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 GST_BOILERPLATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GstElement,
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);
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);
244 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad);
246 /* sinkpad overrides */
247 static gboolean gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps);
248 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
250 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
253 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
255 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
258 /* srcpad overrides */
259 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
262 gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active);
263 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
264 static gboolean gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query);
267 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
269 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
270 gboolean active, guint64 base_time);
273 gst_rtp_jitter_buffer_base_init (gpointer klass)
275 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
277 gst_element_class_add_static_pad_template (element_class,
278 &gst_rtp_jitter_buffer_src_template);
279 gst_element_class_add_static_pad_template (element_class,
280 &gst_rtp_jitter_buffer_sink_template);
281 gst_element_class_add_static_pad_template (element_class,
282 &gst_rtp_jitter_buffer_sink_rtcp_template);
284 gst_element_class_set_details_simple (element_class,
285 "RTP packet jitter-buffer", "Filter/Network/RTP",
286 "A buffer that deals with network jitter and other transmission faults",
287 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
288 "Wim Taymans <wim.taymans@gmail.com>");
292 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
294 GObjectClass *gobject_class;
295 GstElementClass *gstelement_class;
297 gobject_class = (GObjectClass *) klass;
298 gstelement_class = (GstElementClass *) klass;
300 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
302 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
304 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
305 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
308 * GstRtpJitterBuffer::latency:
310 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
311 * for at most this time.
313 g_object_class_install_property (gobject_class, PROP_LATENCY,
314 g_param_spec_uint ("latency", "Buffer latency in ms",
315 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
316 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
318 * GstRtpJitterBuffer::drop-on-latency:
320 * Drop oldest buffers when the queue is completely filled.
322 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
323 g_param_spec_boolean ("drop-on-latency",
324 "Drop buffers when maximum latency is reached",
325 "Tells the jitterbuffer to never exceed the given latency in size",
326 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
328 * GstRtpJitterBuffer::ts-offset:
330 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
331 * This is mainly used to ensure interstream synchronisation.
333 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
334 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
335 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
336 G_MAXINT64, DEFAULT_TS_OFFSET,
337 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
340 * GstRtpJitterBuffer::do-lost:
342 * Send out a GstRTPPacketLost event downstream when a packet is considered
345 g_object_class_install_property (gobject_class, PROP_DO_LOST,
346 g_param_spec_boolean ("do-lost", "Do Lost",
347 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
348 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
351 * GstRtpJitterBuffer::mode:
353 * Control the buffering and timestamping mode used by the jitterbuffer.
355 g_object_class_install_property (gobject_class, PROP_MODE,
356 g_param_spec_enum ("mode", "Mode",
357 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
358 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 * GstRtpJitterBuffer::percent:
362 * The percent of the jitterbuffer that is filled.
366 g_object_class_install_property (gobject_class, PROP_PERCENT,
367 g_param_spec_int ("percent", "percent",
368 "The buffer filled percent", 0, 100,
369 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
371 * GstRtpJitterBuffer::request-pt-map:
372 * @buffer: the object which received the signal
375 * Request the payload type as #GstCaps for @pt.
377 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
378 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
379 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
380 request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
381 GST_TYPE_CAPS, 1, G_TYPE_UINT);
383 * GstRtpJitterBuffer::handle-sync:
384 * @buffer: the object which received the signal
385 * @struct: a GstStructure containing sync values.
387 * Be notified of new sync values.
389 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
390 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
391 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
392 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
393 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
396 * GstRtpJitterBuffer::on-npt-stop
397 * @buffer: the object which received the signal
399 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
400 * the npt-stop position.
402 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
403 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
404 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
405 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
406 G_TYPE_NONE, 0, G_TYPE_NONE);
409 * GstRtpJitterBuffer::clear-pt-map:
410 * @buffer: the object which received the signal
412 * Invalidate the clock-rate as obtained with the
413 * #GstRtpJitterBuffer::request-pt-map signal.
415 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
416 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
417 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
418 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
419 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
422 * GstRtpJitterBuffer::set-active:
423 * @buffer: the object which received the signal
425 * Start pushing out packets with the given base time. This signal is only
426 * useful in buffering mode.
428 * Returns: the time of the last pushed packet.
432 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
433 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
434 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
435 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
436 gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
439 gstelement_class->change_state =
440 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
441 gstelement_class->request_new_pad =
442 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
443 gstelement_class->release_pad =
444 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
445 gstelement_class->provide_clock =
446 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
448 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
449 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
451 GST_DEBUG_CATEGORY_INIT
452 (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
456 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer,
457 GstRtpJitterBufferClass * klass)
459 GstRtpJitterBufferPrivate *priv;
461 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
462 jitterbuffer->priv = priv;
464 priv->latency_ms = DEFAULT_LATENCY_MS;
465 priv->latency_ns = priv->latency_ms * GST_MSECOND;
466 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
467 priv->do_lost = DEFAULT_DO_LOST;
469 priv->jbuf = rtp_jitter_buffer_new ();
470 priv->jbuf_lock = g_mutex_new ();
471 priv->jbuf_cond = g_cond_new ();
473 /* reset skew detection initialy */
474 rtp_jitter_buffer_reset_skew (priv->jbuf);
475 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
476 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
480 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
483 gst_pad_set_activatepush_function (priv->srcpad,
484 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_push));
485 gst_pad_set_query_function (priv->srcpad,
486 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_query));
487 gst_pad_set_getcaps_function (priv->srcpad,
488 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
489 gst_pad_set_event_function (priv->srcpad,
490 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
493 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
496 gst_pad_set_chain_function (priv->sinkpad,
497 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
498 gst_pad_set_event_function (priv->sinkpad,
499 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
500 gst_pad_set_setcaps_function (priv->sinkpad,
501 GST_DEBUG_FUNCPTR (gst_jitter_buffer_sink_setcaps));
502 gst_pad_set_getcaps_function (priv->sinkpad,
503 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_getcaps));
505 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
506 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
510 gst_rtp_jitter_buffer_finalize (GObject * object)
512 GstRtpJitterBuffer *jitterbuffer;
514 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
516 g_mutex_free (jitterbuffer->priv->jbuf_lock);
517 g_cond_free (jitterbuffer->priv->jbuf_cond);
519 g_object_unref (jitterbuffer->priv->jbuf);
521 G_OBJECT_CLASS (parent_class)->finalize (object);
525 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad)
527 GstRtpJitterBuffer *jitterbuffer;
528 GstPad *otherpad = NULL;
531 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
533 if (pad == jitterbuffer->priv->sinkpad) {
534 otherpad = jitterbuffer->priv->srcpad;
535 } else if (pad == jitterbuffer->priv->srcpad) {
536 otherpad = jitterbuffer->priv->sinkpad;
537 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
541 it = gst_iterator_new_single (GST_TYPE_PAD, otherpad,
542 (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref);
544 gst_object_unref (jitterbuffer);
550 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
552 GstRtpJitterBufferPrivate *priv;
554 priv = jitterbuffer->priv;
556 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
559 gst_pad_new_from_static_template
560 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
561 gst_pad_set_chain_function (priv->rtcpsinkpad,
562 gst_rtp_jitter_buffer_chain_rtcp);
563 gst_pad_set_event_function (priv->rtcpsinkpad,
564 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
565 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
566 gst_rtp_jitter_buffer_iterate_internal_links);
567 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
568 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
570 return priv->rtcpsinkpad;
574 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
576 GstRtpJitterBufferPrivate *priv;
578 priv = jitterbuffer->priv;
580 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
582 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
584 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
585 priv->rtcpsinkpad = NULL;
589 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
590 GstPadTemplate * templ, const gchar * name)
592 GstRtpJitterBuffer *jitterbuffer;
593 GstElementClass *klass;
595 GstRtpJitterBufferPrivate *priv;
597 g_return_val_if_fail (templ != NULL, NULL);
598 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
600 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
601 priv = jitterbuffer->priv;
602 klass = GST_ELEMENT_GET_CLASS (element);
604 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
606 /* figure out the template */
607 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
608 if (priv->rtcpsinkpad != NULL)
611 result = create_rtcp_sink (jitterbuffer);
620 g_warning ("gstrtpjitterbuffer: this is not our template");
625 g_warning ("gstrtpjitterbuffer: pad already requested");
631 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
633 GstRtpJitterBuffer *jitterbuffer;
634 GstRtpJitterBufferPrivate *priv;
636 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
637 g_return_if_fail (GST_IS_PAD (pad));
639 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
640 priv = jitterbuffer->priv;
642 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
644 if (priv->rtcpsinkpad == pad) {
645 remove_rtcp_sink (jitterbuffer);
654 g_warning ("gstjitterbuffer: asked to release an unknown pad");
660 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
662 return gst_system_clock_obtain ();
666 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
668 GstRtpJitterBufferPrivate *priv;
670 priv = jitterbuffer->priv;
672 /* this will trigger a new pt-map request signal, FIXME, do something better. */
675 priv->clock_rate = -1;
676 /* do not clear current content, but refresh state for new arrival */
677 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
678 rtp_jitter_buffer_reset_skew (priv->jbuf);
679 priv->last_popped_seqnum = -1;
680 priv->next_seqnum = -1;
685 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
688 GstRtpJitterBufferPrivate *priv;
689 GstClockTime last_out;
695 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
696 active, GST_TIME_ARGS (offset));
698 if (active != priv->active) {
699 /* add the amount of time spent in paused to the output offset. All
700 * outgoing buffers will have this offset applied to their timestamps in
701 * order to make them arrive in time in the sink. */
702 priv->out_offset = offset;
703 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
704 GST_TIME_ARGS (priv->out_offset));
705 priv->active = active;
709 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
711 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
712 /* head buffer timestamp and offset gives our output time */
713 last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset;
715 /* use last known time when the buffer is empty */
716 last_out = priv->last_out_time;
724 gst_rtp_jitter_buffer_getcaps (GstPad * pad)
726 GstRtpJitterBuffer *jitterbuffer;
727 GstRtpJitterBufferPrivate *priv;
730 const GstCaps *templ;
732 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
733 priv = jitterbuffer->priv;
735 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
737 caps = gst_pad_peer_get_caps (other);
739 templ = gst_pad_get_pad_template_caps (pad);
741 GST_DEBUG_OBJECT (jitterbuffer, "copy template");
742 caps = gst_caps_copy (templ);
746 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
748 intersect = gst_caps_intersect (caps, templ);
749 gst_caps_unref (caps);
753 gst_object_unref (jitterbuffer);
759 * Must be called with JBUF_LOCK held
763 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
766 GstRtpJitterBufferPrivate *priv;
767 GstStructure *caps_struct;
771 priv = jitterbuffer->priv;
773 /* first parse the caps */
774 caps_struct = gst_caps_get_structure (caps, 0);
776 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
778 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
779 * measure the amount of data in the buffer */
780 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
783 if (priv->clock_rate <= 0)
786 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
788 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
789 * can use this to track the amount of time elapsed on the sender. */
790 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
791 priv->clock_base = val;
793 priv->clock_base = -1;
795 priv->ext_timestamp = priv->clock_base;
797 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
800 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
801 /* first expected seqnum, only update when we didn't have a previous base. */
802 if (priv->next_in_seqnum == -1)
803 priv->next_in_seqnum = val;
804 if (priv->next_seqnum == -1)
805 priv->next_seqnum = val;
808 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
810 /* the start and stop times. The seqnum-base corresponds to the start time. We
811 * will keep track of the seqnums on the output and when we reach the one
812 * corresponding to npt-stop, we emit the npt-stop-reached signal */
813 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
814 priv->npt_start = tval;
818 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
819 priv->npt_stop = tval;
823 GST_DEBUG_OBJECT (jitterbuffer,
824 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
825 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
832 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
837 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
843 gst_jitter_buffer_sink_setcaps (GstPad * pad, GstCaps * caps)
845 GstRtpJitterBuffer *jitterbuffer;
846 GstRtpJitterBufferPrivate *priv;
849 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
850 priv = jitterbuffer->priv;
853 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
856 /* set same caps on srcpad on success */
858 gst_pad_set_caps (priv->srcpad, caps);
860 gst_object_unref (jitterbuffer);
866 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
868 GstRtpJitterBufferPrivate *priv;
870 priv = jitterbuffer->priv;
873 /* mark ourselves as flushing */
874 priv->srcresult = GST_FLOW_WRONG_STATE;
875 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
876 /* this unblocks any waiting pops on the src pad task */
878 /* unlock clock, we just unschedule, the entry will be released by the
879 * locking streaming thread. */
880 if (priv->clock_id) {
881 gst_clock_id_unschedule (priv->clock_id);
882 priv->unscheduled = TRUE;
888 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
890 GstRtpJitterBufferPrivate *priv;
892 priv = jitterbuffer->priv;
895 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
896 /* Mark as non flushing */
897 priv->srcresult = GST_FLOW_OK;
898 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
899 priv->last_popped_seqnum = -1;
900 priv->last_out_time = -1;
901 priv->next_seqnum = -1;
902 priv->next_in_seqnum = -1;
903 priv->clock_rate = -1;
905 priv->estimated_eos = -1;
906 priv->last_elapsed = 0;
907 priv->reached_npt_stop = FALSE;
908 priv->ext_timestamp = -1;
909 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
910 rtp_jitter_buffer_flush (priv->jbuf);
911 rtp_jitter_buffer_reset_skew (priv->jbuf);
916 gst_rtp_jitter_buffer_src_activate_push (GstPad * pad, gboolean active)
918 gboolean result = TRUE;
919 GstRtpJitterBuffer *jitterbuffer = NULL;
921 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
924 /* allow data processing */
925 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
927 /* start pushing out buffers */
928 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
929 gst_pad_start_task (jitterbuffer->priv->srcpad,
930 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer);
932 /* make sure all data processing stops ASAP */
933 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
935 /* NOTE this will hardlock if the state change is called from the src pad
936 * task thread because we will _join() the thread. */
937 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
938 result = gst_pad_stop_task (pad);
941 gst_object_unref (jitterbuffer);
946 static GstStateChangeReturn
947 gst_rtp_jitter_buffer_change_state (GstElement * element,
948 GstStateChange transition)
950 GstRtpJitterBuffer *jitterbuffer;
951 GstRtpJitterBufferPrivate *priv;
952 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
954 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
955 priv = jitterbuffer->priv;
957 switch (transition) {
958 case GST_STATE_CHANGE_NULL_TO_READY:
960 case GST_STATE_CHANGE_READY_TO_PAUSED:
962 /* reset negotiated values */
963 priv->clock_rate = -1;
964 priv->clock_base = -1;
965 priv->peer_latency = 0;
967 /* block until we go to PLAYING */
968 priv->blocked = TRUE;
971 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
973 /* unblock to allow streaming in PLAYING */
974 priv->blocked = FALSE;
982 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
984 switch (transition) {
985 case GST_STATE_CHANGE_READY_TO_PAUSED:
986 /* we are a live element because we sync to the clock, which we can only
987 * do in the PLAYING state */
988 if (ret != GST_STATE_CHANGE_FAILURE)
989 ret = GST_STATE_CHANGE_NO_PREROLL;
991 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
993 /* block to stop streaming when PAUSED */
994 priv->blocked = TRUE;
996 if (ret != GST_STATE_CHANGE_FAILURE)
997 ret = GST_STATE_CHANGE_NO_PREROLL;
999 case GST_STATE_CHANGE_PAUSED_TO_READY:
1001 case GST_STATE_CHANGE_READY_TO_NULL:
1011 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstEvent * event)
1013 gboolean ret = TRUE;
1014 GstRtpJitterBuffer *jitterbuffer;
1015 GstRtpJitterBufferPrivate *priv;
1017 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1018 if (G_UNLIKELY (jitterbuffer == NULL)) {
1019 gst_event_unref (event);
1022 priv = jitterbuffer->priv;
1024 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1026 switch (GST_EVENT_TYPE (event)) {
1027 case GST_EVENT_LATENCY:
1029 GstClockTime latency;
1031 gst_event_parse_latency (event, &latency);
1034 /* adjust the overall buffer delay to the total pipeline latency in
1035 * buffering mode because if downstream consumes too fast (because of
1036 * large latency or queues, we would start rebuffering again. */
1037 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1038 RTP_JITTER_BUFFER_MODE_BUFFER) {
1039 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1043 ret = gst_pad_push_event (priv->sinkpad, event);
1047 ret = gst_pad_push_event (priv->sinkpad, event);
1050 gst_object_unref (jitterbuffer);
1056 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstEvent * event)
1058 gboolean ret = TRUE;
1059 GstRtpJitterBuffer *jitterbuffer;
1060 GstRtpJitterBufferPrivate *priv;
1062 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1063 if (G_UNLIKELY (jitterbuffer == NULL)) {
1064 gst_event_unref (event);
1067 priv = jitterbuffer->priv;
1069 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1071 switch (GST_EVENT_TYPE (event)) {
1072 case GST_EVENT_NEWSEGMENT:
1075 gdouble rate, arate;
1076 gint64 start, stop, time;
1079 gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
1080 &start, &stop, &time);
1082 /* we need time for now */
1083 if (format != GST_FORMAT_TIME)
1084 goto newseg_wrong_format;
1086 GST_DEBUG_OBJECT (jitterbuffer,
1087 "newsegment: update %d, rate %g, arate %g, start %" GST_TIME_FORMAT
1088 ", stop %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT,
1089 update, rate, arate, GST_TIME_ARGS (start), GST_TIME_ARGS (stop),
1090 GST_TIME_ARGS (time));
1092 /* now configure the values, we need these to time the release of the
1093 * buffers on the srcpad. */
1094 gst_segment_set_newsegment_full (&priv->segment, update,
1095 rate, arate, format, start, stop, time);
1097 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1098 ret = gst_pad_push_event (priv->srcpad, event);
1101 case GST_EVENT_FLUSH_START:
1102 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1103 ret = gst_pad_push_event (priv->srcpad, event);
1105 case GST_EVENT_FLUSH_STOP:
1106 ret = gst_pad_push_event (priv->srcpad, event);
1107 ret = gst_rtp_jitter_buffer_src_activate_push (priv->srcpad, TRUE);
1111 /* push EOS in queue. We always push it at the head */
1113 /* check for flushing, we need to discard the event and return FALSE when
1114 * we are flushing */
1115 ret = priv->srcresult == GST_FLOW_OK;
1116 if (ret && !priv->eos) {
1117 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1120 } else if (priv->eos) {
1121 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1123 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1124 gst_flow_get_name (priv->srcresult));
1127 gst_event_unref (event);
1131 ret = gst_pad_push_event (priv->srcpad, event);
1136 gst_object_unref (jitterbuffer);
1141 newseg_wrong_format:
1143 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1145 gst_event_unref (event);
1151 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstEvent * event)
1153 GstRtpJitterBuffer *jitterbuffer;
1155 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1157 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1159 switch (GST_EVENT_TYPE (event)) {
1160 case GST_EVENT_FLUSH_START:
1162 case GST_EVENT_FLUSH_STOP:
1167 gst_event_unref (event);
1168 gst_object_unref (jitterbuffer);
1174 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1175 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1176 * GST_FLOW_WRONG_STATE when the element is shutting down. On success
1177 * GST_FLOW_OK is returned.
1179 static GstFlowReturn
1180 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1184 GValue args[2] = { {0}, {0} };
1188 g_value_init (&args[0], GST_TYPE_ELEMENT);
1189 g_value_set_object (&args[0], jitterbuffer);
1190 g_value_init (&args[1], G_TYPE_UINT);
1191 g_value_set_uint (&args[1], pt);
1193 g_value_init (&ret, GST_TYPE_CAPS);
1194 g_value_set_boxed (&ret, NULL);
1196 JBUF_UNLOCK (jitterbuffer->priv);
1197 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1199 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1201 g_value_unset (&args[0]);
1202 g_value_unset (&args[1]);
1203 caps = (GstCaps *) g_value_dup_boxed (&ret);
1204 g_value_unset (&ret);
1208 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1209 gst_caps_unref (caps);
1211 if (G_UNLIKELY (!res))
1219 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1220 return GST_FLOW_ERROR;
1224 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1225 return GST_FLOW_WRONG_STATE;
1229 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1230 return GST_FLOW_ERROR;
1234 /* call with jbuf lock held */
1236 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1238 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1240 /* too short a stream, or too close to EOS will never really fill buffer */
1241 if (*percent != -1 && priv->npt_stop != -1 &&
1242 priv->npt_stop - priv->npt_start <=
1243 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1244 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1245 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1251 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1253 GstMessage *message;
1255 /* Post a buffering message */
1256 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1257 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1259 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1262 static GstFlowReturn
1263 gst_rtp_jitter_buffer_chain (GstPad * pad, GstBuffer * buffer)
1265 GstRtpJitterBuffer *jitterbuffer;
1266 GstRtpJitterBufferPrivate *priv;
1268 GstFlowReturn ret = GST_FLOW_OK;
1269 GstClockTime timestamp;
1275 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1277 if (G_UNLIKELY (!gst_rtp_buffer_validate (buffer)))
1278 goto invalid_buffer;
1280 priv = jitterbuffer->priv;
1282 pt = gst_rtp_buffer_get_payload_type (buffer);
1284 /* take the timestamp of the buffer. This is the time when the packet was
1285 * received and is used to calculate jitter and clock skew. We will adjust
1286 * this timestamp with the smoothed value after processing it in the
1288 timestamp = GST_BUFFER_TIMESTAMP (buffer);
1289 /* bring to running time */
1290 timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
1293 seqnum = gst_rtp_buffer_get_seq (buffer);
1295 GST_DEBUG_OBJECT (jitterbuffer,
1296 "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
1297 GST_TIME_ARGS (timestamp));
1299 JBUF_LOCK_CHECK (priv, out_flushing);
1301 if (G_UNLIKELY (priv->last_pt != pt)) {
1304 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1308 /* reset clock-rate so that we get a new one */
1309 priv->clock_rate = -1;
1310 /* Try to get the clock-rate from the caps first if we can. If there are no
1311 * caps we must fire the signal to get the clock-rate. */
1312 if ((caps = GST_BUFFER_CAPS (buffer))) {
1313 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1317 if (G_UNLIKELY (priv->clock_rate == -1)) {
1318 /* no clock rate given on the caps, try to get one with the signal */
1319 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1320 pt) == GST_FLOW_WRONG_STATE)
1323 if (G_UNLIKELY (priv->clock_rate == -1))
1327 /* don't accept more data on EOS */
1328 if (G_UNLIKELY (priv->eos))
1331 /* now check against our expected seqnum */
1332 if (G_LIKELY (priv->next_in_seqnum != -1)) {
1334 gboolean reset = FALSE;
1336 gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
1337 if (G_UNLIKELY (gap != 0)) {
1338 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1339 priv->next_in_seqnum, seqnum, gap);
1340 /* priv->next_in_seqnum >= seqnum, this packet is too late or the
1341 * sender might have been restarted with different seqnum. */
1342 if (gap < -RTP_MAX_MISORDER) {
1343 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
1346 /* priv->next_in_seqnum < seqnum, this is a new packet */
1347 else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1348 GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
1352 GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
1355 if (G_UNLIKELY (reset)) {
1356 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1357 rtp_jitter_buffer_flush (priv->jbuf);
1358 rtp_jitter_buffer_reset_skew (priv->jbuf);
1359 priv->last_popped_seqnum = -1;
1360 priv->next_seqnum = seqnum;
1363 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1365 /* let's check if this buffer is too late, we can only accept packets with
1366 * bigger seqnum than the one we last pushed. */
1367 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1370 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1372 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1373 if (G_UNLIKELY (gap <= 0))
1377 /* let's drop oldest packet if the queue is already full and drop-on-latency
1378 * is set. We can only do this when there actually is a latency. When no
1379 * latency is set, we just pump it in the queue and let the other end push it
1380 * out as fast as possible. */
1381 if (priv->latency_ms && priv->drop_on_latency) {
1383 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1385 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1388 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1390 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet #%d",
1391 gst_rtp_buffer_get_seq (old_buf));
1393 gst_buffer_unref (old_buf);
1397 /* we need to make the metadata writable before pushing it in the jitterbuffer
1398 * because the jitterbuffer will update the timestamp */
1399 buffer = gst_buffer_make_metadata_writable (buffer);
1401 /* now insert the packet into the queue in sorted order. This function returns
1402 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1403 * have a duplicate. */
1404 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
1405 priv->clock_rate, &tail, &percent)))
1408 /* signal addition of new buffer when the _loop is waiting. */
1412 /* let's unschedule and unblock any waiting buffers. We only want to do this
1413 * when the tail buffer changed */
1414 if (G_UNLIKELY (priv->clock_id && tail)) {
1415 GST_DEBUG_OBJECT (jitterbuffer,
1416 "Unscheduling waiting buffer, new tail buffer");
1417 gst_clock_id_unschedule (priv->clock_id);
1418 priv->unscheduled = TRUE;
1421 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
1422 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
1424 check_buffering_percent (jitterbuffer, &percent);
1430 post_buffering_percent (jitterbuffer, percent);
1432 gst_object_unref (jitterbuffer);
1439 /* this is not fatal but should be filtered earlier */
1440 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
1441 ("Received invalid RTP payload, dropping"));
1442 gst_buffer_unref (buffer);
1443 gst_object_unref (jitterbuffer);
1448 GST_WARNING_OBJECT (jitterbuffer,
1449 "No clock-rate in caps!, dropping buffer");
1450 gst_buffer_unref (buffer);
1455 ret = priv->srcresult;
1456 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
1457 gst_buffer_unref (buffer);
1462 ret = GST_FLOW_UNEXPECTED;
1463 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
1464 gst_buffer_unref (buffer);
1469 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
1470 " popped, dropping", seqnum, priv->last_popped_seqnum);
1472 gst_buffer_unref (buffer);
1477 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
1479 priv->num_duplicates++;
1480 gst_buffer_unref (buffer);
1486 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1488 GstRtpJitterBufferPrivate *priv;
1490 priv = jitterbuffer->priv;
1492 if (timestamp == -1)
1495 /* apply the timestamp offset, this is used for inter stream sync */
1496 timestamp += priv->ts_offset;
1497 /* add the offset, this is used when buffering */
1498 timestamp += priv->out_offset;
1504 get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1506 GstClockTime result;
1507 GstRtpJitterBufferPrivate *priv;
1509 priv = jitterbuffer->priv;
1511 result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
1512 /* add latency, this includes our own latency and the peer latency. */
1513 result += priv->latency_ns;
1514 result += priv->peer_latency;
1520 eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
1521 GstRtpJitterBuffer * jitterbuffer)
1523 GstRtpJitterBufferPrivate *priv;
1525 priv = jitterbuffer->priv;
1527 JBUF_LOCK_CHECK (priv, flushing);
1528 if (priv->waiting) {
1529 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
1530 priv->reached_npt_stop = TRUE;
1546 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1548 guint64 ext_time, elapsed;
1550 GstRtpJitterBufferPrivate *priv;
1552 priv = jitterbuffer->priv;
1553 rtp_time = gst_rtp_buffer_get_timestamp (outbuf);
1555 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
1556 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
1558 if (rtp_time < priv->ext_timestamp) {
1559 ext_time = priv->ext_timestamp;
1561 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
1564 if (ext_time > priv->clock_base)
1565 elapsed = ext_time - priv->clock_base;
1569 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
1574 * This funcion will push out buffers on the source pad.
1576 * For each pushed buffer, the seqnum is recorded, if the next buffer B has a
1577 * different seqnum (missing packets before B), this function will wait for the
1578 * missing packet to arrive up to the timestamp of buffer B.
1581 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
1583 GstRtpJitterBufferPrivate *priv;
1585 GstFlowReturn result;
1587 guint32 next_seqnum;
1588 GstClockTime timestamp, out_time;
1589 gboolean discont = FALSE;
1593 GstClockTime sync_time;
1596 priv = jitterbuffer->priv;
1598 JBUF_LOCK_CHECK (priv, flushing);
1600 GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
1603 /* always wait if we are blocked */
1604 if (G_LIKELY (!priv->blocked)) {
1605 /* we're buffering but not EOS, wait. */
1606 if (!priv->eos && (!priv->active
1607 || rtp_jitter_buffer_is_buffering (priv->jbuf))) {
1608 GstClockTime elapsed, delay, left;
1610 if (priv->estimated_eos == -1)
1613 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1614 if (outbuf != NULL) {
1615 elapsed = compute_elapsed (jitterbuffer, outbuf);
1616 if (GST_BUFFER_DURATION_IS_VALID (outbuf))
1617 elapsed += GST_BUFFER_DURATION (outbuf);
1619 GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed");
1620 elapsed = priv->last_elapsed;
1623 delay = rtp_jitter_buffer_get_delay (priv->jbuf);
1625 if (priv->estimated_eos > elapsed)
1626 left = priv->estimated_eos - elapsed;
1630 GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT
1631 " estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT
1632 " delay %" GST_TIME_FORMAT,
1633 GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos),
1634 GST_TIME_ARGS (left), GST_TIME_ARGS (delay));
1638 /* if we have a packet, we can exit the loop and grab it */
1639 if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
1641 /* no packets but we are EOS, do eos logic */
1642 if (G_UNLIKELY (priv->eos))
1644 /* underrun, wait for packets or flushing now if we are expecting an EOS
1645 * timeout, set the async timer for it too */
1646 if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
1647 sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
1649 GST_OBJECT_LOCK (jitterbuffer);
1650 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1652 GST_INFO_OBJECT (jitterbuffer, "scheduling timeout");
1653 id = gst_clock_new_single_shot_id (clock, sync_time);
1654 gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
1657 GST_OBJECT_UNLOCK (jitterbuffer);
1662 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
1663 priv->waiting = TRUE;
1665 priv->waiting = FALSE;
1666 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
1669 /* unschedule any pending async notifications we might have */
1670 gst_clock_id_unschedule (id);
1671 gst_clock_id_unref (id);
1673 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
1676 if (id && priv->reached_npt_stop) {
1681 /* peek a buffer, we're just looking at the timestamp and the sequence number.
1682 * If all is fine, we'll pop and push it. If the sequence number is wrong we
1683 * wait on the timestamp. In the chain function we will unlock the wait when a
1684 * new buffer is available. The peeked buffer is valid for as long as we hold
1685 * the jitterbuffer lock. */
1686 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1688 /* get the seqnum and the next expected seqnum */
1689 seqnum = gst_rtp_buffer_get_seq (outbuf);
1690 next_seqnum = priv->next_seqnum;
1692 /* get the timestamp, this is already corrected for clock skew by the
1694 timestamp = GST_BUFFER_TIMESTAMP (outbuf);
1696 GST_DEBUG_OBJECT (jitterbuffer,
1697 "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
1698 ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
1699 rtp_jitter_buffer_num_packets (priv->jbuf));
1701 /* apply our timestamp offset to the incomming buffer, this will be our output
1703 out_time = apply_offset (jitterbuffer, timestamp);
1705 /* get the gap between this and the previous packet. If we don't know the
1706 * previous packet seqnum assume no gap. */
1707 if (G_LIKELY (next_seqnum != -1)) {
1708 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
1710 /* if we have a packet that we already pushed or considered dropped, pop it
1711 * off and get the next packet */
1712 if (G_UNLIKELY (gap < 0)) {
1713 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
1714 seqnum, next_seqnum);
1715 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1716 gst_buffer_unref (outbuf);
1720 GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
1724 /* If we don't know what the next seqnum should be (== -1) we have to wait
1725 * because it might be possible that we are not receiving this buffer in-order,
1726 * a buffer with a lower seqnum could arrive later and we want to push that
1727 * earlier buffer before this buffer then.
1728 * If we know the expected seqnum, we can compare it to the current seqnum to
1729 * determine if we have missing a packet. If we have a missing packet (which
1730 * must be before this packet) we can wait for it until the deadline for this
1731 * packet expires. */
1732 if (G_UNLIKELY (gap != 0 && out_time != -1)) {
1734 GstClockTime duration = GST_CLOCK_TIME_NONE;
1738 GST_DEBUG_OBJECT (jitterbuffer,
1739 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
1740 next_seqnum, seqnum, gap);
1742 if (priv->last_out_time != -1) {
1743 GST_DEBUG_OBJECT (jitterbuffer,
1744 "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1745 GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
1746 /* interpolate between the current time and the last time based on
1747 * number of packets we are missing, this is the estimated duration
1748 * for the missing packet based on equidistant packet spacing. Also make
1749 * sure we never go negative. */
1750 if (out_time >= priv->last_out_time)
1751 duration = (out_time - priv->last_out_time) / (gap + 1);
1755 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1756 GST_TIME_ARGS (duration));
1757 /* add this duration to the timestamp of the last packet we pushed */
1758 out_time = (priv->last_out_time + duration);
1761 /* we don't know what the next_seqnum should be, wait for the last
1762 * possible moment to push this buffer, maybe we get an earlier seqnum
1764 GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
1767 GST_OBJECT_LOCK (jitterbuffer);
1768 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1770 GST_OBJECT_UNLOCK (jitterbuffer);
1771 /* let's just push if there is no clock */
1772 GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away");
1776 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT,
1777 GST_TIME_ARGS (out_time));
1779 /* prepare for sync against clock */
1780 sync_time = get_sync_time (jitterbuffer, out_time);
1782 /* create an entry for the clock */
1783 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
1784 priv->unscheduled = FALSE;
1785 GST_OBJECT_UNLOCK (jitterbuffer);
1787 /* release the lock so that the other end can push stuff or unlock */
1790 ret = gst_clock_id_wait (id, NULL);
1793 /* and free the entry */
1794 gst_clock_id_unref (id);
1795 priv->clock_id = NULL;
1797 /* at this point, the clock could have been unlocked by a timeout, a new
1798 * tail element was added to the queue or because we are shutting down. Check
1799 * for shutdown first. */
1801 ((priv->srcresult != GST_FLOW_OK))
1804 /* if we got unscheduled and we are not flushing, it's because a new tail
1805 * element became available in the queue or we flushed the queue.
1806 * Grab it and try to push or sync. */
1807 if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
1808 GST_DEBUG_OBJECT (jitterbuffer,
1809 "Wait got unscheduled, will retry to push with new buffer");
1814 /* we now timed out, this means we lost a packet or finished synchronizing
1815 * on the first buffer. */
1819 /* we had a gap and thus we lost a packet. Create an event for this. */
1820 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
1824 /* update our expected next packet */
1825 priv->last_popped_seqnum = next_seqnum;
1826 priv->last_out_time = out_time;
1827 priv->next_seqnum = (next_seqnum + 1) & 0xffff;
1829 if (priv->do_lost) {
1830 /* create paket lost event */
1831 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1832 gst_structure_new ("GstRTPPacketLost",
1833 "seqnum", G_TYPE_UINT, (guint) next_seqnum,
1834 "timestamp", G_TYPE_UINT64, out_time,
1835 "duration", G_TYPE_UINT64, duration, NULL));
1838 gst_pad_push_event (priv->srcpad, event);
1839 JBUF_LOCK_CHECK (priv, flushing);
1841 /* look for next packet */
1845 /* there was no known gap,just the first packet, exit the loop and push */
1846 GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
1848 /* get new timestamp, latency might have changed */
1849 out_time = apply_offset (jitterbuffer, timestamp);
1853 /* when we get here we are ready to pop and push the buffer */
1854 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1856 check_buffering_percent (jitterbuffer, &percent);
1858 if (G_UNLIKELY (discont || priv->discont)) {
1859 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
1860 * into the jitterbuffer so we can modify now. */
1861 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
1862 priv->discont = FALSE;
1865 /* apply timestamp with offset to buffer now */
1866 GST_BUFFER_TIMESTAMP (outbuf) = out_time;
1868 /* update the elapsed time when we need to check against the npt stop time. */
1869 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
1870 && priv->clock_base != -1 && priv->clock_rate > 0) {
1871 guint64 elapsed, estimated;
1873 elapsed = compute_elapsed (jitterbuffer, outbuf);
1875 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
1878 priv->last_elapsed = elapsed;
1880 left = priv->npt_stop - priv->npt_start;
1881 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
1882 GST_TIME_ARGS (left));
1885 estimated = gst_util_uint64_scale (out_time, left, elapsed);
1887 /* if there is almost nothing left,
1888 * we may never advance enough to end up in the above case */
1889 if (left < GST_SECOND)
1890 estimated = GST_SECOND;
1895 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
1896 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
1898 priv->estimated_eos = estimated;
1902 /* now we are ready to push the buffer. Save the seqnum and release the lock
1903 * so the other end can push stuff in the queue again. */
1904 priv->last_popped_seqnum = seqnum;
1905 priv->last_out_time = out_time;
1906 priv->next_seqnum = (seqnum + 1) & 0xffff;
1910 post_buffering_percent (jitterbuffer, percent);
1913 GST_DEBUG_OBJECT (jitterbuffer,
1914 "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
1915 GST_TIME_ARGS (out_time));
1916 result = gst_pad_push (priv->srcpad, outbuf);
1917 if (G_UNLIKELY (result != GST_FLOW_OK))
1925 /* store result, we are flushing now */
1926 GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
1927 priv->srcresult = GST_FLOW_UNEXPECTED;
1928 gst_pad_pause_task (priv->srcpad);
1930 gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
1935 /* store result, we are flushing now */
1936 GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
1939 g_signal_emit (jitterbuffer,
1940 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
1945 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1946 gst_pad_pause_task (priv->srcpad);
1952 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
1953 gst_flow_get_name (result));
1957 priv->srcresult = result;
1958 /* we don't post errors or anything because upstream will do that for us
1959 * when we pass the return value upstream. */
1960 gst_pad_pause_task (priv->srcpad);
1966 static GstFlowReturn
1967 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstBuffer * buffer)
1969 GstRtpJitterBuffer *jitterbuffer;
1970 GstRtpJitterBufferPrivate *priv;
1971 GstFlowReturn ret = GST_FLOW_OK;
1972 guint64 base_rtptime, base_time;
1974 guint64 last_rtptime;
1976 GstRTCPPacket packet;
1977 guint64 ext_rtptime, diff;
1979 gboolean drop = FALSE;
1982 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1984 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
1985 goto invalid_buffer;
1987 priv = jitterbuffer->priv;
1989 if (!gst_rtcp_buffer_get_first_packet (buffer, &packet))
1990 goto invalid_buffer;
1992 /* first packet must be SR or RR or else the validate would have failed */
1993 switch (gst_rtcp_packet_get_type (&packet)) {
1994 case GST_RTCP_TYPE_SR:
1995 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
2002 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
2005 /* convert the RTP timestamp to our extended timestamp, using the same offset
2006 * we used in the jitterbuffer */
2007 ext_rtptime = priv->jbuf->ext_rtptime;
2008 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2010 /* get the last values from the jitterbuffer */
2011 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2012 &clock_rate, &last_rtptime);
2014 clock_base = priv->clock_base;
2016 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2017 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2018 ", clock-base %" G_GUINT64_FORMAT,
2019 ext_rtptime, base_rtptime, clock_rate, clock_base);
2021 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2022 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2025 /* we can't accept anything that happened before we did the last resync */
2026 if (base_rtptime > ext_rtptime) {
2027 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2030 /* the SR RTP timestamp must be something close to what we last observed
2031 * in the jitterbuffer */
2032 if (ext_rtptime > last_rtptime) {
2033 /* check how far ahead it is to our RTP timestamps */
2034 diff = ext_rtptime - last_rtptime;
2035 /* if bigger than 1 second, we drop it */
2036 if (diff > clock_rate) {
2037 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2038 /* should drop this, but some RTSP servers end up with bogus
2039 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2040 * so still trigger rptbin sync but invalidate RTCP data
2041 * (sync might use other methods) */
2044 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2045 G_GUINT64_FORMAT, last_rtptime, diff);
2054 s = gst_structure_new ("application/x-rtp-sync",
2055 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2056 "base-time", G_TYPE_UINT64, base_time,
2057 "clock-rate", G_TYPE_UINT, clock_rate,
2058 "clock-base", G_TYPE_UINT64, clock_base,
2059 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2060 "sr-buffer", GST_TYPE_BUFFER, buffer, NULL);
2062 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2063 g_signal_emit (jitterbuffer,
2064 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2065 gst_structure_free (s);
2067 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2072 gst_buffer_unref (buffer);
2073 gst_object_unref (jitterbuffer);
2079 /* this is not fatal but should be filtered earlier */
2080 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2081 ("Received invalid RTCP payload, dropping"));
2087 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2094 gst_rtp_jitter_buffer_query (GstPad * pad, GstQuery * query)
2096 GstRtpJitterBuffer *jitterbuffer;
2097 GstRtpJitterBufferPrivate *priv;
2098 gboolean res = FALSE;
2100 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
2101 if (G_UNLIKELY (jitterbuffer == NULL))
2103 priv = jitterbuffer->priv;
2105 switch (GST_QUERY_TYPE (query)) {
2106 case GST_QUERY_LATENCY:
2108 /* We need to send the query upstream and add the returned latency to our
2110 GstClockTime min_latency, max_latency;
2112 GstClockTime our_latency;
2114 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2115 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2117 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2118 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2119 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2121 /* store this so that we can safely sync on the peer buffers. */
2123 priv->peer_latency = min_latency;
2124 our_latency = priv->latency_ns;
2127 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2128 GST_TIME_ARGS (our_latency));
2130 /* we add some latency but can buffer an infinite amount of time */
2131 min_latency += our_latency;
2134 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2135 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2136 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2138 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2142 case GST_QUERY_POSITION:
2144 GstClockTime start, last_out;
2147 gst_query_parse_position (query, &fmt, NULL);
2148 if (fmt != GST_FORMAT_TIME) {
2149 res = gst_pad_query_default (pad, query);
2154 start = priv->npt_start;
2155 last_out = priv->last_out_time;
2158 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2159 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2160 GST_TIME_ARGS (last_out));
2162 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2163 /* bring 0-based outgoing time to stream time */
2164 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2167 res = gst_pad_query_default (pad, query);
2172 res = gst_pad_query_default (pad, query);
2176 gst_object_unref (jitterbuffer);
2182 gst_rtp_jitter_buffer_set_property (GObject * object,
2183 guint prop_id, const GValue * value, GParamSpec * pspec)
2185 GstRtpJitterBuffer *jitterbuffer;
2186 GstRtpJitterBufferPrivate *priv;
2188 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2189 priv = jitterbuffer->priv;
2194 guint new_latency, old_latency;
2196 new_latency = g_value_get_uint (value);
2199 old_latency = priv->latency_ms;
2200 priv->latency_ms = new_latency;
2201 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2202 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2205 /* post message if latency changed, this will inform the parent pipeline
2206 * that a latency reconfiguration is possible/needed. */
2207 if (new_latency != old_latency) {
2208 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2209 GST_TIME_ARGS (new_latency * GST_MSECOND));
2211 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2212 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2216 case PROP_DROP_ON_LATENCY:
2218 priv->drop_on_latency = g_value_get_boolean (value);
2221 case PROP_TS_OFFSET:
2223 priv->ts_offset = g_value_get_int64 (value);
2224 /* FIXME, we don't really have a method for signaling a timestamp
2225 * DISCONT without also making this a data discont. */
2226 /* priv->discont = TRUE; */
2231 priv->do_lost = g_value_get_boolean (value);
2236 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2240 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2246 gst_rtp_jitter_buffer_get_property (GObject * object,
2247 guint prop_id, GValue * value, GParamSpec * pspec)
2249 GstRtpJitterBuffer *jitterbuffer;
2250 GstRtpJitterBufferPrivate *priv;
2252 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2253 priv = jitterbuffer->priv;
2258 g_value_set_uint (value, priv->latency_ms);
2261 case PROP_DROP_ON_LATENCY:
2263 g_value_set_boolean (value, priv->drop_on_latency);
2266 case PROP_TS_OFFSET:
2268 g_value_set_int64 (value, priv->ts_offset);
2273 g_value_set_boolean (value, priv->do_lost);
2278 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
2286 if (priv->srcresult != GST_FLOW_OK)
2289 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
2291 g_value_set_int (value, percent);
2296 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);