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;
141 gboolean drop_on_latency;
145 /* the last seqnum we pushed out */
146 guint32 last_popped_seqnum;
147 /* the next expected seqnum we push */
149 /* last output time */
150 GstClockTime last_out_time;
151 GstClockTime last_out_dts;
152 GstClockTime last_out_pts;
153 /* last valid input timestamp and rtptime pair */
154 GstClockTime last_in_dts;
155 GstClockTime last_in_rtptime;
156 GstClockTime packet_spacing;
157 /* the next expected seqnum we receive */
158 guint32 next_in_seqnum;
162 /* start and stop ranges */
163 GstClockTime npt_start;
164 GstClockTime npt_stop;
165 guint64 ext_timestamp;
166 guint64 last_elapsed;
167 guint64 estimated_eos;
173 /* clock rate and rtp timestamp offset */
177 gint64 prev_ts_offset;
179 /* when we are shutting down */
180 GstFlowReturn srcresult;
186 GstClockTime timer_timeout;
187 guint16 timer_seqnum;
188 gboolean unscheduled;
189 /* the latency of the upstream peer, we have to take this into account when
190 * synchronizing the buffers. */
191 GstClockTime peer_latency;
195 /* some accounting */
197 guint64 num_duplicates;
213 GstClockTime timeout;
216 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
217 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
218 GstRtpJitterBufferPrivate))
220 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
221 GST_STATIC_PAD_TEMPLATE ("sink",
224 GST_STATIC_CAPS ("application/x-rtp, "
225 "clock-rate = (int) [ 1, 2147483647 ]"
226 /* "payload = (int) , "
227 * "encoding-name = (string) "
231 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
232 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
235 GST_STATIC_CAPS ("application/x-rtcp")
238 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
239 GST_STATIC_PAD_TEMPLATE ("src",
242 GST_STATIC_CAPS ("application/x-rtp"
243 /* "payload = (int) , "
244 * "clock-rate = (int) , "
245 * "encoding-name = (string) "
249 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
251 #define gst_rtp_jitter_buffer_parent_class parent_class
252 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
254 /* object overrides */
255 static void gst_rtp_jitter_buffer_set_property (GObject * object,
256 guint prop_id, const GValue * value, GParamSpec * pspec);
257 static void gst_rtp_jitter_buffer_get_property (GObject * object,
258 guint prop_id, GValue * value, GParamSpec * pspec);
259 static void gst_rtp_jitter_buffer_finalize (GObject * object);
261 /* element overrides */
262 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
263 * element, GstStateChange transition);
264 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
265 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
266 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
268 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
271 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
272 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
275 /* sinkpad overrides */
276 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
277 GstObject * parent, GstEvent * event);
278 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
279 GstObject * parent, GstBuffer * buffer);
281 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
282 GstObject * parent, GstEvent * event);
283 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
284 GstObject * parent, GstBuffer * buffer);
286 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
287 GstObject * parent, GstQuery * query);
289 /* srcpad overrides */
290 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
291 GstObject * parent, GstEvent * event);
292 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
293 GstObject * parent, GstPadMode mode, gboolean active);
294 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
295 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
296 GstObject * parent, GstQuery * query);
299 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
301 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
302 gboolean active, guint64 base_time);
303 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
305 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
306 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
309 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
311 GObjectClass *gobject_class;
312 GstElementClass *gstelement_class;
314 gobject_class = (GObjectClass *) klass;
315 gstelement_class = (GstElementClass *) klass;
317 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
319 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
321 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
322 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
325 * GstRtpJitterBuffer::latency:
327 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
328 * for at most this time.
330 g_object_class_install_property (gobject_class, PROP_LATENCY,
331 g_param_spec_uint ("latency", "Buffer latency in ms",
332 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
333 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 * GstRtpJitterBuffer::drop-on-latency:
337 * Drop oldest buffers when the queue is completely filled.
339 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
340 g_param_spec_boolean ("drop-on-latency",
341 "Drop buffers when maximum latency is reached",
342 "Tells the jitterbuffer to never exceed the given latency in size",
343 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
345 * GstRtpJitterBuffer::ts-offset:
347 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
348 * This is mainly used to ensure interstream synchronisation.
350 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
351 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
352 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
353 G_MAXINT64, DEFAULT_TS_OFFSET,
354 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
357 * GstRtpJitterBuffer::do-lost:
359 * Send out a GstRTPPacketLost event downstream when a packet is considered
362 g_object_class_install_property (gobject_class, PROP_DO_LOST,
363 g_param_spec_boolean ("do-lost", "Do Lost",
364 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
365 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
368 * GstRtpJitterBuffer::mode:
370 * Control the buffering and timestamping mode used by the jitterbuffer.
372 g_object_class_install_property (gobject_class, PROP_MODE,
373 g_param_spec_enum ("mode", "Mode",
374 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
375 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
377 * GstRtpJitterBuffer::percent:
379 * The percent of the jitterbuffer that is filled.
383 g_object_class_install_property (gobject_class, PROP_PERCENT,
384 g_param_spec_int ("percent", "percent",
385 "The buffer filled percent", 0, 100,
386 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
388 * GstRtpJitterBuffer::request-pt-map:
389 * @buffer: the object which received the signal
392 * Request the payload type as #GstCaps for @pt.
394 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
395 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
396 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
397 request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT,
398 GST_TYPE_CAPS, 1, G_TYPE_UINT);
400 * GstRtpJitterBuffer::handle-sync:
401 * @buffer: the object which received the signal
402 * @struct: a GstStructure containing sync values.
404 * Be notified of new sync values.
406 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
407 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
408 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
409 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
410 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
413 * GstRtpJitterBuffer::on-npt-stop
414 * @buffer: the object which received the signal
416 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
417 * the npt-stop position.
419 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
420 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
421 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
422 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
423 G_TYPE_NONE, 0, G_TYPE_NONE);
426 * GstRtpJitterBuffer::clear-pt-map:
427 * @buffer: the object which received the signal
429 * Invalidate the clock-rate as obtained with the
430 * #GstRtpJitterBuffer::request-pt-map signal.
432 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
433 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
434 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
435 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
436 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
439 * GstRtpJitterBuffer::set-active:
440 * @buffer: the object which received the signal
442 * Start pushing out packets with the given base time. This signal is only
443 * useful in buffering mode.
445 * Returns: the time of the last pushed packet.
449 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
450 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
451 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
452 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
453 gst_rtp_bin_marshal_UINT64__BOOL_UINT64, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
456 gstelement_class->change_state =
457 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
458 gstelement_class->request_new_pad =
459 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
460 gstelement_class->release_pad =
461 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
462 gstelement_class->provide_clock =
463 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
465 gst_element_class_add_pad_template (gstelement_class,
466 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
467 gst_element_class_add_pad_template (gstelement_class,
468 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
469 gst_element_class_add_pad_template (gstelement_class,
470 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
472 gst_element_class_set_static_metadata (gstelement_class,
473 "RTP packet jitter-buffer", "Filter/Network/RTP",
474 "A buffer that deals with network jitter and other transmission faults",
475 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
476 "Wim Taymans <wim.taymans@gmail.com>");
478 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
479 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
481 GST_DEBUG_CATEGORY_INIT
482 (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
486 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
488 GstRtpJitterBufferPrivate *priv;
490 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
491 jitterbuffer->priv = priv;
493 priv->latency_ms = DEFAULT_LATENCY_MS;
494 priv->latency_ns = priv->latency_ms * GST_MSECOND;
495 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
496 priv->do_lost = DEFAULT_DO_LOST;
497 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
499 priv->jbuf = rtp_jitter_buffer_new ();
500 g_mutex_init (&priv->jbuf_lock);
501 g_cond_init (&priv->jbuf_cond);
503 /* reset skew detection initialy */
504 rtp_jitter_buffer_reset_skew (priv->jbuf);
505 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
506 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
510 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
513 gst_pad_set_activatemode_function (priv->srcpad,
514 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
515 gst_pad_set_query_function (priv->srcpad,
516 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
517 gst_pad_set_event_function (priv->srcpad,
518 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
521 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
524 gst_pad_set_chain_function (priv->sinkpad,
525 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
526 gst_pad_set_event_function (priv->sinkpad,
527 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
528 gst_pad_set_query_function (priv->sinkpad,
529 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
531 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
532 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
534 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
538 gst_rtp_jitter_buffer_finalize (GObject * object)
540 GstRtpJitterBuffer *jitterbuffer;
542 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
544 g_array_free (jitterbuffer->priv->timers, TRUE);
545 g_mutex_clear (&jitterbuffer->priv->jbuf_lock);
546 g_cond_clear (&jitterbuffer->priv->jbuf_cond);
548 g_object_unref (jitterbuffer->priv->jbuf);
550 G_OBJECT_CLASS (parent_class)->finalize (object);
554 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
556 GstRtpJitterBuffer *jitterbuffer;
557 GstPad *otherpad = NULL;
561 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
563 if (pad == jitterbuffer->priv->sinkpad) {
564 otherpad = jitterbuffer->priv->srcpad;
565 } else if (pad == jitterbuffer->priv->srcpad) {
566 otherpad = jitterbuffer->priv->sinkpad;
567 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
571 g_value_init (&val, GST_TYPE_PAD);
572 g_value_set_object (&val, otherpad);
573 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
574 g_value_unset (&val);
580 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
582 GstRtpJitterBufferPrivate *priv;
584 priv = jitterbuffer->priv;
586 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
589 gst_pad_new_from_static_template
590 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
591 gst_pad_set_chain_function (priv->rtcpsinkpad,
592 gst_rtp_jitter_buffer_chain_rtcp);
593 gst_pad_set_event_function (priv->rtcpsinkpad,
594 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
595 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
596 gst_rtp_jitter_buffer_iterate_internal_links);
597 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
598 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
600 return priv->rtcpsinkpad;
604 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
606 GstRtpJitterBufferPrivate *priv;
608 priv = jitterbuffer->priv;
610 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
612 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
614 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
615 priv->rtcpsinkpad = NULL;
619 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
620 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
622 GstRtpJitterBuffer *jitterbuffer;
623 GstElementClass *klass;
625 GstRtpJitterBufferPrivate *priv;
627 g_return_val_if_fail (templ != NULL, NULL);
628 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
630 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
631 priv = jitterbuffer->priv;
632 klass = GST_ELEMENT_GET_CLASS (element);
634 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
636 /* figure out the template */
637 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
638 if (priv->rtcpsinkpad != NULL)
641 result = create_rtcp_sink (jitterbuffer);
650 g_warning ("gstrtpjitterbuffer: this is not our template");
655 g_warning ("gstrtpjitterbuffer: pad already requested");
661 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
663 GstRtpJitterBuffer *jitterbuffer;
664 GstRtpJitterBufferPrivate *priv;
666 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
667 g_return_if_fail (GST_IS_PAD (pad));
669 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
670 priv = jitterbuffer->priv;
672 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
674 if (priv->rtcpsinkpad == pad) {
675 remove_rtcp_sink (jitterbuffer);
684 g_warning ("gstjitterbuffer: asked to release an unknown pad");
690 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
692 return gst_system_clock_obtain ();
696 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
698 GstRtpJitterBufferPrivate *priv;
700 priv = jitterbuffer->priv;
702 /* this will trigger a new pt-map request signal, FIXME, do something better. */
705 priv->clock_rate = -1;
706 /* do not clear current content, but refresh state for new arrival */
707 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
708 rtp_jitter_buffer_reset_skew (priv->jbuf);
709 priv->last_popped_seqnum = -1;
710 priv->next_seqnum = -1;
715 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
718 GstRtpJitterBufferPrivate *priv;
719 GstClockTime last_out;
725 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
726 active, GST_TIME_ARGS (offset));
728 if (active != priv->active) {
729 /* add the amount of time spent in paused to the output offset. All
730 * outgoing buffers will have this offset applied to their timestamps in
731 * order to make them arrive in time in the sink. */
732 priv->out_offset = offset;
733 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
734 GST_TIME_ARGS (priv->out_offset));
735 priv->active = active;
739 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
741 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
742 /* head buffer timestamp and offset gives our output time */
743 last_out = GST_BUFFER_DTS (head) + priv->ts_offset;
745 /* use last known time when the buffer is empty */
746 last_out = priv->last_out_time;
754 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
756 GstRtpJitterBuffer *jitterbuffer;
757 GstRtpJitterBufferPrivate *priv;
762 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
763 priv = jitterbuffer->priv;
765 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
767 caps = gst_pad_peer_query_caps (other, filter);
769 templ = gst_pad_get_pad_template_caps (pad);
771 GST_DEBUG_OBJECT (jitterbuffer, "use template");
776 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
778 intersect = gst_caps_intersect (caps, templ);
779 gst_caps_unref (caps);
780 gst_caps_unref (templ);
784 gst_object_unref (jitterbuffer);
790 * Must be called with JBUF_LOCK held
794 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
797 GstRtpJitterBufferPrivate *priv;
798 GstStructure *caps_struct;
802 priv = jitterbuffer->priv;
804 /* first parse the caps */
805 caps_struct = gst_caps_get_structure (caps, 0);
807 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
809 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
810 * measure the amount of data in the buffer */
811 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
814 if (priv->clock_rate <= 0)
817 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
819 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
820 * can use this to track the amount of time elapsed on the sender. */
821 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
822 priv->clock_base = val;
824 priv->clock_base = -1;
826 priv->ext_timestamp = priv->clock_base;
828 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
831 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
832 /* first expected seqnum, only update when we didn't have a previous base. */
833 if (priv->next_in_seqnum == -1)
834 priv->next_in_seqnum = val;
835 if (priv->next_seqnum == -1)
836 priv->next_seqnum = val;
839 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
841 /* the start and stop times. The seqnum-base corresponds to the start time. We
842 * will keep track of the seqnums on the output and when we reach the one
843 * corresponding to npt-stop, we emit the npt-stop-reached signal */
844 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
845 priv->npt_start = tval;
849 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
850 priv->npt_stop = tval;
854 GST_DEBUG_OBJECT (jitterbuffer,
855 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
856 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
863 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
868 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
874 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
876 GstRtpJitterBufferPrivate *priv;
878 priv = jitterbuffer->priv;
881 /* mark ourselves as flushing */
882 priv->srcresult = GST_FLOW_FLUSHING;
883 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
884 /* this unblocks any waiting pops on the src pad task */
886 /* unlock clock, we just unschedule, the entry will be released by the
887 * locking streaming thread. */
888 unschedule_current_timer (jitterbuffer);
893 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
895 GstRtpJitterBufferPrivate *priv;
897 priv = jitterbuffer->priv;
900 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
901 /* Mark as non flushing */
902 priv->srcresult = GST_FLOW_OK;
903 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
904 priv->last_popped_seqnum = -1;
905 priv->last_out_time = -1;
906 priv->last_out_dts = -1;
907 priv->last_out_pts = -1;
908 priv->next_seqnum = -1;
909 priv->last_in_rtptime = -1;
910 priv->last_in_dts = GST_CLOCK_TIME_NONE;
911 priv->packet_spacing = 0;
912 priv->next_in_seqnum = -1;
913 priv->clock_rate = -1;
915 priv->estimated_eos = -1;
916 priv->last_elapsed = 0;
917 priv->ext_timestamp = -1;
918 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
919 rtp_jitter_buffer_flush (priv->jbuf);
920 rtp_jitter_buffer_reset_skew (priv->jbuf);
921 remove_all_timers (jitterbuffer);
926 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
927 GstPadMode mode, gboolean active)
930 GstRtpJitterBuffer *jitterbuffer = NULL;
932 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
935 case GST_PAD_MODE_PUSH:
937 /* allow data processing */
938 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
940 /* start pushing out buffers */
941 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
942 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
943 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
945 /* make sure all data processing stops ASAP */
946 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
948 /* NOTE this will hardlock if the state change is called from the src pad
949 * task thread because we will _join() the thread. */
950 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
951 result = gst_pad_stop_task (pad);
961 static GstStateChangeReturn
962 gst_rtp_jitter_buffer_change_state (GstElement * element,
963 GstStateChange transition)
965 GstRtpJitterBuffer *jitterbuffer;
966 GstRtpJitterBufferPrivate *priv;
967 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
969 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
970 priv = jitterbuffer->priv;
972 switch (transition) {
973 case GST_STATE_CHANGE_NULL_TO_READY:
975 case GST_STATE_CHANGE_READY_TO_PAUSED:
977 /* reset negotiated values */
978 priv->clock_rate = -1;
979 priv->clock_base = -1;
980 priv->peer_latency = 0;
982 /* block until we go to PLAYING */
983 priv->blocked = TRUE;
986 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
988 /* unblock to allow streaming in PLAYING */
989 priv->blocked = FALSE;
997 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
999 switch (transition) {
1000 case GST_STATE_CHANGE_READY_TO_PAUSED:
1001 /* we are a live element because we sync to the clock, which we can only
1002 * do in the PLAYING state */
1003 if (ret != GST_STATE_CHANGE_FAILURE)
1004 ret = GST_STATE_CHANGE_NO_PREROLL;
1006 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1008 /* block to stop streaming when PAUSED */
1009 priv->blocked = TRUE;
1011 if (ret != GST_STATE_CHANGE_FAILURE)
1012 ret = GST_STATE_CHANGE_NO_PREROLL;
1014 case GST_STATE_CHANGE_PAUSED_TO_READY:
1015 gst_buffer_replace (&priv->last_sr, NULL);
1017 case GST_STATE_CHANGE_READY_TO_NULL:
1027 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1030 gboolean ret = TRUE;
1031 GstRtpJitterBuffer *jitterbuffer;
1032 GstRtpJitterBufferPrivate *priv;
1034 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1035 priv = jitterbuffer->priv;
1037 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1039 switch (GST_EVENT_TYPE (event)) {
1040 case GST_EVENT_LATENCY:
1042 GstClockTime latency;
1044 gst_event_parse_latency (event, &latency);
1046 GST_DEBUG_OBJECT (jitterbuffer,
1047 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1050 /* adjust the overall buffer delay to the total pipeline latency in
1051 * buffering mode because if downstream consumes too fast (because of
1052 * large latency or queues, we would start rebuffering again. */
1053 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1054 RTP_JITTER_BUFFER_MODE_BUFFER) {
1055 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1059 ret = gst_pad_push_event (priv->sinkpad, event);
1063 ret = gst_pad_push_event (priv->sinkpad, event);
1071 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1074 gboolean ret = TRUE;
1075 GstRtpJitterBuffer *jitterbuffer;
1076 GstRtpJitterBufferPrivate *priv;
1078 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1079 priv = jitterbuffer->priv;
1081 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1083 switch (GST_EVENT_TYPE (event)) {
1084 case GST_EVENT_CAPS:
1088 gst_event_parse_caps (event, &caps);
1091 ret = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1094 /* set same caps on srcpad on success */
1096 ret = gst_pad_push_event (priv->srcpad, event);
1098 gst_event_unref (event);
1101 case GST_EVENT_SEGMENT:
1103 gst_event_copy_segment (event, &priv->segment);
1105 /* we need time for now */
1106 if (priv->segment.format != GST_FORMAT_TIME)
1107 goto newseg_wrong_format;
1109 GST_DEBUG_OBJECT (jitterbuffer,
1110 "newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
1112 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1113 ret = gst_pad_push_event (priv->srcpad, event);
1116 case GST_EVENT_FLUSH_START:
1117 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1118 ret = gst_pad_push_event (priv->srcpad, event);
1120 case GST_EVENT_FLUSH_STOP:
1121 ret = gst_pad_push_event (priv->srcpad, event);
1123 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1124 GST_PAD_MODE_PUSH, TRUE);
1128 /* push EOS in queue. We always push it at the head */
1130 /* check for flushing, we need to discard the event and return FALSE when
1131 * we are flushing */
1132 ret = priv->srcresult == GST_FLOW_OK;
1133 if (ret && !priv->eos) {
1134 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1137 } else if (priv->eos) {
1138 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1140 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1141 gst_flow_get_name (priv->srcresult));
1144 gst_event_unref (event);
1148 ret = gst_pad_push_event (priv->srcpad, event);
1157 newseg_wrong_format:
1159 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1161 gst_event_unref (event);
1167 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1170 gboolean ret = TRUE;
1171 GstRtpJitterBuffer *jitterbuffer;
1173 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1175 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1177 switch (GST_EVENT_TYPE (event)) {
1178 case GST_EVENT_FLUSH_START:
1179 gst_event_unref (event);
1181 case GST_EVENT_FLUSH_STOP:
1182 gst_event_unref (event);
1185 ret = gst_pad_event_default (pad, parent, event);
1193 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1194 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1195 * GST_FLOW_FLUSHING when the element is shutting down. On success
1196 * GST_FLOW_OK is returned.
1198 static GstFlowReturn
1199 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1203 GValue args[2] = { {0}, {0} };
1207 g_value_init (&args[0], GST_TYPE_ELEMENT);
1208 g_value_set_object (&args[0], jitterbuffer);
1209 g_value_init (&args[1], G_TYPE_UINT);
1210 g_value_set_uint (&args[1], pt);
1212 g_value_init (&ret, GST_TYPE_CAPS);
1213 g_value_set_boxed (&ret, NULL);
1215 JBUF_UNLOCK (jitterbuffer->priv);
1216 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1218 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1220 g_value_unset (&args[0]);
1221 g_value_unset (&args[1]);
1222 caps = (GstCaps *) g_value_dup_boxed (&ret);
1223 g_value_unset (&ret);
1227 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1228 gst_caps_unref (caps);
1230 if (G_UNLIKELY (!res))
1238 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1239 return GST_FLOW_ERROR;
1243 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1244 return GST_FLOW_FLUSHING;
1248 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1249 return GST_FLOW_ERROR;
1253 /* call with jbuf lock held */
1255 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1257 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1259 /* too short a stream, or too close to EOS will never really fill buffer */
1260 if (*percent != -1 && priv->npt_stop != -1 &&
1261 priv->npt_stop - priv->npt_start <=
1262 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1263 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1264 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1270 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1272 GstMessage *message;
1274 /* Post a buffering message */
1275 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1276 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1278 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1282 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1284 GstRtpJitterBufferPrivate *priv;
1286 priv = jitterbuffer->priv;
1288 if (timestamp == -1)
1291 /* apply the timestamp offset, this is used for inter stream sync */
1292 timestamp += priv->ts_offset;
1293 /* add the offset, this is used when buffering */
1294 timestamp += priv->out_offset;
1300 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1302 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1303 TimerData *timer = NULL;
1306 len = priv->timers->len;
1307 for (i = 0; i < len; i++) {
1308 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1309 if (test->seqnum == seqnum && test->type == type) {
1318 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1320 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1322 if (priv->clock_id) {
1323 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1324 gst_clock_id_unschedule (priv->clock_id);
1325 priv->unscheduled = TRUE;
1330 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1332 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1333 GstClockTime test_timeout;
1335 if ((test_timeout = timer->timeout) == -1)
1338 if (timer->type != TIMER_TYPE_EXPECTED) {
1339 /* add our latency and offset to get output times. */
1340 test_timeout = apply_offset (jitterbuffer, test_timeout);
1341 test_timeout += priv->latency_ns;
1343 return test_timeout;
1347 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1349 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1351 if (priv->clock_id) {
1352 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1354 if (timeout == -1 || timeout < priv->timer_timeout)
1355 unschedule_current_timer (jitterbuffer);
1360 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1361 guint16 seqnum, GstClockTime timeout)
1363 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1367 GST_DEBUG_OBJECT (jitterbuffer,
1368 "add timer for seqnum %d to %" GST_TIME_FORMAT,
1369 seqnum, GST_TIME_ARGS (timeout));
1371 len = priv->timers->len;
1372 g_array_set_size (priv->timers, len + 1);
1373 timer = &g_array_index (priv->timers, TimerData, len);
1376 timer->seqnum = seqnum;
1377 timer->timeout = timeout;
1379 recalculate_timer (jitterbuffer, timer);
1385 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1386 guint16 seqnum, GstClockTime timeout)
1388 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1389 gboolean seqchange, timechange;
1392 seqchange = timer->seqnum != seqnum;
1393 timechange = timer->timeout != timeout;
1395 if (!seqchange && !timechange)
1398 oldseq = timer->seqnum;
1400 GST_DEBUG_OBJECT (jitterbuffer,
1401 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1402 oldseq, seqnum, GST_TIME_ARGS (timeout));
1404 timer->timeout = timeout;
1405 timer->seqnum = seqnum;
1407 if (priv->clock_id) {
1408 /* we changed the seqnum and there is a timer currently waiting with this
1409 * seqnum, unschedule it */
1410 if (seqchange && priv->timer_seqnum == oldseq)
1411 unschedule_current_timer (jitterbuffer);
1412 /* we changed the time, check if it is earlier than what we are waiting
1413 * for and unschedule if so */
1414 else if (timechange)
1415 recalculate_timer (jitterbuffer, timer);
1420 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1421 guint16 seqnum, GstClockTime timeout)
1425 /* find the seqnum timer */
1426 timer = find_timer (jitterbuffer, type, seqnum);
1427 if (timer == NULL) {
1428 timer = add_timer (jitterbuffer, type, seqnum, timeout);
1430 reschedule_timer (jitterbuffer, timer, seqnum, timeout);
1436 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1438 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1441 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1442 unschedule_current_timer (jitterbuffer);
1445 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1446 g_array_remove_index_fast (priv->timers, idx);
1451 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1453 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1454 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1455 g_array_set_size (priv->timers, 0);
1456 unschedule_current_timer (jitterbuffer);
1459 /* we just received a packet with seqnum and dts.
1461 * First check for old seqnum that we are still expecting. If the gap with the
1462 * current timestamp is too big, unschedule the timeouts.
1464 * If we have a valid packet spacing estimate we can set a timer for when we
1465 * should receive the next packet.
1466 * If we don't have a valid estimate, we remove any timer we might have
1467 * had for this packet.
1470 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1473 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1474 TimerData *timer = NULL;
1477 /* go through all timers and unschedule the ones with a large gap, also find
1478 * the timer for the seqnum */
1479 len = priv->timers->len;
1480 for (i = 0; i < len; i++) {
1481 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1484 if (test->type != TIMER_TYPE_EXPECTED)
1487 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1489 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d", i,
1490 test->seqnum, seqnum, gap);
1493 /* the timer for the current seqnum */
1495 } else if (gap > 5) {
1496 /* max gap, we exceeded the max reorder distance and we don't expect the
1497 * missing packet to be this reordered */
1498 reschedule_timer (jitterbuffer, test, test->seqnum, -1);
1502 if (priv->packet_spacing > 0) {
1503 GstClockTime expected;
1505 /* calculate expected arrival time of the next seqnum */
1506 expected = dts + priv->packet_spacing + 20 * GST_MSECOND;
1507 /* and update/install timer for next seqnum */
1509 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected);
1511 timer = add_timer (jitterbuffer, TIMER_TYPE_EXPECTED,
1512 priv->next_in_seqnum, expected);
1514 /* if we had a timer, remove it, we don't know when to expect the next
1516 remove_timer (jitterbuffer, timer);
1520 static GstFlowReturn
1521 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
1524 GstRtpJitterBuffer *jitterbuffer;
1525 GstRtpJitterBufferPrivate *priv;
1528 GstFlowReturn ret = GST_FLOW_OK;
1529 GstClockTime dts, pts;
1534 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1536 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1538 priv = jitterbuffer->priv;
1540 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
1541 goto invalid_buffer;
1543 pt = gst_rtp_buffer_get_payload_type (&rtp);
1544 seqnum = gst_rtp_buffer_get_seq (&rtp);
1545 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1546 gst_rtp_buffer_unmap (&rtp);
1548 /* make sure we have PTS and DTS set */
1549 pts = GST_BUFFER_PTS (buffer);
1550 dts = GST_BUFFER_DTS (buffer);
1556 /* take the DTS of the buffer. This is the time when the packet was
1557 * received and is used to calculate jitter and clock skew. We will adjust
1558 * this PTS with the smoothed value after processing it in the
1559 * jitterbuffer and assign it as the PTS. */
1560 /* bring to running time */
1561 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
1563 GST_DEBUG_OBJECT (jitterbuffer,
1564 "Received packet #%d at time %" GST_TIME_FORMAT, seqnum,
1565 GST_TIME_ARGS (dts));
1567 JBUF_LOCK_CHECK (priv, out_flushing);
1569 if (G_UNLIKELY (priv->last_pt != pt)) {
1572 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1576 /* reset clock-rate so that we get a new one */
1577 priv->clock_rate = -1;
1579 /* Try to get the clock-rate from the caps first if we can. If there are no
1580 * caps we must fire the signal to get the clock-rate. */
1581 if ((caps = gst_pad_get_current_caps (pad))) {
1582 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1583 gst_caps_unref (caps);
1587 if (G_UNLIKELY (priv->clock_rate == -1)) {
1588 /* no clock rate given on the caps, try to get one with the signal */
1589 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1590 pt) == GST_FLOW_FLUSHING)
1593 if (G_UNLIKELY (priv->clock_rate == -1))
1597 /* don't accept more data on EOS */
1598 if (G_UNLIKELY (priv->eos))
1601 /* now check against our expected seqnum */
1602 if (G_LIKELY (priv->next_in_seqnum != -1)) {
1605 gap = gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, seqnum);
1606 if (G_UNLIKELY (gap != 0)) {
1607 gboolean reset = FALSE;
1609 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1610 priv->next_in_seqnum, seqnum, gap);
1611 /* priv->next_in_seqnum >= seqnum, this packet is too late or the
1612 * sender might have been restarted with different seqnum. */
1613 if (gap < -RTP_MAX_MISORDER) {
1614 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d", gap);
1617 /* priv->next_in_seqnum < seqnum, this is a new packet */
1618 else if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1619 GST_DEBUG_OBJECT (jitterbuffer, "reset: too many dropped packets %d",
1623 GST_DEBUG_OBJECT (jitterbuffer, "tolerable gap");
1625 if (G_UNLIKELY (reset)) {
1626 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1627 rtp_jitter_buffer_flush (priv->jbuf);
1628 rtp_jitter_buffer_reset_skew (priv->jbuf);
1629 remove_all_timers (jitterbuffer);
1630 priv->last_popped_seqnum = -1;
1631 priv->next_seqnum = seqnum;
1633 /* reset spacing estimation when gap */
1634 priv->last_in_rtptime = -1;
1635 priv->last_in_dts = -1;
1637 /* packet is expected, we need consecutive seqnums with a different
1638 * rtptime to estimate the packet spacing. */
1639 if (priv->last_in_rtptime != rtptime) {
1640 /* rtptime changed, check dts diff */
1641 if (priv->last_in_dts != -1 && dts != -1 && dts > priv->last_in_dts) {
1642 priv->packet_spacing = dts - priv->last_in_dts;
1643 GST_DEBUG_OBJECT (jitterbuffer,
1644 "new packet spacing %" GST_TIME_FORMAT,
1645 GST_TIME_ARGS (priv->packet_spacing));
1647 priv->last_in_rtptime = rtptime;
1648 priv->last_in_dts = dts;
1652 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1654 /* let's check if this buffer is too late, we can only accept packets with
1655 * bigger seqnum than the one we last pushed. */
1656 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1659 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1661 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1662 if (G_UNLIKELY (gap <= 0))
1666 /* let's drop oldest packet if the queue is already full and drop-on-latency
1667 * is set. We can only do this when there actually is a latency. When no
1668 * latency is set, we just pump it in the queue and let the other end push it
1669 * out as fast as possible. */
1670 if (priv->latency_ms && priv->drop_on_latency) {
1672 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1674 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1677 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1679 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
1682 gst_buffer_unref (old_buf);
1686 /* we need to make the metadata writable before pushing it in the jitterbuffer
1687 * because the jitterbuffer will update the PTS */
1688 buffer = gst_buffer_make_writable (buffer);
1689 GST_BUFFER_DTS (buffer) = dts;
1690 GST_BUFFER_PTS (buffer) = pts;
1692 /* now insert the packet into the queue in sorted order. This function returns
1693 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1694 * have a duplicate. */
1695 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, dts,
1696 priv->clock_rate, &tail, &percent)))
1700 update_timers (jitterbuffer, seqnum, dts);
1702 /* we had an unhandled SR, handle it now */
1704 do_handle_sync (jitterbuffer);
1706 /* signal addition of new buffer when the _loop is waiting. */
1707 if (priv->waiting && priv->active)
1710 /* let's unschedule and unblock any waiting buffers. We only want to do this
1711 * when the tail buffer changed */
1712 if (G_UNLIKELY (priv->clock_id && tail)) {
1713 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
1714 unschedule_current_timer (jitterbuffer);
1717 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
1718 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
1720 check_buffering_percent (jitterbuffer, &percent);
1726 post_buffering_percent (jitterbuffer, percent);
1733 /* this is not fatal but should be filtered earlier */
1734 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
1735 ("Received invalid RTP payload, dropping"));
1736 gst_buffer_unref (buffer);
1741 GST_WARNING_OBJECT (jitterbuffer,
1742 "No clock-rate in caps!, dropping buffer");
1743 gst_buffer_unref (buffer);
1748 ret = priv->srcresult;
1749 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
1750 gst_buffer_unref (buffer);
1756 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
1757 gst_buffer_unref (buffer);
1762 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
1763 " popped, dropping", seqnum, priv->last_popped_seqnum);
1765 gst_buffer_unref (buffer);
1770 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
1772 priv->num_duplicates++;
1773 gst_buffer_unref (buffer);
1779 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1781 guint64 ext_time, elapsed;
1783 GstRtpJitterBufferPrivate *priv;
1784 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1786 priv = jitterbuffer->priv;
1787 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1788 rtp_time = gst_rtp_buffer_get_timestamp (&rtp);
1789 gst_rtp_buffer_unmap (&rtp);
1791 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
1792 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
1794 if (rtp_time < priv->ext_timestamp) {
1795 ext_time = priv->ext_timestamp;
1797 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
1800 if (ext_time > priv->clock_base)
1801 elapsed = ext_time - priv->clock_base;
1805 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
1810 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
1812 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1814 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
1815 && priv->clock_base != -1 && priv->clock_rate > 0) {
1816 guint64 elapsed, estimated;
1818 elapsed = compute_elapsed (jitterbuffer, outbuf);
1820 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
1822 GstClockTime out_time;
1824 priv->last_elapsed = elapsed;
1826 left = priv->npt_stop - priv->npt_start;
1827 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
1828 GST_TIME_ARGS (left));
1830 out_time = GST_BUFFER_DTS (outbuf);
1833 estimated = gst_util_uint64_scale (out_time, left, elapsed);
1835 /* if there is almost nothing left,
1836 * we may never advance enough to end up in the above case */
1837 if (left < GST_SECOND)
1838 estimated = GST_SECOND;
1843 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
1844 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
1846 if (estimated != -1 && priv->estimated_eos != estimated) {
1847 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
1848 priv->estimated_eos = estimated;
1854 /* take a buffer from the queue and push it */
1855 static GstFlowReturn
1856 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
1858 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1859 GstFlowReturn result;
1861 GstClockTime dts, pts;
1864 /* when we get here we are ready to pop and push the buffer */
1865 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1867 check_buffering_percent (jitterbuffer, &percent);
1869 if (G_UNLIKELY (priv->discont)) {
1870 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
1871 * into the jitterbuffer so we can modify now. */
1872 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
1873 priv->discont = FALSE;
1875 if (G_UNLIKELY (priv->ts_discont)) {
1876 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
1877 priv->ts_discont = FALSE;
1880 dts = GST_BUFFER_DTS (outbuf);
1881 pts = GST_BUFFER_PTS (outbuf);
1883 /* apply timestamp with offset to buffer now */
1884 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
1885 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
1887 /* update the elapsed time when we need to check against the npt stop time. */
1888 update_estimated_eos (jitterbuffer, outbuf);
1890 /* now we are ready to push the buffer. Save the seqnum and release the lock
1891 * so the other end can push stuff in the queue again. */
1892 priv->last_popped_seqnum = seqnum;
1893 priv->last_out_time = GST_BUFFER_PTS (outbuf);
1894 priv->last_out_dts = dts;
1895 priv->last_out_pts = pts;
1896 priv->next_seqnum = (seqnum + 1) & 0xffff;
1900 post_buffering_percent (jitterbuffer, percent);
1903 GST_DEBUG_OBJECT (jitterbuffer,
1904 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
1905 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
1906 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
1908 result = gst_pad_push (priv->srcpad, outbuf);
1910 JBUF_LOCK_CHECK (priv, out_flushing);
1917 return priv->srcresult;
1922 estimate_dts (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts, gint gap)
1924 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1925 GstClockTime duration;
1927 if (dts == -1 || priv->last_out_dts == -1)
1930 GST_DEBUG_OBJECT (jitterbuffer,
1931 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1932 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_out_dts));
1934 /* interpolate between the current time and the last time based on
1935 * number of packets we are missing, this is the estimated duration
1936 * for the missing packet based on equidistant packet spacing. Also make
1937 * sure we never go negative. */
1938 if (dts >= priv->last_out_dts)
1939 duration = (dts - priv->last_out_dts) / (gap + 1);
1941 /* packet already lost, timer will timeout quickly */
1944 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1945 GST_TIME_ARGS (duration));
1947 /* add this duration to the timestamp of the last packet we pushed */
1948 dts = (priv->last_out_dts + duration);
1953 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
1955 /* Peek a buffer and compare the seqnum to the expected seqnum.
1956 * If all is fine, the buffer is pushed.
1957 * If something is wrong, a timeout is set. We set 2 kinds of timeouts:
1958 * * deadline: to the ultimate time we can still push the packet. We
1959 * do this for the first packet to make sure we have the previous
1961 * * lost: the ultimate time we can receive a packet before we have
1962 * to consider it lost. We estimate this based on the packet
1965 static GstFlowReturn
1966 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
1968 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1969 GstFlowReturn result = GST_FLOW_OK;
1973 guint32 next_seqnum;
1975 GstRTPBuffer rtp = { NULL, };
1977 /* only push buffers when PLAYING and active and not buffering */
1978 if (priv->blocked || !priv->active ||
1979 rtp_jitter_buffer_is_buffering (priv->jbuf))
1980 return GST_FLOW_WAIT;
1983 /* peek a buffer, we're just looking at the sequence number.
1984 * If all is fine, we'll pop and push it. If the sequence number is wrong we
1985 * wait on the DTS. In the chain function we will unlock the wait when a
1986 * new buffer is available. The peeked buffer is valid for as long as we hold
1987 * the jitterbuffer lock. */
1988 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
1992 /* get the seqnum and the next expected seqnum */
1993 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
1994 seqnum = gst_rtp_buffer_get_seq (&rtp);
1995 gst_rtp_buffer_unmap (&rtp);
1997 next_seqnum = priv->next_seqnum;
1999 dts = GST_BUFFER_DTS (outbuf);
2001 /* get the gap between this and the previous packet. If we don't know the
2002 * previous packet seqnum assume no gap. */
2003 if (G_UNLIKELY (next_seqnum == -1)) {
2004 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2005 /* we don't know what the next_seqnum should be, wait for the last
2006 * possible moment to push this buffer, maybe we get an earlier seqnum
2008 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2009 result = GST_FLOW_WAIT;
2011 /* else calculate GAP */
2012 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2014 if (G_LIKELY (gap == 0)) {
2015 /* no missing packet, pop and push */
2016 result = pop_and_push_next (jitterbuffer, seqnum);
2017 } else if (G_UNLIKELY (gap < 0)) {
2018 /* if we have a packet that we already pushed or considered dropped, pop it
2019 * off and get the next packet */
2020 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2021 seqnum, next_seqnum);
2022 outbuf = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2023 gst_buffer_unref (outbuf);
2026 GST_DEBUG_OBJECT (jitterbuffer,
2027 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2028 next_seqnum, seqnum, gap);
2029 /* packet missing, estimate when we should ultimately push this packet */
2030 dts = estimate_dts (jitterbuffer, dts, gap);
2031 /* and set a timer for it */
2032 set_timer (jitterbuffer, TIMER_TYPE_LOST, next_seqnum, dts);
2033 result = GST_FLOW_WAIT;
2040 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
2041 return GST_FLOW_WAIT;
2045 /* a packet is lost */
2046 static GstFlowReturn
2047 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2048 GstClockTimeDiff clock_jitter)
2050 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2051 GstClockTime duration = GST_CLOCK_TIME_NONE;
2052 guint32 lost_packets = 1;
2053 gboolean lost_packets_late = FALSE;
2056 if (clock_jitter > 0
2057 && clock_jitter > (priv->latency_ns + priv->peer_latency)) {
2058 GstClockTimeDiff total_duration;
2059 GstClockTime out_time_diff;
2062 apply_offset (jitterbuffer, timer->timeout) - timer->timeout;
2063 total_duration = MIN (out_time_diff, clock_jitter);
2066 lost_packets = total_duration / duration;
2069 total_duration = lost_packets * duration;
2071 GST_DEBUG_OBJECT (jitterbuffer,
2072 "Current sync_time has expired a long time ago (+%" GST_TIME_FORMAT
2073 ") Cover up %d lost packets with duration %" GST_TIME_FORMAT,
2074 GST_TIME_ARGS (clock_jitter),
2075 lost_packets, GST_TIME_ARGS (total_duration));
2077 duration = total_duration;
2078 lost_packets_late = TRUE;
2082 /* we had a gap and thus we lost some packets. Create an event for this. */
2083 if (lost_packets > 1)
2084 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", timer->seqnum,
2085 timer->seqnum + lost_packets - 1);
2087 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", timer->seqnum);
2089 priv->num_late += lost_packets;
2090 priv->discont = TRUE;
2092 /* update our expected next packet */
2093 priv->last_popped_seqnum = timer->seqnum;
2094 priv->last_out_time = apply_offset (jitterbuffer, timer->timeout);
2095 priv->last_out_dts = timer->timeout;
2096 priv->last_out_pts = timer->timeout;
2097 priv->next_seqnum = (timer->seqnum + lost_packets) & 0xffff;
2098 /* remove timer now */
2099 remove_timer (jitterbuffer, timer);
2101 if (priv->do_lost) {
2104 /* create paket lost event */
2105 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
2106 gst_structure_new ("GstRTPPacketLost",
2107 "seqnum", G_TYPE_UINT, (guint) priv->last_popped_seqnum,
2108 "timestamp", G_TYPE_UINT64, priv->last_out_time,
2109 "duration", G_TYPE_UINT64, duration,
2110 "late", G_TYPE_BOOLEAN, lost_packets_late, NULL));
2112 gst_pad_push_event (priv->srcpad, event);
2113 JBUF_LOCK_CHECK (priv, flushing);
2120 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
2121 return priv->srcresult;
2125 static GstFlowReturn
2126 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2128 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
2129 remove_timer (jitterbuffer, timer);
2131 return GST_FLOW_EOS;
2134 /* called when we need to wait for the next timeout.
2136 * We loop over the array of recorded timeouts and wait for the earliest one.
2137 * When it timed out, do the logic associated with the timer.
2139 * If there are no timers, we wait on a gcond until something new happens.
2141 static GstFlowReturn
2142 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
2144 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2145 GstFlowReturn result = GST_FLOW_OK;
2147 TimerData *timer = NULL;
2148 GstClockTime timer_timeout = -1;
2151 len = priv->timers->len;
2152 for (i = 0; i < len; i++) {
2153 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2154 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
2156 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %" GST_TIME_FORMAT,
2157 i, test->seqnum, GST_TIME_ARGS (test_timeout));
2159 /* find the smallest timeout */
2160 if (timer == NULL || test_timeout == -1 || test_timeout < timer_timeout) {
2162 timer_timeout = test_timeout;
2163 if (timer_timeout == -1)
2169 GstClockTime sync_time;
2172 GstClockTimeDiff clock_jitter;
2174 /* no timestamp, timeout immeditately */
2175 if (timer_timeout == -1)
2178 GST_OBJECT_LOCK (jitterbuffer);
2179 clock = GST_ELEMENT_CLOCK (jitterbuffer);
2181 GST_OBJECT_UNLOCK (jitterbuffer);
2182 /* let's just push if there is no clock */
2183 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
2187 /* prepare for sync against clock */
2188 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
2189 /* add latency of peer to get input time */
2190 sync_time += priv->peer_latency;
2192 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
2193 " with sync time %" GST_TIME_FORMAT,
2194 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
2196 /* create an entry for the clock */
2197 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
2198 priv->unscheduled = FALSE;
2199 priv->timer_timeout = timer_timeout;
2200 priv->timer_seqnum = timer->seqnum;
2201 timer_idx = timer->idx;
2202 GST_OBJECT_UNLOCK (jitterbuffer);
2204 /* release the lock so that the other end can push stuff or unlock */
2207 ret = gst_clock_id_wait (id, &clock_jitter);
2210 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
2211 ret, priv->timer_seqnum, clock_jitter);
2212 /* and free the entry */
2213 gst_clock_id_unref (id);
2214 priv->clock_id = NULL;
2216 /* at this point, the clock could have been unlocked by a timeout, a new
2217 * tail element was added to the queue or because we are shutting down. Check
2218 * for shutdown first. */
2220 ((priv->srcresult != GST_FLOW_OK))
2223 /* we released the lock, the array might have changed */
2224 timer = &g_array_index (priv->timers, TimerData, timer_idx);
2225 /* if changed to timeout immediately, do so */
2226 if (timer->timeout == -1)
2229 /* if we got unscheduled and we are not flushing, it's because a new tail
2230 * element became available in the queue or we flushed the queue.
2231 * Grab it and try to push or sync. */
2232 if (ret == GST_CLOCK_UNSCHEDULED || priv->unscheduled) {
2233 GST_DEBUG_OBJECT (jitterbuffer, "Wait got unscheduled");
2238 switch (timer->type) {
2239 case TIMER_TYPE_EXPECTED:
2240 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive",
2242 remove_timer (jitterbuffer, timer);
2244 case TIMER_TYPE_LOST:
2245 result = do_lost_timeout (jitterbuffer, timer, clock_jitter);
2247 case TIMER_TYPE_DEADLINE:
2248 priv->next_seqnum = timer->seqnum;
2249 remove_timer (jitterbuffer, timer);
2251 case TIMER_TYPE_EOS:
2252 result = do_eos_timeout (jitterbuffer, timer);
2256 /* no timers, wait for activity */
2257 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
2258 priv->waiting = TRUE;
2259 JBUF_WAIT_CHECK (priv, flushing);
2260 priv->waiting = FALSE;
2261 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
2269 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
2270 return priv->srcresult;
2275 * This funcion implements the main pushing loop on the source pad.
2277 * It first tries to push as many buffers as possible. If there is a seqnum
2278 * mismatch, a timeout is created and this function goes waiting for the
2282 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
2284 GstRtpJitterBufferPrivate *priv;
2285 GstFlowReturn result;
2287 priv = jitterbuffer->priv;
2289 JBUF_LOCK_CHECK (priv, flushing);
2291 result = handle_next_buffer (jitterbuffer);
2292 if (G_LIKELY (result == GST_FLOW_WAIT))
2293 /* now wait for the next event */
2294 result = wait_next_timeout (jitterbuffer);
2296 while (result == GST_FLOW_OK);
2299 /* if we get here we need to pause */
2305 result = priv->srcresult;
2311 const gchar *reason = gst_flow_get_name (result);
2314 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
2315 gst_pad_pause_task (priv->srcpad);
2316 if (result == GST_FLOW_EOS) {
2317 event = gst_event_new_eos ();
2318 gst_pad_push_event (priv->srcpad, event);
2324 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
2325 * some sanity checks and then emit the handle-sync signal with the parameters.
2326 * This function must be called with the LOCK */
2328 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
2330 GstRtpJitterBufferPrivate *priv;
2331 guint64 base_rtptime, base_time;
2333 guint64 last_rtptime;
2335 guint64 ext_rtptime, diff;
2336 gboolean drop = FALSE;
2338 priv = jitterbuffer->priv;
2340 /* get the last values from the jitterbuffer */
2341 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2342 &clock_rate, &last_rtptime);
2344 clock_base = priv->clock_base;
2345 ext_rtptime = priv->ext_rtptime;
2347 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2348 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2349 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
2350 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
2352 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2353 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2356 /* we can't accept anything that happened before we did the last resync */
2357 if (base_rtptime > ext_rtptime) {
2358 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2361 /* the SR RTP timestamp must be something close to what we last observed
2362 * in the jitterbuffer */
2363 if (ext_rtptime > last_rtptime) {
2364 /* check how far ahead it is to our RTP timestamps */
2365 diff = ext_rtptime - last_rtptime;
2366 /* if bigger than 1 second, we drop it */
2367 if (diff > clock_rate) {
2368 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2369 /* should drop this, but some RTSP servers end up with bogus
2370 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2371 * so still trigger rptbin sync but invalidate RTCP data
2372 * (sync might use other methods) */
2375 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2376 G_GUINT64_FORMAT, last_rtptime, diff);
2384 s = gst_structure_new ("application/x-rtp-sync",
2385 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2386 "base-time", G_TYPE_UINT64, base_time,
2387 "clock-rate", G_TYPE_UINT, clock_rate,
2388 "clock-base", G_TYPE_UINT64, clock_base,
2389 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2390 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
2392 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2393 gst_buffer_replace (&priv->last_sr, NULL);
2395 g_signal_emit (jitterbuffer,
2396 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2398 gst_structure_free (s);
2400 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2404 static GstFlowReturn
2405 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
2408 GstRtpJitterBuffer *jitterbuffer;
2409 GstRtpJitterBufferPrivate *priv;
2410 GstFlowReturn ret = GST_FLOW_OK;
2412 GstRTCPPacket packet;
2413 guint64 ext_rtptime;
2415 GstRTCPBuffer rtcp = { NULL, };
2417 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2419 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
2420 goto invalid_buffer;
2422 priv = jitterbuffer->priv;
2424 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2426 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
2429 /* first packet must be SR or RR or else the validate would have failed */
2430 switch (gst_rtcp_packet_get_type (&packet)) {
2431 case GST_RTCP_TYPE_SR:
2432 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
2438 gst_rtcp_buffer_unmap (&rtcp);
2440 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
2443 /* convert the RTP timestamp to our extended timestamp, using the same offset
2444 * we used in the jitterbuffer */
2445 ext_rtptime = priv->jbuf->ext_rtptime;
2446 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2448 priv->ext_rtptime = ext_rtptime;
2449 gst_buffer_replace (&priv->last_sr, buffer);
2451 do_handle_sync (jitterbuffer);
2455 gst_buffer_unref (buffer);
2461 /* this is not fatal but should be filtered earlier */
2462 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2463 ("Received invalid RTCP payload, dropping"));
2469 /* this is not fatal but should be filtered earlier */
2470 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2471 ("Received empty RTCP payload, dropping"));
2472 gst_rtcp_buffer_unmap (&rtcp);
2478 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2479 gst_rtcp_buffer_unmap (&rtcp);
2486 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
2489 gboolean res = FALSE;
2491 switch (GST_QUERY_TYPE (query)) {
2492 case GST_QUERY_CAPS:
2494 GstCaps *filter, *caps;
2496 gst_query_parse_caps (query, &filter);
2497 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2498 gst_query_set_caps_result (query, caps);
2499 gst_caps_unref (caps);
2504 if (GST_QUERY_IS_SERIALIZED (query)) {
2505 GST_WARNING_OBJECT (pad, "unhandled serialized query");
2508 res = gst_pad_query_default (pad, parent, query);
2516 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
2519 GstRtpJitterBuffer *jitterbuffer;
2520 GstRtpJitterBufferPrivate *priv;
2521 gboolean res = FALSE;
2523 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2524 priv = jitterbuffer->priv;
2526 switch (GST_QUERY_TYPE (query)) {
2527 case GST_QUERY_LATENCY:
2529 /* We need to send the query upstream and add the returned latency to our
2531 GstClockTime min_latency, max_latency;
2533 GstClockTime our_latency;
2535 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2536 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2538 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2539 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2540 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2542 /* store this so that we can safely sync on the peer buffers. */
2544 priv->peer_latency = min_latency;
2545 our_latency = priv->latency_ns;
2548 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2549 GST_TIME_ARGS (our_latency));
2551 /* we add some latency but can buffer an infinite amount of time */
2552 min_latency += our_latency;
2555 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2556 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2557 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2559 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2563 case GST_QUERY_POSITION:
2565 GstClockTime start, last_out;
2568 gst_query_parse_position (query, &fmt, NULL);
2569 if (fmt != GST_FORMAT_TIME) {
2570 res = gst_pad_query_default (pad, parent, query);
2575 start = priv->npt_start;
2576 last_out = priv->last_out_time;
2579 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2580 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2581 GST_TIME_ARGS (last_out));
2583 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2584 /* bring 0-based outgoing time to stream time */
2585 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2588 res = gst_pad_query_default (pad, parent, query);
2592 case GST_QUERY_CAPS:
2594 GstCaps *filter, *caps;
2596 gst_query_parse_caps (query, &filter);
2597 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2598 gst_query_set_caps_result (query, caps);
2599 gst_caps_unref (caps);
2604 res = gst_pad_query_default (pad, parent, query);
2612 gst_rtp_jitter_buffer_set_property (GObject * object,
2613 guint prop_id, const GValue * value, GParamSpec * pspec)
2615 GstRtpJitterBuffer *jitterbuffer;
2616 GstRtpJitterBufferPrivate *priv;
2618 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2619 priv = jitterbuffer->priv;
2624 guint new_latency, old_latency;
2626 new_latency = g_value_get_uint (value);
2629 old_latency = priv->latency_ms;
2630 priv->latency_ms = new_latency;
2631 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2632 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2635 /* post message if latency changed, this will inform the parent pipeline
2636 * that a latency reconfiguration is possible/needed. */
2637 if (new_latency != old_latency) {
2638 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2639 GST_TIME_ARGS (new_latency * GST_MSECOND));
2641 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2642 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2646 case PROP_DROP_ON_LATENCY:
2648 priv->drop_on_latency = g_value_get_boolean (value);
2651 case PROP_TS_OFFSET:
2653 priv->ts_offset = g_value_get_int64 (value);
2654 priv->ts_discont = TRUE;
2659 priv->do_lost = g_value_get_boolean (value);
2664 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2668 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2674 gst_rtp_jitter_buffer_get_property (GObject * object,
2675 guint prop_id, GValue * value, GParamSpec * pspec)
2677 GstRtpJitterBuffer *jitterbuffer;
2678 GstRtpJitterBufferPrivate *priv;
2680 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2681 priv = jitterbuffer->priv;
2686 g_value_set_uint (value, priv->latency_ms);
2689 case PROP_DROP_ON_LATENCY:
2691 g_value_set_boolean (value, priv->drop_on_latency);
2694 case PROP_TS_OFFSET:
2696 g_value_set_int64 (value, priv->ts_offset);
2701 g_value_set_boolean (value, priv->do_lost);
2706 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
2714 if (priv->srcresult != GST_FLOW_OK)
2717 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
2719 g_value_set_int (value, percent);
2724 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);