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_QUEUE(priv) G_STMT_START { \
204 GST_DEBUG ("waiting queue"); \
205 (priv)->waiting_queue++; \
206 g_cond_wait (&(priv)->jbuf_queue, &(priv)->jbuf_lock); \
207 (priv)->waiting_queue--; \
208 GST_DEBUG ("waiting queue done"); \
210 #define JBUF_SIGNAL_QUEUE(priv) G_STMT_START { \
211 if (G_UNLIKELY ((priv)->waiting_queue)) { \
212 GST_DEBUG ("signal queue, %d waiters", (priv)->waiting_queue); \
213 g_cond_signal (&(priv)->jbuf_queue); \
217 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
218 GST_DEBUG ("waiting timer"); \
219 (priv)->waiting_timer++; \
220 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
221 (priv)->waiting_timer--; \
222 GST_DEBUG ("waiting timer done"); \
224 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
225 if (G_UNLIKELY ((priv)->waiting_timer)) { \
226 GST_DEBUG ("signal timer, %d waiters", (priv)->waiting_timer); \
227 g_cond_signal (&(priv)->jbuf_timer); \
231 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
232 GST_DEBUG ("waiting event"); \
233 (priv)->waiting_event = TRUE; \
234 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
235 (priv)->waiting_event = FALSE; \
236 GST_DEBUG ("waiting event done"); \
237 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
240 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
241 if (G_UNLIKELY ((priv)->waiting_event)) { \
242 GST_DEBUG ("signal event"); \
243 g_cond_signal (&(priv)->jbuf_event); \
247 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
248 GST_DEBUG ("waiting query"); \
249 (priv)->waiting_query = TRUE; \
250 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
251 (priv)->waiting_query = FALSE; \
252 GST_DEBUG ("waiting query done"); \
253 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
256 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
257 (priv)->last_query = res; \
258 if (G_UNLIKELY ((priv)->waiting_query)) { \
259 GST_DEBUG ("signal query"); \
260 g_cond_signal (&(priv)->jbuf_query); \
264 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
265 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
267 typedef struct TimerQueue
270 GHashTable *hashtable;
273 struct _GstRtpJitterBufferPrivate
275 GstPad *sinkpad, *srcpad;
278 RTPJitterBuffer *jbuf;
280 gboolean waiting_queue;
282 gboolean waiting_timer;
284 gboolean waiting_event;
286 gboolean waiting_query;
293 guint32 segment_seqnum;
295 gboolean timer_running;
296 GThread *timer_thread;
301 gboolean drop_on_latency;
303 guint64 max_ts_offset_adjustment;
305 gboolean do_retransmission;
306 gboolean rtx_next_seqnum;
309 gint rtx_delay_reorder;
310 gint rtx_retry_timeout;
311 gint rtx_min_retry_timeout;
312 gint rtx_retry_period;
313 gint rtx_max_retries;
314 guint rtx_stats_timeout;
315 gint rtx_deadline_ms;
316 gint max_rtcp_rtp_time_diff;
317 guint32 max_dropout_time;
318 guint32 max_misorder_time;
319 guint faststart_min_packets;
321 /* the last seqnum we pushed out */
322 guint32 last_popped_seqnum;
323 /* the next expected seqnum we push */
325 /* seqnum-base, if known */
327 /* last output time */
328 GstClockTime last_out_time;
329 /* last valid input timestamp and rtptime pair */
330 GstClockTime ips_pts;
332 GstClockTime packet_spacing;
337 /* the next expected seqnum we receive */
338 GstClockTime last_in_pts;
339 guint32 next_in_seqnum;
342 TimerQueue *rtx_stats_timers;
344 /* start and stop ranges */
345 GstClockTime npt_start;
346 GstClockTime npt_stop;
347 guint64 ext_timestamp;
348 guint64 last_elapsed;
349 guint64 estimated_eos;
356 /* clock rate and rtp timestamp offset */
360 gint64 ts_offset_remainder;
362 /* when we are shutting down */
363 GstFlowReturn srcresult;
369 GstClockTime timer_timeout;
370 guint16 timer_seqnum;
371 /* the latency of the upstream peer, we have to take this into account when
372 * synchronizing the buffers. */
373 GstClockTime peer_latency;
377 /* some accounting */
381 guint64 num_duplicates;
382 guint64 num_rtx_requests;
383 guint64 num_rtx_success;
384 guint64 num_rtx_failed;
387 RTPPacketRateCtx packet_rate_ctx;
390 GstClockTime last_dts;
391 GstClockTime last_pts;
392 guint64 last_rtptime;
393 GstClockTime avg_jitter;
410 GstClockTime timeout;
411 GstClockTime duration;
412 GstClockTime rtx_base;
413 GstClockTime rtx_delay;
414 GstClockTime rtx_retry;
415 GstClockTime rtx_last;
417 guint num_rtx_received;
420 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
421 GST_STATIC_PAD_TEMPLATE ("sink",
424 GST_STATIC_CAPS ("application/x-rtp"
425 /* "clock-rate = (int) [ 1, 2147483647 ], "
426 * "payload = (int) , "
427 * "encoding-name = (string) "
431 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
432 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
435 GST_STATIC_CAPS ("application/x-rtcp")
438 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
439 GST_STATIC_PAD_TEMPLATE ("src",
442 GST_STATIC_CAPS ("application/x-rtp"
443 /* "payload = (int) , "
444 * "clock-rate = (int) , "
445 * "encoding-name = (string) "
449 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
451 #define gst_rtp_jitter_buffer_parent_class parent_class
452 G_DEFINE_TYPE_WITH_PRIVATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer,
455 /* object overrides */
456 static void gst_rtp_jitter_buffer_set_property (GObject * object,
457 guint prop_id, const GValue * value, GParamSpec * pspec);
458 static void gst_rtp_jitter_buffer_get_property (GObject * object,
459 guint prop_id, GValue * value, GParamSpec * pspec);
460 static void gst_rtp_jitter_buffer_finalize (GObject * object);
462 /* element overrides */
463 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
464 * element, GstStateChange transition);
465 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
466 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
467 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
469 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
470 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
474 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
475 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
478 /* sinkpad overrides */
479 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
480 GstObject * parent, GstEvent * event);
481 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
482 GstObject * parent, GstBuffer * buffer);
483 static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad,
484 GstObject * parent, GstBufferList * buffer_list);
486 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
487 GstObject * parent, GstEvent * event);
488 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
489 GstObject * parent, GstBuffer * buffer);
491 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
492 GstObject * parent, GstQuery * query);
494 /* srcpad overrides */
495 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
496 GstObject * parent, GstEvent * event);
497 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
498 GstObject * parent, GstPadMode mode, gboolean active);
499 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
500 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
501 GstObject * parent, GstQuery * query);
504 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
506 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
507 gboolean active, guint64 base_time);
508 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
510 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
511 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
513 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
515 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
518 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
519 TimerData * timer, GstClockTime dts, gboolean success);
521 static TimerQueue *timer_queue_new (void);
522 static void timer_queue_free (TimerQueue * queue);
525 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
527 GObjectClass *gobject_class;
528 GstElementClass *gstelement_class;
530 gobject_class = (GObjectClass *) klass;
531 gstelement_class = (GstElementClass *) klass;
533 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
535 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
536 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
539 * GstRtpJitterBuffer:latency:
541 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
542 * for at most this time.
544 g_object_class_install_property (gobject_class, PROP_LATENCY,
545 g_param_spec_uint ("latency", "Buffer latency in ms",
546 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
547 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
549 * GstRtpJitterBuffer:drop-on-latency:
551 * Drop oldest buffers when the queue is completely filled.
553 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
554 g_param_spec_boolean ("drop-on-latency",
555 "Drop buffers when maximum latency is reached",
556 "Tells the jitterbuffer to never exceed the given latency in size",
557 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
559 * GstRtpJitterBuffer:ts-offset:
561 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
562 * This is mainly used to ensure interstream synchronisation.
564 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
565 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
566 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
567 G_MAXINT64, DEFAULT_TS_OFFSET,
568 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
571 * GstRtpJitterBuffer:max-ts-offset-adjustment:
573 * The maximum number of nanoseconds per frame that time offset may be
574 * adjusted with. This is used to avoid sudden large changes to time stamps.
576 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
577 g_param_spec_uint64 ("max-ts-offset-adjustment",
578 "Max Timestamp Offset Adjustment",
579 "The maximum number of nanoseconds per frame that time stamp "
580 "offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64,
581 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
582 G_PARAM_STATIC_STRINGS));
585 * GstRtpJitterBuffer:do-lost:
587 * Send out a GstRTPPacketLost event downstream when a packet is considered
590 g_object_class_install_property (gobject_class, PROP_DO_LOST,
591 g_param_spec_boolean ("do-lost", "Do Lost",
592 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
593 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
596 * GstRtpJitterBuffer:mode:
598 * Control the buffering and timestamping mode used by the jitterbuffer.
600 g_object_class_install_property (gobject_class, PROP_MODE,
601 g_param_spec_enum ("mode", "Mode",
602 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
603 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
605 * GstRtpJitterBuffer:percent:
607 * The percent of the jitterbuffer that is filled.
609 g_object_class_install_property (gobject_class, PROP_PERCENT,
610 g_param_spec_int ("percent", "percent",
611 "The buffer filled percent", 0, 100,
612 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
614 * GstRtpJitterBuffer:do-retransmission:
616 * Send out a GstRTPRetransmission event upstream when a packet is considered
617 * late and should be retransmitted.
621 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
622 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
623 "Send retransmission events upstream when a packet is late",
624 DEFAULT_DO_RETRANSMISSION,
625 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
628 * GstRtpJitterBuffer:rtx-next-seqnum
630 * Estimate when the next packet should arrive and schedule a retransmission
632 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
633 * for packet N+1. So it will be requested if it does not arrive at the expected time.
634 * The expected time is calculated using the dts of N and the packet spacing.
638 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
639 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
640 "Estimate when the next packet should arrive and schedule a "
641 "retransmission request for it.",
642 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
645 * GstRtpJitterBuffer:rtx-delay:
647 * When a packet did not arrive at the expected time, wait this extra amount
648 * of time before sending a retransmission event.
650 * When -1 is used, the max jitter will be used as extra delay.
654 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
655 g_param_spec_int ("rtx-delay", "RTX Delay",
656 "Extra time in ms to wait before sending retransmission "
657 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
658 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
661 * GstRtpJitterBuffer:rtx-min-delay:
663 * When a packet did not arrive at the expected time, wait at least this extra amount
664 * of time before sending a retransmission event.
668 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
669 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
670 "Minimum time in ms to wait before sending retransmission "
671 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
672 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
674 * GstRtpJitterBuffer:rtx-delay-reorder:
676 * Assume that a retransmission event should be sent when we see
677 * this much packet reordering.
679 * When -1 is used, the value will be estimated based on observed packet
680 * reordering. When 0 is used packet reordering alone will not cause a
681 * retransmission event (Since 1.10).
685 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
686 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
687 "Sending retransmission event when this much reordering "
689 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
690 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
692 * GstRtpJitterBuffer::rtx-retry-timeout:
694 * When no packet has been received after sending a retransmission event
695 * for this time, retry sending a retransmission event.
697 * When -1 is used, the value will be estimated based on observed round
702 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
703 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
704 "Retry sending a transmission event after this timeout in "
705 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
706 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
708 * GstRtpJitterBuffer::rtx-min-retry-timeout:
710 * The minimum amount of time between retry timeouts. When
711 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
712 * minimum interval between retry timeouts.
714 * When -1 is used, the value will be estimated based on the
719 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
720 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
721 "Minimum timeout between sending a transmission event in "
722 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
723 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
725 * GstRtpJitterBuffer:rtx-retry-period:
727 * The amount of time to try to get a retransmission.
729 * When -1 is used, the value will be estimated based on the jitterbuffer
730 * latency and the observed round trip time.
734 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
735 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
736 "Try to get a retransmission for this many ms "
737 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
738 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
740 * GstRtpJitterBuffer:rtx-max-retries:
742 * The maximum number of retries to request a retransmission.
744 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
745 * When -1 is used, the number of retransmission request will not be limited.
749 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
750 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
751 "The maximum number of retries to request a retransmission. "
752 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
753 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
755 * GstRtpJitterBuffer:rtx-deadline:
757 * The deadline for a valid RTX request in ms.
759 * How long the RTX RTCP will be valid for.
760 * When -1 is used, the size of the jitterbuffer will be used.
764 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
765 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
766 "The deadline for a valid RTX request in milliseconds. "
767 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
768 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
770 * GstRtpJitterBuffer::rtx-stats-timeout:
772 * The time to wait for a retransmitted packet after it has been
773 * considered lost in order to collect RTX statistics.
777 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
778 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
779 "The time to wait for a retransmitted packet after it has been "
780 "considered lost in order to collect statistics (ms)",
781 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
782 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
784 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
785 g_param_spec_uint ("max-dropout-time", "Max dropout time",
786 "The maximum time (milliseconds) of missing packets tolerated.",
787 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
788 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
790 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
791 g_param_spec_uint ("max-misorder-time", "Max misorder time",
792 "The maximum time (milliseconds) of misordered packets tolerated.",
793 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
794 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
796 * GstRtpJitterBuffer:stats:
798 * Various jitterbuffer statistics. This property returns a GstStructure
799 * with name application/x-rtp-jitterbuffer-stats with the following fields:
805 * <classname>"num-pushed"</classname>:
806 * the number of packets pushed out.
812 * <classname>"num-lost"</classname>:
813 * the number of packets considered lost.
819 * <classname>"num-late"</classname>:
820 * the number of packets arriving too late.
826 * <classname>"num-duplicates"</classname>:
827 * the number of duplicate packets.
833 * <classname>"rtx-count"</classname>:
834 * the number of retransmissions requested.
840 * <classname>"rtx-success-count"</classname>:
841 * the number of successful retransmissions.
847 * <classname>"rtx-per-packet"</classname>:
848 * average number of RTX per packet.
854 * <classname>"rtx-rtt"</classname>:
855 * average round trip time per RTX.
862 g_object_class_install_property (gobject_class, PROP_STATS,
863 g_param_spec_boxed ("stats", "Statistics",
864 "Various statistics", GST_TYPE_STRUCTURE,
865 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
868 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
870 * The maximum amount of time in ms that the RTP time in the RTCP SRs
871 * is allowed to be ahead of the last RTP packet we received. Use
872 * -1 to disable ignoring of RTCP packets.
876 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
877 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
878 "Maximum amount of time in ms that the RTP time in RTCP SRs "
879 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
880 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
881 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
883 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
884 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
885 "Synchronize received streams to the RFC7273 clock "
886 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
887 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
890 * GstRtpJitterBuffer:faststart-min-packets
892 * The number of consecutive packets needed to start (set to 0 to
893 * disable faststart. The jitterbuffer will by default start after the
894 * latency has elapsed)
898 g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
899 g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
900 "The number of consecutive packets needed to start (set to 0 to "
901 "disable faststart. The jitterbuffer will by default start after "
902 "the latency has elapsed)",
903 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
904 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
907 * GstRtpJitterBuffer::request-pt-map:
908 * @buffer: the object which received the signal
911 * Request the payload type as #GstCaps for @pt.
913 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
914 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
915 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
916 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
917 GST_TYPE_CAPS, 1, G_TYPE_UINT);
919 * GstRtpJitterBuffer::handle-sync:
920 * @buffer: the object which received the signal
921 * @struct: a GstStructure containing sync values.
923 * Be notified of new sync values.
925 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
926 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
927 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
928 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
929 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
932 * GstRtpJitterBuffer::on-npt-stop:
933 * @buffer: the object which received the signal
935 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
936 * the npt-stop position.
938 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
939 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
940 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
941 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
942 G_TYPE_NONE, 0, G_TYPE_NONE);
945 * GstRtpJitterBuffer::clear-pt-map:
946 * @buffer: the object which received the signal
948 * Invalidate the clock-rate as obtained with the
949 * #GstRtpJitterBuffer::request-pt-map signal.
951 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
952 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
953 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
954 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
955 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
958 * GstRtpJitterBuffer::set-active:
959 * @buffer: the object which received the signal
961 * Start pushing out packets with the given base time. This signal is only
962 * useful in buffering mode.
964 * Returns: the time of the last pushed packet.
966 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
967 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
968 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
969 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
970 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
973 gstelement_class->change_state =
974 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
975 gstelement_class->request_new_pad =
976 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
977 gstelement_class->release_pad =
978 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
979 gstelement_class->provide_clock =
980 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
981 gstelement_class->set_clock =
982 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
984 gst_element_class_add_static_pad_template (gstelement_class,
985 &gst_rtp_jitter_buffer_src_template);
986 gst_element_class_add_static_pad_template (gstelement_class,
987 &gst_rtp_jitter_buffer_sink_template);
988 gst_element_class_add_static_pad_template (gstelement_class,
989 &gst_rtp_jitter_buffer_sink_rtcp_template);
991 gst_element_class_set_static_metadata (gstelement_class,
992 "RTP packet jitter-buffer", "Filter/Network/RTP",
993 "A buffer that deals with network jitter and other transmission faults",
994 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
995 "Wim Taymans <wim.taymans@gmail.com>");
997 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
998 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
1000 GST_DEBUG_CATEGORY_INIT
1001 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
1005 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
1007 GstRtpJitterBufferPrivate *priv;
1009 priv = gst_rtp_jitter_buffer_get_instance_private (jitterbuffer);
1010 jitterbuffer->priv = priv;
1012 priv->latency_ms = DEFAULT_LATENCY_MS;
1013 priv->latency_ns = priv->latency_ms * GST_MSECOND;
1014 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
1015 priv->ts_offset = DEFAULT_TS_OFFSET;
1016 priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
1017 priv->do_lost = DEFAULT_DO_LOST;
1018 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
1019 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
1020 priv->rtx_delay = DEFAULT_RTX_DELAY;
1021 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
1022 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
1023 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
1024 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
1025 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
1026 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
1027 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
1028 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
1029 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
1030 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
1031 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
1032 priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
1034 priv->ts_offset_remainder = 0;
1035 priv->last_dts = -1;
1036 priv->last_pts = -1;
1037 priv->last_rtptime = -1;
1038 priv->avg_jitter = 0;
1039 priv->segment_seqnum = GST_SEQNUM_INVALID;
1040 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
1041 priv->rtx_stats_timers = timer_queue_new ();
1042 priv->jbuf = rtp_jitter_buffer_new ();
1043 g_mutex_init (&priv->jbuf_lock);
1044 g_cond_init (&priv->jbuf_queue);
1045 g_cond_init (&priv->jbuf_timer);
1046 g_cond_init (&priv->jbuf_event);
1047 g_cond_init (&priv->jbuf_query);
1048 g_queue_init (&priv->gap_packets);
1049 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1051 /* reset skew detection initialy */
1052 rtp_jitter_buffer_reset_skew (priv->jbuf);
1053 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
1054 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1055 priv->active = TRUE;
1058 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1061 gst_pad_set_activatemode_function (priv->srcpad,
1062 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1063 gst_pad_set_query_function (priv->srcpad,
1064 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1065 gst_pad_set_event_function (priv->srcpad,
1066 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1069 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1072 gst_pad_set_chain_function (priv->sinkpad,
1073 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1074 gst_pad_set_chain_list_function (priv->sinkpad,
1075 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list));
1076 gst_pad_set_event_function (priv->sinkpad,
1077 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1078 gst_pad_set_query_function (priv->sinkpad,
1079 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1081 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1082 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1084 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1087 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1089 #define ITEM_TYPE_BUFFER 0
1090 #define ITEM_TYPE_LOST 1
1091 #define ITEM_TYPE_EVENT 2
1092 #define ITEM_TYPE_QUERY 3
1094 static RTPJitterBufferItem *
1095 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1096 guint seqnum, guint count, guint rtptime)
1098 RTPJitterBufferItem *item;
1100 item = g_slice_new (RTPJitterBufferItem);
1107 item->seqnum = seqnum;
1108 item->count = count;
1109 item->rtptime = rtptime;
1115 free_item (RTPJitterBufferItem * item)
1117 g_return_if_fail (item != NULL);
1119 if (item->data && item->type != ITEM_TYPE_QUERY)
1120 gst_mini_object_unref (item->data);
1121 g_slice_free (RTPJitterBufferItem, item);
1125 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1127 GList **l = user_data;
1129 if (item->data && item->type == ITEM_TYPE_EVENT
1130 && GST_EVENT_IS_STICKY (item->data)) {
1131 *l = g_list_prepend (*l, item->data);
1132 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1133 gst_mini_object_unref (item->data);
1135 g_slice_free (RTPJitterBufferItem, item);
1139 gst_rtp_jitter_buffer_finalize (GObject * object)
1141 GstRtpJitterBuffer *jitterbuffer;
1142 GstRtpJitterBufferPrivate *priv;
1144 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1145 priv = jitterbuffer->priv;
1147 g_array_free (priv->timers, TRUE);
1148 timer_queue_free (priv->rtx_stats_timers);
1149 g_mutex_clear (&priv->jbuf_lock);
1150 g_cond_clear (&priv->jbuf_queue);
1151 g_cond_clear (&priv->jbuf_timer);
1152 g_cond_clear (&priv->jbuf_event);
1153 g_cond_clear (&priv->jbuf_query);
1155 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1156 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1157 g_queue_clear (&priv->gap_packets);
1158 g_object_unref (priv->jbuf);
1160 G_OBJECT_CLASS (parent_class)->finalize (object);
1163 static GstIterator *
1164 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1166 GstRtpJitterBuffer *jitterbuffer;
1167 GstPad *otherpad = NULL;
1168 GstIterator *it = NULL;
1169 GValue val = { 0, };
1171 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1173 if (pad == jitterbuffer->priv->sinkpad) {
1174 otherpad = jitterbuffer->priv->srcpad;
1175 } else if (pad == jitterbuffer->priv->srcpad) {
1176 otherpad = jitterbuffer->priv->sinkpad;
1177 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1178 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1182 g_value_init (&val, GST_TYPE_PAD);
1183 g_value_set_object (&val, otherpad);
1184 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1185 g_value_unset (&val);
1192 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1194 GstRtpJitterBufferPrivate *priv;
1196 priv = jitterbuffer->priv;
1198 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1201 gst_pad_new_from_static_template
1202 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1203 gst_pad_set_chain_function (priv->rtcpsinkpad,
1204 gst_rtp_jitter_buffer_chain_rtcp);
1205 gst_pad_set_event_function (priv->rtcpsinkpad,
1206 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1207 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1208 gst_rtp_jitter_buffer_iterate_internal_links);
1209 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1210 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1212 return priv->rtcpsinkpad;
1216 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1218 GstRtpJitterBufferPrivate *priv;
1220 priv = jitterbuffer->priv;
1222 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1224 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1226 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1227 priv->rtcpsinkpad = NULL;
1231 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1232 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1234 GstRtpJitterBuffer *jitterbuffer;
1235 GstElementClass *klass;
1237 GstRtpJitterBufferPrivate *priv;
1239 g_return_val_if_fail (templ != NULL, NULL);
1240 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1242 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1243 priv = jitterbuffer->priv;
1244 klass = GST_ELEMENT_GET_CLASS (element);
1246 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1248 /* figure out the template */
1249 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1250 if (priv->rtcpsinkpad != NULL)
1253 result = create_rtcp_sink (jitterbuffer);
1255 goto wrong_template;
1262 g_warning ("rtpjitterbuffer: this is not our template");
1267 g_warning ("rtpjitterbuffer: pad already requested");
1273 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1275 GstRtpJitterBuffer *jitterbuffer;
1276 GstRtpJitterBufferPrivate *priv;
1278 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1279 g_return_if_fail (GST_IS_PAD (pad));
1281 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1282 priv = jitterbuffer->priv;
1284 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1286 if (priv->rtcpsinkpad == pad) {
1287 remove_rtcp_sink (jitterbuffer);
1296 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1302 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1304 return gst_system_clock_obtain ();
1308 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1310 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1312 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1314 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1318 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1320 GstRtpJitterBufferPrivate *priv;
1322 priv = jitterbuffer->priv;
1324 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1327 priv->clock_rate = -1;
1328 /* do not clear current content, but refresh state for new arrival */
1329 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1330 rtp_jitter_buffer_reset_skew (priv->jbuf);
1335 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1338 GstRtpJitterBufferPrivate *priv;
1339 GstClockTime last_out;
1340 RTPJitterBufferItem *item;
1345 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1346 active, GST_TIME_ARGS (offset));
1348 if (active != priv->active) {
1349 /* add the amount of time spent in paused to the output offset. All
1350 * outgoing buffers will have this offset applied to their timestamps in
1351 * order to make them arrive in time in the sink. */
1352 priv->out_offset = offset;
1353 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1354 GST_TIME_ARGS (priv->out_offset));
1355 priv->active = active;
1356 JBUF_SIGNAL_EVENT (priv);
1359 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1361 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1362 /* head buffer timestamp and offset gives our output time */
1363 last_out = item->pts + priv->ts_offset;
1365 /* use last known time when the buffer is empty */
1366 last_out = priv->last_out_time;
1374 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1376 GstRtpJitterBuffer *jitterbuffer;
1377 GstRtpJitterBufferPrivate *priv;
1382 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1383 priv = jitterbuffer->priv;
1385 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1387 caps = gst_pad_peer_query_caps (other, filter);
1389 templ = gst_pad_get_pad_template_caps (pad);
1391 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1396 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1398 intersect = gst_caps_intersect (caps, templ);
1399 gst_caps_unref (caps);
1400 gst_caps_unref (templ);
1404 gst_object_unref (jitterbuffer);
1410 * Must be called with JBUF_LOCK held
1414 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1415 GstCaps * caps, gint pt)
1417 GstRtpJitterBufferPrivate *priv;
1418 GstStructure *caps_struct;
1422 const gchar *ts_refclk, *mediaclk;
1424 priv = jitterbuffer->priv;
1426 /* first parse the caps */
1427 caps_struct = gst_caps_get_structure (caps, 0);
1429 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1431 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1433 GST_ERROR_OBJECT (jitterbuffer,
1434 "Got caps with wrong payload type (got %d, expected %d)", pt, payload);
1438 if (payload != -1) {
1439 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1440 priv->last_pt = payload;
1443 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1444 * measure the amount of data in the buffer */
1445 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1448 if (priv->clock_rate <= 0)
1451 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1453 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1455 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1457 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1458 * can use this to track the amount of time elapsed on the sender. */
1459 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1460 priv->clock_base = val;
1462 priv->clock_base = -1;
1464 priv->ext_timestamp = priv->clock_base;
1466 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1469 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1470 /* first expected seqnum, only update when we didn't have a previous base. */
1471 if (priv->next_in_seqnum == -1)
1472 priv->next_in_seqnum = val;
1473 if (priv->next_seqnum == -1) {
1474 priv->next_seqnum = val;
1475 JBUF_SIGNAL_EVENT (priv);
1477 priv->seqnum_base = val;
1479 priv->seqnum_base = -1;
1482 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1484 /* the start and stop times. The seqnum-base corresponds to the start time. We
1485 * will keep track of the seqnums on the output and when we reach the one
1486 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1487 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1488 priv->npt_start = tval;
1490 priv->npt_start = 0;
1492 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1493 priv->npt_stop = tval;
1495 priv->npt_stop = -1;
1497 GST_DEBUG_OBJECT (jitterbuffer,
1498 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1499 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1501 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1502 GstClock *clock = NULL;
1503 guint64 clock_offset = -1;
1505 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1508 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1509 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1510 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1512 const gchar *host, *portstr;
1516 host = ts_refclk + sizeof ("ntp=") - 1;
1517 if (host[0] == '[') {
1519 portstr = strchr (host, ']');
1520 if (portstr && portstr[1] == ':')
1521 portstr = portstr + 1;
1525 portstr = strrchr (host, ':');
1529 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1533 hostname = g_strndup (host, (portstr - host));
1535 hostname = g_strdup (host);
1537 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1540 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1541 const gchar *domainstr =
1542 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1545 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1548 clock = gst_ptp_clock_new (NULL, domain);
1550 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1553 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1554 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1556 if (!g_str_has_prefix (mediaclk, "direct=")
1557 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1558 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1559 if (strstr (mediaclk, "rate=") != NULL) {
1560 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1565 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1567 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1575 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1580 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1586 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1588 GstRtpJitterBufferPrivate *priv;
1590 priv = jitterbuffer->priv;
1593 /* mark ourselves as flushing */
1594 priv->srcresult = GST_FLOW_FLUSHING;
1595 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1596 /* this unblocks any waiting pops on the src pad task */
1597 JBUF_SIGNAL_EVENT (priv);
1598 JBUF_SIGNAL_QUERY (priv, FALSE);
1599 JBUF_SIGNAL_QUEUE (priv);
1604 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1606 GstRtpJitterBufferPrivate *priv;
1608 priv = jitterbuffer->priv;
1611 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1612 /* Mark as non flushing */
1613 priv->srcresult = GST_FLOW_OK;
1614 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1615 priv->last_popped_seqnum = -1;
1616 priv->last_out_time = GST_CLOCK_TIME_NONE;
1617 priv->next_seqnum = -1;
1618 priv->seqnum_base = -1;
1619 priv->ips_rtptime = -1;
1620 priv->ips_pts = GST_CLOCK_TIME_NONE;
1621 priv->packet_spacing = 0;
1622 priv->next_in_seqnum = -1;
1623 priv->clock_rate = -1;
1626 priv->estimated_eos = -1;
1627 priv->last_elapsed = 0;
1628 priv->ext_timestamp = -1;
1629 priv->avg_jitter = 0;
1630 priv->last_dts = -1;
1631 priv->last_rtptime = -1;
1632 priv->last_in_pts = 0;
1633 priv->equidistant = 0;
1634 priv->segment_seqnum = GST_SEQNUM_INVALID;
1635 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1636 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1637 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1638 rtp_jitter_buffer_reset_skew (priv->jbuf);
1639 remove_all_timers (jitterbuffer);
1640 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1641 g_queue_clear (&priv->gap_packets);
1646 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1647 GstPadMode mode, gboolean active)
1650 GstRtpJitterBuffer *jitterbuffer = NULL;
1652 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1655 case GST_PAD_MODE_PUSH:
1657 /* allow data processing */
1658 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1660 /* start pushing out buffers */
1661 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1662 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1663 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1665 /* make sure all data processing stops ASAP */
1666 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1668 /* NOTE this will hardlock if the state change is called from the src pad
1669 * task thread because we will _join() the thread. */
1670 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1671 result = gst_pad_stop_task (pad);
1681 static GstStateChangeReturn
1682 gst_rtp_jitter_buffer_change_state (GstElement * element,
1683 GstStateChange transition)
1685 GstRtpJitterBuffer *jitterbuffer;
1686 GstRtpJitterBufferPrivate *priv;
1687 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1689 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1690 priv = jitterbuffer->priv;
1692 switch (transition) {
1693 case GST_STATE_CHANGE_NULL_TO_READY:
1695 case GST_STATE_CHANGE_READY_TO_PAUSED:
1697 /* reset negotiated values */
1698 priv->clock_rate = -1;
1699 priv->clock_base = -1;
1700 priv->peer_latency = 0;
1702 /* block until we go to PLAYING */
1703 priv->blocked = TRUE;
1704 priv->timer_running = TRUE;
1705 priv->srcresult = GST_FLOW_OK;
1706 priv->timer_thread =
1707 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1710 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1712 /* unblock to allow streaming in PLAYING */
1713 priv->blocked = FALSE;
1714 JBUF_SIGNAL_EVENT (priv);
1715 JBUF_SIGNAL_TIMER (priv);
1722 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1724 switch (transition) {
1725 case GST_STATE_CHANGE_READY_TO_PAUSED:
1726 /* we are a live element because we sync to the clock, which we can only
1727 * do in the PLAYING state */
1728 if (ret != GST_STATE_CHANGE_FAILURE)
1729 ret = GST_STATE_CHANGE_NO_PREROLL;
1731 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1733 /* block to stop streaming when PAUSED */
1734 priv->blocked = TRUE;
1735 unschedule_current_timer (jitterbuffer);
1737 if (ret != GST_STATE_CHANGE_FAILURE)
1738 ret = GST_STATE_CHANGE_NO_PREROLL;
1740 case GST_STATE_CHANGE_PAUSED_TO_READY:
1742 gst_buffer_replace (&priv->last_sr, NULL);
1743 priv->timer_running = FALSE;
1744 priv->srcresult = GST_FLOW_FLUSHING;
1745 unschedule_current_timer (jitterbuffer);
1746 JBUF_SIGNAL_TIMER (priv);
1747 JBUF_SIGNAL_QUERY (priv, FALSE);
1748 JBUF_SIGNAL_QUEUE (priv);
1750 g_thread_join (priv->timer_thread);
1751 priv->timer_thread = NULL;
1753 case GST_STATE_CHANGE_READY_TO_NULL:
1763 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1766 gboolean ret = TRUE;
1767 GstRtpJitterBuffer *jitterbuffer;
1768 GstRtpJitterBufferPrivate *priv;
1770 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1771 priv = jitterbuffer->priv;
1773 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1775 switch (GST_EVENT_TYPE (event)) {
1776 case GST_EVENT_LATENCY:
1778 GstClockTime latency;
1780 gst_event_parse_latency (event, &latency);
1782 GST_DEBUG_OBJECT (jitterbuffer,
1783 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1786 /* adjust the overall buffer delay to the total pipeline latency in
1787 * buffering mode because if downstream consumes too fast (because of
1788 * large latency or queues, we would start rebuffering again. */
1789 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1790 RTP_JITTER_BUFFER_MODE_BUFFER) {
1791 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1795 ret = gst_pad_push_event (priv->sinkpad, event);
1799 ret = gst_pad_push_event (priv->sinkpad, event);
1806 /* handles and stores the event in the jitterbuffer, must be called with
1809 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1811 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1812 RTPJitterBufferItem *item;
1815 switch (GST_EVENT_TYPE (event)) {
1816 case GST_EVENT_CAPS:
1820 gst_event_parse_caps (event, &caps);
1821 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1824 case GST_EVENT_SEGMENT:
1827 gst_event_copy_segment (event, &segment);
1829 priv->segment_seqnum = gst_event_get_seqnum (event);
1831 /* we need time for now */
1832 if (segment.format != GST_FORMAT_TIME) {
1833 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1834 gst_event_unref (event);
1836 gst_segment_init (&segment, GST_FORMAT_TIME);
1837 event = gst_event_new_segment (&segment);
1838 gst_event_set_seqnum (event, priv->segment_seqnum);
1841 priv->segment = segment;
1846 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1853 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1854 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1855 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1856 if (head || priv->eos)
1857 JBUF_SIGNAL_EVENT (priv);
1863 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1866 gboolean ret = TRUE;
1867 GstRtpJitterBuffer *jitterbuffer;
1868 GstRtpJitterBufferPrivate *priv;
1870 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1871 priv = jitterbuffer->priv;
1873 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1875 switch (GST_EVENT_TYPE (event)) {
1876 case GST_EVENT_FLUSH_START:
1877 ret = gst_pad_push_event (priv->srcpad, event);
1878 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1879 /* wait for the loop to go into PAUSED */
1880 gst_pad_pause_task (priv->srcpad);
1882 case GST_EVENT_FLUSH_STOP:
1883 ret = gst_pad_push_event (priv->srcpad, event);
1885 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1886 GST_PAD_MODE_PUSH, TRUE);
1889 if (GST_EVENT_IS_SERIALIZED (event)) {
1890 /* serialized events go in the queue */
1892 if (priv->srcresult != GST_FLOW_OK) {
1893 /* Errors in sticky event pushing are no problem and ignored here
1894 * as they will cause more meaningful errors during data flow.
1895 * For EOS events, that are not followed by data flow, we still
1896 * return FALSE here though.
1898 if (!GST_EVENT_IS_STICKY (event) ||
1899 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1900 goto out_flow_error;
1902 /* refuse more events on EOS */
1905 ret = queue_event (jitterbuffer, event);
1908 /* non-serialized events are forwarded downstream immediately */
1909 ret = gst_pad_push_event (priv->srcpad, event);
1918 GST_DEBUG_OBJECT (jitterbuffer,
1919 "refusing event, we have a downstream flow error: %s",
1920 gst_flow_get_name (priv->srcresult));
1922 gst_event_unref (event);
1927 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1929 gst_event_unref (event);
1935 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1938 gboolean ret = TRUE;
1939 GstRtpJitterBuffer *jitterbuffer;
1941 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1943 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1945 switch (GST_EVENT_TYPE (event)) {
1946 case GST_EVENT_FLUSH_START:
1947 gst_event_unref (event);
1949 case GST_EVENT_FLUSH_STOP:
1950 gst_event_unref (event);
1953 ret = gst_pad_event_default (pad, parent, event);
1961 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1962 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1963 * GST_FLOW_FLUSHING when the element is shutting down. On success
1964 * GST_FLOW_OK is returned.
1966 static GstFlowReturn
1967 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1971 GValue args[2] = { {0}, {0} };
1975 g_value_init (&args[0], GST_TYPE_ELEMENT);
1976 g_value_set_object (&args[0], jitterbuffer);
1977 g_value_init (&args[1], G_TYPE_UINT);
1978 g_value_set_uint (&args[1], pt);
1980 g_value_init (&ret, GST_TYPE_CAPS);
1981 g_value_set_boxed (&ret, NULL);
1983 JBUF_UNLOCK (jitterbuffer->priv);
1984 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1986 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1988 g_value_unset (&args[0]);
1989 g_value_unset (&args[1]);
1990 caps = (GstCaps *) g_value_dup_boxed (&ret);
1991 g_value_unset (&ret);
1995 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1996 gst_caps_unref (caps);
1998 if (G_UNLIKELY (!res))
2006 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
2007 return GST_FLOW_ERROR;
2011 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
2012 return GST_FLOW_FLUSHING;
2016 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
2017 return GST_FLOW_ERROR;
2021 /* call with jbuf lock held */
2023 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
2025 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2026 GstMessage *message = NULL;
2031 /* Post a buffering message */
2032 if (priv->last_percent != percent) {
2033 priv->last_percent = percent;
2035 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
2036 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
2043 update_offset (GstRtpJitterBuffer * jitterbuffer)
2045 GstRtpJitterBufferPrivate *priv;
2047 priv = jitterbuffer->priv;
2049 if (priv->ts_offset_remainder != 0) {
2050 GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT
2051 " off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment,
2052 priv->ts_offset_remainder, priv->ts_offset);
2053 if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) {
2054 if (priv->ts_offset_remainder > 0) {
2055 priv->ts_offset += priv->max_ts_offset_adjustment;
2056 priv->ts_offset_remainder -= priv->max_ts_offset_adjustment;
2058 priv->ts_offset -= priv->max_ts_offset_adjustment;
2059 priv->ts_offset_remainder += priv->max_ts_offset_adjustment;
2062 priv->ts_offset += priv->ts_offset_remainder;
2063 priv->ts_offset_remainder = 0;
2069 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
2071 GstRtpJitterBufferPrivate *priv;
2073 priv = jitterbuffer->priv;
2075 if (timestamp == -1)
2078 /* apply the timestamp offset, this is used for inter stream sync */
2079 timestamp += priv->ts_offset;
2080 /* add the offset, this is used when buffering */
2081 timestamp += priv->out_offset;
2087 timer_queue_new (void)
2091 queue = g_slice_new (TimerQueue);
2092 queue->timers = g_queue_new ();
2093 queue->hashtable = g_hash_table_new (NULL, NULL);
2099 timer_queue_free (TimerQueue * queue)
2104 g_hash_table_destroy (queue->hashtable);
2105 g_queue_free_full (queue->timers, g_free);
2106 g_slice_free (TimerQueue, queue);
2110 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2111 GstClockTime timeout, gboolean lost)
2115 copy = g_memdup (timer, sizeof (*timer));
2116 copy->timeout = timeout;
2117 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2120 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2121 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2122 g_queue_push_tail (queue->timers, copy);
2123 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2127 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2131 test = g_queue_peek_head (queue->timers);
2132 while (test && test->timeout < timeout) {
2133 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2134 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2135 GST_TIME_ARGS (timeout));
2136 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2137 g_free (g_queue_pop_head (queue->timers));
2138 test = g_queue_peek_head (queue->timers);
2143 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2145 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2149 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2151 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2152 TimerData *timer = NULL;
2155 len = priv->timers->len;
2156 for (i = 0; i < len; i++) {
2157 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2158 if (test->seqnum == seqnum) {
2167 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2169 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2171 if (priv->clock_id) {
2172 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2173 gst_clock_id_unschedule (priv->clock_id);
2174 priv->clock_id = NULL;
2179 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2181 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2182 GstClockTime test_timeout;
2184 if ((test_timeout = timer->timeout) == -1)
2187 if (timer->type != TIMER_TYPE_EXPECTED) {
2188 /* add our latency and offset to get output times. */
2189 test_timeout = apply_offset (jitterbuffer, test_timeout);
2190 test_timeout += priv->latency_ns;
2192 return test_timeout;
2196 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2198 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2200 if (priv->clock_id) {
2201 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2203 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2204 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2206 if (timeout == -1 || timeout < priv->timer_timeout)
2207 unschedule_current_timer (jitterbuffer);
2212 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2213 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2214 GstClockTime duration)
2216 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2220 GST_DEBUG_OBJECT (jitterbuffer,
2221 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2222 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2223 GST_TIME_ARGS (delay));
2225 len = priv->timers->len;
2226 g_array_set_size (priv->timers, len + 1);
2227 timer = &g_array_index (priv->timers, TimerData, len);
2230 timer->seqnum = seqnum;
2232 timer->timeout = timeout + delay;
2233 timer->duration = duration;
2234 if (type == TIMER_TYPE_EXPECTED) {
2235 timer->rtx_base = timeout;
2236 timer->rtx_delay = delay;
2237 timer->rtx_retry = 0;
2239 timer->rtx_last = GST_CLOCK_TIME_NONE;
2240 timer->num_rtx_retry = 0;
2241 timer->num_rtx_received = 0;
2242 recalculate_timer (jitterbuffer, timer);
2243 JBUF_SIGNAL_TIMER (priv);
2249 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2250 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2252 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2253 gboolean seqchange, timechange;
2255 GstClockTime new_timeout;
2257 oldseq = timer->seqnum;
2258 new_timeout = timeout + delay;
2259 seqchange = oldseq != seqnum;
2260 timechange = timer->timeout != new_timeout;
2262 if (!seqchange && !timechange) {
2263 GST_DEBUG_OBJECT (jitterbuffer,
2264 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2265 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2269 GST_DEBUG_OBJECT (jitterbuffer,
2270 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2271 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2272 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2274 timer->timeout = new_timeout;
2275 timer->seqnum = seqnum;
2277 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2278 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2279 GST_TIME_ARGS (delay));
2280 timer->rtx_base = timeout;
2281 timer->rtx_delay = delay;
2282 timer->rtx_retry = 0;
2285 timer->num_rtx_retry = 0;
2286 timer->num_rtx_received = 0;
2289 if (priv->clock_id) {
2290 /* we changed the seqnum and there is a timer currently waiting with this
2291 * seqnum, unschedule it */
2292 if (seqchange && priv->timer_seqnum == oldseq)
2293 unschedule_current_timer (jitterbuffer);
2294 /* we changed the time, check if it is earlier than what we are waiting
2295 * for and unschedule if so */
2296 else if (timechange)
2297 recalculate_timer (jitterbuffer, timer);
2302 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2303 guint16 seqnum, GstClockTime timeout)
2307 /* find the seqnum timer */
2308 timer = find_timer (jitterbuffer, seqnum);
2309 if (timer == NULL) {
2310 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2312 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2318 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2320 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2323 if (timer->idx == -1)
2326 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2327 unschedule_current_timer (jitterbuffer);
2330 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2331 g_array_remove_index_fast (priv->timers, idx);
2334 JBUF_SIGNAL_TIMER (priv);
2338 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2340 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2341 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2342 g_array_set_size (priv->timers, 0);
2343 unschedule_current_timer (jitterbuffer);
2344 JBUF_SIGNAL_TIMER (priv);
2347 /* get the extra delay to wait before sending RTX */
2349 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2353 if (priv->rtx_delay == -1) {
2354 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2355 delay = DEFAULT_AUTO_RTX_DELAY;
2357 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2358 * packet spacing is a good margin */
2359 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2362 delay = priv->rtx_delay * GST_MSECOND;
2364 if (priv->rtx_min_delay > 0)
2365 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2370 /* Check if packet with seqnum is already considered definitely lost by being
2371 * part of a "lost timer" for multiple packets */
2373 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2375 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2378 len = priv->timers->len;
2379 for (i = 0; i < len; i++) {
2380 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2381 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2383 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2385 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2386 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2394 /* we just received a packet with seqnum and dts.
2396 * First check for old seqnum that we are still expecting. If the gap with the
2397 * current seqnum is too big, unschedule the timeouts.
2399 * If we have a valid packet spacing estimate we can set a timer for when we
2400 * should receive the next packet.
2401 * If we don't have a valid estimate, we remove any timer we might have
2402 * had for this packet.
2405 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2406 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2407 gboolean is_rtx, TimerData * timer)
2409 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2411 /* go through all timers and unschedule the ones with a large gap */
2412 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2414 len = priv->timers->len;
2415 for (i = 0; i < len; i++) {
2416 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2419 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2421 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2422 test->type, test->seqnum, seqnum, gap);
2424 if (gap > priv->rtx_delay_reorder) {
2425 /* max gap, we exceeded the max reorder distance and we don't expect the
2426 * missing packet to be this reordered */
2427 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2428 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2433 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2434 && priv->do_retransmission && priv->rtx_next_seqnum;
2436 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2437 if (timer->num_rtx_retry > 0) {
2439 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2440 /* don't try to estimate the next seqnum because this is a retransmitted
2441 * packet and it probably did not arrive with the expected packet
2443 do_next_seqnum = FALSE;
2446 if (!is_rtx || timer->num_rtx_retry > 1) {
2447 /* Store timer in order to record stats when/if the retransmitted
2448 * packet arrives. We should also store timer information if we've
2449 * requested retransmission more than once since we may receive
2450 * several retransmitted packets. For accuracy we should update the
2451 * stats also when the redundant retransmitted packets arrives. */
2452 timer_queue_append (priv->rtx_stats_timers, timer,
2453 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2458 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2459 GstClockTime expected, delay;
2461 /* calculate expected arrival time of the next seqnum */
2462 expected = pts + priv->packet_spacing;
2464 delay = get_rtx_delay (priv);
2466 /* and update/install timer for next seqnum */
2467 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2468 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2469 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2470 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2471 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2474 timer->type = TIMER_TYPE_EXPECTED;
2475 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2478 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2479 expected, delay, priv->packet_spacing);
2481 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2482 /* if we had a timer, remove it, we don't know when to expect the next
2484 remove_timer (jitterbuffer, timer);
2489 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2492 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2494 /* we need consecutive seqnums with a different
2495 * rtptime to estimate the packet spacing. */
2496 if (priv->ips_rtptime != rtptime) {
2497 /* rtptime changed, check pts diff */
2498 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2499 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2500 GstClockTime old_packet_spacing = priv->packet_spacing;
2502 /* Biased towards bigger packet spacings to prevent
2503 * too many unneeded retransmission requests for next
2504 * packets that just arrive a little later than we would
2506 if (old_packet_spacing > new_packet_spacing)
2507 priv->packet_spacing =
2508 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2509 else if (old_packet_spacing > 0)
2510 priv->packet_spacing =
2511 (3 * new_packet_spacing + old_packet_spacing) / 4;
2513 priv->packet_spacing = new_packet_spacing;
2515 GST_DEBUG_OBJECT (jitterbuffer,
2516 "new packet spacing %" GST_TIME_FORMAT
2517 " old packet spacing %" GST_TIME_FORMAT
2518 " combined to %" GST_TIME_FORMAT,
2519 GST_TIME_ARGS (new_packet_spacing),
2520 GST_TIME_ARGS (old_packet_spacing),
2521 GST_TIME_ARGS (priv->packet_spacing));
2523 priv->ips_rtptime = rtptime;
2524 priv->ips_pts = pts;
2529 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2530 guint16 seqnum, GstClockTime pts, gint gap)
2532 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2533 GstClockTime duration, expected_pts, delay;
2535 gboolean equidistant = priv->equidistant > 0;
2537 GST_DEBUG_OBJECT (jitterbuffer,
2538 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2539 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2541 if (pts == GST_CLOCK_TIME_NONE) {
2542 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2547 GstClockTime total_duration;
2548 /* the total duration spanned by the missing packets */
2549 if (pts >= priv->last_in_pts)
2550 total_duration = pts - priv->last_in_pts;
2554 /* interpolate between the current time and the last time based on
2555 * number of packets we are missing, this is the estimated duration
2556 * for the missing packet based on equidistant packet spacing. */
2557 duration = total_duration / (gap + 1);
2559 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2560 GST_TIME_ARGS (duration));
2562 if (total_duration > priv->latency_ns) {
2563 GstClockTime gap_time;
2567 GstClockTime gap_dur = gap * duration;
2568 if (gap_dur > priv->latency_ns)
2569 gap_time = gap_dur - priv->latency_ns;
2572 lost_packets = gap_time / duration;
2574 gap_time = total_duration - priv->latency_ns;
2578 /* too many lost packets, some of the missing packets are already
2579 * too late and we can generate lost packet events for them. */
2580 GST_INFO_OBJECT (jitterbuffer,
2581 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2582 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2583 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2584 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2585 GST_TIME_ARGS (gap_time));
2587 /* this timer will fire immediately and the lost event will be pushed from
2588 * the timer thread */
2589 if (lost_packets > 0) {
2590 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2591 priv->last_in_pts + duration, 0, gap_time);
2592 expected += lost_packets;
2593 priv->last_in_pts += gap_time;
2597 expected_pts = priv->last_in_pts + duration;
2599 /* If we cannot assume equidistant packet spacing, the only thing we now
2600 * for sure is that the missing packets have expected pts not later than
2601 * the last received pts. */
2608 if (priv->do_retransmission) {
2609 TimerData *timer = find_timer (jitterbuffer, expected);
2611 type = TIMER_TYPE_EXPECTED;
2612 delay = get_rtx_delay (priv);
2614 /* if we had a timer for the first missing packet, update it. */
2615 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2616 GstClockTime timeout = timer->timeout;
2618 timer->duration = duration;
2619 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2620 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2624 expected_pts += duration;
2627 type = TIMER_TYPE_LOST;
2630 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2631 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2632 expected_pts += duration;
2638 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2642 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2643 GstRtpJitterBufferPrivate *priv;
2645 priv = jitterbuffer->priv;
2647 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2650 if (priv->last_dts != -1)
2651 dtsdiff = dts - priv->last_dts;
2655 if (priv->last_rtptime != -1)
2656 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2660 /* Guess whether stream currently uses equidistant packet spacing. If we
2661 * often see identical timestamps it means the packets are not
2663 if (rtptime == priv->last_rtptime)
2664 priv->equidistant -= 2;
2666 priv->equidistant += 1;
2667 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2669 priv->last_dts = dts;
2670 priv->last_rtptime = rtptime;
2674 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2677 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2679 diff = ABS (dtsdiff - rtpdiffns);
2681 /* jitter is stored in nanoseconds */
2682 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2684 GST_LOG_OBJECT (jitterbuffer,
2685 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2686 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2687 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2688 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2695 GST_DEBUG_OBJECT (jitterbuffer,
2696 "no dts or no clock-rate, can't calculate jitter");
2702 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2704 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2705 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2708 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2709 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2710 gst_rtp_buffer_unmap (&rtp_a);
2712 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2713 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2714 gst_rtp_buffer_unmap (&rtp_b);
2716 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2720 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2721 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2723 GstRtpJitterBufferPrivate *priv;
2724 guint gap_packets_length;
2725 gboolean reset = FALSE;
2726 gboolean future = gap > 0;
2728 priv = jitterbuffer->priv;
2730 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2732 guint32 prev_gap_seq = -1;
2733 gboolean all_consecutive = TRUE;
2735 g_queue_insert_sorted (&priv->gap_packets, buffer,
2736 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2738 for (l = priv->gap_packets.head; l; l = l->next) {
2739 GstBuffer *gap_buffer = l->data;
2740 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2743 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2745 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2747 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2748 if (prev_gap_seq == -1)
2749 prev_gap_seq = gap_seq;
2750 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2751 all_consecutive = FALSE;
2753 prev_gap_seq = gap_seq;
2755 gst_rtp_buffer_unmap (&gap_rtp);
2756 if (!all_consecutive)
2760 if (all_consecutive && gap_packets_length > 3) {
2761 GST_DEBUG_OBJECT (jitterbuffer,
2762 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2763 (future ? "new" : "old"), gap,
2764 (future ? max_dropout : -max_misorder));
2766 } else if (!all_consecutive) {
2767 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2768 g_queue_clear (&priv->gap_packets);
2769 GST_DEBUG_OBJECT (jitterbuffer,
2770 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2771 (future ? "new" : "old"), gap,
2772 (future ? max_dropout : -max_misorder));
2775 GST_DEBUG_OBJECT (jitterbuffer,
2776 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2777 (future ? "new" : "old"), gap,
2778 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2782 GST_DEBUG_OBJECT (jitterbuffer,
2783 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2784 gap, -max_misorder);
2785 g_queue_push_tail (&priv->gap_packets, buffer);
2793 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2795 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2796 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2799 GstClockTime base_time =
2800 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2801 GstClockTime clock_time = gst_clock_get_time (clock);
2803 if (clock_time > base_time)
2804 running_time = clock_time - base_time;
2808 gst_object_unref (clock);
2811 return running_time;
2814 static GstFlowReturn
2815 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2816 GstPad * pad, GstObject * parent, guint16 seqnum)
2818 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2819 GstFlowReturn ret = GST_FLOW_OK;
2820 GList *events = NULL, *l;
2824 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2825 rtp_jitter_buffer_flush (priv->jbuf,
2826 (GFunc) free_item_and_retain_events, &events);
2827 rtp_jitter_buffer_reset_skew (priv->jbuf);
2828 remove_all_timers (jitterbuffer);
2829 priv->discont = TRUE;
2830 priv->last_popped_seqnum = -1;
2832 if (priv->gap_packets.head) {
2833 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2834 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2836 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2837 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2838 gst_rtp_buffer_unmap (&gap_rtp);
2840 priv->next_seqnum = seqnum;
2843 priv->last_in_pts = -1;
2844 priv->next_in_seqnum = -1;
2846 /* Insert all sticky events again in order, otherwise we would
2847 * potentially loose STREAM_START, CAPS or SEGMENT events
2849 events = g_list_reverse (events);
2850 for (l = events; l; l = l->next) {
2851 RTPJitterBufferItem *item;
2853 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2854 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2856 g_list_free (events);
2858 JBUF_SIGNAL_EVENT (priv);
2860 /* reset spacing estimation when gap */
2861 priv->ips_rtptime = -1;
2862 priv->ips_pts = GST_CLOCK_TIME_NONE;
2864 buffers = g_list_copy (priv->gap_packets.head);
2865 g_queue_clear (&priv->gap_packets);
2867 priv->ips_rtptime = -1;
2868 priv->ips_pts = GST_CLOCK_TIME_NONE;
2869 JBUF_UNLOCK (jitterbuffer->priv);
2871 for (l = buffers; l; l = l->next) {
2872 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2874 if (ret != GST_FLOW_OK) {
2879 for (; l; l = l->next)
2880 gst_buffer_unref (l->data);
2881 g_list_free (buffers);
2887 gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
2889 GstRtpJitterBufferPrivate *priv;
2890 RTPJitterBufferItem *item;
2893 priv = jitterbuffer->priv;
2895 if (priv->faststart_min_packets == 0)
2898 item = rtp_jitter_buffer_peek (priv->jbuf);
2902 timer = find_timer (jitterbuffer, item->seqnum);
2903 if (!timer || timer->type != TIMER_TYPE_DEADLINE)
2906 if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
2907 priv->faststart_min_packets)) {
2908 GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
2909 priv->faststart_min_packets);
2910 timer->timeout = -1;
2917 static GstFlowReturn
2918 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2921 GstRtpJitterBuffer *jitterbuffer;
2922 GstRtpJitterBufferPrivate *priv;
2924 guint32 expected, rtptime;
2925 GstFlowReturn ret = GST_FLOW_OK;
2926 GstClockTime dts, pts;
2931 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2932 gboolean do_next_seqnum = FALSE;
2933 RTPJitterBufferItem *item;
2934 GstMessage *msg = NULL;
2935 gboolean estimated_dts = FALSE;
2936 gint32 packet_rate, max_dropout, max_misorder;
2937 TimerData *timer = NULL;
2939 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2941 priv = jitterbuffer->priv;
2943 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2944 goto invalid_buffer;
2946 pt = gst_rtp_buffer_get_payload_type (&rtp);
2947 seqnum = gst_rtp_buffer_get_seq (&rtp);
2948 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2949 gst_rtp_buffer_unmap (&rtp);
2951 /* make sure we have PTS and DTS set */
2952 pts = GST_BUFFER_PTS (buffer);
2953 dts = GST_BUFFER_DTS (buffer);
2960 /* If we have no DTS here, i.e. no capture time, get one from the
2961 * clock now to have something to calculate with in the future. */
2962 dts = get_current_running_time (jitterbuffer);
2965 /* Remember that we estimated the DTS if we are running already
2966 * and this is not our first packet (or first packet after a reset).
2967 * If it's the first packet, we somehow must generate a timestamp for
2968 * everything, otherwise we can't calculate any times
2970 estimated_dts = (priv->next_in_seqnum != -1);
2972 /* take the DTS of the buffer. This is the time when the packet was
2973 * received and is used to calculate jitter and clock skew. We will adjust
2974 * this DTS with the smoothed value after processing it in the
2975 * jitterbuffer and assign it as the PTS. */
2976 /* bring to running time */
2977 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2980 GST_DEBUG_OBJECT (jitterbuffer,
2981 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2982 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2983 GST_BUFFER_IS_RETRANSMISSION (buffer));
2985 JBUF_LOCK_CHECK (priv, out_flushing);
2987 if (G_UNLIKELY (priv->last_pt != pt)) {
2990 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2994 /* reset clock-rate so that we get a new one */
2995 priv->clock_rate = -1;
2997 /* Try to get the clock-rate from the caps first if we can. If there are no
2998 * caps we must fire the signal to get the clock-rate. */
2999 if ((caps = gst_pad_get_current_caps (pad))) {
3000 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
3001 gst_caps_unref (caps);
3005 if (G_UNLIKELY (priv->clock_rate == -1)) {
3006 /* no clock rate given on the caps, try to get one with the signal */
3007 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
3008 pt) == GST_FLOW_FLUSHING)
3011 if (G_UNLIKELY (priv->clock_rate == -1))
3014 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
3017 /* don't accept more data on EOS */
3018 if (G_UNLIKELY (priv->eos))
3021 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
3022 calculate_jitter (jitterbuffer, dts, rtptime);
3024 if (priv->seqnum_base != -1) {
3027 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
3030 GST_DEBUG_OBJECT (jitterbuffer,
3031 "packet seqnum #%d before seqnum-base #%d", seqnum,
3033 gst_buffer_unref (buffer);
3035 } else if (gap > 16384) {
3036 /* From now on don't compare against the seqnum base anymore as
3037 * at some point in the future we will wrap around and also that
3038 * much reordering is very unlikely */
3039 priv->seqnum_base = -1;
3043 expected = priv->next_in_seqnum;
3046 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
3048 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
3049 priv->max_dropout_time);
3051 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
3052 priv->max_misorder_time);
3053 GST_TRACE_OBJECT (jitterbuffer,
3054 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
3055 max_dropout, max_misorder);
3057 /* now check against our expected seqnum */
3058 if (G_UNLIKELY (expected == -1)) {
3059 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3061 /* calculate a pts based on rtptime and arrival time (dts) */
3063 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3064 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3066 /* we don't know what the next_in_seqnum should be, wait for the last
3067 * possible moment to push this buffer, maybe we get an earlier seqnum
3069 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
3071 do_next_seqnum = TRUE;
3072 /* take rtptime and pts to calculate packet spacing */
3073 priv->ips_rtptime = rtptime;
3074 priv->ips_pts = pts;
3078 /* now calculate gap */
3079 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
3080 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
3081 expected, seqnum, gap);
3083 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
3084 /* If we have timers for more than RTP_MAX_DROPOUT packets
3085 * pending this means that we have a huge gap overall. We can
3086 * reset the jitterbuffer at this point because there's
3087 * just too much data missing to be able to do anything
3088 * sensible with the past data. Just try again from the
3090 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
3091 priv->timers->len, max_dropout);
3092 gst_buffer_unref (buffer);
3093 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3096 /* Special handling of large gaps */
3097 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
3098 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
3099 gap, max_dropout, max_misorder);
3101 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3103 GST_DEBUG_OBJECT (jitterbuffer,
3104 "Had big gap, waiting for more consecutive packets");
3109 /* We had no huge gap, let's drop all the gap packets */
3110 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
3111 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3112 g_queue_clear (&priv->gap_packets);
3114 /* calculate a pts based on rtptime and arrival time (dts) */
3115 /* If we estimated the DTS, don't consider it in the clock skew calculations */
3117 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3118 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3120 if (G_LIKELY (gap == 0)) {
3121 /* packet is expected */
3122 calculate_packet_spacing (jitterbuffer, rtptime, pts);
3123 do_next_seqnum = TRUE;
3128 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
3129 /* fill in the gap with EXPECTED timers */
3130 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
3131 do_next_seqnum = TRUE;
3133 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3134 do_next_seqnum = FALSE;
3137 /* reset spacing estimation when gap */
3138 priv->ips_rtptime = -1;
3139 priv->ips_pts = GST_CLOCK_TIME_NONE;
3143 if (do_next_seqnum) {
3144 priv->last_in_pts = pts;
3145 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3148 timer = find_timer (jitterbuffer, seqnum);
3149 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3151 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3153 timer->num_rtx_received++;
3156 /* At 2^15, we would detect a seqnum rollover too early, therefore
3157 * limit the queue size. But let's not limit it to a number that is
3158 * too small to avoid emptying it needlessly if there is a spurious huge
3159 * sequence number, let's allow at least 10k packets in any case. */
3160 while (rtp_jitter_buffer_get_seqnum_diff (priv->jbuf) >= 32765 &&
3161 rtp_jitter_buffer_num_packets (priv->jbuf) > 10000 &&
3162 priv->srcresult == GST_FLOW_OK)
3163 JBUF_WAIT_QUEUE (priv);
3164 if (priv->srcresult != GST_FLOW_OK)
3167 /* let's check if this buffer is too late, we can only accept packets with
3168 * bigger seqnum than the one we last pushed. */
3169 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3172 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3174 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3175 if (G_UNLIKELY (gap <= 0)) {
3176 if (priv->do_retransmission) {
3177 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3178 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3179 /* Only count the retranmitted packet too late if it has been
3180 * considered lost. If the original packet arrived before the
3181 * retransmitted we just count it as a duplicate. */
3182 if (timer->type != TIMER_TYPE_LOST)
3190 if (already_lost (jitterbuffer, seqnum))
3193 /* let's drop oldest packet if the queue is already full and drop-on-latency
3194 * is set. We can only do this when there actually is a latency. When no
3195 * latency is set, we just pump it in the queue and let the other end push it
3196 * out as fast as possible. */
3197 if (priv->latency_ms && priv->drop_on_latency) {
3199 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3201 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3202 RTPJitterBufferItem *old_item;
3204 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3206 if (IS_DROPABLE (old_item)) {
3207 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3208 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3210 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3211 free_item (old_item);
3213 /* we might have removed some head buffers, signal the pushing thread to
3214 * see if it can push now */
3215 JBUF_SIGNAL_EVENT (priv);
3219 /* If we estimated the DTS, don't consider it in the clock skew calculations
3220 * later. The code above always sets dts to pts or the other way around if
3221 * any of those is valid in the buffer, so we know that if we estimated the
3222 * dts that both are unknown */
3225 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3226 pts, seqnum, 1, rtptime);
3228 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3230 /* now insert the packet into the queue in sorted order. This function returns
3231 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3232 * have a duplicate. */
3233 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3235 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3236 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3240 /* Trigger fast start if needed */
3241 if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
3245 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3246 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3248 /* we had an unhandled SR, handle it now */
3250 do_handle_sync (jitterbuffer);
3252 if (G_UNLIKELY (head)) {
3253 /* signal addition of new buffer when the _loop is waiting. */
3254 if (G_LIKELY (priv->active))
3255 JBUF_SIGNAL_EVENT (priv);
3257 /* let's unschedule and unblock any waiting buffers. We only want to do this
3258 * when the head buffer changed */
3259 if (G_UNLIKELY (priv->clock_id)) {
3260 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3261 unschedule_current_timer (jitterbuffer);
3265 GST_DEBUG_OBJECT (jitterbuffer,
3266 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3267 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3269 msg = check_buffering_percent (jitterbuffer, percent);
3275 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3282 /* this is not fatal but should be filtered earlier */
3283 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3284 ("Received invalid RTP payload, dropping"));
3285 gst_buffer_unref (buffer);
3290 GST_WARNING_OBJECT (jitterbuffer,
3291 "No clock-rate in caps!, dropping buffer");
3292 gst_buffer_unref (buffer);
3297 ret = priv->srcresult;
3298 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3299 gst_buffer_unref (buffer);
3305 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3306 gst_buffer_unref (buffer);
3311 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3312 " popped, dropping", seqnum, priv->last_popped_seqnum);
3314 gst_buffer_unref (buffer);
3319 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3320 "considered lost", seqnum);
3322 gst_buffer_unref (buffer);
3327 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3329 priv->num_duplicates++;
3335 GST_DEBUG_OBJECT (jitterbuffer,
3336 "Duplicate RTX packet #%d detected, dropping", seqnum);
3337 priv->num_duplicates++;
3338 gst_buffer_unref (buffer);
3343 /* FIXME: hopefully we can do something more efficient here, especially when
3344 * all packets are in order and/or outside of the currently cached range.
3345 * Still worthwhile to have it, avoids taking/releasing object lock and pad
3346 * stream lock for every single buffer in the default chain_list fallback. */
3347 static GstFlowReturn
3348 gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent,
3349 GstBufferList * buffer_list)
3351 GstFlowReturn flow_ret = GST_FLOW_OK;
3354 n = gst_buffer_list_length (buffer_list);
3355 for (i = 0; i < n; ++i) {
3356 GstBuffer *buf = gst_buffer_list_get (buffer_list, i);
3358 flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf));
3360 if (flow_ret != GST_FLOW_OK)
3363 gst_buffer_list_unref (buffer_list);
3369 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3371 guint64 ext_time, elapsed;
3373 GstRtpJitterBufferPrivate *priv;
3375 priv = jitterbuffer->priv;
3376 rtp_time = item->rtptime;
3378 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3379 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3381 ext_time = priv->ext_timestamp;
3382 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3383 if (ext_time < priv->ext_timestamp) {
3384 ext_time = priv->ext_timestamp;
3386 priv->ext_timestamp = ext_time;
3389 if (ext_time > priv->clock_base)
3390 elapsed = ext_time - priv->clock_base;
3394 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3399 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3400 RTPJitterBufferItem * item)
3402 guint64 total, elapsed, left, estimated;
3403 GstClockTime out_time;
3404 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3406 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3407 || priv->clock_base == -1 || priv->clock_rate <= 0)
3410 /* compute the elapsed time */
3411 elapsed = compute_elapsed (jitterbuffer, item);
3413 /* do nothing if elapsed time doesn't increment */
3414 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3417 priv->last_elapsed = elapsed;
3419 /* this is the total time we need to play */
3420 total = priv->npt_stop - priv->npt_start;
3421 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3422 GST_TIME_ARGS (total));
3424 /* this is how much time there is left */
3425 if (total > elapsed)
3426 left = total - elapsed;
3430 /* if we have less time left that the size of the buffer, we will not
3431 * be able to keep it filled, disabled buffering then */
3432 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3433 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3434 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3435 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3438 /* this is the current time as running-time */
3439 out_time = item->pts;
3442 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3444 /* if there is almost nothing left,
3445 * we may never advance enough to end up in the above case */
3446 if (total < GST_SECOND)
3447 estimated = GST_SECOND;
3451 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3452 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3454 if (estimated != -1 && priv->estimated_eos != estimated) {
3455 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3456 priv->estimated_eos = estimated;
3460 /* take a buffer from the queue and push it */
3461 static GstFlowReturn
3462 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3464 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3465 GstFlowReturn result = GST_FLOW_OK;
3466 RTPJitterBufferItem *item;
3467 GstBuffer *outbuf = NULL;
3468 GstEvent *outevent = NULL;
3469 GstQuery *outquery = NULL;
3470 GstClockTime dts, pts;
3472 gboolean do_push = TRUE;
3476 /* when we get here we are ready to pop and push the buffer */
3477 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3481 case ITEM_TYPE_BUFFER:
3483 /* we need to make writable to change the flags and timestamps */
3484 outbuf = gst_buffer_make_writable (item->data);
3486 if (G_UNLIKELY (priv->discont)) {
3487 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3488 * into the jitterbuffer so we can modify now. */
3489 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3490 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3491 priv->discont = FALSE;
3493 if (G_UNLIKELY (priv->ts_discont)) {
3494 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3495 priv->ts_discont = FALSE;
3499 gst_segment_position_from_running_time (&priv->segment,
3500 GST_FORMAT_TIME, item->dts);
3502 gst_segment_position_from_running_time (&priv->segment,
3503 GST_FORMAT_TIME, item->pts);
3505 /* if this is a new frame, check if ts_offset needs to be updated */
3506 if (pts != priv->last_pts) {
3507 update_offset (jitterbuffer);
3510 /* apply timestamp with offset to buffer now */
3511 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3512 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3514 /* update the elapsed time when we need to check against the npt stop time. */
3515 update_estimated_eos (jitterbuffer, item);
3517 priv->last_pts = pts;
3518 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3520 case ITEM_TYPE_LOST:
3521 priv->discont = TRUE;
3525 case ITEM_TYPE_EVENT:
3526 outevent = item->data;
3528 case ITEM_TYPE_QUERY:
3529 outquery = item->data;
3533 /* now we are ready to push the buffer. Save the seqnum and release the lock
3534 * so the other end can push stuff in the queue again. */
3536 priv->last_popped_seqnum = seqnum;
3537 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3539 msg = check_buffering_percent (jitterbuffer, percent);
3541 if (type == ITEM_TYPE_EVENT && outevent &&
3542 GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3543 g_assert (priv->eos);
3544 while (priv->timers->len > 0) {
3545 /* Stopping timers */
3546 unschedule_current_timer (jitterbuffer);
3547 JBUF_WAIT_TIMER (priv);
3557 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3560 case ITEM_TYPE_BUFFER:
3562 GST_DEBUG_OBJECT (jitterbuffer,
3563 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3564 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3565 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3567 result = gst_pad_push (priv->srcpad, outbuf);
3569 JBUF_LOCK_CHECK (priv, out_flushing);
3571 case ITEM_TYPE_LOST:
3572 case ITEM_TYPE_EVENT:
3573 /* We got not enough consecutive packets with a huge gap, we can
3574 * as well just drop them here now on EOS */
3575 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3576 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3577 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3578 g_queue_clear (&priv->gap_packets);
3581 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3582 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3585 gst_pad_push_event (priv->srcpad, outevent);
3587 gst_event_unref (outevent);
3589 result = GST_FLOW_OK;
3591 JBUF_LOCK_CHECK (priv, out_flushing);
3593 case ITEM_TYPE_QUERY:
3597 res = gst_pad_peer_query (priv->srcpad, outquery);
3599 JBUF_LOCK_CHECK (priv, out_flushing);
3600 result = GST_FLOW_OK;
3601 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3602 JBUF_SIGNAL_QUERY (priv, res);
3611 return priv->srcresult;
3615 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3617 /* Peek a buffer and compare the seqnum to the expected seqnum.
3618 * If all is fine, the buffer is pushed.
3619 * If something is wrong, we wait for some event
3621 static GstFlowReturn
3622 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3624 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3625 GstFlowReturn result;
3626 RTPJitterBufferItem *item;
3628 guint32 next_seqnum;
3630 /* only push buffers when PLAYING and active and not buffering */
3631 if (priv->blocked || !priv->active ||
3632 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3633 return GST_FLOW_WAIT;
3636 /* peek a buffer, we're just looking at the sequence number.
3637 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3638 * wait for a timeout or something to change.
3639 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3640 item = rtp_jitter_buffer_peek (priv->jbuf);
3645 /* get the seqnum and the next expected seqnum */
3646 seqnum = item->seqnum;
3648 return pop_and_push_next (jitterbuffer, seqnum);
3651 next_seqnum = priv->next_seqnum;
3653 /* get the gap between this and the previous packet. If we don't know the
3654 * previous packet seqnum assume no gap. */
3655 if (G_UNLIKELY (next_seqnum == -1)) {
3656 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3657 /* we don't know what the next_seqnum should be, the chain function should
3658 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3659 * fires, so wait for that */
3660 result = GST_FLOW_WAIT;
3662 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3664 if (G_LIKELY (gap == 0)) {
3665 /* no missing packet, pop and push */
3666 result = pop_and_push_next (jitterbuffer, seqnum);
3667 } else if (G_UNLIKELY (gap < 0)) {
3668 /* if we have a packet that we already pushed or considered dropped, pop it
3669 * off and get the next packet */
3670 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3671 seqnum, next_seqnum);
3672 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3674 result = GST_FLOW_OK;
3676 /* the chain function has scheduled timers to request retransmission or
3677 * when to consider the packet lost, wait for that */
3678 GST_DEBUG_OBJECT (jitterbuffer,
3679 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3680 next_seqnum, seqnum, gap);
3681 /* if we have reached EOS, just keep processing */
3683 result = pop_and_push_next (jitterbuffer, seqnum);
3684 result = GST_FLOW_OK;
3686 result = GST_FLOW_WAIT;
3695 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3697 return GST_FLOW_EOS;
3699 return GST_FLOW_WAIT;
3705 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3707 GstClockTime rtx_retry_timeout;
3708 GstClockTime rtx_min_retry_timeout;
3710 if (priv->rtx_retry_timeout == -1) {
3711 if (priv->avg_rtx_rtt == 0)
3712 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3714 /* we want to ask for a retransmission after we waited for a
3715 * complete RTT and the additional jitter */
3716 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3718 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3720 /* make sure we don't retry too often. On very low latency networks,
3721 * the RTT and jitter can be very low. */
3722 if (priv->rtx_min_retry_timeout == -1) {
3723 rtx_min_retry_timeout = priv->packet_spacing;
3725 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3727 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3729 return rtx_retry_timeout;
3733 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3734 GstClockTime rtx_retry_timeout)
3736 GstClockTime rtx_retry_period;
3738 if (priv->rtx_retry_period == -1) {
3739 /* we retry up to the configured jitterbuffer size but leaving some
3740 * room for the retransmission to arrive in time */
3741 if (rtx_retry_timeout > priv->latency_ns) {
3742 rtx_retry_period = 0;
3744 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3747 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3749 return rtx_retry_period;
3753 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3754 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3755 3. For very large measurements (> avg * 2), consider them "outliers"
3756 and count them a lot less (1/48th)
3759 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3763 if (priv->avg_rtx_rtt == 0) {
3764 priv->avg_rtx_rtt = rtt;
3768 if (rtt > 2 * priv->avg_rtx_rtt)
3770 else if (rtt > priv->avg_rtx_rtt)
3775 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3779 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3780 GstClockTime dts, gboolean success)
3782 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3786 /* we scheduled a retry for this packet and now we have it */
3787 priv->num_rtx_success++;
3788 /* all the previous retry attempts failed */
3789 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3791 /* All retries failed or was too late */
3792 priv->num_rtx_failed += timer->num_rtx_retry;
3795 /* number of retries before (hopefully) receiving the packet */
3796 if (priv->avg_rtx_num == 0.0)
3797 priv->avg_rtx_num = timer->num_rtx_retry;
3799 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3801 /* Calculate the delay between retransmission request and receiving this
3802 * packet. We have a valid delay if and only if this packet is a response to
3803 * our last request. If not we don't know if this is a response to an
3804 * earlier request and delay could be way off. For RTT is more important
3805 * with correct values than to update for every packet. */
3806 if (timer->num_rtx_retry == timer->num_rtx_received &&
3807 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3808 delay = dts - timer->rtx_last;
3809 update_avg_rtx_rtt (priv, delay);
3814 GST_LOG_OBJECT (jitterbuffer,
3815 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3816 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3817 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3818 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3819 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3820 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3821 GST_TIME_ARGS (priv->avg_rtx_rtt));
3824 /* the timeout for when we expected a packet expired */
3826 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3829 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3831 guint delay, delay_ms, avg_rtx_rtt_ms;
3832 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3833 guint rtx_deadline_ms;
3834 GstClockTime rtx_retry_period;
3835 GstClockTime rtx_retry_timeout;
3838 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3839 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3841 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3842 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3844 delay = timer->rtx_delay + timer->rtx_retry;
3846 delay_ms = GST_TIME_AS_MSECONDS (delay);
3847 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3848 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3849 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3851 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3853 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3854 gst_structure_new ("GstRTPRetransmissionRequest",
3855 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3856 "running-time", G_TYPE_UINT64, timer->rtx_base,
3857 "delay", G_TYPE_UINT, delay_ms,
3858 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3859 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3860 "period", G_TYPE_UINT, rtx_retry_period_ms,
3861 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3862 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3863 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3864 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3866 priv->num_rtx_requests++;
3867 timer->num_rtx_retry++;
3869 GST_OBJECT_LOCK (jitterbuffer);
3870 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3871 timer->rtx_last = gst_clock_get_time (clock);
3872 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3874 timer->rtx_last = now;
3876 GST_OBJECT_UNLOCK (jitterbuffer);
3878 /* calculate the timeout for the next retransmission attempt */
3879 timer->rtx_retry += rtx_retry_timeout;
3880 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3881 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3882 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3883 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3884 if ((priv->rtx_max_retries != -1
3885 && timer->num_rtx_retry >= priv->rtx_max_retries)
3886 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3887 || (timer->rtx_base + rtx_retry_period < now)) {
3888 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3889 /* too many retransmission request, we now convert the timer
3890 * to a lost timer, leave the num_rtx_retry as it is for stats */
3891 timer->type = TIMER_TYPE_LOST;
3892 timer->rtx_delay = 0;
3893 timer->rtx_retry = 0;
3895 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3896 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3899 gst_pad_push_event (priv->sinkpad, event);
3905 /* a packet is lost */
3907 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3910 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3911 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3913 GstEvent *event = NULL;
3914 RTPJitterBufferItem *item;
3916 seqnum = timer->seqnum;
3917 lost_packets = MAX (timer->num, 1);
3918 num_rtx_retry = timer->num_rtx_retry;
3920 /* we had a gap and thus we lost some packets. Create an event for this. */
3921 if (lost_packets > 1)
3922 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3923 seqnum + lost_packets - 1);
3925 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3927 priv->num_lost += lost_packets;
3928 priv->num_rtx_failed += num_rtx_retry;
3930 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3932 /* we now only accept seqnum bigger than this */
3933 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3934 priv->next_in_seqnum = next_in_seqnum;
3935 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3938 /* Avoid creating events if we don't need it. Note that we still need to create
3939 * the lost *ITEM* since it will be used to notify the outgoing thread of
3940 * lost items (so that we can set discont flags and such) */
3941 if (priv->do_lost) {
3942 GstClockTime duration, timestamp;
3943 /* create paket lost event */
3944 timestamp = apply_offset (jitterbuffer, timer->timeout);
3945 duration = timer->duration;
3946 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3947 duration = priv->packet_spacing;
3948 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3949 gst_structure_new ("GstRTPPacketLost",
3950 "seqnum", G_TYPE_UINT, (guint) seqnum,
3951 "timestamp", G_TYPE_UINT64, timestamp,
3952 "duration", G_TYPE_UINT64, duration,
3953 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3955 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3956 if (!rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL))
3960 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3961 /* Store info to update stats if the packet arrives too late */
3962 timer_queue_append (priv->rtx_stats_timers, timer,
3963 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3965 remove_timer (jitterbuffer, timer);
3968 JBUF_SIGNAL_EVENT (priv);
3974 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3977 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3979 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3980 remove_timer (jitterbuffer, timer);
3984 /* there was no EOS in the buffer, put one in there now */
3985 event = gst_event_new_eos ();
3986 if (priv->segment_seqnum != GST_SEQNUM_INVALID)
3987 gst_event_set_seqnum (event, priv->segment_seqnum);
3988 queue_event (jitterbuffer, event);
3990 JBUF_SIGNAL_EVENT (priv);
3996 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3999 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
4001 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
4003 /* timer seqnum might have been obsoleted by caps seqnum-base,
4004 * only mess with current ongoing seqnum if still unknown */
4005 if (priv->next_seqnum == -1)
4006 priv->next_seqnum = timer->seqnum;
4007 remove_timer (jitterbuffer, timer);
4008 JBUF_SIGNAL_EVENT (priv);
4014 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
4017 gboolean removed = FALSE;
4019 switch (timer->type) {
4020 case TIMER_TYPE_EXPECTED:
4021 removed = do_expected_timeout (jitterbuffer, timer, now);
4023 case TIMER_TYPE_LOST:
4024 removed = do_lost_timeout (jitterbuffer, timer, now);
4026 case TIMER_TYPE_DEADLINE:
4027 removed = do_deadline_timeout (jitterbuffer, timer, now);
4029 case TIMER_TYPE_EOS:
4030 removed = do_eos_timeout (jitterbuffer, timer, now);
4036 /* called when we need to wait for the next timeout.
4038 * We loop over the array of recorded timeouts and wait for the earliest one.
4039 * When it timed out, do the logic associated with the timer.
4041 * If there are no timers, we wait on a gcond until something new happens.
4044 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
4046 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
4047 GstClockTime now = 0;
4050 while (priv->timer_running) {
4051 TimerData *timer = NULL;
4052 GstClockTime timer_timeout = -1;
4055 /* If we have a clock, update "now" now with the very
4056 * latest running time we have. If timers are unscheduled below we
4057 * otherwise wouldn't update now (it's only updated when timers
4058 * expire), and also for the very first loop iteration now would
4059 * otherwise always be 0
4061 GST_OBJECT_LOCK (jitterbuffer);
4063 now = GST_CLOCK_TIME_NONE;
4064 } else if (GST_ELEMENT_CLOCK (jitterbuffer)) {
4066 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
4067 GST_ELEMENT_CAST (jitterbuffer)->base_time;
4069 GST_OBJECT_UNLOCK (jitterbuffer);
4071 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
4072 GST_TIME_ARGS (now));
4074 /* Clear expired rtx-stats timers */
4075 if (priv->do_retransmission)
4076 timer_queue_clear_until (priv->rtx_stats_timers, now);
4078 /* Iterate "normal" timers */
4079 len = priv->timers->len;
4080 for (i = 0; i < len;) {
4081 TimerData *test = &g_array_index (priv->timers, TimerData, i);
4082 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
4083 gboolean save_best = FALSE;
4085 GST_DEBUG_OBJECT (jitterbuffer,
4086 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
4087 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
4088 GST_STIME_ARGS ((gint64) (test_timeout - now)));
4090 /* Weed out anything too late */
4091 if (test->type == TIMER_TYPE_LOST &&
4092 (test_timeout == -1 || test_timeout <= now)) {
4093 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
4094 do_lost_timeout (jitterbuffer, test, now);
4095 if (!priv->timer_running)
4097 /* We don't move the iterator forward since we just removed the current entry,
4098 * but we update the termination condition */
4099 len = priv->timers->len;
4101 /* find the smallest timeout */
4102 if (timer == NULL) {
4104 } else if (timer_timeout == -1) {
4105 /* we already have an immediate timeout, the new timer must be an
4106 * immediate timer with smaller seqnum to become the best */
4107 if (test_timeout == -1
4108 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4109 timer->seqnum) > 0))
4111 } else if (test_timeout == -1) {
4112 /* first immediate timer */
4114 } else if (test_timeout < timer_timeout) {
4117 } else if (test_timeout == timer_timeout
4118 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4119 timer->seqnum) > 0)) {
4120 /* same timer, smaller seqnum */
4125 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
4127 timer_timeout = test_timeout;
4132 if (timer && !priv->blocked) {
4134 GstClockTime sync_time;
4137 GstClockTimeDiff clock_jitter;
4139 if (timer_timeout == -1 || timer_timeout <= now || priv->eos) {
4140 /* We have normally removed all lost timers in the loop above */
4141 g_assert (timer->type != TIMER_TYPE_LOST);
4143 do_timeout (jitterbuffer, timer, now);
4144 /* check here, do_timeout could have released the lock */
4145 if (!priv->timer_running)
4150 GST_OBJECT_LOCK (jitterbuffer);
4151 clock = GST_ELEMENT_CLOCK (jitterbuffer);
4153 GST_OBJECT_UNLOCK (jitterbuffer);
4154 /* let's just push if there is no clock */
4155 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
4156 now = timer_timeout;
4160 /* prepare for sync against clock */
4161 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
4162 /* add latency of peer to get input time */
4163 sync_time += priv->peer_latency;
4165 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
4166 " with sync time %" GST_TIME_FORMAT,
4167 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
4169 /* create an entry for the clock */
4170 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
4171 priv->timer_timeout = timer_timeout;
4172 priv->timer_seqnum = timer->seqnum;
4173 GST_OBJECT_UNLOCK (jitterbuffer);
4175 /* release the lock so that the other end can push stuff or unlock */
4178 ret = gst_clock_id_wait (id, &clock_jitter);
4181 if (!priv->timer_running) {
4182 gst_clock_id_unref (id);
4183 priv->clock_id = NULL;
4187 if (ret != GST_CLOCK_UNSCHEDULED) {
4188 now = timer_timeout + MAX (clock_jitter, 0);
4189 GST_DEBUG_OBJECT (jitterbuffer,
4190 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
4191 GST_STIME_ARGS (clock_jitter));
4193 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
4195 /* and free the entry */
4196 gst_clock_id_unref (id);
4197 priv->clock_id = NULL;
4199 /* no timers, wait for activity */
4200 JBUF_WAIT_TIMER (priv);
4205 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4210 * This funcion implements the main pushing loop on the source pad.
4212 * It first tries to push as many buffers as possible. If there is a seqnum
4213 * mismatch, we wait for the next timeouts.
4216 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4218 GstRtpJitterBufferPrivate *priv;
4219 GstFlowReturn result = GST_FLOW_OK;
4221 priv = jitterbuffer->priv;
4223 JBUF_LOCK_CHECK (priv, flushing);
4225 result = handle_next_buffer (jitterbuffer);
4226 JBUF_SIGNAL_QUEUE (priv);
4227 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4228 /* now wait for the next event */
4229 JBUF_WAIT_EVENT (priv, flushing);
4230 result = GST_FLOW_OK;
4232 } while (result == GST_FLOW_OK);
4233 /* store result for upstream */
4234 priv->srcresult = result;
4235 /* if we get here we need to pause */
4241 result = priv->srcresult;
4248 JBUF_SIGNAL_QUERY (priv, FALSE);
4251 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4252 gst_flow_get_name (result));
4253 gst_pad_pause_task (priv->srcpad);
4254 if (result == GST_FLOW_EOS) {
4255 event = gst_event_new_eos ();
4256 if (priv->segment_seqnum != GST_SEQNUM_INVALID)
4257 gst_event_set_seqnum (event, priv->segment_seqnum);
4258 gst_pad_push_event (priv->srcpad, event);
4264 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4265 * some sanity checks and then emit the handle-sync signal with the parameters.
4266 * This function must be called with the LOCK */
4268 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4270 GstRtpJitterBufferPrivate *priv;
4271 guint64 base_rtptime, base_time;
4273 guint64 last_rtptime;
4275 guint64 ext_rtptime, diff;
4276 gboolean valid = TRUE, keep = FALSE;
4278 priv = jitterbuffer->priv;
4280 /* get the last values from the jitterbuffer */
4281 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4282 &clock_rate, &last_rtptime);
4284 clock_base = priv->clock_base;
4285 ext_rtptime = priv->ext_rtptime;
4287 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4288 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4289 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4290 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4292 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4293 /* we keep this SR packet for later. When we get a valid RTP packet the
4294 * above values will be set and we can try to use the SR packet */
4295 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4298 /* we can't accept anything that happened before we did the last resync */
4299 if (base_rtptime > ext_rtptime) {
4300 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4303 /* the SR RTP timestamp must be something close to what we last observed
4304 * in the jitterbuffer */
4305 if (ext_rtptime > last_rtptime) {
4306 /* check how far ahead it is to our RTP timestamps */
4307 diff = ext_rtptime - last_rtptime;
4308 /* if bigger than 1 second, we drop it */
4309 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4311 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4312 clock_rate, 1000)) {
4313 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4314 /* should drop this, but some RTSP servers end up with bogus
4315 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4316 * so still trigger rptbin sync but invalidate RTCP data
4317 * (sync might use other methods) */
4320 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4321 G_GUINT64_FORMAT, last_rtptime, diff);
4327 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4331 s = gst_structure_new ("application/x-rtp-sync",
4332 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4333 "base-time", G_TYPE_UINT64, base_time,
4334 "clock-rate", G_TYPE_UINT, clock_rate,
4335 "clock-base", G_TYPE_UINT64, clock_base,
4336 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4337 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4339 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4340 gst_buffer_replace (&priv->last_sr, NULL);
4342 g_signal_emit (jitterbuffer,
4343 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4345 gst_structure_free (s);
4347 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4348 gst_buffer_replace (&priv->last_sr, NULL);
4352 static GstFlowReturn
4353 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4356 GstRtpJitterBuffer *jitterbuffer;
4357 GstRtpJitterBufferPrivate *priv;
4358 GstFlowReturn ret = GST_FLOW_OK;
4360 GstRTCPPacket packet;
4361 guint64 ext_rtptime;
4363 GstRTCPBuffer rtcp = { NULL, };
4365 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4367 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4368 goto invalid_buffer;
4370 priv = jitterbuffer->priv;
4372 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4374 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4377 /* first packet must be SR or RR or else the validate would have failed */
4378 switch (gst_rtcp_packet_get_type (&packet)) {
4379 case GST_RTCP_TYPE_SR:
4380 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4386 gst_rtcp_buffer_unmap (&rtcp);
4388 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4391 /* convert the RTP timestamp to our extended timestamp, using the same offset
4392 * we used in the jitterbuffer */
4393 ext_rtptime = priv->jbuf->ext_rtptime;
4394 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4396 priv->ext_rtptime = ext_rtptime;
4397 gst_buffer_replace (&priv->last_sr, buffer);
4399 do_handle_sync (jitterbuffer);
4403 gst_buffer_unref (buffer);
4409 /* this is not fatal but should be filtered earlier */
4410 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4411 ("Received invalid RTCP payload, dropping"));
4417 /* this is not fatal but should be filtered earlier */
4418 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4419 ("Received empty RTCP payload, dropping"));
4420 gst_rtcp_buffer_unmap (&rtcp);
4426 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4427 gst_rtcp_buffer_unmap (&rtcp);
4434 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4437 gboolean res = FALSE;
4438 GstRtpJitterBuffer *jitterbuffer;
4439 GstRtpJitterBufferPrivate *priv;
4441 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4442 priv = jitterbuffer->priv;
4444 switch (GST_QUERY_TYPE (query)) {
4445 case GST_QUERY_CAPS:
4447 GstCaps *filter, *caps;
4449 gst_query_parse_caps (query, &filter);
4450 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4451 gst_query_set_caps_result (query, caps);
4452 gst_caps_unref (caps);
4457 if (GST_QUERY_IS_SERIALIZED (query)) {
4458 RTPJitterBufferItem *item;
4461 JBUF_LOCK_CHECK (priv, out_flushing);
4462 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4463 RTP_JITTER_BUFFER_MODE_BUFFER) {
4464 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4465 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4466 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4468 JBUF_SIGNAL_EVENT (priv);
4469 JBUF_WAIT_QUERY (priv, out_flushing);
4470 res = priv->last_query;
4472 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4477 res = gst_pad_query_default (pad, parent, query);
4485 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4493 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4496 GstRtpJitterBuffer *jitterbuffer;
4497 GstRtpJitterBufferPrivate *priv;
4498 gboolean res = FALSE;
4500 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4501 priv = jitterbuffer->priv;
4503 switch (GST_QUERY_TYPE (query)) {
4504 case GST_QUERY_LATENCY:
4506 /* We need to send the query upstream and add the returned latency to our
4508 GstClockTime min_latency, max_latency;
4510 GstClockTime our_latency;
4512 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4513 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4515 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4516 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4517 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4519 /* store this so that we can safely sync on the peer buffers. */
4521 priv->peer_latency = min_latency;
4522 our_latency = priv->latency_ns;
4525 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4526 GST_TIME_ARGS (our_latency));
4528 /* we add some latency but can buffer an infinite amount of time */
4529 min_latency += our_latency;
4532 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4533 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4534 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4536 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4540 case GST_QUERY_POSITION:
4542 GstClockTime start, last_out;
4545 gst_query_parse_position (query, &fmt, NULL);
4546 if (fmt != GST_FORMAT_TIME) {
4547 res = gst_pad_query_default (pad, parent, query);
4552 start = priv->npt_start;
4553 last_out = priv->last_out_time;
4556 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4557 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4558 GST_TIME_ARGS (last_out));
4560 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4561 /* bring 0-based outgoing time to stream time */
4562 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4565 res = gst_pad_query_default (pad, parent, query);
4569 case GST_QUERY_CAPS:
4571 GstCaps *filter, *caps;
4573 gst_query_parse_caps (query, &filter);
4574 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4575 gst_query_set_caps_result (query, caps);
4576 gst_caps_unref (caps);
4581 res = gst_pad_query_default (pad, parent, query);
4589 gst_rtp_jitter_buffer_set_property (GObject * object,
4590 guint prop_id, const GValue * value, GParamSpec * pspec)
4592 GstRtpJitterBuffer *jitterbuffer;
4593 GstRtpJitterBufferPrivate *priv;
4595 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4596 priv = jitterbuffer->priv;
4601 guint new_latency, old_latency;
4603 new_latency = g_value_get_uint (value);
4606 old_latency = priv->latency_ms;
4607 priv->latency_ms = new_latency;
4608 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4609 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4612 /* post message if latency changed, this will inform the parent pipeline
4613 * that a latency reconfiguration is possible/needed. */
4614 if (new_latency != old_latency) {
4615 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4616 GST_TIME_ARGS (new_latency * GST_MSECOND));
4618 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4619 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4623 case PROP_DROP_ON_LATENCY:
4625 priv->drop_on_latency = g_value_get_boolean (value);
4628 case PROP_TS_OFFSET:
4630 if (priv->max_ts_offset_adjustment != 0) {
4631 gint64 new_offset = g_value_get_int64 (value);
4633 if (new_offset > priv->ts_offset) {
4634 priv->ts_offset_remainder = new_offset - priv->ts_offset;
4636 priv->ts_offset_remainder = -(priv->ts_offset - new_offset);
4639 priv->ts_offset = g_value_get_int64 (value);
4640 priv->ts_offset_remainder = 0;
4642 priv->ts_discont = TRUE;
4645 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4647 priv->max_ts_offset_adjustment = g_value_get_uint64 (value);
4652 priv->do_lost = g_value_get_boolean (value);
4657 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4660 case PROP_DO_RETRANSMISSION:
4662 priv->do_retransmission = g_value_get_boolean (value);
4665 case PROP_RTX_NEXT_SEQNUM:
4667 priv->rtx_next_seqnum = g_value_get_boolean (value);
4670 case PROP_RTX_DELAY:
4672 priv->rtx_delay = g_value_get_int (value);
4675 case PROP_RTX_MIN_DELAY:
4677 priv->rtx_min_delay = g_value_get_uint (value);
4680 case PROP_RTX_DELAY_REORDER:
4682 priv->rtx_delay_reorder = g_value_get_int (value);
4685 case PROP_RTX_RETRY_TIMEOUT:
4687 priv->rtx_retry_timeout = g_value_get_int (value);
4690 case PROP_RTX_MIN_RETRY_TIMEOUT:
4692 priv->rtx_min_retry_timeout = g_value_get_int (value);
4695 case PROP_RTX_RETRY_PERIOD:
4697 priv->rtx_retry_period = g_value_get_int (value);
4700 case PROP_RTX_MAX_RETRIES:
4702 priv->rtx_max_retries = g_value_get_int (value);
4705 case PROP_RTX_DEADLINE:
4707 priv->rtx_deadline_ms = g_value_get_int (value);
4710 case PROP_RTX_STATS_TIMEOUT:
4712 priv->rtx_stats_timeout = g_value_get_uint (value);
4715 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4717 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4720 case PROP_MAX_DROPOUT_TIME:
4722 priv->max_dropout_time = g_value_get_uint (value);
4725 case PROP_MAX_MISORDER_TIME:
4727 priv->max_misorder_time = g_value_get_uint (value);
4730 case PROP_RFC7273_SYNC:
4732 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4733 g_value_get_boolean (value));
4736 case PROP_FASTSTART_MIN_PACKETS:
4738 priv->faststart_min_packets = g_value_get_uint (value);
4742 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4748 gst_rtp_jitter_buffer_get_property (GObject * object,
4749 guint prop_id, GValue * value, GParamSpec * pspec)
4751 GstRtpJitterBuffer *jitterbuffer;
4752 GstRtpJitterBufferPrivate *priv;
4754 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4755 priv = jitterbuffer->priv;
4760 g_value_set_uint (value, priv->latency_ms);
4763 case PROP_DROP_ON_LATENCY:
4765 g_value_set_boolean (value, priv->drop_on_latency);
4768 case PROP_TS_OFFSET:
4770 g_value_set_int64 (value, priv->ts_offset);
4773 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4775 g_value_set_uint64 (value, priv->max_ts_offset_adjustment);
4780 g_value_set_boolean (value, priv->do_lost);
4785 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4793 if (priv->srcresult != GST_FLOW_OK)
4796 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4798 g_value_set_int (value, percent);
4802 case PROP_DO_RETRANSMISSION:
4804 g_value_set_boolean (value, priv->do_retransmission);
4807 case PROP_RTX_NEXT_SEQNUM:
4809 g_value_set_boolean (value, priv->rtx_next_seqnum);
4812 case PROP_RTX_DELAY:
4814 g_value_set_int (value, priv->rtx_delay);
4817 case PROP_RTX_MIN_DELAY:
4819 g_value_set_uint (value, priv->rtx_min_delay);
4822 case PROP_RTX_DELAY_REORDER:
4824 g_value_set_int (value, priv->rtx_delay_reorder);
4827 case PROP_RTX_RETRY_TIMEOUT:
4829 g_value_set_int (value, priv->rtx_retry_timeout);
4832 case PROP_RTX_MIN_RETRY_TIMEOUT:
4834 g_value_set_int (value, priv->rtx_min_retry_timeout);
4837 case PROP_RTX_RETRY_PERIOD:
4839 g_value_set_int (value, priv->rtx_retry_period);
4842 case PROP_RTX_MAX_RETRIES:
4844 g_value_set_int (value, priv->rtx_max_retries);
4847 case PROP_RTX_DEADLINE:
4849 g_value_set_int (value, priv->rtx_deadline_ms);
4852 case PROP_RTX_STATS_TIMEOUT:
4854 g_value_set_uint (value, priv->rtx_stats_timeout);
4858 g_value_take_boxed (value,
4859 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4861 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4863 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4866 case PROP_MAX_DROPOUT_TIME:
4868 g_value_set_uint (value, priv->max_dropout_time);
4871 case PROP_MAX_MISORDER_TIME:
4873 g_value_set_uint (value, priv->max_misorder_time);
4876 case PROP_RFC7273_SYNC:
4878 g_value_set_boolean (value,
4879 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4882 case PROP_FASTSTART_MIN_PACKETS:
4884 g_value_set_uint (value, priv->faststart_min_packets);
4888 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4893 static GstStructure *
4894 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4896 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4900 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4901 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4902 "num-lost", G_TYPE_UINT64, priv->num_lost,
4903 "num-late", G_TYPE_UINT64, priv->num_late,
4904 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4905 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4906 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4907 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4908 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4909 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);