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;
708 gst_query_unref (query);
711 /* must be called with state lock */
713 check_complete (GstRTSPMedia * media)
715 GstRTSPMediaPrivate *priv = media->priv;
717 guint i, n = priv->streams->len;
719 for (i = 0; i < n; i++) {
720 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
722 if (gst_rtsp_stream_is_complete (stream))
729 /* must be called with state lock */
731 collect_media_stats (GstRTSPMedia * media)
733 GstRTSPMediaPrivate *priv = media->priv;
734 gint64 position = 0, stop = -1;
736 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
737 priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
740 priv->range.unit = GST_RTSP_RANGE_NPT;
742 GST_INFO ("collect media stats");
745 priv->range.min.type = GST_RTSP_TIME_NOW;
746 priv->range.min.seconds = -1;
747 priv->range_start = -1;
748 priv->range.max.type = GST_RTSP_TIME_END;
749 priv->range.max.seconds = -1;
750 priv->range_stop = -1;
752 GstRTSPMediaClass *klass;
755 klass = GST_RTSP_MEDIA_GET_CLASS (media);
757 /* get the position */
759 if (klass->query_position)
760 ret = klass->query_position (media, &position);
763 GST_INFO ("position query failed");
767 /* get the current segment stop */
769 if (klass->query_stop)
770 ret = klass->query_stop (media, &stop);
773 GST_INFO ("stop query failed");
777 GST_INFO ("stats: position %" GST_TIME_FORMAT ", stop %"
778 GST_TIME_FORMAT, GST_TIME_ARGS (position), GST_TIME_ARGS (stop));
780 if (position == -1) {
781 priv->range.min.type = GST_RTSP_TIME_NOW;
782 priv->range.min.seconds = -1;
783 priv->range_start = -1;
785 priv->range.min.type = GST_RTSP_TIME_SECONDS;
786 priv->range.min.seconds = ((gdouble) position) / GST_SECOND;
787 priv->range_start = position;
790 priv->range.max.type = GST_RTSP_TIME_END;
791 priv->range.max.seconds = -1;
792 priv->range_stop = -1;
794 priv->range.max.type = GST_RTSP_TIME_SECONDS;
795 priv->range.max.seconds = ((gdouble) stop) / GST_SECOND;
796 priv->range_stop = stop;
799 check_seekable (media);
804 * gst_rtsp_media_new:
805 * @element: (transfer full): a #GstElement
807 * Create a new #GstRTSPMedia instance. @element is the bin element that
808 * provides the different streams. The #GstRTSPMedia object contains the
809 * element to produce RTP data for one or more related (audio/video/..)
812 * Ownership is taken of @element.
814 * Returns: (transfer full): a new #GstRTSPMedia object.
817 gst_rtsp_media_new (GstElement * element)
819 GstRTSPMedia *result;
821 g_return_val_if_fail (GST_IS_ELEMENT (element), NULL);
823 result = g_object_new (GST_TYPE_RTSP_MEDIA, "element", element, NULL);
829 * gst_rtsp_media_get_element:
830 * @media: a #GstRTSPMedia
832 * Get the element that was used when constructing @media.
834 * Returns: (transfer full): a #GstElement. Unref after usage.
837 gst_rtsp_media_get_element (GstRTSPMedia * media)
839 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
841 return gst_object_ref (media->priv->element);
845 * gst_rtsp_media_take_pipeline:
846 * @media: a #GstRTSPMedia
847 * @pipeline: (transfer full): a #GstPipeline
849 * Set @pipeline as the #GstPipeline for @media. Ownership is
850 * taken of @pipeline.
853 gst_rtsp_media_take_pipeline (GstRTSPMedia * media, GstPipeline * pipeline)
855 GstRTSPMediaPrivate *priv;
857 GstNetTimeProvider *nettime;
859 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
860 g_return_if_fail (GST_IS_PIPELINE (pipeline));
864 g_mutex_lock (&priv->lock);
865 old = priv->pipeline;
866 priv->pipeline = GST_ELEMENT_CAST (pipeline);
867 nettime = priv->nettime;
868 priv->nettime = NULL;
869 g_mutex_unlock (&priv->lock);
872 gst_object_unref (old);
875 gst_object_unref (nettime);
877 gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
881 * gst_rtsp_media_set_permissions:
882 * @media: a #GstRTSPMedia
883 * @permissions: (transfer none): a #GstRTSPPermissions
885 * Set @permissions on @media.
888 gst_rtsp_media_set_permissions (GstRTSPMedia * media,
889 GstRTSPPermissions * permissions)
891 GstRTSPMediaPrivate *priv;
893 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
897 g_mutex_lock (&priv->lock);
898 if (priv->permissions)
899 gst_rtsp_permissions_unref (priv->permissions);
900 if ((priv->permissions = permissions))
901 gst_rtsp_permissions_ref (permissions);
902 g_mutex_unlock (&priv->lock);
906 * gst_rtsp_media_get_permissions:
907 * @media: a #GstRTSPMedia
909 * Get the permissions object from @media.
911 * Returns: (transfer full): a #GstRTSPPermissions object, unref after usage.
914 gst_rtsp_media_get_permissions (GstRTSPMedia * media)
916 GstRTSPMediaPrivate *priv;
917 GstRTSPPermissions *result;
919 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
923 g_mutex_lock (&priv->lock);
924 if ((result = priv->permissions))
925 gst_rtsp_permissions_ref (result);
926 g_mutex_unlock (&priv->lock);
932 * gst_rtsp_media_set_suspend_mode:
933 * @media: a #GstRTSPMedia
934 * @mode: the new #GstRTSPSuspendMode
936 * Control how @ media will be suspended after the SDP has been generated and
937 * after a PAUSE request has been performed.
939 * Media must be unprepared when setting the suspend mode.
942 gst_rtsp_media_set_suspend_mode (GstRTSPMedia * media, GstRTSPSuspendMode mode)
944 GstRTSPMediaPrivate *priv;
946 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
950 g_rec_mutex_lock (&priv->state_lock);
951 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED)
953 priv->suspend_mode = mode;
954 g_rec_mutex_unlock (&priv->state_lock);
961 GST_WARNING ("media %p was prepared", media);
962 g_rec_mutex_unlock (&priv->state_lock);
967 * gst_rtsp_media_get_suspend_mode:
968 * @media: a #GstRTSPMedia
970 * Get how @media will be suspended.
972 * Returns: #GstRTSPSuspendMode.
975 gst_rtsp_media_get_suspend_mode (GstRTSPMedia * media)
977 GstRTSPMediaPrivate *priv;
978 GstRTSPSuspendMode res;
980 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_SUSPEND_MODE_NONE);
984 g_rec_mutex_lock (&priv->state_lock);
985 res = priv->suspend_mode;
986 g_rec_mutex_unlock (&priv->state_lock);
992 * gst_rtsp_media_set_shared:
993 * @media: a #GstRTSPMedia
994 * @shared: the new value
996 * Set or unset if the pipeline for @media can be shared will multiple clients.
997 * When @shared is %TRUE, client requests for this media will share the media
1001 gst_rtsp_media_set_shared (GstRTSPMedia * media, gboolean shared)
1003 GstRTSPMediaPrivate *priv;
1005 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1009 g_mutex_lock (&priv->lock);
1010 priv->shared = shared;
1011 g_mutex_unlock (&priv->lock);
1015 * gst_rtsp_media_is_shared:
1016 * @media: a #GstRTSPMedia
1018 * Check if the pipeline for @media can be shared between multiple clients.
1020 * Returns: %TRUE if the media can be shared between clients.
1023 gst_rtsp_media_is_shared (GstRTSPMedia * media)
1025 GstRTSPMediaPrivate *priv;
1028 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1032 g_mutex_lock (&priv->lock);
1034 g_mutex_unlock (&priv->lock);
1040 * gst_rtsp_media_set_reusable:
1041 * @media: a #GstRTSPMedia
1042 * @reusable: the new value
1044 * Set or unset if the pipeline for @media can be reused after the pipeline has
1048 gst_rtsp_media_set_reusable (GstRTSPMedia * media, gboolean reusable)
1050 GstRTSPMediaPrivate *priv;
1052 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1056 g_mutex_lock (&priv->lock);
1057 priv->reusable = reusable;
1058 g_mutex_unlock (&priv->lock);
1062 * gst_rtsp_media_is_reusable:
1063 * @media: a #GstRTSPMedia
1065 * Check if the pipeline for @media can be reused after an unprepare.
1067 * Returns: %TRUE if the media can be reused
1070 gst_rtsp_media_is_reusable (GstRTSPMedia * media)
1072 GstRTSPMediaPrivate *priv;
1075 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1079 g_mutex_lock (&priv->lock);
1080 res = priv->reusable;
1081 g_mutex_unlock (&priv->lock);
1087 do_set_profiles (GstRTSPStream * stream, GstRTSPProfile * profiles)
1089 gst_rtsp_stream_set_profiles (stream, *profiles);
1093 * gst_rtsp_media_set_profiles:
1094 * @media: a #GstRTSPMedia
1095 * @profiles: the new flags
1097 * Configure the allowed lower transport for @media.
1100 gst_rtsp_media_set_profiles (GstRTSPMedia * media, GstRTSPProfile profiles)
1102 GstRTSPMediaPrivate *priv;
1104 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1108 g_mutex_lock (&priv->lock);
1109 priv->profiles = profiles;
1110 g_ptr_array_foreach (priv->streams, (GFunc) do_set_profiles, &profiles);
1111 g_mutex_unlock (&priv->lock);
1115 * gst_rtsp_media_get_profiles:
1116 * @media: a #GstRTSPMedia
1118 * Get the allowed profiles of @media.
1120 * Returns: a #GstRTSPProfile
1123 gst_rtsp_media_get_profiles (GstRTSPMedia * media)
1125 GstRTSPMediaPrivate *priv;
1128 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_RTSP_PROFILE_UNKNOWN);
1132 g_mutex_lock (&priv->lock);
1133 res = priv->profiles;
1134 g_mutex_unlock (&priv->lock);
1140 do_set_protocols (GstRTSPStream * stream, GstRTSPLowerTrans * protocols)
1142 gst_rtsp_stream_set_protocols (stream, *protocols);
1146 * gst_rtsp_media_set_protocols:
1147 * @media: a #GstRTSPMedia
1148 * @protocols: the new flags
1150 * Configure the allowed lower transport for @media.
1153 gst_rtsp_media_set_protocols (GstRTSPMedia * media, GstRTSPLowerTrans protocols)
1155 GstRTSPMediaPrivate *priv;
1157 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1161 g_mutex_lock (&priv->lock);
1162 priv->protocols = protocols;
1163 g_ptr_array_foreach (priv->streams, (GFunc) do_set_protocols, &protocols);
1164 g_mutex_unlock (&priv->lock);
1168 * gst_rtsp_media_get_protocols:
1169 * @media: a #GstRTSPMedia
1171 * Get the allowed protocols of @media.
1173 * Returns: a #GstRTSPLowerTrans
1176 gst_rtsp_media_get_protocols (GstRTSPMedia * media)
1178 GstRTSPMediaPrivate *priv;
1179 GstRTSPLowerTrans res;
1181 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media),
1182 GST_RTSP_LOWER_TRANS_UNKNOWN);
1186 g_mutex_lock (&priv->lock);
1187 res = priv->protocols;
1188 g_mutex_unlock (&priv->lock);
1194 * gst_rtsp_media_set_eos_shutdown:
1195 * @media: a #GstRTSPMedia
1196 * @eos_shutdown: the new value
1198 * Set or unset if an EOS event will be sent to the pipeline for @media before
1202 gst_rtsp_media_set_eos_shutdown (GstRTSPMedia * media, gboolean eos_shutdown)
1204 GstRTSPMediaPrivate *priv;
1206 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1210 g_mutex_lock (&priv->lock);
1211 priv->eos_shutdown = eos_shutdown;
1212 g_mutex_unlock (&priv->lock);
1216 * gst_rtsp_media_is_eos_shutdown:
1217 * @media: a #GstRTSPMedia
1219 * Check if the pipeline for @media will send an EOS down the pipeline before
1222 * Returns: %TRUE if the media will send EOS before unpreparing.
1225 gst_rtsp_media_is_eos_shutdown (GstRTSPMedia * media)
1227 GstRTSPMediaPrivate *priv;
1230 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1234 g_mutex_lock (&priv->lock);
1235 res = priv->eos_shutdown;
1236 g_mutex_unlock (&priv->lock);
1242 * gst_rtsp_media_set_buffer_size:
1243 * @media: a #GstRTSPMedia
1244 * @size: the new value
1246 * Set the kernel UDP buffer size.
1249 gst_rtsp_media_set_buffer_size (GstRTSPMedia * media, guint size)
1251 GstRTSPMediaPrivate *priv;
1254 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1256 GST_LOG_OBJECT (media, "set buffer size %u", size);
1260 g_mutex_lock (&priv->lock);
1261 priv->buffer_size = size;
1263 for (i = 0; i < priv->streams->len; i++) {
1264 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1265 gst_rtsp_stream_set_buffer_size (stream, size);
1267 g_mutex_unlock (&priv->lock);
1271 * gst_rtsp_media_get_buffer_size:
1272 * @media: a #GstRTSPMedia
1274 * Get the kernel UDP buffer size.
1276 * Returns: the kernel UDP buffer size.
1279 gst_rtsp_media_get_buffer_size (GstRTSPMedia * media)
1281 GstRTSPMediaPrivate *priv;
1284 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1288 g_mutex_lock (&priv->lock);
1289 res = priv->buffer_size;
1290 g_mutex_unlock (&priv->lock);
1296 * gst_rtsp_media_set_stop_on_disconnect:
1297 * @media: a #GstRTSPMedia
1298 * @stop_on_disconnect: the new value
1300 * Set or unset if the pipeline for @media should be stopped when a
1301 * client disconnects without sending TEARDOWN.
1304 gst_rtsp_media_set_stop_on_disconnect (GstRTSPMedia * media,
1305 gboolean stop_on_disconnect)
1307 GstRTSPMediaPrivate *priv;
1309 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1313 g_mutex_lock (&priv->lock);
1314 priv->stop_on_disconnect = stop_on_disconnect;
1315 g_mutex_unlock (&priv->lock);
1319 * gst_rtsp_media_is_stop_on_disconnect:
1320 * @media: a #GstRTSPMedia
1322 * Check if the pipeline for @media will be stopped when a client disconnects
1323 * without sending TEARDOWN.
1325 * Returns: %TRUE if the media will be stopped when a client disconnects
1326 * without sending TEARDOWN.
1329 gst_rtsp_media_is_stop_on_disconnect (GstRTSPMedia * media)
1331 GstRTSPMediaPrivate *priv;
1334 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), TRUE);
1338 g_mutex_lock (&priv->lock);
1339 res = priv->stop_on_disconnect;
1340 g_mutex_unlock (&priv->lock);
1346 * gst_rtsp_media_set_retransmission_time:
1347 * @media: a #GstRTSPMedia
1348 * @time: the new value
1350 * Set the amount of time to store retransmission packets.
1353 gst_rtsp_media_set_retransmission_time (GstRTSPMedia * media, GstClockTime time)
1355 GstRTSPMediaPrivate *priv;
1358 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1360 GST_LOG_OBJECT (media, "set retransmission time %" G_GUINT64_FORMAT, time);
1364 g_mutex_lock (&priv->lock);
1365 priv->rtx_time = time;
1366 for (i = 0; i < priv->streams->len; i++) {
1367 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1369 gst_rtsp_stream_set_retransmission_time (stream, time);
1373 g_object_set (priv->rtpbin, "do-retransmission", time > 0, NULL);
1374 g_mutex_unlock (&priv->lock);
1378 * gst_rtsp_media_get_retransmission_time:
1379 * @media: a #GstRTSPMedia
1381 * Get the amount of time to store retransmission data.
1383 * Returns: the amount of time to store retransmission data.
1386 gst_rtsp_media_get_retransmission_time (GstRTSPMedia * media)
1388 GstRTSPMediaPrivate *priv;
1391 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1395 g_mutex_lock (&priv->lock);
1396 res = priv->rtx_time;
1397 g_mutex_unlock (&priv->lock);
1403 * gst_rtsp_media_set_latency:
1404 * @media: a #GstRTSPMedia
1405 * @latency: latency in milliseconds
1407 * Configure the latency used for receiving media.
1410 gst_rtsp_media_set_latency (GstRTSPMedia * media, guint latency)
1412 GstRTSPMediaPrivate *priv;
1414 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1416 GST_LOG_OBJECT (media, "set latency %ums", latency);
1420 g_mutex_lock (&priv->lock);
1421 priv->latency = latency;
1423 g_object_set (priv->rtpbin, "latency", latency, NULL);
1424 g_mutex_unlock (&priv->lock);
1428 * gst_rtsp_media_get_latency:
1429 * @media: a #GstRTSPMedia
1431 * Get the latency that is used for receiving media.
1433 * Returns: latency in milliseconds
1436 gst_rtsp_media_get_latency (GstRTSPMedia * media)
1438 GstRTSPMediaPrivate *priv;
1441 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1445 g_mutex_lock (&priv->lock);
1446 res = priv->latency;
1447 g_mutex_unlock (&priv->lock);
1453 * gst_rtsp_media_use_time_provider:
1454 * @media: a #GstRTSPMedia
1455 * @time_provider: if a #GstNetTimeProvider should be used
1457 * Set @media to provide a #GstNetTimeProvider.
1460 gst_rtsp_media_use_time_provider (GstRTSPMedia * media, gboolean time_provider)
1462 GstRTSPMediaPrivate *priv;
1464 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1468 g_mutex_lock (&priv->lock);
1469 priv->time_provider = time_provider;
1470 g_mutex_unlock (&priv->lock);
1474 * gst_rtsp_media_is_time_provider:
1475 * @media: a #GstRTSPMedia
1477 * Check if @media can provide a #GstNetTimeProvider for its pipeline clock.
1479 * Use gst_rtsp_media_get_time_provider() to get the network clock.
1481 * Returns: %TRUE if @media can provide a #GstNetTimeProvider.
1484 gst_rtsp_media_is_time_provider (GstRTSPMedia * media)
1486 GstRTSPMediaPrivate *priv;
1489 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
1493 g_mutex_lock (&priv->lock);
1494 res = priv->time_provider;
1495 g_mutex_unlock (&priv->lock);
1501 * gst_rtsp_media_set_clock:
1502 * @media: a #GstRTSPMedia
1503 * @clock: #GstClock to be used
1505 * Configure the clock used for the media.
1508 gst_rtsp_media_set_clock (GstRTSPMedia * media, GstClock * clock)
1510 GstRTSPMediaPrivate *priv;
1512 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1513 g_return_if_fail (GST_IS_CLOCK (clock) || clock == NULL);
1515 GST_LOG_OBJECT (media, "setting clock %" GST_PTR_FORMAT, clock);
1519 g_mutex_lock (&priv->lock);
1521 gst_object_unref (priv->clock);
1522 priv->clock = clock ? gst_object_ref (clock) : NULL;
1523 if (priv->pipeline) {
1525 gst_pipeline_use_clock (GST_PIPELINE_CAST (priv->pipeline), clock);
1527 gst_pipeline_auto_clock (GST_PIPELINE_CAST (priv->pipeline));
1530 g_mutex_unlock (&priv->lock);
1534 * gst_rtsp_media_set_publish_clock_mode:
1535 * @media: a #GstRTSPMedia
1536 * @mode: the clock publish mode
1538 * Sets if and how the media clock should be published according to RFC7273.
1543 gst_rtsp_media_set_publish_clock_mode (GstRTSPMedia * media,
1544 GstRTSPPublishClockMode mode)
1546 GstRTSPMediaPrivate *priv;
1550 g_mutex_lock (&priv->lock);
1551 priv->publish_clock_mode = mode;
1553 n = priv->streams->len;
1554 for (i = 0; i < n; i++) {
1555 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1557 gst_rtsp_stream_set_publish_clock_mode (stream, mode);
1559 g_mutex_unlock (&priv->lock);
1563 * gst_rtsp_media_get_publish_clock_mode:
1564 * @media: a #GstRTSPMedia
1566 * Gets if and how the media clock should be published according to RFC7273.
1568 * Returns: The GstRTSPPublishClockMode
1572 GstRTSPPublishClockMode
1573 gst_rtsp_media_get_publish_clock_mode (GstRTSPMedia * media)
1575 GstRTSPMediaPrivate *priv;
1576 GstRTSPPublishClockMode ret;
1579 g_mutex_lock (&priv->lock);
1580 ret = priv->publish_clock_mode;
1581 g_mutex_unlock (&priv->lock);
1587 * gst_rtsp_media_set_address_pool:
1588 * @media: a #GstRTSPMedia
1589 * @pool: (transfer none): a #GstRTSPAddressPool
1591 * configure @pool to be used as the address pool of @media.
1594 gst_rtsp_media_set_address_pool (GstRTSPMedia * media,
1595 GstRTSPAddressPool * pool)
1597 GstRTSPMediaPrivate *priv;
1598 GstRTSPAddressPool *old;
1600 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1604 GST_LOG_OBJECT (media, "set address pool %p", pool);
1606 g_mutex_lock (&priv->lock);
1607 if ((old = priv->pool) != pool)
1608 priv->pool = pool ? g_object_ref (pool) : NULL;
1611 g_ptr_array_foreach (priv->streams, (GFunc) gst_rtsp_stream_set_address_pool,
1613 g_mutex_unlock (&priv->lock);
1616 g_object_unref (old);
1620 * gst_rtsp_media_get_address_pool:
1621 * @media: a #GstRTSPMedia
1623 * Get the #GstRTSPAddressPool used as the address pool of @media.
1625 * Returns: (transfer full): the #GstRTSPAddressPool of @media. g_object_unref() after
1628 GstRTSPAddressPool *
1629 gst_rtsp_media_get_address_pool (GstRTSPMedia * media)
1631 GstRTSPMediaPrivate *priv;
1632 GstRTSPAddressPool *result;
1634 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1638 g_mutex_lock (&priv->lock);
1639 if ((result = priv->pool))
1640 g_object_ref (result);
1641 g_mutex_unlock (&priv->lock);
1647 * gst_rtsp_media_set_multicast_iface:
1648 * @media: a #GstRTSPMedia
1649 * @multicast_iface: (transfer none): a multicast interface name
1651 * configure @multicast_iface to be used for @media.
1654 gst_rtsp_media_set_multicast_iface (GstRTSPMedia * media,
1655 const gchar * multicast_iface)
1657 GstRTSPMediaPrivate *priv;
1660 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1664 GST_LOG_OBJECT (media, "set multicast interface %s", multicast_iface);
1666 g_mutex_lock (&priv->lock);
1667 if ((old = priv->multicast_iface) != multicast_iface)
1668 priv->multicast_iface = multicast_iface ? g_strdup (multicast_iface) : NULL;
1671 g_ptr_array_foreach (priv->streams,
1672 (GFunc) gst_rtsp_stream_set_multicast_iface, (gchar *) multicast_iface);
1673 g_mutex_unlock (&priv->lock);
1680 * gst_rtsp_media_get_multicast_iface:
1681 * @media: a #GstRTSPMedia
1683 * Get the multicast interface used for @media.
1685 * Returns: (transfer full): the multicast interface for @media. g_free() after
1689 gst_rtsp_media_get_multicast_iface (GstRTSPMedia * media)
1691 GstRTSPMediaPrivate *priv;
1694 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1698 g_mutex_lock (&priv->lock);
1699 if ((result = priv->multicast_iface))
1700 result = g_strdup (result);
1701 g_mutex_unlock (&priv->lock);
1707 _find_payload_types (GstRTSPMedia * media)
1710 GQueue queue = G_QUEUE_INIT;
1712 n = media->priv->streams->len;
1713 for (i = 0; i < n; i++) {
1714 GstRTSPStream *stream = g_ptr_array_index (media->priv->streams, i);
1715 guint pt = gst_rtsp_stream_get_pt (stream);
1717 g_queue_push_tail (&queue, GUINT_TO_POINTER (pt));
1724 _next_available_pt (GList * payloads)
1728 for (i = 96; i <= 127; i++) {
1729 GList *iter = g_list_find (payloads, GINT_TO_POINTER (i));
1731 return GPOINTER_TO_UINT (i);
1738 * gst_rtsp_media_collect_streams:
1739 * @media: a #GstRTSPMedia
1741 * Find all payloader elements, they should be named pay\%d in the
1742 * element of @media, and create #GstRTSPStreams for them.
1744 * Collect all dynamic elements, named dynpay\%d, and add them to
1745 * the list of dynamic elements.
1747 * Find all depayloader elements, they should be named depay\%d in the
1748 * element of @media, and create #GstRTSPStreams for them.
1751 gst_rtsp_media_collect_streams (GstRTSPMedia * media)
1753 GstRTSPMediaPrivate *priv;
1754 GstElement *element, *elem;
1758 gboolean more_elem_remaining = TRUE;
1759 GstRTSPTransportMode mode = 0;
1761 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
1764 element = priv->element;
1767 for (i = 0; more_elem_remaining; i++) {
1770 more_elem_remaining = FALSE;
1772 name = g_strdup_printf ("pay%d", i);
1773 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1775 GST_INFO ("found stream %d with payloader %p", i, elem);
1777 /* take the pad of the payloader */
1778 pad = gst_element_get_static_pad (elem, "src");
1780 /* find the real payload element in case elem is a GstBin */
1781 pay = find_payload_element (elem);
1783 /* create the stream */
1785 GST_WARNING ("could not find real payloader, using bin");
1786 gst_rtsp_media_create_stream (media, elem, pad);
1788 gst_rtsp_media_create_stream (media, pay, pad);
1789 gst_object_unref (pay);
1792 gst_object_unref (pad);
1793 gst_object_unref (elem);
1796 more_elem_remaining = TRUE;
1797 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1801 name = g_strdup_printf ("dynpay%d", i);
1802 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1803 /* a stream that will dynamically create pads to provide RTP packets */
1804 GST_INFO ("found dynamic element %d, %p", i, elem);
1806 g_mutex_lock (&priv->lock);
1807 priv->dynamic = g_list_prepend (priv->dynamic, elem);
1808 g_mutex_unlock (&priv->lock);
1810 priv->nb_dynamic_elements++;
1813 more_elem_remaining = TRUE;
1814 mode |= GST_RTSP_TRANSPORT_MODE_PLAY;
1818 name = g_strdup_printf ("depay%d", i);
1819 if ((elem = gst_bin_get_by_name (GST_BIN (element), name))) {
1820 GST_INFO ("found stream %d with depayloader %p", i, elem);
1822 /* take the pad of the payloader */
1823 pad = gst_element_get_static_pad (elem, "sink");
1824 /* create the stream */
1825 gst_rtsp_media_create_stream (media, elem, pad);
1826 gst_object_unref (pad);
1827 gst_object_unref (elem);
1830 more_elem_remaining = TRUE;
1831 mode |= GST_RTSP_TRANSPORT_MODE_RECORD;
1837 if (priv->transport_mode != mode)
1838 GST_WARNING ("found different mode than expected (0x%02x != 0x%02d)",
1839 priv->transport_mode, mode);
1844 * gst_rtsp_media_create_stream:
1845 * @media: a #GstRTSPMedia
1846 * @payloader: a #GstElement
1849 * Create a new stream in @media that provides RTP data on @pad.
1850 * @pad should be a pad of an element inside @media->element.
1852 * Returns: (transfer none): a new #GstRTSPStream that remains valid for as long
1856 gst_rtsp_media_create_stream (GstRTSPMedia * media, GstElement * payloader,
1859 GstRTSPMediaPrivate *priv;
1860 GstRTSPStream *stream;
1865 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1866 g_return_val_if_fail (GST_IS_ELEMENT (payloader), NULL);
1867 g_return_val_if_fail (GST_IS_PAD (pad), NULL);
1871 g_mutex_lock (&priv->lock);
1872 idx = priv->streams->len;
1874 GST_DEBUG ("media %p: creating stream with index %d", media, idx);
1876 if (GST_PAD_IS_SRC (pad))
1877 name = g_strdup_printf ("src_%u", idx);
1879 name = g_strdup_printf ("sink_%u", idx);
1881 ghostpad = gst_ghost_pad_new (name, pad);
1882 gst_pad_set_active (ghostpad, TRUE);
1883 gst_element_add_pad (priv->element, ghostpad);
1886 stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
1888 gst_rtsp_stream_set_address_pool (stream, priv->pool);
1889 gst_rtsp_stream_set_multicast_iface (stream, priv->multicast_iface);
1890 gst_rtsp_stream_set_profiles (stream, priv->profiles);
1891 gst_rtsp_stream_set_protocols (stream, priv->protocols);
1892 gst_rtsp_stream_set_retransmission_time (stream, priv->rtx_time);
1893 gst_rtsp_stream_set_buffer_size (stream, priv->buffer_size);
1894 gst_rtsp_stream_set_publish_clock_mode (stream, priv->publish_clock_mode);
1896 g_ptr_array_add (priv->streams, stream);
1898 if (GST_PAD_IS_SRC (pad)) {
1902 g_list_free (priv->payloads);
1903 priv->payloads = _find_payload_types (media);
1905 n = priv->streams->len;
1906 for (i = 0; i < n; i++) {
1907 GstRTSPStream *stream = g_ptr_array_index (priv->streams, i);
1908 guint rtx_pt = _next_available_pt (priv->payloads);
1911 GST_WARNING ("Ran out of space of dynamic payload types");
1915 gst_rtsp_stream_set_retransmission_pt (stream, rtx_pt);
1918 g_list_append (priv->payloads, GUINT_TO_POINTER (rtx_pt));
1921 g_mutex_unlock (&priv->lock);
1923 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STREAM], 0, stream,
1930 gst_rtsp_media_remove_stream (GstRTSPMedia * media, GstRTSPStream * stream)
1932 GstRTSPMediaPrivate *priv;
1937 g_mutex_lock (&priv->lock);
1938 /* remove the ghostpad */
1939 srcpad = gst_rtsp_stream_get_srcpad (stream);
1940 gst_element_remove_pad (priv->element, srcpad);
1941 gst_object_unref (srcpad);
1942 /* now remove the stream */
1943 g_object_ref (stream);
1944 g_ptr_array_remove (priv->streams, stream);
1945 g_mutex_unlock (&priv->lock);
1947 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_REMOVED_STREAM], 0,
1950 g_object_unref (stream);
1954 * gst_rtsp_media_n_streams:
1955 * @media: a #GstRTSPMedia
1957 * Get the number of streams in this media.
1959 * Returns: The number of streams.
1962 gst_rtsp_media_n_streams (GstRTSPMedia * media)
1964 GstRTSPMediaPrivate *priv;
1967 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), 0);
1971 g_mutex_lock (&priv->lock);
1972 res = priv->streams->len;
1973 g_mutex_unlock (&priv->lock);
1979 * gst_rtsp_media_get_stream:
1980 * @media: a #GstRTSPMedia
1981 * @idx: the stream index
1983 * Retrieve the stream with index @idx from @media.
1985 * Returns: (nullable) (transfer none): the #GstRTSPStream at index
1986 * @idx or %NULL when a stream with that index did not exist.
1989 gst_rtsp_media_get_stream (GstRTSPMedia * media, guint idx)
1991 GstRTSPMediaPrivate *priv;
1994 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
1998 g_mutex_lock (&priv->lock);
1999 if (idx < priv->streams->len)
2000 res = g_ptr_array_index (priv->streams, idx);
2003 g_mutex_unlock (&priv->lock);
2009 * gst_rtsp_media_find_stream:
2010 * @media: a #GstRTSPMedia
2011 * @control: the control of the stream
2013 * Find a stream in @media with @control as the control uri.
2015 * Returns: (nullable) (transfer none): the #GstRTSPStream with
2016 * control uri @control or %NULL when a stream with that control did
2020 gst_rtsp_media_find_stream (GstRTSPMedia * media, const gchar * control)
2022 GstRTSPMediaPrivate *priv;
2026 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2027 g_return_val_if_fail (control != NULL, NULL);
2033 g_mutex_lock (&priv->lock);
2034 for (i = 0; i < priv->streams->len; i++) {
2035 GstRTSPStream *test;
2037 test = g_ptr_array_index (priv->streams, i);
2038 if (gst_rtsp_stream_has_control (test, control)) {
2043 g_mutex_unlock (&priv->lock);
2048 /* called with state-lock */
2050 default_convert_range (GstRTSPMedia * media, GstRTSPTimeRange * range,
2051 GstRTSPRangeUnit unit)
2053 return gst_rtsp_range_convert_units (range, unit);
2057 * gst_rtsp_media_get_range_string:
2058 * @media: a #GstRTSPMedia
2059 * @play: for the PLAY request
2060 * @unit: the unit to use for the string
2062 * Get the current range as a string. @media must be prepared with
2063 * gst_rtsp_media_prepare ().
2065 * Returns: (transfer full): The range as a string, g_free() after usage.
2068 gst_rtsp_media_get_range_string (GstRTSPMedia * media, gboolean play,
2069 GstRTSPRangeUnit unit)
2071 GstRTSPMediaClass *klass;
2072 GstRTSPMediaPrivate *priv;
2074 GstRTSPTimeRange range;
2076 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2077 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
2078 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2082 g_rec_mutex_lock (&priv->state_lock);
2083 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
2084 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
2087 g_mutex_lock (&priv->lock);
2089 /* Update the range value with current position/duration */
2090 collect_media_stats (media);
2093 range = priv->range;
2095 if (!play && priv->n_active > 0) {
2096 range.min.type = GST_RTSP_TIME_NOW;
2097 range.min.seconds = -1;
2099 g_mutex_unlock (&priv->lock);
2100 g_rec_mutex_unlock (&priv->state_lock);
2102 if (!klass->convert_range (media, &range, unit))
2103 goto conversion_failed;
2105 result = gst_rtsp_range_to_string (&range);
2112 GST_WARNING ("media %p was not prepared", media);
2113 g_rec_mutex_unlock (&priv->state_lock);
2118 GST_WARNING ("range conversion to unit %d failed", unit);
2124 stream_update_blocked (GstRTSPStream * stream, GstRTSPMedia * media)
2126 gst_rtsp_stream_set_blocked (stream, media->priv->blocked);
2130 media_streams_set_blocked (GstRTSPMedia * media, gboolean blocked)
2132 GstRTSPMediaPrivate *priv = media->priv;
2134 GST_DEBUG ("media %p set blocked %d", media, blocked);
2135 priv->blocked = blocked;
2136 g_ptr_array_foreach (priv->streams, (GFunc) stream_update_blocked, media);
2140 stream_unblock (GstRTSPStream * stream, GstRTSPMedia * media)
2142 gst_rtsp_stream_unblock_linked (stream);
2146 media_unblock_linked (GstRTSPMedia * media)
2148 GstRTSPMediaPrivate *priv = media->priv;
2150 GST_DEBUG ("media %p unblocking linked streams", media);
2151 g_ptr_array_foreach (priv->streams, (GFunc) stream_unblock, media);
2155 gst_rtsp_media_set_status (GstRTSPMedia * media, GstRTSPMediaStatus status)
2157 GstRTSPMediaPrivate *priv = media->priv;
2159 g_mutex_lock (&priv->lock);
2160 priv->status = status;
2161 GST_DEBUG ("setting new status to %d", status);
2162 g_cond_broadcast (&priv->cond);
2163 g_mutex_unlock (&priv->lock);
2167 * gst_rtsp_media_get_status:
2168 * @media: a #GstRTSPMedia
2170 * Get the status of @media. When @media is busy preparing, this function waits
2171 * until @media is prepared or in error.
2173 * Returns: the status of @media.
2176 gst_rtsp_media_get_status (GstRTSPMedia * media)
2178 GstRTSPMediaPrivate *priv = media->priv;
2179 GstRTSPMediaStatus result;
2182 g_mutex_lock (&priv->lock);
2183 end_time = g_get_monotonic_time () + 20 * G_TIME_SPAN_SECOND;
2184 /* while we are preparing, wait */
2185 while (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING) {
2186 GST_DEBUG ("waiting for status change");
2187 if (!g_cond_wait_until (&priv->cond, &priv->lock, end_time)) {
2188 GST_DEBUG ("timeout, assuming error status");
2189 priv->status = GST_RTSP_MEDIA_STATUS_ERROR;
2192 /* could be success or error */
2193 result = priv->status;
2194 GST_DEBUG ("got status %d", result);
2195 g_mutex_unlock (&priv->lock);
2201 * gst_rtsp_media_seek_full:
2202 * @media: a #GstRTSPMedia
2203 * @range: (transfer none): a #GstRTSPTimeRange
2204 * @flags: The minimal set of #GstSeekFlags to use
2206 * Seek the pipeline of @media to @range. @media must be prepared with
2207 * gst_rtsp_media_prepare().
2209 * Returns: %TRUE on success.
2212 gst_rtsp_media_seek_full (GstRTSPMedia * media, GstRTSPTimeRange * range,
2215 GstRTSPMediaClass *klass;
2216 GstRTSPMediaPrivate *priv;
2218 GstClockTime start, stop;
2219 GstSeekType start_type, stop_type;
2220 gint64 current_position;
2222 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2224 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
2225 g_return_val_if_fail (range != NULL, FALSE);
2226 g_return_val_if_fail (klass->convert_range != NULL, FALSE);
2230 g_rec_mutex_lock (&priv->state_lock);
2231 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
2234 /* check if the media pipeline is complete in order to perform a
2235 * seek operation on it */
2236 if (!check_complete (media))
2239 /* Update the seekable state of the pipeline in case it changed */
2240 check_seekable (media);
2242 if (priv->seekable == 0) {
2243 GST_FIXME_OBJECT (media, "Handle going back to 0 for none live"
2244 " not seekable streams.");
2247 } else if (priv->seekable < 0) {
2251 start_type = stop_type = GST_SEEK_TYPE_NONE;
2253 if (!klass->convert_range (media, range, GST_RTSP_RANGE_NPT))
2255 gst_rtsp_range_get_times (range, &start, &stop);
2257 GST_INFO ("got %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2258 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2259 GST_INFO ("current %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2260 GST_TIME_ARGS (priv->range_start), GST_TIME_ARGS (priv->range_stop));
2262 current_position = -1;
2263 if (klass->query_position)
2264 klass->query_position (media, ¤t_position);
2265 GST_INFO ("current media position %" GST_TIME_FORMAT,
2266 GST_TIME_ARGS (current_position));
2268 if (start != GST_CLOCK_TIME_NONE)
2269 start_type = GST_SEEK_TYPE_SET;
2271 if (priv->range_stop == stop)
2272 stop = GST_CLOCK_TIME_NONE;
2273 else if (stop != GST_CLOCK_TIME_NONE)
2274 stop_type = GST_SEEK_TYPE_SET;
2276 if (start != GST_CLOCK_TIME_NONE || stop != GST_CLOCK_TIME_NONE) {
2277 gboolean had_flags = flags != 0;
2279 GST_INFO ("seeking to %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT,
2280 GST_TIME_ARGS (start), GST_TIME_ARGS (stop));
2282 /* depends on the current playing state of the pipeline. We might need to
2283 * queue this until we get EOS. */
2285 flags |= GST_SEEK_FLAG_FLUSH;
2287 flags = GST_SEEK_FLAG_FLUSH;
2290 /* if range start was not supplied we must continue from current position.
2291 * but since we're doing a flushing seek, let us query the current position
2292 * so we end up at exactly the same position after the seek. */
2293 if (range->min.type == GST_RTSP_TIME_END) { /* Yepp, that's right! */
2294 if (current_position == -1) {
2295 GST_WARNING ("current position unknown");
2297 GST_DEBUG ("doing accurate seek to %" GST_TIME_FORMAT,
2298 GST_TIME_ARGS (current_position));
2299 start = current_position;
2300 start_type = GST_SEEK_TYPE_SET;
2302 flags |= GST_SEEK_FLAG_ACCURATE;
2305 /* only set keyframe flag when modifying start */
2306 if (start_type != GST_SEEK_TYPE_NONE)
2308 flags |= GST_SEEK_FLAG_KEY_UNIT;
2311 if (start == current_position && stop_type == GST_SEEK_TYPE_NONE) {
2312 GST_DEBUG ("not seeking because no position change");
2315 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
2317 media_streams_set_blocked (media, TRUE);
2319 /* FIXME, we only do forwards playback, no trick modes yet */
2320 res = gst_element_seek (priv->pipeline, 1.0, GST_FORMAT_TIME,
2321 flags, start_type, start, stop_type, stop);
2323 /* and block for the seek to complete */
2324 GST_INFO ("done seeking %d", res);
2328 g_rec_mutex_unlock (&priv->state_lock);
2330 /* wait until pipeline is prerolled again, this will also collect stats */
2331 if (!wait_preroll (media))
2332 goto preroll_failed;
2334 g_rec_mutex_lock (&priv->state_lock);
2335 GST_INFO ("prerolled again");
2338 GST_INFO ("no seek needed");
2341 g_rec_mutex_unlock (&priv->state_lock);
2348 g_rec_mutex_unlock (&priv->state_lock);
2349 GST_INFO ("media %p is not prepared", media);
2354 g_rec_mutex_unlock (&priv->state_lock);
2355 GST_INFO ("pipeline is not complete");
2360 g_rec_mutex_unlock (&priv->state_lock);
2361 GST_INFO ("pipeline is not seekable");
2366 g_rec_mutex_unlock (&priv->state_lock);
2367 GST_WARNING ("conversion to npt not supported");
2372 g_rec_mutex_unlock (&priv->state_lock);
2373 GST_INFO ("seeking failed");
2374 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2379 GST_WARNING ("failed to preroll after seek");
2386 * gst_rtsp_media_seek:
2387 * @media: a #GstRTSPMedia
2388 * @range: (transfer none): a #GstRTSPTimeRange
2390 * Seek the pipeline of @media to @range. @media must be prepared with
2391 * gst_rtsp_media_prepare().
2393 * Returns: %TRUE on success.
2396 gst_rtsp_media_seek (GstRTSPMedia * media, GstRTSPTimeRange * range)
2398 return gst_rtsp_media_seek_full (media, range, 0);
2403 stream_collect_blocking (GstRTSPStream * stream, gboolean * blocked)
2405 *blocked &= gst_rtsp_stream_is_blocking (stream);
2409 media_streams_blocking (GstRTSPMedia * media)
2411 gboolean blocking = TRUE;
2413 g_ptr_array_foreach (media->priv->streams, (GFunc) stream_collect_blocking,
2419 static GstStateChangeReturn
2420 set_state (GstRTSPMedia * media, GstState state)
2422 GstRTSPMediaPrivate *priv = media->priv;
2423 GstStateChangeReturn ret;
2425 GST_INFO ("set state to %s for media %p", gst_element_state_get_name (state),
2427 ret = gst_element_set_state (priv->pipeline, state);
2432 static GstStateChangeReturn
2433 set_target_state (GstRTSPMedia * media, GstState state, gboolean do_state)
2435 GstRTSPMediaPrivate *priv = media->priv;
2436 GstStateChangeReturn ret;
2438 GST_INFO ("set target state to %s for media %p",
2439 gst_element_state_get_name (state), media);
2440 priv->target_state = state;
2442 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_TARGET_STATE], 0,
2443 priv->target_state, NULL);
2446 ret = set_state (media, state);
2448 ret = GST_STATE_CHANGE_SUCCESS;
2453 /* called with state-lock */
2455 default_handle_message (GstRTSPMedia * media, GstMessage * message)
2457 GstRTSPMediaPrivate *priv = media->priv;
2458 GstMessageType type;
2460 type = GST_MESSAGE_TYPE (message);
2463 case GST_MESSAGE_STATE_CHANGED:
2465 GstState old, new, pending;
2467 if (GST_MESSAGE_SRC (message) != GST_OBJECT (priv->pipeline))
2470 gst_message_parse_state_changed (message, &old, &new, &pending);
2472 GST_DEBUG ("%p: went from %s to %s (pending %s)", media,
2473 gst_element_state_get_name (old), gst_element_state_get_name (new),
2474 gst_element_state_get_name (pending));
2475 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD)
2476 && old == GST_STATE_READY && new == GST_STATE_PAUSED) {
2477 GST_INFO ("%p: went to PAUSED, prepared now", media);
2478 collect_media_stats (media);
2480 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2481 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2486 case GST_MESSAGE_BUFFERING:
2490 gst_message_parse_buffering (message, &percent);
2492 /* no state management needed for live pipelines */
2496 if (percent == 100) {
2497 /* a 100% message means buffering is done */
2498 priv->buffering = FALSE;
2499 /* if the desired state is playing, go back */
2500 if (priv->target_state == GST_STATE_PLAYING) {
2501 GST_INFO ("Buffering done, setting pipeline to PLAYING");
2502 set_state (media, GST_STATE_PLAYING);
2504 GST_INFO ("Buffering done");
2507 /* buffering busy */
2508 if (priv->buffering == FALSE) {
2509 if (priv->target_state == GST_STATE_PLAYING) {
2510 /* we were not buffering but PLAYING, PAUSE the pipeline. */
2511 GST_INFO ("Buffering, setting pipeline to PAUSED ...");
2512 set_state (media, GST_STATE_PAUSED);
2514 GST_INFO ("Buffering ...");
2517 priv->buffering = TRUE;
2521 case GST_MESSAGE_LATENCY:
2523 gst_bin_recalculate_latency (GST_BIN_CAST (priv->pipeline));
2526 case GST_MESSAGE_ERROR:
2531 gst_message_parse_error (message, &gerror, &debug);
2532 GST_WARNING ("%p: got error %s (%s)", media, gerror->message, debug);
2533 g_error_free (gerror);
2536 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2539 case GST_MESSAGE_WARNING:
2544 gst_message_parse_warning (message, &gerror, &debug);
2545 GST_WARNING ("%p: got warning %s (%s)", media, gerror->message, debug);
2546 g_error_free (gerror);
2550 case GST_MESSAGE_ELEMENT:
2552 const GstStructure *s;
2554 s = gst_message_get_structure (message);
2555 if (gst_structure_has_name (s, "GstRTSPStreamBlocking")) {
2556 GST_DEBUG ("media received blocking message");
2557 if (priv->blocked && media_streams_blocking (media) &&
2558 priv->no_more_pads_pending == 0) {
2559 GST_DEBUG_OBJECT (GST_MESSAGE_SRC (message), "media is blocking");
2560 collect_media_stats (media);
2562 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2563 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2568 case GST_MESSAGE_STREAM_STATUS:
2570 case GST_MESSAGE_ASYNC_DONE:
2571 if (priv->complete) {
2572 /* receive the final ASYNC_DONE, that is posted by the media pipeline
2573 * after all the transport parts have been successfully added to
2574 * the media streams. */
2575 GST_DEBUG_OBJECT (media, "got async-done");
2576 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
2577 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
2580 case GST_MESSAGE_EOS:
2581 GST_INFO ("%p: got EOS", media);
2583 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARING) {
2584 GST_DEBUG ("shutting down after EOS");
2585 finish_unprepare (media);
2589 GST_INFO ("%p: got message type %d (%s)", media, type,
2590 gst_message_type_get_name (type));
2597 bus_message (GstBus * bus, GstMessage * message, GstRTSPMedia * media)
2599 GstRTSPMediaPrivate *priv = media->priv;
2600 GstRTSPMediaClass *klass;
2603 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2605 g_rec_mutex_lock (&priv->state_lock);
2606 if (klass->handle_message)
2607 ret = klass->handle_message (media, message);
2610 g_rec_mutex_unlock (&priv->state_lock);
2616 watch_destroyed (GstRTSPMedia * media)
2618 GST_DEBUG_OBJECT (media, "source destroyed");
2619 g_object_unref (media);
2623 find_payload_element (GstElement * payloader)
2625 GstElement *pay = NULL;
2627 if (GST_IS_BIN (payloader)) {
2629 GValue item = { 0 };
2631 iter = gst_bin_iterate_recurse (GST_BIN (payloader));
2632 while (gst_iterator_next (iter, &item) == GST_ITERATOR_OK) {
2633 GstElement *element = (GstElement *) g_value_get_object (&item);
2634 GstElementClass *eclass = GST_ELEMENT_GET_CLASS (element);
2638 gst_element_class_get_metadata (eclass, GST_ELEMENT_METADATA_KLASS);
2642 if (strstr (klass, "Payloader") && strstr (klass, "RTP")) {
2643 pay = gst_object_ref (element);
2644 g_value_unset (&item);
2647 g_value_unset (&item);
2649 gst_iterator_free (iter);
2651 pay = g_object_ref (payloader);
2657 /* called from streaming threads */
2659 pad_added_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2661 GstRTSPMediaPrivate *priv = media->priv;
2662 GstRTSPStream *stream;
2665 /* find the real payload element */
2666 pay = find_payload_element (element);
2667 stream = gst_rtsp_media_create_stream (media, pay, pad);
2668 gst_object_unref (pay);
2670 GST_INFO ("pad added %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2672 g_rec_mutex_lock (&priv->state_lock);
2673 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2676 g_object_set_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream", stream);
2678 /* join the element in the PAUSED state because this callback is
2679 * called from the streaming thread and it is PAUSED */
2680 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2681 priv->rtpbin, GST_STATE_PAUSED)) {
2682 GST_WARNING ("failed to join bin element");
2686 gst_rtsp_stream_set_blocked (stream, TRUE);
2688 g_rec_mutex_unlock (&priv->state_lock);
2695 gst_rtsp_media_remove_stream (media, stream);
2696 g_rec_mutex_unlock (&priv->state_lock);
2697 GST_INFO ("ignore pad because we are not preparing");
2703 pad_removed_cb (GstElement * element, GstPad * pad, GstRTSPMedia * media)
2705 GstRTSPMediaPrivate *priv = media->priv;
2706 GstRTSPStream *stream;
2708 stream = g_object_get_data (G_OBJECT (pad), "gst-rtsp-dynpad-stream");
2712 GST_INFO ("pad removed %s:%s, stream %p", GST_DEBUG_PAD_NAME (pad), stream);
2714 g_rec_mutex_lock (&priv->state_lock);
2715 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
2716 g_rec_mutex_unlock (&priv->state_lock);
2718 gst_rtsp_media_remove_stream (media, stream);
2722 no_more_pads_cb (GstElement * element, GstRTSPMedia * media)
2724 GstRTSPMediaPrivate *priv = media->priv;
2726 GST_INFO_OBJECT (element, "no more pads");
2727 g_mutex_lock (&priv->lock);
2728 priv->no_more_pads_pending--;
2729 g_mutex_unlock (&priv->lock);
2732 typedef struct _DynPaySignalHandlers DynPaySignalHandlers;
2734 struct _DynPaySignalHandlers
2736 gulong pad_added_handler;
2737 gulong pad_removed_handler;
2738 gulong no_more_pads_handler;
2742 start_preroll (GstRTSPMedia * media)
2744 GstRTSPMediaPrivate *priv = media->priv;
2745 GstStateChangeReturn ret;
2747 GST_INFO ("setting pipeline to PAUSED for media %p", media);
2749 /* start blocked since it is possible that there are no sink elements yet */
2750 media_streams_set_blocked (media, TRUE);
2751 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
2754 case GST_STATE_CHANGE_SUCCESS:
2755 GST_INFO ("SUCCESS state change for media %p", media);
2757 case GST_STATE_CHANGE_ASYNC:
2758 GST_INFO ("ASYNC state change for media %p", media);
2760 case GST_STATE_CHANGE_NO_PREROLL:
2761 /* we need to go to PLAYING */
2762 GST_INFO ("NO_PREROLL state change: live media %p", media);
2763 /* FIXME we disable seeking for live streams for now. We should perform a
2764 * seeking query in preroll instead */
2765 priv->seekable = -1;
2766 priv->is_live = TRUE;
2768 ret = set_state (media, GST_STATE_PLAYING);
2769 if (ret == GST_STATE_CHANGE_FAILURE)
2772 case GST_STATE_CHANGE_FAILURE:
2780 GST_WARNING ("failed to preroll pipeline");
2786 wait_preroll (GstRTSPMedia * media)
2788 GstRTSPMediaStatus status;
2790 GST_DEBUG ("wait to preroll pipeline");
2792 /* wait until pipeline is prerolled */
2793 status = gst_rtsp_media_get_status (media);
2794 if (status == GST_RTSP_MEDIA_STATUS_ERROR)
2795 goto preroll_failed;
2801 GST_WARNING ("failed to preroll pipeline");
2807 request_aux_sender (GstElement * rtpbin, guint sessid, GstRTSPMedia * media)
2809 GstRTSPMediaPrivate *priv = media->priv;
2810 GstRTSPStream *stream = NULL;
2813 g_mutex_lock (&priv->lock);
2814 for (i = 0; i < priv->streams->len; i++) {
2815 stream = g_ptr_array_index (priv->streams, i);
2817 if (sessid == gst_rtsp_stream_get_index (stream))
2820 g_mutex_unlock (&priv->lock);
2822 return gst_rtsp_stream_request_aux_sender (stream, sessid);
2826 start_prepare (GstRTSPMedia * media)
2828 GstRTSPMediaPrivate *priv = media->priv;
2832 g_rec_mutex_lock (&priv->state_lock);
2833 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARING)
2834 goto no_longer_preparing;
2836 /* link streams we already have, other streams might appear when we have
2837 * dynamic elements */
2838 for (i = 0; i < priv->streams->len; i++) {
2839 GstRTSPStream *stream;
2841 stream = g_ptr_array_index (priv->streams, i);
2843 if (priv->rtx_time > 0) {
2844 /* enable retransmission by setting rtprtxsend as the "aux" element of rtpbin */
2845 g_signal_connect (priv->rtpbin, "request-aux-sender",
2846 (GCallback) request_aux_sender, media);
2849 if (!gst_rtsp_stream_join_bin (stream, GST_BIN (priv->pipeline),
2850 priv->rtpbin, GST_STATE_NULL)) {
2851 goto join_bin_failed;
2856 g_object_set (priv->rtpbin, "do-retransmission", priv->rtx_time > 0, NULL);
2858 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
2859 GstElement *elem = walk->data;
2860 DynPaySignalHandlers *handlers = g_slice_new (DynPaySignalHandlers);
2862 GST_INFO ("adding callbacks for dynamic element %p", elem);
2864 handlers->pad_added_handler = g_signal_connect (elem, "pad-added",
2865 (GCallback) pad_added_cb, media);
2866 handlers->pad_removed_handler = g_signal_connect (elem, "pad-removed",
2867 (GCallback) pad_removed_cb, media);
2868 handlers->no_more_pads_handler = g_signal_connect (elem, "no-more-pads",
2869 (GCallback) no_more_pads_cb, media);
2871 g_object_set_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers", handlers);
2874 if (!start_preroll (media))
2875 goto preroll_failed;
2877 g_rec_mutex_unlock (&priv->state_lock);
2881 no_longer_preparing:
2883 GST_INFO ("media is no longer preparing");
2884 g_rec_mutex_unlock (&priv->state_lock);
2889 GST_WARNING ("failed to join bin element");
2890 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2891 g_rec_mutex_unlock (&priv->state_lock);
2896 GST_WARNING ("failed to preroll pipeline");
2897 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
2898 g_rec_mutex_unlock (&priv->state_lock);
2904 default_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2906 GstRTSPMediaPrivate *priv;
2907 GstRTSPMediaClass *klass;
2909 GMainContext *context;
2914 klass = GST_RTSP_MEDIA_GET_CLASS (media);
2916 if (!klass->create_rtpbin)
2917 goto no_create_rtpbin;
2919 priv->rtpbin = klass->create_rtpbin (media);
2920 if (priv->rtpbin != NULL) {
2921 gboolean success = TRUE;
2923 g_object_set (priv->rtpbin, "latency", priv->latency, NULL);
2925 if (klass->setup_rtpbin)
2926 success = klass->setup_rtpbin (media, priv->rtpbin);
2928 if (success == FALSE) {
2929 gst_object_unref (priv->rtpbin);
2930 priv->rtpbin = NULL;
2933 if (priv->rtpbin == NULL)
2936 priv->thread = thread;
2937 context = (thread != NULL) ? (thread->context) : NULL;
2939 bus = gst_pipeline_get_bus (GST_PIPELINE_CAST (priv->pipeline));
2941 /* add the pipeline bus to our custom mainloop */
2942 priv->source = gst_bus_create_watch (bus);
2943 gst_object_unref (bus);
2945 g_source_set_callback (priv->source, (GSourceFunc) bus_message,
2946 g_object_ref (media), (GDestroyNotify) watch_destroyed);
2948 priv->id = g_source_attach (priv->source, context);
2950 /* add stuff to the bin */
2951 gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);
2953 /* do remainder in context */
2954 source = g_idle_source_new ();
2955 g_source_set_callback (source, (GSourceFunc) start_prepare,
2956 g_object_ref (media), (GDestroyNotify) g_object_unref);
2957 g_source_attach (source, context);
2958 g_source_unref (source);
2965 GST_ERROR ("no create_rtpbin function");
2966 g_critical ("no create_rtpbin vmethod function set");
2971 GST_WARNING ("no rtpbin element");
2972 g_warning ("failed to create element 'rtpbin', check your installation");
2978 * gst_rtsp_media_prepare:
2979 * @media: a #GstRTSPMedia
2980 * @thread: (transfer full) (allow-none): a #GstRTSPThread to run the
2981 * bus handler or %NULL
2983 * Prepare @media for streaming. This function will create the objects
2984 * to manage the streaming. A pipeline must have been set on @media with
2985 * gst_rtsp_media_take_pipeline().
2987 * It will preroll the pipeline and collect vital information about the streams
2988 * such as the duration.
2990 * Returns: %TRUE on success.
2993 gst_rtsp_media_prepare (GstRTSPMedia * media, GstRTSPThread * thread)
2995 GstRTSPMediaPrivate *priv;
2996 GstRTSPMediaClass *klass;
2998 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3002 g_rec_mutex_lock (&priv->state_lock);
3003 priv->prepare_count++;
3005 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED ||
3006 priv->status == GST_RTSP_MEDIA_STATUS_SUSPENDED)
3009 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARING)
3012 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARED)
3013 goto not_unprepared;
3015 if (!priv->reusable && priv->reused)
3018 GST_INFO ("preparing media %p", media);
3020 /* reset some variables */
3021 priv->is_live = FALSE;
3022 priv->seekable = -1;
3023 priv->buffering = FALSE;
3024 priv->no_more_pads_pending = priv->nb_dynamic_elements;
3026 /* we're preparing now */
3027 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3029 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3030 if (klass->prepare) {
3031 if (!klass->prepare (media, thread))
3032 goto prepare_failed;
3036 g_rec_mutex_unlock (&priv->state_lock);
3038 /* now wait for all pads to be prerolled, FIXME, we should somehow be
3039 * able to do this async so that we don't block the server thread. */
3040 if (!wait_preroll (media))
3041 goto preroll_failed;
3043 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_PREPARED], 0, NULL);
3045 GST_INFO ("object %p is prerolled", media);
3052 /* we are not going to use the giving thread, so stop it. */
3054 gst_rtsp_thread_stop (thread);
3059 GST_LOG ("media %p was prepared", media);
3060 /* we are not going to use the giving thread, so stop it. */
3062 gst_rtsp_thread_stop (thread);
3063 g_rec_mutex_unlock (&priv->state_lock);
3069 /* we are not going to use the giving thread, so stop it. */
3071 gst_rtsp_thread_stop (thread);
3072 GST_WARNING ("media %p was not unprepared", media);
3073 priv->prepare_count--;
3074 g_rec_mutex_unlock (&priv->state_lock);
3079 /* we are not going to use the giving thread, so stop it. */
3081 gst_rtsp_thread_stop (thread);
3082 priv->prepare_count--;
3083 g_rec_mutex_unlock (&priv->state_lock);
3084 GST_WARNING ("can not reuse media %p", media);
3089 /* we are not going to use the giving thread, so stop it. */
3091 gst_rtsp_thread_stop (thread);
3092 priv->prepare_count--;
3093 g_rec_mutex_unlock (&priv->state_lock);
3094 GST_ERROR ("failed to prepare media");
3099 GST_WARNING ("failed to preroll pipeline");
3100 gst_rtsp_media_unprepare (media);
3105 /* must be called with state-lock */
3107 finish_unprepare (GstRTSPMedia * media)
3109 GstRTSPMediaPrivate *priv = media->priv;
3113 GST_DEBUG ("shutting down");
3115 /* release the lock on shutdown, otherwise pad_added_cb might try to
3116 * acquire the lock and then we deadlock */
3117 g_rec_mutex_unlock (&priv->state_lock);
3118 set_state (media, GST_STATE_NULL);
3119 g_rec_mutex_lock (&priv->state_lock);
3121 media_streams_set_blocked (media, FALSE);
3123 if (priv->status != GST_RTSP_MEDIA_STATUS_UNPREPARING)
3126 for (i = 0; i < priv->streams->len; i++) {
3127 GstRTSPStream *stream;
3129 GST_INFO ("Removing elements of stream %d from pipeline", i);
3131 stream = g_ptr_array_index (priv->streams, i);
3133 gst_rtsp_stream_leave_bin (stream, GST_BIN (priv->pipeline), priv->rtpbin);
3136 /* remove the pad signal handlers */
3137 for (walk = priv->dynamic; walk; walk = g_list_next (walk)) {
3138 GstElement *elem = walk->data;
3139 DynPaySignalHandlers *handlers;
3142 g_object_steal_data (G_OBJECT (elem), "gst-rtsp-dynpay-handlers");
3143 g_assert (handlers != NULL);
3145 g_signal_handler_disconnect (G_OBJECT (elem), handlers->pad_added_handler);
3146 g_signal_handler_disconnect (G_OBJECT (elem),
3147 handlers->pad_removed_handler);
3148 g_signal_handler_disconnect (G_OBJECT (elem),
3149 handlers->no_more_pads_handler);
3151 g_slice_free (DynPaySignalHandlers, handlers);
3154 gst_bin_remove (GST_BIN (priv->pipeline), priv->rtpbin);
3155 priv->rtpbin = NULL;
3158 gst_object_unref (priv->nettime);
3159 priv->nettime = NULL;
3161 priv->reused = TRUE;
3162 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARED);
3164 /* when the media is not reusable, this will effectively unref the media and
3166 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_UNPREPARED], 0, NULL);
3168 /* the source has the last ref to the media */
3170 GST_DEBUG ("destroy source");
3171 g_source_destroy (priv->source);
3172 g_source_unref (priv->source);
3175 GST_DEBUG ("stop thread");
3176 gst_rtsp_thread_stop (priv->thread);
3180 /* called with state-lock */
3182 default_unprepare (GstRTSPMedia * media)
3184 GstRTSPMediaPrivate *priv = media->priv;
3186 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3188 if (priv->eos_shutdown) {
3189 GST_DEBUG ("sending EOS for shutdown");
3190 /* ref so that we don't disappear */
3191 gst_element_send_event (priv->pipeline, gst_event_new_eos ());
3192 /* we need to go to playing again for the EOS to propagate, normally in this
3193 * state, nothing is receiving data from us anymore so this is ok. */
3194 set_state (media, GST_STATE_PLAYING);
3196 finish_unprepare (media);
3202 * gst_rtsp_media_unprepare:
3203 * @media: a #GstRTSPMedia
3205 * Unprepare @media. After this call, the media should be prepared again before
3206 * it can be used again. If the media is set to be non-reusable, a new instance
3209 * Returns: %TRUE on success.
3212 gst_rtsp_media_unprepare (GstRTSPMedia * media)
3214 GstRTSPMediaPrivate *priv;
3217 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3221 g_rec_mutex_lock (&priv->state_lock);
3222 if (priv->status == GST_RTSP_MEDIA_STATUS_UNPREPARED)
3223 goto was_unprepared;
3225 priv->prepare_count--;
3226 if (priv->prepare_count > 0)
3229 GST_INFO ("unprepare media %p", media);
3230 set_target_state (media, GST_STATE_NULL, FALSE);
3233 if (priv->status == GST_RTSP_MEDIA_STATUS_PREPARED) {
3234 GstRTSPMediaClass *klass;
3236 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3237 if (klass->unprepare)
3238 success = klass->unprepare (media);
3240 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_UNPREPARING);
3241 finish_unprepare (media);
3243 g_rec_mutex_unlock (&priv->state_lock);
3249 g_rec_mutex_unlock (&priv->state_lock);
3250 GST_INFO ("media %p was already unprepared", media);
3255 GST_INFO ("media %p still prepared %d times", media, priv->prepare_count);
3256 g_rec_mutex_unlock (&priv->state_lock);
3261 /* should be called with state-lock */
3263 get_clock_unlocked (GstRTSPMedia * media)
3265 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED) {
3266 GST_DEBUG_OBJECT (media, "media was not prepared");
3269 return gst_pipeline_get_clock (GST_PIPELINE_CAST (media->priv->pipeline));
3273 * gst_rtsp_media_get_clock:
3274 * @media: a #GstRTSPMedia
3276 * Get the clock that is used by the pipeline in @media.
3278 * @media must be prepared before this method returns a valid clock object.
3280 * Returns: (transfer full): the #GstClock used by @media. unref after usage.
3283 gst_rtsp_media_get_clock (GstRTSPMedia * media)
3286 GstRTSPMediaPrivate *priv;
3288 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3292 g_rec_mutex_lock (&priv->state_lock);
3293 clock = get_clock_unlocked (media);
3294 g_rec_mutex_unlock (&priv->state_lock);
3300 * gst_rtsp_media_get_base_time:
3301 * @media: a #GstRTSPMedia
3303 * Get the base_time that is used by the pipeline in @media.
3305 * @media must be prepared before this method returns a valid base_time.
3307 * Returns: the base_time used by @media.
3310 gst_rtsp_media_get_base_time (GstRTSPMedia * media)
3312 GstClockTime result;
3313 GstRTSPMediaPrivate *priv;
3315 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), GST_CLOCK_TIME_NONE);
3319 g_rec_mutex_lock (&priv->state_lock);
3320 if (media->priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3323 result = gst_element_get_base_time (media->priv->pipeline);
3324 g_rec_mutex_unlock (&priv->state_lock);
3331 g_rec_mutex_unlock (&priv->state_lock);
3332 GST_DEBUG_OBJECT (media, "media was not prepared");
3333 return GST_CLOCK_TIME_NONE;
3338 * gst_rtsp_media_get_time_provider:
3339 * @media: a #GstRTSPMedia
3340 * @address: (allow-none): an address or %NULL
3341 * @port: a port or 0
3343 * Get the #GstNetTimeProvider for the clock used by @media. The time provider
3344 * will listen on @address and @port for client time requests.
3346 * Returns: (transfer full): the #GstNetTimeProvider of @media.
3348 GstNetTimeProvider *
3349 gst_rtsp_media_get_time_provider (GstRTSPMedia * media, const gchar * address,
3352 GstRTSPMediaPrivate *priv;
3353 GstNetTimeProvider *provider = NULL;
3355 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), NULL);
3359 g_rec_mutex_lock (&priv->state_lock);
3360 if (priv->time_provider) {
3361 if ((provider = priv->nettime) == NULL) {
3364 if (priv->time_provider && (clock = get_clock_unlocked (media))) {
3365 provider = gst_net_time_provider_new (clock, address, port);
3366 gst_object_unref (clock);
3368 priv->nettime = provider;
3372 g_rec_mutex_unlock (&priv->state_lock);
3375 gst_object_ref (provider);
3381 default_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp, GstSDPInfo * info)
3383 return gst_rtsp_sdp_from_media (sdp, info, media);
3387 * gst_rtsp_media_setup_sdp:
3388 * @media: a #GstRTSPMedia
3389 * @sdp: (transfer none): a #GstSDPMessage
3390 * @info: (transfer none): a #GstSDPInfo
3392 * Add @media specific info to @sdp. @info is used to configure the connection
3393 * information in the SDP.
3395 * Returns: TRUE on success.
3398 gst_rtsp_media_setup_sdp (GstRTSPMedia * media, GstSDPMessage * sdp,
3401 GstRTSPMediaPrivate *priv;
3402 GstRTSPMediaClass *klass;
3405 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3406 g_return_val_if_fail (sdp != NULL, FALSE);
3407 g_return_val_if_fail (info != NULL, FALSE);
3411 g_rec_mutex_lock (&priv->state_lock);
3413 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3415 if (!klass->setup_sdp)
3418 res = klass->setup_sdp (media, sdp, info);
3420 g_rec_mutex_unlock (&priv->state_lock);
3427 g_rec_mutex_unlock (&priv->state_lock);
3428 GST_ERROR ("no setup_sdp function");
3429 g_critical ("no setup_sdp vmethod function set");
3435 default_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3437 GstRTSPMediaPrivate *priv = media->priv;
3440 medias_len = gst_sdp_message_medias_len (sdp);
3441 if (medias_len != priv->streams->len) {
3442 GST_ERROR ("%p: Media has more or less streams than SDP (%d /= %d)", media,
3443 priv->streams->len, medias_len);
3447 for (i = 0; i < medias_len; i++) {
3449 const GstSDPMedia *sdp_media = gst_sdp_message_get_media (sdp, i);
3450 GstRTSPStream *stream;
3451 gint j, formats_len;
3452 const gchar *control;
3453 GstRTSPProfile profile, profiles;
3455 stream = g_ptr_array_index (priv->streams, i);
3457 /* TODO: Should we do something with the other SDP information? */
3460 proto = gst_sdp_media_get_proto (sdp_media);
3461 if (proto == NULL) {
3462 GST_ERROR ("%p: SDP media %d has no proto", media, i);
3466 if (g_str_equal (proto, "RTP/AVP")) {
3467 profile = GST_RTSP_PROFILE_AVP;
3468 } else if (g_str_equal (proto, "RTP/SAVP")) {
3469 profile = GST_RTSP_PROFILE_SAVP;
3470 } else if (g_str_equal (proto, "RTP/AVPF")) {
3471 profile = GST_RTSP_PROFILE_AVPF;
3472 } else if (g_str_equal (proto, "RTP/SAVPF")) {
3473 profile = GST_RTSP_PROFILE_SAVPF;
3475 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3479 profiles = gst_rtsp_stream_get_profiles (stream);
3480 if ((profiles & profile) == 0) {
3481 GST_ERROR ("%p: unsupported profile '%s' for stream %d", media, proto, i);
3485 formats_len = gst_sdp_media_formats_len (sdp_media);
3486 for (j = 0; j < formats_len; j++) {
3491 pt = atoi (gst_sdp_media_get_format (sdp_media, j));
3493 GST_DEBUG (" looking at %d pt: %d", j, pt);
3496 caps = gst_sdp_media_get_caps_from_media (sdp_media, pt);
3498 GST_WARNING (" skipping pt %d without caps", pt);
3502 /* do some tweaks */
3503 GST_DEBUG ("mapping sdp session level attributes to caps");
3504 gst_sdp_message_attributes_to_caps (sdp, caps);
3505 GST_DEBUG ("mapping sdp media level attributes to caps");
3506 gst_sdp_media_attributes_to_caps (sdp_media, caps);
3508 s = gst_caps_get_structure (caps, 0);
3509 gst_structure_set_name (s, "application/x-rtp");
3511 gst_rtsp_stream_set_pt_map (stream, pt, caps);
3512 gst_caps_unref (caps);
3515 control = gst_sdp_media_get_attribute_val (sdp_media, "control");
3517 gst_rtsp_stream_set_control (stream, control);
3525 * gst_rtsp_media_handle_sdp:
3526 * @media: a #GstRTSPMedia
3527 * @sdp: (transfer none): a #GstSDPMessage
3529 * Configure an SDP on @media for receiving streams
3531 * Returns: TRUE on success.
3534 gst_rtsp_media_handle_sdp (GstRTSPMedia * media, GstSDPMessage * sdp)
3536 GstRTSPMediaPrivate *priv;
3537 GstRTSPMediaClass *klass;
3540 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3541 g_return_val_if_fail (sdp != NULL, FALSE);
3545 g_rec_mutex_lock (&priv->state_lock);
3547 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3549 if (!klass->handle_sdp)
3552 res = klass->handle_sdp (media, sdp);
3554 g_rec_mutex_unlock (&priv->state_lock);
3561 g_rec_mutex_unlock (&priv->state_lock);
3562 GST_ERROR ("no handle_sdp function");
3563 g_critical ("no handle_sdp vmethod function set");
3569 do_set_seqnum (GstRTSPStream * stream)
3572 seq_num = gst_rtsp_stream_get_current_seqnum (stream);
3573 gst_rtsp_stream_set_seqnum_offset (stream, seq_num + 1);
3576 /* call with state_lock */
3578 default_suspend (GstRTSPMedia * media)
3580 GstRTSPMediaPrivate *priv = media->priv;
3581 GstStateChangeReturn ret;
3583 switch (priv->suspend_mode) {
3584 case GST_RTSP_SUSPEND_MODE_NONE:
3585 GST_DEBUG ("media %p no suspend", media);
3587 case GST_RTSP_SUSPEND_MODE_PAUSE:
3588 GST_DEBUG ("media %p suspend to PAUSED", media);
3589 ret = set_target_state (media, GST_STATE_PAUSED, TRUE);
3590 if (ret == GST_STATE_CHANGE_FAILURE)
3593 case GST_RTSP_SUSPEND_MODE_RESET:
3594 GST_DEBUG ("media %p suspend to NULL", media);
3595 ret = set_target_state (media, GST_STATE_NULL, TRUE);
3596 if (ret == GST_STATE_CHANGE_FAILURE)
3598 /* Because payloader needs to set the sequence number as
3599 * monotonic, we need to preserve the sequence number
3600 * after pause. (otherwise going from pause to play, which
3601 * is actually from NULL to PLAY will create a new sequence
3603 g_ptr_array_foreach (priv->streams, (GFunc) do_set_seqnum, NULL);
3614 GST_WARNING ("failed changing pipeline's state for media %p", media);
3620 * gst_rtsp_media_suspend:
3621 * @media: a #GstRTSPMedia
3623 * Suspend @media. The state of the pipeline managed by @media is set to
3624 * GST_STATE_NULL but all streams are kept. @media can be prepared again
3625 * with gst_rtsp_media_unsuspend()
3627 * @media must be prepared with gst_rtsp_media_prepare();
3629 * Returns: %TRUE on success.
3632 gst_rtsp_media_suspend (GstRTSPMedia * media)
3634 GstRTSPMediaPrivate *priv = media->priv;
3635 GstRTSPMediaClass *klass;
3637 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3639 GST_FIXME ("suspend for dynamic pipelines needs fixing");
3641 g_rec_mutex_lock (&priv->state_lock);
3642 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED)
3645 /* don't attempt to suspend when something is busy */
3646 if (priv->n_active > 0)
3649 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3650 if (klass->suspend) {
3651 if (!klass->suspend (media))
3652 goto suspend_failed;
3655 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_SUSPENDED);
3657 g_rec_mutex_unlock (&priv->state_lock);
3664 g_rec_mutex_unlock (&priv->state_lock);
3665 GST_WARNING ("media %p was not prepared", media);
3670 g_rec_mutex_unlock (&priv->state_lock);
3671 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3672 GST_WARNING ("failed to suspend media %p", media);
3677 /* call with state_lock */
3679 default_unsuspend (GstRTSPMedia * media)
3681 GstRTSPMediaPrivate *priv = media->priv;
3682 gboolean preroll_ok;
3684 switch (priv->suspend_mode) {
3685 case GST_RTSP_SUSPEND_MODE_NONE:
3686 if ((priv->transport_mode & GST_RTSP_TRANSPORT_MODE_RECORD))
3688 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3689 /* at this point the media pipeline has been updated and contain all
3690 * specific transport parts: all active streams contain at least one sink
3691 * element and it's safe to unblock any blocked streams that are active */
3692 media_unblock_linked (media);
3693 g_rec_mutex_unlock (&priv->state_lock);
3694 if (gst_rtsp_media_get_status (media) == GST_RTSP_MEDIA_STATUS_ERROR) {
3695 g_rec_mutex_lock (&priv->state_lock);
3696 goto preroll_failed;
3698 g_rec_mutex_lock (&priv->state_lock);
3700 case GST_RTSP_SUSPEND_MODE_PAUSE:
3701 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARED);
3703 case GST_RTSP_SUSPEND_MODE_RESET:
3705 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_PREPARING);
3706 /* at this point the media pipeline has been updated and contain all
3707 * specific transport parts: all active streams contain at least one sink
3708 * element and it's safe to unblock any blocked streams that are active */
3709 media_unblock_linked (media);
3710 if (!start_preroll (media))
3713 g_rec_mutex_unlock (&priv->state_lock);
3714 preroll_ok = wait_preroll (media);
3715 g_rec_mutex_lock (&priv->state_lock);
3718 goto preroll_failed;
3729 GST_WARNING ("failed to preroll pipeline");
3734 GST_WARNING ("failed to preroll pipeline");
3740 * gst_rtsp_media_unsuspend:
3741 * @media: a #GstRTSPMedia
3743 * Unsuspend @media if it was in a suspended state. This method does nothing
3744 * when the media was not in the suspended state.
3746 * Returns: %TRUE on success.
3749 gst_rtsp_media_unsuspend (GstRTSPMedia * media)
3751 GstRTSPMediaPrivate *priv = media->priv;
3752 GstRTSPMediaClass *klass;
3754 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3756 g_rec_mutex_lock (&priv->state_lock);
3757 if (priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3760 klass = GST_RTSP_MEDIA_GET_CLASS (media);
3761 if (klass->unsuspend) {
3762 if (!klass->unsuspend (media))
3763 goto unsuspend_failed;
3767 g_rec_mutex_unlock (&priv->state_lock);
3774 g_rec_mutex_unlock (&priv->state_lock);
3775 GST_WARNING ("failed to unsuspend media %p", media);
3776 gst_rtsp_media_set_status (media, GST_RTSP_MEDIA_STATUS_ERROR);
3781 /* must be called with state-lock */
3783 media_set_pipeline_state_locked (GstRTSPMedia * media, GstState state)
3785 GstRTSPMediaPrivate *priv = media->priv;
3787 if (state == GST_STATE_NULL) {
3788 gst_rtsp_media_unprepare (media);
3790 GST_INFO ("state %s media %p", gst_element_state_get_name (state), media);
3791 set_target_state (media, state, FALSE);
3792 /* when we are buffering, don't update the state yet, this will be done
3793 * when buffering finishes */
3794 if (priv->buffering) {
3795 GST_INFO ("Buffering busy, delay state change");
3797 if (state == GST_STATE_PLAYING)
3798 /* make sure pads are not blocking anymore when going to PLAYING */
3799 media_unblock_linked (media);
3801 set_state (media, state);
3803 /* and suspend after pause */
3804 if (state == GST_STATE_PAUSED)
3805 gst_rtsp_media_suspend (media);
3811 * gst_rtsp_media_set_pipeline_state:
3812 * @media: a #GstRTSPMedia
3813 * @state: the target state of the pipeline
3815 * Set the state of the pipeline managed by @media to @state
3818 gst_rtsp_media_set_pipeline_state (GstRTSPMedia * media, GstState state)
3820 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
3822 g_rec_mutex_lock (&media->priv->state_lock);
3823 media_set_pipeline_state_locked (media, state);
3824 g_rec_mutex_unlock (&media->priv->state_lock);
3828 * gst_rtsp_media_set_state:
3829 * @media: a #GstRTSPMedia
3830 * @state: the target state of the media
3831 * @transports: (transfer none) (element-type GstRtspServer.RTSPStreamTransport):
3832 * a #GPtrArray of #GstRTSPStreamTransport pointers
3834 * Set the state of @media to @state and for the transports in @transports.
3836 * @media must be prepared with gst_rtsp_media_prepare();
3838 * Returns: %TRUE on success.
3841 gst_rtsp_media_set_state (GstRTSPMedia * media, GstState state,
3842 GPtrArray * transports)
3844 GstRTSPMediaPrivate *priv;
3846 gboolean activate, deactivate, do_state;
3849 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
3850 g_return_val_if_fail (transports != NULL, FALSE);
3854 g_rec_mutex_lock (&priv->state_lock);
3855 if (priv->status == GST_RTSP_MEDIA_STATUS_ERROR)
3857 if (priv->status != GST_RTSP_MEDIA_STATUS_PREPARED &&
3858 priv->status != GST_RTSP_MEDIA_STATUS_SUSPENDED)
3861 /* NULL and READY are the same */
3862 if (state == GST_STATE_READY)
3863 state = GST_STATE_NULL;
3865 activate = deactivate = FALSE;
3867 GST_INFO ("going to state %s media %p, target state %s",
3868 gst_element_state_get_name (state), media,
3869 gst_element_state_get_name (priv->target_state));
3872 case GST_STATE_NULL:
3873 /* we're going from PLAYING or PAUSED to READY or NULL, deactivate */
3874 if (priv->target_state >= GST_STATE_PAUSED)
3877 case GST_STATE_PAUSED:
3878 /* we're going from PLAYING to PAUSED, READY or NULL, deactivate */
3879 if (priv->target_state == GST_STATE_PLAYING)
3882 case GST_STATE_PLAYING:
3883 /* we're going to PLAYING, activate */
3889 old_active = priv->n_active;
3891 GST_DEBUG ("%d transports, activate %d, deactivate %d", transports->len,
3892 activate, deactivate);
3893 for (i = 0; i < transports->len; i++) {
3894 GstRTSPStreamTransport *trans;
3896 /* we need a non-NULL entry in the array */
3897 trans = g_ptr_array_index (transports, i);
3902 if (gst_rtsp_stream_transport_set_active (trans, TRUE))
3904 } else if (deactivate) {
3905 if (gst_rtsp_stream_transport_set_active (trans, FALSE))
3910 /* we just activated the first media, do the playing state change */
3911 if (old_active == 0 && activate)
3913 /* if we have no more active media, do the downward state changes */
3914 else if (priv->n_active == 0)
3919 GST_INFO ("state %d active %d media %p do_state %d", state, priv->n_active,
3922 if (priv->target_state != state) {
3924 media_set_pipeline_state_locked (media, state);
3925 g_signal_emit (media, gst_rtsp_media_signals[SIGNAL_NEW_STATE], 0, state,
3930 /* remember where we are */
3931 if (state != GST_STATE_NULL && (state == GST_STATE_PAUSED ||
3932 old_active != priv->n_active))
3933 collect_media_stats (media);
3935 g_rec_mutex_unlock (&priv->state_lock);
3942 GST_WARNING ("media %p was not prepared", media);
3943 g_rec_mutex_unlock (&priv->state_lock);
3948 GST_WARNING ("media %p in error status while changing to state %d",
3950 if (state == GST_STATE_NULL) {
3951 for (i = 0; i < transports->len; i++) {
3952 GstRTSPStreamTransport *trans;
3954 /* we need a non-NULL entry in the array */
3955 trans = g_ptr_array_index (transports, i);
3959 gst_rtsp_stream_transport_set_active (trans, FALSE);
3963 g_rec_mutex_unlock (&priv->state_lock);
3969 * gst_rtsp_media_set_transport_mode:
3970 * @media: a #GstRTSPMedia
3971 * @mode: the new value
3973 * Sets if the media pipeline can work in PLAY or RECORD mode
3976 gst_rtsp_media_set_transport_mode (GstRTSPMedia * media,
3977 GstRTSPTransportMode mode)
3979 GstRTSPMediaPrivate *priv;
3981 g_return_if_fail (GST_IS_RTSP_MEDIA (media));
3985 g_mutex_lock (&priv->lock);
3986 priv->transport_mode = mode;
3987 g_mutex_unlock (&priv->lock);
3991 * gst_rtsp_media_get_transport_mode:
3992 * @media: a #GstRTSPMedia
3994 * Check if the pipeline for @media can be used for PLAY or RECORD methods.
3996 * Returns: The transport mode.
3998 GstRTSPTransportMode
3999 gst_rtsp_media_get_transport_mode (GstRTSPMedia * media)
4001 GstRTSPMediaPrivate *priv;
4002 GstRTSPTransportMode res;
4004 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4008 g_mutex_lock (&priv->lock);
4009 res = priv->transport_mode;
4010 g_mutex_unlock (&priv->lock);
4016 * gst_rtsp_media_get_seekable:
4017 * @media: a #GstRTSPMedia
4019 * Check if the pipeline for @media seek and up to what point in time,
4022 * Returns: -1 if the stream is not seekable, 0 if seekable only to the beginning
4023 * and > 0 to indicate the longest duration between any two random access points.
4024 * %G_MAXINT64 means any value is possible.
4027 gst_rtsp_media_seekable (GstRTSPMedia * media)
4029 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4031 /* Currently we are not able to seek on live streams,
4032 * and no stream is seekable only to the beginning */
4033 return media->priv->seekable;
4037 * gst_rtsp_media_complete_pipeline:
4038 * @media: a #GstRTSPMedia
4039 * @transports: a list of #GstRTSPTransport
4041 * Add a receiver and sender parts to the pipeline based on the transport from
4044 * Returns: %TRUE if the media pipeline has been sucessfully updated.
4047 gst_rtsp_media_complete_pipeline (GstRTSPMedia * media, GPtrArray * transports)
4049 GstRTSPMediaPrivate *priv;
4052 g_return_val_if_fail (GST_IS_RTSP_MEDIA (media), FALSE);
4053 g_return_val_if_fail (transports, FALSE);
4055 GST_DEBUG_OBJECT (media, "complete pipeline");
4059 g_mutex_lock (&priv->lock);
4060 for (i = 0; i < priv->streams->len; i++) {
4061 GstRTSPStreamTransport *transport;
4062 GstRTSPStream *stream;
4063 const GstRTSPTransport *rtsp_transport;
4065 transport = g_ptr_array_index (transports, i);
4069 stream = gst_rtsp_stream_transport_get_stream (transport);
4073 rtsp_transport = gst_rtsp_stream_transport_get_transport (transport);
4075 if (!gst_rtsp_stream_complete_stream (stream, rtsp_transport)) {
4076 g_mutex_unlock (&priv->lock);
4081 priv->complete = TRUE;
4082 g_mutex_unlock (&priv->lock);