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 };
232 #define C_ENUM(v) ((gint) v)
235 gst_rtsp_suspend_mode_get_type (void)
238 static const GEnumValue values[] = {
239 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
240 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
242 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
247 if (g_once_init_enter (&id)) {
248 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
249 g_once_init_leave (&id, tmp);
254 #define C_FLAGS(v) ((guint) v)
257 gst_rtsp_transport_mode_get_type (void)
260 static const GFlagsValue values[] = {
261 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
263 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
268 if (g_once_init_enter (&id)) {
269 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
270 g_once_init_leave (&id, tmp);
276 gst_rtsp_publish_clock_mode_get_type (void)
279 static const GEnumValue values[] = {
280 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
281 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
282 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
283 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
285 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
286 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
291 if (g_once_init_enter (&id)) {
292 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
293 g_once_init_leave (&id, tmp);
298 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
301 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
303 GObjectClass *gobject_class;
305 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
307 gobject_class = G_OBJECT_CLASS (klass);
309 gobject_class->get_property = gst_rtsp_media_get_property;
310 gobject_class->set_property = gst_rtsp_media_set_property;
311 gobject_class->finalize = gst_rtsp_media_finalize;
313 g_object_class_install_property (gobject_class, PROP_SHARED,
314 g_param_spec_boolean ("shared", "Shared",
315 "If this media pipeline can be shared", DEFAULT_SHARED,
316 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
318 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
319 g_param_spec_enum ("suspend-mode", "Suspend Mode",
320 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
321 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
323 g_object_class_install_property (gobject_class, PROP_REUSABLE,
324 g_param_spec_boolean ("reusable", "Reusable",
325 "If this media pipeline can be reused after an unprepare",
326 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
328 g_object_class_install_property (gobject_class, PROP_PROFILES,
329 g_param_spec_flags ("profiles", "Profiles",
330 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
331 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
333 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
334 g_param_spec_flags ("protocols", "Protocols",
335 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
336 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
339 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
340 "Send an EOS event to the pipeline before unpreparing",
341 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
344 g_param_spec_uint ("buffer-size", "Buffer Size",
345 "The kernel UDP buffer size to use", 0, G_MAXUINT,
346 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
348 g_object_class_install_property (gobject_class, PROP_ELEMENT,
349 g_param_spec_object ("element", "The Element",
350 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
351 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
353 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
354 g_param_spec_boolean ("time-provider", "Time Provider",
355 "Use a NetTimeProvider for clients",
356 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
358 g_object_class_install_property (gobject_class, PROP_LATENCY,
359 g_param_spec_uint ("latency", "Latency",
360 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
361 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
363 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
364 g_param_spec_flags ("transport-mode", "Transport Mode",
365 "If this media pipeline can be used for PLAY or RECORD",
366 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
367 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
369 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
370 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
371 "If this media pipeline should be stopped "
372 "when a client disconnects without TEARDOWN",
373 DEFAULT_STOP_ON_DISCONNECT,
374 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_CLOCK,
377 g_param_spec_object ("clock", "Clock",
378 "Clock to be used by the media pipeline",
379 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
381 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
382 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
383 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
384 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
386 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
387 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
388 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
389 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
390 GST_TYPE_RTSP_STREAM);
392 gst_rtsp_media_signals[SIGNAL_PREPARED] =
393 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
394 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
395 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
397 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
398 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
399 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
400 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
402 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
403 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
404 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
405 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
407 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
408 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
409 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
410 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
412 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
414 klass->handle_message = default_handle_message;
415 klass->prepare = default_prepare;
416 klass->unprepare = default_unprepare;
417 klass->suspend = default_suspend;
418 klass->unsuspend = default_unsuspend;
419 klass->convert_range = default_convert_range;
420 klass->query_position = default_query_position;
421 klass->query_stop = default_query_stop;
422 klass->create_rtpbin = default_create_rtpbin;
423 klass->setup_sdp = default_setup_sdp;
424 klass->handle_sdp = default_handle_sdp;
428 gst_rtsp_media_init (GstRTSPMedia * media)
430 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
434 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
435 g_mutex_init (&priv->lock);
436 g_cond_init (&priv->cond);
437 g_rec_mutex_init (&priv->state_lock);
439 priv->shared = DEFAULT_SHARED;
440 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
441 priv->reusable = DEFAULT_REUSABLE;
442 priv->profiles = DEFAULT_PROFILES;
443 priv->protocols = DEFAULT_PROTOCOLS;
444 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
445 priv->buffer_size = DEFAULT_BUFFER_SIZE;
446 priv->time_provider = DEFAULT_TIME_PROVIDER;
447 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
448 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
449 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
450 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
454 gst_rtsp_media_finalize (GObject * obj)
456 GstRTSPMediaPrivate *priv;
459 media = GST_RTSP_MEDIA (obj);
462 GST_INFO ("finalize media %p", media);
464 if (priv->permissions)
465 gst_rtsp_permissions_unref (priv->permissions);
467 g_ptr_array_unref (priv->streams);
469 g_list_free_full (priv->dynamic, gst_object_unref);
470 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
473 gst_object_unref (priv->pipeline);
475 gst_object_unref (priv->nettime);
476 gst_object_unref (priv->element);
478 g_object_unref (priv->pool);
480 g_list_free (priv->payloads);
481 g_free (priv->multicast_iface);
482 g_mutex_clear (&priv->lock);
483 g_cond_clear (&priv->cond);
484 g_rec_mutex_clear (&priv->state_lock);
486 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
490 gst_rtsp_media_get_property (GObject * object, guint propid,
491 GValue * value, GParamSpec * pspec)
493 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
497 g_value_set_object (value, media->priv->element);
500 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
502 case PROP_SUSPEND_MODE:
503 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
506 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
509 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
512 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
514 case PROP_EOS_SHUTDOWN:
515 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
517 case PROP_BUFFER_SIZE:
518 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
520 case PROP_TIME_PROVIDER:
521 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
524 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
526 case PROP_TRANSPORT_MODE:
527 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
529 case PROP_STOP_ON_DISCONNECT:
530 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
533 g_value_take_object (value, gst_rtsp_media_get_clock (media));
536 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
541 gst_rtsp_media_set_property (GObject * object, guint propid,
542 const GValue * value, GParamSpec * pspec)
544 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
548 media->priv->element = g_value_get_object (value);
549 gst_object_ref_sink (media->priv->element);
552 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
554 case PROP_SUSPEND_MODE:
555 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
558 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
561 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
564 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
566 case PROP_EOS_SHUTDOWN:
567 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
569 case PROP_BUFFER_SIZE:
570 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
572 case PROP_TIME_PROVIDER:
573 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
576 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
578 case PROP_TRANSPORT_MODE:
579 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
581 case PROP_STOP_ON_DISCONNECT:
582 gst_rtsp_media_set_stop_on_disconnect (media,
583 g_value_get_boolean (value));
586 gst_rtsp_media_set_clock (media, g_value_get_object (value));
589 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
597 } DoQueryPositionData;
600 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
604 if (gst_rtsp_stream_query_position (stream, &tmp)) {
605 data->position = MIN (data->position, tmp);
609 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
610 GST_TIME_ARGS (data->position));
614 default_query_position (GstRTSPMedia * media, gint64 * position)
616 GstRTSPMediaPrivate *priv;
617 DoQueryPositionData data;
621 data.position = G_MAXINT64;
624 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
627 *position = GST_CLOCK_TIME_NONE;
629 *position = data.position;
641 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
645 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
646 data->stop = MAX (data->stop, tmp);
652 default_query_stop (GstRTSPMedia * media, gint64 * stop)
654 GstRTSPMediaPrivate *priv;
655 DoQueryStopData data;
662 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
670 default_create_rtpbin (GstRTSPMedia * media)
674 rtpbin = gst_element_factory_make ("rtpbin", NULL);
680 is_receive_only (GstRTSPMedia * media)
682 GstRTSPMediaPrivate *priv = media->priv;
683 gboolean recive_only = TRUE;
686 for (i = 0; i < priv->streams->len; i++) {
687 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
688 if (gst_rtsp_stream_is_sender (stream) ||
689 !gst_rtsp_stream_is_receiver (stream)) {
698 /* must be called with state lock */
700 check_seekable (GstRTSPMedia * media)
703 GstRTSPMediaPrivate *priv = media->priv;
705 /* Update the seekable state of the pipeline in case it changed */
706 if (is_receive_only (media)) {
707 /* TODO: Seeking for "receive-only"? */
710 guint i, n = priv->streams->len;
712 for (i = 0; i < n; i++) {
713 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
715 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
716 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
723 query = gst_query_new_seeking (GST_FORMAT_TIME);
724 if (gst_element_query (priv->pipeline, query)) {
729 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
730 priv->seekable = seekable ? G_MAXINT64 : 0;
731 } else if (priv->streams->len) {
732 gboolean seekable = TRUE;
733 guint i, n = priv->streams->len;
735 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
736 for (i = 0; i < n; i++) {
737 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
738 seekable &= gst_rtsp_stream_seekable (stream);
740 priv->seekable = seekable ? G_MAXINT64 : -1;
743 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
745 gst_query_unref (query);
748 /* must be called with state lock */
750 check_complete (GstRTSPMedia * media)
752 GstRTSPMediaPrivate *priv = media->priv;
754 guint i, n = priv->streams->len;
756 for (i = 0; i < n; i++) {
757 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
759 if (gst_rtsp_stream_is_complete (stream))
766 /* must be called with state lock */
768 collect_media_stats (GstRTSPMedia * media)
770 GstRTSPMediaPrivate *priv = media->priv;
771 gint64 position = 0, stop = -1;
773 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
774 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
777 priv->range.unit = GST_RTSP_RANGE_NPT;
779 GST_INFO ("collect media stats");
782 priv->range.min.type = GST_RTSP_TIME_NOW;
783 priv->range.min.seconds = -1;
784 priv->range_start = -1;
785 priv->range.max.type = GST_RTSP_TIME_END;
786 priv->range.max.seconds = -1;
787 priv->range_stop = -1;
789 GstRTSPMediaClass *klass;
792 klass = GST_RTSP_MEDIA_GET_CLASS (media);
794 /* get the position */
796 if (klass->query_position)
797 ret = klass->query_position (media, &position);
800 GST_INFO ("position query failed");
804 /* get the current segment stop */
806 if (klass->query_stop)
807 ret = klass->query_stop (media, &stop);
810 GST_INFO ("stop query failed");
814 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
815 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
817 if (position == -1) {
818 priv->range.min.type = GST_RTSP_TIME_NOW;
819 priv->range.min.seconds = -1;
820 priv->range_start = -1;
822 priv->range.min.type = GST_RTSP_TIME_SECONDS;
823 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
824 priv->range_start = position;
827 priv->range.max.type = GST_RTSP_TIME_END;
828 priv->range.max.seconds = -1;
829 priv->range_stop = -1;
831 priv->range.max.type = GST_RTSP_TIME_SECONDS;
832 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
833 priv->range_stop = stop;
836 check_seekable (media);
841 * gst_rtsp_media_new:
842 * @element: (transfer full): a #GstElement
844 * Create a new #GstRTSPMedia instance. @element is the bin element that
845 * provides the different streams. The #GstRTSPMedia object contains the
846 * element to produce RTP data for one or more related (audio/video/..)
849 * Ownership is taken of @element.
851 * Returns: (transfer full): a new #GstRTSPMedia object.
854 gst_rtsp_media_new (GstElement * element)
856 GstRTSPMedia *result;
858 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
860 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
866 * gst_rtsp_media_get_element:
867 * @media: a #GstRTSPMedia
869 * Get the element that was used when constructing @media.
871 * Returns: (transfer full): a #GstElement. Unref after usage.
874 gst_rtsp_media_get_element (GstRTSPMedia * media)
876 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
878 return gst_object_ref (media->priv->element);
882 * gst_rtsp_media_take_pipeline:
883 * @media: a #GstRTSPMedia
884 * @pipeline: (transfer full): a #GstPipeline
886 * Set @pipeline as the #GstPipeline for @media. Ownership is
887 * taken of @pipeline.
890 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
892 GstRTSPMediaPrivate *priv;
894 GstNetTimeProvider *nettime;
897 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
898 g_return_if_fail (GST_IS_PIPELINE (pipeline));
902 g_mutex_lock (&priv->lock);
903 old = priv->pipeline;
904 priv->pipeline = GST_ELEMENT_CAST (pipeline);
905 nettime = priv->nettime;
906 priv->nettime = NULL;
907 g_mutex_unlock (&priv->lock);
910 gst_object_unref (old);
913 gst_object_unref (nettime);
915 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
917 for (l = priv->pending_pipeline_elements; l; l = l->next) {
918 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
920 g_list_free (priv->pending_pipeline_elements);
921 priv->pending_pipeline_elements = NULL;
925 * gst_rtsp_media_set_permissions:
926 * @media: a #GstRTSPMedia
927 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
929 * Set @permissions on @media.
932 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
933 GstRTSPPermissions * permissions)
935 GstRTSPMediaPrivate *priv;
937 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
941 g_mutex_lock (&priv->lock);
942 if (priv->permissions)
943 gst_rtsp_permissions_unref (priv->permissions);
944 if ((priv->permissions = permissions))
945 gst_rtsp_permissions_ref (permissions);
946 g_mutex_unlock (&priv->lock);
950 * gst_rtsp_media_get_permissions:
951 * @media: a #GstRTSPMedia
953 * Get the permissions object from @media.
955 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
958 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
960 GstRTSPMediaPrivate *priv;
961 GstRTSPPermissions *result;
963 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
967 g_mutex_lock (&priv->lock);
968 if ((result = priv->permissions))
969 gst_rtsp_permissions_ref (result);
970 g_mutex_unlock (&priv->lock);
976 * gst_rtsp_media_set_suspend_mode:
977 * @media: a #GstRTSPMedia
978 * @mode: the new #GstRTSPSuspendMode
980 * Control how @ media will be suspended after the SDP has been generated and
981 * after a PAUSE request has been performed.
983 * Media must be unprepared when setting the suspend mode.
986 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
988 GstRTSPMediaPrivate *priv;
990 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
994 g_rec_mutex_lock (&priv->state_lock);
995 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
997 priv->suspend_mode = mode;
998 g_rec_mutex_unlock (&priv->state_lock);
1005 GST_WARNING ("media %p was prepared", media);
1006 g_rec_mutex_unlock (&priv->state_lock);
1011 * gst_rtsp_media_get_suspend_mode:
1012 * @media: a #GstRTSPMedia
1014 * Get how @media will be suspended.
1016 * Returns: #GstRTSPSuspendMode.
1019 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1021 GstRTSPMediaPrivate *priv;
1022 GstRTSPSuspendMode res;
1024 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1028 g_rec_mutex_lock (&priv->state_lock);
1029 res = priv->suspend_mode;
1030 g_rec_mutex_unlock (&priv->state_lock);
1036 * gst_rtsp_media_set_shared:
1037 * @media: a #GstRTSPMedia
1038 * @shared: the new value
1040 * Set or unset if the pipeline for @media can be shared will multiple clients.
1041 * When @shared is %TRUE, client requests for this media will share the media
1045 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1047 GstRTSPMediaPrivate *priv;
1049 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1053 g_mutex_lock (&priv->lock);
1054 priv->shared = shared;
1055 g_mutex_unlock (&priv->lock);
1059 * gst_rtsp_media_is_shared:
1060 * @media: a #GstRTSPMedia
1062 * Check if the pipeline for @media can be shared between multiple clients.
1064 * Returns: %TRUE if the media can be shared between clients.
1067 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1069 GstRTSPMediaPrivate *priv;
1072 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1076 g_mutex_lock (&priv->lock);
1078 g_mutex_unlock (&priv->lock);
1084 * gst_rtsp_media_set_reusable:
1085 * @media: a #GstRTSPMedia
1086 * @reusable: the new value
1088 * Set or unset if the pipeline for @media can be reused after the pipeline has
1092 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1094 GstRTSPMediaPrivate *priv;
1096 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1100 g_mutex_lock (&priv->lock);
1101 priv->reusable = reusable;
1102 g_mutex_unlock (&priv->lock);
1106 * gst_rtsp_media_is_reusable:
1107 * @media: a #GstRTSPMedia
1109 * Check if the pipeline for @media can be reused after an unprepare.
1111 * Returns: %TRUE if the media can be reused
1114 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1116 GstRTSPMediaPrivate *priv;
1119 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1123 g_mutex_lock (&priv->lock);
1124 res = priv->reusable;
1125 g_mutex_unlock (&priv->lock);
1131 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1133 gst_rtsp_stream_set_profiles (stream, *profiles);
1137 * gst_rtsp_media_set_profiles:
1138 * @media: a #GstRTSPMedia
1139 * @profiles: the new flags
1141 * Configure the allowed lower transport for @media.
1144 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1146 GstRTSPMediaPrivate *priv;
1148 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1152 g_mutex_lock (&priv->lock);
1153 priv->profiles = profiles;
1154 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1155 g_mutex_unlock (&priv->lock);
1159 * gst_rtsp_media_get_profiles:
1160 * @media: a #GstRTSPMedia
1162 * Get the allowed profiles of @media.
1164 * Returns: a #GstRTSPProfile
1167 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1169 GstRTSPMediaPrivate *priv;
1172 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1176 g_mutex_lock (&priv->lock);
1177 res = priv->profiles;
1178 g_mutex_unlock (&priv->lock);
1184 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1186 gst_rtsp_stream_set_protocols (stream, *protocols);
1190 * gst_rtsp_media_set_protocols:
1191 * @media: a #GstRTSPMedia
1192 * @protocols: the new flags
1194 * Configure the allowed lower transport for @media.
1197 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1199 GstRTSPMediaPrivate *priv;
1201 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1205 g_mutex_lock (&priv->lock);
1206 priv->protocols = protocols;
1207 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1208 g_mutex_unlock (&priv->lock);
1212 * gst_rtsp_media_get_protocols:
1213 * @media: a #GstRTSPMedia
1215 * Get the allowed protocols of @media.
1217 * Returns: a #GstRTSPLowerTrans
1220 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1222 GstRTSPMediaPrivate *priv;
1223 GstRTSPLowerTrans res;
1225 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1226 GST_RTSP_LOWER_TRANS_UNKNOWN);
1230 g_mutex_lock (&priv->lock);
1231 res = priv->protocols;
1232 g_mutex_unlock (&priv->lock);
1238 * gst_rtsp_media_set_eos_shutdown:
1239 * @media: a #GstRTSPMedia
1240 * @eos_shutdown: the new value
1242 * Set or unset if an EOS event will be sent to the pipeline for @media before
1246 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1248 GstRTSPMediaPrivate *priv;
1250 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1254 g_mutex_lock (&priv->lock);
1255 priv->eos_shutdown = eos_shutdown;
1256 g_mutex_unlock (&priv->lock);
1260 * gst_rtsp_media_is_eos_shutdown:
1261 * @media: a #GstRTSPMedia
1263 * Check if the pipeline for @media will send an EOS down the pipeline before
1266 * Returns: %TRUE if the media will send EOS before unpreparing.
1269 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1271 GstRTSPMediaPrivate *priv;
1274 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1278 g_mutex_lock (&priv->lock);
1279 res = priv->eos_shutdown;
1280 g_mutex_unlock (&priv->lock);
1286 * gst_rtsp_media_set_buffer_size:
1287 * @media: a #GstRTSPMedia
1288 * @size: the new value
1290 * Set the kernel UDP buffer size.
1293 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1295 GstRTSPMediaPrivate *priv;
1298 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1300 GST_LOG_OBJECT (media, "set buffer size %u", size);
1304 g_mutex_lock (&priv->lock);
1305 priv->buffer_size = size;
1307 for (i = 0; i < priv->streams->len; i++) {
1308 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1309 gst_rtsp_stream_set_buffer_size (stream, size);
1311 g_mutex_unlock (&priv->lock);
1315 * gst_rtsp_media_get_buffer_size:
1316 * @media: a #GstRTSPMedia
1318 * Get the kernel UDP buffer size.
1320 * Returns: the kernel UDP buffer size.
1323 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1325 GstRTSPMediaPrivate *priv;
1328 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1332 g_mutex_lock (&priv->lock);
1333 res = priv->buffer_size;
1334 g_mutex_unlock (&priv->lock);
1340 * gst_rtsp_media_set_stop_on_disconnect:
1341 * @media: a #GstRTSPMedia
1342 * @stop_on_disconnect: the new value
1344 * Set or unset if the pipeline for @media should be stopped when a
1345 * client disconnects without sending TEARDOWN.
1348 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1349 gboolean stop_on_disconnect)
1351 GstRTSPMediaPrivate *priv;
1353 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1357 g_mutex_lock (&priv->lock);
1358 priv->stop_on_disconnect = stop_on_disconnect;
1359 g_mutex_unlock (&priv->lock);
1363 * gst_rtsp_media_is_stop_on_disconnect:
1364 * @media: a #GstRTSPMedia
1366 * Check if the pipeline for @media will be stopped when a client disconnects
1367 * without sending TEARDOWN.
1369 * Returns: %TRUE if the media will be stopped when a client disconnects
1370 * without sending TEARDOWN.
1373 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1375 GstRTSPMediaPrivate *priv;
1378 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1382 g_mutex_lock (&priv->lock);
1383 res = priv->stop_on_disconnect;
1384 g_mutex_unlock (&priv->lock);
1390 * gst_rtsp_media_set_retransmission_time:
1391 * @media: a #GstRTSPMedia
1392 * @time: the new value
1394 * Set the amount of time to store retransmission packets.
1397 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1399 GstRTSPMediaPrivate *priv;
1402 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1404 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1408 g_mutex_lock (&priv->lock);
1409 priv->rtx_time = time;
1410 for (i = 0; i < priv->streams->len; i++) {
1411 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1413 gst_rtsp_stream_set_retransmission_time (stream, time);
1415 g_mutex_unlock (&priv->lock);
1419 * gst_rtsp_media_get_retransmission_time:
1420 * @media: a #GstRTSPMedia
1422 * Get the amount of time to store retransmission data.
1424 * Returns: the amount of time to store retransmission data.
1427 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1429 GstRTSPMediaPrivate *priv;
1432 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1436 g_mutex_lock (&priv->lock);
1437 res = priv->rtx_time;
1438 g_mutex_unlock (&priv->lock);
1444 * gst_rtsp_media_set_do_retransmission:
1446 * Set whether retransmission requests will be sent
1451 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media, gboolean do_retransmission)
1453 GstRTSPMediaPrivate *priv;
1455 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1459 g_mutex_lock (&priv->lock);
1460 priv->do_retransmission = do_retransmission;
1463 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1464 g_mutex_unlock (&priv->lock);
1468 * gst_rtsp_media_get_do_retransmission:
1470 * Returns: Whether retransmission requests will be sent
1475 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1477 GstRTSPMediaPrivate *priv;
1480 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1484 g_mutex_lock (&priv->lock);
1485 res = priv->do_retransmission;
1486 g_mutex_unlock (&priv->lock);
1492 * gst_rtsp_media_set_latency:
1493 * @media: a #GstRTSPMedia
1494 * @latency: latency in milliseconds
1496 * Configure the latency used for receiving media.
1499 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1501 GstRTSPMediaPrivate *priv;
1503 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1505 GST_LOG_OBJECT (media, "set latency %ums", latency);
1509 g_mutex_lock (&priv->lock);
1510 priv->latency = latency;
1512 g_object_set (priv->rtpbin, "latency", latency, NULL);
1513 g_mutex_unlock (&priv->lock);
1517 * gst_rtsp_media_get_latency:
1518 * @media: a #GstRTSPMedia
1520 * Get the latency that is used for receiving media.
1522 * Returns: latency in milliseconds
1525 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1527 GstRTSPMediaPrivate *priv;
1530 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1534 g_mutex_lock (&priv->lock);
1535 res = priv->latency;
1536 g_mutex_unlock (&priv->lock);
1542 * gst_rtsp_media_use_time_provider:
1543 * @media: a #GstRTSPMedia
1544 * @time_provider: if a #GstNetTimeProvider should be used
1546 * Set @media to provide a #GstNetTimeProvider.
1549 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1551 GstRTSPMediaPrivate *priv;
1553 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1557 g_mutex_lock (&priv->lock);
1558 priv->time_provider = time_provider;
1559 g_mutex_unlock (&priv->lock);
1563 * gst_rtsp_media_is_time_provider:
1564 * @media: a #GstRTSPMedia
1566 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1568 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1570 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1573 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1575 GstRTSPMediaPrivate *priv;
1578 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1582 g_mutex_lock (&priv->lock);
1583 res = priv->time_provider;
1584 g_mutex_unlock (&priv->lock);
1590 * gst_rtsp_media_set_clock:
1591 * @media: a #GstRTSPMedia
1592 * @clock: (nullable): #GstClock to be used
1594 * Configure the clock used for the media.
1597 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1599 GstRTSPMediaPrivate *priv;
1601 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1602 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1604 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1608 g_mutex_lock (&priv->lock);
1610 gst_object_unref (priv->clock);
1611 priv->clock = clock ? gst_object_ref (clock) : NULL;
1612 if (priv->pipeline) {
1614 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1616 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1619 g_mutex_unlock (&priv->lock);
1623 * gst_rtsp_media_set_publish_clock_mode:
1624 * @media: a #GstRTSPMedia
1625 * @mode: the clock publish mode
1627 * Sets if and how the media clock should be published according to RFC7273.
1632 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1633 GstRTSPPublishClockMode mode)
1635 GstRTSPMediaPrivate *priv;
1639 g_mutex_lock (&priv->lock);
1640 priv->publish_clock_mode = mode;
1642 n = priv->streams->len;
1643 for (i = 0; i < n; i++) {
1644 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1646 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1648 g_mutex_unlock (&priv->lock);
1652 * gst_rtsp_media_get_publish_clock_mode:
1653 * @media: a #GstRTSPMedia
1655 * Gets if and how the media clock should be published according to RFC7273.
1657 * Returns: The GstRTSPPublishClockMode
1661 GstRTSPPublishClockMode
1662 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1664 GstRTSPMediaPrivate *priv;
1665 GstRTSPPublishClockMode ret;
1668 g_mutex_lock (&priv->lock);
1669 ret = priv->publish_clock_mode;
1670 g_mutex_unlock (&priv->lock);
1676 * gst_rtsp_media_set_address_pool:
1677 * @media: a #GstRTSPMedia
1678 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1680 * configure @pool to be used as the address pool of @media.
1683 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1684 GstRTSPAddressPool * pool)
1686 GstRTSPMediaPrivate *priv;
1687 GstRTSPAddressPool *old;
1689 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1693 GST_LOG_OBJECT (media, "set address pool %p", pool);
1695 g_mutex_lock (&priv->lock);
1696 if ((old = priv->pool) != pool)
1697 priv->pool = pool ? g_object_ref (pool) : NULL;
1700 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1702 g_mutex_unlock (&priv->lock);
1705 g_object_unref (old);
1709 * gst_rtsp_media_get_address_pool:
1710 * @media: a #GstRTSPMedia
1712 * Get the #GstRTSPAddressPool used as the address pool of @media.
1714 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1715 * g_object_unref() after usage.
1717 GstRTSPAddressPool *
1718 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1720 GstRTSPMediaPrivate *priv;
1721 GstRTSPAddressPool *result;
1723 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1727 g_mutex_lock (&priv->lock);
1728 if ((result = priv->pool))
1729 g_object_ref (result);
1730 g_mutex_unlock (&priv->lock);
1736 * gst_rtsp_media_set_multicast_iface:
1737 * @media: a #GstRTSPMedia
1738 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1740 * configure @multicast_iface to be used for @media.
1743 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1744 const gchar * multicast_iface)
1746 GstRTSPMediaPrivate *priv;
1749 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1753 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1755 g_mutex_lock (&priv->lock);
1756 if ((old = priv->multicast_iface) != multicast_iface)
1757 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1760 g_ptr_array_foreach (priv->streams,
1761 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1762 g_mutex_unlock (&priv->lock);
1769 * gst_rtsp_media_get_multicast_iface:
1770 * @media: a #GstRTSPMedia
1772 * Get the multicast interface used for @media.
1774 * Returns: (transfer full) (nullable): the multicast interface for @media.
1775 * g_free() after usage.
1778 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1780 GstRTSPMediaPrivate *priv;
1783 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1787 g_mutex_lock (&priv->lock);
1788 if ((result = priv->multicast_iface))
1789 result = g_strdup (result);
1790 g_mutex_unlock (&priv->lock);
1796 _find_payload_types (GstRTSPMedia * media)
1799 GQueue queue = G_QUEUE_INIT;
1801 n = media->priv->streams->len;
1802 for (i = 0; i < n; i++) {
1803 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1804 guint pt = gst_rtsp_stream_get_pt (stream);
1806 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1813 _next_available_pt (GList * payloads)
1817 for (i = 96; i <= 127; i++) {
1818 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1820 return GPOINTER_TO_UINT (i);
1827 * gst_rtsp_media_collect_streams:
1828 * @media: a #GstRTSPMedia
1830 * Find all payloader elements, they should be named pay\%d in the
1831 * element of @media, and create #GstRTSPStreams for them.
1833 * Collect all dynamic elements, named dynpay\%d, and add them to
1834 * the list of dynamic elements.
1836 * Find all depayloader elements, they should be named depay\%d in the
1837 * element of @media, and create #GstRTSPStreams for them.
1840 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1842 GstRTSPMediaPrivate *priv;
1843 GstElement *element, *elem;
1847 gboolean more_elem_remaining = TRUE;
1848 GstRTSPTransportMode mode = 0;
1850 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1853 element = priv->element;
1856 for (i = 0; more_elem_remaining; i++) {
1859 more_elem_remaining = FALSE;
1861 name = g_strdup_printf ("pay%d", i);
1862 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1864 GST_INFO ("found stream %d with payloader %p", i, elem);
1866 /* take the pad of the payloader */
1867 pad = gst_element_get_static_pad (elem, "src");
1869 /* find the real payload element in case elem is a GstBin */
1870 pay = find_payload_element (elem);
1872 /* create the stream */
1874 GST_WARNING ("could not find real payloader, using bin");
1875 gst_rtsp_media_create_stream (media, elem, pad);
1877 gst_rtsp_media_create_stream (media, pay, pad);
1878 gst_object_unref (pay);
1881 gst_object_unref (pad);
1882 gst_object_unref (elem);
1885 more_elem_remaining = TRUE;
1886 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1890 name = g_strdup_printf ("dynpay%d", i);
1891 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1892 /* a stream that will dynamically create pads to provide RTP packets */
1893 GST_INFO ("found dynamic element %d, %p", i, elem);
1895 g_mutex_lock (&priv->lock);
1896 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1897 g_mutex_unlock (&priv->lock);
1899 priv->nb_dynamic_elements++;
1902 more_elem_remaining = TRUE;
1903 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1907 name = g_strdup_printf ("depay%d", i);
1908 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1909 GST_INFO ("found stream %d with depayloader %p", i, elem);
1911 /* take the pad of the payloader */
1912 pad = gst_element_get_static_pad (elem, "sink");
1913 /* create the stream */
1914 gst_rtsp_media_create_stream (media, elem, pad);
1915 gst_object_unref (pad);
1916 gst_object_unref (elem);
1919 more_elem_remaining = TRUE;
1920 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1926 if (priv->transport_mode != mode)
1927 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1928 priv->transport_mode, mode);
1934 GstElement *appsink, *appsrc;
1935 GstRTSPStream *stream;
1938 static GstFlowReturn
1939 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
1941 AppSinkSrcData *data = user_data;
1945 sample = gst_app_sink_pull_sample (appsink);
1947 return GST_FLOW_FLUSHING;
1950 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
1951 gst_sample_unref (sample);
1955 static GstAppSinkCallbacks appsink_callbacks = {
1961 static GstPadProbeReturn
1962 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
1964 AppSinkSrcData *data = user_data;
1966 if (GST_IS_EVENT (info->data)
1967 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
1968 GstClockTime min, max;
1970 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
1972 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
1973 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
1974 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
1976 } else if (GST_IS_QUERY (info->data)) {
1977 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
1978 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
1979 gst_object_unref (srcpad);
1980 return GST_PAD_PROBE_HANDLED;
1982 gst_object_unref (srcpad);
1985 return GST_PAD_PROBE_OK;
1988 static GstPadProbeReturn
1989 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
1991 AppSinkSrcData *data = user_data;
1993 if (GST_IS_QUERY (info->data)) {
1994 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
1995 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
1996 gst_object_unref (sinkpad);
1997 return GST_PAD_PROBE_HANDLED;
1999 gst_object_unref (sinkpad);
2002 return GST_PAD_PROBE_OK;
2006 * gst_rtsp_media_create_stream:
2007 * @media: a #GstRTSPMedia
2008 * @payloader: a #GstElement
2011 * Create a new stream in @media that provides RTP data on @pad.
2012 * @pad should be a pad of an element inside @media->element.
2014 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2018 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2021 GstRTSPMediaPrivate *priv;
2022 GstRTSPStream *stream;
2026 AppSinkSrcData *data = NULL;
2028 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2029 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2030 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2034 g_mutex_lock (&priv->lock);
2035 idx = priv->streams->len;
2037 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
2039 if (GST_PAD_IS_SRC (pad))
2040 name = g_strdup_printf ("src_%u", idx);
2042 name = g_strdup_printf ("sink_%u", idx);
2044 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2045 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2046 GstElement *appsink, *appsrc;
2047 GstPad *sinkpad, *srcpad;
2049 appsink = gst_element_factory_make ("appsink", NULL);
2050 appsrc = gst_element_factory_make ("appsrc", NULL);
2052 if (GST_PAD_IS_SINK (pad)) {
2053 srcpad = gst_element_get_static_pad (appsrc, "src");
2055 gst_bin_add (GST_BIN (priv->element), appsrc);
2057 gst_pad_link (srcpad, pad);
2058 gst_object_unref (srcpad);
2060 streampad = gst_element_get_static_pad (appsink, "sink");
2062 priv->pending_pipeline_elements =
2063 g_list_prepend (priv->pending_pipeline_elements, appsink);
2065 sinkpad = gst_element_get_static_pad (appsink, "sink");
2067 gst_pad_link (pad, sinkpad);
2068 gst_object_unref (sinkpad);
2070 streampad = gst_element_get_static_pad (appsrc, "src");
2072 priv->pending_pipeline_elements =
2073 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2076 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2078 g_object_set (appsink, "sync", FALSE, "async", FALSE, NULL);
2080 data = g_new0 (AppSinkSrcData, 1);
2081 data->appsink = appsink;
2082 data->appsrc = appsrc;
2084 sinkpad = gst_element_get_static_pad (appsink, "sink");
2085 gst_pad_add_probe (sinkpad,
2086 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2087 appsink_pad_probe, data, NULL);
2088 gst_object_unref (sinkpad);
2090 srcpad = gst_element_get_static_pad (appsrc, "src");
2091 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2092 appsrc_pad_probe, data, NULL);
2093 gst_object_unref (srcpad);
2095 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2097 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2100 streampad = gst_ghost_pad_new (name, pad);
2101 gst_pad_set_active (streampad, TRUE);
2102 gst_element_add_pad (priv->element, streampad);
2106 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2108 data->stream = stream;
2110 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2111 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2112 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2113 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2114 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2115 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2116 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2118 g_ptr_array_add (priv->streams, stream);
2120 if (GST_PAD_IS_SRC (pad)) {
2124 g_list_free (priv->payloads);
2125 priv->payloads = _find_payload_types (media);
2127 n = priv->streams->len;
2128 for (i = 0; i < n; i++) {
2129 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2130 guint rtx_pt = _next_available_pt (priv->payloads);
2133 GST_WARNING ("Ran out of space of dynamic payload types");
2137 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2140 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2143 g_mutex_unlock (&priv->lock);
2145 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2152 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2154 GstRTSPMediaPrivate *priv;
2156 AppSinkSrcData *data;
2160 g_mutex_lock (&priv->lock);
2161 /* remove the ghostpad */
2162 srcpad = gst_rtsp_stream_get_srcpad (stream);
2163 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2165 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2166 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2167 else if (GST_OBJECT_PARENT (data->appsrc) ==
2168 GST_OBJECT_CAST (priv->element))
2169 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2170 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2171 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2172 else if (GST_OBJECT_PARENT (data->appsink) ==
2173 GST_OBJECT_CAST (priv->element))
2174 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2176 gst_element_remove_pad (priv->element, srcpad);
2178 gst_object_unref (srcpad);
2179 /* now remove the stream */
2180 g_object_ref (stream);
2181 g_ptr_array_remove (priv->streams, stream);
2182 g_mutex_unlock (&priv->lock);
2184 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2187 g_object_unref (stream);
2191 * gst_rtsp_media_n_streams:
2192 * @media: a #GstRTSPMedia
2194 * Get the number of streams in this media.
2196 * Returns: The number of streams.
2199 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2201 GstRTSPMediaPrivate *priv;
2204 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2208 g_mutex_lock (&priv->lock);
2209 res = priv->streams->len;
2210 g_mutex_unlock (&priv->lock);
2216 * gst_rtsp_media_get_stream:
2217 * @media: a #GstRTSPMedia
2218 * @idx: the stream index
2220 * Retrieve the stream with index @idx from @media.
2222 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2223 * @idx or %NULL when a stream with that index did not exist.
2226 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2228 GstRTSPMediaPrivate *priv;
2231 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2235 g_mutex_lock (&priv->lock);
2236 if (idx < priv->streams->len)
2237 res = g_ptr_array_index (priv->streams, idx);
2240 g_mutex_unlock (&priv->lock);
2246 * gst_rtsp_media_find_stream:
2247 * @media: a #GstRTSPMedia
2248 * @control: the control of the stream
2250 * Find a stream in @media with @control as the control uri.
2252 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2253 * control uri @control or %NULL when a stream with that control did
2257 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2259 GstRTSPMediaPrivate *priv;
2263 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2264 g_return_val_if_fail (control != NULL, NULL);
2270 g_mutex_lock (&priv->lock);
2271 for (i = 0; i < priv->streams->len; i++) {
2272 GstRTSPStream *test;
2274 test = g_ptr_array_index (priv->streams, i);
2275 if (gst_rtsp_stream_has_control (test, control)) {
2280 g_mutex_unlock (&priv->lock);
2285 /* called with state-lock */
2287 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2288 GstRTSPRangeUnit unit)
2290 return gst_rtsp_range_convert_units (range, unit);
2294 * gst_rtsp_media_get_range_string:
2295 * @media: a #GstRTSPMedia
2296 * @play: for the PLAY request
2297 * @unit: the unit to use for the string
2299 * Get the current range as a string. @media must be prepared with
2300 * gst_rtsp_media_prepare ().
2302 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2305 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2306 GstRTSPRangeUnit unit)
2308 GstRTSPMediaClass *klass;
2309 GstRTSPMediaPrivate *priv;
2311 GstRTSPTimeRange range;
2313 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2314 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2315 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2319 g_rec_mutex_lock (&priv->state_lock);
2320 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2321 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2324 g_mutex_lock (&priv->lock);
2326 /* Update the range value with current position/duration */
2327 collect_media_stats (media);
2330 range = priv->range;
2332 if (!play && priv->n_active > 0) {
2333 range.min.type = GST_RTSP_TIME_NOW;
2334 range.min.seconds = -1;
2336 g_mutex_unlock (&priv->lock);
2337 g_rec_mutex_unlock (&priv->state_lock);
2339 if (!klass->convert_range (media, &range, unit))
2340 goto conversion_failed;
2342 result = gst_rtsp_range_to_string (&range);
2349 GST_WARNING ("media %p was not prepared", media);
2350 g_rec_mutex_unlock (&priv->state_lock);
2355 GST_WARNING ("range conversion to unit %d failed", unit);
2361 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2363 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2367 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2369 GstRTSPMediaPrivate *priv = media->priv;
2371 GST_DEBUG ("media %p set blocked %d", media, blocked);
2372 priv->blocked = blocked;
2373 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2377 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2379 gst_rtsp_stream_unblock_linked (stream);
2383 media_unblock_linked (GstRTSPMedia * media)
2385 GstRTSPMediaPrivate *priv = media->priv;
2387 GST_DEBUG ("media %p unblocking linked streams", media);
2388 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2392 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2394 GstRTSPMediaPrivate *priv = media->priv;
2396 g_mutex_lock (&priv->lock);
2397 priv->status = status;
2398 GST_DEBUG ("setting new status to %d", status);
2399 g_cond_broadcast (&priv->cond);
2400 g_mutex_unlock (&priv->lock);
2404 * gst_rtsp_media_get_status:
2405 * @media: a #GstRTSPMedia
2407 * Get the status of @media. When @media is busy preparing, this function waits
2408 * until @media is prepared or in error.
2410 * Returns: the status of @media.
2413 gst_rtsp_media_get_status (GstRTSPMedia * media)
2415 GstRTSPMediaPrivate *priv = media->priv;
2416 GstRTSPMediaStatus result;
2419 g_mutex_lock (&priv->lock);
2420 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2421 /* while we are preparing, wait */
2422 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2423 GST_DEBUG ("waiting for status change");
2424 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2425 GST_DEBUG ("timeout, assuming error status");
2426 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2429 /* could be success or error */
2430 result = priv->status;
2431 GST_DEBUG ("got status %d", result);
2432 g_mutex_unlock (&priv->lock);
2438 * gst_rtsp_media_seek_full:
2439 * @media: a #GstRTSPMedia
2440 * @range: (transfer none): a #GstRTSPTimeRange
2441 * @flags: The minimal set of #GstSeekFlags to use
2443 * Seek the pipeline of @media to @range. @media must be prepared with
2444 * gst_rtsp_media_prepare().
2446 * Returns: %TRUE on success.
2449 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2452 GstRTSPMediaClass *klass;
2453 GstRTSPMediaPrivate *priv;
2455 GstClockTime start, stop;
2456 GstSeekType start_type, stop_type;
2457 gint64 current_position;
2459 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2461 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2462 g_return_val_if_fail (range != NULL, FALSE);
2463 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2467 g_rec_mutex_lock (&priv->state_lock);
2468 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2471 /* check if the media pipeline is complete in order to perform a
2472 * seek operation on it */
2473 if (!check_complete (media))
2476 /* Update the seekable state of the pipeline in case it changed */
2477 check_seekable (media);
2479 if (priv->seekable == 0) {
2480 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2481 " not seekable streams.");
2484 } else if (priv->seekable < 0) {
2488 start_type = stop_type = GST_SEEK_TYPE_NONE;
2490 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2492 gst_rtsp_range_get_times (range, &start, &stop);
2494 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2495 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2496 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2497 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2499 current_position = -1;
2500 if (klass->query_position)
2501 klass->query_position (media, ¤t_position);
2502 GST_INFO ("current media position %" GST_TIME_FORMAT,
2503 GST_TIME_ARGS (current_position));
2505 if (start != GST_CLOCK_TIME_NONE)
2506 start_type = GST_SEEK_TYPE_SET;
2508 if (priv->range_stop == stop)
2509 stop = GST_CLOCK_TIME_NONE;
2510 else if (stop != GST_CLOCK_TIME_NONE)
2511 stop_type = GST_SEEK_TYPE_SET;
2513 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2514 gboolean had_flags = flags != 0;
2516 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2517 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2519 /* depends on the current playing state of the pipeline. We might need to
2520 * queue this until we get EOS. */
2522 flags |= GST_SEEK_FLAG_FLUSH;
2524 flags = GST_SEEK_FLAG_FLUSH;
2527 /* if range start was not supplied we must continue from current position.
2528 * but since we're doing a flushing seek, let us query the current position
2529 * so we end up at exactly the same position after the seek. */
2530 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2531 if (current_position == -1) {
2532 GST_WARNING ("current position unknown");
2534 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2535 GST_TIME_ARGS (current_position));
2536 start = current_position;
2537 start_type = GST_SEEK_TYPE_SET;
2539 flags |= GST_SEEK_FLAG_ACCURATE;
2542 /* only set keyframe flag when modifying start */
2543 if (start_type != GST_SEEK_TYPE_NONE)
2545 flags |= GST_SEEK_FLAG_KEY_UNIT;
2548 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2549 GST_DEBUG ("not seeking because no position change");
2552 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2554 media_streams_set_blocked (media, TRUE);
2556 /* FIXME, we only do forwards playback, no trick modes yet */
2557 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2558 flags, start_type, start, stop_type, stop);
2560 /* and block for the seek to complete */
2561 GST_INFO ("done seeking %d", res);
2565 g_rec_mutex_unlock (&priv->state_lock);
2567 /* wait until pipeline is prerolled again, this will also collect stats */
2568 if (!wait_preroll (media))
2569 goto preroll_failed;
2571 g_rec_mutex_lock (&priv->state_lock);
2572 GST_INFO ("prerolled again");
2575 GST_INFO ("no seek needed");
2578 g_rec_mutex_unlock (&priv->state_lock);
2585 g_rec_mutex_unlock (&priv->state_lock);
2586 GST_INFO ("media %p is not prepared", media);
2591 g_rec_mutex_unlock (&priv->state_lock);
2592 GST_INFO ("pipeline is not complete");
2597 g_rec_mutex_unlock (&priv->state_lock);
2598 GST_INFO ("pipeline is not seekable");
2603 g_rec_mutex_unlock (&priv->state_lock);
2604 GST_WARNING ("conversion to npt not supported");
2609 g_rec_mutex_unlock (&priv->state_lock);
2610 GST_INFO ("seeking failed");
2611 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2616 GST_WARNING ("failed to preroll after seek");
2623 * gst_rtsp_media_seek:
2624 * @media: a #GstRTSPMedia
2625 * @range: (transfer none): a #GstRTSPTimeRange
2627 * Seek the pipeline of @media to @range. @media must be prepared with
2628 * gst_rtsp_media_prepare().
2630 * Returns: %TRUE on success.
2633 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2635 return gst_rtsp_media_seek_full (media, range, 0);
2640 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2642 *blocked &= gst_rtsp_stream_is_blocking (stream);
2646 media_streams_blocking (GstRTSPMedia * media)
2648 gboolean blocking = TRUE;
2650 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2656 static GstStateChangeReturn
2657 set_state (GstRTSPMedia * media, GstState state)
2659 GstRTSPMediaPrivate *priv = media->priv;
2660 GstStateChangeReturn ret;
2662 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2664 ret = gst_element_set_state (priv->pipeline, state);
2669 static GstStateChangeReturn
2670 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2672 GstRTSPMediaPrivate *priv = media->priv;
2673 GstStateChangeReturn ret;
2675 GST_INFO ("set target state to %s for media %p",
2676 gst_element_state_get_name (state), media);
2677 priv->target_state = state;
2679 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2680 priv->target_state, NULL);
2683 ret = set_state (media, state);
2685 ret = GST_STATE_CHANGE_SUCCESS;
2690 /* called with state-lock */
2692 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2694 GstRTSPMediaPrivate *priv = media->priv;
2695 GstMessageType type;
2697 type = GST_MESSAGE_TYPE (message);
2700 case GST_MESSAGE_STATE_CHANGED:
2702 GstState old, new, pending;
2704 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2707 gst_message_parse_state_changed (message, &old, &new, &pending);
2709 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2710 gst_element_state_get_name (old), gst_element_state_get_name (new),
2711 gst_element_state_get_name (pending));
2712 if (priv->no_more_pads_pending == 0 && is_receive_only (media) &&
2713 old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2714 GST_INFO ("%p: went to PAUSED, prepared now", media);
2715 collect_media_stats (media);
2717 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2718 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2723 case GST_MESSAGE_BUFFERING:
2727 gst_message_parse_buffering (message, &percent);
2729 /* no state management needed for live pipelines */
2733 if (percent == 100) {
2734 /* a 100% message means buffering is done */
2735 priv->buffering = FALSE;
2736 /* if the desired state is playing, go back */
2737 if (priv->target_state == GST_STATE_PLAYING) {
2738 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2739 set_state (media, GST_STATE_PLAYING);
2741 GST_INFO ("Buffering done");
2744 /* buffering busy */
2745 if (priv->buffering == FALSE) {
2746 if (priv->target_state == GST_STATE_PLAYING) {
2747 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2748 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2749 set_state (media, GST_STATE_PAUSED);
2751 GST_INFO ("Buffering ...");
2754 priv->buffering = TRUE;
2758 case GST_MESSAGE_LATENCY:
2760 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2763 case GST_MESSAGE_ERROR:
2768 gst_message_parse_error (message, &gerror, &debug);
2769 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2770 g_error_free (gerror);
2773 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2776 case GST_MESSAGE_WARNING:
2781 gst_message_parse_warning (message, &gerror, &debug);
2782 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2783 g_error_free (gerror);
2787 case GST_MESSAGE_ELEMENT:
2789 const GstStructure *s;
2791 s = gst_message_get_structure (message);
2792 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2793 GST_DEBUG ("media received blocking message");
2794 if (priv->blocked && media_streams_blocking (media) &&
2795 priv->no_more_pads_pending == 0) {
2796 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
2797 collect_media_stats (media);
2799 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2800 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2805 case GST_MESSAGE_STREAM_STATUS:
2807 case GST_MESSAGE_ASYNC_DONE:
2808 if (priv->complete) {
2809 /* receive the final ASYNC_DONE, that is posted by the media pipeline
2810 * after all the transport parts have been successfully added to
2811 * the media streams. */
2812 GST_DEBUG_OBJECT (media, "got async-done");
2813 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2814 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2817 case GST_MESSAGE_EOS:
2818 GST_INFO ("%p: got EOS", media);
2820 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2821 GST_DEBUG ("shutting down after EOS");
2822 finish_unprepare (media);
2826 GST_INFO ("%p: got message type %d (%s)", media, type,
2827 gst_message_type_get_name (type));
2834 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2836 GstRTSPMediaPrivate *priv = media->priv;
2837 GstRTSPMediaClass *klass;
2840 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2842 g_rec_mutex_lock (&priv->state_lock);
2843 if (klass->handle_message)
2844 ret = klass->handle_message (media, message);
2847 g_rec_mutex_unlock (&priv->state_lock);
2853 watch_destroyed (GstRTSPMedia * media)
2855 GST_DEBUG_OBJECT (media, "source destroyed");
2856 g_object_unref (media);
2860 find_payload_element (GstElement * payloader)
2862 GstElement *pay = NULL;
2864 if (GST_IS_BIN (payloader)) {
2866 GValue item = { 0 };
2868 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2869 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2870 GstElement *element = (GstElement *) g_value_get_object (&item);
2871 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2875 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2879 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2880 pay = gst_object_ref (element);
2881 g_value_unset (&item);
2884 g_value_unset (&item);
2886 gst_iterator_free (iter);
2888 pay = g_object_ref (payloader);
2894 /* called from streaming threads */
2896 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2898 GstRTSPMediaPrivate *priv = media->priv;
2899 GstRTSPStream *stream;
2902 /* find the real payload element */
2903 pay = find_payload_element (element);
2904 stream = gst_rtsp_media_create_stream (media, pay, pad);
2905 gst_object_unref (pay);
2907 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2909 g_rec_mutex_lock (&priv->state_lock);
2910 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2913 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2915 /* join the element in the PAUSED state because this callback is
2916 * called from the streaming thread and it is PAUSED */
2917 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2918 priv->rtpbin, GST_STATE_PAUSED)) {
2919 GST_WARNING ("failed to join bin element");
2923 gst_rtsp_stream_set_blocked (stream, TRUE);
2925 g_rec_mutex_unlock (&priv->state_lock);
2932 gst_rtsp_media_remove_stream (media, stream);
2933 g_rec_mutex_unlock (&priv->state_lock);
2934 GST_INFO ("ignore pad because we are not preparing");
2940 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2942 GstRTSPMediaPrivate *priv = media->priv;
2943 GstRTSPStream *stream;
2945 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2949 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2951 g_rec_mutex_lock (&priv->state_lock);
2952 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2953 g_rec_mutex_unlock (&priv->state_lock);
2955 gst_rtsp_media_remove_stream (media, stream);
2959 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2961 GstRTSPMediaPrivate *priv = media->priv;
2963 GST_INFO_OBJECT (element, "no more pads");
2964 g_mutex_lock (&priv->lock);
2965 priv->no_more_pads_pending--;
2966 g_mutex_unlock (&priv->lock);
2969 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2971 struct _DynPaySignalHandlers
2973 gulong pad_added_handler;
2974 gulong pad_removed_handler;
2975 gulong no_more_pads_handler;
2979 start_preroll (GstRTSPMedia * media)
2981 GstRTSPMediaPrivate *priv = media->priv;
2982 GstStateChangeReturn ret;
2984 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2986 /* start blocked since it is possible that there are no sink elements yet */
2987 media_streams_set_blocked (media, TRUE);
2988 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2991 case GST_STATE_CHANGE_SUCCESS:
2992 GST_INFO ("SUCCESS state change for media %p", media);
2994 case GST_STATE_CHANGE_ASYNC:
2995 GST_INFO ("ASYNC state change for media %p", media);
2997 case GST_STATE_CHANGE_NO_PREROLL:
2998 /* we need to go to PLAYING */
2999 GST_INFO ("NO_PREROLL state change: live media %p", media);
3000 /* FIXME we disable seeking for live streams for now. We should perform a
3001 * seeking query in preroll instead */
3002 priv->seekable = -1;
3003 priv->is_live = TRUE;
3005 ret = set_state (media, GST_STATE_PLAYING);
3006 if (ret == GST_STATE_CHANGE_FAILURE)
3009 case GST_STATE_CHANGE_FAILURE:
3017 GST_WARNING ("failed to preroll pipeline");
3023 wait_preroll (GstRTSPMedia * media)
3025 GstRTSPMediaStatus status;
3027 GST_DEBUG ("wait to preroll pipeline");
3029 /* wait until pipeline is prerolled */
3030 status = gst_rtsp_media_get_status (media);
3031 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3032 goto preroll_failed;
3038 GST_WARNING ("failed to preroll pipeline");
3044 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3046 GstRTSPMediaPrivate *priv = media->priv;
3047 GstRTSPStream *stream = NULL;
3049 GstElement *res = NULL;
3051 g_mutex_lock (&priv->lock);
3052 for (i = 0; i < priv->streams->len; i++) {
3053 stream = g_ptr_array_index (priv->streams, i);
3055 if (sessid == gst_rtsp_stream_get_index (stream))
3060 g_mutex_unlock (&priv->lock);
3063 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3069 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3071 GstRTSPMediaPrivate *priv = media->priv;
3072 GstRTSPStream *stream = NULL;
3074 GstElement *res = NULL;
3076 g_mutex_lock (&priv->lock);
3077 for (i = 0; i < priv->streams->len; i++) {
3078 stream = g_ptr_array_index (priv->streams, i);
3080 if (sessid == gst_rtsp_stream_get_index (stream))
3085 g_mutex_unlock (&priv->lock);
3088 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3094 start_prepare (GstRTSPMedia * media)
3096 GstRTSPMediaPrivate *priv = media->priv;
3100 g_rec_mutex_lock (&priv->state_lock);
3101 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3102 goto no_longer_preparing;
3104 /* link streams we already have, other streams might appear when we have
3105 * dynamic elements */
3106 for (i = 0; i < priv->streams->len; i++) {
3107 GstRTSPStream *stream;
3109 stream = g_ptr_array_index (priv->streams, i);
3111 if (priv->rtx_time > 0) {
3112 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3113 g_signal_connect (priv->rtpbin, "request-aux-sender",
3114 (GCallback) request_aux_sender, media);
3117 if (priv->do_retransmission) {
3118 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3119 (GCallback) request_aux_receiver, media);
3122 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3123 priv->rtpbin, GST_STATE_NULL)) {
3124 goto join_bin_failed;
3129 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission, NULL);
3131 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3132 GstElement *elem = walk->data;
3133 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3135 GST_INFO ("adding callbacks for dynamic element %p", elem);
3137 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3138 (GCallback) pad_added_cb, media);
3139 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3140 (GCallback) pad_removed_cb, media);
3141 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3142 (GCallback) no_more_pads_cb, media);
3144 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3147 if (priv->nb_dynamic_elements == 0 && is_receive_only (media)) {
3148 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3149 * a second ASYNC state change failing */
3150 priv->is_live = TRUE;
3151 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3152 } else if (!start_preroll (media)) {
3153 goto preroll_failed;
3156 g_rec_mutex_unlock (&priv->state_lock);
3160 no_longer_preparing:
3162 GST_INFO ("media is no longer preparing");
3163 g_rec_mutex_unlock (&priv->state_lock);
3168 GST_WARNING ("failed to join bin element");
3169 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3170 g_rec_mutex_unlock (&priv->state_lock);
3175 GST_WARNING ("failed to preroll pipeline");
3176 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3177 g_rec_mutex_unlock (&priv->state_lock);
3183 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3185 GstRTSPMediaPrivate *priv;
3186 GstRTSPMediaClass *klass;
3188 GMainContext *context;
3193 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3195 if (!klass->create_rtpbin)
3196 goto no_create_rtpbin;
3198 priv->rtpbin = klass->create_rtpbin (media);
3199 if (priv->rtpbin != NULL) {
3200 gboolean success = TRUE;
3202 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3204 if (klass->setup_rtpbin)
3205 success = klass->setup_rtpbin (media, priv->rtpbin);
3207 if (success == FALSE) {
3208 gst_object_unref (priv->rtpbin);
3209 priv->rtpbin = NULL;
3212 if (priv->rtpbin == NULL)
3215 priv->thread = thread;
3216 context = (thread != NULL) ? (thread->context) : NULL;
3218 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3220 /* add the pipeline bus to our custom mainloop */
3221 priv->source = gst_bus_create_watch (bus);
3222 gst_object_unref (bus);
3224 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3225 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3227 priv->id = g_source_attach (priv->source, context);
3229 /* add stuff to the bin */
3230 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3232 /* do remainder in context */
3233 source = g_idle_source_new ();
3234 g_source_set_callback (source, (GSourceFunc) start_prepare,
3235 g_object_ref (media), (GDestroyNotify) g_object_unref);
3236 g_source_attach (source, context);
3237 g_source_unref (source);
3244 GST_ERROR ("no create_rtpbin function");
3245 g_critical ("no create_rtpbin vmethod function set");
3250 GST_WARNING ("no rtpbin element");
3251 g_warning ("failed to create element 'rtpbin', check your installation");
3257 * gst_rtsp_media_prepare:
3258 * @media: a #GstRTSPMedia
3259 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3260 * bus handler or %NULL
3262 * Prepare @media for streaming. This function will create the objects
3263 * to manage the streaming. A pipeline must have been set on @media with
3264 * gst_rtsp_media_take_pipeline().
3266 * It will preroll the pipeline and collect vital information about the streams
3267 * such as the duration.
3269 * Returns: %TRUE on success.
3272 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3274 GstRTSPMediaPrivate *priv;
3275 GstRTSPMediaClass *klass;
3277 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3281 g_rec_mutex_lock (&priv->state_lock);
3282 priv->prepare_count++;
3284 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3285 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3288 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3291 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3292 goto not_unprepared;
3294 if (!priv->reusable && priv->reused)
3297 GST_INFO ("preparing media %p", media);
3299 /* reset some variables */
3300 priv->is_live = FALSE;
3301 priv->seekable = -1;
3302 priv->buffering = FALSE;
3303 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3305 /* we're preparing now */
3306 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3308 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3309 if (klass->prepare) {
3310 if (!klass->prepare (media, thread))
3311 goto prepare_failed;
3315 g_rec_mutex_unlock (&priv->state_lock);
3317 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3318 * able to do this async so that we don't block the server thread. */
3319 if (!wait_preroll (media))
3320 goto preroll_failed;
3322 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3324 GST_INFO ("object %p is prerolled", media);
3331 /* we are not going to use the giving thread, so stop it. */
3333 gst_rtsp_thread_stop (thread);
3338 GST_LOG ("media %p was prepared", media);
3339 /* we are not going to use the giving thread, so stop it. */
3341 gst_rtsp_thread_stop (thread);
3342 g_rec_mutex_unlock (&priv->state_lock);
3348 /* we are not going to use the giving thread, so stop it. */
3350 gst_rtsp_thread_stop (thread);
3351 GST_WARNING ("media %p was not unprepared", media);
3352 priv->prepare_count--;
3353 g_rec_mutex_unlock (&priv->state_lock);
3358 /* we are not going to use the giving thread, so stop it. */
3360 gst_rtsp_thread_stop (thread);
3361 priv->prepare_count--;
3362 g_rec_mutex_unlock (&priv->state_lock);
3363 GST_WARNING ("can not reuse media %p", media);
3368 /* we are not going to use the giving thread, so stop it. */
3370 gst_rtsp_thread_stop (thread);
3371 priv->prepare_count--;
3372 g_rec_mutex_unlock (&priv->state_lock);
3373 GST_ERROR ("failed to prepare media");
3378 GST_WARNING ("failed to preroll pipeline");
3379 gst_rtsp_media_unprepare (media);
3384 /* must be called with state-lock */
3386 finish_unprepare (GstRTSPMedia * media)
3388 GstRTSPMediaPrivate *priv = media->priv;
3392 GST_DEBUG ("shutting down");
3394 /* release the lock on shutdown, otherwise pad_added_cb might try to
3395 * acquire the lock and then we deadlock */
3396 g_rec_mutex_unlock (&priv->state_lock);
3397 set_state (media, GST_STATE_NULL);
3398 g_rec_mutex_lock (&priv->state_lock);
3400 media_streams_set_blocked (media, FALSE);
3402 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
3405 for (i = 0; i < priv->streams->len; i++) {
3406 GstRTSPStream *stream;
3408 GST_INFO ("Removing elements of stream %d from pipeline", i);
3410 stream = g_ptr_array_index (priv->streams, i);
3412 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3415 /* remove the pad signal handlers */
3416 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3417 GstElement *elem = walk->data;
3418 DynPaySignalHandlers *handlers;
3421 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3422 g_assert (handlers != NULL);
3424 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3425 g_signal_handler_disconnect (G_OBJECT (elem),
3426 handlers->pad_removed_handler);
3427 g_signal_handler_disconnect (G_OBJECT (elem),
3428 handlers->no_more_pads_handler);
3430 g_slice_free (DynPaySignalHandlers, handlers);
3433 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3434 priv->rtpbin = NULL;
3437 gst_object_unref (priv->nettime);
3438 priv->nettime = NULL;
3440 priv->reused = TRUE;
3441 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3443 /* when the media is not reusable, this will effectively unref the media and
3445 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3447 /* the source has the last ref to the media */
3449 GST_DEBUG ("destroy source");
3450 g_source_destroy (priv->source);
3451 g_source_unref (priv->source);
3454 GST_DEBUG ("stop thread");
3455 gst_rtsp_thread_stop (priv->thread);
3459 /* called with state-lock */
3461 default_unprepare (GstRTSPMedia * media)
3463 GstRTSPMediaPrivate *priv = media->priv;
3465 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3467 if (priv->eos_shutdown) {
3468 GST_DEBUG ("sending EOS for shutdown");
3469 /* ref so that we don't disappear */
3470 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3471 /* we need to go to playing again for the EOS to propagate, normally in this
3472 * state, nothing is receiving data from us anymore so this is ok. */
3473 set_state (media, GST_STATE_PLAYING);
3475 finish_unprepare (media);
3481 * gst_rtsp_media_unprepare:
3482 * @media: a #GstRTSPMedia
3484 * Unprepare @media. After this call, the media should be prepared again before
3485 * it can be used again. If the media is set to be non-reusable, a new instance
3488 * Returns: %TRUE on success.
3491 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3493 GstRTSPMediaPrivate *priv;
3496 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3500 g_rec_mutex_lock (&priv->state_lock);
3501 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3502 goto was_unprepared;
3504 priv->prepare_count--;
3505 if (priv->prepare_count > 0)
3508 GST_INFO ("unprepare media %p", media);
3509 set_target_state (media, GST_STATE_NULL, FALSE);
3512 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3513 GstRTSPMediaClass *klass;
3515 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3516 if (klass->unprepare)
3517 success = klass->unprepare (media);
3519 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3520 finish_unprepare (media);
3522 g_rec_mutex_unlock (&priv->state_lock);
3528 g_rec_mutex_unlock (&priv->state_lock);
3529 GST_INFO ("media %p was already unprepared", media);
3534 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3535 g_rec_mutex_unlock (&priv->state_lock);
3540 /* should be called with state-lock */
3542 get_clock_unlocked (GstRTSPMedia * media)
3544 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3545 GST_DEBUG_OBJECT (media, "media was not prepared");
3548 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3552 * gst_rtsp_media_get_clock:
3553 * @media: a #GstRTSPMedia
3555 * Get the clock that is used by the pipeline in @media.
3557 * @media must be prepared before this method returns a valid clock object.
3559 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3562 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3565 GstRTSPMediaPrivate *priv;
3567 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3571 g_rec_mutex_lock (&priv->state_lock);
3572 clock = get_clock_unlocked (media);
3573 g_rec_mutex_unlock (&priv->state_lock);
3579 * gst_rtsp_media_get_base_time:
3580 * @media: a #GstRTSPMedia
3582 * Get the base_time that is used by the pipeline in @media.
3584 * @media must be prepared before this method returns a valid base_time.
3586 * Returns: the base_time used by @media.
3589 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3591 GstClockTime result;
3592 GstRTSPMediaPrivate *priv;
3594 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3598 g_rec_mutex_lock (&priv->state_lock);
3599 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3602 result = gst_element_get_base_time (media->priv->pipeline);
3603 g_rec_mutex_unlock (&priv->state_lock);
3610 g_rec_mutex_unlock (&priv->state_lock);
3611 GST_DEBUG_OBJECT (media, "media was not prepared");
3612 return GST_CLOCK_TIME_NONE;
3617 * gst_rtsp_media_get_time_provider:
3618 * @media: a #GstRTSPMedia
3619 * @address: (allow-none): an address or %NULL
3620 * @port: a port or 0
3622 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3623 * will listen on @address and @port for client time requests.
3625 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3627 GstNetTimeProvider *
3628 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3631 GstRTSPMediaPrivate *priv;
3632 GstNetTimeProvider *provider = NULL;
3634 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3638 g_rec_mutex_lock (&priv->state_lock);
3639 if (priv->time_provider) {
3640 if ((provider = priv->nettime) == NULL) {
3643 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3644 provider = gst_net_time_provider_new (clock, address, port);
3645 gst_object_unref (clock);
3647 priv->nettime = provider;
3651 g_rec_mutex_unlock (&priv->state_lock);
3654 gst_object_ref (provider);
3660 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3662 return gst_rtsp_sdp_from_media (sdp, info, media);
3666 * gst_rtsp_media_setup_sdp:
3667 * @media: a #GstRTSPMedia
3668 * @sdp: (transfer none): a #GstSDPMessage
3669 * @info: (transfer none): a #GstSDPInfo
3671 * Add @media specific info to @sdp. @info is used to configure the connection
3672 * information in the SDP.
3674 * Returns: TRUE on success.
3677 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3680 GstRTSPMediaPrivate *priv;
3681 GstRTSPMediaClass *klass;
3684 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3685 g_return_val_if_fail (sdp != NULL, FALSE);
3686 g_return_val_if_fail (info != NULL, FALSE);
3690 g_rec_mutex_lock (&priv->state_lock);
3692 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3694 if (!klass->setup_sdp)
3697 res = klass->setup_sdp (media, sdp, info);
3699 g_rec_mutex_unlock (&priv->state_lock);
3706 g_rec_mutex_unlock (&priv->state_lock);
3707 GST_ERROR ("no setup_sdp function");
3708 g_critical ("no setup_sdp vmethod function set");
3714 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3716 GstRTSPMediaPrivate *priv = media->priv;
3719 medias_len = gst_sdp_message_medias_len (sdp);
3720 if (medias_len != priv->streams->len) {
3721 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3722 priv->streams->len, medias_len);
3726 for (i = 0; i < medias_len; i++) {
3728 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3729 GstRTSPStream *stream;
3730 gint j, formats_len;
3731 const gchar *control;
3732 GstRTSPProfile profile, profiles;
3734 stream = g_ptr_array_index (priv->streams, i);
3736 /* TODO: Should we do something with the other SDP information? */
3739 proto = gst_sdp_media_get_proto (sdp_media);
3740 if (proto == NULL) {
3741 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3745 if (g_str_equal (proto, "RTP/AVP")) {
3746 profile = GST_RTSP_PROFILE_AVP;
3747 } else if (g_str_equal (proto, "RTP/SAVP")) {
3748 profile = GST_RTSP_PROFILE_SAVP;
3749 } else if (g_str_equal (proto, "RTP/AVPF")) {
3750 profile = GST_RTSP_PROFILE_AVPF;
3751 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3752 profile = GST_RTSP_PROFILE_SAVPF;
3754 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3758 profiles = gst_rtsp_stream_get_profiles (stream);
3759 if ((profiles & profile) == 0) {
3760 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3764 formats_len = gst_sdp_media_formats_len (sdp_media);
3765 for (j = 0; j < formats_len; j++) {
3770 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3772 GST_DEBUG (" looking at %d pt: %d", j, pt);
3775 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
3777 GST_WARNING (" skipping pt %d without caps", pt);
3781 /* do some tweaks */
3782 GST_DEBUG ("mapping sdp session level attributes to caps");
3783 gst_sdp_message_attributes_to_caps (sdp, caps);
3784 GST_DEBUG ("mapping sdp media level attributes to caps");
3785 gst_sdp_media_attributes_to_caps (sdp_media, caps);
3787 s = gst_caps_get_structure (caps, 0);
3788 gst_structure_set_name (s, "application/x-rtp");
3790 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3791 gst_caps_unref (caps);
3794 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3796 gst_rtsp_stream_set_control (stream, control);
3804 * gst_rtsp_media_handle_sdp:
3805 * @media: a #GstRTSPMedia
3806 * @sdp: (transfer none): a #GstSDPMessage
3808 * Configure an SDP on @media for receiving streams
3810 * Returns: TRUE on success.
3813 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3815 GstRTSPMediaPrivate *priv;
3816 GstRTSPMediaClass *klass;
3819 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3820 g_return_val_if_fail (sdp != NULL, FALSE);
3824 g_rec_mutex_lock (&priv->state_lock);
3826 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3828 if (!klass->handle_sdp)
3831 res = klass->handle_sdp (media, sdp);
3833 g_rec_mutex_unlock (&priv->state_lock);
3840 g_rec_mutex_unlock (&priv->state_lock);
3841 GST_ERROR ("no handle_sdp function");
3842 g_critical ("no handle_sdp vmethod function set");
3848 do_set_seqnum (GstRTSPStream * stream)
3851 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3852 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3855 /* call with state_lock */
3857 default_suspend (GstRTSPMedia * media)
3859 GstRTSPMediaPrivate *priv = media->priv;
3860 GstStateChangeReturn ret;
3862 switch (priv->suspend_mode) {
3863 case GST_RTSP_SUSPEND_MODE_NONE:
3864 GST_DEBUG ("media %p no suspend", media);
3866 case GST_RTSP_SUSPEND_MODE_PAUSE:
3867 GST_DEBUG ("media %p suspend to PAUSED", media);
3868 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3869 if (ret == GST_STATE_CHANGE_FAILURE)
3872 case GST_RTSP_SUSPEND_MODE_RESET:
3873 GST_DEBUG ("media %p suspend to NULL", media);
3874 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3875 if (ret == GST_STATE_CHANGE_FAILURE)
3877 /* Because payloader needs to set the sequence number as
3878 * monotonic, we need to preserve the sequence number
3879 * after pause. (otherwise going from pause to play, which
3880 * is actually from NULL to PLAY will create a new sequence
3882 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3893 GST_WARNING ("failed changing pipeline's state for media %p", media);
3899 * gst_rtsp_media_suspend:
3900 * @media: a #GstRTSPMedia
3902 * Suspend @media. The state of the pipeline managed by @media is set to
3903 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3904 * with gst_rtsp_media_unsuspend()
3906 * @media must be prepared with gst_rtsp_media_prepare();
3908 * Returns: %TRUE on success.
3911 gst_rtsp_media_suspend (GstRTSPMedia * media)
3913 GstRTSPMediaPrivate *priv = media->priv;
3914 GstRTSPMediaClass *klass;
3916 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3918 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3920 g_rec_mutex_lock (&priv->state_lock);
3921 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3924 /* don't attempt to suspend when something is busy */
3925 if (priv->n_active > 0)
3928 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3929 if (klass->suspend) {
3930 if (!klass->suspend (media))
3931 goto suspend_failed;
3934 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3936 g_rec_mutex_unlock (&priv->state_lock);
3943 g_rec_mutex_unlock (&priv->state_lock);
3944 GST_WARNING ("media %p was not prepared", media);
3949 g_rec_mutex_unlock (&priv->state_lock);
3950 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3951 GST_WARNING ("failed to suspend media %p", media);
3956 /* call with state_lock */
3958 default_unsuspend (GstRTSPMedia * media)
3960 GstRTSPMediaPrivate *priv = media->priv;
3961 gboolean preroll_ok;
3963 switch (priv->suspend_mode) {
3964 case GST_RTSP_SUSPEND_MODE_NONE:
3965 if (is_receive_only (media))
3967 if (media_streams_blocking (media)) {
3968 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3969 /* at this point the media pipeline has been updated and contain all
3970 * specific transport parts: all active streams contain at least one sink
3971 * element and it's safe to unblock any blocked streams that are active */
3972 media_unblock_linked (media);
3974 /* streams are not blocked and media is suspended from PAUSED */
3975 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3977 g_rec_mutex_unlock (&priv->state_lock);
3978 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
3979 g_rec_mutex_lock (&priv->state_lock);
3980 goto preroll_failed;
3982 g_rec_mutex_lock (&priv->state_lock);
3984 case GST_RTSP_SUSPEND_MODE_PAUSE:
3985 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3987 case GST_RTSP_SUSPEND_MODE_RESET:
3989 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3990 /* at this point the media pipeline has been updated and contain all
3991 * specific transport parts: all active streams contain at least one sink
3992 * element and it's safe to unblock any blocked streams that are active */
3993 media_unblock_linked (media);
3994 if (!start_preroll (media))
3997 g_rec_mutex_unlock (&priv->state_lock);
3998 preroll_ok = wait_preroll (media);
3999 g_rec_mutex_lock (&priv->state_lock);
4002 goto preroll_failed;
4013 GST_WARNING ("failed to preroll pipeline");
4018 GST_WARNING ("failed to preroll pipeline");
4024 * gst_rtsp_media_unsuspend:
4025 * @media: a #GstRTSPMedia
4027 * Unsuspend @media if it was in a suspended state. This method does nothing
4028 * when the media was not in the suspended state.
4030 * Returns: %TRUE on success.
4033 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4035 GstRTSPMediaPrivate *priv = media->priv;
4036 GstRTSPMediaClass *klass;
4038 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4040 g_rec_mutex_lock (&priv->state_lock);
4041 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4044 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4045 if (klass->unsuspend) {
4046 if (!klass->unsuspend (media))
4047 goto unsuspend_failed;
4051 g_rec_mutex_unlock (&priv->state_lock);
4058 g_rec_mutex_unlock (&priv->state_lock);
4059 GST_WARNING ("failed to unsuspend media %p", media);
4060 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4065 /* must be called with state-lock */
4067 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4069 GstRTSPMediaPrivate *priv = media->priv;
4071 if (state == GST_STATE_NULL) {
4072 gst_rtsp_media_unprepare (media);
4074 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4075 set_target_state (media, state, FALSE);
4076 /* when we are buffering, don't update the state yet, this will be done
4077 * when buffering finishes */
4078 if (priv->buffering) {
4079 GST_INFO ("Buffering busy, delay state change");
4081 if (state == GST_STATE_PLAYING)
4082 /* make sure pads are not blocking anymore when going to PLAYING */
4083 media_unblock_linked (media);
4085 set_state (media, state);
4087 /* and suspend after pause */
4088 if (state == GST_STATE_PAUSED)
4089 gst_rtsp_media_suspend (media);
4095 * gst_rtsp_media_set_pipeline_state:
4096 * @media: a #GstRTSPMedia
4097 * @state: the target state of the pipeline
4099 * Set the state of the pipeline managed by @media to @state
4102 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4104 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4106 g_rec_mutex_lock (&media->priv->state_lock);
4107 media_set_pipeline_state_locked (media, state);
4108 g_rec_mutex_unlock (&media->priv->state_lock);
4112 * gst_rtsp_media_set_state:
4113 * @media: a #GstRTSPMedia
4114 * @state: the target state of the media
4115 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4116 * a #GPtrArray of #GstRTSPStreamTransport pointers
4118 * Set the state of @media to @state and for the transports in @transports.
4120 * @media must be prepared with gst_rtsp_media_prepare();
4122 * Returns: %TRUE on success.
4125 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4126 GPtrArray * transports)
4128 GstRTSPMediaPrivate *priv;
4130 gboolean activate, deactivate, do_state;
4133 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4134 g_return_val_if_fail (transports != NULL, FALSE);
4138 g_rec_mutex_lock (&priv->state_lock);
4139 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4141 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4142 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4145 /* NULL and READY are the same */
4146 if (state == GST_STATE_READY)
4147 state = GST_STATE_NULL;
4149 activate = deactivate = FALSE;
4151 GST_INFO ("going to state %s media %p, target state %s",
4152 gst_element_state_get_name (state), media,
4153 gst_element_state_get_name (priv->target_state));
4156 case GST_STATE_NULL:
4157 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4158 if (priv->target_state >= GST_STATE_PAUSED)
4161 case GST_STATE_PAUSED:
4162 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4163 if (priv->target_state == GST_STATE_PLAYING)
4166 case GST_STATE_PLAYING:
4167 /* we're going to PLAYING, activate */
4173 old_active = priv->n_active;
4175 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4176 activate, deactivate);
4177 for (i = 0; i < transports->len; i++) {
4178 GstRTSPStreamTransport *trans;
4180 /* we need a non-NULL entry in the array */
4181 trans = g_ptr_array_index (transports, i);
4186 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4188 } else if (deactivate) {
4189 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4194 /* we just activated the first media, do the playing state change */
4195 if (old_active == 0 && activate)
4197 /* if we have no more active media, do the downward state changes */
4198 else if (priv->n_active == 0)
4203 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4206 if (priv->target_state != state) {
4208 media_set_pipeline_state_locked (media, state);
4209 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4214 /* remember where we are */
4215 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4216 old_active != priv->n_active))
4217 collect_media_stats (media);
4219 g_rec_mutex_unlock (&priv->state_lock);
4226 GST_WARNING ("media %p was not prepared", media);
4227 g_rec_mutex_unlock (&priv->state_lock);
4232 GST_WARNING ("media %p in error status while changing to state %d",
4234 if (state == GST_STATE_NULL) {
4235 for (i = 0; i < transports->len; i++) {
4236 GstRTSPStreamTransport *trans;
4238 /* we need a non-NULL entry in the array */
4239 trans = g_ptr_array_index (transports, i);
4243 gst_rtsp_stream_transport_set_active (trans, FALSE);
4247 g_rec_mutex_unlock (&priv->state_lock);
4253 * gst_rtsp_media_set_transport_mode:
4254 * @media: a #GstRTSPMedia
4255 * @mode: the new value
4257 * Sets if the media pipeline can work in PLAY or RECORD mode
4260 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4261 GstRTSPTransportMode mode)
4263 GstRTSPMediaPrivate *priv;
4265 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4269 g_mutex_lock (&priv->lock);
4270 priv->transport_mode = mode;
4271 g_mutex_unlock (&priv->lock);
4275 * gst_rtsp_media_get_transport_mode:
4276 * @media: a #GstRTSPMedia
4278 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4280 * Returns: The transport mode.
4282 GstRTSPTransportMode
4283 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4285 GstRTSPMediaPrivate *priv;
4286 GstRTSPTransportMode res;
4288 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4292 g_mutex_lock (&priv->lock);
4293 res = priv->transport_mode;
4294 g_mutex_unlock (&priv->lock);
4300 * gst_rtsp_media_get_seekable:
4301 * @media: a #GstRTSPMedia
4303 * Check if the pipeline for @media seek and up to what point in time,
4306 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4307 * and > 0 to indicate the longest duration between any two random access points.
4308 * %G_MAXINT64 means any value is possible.
4311 gst_rtsp_media_seekable (GstRTSPMedia * media)
4313 GstRTSPMediaPrivate *priv;
4314 GstClockTimeDiff res;
4316 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4320 /* Currently we are not able to seek on live streams,
4321 * and no stream is seekable only to the beginning */
4322 g_mutex_lock (&priv->lock);
4323 res = priv->seekable;
4324 g_mutex_unlock (&priv->lock);
4330 * gst_rtsp_media_complete_pipeline:
4331 * @media: a #GstRTSPMedia
4332 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4334 * Add a receiver and sender parts to the pipeline based on the transport from
4337 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4340 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4342 GstRTSPMediaPrivate *priv;
4345 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4346 g_return_val_if_fail (transports, FALSE);
4348 GST_DEBUG_OBJECT (media, "complete pipeline");
4352 g_mutex_lock (&priv->lock);
4353 for (i = 0; i < priv->streams->len; i++) {
4354 GstRTSPStreamTransport *transport;
4355 GstRTSPStream *stream;
4356 const GstRTSPTransport *rtsp_transport;
4358 transport = g_ptr_array_index (transports, i);
4362 stream = gst_rtsp_stream_transport_get_stream (transport);
4366 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4368 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4369 g_mutex_unlock (&priv->lock);
4374 priv->complete = TRUE;
4375 g_mutex_unlock (&priv->lock);