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 "gstrtpjitterbuffer.h"
65 #include "rtpjitterbuffer.h"
68 #include <gst/glib-compat-private.h>
70 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
71 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
73 /* RTPJitterBuffer signals and args */
76 SIGNAL_REQUEST_PT_MAP,
84 #define DEFAULT_LATENCY_MS 200
85 #define DEFAULT_DROP_ON_LATENCY FALSE
86 #define DEFAULT_TS_OFFSET 0
87 #define DEFAULT_DO_LOST FALSE
88 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
89 #define DEFAULT_PERCENT 0
90 #define DEFAULT_DO_RETRANSMISSION FALSE
91 #define DEFAULT_RTX_DELAY 20
92 #define DEFAULT_RTX_DELAY_REORDER 3
93 #define DEFAULT_RTX_RETRY_TIMEOUT 40
94 #define DEFAULT_RTX_RETRY_PERIOD 160
100 PROP_DROP_ON_LATENCY,
105 PROP_DO_RETRANSMISSION,
107 PROP_RTX_DELAY_REORDER,
108 PROP_RTX_RETRY_TIMEOUT,
109 PROP_RTX_RETRY_PERIOD,
113 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
115 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
117 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
120 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
122 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
123 (priv)->waiting_timer = TRUE; \
124 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
125 (priv)->waiting_timer = FALSE; \
127 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
128 if (G_UNLIKELY ((priv)->waiting_timer)) \
129 g_cond_signal (&(priv)->jbuf_timer); \
132 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
133 (priv)->waiting_event = TRUE; \
134 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
135 (priv)->waiting_event = FALSE; \
136 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
139 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
140 if (G_UNLIKELY ((priv)->waiting_event)) \
141 g_cond_signal (&(priv)->jbuf_event); \
144 struct _GstRtpJitterBufferPrivate
146 GstPad *sinkpad, *srcpad;
149 RTPJitterBuffer *jbuf;
151 gboolean waiting_timer;
153 gboolean waiting_event;
160 gboolean timer_running;
161 GThread *timer_thread;
166 gboolean drop_on_latency;
169 gboolean do_retransmission;
171 gint rtx_delay_reorder;
172 gint rtx_retry_timeout;
173 gint rtx_retry_period;
175 /* the last seqnum we pushed out */
176 guint32 last_popped_seqnum;
177 /* the next expected seqnum we push */
179 /* last output time */
180 GstClockTime last_out_time;
181 /* last valid input timestamp and rtptime pair */
182 GstClockTime ips_dts;
184 GstClockTime packet_spacing;
186 /* the next expected seqnum we receive */
187 GstClockTime last_in_dts;
188 guint32 last_in_seqnum;
189 guint32 next_in_seqnum;
193 /* start and stop ranges */
194 GstClockTime npt_start;
195 GstClockTime npt_stop;
196 guint64 ext_timestamp;
197 guint64 last_elapsed;
198 guint64 estimated_eos;
204 /* clock rate and rtp timestamp offset */
208 gint64 prev_ts_offset;
210 /* when we are shutting down */
211 GstFlowReturn srcresult;
217 GstClockTime timer_timeout;
218 guint16 timer_seqnum;
219 /* the latency of the upstream peer, we have to take this into account when
220 * synchronizing the buffers. */
221 GstClockTime peer_latency;
225 /* some accounting */
227 guint64 num_duplicates;
244 GstClockTime timeout;
245 GstClockTime duration;
246 GstClockTime rtx_base;
247 GstClockTime rtx_delay;
248 GstClockTime rtx_retry;
251 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
252 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
253 GstRtpJitterBufferPrivate))
255 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
256 GST_STATIC_PAD_TEMPLATE ("sink",
259 GST_STATIC_CAPS ("application/x-rtp, "
260 "clock-rate = (int) [ 1, 2147483647 ]"
261 /* "payload = (int) , "
262 * "encoding-name = (string) "
266 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
267 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
270 GST_STATIC_CAPS ("application/x-rtcp")
273 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
274 GST_STATIC_PAD_TEMPLATE ("src",
277 GST_STATIC_CAPS ("application/x-rtp"
278 /* "payload = (int) , "
279 * "clock-rate = (int) , "
280 * "encoding-name = (string) "
284 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
286 #define gst_rtp_jitter_buffer_parent_class parent_class
287 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
289 /* object overrides */
290 static void gst_rtp_jitter_buffer_set_property (GObject * object,
291 guint prop_id, const GValue * value, GParamSpec * pspec);
292 static void gst_rtp_jitter_buffer_get_property (GObject * object,
293 guint prop_id, GValue * value, GParamSpec * pspec);
294 static void gst_rtp_jitter_buffer_finalize (GObject * object);
296 /* element overrides */
297 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
298 * element, GstStateChange transition);
299 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
300 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
301 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
303 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
306 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
307 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
310 /* sinkpad overrides */
311 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
312 GstObject * parent, GstEvent * event);
313 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
314 GstObject * parent, GstBuffer * buffer);
316 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
317 GstObject * parent, GstEvent * event);
318 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
319 GstObject * parent, GstBuffer * buffer);
321 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
322 GstObject * parent, GstQuery * query);
324 /* srcpad overrides */
325 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
326 GstObject * parent, GstEvent * event);
327 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
328 GstObject * parent, GstPadMode mode, gboolean active);
329 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
330 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
331 GstObject * parent, GstQuery * query);
334 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
336 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
337 gboolean active, guint64 base_time);
338 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
340 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
341 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
343 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
346 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
348 GObjectClass *gobject_class;
349 GstElementClass *gstelement_class;
351 gobject_class = (GObjectClass *) klass;
352 gstelement_class = (GstElementClass *) klass;
354 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
356 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
358 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
359 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
362 * GstRtpJitterBuffer::latency:
364 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
365 * for at most this time.
367 g_object_class_install_property (gobject_class, PROP_LATENCY,
368 g_param_spec_uint ("latency", "Buffer latency in ms",
369 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
370 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
372 * GstRtpJitterBuffer::drop-on-latency:
374 * Drop oldest buffers when the queue is completely filled.
376 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
377 g_param_spec_boolean ("drop-on-latency",
378 "Drop buffers when maximum latency is reached",
379 "Tells the jitterbuffer to never exceed the given latency in size",
380 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 * GstRtpJitterBuffer::ts-offset:
384 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
385 * This is mainly used to ensure interstream synchronisation.
387 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
388 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
389 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
390 G_MAXINT64, DEFAULT_TS_OFFSET,
391 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 * GstRtpJitterBuffer::do-lost:
396 * Send out a GstRTPPacketLost event downstream when a packet is considered
399 g_object_class_install_property (gobject_class, PROP_DO_LOST,
400 g_param_spec_boolean ("do-lost", "Do Lost",
401 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
402 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
405 * GstRtpJitterBuffer::mode:
407 * Control the buffering and timestamping mode used by the jitterbuffer.
409 g_object_class_install_property (gobject_class, PROP_MODE,
410 g_param_spec_enum ("mode", "Mode",
411 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
412 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
414 * GstRtpJitterBuffer::percent:
416 * The percent of the jitterbuffer that is filled.
420 g_object_class_install_property (gobject_class, PROP_PERCENT,
421 g_param_spec_int ("percent", "percent",
422 "The buffer filled percent", 0, 100,
423 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
425 * GstRtpJitterBuffer::do-retransmission:
427 * Send out a GstRTPRetransmission event upstream when a packet is considered
428 * late and should be retransmitted.
432 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
433 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
434 "Send retransmission events upstream when a packet is late",
435 DEFAULT_DO_RETRANSMISSION,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
439 * GstRtpJitterBuffer::rtx-delay:
441 * When a packet did not arrive at the expected time, wait this extra amount
442 * of time before sending a retransmission event.
444 * When -1 is used, the max jitter will be used as extra delay.
448 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
449 g_param_spec_int ("rtx-delay", "RTX Delay",
450 "Extra time in ms to wait before sending retransmission "
451 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
452 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
454 * GstRtpJitterBuffer::rtx-delay-reorder:
456 * Assume that a retransmission event should be sent when we see
457 * this much packet reordering.
459 * When -1 is used, the value will be estimated based on observed packet
464 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
465 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
466 "Sending retransmission event when this much reordering (-1 automatic)",
467 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
468 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
470 * GstRtpJitterBuffer::rtx-retry-timeout:
472 * When no packet has been received after sending a retransmission event
473 * for this time, retry sending a retransmission event.
475 * When -1 is used, the value will be estimated based on observed round
480 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
481 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
482 "Retry sending a transmission event after this timeout in "
483 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
484 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
486 * GstRtpJitterBuffer::rtx-retry-period:
488 * The amount of time to try to get a retransmission.
490 * When -1 is used, the value will be estimated based on the jitterbuffer
491 * latency and the observed round trip time.
495 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
496 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
497 "Try to get a retransmission for this many ms "
498 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
499 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
502 * GstRtpJitterBuffer::request-pt-map:
503 * @buffer: the object which received the signal
506 * Request the payload type as #GstCaps for @pt.
508 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
509 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
510 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
511 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
512 GST_TYPE_CAPS, 1, G_TYPE_UINT);
514 * GstRtpJitterBuffer::handle-sync:
515 * @buffer: the object which received the signal
516 * @struct: a GstStructure containing sync values.
518 * Be notified of new sync values.
520 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
521 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
522 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
523 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
524 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
527 * GstRtpJitterBuffer::on-npt-stop
528 * @buffer: the object which received the signal
530 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
531 * the npt-stop position.
533 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
534 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
535 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
536 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
537 G_TYPE_NONE, 0, G_TYPE_NONE);
540 * GstRtpJitterBuffer::clear-pt-map:
541 * @buffer: the object which received the signal
543 * Invalidate the clock-rate as obtained with the
544 * #GstRtpJitterBuffer::request-pt-map signal.
546 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
547 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
548 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
549 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
550 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
553 * GstRtpJitterBuffer::set-active:
554 * @buffer: the object which received the signal
556 * Start pushing out packets with the given base time. This signal is only
557 * useful in buffering mode.
559 * Returns: the time of the last pushed packet.
563 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
564 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
565 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
566 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
567 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
570 gstelement_class->change_state =
571 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
572 gstelement_class->request_new_pad =
573 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
574 gstelement_class->release_pad =
575 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
576 gstelement_class->provide_clock =
577 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
579 gst_element_class_add_pad_template (gstelement_class,
580 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
581 gst_element_class_add_pad_template (gstelement_class,
582 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
583 gst_element_class_add_pad_template (gstelement_class,
584 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
586 gst_element_class_set_static_metadata (gstelement_class,
587 "RTP packet jitter-buffer", "Filter/Network/RTP",
588 "A buffer that deals with network jitter and other transmission faults",
589 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
590 "Wim Taymans <wim.taymans@gmail.com>");
592 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
593 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
595 GST_DEBUG_CATEGORY_INIT
596 (rtpjitterbuffer_debug, "gstrtpjitterbuffer", 0, "RTP Jitter Buffer");
600 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
602 GstRtpJitterBufferPrivate *priv;
604 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
605 jitterbuffer->priv = priv;
607 priv->latency_ms = DEFAULT_LATENCY_MS;
608 priv->latency_ns = priv->latency_ms * GST_MSECOND;
609 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
610 priv->do_lost = DEFAULT_DO_LOST;
611 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
612 priv->rtx_delay = DEFAULT_RTX_DELAY;
613 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
614 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
615 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
617 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
618 priv->jbuf = rtp_jitter_buffer_new ();
619 g_mutex_init (&priv->jbuf_lock);
620 g_cond_init (&priv->jbuf_timer);
621 g_cond_init (&priv->jbuf_event);
623 /* reset skew detection initialy */
624 rtp_jitter_buffer_reset_skew (priv->jbuf);
625 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
626 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
630 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
633 gst_pad_set_activatemode_function (priv->srcpad,
634 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
635 gst_pad_set_query_function (priv->srcpad,
636 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
637 gst_pad_set_event_function (priv->srcpad,
638 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
641 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
644 gst_pad_set_chain_function (priv->sinkpad,
645 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
646 gst_pad_set_event_function (priv->sinkpad,
647 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
648 gst_pad_set_query_function (priv->sinkpad,
649 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
651 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
652 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
654 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
658 gst_rtp_jitter_buffer_finalize (GObject * object)
660 GstRtpJitterBuffer *jitterbuffer;
662 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
664 g_array_free (jitterbuffer->priv->timers, TRUE);
665 g_mutex_clear (&jitterbuffer->priv->jbuf_lock);
666 g_cond_clear (&jitterbuffer->priv->jbuf_timer);
667 g_cond_clear (&jitterbuffer->priv->jbuf_event);
669 g_object_unref (jitterbuffer->priv->jbuf);
671 G_OBJECT_CLASS (parent_class)->finalize (object);
675 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
677 GstRtpJitterBuffer *jitterbuffer;
678 GstPad *otherpad = NULL;
682 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
684 if (pad == jitterbuffer->priv->sinkpad) {
685 otherpad = jitterbuffer->priv->srcpad;
686 } else if (pad == jitterbuffer->priv->srcpad) {
687 otherpad = jitterbuffer->priv->sinkpad;
688 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
692 g_value_init (&val, GST_TYPE_PAD);
693 g_value_set_object (&val, otherpad);
694 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
695 g_value_unset (&val);
701 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
703 GstRtpJitterBufferPrivate *priv;
705 priv = jitterbuffer->priv;
707 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
710 gst_pad_new_from_static_template
711 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
712 gst_pad_set_chain_function (priv->rtcpsinkpad,
713 gst_rtp_jitter_buffer_chain_rtcp);
714 gst_pad_set_event_function (priv->rtcpsinkpad,
715 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
716 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
717 gst_rtp_jitter_buffer_iterate_internal_links);
718 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
719 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
721 return priv->rtcpsinkpad;
725 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
727 GstRtpJitterBufferPrivate *priv;
729 priv = jitterbuffer->priv;
731 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
733 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
735 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
736 priv->rtcpsinkpad = NULL;
740 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
741 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
743 GstRtpJitterBuffer *jitterbuffer;
744 GstElementClass *klass;
746 GstRtpJitterBufferPrivate *priv;
748 g_return_val_if_fail (templ != NULL, NULL);
749 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
751 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
752 priv = jitterbuffer->priv;
753 klass = GST_ELEMENT_GET_CLASS (element);
755 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
757 /* figure out the template */
758 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
759 if (priv->rtcpsinkpad != NULL)
762 result = create_rtcp_sink (jitterbuffer);
771 g_warning ("gstrtpjitterbuffer: this is not our template");
776 g_warning ("gstrtpjitterbuffer: pad already requested");
782 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
784 GstRtpJitterBuffer *jitterbuffer;
785 GstRtpJitterBufferPrivate *priv;
787 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
788 g_return_if_fail (GST_IS_PAD (pad));
790 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
791 priv = jitterbuffer->priv;
793 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
795 if (priv->rtcpsinkpad == pad) {
796 remove_rtcp_sink (jitterbuffer);
805 g_warning ("gstjitterbuffer: asked to release an unknown pad");
811 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
813 return gst_system_clock_obtain ();
817 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
819 GstRtpJitterBufferPrivate *priv;
821 priv = jitterbuffer->priv;
823 /* this will trigger a new pt-map request signal, FIXME, do something better. */
826 priv->clock_rate = -1;
827 /* do not clear current content, but refresh state for new arrival */
828 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
829 rtp_jitter_buffer_reset_skew (priv->jbuf);
830 priv->last_popped_seqnum = -1;
831 priv->next_seqnum = -1;
836 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
839 GstRtpJitterBufferPrivate *priv;
840 GstClockTime last_out;
846 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
847 active, GST_TIME_ARGS (offset));
849 if (active != priv->active) {
850 /* add the amount of time spent in paused to the output offset. All
851 * outgoing buffers will have this offset applied to their timestamps in
852 * order to make them arrive in time in the sink. */
853 priv->out_offset = offset;
854 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
855 GST_TIME_ARGS (priv->out_offset));
856 priv->active = active;
857 JBUF_SIGNAL_EVENT (priv);
860 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
862 if ((head = rtp_jitter_buffer_peek (priv->jbuf))) {
863 /* head buffer timestamp and offset gives our output time */
864 last_out = GST_BUFFER_DTS (head) + priv->ts_offset;
866 /* use last known time when the buffer is empty */
867 last_out = priv->last_out_time;
875 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
877 GstRtpJitterBuffer *jitterbuffer;
878 GstRtpJitterBufferPrivate *priv;
883 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
884 priv = jitterbuffer->priv;
886 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
888 caps = gst_pad_peer_query_caps (other, filter);
890 templ = gst_pad_get_pad_template_caps (pad);
892 GST_DEBUG_OBJECT (jitterbuffer, "use template");
897 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
899 intersect = gst_caps_intersect (caps, templ);
900 gst_caps_unref (caps);
901 gst_caps_unref (templ);
905 gst_object_unref (jitterbuffer);
911 * Must be called with JBUF_LOCK held
915 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
918 GstRtpJitterBufferPrivate *priv;
919 GstStructure *caps_struct;
923 priv = jitterbuffer->priv;
925 /* first parse the caps */
926 caps_struct = gst_caps_get_structure (caps, 0);
928 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
930 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
931 * measure the amount of data in the buffer */
932 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
935 if (priv->clock_rate <= 0)
938 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
940 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
941 * can use this to track the amount of time elapsed on the sender. */
942 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
943 priv->clock_base = val;
945 priv->clock_base = -1;
947 priv->ext_timestamp = priv->clock_base;
949 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
952 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
953 /* first expected seqnum, only update when we didn't have a previous base. */
954 if (priv->next_in_seqnum == -1)
955 priv->next_in_seqnum = val;
956 if (priv->next_seqnum == -1)
957 priv->next_seqnum = val;
960 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
962 /* the start and stop times. The seqnum-base corresponds to the start time. We
963 * will keep track of the seqnums on the output and when we reach the one
964 * corresponding to npt-stop, we emit the npt-stop-reached signal */
965 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
966 priv->npt_start = tval;
970 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
971 priv->npt_stop = tval;
975 GST_DEBUG_OBJECT (jitterbuffer,
976 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
977 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
984 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
989 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
995 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
997 GstRtpJitterBufferPrivate *priv;
999 priv = jitterbuffer->priv;
1002 /* mark ourselves as flushing */
1003 priv->srcresult = GST_FLOW_FLUSHING;
1004 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1005 /* this unblocks any waiting pops on the src pad task */
1006 JBUF_SIGNAL_EVENT (priv);
1007 /* unlock clock, we just unschedule, the entry will be released by the
1008 * locking streaming thread. */
1009 unschedule_current_timer (jitterbuffer);
1014 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1016 GstRtpJitterBufferPrivate *priv;
1018 priv = jitterbuffer->priv;
1021 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1022 /* Mark as non flushing */
1023 priv->srcresult = GST_FLOW_OK;
1024 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1025 priv->last_popped_seqnum = -1;
1026 priv->last_out_time = -1;
1027 priv->next_seqnum = -1;
1028 priv->ips_rtptime = -1;
1029 priv->ips_dts = GST_CLOCK_TIME_NONE;
1030 priv->packet_spacing = 0;
1031 priv->next_in_seqnum = -1;
1032 priv->clock_rate = -1;
1034 priv->estimated_eos = -1;
1035 priv->last_elapsed = 0;
1036 priv->ext_timestamp = -1;
1037 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1038 rtp_jitter_buffer_flush (priv->jbuf);
1039 rtp_jitter_buffer_reset_skew (priv->jbuf);
1040 remove_all_timers (jitterbuffer);
1045 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1046 GstPadMode mode, gboolean active)
1049 GstRtpJitterBuffer *jitterbuffer = NULL;
1051 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1054 case GST_PAD_MODE_PUSH:
1056 /* allow data processing */
1057 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1059 /* start pushing out buffers */
1060 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1061 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1062 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1064 /* make sure all data processing stops ASAP */
1065 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1067 /* NOTE this will hardlock if the state change is called from the src pad
1068 * task thread because we will _join() the thread. */
1069 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1070 result = gst_pad_stop_task (pad);
1080 static GstStateChangeReturn
1081 gst_rtp_jitter_buffer_change_state (GstElement * element,
1082 GstStateChange transition)
1084 GstRtpJitterBuffer *jitterbuffer;
1085 GstRtpJitterBufferPrivate *priv;
1086 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1088 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1089 priv = jitterbuffer->priv;
1091 switch (transition) {
1092 case GST_STATE_CHANGE_NULL_TO_READY:
1094 case GST_STATE_CHANGE_READY_TO_PAUSED:
1096 /* reset negotiated values */
1097 priv->clock_rate = -1;
1098 priv->clock_base = -1;
1099 priv->peer_latency = 0;
1101 /* block until we go to PLAYING */
1102 priv->blocked = TRUE;
1103 priv->timer_running = TRUE;
1104 priv->timer_thread =
1105 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1108 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1110 /* unblock to allow streaming in PLAYING */
1111 priv->blocked = FALSE;
1112 JBUF_SIGNAL_EVENT (priv);
1119 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1121 switch (transition) {
1122 case GST_STATE_CHANGE_READY_TO_PAUSED:
1123 /* we are a live element because we sync to the clock, which we can only
1124 * do in the PLAYING state */
1125 if (ret != GST_STATE_CHANGE_FAILURE)
1126 ret = GST_STATE_CHANGE_NO_PREROLL;
1128 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1130 /* block to stop streaming when PAUSED */
1131 priv->blocked = TRUE;
1133 if (ret != GST_STATE_CHANGE_FAILURE)
1134 ret = GST_STATE_CHANGE_NO_PREROLL;
1136 case GST_STATE_CHANGE_PAUSED_TO_READY:
1138 gst_buffer_replace (&priv->last_sr, NULL);
1139 priv->timer_running = FALSE;
1140 JBUF_SIGNAL_TIMER (priv);
1142 g_thread_join (priv->timer_thread);
1143 priv->timer_thread = NULL;
1145 case GST_STATE_CHANGE_READY_TO_NULL:
1155 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1158 gboolean ret = TRUE;
1159 GstRtpJitterBuffer *jitterbuffer;
1160 GstRtpJitterBufferPrivate *priv;
1162 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1163 priv = jitterbuffer->priv;
1165 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1167 switch (GST_EVENT_TYPE (event)) {
1168 case GST_EVENT_LATENCY:
1170 GstClockTime latency;
1172 gst_event_parse_latency (event, &latency);
1174 GST_DEBUG_OBJECT (jitterbuffer,
1175 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1178 /* adjust the overall buffer delay to the total pipeline latency in
1179 * buffering mode because if downstream consumes too fast (because of
1180 * large latency or queues, we would start rebuffering again. */
1181 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1182 RTP_JITTER_BUFFER_MODE_BUFFER) {
1183 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1187 ret = gst_pad_push_event (priv->sinkpad, event);
1191 ret = gst_pad_push_event (priv->sinkpad, event);
1199 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1202 gboolean ret = TRUE;
1203 GstRtpJitterBuffer *jitterbuffer;
1204 GstRtpJitterBufferPrivate *priv;
1206 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1207 priv = jitterbuffer->priv;
1209 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1211 switch (GST_EVENT_TYPE (event)) {
1212 case GST_EVENT_CAPS:
1216 gst_event_parse_caps (event, &caps);
1219 ret = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1222 /* set same caps on srcpad on success */
1224 ret = gst_pad_push_event (priv->srcpad, event);
1226 gst_event_unref (event);
1229 case GST_EVENT_SEGMENT:
1231 gst_event_copy_segment (event, &priv->segment);
1233 /* we need time for now */
1234 if (priv->segment.format != GST_FORMAT_TIME)
1235 goto newseg_wrong_format;
1237 GST_DEBUG_OBJECT (jitterbuffer,
1238 "newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
1240 /* FIXME, push SEGMENT in the queue. Sorting order might be difficult. */
1241 ret = gst_pad_push_event (priv->srcpad, event);
1244 case GST_EVENT_FLUSH_START:
1245 ret = gst_pad_push_event (priv->srcpad, event);
1246 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1248 case GST_EVENT_FLUSH_STOP:
1249 ret = gst_pad_push_event (priv->srcpad, event);
1251 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1252 GST_PAD_MODE_PUSH, TRUE);
1256 /* push EOS in queue. We always push it at the head */
1258 /* check for flushing, we need to discard the event and return FALSE when
1259 * we are flushing */
1260 ret = priv->srcresult == GST_FLOW_OK;
1261 if (ret && !priv->eos) {
1262 GST_INFO_OBJECT (jitterbuffer, "queuing EOS");
1264 JBUF_SIGNAL_EVENT (priv);
1265 } else if (priv->eos) {
1266 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, we are already EOS");
1268 GST_DEBUG_OBJECT (jitterbuffer, "dropping EOS, reason %s",
1269 gst_flow_get_name (priv->srcresult));
1272 gst_event_unref (event);
1276 ret = gst_pad_push_event (priv->srcpad, event);
1285 newseg_wrong_format:
1287 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1289 gst_event_unref (event);
1295 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1298 gboolean ret = TRUE;
1299 GstRtpJitterBuffer *jitterbuffer;
1301 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1303 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1305 switch (GST_EVENT_TYPE (event)) {
1306 case GST_EVENT_FLUSH_START:
1307 gst_event_unref (event);
1309 case GST_EVENT_FLUSH_STOP:
1310 gst_event_unref (event);
1313 ret = gst_pad_event_default (pad, parent, event);
1321 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1322 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1323 * GST_FLOW_FLUSHING when the element is shutting down. On success
1324 * GST_FLOW_OK is returned.
1326 static GstFlowReturn
1327 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1331 GValue args[2] = { {0}, {0} };
1335 g_value_init (&args[0], GST_TYPE_ELEMENT);
1336 g_value_set_object (&args[0], jitterbuffer);
1337 g_value_init (&args[1], G_TYPE_UINT);
1338 g_value_set_uint (&args[1], pt);
1340 g_value_init (&ret, GST_TYPE_CAPS);
1341 g_value_set_boxed (&ret, NULL);
1343 JBUF_UNLOCK (jitterbuffer->priv);
1344 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1346 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1348 g_value_unset (&args[0]);
1349 g_value_unset (&args[1]);
1350 caps = (GstCaps *) g_value_dup_boxed (&ret);
1351 g_value_unset (&ret);
1355 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1356 gst_caps_unref (caps);
1358 if (G_UNLIKELY (!res))
1366 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1367 return GST_FLOW_ERROR;
1371 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1372 return GST_FLOW_FLUSHING;
1376 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1377 return GST_FLOW_ERROR;
1381 /* call with jbuf lock held */
1383 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1385 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1387 /* too short a stream, or too close to EOS will never really fill buffer */
1388 if (*percent != -1 && priv->npt_stop != -1 &&
1389 priv->npt_stop - priv->npt_start <=
1390 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1391 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1392 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1398 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1400 GstMessage *message;
1402 /* Post a buffering message */
1403 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1404 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1406 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1410 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1412 GstRtpJitterBufferPrivate *priv;
1414 priv = jitterbuffer->priv;
1416 if (timestamp == -1)
1419 /* apply the timestamp offset, this is used for inter stream sync */
1420 timestamp += priv->ts_offset;
1421 /* add the offset, this is used when buffering */
1422 timestamp += priv->out_offset;
1428 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1430 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1431 TimerData *timer = NULL;
1434 len = priv->timers->len;
1435 for (i = 0; i < len; i++) {
1436 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1437 if (test->seqnum == seqnum && test->type == type) {
1446 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1448 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1450 if (priv->clock_id) {
1451 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1452 gst_clock_id_unschedule (priv->clock_id);
1457 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1459 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1460 GstClockTime test_timeout;
1462 if ((test_timeout = timer->timeout) == -1)
1465 if (timer->type != TIMER_TYPE_EXPECTED) {
1466 /* add our latency and offset to get output times. */
1467 test_timeout = apply_offset (jitterbuffer, test_timeout);
1468 test_timeout += priv->latency_ns;
1470 return test_timeout;
1474 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1476 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1478 if (priv->clock_id) {
1479 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1481 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1482 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1484 if (timeout == -1 || timeout < priv->timer_timeout)
1485 unschedule_current_timer (jitterbuffer);
1490 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1491 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1492 GstClockTime duration)
1494 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1498 GST_DEBUG_OBJECT (jitterbuffer,
1499 "add timer for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1500 GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (timeout), GST_TIME_ARGS (delay));
1502 len = priv->timers->len;
1503 g_array_set_size (priv->timers, len + 1);
1504 timer = &g_array_index (priv->timers, TimerData, len);
1507 timer->seqnum = seqnum;
1509 timer->timeout = timeout + delay;
1510 timer->duration = duration;
1511 if (type == TIMER_TYPE_EXPECTED) {
1512 timer->rtx_base = timeout;
1513 timer->rtx_delay = delay;
1514 timer->rtx_retry = 0;
1516 recalculate_timer (jitterbuffer, timer);
1517 JBUF_SIGNAL_TIMER (priv);
1523 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1524 guint16 seqnum, GstClockTime timeout, GstClockTime delay)
1526 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1527 gboolean seqchange, timechange;
1530 seqchange = timer->seqnum != seqnum;
1531 timechange = timer->timeout != timeout;
1533 if (!seqchange && !timechange)
1536 oldseq = timer->seqnum;
1538 GST_DEBUG_OBJECT (jitterbuffer,
1539 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1540 oldseq, seqnum, GST_TIME_ARGS (timeout));
1542 timer->timeout = timeout + delay;
1543 timer->seqnum = seqnum;
1544 if (seqchange && timer->type == TIMER_TYPE_EXPECTED) {
1545 timer->rtx_base = timeout;
1546 timer->rtx_delay = delay;
1547 timer->rtx_retry = 0;
1550 if (priv->clock_id) {
1551 /* we changed the seqnum and there is a timer currently waiting with this
1552 * seqnum, unschedule it */
1553 if (seqchange && priv->timer_seqnum == oldseq)
1554 unschedule_current_timer (jitterbuffer);
1555 /* we changed the time, check if it is earlier than what we are waiting
1556 * for and unschedule if so */
1557 else if (timechange)
1558 recalculate_timer (jitterbuffer, timer);
1563 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1564 guint16 seqnum, GstClockTime timeout)
1568 /* find the seqnum timer */
1569 timer = find_timer (jitterbuffer, type, seqnum);
1570 if (timer == NULL) {
1571 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1573 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0);
1579 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1581 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1584 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1585 unschedule_current_timer (jitterbuffer);
1588 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1589 g_array_remove_index_fast (priv->timers, idx);
1594 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1596 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1597 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1598 g_array_set_size (priv->timers, 0);
1599 unschedule_current_timer (jitterbuffer);
1602 /* we just received a packet with seqnum and dts.
1604 * First check for old seqnum that we are still expecting. If the gap with the
1605 * current timestamp is too big, unschedule the timeouts.
1607 * If we have a valid packet spacing estimate we can set a timer for when we
1608 * should receive the next packet.
1609 * If we don't have a valid estimate, we remove any timer we might have
1610 * had for this packet.
1613 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1614 GstClockTime dts, gboolean do_next_seqnum)
1616 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1617 TimerData *timer = NULL;
1620 /* go through all timers and unschedule the ones with a large gap, also find
1621 * the timer for the seqnum */
1622 len = priv->timers->len;
1623 for (i = 0; i < len; i++) {
1624 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1627 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1629 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d", i,
1630 test->seqnum, seqnum, gap);
1633 GST_DEBUG ("found timer for current seqnum");
1634 /* the timer for the current seqnum */
1636 } else if (gap > priv->rtx_delay_reorder) {
1637 /* max gap, we exceeded the max reorder distance and we don't expect the
1638 * missing packet to be this reordered */
1639 if (test->rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1640 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0);
1644 if (priv->packet_spacing > 0 && do_next_seqnum && priv->do_retransmission) {
1645 GstClockTime expected, delay;
1647 /* calculate expected arrival time of the next seqnum */
1648 expected = dts + priv->packet_spacing;
1649 delay = priv->rtx_delay * GST_MSECOND;
1651 /* and update/install timer for next seqnum */
1653 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
1656 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
1657 expected, delay, priv->packet_spacing);
1658 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1659 /* if we had a timer, remove it, we don't know when to expect the next
1661 remove_timer (jitterbuffer, timer);
1662 JBUF_SIGNAL_EVENT (priv);
1667 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
1670 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1672 /* we need consecutive seqnums with a different
1673 * rtptime to estimate the packet spacing. */
1674 if (priv->ips_rtptime != rtptime) {
1675 /* rtptime changed, check dts diff */
1676 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
1677 priv->packet_spacing = dts - priv->ips_dts;
1678 GST_DEBUG_OBJECT (jitterbuffer,
1679 "new packet spacing %" GST_TIME_FORMAT,
1680 GST_TIME_ARGS (priv->packet_spacing));
1682 priv->ips_rtptime = rtptime;
1683 priv->ips_dts = dts;
1688 send_lost_event (GstRtpJitterBuffer * jitterbuffer, guint seqnum,
1689 guint lost_packets, GstClockTime timestamp, GstClockTime duration,
1692 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1694 /* we had a gap and thus we lost some packets. Create an event for this. */
1695 if (lost_packets > 1)
1696 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
1697 seqnum + lost_packets - 1);
1699 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
1701 priv->num_late += lost_packets;
1702 priv->discont = TRUE;
1704 /* update our expected next packet but make sure the seqnum increases */
1705 if (seqnum + lost_packets > priv->next_seqnum) {
1706 priv->next_seqnum = (seqnum + lost_packets) & 0xffff;
1707 priv->last_popped_seqnum = seqnum;
1708 priv->last_out_time = timestamp;
1710 if (priv->do_lost) {
1713 /* create paket lost event */
1714 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1715 gst_structure_new ("GstRTPPacketLost",
1716 "seqnum", G_TYPE_UINT, (guint) seqnum,
1717 "timestamp", G_TYPE_UINT64, timestamp,
1718 "duration", G_TYPE_UINT64, duration,
1719 "late", G_TYPE_BOOLEAN, late, NULL));
1721 gst_pad_push_event (priv->srcpad, event);
1727 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
1728 guint16 seqnum, GstClockTime dts, gint gap)
1730 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1731 GstClockTime total_duration, duration, expected_dts;
1734 GST_DEBUG_OBJECT (jitterbuffer,
1735 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1736 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
1738 /* the total duration spanned by the missing packets */
1739 if (dts >= priv->last_in_dts)
1740 total_duration = dts - priv->last_in_dts;
1744 /* interpolate between the current time and the last time based on
1745 * number of packets we are missing, this is the estimated duration
1746 * for the missing packet based on equidistant packet spacing. */
1747 duration = total_duration / (gap + 1);
1749 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1750 GST_TIME_ARGS (duration));
1752 if (total_duration > priv->latency_ns) {
1753 GstClockTime gap_time;
1756 gap_time = total_duration - priv->latency_ns;
1759 lost_packets = gap_time / duration;
1760 gap_time = lost_packets * duration;
1765 /* too many lost packets, some of the missing packets are already
1766 * too late and we can generate lost packet events for them. */
1767 GST_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
1768 " > %" GST_TIME_FORMAT ", consider %u lost",
1769 GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
1772 /* this timer will fire immediately and the lost event will be pushed from
1773 * the timer thread */
1774 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
1775 priv->last_in_dts + duration, 0, gap_time);
1777 expected += lost_packets;
1778 priv->last_in_dts += gap_time;
1781 expected_dts = priv->last_in_dts + duration;
1783 if (priv->do_retransmission) {
1784 type = TIMER_TYPE_EXPECTED;
1785 /* if we had a timer for the first missing packet, leave it. */
1786 if (find_timer (jitterbuffer, type, expected)) {
1788 expected_dts += duration;
1791 type = TIMER_TYPE_LOST;
1794 while (expected < seqnum) {
1795 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
1796 expected_dts += duration;
1801 static GstFlowReturn
1802 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
1805 GstRtpJitterBuffer *jitterbuffer;
1806 GstRtpJitterBufferPrivate *priv;
1808 guint32 expected, rtptime;
1809 GstFlowReturn ret = GST_FLOW_OK;
1810 GstClockTime dts, pts;
1815 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1816 gboolean do_next_seqnum = FALSE;
1818 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1820 priv = jitterbuffer->priv;
1822 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
1823 goto invalid_buffer;
1825 pt = gst_rtp_buffer_get_payload_type (&rtp);
1826 seqnum = gst_rtp_buffer_get_seq (&rtp);
1827 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1828 gst_rtp_buffer_unmap (&rtp);
1830 /* make sure we have PTS and DTS set */
1831 pts = GST_BUFFER_PTS (buffer);
1832 dts = GST_BUFFER_DTS (buffer);
1838 /* take the DTS of the buffer. This is the time when the packet was
1839 * received and is used to calculate jitter and clock skew. We will adjust
1840 * this DTS with the smoothed value after processing it in the
1841 * jitterbuffer and assign it as the PTS. */
1842 /* bring to running time */
1843 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
1845 GST_DEBUG_OBJECT (jitterbuffer,
1846 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
1847 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
1849 JBUF_LOCK_CHECK (priv, out_flushing);
1851 if (G_UNLIKELY (priv->last_pt != pt)) {
1854 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
1858 /* reset clock-rate so that we get a new one */
1859 priv->clock_rate = -1;
1861 /* Try to get the clock-rate from the caps first if we can. If there are no
1862 * caps we must fire the signal to get the clock-rate. */
1863 if ((caps = gst_pad_get_current_caps (pad))) {
1864 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1865 gst_caps_unref (caps);
1869 if (G_UNLIKELY (priv->clock_rate == -1)) {
1870 /* no clock rate given on the caps, try to get one with the signal */
1871 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
1872 pt) == GST_FLOW_FLUSHING)
1875 if (G_UNLIKELY (priv->clock_rate == -1))
1879 /* don't accept more data on EOS */
1880 if (G_UNLIKELY (priv->eos))
1883 expected = priv->next_in_seqnum;
1885 /* now check against our expected seqnum */
1886 if (G_LIKELY (expected != -1)) {
1889 /* now calculate gap */
1890 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
1892 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
1893 expected, seqnum, gap);
1895 if (G_LIKELY (gap == 0)) {
1896 /* packet is expected */
1897 calculate_packet_spacing (jitterbuffer, rtptime, dts);
1898 do_next_seqnum = TRUE;
1900 gboolean reset = FALSE;
1903 /* we received an old packet */
1904 if (G_UNLIKELY (gap < -RTP_MAX_MISORDER)) {
1905 /* too old packet, reset */
1906 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d < %d", gap,
1910 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
1913 /* new packet, we are missing some packets */
1914 if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
1915 /* packet too far in future, reset */
1916 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too new %d > %d", gap,
1920 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
1921 /* fill in the gap with EXPECTED timers */
1922 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
1924 do_next_seqnum = TRUE;
1927 if (G_UNLIKELY (reset)) {
1928 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1929 rtp_jitter_buffer_flush (priv->jbuf);
1930 rtp_jitter_buffer_reset_skew (priv->jbuf);
1931 remove_all_timers (jitterbuffer);
1932 priv->last_popped_seqnum = -1;
1933 priv->next_seqnum = seqnum;
1934 do_next_seqnum = TRUE;
1936 /* reset spacing estimation when gap */
1937 priv->ips_rtptime = -1;
1938 priv->ips_dts = GST_CLOCK_TIME_NONE;
1941 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
1942 /* we don't know what the next_in_seqnum should be, wait for the last
1943 * possible moment to push this buffer, maybe we get an earlier seqnum
1945 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
1946 do_next_seqnum = TRUE;
1947 /* take rtptime and dts to calculate packet spacing */
1948 priv->ips_rtptime = rtptime;
1949 priv->ips_dts = dts;
1951 if (do_next_seqnum) {
1952 priv->last_in_seqnum = seqnum;
1953 priv->last_in_dts = dts;
1954 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
1957 /* let's check if this buffer is too late, we can only accept packets with
1958 * bigger seqnum than the one we last pushed. */
1959 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
1962 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
1964 /* priv->last_popped_seqnum >= seqnum, we're too late. */
1965 if (G_UNLIKELY (gap <= 0))
1969 /* let's drop oldest packet if the queue is already full and drop-on-latency
1970 * is set. We can only do this when there actually is a latency. When no
1971 * latency is set, we just pump it in the queue and let the other end push it
1972 * out as fast as possible. */
1973 if (priv->latency_ms && priv->drop_on_latency) {
1975 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
1977 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
1980 old_buf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
1982 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
1985 gst_buffer_unref (old_buf);
1989 /* we need to make the metadata writable before pushing it in the jitterbuffer
1990 * because the jitterbuffer will update the PTS */
1991 buffer = gst_buffer_make_writable (buffer);
1992 GST_BUFFER_DTS (buffer) = dts;
1993 GST_BUFFER_PTS (buffer) = pts;
1995 /* now insert the packet into the queue in sorted order. This function returns
1996 * FALSE if a packet with the same seqnum was already in the queue, meaning we
1997 * have a duplicate. */
1998 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, buffer, dts,
1999 priv->clock_rate, &tail, &percent)))
2003 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2005 /* we had an unhandled SR, handle it now */
2007 do_handle_sync (jitterbuffer);
2009 /* signal addition of new buffer when the _loop is waiting. */
2010 if (priv->active && priv->waiting_timer)
2011 JBUF_SIGNAL_EVENT (priv);
2013 /* let's unschedule and unblock any waiting buffers. We only want to do this
2014 * when the tail buffer changed */
2015 if (G_UNLIKELY (priv->clock_id && tail)) {
2016 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2017 unschedule_current_timer (jitterbuffer);
2020 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
2021 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
2023 check_buffering_percent (jitterbuffer, &percent);
2029 post_buffering_percent (jitterbuffer, percent);
2036 /* this is not fatal but should be filtered earlier */
2037 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2038 ("Received invalid RTP payload, dropping"));
2039 gst_buffer_unref (buffer);
2044 GST_WARNING_OBJECT (jitterbuffer,
2045 "No clock-rate in caps!, dropping buffer");
2046 gst_buffer_unref (buffer);
2051 ret = priv->srcresult;
2052 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2053 gst_buffer_unref (buffer);
2059 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2060 gst_buffer_unref (buffer);
2065 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2066 " popped, dropping", seqnum, priv->last_popped_seqnum);
2068 gst_buffer_unref (buffer);
2073 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2075 priv->num_duplicates++;
2076 gst_buffer_unref (buffer);
2082 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
2084 guint64 ext_time, elapsed;
2086 GstRtpJitterBufferPrivate *priv;
2087 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2089 priv = jitterbuffer->priv;
2090 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
2091 rtp_time = gst_rtp_buffer_get_timestamp (&rtp);
2092 gst_rtp_buffer_unmap (&rtp);
2094 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2095 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2097 if (rtp_time < priv->ext_timestamp) {
2098 ext_time = priv->ext_timestamp;
2100 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
2103 if (ext_time > priv->clock_base)
2104 elapsed = ext_time - priv->clock_base;
2108 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2113 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer, GstBuffer * outbuf)
2115 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2117 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
2118 && priv->clock_base != -1 && priv->clock_rate > 0) {
2119 guint64 elapsed, estimated;
2121 elapsed = compute_elapsed (jitterbuffer, outbuf);
2123 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
2125 GstClockTime out_time;
2127 priv->last_elapsed = elapsed;
2129 left = priv->npt_stop - priv->npt_start;
2130 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
2131 GST_TIME_ARGS (left));
2133 out_time = GST_BUFFER_DTS (outbuf);
2136 estimated = gst_util_uint64_scale (out_time, left, elapsed);
2138 /* if there is almost nothing left,
2139 * we may never advance enough to end up in the above case */
2140 if (left < GST_SECOND)
2141 estimated = GST_SECOND;
2146 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2147 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2149 if (estimated != -1 && priv->estimated_eos != estimated) {
2150 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2151 priv->estimated_eos = estimated;
2157 /* take a buffer from the queue and push it */
2158 static GstFlowReturn
2159 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2161 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2162 GstFlowReturn result;
2164 GstClockTime dts, pts;
2167 /* when we get here we are ready to pop and push the buffer */
2168 outbuf = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2170 check_buffering_percent (jitterbuffer, &percent);
2172 if (G_UNLIKELY (priv->discont)) {
2173 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2174 * into the jitterbuffer so we can modify now. */
2175 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2176 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2177 priv->discont = FALSE;
2179 if (G_UNLIKELY (priv->ts_discont)) {
2180 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2181 priv->ts_discont = FALSE;
2184 dts = GST_BUFFER_DTS (outbuf);
2185 pts = GST_BUFFER_PTS (outbuf);
2187 dts = gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, dts);
2188 pts = gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, pts);
2190 /* apply timestamp with offset to buffer now */
2191 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2192 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2194 /* update the elapsed time when we need to check against the npt stop time. */
2195 update_estimated_eos (jitterbuffer, outbuf);
2197 /* now we are ready to push the buffer. Save the seqnum and release the lock
2198 * so the other end can push stuff in the queue again. */
2199 priv->last_popped_seqnum = seqnum;
2200 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2201 priv->next_seqnum = (seqnum + 1) & 0xffff;
2205 post_buffering_percent (jitterbuffer, percent);
2208 GST_DEBUG_OBJECT (jitterbuffer,
2209 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2210 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2211 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2213 result = gst_pad_push (priv->srcpad, outbuf);
2215 JBUF_LOCK_CHECK (priv, out_flushing);
2222 return priv->srcresult;
2226 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2228 /* Peek a buffer and compare the seqnum to the expected seqnum.
2229 * If all is fine, the buffer is pushed.
2230 * If something is wrong, we wait for some event
2232 static GstFlowReturn
2233 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2235 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2236 GstFlowReturn result = GST_FLOW_OK;
2239 guint32 next_seqnum;
2241 GstRTPBuffer rtp = { NULL, };
2243 /* only push buffers when PLAYING and active and not buffering */
2244 if (priv->blocked || !priv->active ||
2245 rtp_jitter_buffer_is_buffering (priv->jbuf))
2246 return GST_FLOW_WAIT;
2249 /* peek a buffer, we're just looking at the sequence number.
2250 * If all is fine, we'll pop and push it. If the sequence number is wrong we
2251 * wait for a timeout or something to change.
2252 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
2253 outbuf = rtp_jitter_buffer_peek (priv->jbuf);
2257 /* get the seqnum and the next expected seqnum */
2258 gst_rtp_buffer_map (outbuf, GST_MAP_READ, &rtp);
2259 seqnum = gst_rtp_buffer_get_seq (&rtp);
2260 gst_rtp_buffer_unmap (&rtp);
2262 next_seqnum = priv->next_seqnum;
2264 /* get the gap between this and the previous packet. If we don't know the
2265 * previous packet seqnum assume no gap. */
2266 if (G_UNLIKELY (next_seqnum == -1)) {
2267 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2268 /* we don't know what the next_seqnum should be, the chain function should
2269 * have scheduled a DEADLINE timer that will increment next_seqnum when it
2270 * fires, so wait for that */
2271 result = GST_FLOW_WAIT;
2273 /* else calculate GAP */
2274 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2276 if (G_LIKELY (gap == 0)) {
2277 /* no missing packet, pop and push */
2278 result = pop_and_push_next (jitterbuffer, seqnum);
2279 } else if (G_UNLIKELY (gap < 0)) {
2280 /* if we have a packet that we already pushed or considered dropped, pop it
2281 * off and get the next packet */
2282 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2283 seqnum, next_seqnum);
2284 outbuf = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2285 gst_buffer_unref (outbuf);
2288 /* the chain function has scheduled timers to request retransmission or
2289 * when to consider the packet lost, wait for that */
2290 GST_DEBUG_OBJECT (jitterbuffer,
2291 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2292 next_seqnum, seqnum, gap);
2293 result = GST_FLOW_WAIT;
2300 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
2302 result = GST_FLOW_EOS;
2304 result = GST_FLOW_WAIT;
2309 /* the timeout for when we expected a packet expired */
2311 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2314 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2317 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive", timer->seqnum);
2319 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2320 gst_structure_new ("GstRTPRetransmissionRequest",
2321 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
2322 "running-time", G_TYPE_UINT64, timer->rtx_base,
2323 "delay", G_TYPE_UINT,
2324 GST_TIME_AS_MSECONDS (timer->rtx_delay + timer->rtx_retry),
2325 "frequency", G_TYPE_UINT, priv->rtx_retry_timeout, "period",
2326 G_TYPE_UINT, priv->rtx_retry_period, "deadline", G_TYPE_UINT,
2327 priv->latency_ms, "packet-spacing", G_TYPE_UINT64,
2328 priv->packet_spacing, NULL));
2331 gst_pad_push_event (priv->sinkpad, event);
2334 /* calculate the timeout for the next retransmission attempt */
2335 timer->rtx_retry += (priv->rtx_retry_timeout * GST_MSECOND);
2336 if (timer->rtx_retry + timer->rtx_delay >
2337 (priv->rtx_retry_period * GST_MSECOND)) {
2338 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
2339 /* too many retransmission request, we now convert the timer
2340 * to a lost timer */
2341 timer->type = TIMER_TYPE_LOST;
2342 timer->rtx_delay = 0;
2343 timer->rtx_retry = 0;
2345 reschedule_timer (jitterbuffer, timer, timer->seqnum,
2346 timer->rtx_base + timer->rtx_retry, timer->rtx_delay);
2351 /* a packet is lost */
2353 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2356 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2357 GstClockTime duration, timestamp;
2361 seqnum = timer->seqnum;
2362 timestamp = apply_offset (jitterbuffer, timer->timeout);
2363 duration = timer->duration;
2364 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
2365 duration = priv->packet_spacing;
2366 num = MAX (timer->num, 1);
2367 late = timer->num > 0;
2369 /* remove timer now */
2370 remove_timer (jitterbuffer, timer);
2371 JBUF_SIGNAL_EVENT (priv);
2373 send_lost_event (jitterbuffer, seqnum, num, timestamp, duration, late);
2379 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2382 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2384 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
2385 remove_timer (jitterbuffer, timer);
2386 JBUF_SIGNAL_EVENT (priv);
2392 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2395 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2397 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
2399 priv->next_seqnum = timer->seqnum;
2400 remove_timer (jitterbuffer, timer);
2401 JBUF_SIGNAL_EVENT (priv);
2407 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2410 gboolean removed = FALSE;
2412 switch (timer->type) {
2413 case TIMER_TYPE_EXPECTED:
2414 removed = do_expected_timeout (jitterbuffer, timer, now);
2416 case TIMER_TYPE_LOST:
2417 removed = do_lost_timeout (jitterbuffer, timer, now);
2419 case TIMER_TYPE_DEADLINE:
2420 removed = do_deadline_timeout (jitterbuffer, timer, now);
2422 case TIMER_TYPE_EOS:
2423 removed = do_eos_timeout (jitterbuffer, timer, now);
2429 /* called when we need to wait for the next timeout.
2431 * We loop over the array of recorded timeouts and wait for the earliest one.
2432 * When it timed out, do the logic associated with the timer.
2434 * If there are no timers, we wait on a gcond until something new happens.
2437 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
2439 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2440 GstClockTime now = 0;
2443 while (priv->timer_running) {
2444 TimerData *timer = NULL;
2445 GstClockTime timer_timeout = -1;
2448 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
2449 GST_TIME_ARGS (now));
2451 len = priv->timers->len;
2452 for (i = 0; i < len; i++) {
2453 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2454 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
2456 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
2457 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
2459 /* no timestamp, timeout immeditately */
2460 if (test_timeout == -1 || test_timeout <= now) {
2461 if (do_timeout (jitterbuffer, test, now))
2464 } else if (timer == NULL || test_timeout < timer_timeout) {
2465 /* find the smallest timeout */
2467 timer_timeout = test_timeout;
2472 GstClockTime sync_time;
2475 GstClockTimeDiff clock_jitter;
2477 GST_OBJECT_LOCK (jitterbuffer);
2478 clock = GST_ELEMENT_CLOCK (jitterbuffer);
2480 GST_OBJECT_UNLOCK (jitterbuffer);
2481 /* let's just push if there is no clock */
2482 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
2483 now = timer_timeout;
2487 /* prepare for sync against clock */
2488 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
2489 /* add latency of peer to get input time */
2490 sync_time += priv->peer_latency;
2492 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
2493 " with sync time %" GST_TIME_FORMAT,
2494 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
2496 /* create an entry for the clock */
2497 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
2498 priv->timer_timeout = timer_timeout;
2499 priv->timer_seqnum = timer->seqnum;
2500 GST_OBJECT_UNLOCK (jitterbuffer);
2502 /* release the lock so that the other end can push stuff or unlock */
2505 ret = gst_clock_id_wait (id, &clock_jitter);
2508 if (!priv->timer_running)
2511 if (ret != GST_CLOCK_UNSCHEDULED) {
2512 now = timer_timeout + MAX (clock_jitter, 0);
2513 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
2514 ret, priv->timer_seqnum, clock_jitter);
2516 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
2518 /* and free the entry */
2519 gst_clock_id_unref (id);
2520 priv->clock_id = NULL;
2522 /* no timers, wait for activity */
2523 GST_DEBUG_OBJECT (jitterbuffer, "waiting");
2524 JBUF_WAIT_TIMER (priv);
2525 GST_DEBUG_OBJECT (jitterbuffer, "waiting done");
2530 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
2535 * This funcion implements the main pushing loop on the source pad.
2537 * It first tries to push as many buffers as possible. If there is a seqnum
2538 * mismatch, we wait for the next timeouts.
2541 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
2543 GstRtpJitterBufferPrivate *priv;
2544 GstFlowReturn result;
2546 priv = jitterbuffer->priv;
2548 JBUF_LOCK_CHECK (priv, flushing);
2550 result = handle_next_buffer (jitterbuffer);
2551 if (G_LIKELY (result == GST_FLOW_WAIT)) {
2552 GST_DEBUG_OBJECT (jitterbuffer, "waiting for event");
2553 /* now wait for the next event */
2554 JBUF_WAIT_EVENT (priv, flushing);
2555 GST_DEBUG_OBJECT (jitterbuffer, "waiting for event done");
2556 result = GST_FLOW_OK;
2559 while (result == GST_FLOW_OK);
2562 /* if we get here we need to pause */
2568 result = priv->srcresult;
2574 const gchar *reason = gst_flow_get_name (result);
2577 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
2578 gst_pad_pause_task (priv->srcpad);
2579 if (result == GST_FLOW_EOS) {
2580 event = gst_event_new_eos ();
2581 gst_pad_push_event (priv->srcpad, event);
2587 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
2588 * some sanity checks and then emit the handle-sync signal with the parameters.
2589 * This function must be called with the LOCK */
2591 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
2593 GstRtpJitterBufferPrivate *priv;
2594 guint64 base_rtptime, base_time;
2596 guint64 last_rtptime;
2598 guint64 ext_rtptime, diff;
2599 gboolean drop = FALSE;
2601 priv = jitterbuffer->priv;
2603 /* get the last values from the jitterbuffer */
2604 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2605 &clock_rate, &last_rtptime);
2607 clock_base = priv->clock_base;
2608 ext_rtptime = priv->ext_rtptime;
2610 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2611 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2612 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
2613 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
2615 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2616 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2619 /* we can't accept anything that happened before we did the last resync */
2620 if (base_rtptime > ext_rtptime) {
2621 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2624 /* the SR RTP timestamp must be something close to what we last observed
2625 * in the jitterbuffer */
2626 if (ext_rtptime > last_rtptime) {
2627 /* check how far ahead it is to our RTP timestamps */
2628 diff = ext_rtptime - last_rtptime;
2629 /* if bigger than 1 second, we drop it */
2630 if (diff > clock_rate) {
2631 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2632 /* should drop this, but some RTSP servers end up with bogus
2633 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2634 * so still trigger rptbin sync but invalidate RTCP data
2635 * (sync might use other methods) */
2638 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2639 G_GUINT64_FORMAT, last_rtptime, diff);
2647 s = gst_structure_new ("application/x-rtp-sync",
2648 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2649 "base-time", G_TYPE_UINT64, base_time,
2650 "clock-rate", G_TYPE_UINT, clock_rate,
2651 "clock-base", G_TYPE_UINT64, clock_base,
2652 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2653 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
2655 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2656 gst_buffer_replace (&priv->last_sr, NULL);
2658 g_signal_emit (jitterbuffer,
2659 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2661 gst_structure_free (s);
2663 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2667 static GstFlowReturn
2668 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
2671 GstRtpJitterBuffer *jitterbuffer;
2672 GstRtpJitterBufferPrivate *priv;
2673 GstFlowReturn ret = GST_FLOW_OK;
2675 GstRTCPPacket packet;
2676 guint64 ext_rtptime;
2678 GstRTCPBuffer rtcp = { NULL, };
2680 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2682 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
2683 goto invalid_buffer;
2685 priv = jitterbuffer->priv;
2687 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2689 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
2692 /* first packet must be SR or RR or else the validate would have failed */
2693 switch (gst_rtcp_packet_get_type (&packet)) {
2694 case GST_RTCP_TYPE_SR:
2695 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
2701 gst_rtcp_buffer_unmap (&rtcp);
2703 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
2706 /* convert the RTP timestamp to our extended timestamp, using the same offset
2707 * we used in the jitterbuffer */
2708 ext_rtptime = priv->jbuf->ext_rtptime;
2709 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2711 priv->ext_rtptime = ext_rtptime;
2712 gst_buffer_replace (&priv->last_sr, buffer);
2714 do_handle_sync (jitterbuffer);
2718 gst_buffer_unref (buffer);
2724 /* this is not fatal but should be filtered earlier */
2725 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2726 ("Received invalid RTCP payload, dropping"));
2732 /* this is not fatal but should be filtered earlier */
2733 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2734 ("Received empty RTCP payload, dropping"));
2735 gst_rtcp_buffer_unmap (&rtcp);
2741 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2742 gst_rtcp_buffer_unmap (&rtcp);
2749 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
2752 gboolean res = FALSE;
2754 switch (GST_QUERY_TYPE (query)) {
2755 case GST_QUERY_CAPS:
2757 GstCaps *filter, *caps;
2759 gst_query_parse_caps (query, &filter);
2760 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2761 gst_query_set_caps_result (query, caps);
2762 gst_caps_unref (caps);
2767 if (GST_QUERY_IS_SERIALIZED (query)) {
2768 GST_WARNING_OBJECT (pad, "unhandled serialized query");
2771 res = gst_pad_query_default (pad, parent, query);
2779 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
2782 GstRtpJitterBuffer *jitterbuffer;
2783 GstRtpJitterBufferPrivate *priv;
2784 gboolean res = FALSE;
2786 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2787 priv = jitterbuffer->priv;
2789 switch (GST_QUERY_TYPE (query)) {
2790 case GST_QUERY_LATENCY:
2792 /* We need to send the query upstream and add the returned latency to our
2794 GstClockTime min_latency, max_latency;
2796 GstClockTime our_latency;
2798 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
2799 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
2801 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
2802 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2803 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2805 /* store this so that we can safely sync on the peer buffers. */
2807 priv->peer_latency = min_latency;
2808 our_latency = priv->latency_ns;
2811 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
2812 GST_TIME_ARGS (our_latency));
2814 /* we add some latency but can buffer an infinite amount of time */
2815 min_latency += our_latency;
2818 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
2819 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
2820 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
2822 gst_query_set_latency (query, TRUE, min_latency, max_latency);
2826 case GST_QUERY_POSITION:
2828 GstClockTime start, last_out;
2831 gst_query_parse_position (query, &fmt, NULL);
2832 if (fmt != GST_FORMAT_TIME) {
2833 res = gst_pad_query_default (pad, parent, query);
2838 start = priv->npt_start;
2839 last_out = priv->last_out_time;
2842 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
2843 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
2844 GST_TIME_ARGS (last_out));
2846 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
2847 /* bring 0-based outgoing time to stream time */
2848 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
2851 res = gst_pad_query_default (pad, parent, query);
2855 case GST_QUERY_CAPS:
2857 GstCaps *filter, *caps;
2859 gst_query_parse_caps (query, &filter);
2860 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2861 gst_query_set_caps_result (query, caps);
2862 gst_caps_unref (caps);
2867 res = gst_pad_query_default (pad, parent, query);
2875 gst_rtp_jitter_buffer_set_property (GObject * object,
2876 guint prop_id, const GValue * value, GParamSpec * pspec)
2878 GstRtpJitterBuffer *jitterbuffer;
2879 GstRtpJitterBufferPrivate *priv;
2881 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2882 priv = jitterbuffer->priv;
2887 guint new_latency, old_latency;
2889 new_latency = g_value_get_uint (value);
2892 old_latency = priv->latency_ms;
2893 priv->latency_ms = new_latency;
2894 priv->latency_ns = priv->latency_ms * GST_MSECOND;
2895 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
2898 /* post message if latency changed, this will inform the parent pipeline
2899 * that a latency reconfiguration is possible/needed. */
2900 if (new_latency != old_latency) {
2901 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
2902 GST_TIME_ARGS (new_latency * GST_MSECOND));
2904 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
2905 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
2909 case PROP_DROP_ON_LATENCY:
2911 priv->drop_on_latency = g_value_get_boolean (value);
2914 case PROP_TS_OFFSET:
2916 priv->ts_offset = g_value_get_int64 (value);
2917 priv->ts_discont = TRUE;
2922 priv->do_lost = g_value_get_boolean (value);
2927 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
2930 case PROP_DO_RETRANSMISSION:
2932 priv->do_retransmission = g_value_get_boolean (value);
2935 case PROP_RTX_DELAY:
2937 priv->rtx_delay = g_value_get_int (value);
2940 case PROP_RTX_DELAY_REORDER:
2942 priv->rtx_delay_reorder = g_value_get_int (value);
2945 case PROP_RTX_RETRY_TIMEOUT:
2947 priv->rtx_retry_timeout = g_value_get_int (value);
2950 case PROP_RTX_RETRY_PERIOD:
2952 priv->rtx_retry_period = g_value_get_int (value);
2956 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2962 gst_rtp_jitter_buffer_get_property (GObject * object,
2963 guint prop_id, GValue * value, GParamSpec * pspec)
2965 GstRtpJitterBuffer *jitterbuffer;
2966 GstRtpJitterBufferPrivate *priv;
2968 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
2969 priv = jitterbuffer->priv;
2974 g_value_set_uint (value, priv->latency_ms);
2977 case PROP_DROP_ON_LATENCY:
2979 g_value_set_boolean (value, priv->drop_on_latency);
2982 case PROP_TS_OFFSET:
2984 g_value_set_int64 (value, priv->ts_offset);
2989 g_value_set_boolean (value, priv->do_lost);
2994 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3002 if (priv->srcresult != GST_FLOW_OK)
3005 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3007 g_value_set_int (value, percent);
3011 case PROP_DO_RETRANSMISSION:
3013 g_value_set_boolean (value, priv->do_retransmission);
3016 case PROP_RTX_DELAY:
3018 g_value_set_int (value, priv->rtx_delay);
3021 case PROP_RTX_DELAY_REORDER:
3023 g_value_set_int (value, priv->rtx_delay_reorder);
3026 case PROP_RTX_RETRY_TIMEOUT:
3028 g_value_set_int (value, priv->rtx_retry_timeout);
3031 case PROP_RTX_RETRY_PERIOD:
3033 g_value_set_int (value, priv->rtx_retry_period);
3037 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);