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., 51 Franklin St, Fifth Floor,
22 * Boston, MA 02110-1301, 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-1.0 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;
185 /* some accounting */
187 guint64 num_duplicates;
190 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
191 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
192 GstRtpJitterBufferPrivate))
194 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
195 GST_STATIC_PAD_TEMPLATE ("sink",
198 GST_STATIC_CAPS ("application/x-rtp, "
199 "clock-rate = (int) [ 1, 2147483647 ]"
200 /* "payload = (int) , "
201 * "encoding-name = (string) "
205 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
206 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
209 GST_STATIC_CAPS ("application/x-rtcp")
212 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
213 GST_STATIC_PAD_TEMPLATE ("src",
216 GST_STATIC_CAPS ("application/x-rtp"
217 /* "payload = (int) , "
218 * "clock-rate = (int) , "
219 * "encoding-name = (string) "
223 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
225 #define gst_rtp_jitter_buffer_parent_class parent_class
226 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
228 /* object overrides */
229 static void gst_rtp_jitter_buffer_set_property (GObject * object,
230 guint prop_id, const GValue * value, GParamSpec * pspec);
231 static void gst_rtp_jitter_buffer_get_property (GObject * object,
232 guint prop_id, GValue * value, GParamSpec * pspec);
233 static void gst_rtp_jitter_buffer_finalize (GObject * object);
235 /* element overrides */
236 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
237 * element, GstStateChange transition);
238 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
239 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
240 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
242 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
245 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
246 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
249 /* sinkpad overrides */
250 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
251 GstObject * parent, GstEvent * event);
252 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
253 GstObject * parent, GstBuffer * buffer);
255 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
256 GstObject * parent, GstEvent * event);
257 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
258 GstObject * parent, GstBuffer * buffer);
260 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
261 GstObject * parent, GstQuery * query);
263 /* srcpad overrides */
264 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
265 GstObject * parent, GstEvent * event);
266 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
267 GstObject * parent, GstPadMode mode, gboolean active);
268 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
269 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
270 GstObject * parent, GstQuery * query);
273 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
275 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
276 gboolean active, guint64 base_time);
279 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
281 GObjectClass *gobject_class;
282 GstElementClass *gstelement_class;
284 gobject_class = (GObjectClass *) klass;
285 gstelement_class = (GstElementClass *) klass;
287 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
289 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
291 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
292 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
295 * GstRtpJitterBuffer::latency:
297 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
298 * for at most this time.
300 g_object_class_install_property (gobject_class, PROP_LATENCY,
301 g_param_spec_uint ("latency", "Buffer latency in ms",
302 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
303 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
305 * GstRtpJitterBuffer::drop-on-latency:
307 * Drop oldest buffers when the queue is completely filled.
309 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
310 g_param_spec_boolean ("drop-on-latency",
311 "Drop buffers when maximum latency is reached",
312 "Tells the jitterbuffer to never exceed the given latency in size",
313 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
315 * GstRtpJitterBuffer::ts-offset:
317 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
318 * This is mainly used to ensure interstream synchronisation.
320 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
321 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
322 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
323 G_MAXINT64, DEFAULT_TS_OFFSET,
324 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
327 * GstRtpJitterBuffer::do-lost:
329 * Send out a GstRTPPacketLost event downstream when a packet is considered
332 g_object_class_install_property (gobject_class, PROP_DO_LOST,
333 g_param_spec_boolean ("do-lost", "Do Lost",
334 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
335 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 * GstRtpJitterBuffer::mode:
340 * Control the buffering and timestamping mode used by the jitterbuffer.
342 g_object_class_install_property (gobject_class, PROP_MODE,
343 g_param_spec_enum ("mode", "Mode",
344 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
345 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
347 * GstRtpJitterBuffer::percent:
349 * The percent of the jitterbuffer that is filled.
353 g_object_class_install_property (gobject_class, PROP_PERCENT,
354 g_param_spec_int ("percent", "percent",
355 "The buffer filled percent", 0, 100,
356 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
358 * GstRtpJitterBuffer::request-pt-map:
359 * @buffer: the object which received the signal
362 * Request the payload type as #GstCaps for @pt.
364 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
365 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
366 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
367 request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
368 GST_TYPE_CAPS, 1, G_TYPE_UINT);
370 * GstRtpJitterBuffer::handle-sync:
371 * @buffer: the object which received the signal
372 * @struct: a GstStructure containing sync values.
374 * Be notified of new sync values.
376 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
377 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
378 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
379 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
380 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
383 * GstRtpJitterBuffer::on-npt-stop
384 * @buffer: the object which received the signal
386 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
387 * the npt-stop position.
389 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
390 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
391 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
392 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
393 G_TYPE_NONE, 0, G_TYPE_NONE);
396 * GstRtpJitterBuffer::clear-pt-map:
397 * @buffer: the object which received the signal
399 * Invalidate the clock-rate as obtained with the
400 * #GstRtpJitterBuffer::request-pt-map signal.
402 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
403 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
404 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
405 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
406 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
409 * GstRtpJitterBuffer::set-active:
410 * @buffer: the object which received the signal
412 * Start pushing out packets with the given base time. This signal is only
413 * useful in buffering mode.
415 * Returns: the time of the last pushed packet.
419 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
420 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
421 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
422 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
423 gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
426 gstelement_class->change_state =
427 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
428 gstelement_class->request_new_pad =
429 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
430 gstelement_class->release_pad =
431 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
432 gstelement_class->provide_clock =
433 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
435 gst_element_class_add_pad_template (gstelement_class,
436 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
437 gst_element_class_add_pad_template (gstelement_class,
438 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
439 gst_element_class_add_pad_template (gstelement_class,
440 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
442 gst_element_class_set_static_metadata (gstelement_class,
443 "RTP packet jitter-buffer", "Filter/Network/RTP",
444 "A buffer that deals with network jitter and other transmission faults",
445 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
446 "Wim Taymans <wim.taymans@gmail.com>");
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)
458 GstRtpJitterBufferPrivate *priv;
460 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
461 jitterbuffer->priv = priv;
463 priv->latency_ms = DEFAULT_LATENCY_MS;
464 priv->latency_ns = priv->latency_ms * GST_MSECOND;
465 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
466 priv->do_lost = DEFAULT_DO_LOST;
468 priv->jbuf = rtp_jitter_buffer_new ();
469 g_mutex_init (&priv->jbuf_lock);
470 g_cond_init (&priv->jbuf_cond);
472 /* reset skew detection initialy */
473 rtp_jitter_buffer_reset_skew (priv->jbuf);
474 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
475 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
479 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
482 gst_pad_set_activatemode_function (priv->srcpad,
483 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
484 gst_pad_set_query_function (priv->srcpad,
485 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
486 gst_pad_set_event_function (priv->srcpad,
487 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
490 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
493 gst_pad_set_chain_function (priv->sinkpad,
494 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
495 gst_pad_set_event_function (priv->sinkpad,
496 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
497 gst_pad_set_query_function (priv->sinkpad,
498 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
500 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
501 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
503 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
507 gst_rtp_jitter_buffer_finalize (GObject * object)
509 GstRtpJitterBuffer *jitterbuffer;
511 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
513 g_mutex_clear (&jitterbuffer->priv->jbuf_lock);
514 g_cond_clear (&jitterbuffer->priv->jbuf_cond);
516 g_object_unref (jitterbuffer->priv->jbuf);
518 G_OBJECT_CLASS (parent_class)->finalize (object);
522 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
524 GstRtpJitterBuffer *jitterbuffer;
525 GstPad *otherpad = NULL;
529 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
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 g_value_init (&val, GST_TYPE_PAD);
540 g_value_set_object (&val, otherpad);
541 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
542 g_value_unset (&val);
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, const GstCaps * filter)
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;
674 /* do not clear current content, but refresh state for new arrival */
675 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
676 rtp_jitter_buffer_reset_skew (priv->jbuf);
677 priv->last_popped_seqnum = -1;
678 priv->next_seqnum = -1;
683 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
686 GstRtpJitterBufferPrivate *priv;
687 GstClockTime last_out;
693 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
694 active, GST_TIME_ARGS (offset));
696 if (active != priv->active) {
697 /* add the amount of time spent in paused to the output offset. All
698 * outgoing buffers will have this offset applied to their timestamps in
699 * order to make them arrive in time in the sink. */
700 priv->out_offset = offset;
701 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
702 GST_TIME_ARGS (priv->out_offset));
703 priv->active = active;
707 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
709 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
710 /* head buffer timestamp and offset gives our output time */
711 last_out = GST_BUFFER_TIMESTAMP (head) + priv->ts_offset;
713 /* use last known time when the buffer is empty */
714 last_out = priv->last_out_time;
722 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
724 GstRtpJitterBuffer *jitterbuffer;
725 GstRtpJitterBufferPrivate *priv;
730 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
731 priv = jitterbuffer->priv;
733 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
735 caps = gst_pad_peer_query_caps (other, filter);
737 templ = gst_pad_get_pad_template_caps (pad);
739 GST_DEBUG_OBJECT (jitterbuffer, "use template");
744 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
746 intersect = gst_caps_intersect (caps, templ);
747 gst_caps_unref (caps);
748 gst_caps_unref (templ);
752 gst_object_unref (jitterbuffer);
758 * Must be called with JBUF_LOCK held
762 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
765 GstRtpJitterBufferPrivate *priv;
766 GstStructure *caps_struct;
770 priv = jitterbuffer->priv;
772 /* first parse the caps */
773 caps_struct = gst_caps_get_structure (caps, 0);
775 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
777 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
778 * measure the amount of data in the buffer */
779 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
782 if (priv->clock_rate <= 0)
785 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
787 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
788 * can use this to track the amount of time elapsed on the sender. */
789 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
790 priv->clock_base = val;
792 priv->clock_base = -1;
794 priv->ext_timestamp = priv->clock_base;
796 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
799 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
800 /* first expected seqnum, only update when we didn't have a previous base. */
801 if (priv->next_in_seqnum == -1)
802 priv->next_in_seqnum = val;
803 if (priv->next_seqnum == -1)
804 priv->next_seqnum = val;
807 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
809 /* the start and stop times. The seqnum-base corresponds to the start time. We
810 * will keep track of the seqnums on the output and when we reach the one
811 * corresponding to npt-stop, we emit the npt-stop-reached signal */
812 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
813 priv->npt_start = tval;
817 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
818 priv->npt_stop = tval;
822 GST_DEBUG_OBJECT (jitterbuffer,
823 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
824 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
831 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
836 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
842 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
844 GstRtpJitterBufferPrivate *priv;
846 priv = jitterbuffer->priv;
849 /* mark ourselves as flushing */
850 priv->srcresult = GST_FLOW_FLUSHING;
851 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
852 /* this unblocks any waiting pops on the src pad task */
854 /* unlock clock, we just unschedule, the entry will be released by the
855 * locking streaming thread. */
856 if (priv->clock_id) {
857 gst_clock_id_unschedule (priv->clock_id);
858 priv->unscheduled = TRUE;
864 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
866 GstRtpJitterBufferPrivate *priv;
868 priv = jitterbuffer->priv;
871 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
872 /* Mark as non flushing */
873 priv->srcresult = GST_FLOW_OK;
874 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
875 priv->last_popped_seqnum = -1;
876 priv->last_out_time = -1;
877 priv->next_seqnum = -1;
878 priv->next_in_seqnum = -1;
879 priv->clock_rate = -1;
881 priv->estimated_eos = -1;
882 priv->last_elapsed = 0;
883 priv->reached_npt_stop = FALSE;
884 priv->ext_timestamp = -1;
885 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
886 rtp_jitter_buffer_flush (priv->jbuf);
887 rtp_jitter_buffer_reset_skew (priv->jbuf);
892 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
893 GstPadMode mode, gboolean active)
896 GstRtpJitterBuffer *jitterbuffer = NULL;
898 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
901 case GST_PAD_MODE_PUSH:
903 /* allow data processing */
904 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
906 /* start pushing out buffers */
907 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
908 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
909 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
911 /* make sure all data processing stops ASAP */
912 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
914 /* NOTE this will hardlock if the state change is called from the src pad
915 * task thread because we will _join() the thread. */
916 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
917 result = gst_pad_stop_task (pad);
927 static GstStateChangeReturn
928 gst_rtp_jitter_buffer_change_state (GstElement * element,
929 GstStateChange transition)
931 GstRtpJitterBuffer *jitterbuffer;
932 GstRtpJitterBufferPrivate *priv;
933 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
935 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
936 priv = jitterbuffer->priv;
938 switch (transition) {
939 case GST_STATE_CHANGE_NULL_TO_READY:
941 case GST_STATE_CHANGE_READY_TO_PAUSED:
943 /* reset negotiated values */
944 priv->clock_rate = -1;
945 priv->clock_base = -1;
946 priv->peer_latency = 0;
948 /* block until we go to PLAYING */
949 priv->blocked = TRUE;
952 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
954 /* unblock to allow streaming in PLAYING */
955 priv->blocked = FALSE;
963 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
965 switch (transition) {
966 case GST_STATE_CHANGE_READY_TO_PAUSED:
967 /* we are a live element because we sync to the clock, which we can only
968 * do in the PLAYING state */
969 if (ret != GST_STATE_CHANGE_FAILURE)
970 ret = GST_STATE_CHANGE_NO_PREROLL;
972 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
974 /* block to stop streaming when PAUSED */
975 priv->blocked = TRUE;
977 if (ret != GST_STATE_CHANGE_FAILURE)
978 ret = GST_STATE_CHANGE_NO_PREROLL;
980 case GST_STATE_CHANGE_PAUSED_TO_READY:
981 gst_buffer_replace (&priv->last_sr, NULL);
983 case GST_STATE_CHANGE_READY_TO_NULL:
993 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
997 GstRtpJitterBuffer *jitterbuffer;
998 GstRtpJitterBufferPrivate *priv;
1000 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1001 priv = jitterbuffer->priv;
1003 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1005 switch (GST_EVENT_TYPE (event)) {
1006 case GST_EVENT_LATENCY:
1008 GstClockTime latency;
1010 gst_event_parse_latency (event, &latency);
1013 /* adjust the overall buffer delay to the total pipeline latency in
1014 * buffering mode because if downstream consumes too fast (because of
1015 * large latency or queues, we would start rebuffering again. */
1016 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1017 RTP_JITTER_BUFFER_MODE_BUFFER) {
1018 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1022 ret = gst_pad_push_event (priv->sinkpad, event);
1026 ret = gst_pad_push_event (priv->sinkpad, event);
1034 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1037 gboolean ret = TRUE;
1038 GstRtpJitterBuffer *jitterbuffer;
1039 GstRtpJitterBufferPrivate *priv;
1041 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1042 priv = jitterbuffer->priv;
1044 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1046 switch (GST_EVENT_TYPE (event)) {
1047 case GST_EVENT_CAPS:
1051 gst_event_parse_caps (event, &caps);
1054 ret = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1057 /* set same caps on srcpad on success */
1059 ret = gst_pad_push_event (priv->srcpad, event);
1061 gst_event_unref (event);
1064 case GST_EVENT_SEGMENT:
1066 gst_event_copy_segment (event, &priv->segment);
1068 /* we need time for now */
1069 if (priv->segment.format != GST_FORMAT_TIME)
1070 goto newseg_wrong_format;
1072 GST_DEBUG_OBJECT (jitterbuffer,
1073 "newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
1075 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1076 ret = gst_pad_push_event (priv->srcpad, event);
1079 case GST_EVENT_FLUSH_START:
1080 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1081 ret = gst_pad_push_event (priv->srcpad, event);
1083 case GST_EVENT_FLUSH_STOP:
1084 ret = gst_pad_push_event (priv->srcpad, event);
1086 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1087 GST_PAD_MODE_PUSH, TRUE);
1091 /* push EOS in queue. We always push it at the head */
1093 /* check for flushing, we need to discard the event and return FALSE when
1094 * we are flushing */
1095 ret = priv->srcresult == GST_FLOW_OK;
1096 if (ret && !priv->eos) {
1097 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1100 } else if (priv->eos) {
1101 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1103 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1104 gst_flow_get_name (priv->srcresult));
1107 gst_event_unref (event);
1111 ret = gst_pad_push_event (priv->srcpad, event);
1120 newseg_wrong_format:
1122 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1124 gst_event_unref (event);
1130 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1133 gboolean ret = TRUE;
1134 GstRtpJitterBuffer *jitterbuffer;
1136 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1138 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1140 switch (GST_EVENT_TYPE (event)) {
1141 case GST_EVENT_FLUSH_START:
1142 gst_event_unref (event);
1144 case GST_EVENT_FLUSH_STOP:
1145 gst_event_unref (event);
1148 ret = gst_pad_event_default (pad, parent, event);
1156 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1157 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1158 * GST_FLOW_FLUSHING when the element is shutting down. On success
1159 * GST_FLOW_OK is returned.
1161 static GstFlowReturn
1162 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1166 GValue args[2] = { {0}, {0} };
1170 g_value_init (&args[0], GST_TYPE_ELEMENT);
1171 g_value_set_object (&args[0], jitterbuffer);
1172 g_value_init (&args[1], G_TYPE_UINT);
1173 g_value_set_uint (&args[1], pt);
1175 g_value_init (&ret, GST_TYPE_CAPS);
1176 g_value_set_boxed (&ret, NULL);
1178 JBUF_UNLOCK (jitterbuffer->priv);
1179 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1181 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1183 g_value_unset (&args[0]);
1184 g_value_unset (&args[1]);
1185 caps = (GstCaps *) g_value_dup_boxed (&ret);
1186 g_value_unset (&ret);
1190 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1191 gst_caps_unref (caps);
1193 if (G_UNLIKELY (!res))
1201 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1202 return GST_FLOW_ERROR;
1206 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1207 return GST_FLOW_FLUSHING;
1211 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1212 return GST_FLOW_ERROR;
1216 /* call with jbuf lock held */
1218 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1220 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1222 /* too short a stream, or too close to EOS will never really fill buffer */
1223 if (*percent != -1 && priv->npt_stop != -1 &&
1224 priv->npt_stop - priv->npt_start <=
1225 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1226 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1227 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1233 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1235 GstMessage *message;
1237 /* Post a buffering message */
1238 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1239 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1241 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1244 static GstFlowReturn
1245 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
1248 GstRtpJitterBuffer *jitterbuffer;
1249 GstRtpJitterBufferPrivate *priv;
1251 GstFlowReturn ret = GST_FLOW_OK;
1252 GstClockTime timestamp;
1257 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1259 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1261 priv = jitterbuffer->priv;
1263 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
1264 goto invalid_buffer;
1266 pt = gst_rtp_buffer_get_payload_type (&rtp);
1267 seqnum = gst_rtp_buffer_get_seq (&rtp);
1268 gst_rtp_buffer_unmap (&rtp);
1270 /* take the timestamp of the buffer. This is the time when the packet was
1271 * received and is used to calculate jitter and clock skew. We will adjust
1272 * this timestamp with the smoothed value after processing it in the
1274 timestamp = GST_BUFFER_TIMESTAMP (buffer);
1275 /* bring to running time */
1276 timestamp = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME,
1279 GST_DEBUG_OBJECT (jitterbuffer,
1280 "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
1281 GST_TIME_ARGS (timestamp));
1283 JBUF_LOCK_CHECK (priv, out_flushing);
1285 if (G_UNLIKELY (priv->last_pt != pt)) {
1288 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1292 /* reset clock-rate so that we get a new one */
1293 priv->clock_rate = -1;
1295 /* Try to get the clock-rate from the caps first if we can. If there are no
1296 * caps we must fire the signal to get the clock-rate. */
1297 if ((caps = gst_pad_get_current_caps (pad))) {
1298 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1299 gst_caps_unref (caps);
1303 if (G_UNLIKELY (priv->clock_rate == -1)) {
1304 /* no clock rate given on the caps, try to get one with the signal */
1305 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1306 pt) == GST_FLOW_FLUSHING)
1309 if (G_UNLIKELY (priv->clock_rate == -1))
1313 /* don't accept more data on EOS */
1314 if (G_UNLIKELY (priv->eos))
1317 /* now check against our expected seqnum */
1318 if (G_LIKELY (priv->next_in_seqnum != -1)) {
1320 gboolean reset = FALSE;
1322 gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
1323 if (G_UNLIKELY (gap != 0)) {
1324 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1325 priv->next_in_seqnum, seqnum, gap);
1326 /* priv->next_in_seqnum >= seqnum, this packet is too late or the
1327 * sender might have been restarted with different seqnum. */
1328 if (gap < -RTP_MAX_MISORDER) {
1329 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
1332 /* priv->next_in_seqnum < seqnum, this is a new packet */
1333 else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1334 GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
1338 GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
1341 if (G_UNLIKELY (reset)) {
1342 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1343 rtp_jitter_buffer_flush (priv->jbuf);
1344 rtp_jitter_buffer_reset_skew (priv->jbuf);
1345 priv->last_popped_seqnum = -1;
1346 priv->next_seqnum = seqnum;
1349 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1351 /* let's check if this buffer is too late, we can only accept packets with
1352 * bigger seqnum than the one we last pushed. */
1353 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1356 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1358 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1359 if (G_UNLIKELY (gap <= 0))
1363 /* let's drop oldest packet if the queue is already full and drop-on-latency
1364 * is set. We can only do this when there actually is a latency. When no
1365 * latency is set, we just pump it in the queue and let the other end push it
1366 * out as fast as possible. */
1367 if (priv->latency_ms && priv->drop_on_latency) {
1369 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1371 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1374 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1376 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
1379 gst_buffer_unref (old_buf);
1383 /* we need to make the metadata writable before pushing it in the jitterbuffer
1384 * because the jitterbuffer will update the timestamp */
1385 buffer = gst_buffer_make_writable (buffer);
1387 /* now insert the packet into the queue in sorted order. This function returns
1388 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1389 * have a duplicate. */
1390 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, timestamp,
1391 priv->clock_rate, &tail, &percent)))
1394 /* signal addition of new buffer when the _loop is waiting. */
1398 /* let's unschedule and unblock any waiting buffers. We only want to do this
1399 * when the tail buffer changed */
1400 if (G_UNLIKELY (priv->clock_id && tail)) {
1401 GST_DEBUG_OBJECT (jitterbuffer,
1402 "Unscheduling waiting buffer, new tail buffer");
1403 gst_clock_id_unschedule (priv->clock_id);
1404 priv->unscheduled = TRUE;
1407 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
1408 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
1410 check_buffering_percent (jitterbuffer, &percent);
1416 post_buffering_percent (jitterbuffer, percent);
1423 /* this is not fatal but should be filtered earlier */
1424 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
1425 ("Received invalid RTP payload, dropping"));
1426 gst_buffer_unref (buffer);
1431 GST_WARNING_OBJECT (jitterbuffer,
1432 "No clock-rate in caps!, dropping buffer");
1433 gst_buffer_unref (buffer);
1438 ret = priv->srcresult;
1439 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
1440 gst_buffer_unref (buffer);
1446 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
1447 gst_buffer_unref (buffer);
1452 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
1453 " popped, dropping", seqnum, priv->last_popped_seqnum);
1455 gst_buffer_unref (buffer);
1460 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
1462 priv->num_duplicates++;
1463 gst_buffer_unref (buffer);
1469 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1471 GstRtpJitterBufferPrivate *priv;
1473 priv = jitterbuffer->priv;
1475 if (timestamp == -1)
1478 /* apply the timestamp offset, this is used for inter stream sync */
1479 timestamp += priv->ts_offset;
1480 /* add the offset, this is used when buffering */
1481 timestamp += priv->out_offset;
1487 get_sync_time (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1489 GstClockTime result;
1490 GstRtpJitterBufferPrivate *priv;
1492 priv = jitterbuffer->priv;
1494 result = timestamp + GST_ELEMENT_CAST (jitterbuffer)->base_time;
1495 /* add latency, this includes our own latency and the peer latency. */
1496 result += priv->latency_ns;
1497 result += priv->peer_latency;
1503 eos_reached (GstClock * clock, GstClockTime time, GstClockID id,
1504 GstRtpJitterBuffer * jitterbuffer)
1506 GstRtpJitterBufferPrivate *priv;
1508 priv = jitterbuffer->priv;
1510 JBUF_LOCK_CHECK (priv, flushing);
1511 if (priv->waiting) {
1512 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
1513 priv->reached_npt_stop = TRUE;
1529 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1531 guint64 ext_time, elapsed;
1533 GstRtpJitterBufferPrivate *priv;
1534 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1536 priv = jitterbuffer->priv;
1537 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1538 rtp_time = gst_rtp_buffer_get_timestamp (&rtp);
1539 gst_rtp_buffer_unmap (&rtp);
1541 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
1542 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
1544 if (rtp_time < priv->ext_timestamp) {
1545 ext_time = priv->ext_timestamp;
1547 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
1550 if (ext_time > priv->clock_base)
1551 elapsed = ext_time - priv->clock_base;
1555 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
1560 * This funcion will push out buffers on the source pad.
1562 * For each pushed buffer, the seqnum is recorded, if the next buffer B has a
1563 * different seqnum (missing packets before B), this function will wait for the
1564 * missing packet to arrive up to the timestamp of buffer B.
1567 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
1569 GstRtpJitterBufferPrivate *priv;
1571 GstFlowReturn result;
1573 guint32 next_seqnum;
1574 GstClockTime timestamp, out_time;
1575 gboolean discont = FALSE;
1579 GstClockTime sync_time;
1581 GstRTPBuffer rtp = { NULL, };
1583 priv = jitterbuffer->priv;
1585 JBUF_LOCK_CHECK (priv, flushing);
1587 GST_DEBUG_OBJECT (jitterbuffer, "Peeking item");
1590 /* always wait if we are blocked */
1591 if (G_LIKELY (!priv->blocked)) {
1592 /* we're buffering but not EOS, wait. */
1593 if (!priv->eos && (!priv->active
1594 || rtp_jitter_buffer_is_buffering (priv->jbuf))) {
1595 GstClockTime elapsed, delay, left;
1597 if (priv->estimated_eos == -1)
1600 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1601 if (outbuf != NULL) {
1602 elapsed = compute_elapsed (jitterbuffer, outbuf);
1603 if (GST_BUFFER_DURATION_IS_VALID (outbuf))
1604 elapsed += GST_BUFFER_DURATION (outbuf);
1606 GST_INFO_OBJECT (jitterbuffer, "no buffer, using last_elapsed");
1607 elapsed = priv->last_elapsed;
1610 delay = rtp_jitter_buffer_get_delay (priv->jbuf);
1612 if (priv->estimated_eos > elapsed)
1613 left = priv->estimated_eos - elapsed;
1617 GST_INFO_OBJECT (jitterbuffer, "buffering, elapsed %" GST_TIME_FORMAT
1618 " estimated_eos %" GST_TIME_FORMAT " left %" GST_TIME_FORMAT
1619 " delay %" GST_TIME_FORMAT,
1620 GST_TIME_ARGS (elapsed), GST_TIME_ARGS (priv->estimated_eos),
1621 GST_TIME_ARGS (left), GST_TIME_ARGS (delay));
1625 /* if we have a packet, we can exit the loop and grab it */
1626 if (rtp_jitter_buffer_num_packets (priv->jbuf) > 0)
1628 /* no packets but we are EOS, do eos logic */
1629 if (G_UNLIKELY (priv->eos))
1631 /* underrun, wait for packets or flushing now if we are expecting an EOS
1632 * timeout, set the async timer for it too */
1633 if (priv->estimated_eos != -1 && !priv->reached_npt_stop) {
1634 sync_time = get_sync_time (jitterbuffer, priv->estimated_eos);
1636 GST_OBJECT_LOCK (jitterbuffer);
1637 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1639 GST_INFO_OBJECT (jitterbuffer, "scheduling timeout");
1640 id = gst_clock_new_single_shot_id (clock, sync_time);
1641 gst_clock_id_wait_async (id, (GstClockCallback) eos_reached,
1642 jitterbuffer, NULL);
1644 GST_OBJECT_UNLOCK (jitterbuffer);
1649 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
1650 priv->waiting = TRUE;
1652 priv->waiting = FALSE;
1653 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
1656 /* unschedule any pending async notifications we might have */
1657 gst_clock_id_unschedule (id);
1658 gst_clock_id_unref (id);
1660 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK))
1663 if (id && priv->reached_npt_stop) {
1668 /* peek a buffer, we're just looking at the timestamp and the sequence number.
1669 * If all is fine, we'll pop and push it. If the sequence number is wrong we
1670 * wait on the timestamp. In the chain function we will unlock the wait when a
1671 * new buffer is available. The peeked buffer is valid for as long as we hold
1672 * the jitterbuffer lock. */
1673 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1675 /* get the seqnum and the next expected seqnum */
1676 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1677 seqnum = gst_rtp_buffer_get_seq (&rtp);
1678 gst_rtp_buffer_unmap (&rtp);
1679 next_seqnum = priv->next_seqnum;
1681 /* get the timestamp, this is already corrected for clock skew by the
1683 timestamp = GST_BUFFER_TIMESTAMP (outbuf);
1685 GST_DEBUG_OBJECT (jitterbuffer,
1686 "Peeked buffer #%d, expect #%d, timestamp %" GST_TIME_FORMAT
1687 ", now %d left", seqnum, next_seqnum, GST_TIME_ARGS (timestamp),
1688 rtp_jitter_buffer_num_packets (priv->jbuf));
1690 /* apply our timestamp offset to the incomming buffer, this will be our output
1692 out_time = apply_offset (jitterbuffer, timestamp);
1694 /* get the gap between this and the previous packet. If we don't know the
1695 * previous packet seqnum assume no gap. */
1696 if (G_LIKELY (next_seqnum != -1)) {
1697 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
1699 /* if we have a packet that we already pushed or considered dropped, pop it
1700 * off and get the next packet */
1701 if (G_UNLIKELY (gap < 0)) {
1702 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
1703 seqnum, next_seqnum);
1704 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1705 gst_buffer_unref (outbuf);
1709 GST_DEBUG_OBJECT (jitterbuffer, "no next seqnum known, first packet");
1713 /* If we don't know what the next seqnum should be (== -1) we have to wait
1714 * because it might be possible that we are not receiving this buffer in-order,
1715 * a buffer with a lower seqnum could arrive later and we want to push that
1716 * earlier buffer before this buffer then.
1717 * If we know the expected seqnum, we can compare it to the current seqnum to
1718 * determine if we have missing a packet. If we have a missing packet (which
1719 * must be before this packet) we can wait for it until the deadline for this
1720 * packet expires. */
1721 if (G_UNLIKELY (gap != 0 && out_time != -1)) {
1723 GstClockTime duration = GST_CLOCK_TIME_NONE;
1724 GstClockTimeDiff clock_jitter;
1725 guint32 lost_packets = 1;
1726 gboolean lost_packets_late = FALSE;
1730 GST_DEBUG_OBJECT (jitterbuffer,
1731 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
1732 next_seqnum, seqnum, gap);
1734 if (priv->last_out_time != -1) {
1735 GST_DEBUG_OBJECT (jitterbuffer,
1736 "out_time %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1737 GST_TIME_ARGS (out_time), GST_TIME_ARGS (priv->last_out_time));
1738 /* interpolate between the current time and the last time based on
1739 * number of packets we are missing, this is the estimated duration
1740 * for the missing packet based on equidistant packet spacing. Also make
1741 * sure we never go negative. */
1742 if (out_time >= priv->last_out_time)
1743 duration = (out_time - priv->last_out_time) / (gap + 1);
1747 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1748 GST_TIME_ARGS (duration));
1749 /* add this duration to the timestamp of the last packet we pushed */
1750 out_time = (priv->last_out_time + duration);
1753 /* we don't know what the next_seqnum should be, wait for the last
1754 * possible moment to push this buffer, maybe we get an earlier seqnum
1756 GST_DEBUG_OBJECT (jitterbuffer, "First buffer %d, do sync", seqnum);
1759 GST_OBJECT_LOCK (jitterbuffer);
1760 clock = GST_ELEMENT_CLOCK (jitterbuffer);
1762 GST_OBJECT_UNLOCK (jitterbuffer);
1763 /* let's just push if there is no clock */
1764 GST_DEBUG_OBJECT (jitterbuffer, "No clock, push right away");
1768 /* prepare for sync against clock */
1769 sync_time = get_sync_time (jitterbuffer, out_time);
1771 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
1772 " with sync time %" GST_TIME_FORMAT,
1773 GST_TIME_ARGS (out_time), GST_TIME_ARGS (sync_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, &clock_jitter);
1785 if (ret == GST_CLOCK_EARLY && gap > 0
1786 && clock_jitter > (priv->latency_ns + priv->peer_latency)) {
1787 GstClockTimeDiff total_duration;
1788 GstClockTime out_time_diff;
1790 out_time_diff = apply_offset (jitterbuffer, timestamp) - out_time;
1791 total_duration = MIN (out_time_diff, clock_jitter);
1794 lost_packets = total_duration / duration;
1797 total_duration = lost_packets * duration;
1799 GST_DEBUG_OBJECT (jitterbuffer,
1800 "Current sync_time has expired a long time ago (+%" GST_TIME_FORMAT
1801 ") Cover up %d lost packets with duration %" GST_TIME_FORMAT,
1802 GST_TIME_ARGS (clock_jitter),
1803 lost_packets, GST_TIME_ARGS (total_duration));
1805 duration = total_duration;
1806 lost_packets_late = TRUE;
1810 /* and free the entry */
1811 gst_clock_id_unref (id);
1812 priv->clock_id = NULL;
1814 /* at this point, the clock could have been unlocked by a timeout, a new
1815 * tail element was added to the queue or because we are shutting down. Check
1816 * for shutdown first. */
1818 ((priv->srcresult != GST_FLOW_OK))
1821 /* if we got unscheduled and we are not flushing, it's because a new tail
1822 * element became available in the queue or we flushed the queue.
1823 * Grab it and try to push or sync. */
1824 if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
1825 GST_DEBUG_OBJECT (jitterbuffer,
1826 "Wait got unscheduled, will retry to push with new buffer");
1831 /* we now timed out, this means we lost a packet or finished synchronizing
1832 * on the first buffer. */
1836 /* we had a gap and thus we lost some packets. Create an event for this. */
1837 if (lost_packets > 1)
1838 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", next_seqnum,
1839 next_seqnum + lost_packets - 1);
1841 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", next_seqnum);
1843 priv->num_late += lost_packets;
1846 /* update our expected next packet */
1847 priv->last_popped_seqnum = next_seqnum;
1848 priv->last_out_time += duration;
1849 priv->next_seqnum = (next_seqnum + lost_packets) & 0xffff;
1851 if (priv->do_lost) {
1852 /* create paket lost event */
1853 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1854 gst_structure_new ("GstRTPPacketLost",
1855 "seqnum", G_TYPE_UINT, (guint) next_seqnum,
1856 "timestamp", G_TYPE_UINT64, out_time,
1857 "duration", G_TYPE_UINT64, duration,
1858 "late", G_TYPE_BOOLEAN, lost_packets_late, NULL));
1860 gst_pad_push_event (priv->srcpad, event);
1861 JBUF_LOCK_CHECK (priv, flushing);
1863 /* look for next packet */
1867 /* there was no known gap,just the first packet, exit the loop and push */
1868 GST_DEBUG_OBJECT (jitterbuffer, "First packet #%d synced", seqnum);
1870 /* get new timestamp, latency might have changed */
1871 out_time = apply_offset (jitterbuffer, timestamp);
1875 /* when we get here we are ready to pop and push the buffer */
1876 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1878 check_buffering_percent (jitterbuffer, &percent);
1880 if (G_UNLIKELY (discont || priv->discont)) {
1881 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
1882 * into the jitterbuffer so we can modify now. */
1883 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
1884 priv->discont = FALSE;
1887 /* apply timestamp with offset to buffer now */
1888 GST_BUFFER_PTS (outbuf) = out_time;
1889 GST_BUFFER_DTS (outbuf) = out_time;
1891 /* update the elapsed time when we need to check against the npt stop time. */
1892 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
1893 && priv->clock_base != -1 && priv->clock_rate > 0) {
1894 guint64 elapsed, estimated;
1896 elapsed = compute_elapsed (jitterbuffer, outbuf);
1898 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
1901 priv->last_elapsed = elapsed;
1903 left = priv->npt_stop - priv->npt_start;
1904 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
1905 GST_TIME_ARGS (left));
1908 estimated = gst_util_uint64_scale (out_time, left, elapsed);
1910 /* if there is almost nothing left,
1911 * we may never advance enough to end up in the above case */
1912 if (left < GST_SECOND)
1913 estimated = GST_SECOND;
1918 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
1919 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
1921 priv->estimated_eos = estimated;
1925 /* now we are ready to push the buffer. Save the seqnum and release the lock
1926 * so the other end can push stuff in the queue again. */
1927 priv->last_popped_seqnum = seqnum;
1928 priv->last_out_time = out_time;
1929 priv->next_seqnum = (seqnum + 1) & 0xffff;
1933 post_buffering_percent (jitterbuffer, percent);
1936 GST_DEBUG_OBJECT (jitterbuffer,
1937 "Pushing buffer %d, timestamp %" GST_TIME_FORMAT, seqnum,
1938 GST_TIME_ARGS (out_time));
1939 result = gst_pad_push (priv->srcpad, outbuf);
1940 if (G_UNLIKELY (result != GST_FLOW_OK))
1948 /* store result, we are flushing now */
1949 GST_DEBUG_OBJECT (jitterbuffer, "We are EOS, pushing EOS downstream");
1950 priv->srcresult = GST_FLOW_EOS;
1951 gst_pad_pause_task (priv->srcpad);
1953 gst_pad_push_event (priv->srcpad, gst_event_new_eos ());
1958 /* store result, we are flushing now */
1959 GST_DEBUG_OBJECT (jitterbuffer, "We reached the NPT stop");
1962 g_signal_emit (jitterbuffer,
1963 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP], 0, NULL);
1968 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1969 gst_pad_pause_task (priv->srcpad);
1975 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
1976 gst_flow_get_name (result));
1980 priv->srcresult = result;
1981 /* we don't post errors or anything because upstream will do that for us
1982 * when we pass the return value upstream. */
1983 gst_pad_pause_task (priv->srcpad);
1989 /* collect the info form the lastest RTCP packet and the jittebuffer sync, do
1990 * some sanity checks and then emit the handle-sync signal with the parameters.
1991 * This function must be called with the LOCK */
1993 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
1995 GstRtpJitterBufferPrivate *priv;
1996 guint64 base_rtptime, base_time;
1998 guint64 last_rtptime;
2000 guint64 ext_rtptime, diff;
2001 gboolean drop = FALSE;
2003 priv = jitterbuffer->priv;
2005 if (priv->last_sr == NULL) {
2006 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no SR RTCP");
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;
2015 ext_rtptime = priv->ext_rtptime;
2017 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2018 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2019 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
2020 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
2022 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2023 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2026 /* we can't accept anything that happened before we did the last resync */
2027 if (base_rtptime > ext_rtptime) {
2028 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2031 /* the SR RTP timestamp must be something close to what we last observed
2032 * in the jitterbuffer */
2033 if (ext_rtptime > last_rtptime) {
2034 /* check how far ahead it is to our RTP timestamps */
2035 diff = ext_rtptime - last_rtptime;
2036 /* if bigger than 1 second, we drop it */
2037 if (diff > clock_rate) {
2038 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2039 /* should drop this, but some RTSP servers end up with bogus
2040 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2041 * so still trigger rptbin sync but invalidate RTCP data
2042 * (sync might use other methods) */
2045 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2046 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, priv->last_sr, NULL);
2062 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2064 g_signal_emit (jitterbuffer,
2065 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2067 gst_structure_free (s);
2069 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2073 static GstFlowReturn
2074 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
2077 GstRtpJitterBuffer *jitterbuffer;
2078 GstRtpJitterBufferPrivate *priv;
2079 GstFlowReturn ret = GST_FLOW_OK;
2081 GstRTCPPacket packet;
2082 guint64 ext_rtptime;
2084 GstRTCPBuffer rtcp = { NULL, };
2086 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2088 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
2089 goto invalid_buffer;
2091 priv = jitterbuffer->priv;
2093 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2095 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
2098 /* first packet must be SR or RR or else the validate would have failed */
2099 switch (gst_rtcp_packet_get_type (&packet)) {
2100 case GST_RTCP_TYPE_SR:
2101 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
2107 gst_rtcp_buffer_unmap (&rtcp);
2109 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
2112 /* convert the RTP timestamp to our extended timestamp, using the same offset
2113 * we used in the jitterbuffer */
2114 ext_rtptime = priv->jbuf->ext_rtptime;
2115 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2117 priv->ext_rtptime = ext_rtptime;
2118 gst_buffer_replace (&priv->last_sr, buffer);
2120 do_handle_sync (jitterbuffer);
2124 gst_buffer_unref (buffer);
2130 /* this is not fatal but should be filtered earlier */
2131 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2132 ("Received invalid RTCP payload, dropping"));
2138 /* this is not fatal but should be filtered earlier */
2139 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2140 ("Received empty RTCP payload, dropping"));
2141 gst_rtcp_buffer_unmap (&rtcp);
2147 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2148 gst_rtcp_buffer_unmap (&rtcp);
2155 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
2158 gboolean res = FALSE;
2160 switch (GST_QUERY_TYPE (query)) {
2161 case GST_QUERY_CAPS:
2163 GstCaps *filter, *caps;
2165 gst_query_parse_caps (query, &filter);
2166 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2167 gst_query_set_caps_result (query, caps);
2168 gst_caps_unref (caps);
2173 if (GST_QUERY_IS_SERIALIZED (query)) {
2174 GST_WARNING_OBJECT (pad, "unhandled serialized query");
2177 res = gst_pad_query_default (pad, parent, query);
2185 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
2188 GstRtpJitterBuffer *jitterbuffer;
2189 GstRtpJitterBufferPrivate *priv;
2190 gboolean res = FALSE;
2192 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2193 priv = jitterbuffer->priv;
2195 switch (GST_QUERY_TYPE (query)) {
2196 case GST_QUERY_LATENCY:
2198 /* We need to send the query upstream and add the returned latency to our
2200 GstClockTime min_latency, max_latency;
2202 GstClockTime our_latency;
2204 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2205 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2207 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2208 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2209 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2211 /* store this so that we can safely sync on the peer buffers. */
2213 priv->peer_latency = min_latency;
2214 our_latency = priv->latency_ns;
2217 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2218 GST_TIME_ARGS (our_latency));
2220 /* we add some latency but can buffer an infinite amount of time */
2221 min_latency += our_latency;
2224 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2225 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2226 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2228 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2232 case GST_QUERY_POSITION:
2234 GstClockTime start, last_out;
2237 gst_query_parse_position (query, &fmt, NULL);
2238 if (fmt != GST_FORMAT_TIME) {
2239 res = gst_pad_query_default (pad, parent, query);
2244 start = priv->npt_start;
2245 last_out = priv->last_out_time;
2248 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2249 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2250 GST_TIME_ARGS (last_out));
2252 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2253 /* bring 0-based outgoing time to stream time */
2254 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2257 res = gst_pad_query_default (pad, parent, query);
2261 case GST_QUERY_CAPS:
2263 GstCaps *filter, *caps;
2265 gst_query_parse_caps (query, &filter);
2266 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2267 gst_query_set_caps_result (query, caps);
2268 gst_caps_unref (caps);
2273 res = gst_pad_query_default (pad, parent, query);
2281 gst_rtp_jitter_buffer_set_property (GObject * object,
2282 guint prop_id, const GValue * value, GParamSpec * pspec)
2284 GstRtpJitterBuffer *jitterbuffer;
2285 GstRtpJitterBufferPrivate *priv;
2287 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2288 priv = jitterbuffer->priv;
2293 guint new_latency, old_latency;
2295 new_latency = g_value_get_uint (value);
2298 old_latency = priv->latency_ms;
2299 priv->latency_ms = new_latency;
2300 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2301 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2304 /* post message if latency changed, this will inform the parent pipeline
2305 * that a latency reconfiguration is possible/needed. */
2306 if (new_latency != old_latency) {
2307 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2308 GST_TIME_ARGS (new_latency * GST_MSECOND));
2310 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2311 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2315 case PROP_DROP_ON_LATENCY:
2317 priv->drop_on_latency = g_value_get_boolean (value);
2320 case PROP_TS_OFFSET:
2322 priv->ts_offset = g_value_get_int64 (value);
2323 /* FIXME, we don't really have a method for signaling a timestamp
2324 * DISCONT without also making this a data discont. */
2325 /* priv->discont = TRUE; */
2330 priv->do_lost = g_value_get_boolean (value);
2335 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2339 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2345 gst_rtp_jitter_buffer_get_property (GObject * object,
2346 guint prop_id, GValue * value, GParamSpec * pspec)
2348 GstRtpJitterBuffer *jitterbuffer;
2349 GstRtpJitterBufferPrivate *priv;
2351 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2352 priv = jitterbuffer->priv;
2357 g_value_set_uint (value, priv->latency_ms);
2360 case PROP_DROP_ON_LATENCY:
2362 g_value_set_boolean (value, priv->drop_on_latency);
2365 case PROP_TS_OFFSET:
2367 g_value_set_int64 (value, priv->ts_offset);
2372 g_value_set_boolean (value, priv->do_lost);
2377 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
2385 if (priv->srcresult != GST_FLOW_OK)
2388 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
2390 g_value_set_int (value, percent);
2395 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);