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 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
71 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
73 /* RTPJitterBuffer signals and args */
76 SIGNAL_REQUEST_PT_MAP,
84 #define DEFAULT_LATENCY_MS 200
85 #define DEFAULT_DROP_ON_LATENCY FALSE
86 #define DEFAULT_TS_OFFSET 0
87 #define DEFAULT_DO_LOST FALSE
88 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
89 #define DEFAULT_PERCENT 0
103 #define JBUF_LOCK(priv) (g_mutex_lock ((priv)->jbuf_lock))
105 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
107 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
111 #define JBUF_UNLOCK(priv) (g_mutex_unlock ((priv)->jbuf_lock))
112 #define JBUF_WAIT(priv) (g_cond_wait ((priv)->jbuf_cond, (priv)->jbuf_lock))
114 #define JBUF_WAIT_CHECK(priv,label) G_STMT_START { \
116 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
120 #define JBUF_SIGNAL(priv) (g_cond_signal ((priv)->jbuf_cond))
122 struct _GstRtpJitterBufferPrivate
124 GstPad *sinkpad, *srcpad;
127 RTPJitterBuffer *jbuf;
138 gboolean drop_on_latency;
142 /* the last seqnum we pushed out */
143 guint32 last_popped_seqnum;
144 /* the next expected seqnum we push */
146 /* last output time */
147 GstClockTime last_out_time;
148 /* the next expected seqnum we receive */
149 guint32 next_in_seqnum;
151 /* start and stop ranges */
152 GstClockTime npt_start;
153 GstClockTime npt_stop;
154 guint64 ext_timestamp;
155 guint64 last_elapsed;
156 guint64 estimated_eos;
158 gboolean reached_npt_stop;
163 /* clock rate and rtp timestamp offset */
167 gint64 prev_ts_offset;
169 /* when we are shutting down */
170 GstFlowReturn srcresult;
176 gboolean unscheduled;
177 /* the latency of the upstream peer, we have to take this into account when
178 * synchronizing the buffers. */
179 GstClockTime peer_latency;
181 /* some accounting */
183 guint64 num_duplicates;
186 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
187 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
188 GstRtpJitterBufferPrivate))
190 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
191 GST_STATIC_PAD_TEMPLATE ("sink",
194 GST_STATIC_CAPS ("application/x-rtp, "
195 "clock-rate = (int) [ 1, 2147483647 ]"
196 /* "payload = (int) , "
197 * "encoding-name = (string) "
201 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
202 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
205 GST_STATIC_CAPS ("application/x-rtcp")
208 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
209 GST_STATIC_PAD_TEMPLATE ("src",
212 GST_STATIC_CAPS ("application/x-rtp"
213 /* "payload = (int) , "
214 * "clock-rate = (int) , "
215 * "encoding-name = (string) "
219 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
221 GST_BOILERPLATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GstElement,
224 /* object overrides */
225 static void gst_rtp_jitter_buffer_set_property (GObject * object,
226 guint prop_id, const GValue * value, GParamSpec * pspec);
227 static void gst_rtp_jitter_buffer_get_property (GObject * object,
228 guint prop_id, GValue * value, GParamSpec * pspec);
229 static void gst_rtp_jitter_buffer_finalize (GObject * object);
231 /* element overrides */
232 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
233 * element, GstStateChange transition);
234 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
235 GstPadTemplate * templ, const gchar * name);
236 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
238 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
241 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad);
242 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad);
244 /* sinkpad overrides */
245 static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps);
246 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
248 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
251 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
253 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
256 /* srcpad overrides */
257 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
260 gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
261 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
262 static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
265 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
267 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
268 gboolean active, guint64 base_time);
271 gst_rtp_jitter_buffer_base_init (gpointer klass)
273 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
275 gst_element_class_add_pad_template (element_class,
276 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
277 gst_element_class_add_pad_template (element_class,
278 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
279 gst_element_class_add_pad_template (element_class,
280 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
282 gst_element_class_set_details_simple (element_class,
283 "RTP packet jitter-buffer", "Filter/Network/RTP",
284 "A buffer that deals with network jitter and other transmission faults",
285 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
286 "Wim Taymans <wim.taymans@gmail.com>");
290 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
292 GObjectClass *gobject_class;
293 GstElementClass *gstelement_class;
295 gobject_class = (GObjectClass *) klass;
296 gstelement_class = (GstElementClass *) klass;
298 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
300 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
302 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
303 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
306 * GstRtpJitterBuffer::latency:
308 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
309 * for at most this time.
311 g_object_class_install_property (gobject_class, PROP_LATENCY,
312 g_param_spec_uint ("latency", "Buffer latency in ms",
313 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
314 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
316 * GstRtpJitterBuffer::drop-on-latency:
318 * Drop oldest buffers when the queue is completely filled.
320 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
321 g_param_spec_boolean ("drop-on-latency",
322 "Drop buffers when maximum latency is reached",
323 "Tells the jitterbuffer to never exceed the given latency in size",
324 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
326 * GstRtpJitterBuffer::ts-offset:
328 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
329 * This is mainly used to ensure interstream synchronisation.
331 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
332 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
333 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
334 G_MAXINT64, DEFAULT_TS_OFFSET,
335 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 * GstRtpJitterBuffer::do-lost:
340 * Send out a GstRTPPacketLost event downstream when a packet is considered
343 g_object_class_install_property (gobject_class, PROP_DO_LOST,
344 g_param_spec_boolean ("do-lost", "Do Lost",
345 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
346 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 * GstRtpJitterBuffer::mode:
351 * Control the buffering and timestamping mode used by the jitterbuffer.
353 g_object_class_install_property (gobject_class, PROP_MODE,
354 g_param_spec_enum ("mode", "Mode",
355 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
356 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
358 * GstRtpJitterBuffer::percent:
360 * The percent of the jitterbuffer that is filled.
364 g_object_class_install_property (gobject_class, PROP_PERCENT,
365 g_param_spec_int ("percent", "percent",
366 "The buffer filled percent", 0, 100,
367 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
369 * GstRtpJitterBuffer::request-pt-map:
370 * @buffer: the object which received the signal
373 * Request the payload type as #GstCaps for @pt.
375 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
376 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
377 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
378 request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
379 GST_TYPE_CAPS, 1, G_TYPE_UINT);
381 * GstRtpJitterBuffer::handle-sync:
382 * @buffer: the object which received the signal
383 * @struct: a GstStructure containing sync values.
385 * Be notified of new sync values.
387 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
388 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
389 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
390 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
391 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
394 * GstRtpJitterBuffer::on-npt-stop
395 * @buffer: the object which received the signal
397 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
398 * the npt-stop position.
400 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
401 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
402 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
403 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
404 G_TYPE_NONE, 0, G_TYPE_NONE);
407 * GstRtpJitterBuffer::clear-pt-map:
408 * @buffer: the object which received the signal
410 * Invalidate the clock-rate as obtained with the
411 * #GstRtpJitterBuffer::request-pt-map signal.
413 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
414 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
415 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
416 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
417 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
420 * GstRtpJitterBuffer::set-active:
421 * @buffer: the object which received the signal
423 * Start pushing out packets with the given base time. This signal is only
424 * useful in buffering mode.
426 * Returns: the time of the last pushed packet.
430 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
431 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
432 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
433 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
434 gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
437 gstelement_class->change_state =
438 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
439 gstelement_class->request_new_pad =
440 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
441 gstelement_class->release_pad =
442 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
443 gstelement_class->provide_clock =
444 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
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,
455 GstRtpJitterBufferClass * klass)
457 GstRtpJitterBufferPrivate *priv;
459 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
460 jitterbuffer->priv = priv;
462 priv->latency_ms = DEFAULT_LATENCY_MS;
463 priv->latency_ns = priv->latency_ms * GST_MSECOND;
464 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
465 priv->do_lost = DEFAULT_DO_LOST;
467 priv->jbuf = rtp_jitter_buffer_new ();
468 priv->jbuf_lock = g_mutex_new ();
469 priv->jbuf_cond = g_cond_new ();
471 /* reset skew detection initialy */
472 rtp_jitter_buffer_reset_skew (priv->jbuf);
473 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
474 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
478 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
481 gst_pad_set_activatepush_function (priv->srcpad,
482 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_push));
483 gst_pad_set_query_function (priv->srcpad,
484 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_query));
485 gst_pad_set_getcaps_function (priv->srcpad,
486 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
487 gst_pad_set_event_function (priv->srcpad,
488 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
491 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
494 gst_pad_set_chain_function (priv->sinkpad,
495 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
496 gst_pad_set_event_function (priv->sinkpad,
497 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
498 gst_pad_set_setcaps_function (priv->sinkpad,
499 GST_DEBUG_FUNCPTR (gst_jitter_buffer_sink_setcaps));
500 gst_pad_set_getcaps_function (priv->sinkpad,
501 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
503 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
504 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
508 gst_rtp_jitter_buffer_finalize (GObject * object)
510 GstRtpJitterBuffer *jitterbuffer;
512 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
514 g_mutex_free (jitterbuffer->priv->jbuf_lock);
515 g_cond_free (jitterbuffer->priv->jbuf_cond);
517 g_object_unref (jitterbuffer->priv->jbuf);
519 G_OBJECT_CLASS (parent_class)->finalize (object);
523 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad)
525 GstRtpJitterBuffer *jitterbuffer;
526 GstPad *otherpad = NULL;
529 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
531 if (pad == jitterbuffer->priv->sinkpad) {
532 otherpad = jitterbuffer->priv->srcpad;
533 } else if (pad == jitterbuffer->priv->srcpad) {
534 otherpad = jitterbuffer->priv->sinkpad;
535 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
539 it = gst_iterator_new_single (GST_TYPE_PAD, otherpad,
540 (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref);
542 gst_object_unref (jitterbuffer);
548 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
550 GstRtpJitterBufferPrivate *priv;
552 priv = jitterbuffer->priv;
554 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
557 gst_pad_new_from_static_template
558 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
559 gst_pad_set_chain_function (priv->rtcpsinkpad,
560 gst_rtp_jitter_buffer_chain_rtcp);
561 gst_pad_set_event_function (priv->rtcpsinkpad,
562 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
563 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
564 gst_rtp_jitter_buffer_iterate_internal_links);
565 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
566 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
568 return priv->rtcpsinkpad;
572 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
574 GstRtpJitterBufferPrivate *priv;
576 priv = jitterbuffer->priv;
578 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
580 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
582 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
583 priv->rtcpsinkpad = NULL;
587 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
588 GstPadTemplate * templ, const gchar * name)
590 GstRtpJitterBuffer *jitterbuffer;
591 GstElementClass *klass;
593 GstRtpJitterBufferPrivate *priv;
595 g_return_val_if_fail (templ != NULL, NULL);
596 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
598 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
599 priv = jitterbuffer->priv;
600 klass = GST_ELEMENT_GET_CLASS (element);
602 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
604 /* figure out the template */
605 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
606 if (priv->rtcpsinkpad != NULL)
609 result = create_rtcp_sink (jitterbuffer);
618 g_warning ("gstrtpjitterbuffer: this is not our template");
623 g_warning ("gstrtpjitterbuffer: pad already requested");
629 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
631 GstRtpJitterBuffer *jitterbuffer;
632 GstRtpJitterBufferPrivate *priv;
634 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
635 g_return_if_fail (GST_IS_PAD (pad));
637 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
638 priv = jitterbuffer->priv;
640 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
642 if (priv->rtcpsinkpad == pad) {
643 remove_rtcp_sink (jitterbuffer);
652 g_warning ("gstjitterbuffer: asked to release an unknown pad");
658 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
660 return gst_system_clock_obtain ();
664 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
666 GstRtpJitterBufferPrivate *priv;
668 priv = jitterbuffer->priv;
670 /* this will trigger a new pt-map request signal, FIXME, do something better. */
673 priv->clock_rate = -1;
678 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
681 GstRtpJitterBufferPrivate *priv;
682 GstClockTime last_out;
688 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
689 active, GST_TIME_ARGS (offset));
691 if (active != priv->active) {
692 /* add the amount of time spent in paused to the output offset. All
693 * outgoing buffers will have this offset applied to their timestamps in
694 * order to make them arrive in time in the sink. */
695 priv->out_offset = offset;
696 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
697 GST_TIME_ARGS (priv->out_offset));
698 priv->active = active;
702 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
704 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
705 /* head buffer timestamp and offset gives our output time */
706 last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset;
708 /* use last known time when the buffer is empty */
709 last_out = priv->last_out_time;
717 gst_rtp_jitter_buffer_getcaps (GstPad * pad)
719 GstRtpJitterBuffer *jitterbuffer;
720 GstRtpJitterBufferPrivate *priv;
723 const GstCaps *templ;
725 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
726 priv = jitterbuffer->priv;
728 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
730 caps = gst_pad_peer_get_caps (other);
732 templ = gst_pad_get_pad_template_caps (pad);
734 GST_DEBUG_OBJECT (jitterbuffer, "copy template");
735 caps = gst_caps_copy (templ);
739 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
741 intersect = gst_caps_intersect (caps, templ);
742 gst_caps_unref (caps);
746 gst_object_unref (jitterbuffer);
752 * Must be called with JBUF_LOCK held
756 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
759 GstRtpJitterBufferPrivate *priv;
760 GstStructure *caps_struct;
764 priv = jitterbuffer->priv;
766 /* first parse the caps */
767 caps_struct = gst_caps_get_structure (caps, 0);
769 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
771 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
772 * measure the amount of data in the buffer */
773 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
776 if (priv->clock_rate <= 0)
779 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
781 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
782 * can use this to track the amount of time elapsed on the sender. */
783 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
784 priv->clock_base = val;
786 priv->clock_base = -1;
788 priv->ext_timestamp = priv->clock_base;
790 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
793 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
794 /* first expected seqnum, only update when we didn't have a previous base. */
795 if (priv->next_in_seqnum == -1)
796 priv->next_in_seqnum = val;
797 if (priv->next_seqnum == -1)
798 priv->next_seqnum = val;
801 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
803 /* the start and stop times. The seqnum-base corresponds to the start time. We
804 * will keep track of the seqnums on the output and when we reach the one
805 * corresponding to npt-stop, we emit the npt-stop-reached signal */
806 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
807 priv->npt_start = tval;
811 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
812 priv->npt_stop = tval;
816 GST_DEBUG_OBJECT (jitterbuffer,
817 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
818 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
825 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
830 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
836 gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps)
838 GstRtpJitterBuffer *jitterbuffer;
839 GstRtpJitterBufferPrivate *priv;
842 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
843 priv = jitterbuffer->priv;
846 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
849 /* set same caps on srcpad on success */
851 gst_pad_set_caps (priv->srcpad, caps);
853 gst_object_unref (jitterbuffer);
859 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
861 GstRtpJitterBufferPrivate *priv;
863 priv = jitterbuffer->priv;
866 /* mark ourselves as flushing */
867 priv->srcresult = GST_FLOW_WRONG_STATE;
868 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
869 /* this unblocks any waiting pops on the src pad task */
871 /* unlock clock, we just unschedule, the entry will be released by the
872 * locking streaming thread. */
873 if (priv->clock_id) {
874 gst_clock_id_unschedule (priv->clock_id);
875 priv->unscheduled = TRUE;
881 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
883 GstRtpJitterBufferPrivate *priv;
885 priv = jitterbuffer->priv;
888 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
889 /* Mark as non flushing */
890 priv->srcresult = GST_FLOW_OK;
891 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
892 priv->last_popped_seqnum = -1;
893 priv->last_out_time = -1;
894 priv->next_seqnum = -1;
895 priv->next_in_seqnum = -1;
896 priv->clock_rate = -1;
898 priv->estimated_eos = -1;
899 priv->last_elapsed = 0;
900 priv->reached_npt_stop = FALSE;
901 priv->ext_timestamp = -1;
902 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
903 rtp_jitter_buffer_flush (priv->jbuf);
904 rtp_jitter_buffer_reset_skew (priv->jbuf);
909 gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active)
911 gboolean result = TRUE;
912 GstRtpJitterBuffer *jitterbuffer = NULL;
914 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
917 /* allow data processing */
918 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
920 /* start pushing out buffers */
921 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
922 gst_pad_start_task (jitterbuffer->priv->srcpad,
923 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer);
925 /* make sure all data processing stops ASAP */
926 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
928 /* NOTE this will hardlock if the state change is called from the src pad
929 * task thread because we will _join() the thread. */
930 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
931 result = gst_pad_stop_task (pad);
934 gst_object_unref (jitterbuffer);
939 static GstStateChangeReturn
940 gst_rtp_jitter_buffer_change_state (GstElement * element,
941 GstStateChange transition)
943 GstRtpJitterBuffer *jitterbuffer;
944 GstRtpJitterBufferPrivate *priv;
945 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
947 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
948 priv = jitterbuffer->priv;
950 switch (transition) {
951 case GST_STATE_CHANGE_NULL_TO_READY:
953 case GST_STATE_CHANGE_READY_TO_PAUSED:
955 /* reset negotiated values */
956 priv->clock_rate = -1;
957 priv->clock_base = -1;
958 priv->peer_latency = 0;
960 /* block until we go to PLAYING */
961 priv->blocked = TRUE;
964 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
966 /* unblock to allow streaming in PLAYING */
967 priv->blocked = FALSE;
975 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
977 switch (transition) {
978 case GST_STATE_CHANGE_READY_TO_PAUSED:
979 /* we are a live element because we sync to the clock, which we can only
980 * do in the PLAYING state */
981 if (ret != GST_STATE_CHANGE_FAILURE)
982 ret = GST_STATE_CHANGE_NO_PREROLL;
984 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
986 /* block to stop streaming when PAUSED */
987 priv->blocked = TRUE;
989 if (ret != GST_STATE_CHANGE_FAILURE)
990 ret = GST_STATE_CHANGE_NO_PREROLL;
992 case GST_STATE_CHANGE_PAUSED_TO_READY:
994 case GST_STATE_CHANGE_READY_TO_NULL:
1004 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
1006 gboolean ret = TRUE;
1007 GstRtpJitterBuffer *jitterbuffer;
1008 GstRtpJitterBufferPrivate *priv;
1010 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1011 if (G_UNLIKELY (jitterbuffer == NULL)) {
1012 gst_event_unref (event);
1015 priv = jitterbuffer->priv;
1017 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1019 switch (GST_EVENT_TYPE (event)) {
1020 case GST_EVENT_LATENCY:
1022 GstClockTime latency;
1024 gst_event_parse_latency (event, &latency);
1027 /* adjust the overall buffer delay to the total pipeline latency in
1028 * buffering mode because if downstream consumes too fast (because of
1029 * large latency or queues, we would start rebuffering again. */
1030 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1031 RTP_JITTER_BUFFER_MODE_BUFFER) {
1032 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1036 ret = gst_pad_push_event (priv->sinkpad, event);
1040 ret = gst_pad_push_event (priv->sinkpad, event);
1043 gst_object_unref (jitterbuffer);
1049 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
1051 gboolean ret = TRUE;
1052 GstRtpJitterBuffer *jitterbuffer;
1053 GstRtpJitterBufferPrivate *priv;
1055 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1056 if (G_UNLIKELY (jitterbuffer == NULL)) {
1057 gst_event_unref (event);
1060 priv = jitterbuffer->priv;
1062 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1064 switch (GST_EVENT_TYPE (event)) {
1065 case GST_EVENT_NEWSEGMENT:
1068 gdouble rate, arate;
1069 gint64 start, stop, time;
1072 gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
1073 &start, &stop, &time);
1075 /* we need time for now */
1076 if (format != GST_FORMAT_TIME)
1077 goto newseg_wrong_format;
1079 GST_DEBUG_OBJECT (jitterbuffer,
1080 "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
1081 ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
1082 update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
1083 GST_TIME_ARGS (time));
1085 /* now configure the values, we need these to time the release of the
1086 * buffers on the srcpad. */
1087 gst_segment_set_newsegment_full (&priv->segment, update,
1088 rate, arate, format, start, stop, time);
1090 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1091 ret = gst_pad_push_event (priv->srcpad, event);
1094 case GST_EVENT_FLUSH_START:
1095 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1096 ret = gst_pad_push_event (priv->srcpad, event);
1098 case GST_EVENT_FLUSH_STOP:
1099 ret = gst_pad_push_event (priv->srcpad, event);
1100 ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
1104 /* push EOS in queue. We always push it at the head */
1106 /* check for flushing, we need to discard the event and return FALSE when
1107 * we are flushing */
1108 ret = priv->srcresult == GST_FLOW_OK;
1109 if (ret && !priv->eos) {
1110 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1113 } else if (priv->eos) {
1114 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1116 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1117 gst_flow_get_name (priv->srcresult));
1120 gst_event_unref (event);
1124 ret = gst_pad_push_event (priv->srcpad, event);
1129 gst_object_unref (jitterbuffer);
1134 newseg_wrong_format:
1136 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1138 gst_event_unref (event);
1144 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event)
1146 GstRtpJitterBuffer *jitterbuffer;
1148 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1150 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1152 switch (GST_EVENT_TYPE (event)) {
1153 case GST_EVENT_FLUSH_START:
1155 case GST_EVENT_FLUSH_STOP:
1160 gst_event_unref (event);
1161 gst_object_unref (jitterbuffer);
1167 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1168 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1169 * GST_FLOW_WRONG_STATE when the element is shutting down. On success
1170 * GST_FLOW_OK is returned.
1172 static GstFlowReturn
1173 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1177 GValue args[2] = { {0}, {0} };
1181 g_value_init (&args[0], GST_TYPE_ELEMENT);
1182 g_value_set_object (&args[0], jitterbuffer);
1183 g_value_init (&args[1], G_TYPE_UINT);
1184 g_value_set_uint (&args[1], pt);
1186 g_value_init (&ret, GST_TYPE_CAPS);
1187 g_value_set_boxed (&ret, NULL);
1189 JBUF_UNLOCK (jitterbuffer->priv);
1190 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1192 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1194 g_value_unset (&args[0]);
1195 g_value_unset (&args[1]);
1196 caps = (GstCaps *) g_value_dup_boxed (&ret);
1197 g_value_unset (&ret);
1201 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1202 gst_caps_unref (caps);
1204 if (G_UNLIKELY (!res))
1212 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1213 return GST_FLOW_ERROR;
1217 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1218 return GST_FLOW_WRONG_STATE;
1222 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1223 return GST_FLOW_ERROR;
1227 /* call with jbuf lock held */
1229 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1231 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1233 /* too short a stream, or too close to EOS will never really fill buffer */
1234 if (*percent != -1 && priv->npt_stop != -1 &&
1235 priv->npt_stop - priv->npt_start <=
1236 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1237 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1238 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1244 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1246 GstMessage *message;
1248 /* Post a buffering message */
1249 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1250 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1252 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1255 static GstFlowReturn
1256 gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
1258 GstRtpJitterBuffer *jitterbuffer;
1259 GstRtpJitterBufferPrivate *priv;
1261 GstFlowReturn ret = GST_FLOW_OK;
1262 GstClockTime timestamp;
1268 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1270 if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
1271 goto invalid_buffer;
1273 priv = jitterbuffer->priv;
1275 pt = gst_rtp_buffer_get_payload_type (buffer);
1277 /* take the timestamp of the buffer. This is the time when the packet was
1278 * received and is used to calculate jitter and clock skew. We will adjust
1279 * this timestamp with the smoothed value after processing it in the
1281 timestamp = GST_BUFFER_TIMESTAMP (buffer);
1282 /* bring to running time */
1283 timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
1286 seqnum = gst_rtp_buffer_get_seq (buffer);
1288 GST_DEBUG_OBJECT (jitterbuffer,
1289 "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
1290 GST_TIME_ARGS (timestamp));
1292 JBUF_LOCK_CHECK (priv, out_flushing);
1294 if (G_UNLIKELY (priv->last_pt != pt)) {
1297 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1301 /* reset clock-rate so that we get a new one */
1302 priv->clock_rate = -1;
1303 /* Try to get the clock-rate from the caps first if we can. If there are no
1304 * caps we must fire the signal to get the clock-rate. */
1305 if ((caps = GST_BUFFER_CAPS (buffer))) {
1306 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1310 if (G_UNLIKELY (priv->clock_rate == -1)) {
1311 /* no clock rate given on the caps, try to get one with the signal */
1312 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1313 pt) == GST_FLOW_WRONG_STATE)
1316 if (G_UNLIKELY (priv->clock_rate == -1))
1320 /* don't accept more data on EOS */
1321 if (G_UNLIKELY (priv->eos))
1324 /* now check against our expected seqnum */
1325 if (G_LIKELY (priv->next_in_seqnum != -1)) {
1327 gboolean reset = FALSE;
1329 gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
1330 if (G_UNLIKELY (gap != 0)) {
1331 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1332 priv->next_in_seqnum, seqnum, gap);
1333 /* priv->next_in_seqnum >= seqnum, this packet is too late or the
1334 * sender might have been restarted with different seqnum. */
1335 if (gap < -RTP_MAX_MISORDER) {
1336 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
1339 /* priv->next_in_seqnum < seqnum, this is a new packet */
1340 else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1341 GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
1345 GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
1348 if (G_UNLIKELY (reset)) {
1349 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1350 rtp_jitter_buffer_flush (priv->jbuf);
1351 rtp_jitter_buffer_reset_skew (priv->jbuf);
1352 priv->last_popped_seqnum = -1;
1353 priv->next_seqnum = seqnum;
1356 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1358 /* let's check if this buffer is too late, we can only accept packets with
1359 * bigger seqnum than the one we last pushed. */
1360 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1363 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1365 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1366 if (G_UNLIKELY (gap <= 0))
1370 /* let's drop oldest packet if the queue is already full and drop-on-latency
1371 * is set. We can only do this when there actually is a latency. When no
1372 * latency is set, we just pump it in the queue and let the other end push it
1373 * out as fast as possible. */
1374 if (priv->latency_ms && priv->drop_on_latency) {
1376 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1378 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1381 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1383 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
1384 gst_rtp_buffer_get_seq (old_buf));
1386 gst_buffer_unref (old_buf);
1390 /* we need to make the metadata writable before pushing it in the jitterbuffer
1391 * because the jitterbuffer will update the timestamp */
1392 buffer = gst_buffer_make_metadata_writable (buffer);
1394 /* now insert the packet into the queue in sorted order. This function returns
1395 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1396 * have a duplicate. */
1397 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
1398 priv->clock_rate, &tail, &percent)))
1401 /* signal addition of new buffer when the _loop is waiting. */
1405 /* let's unschedule and unblock any waiting buffers. We only want to do this
1406 * when the tail buffer changed */
1407 if (G_UNLIKELY (priv->clock_id && tail)) {
1408 GST_DEBUG_OBJECT (jitterbuffer,
1409 "Unscheduling waiting buffer, new tail buffer");
1410 gst_clock_id_unschedule (priv->clock_id);
1411 priv->unscheduled = TRUE;
1414 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
1415 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
1417 check_buffering_percent (jitterbuffer, &percent);
1423 post_buffering_percent (jitterbuffer, percent);
1425 gst_object_unref (jitterbuffer);
1432 /* this is not fatal but should be filtered earlier */
1433 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
1434 ("Received invalid RTP payload, dropping"));
1435 gst_buffer_unref (buffer);
1436 gst_object_unref (jitterbuffer);
1441 GST_WARNING_OBJECT (jitterbuffer,
1442 "No clock-rate in caps!, dropping buffer");
1443 gst_buffer_unref (buffer);
1448 ret = priv->srcresult;
1449 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
1450 gst_buffer_unref (buffer);
1455 ret = GST_FLOW_UNEXPECTED;
1456 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
1457 gst_buffer_unref (buffer);
1462 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
1463 " popped, dropping", seqnum, priv->last_popped_seqnum);
1465 gst_buffer_unref (buffer);
1470 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
1472 priv->num_duplicates++;
1473 gst_buffer_unref (buffer);
1479 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1481 GstRtpJitterBufferPrivate *priv;
1483 priv = jitterbuffer->priv;
1485 if (timestamp == -1)
1488 /* apply the timestamp offset, this is used for inter stream sync */
1489 timestamp += priv->ts_offset;
1490 /* add the offset, this is used when buffering */
1491 timestamp += priv->out_offset;
1497 get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1499 GstClockTime result;
1500 GstRtpJitterBufferPrivate *priv;
1502 priv = jitterbuffer->priv;
1504 result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
1505 /* add latency, this includes our own latency and the peer latency. */
1506 result += priv->latency_ns;
1507 result += priv->peer_latency;
1513 eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
1514 GstRtpJitterBuffer * jitterbuffer)
1516 GstRtpJitterBufferPrivate *priv;
1518 priv = jitterbuffer->priv;
1520 JBUF_LOCK_CHECK (priv, flushing);
1521 if (priv->waiting) {
1522 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
1523 priv->reached_npt_stop = TRUE;
1539 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1541 guint64 ext_time, elapsed;
1543 GstRtpJitterBufferPrivate *priv;
1545 priv = jitterbuffer->priv;
1546 rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
1548 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
1549 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
1551 if (rtp_time < priv->ext_timestamp) {
1552 ext_time = priv->ext_timestamp;
1554 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
1557 if (ext_time > priv->clock_base)
1558 elapsed = ext_time - priv->clock_base;
1562 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
1567 * This funcion will push out buffers on the source pad.
1569 * For each pushed buffer, the seqnum is recorded, if the next buffer B has a
1570 * different seqnum (missing packets before B), this function will wait for the
1571 * missing packet to arrive up to the timestamp of buffer B.
1574 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
1576 GstRtpJitterBufferPrivate *priv;
1578 GstFlowReturn result;
1580 guint32 next_seqnum;
1581 GstClockTime timestamp, out_time;
1582 gboolean discont = FALSE;
1586 GstClockTime sync_time;
1589 priv = jitterbuffer->priv;
1591 JBUF_LOCK_CHECK (priv, flushing);
1593 GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
1596 /* always wait if we are blocked */
1597 if (G_LIKELY (!priv->blocked)) {
1598 /* we're buffering but not EOS, wait. */
1599 if (!priv->eos && (!priv->active
1600 || rtp_jitter_buffer_is_buffering (priv->jbuf))) {
1601 GstClockTime elapsed, delay, left;
1603 if (priv->estimated_eos == -1)
1606 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1607 if (outbuf != NULL) {
1608 elapsed = compute_elapsed (jitterbuffer, outbuf);
1609 if (GST_BUFFER_DURATION_IS_VALID (outbuf))
1610 elapsed += GST_BUFFER_DURATION (outbuf);
1612 GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed");
1613 elapsed = priv->last_elapsed;
1616 delay = rtp_jitter_buffer_get_delay (priv->jbuf);
1618 if (priv->estimated_eos > elapsed)
1619 left = priv->estimated_eos - elapsed;
1623 GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT
1624 " estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT
1625 " delay %" GST_TIME_FORMAT,
1626 GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos),
1627 GST_TIME_ARGS (left), GST_TIME_ARGS (delay));
1631 /* if we have a packet, we can exit the loop and grab it */
1632 if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
1634 /* no packets but we are EOS, do eos logic */
1635 if (G_UNLIKELY (priv->eos))
1637 /* underrun, wait for packets or flushing now if we are expecting an EOS
1638 * timeout, set the async timer for it too */
1639 if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
1640 sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
1642 GST_OBJECT_LOCK (jitterbuffer);
1643 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1645 GST_INFO_OBJECT (jitterbuffer, "scheduling timeout");
1646 id = gst_clock_new_single_shot_id (clock, sync_time);
1647 gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
1650 GST_OBJECT_UNLOCK (jitterbuffer);
1655 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
1656 priv->waiting = TRUE;
1658 priv->waiting = FALSE;
1659 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
1662 /* unschedule any pending async notifications we might have */
1663 gst_clock_id_unschedule (id);
1664 gst_clock_id_unref (id);
1666 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
1669 if (id && priv->reached_npt_stop) {
1674 /* peek a buffer, we're just looking at the timestamp and the sequence number.
1675 * If all is fine, we'll pop and push it. If the sequence number is wrong we
1676 * wait on the timestamp. In the chain function we will unlock the wait when a
1677 * new buffer is available. The peeked buffer is valid for as long as we hold
1678 * the jitterbuffer lock. */
1679 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1681 /* get the seqnum and the next expected seqnum */
1682 seqnum = gst_rtp_buffer_get_seq (outbuf);
1683 next_seqnum = priv->next_seqnum;
1685 /* get the timestamp, this is already corrected for clock skew by the
1687 timestamp = GST_BUFFER_TIMESTAMP (outbuf);
1689 GST_DEBUG_OBJECT (jitterbuffer,
1690 "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
1691 ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
1692 rtp_jitter_buffer_num_packets (priv->jbuf));
1694 /* apply our timestamp offset to the incomming buffer, this will be our output
1696 out_time = apply_offset (jitterbuffer, timestamp);
1698 /* get the gap between this and the previous packet. If we don't know the
1699 * previous packet seqnum assume no gap. */
1700 if (G_LIKELY (next_seqnum != -1)) {
1701 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
1703 /* if we have a packet that we already pushed or considered dropped, pop it
1704 * off and get the next packet */
1705 if (G_UNLIKELY (gap < 0)) {
1706 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
1707 seqnum, next_seqnum);
1708 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1709 gst_buffer_unref (outbuf);
1713 GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
1717 /* If we don't know what the next seqnum should be (== -1) we have to wait
1718 * because it might be possible that we are not receiving this buffer in-order,
1719 * a buffer with a lower seqnum could arrive later and we want to push that
1720 * earlier buffer before this buffer then.
1721 * If we know the expected seqnum, we can compare it to the current seqnum to
1722 * determine if we have missing a packet. If we have a missing packet (which
1723 * must be before this packet) we can wait for it until the deadline for this
1724 * packet expires. */
1725 if (G_UNLIKELY (gap != 0 && out_time != -1)) {
1727 GstClockTime duration = GST_CLOCK_TIME_NONE;
1731 GST_DEBUG_OBJECT (jitterbuffer,
1732 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
1733 next_seqnum, seqnum, gap);
1735 if (priv->last_out_time != -1) {
1736 GST_DEBUG_OBJECT (jitterbuffer,
1737 "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1738 GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
1739 /* interpolate between the current time and the last time based on
1740 * number of packets we are missing, this is the estimated duration
1741 * for the missing packet based on equidistant packet spacing. Also make
1742 * sure we never go negative. */
1743 if (out_time >= priv->last_out_time)
1744 duration = (out_time - priv->last_out_time) / (gap + 1);
1748 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1749 GST_TIME_ARGS (duration));
1750 /* add this duration to the timestamp of the last packet we pushed */
1751 out_time = (priv->last_out_time + duration);
1754 /* we don't know what the next_seqnum should be, wait for the last
1755 * possible moment to push this buffer, maybe we get an earlier seqnum
1757 GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
1760 GST_OBJECT_LOCK (jitterbuffer);
1761 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1763 GST_OBJECT_UNLOCK (jitterbuffer);
1764 /* let's just push if there is no clock */
1765 GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away");
1769 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
1770 GST_TIME_ARGS (out_time));
1772 /* prepare for sync against clock */
1773 sync_time = get_sync_time (jitterbuffer, out_time);
1775 /* create an entry for the clock */
1776 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
1777 priv->unscheduled = FALSE;
1778 GST_OBJECT_UNLOCK (jitterbuffer);
1780 /* release the lock so that the other end can push stuff or unlock */
1783 ret = gst_clock_id_wait (id, NULL);
1786 /* and free the entry */
1787 gst_clock_id_unref (id);
1788 priv->clock_id = NULL;
1790 /* at this point, the clock could have been unlocked by a timeout, a new
1791 * tail element was added to the queue or because we are shutting down. Check
1792 * for shutdown first. */
1794 ((priv->srcresult != GST_FLOW_OK))
1797 /* if we got unscheduled and we are not flushing, it's because a new tail
1798 * element became available in the queue or we flushed the queue.
1799 * Grab it and try to push or sync. */
1800 if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
1801 GST_DEBUG_OBJECT (jitterbuffer,
1802 "Wait got unscheduled, will retry to push with new buffer");
1807 /* we now timed out, this means we lost a packet or finished synchronizing
1808 * on the first buffer. */
1812 /* we had a gap and thus we lost a packet. Create an event for this. */
1813 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
1817 /* update our expected next packet */
1818 priv->last_popped_seqnum = next_seqnum;
1819 priv->last_out_time = out_time;
1820 priv->next_seqnum = (next_seqnum + 1) & 0xffff;
1822 if (priv->do_lost) {
1823 /* create paket lost event */
1824 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1825 gst_structure_new ("GstRTPPacketLost",
1826 "seqnum", G_TYPE_UINT, (guint) next_seqnum,
1827 "timestamp", G_TYPE_UINT64, out_time,
1828 "duration", G_TYPE_UINT64, duration, NULL));
1831 gst_pad_push_event (priv->srcpad, event);
1832 JBUF_LOCK_CHECK (priv, flushing);
1834 /* look for next packet */
1838 /* there was no known gap,just the first packet, exit the loop and push */
1839 GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
1841 /* get new timestamp, latency might have changed */
1842 out_time = apply_offset (jitterbuffer, timestamp);
1846 /* when we get here we are ready to pop and push the buffer */
1847 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1849 check_buffering_percent (jitterbuffer, &percent);
1851 if (G_UNLIKELY (discont || priv->discont)) {
1852 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
1853 * into the jitterbuffer so we can modify now. */
1854 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
1855 priv->discont = FALSE;
1858 /* apply timestamp with offset to buffer now */
1859 GST_BUFFER_TIMESTAMP (outbuf) = out_time;
1861 /* update the elapsed time when we need to check against the npt stop time. */
1862 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
1863 && priv->clock_base != -1 && priv->clock_rate > 0) {
1864 guint64 elapsed, estimated;
1866 elapsed = compute_elapsed (jitterbuffer, outbuf);
1868 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
1871 priv->last_elapsed = elapsed;
1873 left = priv->npt_stop - priv->npt_start;
1874 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
1875 GST_TIME_ARGS (left));
1878 estimated = gst_util_uint64_scale (out_time, left, elapsed);
1880 /* if there is almost nothing left,
1881 * we may never advance enough to end up in the above case */
1882 if (left < GST_SECOND)
1883 estimated = GST_SECOND;
1888 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
1889 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
1891 priv->estimated_eos = estimated;
1895 /* now we are ready to push the buffer. Save the seqnum and release the lock
1896 * so the other end can push stuff in the queue again. */
1897 priv->last_popped_seqnum = seqnum;
1898 priv->last_out_time = out_time;
1899 priv->next_seqnum = (seqnum + 1) & 0xffff;
1903 post_buffering_percent (jitterbuffer, percent);
1906 GST_DEBUG_OBJECT (jitterbuffer,
1907 "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
1908 GST_TIME_ARGS (out_time));
1909 result = gst_pad_push (priv->srcpad, outbuf);
1910 if (G_UNLIKELY (result != GST_FLOW_OK))
1918 /* store result, we are flushing now */
1919 GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
1920 priv->srcresult = GST_FLOW_UNEXPECTED;
1921 gst_pad_pause_task (priv->srcpad);
1923 gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
1928 /* store result, we are flushing now */
1929 GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
1932 g_signal_emit (jitterbuffer,
1933 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
1938 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1939 gst_pad_pause_task (priv->srcpad);
1945 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
1946 gst_flow_get_name (result));
1950 priv->srcresult = result;
1951 /* we don't post errors or anything because upstream will do that for us
1952 * when we pass the return value upstream. */
1953 gst_pad_pause_task (priv->srcpad);
1959 static GstFlowReturn
1960 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer)
1962 GstRtpJitterBuffer *jitterbuffer;
1963 GstRtpJitterBufferPrivate *priv;
1964 GstFlowReturn ret = GST_FLOW_OK;
1965 guint64 base_rtptime, base_time;
1967 guint64 last_rtptime;
1969 GstRTCPPacket packet;
1970 guint64 ext_rtptime, diff;
1972 gboolean drop = FALSE;
1974 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1976 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
1977 goto invalid_buffer;
1979 priv = jitterbuffer->priv;
1981 if (!gst_rtcp_buffer_get_first_packet (buffer, &packet))
1982 goto invalid_buffer;
1984 /* first packet must be SR or RR or else the validate would have failed */
1985 switch (gst_rtcp_packet_get_type (&packet)) {
1986 case GST_RTCP_TYPE_SR:
1987 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
1994 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
1997 /* convert the RTP timestamp to our extended timestamp, using the same offset
1998 * we used in the jitterbuffer */
1999 ext_rtptime = priv->jbuf->ext_rtptime;
2000 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2002 /* get the last values from the jitterbuffer */
2003 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2004 &clock_rate, &last_rtptime);
2006 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2007 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT,
2008 ext_rtptime, base_rtptime, clock_rate);
2010 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2011 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2014 /* we can't accept anything that happened before we did the last resync */
2015 if (base_rtptime > ext_rtptime) {
2016 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2019 /* the SR RTP timestamp must be something close to what we last observed
2020 * in the jitterbuffer */
2021 if (ext_rtptime > last_rtptime) {
2022 /* check how far ahead it is to our RTP timestamps */
2023 diff = ext_rtptime - last_rtptime;
2024 /* if bigger than 1 second, we drop it */
2025 if (diff > clock_rate) {
2026 GST_DEBUG_OBJECT (jitterbuffer, "dropping, too far ahead");
2029 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2030 G_GUINT64_FORMAT, last_rtptime, diff);
2039 s = gst_structure_new ("application/x-rtp-sync",
2040 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2041 "base-time", G_TYPE_UINT64, base_time,
2042 "clock-rate", G_TYPE_UINT, clock_rate,
2043 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2044 "sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
2046 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2047 g_signal_emit (jitterbuffer,
2048 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2049 gst_structure_free (s);
2051 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2056 gst_buffer_unref (buffer);
2057 gst_object_unref (jitterbuffer);
2063 /* this is not fatal but should be filtered earlier */
2064 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2065 ("Received invalid RTCP payload, dropping"));
2071 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2078 gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query)
2080 GstRtpJitterBuffer *jitterbuffer;
2081 GstRtpJitterBufferPrivate *priv;
2082 gboolean res = FALSE;
2084 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
2085 if (G_UNLIKELY (jitterbuffer == NULL))
2087 priv = jitterbuffer->priv;
2089 switch (GST_QUERY_TYPE (query)) {
2090 case GST_QUERY_LATENCY:
2092 /* We need to send the query upstream and add the returned latency to our
2094 GstClockTime min_latency, max_latency;
2096 GstClockTime our_latency;
2098 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2099 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2101 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2102 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2103 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2105 /* store this so that we can safely sync on the peer buffers. */
2107 priv->peer_latency = min_latency;
2108 our_latency = priv->latency_ns;
2111 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2112 GST_TIME_ARGS (our_latency));
2114 /* we add some latency but can buffer an infinite amount of time */
2115 min_latency += our_latency;
2118 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2119 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2120 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2122 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2126 case GST_QUERY_POSITION:
2128 GstClockTime start, last_out;
2131 gst_query_parse_position (query, &fmt, NULL);
2132 if (fmt != GST_FORMAT_TIME) {
2133 res = gst_pad_query_default (pad, query);
2138 start = priv->npt_start;
2139 last_out = priv->last_out_time;
2142 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2143 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2144 GST_TIME_ARGS (last_out));
2146 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2147 /* bring 0-based outgoing time to stream time */
2148 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2151 res = gst_pad_query_default (pad, query);
2156 res = gst_pad_query_default (pad, query);
2160 gst_object_unref (jitterbuffer);
2166 gst_rtp_jitter_buffer_set_property (GObject * object,
2167 guint prop_id, const GValue * value, GParamSpec * pspec)
2169 GstRtpJitterBuffer *jitterbuffer;
2170 GstRtpJitterBufferPrivate *priv;
2172 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2173 priv = jitterbuffer->priv;
2178 guint new_latency, old_latency;
2180 new_latency = g_value_get_uint (value);
2183 old_latency = priv->latency_ms;
2184 priv->latency_ms = new_latency;
2185 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2186 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2189 /* post message if latency changed, this will inform the parent pipeline
2190 * that a latency reconfiguration is possible/needed. */
2191 if (new_latency != old_latency) {
2192 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2193 GST_TIME_ARGS (new_latency * GST_MSECOND));
2195 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2196 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2200 case PROP_DROP_ON_LATENCY:
2202 priv->drop_on_latency = g_value_get_boolean (value);
2205 case PROP_TS_OFFSET:
2207 priv->ts_offset = g_value_get_int64 (value);
2208 /* FIXME, we don't really have a method for signaling a timestamp
2209 * DISCONT without also making this a data discont. */
2210 /* priv->discont = TRUE; */
2215 priv->do_lost = g_value_get_boolean (value);
2220 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2224 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2230 gst_rtp_jitter_buffer_get_property (GObject * object,
2231 guint prop_id, GValue * value, GParamSpec * pspec)
2233 GstRtpJitterBuffer *jitterbuffer;
2234 GstRtpJitterBufferPrivate *priv;
2236 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2237 priv = jitterbuffer->priv;
2242 g_value_set_uint (value, priv->latency_ms);
2245 case PROP_DROP_ON_LATENCY:
2247 g_value_set_boolean (value, priv->drop_on_latency);
2250 case PROP_TS_OFFSET:
2252 g_value_set_int64 (value, priv->ts_offset);
2257 g_value_set_boolean (value, priv->do_lost);
2262 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
2270 if (priv->srcresult != GST_FLOW_OK)
2273 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
2275 g_value_set_int (value, percent);
2280 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);