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
25 * Provides a base class for RTP depayloaders
28 #include "gstbasertpdepayload.h"
30 #ifdef GST_DISABLE_DEPRECATED
31 #define QUEUE_LOCK_INIT(base) (g_static_rec_mutex_init(&base->queuelock))
32 #define QUEUE_LOCK_FREE(base) (g_static_rec_mutex_free(&base->queuelock))
33 #define QUEUE_LOCK(base) (g_static_rec_mutex_lock(&base->queuelock))
34 #define QUEUE_UNLOCK(base) (g_static_rec_mutex_unlock(&base->queuelock))
36 /* otherwise it's already been defined in the header (FIXME 0.11)*/
39 GST_DEBUG_CATEGORY_STATIC (basertpdepayload_debug);
40 #define GST_CAT_DEFAULT (basertpdepayload_debug)
42 #define GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE(obj) \
43 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_BASE_RTP_DEPAYLOAD, GstBaseRTPDepayloadPrivate))
45 struct _GstBaseRTPDepayloadPrivate
47 GstClockTime npt_start;
48 GstClockTime npt_stop;
53 GstClockTime timestamp;
54 GstClockTime duration;
61 /* Filter signals and args */
68 #define DEFAULT_QUEUE_DELAY 0
77 static void gst_base_rtp_depayload_finalize (GObject * object);
78 static void gst_base_rtp_depayload_set_property (GObject * object,
79 guint prop_id, const GValue * value, GParamSpec * pspec);
80 static void gst_base_rtp_depayload_get_property (GObject * object,
81 guint prop_id, GValue * value, GParamSpec * pspec);
83 static gboolean gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps);
84 static GstFlowReturn gst_base_rtp_depayload_chain (GstPad * pad,
86 static gboolean gst_base_rtp_depayload_handle_sink_event (GstPad * pad,
89 static GstStateChangeReturn gst_base_rtp_depayload_change_state (GstElement *
90 element, GstStateChange transition);
92 static void gst_base_rtp_depayload_set_gst_timestamp
93 (GstBaseRTPDepayload * filter, guint32 rtptime, GstBuffer * buf);
94 static gboolean gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload *
95 filter, GstEvent * event);
96 static gboolean gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload *
97 filter, GstEvent * event);
99 GST_BOILERPLATE (GstBaseRTPDepayload, gst_base_rtp_depayload, GstElement,
103 gst_base_rtp_depayload_base_init (gpointer klass)
105 /*GstElementClass *element_class = GST_ELEMENT_CLASS (klass); */
109 gst_base_rtp_depayload_class_init (GstBaseRTPDepayloadClass * klass)
111 GObjectClass *gobject_class;
112 GstElementClass *gstelement_class;
114 gobject_class = G_OBJECT_CLASS (klass);
115 gstelement_class = (GstElementClass *) klass;
116 parent_class = g_type_class_peek_parent (klass);
118 g_type_class_add_private (klass, sizeof (GstBaseRTPDepayloadPrivate));
120 gobject_class->finalize = gst_base_rtp_depayload_finalize;
121 gobject_class->set_property = gst_base_rtp_depayload_set_property;
122 gobject_class->get_property = gst_base_rtp_depayload_get_property;
125 * GstBaseRTPDepayload::queue-delay
127 * Control the amount of packets to buffer.
129 * Deprecated: Use a jitterbuffer or RTP session manager to delay packet
130 * playback. This property has no effect anymore since 0.10.15.
132 #ifndef GST_REMOVE_DEPRECATED
133 g_object_class_install_property (gobject_class, PROP_QUEUE_DELAY,
134 g_param_spec_uint ("queue-delay", "Queue Delay",
135 "Amount of ms to queue/buffer, deprecated", 0, G_MAXUINT,
136 DEFAULT_QUEUE_DELAY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
139 gstelement_class->change_state = gst_base_rtp_depayload_change_state;
141 klass->set_gst_timestamp = gst_base_rtp_depayload_set_gst_timestamp;
142 klass->packet_lost = gst_base_rtp_depayload_packet_lost;
143 klass->handle_event = gst_base_rtp_depayload_handle_event;
145 GST_DEBUG_CATEGORY_INIT (basertpdepayload_debug, "basertpdepayload", 0,
146 "Base class for RTP Depayloaders");
150 gst_base_rtp_depayload_init (GstBaseRTPDepayload * filter,
151 GstBaseRTPDepayloadClass * klass)
153 GstPadTemplate *pad_template;
154 GstBaseRTPDepayloadPrivate *priv;
156 priv = GST_BASE_RTP_DEPAYLOAD_GET_PRIVATE (filter);
159 GST_DEBUG_OBJECT (filter, "init");
162 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "sink");
163 g_return_if_fail (pad_template != NULL);
164 filter->sinkpad = gst_pad_new_from_template (pad_template, "sink");
165 gst_pad_set_setcaps_function (filter->sinkpad,
166 gst_base_rtp_depayload_setcaps);
167 gst_pad_set_chain_function (filter->sinkpad, gst_base_rtp_depayload_chain);
168 gst_pad_set_event_function (filter->sinkpad,
169 gst_base_rtp_depayload_handle_sink_event);
170 gst_element_add_pad (GST_ELEMENT (filter), filter->sinkpad);
173 gst_element_class_get_pad_template (GST_ELEMENT_CLASS (klass), "src");
174 g_return_if_fail (pad_template != NULL);
175 filter->srcpad = gst_pad_new_from_template (pad_template, "src");
176 gst_pad_use_fixed_caps (filter->srcpad);
177 gst_element_add_pad (GST_ELEMENT (filter), filter->srcpad);
179 filter->queue = g_queue_new ();
180 filter->queue_delay = DEFAULT_QUEUE_DELAY;
182 gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
186 gst_base_rtp_depayload_finalize (GObject * object)
188 GstBaseRTPDepayload *filter = GST_BASE_RTP_DEPAYLOAD (object);
190 g_queue_free (filter->queue);
192 G_OBJECT_CLASS (parent_class)->finalize (object);
196 gst_base_rtp_depayload_setcaps (GstPad * pad, GstCaps * caps)
198 GstBaseRTPDepayload *filter;
199 GstBaseRTPDepayloadClass *bclass;
200 GstBaseRTPDepayloadPrivate *priv;
202 GstStructure *caps_struct;
205 filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
208 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
210 GST_DEBUG_OBJECT (filter, "Set caps");
212 caps_struct = gst_caps_get_structure (caps, 0);
214 /* get other values for newsegment */
215 value = gst_structure_get_value (caps_struct, "npt-start");
216 if (value && G_VALUE_HOLDS_UINT64 (value))
217 priv->npt_start = g_value_get_uint64 (value);
220 GST_DEBUG_OBJECT (filter, "NPT start %" G_GUINT64_FORMAT, priv->npt_start);
222 value = gst_structure_get_value (caps_struct, "npt-stop");
223 if (value && G_VALUE_HOLDS_UINT64 (value))
224 priv->npt_stop = g_value_get_uint64 (value);
228 GST_DEBUG_OBJECT (filter, "NPT stop %" G_GUINT64_FORMAT, priv->npt_stop);
230 value = gst_structure_get_value (caps_struct, "play-speed");
231 if (value && G_VALUE_HOLDS_DOUBLE (value))
232 priv->play_speed = g_value_get_double (value);
234 priv->play_speed = 1.0;
236 value = gst_structure_get_value (caps_struct, "play-scale");
237 if (value && G_VALUE_HOLDS_DOUBLE (value))
238 priv->play_scale = g_value_get_double (value);
240 priv->play_scale = 1.0;
242 if (bclass->set_caps) {
243 res = bclass->set_caps (filter, caps);
245 GST_WARNING_OBJECT (filter, "Subclass rejected caps %" GST_PTR_FORMAT,
252 priv->negotiated = res;
254 gst_object_unref (filter);
260 gst_base_rtp_depayload_chain (GstPad * pad, GstBuffer * in)
262 GstBaseRTPDepayload *filter;
263 GstBaseRTPDepayloadPrivate *priv;
264 GstBaseRTPDepayloadClass *bclass;
265 GstFlowReturn ret = GST_FLOW_OK;
267 GstClockTime timestamp;
273 filter = GST_BASE_RTP_DEPAYLOAD (GST_OBJECT_PARENT (pad));
276 /* we must have a setcaps first */
277 if (G_UNLIKELY (!priv->negotiated))
280 /* we must validate, it's possible that this element is plugged right after a
281 * network receiver and we don't want to operate on invalid data */
282 if (G_UNLIKELY (!gst_rtp_buffer_validate (in)))
286 priv->discont = GST_BUFFER_IS_DISCONT (in);
288 timestamp = GST_BUFFER_TIMESTAMP (in);
289 /* convert to running_time and save the timestamp, this is the timestamp
290 * we put on outgoing buffers. */
291 timestamp = gst_segment_to_running_time (&filter->segment, GST_FORMAT_TIME,
293 priv->timestamp = timestamp;
294 priv->duration = GST_BUFFER_DURATION (in);
296 seqnum = gst_rtp_buffer_get_seq (in);
297 rtptime = gst_rtp_buffer_get_timestamp (in);
300 GST_LOG_OBJECT (filter, "discont %d, seqnum %u, rtptime %u, timestamp %"
301 GST_TIME_FORMAT, priv->discont, seqnum, rtptime,
302 GST_TIME_ARGS (timestamp));
304 /* Check seqnum. This is a very simple check that makes sure that the seqnums
305 * are striclty increasing, dropping anything that is out of the ordinary. We
306 * can only do this when the next_seqnum is known. */
307 if (G_LIKELY (priv->next_seqnum != -1)) {
308 gap = gst_rtp_buffer_compare_seqnum (seqnum, priv->next_seqnum);
310 /* if we have no gap, all is fine */
311 if (G_UNLIKELY (gap != 0)) {
312 GST_LOG_OBJECT (filter, "got packet %u, expected %u, gap %d", seqnum,
313 priv->next_seqnum, gap);
315 /* seqnum > next_seqnum, we are missing some packets, this is always a
317 GST_LOG_OBJECT (filter, "%d missing packets", gap);
320 /* seqnum < next_seqnum, we have seen this packet before or the sender
321 * could be restarted. If the packet is not too old, we throw it away as
322 * a duplicate, otherwise we mark discont and continue. 100 misordered
323 * packets is a good threshold. See also RFC 4737. */
327 GST_LOG_OBJECT (filter,
328 "%d > 100, packet too old, sender likely restarted", gap);
333 priv->next_seqnum = (seqnum + 1) & 0xffff;
335 if (G_UNLIKELY (discont && !priv->discont)) {
336 GST_LOG_OBJECT (filter, "mark DISCONT on input buffer");
337 /* we detected a seqnum discont but the buffer was not flagged with a discont,
338 * set the discont flag so that the subclass can throw away old data. */
339 priv->discont = TRUE;
340 in = gst_buffer_make_metadata_writable (in);
341 GST_BUFFER_FLAG_SET (in, GST_BUFFER_FLAG_DISCONT);
344 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
346 if (G_UNLIKELY (bclass->process == NULL))
349 /* let's send it out to processing */
350 out_buf = bclass->process (filter, in);
352 /* we pass rtptime as backward compatibility, in reality, the incomming
353 * buffer timestamp is always applied to the outgoing packet. */
354 ret = gst_base_rtp_depayload_push_ts (filter, rtptime, out_buf);
356 gst_buffer_unref (in);
363 /* this is not fatal but should be filtered earlier */
364 if (GST_BUFFER_CAPS (in) == NULL) {
365 GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
366 ("No RTP format was negotiated."),
367 ("Input buffers need to have RTP caps set on them. This is usually "
368 "achieved by setting the 'caps' property of the upstream source "
369 "element (often udpsrc or appsrc), or by putting a capsfilter "
370 "element before the depayloader and setting the 'caps' property "
371 "on that. Also see http://cgit.freedesktop.org/gstreamer/"
372 "gst-plugins-good/tree/gst/rtp/README"));
374 GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
375 ("No RTP format was negotiated."),
376 ("RTP caps on input buffer were rejected, most likely because they "
377 "were incomplete or contained wrong values. Check the debug log "
378 "for more information."));
380 gst_buffer_unref (in);
381 return GST_FLOW_NOT_NEGOTIATED;
385 /* this is not fatal but should be filtered earlier */
386 GST_ELEMENT_WARNING (filter, STREAM, DECODE, (NULL),
387 ("Received invalid RTP payload, dropping"));
388 gst_buffer_unref (in);
393 GST_WARNING_OBJECT (filter, "%d <= 100, dropping old packet", gap);
394 gst_buffer_unref (in);
399 /* this is not fatal but should be filtered earlier */
400 GST_ELEMENT_ERROR (filter, STREAM, NOT_IMPLEMENTED, (NULL),
401 ("The subclass does not have a process method"));
402 gst_buffer_unref (in);
403 return GST_FLOW_ERROR;
408 gst_base_rtp_depayload_handle_event (GstBaseRTPDepayload * filter,
412 gboolean forward = TRUE;
414 switch (GST_EVENT_TYPE (event)) {
415 case GST_EVENT_FLUSH_STOP:
416 gst_segment_init (&filter->segment, GST_FORMAT_UNDEFINED);
417 filter->need_newsegment = TRUE;
418 filter->priv->next_seqnum = -1;
420 case GST_EVENT_NEWSEGMENT:
425 gint64 start, stop, position;
427 gst_event_parse_new_segment (event, &update, &rate, &fmt, &start, &stop,
430 gst_segment_set_newsegment (&filter->segment, update, rate, fmt,
431 start, stop, position);
433 /* don't pass the event downstream, we generate our own segment including
434 * the NTP time and other things we receive in caps */
438 case GST_EVENT_CUSTOM_DOWNSTREAM:
440 GstBaseRTPDepayloadClass *bclass;
442 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
444 if (gst_event_has_name (event, "GstRTPPacketLost")) {
445 /* we get this event from the jitterbuffer when it considers a packet as
446 * being lost. We send it to our packet_lost vmethod. The default
447 * implementation will make time progress by pushing out a NEWSEGMENT
448 * update event. Subclasses can override and to one of the following:
449 * - Adjust timestamp/duration to something more accurate before
450 * calling the parent (default) packet_lost method.
451 * - do some more advanced error concealing on the already received
452 * (fragmented) packets.
453 * - ignore the packet lost.
455 if (bclass->packet_lost)
456 res = bclass->packet_lost (filter, event);
466 res = gst_pad_push_event (filter->srcpad, event);
468 gst_event_unref (event);
474 gst_base_rtp_depayload_handle_sink_event (GstPad * pad, GstEvent * event)
476 gboolean res = FALSE;
477 GstBaseRTPDepayload *filter;
478 GstBaseRTPDepayloadClass *bclass;
480 filter = GST_BASE_RTP_DEPAYLOAD (gst_pad_get_parent (pad));
481 if (G_UNLIKELY (filter == NULL)) {
482 gst_event_unref (event);
486 bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
487 if (bclass->handle_event)
488 res = bclass->handle_event (filter, event);
490 gst_event_unref (event);
492 gst_object_unref (filter);
497 create_segment_event (GstBaseRTPDepayload * filter, gboolean update,
498 GstClockTime position)
502 GstBaseRTPDepayloadPrivate *priv;
506 if (priv->npt_stop != -1)
507 stop = priv->npt_stop - priv->npt_start;
511 event = gst_event_new_new_segment_full (update, priv->play_speed,
512 priv->play_scale, GST_FORMAT_TIME, position, stop,
513 position + priv->npt_start);
520 GstBaseRTPDepayload *depayload;
521 GstBaseRTPDepayloadClass *bclass;
527 static GstBufferListItem
528 set_headers (GstBuffer ** buffer, guint group, guint idx, HeaderData * data)
530 GstBaseRTPDepayload *depayload = data->depayload;
532 *buffer = gst_buffer_make_metadata_writable (*buffer);
533 gst_buffer_set_caps (*buffer, data->caps);
535 /* set the timestamp if we must and can */
536 if (data->bclass->set_gst_timestamp && data->do_ts)
537 data->bclass->set_gst_timestamp (depayload, data->rtptime, *buffer);
539 if (G_UNLIKELY (depayload->priv->discont)) {
540 GST_LOG_OBJECT (depayload, "Marking DISCONT on output buffer");
541 GST_BUFFER_FLAG_SET (*buffer, GST_BUFFER_FLAG_DISCONT);
542 depayload->priv->discont = FALSE;
545 return GST_BUFFER_LIST_SKIP_GROUP;
549 gst_base_rtp_depayload_prepare_push (GstBaseRTPDepayload * filter,
550 gboolean do_ts, guint32 rtptime, gboolean is_list, gpointer obj)
554 data.depayload = filter;
555 data.caps = GST_PAD_CAPS (filter->srcpad);
556 data.rtptime = rtptime;
558 data.bclass = GST_BASE_RTP_DEPAYLOAD_GET_CLASS (filter);
561 GstBufferList **blist = obj;
562 gst_buffer_list_foreach (*blist, (GstBufferListFunc) set_headers, &data);
564 GstBuffer **buf = obj;
565 set_headers (buf, 0, 0, &data);
568 /* if this is the first buffer send a NEWSEGMENT */
569 if (G_UNLIKELY (filter->need_newsegment)) {
572 event = create_segment_event (filter, FALSE, 0);
574 gst_pad_push_event (filter->srcpad, event);
576 filter->need_newsegment = FALSE;
577 GST_DEBUG_OBJECT (filter, "Pushed newsegment event on this first buffer");
584 * gst_base_rtp_depayload_push_ts:
585 * @filter: a #GstBaseRTPDepayload
586 * @timestamp: an RTP timestamp to apply
587 * @out_buf: a #GstBuffer
589 * Push @out_buf to the peer of @filter. This function takes ownership of
592 * Unlike gst_base_rtp_depayload_push(), this function will by default apply
593 * the last incomming timestamp on the outgoing buffer when it didn't have a
594 * timestamp already. The set_get_timestamp vmethod can be overwritten to change
595 * this behaviour (and take, for example, @timestamp into account).
597 * Returns: a #GstFlowReturn.
600 gst_base_rtp_depayload_push_ts (GstBaseRTPDepayload * filter, guint32 timestamp,
606 gst_base_rtp_depayload_prepare_push (filter, TRUE, timestamp, FALSE,
609 if (G_LIKELY (res == GST_FLOW_OK))
610 res = gst_pad_push (filter->srcpad, out_buf);
612 gst_buffer_unref (out_buf);
618 * gst_base_rtp_depayload_push:
619 * @filter: a #GstBaseRTPDepayload
620 * @out_buf: a #GstBuffer
622 * Push @out_buf to the peer of @filter. This function takes ownership of
625 * Unlike gst_base_rtp_depayload_push_ts(), this function will not apply
626 * any timestamp on the outgoing buffer. Subclasses should therefore timestamp
627 * outgoing buffers themselves.
629 * Returns: a #GstFlowReturn.
632 gst_base_rtp_depayload_push (GstBaseRTPDepayload * filter, GstBuffer * out_buf)
636 res = gst_base_rtp_depayload_prepare_push (filter, FALSE, 0, FALSE, &out_buf);
638 if (G_LIKELY (res == GST_FLOW_OK))
639 res = gst_pad_push (filter->srcpad, out_buf);
641 gst_buffer_unref (out_buf);
647 * gst_base_rtp_depayload_push_list:
648 * @filter: a #GstBaseRTPDepayload
649 * @out_list: a #GstBufferList
651 * Push @out_list to the peer of @filter. This function takes ownership of
654 * Returns: a #GstFlowReturn.
659 gst_base_rtp_depayload_push_list (GstBaseRTPDepayload * filter,
660 GstBufferList * out_list)
664 res = gst_base_rtp_depayload_prepare_push (filter, TRUE, 0, TRUE, &out_list);
666 if (G_LIKELY (res == GST_FLOW_OK))
667 res = gst_pad_push_list (filter->srcpad, out_list);
669 gst_buffer_list_unref (out_list);
674 /* convert the PacketLost event form a jitterbuffer to a segment update.
675 * subclasses can override this. */
677 gst_base_rtp_depayload_packet_lost (GstBaseRTPDepayload * filter,
680 GstClockTime timestamp, duration, position;
682 const GstStructure *s;
684 s = gst_event_get_structure (event);
686 /* first start by parsing the timestamp and duration */
690 gst_structure_get_clock_time (s, "timestamp", ×tamp);
691 gst_structure_get_clock_time (s, "duration", &duration);
693 position = timestamp;
695 position += duration;
697 /* update the current segment with the elapsed time */
698 sevent = create_segment_event (filter, TRUE, position);
700 return gst_pad_push_event (filter->srcpad, sevent);
704 gst_base_rtp_depayload_set_gst_timestamp (GstBaseRTPDepayload * filter,
705 guint32 rtptime, GstBuffer * buf)
707 GstBaseRTPDepayloadPrivate *priv;
708 GstClockTime timestamp, duration;
712 timestamp = GST_BUFFER_TIMESTAMP (buf);
713 duration = GST_BUFFER_DURATION (buf);
715 /* apply last incomming timestamp and duration to outgoing buffer if
716 * not otherwise set. */
717 if (!GST_CLOCK_TIME_IS_VALID (timestamp))
718 GST_BUFFER_TIMESTAMP (buf) = priv->timestamp;
719 if (!GST_CLOCK_TIME_IS_VALID (duration))
720 GST_BUFFER_DURATION (buf) = priv->duration;
723 static GstStateChangeReturn
724 gst_base_rtp_depayload_change_state (GstElement * element,
725 GstStateChange transition)
727 GstBaseRTPDepayload *filter;
728 GstBaseRTPDepayloadPrivate *priv;
729 GstStateChangeReturn ret;
731 filter = GST_BASE_RTP_DEPAYLOAD (element);
734 switch (transition) {
735 case GST_STATE_CHANGE_NULL_TO_READY:
737 case GST_STATE_CHANGE_READY_TO_PAUSED:
738 filter->need_newsegment = TRUE;
741 priv->play_speed = 1.0;
742 priv->play_scale = 1.0;
743 priv->next_seqnum = -1;
744 priv->negotiated = FALSE;
745 priv->discont = FALSE;
747 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
753 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
755 switch (transition) {
756 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
758 case GST_STATE_CHANGE_PAUSED_TO_READY:
760 case GST_STATE_CHANGE_READY_TO_NULL:
769 gst_base_rtp_depayload_set_property (GObject * object, guint prop_id,
770 const GValue * value, GParamSpec * pspec)
772 GstBaseRTPDepayload *filter;
774 filter = GST_BASE_RTP_DEPAYLOAD (object);
777 case PROP_QUEUE_DELAY:
778 filter->queue_delay = g_value_get_uint (value);
781 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
787 gst_base_rtp_depayload_get_property (GObject * object, guint prop_id,
788 GValue * value, GParamSpec * pspec)
790 GstBaseRTPDepayload *filter;
792 filter = GST_BASE_RTP_DEPAYLOAD (object);
795 case PROP_QUEUE_DELAY:
796 g_value_set_uint (value, filter->queue_delay);
799 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);