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 = TRUE; \
206 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
207 (priv)->waiting_timer = FALSE; \
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"); \
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 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
404 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
405 GstRtpJitterBufferPrivate))
407 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
408 GST_STATIC_PAD_TEMPLATE ("sink",
411 GST_STATIC_CAPS ("application/x-rtp"
412 /* "clock-rate = (int) [ 1, 2147483647 ], "
413 * "payload = (int) , "
414 * "encoding-name = (string) "
418 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
419 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
422 GST_STATIC_CAPS ("application/x-rtcp")
425 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
426 GST_STATIC_PAD_TEMPLATE ("src",
429 GST_STATIC_CAPS ("application/x-rtp"
430 /* "payload = (int) , "
431 * "clock-rate = (int) , "
432 * "encoding-name = (string) "
436 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
438 #define gst_rtp_jitter_buffer_parent_class parent_class
439 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
441 /* object overrides */
442 static void gst_rtp_jitter_buffer_set_property (GObject * object,
443 guint prop_id, const GValue * value, GParamSpec * pspec);
444 static void gst_rtp_jitter_buffer_get_property (GObject * object,
445 guint prop_id, GValue * value, GParamSpec * pspec);
446 static void gst_rtp_jitter_buffer_finalize (GObject * object);
448 /* element overrides */
449 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
450 * element, GstStateChange transition);
451 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
452 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
453 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
455 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
456 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
460 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
461 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
464 /* sinkpad overrides */
465 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
466 GstObject * parent, GstEvent * event);
467 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
468 GstObject * parent, GstBuffer * buffer);
469 static GstFlowReturn gst_rtp_jitter_buffer_chain_list (GstPad * pad,
470 GstObject * parent, GstBufferList * buffer_list);
472 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
473 GstObject * parent, GstEvent * event);
474 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
475 GstObject * parent, GstBuffer * buffer);
477 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
478 GstObject * parent, GstQuery * query);
480 /* srcpad overrides */
481 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
482 GstObject * parent, GstEvent * event);
483 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
484 GstObject * parent, GstPadMode mode, gboolean active);
485 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
486 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
487 GstObject * parent, GstQuery * query);
490 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
492 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
493 gboolean active, guint64 base_time);
494 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
496 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
497 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
499 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
501 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
504 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
505 TimerData * timer, GstClockTime dts, gboolean success);
507 static TimerQueue *timer_queue_new (void);
508 static void timer_queue_free (TimerQueue * queue);
511 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
513 GObjectClass *gobject_class;
514 GstElementClass *gstelement_class;
516 gobject_class = (GObjectClass *) klass;
517 gstelement_class = (GstElementClass *) klass;
519 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
521 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
523 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
524 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
527 * GstRtpJitterBuffer:latency:
529 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
530 * for at most this time.
532 g_object_class_install_property (gobject_class, PROP_LATENCY,
533 g_param_spec_uint ("latency", "Buffer latency in ms",
534 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
535 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
537 * GstRtpJitterBuffer:drop-on-latency:
539 * Drop oldest buffers when the queue is completely filled.
541 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
542 g_param_spec_boolean ("drop-on-latency",
543 "Drop buffers when maximum latency is reached",
544 "Tells the jitterbuffer to never exceed the given latency in size",
545 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
547 * GstRtpJitterBuffer:ts-offset:
549 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
550 * This is mainly used to ensure interstream synchronisation.
552 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
553 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
554 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
555 G_MAXINT64, DEFAULT_TS_OFFSET,
556 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
559 * GstRtpJitterBuffer:max-ts-offset-adjustment:
561 * The maximum number of nanoseconds per frame that time offset may be
562 * adjusted with. This is used to avoid sudden large changes to time stamps.
564 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
565 g_param_spec_uint64 ("max-ts-offset-adjustment",
566 "Max Timestamp Offset Adjustment",
567 "The maximum number of nanoseconds per frame that time stamp "
568 "offsets may be adjusted (0 = no limit).", 0, G_MAXUINT64,
569 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
570 G_PARAM_STATIC_STRINGS));
573 * GstRtpJitterBuffer:do-lost:
575 * Send out a GstRTPPacketLost event downstream when a packet is considered
578 g_object_class_install_property (gobject_class, PROP_DO_LOST,
579 g_param_spec_boolean ("do-lost", "Do Lost",
580 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
581 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
584 * GstRtpJitterBuffer:mode:
586 * Control the buffering and timestamping mode used by the jitterbuffer.
588 g_object_class_install_property (gobject_class, PROP_MODE,
589 g_param_spec_enum ("mode", "Mode",
590 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
591 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
593 * GstRtpJitterBuffer:percent:
595 * The percent of the jitterbuffer that is filled.
597 g_object_class_install_property (gobject_class, PROP_PERCENT,
598 g_param_spec_int ("percent", "percent",
599 "The buffer filled percent", 0, 100,
600 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
602 * GstRtpJitterBuffer:do-retransmission:
604 * Send out a GstRTPRetransmission event upstream when a packet is considered
605 * late and should be retransmitted.
609 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
610 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
611 "Send retransmission events upstream when a packet is late",
612 DEFAULT_DO_RETRANSMISSION,
613 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
616 * GstRtpJitterBuffer:rtx-next-seqnum
618 * Estimate when the next packet should arrive and schedule a retransmission
620 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
621 * for packet N+1. So it will be requested if it does not arrive at the expected time.
622 * The expected time is calculated using the dts of N and the packet spacing.
626 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
627 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
628 "Estimate when the next packet should arrive and schedule a "
629 "retransmission request for it.",
630 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
633 * GstRtpJitterBuffer:rtx-delay:
635 * When a packet did not arrive at the expected time, wait this extra amount
636 * of time before sending a retransmission event.
638 * When -1 is used, the max jitter will be used as extra delay.
642 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
643 g_param_spec_int ("rtx-delay", "RTX Delay",
644 "Extra time in ms to wait before sending retransmission "
645 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
646 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
649 * GstRtpJitterBuffer:rtx-min-delay:
651 * When a packet did not arrive at the expected time, wait at least this extra amount
652 * of time before sending a retransmission event.
656 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
657 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
658 "Minimum time in ms to wait before sending retransmission "
659 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
660 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
662 * GstRtpJitterBuffer:rtx-delay-reorder:
664 * Assume that a retransmission event should be sent when we see
665 * this much packet reordering.
667 * When -1 is used, the value will be estimated based on observed packet
668 * reordering. When 0 is used packet reordering alone will not cause a
669 * retransmission event (Since 1.10).
673 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
674 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
675 "Sending retransmission event when this much reordering "
676 "(0 disable, -1 automatic)",
677 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
678 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
680 * GstRtpJitterBuffer::rtx-retry-timeout:
682 * When no packet has been received after sending a retransmission event
683 * for this time, retry sending a retransmission event.
685 * When -1 is used, the value will be estimated based on observed round
690 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
691 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
692 "Retry sending a transmission event after this timeout in "
693 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
694 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
696 * GstRtpJitterBuffer::rtx-min-retry-timeout:
698 * The minimum amount of time between retry timeouts. When
699 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
700 * minimum interval between retry timeouts.
702 * When -1 is used, the value will be estimated based on the
707 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
708 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
709 "Minimum timeout between sending a transmission event in "
710 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
711 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
713 * GstRtpJitterBuffer:rtx-retry-period:
715 * The amount of time to try to get a retransmission.
717 * When -1 is used, the value will be estimated based on the jitterbuffer
718 * latency and the observed round trip time.
722 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
723 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
724 "Try to get a retransmission for this many ms "
725 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
726 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
728 * GstRtpJitterBuffer:rtx-max-retries:
730 * The maximum number of retries to request a retransmission.
732 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
733 * When -1 is used, the number of retransmission request will not be limited.
737 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
738 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
739 "The maximum number of retries to request a retransmission. "
740 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
741 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
743 * GstRtpJitterBuffer:rtx-deadline:
745 * The deadline for a valid RTX request in ms.
747 * How long the RTX RTCP will be valid for.
748 * When -1 is used, the size of the jitterbuffer will be used.
752 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
753 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
754 "The deadline for a valid RTX request in milliseconds. "
755 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
756 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
758 * GstRtpJitterBuffer::rtx-stats-timeout:
760 * The time to wait for a retransmitted packet after it has been
761 * considered lost in order to collect RTX statistics.
765 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
766 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
767 "The time to wait for a retransmitted packet after it has been "
768 "considered lost in order to collect statistics (ms)",
769 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
770 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
772 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
773 g_param_spec_uint ("max-dropout-time", "Max dropout time",
774 "The maximum time (milliseconds) of missing packets tolerated.",
775 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
776 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
778 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
779 g_param_spec_uint ("max-misorder-time", "Max misorder time",
780 "The maximum time (milliseconds) of misordered packets tolerated.",
781 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
782 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
784 * GstRtpJitterBuffer:stats:
786 * Various jitterbuffer statistics. This property returns a GstStructure
787 * with name application/x-rtp-jitterbuffer-stats with the following fields:
793 * <classname>"num-pushed"</classname>:
794 * the number of packets pushed out.
800 * <classname>"num-lost"</classname>:
801 * the number of packets considered lost.
807 * <classname>"num-late"</classname>:
808 * the number of packets arriving too late.
814 * <classname>"num-duplicates"</classname>:
815 * the number of duplicate packets.
821 * <classname>"rtx-count"</classname>:
822 * the number of retransmissions requested.
828 * <classname>"rtx-success-count"</classname>:
829 * the number of successful retransmissions.
835 * <classname>"rtx-per-packet"</classname>:
836 * average number of RTX per packet.
842 * <classname>"rtx-rtt"</classname>:
843 * average round trip time per RTX.
850 g_object_class_install_property (gobject_class, PROP_STATS,
851 g_param_spec_boxed ("stats", "Statistics",
852 "Various statistics", GST_TYPE_STRUCTURE,
853 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
856 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
858 * The maximum amount of time in ms that the RTP time in the RTCP SRs
859 * is allowed to be ahead of the last RTP packet we received. Use
860 * -1 to disable ignoring of RTCP packets.
864 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
865 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
866 "Maximum amount of time in ms that the RTP time in RTCP SRs "
867 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
868 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
869 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
871 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
872 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
873 "Synchronize received streams to the RFC7273 clock "
874 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
875 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
878 * GstRtpJitterBuffer:faststart-min-packets
880 * The number of consecutive packets needed to start (set to 0 to
881 * disable faststart. The jitterbuffer will by default start after the
882 * latency has elapsed)
886 g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
887 g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
888 "The number of consecutive packets needed to start (set to 0 to "
889 "disable faststart. The jitterbuffer will by default start after "
890 "the latency has elapsed)",
891 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
892 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
895 * GstRtpJitterBuffer::request-pt-map:
896 * @buffer: the object which received the signal
899 * Request the payload type as #GstCaps for @pt.
901 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
902 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
903 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
904 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
905 GST_TYPE_CAPS, 1, G_TYPE_UINT);
907 * GstRtpJitterBuffer::handle-sync:
908 * @buffer: the object which received the signal
909 * @struct: a GstStructure containing sync values.
911 * Be notified of new sync values.
913 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
914 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
915 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
916 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
917 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
920 * GstRtpJitterBuffer::on-npt-stop:
921 * @buffer: the object which received the signal
923 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
924 * the npt-stop position.
926 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
927 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
928 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
929 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
930 G_TYPE_NONE, 0, G_TYPE_NONE);
933 * GstRtpJitterBuffer::clear-pt-map:
934 * @buffer: the object which received the signal
936 * Invalidate the clock-rate as obtained with the
937 * #GstRtpJitterBuffer::request-pt-map signal.
939 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
940 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
941 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
942 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
943 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
946 * GstRtpJitterBuffer::set-active:
947 * @buffer: the object which received the signal
949 * Start pushing out packets with the given base time. This signal is only
950 * useful in buffering mode.
952 * Returns: the time of the last pushed packet.
954 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
955 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
956 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
957 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
958 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
961 gstelement_class->change_state =
962 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
963 gstelement_class->request_new_pad =
964 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
965 gstelement_class->release_pad =
966 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
967 gstelement_class->provide_clock =
968 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
969 gstelement_class->set_clock =
970 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
972 gst_element_class_add_static_pad_template (gstelement_class,
973 &gst_rtp_jitter_buffer_src_template);
974 gst_element_class_add_static_pad_template (gstelement_class,
975 &gst_rtp_jitter_buffer_sink_template);
976 gst_element_class_add_static_pad_template (gstelement_class,
977 &gst_rtp_jitter_buffer_sink_rtcp_template);
979 gst_element_class_set_static_metadata (gstelement_class,
980 "RTP packet jitter-buffer", "Filter/Network/RTP",
981 "A buffer that deals with network jitter and other transmission faults",
982 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
983 "Wim Taymans <wim.taymans@gmail.com>");
985 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
986 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
988 GST_DEBUG_CATEGORY_INIT
989 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
993 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
995 GstRtpJitterBufferPrivate *priv;
997 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
998 jitterbuffer->priv = priv;
1000 priv->latency_ms = DEFAULT_LATENCY_MS;
1001 priv->latency_ns = priv->latency_ms * GST_MSECOND;
1002 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
1003 priv->ts_offset = DEFAULT_TS_OFFSET;
1004 priv->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
1005 priv->do_lost = DEFAULT_DO_LOST;
1006 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
1007 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
1008 priv->rtx_delay = DEFAULT_RTX_DELAY;
1009 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
1010 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
1011 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
1012 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
1013 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
1014 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
1015 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
1016 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
1017 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
1018 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
1019 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
1020 priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
1022 priv->ts_offset_remainder = 0;
1023 priv->last_dts = -1;
1024 priv->last_pts = -1;
1025 priv->last_rtptime = -1;
1026 priv->avg_jitter = 0;
1027 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
1028 priv->rtx_stats_timers = timer_queue_new ();
1029 priv->jbuf = rtp_jitter_buffer_new ();
1030 g_mutex_init (&priv->jbuf_lock);
1031 g_cond_init (&priv->jbuf_timer);
1032 g_cond_init (&priv->jbuf_event);
1033 g_cond_init (&priv->jbuf_query);
1034 g_queue_init (&priv->gap_packets);
1035 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1037 /* reset skew detection initialy */
1038 rtp_jitter_buffer_reset_skew (priv->jbuf);
1039 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
1040 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1041 priv->active = TRUE;
1044 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1047 gst_pad_set_activatemode_function (priv->srcpad,
1048 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1049 gst_pad_set_query_function (priv->srcpad,
1050 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1051 gst_pad_set_event_function (priv->srcpad,
1052 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1055 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1058 gst_pad_set_chain_function (priv->sinkpad,
1059 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1060 gst_pad_set_chain_list_function (priv->sinkpad,
1061 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain_list));
1062 gst_pad_set_event_function (priv->sinkpad,
1063 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1064 gst_pad_set_query_function (priv->sinkpad,
1065 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1067 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1068 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1070 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1073 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1075 #define ITEM_TYPE_BUFFER 0
1076 #define ITEM_TYPE_LOST 1
1077 #define ITEM_TYPE_EVENT 2
1078 #define ITEM_TYPE_QUERY 3
1080 static RTPJitterBufferItem *
1081 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1082 guint seqnum, guint count, guint rtptime)
1084 RTPJitterBufferItem *item;
1086 item = g_slice_new (RTPJitterBufferItem);
1093 item->seqnum = seqnum;
1094 item->count = count;
1095 item->rtptime = rtptime;
1101 free_item (RTPJitterBufferItem * item)
1103 g_return_if_fail (item != NULL);
1105 if (item->data && item->type != ITEM_TYPE_QUERY)
1106 gst_mini_object_unref (item->data);
1107 g_slice_free (RTPJitterBufferItem, item);
1111 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1113 GList **l = user_data;
1115 if (item->data && item->type == ITEM_TYPE_EVENT
1116 && GST_EVENT_IS_STICKY (item->data)) {
1117 *l = g_list_prepend (*l, item->data);
1118 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1119 gst_mini_object_unref (item->data);
1121 g_slice_free (RTPJitterBufferItem, item);
1125 gst_rtp_jitter_buffer_finalize (GObject * object)
1127 GstRtpJitterBuffer *jitterbuffer;
1128 GstRtpJitterBufferPrivate *priv;
1130 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1131 priv = jitterbuffer->priv;
1133 g_array_free (priv->timers, TRUE);
1134 timer_queue_free (priv->rtx_stats_timers);
1135 g_mutex_clear (&priv->jbuf_lock);
1136 g_cond_clear (&priv->jbuf_timer);
1137 g_cond_clear (&priv->jbuf_event);
1138 g_cond_clear (&priv->jbuf_query);
1140 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1141 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1142 g_queue_clear (&priv->gap_packets);
1143 g_object_unref (priv->jbuf);
1145 G_OBJECT_CLASS (parent_class)->finalize (object);
1148 static GstIterator *
1149 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1151 GstRtpJitterBuffer *jitterbuffer;
1152 GstPad *otherpad = NULL;
1153 GstIterator *it = NULL;
1154 GValue val = { 0, };
1156 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1158 if (pad == jitterbuffer->priv->sinkpad) {
1159 otherpad = jitterbuffer->priv->srcpad;
1160 } else if (pad == jitterbuffer->priv->srcpad) {
1161 otherpad = jitterbuffer->priv->sinkpad;
1162 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1163 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1167 g_value_init (&val, GST_TYPE_PAD);
1168 g_value_set_object (&val, otherpad);
1169 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1170 g_value_unset (&val);
1177 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1179 GstRtpJitterBufferPrivate *priv;
1181 priv = jitterbuffer->priv;
1183 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1186 gst_pad_new_from_static_template
1187 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1188 gst_pad_set_chain_function (priv->rtcpsinkpad,
1189 gst_rtp_jitter_buffer_chain_rtcp);
1190 gst_pad_set_event_function (priv->rtcpsinkpad,
1191 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1192 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1193 gst_rtp_jitter_buffer_iterate_internal_links);
1194 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1195 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1197 return priv->rtcpsinkpad;
1201 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1203 GstRtpJitterBufferPrivate *priv;
1205 priv = jitterbuffer->priv;
1207 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1209 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1211 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1212 priv->rtcpsinkpad = NULL;
1216 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1217 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1219 GstRtpJitterBuffer *jitterbuffer;
1220 GstElementClass *klass;
1222 GstRtpJitterBufferPrivate *priv;
1224 g_return_val_if_fail (templ != NULL, NULL);
1225 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1227 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1228 priv = jitterbuffer->priv;
1229 klass = GST_ELEMENT_GET_CLASS (element);
1231 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1233 /* figure out the template */
1234 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1235 if (priv->rtcpsinkpad != NULL)
1238 result = create_rtcp_sink (jitterbuffer);
1240 goto wrong_template;
1247 g_warning ("rtpjitterbuffer: this is not our template");
1252 g_warning ("rtpjitterbuffer: pad already requested");
1258 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1260 GstRtpJitterBuffer *jitterbuffer;
1261 GstRtpJitterBufferPrivate *priv;
1263 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1264 g_return_if_fail (GST_IS_PAD (pad));
1266 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1267 priv = jitterbuffer->priv;
1269 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1271 if (priv->rtcpsinkpad == pad) {
1272 remove_rtcp_sink (jitterbuffer);
1281 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1287 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1289 return gst_system_clock_obtain ();
1293 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1295 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1297 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1299 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1303 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1305 GstRtpJitterBufferPrivate *priv;
1307 priv = jitterbuffer->priv;
1309 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1312 priv->clock_rate = -1;
1313 /* do not clear current content, but refresh state for new arrival */
1314 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1315 rtp_jitter_buffer_reset_skew (priv->jbuf);
1320 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1323 GstRtpJitterBufferPrivate *priv;
1324 GstClockTime last_out;
1325 RTPJitterBufferItem *item;
1330 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1331 active, GST_TIME_ARGS (offset));
1333 if (active != priv->active) {
1334 /* add the amount of time spent in paused to the output offset. All
1335 * outgoing buffers will have this offset applied to their timestamps in
1336 * order to make them arrive in time in the sink. */
1337 priv->out_offset = offset;
1338 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1339 GST_TIME_ARGS (priv->out_offset));
1340 priv->active = active;
1341 JBUF_SIGNAL_EVENT (priv);
1344 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1346 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1347 /* head buffer timestamp and offset gives our output time */
1348 last_out = item->pts + priv->ts_offset;
1350 /* use last known time when the buffer is empty */
1351 last_out = priv->last_out_time;
1359 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1361 GstRtpJitterBuffer *jitterbuffer;
1362 GstRtpJitterBufferPrivate *priv;
1367 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1368 priv = jitterbuffer->priv;
1370 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1372 caps = gst_pad_peer_query_caps (other, filter);
1374 templ = gst_pad_get_pad_template_caps (pad);
1376 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1381 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1383 intersect = gst_caps_intersect (caps, templ);
1384 gst_caps_unref (caps);
1385 gst_caps_unref (templ);
1389 gst_object_unref (jitterbuffer);
1395 * Must be called with JBUF_LOCK held
1399 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1400 GstCaps * caps, gint pt)
1402 GstRtpJitterBufferPrivate *priv;
1403 GstStructure *caps_struct;
1407 const gchar *ts_refclk, *mediaclk;
1409 priv = jitterbuffer->priv;
1411 /* first parse the caps */
1412 caps_struct = gst_caps_get_structure (caps, 0);
1414 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1416 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1418 GST_ERROR_OBJECT (jitterbuffer,
1419 "Got caps with wrong payload type (got %d, expected %d)", pt, payload);
1423 if (payload != -1) {
1424 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1425 priv->last_pt = payload;
1428 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1429 * measure the amount of data in the buffer */
1430 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1433 if (priv->clock_rate <= 0)
1436 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1438 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1440 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1442 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1443 * can use this to track the amount of time elapsed on the sender. */
1444 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1445 priv->clock_base = val;
1447 priv->clock_base = -1;
1449 priv->ext_timestamp = priv->clock_base;
1451 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1454 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1455 /* first expected seqnum, only update when we didn't have a previous base. */
1456 if (priv->next_in_seqnum == -1)
1457 priv->next_in_seqnum = val;
1458 if (priv->next_seqnum == -1) {
1459 priv->next_seqnum = val;
1460 JBUF_SIGNAL_EVENT (priv);
1462 priv->seqnum_base = val;
1464 priv->seqnum_base = -1;
1467 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1469 /* the start and stop times. The seqnum-base corresponds to the start time. We
1470 * will keep track of the seqnums on the output and when we reach the one
1471 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1472 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1473 priv->npt_start = tval;
1475 priv->npt_start = 0;
1477 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1478 priv->npt_stop = tval;
1480 priv->npt_stop = -1;
1482 GST_DEBUG_OBJECT (jitterbuffer,
1483 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1484 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1486 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1487 GstClock *clock = NULL;
1488 guint64 clock_offset = -1;
1490 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1493 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1494 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1495 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1497 const gchar *host, *portstr;
1501 host = ts_refclk + sizeof ("ntp=") - 1;
1502 if (host[0] == '[') {
1504 portstr = strchr (host, ']');
1505 if (portstr && portstr[1] == ':')
1506 portstr = portstr + 1;
1510 portstr = strrchr (host, ':');
1514 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1518 hostname = g_strndup (host, (portstr - host));
1520 hostname = g_strdup (host);
1522 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1525 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1526 const gchar *domainstr =
1527 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1530 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1533 clock = gst_ptp_clock_new (NULL, domain);
1535 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1538 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1539 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1541 if (!g_str_has_prefix (mediaclk, "direct=")
1542 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1543 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1544 if (strstr (mediaclk, "rate=") != NULL) {
1545 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1550 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1552 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1560 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1565 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1571 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1573 GstRtpJitterBufferPrivate *priv;
1575 priv = jitterbuffer->priv;
1578 /* mark ourselves as flushing */
1579 priv->srcresult = GST_FLOW_FLUSHING;
1580 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1581 /* this unblocks any waiting pops on the src pad task */
1582 JBUF_SIGNAL_EVENT (priv);
1583 JBUF_SIGNAL_QUERY (priv, FALSE);
1588 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1590 GstRtpJitterBufferPrivate *priv;
1592 priv = jitterbuffer->priv;
1595 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1596 /* Mark as non flushing */
1597 priv->srcresult = GST_FLOW_OK;
1598 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1599 priv->last_popped_seqnum = -1;
1600 priv->last_out_time = GST_CLOCK_TIME_NONE;
1601 priv->next_seqnum = -1;
1602 priv->seqnum_base = -1;
1603 priv->ips_rtptime = -1;
1604 priv->ips_pts = GST_CLOCK_TIME_NONE;
1605 priv->packet_spacing = 0;
1606 priv->next_in_seqnum = -1;
1607 priv->clock_rate = -1;
1610 priv->estimated_eos = -1;
1611 priv->last_elapsed = 0;
1612 priv->ext_timestamp = -1;
1613 priv->avg_jitter = 0;
1614 priv->last_dts = -1;
1615 priv->last_rtptime = -1;
1616 priv->last_in_pts = 0;
1617 priv->equidistant = 0;
1618 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1619 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1620 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1621 rtp_jitter_buffer_reset_skew (priv->jbuf);
1622 remove_all_timers (jitterbuffer);
1623 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1624 g_queue_clear (&priv->gap_packets);
1629 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1630 GstPadMode mode, gboolean active)
1633 GstRtpJitterBuffer *jitterbuffer = NULL;
1635 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1638 case GST_PAD_MODE_PUSH:
1640 /* allow data processing */
1641 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1643 /* start pushing out buffers */
1644 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1645 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1646 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1648 /* make sure all data processing stops ASAP */
1649 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1651 /* NOTE this will hardlock if the state change is called from the src pad
1652 * task thread because we will _join() the thread. */
1653 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1654 result = gst_pad_stop_task (pad);
1664 static GstStateChangeReturn
1665 gst_rtp_jitter_buffer_change_state (GstElement * element,
1666 GstStateChange transition)
1668 GstRtpJitterBuffer *jitterbuffer;
1669 GstRtpJitterBufferPrivate *priv;
1670 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1672 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1673 priv = jitterbuffer->priv;
1675 switch (transition) {
1676 case GST_STATE_CHANGE_NULL_TO_READY:
1678 case GST_STATE_CHANGE_READY_TO_PAUSED:
1680 /* reset negotiated values */
1681 priv->clock_rate = -1;
1682 priv->clock_base = -1;
1683 priv->peer_latency = 0;
1685 /* block until we go to PLAYING */
1686 priv->blocked = TRUE;
1687 priv->timer_running = TRUE;
1688 priv->timer_thread =
1689 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1692 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1694 /* unblock to allow streaming in PLAYING */
1695 priv->blocked = FALSE;
1696 JBUF_SIGNAL_EVENT (priv);
1697 JBUF_SIGNAL_TIMER (priv);
1704 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1706 switch (transition) {
1707 case GST_STATE_CHANGE_READY_TO_PAUSED:
1708 /* we are a live element because we sync to the clock, which we can only
1709 * do in the PLAYING state */
1710 if (ret != GST_STATE_CHANGE_FAILURE)
1711 ret = GST_STATE_CHANGE_NO_PREROLL;
1713 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1715 /* block to stop streaming when PAUSED */
1716 priv->blocked = TRUE;
1717 unschedule_current_timer (jitterbuffer);
1719 if (ret != GST_STATE_CHANGE_FAILURE)
1720 ret = GST_STATE_CHANGE_NO_PREROLL;
1722 case GST_STATE_CHANGE_PAUSED_TO_READY:
1724 gst_buffer_replace (&priv->last_sr, NULL);
1725 priv->timer_running = FALSE;
1726 unschedule_current_timer (jitterbuffer);
1727 JBUF_SIGNAL_TIMER (priv);
1728 JBUF_SIGNAL_QUERY (priv, FALSE);
1730 g_thread_join (priv->timer_thread);
1731 priv->timer_thread = NULL;
1733 case GST_STATE_CHANGE_READY_TO_NULL:
1743 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1746 gboolean ret = TRUE;
1747 GstRtpJitterBuffer *jitterbuffer;
1748 GstRtpJitterBufferPrivate *priv;
1750 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1751 priv = jitterbuffer->priv;
1753 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1755 switch (GST_EVENT_TYPE (event)) {
1756 case GST_EVENT_LATENCY:
1758 GstClockTime latency;
1760 gst_event_parse_latency (event, &latency);
1762 GST_DEBUG_OBJECT (jitterbuffer,
1763 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1766 /* adjust the overall buffer delay to the total pipeline latency in
1767 * buffering mode because if downstream consumes too fast (because of
1768 * large latency or queues, we would start rebuffering again. */
1769 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1770 RTP_JITTER_BUFFER_MODE_BUFFER) {
1771 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1775 ret = gst_pad_push_event (priv->sinkpad, event);
1779 ret = gst_pad_push_event (priv->sinkpad, event);
1786 /* handles and stores the event in the jitterbuffer, must be called with
1789 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1791 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1792 RTPJitterBufferItem *item;
1795 switch (GST_EVENT_TYPE (event)) {
1796 case GST_EVENT_CAPS:
1800 gst_event_parse_caps (event, &caps);
1801 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1804 case GST_EVENT_SEGMENT:
1807 gst_event_copy_segment (event, &segment);
1809 /* we need time for now */
1810 if (segment.format != GST_FORMAT_TIME) {
1811 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1812 gst_event_unref (event);
1814 gst_segment_init (&segment, GST_FORMAT_TIME);
1815 event = gst_event_new_segment (&segment);
1818 priv->segment = segment;
1823 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1830 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1831 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1832 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1834 JBUF_SIGNAL_EVENT (priv);
1840 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1843 gboolean ret = TRUE;
1844 GstRtpJitterBuffer *jitterbuffer;
1845 GstRtpJitterBufferPrivate *priv;
1847 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1848 priv = jitterbuffer->priv;
1850 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1852 switch (GST_EVENT_TYPE (event)) {
1853 case GST_EVENT_FLUSH_START:
1854 ret = gst_pad_push_event (priv->srcpad, event);
1855 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1856 /* wait for the loop to go into PAUSED */
1857 gst_pad_pause_task (priv->srcpad);
1859 case GST_EVENT_FLUSH_STOP:
1860 ret = gst_pad_push_event (priv->srcpad, event);
1862 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1863 GST_PAD_MODE_PUSH, TRUE);
1866 if (GST_EVENT_IS_SERIALIZED (event)) {
1867 /* serialized events go in the queue */
1869 if (priv->srcresult != GST_FLOW_OK) {
1870 /* Errors in sticky event pushing are no problem and ignored here
1871 * as they will cause more meaningful errors during data flow.
1872 * For EOS events, that are not followed by data flow, we still
1873 * return FALSE here though.
1875 if (!GST_EVENT_IS_STICKY (event) ||
1876 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1877 goto out_flow_error;
1879 /* refuse more events on EOS */
1882 ret = queue_event (jitterbuffer, event);
1885 /* non-serialized events are forwarded downstream immediately */
1886 ret = gst_pad_push_event (priv->srcpad, event);
1895 GST_DEBUG_OBJECT (jitterbuffer,
1896 "refusing event, we have a downstream flow error: %s",
1897 gst_flow_get_name (priv->srcresult));
1899 gst_event_unref (event);
1904 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1906 gst_event_unref (event);
1912 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1915 gboolean ret = TRUE;
1916 GstRtpJitterBuffer *jitterbuffer;
1918 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1920 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1922 switch (GST_EVENT_TYPE (event)) {
1923 case GST_EVENT_FLUSH_START:
1924 gst_event_unref (event);
1926 case GST_EVENT_FLUSH_STOP:
1927 gst_event_unref (event);
1930 ret = gst_pad_event_default (pad, parent, event);
1938 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1939 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1940 * GST_FLOW_FLUSHING when the element is shutting down. On success
1941 * GST_FLOW_OK is returned.
1943 static GstFlowReturn
1944 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1948 GValue args[2] = { {0}, {0} };
1952 g_value_init (&args[0], GST_TYPE_ELEMENT);
1953 g_value_set_object (&args[0], jitterbuffer);
1954 g_value_init (&args[1], G_TYPE_UINT);
1955 g_value_set_uint (&args[1], pt);
1957 g_value_init (&ret, GST_TYPE_CAPS);
1958 g_value_set_boxed (&ret, NULL);
1960 JBUF_UNLOCK (jitterbuffer->priv);
1961 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1963 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1965 g_value_unset (&args[0]);
1966 g_value_unset (&args[1]);
1967 caps = (GstCaps *) g_value_dup_boxed (&ret);
1968 g_value_unset (&ret);
1972 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1973 gst_caps_unref (caps);
1975 if (G_UNLIKELY (!res))
1983 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1984 return GST_FLOW_ERROR;
1988 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1989 return GST_FLOW_FLUSHING;
1993 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1994 return GST_FLOW_ERROR;
1998 /* call with jbuf lock held */
2000 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
2002 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2003 GstMessage *message = NULL;
2008 /* Post a buffering message */
2009 if (priv->last_percent != percent) {
2010 priv->last_percent = percent;
2012 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
2013 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
2020 update_offset (GstRtpJitterBuffer * jitterbuffer)
2022 GstRtpJitterBufferPrivate *priv;
2024 priv = jitterbuffer->priv;
2026 if (priv->ts_offset_remainder != 0) {
2027 GST_DEBUG ("adjustment %" G_GUINT64_FORMAT " remain %" G_GINT64_FORMAT
2028 " off %" G_GINT64_FORMAT, priv->max_ts_offset_adjustment,
2029 priv->ts_offset_remainder, priv->ts_offset);
2030 if (ABS (priv->ts_offset_remainder) > priv->max_ts_offset_adjustment) {
2031 if (priv->ts_offset_remainder > 0) {
2032 priv->ts_offset += priv->max_ts_offset_adjustment;
2033 priv->ts_offset_remainder -= priv->max_ts_offset_adjustment;
2035 priv->ts_offset -= priv->max_ts_offset_adjustment;
2036 priv->ts_offset_remainder += priv->max_ts_offset_adjustment;
2039 priv->ts_offset += priv->ts_offset_remainder;
2040 priv->ts_offset_remainder = 0;
2046 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
2048 GstRtpJitterBufferPrivate *priv;
2050 priv = jitterbuffer->priv;
2052 if (timestamp == -1)
2055 /* apply the timestamp offset, this is used for inter stream sync */
2056 timestamp += priv->ts_offset;
2057 /* add the offset, this is used when buffering */
2058 timestamp += priv->out_offset;
2064 timer_queue_new (void)
2068 queue = g_slice_new (TimerQueue);
2069 queue->timers = g_queue_new ();
2070 queue->hashtable = g_hash_table_new (NULL, NULL);
2076 timer_queue_free (TimerQueue * queue)
2081 g_hash_table_destroy (queue->hashtable);
2082 g_queue_free_full (queue->timers, g_free);
2083 g_slice_free (TimerQueue, queue);
2087 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2088 GstClockTime timeout, gboolean lost)
2092 copy = g_memdup (timer, sizeof (*timer));
2093 copy->timeout = timeout;
2094 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2097 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2098 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2099 g_queue_push_tail (queue->timers, copy);
2100 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2104 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2108 test = g_queue_peek_head (queue->timers);
2109 while (test && test->timeout < timeout) {
2110 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2111 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2112 GST_TIME_ARGS (timeout));
2113 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2114 g_free (g_queue_pop_head (queue->timers));
2115 test = g_queue_peek_head (queue->timers);
2120 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2122 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2126 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2128 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2129 TimerData *timer = NULL;
2132 len = priv->timers->len;
2133 for (i = 0; i < len; i++) {
2134 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2135 if (test->seqnum == seqnum) {
2144 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2146 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2148 if (priv->clock_id) {
2149 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2150 gst_clock_id_unschedule (priv->clock_id);
2151 priv->clock_id = NULL;
2156 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2158 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2159 GstClockTime test_timeout;
2161 if ((test_timeout = timer->timeout) == -1)
2164 if (timer->type != TIMER_TYPE_EXPECTED) {
2165 /* add our latency and offset to get output times. */
2166 test_timeout = apply_offset (jitterbuffer, test_timeout);
2167 test_timeout += priv->latency_ns;
2169 return test_timeout;
2173 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2175 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2177 if (priv->clock_id) {
2178 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2180 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2181 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2183 if (timeout == -1 || timeout < priv->timer_timeout)
2184 unschedule_current_timer (jitterbuffer);
2189 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2190 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2191 GstClockTime duration)
2193 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2197 GST_DEBUG_OBJECT (jitterbuffer,
2198 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2199 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2200 GST_TIME_ARGS (delay));
2202 len = priv->timers->len;
2203 g_array_set_size (priv->timers, len + 1);
2204 timer = &g_array_index (priv->timers, TimerData, len);
2207 timer->seqnum = seqnum;
2209 timer->timeout = timeout + delay;
2210 timer->duration = duration;
2211 if (type == TIMER_TYPE_EXPECTED) {
2212 timer->rtx_base = timeout;
2213 timer->rtx_delay = delay;
2214 timer->rtx_retry = 0;
2216 timer->rtx_last = GST_CLOCK_TIME_NONE;
2217 timer->num_rtx_retry = 0;
2218 timer->num_rtx_received = 0;
2219 recalculate_timer (jitterbuffer, timer);
2220 JBUF_SIGNAL_TIMER (priv);
2226 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2227 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2229 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2230 gboolean seqchange, timechange;
2232 GstClockTime new_timeout;
2234 oldseq = timer->seqnum;
2235 new_timeout = timeout + delay;
2236 seqchange = oldseq != seqnum;
2237 timechange = timer->timeout != new_timeout;
2239 if (!seqchange && !timechange) {
2240 GST_DEBUG_OBJECT (jitterbuffer,
2241 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2242 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2246 GST_DEBUG_OBJECT (jitterbuffer,
2247 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2248 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2249 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2251 timer->timeout = new_timeout;
2252 timer->seqnum = seqnum;
2254 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2255 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2256 GST_TIME_ARGS (delay));
2257 timer->rtx_base = timeout;
2258 timer->rtx_delay = delay;
2259 timer->rtx_retry = 0;
2262 timer->num_rtx_retry = 0;
2263 timer->num_rtx_received = 0;
2266 if (priv->clock_id) {
2267 /* we changed the seqnum and there is a timer currently waiting with this
2268 * seqnum, unschedule it */
2269 if (seqchange && priv->timer_seqnum == oldseq)
2270 unschedule_current_timer (jitterbuffer);
2271 /* we changed the time, check if it is earlier than what we are waiting
2272 * for and unschedule if so */
2273 else if (timechange)
2274 recalculate_timer (jitterbuffer, timer);
2279 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2280 guint16 seqnum, GstClockTime timeout)
2284 /* find the seqnum timer */
2285 timer = find_timer (jitterbuffer, seqnum);
2286 if (timer == NULL) {
2287 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2289 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2295 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2297 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2300 if (timer->idx == -1)
2303 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2304 unschedule_current_timer (jitterbuffer);
2307 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2308 g_array_remove_index_fast (priv->timers, idx);
2313 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2315 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2316 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2317 g_array_set_size (priv->timers, 0);
2318 unschedule_current_timer (jitterbuffer);
2321 /* get the extra delay to wait before sending RTX */
2323 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2327 if (priv->rtx_delay == -1) {
2328 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2329 delay = DEFAULT_AUTO_RTX_DELAY;
2331 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2332 * packet spacing is a good margin */
2333 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2336 delay = priv->rtx_delay * GST_MSECOND;
2338 if (priv->rtx_min_delay > 0)
2339 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2344 /* Check if packet with seqnum is already considered definitely lost by being
2345 * part of a "lost timer" for multiple packets */
2347 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2349 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2352 len = priv->timers->len;
2353 for (i = 0; i < len; i++) {
2354 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2355 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2357 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2359 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2360 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2368 /* we just received a packet with seqnum and dts.
2370 * First check for old seqnum that we are still expecting. If the gap with the
2371 * current seqnum is too big, unschedule the timeouts.
2373 * If we have a valid packet spacing estimate we can set a timer for when we
2374 * should receive the next packet.
2375 * If we don't have a valid estimate, we remove any timer we might have
2376 * had for this packet.
2379 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2380 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2381 gboolean is_rtx, TimerData * timer)
2383 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2385 /* go through all timers and unschedule the ones with a large gap */
2386 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2388 len = priv->timers->len;
2389 for (i = 0; i < len; i++) {
2390 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2393 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2395 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2396 test->type, test->seqnum, seqnum, gap);
2398 if (gap > priv->rtx_delay_reorder) {
2399 /* max gap, we exceeded the max reorder distance and we don't expect the
2400 * missing packet to be this reordered */
2401 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2402 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2407 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2408 && priv->do_retransmission && priv->rtx_next_seqnum;
2410 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2411 if (timer->num_rtx_retry > 0) {
2413 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2414 /* don't try to estimate the next seqnum because this is a retransmitted
2415 * packet and it probably did not arrive with the expected packet
2417 do_next_seqnum = FALSE;
2420 if (!is_rtx || timer->num_rtx_retry > 1) {
2421 /* Store timer in order to record stats when/if the retransmitted
2422 * packet arrives. We should also store timer information if we've
2423 * requested retransmission more than once since we may receive
2424 * several retransmitted packets. For accuracy we should update the
2425 * stats also when the redundant retransmitted packets arrives. */
2426 timer_queue_append (priv->rtx_stats_timers, timer,
2427 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2432 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2433 GstClockTime expected, delay;
2435 /* calculate expected arrival time of the next seqnum */
2436 expected = pts + priv->packet_spacing;
2438 delay = get_rtx_delay (priv);
2440 /* and update/install timer for next seqnum */
2441 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2442 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2443 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2444 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2445 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2448 timer->type = TIMER_TYPE_EXPECTED;
2449 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2452 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2453 expected, delay, priv->packet_spacing);
2455 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2456 /* if we had a timer, remove it, we don't know when to expect the next
2458 remove_timer (jitterbuffer, timer);
2463 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2466 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2468 /* we need consecutive seqnums with a different
2469 * rtptime to estimate the packet spacing. */
2470 if (priv->ips_rtptime != rtptime) {
2471 /* rtptime changed, check pts diff */
2472 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2473 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2474 GstClockTime old_packet_spacing = priv->packet_spacing;
2476 /* Biased towards bigger packet spacings to prevent
2477 * too many unneeded retransmission requests for next
2478 * packets that just arrive a little later than we would
2480 if (old_packet_spacing > new_packet_spacing)
2481 priv->packet_spacing =
2482 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2483 else if (old_packet_spacing > 0)
2484 priv->packet_spacing =
2485 (3 * new_packet_spacing + old_packet_spacing) / 4;
2487 priv->packet_spacing = new_packet_spacing;
2489 GST_DEBUG_OBJECT (jitterbuffer,
2490 "new packet spacing %" GST_TIME_FORMAT
2491 " old packet spacing %" GST_TIME_FORMAT
2492 " combined to %" GST_TIME_FORMAT,
2493 GST_TIME_ARGS (new_packet_spacing),
2494 GST_TIME_ARGS (old_packet_spacing),
2495 GST_TIME_ARGS (priv->packet_spacing));
2497 priv->ips_rtptime = rtptime;
2498 priv->ips_pts = pts;
2503 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2504 guint16 seqnum, GstClockTime pts, gint gap)
2506 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2507 GstClockTime duration, expected_pts, delay;
2509 gboolean equidistant = priv->equidistant > 0;
2511 GST_DEBUG_OBJECT (jitterbuffer,
2512 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2513 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2515 if (pts == GST_CLOCK_TIME_NONE) {
2516 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2521 GstClockTime total_duration;
2522 /* the total duration spanned by the missing packets */
2523 if (pts >= priv->last_in_pts)
2524 total_duration = pts - priv->last_in_pts;
2528 /* interpolate between the current time and the last time based on
2529 * number of packets we are missing, this is the estimated duration
2530 * for the missing packet based on equidistant packet spacing. */
2531 duration = total_duration / (gap + 1);
2533 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2534 GST_TIME_ARGS (duration));
2536 if (total_duration > priv->latency_ns) {
2537 GstClockTime gap_time;
2541 GstClockTime gap_dur = gap * duration;
2542 if (gap_dur > priv->latency_ns)
2543 gap_time = gap_dur - priv->latency_ns;
2546 lost_packets = gap_time / duration;
2548 gap_time = total_duration - priv->latency_ns;
2552 /* too many lost packets, some of the missing packets are already
2553 * too late and we can generate lost packet events for them. */
2554 GST_INFO_OBJECT (jitterbuffer,
2555 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2556 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2557 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2558 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2559 GST_TIME_ARGS (gap_time));
2561 /* this timer will fire immediately and the lost event will be pushed from
2562 * the timer thread */
2563 if (lost_packets > 0) {
2564 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2565 priv->last_in_pts + duration, 0, gap_time);
2566 expected += lost_packets;
2567 priv->last_in_pts += gap_time;
2571 expected_pts = priv->last_in_pts + duration;
2573 /* If we cannot assume equidistant packet spacing, the only thing we now
2574 * for sure is that the missing packets have expected pts not later than
2575 * the last received pts. */
2582 if (priv->do_retransmission) {
2583 TimerData *timer = find_timer (jitterbuffer, expected);
2585 type = TIMER_TYPE_EXPECTED;
2586 delay = get_rtx_delay (priv);
2588 /* if we had a timer for the first missing packet, update it. */
2589 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2590 GstClockTime timeout = timer->timeout;
2592 timer->duration = duration;
2593 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2594 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2598 expected_pts += duration;
2601 type = TIMER_TYPE_LOST;
2604 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2605 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2606 expected_pts += duration;
2612 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2616 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2617 GstRtpJitterBufferPrivate *priv;
2619 priv = jitterbuffer->priv;
2621 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2624 if (priv->last_dts != -1)
2625 dtsdiff = dts - priv->last_dts;
2629 if (priv->last_rtptime != -1)
2630 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2634 /* Guess whether stream currently uses equidistant packet spacing. If we
2635 * often see identical timestamps it means the packets are not
2637 if (rtptime == priv->last_rtptime)
2638 priv->equidistant -= 2;
2640 priv->equidistant += 1;
2641 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2643 priv->last_dts = dts;
2644 priv->last_rtptime = rtptime;
2648 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2651 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2653 diff = ABS (dtsdiff - rtpdiffns);
2655 /* jitter is stored in nanoseconds */
2656 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2658 GST_LOG_OBJECT (jitterbuffer,
2659 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2660 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2661 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2662 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2669 GST_DEBUG_OBJECT (jitterbuffer,
2670 "no dts or no clock-rate, can't calculate jitter");
2676 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2678 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2679 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2682 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2683 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2684 gst_rtp_buffer_unmap (&rtp_a);
2686 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2687 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2688 gst_rtp_buffer_unmap (&rtp_b);
2690 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2694 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2695 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2697 GstRtpJitterBufferPrivate *priv;
2698 guint gap_packets_length;
2699 gboolean reset = FALSE;
2700 gboolean future = gap > 0;
2702 priv = jitterbuffer->priv;
2704 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2706 guint32 prev_gap_seq = -1;
2707 gboolean all_consecutive = TRUE;
2709 g_queue_insert_sorted (&priv->gap_packets, buffer,
2710 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2712 for (l = priv->gap_packets.head; l; l = l->next) {
2713 GstBuffer *gap_buffer = l->data;
2714 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2717 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2719 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2721 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2722 if (prev_gap_seq == -1)
2723 prev_gap_seq = gap_seq;
2724 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2725 all_consecutive = FALSE;
2727 prev_gap_seq = gap_seq;
2729 gst_rtp_buffer_unmap (&gap_rtp);
2730 if (!all_consecutive)
2734 if (all_consecutive && gap_packets_length > 3) {
2735 GST_DEBUG_OBJECT (jitterbuffer,
2736 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2737 (future ? "new" : "old"), gap,
2738 (future ? max_dropout : -max_misorder));
2740 } else if (!all_consecutive) {
2741 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2742 g_queue_clear (&priv->gap_packets);
2743 GST_DEBUG_OBJECT (jitterbuffer,
2744 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2745 (future ? "new" : "old"), gap,
2746 (future ? max_dropout : -max_misorder));
2749 GST_DEBUG_OBJECT (jitterbuffer,
2750 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2751 (future ? "new" : "old"), gap,
2752 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2756 GST_DEBUG_OBJECT (jitterbuffer,
2757 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2758 gap, -max_misorder);
2759 g_queue_push_tail (&priv->gap_packets, buffer);
2767 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2769 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2770 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2773 GstClockTime base_time =
2774 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2775 GstClockTime clock_time = gst_clock_get_time (clock);
2777 if (clock_time > base_time)
2778 running_time = clock_time - base_time;
2782 gst_object_unref (clock);
2785 return running_time;
2788 static GstFlowReturn
2789 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2790 GstPad * pad, GstObject * parent, guint16 seqnum)
2792 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2793 GstFlowReturn ret = GST_FLOW_OK;
2794 GList *events = NULL, *l;
2798 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2799 rtp_jitter_buffer_flush (priv->jbuf,
2800 (GFunc) free_item_and_retain_events, &events);
2801 rtp_jitter_buffer_reset_skew (priv->jbuf);
2802 remove_all_timers (jitterbuffer);
2803 priv->discont = TRUE;
2804 priv->last_popped_seqnum = -1;
2806 if (priv->gap_packets.head) {
2807 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2808 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2810 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2811 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2812 gst_rtp_buffer_unmap (&gap_rtp);
2814 priv->next_seqnum = seqnum;
2817 priv->last_in_pts = -1;
2818 priv->next_in_seqnum = -1;
2820 /* Insert all sticky events again in order, otherwise we would
2821 * potentially loose STREAM_START, CAPS or SEGMENT events
2823 events = g_list_reverse (events);
2824 for (l = events; l; l = l->next) {
2825 RTPJitterBufferItem *item;
2827 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2828 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2830 g_list_free (events);
2832 JBUF_SIGNAL_EVENT (priv);
2834 /* reset spacing estimation when gap */
2835 priv->ips_rtptime = -1;
2836 priv->ips_pts = GST_CLOCK_TIME_NONE;
2838 buffers = g_list_copy (priv->gap_packets.head);
2839 g_queue_clear (&priv->gap_packets);
2841 priv->ips_rtptime = -1;
2842 priv->ips_pts = GST_CLOCK_TIME_NONE;
2843 JBUF_UNLOCK (jitterbuffer->priv);
2845 for (l = buffers; l; l = l->next) {
2846 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2848 if (ret != GST_FLOW_OK) {
2853 for (; l; l = l->next)
2854 gst_buffer_unref (l->data);
2855 g_list_free (buffers);
2861 gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
2863 GstRtpJitterBufferPrivate *priv;
2864 RTPJitterBufferItem *item;
2867 priv = jitterbuffer->priv;
2869 if (priv->faststart_min_packets == 0)
2872 item = rtp_jitter_buffer_peek (priv->jbuf);
2876 timer = find_timer (jitterbuffer, item->seqnum);
2877 if (!timer || timer->type != TIMER_TYPE_DEADLINE)
2880 if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
2881 priv->faststart_min_packets)) {
2882 GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
2883 priv->faststart_min_packets);
2884 timer->timeout = -1;
2891 static GstFlowReturn
2892 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2895 GstRtpJitterBuffer *jitterbuffer;
2896 GstRtpJitterBufferPrivate *priv;
2898 guint32 expected, rtptime;
2899 GstFlowReturn ret = GST_FLOW_OK;
2900 GstClockTime dts, pts;
2905 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2906 gboolean do_next_seqnum = FALSE;
2907 RTPJitterBufferItem *item;
2908 GstMessage *msg = NULL;
2909 gboolean estimated_dts = FALSE;
2910 gint32 packet_rate, max_dropout, max_misorder;
2911 TimerData *timer = NULL;
2913 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2915 priv = jitterbuffer->priv;
2917 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2918 goto invalid_buffer;
2920 pt = gst_rtp_buffer_get_payload_type (&rtp);
2921 seqnum = gst_rtp_buffer_get_seq (&rtp);
2922 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2923 gst_rtp_buffer_unmap (&rtp);
2925 /* make sure we have PTS and DTS set */
2926 pts = GST_BUFFER_PTS (buffer);
2927 dts = GST_BUFFER_DTS (buffer);
2934 /* If we have no DTS here, i.e. no capture time, get one from the
2935 * clock now to have something to calculate with in the future. */
2936 dts = get_current_running_time (jitterbuffer);
2939 /* Remember that we estimated the DTS if we are running already
2940 * and this is not our first packet (or first packet after a reset).
2941 * If it's the first packet, we somehow must generate a timestamp for
2942 * everything, otherwise we can't calculate any times
2944 estimated_dts = (priv->next_in_seqnum != -1);
2946 /* take the DTS of the buffer. This is the time when the packet was
2947 * received and is used to calculate jitter and clock skew. We will adjust
2948 * this DTS with the smoothed value after processing it in the
2949 * jitterbuffer and assign it as the PTS. */
2950 /* bring to running time */
2951 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2954 GST_DEBUG_OBJECT (jitterbuffer,
2955 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2956 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2957 GST_BUFFER_IS_RETRANSMISSION (buffer));
2959 JBUF_LOCK_CHECK (priv, out_flushing);
2961 if (G_UNLIKELY (priv->last_pt != pt)) {
2964 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2968 /* reset clock-rate so that we get a new one */
2969 priv->clock_rate = -1;
2971 /* Try to get the clock-rate from the caps first if we can. If there are no
2972 * caps we must fire the signal to get the clock-rate. */
2973 if ((caps = gst_pad_get_current_caps (pad))) {
2974 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2975 gst_caps_unref (caps);
2979 if (G_UNLIKELY (priv->clock_rate == -1)) {
2980 /* no clock rate given on the caps, try to get one with the signal */
2981 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2982 pt) == GST_FLOW_FLUSHING)
2985 if (G_UNLIKELY (priv->clock_rate == -1))
2988 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2991 /* don't accept more data on EOS */
2992 if (G_UNLIKELY (priv->eos))
2995 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2996 calculate_jitter (jitterbuffer, dts, rtptime);
2998 if (priv->seqnum_base != -1) {
3001 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
3004 GST_DEBUG_OBJECT (jitterbuffer,
3005 "packet seqnum #%d before seqnum-base #%d", seqnum,
3007 gst_buffer_unref (buffer);
3009 } else if (gap > 16384) {
3010 /* From now on don't compare against the seqnum base anymore as
3011 * at some point in the future we will wrap around and also that
3012 * much reordering is very unlikely */
3013 priv->seqnum_base = -1;
3017 expected = priv->next_in_seqnum;
3020 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
3022 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
3023 priv->max_dropout_time);
3025 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
3026 priv->max_misorder_time);
3027 GST_TRACE_OBJECT (jitterbuffer,
3028 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
3029 max_dropout, max_misorder);
3031 /* now check against our expected seqnum */
3032 if (G_UNLIKELY (expected == -1)) {
3033 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3035 /* calculate a pts based on rtptime and arrival time (dts) */
3037 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3038 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3040 /* we don't know what the next_in_seqnum should be, wait for the last
3041 * possible moment to push this buffer, maybe we get an earlier seqnum
3043 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
3045 do_next_seqnum = TRUE;
3046 /* take rtptime and pts to calculate packet spacing */
3047 priv->ips_rtptime = rtptime;
3048 priv->ips_pts = pts;
3052 /* now calculate gap */
3053 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
3054 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
3055 expected, seqnum, gap);
3057 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
3058 /* If we have timers for more than RTP_MAX_DROPOUT packets
3059 * pending this means that we have a huge gap overall. We can
3060 * reset the jitterbuffer at this point because there's
3061 * just too much data missing to be able to do anything
3062 * sensible with the past data. Just try again from the
3064 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
3065 priv->timers->len, max_dropout);
3066 gst_buffer_unref (buffer);
3067 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3070 /* Special handling of large gaps */
3071 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
3072 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
3073 gap, max_dropout, max_misorder);
3075 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3077 GST_DEBUG_OBJECT (jitterbuffer,
3078 "Had big gap, waiting for more consecutive packets");
3083 /* We had no huge gap, let's drop all the gap packets */
3084 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
3085 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3086 g_queue_clear (&priv->gap_packets);
3088 /* calculate a pts based on rtptime and arrival time (dts) */
3089 /* If we estimated the DTS, don't consider it in the clock skew calculations */
3091 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3092 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3094 if (G_LIKELY (gap == 0)) {
3095 /* packet is expected */
3096 calculate_packet_spacing (jitterbuffer, rtptime, pts);
3097 do_next_seqnum = TRUE;
3102 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
3103 /* fill in the gap with EXPECTED timers */
3104 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
3105 do_next_seqnum = TRUE;
3107 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3108 do_next_seqnum = FALSE;
3111 /* reset spacing estimation when gap */
3112 priv->ips_rtptime = -1;
3113 priv->ips_pts = GST_CLOCK_TIME_NONE;
3117 if (do_next_seqnum) {
3118 priv->last_in_pts = pts;
3119 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3122 timer = find_timer (jitterbuffer, seqnum);
3123 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3125 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3127 timer->num_rtx_received++;
3130 /* let's check if this buffer is too late, we can only accept packets with
3131 * bigger seqnum than the one we last pushed. */
3132 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3135 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3137 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3138 if (G_UNLIKELY (gap <= 0)) {
3139 if (priv->do_retransmission) {
3140 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3141 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3142 /* Only count the retranmitted packet too late if it has been
3143 * considered lost. If the original packet arrived before the
3144 * retransmitted we just count it as a duplicate. */
3145 if (timer->type != TIMER_TYPE_LOST)
3153 if (already_lost (jitterbuffer, seqnum))
3156 /* let's drop oldest packet if the queue is already full and drop-on-latency
3157 * is set. We can only do this when there actually is a latency. When no
3158 * latency is set, we just pump it in the queue and let the other end push it
3159 * out as fast as possible. */
3160 if (priv->latency_ms && priv->drop_on_latency) {
3162 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3164 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3165 RTPJitterBufferItem *old_item;
3167 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3169 if (IS_DROPABLE (old_item)) {
3170 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3171 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3173 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3174 free_item (old_item);
3176 /* we might have removed some head buffers, signal the pushing thread to
3177 * see if it can push now */
3178 JBUF_SIGNAL_EVENT (priv);
3182 /* If we estimated the DTS, don't consider it in the clock skew calculations
3183 * later. The code above always sets dts to pts or the other way around if
3184 * any of those is valid in the buffer, so we know that if we estimated the
3185 * dts that both are unknown */
3188 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3189 pts, seqnum, 1, rtptime);
3191 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3193 /* now insert the packet into the queue in sorted order. This function returns
3194 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3195 * have a duplicate. */
3196 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3198 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3199 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3203 /* Trigger fast start if needed */
3204 if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
3208 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3209 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3211 /* we had an unhandled SR, handle it now */
3213 do_handle_sync (jitterbuffer);
3215 if (G_UNLIKELY (head)) {
3216 /* signal addition of new buffer when the _loop is waiting. */
3217 if (G_LIKELY (priv->active))
3218 JBUF_SIGNAL_EVENT (priv);
3220 /* let's unschedule and unblock any waiting buffers. We only want to do this
3221 * when the head buffer changed */
3222 if (G_UNLIKELY (priv->clock_id)) {
3223 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3224 unschedule_current_timer (jitterbuffer);
3228 GST_DEBUG_OBJECT (jitterbuffer,
3229 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3230 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3232 msg = check_buffering_percent (jitterbuffer, percent);
3238 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3245 /* this is not fatal but should be filtered earlier */
3246 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3247 ("Received invalid RTP payload, dropping"));
3248 gst_buffer_unref (buffer);
3253 GST_WARNING_OBJECT (jitterbuffer,
3254 "No clock-rate in caps!, dropping buffer");
3255 gst_buffer_unref (buffer);
3260 ret = priv->srcresult;
3261 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3262 gst_buffer_unref (buffer);
3268 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3269 gst_buffer_unref (buffer);
3274 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3275 " popped, dropping", seqnum, priv->last_popped_seqnum);
3277 gst_buffer_unref (buffer);
3282 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3283 "considered lost", seqnum);
3285 gst_buffer_unref (buffer);
3290 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3292 priv->num_duplicates++;
3298 GST_DEBUG_OBJECT (jitterbuffer,
3299 "Duplicate RTX packet #%d detected, dropping", seqnum);
3300 priv->num_duplicates++;
3301 gst_buffer_unref (buffer);
3306 /* FIXME: hopefully we can do something more efficient here, especially when
3307 * all packets are in order and/or outside of the currently cached range.
3308 * Still worthwhile to have it, avoids taking/releasing object lock and pad
3309 * stream lock for every single buffer in the default chain_list fallback. */
3310 static GstFlowReturn
3311 gst_rtp_jitter_buffer_chain_list (GstPad * pad, GstObject * parent,
3312 GstBufferList * buffer_list)
3314 GstFlowReturn flow_ret = GST_FLOW_OK;
3317 n = gst_buffer_list_length (buffer_list);
3318 for (i = 0; i < n; ++i) {
3319 GstBuffer *buf = gst_buffer_list_get (buffer_list, i);
3321 flow_ret = gst_rtp_jitter_buffer_chain (pad, parent, gst_buffer_ref (buf));
3323 if (flow_ret != GST_FLOW_OK)
3326 gst_buffer_list_unref (buffer_list);
3332 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3334 guint64 ext_time, elapsed;
3336 GstRtpJitterBufferPrivate *priv;
3338 priv = jitterbuffer->priv;
3339 rtp_time = item->rtptime;
3341 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3342 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3344 ext_time = priv->ext_timestamp;
3345 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3346 if (ext_time < priv->ext_timestamp) {
3347 ext_time = priv->ext_timestamp;
3349 priv->ext_timestamp = ext_time;
3352 if (ext_time > priv->clock_base)
3353 elapsed = ext_time - priv->clock_base;
3357 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3362 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3363 RTPJitterBufferItem * item)
3365 guint64 total, elapsed, left, estimated;
3366 GstClockTime out_time;
3367 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3369 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3370 || priv->clock_base == -1 || priv->clock_rate <= 0)
3373 /* compute the elapsed time */
3374 elapsed = compute_elapsed (jitterbuffer, item);
3376 /* do nothing if elapsed time doesn't increment */
3377 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3380 priv->last_elapsed = elapsed;
3382 /* this is the total time we need to play */
3383 total = priv->npt_stop - priv->npt_start;
3384 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3385 GST_TIME_ARGS (total));
3387 /* this is how much time there is left */
3388 if (total > elapsed)
3389 left = total - elapsed;
3393 /* if we have less time left that the size of the buffer, we will not
3394 * be able to keep it filled, disabled buffering then */
3395 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3396 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3397 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3398 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3401 /* this is the current time as running-time */
3402 out_time = item->pts;
3405 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3407 /* if there is almost nothing left,
3408 * we may never advance enough to end up in the above case */
3409 if (total < GST_SECOND)
3410 estimated = GST_SECOND;
3414 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3415 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3417 if (estimated != -1 && priv->estimated_eos != estimated) {
3418 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3419 priv->estimated_eos = estimated;
3423 /* take a buffer from the queue and push it */
3424 static GstFlowReturn
3425 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3427 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3428 GstFlowReturn result = GST_FLOW_OK;
3429 RTPJitterBufferItem *item;
3430 GstBuffer *outbuf = NULL;
3431 GstEvent *outevent = NULL;
3432 GstQuery *outquery = NULL;
3433 GstClockTime dts, pts;
3435 gboolean do_push = TRUE;
3439 /* when we get here we are ready to pop and push the buffer */
3440 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3444 case ITEM_TYPE_BUFFER:
3446 /* we need to make writable to change the flags and timestamps */
3447 outbuf = gst_buffer_make_writable (item->data);
3449 if (G_UNLIKELY (priv->discont)) {
3450 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3451 * into the jitterbuffer so we can modify now. */
3452 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3453 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3454 priv->discont = FALSE;
3456 if (G_UNLIKELY (priv->ts_discont)) {
3457 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3458 priv->ts_discont = FALSE;
3462 gst_segment_position_from_running_time (&priv->segment,
3463 GST_FORMAT_TIME, item->dts);
3465 gst_segment_position_from_running_time (&priv->segment,
3466 GST_FORMAT_TIME, item->pts);
3468 /* if this is a new frame, check if ts_offset needs to be updated */
3469 if (pts != priv->last_pts) {
3470 update_offset (jitterbuffer);
3473 /* apply timestamp with offset to buffer now */
3474 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3475 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3477 /* update the elapsed time when we need to check against the npt stop time. */
3478 update_estimated_eos (jitterbuffer, item);
3480 /* verify that an offset has not caused time stamps to go backwards, if so
3481 * handle by reusing the previous timestamp */
3482 if (priv->last_out_time != GST_CLOCK_TIME_NONE &&
3483 GST_BUFFER_PTS (outbuf) < priv->last_out_time) {
3484 GST_DEBUG_OBJECT (jitterbuffer, "buffer PTS %" GST_TIME_FORMAT
3485 " older than preceding PTS %" GST_TIME_FORMAT
3486 " adjusting to %" GST_TIME_FORMAT,
3487 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
3488 GST_TIME_ARGS (priv->last_out_time),
3489 GST_TIME_ARGS (priv->last_out_time));
3490 GST_BUFFER_PTS (outbuf) = priv->last_out_time;
3493 priv->last_pts = pts;
3494 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3496 case ITEM_TYPE_LOST:
3497 priv->discont = TRUE;
3501 case ITEM_TYPE_EVENT:
3502 outevent = item->data;
3504 case ITEM_TYPE_QUERY:
3505 outquery = item->data;
3509 /* now we are ready to push the buffer. Save the seqnum and release the lock
3510 * so the other end can push stuff in the queue again. */
3512 priv->last_popped_seqnum = seqnum;
3513 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3515 msg = check_buffering_percent (jitterbuffer, percent);
3522 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3525 case ITEM_TYPE_BUFFER:
3527 GST_DEBUG_OBJECT (jitterbuffer,
3528 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3529 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3530 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3532 result = gst_pad_push (priv->srcpad, outbuf);
3534 JBUF_LOCK_CHECK (priv, out_flushing);
3536 case ITEM_TYPE_LOST:
3537 case ITEM_TYPE_EVENT:
3538 /* We got not enough consecutive packets with a huge gap, we can
3539 * as well just drop them here now on EOS */
3540 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3541 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3542 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3543 g_queue_clear (&priv->gap_packets);
3546 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3547 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3550 gst_pad_push_event (priv->srcpad, outevent);
3552 gst_event_unref (outevent);
3554 result = GST_FLOW_OK;
3556 JBUF_LOCK_CHECK (priv, out_flushing);
3558 case ITEM_TYPE_QUERY:
3562 res = gst_pad_peer_query (priv->srcpad, outquery);
3564 JBUF_LOCK_CHECK (priv, out_flushing);
3565 result = GST_FLOW_OK;
3566 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3567 JBUF_SIGNAL_QUERY (priv, res);
3576 return priv->srcresult;
3580 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3582 /* Peek a buffer and compare the seqnum to the expected seqnum.
3583 * If all is fine, the buffer is pushed.
3584 * If something is wrong, we wait for some event
3586 static GstFlowReturn
3587 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3589 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3590 GstFlowReturn result;
3591 RTPJitterBufferItem *item;
3593 guint32 next_seqnum;
3595 /* only push buffers when PLAYING and active and not buffering */
3596 if (priv->blocked || !priv->active ||
3597 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3598 return GST_FLOW_WAIT;
3601 /* peek a buffer, we're just looking at the sequence number.
3602 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3603 * wait for a timeout or something to change.
3604 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3605 item = rtp_jitter_buffer_peek (priv->jbuf);
3610 /* get the seqnum and the next expected seqnum */
3611 seqnum = item->seqnum;
3613 return pop_and_push_next (jitterbuffer, seqnum);
3616 next_seqnum = priv->next_seqnum;
3618 /* get the gap between this and the previous packet. If we don't know the
3619 * previous packet seqnum assume no gap. */
3620 if (G_UNLIKELY (next_seqnum == -1)) {
3621 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3622 /* we don't know what the next_seqnum should be, the chain function should
3623 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3624 * fires, so wait for that */
3625 result = GST_FLOW_WAIT;
3627 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3629 if (G_LIKELY (gap == 0)) {
3630 /* no missing packet, pop and push */
3631 result = pop_and_push_next (jitterbuffer, seqnum);
3632 } else if (G_UNLIKELY (gap < 0)) {
3633 /* if we have a packet that we already pushed or considered dropped, pop it
3634 * off and get the next packet */
3635 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3636 seqnum, next_seqnum);
3637 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3639 result = GST_FLOW_OK;
3641 /* the chain function has scheduled timers to request retransmission or
3642 * when to consider the packet lost, wait for that */
3643 GST_DEBUG_OBJECT (jitterbuffer,
3644 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3645 next_seqnum, seqnum, gap);
3646 result = GST_FLOW_WAIT;
3654 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3656 return GST_FLOW_EOS;
3658 return GST_FLOW_WAIT;
3664 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3666 GstClockTime rtx_retry_timeout;
3667 GstClockTime rtx_min_retry_timeout;
3669 if (priv->rtx_retry_timeout == -1) {
3670 if (priv->avg_rtx_rtt == 0)
3671 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3673 /* we want to ask for a retransmission after we waited for a
3674 * complete RTT and the additional jitter */
3675 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3677 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3679 /* make sure we don't retry too often. On very low latency networks,
3680 * the RTT and jitter can be very low. */
3681 if (priv->rtx_min_retry_timeout == -1) {
3682 rtx_min_retry_timeout = priv->packet_spacing;
3684 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3686 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3688 return rtx_retry_timeout;
3692 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3693 GstClockTime rtx_retry_timeout)
3695 GstClockTime rtx_retry_period;
3697 if (priv->rtx_retry_period == -1) {
3698 /* we retry up to the configured jitterbuffer size but leaving some
3699 * room for the retransmission to arrive in time */
3700 if (rtx_retry_timeout > priv->latency_ns) {
3701 rtx_retry_period = 0;
3703 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3706 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3708 return rtx_retry_period;
3712 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3713 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3714 3. For very large measurements (> avg * 2), consider them "outliers"
3715 and count them a lot less (1/48th)
3718 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3722 if (priv->avg_rtx_rtt == 0) {
3723 priv->avg_rtx_rtt = rtt;
3727 if (rtt > 2 * priv->avg_rtx_rtt)
3729 else if (rtt > priv->avg_rtx_rtt)
3734 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3738 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3739 GstClockTime dts, gboolean success)
3741 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3745 /* we scheduled a retry for this packet and now we have it */
3746 priv->num_rtx_success++;
3747 /* all the previous retry attempts failed */
3748 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3750 /* All retries failed or was too late */
3751 priv->num_rtx_failed += timer->num_rtx_retry;
3754 /* number of retries before (hopefully) receiving the packet */
3755 if (priv->avg_rtx_num == 0.0)
3756 priv->avg_rtx_num = timer->num_rtx_retry;
3758 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3760 /* Calculate the delay between retransmission request and receiving this
3761 * packet. We have a valid delay if and only if this packet is a response to
3762 * our last request. If not we don't know if this is a response to an
3763 * earlier request and delay could be way off. For RTT is more important
3764 * with correct values than to update for every packet. */
3765 if (timer->num_rtx_retry == timer->num_rtx_received &&
3766 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3767 delay = dts - timer->rtx_last;
3768 update_avg_rtx_rtt (priv, delay);
3773 GST_LOG_OBJECT (jitterbuffer,
3774 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3775 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3776 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3777 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3778 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3779 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3780 GST_TIME_ARGS (priv->avg_rtx_rtt));
3783 /* the timeout for when we expected a packet expired */
3785 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3788 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3790 guint delay, delay_ms, avg_rtx_rtt_ms;
3791 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3792 guint rtx_deadline_ms;
3793 GstClockTime rtx_retry_period;
3794 GstClockTime rtx_retry_timeout;
3797 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3798 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3800 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3801 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3803 delay = timer->rtx_delay + timer->rtx_retry;
3805 delay_ms = GST_TIME_AS_MSECONDS (delay);
3806 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3807 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3808 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3810 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3812 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3813 gst_structure_new ("GstRTPRetransmissionRequest",
3814 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3815 "running-time", G_TYPE_UINT64, timer->rtx_base,
3816 "delay", G_TYPE_UINT, delay_ms,
3817 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3818 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3819 "period", G_TYPE_UINT, rtx_retry_period_ms,
3820 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3821 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3822 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3823 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3825 priv->num_rtx_requests++;
3826 timer->num_rtx_retry++;
3828 GST_OBJECT_LOCK (jitterbuffer);
3829 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3830 timer->rtx_last = gst_clock_get_time (clock);
3831 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3833 timer->rtx_last = now;
3835 GST_OBJECT_UNLOCK (jitterbuffer);
3837 /* calculate the timeout for the next retransmission attempt */
3838 timer->rtx_retry += rtx_retry_timeout;
3839 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3840 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3841 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3842 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3843 if ((priv->rtx_max_retries != -1
3844 && timer->num_rtx_retry >= priv->rtx_max_retries)
3845 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3846 || (timer->rtx_base + rtx_retry_period < now)) {
3847 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3848 /* too many retransmission request, we now convert the timer
3849 * to a lost timer, leave the num_rtx_retry as it is for stats */
3850 timer->type = TIMER_TYPE_LOST;
3851 timer->rtx_delay = 0;
3852 timer->rtx_retry = 0;
3854 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3855 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3858 gst_pad_push_event (priv->sinkpad, event);
3864 /* a packet is lost */
3866 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3869 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3870 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3872 GstEvent *event = NULL;
3873 RTPJitterBufferItem *item;
3875 seqnum = timer->seqnum;
3876 lost_packets = MAX (timer->num, 1);
3877 num_rtx_retry = timer->num_rtx_retry;
3879 /* we had a gap and thus we lost some packets. Create an event for this. */
3880 if (lost_packets > 1)
3881 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3882 seqnum + lost_packets - 1);
3884 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3886 priv->num_lost += lost_packets;
3887 priv->num_rtx_failed += num_rtx_retry;
3889 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3891 /* we now only accept seqnum bigger than this */
3892 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3893 priv->next_in_seqnum = next_in_seqnum;
3894 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3897 /* Avoid creating events if we don't need it. Note that we still need to create
3898 * the lost *ITEM* since it will be used to notify the outgoing thread of
3899 * lost items (so that we can set discont flags and such) */
3900 if (priv->do_lost) {
3901 GstClockTime duration, timestamp;
3902 /* create paket lost event */
3903 timestamp = apply_offset (jitterbuffer, timer->timeout);
3904 duration = timer->duration;
3905 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3906 duration = priv->packet_spacing;
3907 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3908 gst_structure_new ("GstRTPPacketLost",
3909 "seqnum", G_TYPE_UINT, (guint) seqnum,
3910 "timestamp", G_TYPE_UINT64, timestamp,
3911 "duration", G_TYPE_UINT64, duration,
3912 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3914 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3915 if (!rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL))
3919 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3920 /* Store info to update stats if the packet arrives too late */
3921 timer_queue_append (priv->rtx_stats_timers, timer,
3922 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3924 remove_timer (jitterbuffer, timer);
3927 JBUF_SIGNAL_EVENT (priv);
3933 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3936 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3938 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3939 remove_timer (jitterbuffer, timer);
3941 /* there was no EOS in the buffer, put one in there now */
3942 queue_event (jitterbuffer, gst_event_new_eos ());
3944 JBUF_SIGNAL_EVENT (priv);
3950 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3953 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3955 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3957 /* timer seqnum might have been obsoleted by caps seqnum-base,
3958 * only mess with current ongoing seqnum if still unknown */
3959 if (priv->next_seqnum == -1)
3960 priv->next_seqnum = timer->seqnum;
3961 remove_timer (jitterbuffer, timer);
3962 JBUF_SIGNAL_EVENT (priv);
3968 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3971 gboolean removed = FALSE;
3973 switch (timer->type) {
3974 case TIMER_TYPE_EXPECTED:
3975 removed = do_expected_timeout (jitterbuffer, timer, now);
3977 case TIMER_TYPE_LOST:
3978 removed = do_lost_timeout (jitterbuffer, timer, now);
3980 case TIMER_TYPE_DEADLINE:
3981 removed = do_deadline_timeout (jitterbuffer, timer, now);
3983 case TIMER_TYPE_EOS:
3984 removed = do_eos_timeout (jitterbuffer, timer, now);
3990 /* called when we need to wait for the next timeout.
3992 * We loop over the array of recorded timeouts and wait for the earliest one.
3993 * When it timed out, do the logic associated with the timer.
3995 * If there are no timers, we wait on a gcond until something new happens.
3998 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
4000 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
4001 GstClockTime now = 0;
4004 while (priv->timer_running) {
4005 TimerData *timer = NULL;
4006 GstClockTime timer_timeout = -1;
4009 /* If we have a clock, update "now" now with the very
4010 * latest running time we have. If timers are unscheduled below we
4011 * otherwise wouldn't update now (it's only updated when timers
4012 * expire), and also for the very first loop iteration now would
4013 * otherwise always be 0
4015 GST_OBJECT_LOCK (jitterbuffer);
4016 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
4018 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
4019 GST_ELEMENT_CAST (jitterbuffer)->base_time;
4021 GST_OBJECT_UNLOCK (jitterbuffer);
4023 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
4024 GST_TIME_ARGS (now));
4026 /* Clear expired rtx-stats timers */
4027 if (priv->do_retransmission)
4028 timer_queue_clear_until (priv->rtx_stats_timers, now);
4030 /* Iterate "normal" timers */
4031 len = priv->timers->len;
4032 for (i = 0; i < len;) {
4033 TimerData *test = &g_array_index (priv->timers, TimerData, i);
4034 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
4035 gboolean save_best = FALSE;
4037 GST_DEBUG_OBJECT (jitterbuffer,
4038 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
4039 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
4040 GST_STIME_ARGS ((gint64) (test_timeout - now)));
4042 /* Weed out anything too late */
4043 if (test->type == TIMER_TYPE_LOST &&
4044 (test_timeout == -1 || test_timeout <= now)) {
4045 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
4046 do_lost_timeout (jitterbuffer, test, now);
4047 if (!priv->timer_running)
4049 /* We don't move the iterator forward since we just removed the current entry,
4050 * but we update the termination condition */
4051 len = priv->timers->len;
4053 /* find the smallest timeout */
4054 if (timer == NULL) {
4056 } else if (timer_timeout == -1) {
4057 /* we already have an immediate timeout, the new timer must be an
4058 * immediate timer with smaller seqnum to become the best */
4059 if (test_timeout == -1
4060 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4061 timer->seqnum) > 0))
4063 } else if (test_timeout == -1) {
4064 /* first immediate timer */
4066 } else if (test_timeout < timer_timeout) {
4069 } else if (test_timeout == timer_timeout
4070 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
4071 timer->seqnum) > 0)) {
4072 /* same timer, smaller seqnum */
4077 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
4079 timer_timeout = test_timeout;
4084 if (timer && !priv->blocked) {
4086 GstClockTime sync_time;
4089 GstClockTimeDiff clock_jitter;
4091 if (timer_timeout == -1 || timer_timeout <= now) {
4092 /* We have normally removed all lost timers in the loop above */
4093 g_assert (timer->type != TIMER_TYPE_LOST);
4095 do_timeout (jitterbuffer, timer, now);
4096 /* check here, do_timeout could have released the lock */
4097 if (!priv->timer_running)
4102 GST_OBJECT_LOCK (jitterbuffer);
4103 clock = GST_ELEMENT_CLOCK (jitterbuffer);
4105 GST_OBJECT_UNLOCK (jitterbuffer);
4106 /* let's just push if there is no clock */
4107 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
4108 now = timer_timeout;
4112 /* prepare for sync against clock */
4113 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
4114 /* add latency of peer to get input time */
4115 sync_time += priv->peer_latency;
4117 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
4118 " with sync time %" GST_TIME_FORMAT,
4119 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
4121 /* create an entry for the clock */
4122 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
4123 priv->timer_timeout = timer_timeout;
4124 priv->timer_seqnum = timer->seqnum;
4125 GST_OBJECT_UNLOCK (jitterbuffer);
4127 /* release the lock so that the other end can push stuff or unlock */
4130 ret = gst_clock_id_wait (id, &clock_jitter);
4133 if (!priv->timer_running) {
4134 gst_clock_id_unref (id);
4135 priv->clock_id = NULL;
4139 if (ret != GST_CLOCK_UNSCHEDULED) {
4140 now = timer_timeout + MAX (clock_jitter, 0);
4141 GST_DEBUG_OBJECT (jitterbuffer,
4142 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
4143 GST_STIME_ARGS (clock_jitter));
4145 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
4147 /* and free the entry */
4148 gst_clock_id_unref (id);
4149 priv->clock_id = NULL;
4151 /* no timers, wait for activity */
4152 JBUF_WAIT_TIMER (priv);
4157 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4162 * This funcion implements the main pushing loop on the source pad.
4164 * It first tries to push as many buffers as possible. If there is a seqnum
4165 * mismatch, we wait for the next timeouts.
4168 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4170 GstRtpJitterBufferPrivate *priv;
4171 GstFlowReturn result = GST_FLOW_OK;
4173 priv = jitterbuffer->priv;
4175 JBUF_LOCK_CHECK (priv, flushing);
4177 result = handle_next_buffer (jitterbuffer);
4178 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4179 /* now wait for the next event */
4180 JBUF_WAIT_EVENT (priv, flushing);
4181 result = GST_FLOW_OK;
4183 } while (result == GST_FLOW_OK);
4184 /* store result for upstream */
4185 priv->srcresult = result;
4186 /* if we get here we need to pause */
4192 result = priv->srcresult;
4199 JBUF_SIGNAL_QUERY (priv, FALSE);
4202 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4203 gst_flow_get_name (result));
4204 gst_pad_pause_task (priv->srcpad);
4205 if (result == GST_FLOW_EOS) {
4206 event = gst_event_new_eos ();
4207 gst_pad_push_event (priv->srcpad, event);
4213 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4214 * some sanity checks and then emit the handle-sync signal with the parameters.
4215 * This function must be called with the LOCK */
4217 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4219 GstRtpJitterBufferPrivate *priv;
4220 guint64 base_rtptime, base_time;
4222 guint64 last_rtptime;
4224 guint64 ext_rtptime, diff;
4225 gboolean valid = TRUE, keep = FALSE;
4227 priv = jitterbuffer->priv;
4229 /* get the last values from the jitterbuffer */
4230 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4231 &clock_rate, &last_rtptime);
4233 clock_base = priv->clock_base;
4234 ext_rtptime = priv->ext_rtptime;
4236 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4237 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4238 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4239 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4241 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4242 /* we keep this SR packet for later. When we get a valid RTP packet the
4243 * above values will be set and we can try to use the SR packet */
4244 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4247 /* we can't accept anything that happened before we did the last resync */
4248 if (base_rtptime > ext_rtptime) {
4249 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4252 /* the SR RTP timestamp must be something close to what we last observed
4253 * in the jitterbuffer */
4254 if (ext_rtptime > last_rtptime) {
4255 /* check how far ahead it is to our RTP timestamps */
4256 diff = ext_rtptime - last_rtptime;
4257 /* if bigger than 1 second, we drop it */
4258 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4260 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4261 clock_rate, 1000)) {
4262 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4263 /* should drop this, but some RTSP servers end up with bogus
4264 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4265 * so still trigger rptbin sync but invalidate RTCP data
4266 * (sync might use other methods) */
4269 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4270 G_GUINT64_FORMAT, last_rtptime, diff);
4276 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4280 s = gst_structure_new ("application/x-rtp-sync",
4281 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4282 "base-time", G_TYPE_UINT64, base_time,
4283 "clock-rate", G_TYPE_UINT, clock_rate,
4284 "clock-base", G_TYPE_UINT64, clock_base,
4285 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4286 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4288 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4289 gst_buffer_replace (&priv->last_sr, NULL);
4291 g_signal_emit (jitterbuffer,
4292 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4294 gst_structure_free (s);
4296 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4297 gst_buffer_replace (&priv->last_sr, NULL);
4301 static GstFlowReturn
4302 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4305 GstRtpJitterBuffer *jitterbuffer;
4306 GstRtpJitterBufferPrivate *priv;
4307 GstFlowReturn ret = GST_FLOW_OK;
4309 GstRTCPPacket packet;
4310 guint64 ext_rtptime;
4312 GstRTCPBuffer rtcp = { NULL, };
4314 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4316 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4317 goto invalid_buffer;
4319 priv = jitterbuffer->priv;
4321 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4323 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4326 /* first packet must be SR or RR or else the validate would have failed */
4327 switch (gst_rtcp_packet_get_type (&packet)) {
4328 case GST_RTCP_TYPE_SR:
4329 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4335 gst_rtcp_buffer_unmap (&rtcp);
4337 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4340 /* convert the RTP timestamp to our extended timestamp, using the same offset
4341 * we used in the jitterbuffer */
4342 ext_rtptime = priv->jbuf->ext_rtptime;
4343 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4345 priv->ext_rtptime = ext_rtptime;
4346 gst_buffer_replace (&priv->last_sr, buffer);
4348 do_handle_sync (jitterbuffer);
4352 gst_buffer_unref (buffer);
4358 /* this is not fatal but should be filtered earlier */
4359 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4360 ("Received invalid RTCP payload, dropping"));
4366 /* this is not fatal but should be filtered earlier */
4367 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4368 ("Received empty RTCP payload, dropping"));
4369 gst_rtcp_buffer_unmap (&rtcp);
4375 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4376 gst_rtcp_buffer_unmap (&rtcp);
4383 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4386 gboolean res = FALSE;
4387 GstRtpJitterBuffer *jitterbuffer;
4388 GstRtpJitterBufferPrivate *priv;
4390 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4391 priv = jitterbuffer->priv;
4393 switch (GST_QUERY_TYPE (query)) {
4394 case GST_QUERY_CAPS:
4396 GstCaps *filter, *caps;
4398 gst_query_parse_caps (query, &filter);
4399 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4400 gst_query_set_caps_result (query, caps);
4401 gst_caps_unref (caps);
4406 if (GST_QUERY_IS_SERIALIZED (query)) {
4407 RTPJitterBufferItem *item;
4410 JBUF_LOCK_CHECK (priv, out_flushing);
4411 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4412 RTP_JITTER_BUFFER_MODE_BUFFER) {
4413 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4414 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4415 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4417 JBUF_SIGNAL_EVENT (priv);
4418 JBUF_WAIT_QUERY (priv, out_flushing);
4419 res = priv->last_query;
4421 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4426 res = gst_pad_query_default (pad, parent, query);
4434 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4442 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4445 GstRtpJitterBuffer *jitterbuffer;
4446 GstRtpJitterBufferPrivate *priv;
4447 gboolean res = FALSE;
4449 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4450 priv = jitterbuffer->priv;
4452 switch (GST_QUERY_TYPE (query)) {
4453 case GST_QUERY_LATENCY:
4455 /* We need to send the query upstream and add the returned latency to our
4457 GstClockTime min_latency, max_latency;
4459 GstClockTime our_latency;
4461 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4462 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4464 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4465 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4466 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4468 /* store this so that we can safely sync on the peer buffers. */
4470 priv->peer_latency = min_latency;
4471 our_latency = priv->latency_ns;
4474 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4475 GST_TIME_ARGS (our_latency));
4477 /* we add some latency but can buffer an infinite amount of time */
4478 min_latency += our_latency;
4481 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4482 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4483 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4485 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4489 case GST_QUERY_POSITION:
4491 GstClockTime start, last_out;
4494 gst_query_parse_position (query, &fmt, NULL);
4495 if (fmt != GST_FORMAT_TIME) {
4496 res = gst_pad_query_default (pad, parent, query);
4501 start = priv->npt_start;
4502 last_out = priv->last_out_time;
4505 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4506 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4507 GST_TIME_ARGS (last_out));
4509 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4510 /* bring 0-based outgoing time to stream time */
4511 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4514 res = gst_pad_query_default (pad, parent, query);
4518 case GST_QUERY_CAPS:
4520 GstCaps *filter, *caps;
4522 gst_query_parse_caps (query, &filter);
4523 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4524 gst_query_set_caps_result (query, caps);
4525 gst_caps_unref (caps);
4530 res = gst_pad_query_default (pad, parent, query);
4538 gst_rtp_jitter_buffer_set_property (GObject * object,
4539 guint prop_id, const GValue * value, GParamSpec * pspec)
4541 GstRtpJitterBuffer *jitterbuffer;
4542 GstRtpJitterBufferPrivate *priv;
4544 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4545 priv = jitterbuffer->priv;
4550 guint new_latency, old_latency;
4552 new_latency = g_value_get_uint (value);
4555 old_latency = priv->latency_ms;
4556 priv->latency_ms = new_latency;
4557 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4558 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4561 /* post message if latency changed, this will inform the parent pipeline
4562 * that a latency reconfiguration is possible/needed. */
4563 if (new_latency != old_latency) {
4564 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4565 GST_TIME_ARGS (new_latency * GST_MSECOND));
4567 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4568 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4572 case PROP_DROP_ON_LATENCY:
4574 priv->drop_on_latency = g_value_get_boolean (value);
4577 case PROP_TS_OFFSET:
4579 if (priv->max_ts_offset_adjustment != 0) {
4580 gint64 new_offset = g_value_get_int64 (value);
4582 if (new_offset > priv->ts_offset) {
4583 priv->ts_offset_remainder = new_offset - priv->ts_offset;
4585 priv->ts_offset_remainder = -(priv->ts_offset - new_offset);
4588 priv->ts_offset = g_value_get_int64 (value);
4589 priv->ts_offset_remainder = 0;
4591 priv->ts_discont = TRUE;
4594 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4596 priv->max_ts_offset_adjustment = g_value_get_uint64 (value);
4601 priv->do_lost = g_value_get_boolean (value);
4606 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4609 case PROP_DO_RETRANSMISSION:
4611 priv->do_retransmission = g_value_get_boolean (value);
4614 case PROP_RTX_NEXT_SEQNUM:
4616 priv->rtx_next_seqnum = g_value_get_boolean (value);
4619 case PROP_RTX_DELAY:
4621 priv->rtx_delay = g_value_get_int (value);
4624 case PROP_RTX_MIN_DELAY:
4626 priv->rtx_min_delay = g_value_get_uint (value);
4629 case PROP_RTX_DELAY_REORDER:
4631 priv->rtx_delay_reorder = g_value_get_int (value);
4634 case PROP_RTX_RETRY_TIMEOUT:
4636 priv->rtx_retry_timeout = g_value_get_int (value);
4639 case PROP_RTX_MIN_RETRY_TIMEOUT:
4641 priv->rtx_min_retry_timeout = g_value_get_int (value);
4644 case PROP_RTX_RETRY_PERIOD:
4646 priv->rtx_retry_period = g_value_get_int (value);
4649 case PROP_RTX_MAX_RETRIES:
4651 priv->rtx_max_retries = g_value_get_int (value);
4654 case PROP_RTX_DEADLINE:
4656 priv->rtx_deadline_ms = g_value_get_int (value);
4659 case PROP_RTX_STATS_TIMEOUT:
4661 priv->rtx_stats_timeout = g_value_get_uint (value);
4664 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4666 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4669 case PROP_MAX_DROPOUT_TIME:
4671 priv->max_dropout_time = g_value_get_uint (value);
4674 case PROP_MAX_MISORDER_TIME:
4676 priv->max_misorder_time = g_value_get_uint (value);
4679 case PROP_RFC7273_SYNC:
4681 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4682 g_value_get_boolean (value));
4685 case PROP_FASTSTART_MIN_PACKETS:
4687 priv->faststart_min_packets = g_value_get_uint (value);
4691 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4697 gst_rtp_jitter_buffer_get_property (GObject * object,
4698 guint prop_id, GValue * value, GParamSpec * pspec)
4700 GstRtpJitterBuffer *jitterbuffer;
4701 GstRtpJitterBufferPrivate *priv;
4703 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4704 priv = jitterbuffer->priv;
4709 g_value_set_uint (value, priv->latency_ms);
4712 case PROP_DROP_ON_LATENCY:
4714 g_value_set_boolean (value, priv->drop_on_latency);
4717 case PROP_TS_OFFSET:
4719 g_value_set_int64 (value, priv->ts_offset);
4722 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
4724 g_value_set_uint64 (value, priv->max_ts_offset_adjustment);
4729 g_value_set_boolean (value, priv->do_lost);
4734 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4742 if (priv->srcresult != GST_FLOW_OK)
4745 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4747 g_value_set_int (value, percent);
4751 case PROP_DO_RETRANSMISSION:
4753 g_value_set_boolean (value, priv->do_retransmission);
4756 case PROP_RTX_NEXT_SEQNUM:
4758 g_value_set_boolean (value, priv->rtx_next_seqnum);
4761 case PROP_RTX_DELAY:
4763 g_value_set_int (value, priv->rtx_delay);
4766 case PROP_RTX_MIN_DELAY:
4768 g_value_set_uint (value, priv->rtx_min_delay);
4771 case PROP_RTX_DELAY_REORDER:
4773 g_value_set_int (value, priv->rtx_delay_reorder);
4776 case PROP_RTX_RETRY_TIMEOUT:
4778 g_value_set_int (value, priv->rtx_retry_timeout);
4781 case PROP_RTX_MIN_RETRY_TIMEOUT:
4783 g_value_set_int (value, priv->rtx_min_retry_timeout);
4786 case PROP_RTX_RETRY_PERIOD:
4788 g_value_set_int (value, priv->rtx_retry_period);
4791 case PROP_RTX_MAX_RETRIES:
4793 g_value_set_int (value, priv->rtx_max_retries);
4796 case PROP_RTX_DEADLINE:
4798 g_value_set_int (value, priv->rtx_deadline_ms);
4801 case PROP_RTX_STATS_TIMEOUT:
4803 g_value_set_uint (value, priv->rtx_stats_timeout);
4807 g_value_take_boxed (value,
4808 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4810 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4812 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4815 case PROP_MAX_DROPOUT_TIME:
4817 g_value_set_uint (value, priv->max_dropout_time);
4820 case PROP_MAX_MISORDER_TIME:
4822 g_value_set_uint (value, priv->max_misorder_time);
4825 case PROP_RFC7273_SYNC:
4827 g_value_set_boolean (value,
4828 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4831 case PROP_FASTSTART_MIN_PACKETS:
4833 g_value_set_uint (value, priv->faststart_min_packets);
4837 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4842 static GstStructure *
4843 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4845 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4849 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4850 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4851 "num-lost", G_TYPE_UINT64, priv->num_lost,
4852 "num-late", G_TYPE_UINT64, priv->num_late,
4853 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4854 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4855 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4856 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4857 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4858 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);