2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
10 * Copyright 2016 Pexip AS
11 * @author: Havard Graff <havard@pexip.com>
12 * @author: Stian Selnes <stian@pexip.com>
14 * This library is free software; you can redistribute it and/or
15 * modify it under the terms of the GNU Library General Public
16 * License as published by the Free Software Foundation; either
17 * version 2 of the License, or (at your option) any later version.
19 * This library is distributed in the hope that it will be useful,
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
22 * Library General Public License for more details.
24 * You should have received a copy of the GNU Library General Public
25 * License along with this library; if not, write to the
26 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
27 * Boston, MA 02110-1301, USA.
32 * SECTION:element-rtpjitterbuffer
34 * This element reorders and removes duplicate RTP packets as they are received
35 * from a network source.
37 * The element needs the clock-rate of the RTP payload in order to estimate the
38 * delay. This information is obtained either from the caps on the sink pad or,
39 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
40 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
42 * The rtpjitterbuffer will wait for missing packets up to a configurable time
43 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
44 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
45 * property is set, lost packets will result in a custom serialized downstream
46 * event of name GstRTPPacketLost. The lost packet events are usually used by a
47 * depayloader or other element to create concealment data or some other logic
48 * to gracefully handle the missing packets.
50 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
51 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
54 * The jitterbuffer can also be configured to send early retransmission events
55 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
56 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
57 * sends a custom upstream event named GstRTPRetransmissionRequest when the
58 * packet is considered late. The initial expected packet arrival time is
59 * calculated as follows:
61 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
62 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
63 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
64 * packets with different rtptime.
66 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
67 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
68 * previously scheduled timeout is overwritten.
70 * - If seqnum N arrived, all seqnum older than
71 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
72 * immediately. This is to request fast feedback for abonormally reorder
73 * packets before any of the previous timeouts is triggered.
75 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
76 * event. After the initial timeout expires and the retransmission event is
77 * sent, the timeout is scheduled for
78 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
79 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
80 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
81 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
82 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
83 * retransmission requests are sent and the regular logic is performed to
84 * schedule a lost packet as discussed above.
86 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
89 * This element will automatically be used inside rtpbin.
92 * <title>Example pipelines</title>
94 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
95 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
96 * inserted into the pipeline to smooth out network jitter and to reorder the
97 * out-of-order RTP packets.
108 #include <gst/rtp/gstrtpbuffer.h>
109 #include <gst/net/net.h>
111 #include "gstrtpjitterbuffer.h"
112 #include "rtpjitterbuffer.h"
113 #include "rtpstats.h"
115 #include <gst/glib-compat-private.h>
117 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
118 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
120 /* RTPJitterBuffer signals and args */
123 SIGNAL_REQUEST_PT_MAP,
131 #define DEFAULT_LATENCY_MS 200
132 #define DEFAULT_DROP_ON_LATENCY FALSE
133 #define DEFAULT_TS_OFFSET 0
134 #define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT 0
135 #define DEFAULT_DO_LOST FALSE
136 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
137 #define DEFAULT_PERCENT 0
138 #define DEFAULT_DO_RETRANSMISSION FALSE
139 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
140 #define DEFAULT_RTX_DELAY -1
141 #define DEFAULT_RTX_MIN_DELAY 0
142 #define DEFAULT_RTX_DELAY_REORDER 3
143 #define DEFAULT_RTX_RETRY_TIMEOUT -1
144 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
145 #define DEFAULT_RTX_RETRY_PERIOD -1
146 #define DEFAULT_RTX_MAX_RETRIES -1
147 #define DEFAULT_RTX_DEADLINE -1
148 #define DEFAULT_RTX_STATS_TIMEOUT 1000
149 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
150 #define DEFAULT_MAX_DROPOUT_TIME 60000
151 #define DEFAULT_MAX_MISORDER_TIME 2000
152 #define DEFAULT_RFC7273_SYNC FALSE
153 #define DEFAULT_FASTSTART_MIN_PACKETS 0
155 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
156 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
162 PROP_DROP_ON_LATENCY,
164 PROP_MAX_TS_OFFSET_ADJUSTMENT,
168 PROP_DO_RETRANSMISSION,
169 PROP_RTX_NEXT_SEQNUM,
172 PROP_RTX_DELAY_REORDER,
173 PROP_RTX_RETRY_TIMEOUT,
174 PROP_RTX_MIN_RETRY_TIMEOUT,
175 PROP_RTX_RETRY_PERIOD,
176 PROP_RTX_MAX_RETRIES,
178 PROP_RTX_STATS_TIMEOUT,
180 PROP_MAX_RTCP_RTP_TIME_DIFF,
181 PROP_MAX_DROPOUT_TIME,
182 PROP_MAX_MISORDER_TIME,
184 PROP_FASTSTART_MIN_PACKETS
187 #define JBUF_LOCK(priv) G_STMT_START { \
188 GST_TRACE("Locking from thread %p", g_thread_self()); \
189 (g_mutex_lock (&(priv)->jbuf_lock)); \
190 GST_TRACE("Locked from thread %p", g_thread_self()); \
193 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
195 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
198 #define JBUF_UNLOCK(priv) G_STMT_START { \
199 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
200 (g_mutex_unlock (&(priv)->jbuf_lock)); \
203 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
204 GST_DEBUG ("waiting timer"); \
205 (priv)->waiting_timer++; \
206 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
207 (priv)->waiting_timer--; \
208 GST_DEBUG ("waiting timer done"); \
210 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
211 if (G_UNLIKELY ((priv)->waiting_timer)) { \
212 GST_DEBUG ("signal timer, %d waiters", (priv)->waiting_timer); \
213 g_cond_signal (&(priv)->jbuf_timer); \
217 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
218 GST_DEBUG ("waiting event"); \
219 (priv)->waiting_event = TRUE; \
220 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
221 (priv)->waiting_event = FALSE; \
222 GST_DEBUG ("waiting event done"); \
223 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
226 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
227 if (G_UNLIKELY ((priv)->waiting_event)) { \
228 GST_DEBUG ("signal event"); \
229 g_cond_signal (&(priv)->jbuf_event); \
233 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
234 GST_DEBUG ("waiting query"); \
235 (priv)->waiting_query = TRUE; \
236 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
237 (priv)->waiting_query = FALSE; \
238 GST_DEBUG ("waiting query done"); \
239 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
242 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
243 (priv)->last_query = res; \
244 if (G_UNLIKELY ((priv)->waiting_query)) { \
245 GST_DEBUG ("signal query"); \
246 g_cond_signal (&(priv)->jbuf_query); \
250 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
251 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
253 typedef struct TimerQueue
256 GHashTable *hashtable;
259 struct _GstRtpJitterBufferPrivate
261 GstPad *sinkpad, *srcpad;
264 RTPJitterBuffer *jbuf;
266 gboolean waiting_timer;
268 gboolean waiting_event;
270 gboolean waiting_query;
278 gboolean timer_running;
279 GThread *timer_thread;
284 gboolean drop_on_latency;
286 guint64 max_ts_offset_adjustment;
288 gboolean do_retransmission;
289 gboolean rtx_next_seqnum;
292 gint rtx_delay_reorder;
293 gint rtx_retry_timeout;
294 gint rtx_min_retry_timeout;
295 gint rtx_retry_period;
296 gint rtx_max_retries;
297 guint rtx_stats_timeout;
298 gint rtx_deadline_ms;
299 gint max_rtcp_rtp_time_diff;
300 guint32 max_dropout_time;
301 guint32 max_misorder_time;
302 guint faststart_min_packets;
304 /* the last seqnum we pushed out */
305 guint32 last_popped_seqnum;
306 /* the next expected seqnum we push */
308 /* seqnum-base, if known */
310 /* last output time */
311 GstClockTime last_out_time;
312 /* last valid input timestamp and rtptime pair */
313 GstClockTime ips_pts;
315 GstClockTime packet_spacing;
320 /* the next expected seqnum we receive */
321 GstClockTime last_in_pts;
322 guint32 next_in_seqnum;
325 TimerQueue *rtx_stats_timers;
327 /* start and stop ranges */
328 GstClockTime npt_start;
329 GstClockTime npt_stop;
330 guint64 ext_timestamp;
331 guint64 last_elapsed;
332 guint64 estimated_eos;
339 /* clock rate and rtp timestamp offset */
343 gint64 ts_offset_remainder;
345 /* when we are shutting down */
346 GstFlowReturn srcresult;
352 GstClockTime timer_timeout;
353 guint16 timer_seqnum;
354 /* the latency of the upstream peer, we have to take this into account when
355 * synchronizing the buffers. */
356 GstClockTime peer_latency;
360 /* some accounting */
364 guint64 num_duplicates;
365 guint64 num_rtx_requests;
366 guint64 num_rtx_success;
367 guint64 num_rtx_failed;
370 RTPPacketRateCtx packet_rate_ctx;
373 GstClockTime last_dts;
374 GstClockTime last_pts;
375 guint64 last_rtptime;
376 GstClockTime avg_jitter;
393 GstClockTime timeout;
394 GstClockTime duration;
395 GstClockTime rtx_base;
396 GstClockTime rtx_delay;
397 GstClockTime rtx_retry;
398 GstClockTime rtx_last;
400 guint num_rtx_received;
403 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
404 GST_STATIC_PAD_TEMPLATE ("sink",
407 GST_STATIC_CAPS ("application/x-rtp"
408 /* "clock-rate = (int) [ 1, 2147483647 ], "
409 * "payload = (int) , "
410 * "encoding-name = (string) "
414 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
415 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
418 GST_STATIC_CAPS ("application/x-rtcp")
421 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
422 GST_STATIC_PAD_TEMPLATE ("src",
425 GST_STATIC_CAPS ("application/x-rtp"
426 /* "payload = (int) , "
427 * "clock-rate = (int) , "
428 * "encoding-name = (string) "
432 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
434 #define gst_rtp_jitter_buffer_parent_class parent_class
435 G_DEFINE_TYPE_WITH_PRIVATE (GstRtpJitterBuffer, gst_rtp_jitter_buffer,
438 /* object overrides */
439 static void gst_rtp_jitter_buffer_set_property (GObject * object,
440 guint prop_id, const GValue * value, GParamSpec * pspec);
441 static void gst_rtp_jitter_buffer_get_property (GObject * object,
442 guint prop_id, GValue * value, GParamSpec * pspec);
443 static void gst_rtp_jitter_buffer_finalize (GObject * object);
445 /* element overrides */
446 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
447 * element, GstStateChange transition);
448 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
449 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
450 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
452 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
453 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
457 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
458 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
461 /* sinkpad overrides */
462 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
463 GstObject * parent, GstEvent * event);
464 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
465 GstObject * parent, GstBuffer * buffer);
466 static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad,
467 GstObject * parent, GstBufferList * buffer_list);
469 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
470 GstObject * parent, GstEvent * event);
471 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
472 GstObject * parent, GstBuffer * buffer);
474 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
475 GstObject * parent, GstQuery * query);
477 /* srcpad overrides */
478 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
479 GstObject * parent, GstEvent * event);
480 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
481 GstObject * parent, GstPadMode mode, gboolean active);
482 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
483 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
484 GstObject * parent, GstQuery * query);
487 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
489 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
490 gboolean active, guint64 base_time);
491 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
493 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
494 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
496 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
498 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
501 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
502 TimerData * timer, GstClockTime dts, gboolean success);
504 static TimerQueue *timer_queue_new (void);
505 static void timer_queue_free (TimerQueue * queue);
508 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
510 GObjectClass *gobject_class;
511 GstElementClass *gstelement_class;
513 gobject_class = (GObjectClass *) klass;
514 gstelement_class = (GstElementClass *) klass;
516 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
518 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
519 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
522 * GstRtpJitterBuffer:latency:
524 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
525 * for at most this time.
527 g_object_class_install_property (gobject_class, PROP_LATENCY,
528 g_param_spec_uint ("latency", "Buffer latency in ms",
529 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
530 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
532 * GstRtpJitterBuffer:drop-on-latency:
534 * Drop oldest buffers when the queue is completely filled.
536 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
537 g_param_spec_boolean ("drop-on-latency",
538 "Drop buffers when maximum latency is reached",
539 "Tells the jitterbuffer to never exceed the given latency in size",
540 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
542 * GstRtpJitterBuffer:ts-offset:
544 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
545 * This is mainly used to ensure interstream synchronisation.
547 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
548 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
549 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
550 G_MAXINT64, DEFAULT_TS_OFFSET,
551 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
554 * GstRtpJitterBuffer:max-ts-offset-adjustment:
556 * The maximum number of nanoseconds per frame that time offset may be
557 * adjusted with. This is used to avoid sudden large changes to time stamps.
559 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
560 g_param_spec_uint64 ("max-ts-offset-adjustment",
561 "Max Timestamp Offset Adjustment",
562 "The maximum number of nanoseconds per frame that time stamp "
563 "offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64,
564 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
565 G_PARAM_STATIC_STRINGS));
568 * GstRtpJitterBuffer:do-lost:
570 * Send out a GstRTPPacketLost event downstream when a packet is considered
573 g_object_class_install_property (gobject_class, PROP_DO_LOST,
574 g_param_spec_boolean ("do-lost", "Do Lost",
575 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
576 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
579 * GstRtpJitterBuffer:mode:
581 * Control the buffering and timestamping mode used by the jitterbuffer.
583 g_object_class_install_property (gobject_class, PROP_MODE,
584 g_param_spec_enum ("mode", "Mode",
585 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
586 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
588 * GstRtpJitterBuffer:percent:
590 * The percent of the jitterbuffer that is filled.
592 g_object_class_install_property (gobject_class, PROP_PERCENT,
593 g_param_spec_int ("percent", "percent",
594 "The buffer filled percent", 0, 100,
595 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
597 * GstRtpJitterBuffer:do-retransmission:
599 * Send out a GstRTPRetransmission event upstream when a packet is considered
600 * late and should be retransmitted.
604 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
605 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
606 "Send retransmission events upstream when a packet is late",
607 DEFAULT_DO_RETRANSMISSION,
608 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRtpJitterBuffer:rtx-next-seqnum
613 * Estimate when the next packet should arrive and schedule a retransmission
615 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
616 * for packet N+1. So it will be requested if it does not arrive at the expected time.
617 * The expected time is calculated using the dts of N and the packet spacing.
621 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
622 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
623 "Estimate when the next packet should arrive and schedule a "
624 "retransmission request for it.",
625 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
628 * GstRtpJitterBuffer:rtx-delay:
630 * When a packet did not arrive at the expected time, wait this extra amount
631 * of time before sending a retransmission event.
633 * When -1 is used, the max jitter will be used as extra delay.
637 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
638 g_param_spec_int ("rtx-delay", "RTX Delay",
639 "Extra time in ms to wait before sending retransmission "
640 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
641 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
644 * GstRtpJitterBuffer:rtx-min-delay:
646 * When a packet did not arrive at the expected time, wait at least this extra amount
647 * of time before sending a retransmission event.
651 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
652 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
653 "Minimum time in ms to wait before sending retransmission "
654 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
655 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
657 * GstRtpJitterBuffer:rtx-delay-reorder:
659 * Assume that a retransmission event should be sent when we see
660 * this much packet reordering.
662 * When -1 is used, the value will be estimated based on observed packet
663 * reordering. When 0 is used packet reordering alone will not cause a
664 * retransmission event (Since 1.10).
668 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
669 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
670 "Sending retransmission event when this much reordering "
671 "(0 disable, -1 automatic)",
672 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
673 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
675 * GstRtpJitterBuffer::rtx-retry-timeout:
677 * When no packet has been received after sending a retransmission event
678 * for this time, retry sending a retransmission event.
680 * When -1 is used, the value will be estimated based on observed round
685 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
686 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
687 "Retry sending a transmission event after this timeout in "
688 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
689 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
691 * GstRtpJitterBuffer::rtx-min-retry-timeout:
693 * The minimum amount of time between retry timeouts. When
694 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
695 * minimum interval between retry timeouts.
697 * When -1 is used, the value will be estimated based on the
702 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
703 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
704 "Minimum timeout between sending a transmission event in "
705 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
706 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
708 * GstRtpJitterBuffer:rtx-retry-period:
710 * The amount of time to try to get a retransmission.
712 * When -1 is used, the value will be estimated based on the jitterbuffer
713 * latency and the observed round trip time.
717 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
718 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
719 "Try to get a retransmission for this many ms "
720 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
721 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
723 * GstRtpJitterBuffer:rtx-max-retries:
725 * The maximum number of retries to request a retransmission.
727 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
728 * When -1 is used, the number of retransmission request will not be limited.
732 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
733 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
734 "The maximum number of retries to request a retransmission. "
735 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
736 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
738 * GstRtpJitterBuffer:rtx-deadline:
740 * The deadline for a valid RTX request in ms.
742 * How long the RTX RTCP will be valid for.
743 * When -1 is used, the size of the jitterbuffer will be used.
747 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
748 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
749 "The deadline for a valid RTX request in milliseconds. "
750 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
751 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
753 * GstRtpJitterBuffer::rtx-stats-timeout:
755 * The time to wait for a retransmitted packet after it has been
756 * considered lost in order to collect RTX statistics.
760 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
761 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
762 "The time to wait for a retransmitted packet after it has been "
763 "considered lost in order to collect statistics (ms)",
764 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
765 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
767 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
768 g_param_spec_uint ("max-dropout-time", "Max dropout time",
769 "The maximum time (milliseconds) of missing packets tolerated.",
770 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
771 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
773 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
774 g_param_spec_uint ("max-misorder-time", "Max misorder time",
775 "The maximum time (milliseconds) of misordered packets tolerated.",
776 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
777 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
779 * GstRtpJitterBuffer:stats:
781 * Various jitterbuffer statistics. This property returns a GstStructure
782 * with name application/x-rtp-jitterbuffer-stats with the following fields:
788 * <classname>"num-pushed"</classname>:
789 * the number of packets pushed out.
795 * <classname>"num-lost"</classname>:
796 * the number of packets considered lost.
802 * <classname>"num-late"</classname>:
803 * the number of packets arriving too late.
809 * <classname>"num-duplicates"</classname>:
810 * the number of duplicate packets.
816 * <classname>"rtx-count"</classname>:
817 * the number of retransmissions requested.
823 * <classname>"rtx-success-count"</classname>:
824 * the number of successful retransmissions.
830 * <classname>"rtx-per-packet"</classname>:
831 * average number of RTX per packet.
837 * <classname>"rtx-rtt"</classname>:
838 * average round trip time per RTX.
845 g_object_class_install_property (gobject_class, PROP_STATS,
846 g_param_spec_boxed ("stats", "Statistics",
847 "Various statistics", GST_TYPE_STRUCTURE,
848 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
851 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
853 * The maximum amount of time in ms that the RTP time in the RTCP SRs
854 * is allowed to be ahead of the last RTP packet we received. Use
855 * -1 to disable ignoring of RTCP packets.
859 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
860 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
861 "Maximum amount of time in ms that the RTP time in RTCP SRs "
862 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
863 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
864 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
866 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
867 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
868 "Synchronize received streams to the RFC7273 clock "
869 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
870 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
873 * GstRtpJitterBuffer:faststart-min-packets
875 * The number of consecutive packets needed to start (set to 0 to
876 * disable faststart. The jitterbuffer will by default start after the
877 * latency has elapsed)
881 g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
882 g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
883 "The number of consecutive packets needed to start (set to 0 to "
884 "disable faststart. The jitterbuffer will by default start after "
885 "the latency has elapsed)",
886 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
887 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
890 * GstRtpJitterBuffer::request-pt-map:
891 * @buffer: the object which received the signal
894 * Request the payload type as #GstCaps for @pt.
896 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
897 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
898 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
899 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
900 GST_TYPE_CAPS, 1, G_TYPE_UINT);
902 * GstRtpJitterBuffer::handle-sync:
903 * @buffer: the object which received the signal
904 * @struct: a GstStructure containing sync values.
906 * Be notified of new sync values.
908 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
909 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
910 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
911 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
912 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
915 * GstRtpJitterBuffer::on-npt-stop:
916 * @buffer: the object which received the signal
918 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
919 * the npt-stop position.
921 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
922 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
923 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
924 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
925 G_TYPE_NONE, 0, G_TYPE_NONE);
928 * GstRtpJitterBuffer::clear-pt-map:
929 * @buffer: the object which received the signal
931 * Invalidate the clock-rate as obtained with the
932 * #GstRtpJitterBuffer::request-pt-map signal.
934 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
935 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
936 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
937 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
938 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
941 * GstRtpJitterBuffer::set-active:
942 * @buffer: the object which received the signal
944 * Start pushing out packets with the given base time. This signal is only
945 * useful in buffering mode.
947 * Returns: the time of the last pushed packet.
949 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
950 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
951 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
952 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
953 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
956 gstelement_class->change_state =
957 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
958 gstelement_class->request_new_pad =
959 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
960 gstelement_class->release_pad =
961 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
962 gstelement_class->provide_clock =
963 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
964 gstelement_class->set_clock =
965 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
967 gst_element_class_add_static_pad_template (gstelement_class,
968 &gst_rtp_jitter_buffer_src_template);
969 gst_element_class_add_static_pad_template (gstelement_class,
970 &gst_rtp_jitter_buffer_sink_template);
971 gst_element_class_add_static_pad_template (gstelement_class,
972 &gst_rtp_jitter_buffer_sink_rtcp_template);
974 gst_element_class_set_static_metadata (gstelement_class,
975 "RTP packet jitter-buffer", "Filter/Network/RTP",
976 "A buffer that deals with network jitter and other transmission faults",
977 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
978 "Wim Taymans <wim.taymans@gmail.com>");
980 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
981 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
983 GST_DEBUG_CATEGORY_INIT
984 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
988 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
990 GstRtpJitterBufferPrivate *priv;
992 priv = gst_rtp_jitter_buffer_get_instance_private (jitterbuffer);
993 jitterbuffer->priv = priv;
995 priv->latency_ms = DEFAULT_LATENCY_MS;
996 priv->latency_ns = priv->latency_ms * GST_MSECOND;
997 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
998 priv->ts_offset = DEFAULT_TS_OFFSET;
999 priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
1000 priv->do_lost = DEFAULT_DO_LOST;
1001 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
1002 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
1003 priv->rtx_delay = DEFAULT_RTX_DELAY;
1004 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
1005 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
1006 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
1007 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
1008 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
1009 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
1010 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
1011 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
1012 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
1013 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
1014 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
1015 priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
1017 priv->ts_offset_remainder = 0;
1018 priv->last_dts = -1;
1019 priv->last_pts = -1;
1020 priv->last_rtptime = -1;
1021 priv->avg_jitter = 0;
1022 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
1023 priv->rtx_stats_timers = timer_queue_new ();
1024 priv->jbuf = rtp_jitter_buffer_new ();
1025 g_mutex_init (&priv->jbuf_lock);
1026 g_cond_init (&priv->jbuf_timer);
1027 g_cond_init (&priv->jbuf_event);
1028 g_cond_init (&priv->jbuf_query);
1029 g_queue_init (&priv->gap_packets);
1030 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1032 /* reset skew detection initialy */
1033 rtp_jitter_buffer_reset_skew (priv->jbuf);
1034 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
1035 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1036 priv->active = TRUE;
1039 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1042 gst_pad_set_activatemode_function (priv->srcpad,
1043 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1044 gst_pad_set_query_function (priv->srcpad,
1045 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1046 gst_pad_set_event_function (priv->srcpad,
1047 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1050 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1053 gst_pad_set_chain_function (priv->sinkpad,
1054 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1055 gst_pad_set_chain_list_function (priv->sinkpad,
1056 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list));
1057 gst_pad_set_event_function (priv->sinkpad,
1058 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1059 gst_pad_set_query_function (priv->sinkpad,
1060 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1062 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1063 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1065 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1068 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1070 #define ITEM_TYPE_BUFFER 0
1071 #define ITEM_TYPE_LOST 1
1072 #define ITEM_TYPE_EVENT 2
1073 #define ITEM_TYPE_QUERY 3
1075 static RTPJitterBufferItem *
1076 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1077 guint seqnum, guint count, guint rtptime)
1079 RTPJitterBufferItem *item;
1081 item = g_slice_new (RTPJitterBufferItem);
1088 item->seqnum = seqnum;
1089 item->count = count;
1090 item->rtptime = rtptime;
1096 free_item (RTPJitterBufferItem * item)
1098 g_return_if_fail (item != NULL);
1100 if (item->data && item->type != ITEM_TYPE_QUERY)
1101 gst_mini_object_unref (item->data);
1102 g_slice_free (RTPJitterBufferItem, item);
1106 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1108 GList **l = user_data;
1110 if (item->data && item->type == ITEM_TYPE_EVENT
1111 && GST_EVENT_IS_STICKY (item->data)) {
1112 *l = g_list_prepend (*l, item->data);
1113 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1114 gst_mini_object_unref (item->data);
1116 g_slice_free (RTPJitterBufferItem, item);
1120 gst_rtp_jitter_buffer_finalize (GObject * object)
1122 GstRtpJitterBuffer *jitterbuffer;
1123 GstRtpJitterBufferPrivate *priv;
1125 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1126 priv = jitterbuffer->priv;
1128 g_array_free (priv->timers, TRUE);
1129 timer_queue_free (priv->rtx_stats_timers);
1130 g_mutex_clear (&priv->jbuf_lock);
1131 g_cond_clear (&priv->jbuf_timer);
1132 g_cond_clear (&priv->jbuf_event);
1133 g_cond_clear (&priv->jbuf_query);
1135 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1136 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1137 g_queue_clear (&priv->gap_packets);
1138 g_object_unref (priv->jbuf);
1140 G_OBJECT_CLASS (parent_class)->finalize (object);
1143 static GstIterator *
1144 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1146 GstRtpJitterBuffer *jitterbuffer;
1147 GstPad *otherpad = NULL;
1148 GstIterator *it = NULL;
1149 GValue val = { 0, };
1151 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1153 if (pad == jitterbuffer->priv->sinkpad) {
1154 otherpad = jitterbuffer->priv->srcpad;
1155 } else if (pad == jitterbuffer->priv->srcpad) {
1156 otherpad = jitterbuffer->priv->sinkpad;
1157 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1158 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1162 g_value_init (&val, GST_TYPE_PAD);
1163 g_value_set_object (&val, otherpad);
1164 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1165 g_value_unset (&val);
1172 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1174 GstRtpJitterBufferPrivate *priv;
1176 priv = jitterbuffer->priv;
1178 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1181 gst_pad_new_from_static_template
1182 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1183 gst_pad_set_chain_function (priv->rtcpsinkpad,
1184 gst_rtp_jitter_buffer_chain_rtcp);
1185 gst_pad_set_event_function (priv->rtcpsinkpad,
1186 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1187 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1188 gst_rtp_jitter_buffer_iterate_internal_links);
1189 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1190 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1192 return priv->rtcpsinkpad;
1196 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1198 GstRtpJitterBufferPrivate *priv;
1200 priv = jitterbuffer->priv;
1202 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1204 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1206 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1207 priv->rtcpsinkpad = NULL;
1211 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1212 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1214 GstRtpJitterBuffer *jitterbuffer;
1215 GstElementClass *klass;
1217 GstRtpJitterBufferPrivate *priv;
1219 g_return_val_if_fail (templ != NULL, NULL);
1220 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1222 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1223 priv = jitterbuffer->priv;
1224 klass = GST_ELEMENT_GET_CLASS (element);
1226 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1228 /* figure out the template */
1229 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1230 if (priv->rtcpsinkpad != NULL)
1233 result = create_rtcp_sink (jitterbuffer);
1235 goto wrong_template;
1242 g_warning ("rtpjitterbuffer: this is not our template");
1247 g_warning ("rtpjitterbuffer: pad already requested");
1253 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1255 GstRtpJitterBuffer *jitterbuffer;
1256 GstRtpJitterBufferPrivate *priv;
1258 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1259 g_return_if_fail (GST_IS_PAD (pad));
1261 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1262 priv = jitterbuffer->priv;
1264 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1266 if (priv->rtcpsinkpad == pad) {
1267 remove_rtcp_sink (jitterbuffer);
1276 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1282 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1284 return gst_system_clock_obtain ();
1288 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1290 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1292 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1294 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1298 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1300 GstRtpJitterBufferPrivate *priv;
1302 priv = jitterbuffer->priv;
1304 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1307 priv->clock_rate = -1;
1308 /* do not clear current content, but refresh state for new arrival */
1309 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1310 rtp_jitter_buffer_reset_skew (priv->jbuf);
1315 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1318 GstRtpJitterBufferPrivate *priv;
1319 GstClockTime last_out;
1320 RTPJitterBufferItem *item;
1325 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1326 active, GST_TIME_ARGS (offset));
1328 if (active != priv->active) {
1329 /* add the amount of time spent in paused to the output offset. All
1330 * outgoing buffers will have this offset applied to their timestamps in
1331 * order to make them arrive in time in the sink. */
1332 priv->out_offset = offset;
1333 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1334 GST_TIME_ARGS (priv->out_offset));
1335 priv->active = active;
1336 JBUF_SIGNAL_EVENT (priv);
1339 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1341 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1342 /* head buffer timestamp and offset gives our output time */
1343 last_out = item->pts + priv->ts_offset;
1345 /* use last known time when the buffer is empty */
1346 last_out = priv->last_out_time;
1354 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1356 GstRtpJitterBuffer *jitterbuffer;
1357 GstRtpJitterBufferPrivate *priv;
1362 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1363 priv = jitterbuffer->priv;
1365 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1367 caps = gst_pad_peer_query_caps (other, filter);
1369 templ = gst_pad_get_pad_template_caps (pad);
1371 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1376 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1378 intersect = gst_caps_intersect (caps, templ);
1379 gst_caps_unref (caps);
1380 gst_caps_unref (templ);
1384 gst_object_unref (jitterbuffer);
1390 * Must be called with JBUF_LOCK held
1394 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1395 GstCaps * caps, gint pt)
1397 GstRtpJitterBufferPrivate *priv;
1398 GstStructure *caps_struct;
1402 const gchar *ts_refclk, *mediaclk;
1404 priv = jitterbuffer->priv;
1406 /* first parse the caps */
1407 caps_struct = gst_caps_get_structure (caps, 0);
1409 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1411 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1413 GST_ERROR_OBJECT (jitterbuffer,
1414 "Got caps with wrong payload type (got %d, expected %d)", pt, payload);
1418 if (payload != -1) {
1419 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1420 priv->last_pt = payload;
1423 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1424 * measure the amount of data in the buffer */
1425 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1428 if (priv->clock_rate <= 0)
1431 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1433 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1435 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1437 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1438 * can use this to track the amount of time elapsed on the sender. */
1439 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1440 priv->clock_base = val;
1442 priv->clock_base = -1;
1444 priv->ext_timestamp = priv->clock_base;
1446 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1449 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1450 /* first expected seqnum, only update when we didn't have a previous base. */
1451 if (priv->next_in_seqnum == -1)
1452 priv->next_in_seqnum = val;
1453 if (priv->next_seqnum == -1) {
1454 priv->next_seqnum = val;
1455 JBUF_SIGNAL_EVENT (priv);
1457 priv->seqnum_base = val;
1459 priv->seqnum_base = -1;
1462 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1464 /* the start and stop times. The seqnum-base corresponds to the start time. We
1465 * will keep track of the seqnums on the output and when we reach the one
1466 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1467 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1468 priv->npt_start = tval;
1470 priv->npt_start = 0;
1472 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1473 priv->npt_stop = tval;
1475 priv->npt_stop = -1;
1477 GST_DEBUG_OBJECT (jitterbuffer,
1478 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1479 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1481 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1482 GstClock *clock = NULL;
1483 guint64 clock_offset = -1;
1485 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1488 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1489 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1490 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1492 const gchar *host, *portstr;
1496 host = ts_refclk + sizeof ("ntp=") - 1;
1497 if (host[0] == '[') {
1499 portstr = strchr (host, ']');
1500 if (portstr && portstr[1] == ':')
1501 portstr = portstr + 1;
1505 portstr = strrchr (host, ':');
1509 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1513 hostname = g_strndup (host, (portstr - host));
1515 hostname = g_strdup (host);
1517 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1520 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1521 const gchar *domainstr =
1522 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1525 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1528 clock = gst_ptp_clock_new (NULL, domain);
1530 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1533 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1534 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1536 if (!g_str_has_prefix (mediaclk, "direct=")
1537 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1538 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1539 if (strstr (mediaclk, "rate=") != NULL) {
1540 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1545 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1547 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1555 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1560 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1566 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1568 GstRtpJitterBufferPrivate *priv;
1570 priv = jitterbuffer->priv;
1573 /* mark ourselves as flushing */
1574 priv->srcresult = GST_FLOW_FLUSHING;
1575 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1576 /* this unblocks any waiting pops on the src pad task */
1577 JBUF_SIGNAL_EVENT (priv);
1578 JBUF_SIGNAL_QUERY (priv, FALSE);
1583 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1585 GstRtpJitterBufferPrivate *priv;
1587 priv = jitterbuffer->priv;
1590 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1591 /* Mark as non flushing */
1592 priv->srcresult = GST_FLOW_OK;
1593 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1594 priv->last_popped_seqnum = -1;
1595 priv->last_out_time = GST_CLOCK_TIME_NONE;
1596 priv->next_seqnum = -1;
1597 priv->seqnum_base = -1;
1598 priv->ips_rtptime = -1;
1599 priv->ips_pts = GST_CLOCK_TIME_NONE;
1600 priv->packet_spacing = 0;
1601 priv->next_in_seqnum = -1;
1602 priv->clock_rate = -1;
1605 priv->estimated_eos = -1;
1606 priv->last_elapsed = 0;
1607 priv->ext_timestamp = -1;
1608 priv->avg_jitter = 0;
1609 priv->last_dts = -1;
1610 priv->last_rtptime = -1;
1611 priv->last_in_pts = 0;
1612 priv->equidistant = 0;
1613 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1614 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1615 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1616 rtp_jitter_buffer_reset_skew (priv->jbuf);
1617 remove_all_timers (jitterbuffer);
1618 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1619 g_queue_clear (&priv->gap_packets);
1624 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1625 GstPadMode mode, gboolean active)
1628 GstRtpJitterBuffer *jitterbuffer = NULL;
1630 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1633 case GST_PAD_MODE_PUSH:
1635 /* allow data processing */
1636 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1638 /* start pushing out buffers */
1639 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1640 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1641 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1643 /* make sure all data processing stops ASAP */
1644 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1646 /* NOTE this will hardlock if the state change is called from the src pad
1647 * task thread because we will _join() the thread. */
1648 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1649 result = gst_pad_stop_task (pad);
1659 static GstStateChangeReturn
1660 gst_rtp_jitter_buffer_change_state (GstElement * element,
1661 GstStateChange transition)
1663 GstRtpJitterBuffer *jitterbuffer;
1664 GstRtpJitterBufferPrivate *priv;
1665 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1667 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1668 priv = jitterbuffer->priv;
1670 switch (transition) {
1671 case GST_STATE_CHANGE_NULL_TO_READY:
1673 case GST_STATE_CHANGE_READY_TO_PAUSED:
1675 /* reset negotiated values */
1676 priv->clock_rate = -1;
1677 priv->clock_base = -1;
1678 priv->peer_latency = 0;
1680 /* block until we go to PLAYING */
1681 priv->blocked = TRUE;
1682 priv->timer_running = TRUE;
1683 priv->timer_thread =
1684 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1687 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1689 /* unblock to allow streaming in PLAYING */
1690 priv->blocked = FALSE;
1691 JBUF_SIGNAL_EVENT (priv);
1692 JBUF_SIGNAL_TIMER (priv);
1699 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1701 switch (transition) {
1702 case GST_STATE_CHANGE_READY_TO_PAUSED:
1703 /* we are a live element because we sync to the clock, which we can only
1704 * do in the PLAYING state */
1705 if (ret != GST_STATE_CHANGE_FAILURE)
1706 ret = GST_STATE_CHANGE_NO_PREROLL;
1708 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1710 /* block to stop streaming when PAUSED */
1711 priv->blocked = TRUE;
1712 unschedule_current_timer (jitterbuffer);
1714 if (ret != GST_STATE_CHANGE_FAILURE)
1715 ret = GST_STATE_CHANGE_NO_PREROLL;
1717 case GST_STATE_CHANGE_PAUSED_TO_READY:
1719 gst_buffer_replace (&priv->last_sr, NULL);
1720 priv->timer_running = FALSE;
1721 unschedule_current_timer (jitterbuffer);
1722 JBUF_SIGNAL_TIMER (priv);
1723 JBUF_SIGNAL_QUERY (priv, FALSE);
1725 g_thread_join (priv->timer_thread);
1726 priv->timer_thread = NULL;
1728 case GST_STATE_CHANGE_READY_TO_NULL:
1738 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1741 gboolean ret = TRUE;
1742 GstRtpJitterBuffer *jitterbuffer;
1743 GstRtpJitterBufferPrivate *priv;
1745 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1746 priv = jitterbuffer->priv;
1748 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1750 switch (GST_EVENT_TYPE (event)) {
1751 case GST_EVENT_LATENCY:
1753 GstClockTime latency;
1755 gst_event_parse_latency (event, &latency);
1757 GST_DEBUG_OBJECT (jitterbuffer,
1758 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1761 /* adjust the overall buffer delay to the total pipeline latency in
1762 * buffering mode because if downstream consumes too fast (because of
1763 * large latency or queues, we would start rebuffering again. */
1764 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1765 RTP_JITTER_BUFFER_MODE_BUFFER) {
1766 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1770 ret = gst_pad_push_event (priv->sinkpad, event);
1774 ret = gst_pad_push_event (priv->sinkpad, event);
1781 /* handles and stores the event in the jitterbuffer, must be called with
1784 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1786 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1787 RTPJitterBufferItem *item;
1790 switch (GST_EVENT_TYPE (event)) {
1791 case GST_EVENT_CAPS:
1795 gst_event_parse_caps (event, &caps);
1796 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1799 case GST_EVENT_SEGMENT:
1802 gst_event_copy_segment (event, &segment);
1804 /* we need time for now */
1805 if (segment.format != GST_FORMAT_TIME) {
1806 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1807 gst_event_unref (event);
1809 gst_segment_init (&segment, GST_FORMAT_TIME);
1810 event = gst_event_new_segment (&segment);
1813 priv->segment = segment;
1818 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1825 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1826 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1827 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1828 if (head || priv->eos)
1829 JBUF_SIGNAL_EVENT (priv);
1835 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1838 gboolean ret = TRUE;
1839 GstRtpJitterBuffer *jitterbuffer;
1840 GstRtpJitterBufferPrivate *priv;
1842 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1843 priv = jitterbuffer->priv;
1845 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1847 switch (GST_EVENT_TYPE (event)) {
1848 case GST_EVENT_FLUSH_START:
1849 ret = gst_pad_push_event (priv->srcpad, event);
1850 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1851 /* wait for the loop to go into PAUSED */
1852 gst_pad_pause_task (priv->srcpad);
1854 case GST_EVENT_FLUSH_STOP:
1855 ret = gst_pad_push_event (priv->srcpad, event);
1857 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1858 GST_PAD_MODE_PUSH, TRUE);
1861 if (GST_EVENT_IS_SERIALIZED (event)) {
1862 /* serialized events go in the queue */
1864 if (priv->srcresult != GST_FLOW_OK) {
1865 /* Errors in sticky event pushing are no problem and ignored here
1866 * as they will cause more meaningful errors during data flow.
1867 * For EOS events, that are not followed by data flow, we still
1868 * return FALSE here though.
1870 if (!GST_EVENT_IS_STICKY (event) ||
1871 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1872 goto out_flow_error;
1874 /* refuse more events on EOS */
1877 ret = queue_event (jitterbuffer, event);
1880 /* non-serialized events are forwarded downstream immediately */
1881 ret = gst_pad_push_event (priv->srcpad, event);
1890 GST_DEBUG_OBJECT (jitterbuffer,
1891 "refusing event, we have a downstream flow error: %s",
1892 gst_flow_get_name (priv->srcresult));
1894 gst_event_unref (event);
1899 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1901 gst_event_unref (event);
1907 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1910 gboolean ret = TRUE;
1911 GstRtpJitterBuffer *jitterbuffer;
1913 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1915 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1917 switch (GST_EVENT_TYPE (event)) {
1918 case GST_EVENT_FLUSH_START:
1919 gst_event_unref (event);
1921 case GST_EVENT_FLUSH_STOP:
1922 gst_event_unref (event);
1925 ret = gst_pad_event_default (pad, parent, event);
1933 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1934 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1935 * GST_FLOW_FLUSHING when the element is shutting down. On success
1936 * GST_FLOW_OK is returned.
1938 static GstFlowReturn
1939 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1943 GValue args[2] = { {0}, {0} };
1947 g_value_init (&args[0], GST_TYPE_ELEMENT);
1948 g_value_set_object (&args[0], jitterbuffer);
1949 g_value_init (&args[1], G_TYPE_UINT);
1950 g_value_set_uint (&args[1], pt);
1952 g_value_init (&ret, GST_TYPE_CAPS);
1953 g_value_set_boxed (&ret, NULL);
1955 JBUF_UNLOCK (jitterbuffer->priv);
1956 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1958 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1960 g_value_unset (&args[0]);
1961 g_value_unset (&args[1]);
1962 caps = (GstCaps *) g_value_dup_boxed (&ret);
1963 g_value_unset (&ret);
1967 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1968 gst_caps_unref (caps);
1970 if (G_UNLIKELY (!res))
1978 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1979 return GST_FLOW_ERROR;
1983 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1984 return GST_FLOW_FLUSHING;
1988 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1989 return GST_FLOW_ERROR;
1993 /* call with jbuf lock held */
1995 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1997 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1998 GstMessage *message = NULL;
2003 /* Post a buffering message */
2004 if (priv->last_percent != percent) {
2005 priv->last_percent = percent;
2007 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
2008 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
2015 update_offset (GstRtpJitterBuffer * jitterbuffer)
2017 GstRtpJitterBufferPrivate *priv;
2019 priv = jitterbuffer->priv;
2021 if (priv->ts_offset_remainder != 0) {
2022 GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT
2023 " off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment,
2024 priv->ts_offset_remainder, priv->ts_offset);
2025 if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) {
2026 if (priv->ts_offset_remainder > 0) {
2027 priv->ts_offset += priv->max_ts_offset_adjustment;
2028 priv->ts_offset_remainder -= priv->max_ts_offset_adjustment;
2030 priv->ts_offset -= priv->max_ts_offset_adjustment;
2031 priv->ts_offset_remainder += priv->max_ts_offset_adjustment;
2034 priv->ts_offset += priv->ts_offset_remainder;
2035 priv->ts_offset_remainder = 0;
2041 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
2043 GstRtpJitterBufferPrivate *priv;
2045 priv = jitterbuffer->priv;
2047 if (timestamp == -1)
2050 /* apply the timestamp offset, this is used for inter stream sync */
2051 timestamp += priv->ts_offset;
2052 /* add the offset, this is used when buffering */
2053 timestamp += priv->out_offset;
2059 timer_queue_new (void)
2063 queue = g_slice_new (TimerQueue);
2064 queue->timers = g_queue_new ();
2065 queue->hashtable = g_hash_table_new (NULL, NULL);
2071 timer_queue_free (TimerQueue * queue)
2076 g_hash_table_destroy (queue->hashtable);
2077 g_queue_free_full (queue->timers, g_free);
2078 g_slice_free (TimerQueue, queue);
2082 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2083 GstClockTime timeout, gboolean lost)
2087 copy = g_memdup (timer, sizeof (*timer));
2088 copy->timeout = timeout;
2089 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2092 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2093 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2094 g_queue_push_tail (queue->timers, copy);
2095 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2099 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2103 test = g_queue_peek_head (queue->timers);
2104 while (test && test->timeout < timeout) {
2105 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2106 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2107 GST_TIME_ARGS (timeout));
2108 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2109 g_free (g_queue_pop_head (queue->timers));
2110 test = g_queue_peek_head (queue->timers);
2115 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2117 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2121 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2123 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2124 TimerData *timer = NULL;
2127 len = priv->timers->len;
2128 for (i = 0; i < len; i++) {
2129 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2130 if (test->seqnum == seqnum) {
2139 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2141 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2143 if (priv->clock_id) {
2144 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2145 gst_clock_id_unschedule (priv->clock_id);
2146 priv->clock_id = NULL;
2151 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2153 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2154 GstClockTime test_timeout;
2156 if ((test_timeout = timer->timeout) == -1)
2159 if (timer->type != TIMER_TYPE_EXPECTED) {
2160 /* add our latency and offset to get output times. */
2161 test_timeout = apply_offset (jitterbuffer, test_timeout);
2162 test_timeout += priv->latency_ns;
2164 return test_timeout;
2168 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2170 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2172 if (priv->clock_id) {
2173 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2175 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2176 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2178 if (timeout == -1 || timeout < priv->timer_timeout)
2179 unschedule_current_timer (jitterbuffer);
2184 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2185 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2186 GstClockTime duration)
2188 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2192 GST_DEBUG_OBJECT (jitterbuffer,
2193 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2194 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2195 GST_TIME_ARGS (delay));
2197 len = priv->timers->len;
2198 g_array_set_size (priv->timers, len + 1);
2199 timer = &g_array_index (priv->timers, TimerData, len);
2202 timer->seqnum = seqnum;
2204 timer->timeout = timeout + delay;
2205 timer->duration = duration;
2206 if (type == TIMER_TYPE_EXPECTED) {
2207 timer->rtx_base = timeout;
2208 timer->rtx_delay = delay;
2209 timer->rtx_retry = 0;
2211 timer->rtx_last = GST_CLOCK_TIME_NONE;
2212 timer->num_rtx_retry = 0;
2213 timer->num_rtx_received = 0;
2214 recalculate_timer (jitterbuffer, timer);
2215 JBUF_SIGNAL_TIMER (priv);
2221 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2222 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2224 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2225 gboolean seqchange, timechange;
2227 GstClockTime new_timeout;
2229 oldseq = timer->seqnum;
2230 new_timeout = timeout + delay;
2231 seqchange = oldseq != seqnum;
2232 timechange = timer->timeout != new_timeout;
2234 if (!seqchange && !timechange) {
2235 GST_DEBUG_OBJECT (jitterbuffer,
2236 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2237 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2241 GST_DEBUG_OBJECT (jitterbuffer,
2242 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2243 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2244 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2246 timer->timeout = new_timeout;
2247 timer->seqnum = seqnum;
2249 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2250 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2251 GST_TIME_ARGS (delay));
2252 timer->rtx_base = timeout;
2253 timer->rtx_delay = delay;
2254 timer->rtx_retry = 0;
2257 timer->num_rtx_retry = 0;
2258 timer->num_rtx_received = 0;
2261 if (priv->clock_id) {
2262 /* we changed the seqnum and there is a timer currently waiting with this
2263 * seqnum, unschedule it */
2264 if (seqchange && priv->timer_seqnum == oldseq)
2265 unschedule_current_timer (jitterbuffer);
2266 /* we changed the time, check if it is earlier than what we are waiting
2267 * for and unschedule if so */
2268 else if (timechange)
2269 recalculate_timer (jitterbuffer, timer);
2274 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2275 guint16 seqnum, GstClockTime timeout)
2279 /* find the seqnum timer */
2280 timer = find_timer (jitterbuffer, seqnum);
2281 if (timer == NULL) {
2282 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2284 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2290 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2292 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2295 if (timer->idx == -1)
2298 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2299 unschedule_current_timer (jitterbuffer);
2302 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2303 g_array_remove_index_fast (priv->timers, idx);
2306 JBUF_SIGNAL_TIMER (priv);
2310 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2312 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2313 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2314 g_array_set_size (priv->timers, 0);
2315 unschedule_current_timer (jitterbuffer);
2316 JBUF_SIGNAL_TIMER (priv);
2319 /* get the extra delay to wait before sending RTX */
2321 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2325 if (priv->rtx_delay == -1) {
2326 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2327 delay = DEFAULT_AUTO_RTX_DELAY;
2329 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2330 * packet spacing is a good margin */
2331 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2334 delay = priv->rtx_delay * GST_MSECOND;
2336 if (priv->rtx_min_delay > 0)
2337 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2342 /* Check if packet with seqnum is already considered definitely lost by being
2343 * part of a "lost timer" for multiple packets */
2345 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2347 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2350 len = priv->timers->len;
2351 for (i = 0; i < len; i++) {
2352 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2353 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2355 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2357 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2358 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2366 /* we just received a packet with seqnum and dts.
2368 * First check for old seqnum that we are still expecting. If the gap with the
2369 * current seqnum is too big, unschedule the timeouts.
2371 * If we have a valid packet spacing estimate we can set a timer for when we
2372 * should receive the next packet.
2373 * If we don't have a valid estimate, we remove any timer we might have
2374 * had for this packet.
2377 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2378 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2379 gboolean is_rtx, TimerData * timer)
2381 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2383 /* go through all timers and unschedule the ones with a large gap */
2384 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2386 len = priv->timers->len;
2387 for (i = 0; i < len; i++) {
2388 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2391 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2393 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2394 test->type, test->seqnum, seqnum, gap);
2396 if (gap > priv->rtx_delay_reorder) {
2397 /* max gap, we exceeded the max reorder distance and we don't expect the
2398 * missing packet to be this reordered */
2399 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2400 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2405 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2406 && priv->do_retransmission && priv->rtx_next_seqnum;
2408 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2409 if (timer->num_rtx_retry > 0) {
2411 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2412 /* don't try to estimate the next seqnum because this is a retransmitted
2413 * packet and it probably did not arrive with the expected packet
2415 do_next_seqnum = FALSE;
2418 if (!is_rtx || timer->num_rtx_retry > 1) {
2419 /* Store timer in order to record stats when/if the retransmitted
2420 * packet arrives. We should also store timer information if we've
2421 * requested retransmission more than once since we may receive
2422 * several retransmitted packets. For accuracy we should update the
2423 * stats also when the redundant retransmitted packets arrives. */
2424 timer_queue_append (priv->rtx_stats_timers, timer,
2425 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2430 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2431 GstClockTime expected, delay;
2433 /* calculate expected arrival time of the next seqnum */
2434 expected = pts + priv->packet_spacing;
2436 delay = get_rtx_delay (priv);
2438 /* and update/install timer for next seqnum */
2439 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2440 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2441 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2442 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2443 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2446 timer->type = TIMER_TYPE_EXPECTED;
2447 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2450 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2451 expected, delay, priv->packet_spacing);
2453 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2454 /* if we had a timer, remove it, we don't know when to expect the next
2456 remove_timer (jitterbuffer, timer);
2461 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2464 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2466 /* we need consecutive seqnums with a different
2467 * rtptime to estimate the packet spacing. */
2468 if (priv->ips_rtptime != rtptime) {
2469 /* rtptime changed, check pts diff */
2470 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2471 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2472 GstClockTime old_packet_spacing = priv->packet_spacing;
2474 /* Biased towards bigger packet spacings to prevent
2475 * too many unneeded retransmission requests for next
2476 * packets that just arrive a little later than we would
2478 if (old_packet_spacing > new_packet_spacing)
2479 priv->packet_spacing =
2480 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2481 else if (old_packet_spacing > 0)
2482 priv->packet_spacing =
2483 (3 * new_packet_spacing + old_packet_spacing) / 4;
2485 priv->packet_spacing = new_packet_spacing;
2487 GST_DEBUG_OBJECT (jitterbuffer,
2488 "new packet spacing %" GST_TIME_FORMAT
2489 " old packet spacing %" GST_TIME_FORMAT
2490 " combined to %" GST_TIME_FORMAT,
2491 GST_TIME_ARGS (new_packet_spacing),
2492 GST_TIME_ARGS (old_packet_spacing),
2493 GST_TIME_ARGS (priv->packet_spacing));
2495 priv->ips_rtptime = rtptime;
2496 priv->ips_pts = pts;
2501 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2502 guint16 seqnum, GstClockTime pts, gint gap)
2504 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2505 GstClockTime duration, expected_pts, delay;
2507 gboolean equidistant = priv->equidistant > 0;
2509 GST_DEBUG_OBJECT (jitterbuffer,
2510 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2511 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2513 if (pts == GST_CLOCK_TIME_NONE) {
2514 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2519 GstClockTime total_duration;
2520 /* the total duration spanned by the missing packets */
2521 if (pts >= priv->last_in_pts)
2522 total_duration = pts - priv->last_in_pts;
2526 /* interpolate between the current time and the last time based on
2527 * number of packets we are missing, this is the estimated duration
2528 * for the missing packet based on equidistant packet spacing. */
2529 duration = total_duration / (gap + 1);
2531 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2532 GST_TIME_ARGS (duration));
2534 if (total_duration > priv->latency_ns) {
2535 GstClockTime gap_time;
2539 GstClockTime gap_dur = gap * duration;
2540 if (gap_dur > priv->latency_ns)
2541 gap_time = gap_dur - priv->latency_ns;
2544 lost_packets = gap_time / duration;
2546 gap_time = total_duration - priv->latency_ns;
2550 /* too many lost packets, some of the missing packets are already
2551 * too late and we can generate lost packet events for them. */
2552 GST_INFO_OBJECT (jitterbuffer,
2553 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2554 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2555 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2556 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2557 GST_TIME_ARGS (gap_time));
2559 /* this timer will fire immediately and the lost event will be pushed from
2560 * the timer thread */
2561 if (lost_packets > 0) {
2562 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2563 priv->last_in_pts + duration, 0, gap_time);
2564 expected += lost_packets;
2565 priv->last_in_pts += gap_time;
2569 expected_pts = priv->last_in_pts + duration;
2571 /* If we cannot assume equidistant packet spacing, the only thing we now
2572 * for sure is that the missing packets have expected pts not later than
2573 * the last received pts. */
2580 if (priv->do_retransmission) {
2581 TimerData *timer = find_timer (jitterbuffer, expected);
2583 type = TIMER_TYPE_EXPECTED;
2584 delay = get_rtx_delay (priv);
2586 /* if we had a timer for the first missing packet, update it. */
2587 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2588 GstClockTime timeout = timer->timeout;
2590 timer->duration = duration;
2591 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2592 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2596 expected_pts += duration;
2599 type = TIMER_TYPE_LOST;
2602 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2603 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2604 expected_pts += duration;
2610 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2614 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2615 GstRtpJitterBufferPrivate *priv;
2617 priv = jitterbuffer->priv;
2619 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2622 if (priv->last_dts != -1)
2623 dtsdiff = dts - priv->last_dts;
2627 if (priv->last_rtptime != -1)
2628 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2632 /* Guess whether stream currently uses equidistant packet spacing. If we
2633 * often see identical timestamps it means the packets are not
2635 if (rtptime == priv->last_rtptime)
2636 priv->equidistant -= 2;
2638 priv->equidistant += 1;
2639 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2641 priv->last_dts = dts;
2642 priv->last_rtptime = rtptime;
2646 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2649 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2651 diff = ABS (dtsdiff - rtpdiffns);
2653 /* jitter is stored in nanoseconds */
2654 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2656 GST_LOG_OBJECT (jitterbuffer,
2657 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2658 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2659 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2660 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2667 GST_DEBUG_OBJECT (jitterbuffer,
2668 "no dts or no clock-rate, can't calculate jitter");
2674 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2676 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2677 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2680 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2681 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2682 gst_rtp_buffer_unmap (&rtp_a);
2684 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2685 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2686 gst_rtp_buffer_unmap (&rtp_b);
2688 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2692 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2693 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2695 GstRtpJitterBufferPrivate *priv;
2696 guint gap_packets_length;
2697 gboolean reset = FALSE;
2698 gboolean future = gap > 0;
2700 priv = jitterbuffer->priv;
2702 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2704 guint32 prev_gap_seq = -1;
2705 gboolean all_consecutive = TRUE;
2707 g_queue_insert_sorted (&priv->gap_packets, buffer,
2708 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2710 for (l = priv->gap_packets.head; l; l = l->next) {
2711 GstBuffer *gap_buffer = l->data;
2712 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2715 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2717 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2719 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2720 if (prev_gap_seq == -1)
2721 prev_gap_seq = gap_seq;
2722 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2723 all_consecutive = FALSE;
2725 prev_gap_seq = gap_seq;
2727 gst_rtp_buffer_unmap (&gap_rtp);
2728 if (!all_consecutive)
2732 if (all_consecutive && gap_packets_length > 3) {
2733 GST_DEBUG_OBJECT (jitterbuffer,
2734 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2735 (future ? "new" : "old"), gap,
2736 (future ? max_dropout : -max_misorder));
2738 } else if (!all_consecutive) {
2739 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2740 g_queue_clear (&priv->gap_packets);
2741 GST_DEBUG_OBJECT (jitterbuffer,
2742 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2743 (future ? "new" : "old"), gap,
2744 (future ? max_dropout : -max_misorder));
2747 GST_DEBUG_OBJECT (jitterbuffer,
2748 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2749 (future ? "new" : "old"), gap,
2750 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2754 GST_DEBUG_OBJECT (jitterbuffer,
2755 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2756 gap, -max_misorder);
2757 g_queue_push_tail (&priv->gap_packets, buffer);
2765 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2767 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2768 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2771 GstClockTime base_time =
2772 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2773 GstClockTime clock_time = gst_clock_get_time (clock);
2775 if (clock_time > base_time)
2776 running_time = clock_time - base_time;
2780 gst_object_unref (clock);
2783 return running_time;
2786 static GstFlowReturn
2787 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2788 GstPad * pad, GstObject * parent, guint16 seqnum)
2790 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2791 GstFlowReturn ret = GST_FLOW_OK;
2792 GList *events = NULL, *l;
2796 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2797 rtp_jitter_buffer_flush (priv->jbuf,
2798 (GFunc) free_item_and_retain_events, &events);
2799 rtp_jitter_buffer_reset_skew (priv->jbuf);
2800 remove_all_timers (jitterbuffer);
2801 priv->discont = TRUE;
2802 priv->last_popped_seqnum = -1;
2804 if (priv->gap_packets.head) {
2805 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2806 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2808 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2809 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2810 gst_rtp_buffer_unmap (&gap_rtp);
2812 priv->next_seqnum = seqnum;
2815 priv->last_in_pts = -1;
2816 priv->next_in_seqnum = -1;
2818 /* Insert all sticky events again in order, otherwise we would
2819 * potentially loose STREAM_START, CAPS or SEGMENT events
2821 events = g_list_reverse (events);
2822 for (l = events; l; l = l->next) {
2823 RTPJitterBufferItem *item;
2825 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2826 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2828 g_list_free (events);
2830 JBUF_SIGNAL_EVENT (priv);
2832 /* reset spacing estimation when gap */
2833 priv->ips_rtptime = -1;
2834 priv->ips_pts = GST_CLOCK_TIME_NONE;
2836 buffers = g_list_copy (priv->gap_packets.head);
2837 g_queue_clear (&priv->gap_packets);
2839 priv->ips_rtptime = -1;
2840 priv->ips_pts = GST_CLOCK_TIME_NONE;
2841 JBUF_UNLOCK (jitterbuffer->priv);
2843 for (l = buffers; l; l = l->next) {
2844 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2846 if (ret != GST_FLOW_OK) {
2851 for (; l; l = l->next)
2852 gst_buffer_unref (l->data);
2853 g_list_free (buffers);
2859 gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
2861 GstRtpJitterBufferPrivate *priv;
2862 RTPJitterBufferItem *item;
2865 priv = jitterbuffer->priv;
2867 if (priv->faststart_min_packets == 0)
2870 item = rtp_jitter_buffer_peek (priv->jbuf);
2874 timer = find_timer (jitterbuffer, item->seqnum);
2875 if (!timer || timer->type != TIMER_TYPE_DEADLINE)
2878 if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
2879 priv->faststart_min_packets)) {
2880 GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
2881 priv->faststart_min_packets);
2882 timer->timeout = -1;
2889 static GstFlowReturn
2890 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2893 GstRtpJitterBuffer *jitterbuffer;
2894 GstRtpJitterBufferPrivate *priv;
2896 guint32 expected, rtptime;
2897 GstFlowReturn ret = GST_FLOW_OK;
2898 GstClockTime dts, pts;
2903 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2904 gboolean do_next_seqnum = FALSE;
2905 RTPJitterBufferItem *item;
2906 GstMessage *msg = NULL;
2907 gboolean estimated_dts = FALSE;
2908 gint32 packet_rate, max_dropout, max_misorder;
2909 TimerData *timer = NULL;
2911 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2913 priv = jitterbuffer->priv;
2915 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2916 goto invalid_buffer;
2918 pt = gst_rtp_buffer_get_payload_type (&rtp);
2919 seqnum = gst_rtp_buffer_get_seq (&rtp);
2920 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2921 gst_rtp_buffer_unmap (&rtp);
2923 /* make sure we have PTS and DTS set */
2924 pts = GST_BUFFER_PTS (buffer);
2925 dts = GST_BUFFER_DTS (buffer);
2932 /* If we have no DTS here, i.e. no capture time, get one from the
2933 * clock now to have something to calculate with in the future. */
2934 dts = get_current_running_time (jitterbuffer);
2937 /* Remember that we estimated the DTS if we are running already
2938 * and this is not our first packet (or first packet after a reset).
2939 * If it's the first packet, we somehow must generate a timestamp for
2940 * everything, otherwise we can't calculate any times
2942 estimated_dts = (priv->next_in_seqnum != -1);
2944 /* take the DTS of the buffer. This is the time when the packet was
2945 * received and is used to calculate jitter and clock skew. We will adjust
2946 * this DTS with the smoothed value after processing it in the
2947 * jitterbuffer and assign it as the PTS. */
2948 /* bring to running time */
2949 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2952 GST_DEBUG_OBJECT (jitterbuffer,
2953 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2954 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2955 GST_BUFFER_IS_RETRANSMISSION (buffer));
2957 JBUF_LOCK_CHECK (priv, out_flushing);
2959 if (G_UNLIKELY (priv->last_pt != pt)) {
2962 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2966 /* reset clock-rate so that we get a new one */
2967 priv->clock_rate = -1;
2969 /* Try to get the clock-rate from the caps first if we can. If there are no
2970 * caps we must fire the signal to get the clock-rate. */
2971 if ((caps = gst_pad_get_current_caps (pad))) {
2972 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2973 gst_caps_unref (caps);
2977 if (G_UNLIKELY (priv->clock_rate == -1)) {
2978 /* no clock rate given on the caps, try to get one with the signal */
2979 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2980 pt) == GST_FLOW_FLUSHING)
2983 if (G_UNLIKELY (priv->clock_rate == -1))
2986 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2989 /* don't accept more data on EOS */
2990 if (G_UNLIKELY (priv->eos))
2993 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2994 calculate_jitter (jitterbuffer, dts, rtptime);
2996 if (priv->seqnum_base != -1) {
2999 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
3002 GST_DEBUG_OBJECT (jitterbuffer,
3003 "packet seqnum #%d before seqnum-base #%d", seqnum,
3005 gst_buffer_unref (buffer);
3007 } else if (gap > 16384) {
3008 /* From now on don't compare against the seqnum base anymore as
3009 * at some point in the future we will wrap around and also that
3010 * much reordering is very unlikely */
3011 priv->seqnum_base = -1;
3015 expected = priv->next_in_seqnum;
3018 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
3020 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
3021 priv->max_dropout_time);
3023 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
3024 priv->max_misorder_time);
3025 GST_TRACE_OBJECT (jitterbuffer,
3026 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
3027 max_dropout, max_misorder);
3029 /* now check against our expected seqnum */
3030 if (G_UNLIKELY (expected == -1)) {
3031 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3033 /* calculate a pts based on rtptime and arrival time (dts) */
3035 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3036 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3038 /* we don't know what the next_in_seqnum should be, wait for the last
3039 * possible moment to push this buffer, maybe we get an earlier seqnum
3041 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
3043 do_next_seqnum = TRUE;
3044 /* take rtptime and pts to calculate packet spacing */
3045 priv->ips_rtptime = rtptime;
3046 priv->ips_pts = pts;
3050 /* now calculate gap */
3051 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
3052 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
3053 expected, seqnum, gap);
3055 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
3056 /* If we have timers for more than RTP_MAX_DROPOUT packets
3057 * pending this means that we have a huge gap overall. We can
3058 * reset the jitterbuffer at this point because there's
3059 * just too much data missing to be able to do anything
3060 * sensible with the past data. Just try again from the
3062 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
3063 priv->timers->len, max_dropout);
3064 gst_buffer_unref (buffer);
3065 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3068 /* Special handling of large gaps */
3069 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
3070 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
3071 gap, max_dropout, max_misorder);
3073 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3075 GST_DEBUG_OBJECT (jitterbuffer,
3076 "Had big gap, waiting for more consecutive packets");
3081 /* We had no huge gap, let's drop all the gap packets */
3082 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
3083 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3084 g_queue_clear (&priv->gap_packets);
3086 /* calculate a pts based on rtptime and arrival time (dts) */
3087 /* If we estimated the DTS, don't consider it in the clock skew calculations */
3089 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3090 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3092 if (G_LIKELY (gap == 0)) {
3093 /* packet is expected */
3094 calculate_packet_spacing (jitterbuffer, rtptime, pts);
3095 do_next_seqnum = TRUE;
3100 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
3101 /* fill in the gap with EXPECTED timers */
3102 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
3103 do_next_seqnum = TRUE;
3105 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3106 do_next_seqnum = FALSE;
3109 /* reset spacing estimation when gap */
3110 priv->ips_rtptime = -1;
3111 priv->ips_pts = GST_CLOCK_TIME_NONE;
3115 if (do_next_seqnum) {
3116 priv->last_in_pts = pts;
3117 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3120 timer = find_timer (jitterbuffer, seqnum);
3121 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3123 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3125 timer->num_rtx_received++;
3128 /* let's check if this buffer is too late, we can only accept packets with
3129 * bigger seqnum than the one we last pushed. */
3130 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3133 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3135 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3136 if (G_UNLIKELY (gap <= 0)) {
3137 if (priv->do_retransmission) {
3138 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3139 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3140 /* Only count the retranmitted packet too late if it has been
3141 * considered lost. If the original packet arrived before the
3142 * retransmitted we just count it as a duplicate. */
3143 if (timer->type != TIMER_TYPE_LOST)
3151 if (already_lost (jitterbuffer, seqnum))
3154 /* let's drop oldest packet if the queue is already full and drop-on-latency
3155 * is set. We can only do this when there actually is a latency. When no
3156 * latency is set, we just pump it in the queue and let the other end push it
3157 * out as fast as possible. */
3158 if (priv->latency_ms && priv->drop_on_latency) {
3160 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3162 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3163 RTPJitterBufferItem *old_item;
3165 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3167 if (IS_DROPABLE (old_item)) {
3168 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3169 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3171 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3172 free_item (old_item);
3174 /* we might have removed some head buffers, signal the pushing thread to
3175 * see if it can push now */
3176 JBUF_SIGNAL_EVENT (priv);
3180 /* If we estimated the DTS, don't consider it in the clock skew calculations
3181 * later. The code above always sets dts to pts or the other way around if
3182 * any of those is valid in the buffer, so we know that if we estimated the
3183 * dts that both are unknown */
3186 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3187 pts, seqnum, 1, rtptime);
3189 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3191 /* now insert the packet into the queue in sorted order. This function returns
3192 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3193 * have a duplicate. */
3194 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3196 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3197 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3201 /* Trigger fast start if needed */
3202 if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
3206 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3207 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3209 /* we had an unhandled SR, handle it now */
3211 do_handle_sync (jitterbuffer);
3213 if (G_UNLIKELY (head)) {
3214 /* signal addition of new buffer when the _loop is waiting. */
3215 if (G_LIKELY (priv->active))
3216 JBUF_SIGNAL_EVENT (priv);
3218 /* let's unschedule and unblock any waiting buffers. We only want to do this
3219 * when the head buffer changed */
3220 if (G_UNLIKELY (priv->clock_id)) {
3221 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3222 unschedule_current_timer (jitterbuffer);
3226 GST_DEBUG_OBJECT (jitterbuffer,
3227 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3228 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3230 msg = check_buffering_percent (jitterbuffer, percent);
3236 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3243 /* this is not fatal but should be filtered earlier */
3244 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3245 ("Received invalid RTP payload, dropping"));
3246 gst_buffer_unref (buffer);
3251 GST_WARNING_OBJECT (jitterbuffer,
3252 "No clock-rate in caps!, dropping buffer");
3253 gst_buffer_unref (buffer);
3258 ret = priv->srcresult;
3259 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3260 gst_buffer_unref (buffer);
3266 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3267 gst_buffer_unref (buffer);
3272 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3273 " popped, dropping", seqnum, priv->last_popped_seqnum);
3275 gst_buffer_unref (buffer);
3280 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3281 "considered lost", seqnum);
3283 gst_buffer_unref (buffer);
3288 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3290 priv->num_duplicates++;
3296 GST_DEBUG_OBJECT (jitterbuffer,
3297 "Duplicate RTX packet #%d detected, dropping", seqnum);
3298 priv->num_duplicates++;
3299 gst_buffer_unref (buffer);
3304 /* FIXME: hopefully we can do something more efficient here, especially when
3305 * all packets are in order and/or outside of the currently cached range.
3306 * Still worthwhile to have it, avoids taking/releasing object lock and pad
3307 * stream lock for every single buffer in the default chain_list fallback. */
3308 static GstFlowReturn
3309 gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent,
3310 GstBufferList * buffer_list)
3312 GstFlowReturn flow_ret = GST_FLOW_OK;
3315 n = gst_buffer_list_length (buffer_list);
3316 for (i = 0; i < n; ++i) {
3317 GstBuffer *buf = gst_buffer_list_get (buffer_list, i);
3319 flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf));
3321 if (flow_ret != GST_FLOW_OK)
3324 gst_buffer_list_unref (buffer_list);
3330 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3332 guint64 ext_time, elapsed;
3334 GstRtpJitterBufferPrivate *priv;
3336 priv = jitterbuffer->priv;
3337 rtp_time = item->rtptime;
3339 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3340 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3342 ext_time = priv->ext_timestamp;
3343 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3344 if (ext_time < priv->ext_timestamp) {
3345 ext_time = priv->ext_timestamp;
3347 priv->ext_timestamp = ext_time;
3350 if (ext_time > priv->clock_base)
3351 elapsed = ext_time - priv->clock_base;
3355 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3360 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3361 RTPJitterBufferItem * item)
3363 guint64 total, elapsed, left, estimated;
3364 GstClockTime out_time;
3365 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3367 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3368 || priv->clock_base == -1 || priv->clock_rate <= 0)
3371 /* compute the elapsed time */
3372 elapsed = compute_elapsed (jitterbuffer, item);
3374 /* do nothing if elapsed time doesn't increment */
3375 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3378 priv->last_elapsed = elapsed;
3380 /* this is the total time we need to play */
3381 total = priv->npt_stop - priv->npt_start;
3382 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3383 GST_TIME_ARGS (total));
3385 /* this is how much time there is left */
3386 if (total > elapsed)
3387 left = total - elapsed;
3391 /* if we have less time left that the size of the buffer, we will not
3392 * be able to keep it filled, disabled buffering then */
3393 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3394 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3395 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3396 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3399 /* this is the current time as running-time */
3400 out_time = item->pts;
3403 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3405 /* if there is almost nothing left,
3406 * we may never advance enough to end up in the above case */
3407 if (total < GST_SECOND)
3408 estimated = GST_SECOND;
3412 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3413 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3415 if (estimated != -1 && priv->estimated_eos != estimated) {
3416 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3417 priv->estimated_eos = estimated;
3421 /* take a buffer from the queue and push it */
3422 static GstFlowReturn
3423 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3425 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3426 GstFlowReturn result = GST_FLOW_OK;
3427 RTPJitterBufferItem *item;
3428 GstBuffer *outbuf = NULL;
3429 GstEvent *outevent = NULL;
3430 GstQuery *outquery = NULL;
3431 GstClockTime dts, pts;
3433 gboolean do_push = TRUE;
3437 /* when we get here we are ready to pop and push the buffer */
3438 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3442 case ITEM_TYPE_BUFFER:
3444 /* we need to make writable to change the flags and timestamps */
3445 outbuf = gst_buffer_make_writable (item->data);
3447 if (G_UNLIKELY (priv->discont)) {
3448 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3449 * into the jitterbuffer so we can modify now. */
3450 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3451 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3452 priv->discont = FALSE;
3454 if (G_UNLIKELY (priv->ts_discont)) {
3455 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3456 priv->ts_discont = FALSE;
3460 gst_segment_position_from_running_time (&priv->segment,
3461 GST_FORMAT_TIME, item->dts);
3463 gst_segment_position_from_running_time (&priv->segment,
3464 GST_FORMAT_TIME, item->pts);
3466 /* if this is a new frame, check if ts_offset needs to be updated */
3467 if (pts != priv->last_pts) {
3468 update_offset (jitterbuffer);
3471 /* apply timestamp with offset to buffer now */
3472 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3473 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3475 /* update the elapsed time when we need to check against the npt stop time. */
3476 update_estimated_eos (jitterbuffer, item);
3478 priv->last_pts = pts;
3479 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3481 case ITEM_TYPE_LOST:
3482 priv->discont = TRUE;
3486 case ITEM_TYPE_EVENT:
3487 outevent = item->data;
3489 case ITEM_TYPE_QUERY:
3490 outquery = item->data;
3494 /* now we are ready to push the buffer. Save the seqnum and release the lock
3495 * so the other end can push stuff in the queue again. */
3497 priv->last_popped_seqnum = seqnum;
3498 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3500 msg = check_buffering_percent (jitterbuffer, percent);
3502 if (type == ITEM_TYPE_EVENT && outevent &&
3503 GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3504 g_assert (priv->eos);
3505 while (priv->timers->len > 0) {
3506 /* Stopping timers */
3507 unschedule_current_timer (jitterbuffer);
3508 JBUF_WAIT_TIMER (priv);
3518 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3521 case ITEM_TYPE_BUFFER:
3523 GST_DEBUG_OBJECT (jitterbuffer,
3524 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3525 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3526 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3528 result = gst_pad_push (priv->srcpad, outbuf);
3530 JBUF_LOCK_CHECK (priv, out_flushing);
3532 case ITEM_TYPE_LOST:
3533 case ITEM_TYPE_EVENT:
3534 /* We got not enough consecutive packets with a huge gap, we can
3535 * as well just drop them here now on EOS */
3536 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3537 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3538 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3539 g_queue_clear (&priv->gap_packets);
3542 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3543 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3546 gst_pad_push_event (priv->srcpad, outevent);
3548 gst_event_unref (outevent);
3550 result = GST_FLOW_OK;
3552 JBUF_LOCK_CHECK (priv, out_flushing);
3554 case ITEM_TYPE_QUERY:
3558 res = gst_pad_peer_query (priv->srcpad, outquery);
3560 JBUF_LOCK_CHECK (priv, out_flushing);
3561 result = GST_FLOW_OK;
3562 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3563 JBUF_SIGNAL_QUERY (priv, res);
3572 return priv->srcresult;
3576 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3578 /* Peek a buffer and compare the seqnum to the expected seqnum.
3579 * If all is fine, the buffer is pushed.
3580 * If something is wrong, we wait for some event
3582 static GstFlowReturn
3583 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3585 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3586 GstFlowReturn result;
3587 RTPJitterBufferItem *item;
3589 guint32 next_seqnum;
3591 /* only push buffers when PLAYING and active and not buffering */
3592 if (priv->blocked || !priv->active ||
3593 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3594 return GST_FLOW_WAIT;
3597 /* peek a buffer, we're just looking at the sequence number.
3598 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3599 * wait for a timeout or something to change.
3600 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3601 item = rtp_jitter_buffer_peek (priv->jbuf);
3606 /* get the seqnum and the next expected seqnum */
3607 seqnum = item->seqnum;
3609 return pop_and_push_next (jitterbuffer, seqnum);
3612 next_seqnum = priv->next_seqnum;
3614 /* get the gap between this and the previous packet. If we don't know the
3615 * previous packet seqnum assume no gap. */
3616 if (G_UNLIKELY (next_seqnum == -1)) {
3617 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3618 /* we don't know what the next_seqnum should be, the chain function should
3619 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3620 * fires, so wait for that */
3621 result = GST_FLOW_WAIT;
3623 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3625 if (G_LIKELY (gap == 0)) {
3626 /* no missing packet, pop and push */
3627 result = pop_and_push_next (jitterbuffer, seqnum);
3628 } else if (G_UNLIKELY (gap < 0)) {
3629 /* if we have a packet that we already pushed or considered dropped, pop it
3630 * off and get the next packet */
3631 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3632 seqnum, next_seqnum);
3633 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3635 result = GST_FLOW_OK;
3637 /* the chain function has scheduled timers to request retransmission or
3638 * when to consider the packet lost, wait for that */
3639 GST_DEBUG_OBJECT (jitterbuffer,
3640 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3641 next_seqnum, seqnum, gap);
3642 /* if we have reached EOS, just keep processing */
3644 result = pop_and_push_next (jitterbuffer, seqnum);
3645 result = GST_FLOW_OK;
3647 result = GST_FLOW_WAIT;
3656 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3658 return GST_FLOW_EOS;
3660 return GST_FLOW_WAIT;
3666 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3668 GstClockTime rtx_retry_timeout;
3669 GstClockTime rtx_min_retry_timeout;
3671 if (priv->rtx_retry_timeout == -1) {
3672 if (priv->avg_rtx_rtt == 0)
3673 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3675 /* we want to ask for a retransmission after we waited for a
3676 * complete RTT and the additional jitter */
3677 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3679 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3681 /* make sure we don't retry too often. On very low latency networks,
3682 * the RTT and jitter can be very low. */
3683 if (priv->rtx_min_retry_timeout == -1) {
3684 rtx_min_retry_timeout = priv->packet_spacing;
3686 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3688 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3690 return rtx_retry_timeout;
3694 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3695 GstClockTime rtx_retry_timeout)
3697 GstClockTime rtx_retry_period;
3699 if (priv->rtx_retry_period == -1) {
3700 /* we retry up to the configured jitterbuffer size but leaving some
3701 * room for the retransmission to arrive in time */
3702 if (rtx_retry_timeout > priv->latency_ns) {
3703 rtx_retry_period = 0;
3705 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3708 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3710 return rtx_retry_period;
3714 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3715 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3716 3. For very large measurements (> avg * 2), consider them "outliers"
3717 and count them a lot less (1/48th)
3720 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3724 if (priv->avg_rtx_rtt == 0) {
3725 priv->avg_rtx_rtt = rtt;
3729 if (rtt > 2 * priv->avg_rtx_rtt)
3731 else if (rtt > priv->avg_rtx_rtt)
3736 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3740 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3741 GstClockTime dts, gboolean success)
3743 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3747 /* we scheduled a retry for this packet and now we have it */
3748 priv->num_rtx_success++;
3749 /* all the previous retry attempts failed */
3750 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3752 /* All retries failed or was too late */
3753 priv->num_rtx_failed += timer->num_rtx_retry;
3756 /* number of retries before (hopefully) receiving the packet */
3757 if (priv->avg_rtx_num == 0.0)
3758 priv->avg_rtx_num = timer->num_rtx_retry;
3760 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3762 /* Calculate the delay between retransmission request and receiving this
3763 * packet. We have a valid delay if and only if this packet is a response to
3764 * our last request. If not we don't know if this is a response to an
3765 * earlier request and delay could be way off. For RTT is more important
3766 * with correct values than to update for every packet. */
3767 if (timer->num_rtx_retry == timer->num_rtx_received &&
3768 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3769 delay = dts - timer->rtx_last;
3770 update_avg_rtx_rtt (priv, delay);
3775 GST_LOG_OBJECT (jitterbuffer,
3776 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3777 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3778 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3779 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3780 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3781 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3782 GST_TIME_ARGS (priv->avg_rtx_rtt));
3785 /* the timeout for when we expected a packet expired */
3787 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3790 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3792 guint delay, delay_ms, avg_rtx_rtt_ms;
3793 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3794 guint rtx_deadline_ms;
3795 GstClockTime rtx_retry_period;
3796 GstClockTime rtx_retry_timeout;
3799 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3800 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3802 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3803 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3805 delay = timer->rtx_delay + timer->rtx_retry;
3807 delay_ms = GST_TIME_AS_MSECONDS (delay);
3808 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3809 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3810 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3812 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3814 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3815 gst_structure_new ("GstRTPRetransmissionRequest",
3816 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3817 "running-time", G_TYPE_UINT64, timer->rtx_base,
3818 "delay", G_TYPE_UINT, delay_ms,
3819 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3820 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3821 "period", G_TYPE_UINT, rtx_retry_period_ms,
3822 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3823 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3824 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3825 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3827 priv->num_rtx_requests++;
3828 timer->num_rtx_retry++;
3830 GST_OBJECT_LOCK (jitterbuffer);
3831 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3832 timer->rtx_last = gst_clock_get_time (clock);
3833 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3835 timer->rtx_last = now;
3837 GST_OBJECT_UNLOCK (jitterbuffer);
3839 /* calculate the timeout for the next retransmission attempt */
3840 timer->rtx_retry += rtx_retry_timeout;
3841 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3842 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3843 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3844 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3845 if ((priv->rtx_max_retries != -1
3846 && timer->num_rtx_retry >= priv->rtx_max_retries)
3847 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3848 || (timer->rtx_base + rtx_retry_period < now)) {
3849 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3850 /* too many retransmission request, we now convert the timer
3851 * to a lost timer, leave the num_rtx_retry as it is for stats */
3852 timer->type = TIMER_TYPE_LOST;
3853 timer->rtx_delay = 0;
3854 timer->rtx_retry = 0;
3856 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3857 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3860 gst_pad_push_event (priv->sinkpad, event);
3866 /* a packet is lost */
3868 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3871 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3872 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3874 GstEvent *event = NULL;
3875 RTPJitterBufferItem *item;
3877 seqnum = timer->seqnum;
3878 lost_packets = MAX (timer->num, 1);
3879 num_rtx_retry = timer->num_rtx_retry;
3881 /* we had a gap and thus we lost some packets. Create an event for this. */
3882 if (lost_packets > 1)
3883 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3884 seqnum + lost_packets - 1);
3886 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3888 priv->num_lost += lost_packets;
3889 priv->num_rtx_failed += num_rtx_retry;
3891 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3893 /* we now only accept seqnum bigger than this */
3894 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3895 priv->next_in_seqnum = next_in_seqnum;
3896 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3899 /* Avoid creating events if we don't need it. Note that we still need to create
3900 * the lost *ITEM* since it will be used to notify the outgoing thread of
3901 * lost items (so that we can set discont flags and such) */
3902 if (priv->do_lost) {
3903 GstClockTime duration, timestamp;
3904 /* create paket lost event */
3905 timestamp = apply_offset (jitterbuffer, timer->timeout);
3906 duration = timer->duration;
3907 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3908 duration = priv->packet_spacing;
3909 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3910 gst_structure_new ("GstRTPPacketLost",
3911 "seqnum", G_TYPE_UINT, (guint) seqnum,
3912 "timestamp", G_TYPE_UINT64, timestamp,
3913 "duration", G_TYPE_UINT64, duration,
3914 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3916 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3917 if (!rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL))
3921 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3922 /* Store info to update stats if the packet arrives too late */
3923 timer_queue_append (priv->rtx_stats_timers, timer,
3924 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3926 remove_timer (jitterbuffer, timer);
3929 JBUF_SIGNAL_EVENT (priv);
3935 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3938 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3940 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3941 remove_timer (jitterbuffer, timer);
3943 /* there was no EOS in the buffer, put one in there now */
3944 queue_event (jitterbuffer, gst_event_new_eos ());
3946 JBUF_SIGNAL_EVENT (priv);
3952 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3955 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3957 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3959 /* timer seqnum might have been obsoleted by caps seqnum-base,
3960 * only mess with current ongoing seqnum if still unknown */
3961 if (priv->next_seqnum == -1)
3962 priv->next_seqnum = timer->seqnum;
3963 remove_timer (jitterbuffer, timer);
3964 JBUF_SIGNAL_EVENT (priv);
3970 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3973 gboolean removed = FALSE;
3975 switch (timer->type) {
3976 case TIMER_TYPE_EXPECTED:
3977 removed = do_expected_timeout (jitterbuffer, timer, now);
3979 case TIMER_TYPE_LOST:
3980 removed = do_lost_timeout (jitterbuffer, timer, now);
3982 case TIMER_TYPE_DEADLINE:
3983 removed = do_deadline_timeout (jitterbuffer, timer, now);
3985 case TIMER_TYPE_EOS:
3986 removed = do_eos_timeout (jitterbuffer, timer, now);
3992 /* called when we need to wait for the next timeout.
3994 * We loop over the array of recorded timeouts and wait for the earliest one.
3995 * When it timed out, do the logic associated with the timer.
3997 * If there are no timers, we wait on a gcond until something new happens.
4000 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
4002 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
4003 GstClockTime now = 0;
4006 while (priv->timer_running) {
4007 TimerData *timer = NULL;
4008 GstClockTime timer_timeout = -1;
4011 /* If we have a clock, update "now" now with the very
4012 * latest running time we have. If timers are unscheduled below we
4013 * otherwise wouldn't update now (it's only updated when timers
4014 * expire), and also for the very first loop iteration now would
4015 * otherwise always be 0
4017 GST_OBJECT_LOCK (jitterbuffer);
4019 now = GST_CLOCK_TIME_NONE;
4020 } else if (GST_ELEMENT_CLOCK (jitterbuffer)) {
4022 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
4023 GST_ELEMENT_CAST (jitterbuffer)->base_time;
4025 GST_OBJECT_UNLOCK (jitterbuffer);
4027 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
4028 GST_TIME_ARGS (now));
4030 /* Clear expired rtx-stats timers */
4031 if (priv->do_retransmission)
4032 timer_queue_clear_until (priv->rtx_stats_timers, now);
4034 /* Iterate "normal" timers */
4035 len = priv->timers->len;
4036 for (i = 0; i < len;) {
4037 TimerData *test = &g_array_index (priv->timers, TimerData, i);
4038 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
4039 gboolean save_best = FALSE;
4041 GST_DEBUG_OBJECT (jitterbuffer,
4042 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
4043 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
4044 GST_STIME_ARGS ((gint64) (test_timeout - now)));
4046 /* Weed out anything too late */
4047 if (test->type == TIMER_TYPE_LOST &&
4048 (test_timeout == -1 || test_timeout <= now)) {
4049 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
4050 do_lost_timeout (jitterbuffer, test, now);
4051 if (!priv->timer_running)
4053 /* We don't move the iterator forward since we just removed the current entry,
4054 * but we update the termination condition */
4055 len = priv->timers->len;
4057 /* find the smallest timeout */
4058 if (timer == NULL) {
4060 } else if (timer_timeout == -1) {
4061 /* we already have an immediate timeout, the new timer must be an
4062 * immediate timer with smaller seqnum to become the best */
4063 if (test_timeout == -1
4064 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4065 timer->seqnum) > 0))
4067 } else if (test_timeout == -1) {
4068 /* first immediate timer */
4070 } else if (test_timeout < timer_timeout) {
4073 } else if (test_timeout == timer_timeout
4074 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4075 timer->seqnum) > 0)) {
4076 /* same timer, smaller seqnum */
4081 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
4083 timer_timeout = test_timeout;
4088 if (timer && !priv->blocked) {
4090 GstClockTime sync_time;
4093 GstClockTimeDiff clock_jitter;
4095 if (timer_timeout == -1 || timer_timeout <= now || priv->eos) {
4096 /* We have normally removed all lost timers in the loop above */
4097 g_assert (timer->type != TIMER_TYPE_LOST);
4099 do_timeout (jitterbuffer, timer, now);
4100 /* check here, do_timeout could have released the lock */
4101 if (!priv->timer_running)
4106 GST_OBJECT_LOCK (jitterbuffer);
4107 clock = GST_ELEMENT_CLOCK (jitterbuffer);
4109 GST_OBJECT_UNLOCK (jitterbuffer);
4110 /* let's just push if there is no clock */
4111 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
4112 now = timer_timeout;
4116 /* prepare for sync against clock */
4117 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
4118 /* add latency of peer to get input time */
4119 sync_time += priv->peer_latency;
4121 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
4122 " with sync time %" GST_TIME_FORMAT,
4123 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
4125 /* create an entry for the clock */
4126 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
4127 priv->timer_timeout = timer_timeout;
4128 priv->timer_seqnum = timer->seqnum;
4129 GST_OBJECT_UNLOCK (jitterbuffer);
4131 /* release the lock so that the other end can push stuff or unlock */
4134 ret = gst_clock_id_wait (id, &clock_jitter);
4137 if (!priv->timer_running) {
4138 gst_clock_id_unref (id);
4139 priv->clock_id = NULL;
4143 if (ret != GST_CLOCK_UNSCHEDULED) {
4144 now = timer_timeout + MAX (clock_jitter, 0);
4145 GST_DEBUG_OBJECT (jitterbuffer,
4146 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
4147 GST_STIME_ARGS (clock_jitter));
4149 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
4151 /* and free the entry */
4152 gst_clock_id_unref (id);
4153 priv->clock_id = NULL;
4155 /* no timers, wait for activity */
4156 JBUF_WAIT_TIMER (priv);
4161 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4166 * This funcion implements the main pushing loop on the source pad.
4168 * It first tries to push as many buffers as possible. If there is a seqnum
4169 * mismatch, we wait for the next timeouts.
4172 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4174 GstRtpJitterBufferPrivate *priv;
4175 GstFlowReturn result = GST_FLOW_OK;
4177 priv = jitterbuffer->priv;
4179 JBUF_LOCK_CHECK (priv, flushing);
4181 result = handle_next_buffer (jitterbuffer);
4182 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4183 /* now wait for the next event */
4184 JBUF_WAIT_EVENT (priv, flushing);
4185 result = GST_FLOW_OK;
4187 } while (result == GST_FLOW_OK);
4188 /* store result for upstream */
4189 priv->srcresult = result;
4190 /* if we get here we need to pause */
4196 result = priv->srcresult;
4203 JBUF_SIGNAL_QUERY (priv, FALSE);
4206 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4207 gst_flow_get_name (result));
4208 gst_pad_pause_task (priv->srcpad);
4209 if (result == GST_FLOW_EOS) {
4210 event = gst_event_new_eos ();
4211 gst_pad_push_event (priv->srcpad, event);
4217 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4218 * some sanity checks and then emit the handle-sync signal with the parameters.
4219 * This function must be called with the LOCK */
4221 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4223 GstRtpJitterBufferPrivate *priv;
4224 guint64 base_rtptime, base_time;
4226 guint64 last_rtptime;
4228 guint64 ext_rtptime, diff;
4229 gboolean valid = TRUE, keep = FALSE;
4231 priv = jitterbuffer->priv;
4233 /* get the last values from the jitterbuffer */
4234 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4235 &clock_rate, &last_rtptime);
4237 clock_base = priv->clock_base;
4238 ext_rtptime = priv->ext_rtptime;
4240 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4241 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4242 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4243 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4245 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4246 /* we keep this SR packet for later. When we get a valid RTP packet the
4247 * above values will be set and we can try to use the SR packet */
4248 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4251 /* we can't accept anything that happened before we did the last resync */
4252 if (base_rtptime > ext_rtptime) {
4253 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4256 /* the SR RTP timestamp must be something close to what we last observed
4257 * in the jitterbuffer */
4258 if (ext_rtptime > last_rtptime) {
4259 /* check how far ahead it is to our RTP timestamps */
4260 diff = ext_rtptime - last_rtptime;
4261 /* if bigger than 1 second, we drop it */
4262 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4264 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4265 clock_rate, 1000)) {
4266 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4267 /* should drop this, but some RTSP servers end up with bogus
4268 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4269 * so still trigger rptbin sync but invalidate RTCP data
4270 * (sync might use other methods) */
4273 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4274 G_GUINT64_FORMAT, last_rtptime, diff);
4280 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4284 s = gst_structure_new ("application/x-rtp-sync",
4285 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4286 "base-time", G_TYPE_UINT64, base_time,
4287 "clock-rate", G_TYPE_UINT, clock_rate,
4288 "clock-base", G_TYPE_UINT64, clock_base,
4289 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4290 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4292 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4293 gst_buffer_replace (&priv->last_sr, NULL);
4295 g_signal_emit (jitterbuffer,
4296 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4298 gst_structure_free (s);
4300 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4301 gst_buffer_replace (&priv->last_sr, NULL);
4305 static GstFlowReturn
4306 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4309 GstRtpJitterBuffer *jitterbuffer;
4310 GstRtpJitterBufferPrivate *priv;
4311 GstFlowReturn ret = GST_FLOW_OK;
4313 GstRTCPPacket packet;
4314 guint64 ext_rtptime;
4316 GstRTCPBuffer rtcp = { NULL, };
4318 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4320 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4321 goto invalid_buffer;
4323 priv = jitterbuffer->priv;
4325 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4327 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4330 /* first packet must be SR or RR or else the validate would have failed */
4331 switch (gst_rtcp_packet_get_type (&packet)) {
4332 case GST_RTCP_TYPE_SR:
4333 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4339 gst_rtcp_buffer_unmap (&rtcp);
4341 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4344 /* convert the RTP timestamp to our extended timestamp, using the same offset
4345 * we used in the jitterbuffer */
4346 ext_rtptime = priv->jbuf->ext_rtptime;
4347 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4349 priv->ext_rtptime = ext_rtptime;
4350 gst_buffer_replace (&priv->last_sr, buffer);
4352 do_handle_sync (jitterbuffer);
4356 gst_buffer_unref (buffer);
4362 /* this is not fatal but should be filtered earlier */
4363 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4364 ("Received invalid RTCP payload, dropping"));
4370 /* this is not fatal but should be filtered earlier */
4371 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4372 ("Received empty RTCP payload, dropping"));
4373 gst_rtcp_buffer_unmap (&rtcp);
4379 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4380 gst_rtcp_buffer_unmap (&rtcp);
4387 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4390 gboolean res = FALSE;
4391 GstRtpJitterBuffer *jitterbuffer;
4392 GstRtpJitterBufferPrivate *priv;
4394 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4395 priv = jitterbuffer->priv;
4397 switch (GST_QUERY_TYPE (query)) {
4398 case GST_QUERY_CAPS:
4400 GstCaps *filter, *caps;
4402 gst_query_parse_caps (query, &filter);
4403 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4404 gst_query_set_caps_result (query, caps);
4405 gst_caps_unref (caps);
4410 if (GST_QUERY_IS_SERIALIZED (query)) {
4411 RTPJitterBufferItem *item;
4414 JBUF_LOCK_CHECK (priv, out_flushing);
4415 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4416 RTP_JITTER_BUFFER_MODE_BUFFER) {
4417 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4418 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4419 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4421 JBUF_SIGNAL_EVENT (priv);
4422 JBUF_WAIT_QUERY (priv, out_flushing);
4423 res = priv->last_query;
4425 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4430 res = gst_pad_query_default (pad, parent, query);
4438 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4446 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4449 GstRtpJitterBuffer *jitterbuffer;
4450 GstRtpJitterBufferPrivate *priv;
4451 gboolean res = FALSE;
4453 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4454 priv = jitterbuffer->priv;
4456 switch (GST_QUERY_TYPE (query)) {
4457 case GST_QUERY_LATENCY:
4459 /* We need to send the query upstream and add the returned latency to our
4461 GstClockTime min_latency, max_latency;
4463 GstClockTime our_latency;
4465 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4466 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4468 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4469 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4470 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4472 /* store this so that we can safely sync on the peer buffers. */
4474 priv->peer_latency = min_latency;
4475 our_latency = priv->latency_ns;
4478 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4479 GST_TIME_ARGS (our_latency));
4481 /* we add some latency but can buffer an infinite amount of time */
4482 min_latency += our_latency;
4485 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4486 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4487 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4489 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4493 case GST_QUERY_POSITION:
4495 GstClockTime start, last_out;
4498 gst_query_parse_position (query, &fmt, NULL);
4499 if (fmt != GST_FORMAT_TIME) {
4500 res = gst_pad_query_default (pad, parent, query);
4505 start = priv->npt_start;
4506 last_out = priv->last_out_time;
4509 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4510 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4511 GST_TIME_ARGS (last_out));
4513 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4514 /* bring 0-based outgoing time to stream time */
4515 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4518 res = gst_pad_query_default (pad, parent, query);
4522 case GST_QUERY_CAPS:
4524 GstCaps *filter, *caps;
4526 gst_query_parse_caps (query, &filter);
4527 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4528 gst_query_set_caps_result (query, caps);
4529 gst_caps_unref (caps);
4534 res = gst_pad_query_default (pad, parent, query);
4542 gst_rtp_jitter_buffer_set_property (GObject * object,
4543 guint prop_id, const GValue * value, GParamSpec * pspec)
4545 GstRtpJitterBuffer *jitterbuffer;
4546 GstRtpJitterBufferPrivate *priv;
4548 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4549 priv = jitterbuffer->priv;
4554 guint new_latency, old_latency;
4556 new_latency = g_value_get_uint (value);
4559 old_latency = priv->latency_ms;
4560 priv->latency_ms = new_latency;
4561 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4562 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4565 /* post message if latency changed, this will inform the parent pipeline
4566 * that a latency reconfiguration is possible/needed. */
4567 if (new_latency != old_latency) {
4568 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4569 GST_TIME_ARGS (new_latency * GST_MSECOND));
4571 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4572 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4576 case PROP_DROP_ON_LATENCY:
4578 priv->drop_on_latency = g_value_get_boolean (value);
4581 case PROP_TS_OFFSET:
4583 if (priv->max_ts_offset_adjustment != 0) {
4584 gint64 new_offset = g_value_get_int64 (value);
4586 if (new_offset > priv->ts_offset) {
4587 priv->ts_offset_remainder = new_offset - priv->ts_offset;
4589 priv->ts_offset_remainder = -(priv->ts_offset - new_offset);
4592 priv->ts_offset = g_value_get_int64 (value);
4593 priv->ts_offset_remainder = 0;
4595 priv->ts_discont = TRUE;
4598 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4600 priv->max_ts_offset_adjustment = g_value_get_uint64 (value);
4605 priv->do_lost = g_value_get_boolean (value);
4610 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4613 case PROP_DO_RETRANSMISSION:
4615 priv->do_retransmission = g_value_get_boolean (value);
4618 case PROP_RTX_NEXT_SEQNUM:
4620 priv->rtx_next_seqnum = g_value_get_boolean (value);
4623 case PROP_RTX_DELAY:
4625 priv->rtx_delay = g_value_get_int (value);
4628 case PROP_RTX_MIN_DELAY:
4630 priv->rtx_min_delay = g_value_get_uint (value);
4633 case PROP_RTX_DELAY_REORDER:
4635 priv->rtx_delay_reorder = g_value_get_int (value);
4638 case PROP_RTX_RETRY_TIMEOUT:
4640 priv->rtx_retry_timeout = g_value_get_int (value);
4643 case PROP_RTX_MIN_RETRY_TIMEOUT:
4645 priv->rtx_min_retry_timeout = g_value_get_int (value);
4648 case PROP_RTX_RETRY_PERIOD:
4650 priv->rtx_retry_period = g_value_get_int (value);
4653 case PROP_RTX_MAX_RETRIES:
4655 priv->rtx_max_retries = g_value_get_int (value);
4658 case PROP_RTX_DEADLINE:
4660 priv->rtx_deadline_ms = g_value_get_int (value);
4663 case PROP_RTX_STATS_TIMEOUT:
4665 priv->rtx_stats_timeout = g_value_get_uint (value);
4668 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4670 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4673 case PROP_MAX_DROPOUT_TIME:
4675 priv->max_dropout_time = g_value_get_uint (value);
4678 case PROP_MAX_MISORDER_TIME:
4680 priv->max_misorder_time = g_value_get_uint (value);
4683 case PROP_RFC7273_SYNC:
4685 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4686 g_value_get_boolean (value));
4689 case PROP_FASTSTART_MIN_PACKETS:
4691 priv->faststart_min_packets = g_value_get_uint (value);
4695 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4701 gst_rtp_jitter_buffer_get_property (GObject * object,
4702 guint prop_id, GValue * value, GParamSpec * pspec)
4704 GstRtpJitterBuffer *jitterbuffer;
4705 GstRtpJitterBufferPrivate *priv;
4707 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4708 priv = jitterbuffer->priv;
4713 g_value_set_uint (value, priv->latency_ms);
4716 case PROP_DROP_ON_LATENCY:
4718 g_value_set_boolean (value, priv->drop_on_latency);
4721 case PROP_TS_OFFSET:
4723 g_value_set_int64 (value, priv->ts_offset);
4726 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4728 g_value_set_uint64 (value, priv->max_ts_offset_adjustment);
4733 g_value_set_boolean (value, priv->do_lost);
4738 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4746 if (priv->srcresult != GST_FLOW_OK)
4749 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4751 g_value_set_int (value, percent);
4755 case PROP_DO_RETRANSMISSION:
4757 g_value_set_boolean (value, priv->do_retransmission);
4760 case PROP_RTX_NEXT_SEQNUM:
4762 g_value_set_boolean (value, priv->rtx_next_seqnum);
4765 case PROP_RTX_DELAY:
4767 g_value_set_int (value, priv->rtx_delay);
4770 case PROP_RTX_MIN_DELAY:
4772 g_value_set_uint (value, priv->rtx_min_delay);
4775 case PROP_RTX_DELAY_REORDER:
4777 g_value_set_int (value, priv->rtx_delay_reorder);
4780 case PROP_RTX_RETRY_TIMEOUT:
4782 g_value_set_int (value, priv->rtx_retry_timeout);
4785 case PROP_RTX_MIN_RETRY_TIMEOUT:
4787 g_value_set_int (value, priv->rtx_min_retry_timeout);
4790 case PROP_RTX_RETRY_PERIOD:
4792 g_value_set_int (value, priv->rtx_retry_period);
4795 case PROP_RTX_MAX_RETRIES:
4797 g_value_set_int (value, priv->rtx_max_retries);
4800 case PROP_RTX_DEADLINE:
4802 g_value_set_int (value, priv->rtx_deadline_ms);
4805 case PROP_RTX_STATS_TIMEOUT:
4807 g_value_set_uint (value, priv->rtx_stats_timeout);
4811 g_value_take_boxed (value,
4812 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4814 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4816 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4819 case PROP_MAX_DROPOUT_TIME:
4821 g_value_set_uint (value, priv->max_dropout_time);
4824 case PROP_MAX_MISORDER_TIME:
4826 g_value_set_uint (value, priv->max_misorder_time);
4829 case PROP_RFC7273_SYNC:
4831 g_value_set_boolean (value,
4832 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4835 case PROP_FASTSTART_MIN_PACKETS:
4837 g_value_set_uint (value, priv->faststart_min_packets);
4841 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4846 static GstStructure *
4847 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4849 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4853 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4854 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4855 "num-lost", G_TYPE_UINT64, priv->num_lost,
4856 "num-late", G_TYPE_UINT64, priv->num_late,
4857 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4858 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4859 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4860 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4861 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4862 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);