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 struct _GstRTSPMediaPrivate
89 /* protected by lock */
90 GstRTSPPermissions *permissions;
92 gboolean suspend_mode;
94 GstRTSPProfile profiles;
95 GstRTSPLowerTrans protocols;
97 gboolean eos_shutdown;
99 GstRTSPAddressPool *pool;
100 gchar *multicast_iface;
102 GstRTSPTransportMode transport_mode;
103 gboolean stop_on_disconnect;
106 GRecMutex state_lock; /* locking order: state lock, lock */
107 GPtrArray *streams; /* protected by lock */
108 GList *dynamic; /* protected by lock */
109 GstRTSPMediaStatus status; /* protected by lock */
114 /* the pipeline for the media */
115 GstElement *pipeline;
118 GstRTSPThread *thread;
119 GList *pending_pipeline_elements;
121 gboolean time_provider;
122 GstNetTimeProvider *nettime;
125 GstClockTimeDiff seekable;
127 GstState target_state;
129 /* RTP session manager */
132 /* the range of media */
133 GstRTSPTimeRange range; /* protected by lock */
134 GstClockTime range_start;
135 GstClockTime range_stop;
137 GList *payloads; /* protected by lock */
138 GstClockTime rtx_time; /* protected by lock */
139 gboolean do_retransmission; /* protected by lock */
140 guint latency; /* protected by lock */
141 GstClock *clock; /* protected by lock */
142 GstRTSPPublishClockMode publish_clock_mode;
144 /* Dynamic element handling */
145 guint nb_dynamic_elements;
146 guint no_more_pads_pending;
149 #define DEFAULT_SHARED FALSE
150 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
151 #define DEFAULT_REUSABLE FALSE
152 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
153 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
154 GST_RTSP_LOWER_TRANS_TCP
155 #define DEFAULT_EOS_SHUTDOWN FALSE
156 #define DEFAULT_BUFFER_SIZE 0x80000
157 #define DEFAULT_TIME_PROVIDER FALSE
158 #define DEFAULT_LATENCY 200
159 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
160 #define DEFAULT_STOP_ON_DISCONNECT TRUE
162 #define DEFAULT_DO_RETRANSMISSION FALSE
164 /* define to dump received RTCP packets */
181 PROP_STOP_ON_DISCONNECT,
189 SIGNAL_REMOVED_STREAM,
197 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
198 #define GST_CAT_DEFAULT rtsp_media_debug
200 static void gst_rtsp_media_get_property (GObject * object, guint propid,
201 GValue * value, GParamSpec * pspec);
202 static void gst_rtsp_media_set_property (GObject * object, guint propid,
203 const GValue * value, GParamSpec * pspec);
204 static void gst_rtsp_media_finalize (GObject * obj);
206 static gboolean default_handle_message (GstRTSPMedia * media,
207 GstMessage * message);
208 static void finish_unprepare (GstRTSPMedia * media);
209 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
210 static gboolean default_unprepare (GstRTSPMedia * media);
211 static gboolean default_suspend (GstRTSPMedia * media);
212 static gboolean default_unsuspend (GstRTSPMedia * media);
213 static gboolean default_convert_range (GstRTSPMedia * media,
214 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
215 static gboolean default_query_position (GstRTSPMedia * media,
217 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
218 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
219 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
221 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
223 static gboolean wait_preroll (GstRTSPMedia * media);
225 static GstElement *find_payload_element (GstElement * payloader);
227 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
229 static gboolean check_complete (GstRTSPMedia * media);
231 #define C_ENUM(v) ((gint) v)
234 gst_rtsp_suspend_mode_get_type (void)
237 static const GEnumValue values[] = {
238 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
239 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
241 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
246 if (g_once_init_enter (&id)) {
247 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
248 g_once_init_leave (&id, tmp);
253 #define C_FLAGS(v) ((guint) v)
256 gst_rtsp_transport_mode_get_type (void)
259 static const GFlagsValue values[] = {
260 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
262 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
267 if (g_once_init_enter (&id)) {
268 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
269 g_once_init_leave (&id, tmp);
275 gst_rtsp_publish_clock_mode_get_type (void)
278 static const GEnumValue values[] = {
279 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
280 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
281 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
282 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
284 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
285 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
290 if (g_once_init_enter (&id)) {
291 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
292 g_once_init_leave (&id, tmp);
297 G_DEFINE_TYPE_WITH_PRIVATE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
300 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
302 GObjectClass *gobject_class;
304 gobject_class = G_OBJECT_CLASS (klass);
306 gobject_class->get_property = gst_rtsp_media_get_property;
307 gobject_class->set_property = gst_rtsp_media_set_property;
308 gobject_class->finalize = gst_rtsp_media_finalize;
310 g_object_class_install_property (gobject_class, PROP_SHARED,
311 g_param_spec_boolean ("shared", "Shared",
312 "If this media pipeline can be shared", DEFAULT_SHARED,
313 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
315 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
316 g_param_spec_enum ("suspend-mode", "Suspend Mode",
317 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
318 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
320 g_object_class_install_property (gobject_class, PROP_REUSABLE,
321 g_param_spec_boolean ("reusable", "Reusable",
322 "If this media pipeline can be reused after an unprepare",
323 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
325 g_object_class_install_property (gobject_class, PROP_PROFILES,
326 g_param_spec_flags ("profiles", "Profiles",
327 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
328 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
330 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
331 g_param_spec_flags ("protocols", "Protocols",
332 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
333 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
336 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
337 "Send an EOS event to the pipeline before unpreparing",
338 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
340 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
341 g_param_spec_uint ("buffer-size", "Buffer Size",
342 "The kernel UDP buffer size to use", 0, G_MAXUINT,
343 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 g_object_class_install_property (gobject_class, PROP_ELEMENT,
346 g_param_spec_object ("element", "The Element",
347 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
348 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
350 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
351 g_param_spec_boolean ("time-provider", "Time Provider",
352 "Use a NetTimeProvider for clients",
353 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_LATENCY,
356 g_param_spec_uint ("latency", "Latency",
357 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
358 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
361 g_param_spec_flags ("transport-mode", "Transport Mode",
362 "If this media pipeline can be used for PLAY or RECORD",
363 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
364 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
366 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
367 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
368 "If this media pipeline should be stopped "
369 "when a client disconnects without TEARDOWN",
370 DEFAULT_STOP_ON_DISCONNECT,
371 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
373 g_object_class_install_property (gobject_class, PROP_CLOCK,
374 g_param_spec_object ("clock", "Clock",
375 "Clock to be used by the media pipeline",
376 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
379 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
380 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
381 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
383 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
384 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
385 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
386 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
387 GST_TYPE_RTSP_STREAM);
389 gst_rtsp_media_signals[SIGNAL_PREPARED] =
390 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
391 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
392 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
394 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
395 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
396 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
397 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
399 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
400 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
401 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
402 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
404 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
405 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
406 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
407 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
409 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
411 klass->handle_message = default_handle_message;
412 klass->prepare = default_prepare;
413 klass->unprepare = default_unprepare;
414 klass->suspend = default_suspend;
415 klass->unsuspend = default_unsuspend;
416 klass->convert_range = default_convert_range;
417 klass->query_position = default_query_position;
418 klass->query_stop = default_query_stop;
419 klass->create_rtpbin = default_create_rtpbin;
420 klass->setup_sdp = default_setup_sdp;
421 klass->handle_sdp = default_handle_sdp;
425 gst_rtsp_media_init (GstRTSPMedia * media)
427 GstRTSPMediaPrivate *priv = gst_rtsp_media_get_instance_private (media);
431 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
432 g_mutex_init (&priv->lock);
433 g_cond_init (&priv->cond);
434 g_rec_mutex_init (&priv->state_lock);
436 priv->shared = DEFAULT_SHARED;
437 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
438 priv->reusable = DEFAULT_REUSABLE;
439 priv->profiles = DEFAULT_PROFILES;
440 priv->protocols = DEFAULT_PROTOCOLS;
441 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
442 priv->buffer_size = DEFAULT_BUFFER_SIZE;
443 priv->time_provider = DEFAULT_TIME_PROVIDER;
444 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
445 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
446 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
447 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
451 gst_rtsp_media_finalize (GObject * obj)
453 GstRTSPMediaPrivate *priv;
456 media = GST_RTSP_MEDIA (obj);
459 GST_INFO ("finalize media %p", media);
461 if (priv->permissions)
462 gst_rtsp_permissions_unref (priv->permissions);
464 g_ptr_array_unref (priv->streams);
466 g_list_free_full (priv->dynamic, gst_object_unref);
467 g_list_free_full (priv->pending_pipeline_elements, gst_object_unref);
470 gst_object_unref (priv->pipeline);
472 gst_object_unref (priv->nettime);
473 gst_object_unref (priv->element);
475 g_object_unref (priv->pool);
477 g_list_free (priv->payloads);
479 gst_object_unref (priv->clock);
480 g_free (priv->multicast_iface);
481 g_mutex_clear (&priv->lock);
482 g_cond_clear (&priv->cond);
483 g_rec_mutex_clear (&priv->state_lock);
485 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
489 gst_rtsp_media_get_property (GObject * object, guint propid,
490 GValue * value, GParamSpec * pspec)
492 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
496 g_value_set_object (value, media->priv->element);
499 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
501 case PROP_SUSPEND_MODE:
502 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
505 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
508 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
511 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
513 case PROP_EOS_SHUTDOWN:
514 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
516 case PROP_BUFFER_SIZE:
517 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
519 case PROP_TIME_PROVIDER:
520 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
523 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
525 case PROP_TRANSPORT_MODE:
526 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
528 case PROP_STOP_ON_DISCONNECT:
529 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
532 g_value_take_object (value, gst_rtsp_media_get_clock (media));
535 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
540 gst_rtsp_media_set_property (GObject * object, guint propid,
541 const GValue * value, GParamSpec * pspec)
543 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
547 media->priv->element = g_value_get_object (value);
548 gst_object_ref_sink (media->priv->element);
551 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
553 case PROP_SUSPEND_MODE:
554 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
557 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
560 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
563 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
565 case PROP_EOS_SHUTDOWN:
566 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
568 case PROP_BUFFER_SIZE:
569 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
571 case PROP_TIME_PROVIDER:
572 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
575 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
577 case PROP_TRANSPORT_MODE:
578 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
580 case PROP_STOP_ON_DISCONNECT:
581 gst_rtsp_media_set_stop_on_disconnect (media,
582 g_value_get_boolean (value));
585 gst_rtsp_media_set_clock (media, g_value_get_object (value));
588 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
595 gboolean complete_streams_only;
597 } DoQueryPositionData;
600 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
604 if (!gst_rtsp_stream_is_sender (stream))
607 if (data->complete_streams_only && !gst_rtsp_stream_is_complete (stream)) {
608 GST_DEBUG_OBJECT (stream, "stream not complete, do not query position");
612 if (gst_rtsp_stream_query_position (stream, &tmp)) {
613 data->position = MIN (data->position, tmp);
617 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
618 GST_TIME_ARGS (data->position));
622 default_query_position (GstRTSPMedia * media, gint64 * position)
624 GstRTSPMediaPrivate *priv;
625 DoQueryPositionData data;
629 data.position = G_MAXINT64;
632 /* if the media is complete, i.e. one or more streams have been configured
633 * with sinks, then we want to query the position on those streams only.
634 * a query on an incmplete stream may return a position that originates from
635 * an earlier preroll */
636 if (check_complete (media))
637 data.complete_streams_only = TRUE;
639 data.complete_streams_only = FALSE;
641 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
644 *position = GST_CLOCK_TIME_NONE;
646 *position = data.position;
658 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
662 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
663 data->stop = MAX (data->stop, tmp);
669 default_query_stop (GstRTSPMedia * media, gint64 * stop)
671 GstRTSPMediaPrivate *priv;
672 DoQueryStopData data;
679 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
687 default_create_rtpbin (GstRTSPMedia * media)
691 rtpbin = gst_element_factory_make ("rtpbin", NULL);
697 is_receive_only (GstRTSPMedia * media)
699 GstRTSPMediaPrivate *priv = media->priv;
700 gboolean recive_only = TRUE;
703 for (i = 0; i < priv->streams->len; i++) {
704 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
705 if (gst_rtsp_stream_is_sender (stream) ||
706 !gst_rtsp_stream_is_receiver (stream)) {
715 /* must be called with state lock */
717 check_seekable (GstRTSPMedia * media)
720 GstRTSPMediaPrivate *priv = media->priv;
722 /* Update the seekable state of the pipeline in case it changed */
723 if (is_receive_only (media)) {
724 /* TODO: Seeking for "receive-only"? */
727 guint i, n = priv->streams->len;
729 for (i = 0; i < n; i++) {
730 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
732 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
733 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
740 query = gst_query_new_seeking (GST_FORMAT_TIME);
741 if (gst_element_query (priv->pipeline, query)) {
746 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
747 priv->seekable = seekable ? G_MAXINT64 : 0;
748 } else if (priv->streams->len) {
749 gboolean seekable = TRUE;
750 guint i, n = priv->streams->len;
752 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
753 for (i = 0; i < n; i++) {
754 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
755 seekable &= gst_rtsp_stream_seekable (stream);
757 priv->seekable = seekable ? G_MAXINT64 : -1;
760 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
762 gst_query_unref (query);
765 /* must be called with state lock */
767 check_complete (GstRTSPMedia * media)
769 GstRTSPMediaPrivate *priv = media->priv;
771 guint i, n = priv->streams->len;
773 for (i = 0; i < n; i++) {
774 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
776 if (gst_rtsp_stream_is_complete (stream))
783 /* must be called with state lock */
785 collect_media_stats (GstRTSPMedia * media)
787 GstRTSPMediaPrivate *priv = media->priv;
788 gint64 position = 0, stop = -1;
790 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
791 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
794 priv->range.unit = GST_RTSP_RANGE_NPT;
796 GST_INFO ("collect media stats");
799 priv->range.min.type = GST_RTSP_TIME_NOW;
800 priv->range.min.seconds = -1;
801 priv->range_start = -1;
802 priv->range.max.type = GST_RTSP_TIME_END;
803 priv->range.max.seconds = -1;
804 priv->range_stop = -1;
806 GstRTSPMediaClass *klass;
809 klass = GST_RTSP_MEDIA_GET_CLASS (media);
811 /* get the position */
813 if (klass->query_position)
814 ret = klass->query_position (media, &position);
817 GST_INFO ("position query failed");
821 /* get the current segment stop */
823 if (klass->query_stop)
824 ret = klass->query_stop (media, &stop);
827 GST_INFO ("stop query failed");
831 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
832 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
834 if (position == -1) {
835 priv->range.min.type = GST_RTSP_TIME_NOW;
836 priv->range.min.seconds = -1;
837 priv->range_start = -1;
839 priv->range.min.type = GST_RTSP_TIME_SECONDS;
840 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
841 priv->range_start = position;
844 priv->range.max.type = GST_RTSP_TIME_END;
845 priv->range.max.seconds = -1;
846 priv->range_stop = -1;
848 priv->range.max.type = GST_RTSP_TIME_SECONDS;
849 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
850 priv->range_stop = stop;
853 check_seekable (media);
858 * gst_rtsp_media_new:
859 * @element: (transfer full): a #GstElement
861 * Create a new #GstRTSPMedia instance. @element is the bin element that
862 * provides the different streams. The #GstRTSPMedia object contains the
863 * element to produce RTP data for one or more related (audio/video/..)
866 * Ownership is taken of @element.
868 * Returns: (transfer full): a new #GstRTSPMedia object.
871 gst_rtsp_media_new (GstElement * element)
873 GstRTSPMedia *result;
875 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
877 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
883 * gst_rtsp_media_get_element:
884 * @media: a #GstRTSPMedia
886 * Get the element that was used when constructing @media.
888 * Returns: (transfer full): a #GstElement. Unref after usage.
891 gst_rtsp_media_get_element (GstRTSPMedia * media)
893 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
895 return gst_object_ref (media->priv->element);
899 * gst_rtsp_media_take_pipeline:
900 * @media: a #GstRTSPMedia
901 * @pipeline: (transfer full): a #GstPipeline
903 * Set @pipeline as the #GstPipeline for @media. Ownership is
904 * taken of @pipeline.
907 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
909 GstRTSPMediaPrivate *priv;
911 GstNetTimeProvider *nettime;
914 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
915 g_return_if_fail (GST_IS_PIPELINE (pipeline));
919 g_mutex_lock (&priv->lock);
920 old = priv->pipeline;
921 priv->pipeline = GST_ELEMENT_CAST (pipeline);
922 nettime = priv->nettime;
923 priv->nettime = NULL;
924 g_mutex_unlock (&priv->lock);
927 gst_object_unref (old);
930 gst_object_unref (nettime);
932 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
934 for (l = priv->pending_pipeline_elements; l; l = l->next) {
935 gst_bin_add (GST_BIN_CAST (pipeline), l->data);
937 g_list_free (priv->pending_pipeline_elements);
938 priv->pending_pipeline_elements = NULL;
942 * gst_rtsp_media_set_permissions:
943 * @media: a #GstRTSPMedia
944 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
946 * Set @permissions on @media.
949 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
950 GstRTSPPermissions * permissions)
952 GstRTSPMediaPrivate *priv;
954 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
958 g_mutex_lock (&priv->lock);
959 if (priv->permissions)
960 gst_rtsp_permissions_unref (priv->permissions);
961 if ((priv->permissions = permissions))
962 gst_rtsp_permissions_ref (permissions);
963 g_mutex_unlock (&priv->lock);
967 * gst_rtsp_media_get_permissions:
968 * @media: a #GstRTSPMedia
970 * Get the permissions object from @media.
972 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
975 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
977 GstRTSPMediaPrivate *priv;
978 GstRTSPPermissions *result;
980 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
984 g_mutex_lock (&priv->lock);
985 if ((result = priv->permissions))
986 gst_rtsp_permissions_ref (result);
987 g_mutex_unlock (&priv->lock);
993 * gst_rtsp_media_set_suspend_mode:
994 * @media: a #GstRTSPMedia
995 * @mode: the new #GstRTSPSuspendMode
997 * Control how @ media will be suspended after the SDP has been generated and
998 * after a PAUSE request has been performed.
1000 * Media must be unprepared when setting the suspend mode.
1003 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
1005 GstRTSPMediaPrivate *priv;
1007 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1011 g_rec_mutex_lock (&priv->state_lock);
1012 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
1014 priv->suspend_mode = mode;
1015 g_rec_mutex_unlock (&priv->state_lock);
1022 GST_WARNING ("media %p was prepared", media);
1023 g_rec_mutex_unlock (&priv->state_lock);
1028 * gst_rtsp_media_get_suspend_mode:
1029 * @media: a #GstRTSPMedia
1031 * Get how @media will be suspended.
1033 * Returns: #GstRTSPSuspendMode.
1036 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
1038 GstRTSPMediaPrivate *priv;
1039 GstRTSPSuspendMode res;
1041 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
1045 g_rec_mutex_lock (&priv->state_lock);
1046 res = priv->suspend_mode;
1047 g_rec_mutex_unlock (&priv->state_lock);
1053 * gst_rtsp_media_set_shared:
1054 * @media: a #GstRTSPMedia
1055 * @shared: the new value
1057 * Set or unset if the pipeline for @media can be shared will multiple clients.
1058 * When @shared is %TRUE, client requests for this media will share the media
1062 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1064 GstRTSPMediaPrivate *priv;
1066 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1070 g_mutex_lock (&priv->lock);
1071 priv->shared = shared;
1072 g_mutex_unlock (&priv->lock);
1076 * gst_rtsp_media_is_shared:
1077 * @media: a #GstRTSPMedia
1079 * Check if the pipeline for @media can be shared between multiple clients.
1081 * Returns: %TRUE if the media can be shared between clients.
1084 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1086 GstRTSPMediaPrivate *priv;
1089 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1093 g_mutex_lock (&priv->lock);
1095 g_mutex_unlock (&priv->lock);
1101 * gst_rtsp_media_set_reusable:
1102 * @media: a #GstRTSPMedia
1103 * @reusable: the new value
1105 * Set or unset if the pipeline for @media can be reused after the pipeline has
1109 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1111 GstRTSPMediaPrivate *priv;
1113 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1117 g_mutex_lock (&priv->lock);
1118 priv->reusable = reusable;
1119 g_mutex_unlock (&priv->lock);
1123 * gst_rtsp_media_is_reusable:
1124 * @media: a #GstRTSPMedia
1126 * Check if the pipeline for @media can be reused after an unprepare.
1128 * Returns: %TRUE if the media can be reused
1131 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1133 GstRTSPMediaPrivate *priv;
1136 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1140 g_mutex_lock (&priv->lock);
1141 res = priv->reusable;
1142 g_mutex_unlock (&priv->lock);
1148 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1150 gst_rtsp_stream_set_profiles (stream, *profiles);
1154 * gst_rtsp_media_set_profiles:
1155 * @media: a #GstRTSPMedia
1156 * @profiles: the new flags
1158 * Configure the allowed lower transport for @media.
1161 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1163 GstRTSPMediaPrivate *priv;
1165 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1169 g_mutex_lock (&priv->lock);
1170 priv->profiles = profiles;
1171 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1172 g_mutex_unlock (&priv->lock);
1176 * gst_rtsp_media_get_profiles:
1177 * @media: a #GstRTSPMedia
1179 * Get the allowed profiles of @media.
1181 * Returns: a #GstRTSPProfile
1184 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1186 GstRTSPMediaPrivate *priv;
1189 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1193 g_mutex_lock (&priv->lock);
1194 res = priv->profiles;
1195 g_mutex_unlock (&priv->lock);
1201 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1203 gst_rtsp_stream_set_protocols (stream, *protocols);
1207 * gst_rtsp_media_set_protocols:
1208 * @media: a #GstRTSPMedia
1209 * @protocols: the new flags
1211 * Configure the allowed lower transport for @media.
1214 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1216 GstRTSPMediaPrivate *priv;
1218 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1222 g_mutex_lock (&priv->lock);
1223 priv->protocols = protocols;
1224 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1225 g_mutex_unlock (&priv->lock);
1229 * gst_rtsp_media_get_protocols:
1230 * @media: a #GstRTSPMedia
1232 * Get the allowed protocols of @media.
1234 * Returns: a #GstRTSPLowerTrans
1237 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1239 GstRTSPMediaPrivate *priv;
1240 GstRTSPLowerTrans res;
1242 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1243 GST_RTSP_LOWER_TRANS_UNKNOWN);
1247 g_mutex_lock (&priv->lock);
1248 res = priv->protocols;
1249 g_mutex_unlock (&priv->lock);
1255 * gst_rtsp_media_set_eos_shutdown:
1256 * @media: a #GstRTSPMedia
1257 * @eos_shutdown: the new value
1259 * Set or unset if an EOS event will be sent to the pipeline for @media before
1263 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1265 GstRTSPMediaPrivate *priv;
1267 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1271 g_mutex_lock (&priv->lock);
1272 priv->eos_shutdown = eos_shutdown;
1273 g_mutex_unlock (&priv->lock);
1277 * gst_rtsp_media_is_eos_shutdown:
1278 * @media: a #GstRTSPMedia
1280 * Check if the pipeline for @media will send an EOS down the pipeline before
1283 * Returns: %TRUE if the media will send EOS before unpreparing.
1286 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1288 GstRTSPMediaPrivate *priv;
1291 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1295 g_mutex_lock (&priv->lock);
1296 res = priv->eos_shutdown;
1297 g_mutex_unlock (&priv->lock);
1303 * gst_rtsp_media_set_buffer_size:
1304 * @media: a #GstRTSPMedia
1305 * @size: the new value
1307 * Set the kernel UDP buffer size.
1310 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1312 GstRTSPMediaPrivate *priv;
1315 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1317 GST_LOG_OBJECT (media, "set buffer size %u", size);
1321 g_mutex_lock (&priv->lock);
1322 priv->buffer_size = size;
1324 for (i = 0; i < priv->streams->len; i++) {
1325 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1326 gst_rtsp_stream_set_buffer_size (stream, size);
1328 g_mutex_unlock (&priv->lock);
1332 * gst_rtsp_media_get_buffer_size:
1333 * @media: a #GstRTSPMedia
1335 * Get the kernel UDP buffer size.
1337 * Returns: the kernel UDP buffer size.
1340 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1342 GstRTSPMediaPrivate *priv;
1345 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1349 g_mutex_lock (&priv->lock);
1350 res = priv->buffer_size;
1351 g_mutex_unlock (&priv->lock);
1357 * gst_rtsp_media_set_stop_on_disconnect:
1358 * @media: a #GstRTSPMedia
1359 * @stop_on_disconnect: the new value
1361 * Set or unset if the pipeline for @media should be stopped when a
1362 * client disconnects without sending TEARDOWN.
1365 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1366 gboolean stop_on_disconnect)
1368 GstRTSPMediaPrivate *priv;
1370 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1374 g_mutex_lock (&priv->lock);
1375 priv->stop_on_disconnect = stop_on_disconnect;
1376 g_mutex_unlock (&priv->lock);
1380 * gst_rtsp_media_is_stop_on_disconnect:
1381 * @media: a #GstRTSPMedia
1383 * Check if the pipeline for @media will be stopped when a client disconnects
1384 * without sending TEARDOWN.
1386 * Returns: %TRUE if the media will be stopped when a client disconnects
1387 * without sending TEARDOWN.
1390 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1392 GstRTSPMediaPrivate *priv;
1395 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1399 g_mutex_lock (&priv->lock);
1400 res = priv->stop_on_disconnect;
1401 g_mutex_unlock (&priv->lock);
1407 * gst_rtsp_media_set_retransmission_time:
1408 * @media: a #GstRTSPMedia
1409 * @time: the new value
1411 * Set the amount of time to store retransmission packets.
1414 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1416 GstRTSPMediaPrivate *priv;
1419 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1421 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1425 g_mutex_lock (&priv->lock);
1426 priv->rtx_time = time;
1427 for (i = 0; i < priv->streams->len; i++) {
1428 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1430 gst_rtsp_stream_set_retransmission_time (stream, time);
1432 g_mutex_unlock (&priv->lock);
1436 * gst_rtsp_media_get_retransmission_time:
1437 * @media: a #GstRTSPMedia
1439 * Get the amount of time to store retransmission data.
1441 * Returns: the amount of time to store retransmission data.
1444 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1446 GstRTSPMediaPrivate *priv;
1449 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1453 g_mutex_lock (&priv->lock);
1454 res = priv->rtx_time;
1455 g_mutex_unlock (&priv->lock);
1461 * gst_rtsp_media_set_do_retransmission:
1463 * Set whether retransmission requests will be sent
1468 gst_rtsp_media_set_do_retransmission (GstRTSPMedia * media,
1469 gboolean do_retransmission)
1471 GstRTSPMediaPrivate *priv;
1473 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1477 g_mutex_lock (&priv->lock);
1478 priv->do_retransmission = do_retransmission;
1481 g_object_set (priv->rtpbin, "do-retransmission", do_retransmission, NULL);
1482 g_mutex_unlock (&priv->lock);
1486 * gst_rtsp_media_get_do_retransmission:
1488 * Returns: Whether retransmission requests will be sent
1493 gst_rtsp_media_get_do_retransmission (GstRTSPMedia * media)
1495 GstRTSPMediaPrivate *priv;
1498 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1502 g_mutex_lock (&priv->lock);
1503 res = priv->do_retransmission;
1504 g_mutex_unlock (&priv->lock);
1510 * gst_rtsp_media_set_latency:
1511 * @media: a #GstRTSPMedia
1512 * @latency: latency in milliseconds
1514 * Configure the latency used for receiving media.
1517 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1519 GstRTSPMediaPrivate *priv;
1522 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1524 GST_LOG_OBJECT (media, "set latency %ums", latency);
1528 g_mutex_lock (&priv->lock);
1529 priv->latency = latency;
1531 g_object_set (priv->rtpbin, "latency", latency, NULL);
1533 for (i = 0; i < media->priv->streams->len; i++) {
1534 GObject *storage = NULL;
1536 g_signal_emit_by_name (G_OBJECT (media->priv->rtpbin), "get-storage",
1539 g_object_set (storage, "size-time",
1540 (media->priv->latency + 50) * GST_MSECOND, NULL);
1544 g_mutex_unlock (&priv->lock);
1548 * gst_rtsp_media_get_latency:
1549 * @media: a #GstRTSPMedia
1551 * Get the latency that is used for receiving media.
1553 * Returns: latency in milliseconds
1556 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1558 GstRTSPMediaPrivate *priv;
1561 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1565 g_mutex_lock (&priv->lock);
1566 res = priv->latency;
1567 g_mutex_unlock (&priv->lock);
1573 * gst_rtsp_media_use_time_provider:
1574 * @media: a #GstRTSPMedia
1575 * @time_provider: if a #GstNetTimeProvider should be used
1577 * Set @media to provide a #GstNetTimeProvider.
1580 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1582 GstRTSPMediaPrivate *priv;
1584 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1588 g_mutex_lock (&priv->lock);
1589 priv->time_provider = time_provider;
1590 g_mutex_unlock (&priv->lock);
1594 * gst_rtsp_media_is_time_provider:
1595 * @media: a #GstRTSPMedia
1597 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1599 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1601 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1604 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1606 GstRTSPMediaPrivate *priv;
1609 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1613 g_mutex_lock (&priv->lock);
1614 res = priv->time_provider;
1615 g_mutex_unlock (&priv->lock);
1621 * gst_rtsp_media_set_clock:
1622 * @media: a #GstRTSPMedia
1623 * @clock: (nullable): #GstClock to be used
1625 * Configure the clock used for the media.
1628 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1630 GstRTSPMediaPrivate *priv;
1632 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1633 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1635 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1639 g_mutex_lock (&priv->lock);
1641 gst_object_unref (priv->clock);
1642 priv->clock = clock ? gst_object_ref (clock) : NULL;
1643 if (priv->pipeline) {
1645 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1647 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1650 g_mutex_unlock (&priv->lock);
1654 * gst_rtsp_media_set_publish_clock_mode:
1655 * @media: a #GstRTSPMedia
1656 * @mode: the clock publish mode
1658 * Sets if and how the media clock should be published according to RFC7273.
1663 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1664 GstRTSPPublishClockMode mode)
1666 GstRTSPMediaPrivate *priv;
1670 g_mutex_lock (&priv->lock);
1671 priv->publish_clock_mode = mode;
1673 n = priv->streams->len;
1674 for (i = 0; i < n; i++) {
1675 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1677 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1679 g_mutex_unlock (&priv->lock);
1683 * gst_rtsp_media_get_publish_clock_mode:
1684 * @media: a #GstRTSPMedia
1686 * Gets if and how the media clock should be published according to RFC7273.
1688 * Returns: The GstRTSPPublishClockMode
1692 GstRTSPPublishClockMode
1693 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1695 GstRTSPMediaPrivate *priv;
1696 GstRTSPPublishClockMode ret;
1699 g_mutex_lock (&priv->lock);
1700 ret = priv->publish_clock_mode;
1701 g_mutex_unlock (&priv->lock);
1707 * gst_rtsp_media_set_address_pool:
1708 * @media: a #GstRTSPMedia
1709 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1711 * configure @pool to be used as the address pool of @media.
1714 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1715 GstRTSPAddressPool * pool)
1717 GstRTSPMediaPrivate *priv;
1718 GstRTSPAddressPool *old;
1720 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1724 GST_LOG_OBJECT (media, "set address pool %p", pool);
1726 g_mutex_lock (&priv->lock);
1727 if ((old = priv->pool) != pool)
1728 priv->pool = pool ? g_object_ref (pool) : NULL;
1731 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1733 g_mutex_unlock (&priv->lock);
1736 g_object_unref (old);
1740 * gst_rtsp_media_get_address_pool:
1741 * @media: a #GstRTSPMedia
1743 * Get the #GstRTSPAddressPool used as the address pool of @media.
1745 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1746 * g_object_unref() after usage.
1748 GstRTSPAddressPool *
1749 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1751 GstRTSPMediaPrivate *priv;
1752 GstRTSPAddressPool *result;
1754 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1758 g_mutex_lock (&priv->lock);
1759 if ((result = priv->pool))
1760 g_object_ref (result);
1761 g_mutex_unlock (&priv->lock);
1767 * gst_rtsp_media_set_multicast_iface:
1768 * @media: a #GstRTSPMedia
1769 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1771 * configure @multicast_iface to be used for @media.
1774 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1775 const gchar * multicast_iface)
1777 GstRTSPMediaPrivate *priv;
1780 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1784 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1786 g_mutex_lock (&priv->lock);
1787 if ((old = priv->multicast_iface) != multicast_iface)
1788 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1791 g_ptr_array_foreach (priv->streams,
1792 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1793 g_mutex_unlock (&priv->lock);
1800 * gst_rtsp_media_get_multicast_iface:
1801 * @media: a #GstRTSPMedia
1803 * Get the multicast interface used for @media.
1805 * Returns: (transfer full) (nullable): the multicast interface for @media.
1806 * g_free() after usage.
1809 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1811 GstRTSPMediaPrivate *priv;
1814 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1818 g_mutex_lock (&priv->lock);
1819 if ((result = priv->multicast_iface))
1820 result = g_strdup (result);
1821 g_mutex_unlock (&priv->lock);
1827 _find_payload_types (GstRTSPMedia * media)
1830 GQueue queue = G_QUEUE_INIT;
1832 n = media->priv->streams->len;
1833 for (i = 0; i < n; i++) {
1834 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1835 guint pt = gst_rtsp_stream_get_pt (stream);
1837 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1844 _next_available_pt (GList * payloads)
1848 for (i = 96; i <= 127; i++) {
1849 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1851 return GPOINTER_TO_UINT (i);
1858 * gst_rtsp_media_collect_streams:
1859 * @media: a #GstRTSPMedia
1861 * Find all payloader elements, they should be named pay\%d in the
1862 * element of @media, and create #GstRTSPStreams for them.
1864 * Collect all dynamic elements, named dynpay\%d, and add them to
1865 * the list of dynamic elements.
1867 * Find all depayloader elements, they should be named depay\%d in the
1868 * element of @media, and create #GstRTSPStreams for them.
1871 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1873 GstRTSPMediaPrivate *priv;
1874 GstElement *element, *elem;
1878 gboolean more_elem_remaining = TRUE;
1879 GstRTSPTransportMode mode = 0;
1881 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1884 element = priv->element;
1887 for (i = 0; more_elem_remaining; i++) {
1890 more_elem_remaining = FALSE;
1892 name = g_strdup_printf ("pay%d", i);
1893 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1895 GST_INFO ("found stream %d with payloader %p", i, elem);
1897 /* take the pad of the payloader */
1898 pad = gst_element_get_static_pad (elem, "src");
1900 /* find the real payload element in case elem is a GstBin */
1901 pay = find_payload_element (elem);
1903 /* create the stream */
1905 GST_WARNING ("could not find real payloader, using bin");
1906 gst_rtsp_media_create_stream (media, elem, pad);
1908 gst_rtsp_media_create_stream (media, pay, pad);
1909 gst_object_unref (pay);
1912 gst_object_unref (pad);
1913 gst_object_unref (elem);
1916 more_elem_remaining = TRUE;
1917 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1921 name = g_strdup_printf ("dynpay%d", i);
1922 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1923 /* a stream that will dynamically create pads to provide RTP packets */
1924 GST_INFO ("found dynamic element %d, %p", i, elem);
1926 g_mutex_lock (&priv->lock);
1927 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1928 g_mutex_unlock (&priv->lock);
1930 priv->nb_dynamic_elements++;
1933 more_elem_remaining = TRUE;
1934 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1938 name = g_strdup_printf ("depay%d", i);
1939 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1940 GST_INFO ("found stream %d with depayloader %p", i, elem);
1942 /* take the pad of the payloader */
1943 pad = gst_element_get_static_pad (elem, "sink");
1944 /* create the stream */
1945 gst_rtsp_media_create_stream (media, elem, pad);
1946 gst_object_unref (pad);
1947 gst_object_unref (elem);
1950 more_elem_remaining = TRUE;
1951 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1957 if (priv->transport_mode != mode)
1958 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1959 priv->transport_mode, mode);
1965 GstElement *appsink, *appsrc;
1966 GstRTSPStream *stream;
1969 static GstFlowReturn
1970 appsink_new_sample (GstAppSink * appsink, gpointer user_data)
1972 AppSinkSrcData *data = user_data;
1976 sample = gst_app_sink_pull_sample (appsink);
1978 return GST_FLOW_FLUSHING;
1981 ret = gst_app_src_push_sample (GST_APP_SRC (data->appsrc), sample);
1982 gst_sample_unref (sample);
1986 static GstAppSinkCallbacks appsink_callbacks = {
1992 static GstPadProbeReturn
1993 appsink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
1995 AppSinkSrcData *data = user_data;
1997 if (GST_IS_EVENT (info->data)
1998 && GST_EVENT_TYPE (info->data) == GST_EVENT_LATENCY) {
1999 GstClockTime min, max;
2001 if (gst_base_sink_query_latency (GST_BASE_SINK (data->appsink), NULL, NULL,
2003 g_object_set (data->appsrc, "min-latency", min, "max-latency", max, NULL);
2004 GST_DEBUG ("setting latency to min %" GST_TIME_FORMAT " max %"
2005 GST_TIME_FORMAT, GST_TIME_ARGS (min), GST_TIME_ARGS (max));
2007 } else if (GST_IS_QUERY (info->data)) {
2008 GstPad *srcpad = gst_element_get_static_pad (data->appsrc, "src");
2009 if (gst_pad_peer_query (srcpad, GST_QUERY_CAST (info->data))) {
2010 gst_object_unref (srcpad);
2011 return GST_PAD_PROBE_HANDLED;
2013 gst_object_unref (srcpad);
2016 return GST_PAD_PROBE_OK;
2019 static GstPadProbeReturn
2020 appsrc_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2022 AppSinkSrcData *data = user_data;
2024 if (GST_IS_QUERY (info->data)) {
2025 GstPad *sinkpad = gst_element_get_static_pad (data->appsink, "sink");
2026 if (gst_pad_peer_query (sinkpad, GST_QUERY_CAST (info->data))) {
2027 gst_object_unref (sinkpad);
2028 return GST_PAD_PROBE_HANDLED;
2030 gst_object_unref (sinkpad);
2033 return GST_PAD_PROBE_OK;
2037 * gst_rtsp_media_create_stream:
2038 * @media: a #GstRTSPMedia
2039 * @payloader: a #GstElement
2042 * Create a new stream in @media that provides RTP data on @pad.
2043 * @pad should be a pad of an element inside @media->element.
2045 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
2049 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
2052 GstRTSPMediaPrivate *priv;
2053 GstRTSPStream *stream;
2057 AppSinkSrcData *data = NULL;
2059 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2060 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
2061 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
2065 g_mutex_lock (&priv->lock);
2066 idx = priv->streams->len;
2068 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
2070 if (GST_PAD_IS_SRC (pad))
2071 name = g_strdup_printf ("src_%u", idx);
2073 name = g_strdup_printf ("sink_%u", idx);
2075 if ((GST_PAD_IS_SRC (pad) && priv->element->numsinkpads > 0) ||
2076 (GST_PAD_IS_SINK (pad) && priv->element->numsrcpads > 0)) {
2077 GstElement *appsink, *appsrc;
2078 GstPad *sinkpad, *srcpad;
2080 appsink = gst_element_factory_make ("appsink", NULL);
2081 appsrc = gst_element_factory_make ("appsrc", NULL);
2083 if (GST_PAD_IS_SINK (pad)) {
2084 srcpad = gst_element_get_static_pad (appsrc, "src");
2086 gst_bin_add (GST_BIN (priv->element), appsrc);
2088 gst_pad_link (srcpad, pad);
2089 gst_object_unref (srcpad);
2091 streampad = gst_element_get_static_pad (appsink, "sink");
2093 priv->pending_pipeline_elements =
2094 g_list_prepend (priv->pending_pipeline_elements, appsink);
2096 sinkpad = gst_element_get_static_pad (appsink, "sink");
2098 gst_pad_link (pad, sinkpad);
2099 gst_object_unref (sinkpad);
2101 streampad = gst_element_get_static_pad (appsrc, "src");
2103 priv->pending_pipeline_elements =
2104 g_list_prepend (priv->pending_pipeline_elements, appsrc);
2107 g_object_set (appsrc, "block", TRUE, "format", GST_FORMAT_TIME, "is-live",
2109 g_object_set (appsink, "sync", FALSE, "async", FALSE, NULL);
2111 data = g_new0 (AppSinkSrcData, 1);
2112 data->appsink = appsink;
2113 data->appsrc = appsrc;
2115 sinkpad = gst_element_get_static_pad (appsink, "sink");
2116 gst_pad_add_probe (sinkpad,
2117 GST_PAD_PROBE_TYPE_EVENT_UPSTREAM | GST_PAD_PROBE_TYPE_QUERY_DOWNSTREAM,
2118 appsink_pad_probe, data, NULL);
2119 gst_object_unref (sinkpad);
2121 srcpad = gst_element_get_static_pad (appsrc, "src");
2122 gst_pad_add_probe (srcpad, GST_PAD_PROBE_TYPE_QUERY_UPSTREAM,
2123 appsrc_pad_probe, data, NULL);
2124 gst_object_unref (srcpad);
2126 gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_callbacks,
2128 g_object_set_data_full (G_OBJECT (streampad), "media-appsink-appsrc", data,
2131 streampad = gst_ghost_pad_new (name, pad);
2132 gst_pad_set_active (streampad, TRUE);
2133 gst_element_add_pad (priv->element, streampad);
2137 stream = gst_rtsp_stream_new (idx, payloader, streampad);
2139 data->stream = stream;
2141 gst_rtsp_stream_set_address_pool (stream, priv->pool);
2142 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
2143 gst_rtsp_stream_set_profiles (stream, priv->profiles);
2144 gst_rtsp_stream_set_protocols (stream, priv->protocols);
2145 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
2146 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
2147 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
2149 g_ptr_array_add (priv->streams, stream);
2151 if (GST_PAD_IS_SRC (pad)) {
2155 g_list_free (priv->payloads);
2156 priv->payloads = _find_payload_types (media);
2158 n = priv->streams->len;
2159 for (i = 0; i < n; i++) {
2160 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
2161 guint rtx_pt = _next_available_pt (priv->payloads);
2164 GST_WARNING ("Ran out of space of dynamic payload types");
2168 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
2171 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
2174 g_mutex_unlock (&priv->lock);
2176 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
2183 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
2185 GstRTSPMediaPrivate *priv;
2187 AppSinkSrcData *data;
2191 g_mutex_lock (&priv->lock);
2192 /* remove the ghostpad */
2193 srcpad = gst_rtsp_stream_get_srcpad (stream);
2194 data = g_object_get_data (G_OBJECT (srcpad), "media-appsink-appsrc");
2196 if (GST_OBJECT_PARENT (data->appsrc) == GST_OBJECT_CAST (priv->pipeline))
2197 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsrc);
2198 else if (GST_OBJECT_PARENT (data->appsrc) ==
2199 GST_OBJECT_CAST (priv->element))
2200 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsrc);
2201 if (GST_OBJECT_PARENT (data->appsink) == GST_OBJECT_CAST (priv->pipeline))
2202 gst_bin_remove (GST_BIN_CAST (priv->pipeline), data->appsink);
2203 else if (GST_OBJECT_PARENT (data->appsink) ==
2204 GST_OBJECT_CAST (priv->element))
2205 gst_bin_remove (GST_BIN_CAST (priv->element), data->appsink);
2207 gst_element_remove_pad (priv->element, srcpad);
2209 gst_object_unref (srcpad);
2210 /* now remove the stream */
2211 g_object_ref (stream);
2212 g_ptr_array_remove (priv->streams, stream);
2213 g_mutex_unlock (&priv->lock);
2215 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
2218 g_object_unref (stream);
2222 * gst_rtsp_media_n_streams:
2223 * @media: a #GstRTSPMedia
2225 * Get the number of streams in this media.
2227 * Returns: The number of streams.
2230 gst_rtsp_media_n_streams (GstRTSPMedia * media)
2232 GstRTSPMediaPrivate *priv;
2235 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
2239 g_mutex_lock (&priv->lock);
2240 res = priv->streams->len;
2241 g_mutex_unlock (&priv->lock);
2247 * gst_rtsp_media_get_stream:
2248 * @media: a #GstRTSPMedia
2249 * @idx: the stream index
2251 * Retrieve the stream with index @idx from @media.
2253 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
2254 * @idx or %NULL when a stream with that index did not exist.
2257 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2259 GstRTSPMediaPrivate *priv;
2262 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2266 g_mutex_lock (&priv->lock);
2267 if (idx < priv->streams->len)
2268 res = g_ptr_array_index (priv->streams, idx);
2271 g_mutex_unlock (&priv->lock);
2277 * gst_rtsp_media_find_stream:
2278 * @media: a #GstRTSPMedia
2279 * @control: the control of the stream
2281 * Find a stream in @media with @control as the control uri.
2283 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2284 * control uri @control or %NULL when a stream with that control did
2288 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2290 GstRTSPMediaPrivate *priv;
2294 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2295 g_return_val_if_fail (control != NULL, NULL);
2301 g_mutex_lock (&priv->lock);
2302 for (i = 0; i < priv->streams->len; i++) {
2303 GstRTSPStream *test;
2305 test = g_ptr_array_index (priv->streams, i);
2306 if (gst_rtsp_stream_has_control (test, control)) {
2311 g_mutex_unlock (&priv->lock);
2316 /* called with state-lock */
2318 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2319 GstRTSPRangeUnit unit)
2321 return gst_rtsp_range_convert_units (range, unit);
2325 * gst_rtsp_media_get_range_string:
2326 * @media: a #GstRTSPMedia
2327 * @play: for the PLAY request
2328 * @unit: the unit to use for the string
2330 * Get the current range as a string. @media must be prepared with
2331 * gst_rtsp_media_prepare ().
2333 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2336 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2337 GstRTSPRangeUnit unit)
2339 GstRTSPMediaClass *klass;
2340 GstRTSPMediaPrivate *priv;
2342 GstRTSPTimeRange range;
2344 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2345 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2346 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2350 g_rec_mutex_lock (&priv->state_lock);
2351 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2352 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2355 g_mutex_lock (&priv->lock);
2357 /* Update the range value with current position/duration */
2358 collect_media_stats (media);
2361 range = priv->range;
2363 if (!play && priv->n_active > 0) {
2364 range.min.type = GST_RTSP_TIME_NOW;
2365 range.min.seconds = -1;
2367 g_mutex_unlock (&priv->lock);
2368 g_rec_mutex_unlock (&priv->state_lock);
2370 if (!klass->convert_range (media, &range, unit))
2371 goto conversion_failed;
2373 result = gst_rtsp_range_to_string (&range);
2380 GST_WARNING ("media %p was not prepared", media);
2381 g_rec_mutex_unlock (&priv->state_lock);
2386 GST_WARNING ("range conversion to unit %d failed", unit);
2392 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2394 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2398 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2400 GstRTSPMediaPrivate *priv = media->priv;
2402 GST_DEBUG ("media %p set blocked %d", media, blocked);
2403 priv->blocked = blocked;
2404 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2408 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2410 gst_rtsp_stream_unblock_linked (stream);
2414 media_unblock_linked (GstRTSPMedia * media)
2416 GstRTSPMediaPrivate *priv = media->priv;
2418 GST_DEBUG ("media %p unblocking linked streams", media);
2419 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2423 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2425 GstRTSPMediaPrivate *priv = media->priv;
2427 g_mutex_lock (&priv->lock);
2428 priv->status = status;
2429 GST_DEBUG ("setting new status to %d", status);
2430 g_cond_broadcast (&priv->cond);
2431 g_mutex_unlock (&priv->lock);
2435 * gst_rtsp_media_get_status:
2436 * @media: a #GstRTSPMedia
2438 * Get the status of @media. When @media is busy preparing, this function waits
2439 * until @media is prepared or in error.
2441 * Returns: the status of @media.
2444 gst_rtsp_media_get_status (GstRTSPMedia * media)
2446 GstRTSPMediaPrivate *priv = media->priv;
2447 GstRTSPMediaStatus result;
2450 g_mutex_lock (&priv->lock);
2451 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2452 /* while we are preparing, wait */
2453 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2454 GST_DEBUG ("waiting for status change");
2455 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2456 GST_DEBUG ("timeout, assuming error status");
2457 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2460 /* could be success or error */
2461 result = priv->status;
2462 GST_DEBUG ("got status %d", result);
2463 g_mutex_unlock (&priv->lock);
2469 * gst_rtsp_media_seek_full:
2470 * @media: a #GstRTSPMedia
2471 * @range: (transfer none): a #GstRTSPTimeRange
2472 * @flags: The minimal set of #GstSeekFlags to use
2474 * Seek the pipeline of @media to @range. @media must be prepared with
2475 * gst_rtsp_media_prepare().
2477 * Returns: %TRUE on success.
2480 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2483 GstRTSPMediaClass *klass;
2484 GstRTSPMediaPrivate *priv;
2486 GstClockTime start, stop;
2487 GstSeekType start_type, stop_type;
2488 gint64 current_position;
2490 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2492 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2493 g_return_val_if_fail (range != NULL, FALSE);
2494 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2498 g_rec_mutex_lock (&priv->state_lock);
2499 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2502 /* check if the media pipeline is complete in order to perform a
2503 * seek operation on it */
2504 if (!check_complete (media))
2507 /* Update the seekable state of the pipeline in case it changed */
2508 check_seekable (media);
2510 if (priv->seekable == 0) {
2511 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2512 " not seekable streams.");
2515 } else if (priv->seekable < 0) {
2519 start_type = stop_type = GST_SEEK_TYPE_NONE;
2521 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2523 gst_rtsp_range_get_times (range, &start, &stop);
2525 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2526 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2527 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2528 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2530 current_position = -1;
2531 if (klass->query_position)
2532 klass->query_position (media, ¤t_position);
2533 GST_INFO ("current media position %" GST_TIME_FORMAT,
2534 GST_TIME_ARGS (current_position));
2536 if (start != GST_CLOCK_TIME_NONE)
2537 start_type = GST_SEEK_TYPE_SET;
2539 if (priv->range_stop == stop)
2540 stop = GST_CLOCK_TIME_NONE;
2541 else if (stop != GST_CLOCK_TIME_NONE)
2542 stop_type = GST_SEEK_TYPE_SET;
2544 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2545 gboolean had_flags = flags != 0;
2547 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2548 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2550 /* depends on the current playing state of the pipeline. We might need to
2551 * queue this until we get EOS. */
2553 flags |= GST_SEEK_FLAG_FLUSH;
2555 flags = GST_SEEK_FLAG_FLUSH;
2558 /* if range start was not supplied we must continue from current position.
2559 * but since we're doing a flushing seek, let us query the current position
2560 * so we end up at exactly the same position after the seek. */
2561 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2562 if (current_position == -1) {
2563 GST_WARNING ("current position unknown");
2565 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2566 GST_TIME_ARGS (current_position));
2567 start = current_position;
2568 start_type = GST_SEEK_TYPE_SET;
2570 flags |= GST_SEEK_FLAG_ACCURATE;
2573 /* only set keyframe flag when modifying start */
2574 if (start_type != GST_SEEK_TYPE_NONE)
2576 flags |= GST_SEEK_FLAG_KEY_UNIT;
2579 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2580 GST_DEBUG ("not seeking because no position change");
2583 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2585 media_streams_set_blocked (media, TRUE);
2587 /* FIXME, we only do forwards playback, no trick modes yet */
2588 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2589 flags, start_type, start, stop_type, stop);
2591 /* and block for the seek to complete */
2592 GST_INFO ("done seeking %d", res);
2596 g_rec_mutex_unlock (&priv->state_lock);
2598 /* wait until pipeline is prerolled again, this will also collect stats */
2599 if (!wait_preroll (media))
2600 goto preroll_failed;
2602 g_rec_mutex_lock (&priv->state_lock);
2603 GST_INFO ("prerolled again");
2606 GST_INFO ("no seek needed");
2609 g_rec_mutex_unlock (&priv->state_lock);
2616 g_rec_mutex_unlock (&priv->state_lock);
2617 GST_INFO ("media %p is not prepared", media);
2622 g_rec_mutex_unlock (&priv->state_lock);
2623 GST_INFO ("pipeline is not complete");
2628 g_rec_mutex_unlock (&priv->state_lock);
2629 GST_INFO ("pipeline is not seekable");
2634 g_rec_mutex_unlock (&priv->state_lock);
2635 GST_WARNING ("conversion to npt not supported");
2640 g_rec_mutex_unlock (&priv->state_lock);
2641 GST_INFO ("seeking failed");
2642 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2647 GST_WARNING ("failed to preroll after seek");
2654 * gst_rtsp_media_seek:
2655 * @media: a #GstRTSPMedia
2656 * @range: (transfer none): a #GstRTSPTimeRange
2658 * Seek the pipeline of @media to @range. @media must be prepared with
2659 * gst_rtsp_media_prepare().
2661 * Returns: %TRUE on success.
2664 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2666 return gst_rtsp_media_seek_full (media, range, 0);
2671 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2673 *blocked &= gst_rtsp_stream_is_blocking (stream);
2677 media_streams_blocking (GstRTSPMedia * media)
2679 gboolean blocking = TRUE;
2681 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2687 static GstStateChangeReturn
2688 set_state (GstRTSPMedia * media, GstState state)
2690 GstRTSPMediaPrivate *priv = media->priv;
2691 GstStateChangeReturn ret;
2693 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2695 ret = gst_element_set_state (priv->pipeline, state);
2700 static GstStateChangeReturn
2701 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2703 GstRTSPMediaPrivate *priv = media->priv;
2704 GstStateChangeReturn ret;
2706 GST_INFO ("set target state to %s for media %p",
2707 gst_element_state_get_name (state), media);
2708 priv->target_state = state;
2710 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2711 priv->target_state, NULL);
2714 ret = set_state (media, state);
2716 ret = GST_STATE_CHANGE_SUCCESS;
2721 /* called with state-lock */
2723 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2725 GstRTSPMediaPrivate *priv = media->priv;
2726 GstMessageType type;
2728 type = GST_MESSAGE_TYPE (message);
2731 case GST_MESSAGE_STATE_CHANGED:
2733 GstState old, new, pending;
2735 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2738 gst_message_parse_state_changed (message, &old, &new, &pending);
2740 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2741 gst_element_state_get_name (old), gst_element_state_get_name (new),
2742 gst_element_state_get_name (pending));
2743 if (priv->no_more_pads_pending == 0 && is_receive_only (media) &&
2744 old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2745 GST_INFO ("%p: went to PAUSED, prepared now", media);
2746 collect_media_stats (media);
2748 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2749 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2754 case GST_MESSAGE_BUFFERING:
2758 gst_message_parse_buffering (message, &percent);
2760 /* no state management needed for live pipelines */
2764 if (percent == 100) {
2765 /* a 100% message means buffering is done */
2766 priv->buffering = FALSE;
2767 /* if the desired state is playing, go back */
2768 if (priv->target_state == GST_STATE_PLAYING) {
2769 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2770 set_state (media, GST_STATE_PLAYING);
2772 GST_INFO ("Buffering done");
2775 /* buffering busy */
2776 if (priv->buffering == FALSE) {
2777 if (priv->target_state == GST_STATE_PLAYING) {
2778 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2779 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2780 set_state (media, GST_STATE_PAUSED);
2782 GST_INFO ("Buffering ...");
2785 priv->buffering = TRUE;
2789 case GST_MESSAGE_LATENCY:
2791 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2794 case GST_MESSAGE_ERROR:
2799 gst_message_parse_error (message, &gerror, &debug);
2800 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2801 g_error_free (gerror);
2804 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2807 case GST_MESSAGE_WARNING:
2812 gst_message_parse_warning (message, &gerror, &debug);
2813 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2814 g_error_free (gerror);
2818 case GST_MESSAGE_ELEMENT:
2820 const GstStructure *s;
2822 s = gst_message_get_structure (message);
2823 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2824 GST_DEBUG ("media received blocking message");
2825 if (priv->blocked && media_streams_blocking (media) &&
2826 priv->no_more_pads_pending == 0) {
2827 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
2828 collect_media_stats (media);
2830 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2831 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2836 case GST_MESSAGE_STREAM_STATUS:
2838 case GST_MESSAGE_ASYNC_DONE:
2839 if (priv->complete) {
2840 /* receive the final ASYNC_DONE, that is posted by the media pipeline
2841 * after all the transport parts have been successfully added to
2842 * the media streams. */
2843 GST_DEBUG_OBJECT (media, "got async-done");
2844 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2845 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2848 case GST_MESSAGE_EOS:
2849 GST_INFO ("%p: got EOS", media);
2851 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2852 GST_DEBUG ("shutting down after EOS");
2853 finish_unprepare (media);
2857 GST_INFO ("%p: got message type %d (%s)", media, type,
2858 gst_message_type_get_name (type));
2865 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2867 GstRTSPMediaPrivate *priv = media->priv;
2868 GstRTSPMediaClass *klass;
2871 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2873 g_rec_mutex_lock (&priv->state_lock);
2874 if (klass->handle_message)
2875 ret = klass->handle_message (media, message);
2878 g_rec_mutex_unlock (&priv->state_lock);
2884 watch_destroyed (GstRTSPMedia * media)
2886 GST_DEBUG_OBJECT (media, "source destroyed");
2887 g_object_unref (media);
2891 find_payload_element (GstElement * payloader)
2893 GstElement *pay = NULL;
2895 if (GST_IS_BIN (payloader)) {
2897 GValue item = { 0 };
2899 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2900 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2901 GstElement *element = (GstElement *) g_value_get_object (&item);
2902 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2906 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2910 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2911 pay = gst_object_ref (element);
2912 g_value_unset (&item);
2915 g_value_unset (&item);
2917 gst_iterator_free (iter);
2919 pay = g_object_ref (payloader);
2925 /* called from streaming threads */
2927 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2929 GstRTSPMediaPrivate *priv = media->priv;
2930 GstRTSPStream *stream;
2933 /* find the real payload element */
2934 pay = find_payload_element (element);
2935 stream = gst_rtsp_media_create_stream (media, pay, pad);
2936 gst_object_unref (pay);
2938 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2940 g_rec_mutex_lock (&priv->state_lock);
2941 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2944 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2946 /* join the element in the PAUSED state because this callback is
2947 * called from the streaming thread and it is PAUSED */
2948 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2949 priv->rtpbin, GST_STATE_PAUSED)) {
2950 GST_WARNING ("failed to join bin element");
2954 gst_rtsp_stream_set_blocked (stream, TRUE);
2956 g_rec_mutex_unlock (&priv->state_lock);
2963 gst_rtsp_media_remove_stream (media, stream);
2964 g_rec_mutex_unlock (&priv->state_lock);
2965 GST_INFO ("ignore pad because we are not preparing");
2971 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2973 GstRTSPMediaPrivate *priv = media->priv;
2974 GstRTSPStream *stream;
2976 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2980 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2982 g_rec_mutex_lock (&priv->state_lock);
2983 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2984 g_rec_mutex_unlock (&priv->state_lock);
2986 gst_rtsp_media_remove_stream (media, stream);
2990 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2992 GstRTSPMediaPrivate *priv = media->priv;
2994 GST_INFO_OBJECT (element, "no more pads");
2995 g_mutex_lock (&priv->lock);
2996 priv->no_more_pads_pending--;
2997 g_mutex_unlock (&priv->lock);
3000 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
3002 struct _DynPaySignalHandlers
3004 gulong pad_added_handler;
3005 gulong pad_removed_handler;
3006 gulong no_more_pads_handler;
3010 start_preroll (GstRTSPMedia * media)
3012 GstRTSPMediaPrivate *priv = media->priv;
3013 GstStateChangeReturn ret;
3015 GST_INFO ("setting pipeline to PAUSED for media %p", media);
3017 /* start blocked since it is possible that there are no sink elements yet */
3018 media_streams_set_blocked (media, TRUE);
3019 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3022 case GST_STATE_CHANGE_SUCCESS:
3023 GST_INFO ("SUCCESS state change for media %p", media);
3025 case GST_STATE_CHANGE_ASYNC:
3026 GST_INFO ("ASYNC state change for media %p", media);
3028 case GST_STATE_CHANGE_NO_PREROLL:
3029 /* we need to go to PLAYING */
3030 GST_INFO ("NO_PREROLL state change: live media %p", media);
3031 /* FIXME we disable seeking for live streams for now. We should perform a
3032 * seeking query in preroll instead */
3033 priv->seekable = -1;
3034 priv->is_live = TRUE;
3036 ret = set_state (media, GST_STATE_PLAYING);
3037 if (ret == GST_STATE_CHANGE_FAILURE)
3040 case GST_STATE_CHANGE_FAILURE:
3048 GST_WARNING ("failed to preroll pipeline");
3054 wait_preroll (GstRTSPMedia * media)
3056 GstRTSPMediaStatus status;
3058 GST_DEBUG ("wait to preroll pipeline");
3060 /* wait until pipeline is prerolled */
3061 status = gst_rtsp_media_get_status (media);
3062 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
3063 goto preroll_failed;
3069 GST_WARNING ("failed to preroll pipeline");
3075 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3077 GstRTSPMediaPrivate *priv = media->priv;
3078 GstRTSPStream *stream = NULL;
3080 GstElement *res = NULL;
3082 g_mutex_lock (&priv->lock);
3083 for (i = 0; i < priv->streams->len; i++) {
3084 stream = g_ptr_array_index (priv->streams, i);
3086 if (sessid == gst_rtsp_stream_get_index (stream))
3091 g_mutex_unlock (&priv->lock);
3094 res = gst_rtsp_stream_request_aux_sender (stream, sessid);
3100 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3102 GstRTSPMediaPrivate *priv = media->priv;
3103 GstRTSPStream *stream = NULL;
3105 GstElement *res = NULL;
3107 g_mutex_lock (&priv->lock);
3108 for (i = 0; i < priv->streams->len; i++) {
3109 stream = g_ptr_array_index (priv->streams, i);
3111 if (sessid == gst_rtsp_stream_get_index (stream))
3116 g_mutex_unlock (&priv->lock);
3119 res = gst_rtsp_stream_request_aux_receiver (stream, sessid);
3125 request_fec_decoder (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
3127 GstRTSPMediaPrivate *priv = media->priv;
3128 GstRTSPStream *stream = NULL;
3130 GstElement *res = NULL;
3132 g_mutex_lock (&priv->lock);
3133 for (i = 0; i < priv->streams->len; i++) {
3134 stream = g_ptr_array_index (priv->streams, i);
3136 if (sessid == gst_rtsp_stream_get_index (stream))
3141 g_mutex_unlock (&priv->lock);
3144 res = gst_rtsp_stream_request_ulpfec_decoder (stream, rtpbin, sessid);
3151 new_storage_cb (GstElement * rtpbin, GObject * storage, guint sessid,
3152 GstRTSPMedia * media)
3154 g_object_set (storage, "size-time", (media->priv->latency + 50) * GST_MSECOND,
3159 start_prepare (GstRTSPMedia * media)
3161 GstRTSPMediaPrivate *priv = media->priv;
3165 g_rec_mutex_lock (&priv->state_lock);
3166 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
3167 goto no_longer_preparing;
3169 g_signal_connect (priv->rtpbin, "new-storage", G_CALLBACK (new_storage_cb),
3171 g_signal_connect (priv->rtpbin, "request-fec-decoder",
3172 G_CALLBACK (request_fec_decoder), media);
3174 /* link streams we already have, other streams might appear when we have
3175 * dynamic elements */
3176 for (i = 0; i < priv->streams->len; i++) {
3177 GstRTSPStream *stream;
3179 stream = g_ptr_array_index (priv->streams, i);
3181 if (priv->rtx_time > 0) {
3182 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
3183 g_signal_connect (priv->rtpbin, "request-aux-sender",
3184 (GCallback) request_aux_sender, media);
3187 if (priv->do_retransmission) {
3188 g_signal_connect (priv->rtpbin, "request-aux-receiver",
3189 (GCallback) request_aux_receiver, media);
3192 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
3193 priv->rtpbin, GST_STATE_NULL)) {
3194 goto join_bin_failed;
3199 g_object_set (priv->rtpbin, "do-retransmission", priv->do_retransmission,
3200 "do-lost", TRUE, NULL);
3202 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3203 GstElement *elem = walk->data;
3204 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
3206 GST_INFO ("adding callbacks for dynamic element %p", elem);
3208 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
3209 (GCallback) pad_added_cb, media);
3210 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
3211 (GCallback) pad_removed_cb, media);
3212 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
3213 (GCallback) no_more_pads_cb, media);
3215 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
3218 if (priv->nb_dynamic_elements == 0 && is_receive_only (media)) {
3219 /* If we are receive_only (RECORD), do not try to preroll, to avoid
3220 * a second ASYNC state change failing */
3221 priv->is_live = TRUE;
3222 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3223 } else if (!start_preroll (media)) {
3224 goto preroll_failed;
3227 g_rec_mutex_unlock (&priv->state_lock);
3231 no_longer_preparing:
3233 GST_INFO ("media is no longer preparing");
3234 g_rec_mutex_unlock (&priv->state_lock);
3239 GST_WARNING ("failed to join bin element");
3240 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3241 g_rec_mutex_unlock (&priv->state_lock);
3246 GST_WARNING ("failed to preroll pipeline");
3247 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3248 g_rec_mutex_unlock (&priv->state_lock);
3254 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3256 GstRTSPMediaPrivate *priv;
3257 GstRTSPMediaClass *klass;
3259 GMainContext *context;
3264 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3266 if (!klass->create_rtpbin)
3267 goto no_create_rtpbin;
3269 priv->rtpbin = klass->create_rtpbin (media);
3270 if (priv->rtpbin != NULL) {
3271 gboolean success = TRUE;
3273 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
3275 if (klass->setup_rtpbin)
3276 success = klass->setup_rtpbin (media, priv->rtpbin);
3278 if (success == FALSE) {
3279 gst_object_unref (priv->rtpbin);
3280 priv->rtpbin = NULL;
3283 if (priv->rtpbin == NULL)
3286 priv->thread = thread;
3287 context = (thread != NULL) ? (thread->context) : NULL;
3289 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
3291 /* add the pipeline bus to our custom mainloop */
3292 priv->source = gst_bus_create_watch (bus);
3293 gst_object_unref (bus);
3295 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
3296 g_object_ref (media), (GDestroyNotify) watch_destroyed);
3298 priv->id = g_source_attach (priv->source, context);
3300 /* add stuff to the bin */
3301 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
3303 /* do remainder in context */
3304 source = g_idle_source_new ();
3305 g_source_set_callback (source, (GSourceFunc) start_prepare,
3306 g_object_ref (media), (GDestroyNotify) g_object_unref);
3307 g_source_attach (source, context);
3308 g_source_unref (source);
3315 GST_ERROR ("no create_rtpbin function");
3316 g_critical ("no create_rtpbin vmethod function set");
3321 GST_WARNING ("no rtpbin element");
3322 g_warning ("failed to create element 'rtpbin', check your installation");
3328 * gst_rtsp_media_prepare:
3329 * @media: a #GstRTSPMedia
3330 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
3331 * bus handler or %NULL
3333 * Prepare @media for streaming. This function will create the objects
3334 * to manage the streaming. A pipeline must have been set on @media with
3335 * gst_rtsp_media_take_pipeline().
3337 * It will preroll the pipeline and collect vital information about the streams
3338 * such as the duration.
3340 * Returns: %TRUE on success.
3343 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3345 GstRTSPMediaPrivate *priv;
3346 GstRTSPMediaClass *klass;
3348 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3352 g_rec_mutex_lock (&priv->state_lock);
3353 priv->prepare_count++;
3355 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3356 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3359 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3362 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3363 goto not_unprepared;
3365 if (!priv->reusable && priv->reused)
3368 GST_INFO ("preparing media %p", media);
3370 /* reset some variables */
3371 priv->is_live = FALSE;
3372 priv->seekable = -1;
3373 priv->buffering = FALSE;
3374 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3376 /* we're preparing now */
3377 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3379 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3380 if (klass->prepare) {
3381 if (!klass->prepare (media, thread))
3382 goto prepare_failed;
3386 g_rec_mutex_unlock (&priv->state_lock);
3388 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3389 * able to do this async so that we don't block the server thread. */
3390 if (!wait_preroll (media))
3391 goto preroll_failed;
3393 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3395 GST_INFO ("object %p is prerolled", media);
3402 /* we are not going to use the giving thread, so stop it. */
3404 gst_rtsp_thread_stop (thread);
3409 GST_LOG ("media %p was prepared", media);
3410 /* we are not going to use the giving thread, so stop it. */
3412 gst_rtsp_thread_stop (thread);
3413 g_rec_mutex_unlock (&priv->state_lock);
3419 /* we are not going to use the giving thread, so stop it. */
3421 gst_rtsp_thread_stop (thread);
3422 GST_WARNING ("media %p was not unprepared", media);
3423 priv->prepare_count--;
3424 g_rec_mutex_unlock (&priv->state_lock);
3429 /* we are not going to use the giving thread, so stop it. */
3431 gst_rtsp_thread_stop (thread);
3432 priv->prepare_count--;
3433 g_rec_mutex_unlock (&priv->state_lock);
3434 GST_WARNING ("can not reuse media %p", media);
3439 /* we are not going to use the giving thread, so stop it. */
3441 gst_rtsp_thread_stop (thread);
3442 priv->prepare_count--;
3443 g_rec_mutex_unlock (&priv->state_lock);
3444 GST_ERROR ("failed to prepare media");
3449 GST_WARNING ("failed to preroll pipeline");
3450 gst_rtsp_media_unprepare (media);
3455 /* must be called with state-lock */
3457 finish_unprepare (GstRTSPMedia * media)
3459 GstRTSPMediaPrivate *priv = media->priv;
3463 GST_DEBUG ("shutting down");
3465 /* release the lock on shutdown, otherwise pad_added_cb might try to
3466 * acquire the lock and then we deadlock */
3467 g_rec_mutex_unlock (&priv->state_lock);
3468 set_state (media, GST_STATE_NULL);
3469 g_rec_mutex_lock (&priv->state_lock);
3471 media_streams_set_blocked (media, FALSE);
3473 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
3476 for (i = 0; i < priv->streams->len; i++) {
3477 GstRTSPStream *stream;
3479 GST_INFO ("Removing elements of stream %d from pipeline", i);
3481 stream = g_ptr_array_index (priv->streams, i);
3483 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3486 /* remove the pad signal handlers */
3487 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3488 GstElement *elem = walk->data;
3489 DynPaySignalHandlers *handlers;
3492 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3493 g_assert (handlers != NULL);
3495 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3496 g_signal_handler_disconnect (G_OBJECT (elem),
3497 handlers->pad_removed_handler);
3498 g_signal_handler_disconnect (G_OBJECT (elem),
3499 handlers->no_more_pads_handler);
3501 g_slice_free (DynPaySignalHandlers, handlers);
3504 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3505 priv->rtpbin = NULL;
3508 gst_object_unref (priv->nettime);
3509 priv->nettime = NULL;
3511 priv->reused = TRUE;
3512 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3514 /* when the media is not reusable, this will effectively unref the media and
3516 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3518 /* the source has the last ref to the media */
3520 GST_DEBUG ("destroy source");
3521 g_source_destroy (priv->source);
3522 g_source_unref (priv->source);
3525 GST_DEBUG ("stop thread");
3526 gst_rtsp_thread_stop (priv->thread);
3530 /* called with state-lock */
3532 default_unprepare (GstRTSPMedia * media)
3534 GstRTSPMediaPrivate *priv = media->priv;
3536 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3538 if (priv->eos_shutdown) {
3539 GST_DEBUG ("sending EOS for shutdown");
3540 /* ref so that we don't disappear */
3541 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3542 /* we need to go to playing again for the EOS to propagate, normally in this
3543 * state, nothing is receiving data from us anymore so this is ok. */
3544 set_state (media, GST_STATE_PLAYING);
3546 finish_unprepare (media);
3552 * gst_rtsp_media_unprepare:
3553 * @media: a #GstRTSPMedia
3555 * Unprepare @media. After this call, the media should be prepared again before
3556 * it can be used again. If the media is set to be non-reusable, a new instance
3559 * Returns: %TRUE on success.
3562 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3564 GstRTSPMediaPrivate *priv;
3567 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3571 g_rec_mutex_lock (&priv->state_lock);
3572 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3573 goto was_unprepared;
3575 priv->prepare_count--;
3576 if (priv->prepare_count > 0)
3579 GST_INFO ("unprepare media %p", media);
3580 set_target_state (media, GST_STATE_NULL, FALSE);
3583 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3584 GstRTSPMediaClass *klass;
3586 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3587 if (klass->unprepare)
3588 success = klass->unprepare (media);
3590 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3591 finish_unprepare (media);
3593 g_rec_mutex_unlock (&priv->state_lock);
3599 g_rec_mutex_unlock (&priv->state_lock);
3600 GST_INFO ("media %p was already unprepared", media);
3605 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3606 g_rec_mutex_unlock (&priv->state_lock);
3611 /* should be called with state-lock */
3613 get_clock_unlocked (GstRTSPMedia * media)
3615 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3616 GST_DEBUG_OBJECT (media, "media was not prepared");
3619 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3623 * gst_rtsp_media_get_clock:
3624 * @media: a #GstRTSPMedia
3626 * Get the clock that is used by the pipeline in @media.
3628 * @media must be prepared before this method returns a valid clock object.
3630 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3633 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3636 GstRTSPMediaPrivate *priv;
3638 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3642 g_rec_mutex_lock (&priv->state_lock);
3643 clock = get_clock_unlocked (media);
3644 g_rec_mutex_unlock (&priv->state_lock);
3650 * gst_rtsp_media_get_base_time:
3651 * @media: a #GstRTSPMedia
3653 * Get the base_time that is used by the pipeline in @media.
3655 * @media must be prepared before this method returns a valid base_time.
3657 * Returns: the base_time used by @media.
3660 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3662 GstClockTime result;
3663 GstRTSPMediaPrivate *priv;
3665 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3669 g_rec_mutex_lock (&priv->state_lock);
3670 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3673 result = gst_element_get_base_time (media->priv->pipeline);
3674 g_rec_mutex_unlock (&priv->state_lock);
3681 g_rec_mutex_unlock (&priv->state_lock);
3682 GST_DEBUG_OBJECT (media, "media was not prepared");
3683 return GST_CLOCK_TIME_NONE;
3688 * gst_rtsp_media_get_time_provider:
3689 * @media: a #GstRTSPMedia
3690 * @address: (allow-none): an address or %NULL
3691 * @port: a port or 0
3693 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3694 * will listen on @address and @port for client time requests.
3696 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3698 GstNetTimeProvider *
3699 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3702 GstRTSPMediaPrivate *priv;
3703 GstNetTimeProvider *provider = NULL;
3705 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3709 g_rec_mutex_lock (&priv->state_lock);
3710 if (priv->time_provider) {
3711 if ((provider = priv->nettime) == NULL) {
3714 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3715 provider = gst_net_time_provider_new (clock, address, port);
3716 gst_object_unref (clock);
3718 priv->nettime = provider;
3722 g_rec_mutex_unlock (&priv->state_lock);
3725 gst_object_ref (provider);
3731 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3733 return gst_rtsp_sdp_from_media (sdp, info, media);
3737 * gst_rtsp_media_setup_sdp:
3738 * @media: a #GstRTSPMedia
3739 * @sdp: (transfer none): a #GstSDPMessage
3740 * @info: (transfer none): a #GstSDPInfo
3742 * Add @media specific info to @sdp. @info is used to configure the connection
3743 * information in the SDP.
3745 * Returns: TRUE on success.
3748 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3751 GstRTSPMediaPrivate *priv;
3752 GstRTSPMediaClass *klass;
3755 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3756 g_return_val_if_fail (sdp != NULL, FALSE);
3757 g_return_val_if_fail (info != NULL, FALSE);
3761 g_rec_mutex_lock (&priv->state_lock);
3763 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3765 if (!klass->setup_sdp)
3768 res = klass->setup_sdp (media, sdp, info);
3770 g_rec_mutex_unlock (&priv->state_lock);
3777 g_rec_mutex_unlock (&priv->state_lock);
3778 GST_ERROR ("no setup_sdp function");
3779 g_critical ("no setup_sdp vmethod function set");
3785 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3787 GstRTSPMediaPrivate *priv = media->priv;
3790 medias_len = gst_sdp_message_medias_len (sdp);
3791 if (medias_len != priv->streams->len) {
3792 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3793 priv->streams->len, medias_len);
3797 for (i = 0; i < medias_len; i++) {
3799 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3800 GstRTSPStream *stream;
3801 gint j, formats_len;
3802 const gchar *control;
3803 GstRTSPProfile profile, profiles;
3805 stream = g_ptr_array_index (priv->streams, i);
3807 /* TODO: Should we do something with the other SDP information? */
3810 proto = gst_sdp_media_get_proto (sdp_media);
3811 if (proto == NULL) {
3812 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3816 if (g_str_equal (proto, "RTP/AVP")) {
3817 profile = GST_RTSP_PROFILE_AVP;
3818 } else if (g_str_equal (proto, "RTP/SAVP")) {
3819 profile = GST_RTSP_PROFILE_SAVP;
3820 } else if (g_str_equal (proto, "RTP/AVPF")) {
3821 profile = GST_RTSP_PROFILE_AVPF;
3822 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3823 profile = GST_RTSP_PROFILE_SAVPF;
3825 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3829 profiles = gst_rtsp_stream_get_profiles (stream);
3830 if ((profiles & profile) == 0) {
3831 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3835 formats_len = gst_sdp_media_formats_len (sdp_media);
3836 for (j = 0; j < formats_len; j++) {
3841 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3843 GST_DEBUG (" looking at %d pt: %d", j, pt);
3846 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
3848 GST_WARNING (" skipping pt %d without caps", pt);
3852 /* do some tweaks */
3853 GST_DEBUG ("mapping sdp session level attributes to caps");
3854 gst_sdp_message_attributes_to_caps (sdp, caps);
3855 GST_DEBUG ("mapping sdp media level attributes to caps");
3856 gst_sdp_media_attributes_to_caps (sdp_media, caps);
3858 s = gst_caps_get_structure (caps, 0);
3859 gst_structure_set_name (s, "application/x-rtp");
3861 if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
3862 gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
3864 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3865 gst_caps_unref (caps);
3868 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3870 gst_rtsp_stream_set_control (stream, control);
3878 * gst_rtsp_media_handle_sdp:
3879 * @media: a #GstRTSPMedia
3880 * @sdp: (transfer none): a #GstSDPMessage
3882 * Configure an SDP on @media for receiving streams
3884 * Returns: TRUE on success.
3887 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3889 GstRTSPMediaPrivate *priv;
3890 GstRTSPMediaClass *klass;
3893 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3894 g_return_val_if_fail (sdp != NULL, FALSE);
3898 g_rec_mutex_lock (&priv->state_lock);
3900 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3902 if (!klass->handle_sdp)
3905 res = klass->handle_sdp (media, sdp);
3907 g_rec_mutex_unlock (&priv->state_lock);
3914 g_rec_mutex_unlock (&priv->state_lock);
3915 GST_ERROR ("no handle_sdp function");
3916 g_critical ("no handle_sdp vmethod function set");
3922 do_set_seqnum (GstRTSPStream * stream)
3926 if (gst_rtsp_stream_is_sender (stream)) {
3927 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3928 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3932 /* call with state_lock */
3934 default_suspend (GstRTSPMedia * media)
3936 GstRTSPMediaPrivate *priv = media->priv;
3937 GstStateChangeReturn ret;
3939 switch (priv->suspend_mode) {
3940 case GST_RTSP_SUSPEND_MODE_NONE:
3941 GST_DEBUG ("media %p no suspend", media);
3943 case GST_RTSP_SUSPEND_MODE_PAUSE:
3944 GST_DEBUG ("media %p suspend to PAUSED", media);
3945 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3946 if (ret == GST_STATE_CHANGE_FAILURE)
3949 case GST_RTSP_SUSPEND_MODE_RESET:
3950 GST_DEBUG ("media %p suspend to NULL", media);
3951 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3952 if (ret == GST_STATE_CHANGE_FAILURE)
3954 /* Because payloader needs to set the sequence number as
3955 * monotonic, we need to preserve the sequence number
3956 * after pause. (otherwise going from pause to play, which
3957 * is actually from NULL to PLAY will create a new sequence
3959 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3970 GST_WARNING ("failed changing pipeline's state for media %p", media);
3976 * gst_rtsp_media_suspend:
3977 * @media: a #GstRTSPMedia
3979 * Suspend @media. The state of the pipeline managed by @media is set to
3980 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3981 * with gst_rtsp_media_unsuspend()
3983 * @media must be prepared with gst_rtsp_media_prepare();
3985 * Returns: %TRUE on success.
3988 gst_rtsp_media_suspend (GstRTSPMedia * media)
3990 GstRTSPMediaPrivate *priv = media->priv;
3991 GstRTSPMediaClass *klass;
3993 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3995 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3997 g_rec_mutex_lock (&priv->state_lock);
3998 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
4001 /* don't attempt to suspend when something is busy */
4002 if (priv->n_active > 0)
4005 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4006 if (klass->suspend) {
4007 if (!klass->suspend (media))
4008 goto suspend_failed;
4011 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
4013 g_rec_mutex_unlock (&priv->state_lock);
4020 g_rec_mutex_unlock (&priv->state_lock);
4021 GST_WARNING ("media %p was not prepared", media);
4026 g_rec_mutex_unlock (&priv->state_lock);
4027 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4028 GST_WARNING ("failed to suspend media %p", media);
4033 /* call with state_lock */
4035 default_unsuspend (GstRTSPMedia * media)
4037 GstRTSPMediaPrivate *priv = media->priv;
4038 gboolean preroll_ok;
4040 switch (priv->suspend_mode) {
4041 case GST_RTSP_SUSPEND_MODE_NONE:
4042 if (is_receive_only (media))
4044 if (media_streams_blocking (media)) {
4045 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4046 /* at this point the media pipeline has been updated and contain all
4047 * specific transport parts: all active streams contain at least one sink
4048 * element and it's safe to unblock any blocked streams that are active */
4049 media_unblock_linked (media);
4051 /* streams are not blocked and media is suspended from PAUSED */
4052 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4054 g_rec_mutex_unlock (&priv->state_lock);
4055 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
4056 g_rec_mutex_lock (&priv->state_lock);
4057 goto preroll_failed;
4059 g_rec_mutex_lock (&priv->state_lock);
4061 case GST_RTSP_SUSPEND_MODE_PAUSE:
4062 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
4064 case GST_RTSP_SUSPEND_MODE_RESET:
4066 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
4067 /* at this point the media pipeline has been updated and contain all
4068 * specific transport parts: all active streams contain at least one sink
4069 * element and it's safe to unblock any blocked streams that are active */
4070 media_unblock_linked (media);
4071 if (!start_preroll (media))
4074 g_rec_mutex_unlock (&priv->state_lock);
4075 preroll_ok = wait_preroll (media);
4076 g_rec_mutex_lock (&priv->state_lock);
4079 goto preroll_failed;
4090 GST_WARNING ("failed to preroll pipeline");
4095 GST_WARNING ("failed to preroll pipeline");
4101 * gst_rtsp_media_unsuspend:
4102 * @media: a #GstRTSPMedia
4104 * Unsuspend @media if it was in a suspended state. This method does nothing
4105 * when the media was not in the suspended state.
4107 * Returns: %TRUE on success.
4110 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
4112 GstRTSPMediaPrivate *priv = media->priv;
4113 GstRTSPMediaClass *klass;
4115 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4117 g_rec_mutex_lock (&priv->state_lock);
4118 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4121 klass = GST_RTSP_MEDIA_GET_CLASS (media);
4122 if (klass->unsuspend) {
4123 if (!klass->unsuspend (media))
4124 goto unsuspend_failed;
4128 g_rec_mutex_unlock (&priv->state_lock);
4135 g_rec_mutex_unlock (&priv->state_lock);
4136 GST_WARNING ("failed to unsuspend media %p", media);
4137 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
4142 /* must be called with state-lock */
4144 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
4146 GstRTSPMediaPrivate *priv = media->priv;
4148 if (state == GST_STATE_NULL) {
4149 gst_rtsp_media_unprepare (media);
4151 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
4152 set_target_state (media, state, FALSE);
4153 /* when we are buffering, don't update the state yet, this will be done
4154 * when buffering finishes */
4155 if (priv->buffering) {
4156 GST_INFO ("Buffering busy, delay state change");
4158 if (state == GST_STATE_PLAYING)
4159 /* make sure pads are not blocking anymore when going to PLAYING */
4160 media_unblock_linked (media);
4162 set_state (media, state);
4164 /* and suspend after pause */
4165 if (state == GST_STATE_PAUSED)
4166 gst_rtsp_media_suspend (media);
4172 * gst_rtsp_media_set_pipeline_state:
4173 * @media: a #GstRTSPMedia
4174 * @state: the target state of the pipeline
4176 * Set the state of the pipeline managed by @media to @state
4179 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
4181 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4183 g_rec_mutex_lock (&media->priv->state_lock);
4184 media_set_pipeline_state_locked (media, state);
4185 g_rec_mutex_unlock (&media->priv->state_lock);
4189 * gst_rtsp_media_set_state:
4190 * @media: a #GstRTSPMedia
4191 * @state: the target state of the media
4192 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
4193 * a #GPtrArray of #GstRTSPStreamTransport pointers
4195 * Set the state of @media to @state and for the transports in @transports.
4197 * @media must be prepared with gst_rtsp_media_prepare();
4199 * Returns: %TRUE on success.
4202 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
4203 GPtrArray * transports)
4205 GstRTSPMediaPrivate *priv;
4207 gboolean activate, deactivate, do_state;
4210 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4211 g_return_val_if_fail (transports != NULL, FALSE);
4215 g_rec_mutex_lock (&priv->state_lock);
4216 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
4218 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
4219 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
4222 /* NULL and READY are the same */
4223 if (state == GST_STATE_READY)
4224 state = GST_STATE_NULL;
4226 activate = deactivate = FALSE;
4228 GST_INFO ("going to state %s media %p, target state %s",
4229 gst_element_state_get_name (state), media,
4230 gst_element_state_get_name (priv->target_state));
4233 case GST_STATE_NULL:
4234 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
4235 if (priv->target_state >= GST_STATE_PAUSED)
4238 case GST_STATE_PAUSED:
4239 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
4240 if (priv->target_state == GST_STATE_PLAYING)
4243 case GST_STATE_PLAYING:
4244 /* we're going to PLAYING, activate */
4250 old_active = priv->n_active;
4252 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
4253 activate, deactivate);
4254 for (i = 0; i < transports->len; i++) {
4255 GstRTSPStreamTransport *trans;
4257 /* we need a non-NULL entry in the array */
4258 trans = g_ptr_array_index (transports, i);
4263 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
4265 } else if (deactivate) {
4266 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
4271 /* we just activated the first media, do the playing state change */
4272 if (old_active == 0 && activate)
4274 /* if we have no more active media, do the downward state changes */
4275 else if (priv->n_active == 0)
4280 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
4283 if (priv->target_state != state) {
4285 media_set_pipeline_state_locked (media, state);
4286 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
4291 /* remember where we are */
4292 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
4293 old_active != priv->n_active))
4294 collect_media_stats (media);
4296 g_rec_mutex_unlock (&priv->state_lock);
4303 GST_WARNING ("media %p was not prepared", media);
4304 g_rec_mutex_unlock (&priv->state_lock);
4309 GST_WARNING ("media %p in error status while changing to state %d",
4311 if (state == GST_STATE_NULL) {
4312 for (i = 0; i < transports->len; i++) {
4313 GstRTSPStreamTransport *trans;
4315 /* we need a non-NULL entry in the array */
4316 trans = g_ptr_array_index (transports, i);
4320 gst_rtsp_stream_transport_set_active (trans, FALSE);
4324 g_rec_mutex_unlock (&priv->state_lock);
4330 * gst_rtsp_media_set_transport_mode:
4331 * @media: a #GstRTSPMedia
4332 * @mode: the new value
4334 * Sets if the media pipeline can work in PLAY or RECORD mode
4337 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
4338 GstRTSPTransportMode mode)
4340 GstRTSPMediaPrivate *priv;
4342 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4346 g_mutex_lock (&priv->lock);
4347 priv->transport_mode = mode;
4348 g_mutex_unlock (&priv->lock);
4352 * gst_rtsp_media_get_transport_mode:
4353 * @media: a #GstRTSPMedia
4355 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4357 * Returns: The transport mode.
4359 GstRTSPTransportMode
4360 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4362 GstRTSPMediaPrivate *priv;
4363 GstRTSPTransportMode res;
4365 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4369 g_mutex_lock (&priv->lock);
4370 res = priv->transport_mode;
4371 g_mutex_unlock (&priv->lock);
4377 * gst_rtsp_media_get_seekable:
4378 * @media: a #GstRTSPMedia
4380 * Check if the pipeline for @media seek and up to what point in time,
4383 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4384 * and > 0 to indicate the longest duration between any two random access points.
4385 * %G_MAXINT64 means any value is possible.
4388 gst_rtsp_media_seekable (GstRTSPMedia * media)
4390 GstRTSPMediaPrivate *priv;
4391 GstClockTimeDiff res;
4393 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4397 /* Currently we are not able to seek on live streams,
4398 * and no stream is seekable only to the beginning */
4399 g_mutex_lock (&priv->lock);
4400 res = priv->seekable;
4401 g_mutex_unlock (&priv->lock);
4407 * gst_rtsp_media_complete_pipeline:
4408 * @media: a #GstRTSPMedia
4409 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4411 * Add a receiver and sender parts to the pipeline based on the transport from
4414 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4417 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4419 GstRTSPMediaPrivate *priv;
4422 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4423 g_return_val_if_fail (transports, FALSE);
4425 GST_DEBUG_OBJECT (media, "complete pipeline");
4429 g_mutex_lock (&priv->lock);
4430 for (i = 0; i < priv->streams->len; i++) {
4431 GstRTSPStreamTransport *transport;
4432 GstRTSPStream *stream;
4433 const GstRTSPTransport *rtsp_transport;
4435 transport = g_ptr_array_index (transports, i);
4439 stream = gst_rtsp_stream_transport_get_stream (transport);
4443 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4445 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4446 g_mutex_unlock (&priv->lock);
4451 priv->complete = TRUE;
4452 g_mutex_unlock (&priv->lock);