2 * Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
3 * Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
18 * Boston, MA 02111-1307, USA.
22 * SECTION:gstbasertpdepayload
23 * @short_description: Base class for RTP depayloader
27 * Provides a base class for RTP depayloaders
32 #include "gstbasertpdepayload.h"
34 #ifdef GST_DISABLE_DEPRECATED
35 #define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock))
36 #define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock))
37 #define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock))
38 #define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock))
40 /* otherwise it's already been defined in the header (FIXME 0.11)*/
43 GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
44 #define GST_CAT_DEFAULT (basertpdepayload_debug)
46 #define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
47 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
49 struct _GstBaseRTPDepayloadPrivate
51 GstClockTime npt_start;
52 GstClockTime npt_stop;
57 GstClockTime timestamp;
58 GstClockTime duration;
65 /* Filter signals and args */
72 #define DEFAULT_QUEUE_DELAY 0
81 static void gst_base_rtp_depayload_finalize (GObject * object);
82 static void gst_base_rtp_depayload_set_property (GObject * object,
83 guint prop_id, const GValue * value, GParamSpec * pspec);
84 static void gst_base_rtp_depayload_get_property (GObject * object,
85 guint prop_id, GValue * value, GParamSpec * pspec);
87 static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
88 static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
90 static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
93 static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
94 element, GstStateChange transition);
96 static void gst_base_rtp_depayload_set_gst_timestamp
97 (GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
98 static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload *
99 filter, GstEvent * event);
100 static gboolean gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload *
101 filter, GstEvent * event);
103 GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
107 gst_base_rtp_depayload_base_init (gpointer klass)
109 /*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
113 gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
115 GObjectClass *gobject_class;
116 GstElementClass *gstelement_class;
118 gobject_class = G_OBJECT_CLASS (klass);
119 gstelement_class = (GstElementClass *) klass;
120 parent_class = g_type_class_peek_parent (klass);
122 g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
124 gobject_class->finalize = gst_base_rtp_depayload_finalize;
125 gobject_class->set_property = gst_base_rtp_depayload_set_property;
126 gobject_class->get_property = gst_base_rtp_depayload_get_property;
129 * GstBaseRTPDepayload::queue-delay
131 * Control the amount of packets to buffer.
133 * Deprecated: Use a jitterbuffer or RTP session manager to delay packet
134 * playback. This property has no effect anymore since 0.10.15.
136 #ifndef GST_REMOVE_DEPRECATED
137 g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
138 g_param_spec_uint ("queue-delay", "Queue Delay",
139 "Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
140 DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
143 gstelement_class->change_state = gst_base_rtp_depayload_change_state;
145 klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
146 klass->packet_lost = gst_base_rtp_depayload_packet_lost;
147 klass->handle_event = gst_base_rtp_depayload_handle_event;
149 GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
150 "Base class for RTP Depayloaders");
154 gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
155 GstBaseRTPDepayloadClass * klass)
157 GstPadTemplate *pad_template;
158 GstBaseRTPDepayloadPrivate *priv;
160 priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
163 GST_DEBUG_OBJECT (filter, "init");
166 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
167 g_return_if_fail (pad_template != NULL);
168 filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
169 gst_pad_set_setcaps_function (filter->sinkpad,
170 gst_base_rtp_depayload_setcaps);
171 gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
172 gst_pad_set_event_function (filter->sinkpad,
173 gst_base_rtp_depayload_handle_sink_event);
174 gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
177 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
178 g_return_if_fail (pad_template != NULL);
179 filter->srcpad = gst_pad_new_from_template (pad_template, "src");
180 gst_pad_use_fixed_caps (filter->srcpad);
181 gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
183 filter->queue = g_queue_new ();
184 filter->queue_delay = DEFAULT_QUEUE_DELAY;
186 gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
190 gst_base_rtp_depayload_finalize (GObject * object)
192 GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
194 g_queue_free (filter->queue);
196 G_OBJECT_CLASS (parent_class)->finalize (object);
200 gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
202 GstBaseRTPDepayload *filter;
203 GstBaseRTPDepayloadClass *bclass;
204 GstBaseRTPDepayloadPrivate *priv;
206 GstStructure *caps_struct;
209 filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
212 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
214 GST_DEBUG_OBJECT (filter, "Set caps");
216 caps_struct = gst_caps_get_structure (caps, 0);
218 /* get other values for newsegment */
219 value = gst_structure_get_value (caps_struct, "npt-start");
220 if (value && G_VALUE_HOLDS_UINT64 (value))
221 priv->npt_start = g_value_get_uint64 (value);
224 GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
226 value = gst_structure_get_value (caps_struct, "npt-stop");
227 if (value && G_VALUE_HOLDS_UINT64 (value))
228 priv->npt_stop = g_value_get_uint64 (value);
232 GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
234 value = gst_structure_get_value (caps_struct, "play-speed");
235 if (value && G_VALUE_HOLDS_DOUBLE (value))
236 priv->play_speed = g_value_get_double (value);
238 priv->play_speed = 1.0;
240 value = gst_structure_get_value (caps_struct, "play-scale");
241 if (value && G_VALUE_HOLDS_DOUBLE (value))
242 priv->play_scale = g_value_get_double (value);
244 priv->play_scale = 1.0;
246 if (bclass->set_caps) {
247 res = bclass->set_caps (filter, caps);
249 GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
256 priv->negotiated = res;
258 gst_object_unref (filter);
264 gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
266 GstBaseRTPDepayload *filter;
267 GstBaseRTPDepayloadPrivate *priv;
268 GstBaseRTPDepayloadClass *bclass;
269 GstFlowReturn ret = GST_FLOW_OK;
271 GstClockTime timestamp;
277 filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
280 /* we must have a setcaps first */
281 if (G_UNLIKELY (!priv->negotiated))
284 /* we must validate, it's possible that this element is plugged right after a
285 * network receiver and we don't want to operate on invalid data */
286 if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
290 priv->discont = GST_BUFFER_IS_DISCONT (in);
292 timestamp = GST_BUFFER_TIMESTAMP (in);
293 /* convert to running_time and save the timestamp, this is the timestamp
294 * we put on outgoing buffers. */
295 timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
297 priv->timestamp = timestamp;
298 priv->duration = GST_BUFFER_DURATION (in);
300 seqnum = gst_rtp_buffer_get_seq (in);
301 rtptime = gst_rtp_buffer_get_timestamp (in);
304 GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
305 GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
306 GST_TIME_ARGS (timestamp));
308 /* Check seqnum. This is a very simple check that makes sure that the seqnums
309 * are striclty increasing, dropping anything that is out of the ordinary. We
310 * can only do this when the next_seqnum is known. */
311 if (G_LIKELY (priv->next_seqnum != -1)) {
312 gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
314 /* if we have no gap, all is fine */
315 if (G_UNLIKELY (gap != 0)) {
316 GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
317 priv->next_seqnum, gap);
319 /* seqnum > next_seqnum, we are missing some packets, this is always a
321 GST_LOG_OBJECT (filter, "%d missing packets", gap);
324 /* seqnum < next_seqnum, we have seen this packet before or the sender
325 * could be restarted. If the packet is not too old, we throw it away as
326 * a duplicate, otherwise we mark discont and continue. 100 misordered
327 * packets is a good threshold. See also RFC 4737. */
331 GST_LOG_OBJECT (filter,
332 "%d > 100, packet too old, sender likely restarted", gap);
337 priv->next_seqnum = (seqnum + 1) & 0xffff;
339 if (G_UNLIKELY (discont && !priv->discont)) {
340 GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
341 /* we detected a seqnum discont but the buffer was not flagged with a discont,
342 * set the discont flag so that the subclass can throw away old data. */
343 priv->discont = TRUE;
344 in = gst_buffer_make_metadata_writable (in);
345 GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
348 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
350 if (G_UNLIKELY (bclass->process == NULL))
353 /* let's send it out to processing */
354 out_buf = bclass->process (filter, in);
356 /* we pass rtptime as backward compatibility, in reality, the incomming
357 * buffer timestamp is always applied to the outgoing packet. */
358 ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
360 gst_buffer_unref (in);
367 /* this is not fatal but should be filtered earlier */
368 if (GST_BUFFER_CAPS (in) == NULL) {
369 GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
370 ("No RTP format was negotiated."),
371 ("Input buffers need to have RTP caps set on them. This is usually "
372 "achieved by setting the 'caps' property of the upstream source "
373 "element (often udpsrc or appsrc), or by putting a capsfilter "
374 "element before the depayloader and setting the 'caps' property "
375 "on that. Also see http://cgit.freedesktop.org/gstreamer/"
376 "gst-plugins-good/tree/gst/rtp/README"));
378 GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
379 ("No RTP format was negotiated."),
380 ("RTP caps on input buffer were rejected, most likely because they "
381 "were incomplete or contained wrong values. Check the debug log "
382 "for more information."));
384 gst_buffer_unref (in);
385 return GST_FLOW_NOT_NEGOTIATED;
389 /* this is not fatal but should be filtered earlier */
390 GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
391 ("Received invalid RTP payload, dropping"));
392 gst_buffer_unref (in);
397 GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
398 gst_buffer_unref (in);
403 /* this is not fatal but should be filtered earlier */
404 GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
405 ("The subclass does not have a process method"));
406 gst_buffer_unref (in);
407 return GST_FLOW_ERROR;
412 gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter,
416 gboolean forward = TRUE;
418 switch (GST_EVENT_TYPE (event)) {
419 case GST_EVENT_FLUSH_STOP:
420 gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
421 filter->need_newsegment = TRUE;
422 filter->priv->next_seqnum = -1;
424 case GST_EVENT_NEWSEGMENT:
429 gint64 start, stop, position;
431 gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
434 gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
435 start, stop, position);
437 /* don't pass the event downstream, we generate our own segment including
438 * the NTP time and other things we receive in caps */
442 case GST_EVENT_CUSTOM_DOWNSTREAM:
444 GstBaseRTPDepayloadClass *bclass;
446 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
448 if (gst_event_has_name (event, "GstRTPPacketLost")) {
449 /* we get this event from the jitterbuffer when it considers a packet as
450 * being lost. We send it to our packet_lost vmethod. The default
451 * implementation will make time progress by pushing out a NEWSEGMENT
452 * update event. Subclasses can override and to one of the following:
453 * - Adjust timestamp/duration to something more accurate before
454 * calling the parent (default) packet_lost method.
455 * - do some more advanced error concealing on the already received
456 * (fragmented) packets.
457 * - ignore the packet lost.
459 if (bclass->packet_lost)
460 res = bclass->packet_lost (filter, event);
470 res = gst_pad_push_event (filter->srcpad, event);
472 gst_event_unref (event);
478 gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
480 gboolean res = FALSE;
481 GstBaseRTPDepayload *filter;
482 GstBaseRTPDepayloadClass *bclass;
484 filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
485 if (G_UNLIKELY (filter == NULL)) {
486 gst_event_unref (event);
490 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
491 if (bclass->handle_event)
492 res = bclass->handle_event (filter, event);
494 gst_event_unref (event);
496 gst_object_unref (filter);
501 create_segment_event (GstBaseRTPDepayload * filter, gboolean update,
502 GstClockTime position)
506 GstBaseRTPDepayloadPrivate *priv;
510 if (priv->npt_stop != -1)
511 stop = priv->npt_stop - priv->npt_start;
515 event = gst_event_new_new_segment_full (update, priv->play_speed,
516 priv->play_scale, GST_FORMAT_TIME, position, stop,
517 position + priv->npt_start);
524 GstBaseRTPDepayload *depayload;
525 GstBaseRTPDepayloadClass *bclass;
531 static GstBufferListItem
532 set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
534 GstBaseRTPDepayload *depayload = data->depayload;
536 *buffer = gst_buffer_make_metadata_writable (*buffer);
537 gst_buffer_set_caps (*buffer, data->caps);
539 /* set the timestamp if we must and can */
540 if (data->bclass->set_gst_timestamp && data->do_ts)
541 data->bclass->set_gst_timestamp (depayload, data->rtptime, *buffer);
543 if (G_UNLIKELY (depayload->priv->discont)) {
544 GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
545 GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
546 depayload->priv->discont = FALSE;
549 return GST_BUFFER_LIST_SKIP_GROUP;
553 gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter,
554 gboolean do_ts, guint32 rtptime, gboolean is_list, gpointer obj)
558 data.depayload = filter;
559 data.caps = GST_PAD_CAPS (filter->srcpad);
560 data.rtptime = rtptime;
562 data.bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
565 GstBufferList **blist = obj;
566 gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, &data);
568 GstBuffer **buf = obj;
569 set_headers (buf, 0, 0, &data);
572 /* if this is the first buffer send a NEWSEGMENT */
573 if (G_UNLIKELY (filter->need_newsegment)) {
576 event = create_segment_event (filter, FALSE, 0);
578 gst_pad_push_event (filter->srcpad, event);
580 filter->need_newsegment = FALSE;
581 GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
588 * gst_base_rtp_depayload_push_ts:
589 * @filter: a #GstBaseRTPDepayload
590 * @timestamp: an RTP timestamp to apply
591 * @out_buf: a #GstBuffer
593 * Push @out_buf to the peer of @filter. This function takes ownership of
596 * Unlike gst_base_rtp_depayload_push(), this function will by default apply
597 * the last incomming timestamp on the outgoing buffer when it didn't have a
598 * timestamp already. The set_get_timestamp vmethod can be overwritten to change
599 * this behaviour (and take, for example, @timestamp into account).
601 * Returns: a #GstFlowReturn.
604 gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
610 gst_base_rtp_depayload_prepare_push (filter, TRUE, timestamp, FALSE,
613 if (G_LIKELY (res == GST_FLOW_OK))
614 res = gst_pad_push (filter->srcpad, out_buf);
616 gst_buffer_unref (out_buf);
622 * gst_base_rtp_depayload_push:
623 * @filter: a #GstBaseRTPDepayload
624 * @out_buf: a #GstBuffer
626 * Push @out_buf to the peer of @filter. This function takes ownership of
629 * Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
630 * any timestamp on the outgoing buffer. Subclasses should therefore timestamp
631 * outgoing buffers themselves.
633 * Returns: a #GstFlowReturn.
636 gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
640 res = gst_base_rtp_depayload_prepare_push (filter, FALSE, 0, FALSE, &out_buf);
642 if (G_LIKELY (res == GST_FLOW_OK))
643 res = gst_pad_push (filter->srcpad, out_buf);
645 gst_buffer_unref (out_buf);
651 * gst_base_rtp_depayload_push_list:
652 * @filter: a #GstBaseRTPDepayload
653 * @out_list: a #GstBufferList
655 * Push @out_list to the peer of @filter. This function takes ownership of
658 * Returns: a #GstFlowReturn.
663 gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter,
664 GstBufferList * out_list)
668 res = gst_base_rtp_depayload_prepare_push (filter, TRUE, 0, TRUE, &out_list);
670 if (G_LIKELY (res == GST_FLOW_OK))
671 res = gst_pad_push_list (filter->srcpad, out_list);
673 gst_buffer_list_unref (out_list);
678 /* convert the PacketLost event form a jitterbuffer to a segment update.
679 * subclasses can override this. */
681 gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter,
684 GstClockTime timestamp, duration, position;
686 const GstStructure *s;
688 s = gst_event_get_structure (event);
690 /* first start by parsing the timestamp and duration */
694 gst_structure_get_clock_time (s, "timestamp", ×tamp);
695 gst_structure_get_clock_time (s, "duration", &duration);
697 position = timestamp;
699 position += duration;
701 /* update the current segment with the elapsed time */
702 sevent = create_segment_event (filter, TRUE, position);
704 return gst_pad_push_event (filter->srcpad, sevent);
708 gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
709 guint32 rtptime, GstBuffer * buf)
711 GstBaseRTPDepayloadPrivate *priv;
712 GstClockTime timestamp, duration;
716 timestamp = GST_BUFFER_TIMESTAMP (buf);
717 duration = GST_BUFFER_DURATION (buf);
719 /* apply last incomming timestamp and duration to outgoing buffer if
720 * not otherwise set. */
721 if (!GST_CLOCK_TIME_IS_VALID (timestamp))
722 GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
723 if (!GST_CLOCK_TIME_IS_VALID (duration))
724 GST_BUFFER_DURATION (buf) = priv->duration;
727 static GstStateChangeReturn
728 gst_base_rtp_depayload_change_state (GstElement * element,
729 GstStateChange transition)
731 GstBaseRTPDepayload *filter;
732 GstBaseRTPDepayloadPrivate *priv;
733 GstStateChangeReturn ret;
735 filter = GST_BASE_RTP_DEPAYLOAD (element);
738 switch (transition) {
739 case GST_STATE_CHANGE_NULL_TO_READY:
741 case GST_STATE_CHANGE_READY_TO_PAUSED:
742 filter->need_newsegment = TRUE;
745 priv->play_speed = 1.0;
746 priv->play_scale = 1.0;
747 priv->next_seqnum = -1;
748 priv->negotiated = FALSE;
749 priv->discont = FALSE;
751 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
757 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
759 switch (transition) {
760 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
762 case GST_STATE_CHANGE_PAUSED_TO_READY:
764 case GST_STATE_CHANGE_READY_TO_NULL:
773 gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
774 const GValue * value, GParamSpec * pspec)
776 GstBaseRTPDepayload *filter;
778 filter = GST_BASE_RTP_DEPAYLOAD (object);
781 case PROP_QUEUE_DELAY:
782 filter->queue_delay = g_value_get_uint (value);
785 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
791 gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
792 GValue * value, GParamSpec * pspec)
794 GstBaseRTPDepayload *filter;
796 filter = GST_BASE_RTP_DEPAYLOAD (object);
799 case PROP_QUEUE_DELAY:
800 g_value_set_uint (value, filter->queue_delay);
803 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);