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>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
10 * Copyright 2016 Pexip AS
11 * @author: Havard Graff <havard@pexip.com>
12 * @author: Stian Selnes <stian@pexip.com>
14 * This library is free software; you can redistribute it and/or
15 * modify it under the terms of the GNU Library General Public
16 * License as published by the Free Software Foundation; either
17 * version 2 of the License, or (at your option) any later version.
19 * This library is distributed in the hope that it will be useful,
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
22 * Library General Public License for more details.
24 * You should have received a copy of the GNU Library General Public
25 * License along with this library; if not, write to the
26 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
27 * Boston, MA 02110-1301, USA.
32 * SECTION:element-rtpjitterbuffer
34 * This element reorders and removes duplicate RTP packets as they are received
35 * from a network source.
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 * The rtpjitterbuffer will wait for missing packets up to a configurable time
43 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
44 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
45 * property is set, lost packets will result in a custom serialized downstream
46 * event of name GstRTPPacketLost. The lost packet events are usually used by a
47 * depayloader or other element to create concealment data or some other logic
48 * to gracefully handle the missing packets.
50 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
51 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
54 * The jitterbuffer can also be configured to send early retransmission events
55 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
56 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
57 * sends a custom upstream event named GstRTPRetransmissionRequest when the
58 * packet is considered late. The initial expected packet arrival time is
59 * calculated as follows:
61 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
62 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
63 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
64 * packets with different rtptime.
66 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
67 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
68 * previously scheduled timeout is overwritten.
70 * - If seqnum N arrived, all seqnum older than
71 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
72 * immediately. This is to request fast feedback for abonormally reorder
73 * packets before any of the previous timeouts is triggered.
75 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
76 * event. After the initial timeout expires and the retransmission event is
77 * sent, the timeout is scheduled for
78 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
79 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
80 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
81 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
82 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
83 * retransmission requests are sent and the regular logic is performed to
84 * schedule a lost packet as discussed above.
86 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
89 * This element will automatically be used inside rtpbin.
92 * <title>Example pipelines</title>
94 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
95 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
96 * inserted into the pipeline to smooth out network jitter and to reorder the
97 * out-of-order RTP packets.
108 #include <gst/rtp/gstrtpbuffer.h>
109 #include <gst/net/net.h>
111 #include "gstrtpjitterbuffer.h"
112 #include "rtpjitterbuffer.h"
113 #include "rtpstats.h"
115 #include <gst/glib-compat-private.h>
117 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
118 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
120 /* RTPJitterBuffer signals and args */
123 SIGNAL_REQUEST_PT_MAP,
131 #define DEFAULT_LATENCY_MS 200
132 #define DEFAULT_DROP_ON_LATENCY FALSE
133 #define DEFAULT_TS_OFFSET 0
134 #define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT 0
135 #define DEFAULT_DO_LOST FALSE
136 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
137 #define DEFAULT_PERCENT 0
138 #define DEFAULT_DO_RETRANSMISSION FALSE
139 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
140 #define DEFAULT_RTX_DELAY -1
141 #define DEFAULT_RTX_MIN_DELAY 0
142 #define DEFAULT_RTX_DELAY_REORDER 3
143 #define DEFAULT_RTX_RETRY_TIMEOUT -1
144 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
145 #define DEFAULT_RTX_RETRY_PERIOD -1
146 #define DEFAULT_RTX_MAX_RETRIES -1
147 #define DEFAULT_RTX_DEADLINE -1
148 #define DEFAULT_RTX_STATS_TIMEOUT 1000
149 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
150 #define DEFAULT_MAX_DROPOUT_TIME 60000
151 #define DEFAULT_MAX_MISORDER_TIME 2000
152 #define DEFAULT_RFC7273_SYNC FALSE
153 #define DEFAULT_FASTSTART_MIN_PACKETS 0
155 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
156 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
162 PROP_DROP_ON_LATENCY,
164 PROP_MAX_TS_OFFSET_ADJUSTMENT,
168 PROP_DO_RETRANSMISSION,
169 PROP_RTX_NEXT_SEQNUM,
172 PROP_RTX_DELAY_REORDER,
173 PROP_RTX_RETRY_TIMEOUT,
174 PROP_RTX_MIN_RETRY_TIMEOUT,
175 PROP_RTX_RETRY_PERIOD,
176 PROP_RTX_MAX_RETRIES,
178 PROP_RTX_STATS_TIMEOUT,
180 PROP_MAX_RTCP_RTP_TIME_DIFF,
181 PROP_MAX_DROPOUT_TIME,
182 PROP_MAX_MISORDER_TIME,
184 PROP_FASTSTART_MIN_PACKETS
187 #define JBUF_LOCK(priv) G_STMT_START { \
188 GST_TRACE("Locking from thread %p", g_thread_self()); \
189 (g_mutex_lock (&(priv)->jbuf_lock)); \
190 GST_TRACE("Locked from thread %p", g_thread_self()); \
193 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
195 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
198 #define JBUF_UNLOCK(priv) G_STMT_START { \
199 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
200 (g_mutex_unlock (&(priv)->jbuf_lock)); \
203 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
204 GST_DEBUG ("waiting timer"); \
205 (priv)->waiting_timer++; \
206 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
207 (priv)->waiting_timer--; \
208 GST_DEBUG ("waiting timer done"); \
210 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
211 if (G_UNLIKELY ((priv)->waiting_timer)) { \
212 GST_DEBUG ("signal timer, %d waiters", (priv)->waiting_timer); \
213 g_cond_signal (&(priv)->jbuf_timer); \
217 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
218 GST_DEBUG ("waiting event"); \
219 (priv)->waiting_event = TRUE; \
220 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
221 (priv)->waiting_event = FALSE; \
222 GST_DEBUG ("waiting event done"); \
223 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
226 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
227 if (G_UNLIKELY ((priv)->waiting_event)) { \
228 GST_DEBUG ("signal event"); \
229 g_cond_signal (&(priv)->jbuf_event); \
233 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
234 GST_DEBUG ("waiting query"); \
235 (priv)->waiting_query = TRUE; \
236 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
237 (priv)->waiting_query = FALSE; \
238 GST_DEBUG ("waiting query done"); \
239 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
242 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
243 (priv)->last_query = res; \
244 if (G_UNLIKELY ((priv)->waiting_query)) { \
245 GST_DEBUG ("signal query"); \
246 g_cond_signal (&(priv)->jbuf_query); \
250 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
251 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
253 typedef struct TimerQueue
256 GHashTable *hashtable;
259 struct _GstRtpJitterBufferPrivate
261 GstPad *sinkpad, *srcpad;
264 RTPJitterBuffer *jbuf;
266 gboolean waiting_timer;
268 gboolean waiting_event;
270 gboolean waiting_query;
277 guint32 segment_seqnum;
279 gboolean timer_running;
280 GThread *timer_thread;
285 gboolean drop_on_latency;
287 guint64 max_ts_offset_adjustment;
289 gboolean do_retransmission;
290 gboolean rtx_next_seqnum;
293 gint rtx_delay_reorder;
294 gint rtx_retry_timeout;
295 gint rtx_min_retry_timeout;
296 gint rtx_retry_period;
297 gint rtx_max_retries;
298 guint rtx_stats_timeout;
299 gint rtx_deadline_ms;
300 gint max_rtcp_rtp_time_diff;
301 guint32 max_dropout_time;
302 guint32 max_misorder_time;
303 guint faststart_min_packets;
305 /* the last seqnum we pushed out */
306 guint32 last_popped_seqnum;
307 /* the next expected seqnum we push */
309 /* seqnum-base, if known */
311 /* last output time */
312 GstClockTime last_out_time;
313 /* last valid input timestamp and rtptime pair */
314 GstClockTime ips_pts;
316 GstClockTime packet_spacing;
321 /* the next expected seqnum we receive */
322 GstClockTime last_in_pts;
323 guint32 next_in_seqnum;
326 TimerQueue *rtx_stats_timers;
328 /* start and stop ranges */
329 GstClockTime npt_start;
330 GstClockTime npt_stop;
331 guint64 ext_timestamp;
332 guint64 last_elapsed;
333 guint64 estimated_eos;
340 /* clock rate and rtp timestamp offset */
344 gint64 ts_offset_remainder;
346 /* when we are shutting down */
347 GstFlowReturn srcresult;
353 GstClockTime timer_timeout;
354 guint16 timer_seqnum;
355 /* the latency of the upstream peer, we have to take this into account when
356 * synchronizing the buffers. */
357 GstClockTime peer_latency;
361 /* some accounting */
365 guint64 num_duplicates;
366 guint64 num_rtx_requests;
367 guint64 num_rtx_success;
368 guint64 num_rtx_failed;
371 RTPPacketRateCtx packet_rate_ctx;
374 GstClockTime last_dts;
375 GstClockTime last_pts;
376 guint64 last_rtptime;
377 GstClockTime avg_jitter;
394 GstClockTime timeout;
395 GstClockTime duration;
396 GstClockTime rtx_base;
397 GstClockTime rtx_delay;
398 GstClockTime rtx_retry;
399 GstClockTime rtx_last;
401 guint num_rtx_received;
404 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
405 GST_STATIC_PAD_TEMPLATE ("sink",
408 GST_STATIC_CAPS ("application/x-rtp"
409 /* "clock-rate = (int) [ 1, 2147483647 ], "
410 * "payload = (int) , "
411 * "encoding-name = (string) "
415 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
416 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
419 GST_STATIC_CAPS ("application/x-rtcp")
422 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
423 GST_STATIC_PAD_TEMPLATE ("src",
426 GST_STATIC_CAPS ("application/x-rtp"
427 /* "payload = (int) , "
428 * "clock-rate = (int) , "
429 * "encoding-name = (string) "
433 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
435 #define gst_rtp_jitter_buffer_parent_class parent_class
436 G_DEFINE_TYPE_WITH_PRIVATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer,
439 /* object overrides */
440 static void gst_rtp_jitter_buffer_set_property (GObject * object,
441 guint prop_id, const GValue * value, GParamSpec * pspec);
442 static void gst_rtp_jitter_buffer_get_property (GObject * object,
443 guint prop_id, GValue * value, GParamSpec * pspec);
444 static void gst_rtp_jitter_buffer_finalize (GObject * object);
446 /* element overrides */
447 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
448 * element, GstStateChange transition);
449 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
450 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
451 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
453 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
454 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
458 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
459 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
462 /* sinkpad overrides */
463 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
464 GstObject * parent, GstEvent * event);
465 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
466 GstObject * parent, GstBuffer * buffer);
467 static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad,
468 GstObject * parent, GstBufferList * buffer_list);
470 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
471 GstObject * parent, GstEvent * event);
472 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
473 GstObject * parent, GstBuffer * buffer);
475 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
476 GstObject * parent, GstQuery * query);
478 /* srcpad overrides */
479 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
480 GstObject * parent, GstEvent * event);
481 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
482 GstObject * parent, GstPadMode mode, gboolean active);
483 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
484 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
485 GstObject * parent, GstQuery * query);
488 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
490 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
491 gboolean active, guint64 base_time);
492 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
494 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
495 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
497 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
499 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
502 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
503 TimerData * timer, GstClockTime dts, gboolean success);
505 static TimerQueue *timer_queue_new (void);
506 static void timer_queue_free (TimerQueue * queue);
509 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
511 GObjectClass *gobject_class;
512 GstElementClass *gstelement_class;
514 gobject_class = (GObjectClass *) klass;
515 gstelement_class = (GstElementClass *) klass;
517 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
519 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
520 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
523 * GstRtpJitterBuffer:latency:
525 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
526 * for at most this time.
528 g_object_class_install_property (gobject_class, PROP_LATENCY,
529 g_param_spec_uint ("latency", "Buffer latency in ms",
530 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
531 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
533 * GstRtpJitterBuffer:drop-on-latency:
535 * Drop oldest buffers when the queue is completely filled.
537 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
538 g_param_spec_boolean ("drop-on-latency",
539 "Drop buffers when maximum latency is reached",
540 "Tells the jitterbuffer to never exceed the given latency in size",
541 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
543 * GstRtpJitterBuffer:ts-offset:
545 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
546 * This is mainly used to ensure interstream synchronisation.
548 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
549 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
550 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
551 G_MAXINT64, DEFAULT_TS_OFFSET,
552 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
555 * GstRtpJitterBuffer:max-ts-offset-adjustment:
557 * The maximum number of nanoseconds per frame that time offset may be
558 * adjusted with. This is used to avoid sudden large changes to time stamps.
560 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
561 g_param_spec_uint64 ("max-ts-offset-adjustment",
562 "Max Timestamp Offset Adjustment",
563 "The maximum number of nanoseconds per frame that time stamp "
564 "offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64,
565 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
566 G_PARAM_STATIC_STRINGS));
569 * GstRtpJitterBuffer:do-lost:
571 * Send out a GstRTPPacketLost event downstream when a packet is considered
574 g_object_class_install_property (gobject_class, PROP_DO_LOST,
575 g_param_spec_boolean ("do-lost", "Do Lost",
576 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
577 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
580 * GstRtpJitterBuffer:mode:
582 * Control the buffering and timestamping mode used by the jitterbuffer.
584 g_object_class_install_property (gobject_class, PROP_MODE,
585 g_param_spec_enum ("mode", "Mode",
586 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
587 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
589 * GstRtpJitterBuffer:percent:
591 * The percent of the jitterbuffer that is filled.
593 g_object_class_install_property (gobject_class, PROP_PERCENT,
594 g_param_spec_int ("percent", "percent",
595 "The buffer filled percent", 0, 100,
596 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
598 * GstRtpJitterBuffer:do-retransmission:
600 * Send out a GstRTPRetransmission event upstream when a packet is considered
601 * late and should be retransmitted.
605 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
606 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
607 "Send retransmission events upstream when a packet is late",
608 DEFAULT_DO_RETRANSMISSION,
609 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
612 * GstRtpJitterBuffer:rtx-next-seqnum
614 * Estimate when the next packet should arrive and schedule a retransmission
616 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
617 * for packet N+1. So it will be requested if it does not arrive at the expected time.
618 * The expected time is calculated using the dts of N and the packet spacing.
622 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
623 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
624 "Estimate when the next packet should arrive and schedule a "
625 "retransmission request for it.",
626 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
629 * GstRtpJitterBuffer:rtx-delay:
631 * When a packet did not arrive at the expected time, wait this extra amount
632 * of time before sending a retransmission event.
634 * When -1 is used, the max jitter will be used as extra delay.
638 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
639 g_param_spec_int ("rtx-delay", "RTX Delay",
640 "Extra time in ms to wait before sending retransmission "
641 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
642 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
645 * GstRtpJitterBuffer:rtx-min-delay:
647 * When a packet did not arrive at the expected time, wait at least this extra amount
648 * of time before sending a retransmission event.
652 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
653 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
654 "Minimum time in ms to wait before sending retransmission "
655 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
656 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
658 * GstRtpJitterBuffer:rtx-delay-reorder:
660 * Assume that a retransmission event should be sent when we see
661 * this much packet reordering.
663 * When -1 is used, the value will be estimated based on observed packet
664 * reordering. When 0 is used packet reordering alone will not cause a
665 * retransmission event (Since 1.10).
669 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
670 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
671 "Sending retransmission event when this much reordering "
673 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
674 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
676 * GstRtpJitterBuffer::rtx-retry-timeout:
678 * When no packet has been received after sending a retransmission event
679 * for this time, retry sending a retransmission event.
681 * When -1 is used, the value will be estimated based on observed round
686 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
687 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
688 "Retry sending a transmission event after this timeout in "
689 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
690 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
692 * GstRtpJitterBuffer::rtx-min-retry-timeout:
694 * The minimum amount of time between retry timeouts. When
695 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
696 * minimum interval between retry timeouts.
698 * When -1 is used, the value will be estimated based on the
703 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
704 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
705 "Minimum timeout between sending a transmission event in "
706 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
707 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
709 * GstRtpJitterBuffer:rtx-retry-period:
711 * The amount of time to try to get a retransmission.
713 * When -1 is used, the value will be estimated based on the jitterbuffer
714 * latency and the observed round trip time.
718 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
719 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
720 "Try to get a retransmission for this many ms "
721 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
722 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
724 * GstRtpJitterBuffer:rtx-max-retries:
726 * The maximum number of retries to request a retransmission.
728 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
729 * When -1 is used, the number of retransmission request will not be limited.
733 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
734 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
735 "The maximum number of retries to request a retransmission. "
736 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
737 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
739 * GstRtpJitterBuffer:rtx-deadline:
741 * The deadline for a valid RTX request in ms.
743 * How long the RTX RTCP will be valid for.
744 * When -1 is used, the size of the jitterbuffer will be used.
748 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
749 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
750 "The deadline for a valid RTX request in milliseconds. "
751 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
752 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
754 * GstRtpJitterBuffer::rtx-stats-timeout:
756 * The time to wait for a retransmitted packet after it has been
757 * considered lost in order to collect RTX statistics.
761 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
762 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
763 "The time to wait for a retransmitted packet after it has been "
764 "considered lost in order to collect statistics (ms)",
765 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
766 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
768 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
769 g_param_spec_uint ("max-dropout-time", "Max dropout time",
770 "The maximum time (milliseconds) of missing packets tolerated.",
771 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
772 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
774 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
775 g_param_spec_uint ("max-misorder-time", "Max misorder time",
776 "The maximum time (milliseconds) of misordered packets tolerated.",
777 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
778 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
780 * GstRtpJitterBuffer:stats:
782 * Various jitterbuffer statistics. This property returns a GstStructure
783 * with name application/x-rtp-jitterbuffer-stats with the following fields:
789 * <classname>"num-pushed"</classname>:
790 * the number of packets pushed out.
796 * <classname>"num-lost"</classname>:
797 * the number of packets considered lost.
803 * <classname>"num-late"</classname>:
804 * the number of packets arriving too late.
810 * <classname>"num-duplicates"</classname>:
811 * the number of duplicate packets.
817 * <classname>"rtx-count"</classname>:
818 * the number of retransmissions requested.
824 * <classname>"rtx-success-count"</classname>:
825 * the number of successful retransmissions.
831 * <classname>"rtx-per-packet"</classname>:
832 * average number of RTX per packet.
838 * <classname>"rtx-rtt"</classname>:
839 * average round trip time per RTX.
846 g_object_class_install_property (gobject_class, PROP_STATS,
847 g_param_spec_boxed ("stats", "Statistics",
848 "Various statistics", GST_TYPE_STRUCTURE,
849 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
852 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
854 * The maximum amount of time in ms that the RTP time in the RTCP SRs
855 * is allowed to be ahead of the last RTP packet we received. Use
856 * -1 to disable ignoring of RTCP packets.
860 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
861 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
862 "Maximum amount of time in ms that the RTP time in RTCP SRs "
863 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
864 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
865 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
867 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
868 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
869 "Synchronize received streams to the RFC7273 clock "
870 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
871 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
874 * GstRtpJitterBuffer:faststart-min-packets
876 * The number of consecutive packets needed to start (set to 0 to
877 * disable faststart. The jitterbuffer will by default start after the
878 * latency has elapsed)
882 g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
883 g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
884 "The number of consecutive packets needed to start (set to 0 to "
885 "disable faststart. The jitterbuffer will by default start after "
886 "the latency has elapsed)",
887 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
888 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
891 * GstRtpJitterBuffer::request-pt-map:
892 * @buffer: the object which received the signal
895 * Request the payload type as #GstCaps for @pt.
897 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
898 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
899 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
900 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
901 GST_TYPE_CAPS, 1, G_TYPE_UINT);
903 * GstRtpJitterBuffer::handle-sync:
904 * @buffer: the object which received the signal
905 * @struct: a GstStructure containing sync values.
907 * Be notified of new sync values.
909 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
910 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
911 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
912 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
913 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
916 * GstRtpJitterBuffer::on-npt-stop:
917 * @buffer: the object which received the signal
919 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
920 * the npt-stop position.
922 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
923 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
924 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
925 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
926 G_TYPE_NONE, 0, G_TYPE_NONE);
929 * GstRtpJitterBuffer::clear-pt-map:
930 * @buffer: the object which received the signal
932 * Invalidate the clock-rate as obtained with the
933 * #GstRtpJitterBuffer::request-pt-map signal.
935 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
936 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
937 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
938 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
939 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
942 * GstRtpJitterBuffer::set-active:
943 * @buffer: the object which received the signal
945 * Start pushing out packets with the given base time. This signal is only
946 * useful in buffering mode.
948 * Returns: the time of the last pushed packet.
950 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
951 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
952 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
953 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
954 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
957 gstelement_class->change_state =
958 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
959 gstelement_class->request_new_pad =
960 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
961 gstelement_class->release_pad =
962 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
963 gstelement_class->provide_clock =
964 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
965 gstelement_class->set_clock =
966 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
968 gst_element_class_add_static_pad_template (gstelement_class,
969 &gst_rtp_jitter_buffer_src_template);
970 gst_element_class_add_static_pad_template (gstelement_class,
971 &gst_rtp_jitter_buffer_sink_template);
972 gst_element_class_add_static_pad_template (gstelement_class,
973 &gst_rtp_jitter_buffer_sink_rtcp_template);
975 gst_element_class_set_static_metadata (gstelement_class,
976 "RTP packet jitter-buffer", "Filter/Network/RTP",
977 "A buffer that deals with network jitter and other transmission faults",
978 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
979 "Wim Taymans <wim.taymans@gmail.com>");
981 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
982 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
984 GST_DEBUG_CATEGORY_INIT
985 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
989 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
991 GstRtpJitterBufferPrivate *priv;
993 priv = gst_rtp_jitter_buffer_get_instance_private (jitterbuffer);
994 jitterbuffer->priv = priv;
996 priv->latency_ms = DEFAULT_LATENCY_MS;
997 priv->latency_ns = priv->latency_ms * GST_MSECOND;
998 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
999 priv->ts_offset = DEFAULT_TS_OFFSET;
1000 priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
1001 priv->do_lost = DEFAULT_DO_LOST;
1002 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
1003 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
1004 priv->rtx_delay = DEFAULT_RTX_DELAY;
1005 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
1006 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
1007 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
1008 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
1009 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
1010 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
1011 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
1012 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
1013 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
1014 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
1015 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
1016 priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
1018 priv->ts_offset_remainder = 0;
1019 priv->last_dts = -1;
1020 priv->last_pts = -1;
1021 priv->last_rtptime = -1;
1022 priv->avg_jitter = 0;
1023 priv->segment_seqnum = GST_SEQNUM_INVALID;
1024 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
1025 priv->rtx_stats_timers = timer_queue_new ();
1026 priv->jbuf = rtp_jitter_buffer_new ();
1027 g_mutex_init (&priv->jbuf_lock);
1028 g_cond_init (&priv->jbuf_timer);
1029 g_cond_init (&priv->jbuf_event);
1030 g_cond_init (&priv->jbuf_query);
1031 g_queue_init (&priv->gap_packets);
1032 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1034 /* reset skew detection initialy */
1035 rtp_jitter_buffer_reset_skew (priv->jbuf);
1036 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
1037 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1038 priv->active = TRUE;
1041 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1044 gst_pad_set_activatemode_function (priv->srcpad,
1045 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1046 gst_pad_set_query_function (priv->srcpad,
1047 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1048 gst_pad_set_event_function (priv->srcpad,
1049 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1052 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1055 gst_pad_set_chain_function (priv->sinkpad,
1056 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1057 gst_pad_set_chain_list_function (priv->sinkpad,
1058 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list));
1059 gst_pad_set_event_function (priv->sinkpad,
1060 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1061 gst_pad_set_query_function (priv->sinkpad,
1062 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1064 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1065 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1067 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1070 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1072 #define ITEM_TYPE_BUFFER 0
1073 #define ITEM_TYPE_LOST 1
1074 #define ITEM_TYPE_EVENT 2
1075 #define ITEM_TYPE_QUERY 3
1077 static RTPJitterBufferItem *
1078 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1079 guint seqnum, guint count, guint rtptime)
1081 RTPJitterBufferItem *item;
1083 item = g_slice_new (RTPJitterBufferItem);
1090 item->seqnum = seqnum;
1091 item->count = count;
1092 item->rtptime = rtptime;
1098 free_item (RTPJitterBufferItem * item)
1100 g_return_if_fail (item != NULL);
1102 if (item->data && item->type != ITEM_TYPE_QUERY)
1103 gst_mini_object_unref (item->data);
1104 g_slice_free (RTPJitterBufferItem, item);
1108 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1110 GList **l = user_data;
1112 if (item->data && item->type == ITEM_TYPE_EVENT
1113 && GST_EVENT_IS_STICKY (item->data)) {
1114 *l = g_list_prepend (*l, item->data);
1115 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1116 gst_mini_object_unref (item->data);
1118 g_slice_free (RTPJitterBufferItem, item);
1122 gst_rtp_jitter_buffer_finalize (GObject * object)
1124 GstRtpJitterBuffer *jitterbuffer;
1125 GstRtpJitterBufferPrivate *priv;
1127 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1128 priv = jitterbuffer->priv;
1130 g_array_free (priv->timers, TRUE);
1131 timer_queue_free (priv->rtx_stats_timers);
1132 g_mutex_clear (&priv->jbuf_lock);
1133 g_cond_clear (&priv->jbuf_timer);
1134 g_cond_clear (&priv->jbuf_event);
1135 g_cond_clear (&priv->jbuf_query);
1137 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1138 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1139 g_queue_clear (&priv->gap_packets);
1140 g_object_unref (priv->jbuf);
1142 G_OBJECT_CLASS (parent_class)->finalize (object);
1145 static GstIterator *
1146 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1148 GstRtpJitterBuffer *jitterbuffer;
1149 GstPad *otherpad = NULL;
1150 GstIterator *it = NULL;
1151 GValue val = { 0, };
1153 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1155 if (pad == jitterbuffer->priv->sinkpad) {
1156 otherpad = jitterbuffer->priv->srcpad;
1157 } else if (pad == jitterbuffer->priv->srcpad) {
1158 otherpad = jitterbuffer->priv->sinkpad;
1159 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1160 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1164 g_value_init (&val, GST_TYPE_PAD);
1165 g_value_set_object (&val, otherpad);
1166 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1167 g_value_unset (&val);
1174 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1176 GstRtpJitterBufferPrivate *priv;
1178 priv = jitterbuffer->priv;
1180 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1183 gst_pad_new_from_static_template
1184 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1185 gst_pad_set_chain_function (priv->rtcpsinkpad,
1186 gst_rtp_jitter_buffer_chain_rtcp);
1187 gst_pad_set_event_function (priv->rtcpsinkpad,
1188 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1189 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1190 gst_rtp_jitter_buffer_iterate_internal_links);
1191 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1192 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1194 return priv->rtcpsinkpad;
1198 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1200 GstRtpJitterBufferPrivate *priv;
1202 priv = jitterbuffer->priv;
1204 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1206 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1208 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1209 priv->rtcpsinkpad = NULL;
1213 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1214 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1216 GstRtpJitterBuffer *jitterbuffer;
1217 GstElementClass *klass;
1219 GstRtpJitterBufferPrivate *priv;
1221 g_return_val_if_fail (templ != NULL, NULL);
1222 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1224 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1225 priv = jitterbuffer->priv;
1226 klass = GST_ELEMENT_GET_CLASS (element);
1228 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1230 /* figure out the template */
1231 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1232 if (priv->rtcpsinkpad != NULL)
1235 result = create_rtcp_sink (jitterbuffer);
1237 goto wrong_template;
1244 g_warning ("rtpjitterbuffer: this is not our template");
1249 g_warning ("rtpjitterbuffer: pad already requested");
1255 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1257 GstRtpJitterBuffer *jitterbuffer;
1258 GstRtpJitterBufferPrivate *priv;
1260 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1261 g_return_if_fail (GST_IS_PAD (pad));
1263 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1264 priv = jitterbuffer->priv;
1266 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1268 if (priv->rtcpsinkpad == pad) {
1269 remove_rtcp_sink (jitterbuffer);
1278 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1284 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1286 return gst_system_clock_obtain ();
1290 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1292 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1294 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1296 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1300 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1302 GstRtpJitterBufferPrivate *priv;
1304 priv = jitterbuffer->priv;
1306 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1309 priv->clock_rate = -1;
1310 /* do not clear current content, but refresh state for new arrival */
1311 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1312 rtp_jitter_buffer_reset_skew (priv->jbuf);
1317 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1320 GstRtpJitterBufferPrivate *priv;
1321 GstClockTime last_out;
1322 RTPJitterBufferItem *item;
1327 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1328 active, GST_TIME_ARGS (offset));
1330 if (active != priv->active) {
1331 /* add the amount of time spent in paused to the output offset. All
1332 * outgoing buffers will have this offset applied to their timestamps in
1333 * order to make them arrive in time in the sink. */
1334 priv->out_offset = offset;
1335 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1336 GST_TIME_ARGS (priv->out_offset));
1337 priv->active = active;
1338 JBUF_SIGNAL_EVENT (priv);
1341 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1343 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1344 /* head buffer timestamp and offset gives our output time */
1345 last_out = item->pts + priv->ts_offset;
1347 /* use last known time when the buffer is empty */
1348 last_out = priv->last_out_time;
1356 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1358 GstRtpJitterBuffer *jitterbuffer;
1359 GstRtpJitterBufferPrivate *priv;
1364 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1365 priv = jitterbuffer->priv;
1367 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1369 caps = gst_pad_peer_query_caps (other, filter);
1371 templ = gst_pad_get_pad_template_caps (pad);
1373 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1378 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1380 intersect = gst_caps_intersect (caps, templ);
1381 gst_caps_unref (caps);
1382 gst_caps_unref (templ);
1386 gst_object_unref (jitterbuffer);
1392 * Must be called with JBUF_LOCK held
1396 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1397 GstCaps * caps, gint pt)
1399 GstRtpJitterBufferPrivate *priv;
1400 GstStructure *caps_struct;
1404 const gchar *ts_refclk, *mediaclk;
1406 priv = jitterbuffer->priv;
1408 /* first parse the caps */
1409 caps_struct = gst_caps_get_structure (caps, 0);
1411 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1413 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1415 GST_ERROR_OBJECT (jitterbuffer,
1416 "Got caps with wrong payload type (got %d, expected %d)", pt, payload);
1420 if (payload != -1) {
1421 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1422 priv->last_pt = payload;
1425 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1426 * measure the amount of data in the buffer */
1427 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1430 if (priv->clock_rate <= 0)
1433 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1435 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1437 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1439 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1440 * can use this to track the amount of time elapsed on the sender. */
1441 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1442 priv->clock_base = val;
1444 priv->clock_base = -1;
1446 priv->ext_timestamp = priv->clock_base;
1448 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1451 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1452 /* first expected seqnum, only update when we didn't have a previous base. */
1453 if (priv->next_in_seqnum == -1)
1454 priv->next_in_seqnum = val;
1455 if (priv->next_seqnum == -1) {
1456 priv->next_seqnum = val;
1457 JBUF_SIGNAL_EVENT (priv);
1459 priv->seqnum_base = val;
1461 priv->seqnum_base = -1;
1464 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1466 /* the start and stop times. The seqnum-base corresponds to the start time. We
1467 * will keep track of the seqnums on the output and when we reach the one
1468 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1469 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1470 priv->npt_start = tval;
1472 priv->npt_start = 0;
1474 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1475 priv->npt_stop = tval;
1477 priv->npt_stop = -1;
1479 GST_DEBUG_OBJECT (jitterbuffer,
1480 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1481 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1483 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1484 GstClock *clock = NULL;
1485 guint64 clock_offset = -1;
1487 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1490 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1491 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1492 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1494 const gchar *host, *portstr;
1498 host = ts_refclk + sizeof ("ntp=") - 1;
1499 if (host[0] == '[') {
1501 portstr = strchr (host, ']');
1502 if (portstr && portstr[1] == ':')
1503 portstr = portstr + 1;
1507 portstr = strrchr (host, ':');
1511 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1515 hostname = g_strndup (host, (portstr - host));
1517 hostname = g_strdup (host);
1519 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1522 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1523 const gchar *domainstr =
1524 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1527 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1530 clock = gst_ptp_clock_new (NULL, domain);
1532 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1535 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1536 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1538 if (!g_str_has_prefix (mediaclk, "direct=")
1539 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1540 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1541 if (strstr (mediaclk, "rate=") != NULL) {
1542 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1547 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1549 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1557 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1562 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1568 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1570 GstRtpJitterBufferPrivate *priv;
1572 priv = jitterbuffer->priv;
1575 /* mark ourselves as flushing */
1576 priv->srcresult = GST_FLOW_FLUSHING;
1577 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1578 /* this unblocks any waiting pops on the src pad task */
1579 JBUF_SIGNAL_EVENT (priv);
1580 JBUF_SIGNAL_QUERY (priv, FALSE);
1585 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1587 GstRtpJitterBufferPrivate *priv;
1589 priv = jitterbuffer->priv;
1592 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1593 /* Mark as non flushing */
1594 priv->srcresult = GST_FLOW_OK;
1595 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1596 priv->last_popped_seqnum = -1;
1597 priv->last_out_time = GST_CLOCK_TIME_NONE;
1598 priv->next_seqnum = -1;
1599 priv->seqnum_base = -1;
1600 priv->ips_rtptime = -1;
1601 priv->ips_pts = GST_CLOCK_TIME_NONE;
1602 priv->packet_spacing = 0;
1603 priv->next_in_seqnum = -1;
1604 priv->clock_rate = -1;
1607 priv->estimated_eos = -1;
1608 priv->last_elapsed = 0;
1609 priv->ext_timestamp = -1;
1610 priv->avg_jitter = 0;
1611 priv->last_dts = -1;
1612 priv->last_rtptime = -1;
1613 priv->last_in_pts = 0;
1614 priv->equidistant = 0;
1615 priv->segment_seqnum = GST_SEQNUM_INVALID;
1616 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1617 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1618 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1619 rtp_jitter_buffer_reset_skew (priv->jbuf);
1620 remove_all_timers (jitterbuffer);
1621 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1622 g_queue_clear (&priv->gap_packets);
1627 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1628 GstPadMode mode, gboolean active)
1631 GstRtpJitterBuffer *jitterbuffer = NULL;
1633 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1636 case GST_PAD_MODE_PUSH:
1638 /* allow data processing */
1639 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1641 /* start pushing out buffers */
1642 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1643 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1644 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1646 /* make sure all data processing stops ASAP */
1647 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1649 /* NOTE this will hardlock if the state change is called from the src pad
1650 * task thread because we will _join() the thread. */
1651 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1652 result = gst_pad_stop_task (pad);
1662 static GstStateChangeReturn
1663 gst_rtp_jitter_buffer_change_state (GstElement * element,
1664 GstStateChange transition)
1666 GstRtpJitterBuffer *jitterbuffer;
1667 GstRtpJitterBufferPrivate *priv;
1668 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1670 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1671 priv = jitterbuffer->priv;
1673 switch (transition) {
1674 case GST_STATE_CHANGE_NULL_TO_READY:
1676 case GST_STATE_CHANGE_READY_TO_PAUSED:
1678 /* reset negotiated values */
1679 priv->clock_rate = -1;
1680 priv->clock_base = -1;
1681 priv->peer_latency = 0;
1683 /* block until we go to PLAYING */
1684 priv->blocked = TRUE;
1685 priv->timer_running = TRUE;
1686 priv->timer_thread =
1687 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1690 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1692 /* unblock to allow streaming in PLAYING */
1693 priv->blocked = FALSE;
1694 JBUF_SIGNAL_EVENT (priv);
1695 JBUF_SIGNAL_TIMER (priv);
1702 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1704 switch (transition) {
1705 case GST_STATE_CHANGE_READY_TO_PAUSED:
1706 /* we are a live element because we sync to the clock, which we can only
1707 * do in the PLAYING state */
1708 if (ret != GST_STATE_CHANGE_FAILURE)
1709 ret = GST_STATE_CHANGE_NO_PREROLL;
1711 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1713 /* block to stop streaming when PAUSED */
1714 priv->blocked = TRUE;
1715 unschedule_current_timer (jitterbuffer);
1717 if (ret != GST_STATE_CHANGE_FAILURE)
1718 ret = GST_STATE_CHANGE_NO_PREROLL;
1720 case GST_STATE_CHANGE_PAUSED_TO_READY:
1722 gst_buffer_replace (&priv->last_sr, NULL);
1723 priv->timer_running = FALSE;
1724 unschedule_current_timer (jitterbuffer);
1725 JBUF_SIGNAL_TIMER (priv);
1726 JBUF_SIGNAL_QUERY (priv, FALSE);
1728 g_thread_join (priv->timer_thread);
1729 priv->timer_thread = NULL;
1731 case GST_STATE_CHANGE_READY_TO_NULL:
1741 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1744 gboolean ret = TRUE;
1745 GstRtpJitterBuffer *jitterbuffer;
1746 GstRtpJitterBufferPrivate *priv;
1748 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1749 priv = jitterbuffer->priv;
1751 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1753 switch (GST_EVENT_TYPE (event)) {
1754 case GST_EVENT_LATENCY:
1756 GstClockTime latency;
1758 gst_event_parse_latency (event, &latency);
1760 GST_DEBUG_OBJECT (jitterbuffer,
1761 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1764 /* adjust the overall buffer delay to the total pipeline latency in
1765 * buffering mode because if downstream consumes too fast (because of
1766 * large latency or queues, we would start rebuffering again. */
1767 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1768 RTP_JITTER_BUFFER_MODE_BUFFER) {
1769 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1773 ret = gst_pad_push_event (priv->sinkpad, event);
1777 ret = gst_pad_push_event (priv->sinkpad, event);
1784 /* handles and stores the event in the jitterbuffer, must be called with
1787 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1789 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1790 RTPJitterBufferItem *item;
1793 switch (GST_EVENT_TYPE (event)) {
1794 case GST_EVENT_CAPS:
1798 gst_event_parse_caps (event, &caps);
1799 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1802 case GST_EVENT_SEGMENT:
1805 gst_event_copy_segment (event, &segment);
1807 priv->segment_seqnum = gst_event_get_seqnum (event);
1809 /* we need time for now */
1810 if (segment.format != GST_FORMAT_TIME) {
1811 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1812 gst_event_unref (event);
1814 gst_segment_init (&segment, GST_FORMAT_TIME);
1815 event = gst_event_new_segment (&segment);
1816 gst_event_set_seqnum (event, priv->segment_seqnum);
1819 priv->segment = segment;
1824 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1831 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1832 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1833 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1834 if (head || priv->eos)
1835 JBUF_SIGNAL_EVENT (priv);
1841 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1844 gboolean ret = TRUE;
1845 GstRtpJitterBuffer *jitterbuffer;
1846 GstRtpJitterBufferPrivate *priv;
1848 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1849 priv = jitterbuffer->priv;
1851 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1853 switch (GST_EVENT_TYPE (event)) {
1854 case GST_EVENT_FLUSH_START:
1855 ret = gst_pad_push_event (priv->srcpad, event);
1856 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1857 /* wait for the loop to go into PAUSED */
1858 gst_pad_pause_task (priv->srcpad);
1860 case GST_EVENT_FLUSH_STOP:
1861 ret = gst_pad_push_event (priv->srcpad, event);
1863 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1864 GST_PAD_MODE_PUSH, TRUE);
1867 if (GST_EVENT_IS_SERIALIZED (event)) {
1868 /* serialized events go in the queue */
1870 if (priv->srcresult != GST_FLOW_OK) {
1871 /* Errors in sticky event pushing are no problem and ignored here
1872 * as they will cause more meaningful errors during data flow.
1873 * For EOS events, that are not followed by data flow, we still
1874 * return FALSE here though.
1876 if (!GST_EVENT_IS_STICKY (event) ||
1877 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1878 goto out_flow_error;
1880 /* refuse more events on EOS */
1883 ret = queue_event (jitterbuffer, event);
1886 /* non-serialized events are forwarded downstream immediately */
1887 ret = gst_pad_push_event (priv->srcpad, event);
1896 GST_DEBUG_OBJECT (jitterbuffer,
1897 "refusing event, we have a downstream flow error: %s",
1898 gst_flow_get_name (priv->srcresult));
1900 gst_event_unref (event);
1905 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1907 gst_event_unref (event);
1913 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1916 gboolean ret = TRUE;
1917 GstRtpJitterBuffer *jitterbuffer;
1919 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1921 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1923 switch (GST_EVENT_TYPE (event)) {
1924 case GST_EVENT_FLUSH_START:
1925 gst_event_unref (event);
1927 case GST_EVENT_FLUSH_STOP:
1928 gst_event_unref (event);
1931 ret = gst_pad_event_default (pad, parent, event);
1939 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1940 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1941 * GST_FLOW_FLUSHING when the element is shutting down. On success
1942 * GST_FLOW_OK is returned.
1944 static GstFlowReturn
1945 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1949 GValue args[2] = { {0}, {0} };
1953 g_value_init (&args[0], GST_TYPE_ELEMENT);
1954 g_value_set_object (&args[0], jitterbuffer);
1955 g_value_init (&args[1], G_TYPE_UINT);
1956 g_value_set_uint (&args[1], pt);
1958 g_value_init (&ret, GST_TYPE_CAPS);
1959 g_value_set_boxed (&ret, NULL);
1961 JBUF_UNLOCK (jitterbuffer->priv);
1962 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1964 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1966 g_value_unset (&args[0]);
1967 g_value_unset (&args[1]);
1968 caps = (GstCaps *) g_value_dup_boxed (&ret);
1969 g_value_unset (&ret);
1973 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1974 gst_caps_unref (caps);
1976 if (G_UNLIKELY (!res))
1984 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1985 return GST_FLOW_ERROR;
1989 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1990 return GST_FLOW_FLUSHING;
1994 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1995 return GST_FLOW_ERROR;
1999 /* call with jbuf lock held */
2001 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
2003 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2004 GstMessage *message = NULL;
2009 /* Post a buffering message */
2010 if (priv->last_percent != percent) {
2011 priv->last_percent = percent;
2013 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
2014 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
2021 update_offset (GstRtpJitterBuffer * jitterbuffer)
2023 GstRtpJitterBufferPrivate *priv;
2025 priv = jitterbuffer->priv;
2027 if (priv->ts_offset_remainder != 0) {
2028 GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT
2029 " off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment,
2030 priv->ts_offset_remainder, priv->ts_offset);
2031 if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) {
2032 if (priv->ts_offset_remainder > 0) {
2033 priv->ts_offset += priv->max_ts_offset_adjustment;
2034 priv->ts_offset_remainder -= priv->max_ts_offset_adjustment;
2036 priv->ts_offset -= priv->max_ts_offset_adjustment;
2037 priv->ts_offset_remainder += priv->max_ts_offset_adjustment;
2040 priv->ts_offset += priv->ts_offset_remainder;
2041 priv->ts_offset_remainder = 0;
2047 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
2049 GstRtpJitterBufferPrivate *priv;
2051 priv = jitterbuffer->priv;
2053 if (timestamp == -1)
2056 /* apply the timestamp offset, this is used for inter stream sync */
2057 timestamp += priv->ts_offset;
2058 /* add the offset, this is used when buffering */
2059 timestamp += priv->out_offset;
2065 timer_queue_new (void)
2069 queue = g_slice_new (TimerQueue);
2070 queue->timers = g_queue_new ();
2071 queue->hashtable = g_hash_table_new (NULL, NULL);
2077 timer_queue_free (TimerQueue * queue)
2082 g_hash_table_destroy (queue->hashtable);
2083 g_queue_free_full (queue->timers, g_free);
2084 g_slice_free (TimerQueue, queue);
2088 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2089 GstClockTime timeout, gboolean lost)
2093 copy = g_memdup (timer, sizeof (*timer));
2094 copy->timeout = timeout;
2095 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2098 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2099 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2100 g_queue_push_tail (queue->timers, copy);
2101 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2105 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2109 test = g_queue_peek_head (queue->timers);
2110 while (test && test->timeout < timeout) {
2111 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2112 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2113 GST_TIME_ARGS (timeout));
2114 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2115 g_free (g_queue_pop_head (queue->timers));
2116 test = g_queue_peek_head (queue->timers);
2121 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2123 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2127 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2129 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2130 TimerData *timer = NULL;
2133 len = priv->timers->len;
2134 for (i = 0; i < len; i++) {
2135 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2136 if (test->seqnum == seqnum) {
2145 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2147 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2149 if (priv->clock_id) {
2150 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2151 gst_clock_id_unschedule (priv->clock_id);
2152 priv->clock_id = NULL;
2157 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2159 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2160 GstClockTime test_timeout;
2162 if ((test_timeout = timer->timeout) == -1)
2165 if (timer->type != TIMER_TYPE_EXPECTED) {
2166 /* add our latency and offset to get output times. */
2167 test_timeout = apply_offset (jitterbuffer, test_timeout);
2168 test_timeout += priv->latency_ns;
2170 return test_timeout;
2174 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2176 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2178 if (priv->clock_id) {
2179 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2181 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2182 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2184 if (timeout == -1 || timeout < priv->timer_timeout)
2185 unschedule_current_timer (jitterbuffer);
2190 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2191 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2192 GstClockTime duration)
2194 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2198 GST_DEBUG_OBJECT (jitterbuffer,
2199 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2200 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2201 GST_TIME_ARGS (delay));
2203 len = priv->timers->len;
2204 g_array_set_size (priv->timers, len + 1);
2205 timer = &g_array_index (priv->timers, TimerData, len);
2208 timer->seqnum = seqnum;
2210 timer->timeout = timeout + delay;
2211 timer->duration = duration;
2212 if (type == TIMER_TYPE_EXPECTED) {
2213 timer->rtx_base = timeout;
2214 timer->rtx_delay = delay;
2215 timer->rtx_retry = 0;
2217 timer->rtx_last = GST_CLOCK_TIME_NONE;
2218 timer->num_rtx_retry = 0;
2219 timer->num_rtx_received = 0;
2220 recalculate_timer (jitterbuffer, timer);
2221 JBUF_SIGNAL_TIMER (priv);
2227 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2228 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2230 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2231 gboolean seqchange, timechange;
2233 GstClockTime new_timeout;
2235 oldseq = timer->seqnum;
2236 new_timeout = timeout + delay;
2237 seqchange = oldseq != seqnum;
2238 timechange = timer->timeout != new_timeout;
2240 if (!seqchange && !timechange) {
2241 GST_DEBUG_OBJECT (jitterbuffer,
2242 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2243 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2247 GST_DEBUG_OBJECT (jitterbuffer,
2248 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2249 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2250 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2252 timer->timeout = new_timeout;
2253 timer->seqnum = seqnum;
2255 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2256 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2257 GST_TIME_ARGS (delay));
2258 timer->rtx_base = timeout;
2259 timer->rtx_delay = delay;
2260 timer->rtx_retry = 0;
2263 timer->num_rtx_retry = 0;
2264 timer->num_rtx_received = 0;
2267 if (priv->clock_id) {
2268 /* we changed the seqnum and there is a timer currently waiting with this
2269 * seqnum, unschedule it */
2270 if (seqchange && priv->timer_seqnum == oldseq)
2271 unschedule_current_timer (jitterbuffer);
2272 /* we changed the time, check if it is earlier than what we are waiting
2273 * for and unschedule if so */
2274 else if (timechange)
2275 recalculate_timer (jitterbuffer, timer);
2280 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2281 guint16 seqnum, GstClockTime timeout)
2285 /* find the seqnum timer */
2286 timer = find_timer (jitterbuffer, seqnum);
2287 if (timer == NULL) {
2288 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2290 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2296 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2298 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2301 if (timer->idx == -1)
2304 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2305 unschedule_current_timer (jitterbuffer);
2308 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2309 g_array_remove_index_fast (priv->timers, idx);
2312 JBUF_SIGNAL_TIMER (priv);
2316 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2318 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2319 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2320 g_array_set_size (priv->timers, 0);
2321 unschedule_current_timer (jitterbuffer);
2322 JBUF_SIGNAL_TIMER (priv);
2325 /* get the extra delay to wait before sending RTX */
2327 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2331 if (priv->rtx_delay == -1) {
2332 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2333 delay = DEFAULT_AUTO_RTX_DELAY;
2335 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2336 * packet spacing is a good margin */
2337 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2340 delay = priv->rtx_delay * GST_MSECOND;
2342 if (priv->rtx_min_delay > 0)
2343 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2348 /* Check if packet with seqnum is already considered definitely lost by being
2349 * part of a "lost timer" for multiple packets */
2351 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2353 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2356 len = priv->timers->len;
2357 for (i = 0; i < len; i++) {
2358 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2359 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2361 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2363 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2364 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2372 /* we just received a packet with seqnum and dts.
2374 * First check for old seqnum that we are still expecting. If the gap with the
2375 * current seqnum is too big, unschedule the timeouts.
2377 * If we have a valid packet spacing estimate we can set a timer for when we
2378 * should receive the next packet.
2379 * If we don't have a valid estimate, we remove any timer we might have
2380 * had for this packet.
2383 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2384 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2385 gboolean is_rtx, TimerData * timer)
2387 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2389 /* go through all timers and unschedule the ones with a large gap */
2390 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2392 len = priv->timers->len;
2393 for (i = 0; i < len; i++) {
2394 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2397 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2399 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2400 test->type, test->seqnum, seqnum, gap);
2402 if (gap > priv->rtx_delay_reorder) {
2403 /* max gap, we exceeded the max reorder distance and we don't expect the
2404 * missing packet to be this reordered */
2405 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2406 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2411 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2412 && priv->do_retransmission && priv->rtx_next_seqnum;
2414 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2415 if (timer->num_rtx_retry > 0) {
2417 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2418 /* don't try to estimate the next seqnum because this is a retransmitted
2419 * packet and it probably did not arrive with the expected packet
2421 do_next_seqnum = FALSE;
2424 if (!is_rtx || timer->num_rtx_retry > 1) {
2425 /* Store timer in order to record stats when/if the retransmitted
2426 * packet arrives. We should also store timer information if we've
2427 * requested retransmission more than once since we may receive
2428 * several retransmitted packets. For accuracy we should update the
2429 * stats also when the redundant retransmitted packets arrives. */
2430 timer_queue_append (priv->rtx_stats_timers, timer,
2431 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2436 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2437 GstClockTime expected, delay;
2439 /* calculate expected arrival time of the next seqnum */
2440 expected = pts + priv->packet_spacing;
2442 delay = get_rtx_delay (priv);
2444 /* and update/install timer for next seqnum */
2445 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2446 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2447 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2448 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2449 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2452 timer->type = TIMER_TYPE_EXPECTED;
2453 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2456 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2457 expected, delay, priv->packet_spacing);
2459 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2460 /* if we had a timer, remove it, we don't know when to expect the next
2462 remove_timer (jitterbuffer, timer);
2467 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2470 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2472 /* we need consecutive seqnums with a different
2473 * rtptime to estimate the packet spacing. */
2474 if (priv->ips_rtptime != rtptime) {
2475 /* rtptime changed, check pts diff */
2476 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2477 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2478 GstClockTime old_packet_spacing = priv->packet_spacing;
2480 /* Biased towards bigger packet spacings to prevent
2481 * too many unneeded retransmission requests for next
2482 * packets that just arrive a little later than we would
2484 if (old_packet_spacing > new_packet_spacing)
2485 priv->packet_spacing =
2486 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2487 else if (old_packet_spacing > 0)
2488 priv->packet_spacing =
2489 (3 * new_packet_spacing + old_packet_spacing) / 4;
2491 priv->packet_spacing = new_packet_spacing;
2493 GST_DEBUG_OBJECT (jitterbuffer,
2494 "new packet spacing %" GST_TIME_FORMAT
2495 " old packet spacing %" GST_TIME_FORMAT
2496 " combined to %" GST_TIME_FORMAT,
2497 GST_TIME_ARGS (new_packet_spacing),
2498 GST_TIME_ARGS (old_packet_spacing),
2499 GST_TIME_ARGS (priv->packet_spacing));
2501 priv->ips_rtptime = rtptime;
2502 priv->ips_pts = pts;
2507 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2508 guint16 seqnum, GstClockTime pts, gint gap)
2510 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2511 GstClockTime duration, expected_pts, delay;
2513 gboolean equidistant = priv->equidistant > 0;
2515 GST_DEBUG_OBJECT (jitterbuffer,
2516 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2517 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2519 if (pts == GST_CLOCK_TIME_NONE) {
2520 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2525 GstClockTime total_duration;
2526 /* the total duration spanned by the missing packets */
2527 if (pts >= priv->last_in_pts)
2528 total_duration = pts - priv->last_in_pts;
2532 /* interpolate between the current time and the last time based on
2533 * number of packets we are missing, this is the estimated duration
2534 * for the missing packet based on equidistant packet spacing. */
2535 duration = total_duration / (gap + 1);
2537 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2538 GST_TIME_ARGS (duration));
2540 if (total_duration > priv->latency_ns) {
2541 GstClockTime gap_time;
2545 GstClockTime gap_dur = gap * duration;
2546 if (gap_dur > priv->latency_ns)
2547 gap_time = gap_dur - priv->latency_ns;
2550 lost_packets = gap_time / duration;
2552 gap_time = total_duration - priv->latency_ns;
2556 /* too many lost packets, some of the missing packets are already
2557 * too late and we can generate lost packet events for them. */
2558 GST_INFO_OBJECT (jitterbuffer,
2559 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2560 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2561 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2562 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2563 GST_TIME_ARGS (gap_time));
2565 /* this timer will fire immediately and the lost event will be pushed from
2566 * the timer thread */
2567 if (lost_packets > 0) {
2568 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2569 priv->last_in_pts + duration, 0, gap_time);
2570 expected += lost_packets;
2571 priv->last_in_pts += gap_time;
2575 expected_pts = priv->last_in_pts + duration;
2577 /* If we cannot assume equidistant packet spacing, the only thing we now
2578 * for sure is that the missing packets have expected pts not later than
2579 * the last received pts. */
2586 if (priv->do_retransmission) {
2587 TimerData *timer = find_timer (jitterbuffer, expected);
2589 type = TIMER_TYPE_EXPECTED;
2590 delay = get_rtx_delay (priv);
2592 /* if we had a timer for the first missing packet, update it. */
2593 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2594 GstClockTime timeout = timer->timeout;
2596 timer->duration = duration;
2597 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2598 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2602 expected_pts += duration;
2605 type = TIMER_TYPE_LOST;
2608 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2609 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2610 expected_pts += duration;
2616 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2620 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2621 GstRtpJitterBufferPrivate *priv;
2623 priv = jitterbuffer->priv;
2625 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2628 if (priv->last_dts != -1)
2629 dtsdiff = dts - priv->last_dts;
2633 if (priv->last_rtptime != -1)
2634 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2638 /* Guess whether stream currently uses equidistant packet spacing. If we
2639 * often see identical timestamps it means the packets are not
2641 if (rtptime == priv->last_rtptime)
2642 priv->equidistant -= 2;
2644 priv->equidistant += 1;
2645 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2647 priv->last_dts = dts;
2648 priv->last_rtptime = rtptime;
2652 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2655 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2657 diff = ABS (dtsdiff - rtpdiffns);
2659 /* jitter is stored in nanoseconds */
2660 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2662 GST_LOG_OBJECT (jitterbuffer,
2663 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2664 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2665 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2666 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2673 GST_DEBUG_OBJECT (jitterbuffer,
2674 "no dts or no clock-rate, can't calculate jitter");
2680 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2682 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2683 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2686 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2687 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2688 gst_rtp_buffer_unmap (&rtp_a);
2690 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2691 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2692 gst_rtp_buffer_unmap (&rtp_b);
2694 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2698 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2699 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2701 GstRtpJitterBufferPrivate *priv;
2702 guint gap_packets_length;
2703 gboolean reset = FALSE;
2704 gboolean future = gap > 0;
2706 priv = jitterbuffer->priv;
2708 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2710 guint32 prev_gap_seq = -1;
2711 gboolean all_consecutive = TRUE;
2713 g_queue_insert_sorted (&priv->gap_packets, buffer,
2714 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2716 for (l = priv->gap_packets.head; l; l = l->next) {
2717 GstBuffer *gap_buffer = l->data;
2718 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2721 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2723 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2725 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2726 if (prev_gap_seq == -1)
2727 prev_gap_seq = gap_seq;
2728 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2729 all_consecutive = FALSE;
2731 prev_gap_seq = gap_seq;
2733 gst_rtp_buffer_unmap (&gap_rtp);
2734 if (!all_consecutive)
2738 if (all_consecutive && gap_packets_length > 3) {
2739 GST_DEBUG_OBJECT (jitterbuffer,
2740 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2741 (future ? "new" : "old"), gap,
2742 (future ? max_dropout : -max_misorder));
2744 } else if (!all_consecutive) {
2745 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2746 g_queue_clear (&priv->gap_packets);
2747 GST_DEBUG_OBJECT (jitterbuffer,
2748 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2749 (future ? "new" : "old"), gap,
2750 (future ? max_dropout : -max_misorder));
2753 GST_DEBUG_OBJECT (jitterbuffer,
2754 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2755 (future ? "new" : "old"), gap,
2756 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2760 GST_DEBUG_OBJECT (jitterbuffer,
2761 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2762 gap, -max_misorder);
2763 g_queue_push_tail (&priv->gap_packets, buffer);
2771 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2773 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2774 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2777 GstClockTime base_time =
2778 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2779 GstClockTime clock_time = gst_clock_get_time (clock);
2781 if (clock_time > base_time)
2782 running_time = clock_time - base_time;
2786 gst_object_unref (clock);
2789 return running_time;
2792 static GstFlowReturn
2793 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2794 GstPad * pad, GstObject * parent, guint16 seqnum)
2796 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2797 GstFlowReturn ret = GST_FLOW_OK;
2798 GList *events = NULL, *l;
2802 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2803 rtp_jitter_buffer_flush (priv->jbuf,
2804 (GFunc) free_item_and_retain_events, &events);
2805 rtp_jitter_buffer_reset_skew (priv->jbuf);
2806 remove_all_timers (jitterbuffer);
2807 priv->discont = TRUE;
2808 priv->last_popped_seqnum = -1;
2810 if (priv->gap_packets.head) {
2811 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2812 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2814 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2815 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2816 gst_rtp_buffer_unmap (&gap_rtp);
2818 priv->next_seqnum = seqnum;
2821 priv->last_in_pts = -1;
2822 priv->next_in_seqnum = -1;
2824 /* Insert all sticky events again in order, otherwise we would
2825 * potentially loose STREAM_START, CAPS or SEGMENT events
2827 events = g_list_reverse (events);
2828 for (l = events; l; l = l->next) {
2829 RTPJitterBufferItem *item;
2831 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2832 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2834 g_list_free (events);
2836 JBUF_SIGNAL_EVENT (priv);
2838 /* reset spacing estimation when gap */
2839 priv->ips_rtptime = -1;
2840 priv->ips_pts = GST_CLOCK_TIME_NONE;
2842 buffers = g_list_copy (priv->gap_packets.head);
2843 g_queue_clear (&priv->gap_packets);
2845 priv->ips_rtptime = -1;
2846 priv->ips_pts = GST_CLOCK_TIME_NONE;
2847 JBUF_UNLOCK (jitterbuffer->priv);
2849 for (l = buffers; l; l = l->next) {
2850 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2852 if (ret != GST_FLOW_OK) {
2857 for (; l; l = l->next)
2858 gst_buffer_unref (l->data);
2859 g_list_free (buffers);
2865 gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
2867 GstRtpJitterBufferPrivate *priv;
2868 RTPJitterBufferItem *item;
2871 priv = jitterbuffer->priv;
2873 if (priv->faststart_min_packets == 0)
2876 item = rtp_jitter_buffer_peek (priv->jbuf);
2880 timer = find_timer (jitterbuffer, item->seqnum);
2881 if (!timer || timer->type != TIMER_TYPE_DEADLINE)
2884 if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
2885 priv->faststart_min_packets)) {
2886 GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
2887 priv->faststart_min_packets);
2888 timer->timeout = -1;
2895 static GstFlowReturn
2896 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2899 GstRtpJitterBuffer *jitterbuffer;
2900 GstRtpJitterBufferPrivate *priv;
2902 guint32 expected, rtptime;
2903 GstFlowReturn ret = GST_FLOW_OK;
2904 GstClockTime dts, pts;
2909 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2910 gboolean do_next_seqnum = FALSE;
2911 RTPJitterBufferItem *item;
2912 GstMessage *msg = NULL;
2913 gboolean estimated_dts = FALSE;
2914 gint32 packet_rate, max_dropout, max_misorder;
2915 TimerData *timer = NULL;
2917 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2919 priv = jitterbuffer->priv;
2921 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2922 goto invalid_buffer;
2924 pt = gst_rtp_buffer_get_payload_type (&rtp);
2925 seqnum = gst_rtp_buffer_get_seq (&rtp);
2926 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2927 gst_rtp_buffer_unmap (&rtp);
2929 /* make sure we have PTS and DTS set */
2930 pts = GST_BUFFER_PTS (buffer);
2931 dts = GST_BUFFER_DTS (buffer);
2938 /* If we have no DTS here, i.e. no capture time, get one from the
2939 * clock now to have something to calculate with in the future. */
2940 dts = get_current_running_time (jitterbuffer);
2943 /* Remember that we estimated the DTS if we are running already
2944 * and this is not our first packet (or first packet after a reset).
2945 * If it's the first packet, we somehow must generate a timestamp for
2946 * everything, otherwise we can't calculate any times
2948 estimated_dts = (priv->next_in_seqnum != -1);
2950 /* take the DTS of the buffer. This is the time when the packet was
2951 * received and is used to calculate jitter and clock skew. We will adjust
2952 * this DTS with the smoothed value after processing it in the
2953 * jitterbuffer and assign it as the PTS. */
2954 /* bring to running time */
2955 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2958 GST_DEBUG_OBJECT (jitterbuffer,
2959 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2960 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2961 GST_BUFFER_IS_RETRANSMISSION (buffer));
2963 JBUF_LOCK_CHECK (priv, out_flushing);
2965 if (G_UNLIKELY (priv->last_pt != pt)) {
2968 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2972 /* reset clock-rate so that we get a new one */
2973 priv->clock_rate = -1;
2975 /* Try to get the clock-rate from the caps first if we can. If there are no
2976 * caps we must fire the signal to get the clock-rate. */
2977 if ((caps = gst_pad_get_current_caps (pad))) {
2978 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2979 gst_caps_unref (caps);
2983 if (G_UNLIKELY (priv->clock_rate == -1)) {
2984 /* no clock rate given on the caps, try to get one with the signal */
2985 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2986 pt) == GST_FLOW_FLUSHING)
2989 if (G_UNLIKELY (priv->clock_rate == -1))
2992 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2995 /* don't accept more data on EOS */
2996 if (G_UNLIKELY (priv->eos))
2999 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
3000 calculate_jitter (jitterbuffer, dts, rtptime);
3002 if (priv->seqnum_base != -1) {
3005 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
3008 GST_DEBUG_OBJECT (jitterbuffer,
3009 "packet seqnum #%d before seqnum-base #%d", seqnum,
3011 gst_buffer_unref (buffer);
3013 } else if (gap > 16384) {
3014 /* From now on don't compare against the seqnum base anymore as
3015 * at some point in the future we will wrap around and also that
3016 * much reordering is very unlikely */
3017 priv->seqnum_base = -1;
3021 expected = priv->next_in_seqnum;
3024 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
3026 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
3027 priv->max_dropout_time);
3029 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
3030 priv->max_misorder_time);
3031 GST_TRACE_OBJECT (jitterbuffer,
3032 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
3033 max_dropout, max_misorder);
3035 /* now check against our expected seqnum */
3036 if (G_UNLIKELY (expected == -1)) {
3037 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3039 /* calculate a pts based on rtptime and arrival time (dts) */
3041 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3042 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3044 /* we don't know what the next_in_seqnum should be, wait for the last
3045 * possible moment to push this buffer, maybe we get an earlier seqnum
3047 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
3049 do_next_seqnum = TRUE;
3050 /* take rtptime and pts to calculate packet spacing */
3051 priv->ips_rtptime = rtptime;
3052 priv->ips_pts = pts;
3056 /* now calculate gap */
3057 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
3058 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
3059 expected, seqnum, gap);
3061 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
3062 /* If we have timers for more than RTP_MAX_DROPOUT packets
3063 * pending this means that we have a huge gap overall. We can
3064 * reset the jitterbuffer at this point because there's
3065 * just too much data missing to be able to do anything
3066 * sensible with the past data. Just try again from the
3068 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
3069 priv->timers->len, max_dropout);
3070 gst_buffer_unref (buffer);
3071 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3074 /* Special handling of large gaps */
3075 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
3076 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
3077 gap, max_dropout, max_misorder);
3079 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3081 GST_DEBUG_OBJECT (jitterbuffer,
3082 "Had big gap, waiting for more consecutive packets");
3087 /* We had no huge gap, let's drop all the gap packets */
3088 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
3089 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3090 g_queue_clear (&priv->gap_packets);
3092 /* calculate a pts based on rtptime and arrival time (dts) */
3093 /* If we estimated the DTS, don't consider it in the clock skew calculations */
3095 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3096 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3098 if (G_LIKELY (gap == 0)) {
3099 /* packet is expected */
3100 calculate_packet_spacing (jitterbuffer, rtptime, pts);
3101 do_next_seqnum = TRUE;
3106 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
3107 /* fill in the gap with EXPECTED timers */
3108 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
3109 do_next_seqnum = TRUE;
3111 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3112 do_next_seqnum = FALSE;
3115 /* reset spacing estimation when gap */
3116 priv->ips_rtptime = -1;
3117 priv->ips_pts = GST_CLOCK_TIME_NONE;
3121 if (do_next_seqnum) {
3122 priv->last_in_pts = pts;
3123 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3126 timer = find_timer (jitterbuffer, seqnum);
3127 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3129 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3131 timer->num_rtx_received++;
3134 /* let's check if this buffer is too late, we can only accept packets with
3135 * bigger seqnum than the one we last pushed. */
3136 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3139 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3141 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3142 if (G_UNLIKELY (gap <= 0)) {
3143 if (priv->do_retransmission) {
3144 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3145 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3146 /* Only count the retranmitted packet too late if it has been
3147 * considered lost. If the original packet arrived before the
3148 * retransmitted we just count it as a duplicate. */
3149 if (timer->type != TIMER_TYPE_LOST)
3157 if (already_lost (jitterbuffer, seqnum))
3160 /* let's drop oldest packet if the queue is already full and drop-on-latency
3161 * is set. We can only do this when there actually is a latency. When no
3162 * latency is set, we just pump it in the queue and let the other end push it
3163 * out as fast as possible. */
3164 if (priv->latency_ms && priv->drop_on_latency) {
3166 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3168 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3169 RTPJitterBufferItem *old_item;
3171 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3173 if (IS_DROPABLE (old_item)) {
3174 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3175 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3177 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3178 free_item (old_item);
3180 /* we might have removed some head buffers, signal the pushing thread to
3181 * see if it can push now */
3182 JBUF_SIGNAL_EVENT (priv);
3186 /* If we estimated the DTS, don't consider it in the clock skew calculations
3187 * later. The code above always sets dts to pts or the other way around if
3188 * any of those is valid in the buffer, so we know that if we estimated the
3189 * dts that both are unknown */
3192 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3193 pts, seqnum, 1, rtptime);
3195 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3197 /* now insert the packet into the queue in sorted order. This function returns
3198 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3199 * have a duplicate. */
3200 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3202 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3203 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3207 /* Trigger fast start if needed */
3208 if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
3212 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3213 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3215 /* we had an unhandled SR, handle it now */
3217 do_handle_sync (jitterbuffer);
3219 if (G_UNLIKELY (head)) {
3220 /* signal addition of new buffer when the _loop is waiting. */
3221 if (G_LIKELY (priv->active))
3222 JBUF_SIGNAL_EVENT (priv);
3224 /* let's unschedule and unblock any waiting buffers. We only want to do this
3225 * when the head buffer changed */
3226 if (G_UNLIKELY (priv->clock_id)) {
3227 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3228 unschedule_current_timer (jitterbuffer);
3232 GST_DEBUG_OBJECT (jitterbuffer,
3233 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3234 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3236 msg = check_buffering_percent (jitterbuffer, percent);
3242 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3249 /* this is not fatal but should be filtered earlier */
3250 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3251 ("Received invalid RTP payload, dropping"));
3252 gst_buffer_unref (buffer);
3257 GST_WARNING_OBJECT (jitterbuffer,
3258 "No clock-rate in caps!, dropping buffer");
3259 gst_buffer_unref (buffer);
3264 ret = priv->srcresult;
3265 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3266 gst_buffer_unref (buffer);
3272 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3273 gst_buffer_unref (buffer);
3278 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3279 " popped, dropping", seqnum, priv->last_popped_seqnum);
3281 gst_buffer_unref (buffer);
3286 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3287 "considered lost", seqnum);
3289 gst_buffer_unref (buffer);
3294 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3296 priv->num_duplicates++;
3302 GST_DEBUG_OBJECT (jitterbuffer,
3303 "Duplicate RTX packet #%d detected, dropping", seqnum);
3304 priv->num_duplicates++;
3305 gst_buffer_unref (buffer);
3310 /* FIXME: hopefully we can do something more efficient here, especially when
3311 * all packets are in order and/or outside of the currently cached range.
3312 * Still worthwhile to have it, avoids taking/releasing object lock and pad
3313 * stream lock for every single buffer in the default chain_list fallback. */
3314 static GstFlowReturn
3315 gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent,
3316 GstBufferList * buffer_list)
3318 GstFlowReturn flow_ret = GST_FLOW_OK;
3321 n = gst_buffer_list_length (buffer_list);
3322 for (i = 0; i < n; ++i) {
3323 GstBuffer *buf = gst_buffer_list_get (buffer_list, i);
3325 flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf));
3327 if (flow_ret != GST_FLOW_OK)
3330 gst_buffer_list_unref (buffer_list);
3336 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3338 guint64 ext_time, elapsed;
3340 GstRtpJitterBufferPrivate *priv;
3342 priv = jitterbuffer->priv;
3343 rtp_time = item->rtptime;
3345 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3346 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3348 ext_time = priv->ext_timestamp;
3349 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3350 if (ext_time < priv->ext_timestamp) {
3351 ext_time = priv->ext_timestamp;
3353 priv->ext_timestamp = ext_time;
3356 if (ext_time > priv->clock_base)
3357 elapsed = ext_time - priv->clock_base;
3361 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3366 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3367 RTPJitterBufferItem * item)
3369 guint64 total, elapsed, left, estimated;
3370 GstClockTime out_time;
3371 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3373 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3374 || priv->clock_base == -1 || priv->clock_rate <= 0)
3377 /* compute the elapsed time */
3378 elapsed = compute_elapsed (jitterbuffer, item);
3380 /* do nothing if elapsed time doesn't increment */
3381 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3384 priv->last_elapsed = elapsed;
3386 /* this is the total time we need to play */
3387 total = priv->npt_stop - priv->npt_start;
3388 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3389 GST_TIME_ARGS (total));
3391 /* this is how much time there is left */
3392 if (total > elapsed)
3393 left = total - elapsed;
3397 /* if we have less time left that the size of the buffer, we will not
3398 * be able to keep it filled, disabled buffering then */
3399 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3400 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3401 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3402 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3405 /* this is the current time as running-time */
3406 out_time = item->pts;
3409 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3411 /* if there is almost nothing left,
3412 * we may never advance enough to end up in the above case */
3413 if (total < GST_SECOND)
3414 estimated = GST_SECOND;
3418 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3419 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3421 if (estimated != -1 && priv->estimated_eos != estimated) {
3422 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3423 priv->estimated_eos = estimated;
3427 /* take a buffer from the queue and push it */
3428 static GstFlowReturn
3429 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3431 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3432 GstFlowReturn result = GST_FLOW_OK;
3433 RTPJitterBufferItem *item;
3434 GstBuffer *outbuf = NULL;
3435 GstEvent *outevent = NULL;
3436 GstQuery *outquery = NULL;
3437 GstClockTime dts, pts;
3439 gboolean do_push = TRUE;
3443 /* when we get here we are ready to pop and push the buffer */
3444 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3448 case ITEM_TYPE_BUFFER:
3450 /* we need to make writable to change the flags and timestamps */
3451 outbuf = gst_buffer_make_writable (item->data);
3453 if (G_UNLIKELY (priv->discont)) {
3454 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3455 * into the jitterbuffer so we can modify now. */
3456 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3457 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3458 priv->discont = FALSE;
3460 if (G_UNLIKELY (priv->ts_discont)) {
3461 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3462 priv->ts_discont = FALSE;
3466 gst_segment_position_from_running_time (&priv->segment,
3467 GST_FORMAT_TIME, item->dts);
3469 gst_segment_position_from_running_time (&priv->segment,
3470 GST_FORMAT_TIME, item->pts);
3472 /* if this is a new frame, check if ts_offset needs to be updated */
3473 if (pts != priv->last_pts) {
3474 update_offset (jitterbuffer);
3477 /* apply timestamp with offset to buffer now */
3478 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3479 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3481 /* update the elapsed time when we need to check against the npt stop time. */
3482 update_estimated_eos (jitterbuffer, item);
3484 priv->last_pts = pts;
3485 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3487 case ITEM_TYPE_LOST:
3488 priv->discont = TRUE;
3492 case ITEM_TYPE_EVENT:
3493 outevent = item->data;
3495 case ITEM_TYPE_QUERY:
3496 outquery = item->data;
3500 /* now we are ready to push the buffer. Save the seqnum and release the lock
3501 * so the other end can push stuff in the queue again. */
3503 priv->last_popped_seqnum = seqnum;
3504 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3506 msg = check_buffering_percent (jitterbuffer, percent);
3508 if (type == ITEM_TYPE_EVENT && outevent &&
3509 GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3510 g_assert (priv->eos);
3511 while (priv->timers->len > 0) {
3512 /* Stopping timers */
3513 unschedule_current_timer (jitterbuffer);
3514 JBUF_WAIT_TIMER (priv);
3524 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3527 case ITEM_TYPE_BUFFER:
3529 GST_DEBUG_OBJECT (jitterbuffer,
3530 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3531 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3532 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3534 result = gst_pad_push (priv->srcpad, outbuf);
3536 JBUF_LOCK_CHECK (priv, out_flushing);
3538 case ITEM_TYPE_LOST:
3539 case ITEM_TYPE_EVENT:
3540 /* We got not enough consecutive packets with a huge gap, we can
3541 * as well just drop them here now on EOS */
3542 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3543 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3544 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3545 g_queue_clear (&priv->gap_packets);
3548 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3549 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3552 gst_pad_push_event (priv->srcpad, outevent);
3554 gst_event_unref (outevent);
3556 result = GST_FLOW_OK;
3558 JBUF_LOCK_CHECK (priv, out_flushing);
3560 case ITEM_TYPE_QUERY:
3564 res = gst_pad_peer_query (priv->srcpad, outquery);
3566 JBUF_LOCK_CHECK (priv, out_flushing);
3567 result = GST_FLOW_OK;
3568 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3569 JBUF_SIGNAL_QUERY (priv, res);
3578 return priv->srcresult;
3582 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3584 /* Peek a buffer and compare the seqnum to the expected seqnum.
3585 * If all is fine, the buffer is pushed.
3586 * If something is wrong, we wait for some event
3588 static GstFlowReturn
3589 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3591 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3592 GstFlowReturn result;
3593 RTPJitterBufferItem *item;
3595 guint32 next_seqnum;
3597 /* only push buffers when PLAYING and active and not buffering */
3598 if (priv->blocked || !priv->active ||
3599 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3600 return GST_FLOW_WAIT;
3603 /* peek a buffer, we're just looking at the sequence number.
3604 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3605 * wait for a timeout or something to change.
3606 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3607 item = rtp_jitter_buffer_peek (priv->jbuf);
3612 /* get the seqnum and the next expected seqnum */
3613 seqnum = item->seqnum;
3615 return pop_and_push_next (jitterbuffer, seqnum);
3618 next_seqnum = priv->next_seqnum;
3620 /* get the gap between this and the previous packet. If we don't know the
3621 * previous packet seqnum assume no gap. */
3622 if (G_UNLIKELY (next_seqnum == -1)) {
3623 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3624 /* we don't know what the next_seqnum should be, the chain function should
3625 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3626 * fires, so wait for that */
3627 result = GST_FLOW_WAIT;
3629 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3631 if (G_LIKELY (gap == 0)) {
3632 /* no missing packet, pop and push */
3633 result = pop_and_push_next (jitterbuffer, seqnum);
3634 } else if (G_UNLIKELY (gap < 0)) {
3635 /* if we have a packet that we already pushed or considered dropped, pop it
3636 * off and get the next packet */
3637 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3638 seqnum, next_seqnum);
3639 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3641 result = GST_FLOW_OK;
3643 /* the chain function has scheduled timers to request retransmission or
3644 * when to consider the packet lost, wait for that */
3645 GST_DEBUG_OBJECT (jitterbuffer,
3646 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3647 next_seqnum, seqnum, gap);
3648 /* if we have reached EOS, just keep processing */
3650 result = pop_and_push_next (jitterbuffer, seqnum);
3651 result = GST_FLOW_OK;
3653 result = GST_FLOW_WAIT;
3662 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3664 return GST_FLOW_EOS;
3666 return GST_FLOW_WAIT;
3672 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3674 GstClockTime rtx_retry_timeout;
3675 GstClockTime rtx_min_retry_timeout;
3677 if (priv->rtx_retry_timeout == -1) {
3678 if (priv->avg_rtx_rtt == 0)
3679 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3681 /* we want to ask for a retransmission after we waited for a
3682 * complete RTT and the additional jitter */
3683 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3685 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3687 /* make sure we don't retry too often. On very low latency networks,
3688 * the RTT and jitter can be very low. */
3689 if (priv->rtx_min_retry_timeout == -1) {
3690 rtx_min_retry_timeout = priv->packet_spacing;
3692 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3694 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3696 return rtx_retry_timeout;
3700 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3701 GstClockTime rtx_retry_timeout)
3703 GstClockTime rtx_retry_period;
3705 if (priv->rtx_retry_period == -1) {
3706 /* we retry up to the configured jitterbuffer size but leaving some
3707 * room for the retransmission to arrive in time */
3708 if (rtx_retry_timeout > priv->latency_ns) {
3709 rtx_retry_period = 0;
3711 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3714 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3716 return rtx_retry_period;
3720 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3721 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3722 3. For very large measurements (> avg * 2), consider them "outliers"
3723 and count them a lot less (1/48th)
3726 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3730 if (priv->avg_rtx_rtt == 0) {
3731 priv->avg_rtx_rtt = rtt;
3735 if (rtt > 2 * priv->avg_rtx_rtt)
3737 else if (rtt > priv->avg_rtx_rtt)
3742 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3746 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3747 GstClockTime dts, gboolean success)
3749 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3753 /* we scheduled a retry for this packet and now we have it */
3754 priv->num_rtx_success++;
3755 /* all the previous retry attempts failed */
3756 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3758 /* All retries failed or was too late */
3759 priv->num_rtx_failed += timer->num_rtx_retry;
3762 /* number of retries before (hopefully) receiving the packet */
3763 if (priv->avg_rtx_num == 0.0)
3764 priv->avg_rtx_num = timer->num_rtx_retry;
3766 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3768 /* Calculate the delay between retransmission request and receiving this
3769 * packet. We have a valid delay if and only if this packet is a response to
3770 * our last request. If not we don't know if this is a response to an
3771 * earlier request and delay could be way off. For RTT is more important
3772 * with correct values than to update for every packet. */
3773 if (timer->num_rtx_retry == timer->num_rtx_received &&
3774 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3775 delay = dts - timer->rtx_last;
3776 update_avg_rtx_rtt (priv, delay);
3781 GST_LOG_OBJECT (jitterbuffer,
3782 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3783 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3784 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3785 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3786 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3787 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3788 GST_TIME_ARGS (priv->avg_rtx_rtt));
3791 /* the timeout for when we expected a packet expired */
3793 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3796 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3798 guint delay, delay_ms, avg_rtx_rtt_ms;
3799 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3800 guint rtx_deadline_ms;
3801 GstClockTime rtx_retry_period;
3802 GstClockTime rtx_retry_timeout;
3805 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3806 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3808 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3809 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3811 delay = timer->rtx_delay + timer->rtx_retry;
3813 delay_ms = GST_TIME_AS_MSECONDS (delay);
3814 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3815 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3816 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3818 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3820 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3821 gst_structure_new ("GstRTPRetransmissionRequest",
3822 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3823 "running-time", G_TYPE_UINT64, timer->rtx_base,
3824 "delay", G_TYPE_UINT, delay_ms,
3825 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3826 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3827 "period", G_TYPE_UINT, rtx_retry_period_ms,
3828 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3829 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3830 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3831 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3833 priv->num_rtx_requests++;
3834 timer->num_rtx_retry++;
3836 GST_OBJECT_LOCK (jitterbuffer);
3837 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3838 timer->rtx_last = gst_clock_get_time (clock);
3839 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3841 timer->rtx_last = now;
3843 GST_OBJECT_UNLOCK (jitterbuffer);
3845 /* calculate the timeout for the next retransmission attempt */
3846 timer->rtx_retry += rtx_retry_timeout;
3847 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3848 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3849 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3850 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3851 if ((priv->rtx_max_retries != -1
3852 && timer->num_rtx_retry >= priv->rtx_max_retries)
3853 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3854 || (timer->rtx_base + rtx_retry_period < now)) {
3855 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3856 /* too many retransmission request, we now convert the timer
3857 * to a lost timer, leave the num_rtx_retry as it is for stats */
3858 timer->type = TIMER_TYPE_LOST;
3859 timer->rtx_delay = 0;
3860 timer->rtx_retry = 0;
3862 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3863 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3866 gst_pad_push_event (priv->sinkpad, event);
3872 /* a packet is lost */
3874 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3877 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3878 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3880 GstEvent *event = NULL;
3881 RTPJitterBufferItem *item;
3883 seqnum = timer->seqnum;
3884 lost_packets = MAX (timer->num, 1);
3885 num_rtx_retry = timer->num_rtx_retry;
3887 /* we had a gap and thus we lost some packets. Create an event for this. */
3888 if (lost_packets > 1)
3889 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3890 seqnum + lost_packets - 1);
3892 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3894 priv->num_lost += lost_packets;
3895 priv->num_rtx_failed += num_rtx_retry;
3897 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3899 /* we now only accept seqnum bigger than this */
3900 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3901 priv->next_in_seqnum = next_in_seqnum;
3902 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3905 /* Avoid creating events if we don't need it. Note that we still need to create
3906 * the lost *ITEM* since it will be used to notify the outgoing thread of
3907 * lost items (so that we can set discont flags and such) */
3908 if (priv->do_lost) {
3909 GstClockTime duration, timestamp;
3910 /* create paket lost event */
3911 timestamp = apply_offset (jitterbuffer, timer->timeout);
3912 duration = timer->duration;
3913 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3914 duration = priv->packet_spacing;
3915 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3916 gst_structure_new ("GstRTPPacketLost",
3917 "seqnum", G_TYPE_UINT, (guint) seqnum,
3918 "timestamp", G_TYPE_UINT64, timestamp,
3919 "duration", G_TYPE_UINT64, duration,
3920 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3922 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3923 if (!rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL))
3927 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3928 /* Store info to update stats if the packet arrives too late */
3929 timer_queue_append (priv->rtx_stats_timers, timer,
3930 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3932 remove_timer (jitterbuffer, timer);
3935 JBUF_SIGNAL_EVENT (priv);
3941 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3944 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3946 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3947 remove_timer (jitterbuffer, timer);
3951 /* there was no EOS in the buffer, put one in there now */
3952 event = gst_event_new_eos ();
3953 if (priv->segment_seqnum != GST_SEQNUM_INVALID)
3954 gst_event_set_seqnum (event, priv->segment_seqnum);
3955 queue_event (jitterbuffer, event);
3957 JBUF_SIGNAL_EVENT (priv);
3963 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3966 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3968 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3970 /* timer seqnum might have been obsoleted by caps seqnum-base,
3971 * only mess with current ongoing seqnum if still unknown */
3972 if (priv->next_seqnum == -1)
3973 priv->next_seqnum = timer->seqnum;
3974 remove_timer (jitterbuffer, timer);
3975 JBUF_SIGNAL_EVENT (priv);
3981 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3984 gboolean removed = FALSE;
3986 switch (timer->type) {
3987 case TIMER_TYPE_EXPECTED:
3988 removed = do_expected_timeout (jitterbuffer, timer, now);
3990 case TIMER_TYPE_LOST:
3991 removed = do_lost_timeout (jitterbuffer, timer, now);
3993 case TIMER_TYPE_DEADLINE:
3994 removed = do_deadline_timeout (jitterbuffer, timer, now);
3996 case TIMER_TYPE_EOS:
3997 removed = do_eos_timeout (jitterbuffer, timer, now);
4003 /* called when we need to wait for the next timeout.
4005 * We loop over the array of recorded timeouts and wait for the earliest one.
4006 * When it timed out, do the logic associated with the timer.
4008 * If there are no timers, we wait on a gcond until something new happens.
4011 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
4013 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
4014 GstClockTime now = 0;
4017 while (priv->timer_running) {
4018 TimerData *timer = NULL;
4019 GstClockTime timer_timeout = -1;
4022 /* If we have a clock, update "now" now with the very
4023 * latest running time we have. If timers are unscheduled below we
4024 * otherwise wouldn't update now (it's only updated when timers
4025 * expire), and also for the very first loop iteration now would
4026 * otherwise always be 0
4028 GST_OBJECT_LOCK (jitterbuffer);
4030 now = GST_CLOCK_TIME_NONE;
4031 } else if (GST_ELEMENT_CLOCK (jitterbuffer)) {
4033 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
4034 GST_ELEMENT_CAST (jitterbuffer)->base_time;
4036 GST_OBJECT_UNLOCK (jitterbuffer);
4038 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
4039 GST_TIME_ARGS (now));
4041 /* Clear expired rtx-stats timers */
4042 if (priv->do_retransmission)
4043 timer_queue_clear_until (priv->rtx_stats_timers, now);
4045 /* Iterate "normal" timers */
4046 len = priv->timers->len;
4047 for (i = 0; i < len;) {
4048 TimerData *test = &g_array_index (priv->timers, TimerData, i);
4049 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
4050 gboolean save_best = FALSE;
4052 GST_DEBUG_OBJECT (jitterbuffer,
4053 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
4054 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
4055 GST_STIME_ARGS ((gint64) (test_timeout - now)));
4057 /* Weed out anything too late */
4058 if (test->type == TIMER_TYPE_LOST &&
4059 (test_timeout == -1 || test_timeout <= now)) {
4060 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
4061 do_lost_timeout (jitterbuffer, test, now);
4062 if (!priv->timer_running)
4064 /* We don't move the iterator forward since we just removed the current entry,
4065 * but we update the termination condition */
4066 len = priv->timers->len;
4068 /* find the smallest timeout */
4069 if (timer == NULL) {
4071 } else if (timer_timeout == -1) {
4072 /* we already have an immediate timeout, the new timer must be an
4073 * immediate timer with smaller seqnum to become the best */
4074 if (test_timeout == -1
4075 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4076 timer->seqnum) > 0))
4078 } else if (test_timeout == -1) {
4079 /* first immediate timer */
4081 } else if (test_timeout < timer_timeout) {
4084 } else if (test_timeout == timer_timeout
4085 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4086 timer->seqnum) > 0)) {
4087 /* same timer, smaller seqnum */
4092 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
4094 timer_timeout = test_timeout;
4099 if (timer && !priv->blocked) {
4101 GstClockTime sync_time;
4104 GstClockTimeDiff clock_jitter;
4106 if (timer_timeout == -1 || timer_timeout <= now || priv->eos) {
4107 /* We have normally removed all lost timers in the loop above */
4108 g_assert (timer->type != TIMER_TYPE_LOST);
4110 do_timeout (jitterbuffer, timer, now);
4111 /* check here, do_timeout could have released the lock */
4112 if (!priv->timer_running)
4117 GST_OBJECT_LOCK (jitterbuffer);
4118 clock = GST_ELEMENT_CLOCK (jitterbuffer);
4120 GST_OBJECT_UNLOCK (jitterbuffer);
4121 /* let's just push if there is no clock */
4122 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
4123 now = timer_timeout;
4127 /* prepare for sync against clock */
4128 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
4129 /* add latency of peer to get input time */
4130 sync_time += priv->peer_latency;
4132 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
4133 " with sync time %" GST_TIME_FORMAT,
4134 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
4136 /* create an entry for the clock */
4137 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
4138 priv->timer_timeout = timer_timeout;
4139 priv->timer_seqnum = timer->seqnum;
4140 GST_OBJECT_UNLOCK (jitterbuffer);
4142 /* release the lock so that the other end can push stuff or unlock */
4145 ret = gst_clock_id_wait (id, &clock_jitter);
4148 if (!priv->timer_running) {
4149 gst_clock_id_unref (id);
4150 priv->clock_id = NULL;
4154 if (ret != GST_CLOCK_UNSCHEDULED) {
4155 now = timer_timeout + MAX (clock_jitter, 0);
4156 GST_DEBUG_OBJECT (jitterbuffer,
4157 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
4158 GST_STIME_ARGS (clock_jitter));
4160 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
4162 /* and free the entry */
4163 gst_clock_id_unref (id);
4164 priv->clock_id = NULL;
4166 /* no timers, wait for activity */
4167 JBUF_WAIT_TIMER (priv);
4172 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4177 * This funcion implements the main pushing loop on the source pad.
4179 * It first tries to push as many buffers as possible. If there is a seqnum
4180 * mismatch, we wait for the next timeouts.
4183 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4185 GstRtpJitterBufferPrivate *priv;
4186 GstFlowReturn result = GST_FLOW_OK;
4188 priv = jitterbuffer->priv;
4190 JBUF_LOCK_CHECK (priv, flushing);
4192 result = handle_next_buffer (jitterbuffer);
4193 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4194 /* now wait for the next event */
4195 JBUF_WAIT_EVENT (priv, flushing);
4196 result = GST_FLOW_OK;
4198 } while (result == GST_FLOW_OK);
4199 /* store result for upstream */
4200 priv->srcresult = result;
4201 /* if we get here we need to pause */
4207 result = priv->srcresult;
4214 JBUF_SIGNAL_QUERY (priv, FALSE);
4217 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4218 gst_flow_get_name (result));
4219 gst_pad_pause_task (priv->srcpad);
4220 if (result == GST_FLOW_EOS) {
4221 event = gst_event_new_eos ();
4222 if (priv->segment_seqnum != GST_SEQNUM_INVALID)
4223 gst_event_set_seqnum (event, priv->segment_seqnum);
4224 gst_pad_push_event (priv->srcpad, event);
4230 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4231 * some sanity checks and then emit the handle-sync signal with the parameters.
4232 * This function must be called with the LOCK */
4234 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4236 GstRtpJitterBufferPrivate *priv;
4237 guint64 base_rtptime, base_time;
4239 guint64 last_rtptime;
4241 guint64 ext_rtptime, diff;
4242 gboolean valid = TRUE, keep = FALSE;
4244 priv = jitterbuffer->priv;
4246 /* get the last values from the jitterbuffer */
4247 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4248 &clock_rate, &last_rtptime);
4250 clock_base = priv->clock_base;
4251 ext_rtptime = priv->ext_rtptime;
4253 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4254 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4255 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4256 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4258 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4259 /* we keep this SR packet for later. When we get a valid RTP packet the
4260 * above values will be set and we can try to use the SR packet */
4261 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4264 /* we can't accept anything that happened before we did the last resync */
4265 if (base_rtptime > ext_rtptime) {
4266 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4269 /* the SR RTP timestamp must be something close to what we last observed
4270 * in the jitterbuffer */
4271 if (ext_rtptime > last_rtptime) {
4272 /* check how far ahead it is to our RTP timestamps */
4273 diff = ext_rtptime - last_rtptime;
4274 /* if bigger than 1 second, we drop it */
4275 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4277 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4278 clock_rate, 1000)) {
4279 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4280 /* should drop this, but some RTSP servers end up with bogus
4281 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4282 * so still trigger rptbin sync but invalidate RTCP data
4283 * (sync might use other methods) */
4286 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4287 G_GUINT64_FORMAT, last_rtptime, diff);
4293 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4297 s = gst_structure_new ("application/x-rtp-sync",
4298 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4299 "base-time", G_TYPE_UINT64, base_time,
4300 "clock-rate", G_TYPE_UINT, clock_rate,
4301 "clock-base", G_TYPE_UINT64, clock_base,
4302 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4303 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4305 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4306 gst_buffer_replace (&priv->last_sr, NULL);
4308 g_signal_emit (jitterbuffer,
4309 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4311 gst_structure_free (s);
4313 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4314 gst_buffer_replace (&priv->last_sr, NULL);
4318 static GstFlowReturn
4319 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4322 GstRtpJitterBuffer *jitterbuffer;
4323 GstRtpJitterBufferPrivate *priv;
4324 GstFlowReturn ret = GST_FLOW_OK;
4326 GstRTCPPacket packet;
4327 guint64 ext_rtptime;
4329 GstRTCPBuffer rtcp = { NULL, };
4331 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4333 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4334 goto invalid_buffer;
4336 priv = jitterbuffer->priv;
4338 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4340 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4343 /* first packet must be SR or RR or else the validate would have failed */
4344 switch (gst_rtcp_packet_get_type (&packet)) {
4345 case GST_RTCP_TYPE_SR:
4346 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4352 gst_rtcp_buffer_unmap (&rtcp);
4354 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4357 /* convert the RTP timestamp to our extended timestamp, using the same offset
4358 * we used in the jitterbuffer */
4359 ext_rtptime = priv->jbuf->ext_rtptime;
4360 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4362 priv->ext_rtptime = ext_rtptime;
4363 gst_buffer_replace (&priv->last_sr, buffer);
4365 do_handle_sync (jitterbuffer);
4369 gst_buffer_unref (buffer);
4375 /* this is not fatal but should be filtered earlier */
4376 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4377 ("Received invalid RTCP payload, dropping"));
4383 /* this is not fatal but should be filtered earlier */
4384 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4385 ("Received empty RTCP payload, dropping"));
4386 gst_rtcp_buffer_unmap (&rtcp);
4392 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4393 gst_rtcp_buffer_unmap (&rtcp);
4400 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4403 gboolean res = FALSE;
4404 GstRtpJitterBuffer *jitterbuffer;
4405 GstRtpJitterBufferPrivate *priv;
4407 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4408 priv = jitterbuffer->priv;
4410 switch (GST_QUERY_TYPE (query)) {
4411 case GST_QUERY_CAPS:
4413 GstCaps *filter, *caps;
4415 gst_query_parse_caps (query, &filter);
4416 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4417 gst_query_set_caps_result (query, caps);
4418 gst_caps_unref (caps);
4423 if (GST_QUERY_IS_SERIALIZED (query)) {
4424 RTPJitterBufferItem *item;
4427 JBUF_LOCK_CHECK (priv, out_flushing);
4428 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4429 RTP_JITTER_BUFFER_MODE_BUFFER) {
4430 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4431 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4432 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4434 JBUF_SIGNAL_EVENT (priv);
4435 JBUF_WAIT_QUERY (priv, out_flushing);
4436 res = priv->last_query;
4438 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4443 res = gst_pad_query_default (pad, parent, query);
4451 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4459 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4462 GstRtpJitterBuffer *jitterbuffer;
4463 GstRtpJitterBufferPrivate *priv;
4464 gboolean res = FALSE;
4466 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4467 priv = jitterbuffer->priv;
4469 switch (GST_QUERY_TYPE (query)) {
4470 case GST_QUERY_LATENCY:
4472 /* We need to send the query upstream and add the returned latency to our
4474 GstClockTime min_latency, max_latency;
4476 GstClockTime our_latency;
4478 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4479 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4481 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4482 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4483 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4485 /* store this so that we can safely sync on the peer buffers. */
4487 priv->peer_latency = min_latency;
4488 our_latency = priv->latency_ns;
4491 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4492 GST_TIME_ARGS (our_latency));
4494 /* we add some latency but can buffer an infinite amount of time */
4495 min_latency += our_latency;
4498 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4499 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4500 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4502 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4506 case GST_QUERY_POSITION:
4508 GstClockTime start, last_out;
4511 gst_query_parse_position (query, &fmt, NULL);
4512 if (fmt != GST_FORMAT_TIME) {
4513 res = gst_pad_query_default (pad, parent, query);
4518 start = priv->npt_start;
4519 last_out = priv->last_out_time;
4522 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4523 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4524 GST_TIME_ARGS (last_out));
4526 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4527 /* bring 0-based outgoing time to stream time */
4528 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4531 res = gst_pad_query_default (pad, parent, query);
4535 case GST_QUERY_CAPS:
4537 GstCaps *filter, *caps;
4539 gst_query_parse_caps (query, &filter);
4540 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4541 gst_query_set_caps_result (query, caps);
4542 gst_caps_unref (caps);
4547 res = gst_pad_query_default (pad, parent, query);
4555 gst_rtp_jitter_buffer_set_property (GObject * object,
4556 guint prop_id, const GValue * value, GParamSpec * pspec)
4558 GstRtpJitterBuffer *jitterbuffer;
4559 GstRtpJitterBufferPrivate *priv;
4561 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4562 priv = jitterbuffer->priv;
4567 guint new_latency, old_latency;
4569 new_latency = g_value_get_uint (value);
4572 old_latency = priv->latency_ms;
4573 priv->latency_ms = new_latency;
4574 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4575 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4578 /* post message if latency changed, this will inform the parent pipeline
4579 * that a latency reconfiguration is possible/needed. */
4580 if (new_latency != old_latency) {
4581 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4582 GST_TIME_ARGS (new_latency * GST_MSECOND));
4584 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4585 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4589 case PROP_DROP_ON_LATENCY:
4591 priv->drop_on_latency = g_value_get_boolean (value);
4594 case PROP_TS_OFFSET:
4596 if (priv->max_ts_offset_adjustment != 0) {
4597 gint64 new_offset = g_value_get_int64 (value);
4599 if (new_offset > priv->ts_offset) {
4600 priv->ts_offset_remainder = new_offset - priv->ts_offset;
4602 priv->ts_offset_remainder = -(priv->ts_offset - new_offset);
4605 priv->ts_offset = g_value_get_int64 (value);
4606 priv->ts_offset_remainder = 0;
4608 priv->ts_discont = TRUE;
4611 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4613 priv->max_ts_offset_adjustment = g_value_get_uint64 (value);
4618 priv->do_lost = g_value_get_boolean (value);
4623 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4626 case PROP_DO_RETRANSMISSION:
4628 priv->do_retransmission = g_value_get_boolean (value);
4631 case PROP_RTX_NEXT_SEQNUM:
4633 priv->rtx_next_seqnum = g_value_get_boolean (value);
4636 case PROP_RTX_DELAY:
4638 priv->rtx_delay = g_value_get_int (value);
4641 case PROP_RTX_MIN_DELAY:
4643 priv->rtx_min_delay = g_value_get_uint (value);
4646 case PROP_RTX_DELAY_REORDER:
4648 priv->rtx_delay_reorder = g_value_get_int (value);
4651 case PROP_RTX_RETRY_TIMEOUT:
4653 priv->rtx_retry_timeout = g_value_get_int (value);
4656 case PROP_RTX_MIN_RETRY_TIMEOUT:
4658 priv->rtx_min_retry_timeout = g_value_get_int (value);
4661 case PROP_RTX_RETRY_PERIOD:
4663 priv->rtx_retry_period = g_value_get_int (value);
4666 case PROP_RTX_MAX_RETRIES:
4668 priv->rtx_max_retries = g_value_get_int (value);
4671 case PROP_RTX_DEADLINE:
4673 priv->rtx_deadline_ms = g_value_get_int (value);
4676 case PROP_RTX_STATS_TIMEOUT:
4678 priv->rtx_stats_timeout = g_value_get_uint (value);
4681 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4683 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4686 case PROP_MAX_DROPOUT_TIME:
4688 priv->max_dropout_time = g_value_get_uint (value);
4691 case PROP_MAX_MISORDER_TIME:
4693 priv->max_misorder_time = g_value_get_uint (value);
4696 case PROP_RFC7273_SYNC:
4698 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4699 g_value_get_boolean (value));
4702 case PROP_FASTSTART_MIN_PACKETS:
4704 priv->faststart_min_packets = g_value_get_uint (value);
4708 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4714 gst_rtp_jitter_buffer_get_property (GObject * object,
4715 guint prop_id, GValue * value, GParamSpec * pspec)
4717 GstRtpJitterBuffer *jitterbuffer;
4718 GstRtpJitterBufferPrivate *priv;
4720 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4721 priv = jitterbuffer->priv;
4726 g_value_set_uint (value, priv->latency_ms);
4729 case PROP_DROP_ON_LATENCY:
4731 g_value_set_boolean (value, priv->drop_on_latency);
4734 case PROP_TS_OFFSET:
4736 g_value_set_int64 (value, priv->ts_offset);
4739 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4741 g_value_set_uint64 (value, priv->max_ts_offset_adjustment);
4746 g_value_set_boolean (value, priv->do_lost);
4751 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4759 if (priv->srcresult != GST_FLOW_OK)
4762 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4764 g_value_set_int (value, percent);
4768 case PROP_DO_RETRANSMISSION:
4770 g_value_set_boolean (value, priv->do_retransmission);
4773 case PROP_RTX_NEXT_SEQNUM:
4775 g_value_set_boolean (value, priv->rtx_next_seqnum);
4778 case PROP_RTX_DELAY:
4780 g_value_set_int (value, priv->rtx_delay);
4783 case PROP_RTX_MIN_DELAY:
4785 g_value_set_uint (value, priv->rtx_min_delay);
4788 case PROP_RTX_DELAY_REORDER:
4790 g_value_set_int (value, priv->rtx_delay_reorder);
4793 case PROP_RTX_RETRY_TIMEOUT:
4795 g_value_set_int (value, priv->rtx_retry_timeout);
4798 case PROP_RTX_MIN_RETRY_TIMEOUT:
4800 g_value_set_int (value, priv->rtx_min_retry_timeout);
4803 case PROP_RTX_RETRY_PERIOD:
4805 g_value_set_int (value, priv->rtx_retry_period);
4808 case PROP_RTX_MAX_RETRIES:
4810 g_value_set_int (value, priv->rtx_max_retries);
4813 case PROP_RTX_DEADLINE:
4815 g_value_set_int (value, priv->rtx_deadline_ms);
4818 case PROP_RTX_STATS_TIMEOUT:
4820 g_value_set_uint (value, priv->rtx_stats_timeout);
4824 g_value_take_boxed (value,
4825 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4827 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4829 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4832 case PROP_MAX_DROPOUT_TIME:
4834 g_value_set_uint (value, priv->max_dropout_time);
4837 case PROP_MAX_MISORDER_TIME:
4839 g_value_set_uint (value, priv->max_misorder_time);
4842 case PROP_RFC7273_SYNC:
4844 g_value_set_boolean (value,
4845 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4848 case PROP_FASTSTART_MIN_PACKETS:
4850 g_value_set_uint (value, priv->faststart_min_packets);
4854 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4859 static GstStructure *
4860 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4862 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4866 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4867 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4868 "num-lost", G_TYPE_UINT64, priv->num_lost,
4869 "num-late", G_TYPE_UINT64, priv->num_late,
4870 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4871 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4872 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4873 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4874 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4875 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);