2 * Copyright (C) 2008 Wim Taymans <wim.taymans at gmail.com>
3 * Copyright (C) 2015 Centricular Ltd
4 * Author: Sebastian Dröge <sebastian@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * @short_description: The media pipeline
24 * @see_also: #GstRTSPMediaFactory, #GstRTSPStream, #GstRTSPSession,
25 * #GstRTSPSessionMedia
27 * a #GstRTSPMedia contains the complete GStreamer pipeline to manage the
28 * streaming to the clients. The actual data transfer is done by the
29 * #GstRTSPStream objects that are created and exposed by the #GstRTSPMedia.
31 * The #GstRTSPMedia is usually created from a #GstRTSPMediaFactory when the
32 * client does a DESCRIBE or SETUP of a resource.
34 * A media is created with gst_rtsp_media_new() that takes the element that will
35 * provide the streaming elements. For each of the streams, a new #GstRTSPStream
36 * object needs to be made with the gst_rtsp_media_create_stream() which takes
37 * the payloader element and the source pad that produces the RTP stream.
39 * The pipeline of the media is set to PAUSED with gst_rtsp_media_prepare(). The
40 * prepare method will add rtpbin and sinks and sources to send and receive RTP
41 * and RTCP packets from the clients. Each stream srcpad is connected to an
42 * input into the internal rtpbin.
44 * It is also possible to dynamically create #GstRTSPStream objects during the
45 * prepare phase. With gst_rtsp_media_get_status() you can check the status of
48 * After the media is prepared, it is ready for streaming. It will usually be
49 * managed in a session with gst_rtsp_session_manage_media(). See
50 * #GstRTSPSession and #GstRTSPSessionMedia.
52 * The state of the media can be controlled with gst_rtsp_media_set_state ().
53 * Seeking can be done with gst_rtsp_media_seek().
55 * With gst_rtsp_media_unprepare() the pipeline is stopped and shut down. When
56 * gst_rtsp_media_set_eos_shutdown() an EOS will be sent to the pipeline to
59 * With gst_rtsp_media_set_shared(), the media can be shared between multiple
60 * clients. With gst_rtsp_media_set_reusable() you can control if the pipeline
61 * can be prepared again after an unprepare.
63 * Last reviewed on 2013-07-11 (1.0.0)
70 #include <gst/app/gstappsrc.h>
71 #include <gst/app/gstappsink.h>
73 #include <gst/sdp/gstmikey.h>
74 #include <gst/rtp/gstrtppayloads.h>
76 #define AES_128_KEY_LEN 16
77 #define AES_256_KEY_LEN 32
79 #define HMAC_32_KEY_LEN 4
80 #define HMAC_80_KEY_LEN 10
82 #include "rtsp-media.h"
84 #define GST_RTSP_MEDIA_GET_PRIVATE(obj) \
85 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTSP_MEDIA, GstRTSPMediaPrivate))
87 struct _GstRTSPMediaPrivate
92 /* protected by lock */
93 GstRTSPPermissions *permissions;
95 gboolean suspend_mode;
97 GstRTSPProfile profiles;
98 GstRTSPLowerTrans protocols;
100 gboolean eos_shutdown;
102 GstRTSPAddressPool *pool;
103 gchar *multicast_iface;
105 GstRTSPTransportMode transport_mode;
106 gboolean stop_on_disconnect;
109 GRecMutex state_lock; /* locking order: state lock, lock */
110 GPtrArray *streams; /* protected by lock */
111 GList *dynamic; /* protected by lock */
112 GstRTSPMediaStatus status; /* protected by lock */
117 /* the pipeline for the media */
118 GstElement *pipeline;
121 GstRTSPThread *thread;
123 gboolean time_provider;
124 GstNetTimeProvider *nettime;
127 GstClockTimeDiff seekable;
129 GstState target_state;
131 /* RTP session manager */
134 /* the range of media */
135 GstRTSPTimeRange range; /* protected by lock */
136 GstClockTime range_start;
137 GstClockTime range_stop;
139 GList *payloads; /* protected by lock */
140 GstClockTime rtx_time; /* protected by lock */
141 guint latency; /* protected by lock */
142 GstClock *clock; /* protected by lock */
143 GstRTSPPublishClockMode publish_clock_mode;
145 /* Dynamic element handling */
146 guint nb_dynamic_elements;
147 guint no_more_pads_pending;
150 #define DEFAULT_SHARED FALSE
151 #define DEFAULT_SUSPEND_MODE GST_RTSP_SUSPEND_MODE_NONE
152 #define DEFAULT_REUSABLE FALSE
153 #define DEFAULT_PROFILES GST_RTSP_PROFILE_AVP
154 #define DEFAULT_PROTOCOLS GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | \
155 GST_RTSP_LOWER_TRANS_TCP
156 #define DEFAULT_EOS_SHUTDOWN FALSE
157 #define DEFAULT_BUFFER_SIZE 0x80000
158 #define DEFAULT_TIME_PROVIDER FALSE
159 #define DEFAULT_LATENCY 200
160 #define DEFAULT_TRANSPORT_MODE GST_RTSP_TRANSPORT_MODE_PLAY
161 #define DEFAULT_STOP_ON_DISCONNECT TRUE
163 /* define to dump received RTCP packets */
180 PROP_STOP_ON_DISCONNECT,
188 SIGNAL_REMOVED_STREAM,
196 GST_DEBUG_CATEGORY_STATIC (rtsp_media_debug);
197 #define GST_CAT_DEFAULT rtsp_media_debug
199 static void gst_rtsp_media_get_property (GObject * object, guint propid,
200 GValue * value, GParamSpec * pspec);
201 static void gst_rtsp_media_set_property (GObject * object, guint propid,
202 const GValue * value, GParamSpec * pspec);
203 static void gst_rtsp_media_finalize (GObject * obj);
205 static gboolean default_handle_message (GstRTSPMedia * media,
206 GstMessage * message);
207 static void finish_unprepare (GstRTSPMedia * media);
208 static gboolean default_prepare (GstRTSPMedia * media, GstRTSPThread * thread);
209 static gboolean default_unprepare (GstRTSPMedia * media);
210 static gboolean default_suspend (GstRTSPMedia * media);
211 static gboolean default_unsuspend (GstRTSPMedia * media);
212 static gboolean default_convert_range (GstRTSPMedia * media,
213 GstRTSPTimeRange * range, GstRTSPRangeUnit unit);
214 static gboolean default_query_position (GstRTSPMedia * media,
216 static gboolean default_query_stop (GstRTSPMedia * media, gint64 * stop);
217 static GstElement *default_create_rtpbin (GstRTSPMedia * media);
218 static gboolean default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
220 static gboolean default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp);
222 static gboolean wait_preroll (GstRTSPMedia * media);
224 static GstElement *find_payload_element (GstElement * payloader);
226 static guint gst_rtsp_media_signals[SIGNAL_LAST] = { 0 };
228 #define C_ENUM(v) ((gint) v)
231 gst_rtsp_suspend_mode_get_type (void)
234 static const GEnumValue values[] = {
235 {C_ENUM (GST_RTSP_SUSPEND_MODE_NONE), "GST_RTSP_SUSPEND_MODE_NONE", "none"},
236 {C_ENUM (GST_RTSP_SUSPEND_MODE_PAUSE), "GST_RTSP_SUSPEND_MODE_PAUSE",
238 {C_ENUM (GST_RTSP_SUSPEND_MODE_RESET), "GST_RTSP_SUSPEND_MODE_RESET",
243 if (g_once_init_enter (&id)) {
244 GType tmp = g_enum_register_static ("GstRTSPSuspendMode", values);
245 g_once_init_leave (&id, tmp);
250 #define C_FLAGS(v) ((guint) v)
253 gst_rtsp_transport_mode_get_type (void)
256 static const GFlagsValue values[] = {
257 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_PLAY), "GST_RTSP_TRANSPORT_MODE_PLAY",
259 {C_FLAGS (GST_RTSP_TRANSPORT_MODE_RECORD), "GST_RTSP_TRANSPORT_MODE_RECORD",
264 if (g_once_init_enter (&id)) {
265 GType tmp = g_flags_register_static ("GstRTSPTransportMode", values);
266 g_once_init_leave (&id, tmp);
272 gst_rtsp_publish_clock_mode_get_type (void)
275 static const GEnumValue values[] = {
276 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_NONE),
277 "GST_RTSP_PUBLISH_CLOCK_MODE_NONE", "none"},
278 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK),
279 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK",
281 {C_ENUM (GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET),
282 "GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET",
287 if (g_once_init_enter (&id)) {
288 GType tmp = g_enum_register_static ("GstRTSPPublishClockMode", values);
289 g_once_init_leave (&id, tmp);
294 G_DEFINE_TYPE (GstRTSPMedia, gst_rtsp_media, G_TYPE_OBJECT);
297 gst_rtsp_media_class_init (GstRTSPMediaClass * klass)
299 GObjectClass *gobject_class;
301 g_type_class_add_private (klass, sizeof (GstRTSPMediaPrivate));
303 gobject_class = G_OBJECT_CLASS (klass);
305 gobject_class->get_property = gst_rtsp_media_get_property;
306 gobject_class->set_property = gst_rtsp_media_set_property;
307 gobject_class->finalize = gst_rtsp_media_finalize;
309 g_object_class_install_property (gobject_class, PROP_SHARED,
310 g_param_spec_boolean ("shared", "Shared",
311 "If this media pipeline can be shared", DEFAULT_SHARED,
312 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
314 g_object_class_install_property (gobject_class, PROP_SUSPEND_MODE,
315 g_param_spec_enum ("suspend-mode", "Suspend Mode",
316 "How to suspend the media in PAUSED", GST_TYPE_RTSP_SUSPEND_MODE,
317 DEFAULT_SUSPEND_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
319 g_object_class_install_property (gobject_class, PROP_REUSABLE,
320 g_param_spec_boolean ("reusable", "Reusable",
321 "If this media pipeline can be reused after an unprepare",
322 DEFAULT_REUSABLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
324 g_object_class_install_property (gobject_class, PROP_PROFILES,
325 g_param_spec_flags ("profiles", "Profiles",
326 "Allowed transfer profiles", GST_TYPE_RTSP_PROFILE,
327 DEFAULT_PROFILES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
329 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
330 g_param_spec_flags ("protocols", "Protocols",
331 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
332 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
334 g_object_class_install_property (gobject_class, PROP_EOS_SHUTDOWN,
335 g_param_spec_boolean ("eos-shutdown", "EOS Shutdown",
336 "Send an EOS event to the pipeline before unpreparing",
337 DEFAULT_EOS_SHUTDOWN, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
339 g_object_class_install_property (gobject_class, PROP_BUFFER_SIZE,
340 g_param_spec_uint ("buffer-size", "Buffer Size",
341 "The kernel UDP buffer size to use", 0, G_MAXUINT,
342 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
344 g_object_class_install_property (gobject_class, PROP_ELEMENT,
345 g_param_spec_object ("element", "The Element",
346 "The GstBin to use for streaming the media", GST_TYPE_ELEMENT,
347 G_PARAM_CONSTRUCT_ONLY | G_PARAM_READWRITE));
349 g_object_class_install_property (gobject_class, PROP_TIME_PROVIDER,
350 g_param_spec_boolean ("time-provider", "Time Provider",
351 "Use a NetTimeProvider for clients",
352 DEFAULT_TIME_PROVIDER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
354 g_object_class_install_property (gobject_class, PROP_LATENCY,
355 g_param_spec_uint ("latency", "Latency",
356 "Latency used for receiving media in milliseconds", 0, G_MAXUINT,
357 DEFAULT_BUFFER_SIZE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
359 g_object_class_install_property (gobject_class, PROP_TRANSPORT_MODE,
360 g_param_spec_flags ("transport-mode", "Transport Mode",
361 "If this media pipeline can be used for PLAY or RECORD",
362 GST_TYPE_RTSP_TRANSPORT_MODE, DEFAULT_TRANSPORT_MODE,
363 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
365 g_object_class_install_property (gobject_class, PROP_STOP_ON_DISCONNECT,
366 g_param_spec_boolean ("stop-on-disconnect", "Stop On Disconnect",
367 "If this media pipeline should be stopped "
368 "when a client disconnects without TEARDOWN",
369 DEFAULT_STOP_ON_DISCONNECT,
370 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
372 g_object_class_install_property (gobject_class, PROP_CLOCK,
373 g_param_spec_object ("clock", "Clock",
374 "Clock to be used by the media pipeline",
375 GST_TYPE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
377 gst_rtsp_media_signals[SIGNAL_NEW_STREAM] =
378 g_signal_new ("new-stream", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
379 G_STRUCT_OFFSET (GstRTSPMediaClass, new_stream), NULL, NULL,
380 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_RTSP_STREAM);
382 gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM] =
383 g_signal_new ("removed-stream", G_TYPE_FROM_CLASS (klass),
384 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, removed_stream),
385 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
386 GST_TYPE_RTSP_STREAM);
388 gst_rtsp_media_signals[SIGNAL_PREPARED] =
389 g_signal_new ("prepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
390 G_STRUCT_OFFSET (GstRTSPMediaClass, prepared), NULL, NULL,
391 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
393 gst_rtsp_media_signals[SIGNAL_UNPREPARED] =
394 g_signal_new ("unprepared", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
395 G_STRUCT_OFFSET (GstRTSPMediaClass, unprepared), NULL, NULL,
396 g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
398 gst_rtsp_media_signals[SIGNAL_TARGET_STATE] =
399 g_signal_new ("target-state", G_TYPE_FROM_CLASS (klass),
400 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTSPMediaClass, target_state),
401 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
403 gst_rtsp_media_signals[SIGNAL_NEW_STATE] =
404 g_signal_new ("new-state", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
405 G_STRUCT_OFFSET (GstRTSPMediaClass, new_state), NULL, NULL,
406 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_INT);
408 GST_DEBUG_CATEGORY_INIT (rtsp_media_debug, "rtspmedia", 0, "GstRTSPMedia");
410 klass->handle_message = default_handle_message;
411 klass->prepare = default_prepare;
412 klass->unprepare = default_unprepare;
413 klass->suspend = default_suspend;
414 klass->unsuspend = default_unsuspend;
415 klass->convert_range = default_convert_range;
416 klass->query_position = default_query_position;
417 klass->query_stop = default_query_stop;
418 klass->create_rtpbin = default_create_rtpbin;
419 klass->setup_sdp = default_setup_sdp;
420 klass->handle_sdp = default_handle_sdp;
424 gst_rtsp_media_init (GstRTSPMedia * media)
426 GstRTSPMediaPrivate *priv = GST_RTSP_MEDIA_GET_PRIVATE (media);
430 priv->streams = g_ptr_array_new_with_free_func (g_object_unref);
431 g_mutex_init (&priv->lock);
432 g_cond_init (&priv->cond);
433 g_rec_mutex_init (&priv->state_lock);
435 priv->shared = DEFAULT_SHARED;
436 priv->suspend_mode = DEFAULT_SUSPEND_MODE;
437 priv->reusable = DEFAULT_REUSABLE;
438 priv->profiles = DEFAULT_PROFILES;
439 priv->protocols = DEFAULT_PROTOCOLS;
440 priv->eos_shutdown = DEFAULT_EOS_SHUTDOWN;
441 priv->buffer_size = DEFAULT_BUFFER_SIZE;
442 priv->time_provider = DEFAULT_TIME_PROVIDER;
443 priv->transport_mode = DEFAULT_TRANSPORT_MODE;
444 priv->stop_on_disconnect = DEFAULT_STOP_ON_DISCONNECT;
445 priv->publish_clock_mode = GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK;
449 gst_rtsp_media_finalize (GObject * obj)
451 GstRTSPMediaPrivate *priv;
454 media = GST_RTSP_MEDIA (obj);
457 GST_INFO ("finalize media %p", media);
459 if (priv->permissions)
460 gst_rtsp_permissions_unref (priv->permissions);
462 g_ptr_array_unref (priv->streams);
464 g_list_free_full (priv->dynamic, gst_object_unref);
467 gst_object_unref (priv->pipeline);
469 gst_object_unref (priv->nettime);
470 gst_object_unref (priv->element);
472 g_object_unref (priv->pool);
474 g_list_free (priv->payloads);
475 g_free (priv->multicast_iface);
476 g_mutex_clear (&priv->lock);
477 g_cond_clear (&priv->cond);
478 g_rec_mutex_clear (&priv->state_lock);
480 G_OBJECT_CLASS (gst_rtsp_media_parent_class)->finalize (obj);
484 gst_rtsp_media_get_property (GObject * object, guint propid,
485 GValue * value, GParamSpec * pspec)
487 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
491 g_value_set_object (value, media->priv->element);
494 g_value_set_boolean (value, gst_rtsp_media_is_shared (media));
496 case PROP_SUSPEND_MODE:
497 g_value_set_enum (value, gst_rtsp_media_get_suspend_mode (media));
500 g_value_set_boolean (value, gst_rtsp_media_is_reusable (media));
503 g_value_set_flags (value, gst_rtsp_media_get_profiles (media));
506 g_value_set_flags (value, gst_rtsp_media_get_protocols (media));
508 case PROP_EOS_SHUTDOWN:
509 g_value_set_boolean (value, gst_rtsp_media_is_eos_shutdown (media));
511 case PROP_BUFFER_SIZE:
512 g_value_set_uint (value, gst_rtsp_media_get_buffer_size (media));
514 case PROP_TIME_PROVIDER:
515 g_value_set_boolean (value, gst_rtsp_media_is_time_provider (media));
518 g_value_set_uint (value, gst_rtsp_media_get_latency (media));
520 case PROP_TRANSPORT_MODE:
521 g_value_set_flags (value, gst_rtsp_media_get_transport_mode (media));
523 case PROP_STOP_ON_DISCONNECT:
524 g_value_set_boolean (value, gst_rtsp_media_is_stop_on_disconnect (media));
527 g_value_take_object (value, gst_rtsp_media_get_clock (media));
530 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
535 gst_rtsp_media_set_property (GObject * object, guint propid,
536 const GValue * value, GParamSpec * pspec)
538 GstRTSPMedia *media = GST_RTSP_MEDIA (object);
542 media->priv->element = g_value_get_object (value);
543 gst_object_ref_sink (media->priv->element);
546 gst_rtsp_media_set_shared (media, g_value_get_boolean (value));
548 case PROP_SUSPEND_MODE:
549 gst_rtsp_media_set_suspend_mode (media, g_value_get_enum (value));
552 gst_rtsp_media_set_reusable (media, g_value_get_boolean (value));
555 gst_rtsp_media_set_profiles (media, g_value_get_flags (value));
558 gst_rtsp_media_set_protocols (media, g_value_get_flags (value));
560 case PROP_EOS_SHUTDOWN:
561 gst_rtsp_media_set_eos_shutdown (media, g_value_get_boolean (value));
563 case PROP_BUFFER_SIZE:
564 gst_rtsp_media_set_buffer_size (media, g_value_get_uint (value));
566 case PROP_TIME_PROVIDER:
567 gst_rtsp_media_use_time_provider (media, g_value_get_boolean (value));
570 gst_rtsp_media_set_latency (media, g_value_get_uint (value));
572 case PROP_TRANSPORT_MODE:
573 gst_rtsp_media_set_transport_mode (media, g_value_get_flags (value));
575 case PROP_STOP_ON_DISCONNECT:
576 gst_rtsp_media_set_stop_on_disconnect (media,
577 g_value_get_boolean (value));
580 gst_rtsp_media_set_clock (media, g_value_get_object (value));
583 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, propid, pspec);
591 } DoQueryPositionData;
594 do_query_position (GstRTSPStream * stream, DoQueryPositionData * data)
598 if (gst_rtsp_stream_query_position (stream, &tmp)) {
599 data->position = MIN (data->position, tmp);
603 GST_INFO_OBJECT (stream, "media position: %" GST_TIME_FORMAT,
604 GST_TIME_ARGS (data->position));
608 default_query_position (GstRTSPMedia * media, gint64 * position)
610 GstRTSPMediaPrivate *priv;
611 DoQueryPositionData data;
615 data.position = G_MAXINT64;
618 g_ptr_array_foreach (priv->streams, (GFunc) do_query_position, &data);
621 *position = GST_CLOCK_TIME_NONE;
623 *position = data.position;
635 do_query_stop (GstRTSPStream * stream, DoQueryStopData * data)
639 if (gst_rtsp_stream_query_stop (stream, &tmp)) {
640 data->stop = MAX (data->stop, tmp);
646 default_query_stop (GstRTSPMedia * media, gint64 * stop)
648 GstRTSPMediaPrivate *priv;
649 DoQueryStopData data;
656 g_ptr_array_foreach (priv->streams, (GFunc) do_query_stop, &data);
664 default_create_rtpbin (GstRTSPMedia * media)
668 rtpbin = gst_element_factory_make ("rtpbin", NULL);
673 /* must be called with state lock */
675 check_seekable (GstRTSPMedia * media)
678 GstRTSPMediaPrivate *priv = media->priv;
680 /* Update the seekable state of the pipeline in case it changed */
681 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)) {
682 /* TODO: Seeking for RECORD? */
685 guint i, n = priv->streams->len;
687 for (i = 0; i < n; i++) {
688 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
690 if (gst_rtsp_stream_get_publish_clock_mode (stream) ==
691 GST_RTSP_PUBLISH_CLOCK_MODE_CLOCK_AND_OFFSET) {
698 query = gst_query_new_seeking (GST_FORMAT_TIME);
699 if (gst_element_query (priv->pipeline, query)) {
704 gst_query_parse_seeking (query, &format, &seekable, &start, &end);
705 priv->seekable = seekable ? G_MAXINT64 : 0;
706 } else if (priv->streams->len) {
707 gboolean seekable = TRUE;
708 guint i, n = priv->streams->len;
710 GST_DEBUG_OBJECT (media, "Checking %d streams", n);
711 for (i = 0; i < n; i++) {
712 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
713 seekable &= gst_rtsp_stream_seekable (stream);
715 priv->seekable = seekable ? G_MAXINT64 : -1;
718 GST_DEBUG_OBJECT (media, "seekable:%" G_GINT64_FORMAT, priv->seekable);
720 gst_query_unref (query);
723 /* must be called with state lock */
725 check_complete (GstRTSPMedia * media)
727 GstRTSPMediaPrivate *priv = media->priv;
729 guint i, n = priv->streams->len;
731 for (i = 0; i < n; i++) {
732 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
734 if (gst_rtsp_stream_is_complete (stream))
741 /* must be called with state lock */
743 collect_media_stats (GstRTSPMedia * media)
745 GstRTSPMediaPrivate *priv = media->priv;
746 gint64 position = 0, stop = -1;
748 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
749 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
752 priv->range.unit = GST_RTSP_RANGE_NPT;
754 GST_INFO ("collect media stats");
757 priv->range.min.type = GST_RTSP_TIME_NOW;
758 priv->range.min.seconds = -1;
759 priv->range_start = -1;
760 priv->range.max.type = GST_RTSP_TIME_END;
761 priv->range.max.seconds = -1;
762 priv->range_stop = -1;
764 GstRTSPMediaClass *klass;
767 klass = GST_RTSP_MEDIA_GET_CLASS (media);
769 /* get the position */
771 if (klass->query_position)
772 ret = klass->query_position (media, &position);
775 GST_INFO ("position query failed");
779 /* get the current segment stop */
781 if (klass->query_stop)
782 ret = klass->query_stop (media, &stop);
785 GST_INFO ("stop query failed");
789 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
790 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
792 if (position == -1) {
793 priv->range.min.type = GST_RTSP_TIME_NOW;
794 priv->range.min.seconds = -1;
795 priv->range_start = -1;
797 priv->range.min.type = GST_RTSP_TIME_SECONDS;
798 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
799 priv->range_start = position;
802 priv->range.max.type = GST_RTSP_TIME_END;
803 priv->range.max.seconds = -1;
804 priv->range_stop = -1;
806 priv->range.max.type = GST_RTSP_TIME_SECONDS;
807 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
808 priv->range_stop = stop;
811 check_seekable (media);
816 * gst_rtsp_media_new:
817 * @element: (transfer full): a #GstElement
819 * Create a new #GstRTSPMedia instance. @element is the bin element that
820 * provides the different streams. The #GstRTSPMedia object contains the
821 * element to produce RTP data for one or more related (audio/video/..)
824 * Ownership is taken of @element.
826 * Returns: (transfer full): a new #GstRTSPMedia object.
829 gst_rtsp_media_new (GstElement * element)
831 GstRTSPMedia *result;
833 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
835 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
841 * gst_rtsp_media_get_element:
842 * @media: a #GstRTSPMedia
844 * Get the element that was used when constructing @media.
846 * Returns: (transfer full): a #GstElement. Unref after usage.
849 gst_rtsp_media_get_element (GstRTSPMedia * media)
851 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
853 return gst_object_ref (media->priv->element);
857 * gst_rtsp_media_take_pipeline:
858 * @media: a #GstRTSPMedia
859 * @pipeline: (transfer full): a #GstPipeline
861 * Set @pipeline as the #GstPipeline for @media. Ownership is
862 * taken of @pipeline.
865 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
867 GstRTSPMediaPrivate *priv;
869 GstNetTimeProvider *nettime;
871 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
872 g_return_if_fail (GST_IS_PIPELINE (pipeline));
876 g_mutex_lock (&priv->lock);
877 old = priv->pipeline;
878 priv->pipeline = GST_ELEMENT_CAST (pipeline);
879 nettime = priv->nettime;
880 priv->nettime = NULL;
881 g_mutex_unlock (&priv->lock);
884 gst_object_unref (old);
887 gst_object_unref (nettime);
889 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
893 * gst_rtsp_media_set_permissions:
894 * @media: a #GstRTSPMedia
895 * @permissions: (transfer none) (nullable): a #GstRTSPPermissions
897 * Set @permissions on @media.
900 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
901 GstRTSPPermissions * permissions)
903 GstRTSPMediaPrivate *priv;
905 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
909 g_mutex_lock (&priv->lock);
910 if (priv->permissions)
911 gst_rtsp_permissions_unref (priv->permissions);
912 if ((priv->permissions = permissions))
913 gst_rtsp_permissions_ref (permissions);
914 g_mutex_unlock (&priv->lock);
918 * gst_rtsp_media_get_permissions:
919 * @media: a #GstRTSPMedia
921 * Get the permissions object from @media.
923 * Returns: (transfer full) (nullable): a #GstRTSPPermissions object, unref after usage.
926 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
928 GstRTSPMediaPrivate *priv;
929 GstRTSPPermissions *result;
931 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
935 g_mutex_lock (&priv->lock);
936 if ((result = priv->permissions))
937 gst_rtsp_permissions_ref (result);
938 g_mutex_unlock (&priv->lock);
944 * gst_rtsp_media_set_suspend_mode:
945 * @media: a #GstRTSPMedia
946 * @mode: the new #GstRTSPSuspendMode
948 * Control how @ media will be suspended after the SDP has been generated and
949 * after a PAUSE request has been performed.
951 * Media must be unprepared when setting the suspend mode.
954 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
956 GstRTSPMediaPrivate *priv;
958 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
962 g_rec_mutex_lock (&priv->state_lock);
963 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
965 priv->suspend_mode = mode;
966 g_rec_mutex_unlock (&priv->state_lock);
973 GST_WARNING ("media %p was prepared", media);
974 g_rec_mutex_unlock (&priv->state_lock);
979 * gst_rtsp_media_get_suspend_mode:
980 * @media: a #GstRTSPMedia
982 * Get how @media will be suspended.
984 * Returns: #GstRTSPSuspendMode.
987 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
989 GstRTSPMediaPrivate *priv;
990 GstRTSPSuspendMode res;
992 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
996 g_rec_mutex_lock (&priv->state_lock);
997 res = priv->suspend_mode;
998 g_rec_mutex_unlock (&priv->state_lock);
1004 * gst_rtsp_media_set_shared:
1005 * @media: a #GstRTSPMedia
1006 * @shared: the new value
1008 * Set or unset if the pipeline for @media can be shared will multiple clients.
1009 * When @shared is %TRUE, client requests for this media will share the media
1013 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1015 GstRTSPMediaPrivate *priv;
1017 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1021 g_mutex_lock (&priv->lock);
1022 priv->shared = shared;
1023 g_mutex_unlock (&priv->lock);
1027 * gst_rtsp_media_is_shared:
1028 * @media: a #GstRTSPMedia
1030 * Check if the pipeline for @media can be shared between multiple clients.
1032 * Returns: %TRUE if the media can be shared between clients.
1035 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1037 GstRTSPMediaPrivate *priv;
1040 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1044 g_mutex_lock (&priv->lock);
1046 g_mutex_unlock (&priv->lock);
1052 * gst_rtsp_media_set_reusable:
1053 * @media: a #GstRTSPMedia
1054 * @reusable: the new value
1056 * Set or unset if the pipeline for @media can be reused after the pipeline has
1060 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1062 GstRTSPMediaPrivate *priv;
1064 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1068 g_mutex_lock (&priv->lock);
1069 priv->reusable = reusable;
1070 g_mutex_unlock (&priv->lock);
1074 * gst_rtsp_media_is_reusable:
1075 * @media: a #GstRTSPMedia
1077 * Check if the pipeline for @media can be reused after an unprepare.
1079 * Returns: %TRUE if the media can be reused
1082 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1084 GstRTSPMediaPrivate *priv;
1087 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1091 g_mutex_lock (&priv->lock);
1092 res = priv->reusable;
1093 g_mutex_unlock (&priv->lock);
1099 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1101 gst_rtsp_stream_set_profiles (stream, *profiles);
1105 * gst_rtsp_media_set_profiles:
1106 * @media: a #GstRTSPMedia
1107 * @profiles: the new flags
1109 * Configure the allowed lower transport for @media.
1112 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1114 GstRTSPMediaPrivate *priv;
1116 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1120 g_mutex_lock (&priv->lock);
1121 priv->profiles = profiles;
1122 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1123 g_mutex_unlock (&priv->lock);
1127 * gst_rtsp_media_get_profiles:
1128 * @media: a #GstRTSPMedia
1130 * Get the allowed profiles of @media.
1132 * Returns: a #GstRTSPProfile
1135 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1137 GstRTSPMediaPrivate *priv;
1140 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1144 g_mutex_lock (&priv->lock);
1145 res = priv->profiles;
1146 g_mutex_unlock (&priv->lock);
1152 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1154 gst_rtsp_stream_set_protocols (stream, *protocols);
1158 * gst_rtsp_media_set_protocols:
1159 * @media: a #GstRTSPMedia
1160 * @protocols: the new flags
1162 * Configure the allowed lower transport for @media.
1165 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1167 GstRTSPMediaPrivate *priv;
1169 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1173 g_mutex_lock (&priv->lock);
1174 priv->protocols = protocols;
1175 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1176 g_mutex_unlock (&priv->lock);
1180 * gst_rtsp_media_get_protocols:
1181 * @media: a #GstRTSPMedia
1183 * Get the allowed protocols of @media.
1185 * Returns: a #GstRTSPLowerTrans
1188 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1190 GstRTSPMediaPrivate *priv;
1191 GstRTSPLowerTrans res;
1193 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1194 GST_RTSP_LOWER_TRANS_UNKNOWN);
1198 g_mutex_lock (&priv->lock);
1199 res = priv->protocols;
1200 g_mutex_unlock (&priv->lock);
1206 * gst_rtsp_media_set_eos_shutdown:
1207 * @media: a #GstRTSPMedia
1208 * @eos_shutdown: the new value
1210 * Set or unset if an EOS event will be sent to the pipeline for @media before
1214 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1216 GstRTSPMediaPrivate *priv;
1218 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1222 g_mutex_lock (&priv->lock);
1223 priv->eos_shutdown = eos_shutdown;
1224 g_mutex_unlock (&priv->lock);
1228 * gst_rtsp_media_is_eos_shutdown:
1229 * @media: a #GstRTSPMedia
1231 * Check if the pipeline for @media will send an EOS down the pipeline before
1234 * Returns: %TRUE if the media will send EOS before unpreparing.
1237 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1239 GstRTSPMediaPrivate *priv;
1242 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1246 g_mutex_lock (&priv->lock);
1247 res = priv->eos_shutdown;
1248 g_mutex_unlock (&priv->lock);
1254 * gst_rtsp_media_set_buffer_size:
1255 * @media: a #GstRTSPMedia
1256 * @size: the new value
1258 * Set the kernel UDP buffer size.
1261 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1263 GstRTSPMediaPrivate *priv;
1266 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1268 GST_LOG_OBJECT (media, "set buffer size %u", size);
1272 g_mutex_lock (&priv->lock);
1273 priv->buffer_size = size;
1275 for (i = 0; i < priv->streams->len; i++) {
1276 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1277 gst_rtsp_stream_set_buffer_size (stream, size);
1279 g_mutex_unlock (&priv->lock);
1283 * gst_rtsp_media_get_buffer_size:
1284 * @media: a #GstRTSPMedia
1286 * Get the kernel UDP buffer size.
1288 * Returns: the kernel UDP buffer size.
1291 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1293 GstRTSPMediaPrivate *priv;
1296 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1300 g_mutex_lock (&priv->lock);
1301 res = priv->buffer_size;
1302 g_mutex_unlock (&priv->lock);
1308 * gst_rtsp_media_set_stop_on_disconnect:
1309 * @media: a #GstRTSPMedia
1310 * @stop_on_disconnect: the new value
1312 * Set or unset if the pipeline for @media should be stopped when a
1313 * client disconnects without sending TEARDOWN.
1316 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1317 gboolean stop_on_disconnect)
1319 GstRTSPMediaPrivate *priv;
1321 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1325 g_mutex_lock (&priv->lock);
1326 priv->stop_on_disconnect = stop_on_disconnect;
1327 g_mutex_unlock (&priv->lock);
1331 * gst_rtsp_media_is_stop_on_disconnect:
1332 * @media: a #GstRTSPMedia
1334 * Check if the pipeline for @media will be stopped when a client disconnects
1335 * without sending TEARDOWN.
1337 * Returns: %TRUE if the media will be stopped when a client disconnects
1338 * without sending TEARDOWN.
1341 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1343 GstRTSPMediaPrivate *priv;
1346 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1350 g_mutex_lock (&priv->lock);
1351 res = priv->stop_on_disconnect;
1352 g_mutex_unlock (&priv->lock);
1358 * gst_rtsp_media_set_retransmission_time:
1359 * @media: a #GstRTSPMedia
1360 * @time: the new value
1362 * Set the amount of time to store retransmission packets.
1365 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1367 GstRTSPMediaPrivate *priv;
1370 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1372 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1376 g_mutex_lock (&priv->lock);
1377 priv->rtx_time = time;
1378 for (i = 0; i < priv->streams->len; i++) {
1379 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1381 gst_rtsp_stream_set_retransmission_time (stream, time);
1385 g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
1386 g_mutex_unlock (&priv->lock);
1390 * gst_rtsp_media_get_retransmission_time:
1391 * @media: a #GstRTSPMedia
1393 * Get the amount of time to store retransmission data.
1395 * Returns: the amount of time to store retransmission data.
1398 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1400 GstRTSPMediaPrivate *priv;
1403 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1407 g_mutex_lock (&priv->lock);
1408 res = priv->rtx_time;
1409 g_mutex_unlock (&priv->lock);
1415 * gst_rtsp_media_set_latency:
1416 * @media: a #GstRTSPMedia
1417 * @latency: latency in milliseconds
1419 * Configure the latency used for receiving media.
1422 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1424 GstRTSPMediaPrivate *priv;
1426 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1428 GST_LOG_OBJECT (media, "set latency %ums", latency);
1432 g_mutex_lock (&priv->lock);
1433 priv->latency = latency;
1435 g_object_set (priv->rtpbin, "latency", latency, NULL);
1436 g_mutex_unlock (&priv->lock);
1440 * gst_rtsp_media_get_latency:
1441 * @media: a #GstRTSPMedia
1443 * Get the latency that is used for receiving media.
1445 * Returns: latency in milliseconds
1448 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1450 GstRTSPMediaPrivate *priv;
1453 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1457 g_mutex_lock (&priv->lock);
1458 res = priv->latency;
1459 g_mutex_unlock (&priv->lock);
1465 * gst_rtsp_media_use_time_provider:
1466 * @media: a #GstRTSPMedia
1467 * @time_provider: if a #GstNetTimeProvider should be used
1469 * Set @media to provide a #GstNetTimeProvider.
1472 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1474 GstRTSPMediaPrivate *priv;
1476 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1480 g_mutex_lock (&priv->lock);
1481 priv->time_provider = time_provider;
1482 g_mutex_unlock (&priv->lock);
1486 * gst_rtsp_media_is_time_provider:
1487 * @media: a #GstRTSPMedia
1489 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1491 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1493 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1496 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1498 GstRTSPMediaPrivate *priv;
1501 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1505 g_mutex_lock (&priv->lock);
1506 res = priv->time_provider;
1507 g_mutex_unlock (&priv->lock);
1513 * gst_rtsp_media_set_clock:
1514 * @media: a #GstRTSPMedia
1515 * @clock: (nullable): #GstClock to be used
1517 * Configure the clock used for the media.
1520 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1522 GstRTSPMediaPrivate *priv;
1524 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1525 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1527 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1531 g_mutex_lock (&priv->lock);
1533 gst_object_unref (priv->clock);
1534 priv->clock = clock ? gst_object_ref (clock) : NULL;
1535 if (priv->pipeline) {
1537 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1539 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1542 g_mutex_unlock (&priv->lock);
1546 * gst_rtsp_media_set_publish_clock_mode:
1547 * @media: a #GstRTSPMedia
1548 * @mode: the clock publish mode
1550 * Sets if and how the media clock should be published according to RFC7273.
1555 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1556 GstRTSPPublishClockMode mode)
1558 GstRTSPMediaPrivate *priv;
1562 g_mutex_lock (&priv->lock);
1563 priv->publish_clock_mode = mode;
1565 n = priv->streams->len;
1566 for (i = 0; i < n; i++) {
1567 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1569 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1571 g_mutex_unlock (&priv->lock);
1575 * gst_rtsp_media_get_publish_clock_mode:
1576 * @media: a #GstRTSPMedia
1578 * Gets if and how the media clock should be published according to RFC7273.
1580 * Returns: The GstRTSPPublishClockMode
1584 GstRTSPPublishClockMode
1585 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1587 GstRTSPMediaPrivate *priv;
1588 GstRTSPPublishClockMode ret;
1591 g_mutex_lock (&priv->lock);
1592 ret = priv->publish_clock_mode;
1593 g_mutex_unlock (&priv->lock);
1599 * gst_rtsp_media_set_address_pool:
1600 * @media: a #GstRTSPMedia
1601 * @pool: (transfer none) (nullable): a #GstRTSPAddressPool
1603 * configure @pool to be used as the address pool of @media.
1606 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1607 GstRTSPAddressPool * pool)
1609 GstRTSPMediaPrivate *priv;
1610 GstRTSPAddressPool *old;
1612 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1616 GST_LOG_OBJECT (media, "set address pool %p", pool);
1618 g_mutex_lock (&priv->lock);
1619 if ((old = priv->pool) != pool)
1620 priv->pool = pool ? g_object_ref (pool) : NULL;
1623 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1625 g_mutex_unlock (&priv->lock);
1628 g_object_unref (old);
1632 * gst_rtsp_media_get_address_pool:
1633 * @media: a #GstRTSPMedia
1635 * Get the #GstRTSPAddressPool used as the address pool of @media.
1637 * Returns: (transfer full) (nullable): the #GstRTSPAddressPool of @media.
1638 * g_object_unref() after usage.
1640 GstRTSPAddressPool *
1641 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1643 GstRTSPMediaPrivate *priv;
1644 GstRTSPAddressPool *result;
1646 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1650 g_mutex_lock (&priv->lock);
1651 if ((result = priv->pool))
1652 g_object_ref (result);
1653 g_mutex_unlock (&priv->lock);
1659 * gst_rtsp_media_set_multicast_iface:
1660 * @media: a #GstRTSPMedia
1661 * @multicast_iface: (transfer none) (nullable): a multicast interface name
1663 * configure @multicast_iface to be used for @media.
1666 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1667 const gchar * multicast_iface)
1669 GstRTSPMediaPrivate *priv;
1672 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1676 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1678 g_mutex_lock (&priv->lock);
1679 if ((old = priv->multicast_iface) != multicast_iface)
1680 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1683 g_ptr_array_foreach (priv->streams,
1684 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1685 g_mutex_unlock (&priv->lock);
1692 * gst_rtsp_media_get_multicast_iface:
1693 * @media: a #GstRTSPMedia
1695 * Get the multicast interface used for @media.
1697 * Returns: (transfer full) (nullable): the multicast interface for @media.
1698 * g_free() after usage.
1701 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1703 GstRTSPMediaPrivate *priv;
1706 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1710 g_mutex_lock (&priv->lock);
1711 if ((result = priv->multicast_iface))
1712 result = g_strdup (result);
1713 g_mutex_unlock (&priv->lock);
1719 _find_payload_types (GstRTSPMedia * media)
1722 GQueue queue = G_QUEUE_INIT;
1724 n = media->priv->streams->len;
1725 for (i = 0; i < n; i++) {
1726 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1727 guint pt = gst_rtsp_stream_get_pt (stream);
1729 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1736 _next_available_pt (GList * payloads)
1740 for (i = 96; i <= 127; i++) {
1741 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1743 return GPOINTER_TO_UINT (i);
1750 * gst_rtsp_media_collect_streams:
1751 * @media: a #GstRTSPMedia
1753 * Find all payloader elements, they should be named pay\%d in the
1754 * element of @media, and create #GstRTSPStreams for them.
1756 * Collect all dynamic elements, named dynpay\%d, and add them to
1757 * the list of dynamic elements.
1759 * Find all depayloader elements, they should be named depay\%d in the
1760 * element of @media, and create #GstRTSPStreams for them.
1763 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1765 GstRTSPMediaPrivate *priv;
1766 GstElement *element, *elem;
1770 gboolean more_elem_remaining = TRUE;
1771 GstRTSPTransportMode mode = 0;
1773 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1776 element = priv->element;
1779 for (i = 0; more_elem_remaining; i++) {
1782 more_elem_remaining = FALSE;
1784 name = g_strdup_printf ("pay%d", i);
1785 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1787 GST_INFO ("found stream %d with payloader %p", i, elem);
1789 /* take the pad of the payloader */
1790 pad = gst_element_get_static_pad (elem, "src");
1792 /* find the real payload element in case elem is a GstBin */
1793 pay = find_payload_element (elem);
1795 /* create the stream */
1797 GST_WARNING ("could not find real payloader, using bin");
1798 gst_rtsp_media_create_stream (media, elem, pad);
1800 gst_rtsp_media_create_stream (media, pay, pad);
1801 gst_object_unref (pay);
1804 gst_object_unref (pad);
1805 gst_object_unref (elem);
1808 more_elem_remaining = TRUE;
1809 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1813 name = g_strdup_printf ("dynpay%d", i);
1814 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1815 /* a stream that will dynamically create pads to provide RTP packets */
1816 GST_INFO ("found dynamic element %d, %p", i, elem);
1818 g_mutex_lock (&priv->lock);
1819 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1820 g_mutex_unlock (&priv->lock);
1822 priv->nb_dynamic_elements++;
1825 more_elem_remaining = TRUE;
1826 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1830 name = g_strdup_printf ("depay%d", i);
1831 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1832 GST_INFO ("found stream %d with depayloader %p", i, elem);
1834 /* take the pad of the payloader */
1835 pad = gst_element_get_static_pad (elem, "sink");
1836 /* create the stream */
1837 gst_rtsp_media_create_stream (media, elem, pad);
1838 gst_object_unref (pad);
1839 gst_object_unref (elem);
1842 more_elem_remaining = TRUE;
1843 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1849 if (priv->transport_mode != mode)
1850 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1851 priv->transport_mode, mode);
1856 * gst_rtsp_media_create_stream:
1857 * @media: a #GstRTSPMedia
1858 * @payloader: a #GstElement
1861 * Create a new stream in @media that provides RTP data on @pad.
1862 * @pad should be a pad of an element inside @media->element.
1864 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1868 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1871 GstRTSPMediaPrivate *priv;
1872 GstRTSPStream *stream;
1877 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1878 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1879 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1883 g_mutex_lock (&priv->lock);
1884 idx = priv->streams->len;
1886 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1888 if (GST_PAD_IS_SRC (pad))
1889 name = g_strdup_printf ("src_%u", idx);
1891 name = g_strdup_printf ("sink_%u", idx);
1893 ghostpad = gst_ghost_pad_new (name, pad);
1894 gst_pad_set_active (ghostpad, TRUE);
1895 gst_element_add_pad (priv->element, ghostpad);
1898 stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
1900 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1901 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
1902 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1903 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1904 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
1905 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
1906 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
1908 g_ptr_array_add (priv->streams, stream);
1910 if (GST_PAD_IS_SRC (pad)) {
1914 g_list_free (priv->payloads);
1915 priv->payloads = _find_payload_types (media);
1917 n = priv->streams->len;
1918 for (i = 0; i < n; i++) {
1919 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1920 guint rtx_pt = _next_available_pt (priv->payloads);
1923 GST_WARNING ("Ran out of space of dynamic payload types");
1927 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
1930 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
1933 g_mutex_unlock (&priv->lock);
1935 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1942 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1944 GstRTSPMediaPrivate *priv;
1949 g_mutex_lock (&priv->lock);
1950 /* remove the ghostpad */
1951 srcpad = gst_rtsp_stream_get_srcpad (stream);
1952 gst_element_remove_pad (priv->element, srcpad);
1953 gst_object_unref (srcpad);
1954 /* now remove the stream */
1955 g_object_ref (stream);
1956 g_ptr_array_remove (priv->streams, stream);
1957 g_mutex_unlock (&priv->lock);
1959 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1962 g_object_unref (stream);
1966 * gst_rtsp_media_n_streams:
1967 * @media: a #GstRTSPMedia
1969 * Get the number of streams in this media.
1971 * Returns: The number of streams.
1974 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1976 GstRTSPMediaPrivate *priv;
1979 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1983 g_mutex_lock (&priv->lock);
1984 res = priv->streams->len;
1985 g_mutex_unlock (&priv->lock);
1991 * gst_rtsp_media_get_stream:
1992 * @media: a #GstRTSPMedia
1993 * @idx: the stream index
1995 * Retrieve the stream with index @idx from @media.
1997 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1998 * @idx or %NULL when a stream with that index did not exist.
2001 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
2003 GstRTSPMediaPrivate *priv;
2006 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2010 g_mutex_lock (&priv->lock);
2011 if (idx < priv->streams->len)
2012 res = g_ptr_array_index (priv->streams, idx);
2015 g_mutex_unlock (&priv->lock);
2021 * gst_rtsp_media_find_stream:
2022 * @media: a #GstRTSPMedia
2023 * @control: the control of the stream
2025 * Find a stream in @media with @control as the control uri.
2027 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2028 * control uri @control or %NULL when a stream with that control did
2032 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2034 GstRTSPMediaPrivate *priv;
2038 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2039 g_return_val_if_fail (control != NULL, NULL);
2045 g_mutex_lock (&priv->lock);
2046 for (i = 0; i < priv->streams->len; i++) {
2047 GstRTSPStream *test;
2049 test = g_ptr_array_index (priv->streams, i);
2050 if (gst_rtsp_stream_has_control (test, control)) {
2055 g_mutex_unlock (&priv->lock);
2060 /* called with state-lock */
2062 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2063 GstRTSPRangeUnit unit)
2065 return gst_rtsp_range_convert_units (range, unit);
2069 * gst_rtsp_media_get_range_string:
2070 * @media: a #GstRTSPMedia
2071 * @play: for the PLAY request
2072 * @unit: the unit to use for the string
2074 * Get the current range as a string. @media must be prepared with
2075 * gst_rtsp_media_prepare ().
2077 * Returns: (transfer full) (nullable): The range as a string, g_free() after usage.
2080 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2081 GstRTSPRangeUnit unit)
2083 GstRTSPMediaClass *klass;
2084 GstRTSPMediaPrivate *priv;
2086 GstRTSPTimeRange range;
2088 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2089 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2090 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2094 g_rec_mutex_lock (&priv->state_lock);
2095 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2096 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2099 g_mutex_lock (&priv->lock);
2101 /* Update the range value with current position/duration */
2102 collect_media_stats (media);
2105 range = priv->range;
2107 if (!play && priv->n_active > 0) {
2108 range.min.type = GST_RTSP_TIME_NOW;
2109 range.min.seconds = -1;
2111 g_mutex_unlock (&priv->lock);
2112 g_rec_mutex_unlock (&priv->state_lock);
2114 if (!klass->convert_range (media, &range, unit))
2115 goto conversion_failed;
2117 result = gst_rtsp_range_to_string (&range);
2124 GST_WARNING ("media %p was not prepared", media);
2125 g_rec_mutex_unlock (&priv->state_lock);
2130 GST_WARNING ("range conversion to unit %d failed", unit);
2136 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2138 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2142 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2144 GstRTSPMediaPrivate *priv = media->priv;
2146 GST_DEBUG ("media %p set blocked %d", media, blocked);
2147 priv->blocked = blocked;
2148 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2152 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2154 gst_rtsp_stream_unblock_linked (stream);
2158 media_unblock_linked (GstRTSPMedia * media)
2160 GstRTSPMediaPrivate *priv = media->priv;
2162 GST_DEBUG ("media %p unblocking linked streams", media);
2163 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2167 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2169 GstRTSPMediaPrivate *priv = media->priv;
2171 g_mutex_lock (&priv->lock);
2172 priv->status = status;
2173 GST_DEBUG ("setting new status to %d", status);
2174 g_cond_broadcast (&priv->cond);
2175 g_mutex_unlock (&priv->lock);
2179 * gst_rtsp_media_get_status:
2180 * @media: a #GstRTSPMedia
2182 * Get the status of @media. When @media is busy preparing, this function waits
2183 * until @media is prepared or in error.
2185 * Returns: the status of @media.
2188 gst_rtsp_media_get_status (GstRTSPMedia * media)
2190 GstRTSPMediaPrivate *priv = media->priv;
2191 GstRTSPMediaStatus result;
2194 g_mutex_lock (&priv->lock);
2195 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2196 /* while we are preparing, wait */
2197 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2198 GST_DEBUG ("waiting for status change");
2199 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2200 GST_DEBUG ("timeout, assuming error status");
2201 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2204 /* could be success or error */
2205 result = priv->status;
2206 GST_DEBUG ("got status %d", result);
2207 g_mutex_unlock (&priv->lock);
2213 * gst_rtsp_media_seek_full:
2214 * @media: a #GstRTSPMedia
2215 * @range: (transfer none): a #GstRTSPTimeRange
2216 * @flags: The minimal set of #GstSeekFlags to use
2218 * Seek the pipeline of @media to @range. @media must be prepared with
2219 * gst_rtsp_media_prepare().
2221 * Returns: %TRUE on success.
2224 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2227 GstRTSPMediaClass *klass;
2228 GstRTSPMediaPrivate *priv;
2230 GstClockTime start, stop;
2231 GstSeekType start_type, stop_type;
2232 gint64 current_position;
2234 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2236 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2237 g_return_val_if_fail (range != NULL, FALSE);
2238 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2242 g_rec_mutex_lock (&priv->state_lock);
2243 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2246 /* check if the media pipeline is complete in order to perform a
2247 * seek operation on it */
2248 if (!check_complete (media))
2251 /* Update the seekable state of the pipeline in case it changed */
2252 check_seekable (media);
2254 if (priv->seekable == 0) {
2255 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2256 " not seekable streams.");
2259 } else if (priv->seekable < 0) {
2263 start_type = stop_type = GST_SEEK_TYPE_NONE;
2265 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2267 gst_rtsp_range_get_times (range, &start, &stop);
2269 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2270 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2271 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2272 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2274 current_position = -1;
2275 if (klass->query_position)
2276 klass->query_position (media, ¤t_position);
2277 GST_INFO ("current media position %" GST_TIME_FORMAT,
2278 GST_TIME_ARGS (current_position));
2280 if (start != GST_CLOCK_TIME_NONE)
2281 start_type = GST_SEEK_TYPE_SET;
2283 if (priv->range_stop == stop)
2284 stop = GST_CLOCK_TIME_NONE;
2285 else if (stop != GST_CLOCK_TIME_NONE)
2286 stop_type = GST_SEEK_TYPE_SET;
2288 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2289 gboolean had_flags = flags != 0;
2291 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2292 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2294 /* depends on the current playing state of the pipeline. We might need to
2295 * queue this until we get EOS. */
2297 flags |= GST_SEEK_FLAG_FLUSH;
2299 flags = GST_SEEK_FLAG_FLUSH;
2302 /* if range start was not supplied we must continue from current position.
2303 * but since we're doing a flushing seek, let us query the current position
2304 * so we end up at exactly the same position after the seek. */
2305 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2306 if (current_position == -1) {
2307 GST_WARNING ("current position unknown");
2309 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2310 GST_TIME_ARGS (current_position));
2311 start = current_position;
2312 start_type = GST_SEEK_TYPE_SET;
2314 flags |= GST_SEEK_FLAG_ACCURATE;
2317 /* only set keyframe flag when modifying start */
2318 if (start_type != GST_SEEK_TYPE_NONE)
2320 flags |= GST_SEEK_FLAG_KEY_UNIT;
2323 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2324 GST_DEBUG ("not seeking because no position change");
2327 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2329 media_streams_set_blocked (media, TRUE);
2331 /* FIXME, we only do forwards playback, no trick modes yet */
2332 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2333 flags, start_type, start, stop_type, stop);
2335 /* and block for the seek to complete */
2336 GST_INFO ("done seeking %d", res);
2340 g_rec_mutex_unlock (&priv->state_lock);
2342 /* wait until pipeline is prerolled again, this will also collect stats */
2343 if (!wait_preroll (media))
2344 goto preroll_failed;
2346 g_rec_mutex_lock (&priv->state_lock);
2347 GST_INFO ("prerolled again");
2350 GST_INFO ("no seek needed");
2353 g_rec_mutex_unlock (&priv->state_lock);
2360 g_rec_mutex_unlock (&priv->state_lock);
2361 GST_INFO ("media %p is not prepared", media);
2366 g_rec_mutex_unlock (&priv->state_lock);
2367 GST_INFO ("pipeline is not complete");
2372 g_rec_mutex_unlock (&priv->state_lock);
2373 GST_INFO ("pipeline is not seekable");
2378 g_rec_mutex_unlock (&priv->state_lock);
2379 GST_WARNING ("conversion to npt not supported");
2384 g_rec_mutex_unlock (&priv->state_lock);
2385 GST_INFO ("seeking failed");
2386 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2391 GST_WARNING ("failed to preroll after seek");
2398 * gst_rtsp_media_seek:
2399 * @media: a #GstRTSPMedia
2400 * @range: (transfer none): a #GstRTSPTimeRange
2402 * Seek the pipeline of @media to @range. @media must be prepared with
2403 * gst_rtsp_media_prepare().
2405 * Returns: %TRUE on success.
2408 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2410 return gst_rtsp_media_seek_full (media, range, 0);
2415 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2417 *blocked &= gst_rtsp_stream_is_blocking (stream);
2421 media_streams_blocking (GstRTSPMedia * media)
2423 gboolean blocking = TRUE;
2425 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2431 static GstStateChangeReturn
2432 set_state (GstRTSPMedia * media, GstState state)
2434 GstRTSPMediaPrivate *priv = media->priv;
2435 GstStateChangeReturn ret;
2437 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2439 ret = gst_element_set_state (priv->pipeline, state);
2444 static GstStateChangeReturn
2445 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2447 GstRTSPMediaPrivate *priv = media->priv;
2448 GstStateChangeReturn ret;
2450 GST_INFO ("set target state to %s for media %p",
2451 gst_element_state_get_name (state), media);
2452 priv->target_state = state;
2454 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2455 priv->target_state, NULL);
2458 ret = set_state (media, state);
2460 ret = GST_STATE_CHANGE_SUCCESS;
2465 /* called with state-lock */
2467 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2469 GstRTSPMediaPrivate *priv = media->priv;
2470 GstMessageType type;
2472 type = GST_MESSAGE_TYPE (message);
2475 case GST_MESSAGE_STATE_CHANGED:
2477 GstState old, new, pending;
2479 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2482 gst_message_parse_state_changed (message, &old, &new, &pending);
2484 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2485 gst_element_state_get_name (old), gst_element_state_get_name (new),
2486 gst_element_state_get_name (pending));
2487 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
2488 && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2489 GST_INFO ("%p: went to PAUSED, prepared now", media);
2490 collect_media_stats (media);
2492 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2493 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2498 case GST_MESSAGE_BUFFERING:
2502 gst_message_parse_buffering (message, &percent);
2504 /* no state management needed for live pipelines */
2508 if (percent == 100) {
2509 /* a 100% message means buffering is done */
2510 priv->buffering = FALSE;
2511 /* if the desired state is playing, go back */
2512 if (priv->target_state == GST_STATE_PLAYING) {
2513 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2514 set_state (media, GST_STATE_PLAYING);
2516 GST_INFO ("Buffering done");
2519 /* buffering busy */
2520 if (priv->buffering == FALSE) {
2521 if (priv->target_state == GST_STATE_PLAYING) {
2522 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2523 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2524 set_state (media, GST_STATE_PAUSED);
2526 GST_INFO ("Buffering ...");
2529 priv->buffering = TRUE;
2533 case GST_MESSAGE_LATENCY:
2535 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2538 case GST_MESSAGE_ERROR:
2543 gst_message_parse_error (message, &gerror, &debug);
2544 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2545 g_error_free (gerror);
2548 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2551 case GST_MESSAGE_WARNING:
2556 gst_message_parse_warning (message, &gerror, &debug);
2557 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2558 g_error_free (gerror);
2562 case GST_MESSAGE_ELEMENT:
2564 const GstStructure *s;
2566 s = gst_message_get_structure (message);
2567 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2568 GST_DEBUG ("media received blocking message");
2569 if (priv->blocked && media_streams_blocking (media) &&
2570 priv->no_more_pads_pending == 0) {
2571 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
2572 collect_media_stats (media);
2574 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2575 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2580 case GST_MESSAGE_STREAM_STATUS:
2582 case GST_MESSAGE_ASYNC_DONE:
2583 if (priv->complete) {
2584 /* receive the final ASYNC_DONE, that is posted by the media pipeline
2585 * after all the transport parts have been successfully added to
2586 * the media streams. */
2587 GST_DEBUG_OBJECT (media, "got async-done");
2588 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2589 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2592 case GST_MESSAGE_EOS:
2593 GST_INFO ("%p: got EOS", media);
2595 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2596 GST_DEBUG ("shutting down after EOS");
2597 finish_unprepare (media);
2601 GST_INFO ("%p: got message type %d (%s)", media, type,
2602 gst_message_type_get_name (type));
2609 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2611 GstRTSPMediaPrivate *priv = media->priv;
2612 GstRTSPMediaClass *klass;
2615 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2617 g_rec_mutex_lock (&priv->state_lock);
2618 if (klass->handle_message)
2619 ret = klass->handle_message (media, message);
2622 g_rec_mutex_unlock (&priv->state_lock);
2628 watch_destroyed (GstRTSPMedia * media)
2630 GST_DEBUG_OBJECT (media, "source destroyed");
2631 g_object_unref (media);
2635 find_payload_element (GstElement * payloader)
2637 GstElement *pay = NULL;
2639 if (GST_IS_BIN (payloader)) {
2641 GValue item = { 0 };
2643 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2644 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2645 GstElement *element = (GstElement *) g_value_get_object (&item);
2646 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2650 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2654 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2655 pay = gst_object_ref (element);
2656 g_value_unset (&item);
2659 g_value_unset (&item);
2661 gst_iterator_free (iter);
2663 pay = g_object_ref (payloader);
2669 /* called from streaming threads */
2671 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2673 GstRTSPMediaPrivate *priv = media->priv;
2674 GstRTSPStream *stream;
2677 /* find the real payload element */
2678 pay = find_payload_element (element);
2679 stream = gst_rtsp_media_create_stream (media, pay, pad);
2680 gst_object_unref (pay);
2682 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2684 g_rec_mutex_lock (&priv->state_lock);
2685 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2688 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2690 /* join the element in the PAUSED state because this callback is
2691 * called from the streaming thread and it is PAUSED */
2692 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2693 priv->rtpbin, GST_STATE_PAUSED)) {
2694 GST_WARNING ("failed to join bin element");
2698 gst_rtsp_stream_set_blocked (stream, TRUE);
2700 g_rec_mutex_unlock (&priv->state_lock);
2707 gst_rtsp_media_remove_stream (media, stream);
2708 g_rec_mutex_unlock (&priv->state_lock);
2709 GST_INFO ("ignore pad because we are not preparing");
2715 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2717 GstRTSPMediaPrivate *priv = media->priv;
2718 GstRTSPStream *stream;
2720 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2724 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2726 g_rec_mutex_lock (&priv->state_lock);
2727 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2728 g_rec_mutex_unlock (&priv->state_lock);
2730 gst_rtsp_media_remove_stream (media, stream);
2734 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2736 GstRTSPMediaPrivate *priv = media->priv;
2738 GST_INFO_OBJECT (element, "no more pads");
2739 g_mutex_lock (&priv->lock);
2740 priv->no_more_pads_pending--;
2741 g_mutex_unlock (&priv->lock);
2744 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2746 struct _DynPaySignalHandlers
2748 gulong pad_added_handler;
2749 gulong pad_removed_handler;
2750 gulong no_more_pads_handler;
2754 start_preroll (GstRTSPMedia * media)
2756 GstRTSPMediaPrivate *priv = media->priv;
2757 GstStateChangeReturn ret;
2759 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2761 /* start blocked since it is possible that there are no sink elements yet */
2762 media_streams_set_blocked (media, TRUE);
2763 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2766 case GST_STATE_CHANGE_SUCCESS:
2767 GST_INFO ("SUCCESS state change for media %p", media);
2769 case GST_STATE_CHANGE_ASYNC:
2770 GST_INFO ("ASYNC state change for media %p", media);
2772 case GST_STATE_CHANGE_NO_PREROLL:
2773 /* we need to go to PLAYING */
2774 GST_INFO ("NO_PREROLL state change: live media %p", media);
2775 /* FIXME we disable seeking for live streams for now. We should perform a
2776 * seeking query in preroll instead */
2777 priv->seekable = -1;
2778 priv->is_live = TRUE;
2780 ret = set_state (media, GST_STATE_PLAYING);
2781 if (ret == GST_STATE_CHANGE_FAILURE)
2784 case GST_STATE_CHANGE_FAILURE:
2792 GST_WARNING ("failed to preroll pipeline");
2798 wait_preroll (GstRTSPMedia * media)
2800 GstRTSPMediaStatus status;
2802 GST_DEBUG ("wait to preroll pipeline");
2804 /* wait until pipeline is prerolled */
2805 status = gst_rtsp_media_get_status (media);
2806 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2807 goto preroll_failed;
2813 GST_WARNING ("failed to preroll pipeline");
2819 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
2821 GstRTSPMediaPrivate *priv = media->priv;
2822 GstRTSPStream *stream = NULL;
2825 g_mutex_lock (&priv->lock);
2826 for (i = 0; i < priv->streams->len; i++) {
2827 stream = g_ptr_array_index (priv->streams, i);
2829 if (sessid == gst_rtsp_stream_get_index (stream))
2832 g_mutex_unlock (&priv->lock);
2834 return gst_rtsp_stream_request_aux_sender (stream, sessid);
2838 start_prepare (GstRTSPMedia * media)
2840 GstRTSPMediaPrivate *priv = media->priv;
2844 g_rec_mutex_lock (&priv->state_lock);
2845 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2846 goto no_longer_preparing;
2848 /* link streams we already have, other streams might appear when we have
2849 * dynamic elements */
2850 for (i = 0; i < priv->streams->len; i++) {
2851 GstRTSPStream *stream;
2853 stream = g_ptr_array_index (priv->streams, i);
2855 if (priv->rtx_time > 0) {
2856 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2857 g_signal_connect (priv->rtpbin, "request-aux-sender",
2858 (GCallback) request_aux_sender, media);
2861 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2862 priv->rtpbin, GST_STATE_NULL)) {
2863 goto join_bin_failed;
2868 g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
2870 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2871 GstElement *elem = walk->data;
2872 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2874 GST_INFO ("adding callbacks for dynamic element %p", elem);
2876 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2877 (GCallback) pad_added_cb, media);
2878 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2879 (GCallback) pad_removed_cb, media);
2880 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2881 (GCallback) no_more_pads_cb, media);
2883 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2886 if (!start_preroll (media))
2887 goto preroll_failed;
2889 g_rec_mutex_unlock (&priv->state_lock);
2893 no_longer_preparing:
2895 GST_INFO ("media is no longer preparing");
2896 g_rec_mutex_unlock (&priv->state_lock);
2901 GST_WARNING ("failed to join bin element");
2902 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2903 g_rec_mutex_unlock (&priv->state_lock);
2908 GST_WARNING ("failed to preroll pipeline");
2909 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2910 g_rec_mutex_unlock (&priv->state_lock);
2916 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2918 GstRTSPMediaPrivate *priv;
2919 GstRTSPMediaClass *klass;
2921 GMainContext *context;
2926 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2928 if (!klass->create_rtpbin)
2929 goto no_create_rtpbin;
2931 priv->rtpbin = klass->create_rtpbin (media);
2932 if (priv->rtpbin != NULL) {
2933 gboolean success = TRUE;
2935 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
2937 if (klass->setup_rtpbin)
2938 success = klass->setup_rtpbin (media, priv->rtpbin);
2940 if (success == FALSE) {
2941 gst_object_unref (priv->rtpbin);
2942 priv->rtpbin = NULL;
2945 if (priv->rtpbin == NULL)
2948 priv->thread = thread;
2949 context = (thread != NULL) ? (thread->context) : NULL;
2951 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2953 /* add the pipeline bus to our custom mainloop */
2954 priv->source = gst_bus_create_watch (bus);
2955 gst_object_unref (bus);
2957 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2958 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2960 priv->id = g_source_attach (priv->source, context);
2962 /* add stuff to the bin */
2963 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2965 /* do remainder in context */
2966 source = g_idle_source_new ();
2967 g_source_set_callback (source, (GSourceFunc) start_prepare,
2968 g_object_ref (media), (GDestroyNotify) g_object_unref);
2969 g_source_attach (source, context);
2970 g_source_unref (source);
2977 GST_ERROR ("no create_rtpbin function");
2978 g_critical ("no create_rtpbin vmethod function set");
2983 GST_WARNING ("no rtpbin element");
2984 g_warning ("failed to create element 'rtpbin', check your installation");
2990 * gst_rtsp_media_prepare:
2991 * @media: a #GstRTSPMedia
2992 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2993 * bus handler or %NULL
2995 * Prepare @media for streaming. This function will create the objects
2996 * to manage the streaming. A pipeline must have been set on @media with
2997 * gst_rtsp_media_take_pipeline().
2999 * It will preroll the pipeline and collect vital information about the streams
3000 * such as the duration.
3002 * Returns: %TRUE on success.
3005 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
3007 GstRTSPMediaPrivate *priv;
3008 GstRTSPMediaClass *klass;
3010 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3014 g_rec_mutex_lock (&priv->state_lock);
3015 priv->prepare_count++;
3017 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3018 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3021 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3024 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3025 goto not_unprepared;
3027 if (!priv->reusable && priv->reused)
3030 GST_INFO ("preparing media %p", media);
3032 /* reset some variables */
3033 priv->is_live = FALSE;
3034 priv->seekable = -1;
3035 priv->buffering = FALSE;
3036 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3038 /* we're preparing now */
3039 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3041 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3042 if (klass->prepare) {
3043 if (!klass->prepare (media, thread))
3044 goto prepare_failed;
3048 g_rec_mutex_unlock (&priv->state_lock);
3050 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3051 * able to do this async so that we don't block the server thread. */
3052 if (!wait_preroll (media))
3053 goto preroll_failed;
3055 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3057 GST_INFO ("object %p is prerolled", media);
3064 /* we are not going to use the giving thread, so stop it. */
3066 gst_rtsp_thread_stop (thread);
3071 GST_LOG ("media %p was prepared", media);
3072 /* we are not going to use the giving thread, so stop it. */
3074 gst_rtsp_thread_stop (thread);
3075 g_rec_mutex_unlock (&priv->state_lock);
3081 /* we are not going to use the giving thread, so stop it. */
3083 gst_rtsp_thread_stop (thread);
3084 GST_WARNING ("media %p was not unprepared", media);
3085 priv->prepare_count--;
3086 g_rec_mutex_unlock (&priv->state_lock);
3091 /* we are not going to use the giving thread, so stop it. */
3093 gst_rtsp_thread_stop (thread);
3094 priv->prepare_count--;
3095 g_rec_mutex_unlock (&priv->state_lock);
3096 GST_WARNING ("can not reuse media %p", media);
3101 /* we are not going to use the giving thread, so stop it. */
3103 gst_rtsp_thread_stop (thread);
3104 priv->prepare_count--;
3105 g_rec_mutex_unlock (&priv->state_lock);
3106 GST_ERROR ("failed to prepare media");
3111 GST_WARNING ("failed to preroll pipeline");
3112 gst_rtsp_media_unprepare (media);
3117 /* must be called with state-lock */
3119 finish_unprepare (GstRTSPMedia * media)
3121 GstRTSPMediaPrivate *priv = media->priv;
3125 GST_DEBUG ("shutting down");
3127 /* release the lock on shutdown, otherwise pad_added_cb might try to
3128 * acquire the lock and then we deadlock */
3129 g_rec_mutex_unlock (&priv->state_lock);
3130 set_state (media, GST_STATE_NULL);
3131 g_rec_mutex_lock (&priv->state_lock);
3133 media_streams_set_blocked (media, FALSE);
3135 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
3138 for (i = 0; i < priv->streams->len; i++) {
3139 GstRTSPStream *stream;
3141 GST_INFO ("Removing elements of stream %d from pipeline", i);
3143 stream = g_ptr_array_index (priv->streams, i);
3145 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3148 /* remove the pad signal handlers */
3149 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3150 GstElement *elem = walk->data;
3151 DynPaySignalHandlers *handlers;
3154 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3155 g_assert (handlers != NULL);
3157 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3158 g_signal_handler_disconnect (G_OBJECT (elem),
3159 handlers->pad_removed_handler);
3160 g_signal_handler_disconnect (G_OBJECT (elem),
3161 handlers->no_more_pads_handler);
3163 g_slice_free (DynPaySignalHandlers, handlers);
3166 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3167 priv->rtpbin = NULL;
3170 gst_object_unref (priv->nettime);
3171 priv->nettime = NULL;
3173 priv->reused = TRUE;
3174 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3176 /* when the media is not reusable, this will effectively unref the media and
3178 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3180 /* the source has the last ref to the media */
3182 GST_DEBUG ("destroy source");
3183 g_source_destroy (priv->source);
3184 g_source_unref (priv->source);
3187 GST_DEBUG ("stop thread");
3188 gst_rtsp_thread_stop (priv->thread);
3192 /* called with state-lock */
3194 default_unprepare (GstRTSPMedia * media)
3196 GstRTSPMediaPrivate *priv = media->priv;
3198 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3200 if (priv->eos_shutdown) {
3201 GST_DEBUG ("sending EOS for shutdown");
3202 /* ref so that we don't disappear */
3203 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3204 /* we need to go to playing again for the EOS to propagate, normally in this
3205 * state, nothing is receiving data from us anymore so this is ok. */
3206 set_state (media, GST_STATE_PLAYING);
3208 finish_unprepare (media);
3214 * gst_rtsp_media_unprepare:
3215 * @media: a #GstRTSPMedia
3217 * Unprepare @media. After this call, the media should be prepared again before
3218 * it can be used again. If the media is set to be non-reusable, a new instance
3221 * Returns: %TRUE on success.
3224 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3226 GstRTSPMediaPrivate *priv;
3229 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3233 g_rec_mutex_lock (&priv->state_lock);
3234 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3235 goto was_unprepared;
3237 priv->prepare_count--;
3238 if (priv->prepare_count > 0)
3241 GST_INFO ("unprepare media %p", media);
3242 set_target_state (media, GST_STATE_NULL, FALSE);
3245 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3246 GstRTSPMediaClass *klass;
3248 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3249 if (klass->unprepare)
3250 success = klass->unprepare (media);
3252 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3253 finish_unprepare (media);
3255 g_rec_mutex_unlock (&priv->state_lock);
3261 g_rec_mutex_unlock (&priv->state_lock);
3262 GST_INFO ("media %p was already unprepared", media);
3267 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3268 g_rec_mutex_unlock (&priv->state_lock);
3273 /* should be called with state-lock */
3275 get_clock_unlocked (GstRTSPMedia * media)
3277 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3278 GST_DEBUG_OBJECT (media, "media was not prepared");
3281 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3285 * gst_rtsp_media_get_clock:
3286 * @media: a #GstRTSPMedia
3288 * Get the clock that is used by the pipeline in @media.
3290 * @media must be prepared before this method returns a valid clock object.
3292 * Returns: (transfer full) (nullable): the #GstClock used by @media. unref after usage.
3295 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3298 GstRTSPMediaPrivate *priv;
3300 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3304 g_rec_mutex_lock (&priv->state_lock);
3305 clock = get_clock_unlocked (media);
3306 g_rec_mutex_unlock (&priv->state_lock);
3312 * gst_rtsp_media_get_base_time:
3313 * @media: a #GstRTSPMedia
3315 * Get the base_time that is used by the pipeline in @media.
3317 * @media must be prepared before this method returns a valid base_time.
3319 * Returns: the base_time used by @media.
3322 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3324 GstClockTime result;
3325 GstRTSPMediaPrivate *priv;
3327 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3331 g_rec_mutex_lock (&priv->state_lock);
3332 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3335 result = gst_element_get_base_time (media->priv->pipeline);
3336 g_rec_mutex_unlock (&priv->state_lock);
3343 g_rec_mutex_unlock (&priv->state_lock);
3344 GST_DEBUG_OBJECT (media, "media was not prepared");
3345 return GST_CLOCK_TIME_NONE;
3350 * gst_rtsp_media_get_time_provider:
3351 * @media: a #GstRTSPMedia
3352 * @address: (allow-none): an address or %NULL
3353 * @port: a port or 0
3355 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3356 * will listen on @address and @port for client time requests.
3358 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3360 GstNetTimeProvider *
3361 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3364 GstRTSPMediaPrivate *priv;
3365 GstNetTimeProvider *provider = NULL;
3367 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3371 g_rec_mutex_lock (&priv->state_lock);
3372 if (priv->time_provider) {
3373 if ((provider = priv->nettime) == NULL) {
3376 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3377 provider = gst_net_time_provider_new (clock, address, port);
3378 gst_object_unref (clock);
3380 priv->nettime = provider;
3384 g_rec_mutex_unlock (&priv->state_lock);
3387 gst_object_ref (provider);
3393 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3395 return gst_rtsp_sdp_from_media (sdp, info, media);
3399 * gst_rtsp_media_setup_sdp:
3400 * @media: a #GstRTSPMedia
3401 * @sdp: (transfer none): a #GstSDPMessage
3402 * @info: (transfer none): a #GstSDPInfo
3404 * Add @media specific info to @sdp. @info is used to configure the connection
3405 * information in the SDP.
3407 * Returns: TRUE on success.
3410 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3413 GstRTSPMediaPrivate *priv;
3414 GstRTSPMediaClass *klass;
3417 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3418 g_return_val_if_fail (sdp != NULL, FALSE);
3419 g_return_val_if_fail (info != NULL, FALSE);
3423 g_rec_mutex_lock (&priv->state_lock);
3425 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3427 if (!klass->setup_sdp)
3430 res = klass->setup_sdp (media, sdp, info);
3432 g_rec_mutex_unlock (&priv->state_lock);
3439 g_rec_mutex_unlock (&priv->state_lock);
3440 GST_ERROR ("no setup_sdp function");
3441 g_critical ("no setup_sdp vmethod function set");
3447 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3449 GstRTSPMediaPrivate *priv = media->priv;
3452 medias_len = gst_sdp_message_medias_len (sdp);
3453 if (medias_len != priv->streams->len) {
3454 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3455 priv->streams->len, medias_len);
3459 for (i = 0; i < medias_len; i++) {
3461 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3462 GstRTSPStream *stream;
3463 gint j, formats_len;
3464 const gchar *control;
3465 GstRTSPProfile profile, profiles;
3467 stream = g_ptr_array_index (priv->streams, i);
3469 /* TODO: Should we do something with the other SDP information? */
3472 proto = gst_sdp_media_get_proto (sdp_media);
3473 if (proto == NULL) {
3474 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3478 if (g_str_equal (proto, "RTP/AVP")) {
3479 profile = GST_RTSP_PROFILE_AVP;
3480 } else if (g_str_equal (proto, "RTP/SAVP")) {
3481 profile = GST_RTSP_PROFILE_SAVP;
3482 } else if (g_str_equal (proto, "RTP/AVPF")) {
3483 profile = GST_RTSP_PROFILE_AVPF;
3484 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3485 profile = GST_RTSP_PROFILE_SAVPF;
3487 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3491 profiles = gst_rtsp_stream_get_profiles (stream);
3492 if ((profiles & profile) == 0) {
3493 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3497 formats_len = gst_sdp_media_formats_len (sdp_media);
3498 for (j = 0; j < formats_len; j++) {
3503 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3505 GST_DEBUG (" looking at %d pt: %d", j, pt);
3508 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
3510 GST_WARNING (" skipping pt %d without caps", pt);
3514 /* do some tweaks */
3515 GST_DEBUG ("mapping sdp session level attributes to caps");
3516 gst_sdp_message_attributes_to_caps (sdp, caps);
3517 GST_DEBUG ("mapping sdp media level attributes to caps");
3518 gst_sdp_media_attributes_to_caps (sdp_media, caps);
3520 s = gst_caps_get_structure (caps, 0);
3521 gst_structure_set_name (s, "application/x-rtp");
3523 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3524 gst_caps_unref (caps);
3527 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3529 gst_rtsp_stream_set_control (stream, control);
3537 * gst_rtsp_media_handle_sdp:
3538 * @media: a #GstRTSPMedia
3539 * @sdp: (transfer none): a #GstSDPMessage
3541 * Configure an SDP on @media for receiving streams
3543 * Returns: TRUE on success.
3546 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3548 GstRTSPMediaPrivate *priv;
3549 GstRTSPMediaClass *klass;
3552 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3553 g_return_val_if_fail (sdp != NULL, FALSE);
3557 g_rec_mutex_lock (&priv->state_lock);
3559 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3561 if (!klass->handle_sdp)
3564 res = klass->handle_sdp (media, sdp);
3566 g_rec_mutex_unlock (&priv->state_lock);
3573 g_rec_mutex_unlock (&priv->state_lock);
3574 GST_ERROR ("no handle_sdp function");
3575 g_critical ("no handle_sdp vmethod function set");
3581 do_set_seqnum (GstRTSPStream * stream)
3584 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3585 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3588 /* call with state_lock */
3590 default_suspend (GstRTSPMedia * media)
3592 GstRTSPMediaPrivate *priv = media->priv;
3593 GstStateChangeReturn ret;
3595 switch (priv->suspend_mode) {
3596 case GST_RTSP_SUSPEND_MODE_NONE:
3597 GST_DEBUG ("media %p no suspend", media);
3599 case GST_RTSP_SUSPEND_MODE_PAUSE:
3600 GST_DEBUG ("media %p suspend to PAUSED", media);
3601 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3602 if (ret == GST_STATE_CHANGE_FAILURE)
3605 case GST_RTSP_SUSPEND_MODE_RESET:
3606 GST_DEBUG ("media %p suspend to NULL", media);
3607 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3608 if (ret == GST_STATE_CHANGE_FAILURE)
3610 /* Because payloader needs to set the sequence number as
3611 * monotonic, we need to preserve the sequence number
3612 * after pause. (otherwise going from pause to play, which
3613 * is actually from NULL to PLAY will create a new sequence
3615 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3626 GST_WARNING ("failed changing pipeline's state for media %p", media);
3632 * gst_rtsp_media_suspend:
3633 * @media: a #GstRTSPMedia
3635 * Suspend @media. The state of the pipeline managed by @media is set to
3636 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3637 * with gst_rtsp_media_unsuspend()
3639 * @media must be prepared with gst_rtsp_media_prepare();
3641 * Returns: %TRUE on success.
3644 gst_rtsp_media_suspend (GstRTSPMedia * media)
3646 GstRTSPMediaPrivate *priv = media->priv;
3647 GstRTSPMediaClass *klass;
3649 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3651 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3653 g_rec_mutex_lock (&priv->state_lock);
3654 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3657 /* don't attempt to suspend when something is busy */
3658 if (priv->n_active > 0)
3661 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3662 if (klass->suspend) {
3663 if (!klass->suspend (media))
3664 goto suspend_failed;
3667 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3669 g_rec_mutex_unlock (&priv->state_lock);
3676 g_rec_mutex_unlock (&priv->state_lock);
3677 GST_WARNING ("media %p was not prepared", media);
3682 g_rec_mutex_unlock (&priv->state_lock);
3683 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3684 GST_WARNING ("failed to suspend media %p", media);
3689 /* call with state_lock */
3691 default_unsuspend (GstRTSPMedia * media)
3693 GstRTSPMediaPrivate *priv = media->priv;
3694 gboolean preroll_ok;
3696 switch (priv->suspend_mode) {
3697 case GST_RTSP_SUSPEND_MODE_NONE:
3698 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD))
3700 if (media_streams_blocking (media)) {
3701 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3702 /* at this point the media pipeline has been updated and contain all
3703 * specific transport parts: all active streams contain at least one sink
3704 * element and it's safe to unblock any blocked streams that are active */
3705 media_unblock_linked (media);
3707 /* streams are not blocked and media is suspended from PAUSED */
3708 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3710 g_rec_mutex_unlock (&priv->state_lock);
3711 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
3712 g_rec_mutex_lock (&priv->state_lock);
3713 goto preroll_failed;
3715 g_rec_mutex_lock (&priv->state_lock);
3717 case GST_RTSP_SUSPEND_MODE_PAUSE:
3718 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3720 case GST_RTSP_SUSPEND_MODE_RESET:
3722 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3723 /* at this point the media pipeline has been updated and contain all
3724 * specific transport parts: all active streams contain at least one sink
3725 * element and it's safe to unblock any blocked streams that are active */
3726 media_unblock_linked (media);
3727 if (!start_preroll (media))
3730 g_rec_mutex_unlock (&priv->state_lock);
3731 preroll_ok = wait_preroll (media);
3732 g_rec_mutex_lock (&priv->state_lock);
3735 goto preroll_failed;
3746 GST_WARNING ("failed to preroll pipeline");
3751 GST_WARNING ("failed to preroll pipeline");
3757 * gst_rtsp_media_unsuspend:
3758 * @media: a #GstRTSPMedia
3760 * Unsuspend @media if it was in a suspended state. This method does nothing
3761 * when the media was not in the suspended state.
3763 * Returns: %TRUE on success.
3766 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
3768 GstRTSPMediaPrivate *priv = media->priv;
3769 GstRTSPMediaClass *klass;
3771 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3773 g_rec_mutex_lock (&priv->state_lock);
3774 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3777 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3778 if (klass->unsuspend) {
3779 if (!klass->unsuspend (media))
3780 goto unsuspend_failed;
3784 g_rec_mutex_unlock (&priv->state_lock);
3791 g_rec_mutex_unlock (&priv->state_lock);
3792 GST_WARNING ("failed to unsuspend media %p", media);
3793 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3798 /* must be called with state-lock */
3800 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
3802 GstRTSPMediaPrivate *priv = media->priv;
3804 if (state == GST_STATE_NULL) {
3805 gst_rtsp_media_unprepare (media);
3807 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
3808 set_target_state (media, state, FALSE);
3809 /* when we are buffering, don't update the state yet, this will be done
3810 * when buffering finishes */
3811 if (priv->buffering) {
3812 GST_INFO ("Buffering busy, delay state change");
3814 if (state == GST_STATE_PLAYING)
3815 /* make sure pads are not blocking anymore when going to PLAYING */
3816 media_unblock_linked (media);
3818 set_state (media, state);
3820 /* and suspend after pause */
3821 if (state == GST_STATE_PAUSED)
3822 gst_rtsp_media_suspend (media);
3828 * gst_rtsp_media_set_pipeline_state:
3829 * @media: a #GstRTSPMedia
3830 * @state: the target state of the pipeline
3832 * Set the state of the pipeline managed by @media to @state
3835 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
3837 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
3839 g_rec_mutex_lock (&media->priv->state_lock);
3840 media_set_pipeline_state_locked (media, state);
3841 g_rec_mutex_unlock (&media->priv->state_lock);
3845 * gst_rtsp_media_set_state:
3846 * @media: a #GstRTSPMedia
3847 * @state: the target state of the media
3848 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
3849 * a #GPtrArray of #GstRTSPStreamTransport pointers
3851 * Set the state of @media to @state and for the transports in @transports.
3853 * @media must be prepared with gst_rtsp_media_prepare();
3855 * Returns: %TRUE on success.
3858 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
3859 GPtrArray * transports)
3861 GstRTSPMediaPrivate *priv;
3863 gboolean activate, deactivate, do_state;
3866 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3867 g_return_val_if_fail (transports != NULL, FALSE);
3871 g_rec_mutex_lock (&priv->state_lock);
3872 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
3874 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
3875 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3878 /* NULL and READY are the same */
3879 if (state == GST_STATE_READY)
3880 state = GST_STATE_NULL;
3882 activate = deactivate = FALSE;
3884 GST_INFO ("going to state %s media %p, target state %s",
3885 gst_element_state_get_name (state), media,
3886 gst_element_state_get_name (priv->target_state));
3889 case GST_STATE_NULL:
3890 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
3891 if (priv->target_state >= GST_STATE_PAUSED)
3894 case GST_STATE_PAUSED:
3895 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
3896 if (priv->target_state == GST_STATE_PLAYING)
3899 case GST_STATE_PLAYING:
3900 /* we're going to PLAYING, activate */
3906 old_active = priv->n_active;
3908 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
3909 activate, deactivate);
3910 for (i = 0; i < transports->len; i++) {
3911 GstRTSPStreamTransport *trans;
3913 /* we need a non-NULL entry in the array */
3914 trans = g_ptr_array_index (transports, i);
3919 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
3921 } else if (deactivate) {
3922 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
3927 /* we just activated the first media, do the playing state change */
3928 if (old_active == 0 && activate)
3930 /* if we have no more active media, do the downward state changes */
3931 else if (priv->n_active == 0)
3936 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
3939 if (priv->target_state != state) {
3941 media_set_pipeline_state_locked (media, state);
3942 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
3947 /* remember where we are */
3948 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
3949 old_active != priv->n_active))
3950 collect_media_stats (media);
3952 g_rec_mutex_unlock (&priv->state_lock);
3959 GST_WARNING ("media %p was not prepared", media);
3960 g_rec_mutex_unlock (&priv->state_lock);
3965 GST_WARNING ("media %p in error status while changing to state %d",
3967 if (state == GST_STATE_NULL) {
3968 for (i = 0; i < transports->len; i++) {
3969 GstRTSPStreamTransport *trans;
3971 /* we need a non-NULL entry in the array */
3972 trans = g_ptr_array_index (transports, i);
3976 gst_rtsp_stream_transport_set_active (trans, FALSE);
3980 g_rec_mutex_unlock (&priv->state_lock);
3986 * gst_rtsp_media_set_transport_mode:
3987 * @media: a #GstRTSPMedia
3988 * @mode: the new value
3990 * Sets if the media pipeline can work in PLAY or RECORD mode
3993 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
3994 GstRTSPTransportMode mode)
3996 GstRTSPMediaPrivate *priv;
3998 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
4002 g_mutex_lock (&priv->lock);
4003 priv->transport_mode = mode;
4004 g_mutex_unlock (&priv->lock);
4008 * gst_rtsp_media_get_transport_mode:
4009 * @media: a #GstRTSPMedia
4011 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
4013 * Returns: The transport mode.
4015 GstRTSPTransportMode
4016 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4018 GstRTSPMediaPrivate *priv;
4019 GstRTSPTransportMode res;
4021 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4025 g_mutex_lock (&priv->lock);
4026 res = priv->transport_mode;
4027 g_mutex_unlock (&priv->lock);
4033 * gst_rtsp_media_get_seekable:
4034 * @media: a #GstRTSPMedia
4036 * Check if the pipeline for @media seek and up to what point in time,
4039 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4040 * and > 0 to indicate the longest duration between any two random access points.
4041 * %G_MAXINT64 means any value is possible.
4044 gst_rtsp_media_seekable (GstRTSPMedia * media)
4046 GstRTSPMediaPrivate *priv;
4047 GstClockTimeDiff res;
4049 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4053 /* Currently we are not able to seek on live streams,
4054 * and no stream is seekable only to the beginning */
4055 g_mutex_lock (&priv->lock);
4056 res = priv->seekable;
4057 g_mutex_unlock (&priv->lock);
4063 * gst_rtsp_media_complete_pipeline:
4064 * @media: a #GstRTSPMedia
4065 * @transports: (element-type GstRTSPTransport): a list of #GstRTSPTransport
4067 * Add a receiver and sender parts to the pipeline based on the transport from
4070 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4073 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4075 GstRTSPMediaPrivate *priv;
4078 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4079 g_return_val_if_fail (transports, FALSE);
4081 GST_DEBUG_OBJECT (media, "complete pipeline");
4085 g_mutex_lock (&priv->lock);
4086 for (i = 0; i < priv->streams->len; i++) {
4087 GstRTSPStreamTransport *transport;
4088 GstRTSPStream *stream;
4089 const GstRTSPTransport *rtsp_transport;
4091 transport = g_ptr_array_index (transports, i);
4095 stream = gst_rtsp_stream_transport_get_stream (transport);
4099 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4101 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4102 g_mutex_unlock (&priv->lock);
4107 priv->complete = TRUE;
4108 g_mutex_unlock (&priv->lock);