2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
9 * This library is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Library General Public
11 * License as published by the Free Software Foundation; either
12 * version 2 of the License, or (at your option) any later version.
14 * This library is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Library General Public License for more details.
19 * You should have received a copy of the GNU Library General Public
20 * License along with this library; if not, write to the
21 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
22 * Boston, MA 02111-1307, USA.
27 * SECTION:element-gstrtpjitterbuffer
29 * This element reorders and removes duplicate RTP packets as they are received
30 * from a network source. It will also wait for missing packets up to a
31 * configurable time limit using the #GstRtpJitterBuffer:latency property.
32 * Packets arriving too late are considered to be lost packets.
34 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
37 * The element needs the clock-rate of the RTP payload in order to estimate the
38 * delay. This information is obtained either from the caps on the sink pad or,
39 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
40 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
42 * This element will automatically be used inside gstrtpbin.
45 * <title>Example pipelines</title>
47 * gst-launch rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! gstrtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
48 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
49 * inserted into the pipeline to smooth out network jitter and to reorder the
50 * out-of-order RTP packets.
53 * Last reviewed on 2007-05-28 (0.10.5)
62 #include <gst/rtp/gstrtpbuffer.h>
64 #include "gstrtpbin-marshal.h"
66 #include "gstrtpjitterbuffer.h"
67 #include "rtpjitterbuffer.h"
70 #include <gst/glib-compat-private.h>
72 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
73 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
75 /* RTPJitterBuffer signals and args */
78 SIGNAL_REQUEST_PT_MAP,
86 #define DEFAULT_LATENCY_MS 200
87 #define DEFAULT_DROP_ON_LATENCY FALSE
88 #define DEFAULT_TS_OFFSET 0
89 #define DEFAULT_DO_LOST FALSE
90 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
91 #define DEFAULT_PERCENT 0
105 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
107 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
109 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
113 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
114 #define JBUF_WAIT(priv) (g_cond_wait (&(priv)->jbuf_cond, &(priv)->jbuf_lock))
116 #define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \
118 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
122 #define JBUF_SIGNAL(priv) (g_cond_signal (&(priv)->jbuf_cond))
124 struct _GstRtpJitterBufferPrivate
126 GstPad *sinkpad, *srcpad;
129 RTPJitterBuffer *jbuf;
140 gboolean drop_on_latency;
144 /* the last seqnum we pushed out */
145 guint32 last_popped_seqnum;
146 /* the next expected seqnum we push */
148 /* last output time */
149 GstClockTime last_out_time;
150 /* the next expected seqnum we receive */
151 guint32 next_in_seqnum;
153 /* start and stop ranges */
154 GstClockTime npt_start;
155 GstClockTime npt_stop;
156 guint64 ext_timestamp;
157 guint64 last_elapsed;
158 guint64 estimated_eos;
160 gboolean reached_npt_stop;
165 /* clock rate and rtp timestamp offset */
169 gint64 prev_ts_offset;
171 /* when we are shutting down */
172 GstFlowReturn srcresult;
178 gboolean unscheduled;
179 /* the latency of the upstream peer, we have to take this into account when
180 * synchronizing the buffers. */
181 GstClockTime peer_latency;
183 /* some accounting */
185 guint64 num_duplicates;
188 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
189 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
190 GstRtpJitterBufferPrivate))
192 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
193 GST_STATIC_PAD_TEMPLATE ("sink",
196 GST_STATIC_CAPS ("application/x-rtp, "
197 "clock-rate = (int) [ 1, 2147483647 ]"
198 /* "payload = (int) , "
199 * "encoding-name = (string) "
203 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
204 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
207 GST_STATIC_CAPS ("application/x-rtcp")
210 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
211 GST_STATIC_PAD_TEMPLATE ("src",
214 GST_STATIC_CAPS ("application/x-rtp"
215 /* "payload = (int) , "
216 * "clock-rate = (int) , "
217 * "encoding-name = (string) "
221 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
223 #define gst_rtp_jitter_buffer_parent_class parent_class
224 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
226 /* object overrides */
227 static void gst_rtp_jitter_buffer_set_property (GObject * object,
228 guint prop_id, const GValue * value, GParamSpec * pspec);
229 static void gst_rtp_jitter_buffer_get_property (GObject * object,
230 guint prop_id, GValue * value, GParamSpec * pspec);
231 static void gst_rtp_jitter_buffer_finalize (GObject * object);
233 /* element overrides */
234 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
235 * element, GstStateChange transition);
236 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
237 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
238 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
240 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
243 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
244 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
247 /* sinkpad overrides */
248 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
249 GstObject * parent, GstEvent * event);
250 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
251 GstObject * parent, GstBuffer * buffer);
253 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
254 GstObject * parent, GstEvent * event);
255 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
256 GstObject * parent, GstBuffer * buffer);
258 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
259 GstObject * parent, GstQuery * query);
261 /* srcpad overrides */
262 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
263 GstObject * parent, GstEvent * event);
264 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
265 GstObject * parent, GstPadMode mode, gboolean active);
266 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
267 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
268 GstObject * parent, GstQuery * query);
271 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
273 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
274 gboolean active, guint64 base_time);
277 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
279 GObjectClass *gobject_class;
280 GstElementClass *gstelement_class;
282 gobject_class = (GObjectClass *) klass;
283 gstelement_class = (GstElementClass *) klass;
285 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
287 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
289 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
290 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
293 * GstRtpJitterBuffer::latency:
295 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
296 * for at most this time.
298 g_object_class_install_property (gobject_class, PROP_LATENCY,
299 g_param_spec_uint ("latency", "Buffer latency in ms",
300 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
301 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
303 * GstRtpJitterBuffer::drop-on-latency:
305 * Drop oldest buffers when the queue is completely filled.
307 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
308 g_param_spec_boolean ("drop-on-latency",
309 "Drop buffers when maximum latency is reached",
310 "Tells the jitterbuffer to never exceed the given latency in size",
311 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
313 * GstRtpJitterBuffer::ts-offset:
315 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
316 * This is mainly used to ensure interstream synchronisation.
318 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
319 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
320 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
321 G_MAXINT64, DEFAULT_TS_OFFSET,
322 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
325 * GstRtpJitterBuffer::do-lost:
327 * Send out a GstRTPPacketLost event downstream when a packet is considered
330 g_object_class_install_property (gobject_class, PROP_DO_LOST,
331 g_param_spec_boolean ("do-lost", "Do Lost",
332 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
333 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
336 * GstRtpJitterBuffer::mode:
338 * Control the buffering and timestamping mode used by the jitterbuffer.
340 g_object_class_install_property (gobject_class, PROP_MODE,
341 g_param_spec_enum ("mode", "Mode",
342 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
343 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 * GstRtpJitterBuffer::percent:
347 * The percent of the jitterbuffer that is filled.
351 g_object_class_install_property (gobject_class, PROP_PERCENT,
352 g_param_spec_int ("percent", "percent",
353 "The buffer filled percent", 0, 100,
354 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
356 * GstRtpJitterBuffer::request-pt-map:
357 * @buffer: the object which received the signal
360 * Request the payload type as #GstCaps for @pt.
362 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
363 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
364 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
365 request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
366 GST_TYPE_CAPS, 1, G_TYPE_UINT);
368 * GstRtpJitterBuffer::handle-sync:
369 * @buffer: the object which received the signal
370 * @struct: a GstStructure containing sync values.
372 * Be notified of new sync values.
374 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
375 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
376 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
377 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
378 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
381 * GstRtpJitterBuffer::on-npt-stop
382 * @buffer: the object which received the signal
384 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
385 * the npt-stop position.
387 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
388 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
389 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
390 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
391 G_TYPE_NONE, 0, G_TYPE_NONE);
394 * GstRtpJitterBuffer::clear-pt-map:
395 * @buffer: the object which received the signal
397 * Invalidate the clock-rate as obtained with the
398 * #GstRtpJitterBuffer::request-pt-map signal.
400 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
401 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
402 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
403 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
404 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
407 * GstRtpJitterBuffer::set-active:
408 * @buffer: the object which received the signal
410 * Start pushing out packets with the given base time. This signal is only
411 * useful in buffering mode.
413 * Returns: the time of the last pushed packet.
417 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
418 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
419 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
420 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
421 gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
424 gstelement_class->change_state =
425 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
426 gstelement_class->request_new_pad =
427 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
428 gstelement_class->release_pad =
429 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
430 gstelement_class->provide_clock =
431 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
433 gst_element_class_add_pad_template (gstelement_class,
434 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
435 gst_element_class_add_pad_template (gstelement_class,
436 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
437 gst_element_class_add_pad_template (gstelement_class,
438 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
440 gst_element_class_set_details_simple (gstelement_class,
441 "RTP packet jitter-buffer", "Filter/Network/RTP",
442 "A buffer that deals with network jitter and other transmission faults",
443 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
444 "Wim Taymans <wim.taymans@gmail.com>");
446 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
447 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
449 GST_DEBUG_CATEGORY_INIT
450 (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
454 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
456 GstRtpJitterBufferPrivate *priv;
458 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
459 jitterbuffer->priv = priv;
461 priv->latency_ms = DEFAULT_LATENCY_MS;
462 priv->latency_ns = priv->latency_ms * GST_MSECOND;
463 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
464 priv->do_lost = DEFAULT_DO_LOST;
466 priv->jbuf = rtp_jitter_buffer_new ();
467 g_mutex_init (&priv->jbuf_lock);
468 g_cond_init (&priv->jbuf_cond);
470 /* reset skew detection initialy */
471 rtp_jitter_buffer_reset_skew (priv->jbuf);
472 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
473 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
477 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
480 gst_pad_set_activatemode_function (priv->srcpad,
481 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
482 gst_pad_set_query_function (priv->srcpad,
483 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
484 gst_pad_set_event_function (priv->srcpad,
485 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
488 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
491 gst_pad_set_chain_function (priv->sinkpad,
492 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
493 gst_pad_set_event_function (priv->sinkpad,
494 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
495 gst_pad_set_query_function (priv->sinkpad,
496 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
498 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
499 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
501 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
505 gst_rtp_jitter_buffer_finalize (GObject * object)
507 GstRtpJitterBuffer *jitterbuffer;
509 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
511 g_mutex_clear (&jitterbuffer->priv->jbuf_lock);
512 g_cond_clear (&jitterbuffer->priv->jbuf_cond);
514 g_object_unref (jitterbuffer->priv->jbuf);
516 G_OBJECT_CLASS (parent_class)->finalize (object);
520 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
522 GstRtpJitterBuffer *jitterbuffer;
523 GstPad *otherpad = NULL;
527 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
529 if (pad == jitterbuffer->priv->sinkpad) {
530 otherpad = jitterbuffer->priv->srcpad;
531 } else if (pad == jitterbuffer->priv->srcpad) {
532 otherpad = jitterbuffer->priv->sinkpad;
533 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
537 g_value_init (&val, GST_TYPE_PAD);
538 g_value_set_object (&val, otherpad);
539 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
540 g_value_unset (&val);
546 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
548 GstRtpJitterBufferPrivate *priv;
550 priv = jitterbuffer->priv;
552 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
555 gst_pad_new_from_static_template
556 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
557 gst_pad_set_chain_function (priv->rtcpsinkpad,
558 gst_rtp_jitter_buffer_chain_rtcp);
559 gst_pad_set_event_function (priv->rtcpsinkpad,
560 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
561 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
562 gst_rtp_jitter_buffer_iterate_internal_links);
563 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
564 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
566 return priv->rtcpsinkpad;
570 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
572 GstRtpJitterBufferPrivate *priv;
574 priv = jitterbuffer->priv;
576 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
578 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
580 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
581 priv->rtcpsinkpad = NULL;
585 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
586 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
588 GstRtpJitterBuffer *jitterbuffer;
589 GstElementClass *klass;
591 GstRtpJitterBufferPrivate *priv;
593 g_return_val_if_fail (templ != NULL, NULL);
594 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
596 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
597 priv = jitterbuffer->priv;
598 klass = GST_ELEMENT_GET_CLASS (element);
600 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
602 /* figure out the template */
603 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
604 if (priv->rtcpsinkpad != NULL)
607 result = create_rtcp_sink (jitterbuffer);
616 g_warning ("gstrtpjitterbuffer: this is not our template");
621 g_warning ("gstrtpjitterbuffer: pad already requested");
627 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
629 GstRtpJitterBuffer *jitterbuffer;
630 GstRtpJitterBufferPrivate *priv;
632 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
633 g_return_if_fail (GST_IS_PAD (pad));
635 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
636 priv = jitterbuffer->priv;
638 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
640 if (priv->rtcpsinkpad == pad) {
641 remove_rtcp_sink (jitterbuffer);
650 g_warning ("gstjitterbuffer: asked to release an unknown pad");
656 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
658 return gst_system_clock_obtain ();
662 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
664 GstRtpJitterBufferPrivate *priv;
666 priv = jitterbuffer->priv;
668 /* this will trigger a new pt-map request signal, FIXME, do something better. */
671 priv->clock_rate = -1;
672 /* do not clear current content, but refresh state for new arrival */
673 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
674 rtp_jitter_buffer_reset_skew (priv->jbuf);
675 priv->last_popped_seqnum = -1;
676 priv->next_seqnum = -1;
681 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
684 GstRtpJitterBufferPrivate *priv;
685 GstClockTime last_out;
691 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
692 active, GST_TIME_ARGS (offset));
694 if (active != priv->active) {
695 /* add the amount of time spent in paused to the output offset. All
696 * outgoing buffers will have this offset applied to their timestamps in
697 * order to make them arrive in time in the sink. */
698 priv->out_offset = offset;
699 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
700 GST_TIME_ARGS (priv->out_offset));
701 priv->active = active;
705 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
707 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
708 /* head buffer timestamp and offset gives our output time */
709 last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset;
711 /* use last known time when the buffer is empty */
712 last_out = priv->last_out_time;
720 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
722 GstRtpJitterBuffer *jitterbuffer;
723 GstRtpJitterBufferPrivate *priv;
728 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
729 priv = jitterbuffer->priv;
731 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
733 caps = gst_pad_peer_query_caps (other, filter);
735 templ = gst_pad_get_pad_template_caps (pad);
737 GST_DEBUG_OBJECT (jitterbuffer, "use template");
742 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
744 intersect = gst_caps_intersect (caps, templ);
745 gst_caps_unref (caps);
746 gst_caps_unref (templ);
750 gst_object_unref (jitterbuffer);
756 * Must be called with JBUF_LOCK held
760 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
763 GstRtpJitterBufferPrivate *priv;
764 GstStructure *caps_struct;
768 priv = jitterbuffer->priv;
770 /* first parse the caps */
771 caps_struct = gst_caps_get_structure (caps, 0);
773 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
775 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
776 * measure the amount of data in the buffer */
777 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
780 if (priv->clock_rate <= 0)
783 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
785 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
786 * can use this to track the amount of time elapsed on the sender. */
787 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
788 priv->clock_base = val;
790 priv->clock_base = -1;
792 priv->ext_timestamp = priv->clock_base;
794 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
797 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
798 /* first expected seqnum, only update when we didn't have a previous base. */
799 if (priv->next_in_seqnum == -1)
800 priv->next_in_seqnum = val;
801 if (priv->next_seqnum == -1)
802 priv->next_seqnum = val;
805 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
807 /* the start and stop times. The seqnum-base corresponds to the start time. We
808 * will keep track of the seqnums on the output and when we reach the one
809 * corresponding to npt-stop, we emit the npt-stop-reached signal */
810 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
811 priv->npt_start = tval;
815 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
816 priv->npt_stop = tval;
820 GST_DEBUG_OBJECT (jitterbuffer,
821 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
822 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
829 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
834 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
840 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
842 GstRtpJitterBufferPrivate *priv;
844 priv = jitterbuffer->priv;
847 /* mark ourselves as flushing */
848 priv->srcresult = GST_FLOW_FLUSHING;
849 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
850 /* this unblocks any waiting pops on the src pad task */
852 /* unlock clock, we just unschedule, the entry will be released by the
853 * locking streaming thread. */
854 if (priv->clock_id) {
855 gst_clock_id_unschedule (priv->clock_id);
856 priv->unscheduled = TRUE;
862 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
864 GstRtpJitterBufferPrivate *priv;
866 priv = jitterbuffer->priv;
869 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
870 /* Mark as non flushing */
871 priv->srcresult = GST_FLOW_OK;
872 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
873 priv->last_popped_seqnum = -1;
874 priv->last_out_time = -1;
875 priv->next_seqnum = -1;
876 priv->next_in_seqnum = -1;
877 priv->clock_rate = -1;
879 priv->estimated_eos = -1;
880 priv->last_elapsed = 0;
881 priv->reached_npt_stop = FALSE;
882 priv->ext_timestamp = -1;
883 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
884 rtp_jitter_buffer_flush (priv->jbuf);
885 rtp_jitter_buffer_reset_skew (priv->jbuf);
890 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
891 GstPadMode mode, gboolean active)
894 GstRtpJitterBuffer *jitterbuffer = NULL;
896 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
899 case GST_PAD_MODE_PUSH:
901 /* allow data processing */
902 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
904 /* start pushing out buffers */
905 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
906 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
907 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer);
909 /* make sure all data processing stops ASAP */
910 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
912 /* NOTE this will hardlock if the state change is called from the src pad
913 * task thread because we will _join() the thread. */
914 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
915 result = gst_pad_stop_task (pad);
925 static GstStateChangeReturn
926 gst_rtp_jitter_buffer_change_state (GstElement * element,
927 GstStateChange transition)
929 GstRtpJitterBuffer *jitterbuffer;
930 GstRtpJitterBufferPrivate *priv;
931 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
933 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
934 priv = jitterbuffer->priv;
936 switch (transition) {
937 case GST_STATE_CHANGE_NULL_TO_READY:
939 case GST_STATE_CHANGE_READY_TO_PAUSED:
941 /* reset negotiated values */
942 priv->clock_rate = -1;
943 priv->clock_base = -1;
944 priv->peer_latency = 0;
946 /* block until we go to PLAYING */
947 priv->blocked = TRUE;
950 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
952 /* unblock to allow streaming in PLAYING */
953 priv->blocked = FALSE;
961 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
963 switch (transition) {
964 case GST_STATE_CHANGE_READY_TO_PAUSED:
965 /* we are a live element because we sync to the clock, which we can only
966 * do in the PLAYING state */
967 if (ret != GST_STATE_CHANGE_FAILURE)
968 ret = GST_STATE_CHANGE_NO_PREROLL;
970 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
972 /* block to stop streaming when PAUSED */
973 priv->blocked = TRUE;
975 if (ret != GST_STATE_CHANGE_FAILURE)
976 ret = GST_STATE_CHANGE_NO_PREROLL;
978 case GST_STATE_CHANGE_PAUSED_TO_READY:
980 case GST_STATE_CHANGE_READY_TO_NULL:
990 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
994 GstRtpJitterBuffer *jitterbuffer;
995 GstRtpJitterBufferPrivate *priv;
997 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
998 priv = jitterbuffer->priv;
1000 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1002 switch (GST_EVENT_TYPE (event)) {
1003 case GST_EVENT_LATENCY:
1005 GstClockTime latency;
1007 gst_event_parse_latency (event, &latency);
1010 /* adjust the overall buffer delay to the total pipeline latency in
1011 * buffering mode because if downstream consumes too fast (because of
1012 * large latency or queues, we would start rebuffering again. */
1013 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1014 RTP_JITTER_BUFFER_MODE_BUFFER) {
1015 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1019 ret = gst_pad_push_event (priv->sinkpad, event);
1023 ret = gst_pad_push_event (priv->sinkpad, event);
1031 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1034 gboolean ret = TRUE;
1035 GstRtpJitterBuffer *jitterbuffer;
1036 GstRtpJitterBufferPrivate *priv;
1038 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1039 priv = jitterbuffer->priv;
1041 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1043 switch (GST_EVENT_TYPE (event)) {
1044 case GST_EVENT_CAPS:
1048 gst_event_parse_caps (event, &caps);
1051 ret = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1054 /* set same caps on srcpad on success */
1056 ret = gst_pad_push_event (priv->srcpad, event);
1058 gst_event_unref (event);
1061 case GST_EVENT_SEGMENT:
1063 gst_event_copy_segment (event, &priv->segment);
1065 /* we need time for now */
1066 if (priv->segment.format != GST_FORMAT_TIME)
1067 goto newseg_wrong_format;
1069 GST_DEBUG_OBJECT (jitterbuffer,
1070 "newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
1072 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1073 ret = gst_pad_push_event (priv->srcpad, event);
1076 case GST_EVENT_FLUSH_START:
1077 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1078 ret = gst_pad_push_event (priv->srcpad, event);
1080 case GST_EVENT_FLUSH_STOP:
1081 ret = gst_pad_push_event (priv->srcpad, event);
1083 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1084 GST_PAD_MODE_PUSH, TRUE);
1088 /* push EOS in queue. We always push it at the head */
1090 /* check for flushing, we need to discard the event and return FALSE when
1091 * we are flushing */
1092 ret = priv->srcresult == GST_FLOW_OK;
1093 if (ret && !priv->eos) {
1094 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1097 } else if (priv->eos) {
1098 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1100 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1101 gst_flow_get_name (priv->srcresult));
1104 gst_event_unref (event);
1108 ret = gst_pad_push_event (priv->srcpad, event);
1117 newseg_wrong_format:
1119 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1121 gst_event_unref (event);
1127 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1130 gboolean ret = TRUE;
1131 GstRtpJitterBuffer *jitterbuffer;
1133 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1135 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1137 switch (GST_EVENT_TYPE (event)) {
1138 case GST_EVENT_FLUSH_START:
1139 gst_event_unref (event);
1141 case GST_EVENT_FLUSH_STOP:
1142 gst_event_unref (event);
1145 ret = gst_pad_event_default (pad, parent, event);
1153 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1154 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1155 * GST_FLOW_FLUSHING when the element is shutting down. On success
1156 * GST_FLOW_OK is returned.
1158 static GstFlowReturn
1159 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1163 GValue args[2] = { {0}, {0} };
1167 g_value_init (&args[0], GST_TYPE_ELEMENT);
1168 g_value_set_object (&args[0], jitterbuffer);
1169 g_value_init (&args[1], G_TYPE_UINT);
1170 g_value_set_uint (&args[1], pt);
1172 g_value_init (&ret, GST_TYPE_CAPS);
1173 g_value_set_boxed (&ret, NULL);
1175 JBUF_UNLOCK (jitterbuffer->priv);
1176 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1178 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1180 g_value_unset (&args[0]);
1181 g_value_unset (&args[1]);
1182 caps = (GstCaps *) g_value_dup_boxed (&ret);
1183 g_value_unset (&ret);
1187 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1188 gst_caps_unref (caps);
1190 if (G_UNLIKELY (!res))
1198 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1199 return GST_FLOW_ERROR;
1203 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1204 return GST_FLOW_FLUSHING;
1208 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1209 return GST_FLOW_ERROR;
1213 /* call with jbuf lock held */
1215 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1217 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1219 /* too short a stream, or too close to EOS will never really fill buffer */
1220 if (*percent != -1 && priv->npt_stop != -1 &&
1221 priv->npt_stop - priv->npt_start <=
1222 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1223 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1224 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1230 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1232 GstMessage *message;
1234 /* Post a buffering message */
1235 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1236 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1238 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1241 static GstFlowReturn
1242 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
1245 GstRtpJitterBuffer *jitterbuffer;
1246 GstRtpJitterBufferPrivate *priv;
1248 GstFlowReturn ret = GST_FLOW_OK;
1249 GstClockTime timestamp;
1254 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1256 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1258 if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
1259 goto invalid_buffer;
1261 priv = jitterbuffer->priv;
1263 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp);
1264 pt = gst_rtp_buffer_get_payload_type (&rtp);
1265 seqnum = gst_rtp_buffer_get_seq (&rtp);
1266 gst_rtp_buffer_unmap (&rtp);
1268 /* take the timestamp of the buffer. This is the time when the packet was
1269 * received and is used to calculate jitter and clock skew. We will adjust
1270 * this timestamp with the smoothed value after processing it in the
1272 timestamp = GST_BUFFER_TIMESTAMP (buffer);
1273 /* bring to running time */
1274 timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
1277 GST_DEBUG_OBJECT (jitterbuffer,
1278 "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
1279 GST_TIME_ARGS (timestamp));
1281 JBUF_LOCK_CHECK (priv, out_flushing);
1283 if (G_UNLIKELY (priv->last_pt != pt)) {
1284 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1288 /* reset clock-rate so that we get a new one */
1289 priv->clock_rate = -1;
1291 /* Try to get the clock-rate from the caps first if we can. If there are no
1292 * caps we must fire the signal to get the clock-rate. */
1293 if ((caps = gst_pad_get_current_caps (pad))) {
1294 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1295 gst_caps_unref (caps);
1299 if (G_UNLIKELY (priv->clock_rate == -1)) {
1300 /* no clock rate given on the caps, try to get one with the signal */
1301 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1302 pt) == GST_FLOW_FLUSHING)
1305 if (G_UNLIKELY (priv->clock_rate == -1))
1309 /* don't accept more data on EOS */
1310 if (G_UNLIKELY (priv->eos))
1313 /* now check against our expected seqnum */
1314 if (G_LIKELY (priv->next_in_seqnum != -1)) {
1316 gboolean reset = FALSE;
1318 gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
1319 if (G_UNLIKELY (gap != 0)) {
1320 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1321 priv->next_in_seqnum, seqnum, gap);
1322 /* priv->next_in_seqnum >= seqnum, this packet is too late or the
1323 * sender might have been restarted with different seqnum. */
1324 if (gap < -RTP_MAX_MISORDER) {
1325 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
1328 /* priv->next_in_seqnum < seqnum, this is a new packet */
1329 else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1330 GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
1334 GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
1337 if (G_UNLIKELY (reset)) {
1338 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1339 rtp_jitter_buffer_flush (priv->jbuf);
1340 rtp_jitter_buffer_reset_skew (priv->jbuf);
1341 priv->last_popped_seqnum = -1;
1342 priv->next_seqnum = seqnum;
1345 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1347 /* let's check if this buffer is too late, we can only accept packets with
1348 * bigger seqnum than the one we last pushed. */
1349 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1352 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1354 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1355 if (G_UNLIKELY (gap <= 0))
1359 /* let's drop oldest packet if the queue is already full and drop-on-latency
1360 * is set. We can only do this when there actually is a latency. When no
1361 * latency is set, we just pump it in the queue and let the other end push it
1362 * out as fast as possible. */
1363 if (priv->latency_ms && priv->drop_on_latency) {
1365 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1367 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1370 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1372 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
1375 gst_buffer_unref (old_buf);
1379 /* we need to make the metadata writable before pushing it in the jitterbuffer
1380 * because the jitterbuffer will update the timestamp */
1381 buffer = gst_buffer_make_writable (buffer);
1383 /* now insert the packet into the queue in sorted order. This function returns
1384 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1385 * have a duplicate. */
1386 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
1387 priv->clock_rate, &tail, &percent)))
1390 /* signal addition of new buffer when the _loop is waiting. */
1394 /* let's unschedule and unblock any waiting buffers. We only want to do this
1395 * when the tail buffer changed */
1396 if (G_UNLIKELY (priv->clock_id && tail)) {
1397 GST_DEBUG_OBJECT (jitterbuffer,
1398 "Unscheduling waiting buffer, new tail buffer");
1399 gst_clock_id_unschedule (priv->clock_id);
1400 priv->unscheduled = TRUE;
1403 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
1404 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
1406 check_buffering_percent (jitterbuffer, &percent);
1412 post_buffering_percent (jitterbuffer, percent);
1419 /* this is not fatal but should be filtered earlier */
1420 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
1421 ("Received invalid RTP payload, dropping"));
1422 gst_buffer_unref (buffer);
1427 GST_WARNING_OBJECT (jitterbuffer,
1428 "No clock-rate in caps!, dropping buffer");
1429 gst_buffer_unref (buffer);
1434 ret = priv->srcresult;
1435 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
1436 gst_buffer_unref (buffer);
1442 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
1443 gst_buffer_unref (buffer);
1448 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
1449 " popped, dropping", seqnum, priv->last_popped_seqnum);
1451 gst_buffer_unref (buffer);
1456 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
1458 priv->num_duplicates++;
1459 gst_buffer_unref (buffer);
1465 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1467 GstRtpJitterBufferPrivate *priv;
1469 priv = jitterbuffer->priv;
1471 if (timestamp == -1)
1474 /* apply the timestamp offset, this is used for inter stream sync */
1475 timestamp += priv->ts_offset;
1476 /* add the offset, this is used when buffering */
1477 timestamp += priv->out_offset;
1483 get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1485 GstClockTime result;
1486 GstRtpJitterBufferPrivate *priv;
1488 priv = jitterbuffer->priv;
1490 result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
1491 /* add latency, this includes our own latency and the peer latency. */
1492 result += priv->latency_ns;
1493 result += priv->peer_latency;
1499 eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
1500 GstRtpJitterBuffer * jitterbuffer)
1502 GstRtpJitterBufferPrivate *priv;
1504 priv = jitterbuffer->priv;
1506 JBUF_LOCK_CHECK (priv, flushing);
1507 if (priv->waiting) {
1508 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
1509 priv->reached_npt_stop = TRUE;
1525 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1527 guint64 ext_time, elapsed;
1529 GstRtpJitterBufferPrivate *priv;
1530 GstRTPBuffer rtp = { NULL, };
1532 priv = jitterbuffer->priv;
1533 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1534 rtp_time = gst_rtp_buffer_get_timestamp (&rtp);
1535 gst_rtp_buffer_unmap (&rtp);
1537 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
1538 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
1540 if (rtp_time < priv->ext_timestamp) {
1541 ext_time = priv->ext_timestamp;
1543 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
1546 if (ext_time > priv->clock_base)
1547 elapsed = ext_time - priv->clock_base;
1551 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
1556 * This funcion will push out buffers on the source pad.
1558 * For each pushed buffer, the seqnum is recorded, if the next buffer B has a
1559 * different seqnum (missing packets before B), this function will wait for the
1560 * missing packet to arrive up to the timestamp of buffer B.
1563 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
1565 GstRtpJitterBufferPrivate *priv;
1567 GstFlowReturn result;
1569 guint32 next_seqnum;
1570 GstClockTime timestamp, out_time;
1571 gboolean discont = FALSE;
1575 GstClockTime sync_time;
1577 GstRTPBuffer rtp = { NULL, };
1579 priv = jitterbuffer->priv;
1581 JBUF_LOCK_CHECK (priv, flushing);
1583 GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
1586 /* always wait if we are blocked */
1587 if (G_LIKELY (!priv->blocked)) {
1588 /* we're buffering but not EOS, wait. */
1589 if (!priv->eos && (!priv->active
1590 || rtp_jitter_buffer_is_buffering (priv->jbuf))) {
1591 GstClockTime elapsed, delay, left;
1593 if (priv->estimated_eos == -1)
1596 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1597 if (outbuf != NULL) {
1598 elapsed = compute_elapsed (jitterbuffer, outbuf);
1599 if (GST_BUFFER_DURATION_IS_VALID (outbuf))
1600 elapsed += GST_BUFFER_DURATION (outbuf);
1602 GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed");
1603 elapsed = priv->last_elapsed;
1606 delay = rtp_jitter_buffer_get_delay (priv->jbuf);
1608 if (priv->estimated_eos > elapsed)
1609 left = priv->estimated_eos - elapsed;
1613 GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT
1614 " estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT
1615 " delay %" GST_TIME_FORMAT,
1616 GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos),
1617 GST_TIME_ARGS (left), GST_TIME_ARGS (delay));
1621 /* if we have a packet, we can exit the loop and grab it */
1622 if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
1624 /* no packets but we are EOS, do eos logic */
1625 if (G_UNLIKELY (priv->eos))
1627 /* underrun, wait for packets or flushing now if we are expecting an EOS
1628 * timeout, set the async timer for it too */
1629 if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
1630 sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
1632 GST_OBJECT_LOCK (jitterbuffer);
1633 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1635 GST_INFO_OBJECT (jitterbuffer, "scheduling timeout");
1636 id = gst_clock_new_single_shot_id (clock, sync_time);
1637 gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
1640 GST_OBJECT_UNLOCK (jitterbuffer);
1645 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
1646 priv->waiting = TRUE;
1648 priv->waiting = FALSE;
1649 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
1652 /* unschedule any pending async notifications we might have */
1653 gst_clock_id_unschedule (id);
1654 gst_clock_id_unref (id);
1656 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
1659 if (id && priv->reached_npt_stop) {
1664 /* peek a buffer, we're just looking at the timestamp and the sequence number.
1665 * If all is fine, we'll pop and push it. If the sequence number is wrong we
1666 * wait on the timestamp. In the chain function we will unlock the wait when a
1667 * new buffer is available. The peeked buffer is valid for as long as we hold
1668 * the jitterbuffer lock. */
1669 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1671 /* get the seqnum and the next expected seqnum */
1672 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1673 seqnum = gst_rtp_buffer_get_seq (&rtp);
1674 gst_rtp_buffer_unmap (&rtp);
1675 next_seqnum = priv->next_seqnum;
1677 /* get the timestamp, this is already corrected for clock skew by the
1679 timestamp = GST_BUFFER_TIMESTAMP (outbuf);
1681 GST_DEBUG_OBJECT (jitterbuffer,
1682 "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
1683 ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
1684 rtp_jitter_buffer_num_packets (priv->jbuf));
1686 /* apply our timestamp offset to the incomming buffer, this will be our output
1688 out_time = apply_offset (jitterbuffer, timestamp);
1690 /* get the gap between this and the previous packet. If we don't know the
1691 * previous packet seqnum assume no gap. */
1692 if (G_LIKELY (next_seqnum != -1)) {
1693 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
1695 /* if we have a packet that we already pushed or considered dropped, pop it
1696 * off and get the next packet */
1697 if (G_UNLIKELY (gap < 0)) {
1698 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
1699 seqnum, next_seqnum);
1700 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1701 gst_buffer_unref (outbuf);
1705 GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
1709 /* If we don't know what the next seqnum should be (== -1) we have to wait
1710 * because it might be possible that we are not receiving this buffer in-order,
1711 * a buffer with a lower seqnum could arrive later and we want to push that
1712 * earlier buffer before this buffer then.
1713 * If we know the expected seqnum, we can compare it to the current seqnum to
1714 * determine if we have missing a packet. If we have a missing packet (which
1715 * must be before this packet) we can wait for it until the deadline for this
1716 * packet expires. */
1717 if (G_UNLIKELY (gap != 0 && out_time != -1)) {
1719 GstClockTime duration = GST_CLOCK_TIME_NONE;
1723 GST_DEBUG_OBJECT (jitterbuffer,
1724 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
1725 next_seqnum, seqnum, gap);
1727 if (priv->last_out_time != -1) {
1728 GST_DEBUG_OBJECT (jitterbuffer,
1729 "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1730 GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
1731 /* interpolate between the current time and the last time based on
1732 * number of packets we are missing, this is the estimated duration
1733 * for the missing packet based on equidistant packet spacing. Also make
1734 * sure we never go negative. */
1735 if (out_time >= priv->last_out_time)
1736 duration = (out_time - priv->last_out_time) / (gap + 1);
1740 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1741 GST_TIME_ARGS (duration));
1742 /* add this duration to the timestamp of the last packet we pushed */
1743 out_time = (priv->last_out_time + duration);
1746 /* we don't know what the next_seqnum should be, wait for the last
1747 * possible moment to push this buffer, maybe we get an earlier seqnum
1749 GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
1752 GST_OBJECT_LOCK (jitterbuffer);
1753 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1755 GST_OBJECT_UNLOCK (jitterbuffer);
1756 /* let's just push if there is no clock */
1757 GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away");
1761 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
1762 GST_TIME_ARGS (out_time));
1764 /* prepare for sync against clock */
1765 sync_time = get_sync_time (jitterbuffer, out_time);
1767 /* create an entry for the clock */
1768 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
1769 priv->unscheduled = FALSE;
1770 GST_OBJECT_UNLOCK (jitterbuffer);
1772 /* release the lock so that the other end can push stuff or unlock */
1775 ret = gst_clock_id_wait (id, NULL);
1778 /* and free the entry */
1779 gst_clock_id_unref (id);
1780 priv->clock_id = NULL;
1782 /* at this point, the clock could have been unlocked by a timeout, a new
1783 * tail element was added to the queue or because we are shutting down. Check
1784 * for shutdown first. */
1786 ((priv->srcresult != GST_FLOW_OK))
1789 /* if we got unscheduled and we are not flushing, it's because a new tail
1790 * element became available in the queue or we flushed the queue.
1791 * Grab it and try to push or sync. */
1792 if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
1793 GST_DEBUG_OBJECT (jitterbuffer,
1794 "Wait got unscheduled, will retry to push with new buffer");
1799 /* we now timed out, this means we lost a packet or finished synchronizing
1800 * on the first buffer. */
1804 /* we had a gap and thus we lost a packet. Create an event for this. */
1805 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
1809 /* update our expected next packet */
1810 priv->last_popped_seqnum = next_seqnum;
1811 priv->last_out_time = out_time;
1812 priv->next_seqnum = (next_seqnum + 1) & 0xffff;
1814 if (priv->do_lost) {
1815 /* create paket lost event */
1816 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1817 gst_structure_new ("GstRTPPacketLost",
1818 "seqnum", G_TYPE_UINT, (guint) next_seqnum,
1819 "timestamp", G_TYPE_UINT64, out_time,
1820 "duration", G_TYPE_UINT64, duration, NULL));
1823 gst_pad_push_event (priv->srcpad, event);
1824 JBUF_LOCK_CHECK (priv, flushing);
1826 /* look for next packet */
1830 /* there was no known gap,just the first packet, exit the loop and push */
1831 GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
1833 /* get new timestamp, latency might have changed */
1834 out_time = apply_offset (jitterbuffer, timestamp);
1838 /* when we get here we are ready to pop and push the buffer */
1839 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1841 check_buffering_percent (jitterbuffer, &percent);
1843 if (G_UNLIKELY (discont || priv->discont)) {
1844 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
1845 * into the jitterbuffer so we can modify now. */
1846 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
1847 priv->discont = FALSE;
1850 /* apply timestamp with offset to buffer now */
1851 GST_BUFFER_TIMESTAMP (outbuf) = out_time;
1853 /* update the elapsed time when we need to check against the npt stop time. */
1854 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
1855 && priv->clock_base != -1 && priv->clock_rate > 0) {
1856 guint64 elapsed, estimated;
1858 elapsed = compute_elapsed (jitterbuffer, outbuf);
1860 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
1863 priv->last_elapsed = elapsed;
1865 left = priv->npt_stop - priv->npt_start;
1866 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
1867 GST_TIME_ARGS (left));
1870 estimated = gst_util_uint64_scale (out_time, left, elapsed);
1872 /* if there is almost nothing left,
1873 * we may never advance enough to end up in the above case */
1874 if (left < GST_SECOND)
1875 estimated = GST_SECOND;
1880 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
1881 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
1883 priv->estimated_eos = estimated;
1887 /* now we are ready to push the buffer. Save the seqnum and release the lock
1888 * so the other end can push stuff in the queue again. */
1889 priv->last_popped_seqnum = seqnum;
1890 priv->last_out_time = out_time;
1891 priv->next_seqnum = (seqnum + 1) & 0xffff;
1895 post_buffering_percent (jitterbuffer, percent);
1898 GST_DEBUG_OBJECT (jitterbuffer,
1899 "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
1900 GST_TIME_ARGS (out_time));
1901 result = gst_pad_push (priv->srcpad, outbuf);
1902 if (G_UNLIKELY (result != GST_FLOW_OK))
1910 /* store result, we are flushing now */
1911 GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
1912 priv->srcresult = GST_FLOW_EOS;
1913 gst_pad_pause_task (priv->srcpad);
1915 gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
1920 /* store result, we are flushing now */
1921 GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
1924 g_signal_emit (jitterbuffer,
1925 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
1930 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1931 gst_pad_pause_task (priv->srcpad);
1937 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
1938 gst_flow_get_name (result));
1942 priv->srcresult = result;
1943 /* we don't post errors or anything because upstream will do that for us
1944 * when we pass the return value upstream. */
1945 gst_pad_pause_task (priv->srcpad);
1951 static GstFlowReturn
1952 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
1955 GstRtpJitterBuffer *jitterbuffer;
1956 GstRtpJitterBufferPrivate *priv;
1957 GstFlowReturn ret = GST_FLOW_OK;
1958 guint64 base_rtptime, base_time;
1960 guint64 last_rtptime;
1962 GstRTCPPacket packet;
1963 guint64 ext_rtptime, diff;
1965 gboolean drop = FALSE;
1966 GstRTCPBuffer rtcp = { NULL, };
1969 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1971 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
1972 goto invalid_buffer;
1974 priv = jitterbuffer->priv;
1976 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
1978 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
1981 /* first packet must be SR or RR or else the validate would have failed */
1982 switch (gst_rtcp_packet_get_type (&packet)) {
1983 case GST_RTCP_TYPE_SR:
1984 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
1990 gst_rtcp_buffer_unmap (&rtcp);
1992 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
1995 /* convert the RTP timestamp to our extended timestamp, using the same offset
1996 * we used in the jitterbuffer */
1997 ext_rtptime = priv->jbuf->ext_rtptime;
1998 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2000 /* get the last values from the jitterbuffer */
2001 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2002 &clock_rate, &last_rtptime);
2004 clock_base = priv->clock_base;
2006 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2007 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2008 ", clock-base %" G_GUINT64_FORMAT,
2009 ext_rtptime, base_rtptime, clock_rate, clock_base);
2011 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2012 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2015 /* we can't accept anything that happened before we did the last resync */
2016 if (base_rtptime > ext_rtptime) {
2017 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2020 /* the SR RTP timestamp must be something close to what we last observed
2021 * in the jitterbuffer */
2022 if (ext_rtptime > last_rtptime) {
2023 /* check how far ahead it is to our RTP timestamps */
2024 diff = ext_rtptime - last_rtptime;
2025 /* if bigger than 1 second, we drop it */
2026 if (diff > clock_rate) {
2027 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2028 /* should drop this, but some RTSP servers end up with bogus
2029 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2030 * so still trigger rptbin sync but invalidate RTCP data
2031 * (sync might use other methods) */
2034 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2035 G_GUINT64_FORMAT, last_rtptime, diff);
2044 s = gst_structure_new ("application/x-rtp-sync",
2045 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2046 "base-time", G_TYPE_UINT64, base_time,
2047 "clock-rate", G_TYPE_UINT, clock_rate,
2048 "clock-base", G_TYPE_UINT64, clock_base,
2049 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2050 "sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
2052 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2053 g_signal_emit (jitterbuffer,
2054 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2055 gst_structure_free (s);
2057 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2062 gst_buffer_unref (buffer);
2068 /* this is not fatal but should be filtered earlier */
2069 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2070 ("Received invalid RTCP payload, dropping"));
2076 /* this is not fatal but should be filtered earlier */
2077 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2078 ("Received empty RTCP payload, dropping"));
2079 gst_rtcp_buffer_unmap (&rtcp);
2085 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2086 gst_rtcp_buffer_unmap (&rtcp);
2093 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
2096 gboolean res = FALSE;
2098 switch (GST_QUERY_TYPE (query)) {
2099 case GST_QUERY_CAPS:
2101 GstCaps *filter, *caps;
2103 gst_query_parse_caps (query, &filter);
2104 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2105 gst_query_set_caps_result (query, caps);
2106 gst_caps_unref (caps);
2111 res = gst_pad_query_default (pad, parent, query);
2119 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
2122 GstRtpJitterBuffer *jitterbuffer;
2123 GstRtpJitterBufferPrivate *priv;
2124 gboolean res = FALSE;
2126 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2127 priv = jitterbuffer->priv;
2129 switch (GST_QUERY_TYPE (query)) {
2130 case GST_QUERY_LATENCY:
2132 /* We need to send the query upstream and add the returned latency to our
2134 GstClockTime min_latency, max_latency;
2136 GstClockTime our_latency;
2138 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2139 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2141 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2142 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2143 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2145 /* store this so that we can safely sync on the peer buffers. */
2147 priv->peer_latency = min_latency;
2148 our_latency = priv->latency_ns;
2151 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2152 GST_TIME_ARGS (our_latency));
2154 /* we add some latency but can buffer an infinite amount of time */
2155 min_latency += our_latency;
2158 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2159 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2160 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2162 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2166 case GST_QUERY_POSITION:
2168 GstClockTime start, last_out;
2171 gst_query_parse_position (query, &fmt, NULL);
2172 if (fmt != GST_FORMAT_TIME) {
2173 res = gst_pad_query_default (pad, parent, query);
2178 start = priv->npt_start;
2179 last_out = priv->last_out_time;
2182 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2183 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2184 GST_TIME_ARGS (last_out));
2186 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2187 /* bring 0-based outgoing time to stream time */
2188 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2191 res = gst_pad_query_default (pad, parent, query);
2195 case GST_QUERY_CAPS:
2197 GstCaps *filter, *caps;
2199 gst_query_parse_caps (query, &filter);
2200 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2201 gst_query_set_caps_result (query, caps);
2202 gst_caps_unref (caps);
2207 res = gst_pad_query_default (pad, parent, query);
2215 gst_rtp_jitter_buffer_set_property (GObject * object,
2216 guint prop_id, const GValue * value, GParamSpec * pspec)
2218 GstRtpJitterBuffer *jitterbuffer;
2219 GstRtpJitterBufferPrivate *priv;
2221 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2222 priv = jitterbuffer->priv;
2227 guint new_latency, old_latency;
2229 new_latency = g_value_get_uint (value);
2232 old_latency = priv->latency_ms;
2233 priv->latency_ms = new_latency;
2234 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2235 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2238 /* post message if latency changed, this will inform the parent pipeline
2239 * that a latency reconfiguration is possible/needed. */
2240 if (new_latency != old_latency) {
2241 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2242 GST_TIME_ARGS (new_latency * GST_MSECOND));
2244 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2245 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2249 case PROP_DROP_ON_LATENCY:
2251 priv->drop_on_latency = g_value_get_boolean (value);
2254 case PROP_TS_OFFSET:
2256 priv->ts_offset = g_value_get_int64 (value);
2257 /* FIXME, we don't really have a method for signaling a timestamp
2258 * DISCONT without also making this a data discont. */
2259 /* priv->discont = TRUE; */
2264 priv->do_lost = g_value_get_boolean (value);
2269 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2273 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2279 gst_rtp_jitter_buffer_get_property (GObject * object,
2280 guint prop_id, GValue * value, GParamSpec * pspec)
2282 GstRtpJitterBuffer *jitterbuffer;
2283 GstRtpJitterBufferPrivate *priv;
2285 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2286 priv = jitterbuffer->priv;
2291 g_value_set_uint (value, priv->latency_ms);
2294 case PROP_DROP_ON_LATENCY:
2296 g_value_set_boolean (value, priv->drop_on_latency);
2299 case PROP_TS_OFFSET:
2301 g_value_set_int64 (value, priv->ts_offset);
2306 g_value_set_boolean (value, priv->do_lost);
2311 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
2319 if (priv->srcresult != GST_FLOW_OK)
2322 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
2324 g_value_set_int (value, percent);
2329 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);