_add (GST_BIN (priv->element), appsrc);
2074 gst_pad_link (srcpad, pad);
2075 gst_object_unref (srcpad);
2077 streampad = gst_element_get_static_pad (appsink, "sink");
2079 priv->pending_pipeline_elements =
2080 g_list_prepend (priv->pending_pipeline_elements, appsink);
2082 sinkpad = gst_element_get_static_pad (appsink, "sink");
2084 gst_pad_link (pad, sinkpad);
2085 gst_object_unref (sinkpad);
2087 streampad = gst_element_get_static_pad (appsrc, "src");
2089 priv->pending_pipeline_elements =
2090 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2093 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2095 g_object_set (appsink, "sync", FALSE, "async", FALSE, NULL);
2097 data = g_new0 (AppSinkSrcData, 1);
2098 data->appsink = appsink;
2099 data->appsrc = appsrc;
2101 sinkpad = gst_element_get_static_pad (appsink, "sink");
2102 gst_pad_add_probe (sinkpad,
2103 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2104 appsink_pad_probe, data, NULL);
2105 gst_object_unref (sinkpad);
2107 srcpad = gst_element_get_static_pad (appsrc, "src");
2108 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2109 appsrc_pad_probe, data, NULL);
2110 gst_object_unref (srcpad);
2112 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2114 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2117 streampad = gst_ghost_pad_new (name, pad);
2118 gst_pad_set_active (streampad, TRUE);
2119 gst_element_add_pad (priv->element, streampad);
2123 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2125 data->stream = stream;
2127 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2128 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2129 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2130 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2131 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2132 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2133 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2135 g_ptr_array_add (priv->streams, stream);
2137 if (GST_PAD_IS_SRC (pad)) {
2141 g_list_free (priv->payloads);
2142 priv->payloads = _find_payload_types (media);
2144 n = priv->streams->len;
2145 for (i = 0; i < n; i++) {
2146 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2147 guint rtx_pt = _next_available_pt (priv->payloads);
2150 GST_WARNING ("Ran out of space of dynamic payload types");
2154 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2157 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2160 g_mutex_unlock (&priv->lock);
2162 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2169 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2171 GstRTSPMediaPrivate *priv;
2173 AppSinkSrcData *data;
2177 g_mutex_lock (&priv->lock);
2178 /* remove the ghostpad */
2179 srcpad = gst_rtsp_stream_get_srcpad (stream);
2180 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2182 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2183 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2184 else if (GST_OBJECT_PARENT (data->appsrc) ==
2185 GST_OBJECT_CAST (priv->element))
2186 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2187 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2188 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2189 else if (GST_OBJECT_PARENT (data->appsink) ==
2190 GST_OBJECT_CAST (priv->element))
2191 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2193 gst_element_remove_pad (priv->element, srcpad);
2195 gst_object_unref (srcpad);
2196 /* now remove the stream */
2197 g_object_ref (stream);
2198 g_ptr_array_remove (priv->streams, stream);
2199 g_mutex_unlock (&priv->lock);
2201 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2204 g_object_unref (stream);
2208 * gst_rtsp_media_n_streams:
2209 * @media: a #GstRTSPMedia
2211 * Get the number of streams in this media.
2213 * Returns: The number of streams.
2216 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2218 GstRTSPMediaPrivate *priv;
2221 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2225 g_mutex_lock (&priv->lock);
2226 res = priv->streams->len;
2227 g_mutex_unlock (&priv->lock);
2233 * gst_rtsp_media_get_stream:
2234 * @media: a #GstRTSPMedia
2235 * @idx: the stream index
2237 * Retrieve the stream with index @idx from @media.
2239 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2240 * @idx or %NULL when a stream with that index did not exist.
2243 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2245 GstRTSPMediaPrivate *priv;
2248 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2252 g_mutex_lock (&priv->lock);
2253 if (idx < priv->streams->len)
2254 res = g_ptr_array_index (priv->streams, idx);
2257 g_mutex_unlock (&priv->lock);
2263 * gst_rtsp_media_find_stream:
2264 * @media: a #GstRTSPMedia
2265 * @control: the control of the stream
2267 * Find a stream in @media with @control as the control uri.
2269 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2270 * control uri @control or %NULL when a stream with that control did
2274 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2276 GstRTSPMediaPrivate *priv;
2280 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2281 g_return_val_if_fail (control != NULL, NULL);
2287 g_mutex_lock (&priv->lock);
2288 for (i = 0; i < priv->streams->len; i++) {
2289 GstRTSPStream *test;
2291 test = g_ptr_array_index (priv->streams, i);
2292 if (gst_rtsp_stream_has_control (test, control)) {
2297 g_mutex_unlock (&priv->lock);
2302 /* called with state-lock */
2304 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2305 GstRTSPRangeUnit unit)
2307 return gst_rtsp_range_convert_units (range, unit);
2311 * gst_rtsp_media_get_range_string:
2312 * @media: a #GstRTSPMedia
2313 * @play: for the PLAY request
2314 * @unit: the unit to use for the string
2316 * Get the current range as a string. @media must be prepared with
2317 * gst_rtsp_media_prepare ().
2319 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2322 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2323 GstRTSPRangeUnit unit)
2325 GstRTSPMediaClass *klass;
2326 GstRTSPMediaPrivate *priv;
2328 GstRTSPTimeRange range;
2330 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2331 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2332 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2336 g_rec_mutex_lock (&priv->state_lock);
2337 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2338 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2341 g_mutex_lock (&priv->lock);
2343 /* Update the range value with current position/duration */
2344 collect_media_stats (media);
2347 range = priv->range;
2349 if (!play && priv->n_active > 0) {
2350 range.min.type = GST_RTSP_TIME_NOW;
2351 range.min.seconds = -1;
2353 g_mutex_unlock (&priv->lock);
2354 g_rec_mutex_unlock (&priv->state_lock);
2356 if (!klass->convert_range (media, &range, unit))
2357 goto conversion_failed;
2359 result = gst_rtsp_range_to_string (&range);
2366 GST_WARNING ("media %p was not prepared", media);
2367 g_rec_mutex_unlock (&priv->state_lock);
2372 GST_WARNING ("range conversion to unit %d failed", unit);
2378 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2380 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2384 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2386 GstRTSPMediaPrivate *priv = media->priv;
2388 GST_DEBUG ("media %p set blocked %d", media, blocked);
2389 priv->blocked = blocked;
2390 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2394 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2396 gst_rtsp_stream_unblock_linked (stream);
2400 media_unblock_linked (GstRTSPMedia * media)
2402 GstRTSPMediaPrivate *priv = media->priv;
2404 GST_DEBUG ("media %p unblocking linked streams", media);
2405 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2409 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2411 GstRTSPMediaPrivate *priv = media->priv;
2413 g_mutex_lock (&priv->lock);
2414 priv->status = status;
2415 GST_DEBUG ("setting new status to %d", status);
2416 g_cond_broadcast (&priv->cond);
2417 g_mutex_unlock (&priv->lock);
2421 * gst_rtsp_media_get_status:
2422 * @media: a #GstRTSPMedia
2424 * Get the status of @media. When @media is busy preparing, this function waits
2425 * until @media is prepared or in error.
2427 * Returns: the status of @media.
2430 gst_rtsp_media_get_status (GstRTSPMedia * media)
2432 GstRTSPMediaPrivate *priv = media->priv;
2433 GstRTSPMediaStatus result;
2436 g_mutex_lock (&priv->lock);
2437 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2438 /* while we are preparing, wait */
2439 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2440 GST_DEBUG ("waiting for status change");
2441 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2442 GST_DEBUG ("timeout, assuming error status");
2443 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2446 /* could be success or error */
2447 result = priv->status;
2448 GST_DEBUG ("got status %d", result);
2449 g_mutex_unlock (&priv->lock);
2455 * gst_rtsp_media_seek_full:
2456 * @media: a #GstRTSPMedia
2457 * @range: (transfer none): a #GstRTSPTimeRange
2458 * @flags: The minimal set of #GstSeekFlags to use
2460 * Seek the pipeline of @media to @range. @media must be prepared with
2461 * gst_rtsp_media_prepare().
2463 * Returns: %TRUE on success.
2466 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2469 GstRTSPMediaClass *klass;
2470 GstRTSPMediaPrivate *priv;
2472 GstClockTime start, stop;
2473 GstSeekType start_type, stop_type;
2474 gint64 current_position;
2476 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2478 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2479 g_return_val_if_fail (range != NULL, FALSE);
2480 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2484 g_rec_mutex_lock (&priv->state_lock);
2485 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2488 /* check if the media pipeline is complete in order to perform a
2489 * seek operation on it */
2490 if (!check_complete (media))
2493 /* Update the seekable state of the pipeline in case it changed */
2494 check_seekable (media);
2496 if (priv->seekable == 0) {
2497 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2498 " not seekable streams.");
2501 } else if (priv->seekable < 0) {
2505 start_type = stop_type = GST_SEEK_TYPE_NONE;
2507 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2509 gst_rtsp_range_get_times (range, &start, &stop);
2511 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2512 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2513 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2514 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2516 current_position = -1;
2517 if (klass->query_position)
2518 klass->query_position (media, ¤t_position);
2519 GST_INFO ("current media position %" GST_TIME_FORMAT,
2520 GST_TIME_ARGS (current_position));
2522 if (start != GST_CLOCK_TIME_NONE)
2523 start_type = GST_SEEK_TYPE_SET;
2525 if (priv->range_stop == stop)
2526 stop = GST_CLOCK_TIME_NONE;
2527 else if (stop != GST_CLOCK_TIME_NONE)
2528 stop_type = GST_SEEK_TYPE_SET;
2530 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2531 gboolean had_flags = flags != 0;
2533 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2534 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2536 /* depends on the current playing state of the pipeline. We might need to
2537 * queue this until we get EOS. */
2539 flags |= GST_SEEK_FLAG_FLUSH;
2541 flags = GST_SEEK_FLAG_FLUSH;
2544 /* if range start was not supplied we must continue from current position.
2545 * but since we're doing a flushing seek, let us query the current position
2546 * so we end up at exactly the same position after the seek. */
2547 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2548 if (current_position == -1) {
2549 GST_WARNING ("current position unknown");
2551 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2552 GST_TIME_ARGS (current_position));
2553 start = current_position;
2554 start_type = GST_SEEK_TYPE_SET;
2556 flags |= GST_SEEK_FLAG_ACCURATE;
2559 /* only set keyframe flag when modifying start */
2560 if (start_type != GST_SEEK_TYPE_NONE)
2562 flags |= GST_SEEK_FLAG_KEY_UNIT;
2565 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2566 GST_DEBUG ("not seeking because no position change");
2569 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2571 media_streams_set_blocked (media, TRUE);
2573 /* FIXME, we only do forwards playback, no trick modes yet */
2574 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2575 flags, start_type, start, stop_type, stop);
2577 /* and block for the seek to complete */
2578 GST_INFO ("done seeking %d", res);
2582 g_rec_mutex_unlock (&priv->state_lock);
2584 /* wait until pipeline is prerolled again, this will also collect stats */
2585 if (!wait_preroll (media))
2586 goto preroll_failed;
2588 g_rec_mutex_lock (&priv->state_lock);
2589 GST_INFO ("prerolled again");
2592 GST_INFO ("no seek needed");
2595 g_rec_mutex_unlock (&priv->state_lock);
2602 g_rec_mutex_unlock (&priv->state_lock);
2603 GST_INFO ("media %p is not prepared", media);
2608 g_rec_mutex_unlock (&priv->state_lock);
2609 GST_INFO ("pipeline is not complete");
2614 g_rec_mutex_unlock (&priv->state_lock);
2615 GST_INFO ("pipeline is not seekable");
2620 g_rec_mutex_unlock (&priv->state_lock);
2621 GST_WARNING ("conversion to npt not supported");
2626 g_rec_mutex_unlock (&priv->state_lock);
2627 GST_INFO ("seeking failed");
2628 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2633 GST_WARNING ("failed to preroll after seek");
2640 * gst_rtsp_media_seek:
2641 * @media: a #GstRTSPMedia
2642 * @range: (transfer none): a #GstRTSPTimeRange
2644 * Seek the pipeline of @media to @range. @media must be prepared with
2645 * gst_rtsp_media_prepare().
2647 * Returns: %TRUE on success.
2650 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2652 return gst_rtsp_media_seek_full (media, range, 0);
2657 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2659 *blocked &= gst_rtsp_stream_is_blocking (stream);
2663 media_streams_blocking (GstRTSPMedia * media)
2665 gboolean blocking = TRUE;
2667 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2673 static GstStateChangeReturn
2674 set_state (GstRTSPMedia * media, GstState state)
2676 GstRTSPMediaPrivate *priv = media->priv;
2677 GstStateChangeReturn ret;
2679 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2681 ret = gst_element_set_state (priv->pipeline, state);
2686 static GstStateChangeReturn
2687 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2689 GstRTSPMediaPrivate *priv = media->priv;
2690 GstStateChangeReturn ret;
2692 GST_INFO ("set target state to %s for media %p",
2693 gst_element_state_get_name (state), media);
2694 priv->target_state = state;
2696 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2697 priv->target_state, NULL);
2700 ret = set_state (media, state);
2702 ret = GST_STATE_CHANGE_SUCCESS;
2707 /* called with state-lock */
2709 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2711 GstRTSPMediaPrivate *priv = media->priv;
2712 GstMessageType type;
2714 type = GST_MESSAGE_TYPE (message);
2717 case GST_MESSAGE_STATE_CHANGED:
2719 GstState old, new, pending;
2721 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2724 gst_message_parse_state_changed (message, &old, &new, &pending);
2726 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2727 gst_element_state_get_name (old), gst_element_state_get_name (new),
2728 gst_element_state_get_name (pending));
2729 if (priv->no_more_pads_pending == 0 && is_receive_only (media) &&
2730 old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2731 GST_INFO ("%p: went to PAUSED, prepared now", media);
2732 collect_media_stats (media);
2734 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2735 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2740 case GST_MESSAGE_BUFFERING:
2744 gst_message_parse_buffering (message, &percent);
2746 /* no state management needed for live pipelines */
2750 if (percent == 100) {
2751 /* a 100% message means buffering is done */
2752 priv->buffering = FALSE;
2753 /* if the desired state is playing, go back */
2754 if (priv->target_state == GST_STATE_PLAYING) {
2755 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2756 set_state (media, GST_STATE_PLAYING);
2758 GST_INFO ("Buffering done");
2761 /* buffering busy */
2762 if (priv->buffering == FALSE) {
2763 if (priv->target_state == GST_STATE_PLAYING) {
2764 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2765 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2766 set_state (media, GST_STATE_PAUSED);
2768 GST_INFO ("Buffering ...");
2771 priv->buffering = TRUE;
2775 case GST_MESSAGE_LATENCY:
2777 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2780 case GST_MESSAGE_ERROR:
2785 gst_message_parse_error (message, &gerror, &debug);
2786 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2787 g_error_free (gerror);
2790 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2793 case GST_MESSAGE_WARNING:
2798 gst_message_parse_warning (message, &gerror, &debug);
2799 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2800 g_error_free (gerror);
2804 case GST_MESSAGE_ELEMENT:
2806 const GstStructure *s;
2808 s = gst_message_get_structure (message);
2809 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2810 GST_DEBUG ("media received blocking message");
2811 if (priv->blocked && media_streams_blocking (media) &&
2812 priv->no_more_pads_pending == 0) {
2813 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
2814 collect_media_stats (media);
2816 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2817 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2822 case GST_MESSAGE_STREAM_STATUS:
2824 case GST_MESSAGE_ASYNC_DONE:
2825 if (priv->complete) {
2826 /* receive the final ASYNC_DONE, that is posted by the media pipeline
2827 * after all the transport parts have been successfully added to
2828 * the media streams. */
2829 GST_DEBUG_OBJECT (media, "got async-done");
2830 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2831 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2834 case GST_MESSAGE_EOS:
2835 GST_INFO ("%p: got EOS", media);
2837 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2838 GST_DEBUG ("shutting down after EOS");
2839 finish_unprepare (media);
2843 GST_INFO ("%p: got message type %d (%s)", media, type,
2844 gst_message_type_get_name (type));
2851 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2853 GstRTSPMediaPrivate *priv = media->priv;
2854 GstRTSPMediaClass *klass;
2857 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2859 g_rec_mutex_lock (&priv->state_lock);
2860 if (klass->handle_message)
2861 ret = klass->handle_message (media, message);
2864 g_rec_mutex_unlock (&priv->state_lock);
2870 watch_destroyed (GstRTSPMedia * media)
2872 GST_DEBUG_OBJECT (media, "source destroyed");
2873 g_object_unref (media);
2877 find_payload_element (GstElement * payloader)
2879 GstElement *pay = NULL;
2881 if (GST_IS_BIN (payloader)) {
2883 GValue item = { 0 };
2885 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2886 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2887 GstElement *element = (GstElement *) g_value_get_object (&item);
2888 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2892 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2896 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2897 pay = gst_object_ref (element);
2898 g_value_unset (&item);
2901 g_value_unset (&item);
2903 gst_iterator_free (iter);
2905 pay = g_object_ref (payloader);
2911 /* called from streaming threads */
2913 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2915 GstRTSPMediaPrivate *priv = media->priv;
2916 GstRTSPStream *stream;
2919 /* find the real payload element */
2920 pay = find_payload_element (element);
2921 stream = gst_rtsp_media_create_stream (media, pay, pad);
2922 gst_object_unref (pay);
2924 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2926 g_rec_mutex_lock (&priv->state_lock);
2927 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2930 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2932 /* join the element in the PAUSED state because this callback is
2933 * called from the streaming thread and it is PAUSED */
2934 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2935 priv->rtpbin, GST_STATE_PAUSED)) {
2936 GST_WARNING ("failed to join bin element");
2940 gst_rtsp_stream_set_blocked (stream, TRUE);
2942 g_rec_mutex_unlock (&priv->state_lock);
2949 gst_rtsp_media_remove_stream (media, stream);
2950 g_rec_mutex_unlock (&priv->state_lock);
2951 GST_INFO ("ignore pad because we are not preparing");
2957 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2959 GstRTSPMediaPrivate *priv = media->priv;
2960 GstRTSPStream *stream;
2962 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2966 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2968 g_rec_mutex_lock (&priv->state_lock);
2969 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2970 g_rec_mutex_unlock (&priv->state_lock);
2972 gst_rtsp_media_remove_stream (media, stream);
2976 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2978 GstRTSPMediaPrivate *priv = media->priv;
2980 GST_INFO_OBJECT (element, "no more pads");
2981 g_mutex_lock (&priv->lock);
2982 priv->no_more_pads_pending--;
2983 g_mutex_unlock (&priv->lock);
2986 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2988 struct _DynPaySignalHandlers
2990 gulong pad_added_handler;
2991 gulong pad_removed_handler;
2992 gulong no_more_pads_handler;
2996 start_preroll (GstRTSPMedia * media)
2998 GstRTSPMediaPrivate *priv = media->priv;
2999 GstStateChangeReturn ret;
3001 GST_INFO ("setting pipeline to PAUSED for media %p", media);
3003 /* start blocked since it is possible that there are no sink elements yet */
3004 media_streams_set_blocked (media, TRUE);
3005 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3008 case GST_STATE_CHANGE_SUCCESS:
3009 GST_INFO ("SUCCESS state change for media %p", media);
3011 case GST_STATE_CHANGE_ASYNC:
3012 GST_INFO ("ASYNC state change for media %p", media);
3014 case GST_STATE_CHANGE_NO_PREROLL:
3015 /* we need to go to PLAYING */
3016 GST_INFO ("NO_PREROLL state change: live media %p", media);
3017 /* FIXME we disable seeking for live streams for now. We should perform a
3018 * seeking query in preroll instead */
3019 priv->seekable = -1;
3020 priv->is_live = TRUE;
3022 ret = set_state (media, GST_STATE_PLAYING);
3023 if (ret == GST_STATE_CHANGE_FAILURE)
3026 case GST_STATE_CHANGE_FAILURE:
3034 GST_WARNING ("failed to preroll pipeline");
3040 wait_preroll (GstRTSPMedia * media)
3042 GstRTSPMediaStatus status;
3044 GST_DEBUG ("wait to preroll pipeline");
3046 /* wait until pipeline is prerolled */
3047 status = gst_rtsp_media_get_status (media);
3048 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3049 goto preroll_failed;
3055 GST_WARNING ("failed to preroll pipeline");
3061 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3063 GstRTSPMediaPrivate *priv = media->priv;
3064 GstRTSPStream *stream = NULL;
3066 GstElement *res = NULL;
3068 g_mutex_lock (&priv->lock);
3069 for (i = 0; i < priv->streams->len; i++) {
3070 stream = g_ptr_array_index (priv->streams, i);
3072 if (sessid == gst_rtsp_stream_get_index (stream))
3077 g_mutex_unlock (&priv->lock);
3080 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3086 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3088 GstRTSPMediaPrivate *priv = media->priv;
3089 GstRTSPStream *stream = NULL;
3091 GstElement *res = NULL;
3093 g_mutex_lock (&priv->lock);
3094 for (i = 0; i < priv->streams->len; i++) {
3095 stream = g_ptr_array_index (priv->streams, i);
3097 if (sessid == gst_rtsp_stream_get_index (stream))
3102 g_mutex_unlock (&priv->lock);
3105 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3111 start_prepare (GstRTSPMedia * media)
3113 GstRTSPMediaPrivate *priv = media->priv;
3117 g_rec_mutex_lock (&priv->state_lock);
3118 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3119 goto no_longer_preparing;
3121 /* link streams we already have, other streams might appear when we have
3122 * dynamic elements */
3123 for (i = 0; i < priv->streams->len; i++) {
3124 GstRTSPStream *stream;
3126 stream = g_ptr_array_index (priv->streams, i);
3128 if (priv->rtx_time > 0) {
3129 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3130 g_signal_connect (priv->rtpbin, "request-aux-sender",
3131 (GCallback) request_aux_sender, media);
3134 if (priv->do_retransmission) {
3135 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3136 (GCallback) request_aux_receiver, media);
3139 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3140 priv->rtpbin, GST_STATE_NULL)) {
3141 goto join_bin_failed;
3146 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission, NULL);
3148 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3149 GstElement *elem = walk->data;
3150 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3152 GST_INFO ("adding callbacks for dynamic element %p", elem);
3154 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3155 (GCallback) pad_added_cb, media);
3156 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3157 (GCallback) pad_removed_cb, media);
3158 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3159 (GCallback) no_more_pads_cb, media);
3161 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3164 if (priv->nb_dynamic_elements == 0 && is_receive_only (media)) {
3165 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3166 * a second ASYNC state change failing */
3167 priv->is_live = TRUE;
3168 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3169 } else if (!start_preroll (media)) {
3170 goto preroll_failed;
3173 g_rec_mutex_unlock (&priv->state_lock);
3177 no_longer_preparing:
3179 GST_INFO ("media is no longer preparing");
3180 g_rec_mutex_unlock (&priv->state_lock);
3185 GST_WARNING ("failed to join bin element");
3186 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3187 g_rec_mutex_unlock (&priv->state_lock);
3192 GST_WARNING ("failed to preroll pipeline");
3193 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3194 g_rec_mutex_unlock (&priv->state_lock);
3200 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3202 GstRTSPMediaPrivate *priv;
3203 GstRTSPMediaClass *klass;
3205 GMainContext *context;
3210 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3212 if (!klass->create_rtpbin)
3213 goto no_create_rtpbin;
3215 priv->rtpbin = klass->create_rtpbin (media);
3216 if (priv->rtpbin != NULL) {
3217 gboolean success = TRUE;
3219 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3221 if (klass->setup_rtpbin)
3222 success = klass->setup_rtpbin (media, priv->rtpbin);
3224 if (success == FALSE) {
3225 gst_object_unref (priv->rtpbin);
3226 priv->rtpbin = NULL;
3229 if (priv->rtpbin == NULL)
3232 priv->thread = thread;
3233 context = (thread != NULL) ? (thread->context) : NULL;
3235 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3237 /* add the pipeline bus to our custom mainloop */
3238 priv->source = gst_bus_create_watch (bus);
3239 gst_object_unref (bus);
3241 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3242 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3244 priv->id = g_source_attach (priv->source, context);
3246 /* add stuff to the bin */
3247 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3249 /* do remainder in context */
3250 source = g_idle_source_new ();
3251 g_source_set_callback (source, (GSourceFunc) start_prepare,
3252 g_object_ref (media), (GDestroyNotify) g_object_unref);
3253 g_source_attach (source, context);
3254 g_source_unref (source);
3261 GST_ERROR ("no create_rtpbin function");
3262 g_critical ("no create_rtpbin vmethod function set");
3267 GST_WARNING ("no rtpbin element");
3268 g_warning ("failed to create element 'rtpbin', check your installation");
3274 * gst_rtsp_media_prepare:
3275 * @media: a #GstRTSPMedia
3276 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3277 * bus handler or %NULL
3279 * Prepare @media for streaming. This function will create the objects
3280 * to manage the streaming. A pipeline must have been set on @media with
3281 * gst_rtsp_media_take_pipeline().
3283 * It will preroll the pipeline and collect vital information about the streams
3284 * such as the duration.
3286 * Returns: %TRUE on success.
3289 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3291 GstRTSPMediaPrivate *priv;
3292 GstRTSPMediaClass *klass;
3294 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3298 g_rec_mutex_lock (&priv->state_lock);
3299 priv->prepare_count++;
3301 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3302 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3305 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3308 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3309 goto not_unprepared;
3311 if (!priv->reusable && priv->reused)
3314 GST_INFO ("preparing media %p", media);
3316 /* reset some variables */
3317 priv->is_live = FALSE;
3318 priv->seekable = -1;
3319 priv->buffering = FALSE;
3320 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3322 /* we're preparing now */
3323 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3325 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3326 if (klass->prepare) {
3327 if (!klass->prepare (media, thread))
3328 goto prepare_failed;
3332 g_rec_mutex_unlock (&priv->state_lock);
3334 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3335 * able to do this async so that we don't block the server thread. */
3336 if (!wait_preroll (media))
3337 goto preroll_failed;
3339 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3341 GST_INFO ("object %p is prerolled", media);
3348 /* we are not going to use the giving thread, so stop it. */
3350 gst_rtsp_thread_stop (thread);
3355 GST_LOG ("media %p was prepared", media);
3356 /* we are not going to use the giving thread, so stop it. */
3358 gst_rtsp_thread_stop (thread);
3359 g_rec_mutex_unlock (&priv->state_lock);
3365 /* we are not going to use the giving thread, so stop it. */
3367 gst_rtsp_thread_stop (thread);
3368 GST_WARNING ("media %p was not unprepared", media);
3369 priv->prepare_count--;
3370 g_rec_mutex_unlock (&priv->state_lock);
3375 /* we are not going to use the giving thread, so stop it. */
3377 gst_rtsp_thread_stop (thread);
3378 priv->prepare_count--;
3379 g_rec_mutex_unlock (&priv->state_lock);
3380 GST_WARNING ("can not reuse media %p", media);
3385 /* we are not going to use the giving thread, so stop it. */
3387 gst_rtsp_thread_stop (thread);
3388 priv->prepare_count--;
3389 g_rec_mutex_unlock (&priv->state_lock);
3390 GST_ERROR ("failed to prepare media");
3395 GST_WARNING ("failed to preroll pipeline");
3396 gst_rtsp_media_unprepare (media);
3401 /* must be called with state-lock */
3403 finish_unprepare (GstRTSPMedia * media)
3405 GstRTSPMediaPrivate *priv = media->priv;
3409 GST_DEBUG ("shutting down");
3411 /* release the lock on shutdown, otherwise pad_added_cb might try to
3412 * acquire the lock and then we deadlock */
3413 g_rec_mutex_unlock (&priv->state_lock);
3414 set_state (media, GST_STATE_NULL);
3415 g_rec_mutex_lock (&priv->state_lock);
3417 media_streams_set_blocked (media, FALSE);
3419 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
3422 for (i = 0; i < priv->streams->len; i++) {
3423 GstRTSPStream *stream;
3425 GST_INFO ("Removing elements of stream %d from pipeline", i);
3427 stream = g_ptr_array_index (priv->streams, i);
3429 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3432 /* remove the pad signal handlers */
3433 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3434 GstElement *elem = walk->data;
3435 DynPaySignalHandlers *handlers;
3438 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3439 g_assert (handlers != NULL);
3441 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3442 g_signal_handler_disconnect (G_OBJECT (elem),
3443 handlers->pad_removed_handler);
3444 g_signal_handler_disconnect (G_OBJECT (elem),
3445 handlers->no_more_pads_handler);
3447 g_slice_free (DynPaySignalHandlers, handlers);
3450 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3451 priv->rtpbin = NULL;
3454 gst_object_unref (priv->nettime);
3455 priv->nettime = NULL;
3457 priv->reused = TRUE;
3458 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3460 /* when the media is not reusable, this will effectively unref the media and
3462 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3464 /* the source has the last ref to the media */
3466 GST_DEBUG ("destroy source");
3467 g_source_destroy (priv->source);
3468 g_source_unref (priv->source);
3471 GST_DEBUG ("stop thread");
3472 gst_rtsp_thread_stop (priv->thread);
3476 /* called with state-lock */
3478 default_unprepare (GstRTSPMedia * media)
3480 GstRTSPMediaPrivate *priv = media->priv;
3482 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3484 if (priv->eos_shutdown) {
3485 GST_DEBUG ("sending EOS for shutdown");
3486 /* ref so that we don't disappear */
3487 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3488 /* we need to go to playing again for the EOS to propagate, normally in this
3489 * state, nothing is receiving data from us anymore so this is ok. */
3490 set_state (media, GST_STATE_PLAYING);
3492 finish_unprepare (media);
3498 * gst_rtsp_media_unprepare:
3499 * @media: a #GstRTSPMedia
3501 * Unprepare @media. After this call, the media should be prepared again before
3502 * it can be used again. If the media is set to be non-reusable, a new instance
3505 * Returns: %TRUE on success.
3508 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3510 GstRTSPMediaPrivate *priv;
3513 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3517 g_rec_mutex_lock (&priv->state_lock);
3518 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3519 goto was_unprepared;
3521 priv->prepare_count--;
3522 if (priv->prepare_count > 0)
3525 GST_INFO ("unprepare media %p", media);
3526 set_target_state (media, GST_STATE_NULL, FALSE);
3529 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3530 GstRTSPMediaClass *klass;
3532 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3533 if (klass->unprepare)
3534 success = klass->unprepare (media);
3536 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3537 finish_unprepare (media);
3539 g_rec_mutex_unlock (&priv->state_lock);
3545 g_rec_mutex_unlock (&priv->state_lock);
3546 GST_INFO ("media %p was already unprepared", media);
3551 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3552 g_rec_mutex_unlock (&priv->state_lock);
3557 /* should be called with state-lock */
3559 get_clock_unlocked (GstRTSPMedia * media)
3561 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3562 GST_DEBUG_OBJECT (media, "media was not prepared");
3565 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3569 * gst_rtsp_media_get_clock:
3570 * @media: a #GstRTSPMedia
3572 * Get the clock that is used by the pipeline in @media.
3574 * @media must be prepared before this method returns a valid clock object.
3576 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3579 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3582 GstRTSPMediaPrivate *priv;
3584 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3588 g_rec_mutex_lock (&priv->state_lock);
3589 clock = get_clock_unlocked (media);
3590 g_rec_mutex_unlock (&priv->state_lock);
3596 * gst_rtsp_media_get_base_time:
3597 * @media: a #GstRTSPMedia
3599 * Get the base_time that is used by the pipeline in @media.
3601 * @media must be prepared before this method returns a valid base_time.
3603 * Returns: the base_time used by @media.
3606 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3608 GstClockTime result;
3609 GstRTSPMediaPrivate *priv;
3611 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3615 g_rec_mutex_lock (&priv->state_lock);
3616 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3619 result = gst_element_get_base_time (media->priv->pipeline);
3620 g_rec_mutex_unlock (&priv->state_lock);
3627 g_rec_mutex_unlock (&priv->state_lock);
3628 GST_DEBUG_OBJECT (media, "media was not prepared");
3629 return GST_CLOCK_TIME_NONE;
3634 * gst_rtsp_media_get_time_provider:
3635 * @media: a #GstRTSPMedia
3636 * @address: (allow-none): an address or %NULL
3637 * @port: a port or 0
3639 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3640 * will listen on @address and @port for client time requests.
3642 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3644 GstNetTimeProvider *
3645 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3648 GstRTSPMediaPrivate *priv;
3649 GstNetTimeProvider *provider = NULL;
3651 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3655 g_rec_mutex_lock (&priv->state_lock);
3656 if (priv->time_provider) {
3657 if ((provider = priv->nettime) == NULL) {
3660 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3661 provider = gst_net_time_provider_new (clock, address, port);
3662 gst_object_unref (clock);
3664 priv->nettime = provider;
3668 g_rec_mutex_unlock (&priv->state_lock);
3671 gst_object_ref (provider);
3677 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3679 return gst_rtsp_sdp_from_media (sdp, info, media);
3683 * gst_rtsp_media_setup_sdp:
3684 * @media: a #GstRTSPMedia
3685 * @sdp: (transfer none): a #GstSDPMessage
3686 * @info: (transfer none): a #GstSDPInfo
3688 * Add @media specific info to @sdp. @info is used to configure the connection
3689 * information in the SDP.
3691 * Returns: TRUE on success.
3694 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3697 GstRTSPMediaPrivate *priv;
3698 GstRTSPMediaClass *klass;
3701 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3702 g_return_val_if_fail (sdp != NULL, FALSE);
3703 g_return_val_if_fail (info != NULL, FALSE);
3707 g_rec_mutex_lock (&priv->state_lock);
3709 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3711 if (!klass->setup_sdp)
3714 res = klass->setup_sdp (media, sdp, info);
3716 g_rec_mutex_unlock (&priv->state_lock);
3723 g_rec_mutex_unlock (&priv->state_lock);
3724 GST_ERROR ("no setup_sdp function");
3725 g_critical ("no setup_sdp vmethod function set");
3731 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3733 GstRTSPMediaPrivate *priv = media->priv;
3736 medias_len = gst_sdp_message_medias_len (sdp);
3737 if (medias_len != priv->streams->len) {
3738 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3739 priv->streams->len, medias_len);
3743 for (i = 0; i < medias_len; i++) {
3745 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3746 GstRTSPStream *stream;
3747 gint j, formats_len;
3748 const gchar *control;
3749 GstRTSPProfile profile, profiles;
3751 stream = g_ptr_array_index (priv->streams, i);
3753 /* TODO: Should we do something with the other SDP information? */
3756 proto = gst_sdp_media_get_proto (sdp_media);
3757 if (proto == NULL) {
3758 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3762 if (g_str_equal (proto, "RTP/AVP")) {
3763 profile = GST_RTSP_PROFILE_AVP;
3764 } else if (g_str_equal (proto, "RTP/SAVP")) {
3765 profile = GST_RTSP_PROFILE_SAVP;
3766 } else if (g_str_equal (proto, "RTP/AVPF")) {
3767 profile = GST_RTSP_PROFILE_AVPF;
3768 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3769 profile = GST_RTSP_PROFILE_SAVPF;
3771 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3775 profiles = gst_rtsp_stream_get_profiles (stream);
3776 if ((profiles & profile) == 0) {
3777 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3781 formats_len = gst_sdp_media_formats_len (sdp_media);
3782 for (j = 0; j < formats_len; j++) {
3787 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3789 GST_DEBUG (" looking at %d pt: %d", j, pt);
3792 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
3794 GST_WARNING (" skipping pt %d without caps", pt);
3798 /* do some tweaks */
3799 GST_DEBUG ("mapping sdp session level attributes to caps");
3800 gst_sdp_message_attributes_to_caps (sdp, caps);
3801 GST_DEBUG ("mapping sdp media level attributes to caps");
3802 gst_sdp_media_attributes_to_caps (sdp_media, caps);
3804 s = gst_caps_get_structure (caps, 0);
3805 gst_structure_set_name (s, "application/x-rtp");
3807 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3808 gst_caps_unref (caps);
3811 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3813 gst_rtsp_stream_set_control (stream, control);
3821 * gst_rtsp_media_handle_sdp:
3822 * @media: a #GstRTSPMedia
3823 * @sdp: (transfer none): a #GstSDPMessage
3825 * Configure an SDP on @media for receiving streams
3827 * Returns: TRUE on success.
3830 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3832 GstRTSPMediaPrivate *priv;
3833 GstRTSPMediaClass *klass;
3836 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3837 g_return_val_if_fail (sdp != NULL, FALSE);
3841 g_rec_mutex_lock (&priv->state_lock);
3843 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3845 if (!klass->handle_sdp)
3848 res = klass->handle_sdp (media, sdp);
3850 g_rec_mutex_unlock (&priv->state_lock);
3857 g_rec_mutex_unlock (&priv->state_lock);
3858 GST_ERROR ("no handle_sdp function");
3859 g_critical ("no handle_sdp vmethod function set");
3865 do_set_seqnum (GstRTSPStream * stream)
3868 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3869 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3872 /* call with state_lock */
3874 default_suspend (GstRTSPMedia * media)
3876 GstRTSPMediaPrivate *priv = media->priv;
3877 GstStateChangeReturn ret;
3879 switch (priv->suspend_mode) {
3880 case GST_RTSP_SUSPEND_MODE_NONE:
3881 GST_DEBUG ("media %p no suspend", media);
3883 case GST_RTSP_SUSPEND_MODE_PAUSE:
3884 GST_DEBUG ("media %p suspend to PAUSED", media);
3885 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3886 if (ret == GST_STATE_CHANGE_FAILURE)
3889 case GST_RTSP_SUSPEND_MODE_RESET:
3890 GST_DEBUG ("media %p suspend to NULL", media);
3891 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3892 if (ret == GST_STATE_CHANGE_FAILURE)
3894 /* Because payloader needs to set the sequence number as
3895 * monotonic, we need to preserve the sequence number
3896 * after pause. (otherwise going from pause to play, which
3897 * is actually from NULL to PLAY will create a new sequence
3899 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3910 GST_WARNING ("failed changing pipeline's state for media %p", media);
3916 * gst_rtsp_media_suspend:
3917 * @media: a #GstRTSPMedia
3919 * Suspend @media. The state of the pipeline managed by @media is set to
3920 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3921 * with gst_rtsp_media_unsuspend()
3923 * @media must be prepared with gst_rtsp_media_prepare();
3925 * Returns: %TRUE on success.
3928 gst_rtsp_media_suspend (GstRTSPMedia * media)
3930 GstRTSPMediaPrivate *priv = media->priv;
3931 GstRTSPMediaClass *klass;
3933 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3935 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3937 g_rec_mutex_lock (&priv->state_lock);
3938 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3941 /* don't attempt to suspend when something is busy */
3942 if (priv->n_active > 0)
3945 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3946 if (klass->suspend) {
3947 if (!klass->suspend (media))
3948 goto suspend_failed;
3951 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3953 g_rec_mutex_unlock (&priv->state_lock);
3960 g_rec_mutex_unlock (&priv->state_lock);
3961 GST_WARNING ("media %p was not prepared", media);
3966 g_rec_mutex_unlock (&priv->state_lock);
3967 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3968 GST_WARNING ("failed to suspend media %p", media);
3973 /* call with state_lock */
3975 default_unsuspend (GstRTSPMedia * media)
3977 GstRTSPMediaPrivate *priv = media->priv;
3978 gboolean preroll_ok;
3980 switch (priv->suspend_mode) {
3981 case GST_RTSP_SUSPEND_MODE_NONE:
3982 if (is_receive_only (media))
3984 if (media_streams_blocking (media)) {
3985 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3986 /* at this point the media pipeline has been updated and contain all
3987 * specific transport parts: all active streams contain at least one sink
3988 * element and it's safe to unblock any blocked streams that are active */
3989 media_unblock_linked (media);
3991 /* streams are not blocked and media is suspended from PAUSED */
3992 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3994 g_rec_mutex_unlock (&priv->state_lock);
3995 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
3996 g_rec_mutex_lock (&priv->state_lock);
3997 goto preroll_failed;
3999 g_rec_mutex_lock (&priv->state_lock);
4001 case GST_RTSP_SUSPEND_MODE_PAUSE:
4002 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4004 case GST_RTSP_SUSPEND_MODE_RESET:
4006 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4007 /* at this point the media pipeline has been updated and contain all
4008 * specific transport parts: all active streams contain at least one sink
4009 * element and it's safe to unblock any blocked streams that are active */
4010 media_unblock_linked (media);
4011 if (!start_preroll (media))
4014 g_rec_mutex_unlock (&priv->state_lock);
4015 preroll_ok = wait_preroll (media);
4016 g_rec_mutex_lock (&priv->state_lock);
4019 goto preroll_failed;
4030 GST_WARNING ("failed to preroll pipeline");
4035 GST_WARNING ("failed to preroll pipeline");
4041 * gst_rtsp_media_unsuspend:
4042 * @media: a #GstRTSPMedia
4044 * Unsuspend @media if it was in a suspended state. This method does nothing
4045 * when the media was not in the suspended state.
4047 * Returns: %TRUE on success.
4050 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4052 GstRTSPMediaPrivate *priv = media->priv;
4053 GstRTSPMediaClass *klass;
4055 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4057 g_rec_mutex_lock (&priv->state_lock);
4058 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4061 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4062 if (klass->unsuspend) {
4063 if (!klass->unsuspend (media))
4064 goto unsuspend_failed;
4068 g_rec_mutex_unlock (&priv->state_lock);
4075 g_rec_mutex_unlock (&priv->state_lock);
4076 GST_WARNING ("failed to unsuspend media %p", media);
4077 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4082 /* must be called with state-lock */
4084 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4086 GstRTSPMediaPrivate *priv = media->priv;
4088 if (state == GST_STATE_NULL) {
4089 gst_rtsp_media_unprepare (media);
4091 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4092 set_target_state (media, state, FALSE);
4093 /* when we are buffering, don't update the state yet, this will be done
4094 * when buffering finishes */
4095 if (priv->buffering) {
4096 GST_INFO ("Buffering busy, delay state change");
4098 if (state == GST_STATE_PLAYING)
4099 /* make sure pads are not blocking anymore when going to PLAYING */
4100 media_unblock_linked (media);
4102 set_state (media, state);
4104 /* and suspend after pause */
4105 if (state == GST_STATE_PAUSED)
4106 gst_rtsp_media_suspend (media);
4112 * gst_rtsp_media_set_pipeline_state:
4113 * @media: a #GstRTSPMedia
4114 * @state: the target state of the pipeline
4116 * Set the state of the pipeline managed by @media to @state
4119 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4121 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4123 g_rec_mutex_lock (&media->priv->state_lock);
4124 media_set_pipeline_state_locked (media, state);
4125 g_rec_mutex_unlock (&media->priv->state_lock);
4129 * gst_rtsp_media_set_state:
4130 * @media: a #GstRTSPMedia
4131 * @state: the target state of the media
4132 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4133 * a #GPtrArray of #GstRTSPStreamTransport pointers
4135 * Set the state of @media to @state and for the transports in @transports.
4137 * @media must be prepared with gst_rtsp_media_prepare();
4139 * Returns: %TRUE on success.
4142 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4143 GPtrArray * transports)
4145 GstRTSPMediaPrivate *priv;
4147 gboolean activate, deactivate, do_state;
4150 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4151 g_return_val_if_fail (transports != NULL, FALSE);
4155 g_rec_mutex_lock (&priv->state_lock);
4156 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4158 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4159 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4162 /* NULL and READY are the same */
4163 if (state == GST_STATE_READY)
4164 state = GST_STATE_NULL;
4166 activate = deactivate = FALSE;
4168 GST_INFO ("going to state %s media %p, target state %s",
4169 gst_element_state_get_name (state), media,
4170 gst_element_state_get_name (priv->target_state));
4173 case GST_STATE_NULL:
4174 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4175 if (priv->target_state >= GST_STATE_PAUSED)
4178 case GST_STATE_PAUSED:
4179 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4180 if (priv->target_state == GST_STATE_PLAYING)
4183 case GST_STATE_PLAYING:
4184 /* we're going to PLAYING, activate */
4190 old_active = priv->n_active;
4192 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4193 activate, deactivate);
4194 for (i = 0; i < transports->len; i++) {
4195 GstRTSPStreamTransport *trans;
4197 /* we need a non-NULL entry in the array */
4198 trans = g_ptr_array_index (transports, i);
4203 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4205 } else if (deactivate) {
4206 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4211 /* we just activated the first media, do the playing state change */
4212 if (old_active == 0 && activate)
4214 /* if we have no more active media, do the downward state changes */
4215 else if (priv->n_active == 0)
4220 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4223 if (priv->target_state != state) {
4225 media_set_pipeline_state_locked (media, state);
4226 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4231 /* remember where we are */
4232 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4233 old_active != priv->n_active))
4234 collect_media_stats (media);
4236 g_rec_mutex_unlock (&priv->state_lock);
4243 GST_WARNING ("media %p was not prepared", media);
4244 g_rec_mutex_unlock (&priv->state_lock);
4249 GST_WARNING ("media %p in error status while changing to state %d",
4251 if (state == GST_STATE_NULL) {
4252 for (i = 0; i < transports->len; i++) {
4253 GstRTSPStreamTransport *trans;
4255 /* we need a non-NULL entry in the array */
4256 trans = g_ptr_array_index (transports, i);
4260 gst_rtsp_stream_transport_set_active (trans, FALSE);
4264 g_rec_mutex_unlock (&priv->state_lock);
4270 * gst_rtsp_media_set_transport_mode:
4271 * @media: a #GstRTSPMedia
4272 * @mode: the new value
4274 * Sets if the media pipeline can work in PLAY or RECORD mode
4277 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4278 GstRTSPTransportMode mode)
4280 GstRTSPMediaPrivate *priv;
4282 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4286 g_mutex_lock (&priv->lock);
4287 priv->transport_mode = mode;
4288 g_mutex_unlock (&priv->lock);
4292 * gst_rtsp_media_get_transport_mode:
4293 * @media: a #GstRTSPMedia
4295 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4297 * Returns: The transport mode.
4299 GstRTSPTransportMode
4300 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4302 GstRTSPMediaPrivate *priv;
4303 GstRTSPTransportMode res;
4305 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4309 g_mutex_lock (&priv->lock);
4310 res = priv->transport_mode;
4311 g_mutex_unlock (&priv->lock);
4317 * gst_rtsp_media_get_seekable:
4318 * @media: a #GstRTSPMedia
4320 * Check if the pipeline for @media seek and up to what point in time,
4323 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4324 * and > 0 to indicate the longest duration between any two random access points.
4325 * %G_MAXINT64 means any value is possible.
4328 gst_rtsp_media_seekable (GstRTSPMedia * media)
4330 GstRTSPMediaPrivate *priv;
4331 GstClockTimeDiff res;
4333 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4337 /* Currently we are not able to seek on live streams,
4338 * and no stream is seekable only to the beginning */
4339 g_mutex_lock (&priv->lock);
4340 res = priv->seekable;
4341 g_mutex_unlock (&priv->lock);
4347 * gst_rtsp_media_complete_pipeline:
4348 * @media: a #GstRTSPMedia
4349 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4351 * Add a receiver and sender parts to the pipeline based on the transport from
4354 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4357 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4359 GstRTSPMediaPrivate *priv;
4362 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4363 g_return_val_if_fail (transports, FALSE);
4365 GST_DEBUG_OBJECT (media, "complete pipeline");
4369 g_mutex_lock (&priv->lock);
4370 for (i = 0; i < priv->streams->len; i++) {
4371 GstRTSPStreamTransport *transport;
4372 GstRTSPStream *stream;
4373 const GstRTSPTransport *rtsp_transport;
4375 transport = g_ptr_array_index (transports, i);
4379 stream = gst_rtsp_stream_transport_get_stream (transport);
4383 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4385 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4386 g_mutex_unlock (&priv->lock);
4391 priv->complete = TRUE;
4392 g_mutex_unlock (&priv->lock);