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_DO_LOST FALSE
135 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
136 #define DEFAULT_PERCENT 0
137 #define DEFAULT_DO_RETRANSMISSION FALSE
138 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
139 #define DEFAULT_RTX_DELAY -1
140 #define DEFAULT_RTX_MIN_DELAY 0
141 #define DEFAULT_RTX_DELAY_REORDER 3
142 #define DEFAULT_RTX_RETRY_TIMEOUT -1
143 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
144 #define DEFAULT_RTX_RETRY_PERIOD -1
145 #define DEFAULT_RTX_MAX_RETRIES -1
146 #define DEFAULT_RTX_DEADLINE -1
147 #define DEFAULT_RTX_STATS_TIMEOUT 1000
148 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
149 #define DEFAULT_MAX_DROPOUT_TIME 60000
150 #define DEFAULT_MAX_MISORDER_TIME 2000
151 #define DEFAULT_RFC7273_SYNC FALSE
153 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
154 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
160 PROP_DROP_ON_LATENCY,
165 PROP_DO_RETRANSMISSION,
166 PROP_RTX_NEXT_SEQNUM,
169 PROP_RTX_DELAY_REORDER,
170 PROP_RTX_RETRY_TIMEOUT,
171 PROP_RTX_MIN_RETRY_TIMEOUT,
172 PROP_RTX_RETRY_PERIOD,
173 PROP_RTX_MAX_RETRIES,
175 PROP_RTX_STATS_TIMEOUT,
177 PROP_MAX_RTCP_RTP_TIME_DIFF,
178 PROP_MAX_DROPOUT_TIME,
179 PROP_MAX_MISORDER_TIME,
183 #define JBUF_LOCK(priv) G_STMT_START { \
184 GST_TRACE("Locking from thread %p", g_thread_self()); \
185 (g_mutex_lock (&(priv)->jbuf_lock)); \
186 GST_TRACE("Locked from thread %p", g_thread_self()); \
189 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
191 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
194 #define JBUF_UNLOCK(priv) G_STMT_START { \
195 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
196 (g_mutex_unlock (&(priv)->jbuf_lock)); \
199 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
200 GST_DEBUG ("waiting timer"); \
201 (priv)->waiting_timer = TRUE; \
202 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
203 (priv)->waiting_timer = FALSE; \
204 GST_DEBUG ("waiting timer done"); \
206 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
207 if (G_UNLIKELY ((priv)->waiting_timer)) { \
208 GST_DEBUG ("signal timer"); \
209 g_cond_signal (&(priv)->jbuf_timer); \
213 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
214 GST_DEBUG ("waiting event"); \
215 (priv)->waiting_event = TRUE; \
216 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
217 (priv)->waiting_event = FALSE; \
218 GST_DEBUG ("waiting event done"); \
219 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
222 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
223 if (G_UNLIKELY ((priv)->waiting_event)) { \
224 GST_DEBUG ("signal event"); \
225 g_cond_signal (&(priv)->jbuf_event); \
229 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
230 GST_DEBUG ("waiting query"); \
231 (priv)->waiting_query = TRUE; \
232 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
233 (priv)->waiting_query = FALSE; \
234 GST_DEBUG ("waiting query done"); \
235 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
238 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
239 (priv)->last_query = res; \
240 if (G_UNLIKELY ((priv)->waiting_query)) { \
241 GST_DEBUG ("signal query"); \
242 g_cond_signal (&(priv)->jbuf_query); \
246 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
247 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
249 typedef struct TimerQueue
252 GHashTable *hashtable;
255 struct _GstRtpJitterBufferPrivate
257 GstPad *sinkpad, *srcpad;
260 RTPJitterBuffer *jbuf;
262 gboolean waiting_timer;
264 gboolean waiting_event;
266 gboolean waiting_query;
274 gboolean timer_running;
275 GThread *timer_thread;
280 gboolean drop_on_latency;
283 gboolean do_retransmission;
284 gboolean rtx_next_seqnum;
287 gint rtx_delay_reorder;
288 gint rtx_retry_timeout;
289 gint rtx_min_retry_timeout;
290 gint rtx_retry_period;
291 gint rtx_max_retries;
292 guint rtx_stats_timeout;
293 gint rtx_deadline_ms;
294 gint max_rtcp_rtp_time_diff;
295 guint32 max_dropout_time;
296 guint32 max_misorder_time;
298 /* the last seqnum we pushed out */
299 guint32 last_popped_seqnum;
300 /* the next expected seqnum we push */
302 /* seqnum-base, if known */
304 /* last output time */
305 GstClockTime last_out_time;
306 /* last valid input timestamp and rtptime pair */
307 GstClockTime ips_pts;
309 GstClockTime packet_spacing;
314 /* the next expected seqnum we receive */
315 GstClockTime last_in_pts;
316 guint32 next_in_seqnum;
319 TimerQueue *rtx_stats_timers;
321 /* start and stop ranges */
322 GstClockTime npt_start;
323 GstClockTime npt_stop;
324 guint64 ext_timestamp;
325 guint64 last_elapsed;
326 guint64 estimated_eos;
333 /* clock rate and rtp timestamp offset */
337 gint64 prev_ts_offset;
339 /* when we are shutting down */
340 GstFlowReturn srcresult;
346 GstClockTime timer_timeout;
347 guint16 timer_seqnum;
348 /* the latency of the upstream peer, we have to take this into account when
349 * synchronizing the buffers. */
350 GstClockTime peer_latency;
354 /* some accounting */
358 guint64 num_duplicates;
359 guint64 num_rtx_requests;
360 guint64 num_rtx_success;
361 guint64 num_rtx_failed;
364 RTPPacketRateCtx packet_rate_ctx;
367 GstClockTime last_dts;
368 guint64 last_rtptime;
369 GstClockTime avg_jitter;
386 GstClockTime timeout;
387 GstClockTime duration;
388 GstClockTime rtx_base;
389 GstClockTime rtx_delay;
390 GstClockTime rtx_retry;
391 GstClockTime rtx_last;
393 guint num_rtx_received;
396 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
397 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
398 GstRtpJitterBufferPrivate))
400 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
401 GST_STATIC_PAD_TEMPLATE ("sink",
404 GST_STATIC_CAPS ("application/x-rtp"
405 /* "clock-rate = (int) [ 1, 2147483647 ], "
406 * "payload = (int) , "
407 * "encoding-name = (string) "
411 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
412 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
415 GST_STATIC_CAPS ("application/x-rtcp")
418 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
419 GST_STATIC_PAD_TEMPLATE ("src",
422 GST_STATIC_CAPS ("application/x-rtp"
423 /* "payload = (int) , "
424 * "clock-rate = (int) , "
425 * "encoding-name = (string) "
429 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
431 #define gst_rtp_jitter_buffer_parent_class parent_class
432 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
434 /* object overrides */
435 static void gst_rtp_jitter_buffer_set_property (GObject * object,
436 guint prop_id, const GValue * value, GParamSpec * pspec);
437 static void gst_rtp_jitter_buffer_get_property (GObject * object,
438 guint prop_id, GValue * value, GParamSpec * pspec);
439 static void gst_rtp_jitter_buffer_finalize (GObject * object);
441 /* element overrides */
442 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
443 * element, GstStateChange transition);
444 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
445 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
446 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
448 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
449 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
453 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
454 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
457 /* sinkpad overrides */
458 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
459 GstObject * parent, GstEvent * event);
460 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
461 GstObject * parent, GstBuffer * buffer);
463 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
464 GstObject * parent, GstEvent * event);
465 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
466 GstObject * parent, GstBuffer * buffer);
468 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
469 GstObject * parent, GstQuery * query);
471 /* srcpad overrides */
472 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
473 GstObject * parent, GstEvent * event);
474 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
475 GstObject * parent, GstPadMode mode, gboolean active);
476 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
477 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
478 GstObject * parent, GstQuery * query);
481 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
483 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
484 gboolean active, guint64 base_time);
485 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
487 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
488 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
490 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
492 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
495 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
496 TimerData * timer, GstClockTime dts, gboolean success);
498 static TimerQueue *timer_queue_new (void);
499 static void timer_queue_free (TimerQueue * queue);
502 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
504 GObjectClass *gobject_class;
505 GstElementClass *gstelement_class;
507 gobject_class = (GObjectClass *) klass;
508 gstelement_class = (GstElementClass *) klass;
510 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
512 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
514 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
515 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
518 * GstRtpJitterBuffer:latency:
520 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
521 * for at most this time.
523 g_object_class_install_property (gobject_class, PROP_LATENCY,
524 g_param_spec_uint ("latency", "Buffer latency in ms",
525 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
526 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
528 * GstRtpJitterBuffer:drop-on-latency:
530 * Drop oldest buffers when the queue is completely filled.
532 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
533 g_param_spec_boolean ("drop-on-latency",
534 "Drop buffers when maximum latency is reached",
535 "Tells the jitterbuffer to never exceed the given latency in size",
536 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
538 * GstRtpJitterBuffer:ts-offset:
540 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
541 * This is mainly used to ensure interstream synchronisation.
543 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
544 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
545 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
546 G_MAXINT64, DEFAULT_TS_OFFSET,
547 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
550 * GstRtpJitterBuffer:do-lost:
552 * Send out a GstRTPPacketLost event downstream when a packet is considered
555 g_object_class_install_property (gobject_class, PROP_DO_LOST,
556 g_param_spec_boolean ("do-lost", "Do Lost",
557 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
558 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
561 * GstRtpJitterBuffer:mode:
563 * Control the buffering and timestamping mode used by the jitterbuffer.
565 g_object_class_install_property (gobject_class, PROP_MODE,
566 g_param_spec_enum ("mode", "Mode",
567 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
568 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
570 * GstRtpJitterBuffer:percent:
572 * The percent of the jitterbuffer that is filled.
574 g_object_class_install_property (gobject_class, PROP_PERCENT,
575 g_param_spec_int ("percent", "percent",
576 "The buffer filled percent", 0, 100,
577 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
579 * GstRtpJitterBuffer:do-retransmission:
581 * Send out a GstRTPRetransmission event upstream when a packet is considered
582 * late and should be retransmitted.
586 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
587 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
588 "Send retransmission events upstream when a packet is late",
589 DEFAULT_DO_RETRANSMISSION,
590 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
593 * GstRtpJitterBuffer:rtx-next-seqnum
595 * Estimate when the next packet should arrive and schedule a retransmission
597 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
598 * for packet N+1. So it will be requested if it does not arrive at the expected time.
599 * The expected time is calculated using the dts of N and the packet spacing.
603 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
604 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
605 "Estimate when the next packet should arrive and schedule a "
606 "retransmission request for it.",
607 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
610 * GstRtpJitterBuffer:rtx-delay:
612 * When a packet did not arrive at the expected time, wait this extra amount
613 * of time before sending a retransmission event.
615 * When -1 is used, the max jitter will be used as extra delay.
619 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
620 g_param_spec_int ("rtx-delay", "RTX Delay",
621 "Extra time in ms to wait before sending retransmission "
622 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
623 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
626 * GstRtpJitterBuffer:rtx-min-delay:
628 * When a packet did not arrive at the expected time, wait at least this extra amount
629 * of time before sending a retransmission event.
633 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
634 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
635 "Minimum time in ms to wait before sending retransmission "
636 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
637 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
639 * GstRtpJitterBuffer:rtx-delay-reorder:
641 * Assume that a retransmission event should be sent when we see
642 * this much packet reordering.
644 * When -1 is used, the value will be estimated based on observed packet
645 * reordering. When 0 is used packet reordering alone will not cause a
646 * retransmission event (Since 1.10).
650 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
651 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
652 "Sending retransmission event when this much reordering "
653 "(0 disable, -1 automatic)",
654 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
655 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
657 * GstRtpJitterBuffer::rtx-retry-timeout:
659 * When no packet has been received after sending a retransmission event
660 * for this time, retry sending a retransmission event.
662 * When -1 is used, the value will be estimated based on observed round
667 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
668 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
669 "Retry sending a transmission event after this timeout in "
670 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
671 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
673 * GstRtpJitterBuffer::rtx-min-retry-timeout:
675 * The minimum amount of time between retry timeouts. When
676 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
677 * minimum interval between retry timeouts.
679 * When -1 is used, the value will be estimated based on the
684 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
685 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
686 "Minimum timeout between sending a transmission event in "
687 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
688 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
690 * GstRtpJitterBuffer:rtx-retry-period:
692 * The amount of time to try to get a retransmission.
694 * When -1 is used, the value will be estimated based on the jitterbuffer
695 * latency and the observed round trip time.
699 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
700 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
701 "Try to get a retransmission for this many ms "
702 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
703 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
705 * GstRtpJitterBuffer:rtx-max-retries:
707 * The maximum number of retries to request a retransmission.
709 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
710 * When -1 is used, the number of retransmission request will not be limited.
714 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
715 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
716 "The maximum number of retries to request a retransmission. "
717 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
718 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
720 * GstRtpJitterBuffer:rtx-deadline:
722 * The deadline for a valid RTX request in ms.
724 * How long the RTX RTCP will be valid for.
725 * When -1 is used, the size of the jitterbuffer will be used.
729 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
730 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
731 "The deadline for a valid RTX request in milliseconds. "
732 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
733 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
735 * GstRtpJitterBuffer::rtx-stats-timeout:
737 * The time to wait for a retransmitted packet after it has been
738 * considered lost in order to collect RTX statistics.
742 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
743 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
744 "The time to wait for a retransmitted packet after it has been "
745 "considered lost in order to collect statistics (ms)",
746 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
747 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
749 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
750 g_param_spec_uint ("max-dropout-time", "Max dropout time",
751 "The maximum time (milliseconds) of missing packets tolerated.",
752 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
753 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
755 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
756 g_param_spec_uint ("max-misorder-time", "Max misorder time",
757 "The maximum time (milliseconds) of misordered packets tolerated.",
758 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
759 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
761 * GstRtpJitterBuffer:stats:
763 * Various jitterbuffer statistics. This property returns a GstStructure
764 * with name application/x-rtp-jitterbuffer-stats with the following fields:
770 * <classname>"num-pushed"</classname>:
771 * the number of packets pushed out.
777 * <classname>"num-lost"</classname>:
778 * the number of packets considered lost.
784 * <classname>"num-late"</classname>:
785 * the number of packets arriving too late.
791 * <classname>"num-duplicates"</classname>:
792 * the number of duplicate packets.
798 * <classname>"rtx-count"</classname>:
799 * the number of retransmissions requested.
805 * <classname>"rtx-success-count"</classname>:
806 * the number of successful retransmissions.
812 * <classname>"rtx-per-packet"</classname>:
813 * average number of RTX per packet.
819 * <classname>"rtx-rtt"</classname>:
820 * average round trip time per RTX.
827 g_object_class_install_property (gobject_class, PROP_STATS,
828 g_param_spec_boxed ("stats", "Statistics",
829 "Various statistics", GST_TYPE_STRUCTURE,
830 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
833 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
835 * The maximum amount of time in ms that the RTP time in the RTCP SRs
836 * is allowed to be ahead of the last RTP packet we received. Use
837 * -1 to disable ignoring of RTCP packets.
841 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
842 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
843 "Maximum amount of time in ms that the RTP time in RTCP SRs "
844 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
845 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
846 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
848 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
849 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
850 "Synchronize received streams to the RFC7273 clock "
851 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
852 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
855 * GstRtpJitterBuffer::request-pt-map:
856 * @buffer: the object which received the signal
859 * Request the payload type as #GstCaps for @pt.
861 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
862 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
863 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
864 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
865 GST_TYPE_CAPS, 1, G_TYPE_UINT);
867 * GstRtpJitterBuffer::handle-sync:
868 * @buffer: the object which received the signal
869 * @struct: a GstStructure containing sync values.
871 * Be notified of new sync values.
873 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
874 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
875 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
876 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
877 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
880 * GstRtpJitterBuffer::on-npt-stop:
881 * @buffer: the object which received the signal
883 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
884 * the npt-stop position.
886 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
887 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
888 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
889 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
890 G_TYPE_NONE, 0, G_TYPE_NONE);
893 * GstRtpJitterBuffer::clear-pt-map:
894 * @buffer: the object which received the signal
896 * Invalidate the clock-rate as obtained with the
897 * #GstRtpJitterBuffer::request-pt-map signal.
899 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
900 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
901 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
902 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
903 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
906 * GstRtpJitterBuffer::set-active:
907 * @buffer: the object which received the signal
909 * Start pushing out packets with the given base time. This signal is only
910 * useful in buffering mode.
912 * Returns: the time of the last pushed packet.
914 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
915 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
916 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
917 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
918 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
921 gstelement_class->change_state =
922 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
923 gstelement_class->request_new_pad =
924 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
925 gstelement_class->release_pad =
926 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
927 gstelement_class->provide_clock =
928 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
929 gstelement_class->set_clock =
930 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
932 gst_element_class_add_static_pad_template (gstelement_class,
933 &gst_rtp_jitter_buffer_src_template);
934 gst_element_class_add_static_pad_template (gstelement_class,
935 &gst_rtp_jitter_buffer_sink_template);
936 gst_element_class_add_static_pad_template (gstelement_class,
937 &gst_rtp_jitter_buffer_sink_rtcp_template);
939 gst_element_class_set_static_metadata (gstelement_class,
940 "RTP packet jitter-buffer", "Filter/Network/RTP",
941 "A buffer that deals with network jitter and other transmission faults",
942 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
943 "Wim Taymans <wim.taymans@gmail.com>");
945 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
946 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
948 GST_DEBUG_CATEGORY_INIT
949 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
953 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
955 GstRtpJitterBufferPrivate *priv;
957 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
958 jitterbuffer->priv = priv;
960 priv->latency_ms = DEFAULT_LATENCY_MS;
961 priv->latency_ns = priv->latency_ms * GST_MSECOND;
962 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
963 priv->do_lost = DEFAULT_DO_LOST;
964 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
965 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
966 priv->rtx_delay = DEFAULT_RTX_DELAY;
967 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
968 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
969 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
970 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
971 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
972 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
973 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
974 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
975 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
976 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
977 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
980 priv->last_rtptime = -1;
981 priv->avg_jitter = 0;
982 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
983 priv->rtx_stats_timers = timer_queue_new ();
984 priv->jbuf = rtp_jitter_buffer_new ();
985 g_mutex_init (&priv->jbuf_lock);
986 g_cond_init (&priv->jbuf_timer);
987 g_cond_init (&priv->jbuf_event);
988 g_cond_init (&priv->jbuf_query);
989 g_queue_init (&priv->gap_packets);
990 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
992 /* reset skew detection initialy */
993 rtp_jitter_buffer_reset_skew (priv->jbuf);
994 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
995 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
999 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1002 gst_pad_set_activatemode_function (priv->srcpad,
1003 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1004 gst_pad_set_query_function (priv->srcpad,
1005 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1006 gst_pad_set_event_function (priv->srcpad,
1007 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1010 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1013 gst_pad_set_chain_function (priv->sinkpad,
1014 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1015 gst_pad_set_event_function (priv->sinkpad,
1016 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1017 gst_pad_set_query_function (priv->sinkpad,
1018 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1020 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1021 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1023 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1026 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1028 #define ITEM_TYPE_BUFFER 0
1029 #define ITEM_TYPE_LOST 1
1030 #define ITEM_TYPE_EVENT 2
1031 #define ITEM_TYPE_QUERY 3
1033 static RTPJitterBufferItem *
1034 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1035 guint seqnum, guint count, guint rtptime)
1037 RTPJitterBufferItem *item;
1039 item = g_slice_new (RTPJitterBufferItem);
1046 item->seqnum = seqnum;
1047 item->count = count;
1048 item->rtptime = rtptime;
1054 free_item (RTPJitterBufferItem * item)
1056 g_return_if_fail (item != NULL);
1058 if (item->data && item->type != ITEM_TYPE_QUERY)
1059 gst_mini_object_unref (item->data);
1060 g_slice_free (RTPJitterBufferItem, item);
1064 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1066 GList **l = user_data;
1068 if (item->data && item->type == ITEM_TYPE_EVENT
1069 && GST_EVENT_IS_STICKY (item->data)) {
1070 *l = g_list_prepend (*l, item->data);
1071 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1072 gst_mini_object_unref (item->data);
1074 g_slice_free (RTPJitterBufferItem, item);
1078 gst_rtp_jitter_buffer_finalize (GObject * object)
1080 GstRtpJitterBuffer *jitterbuffer;
1081 GstRtpJitterBufferPrivate *priv;
1083 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1084 priv = jitterbuffer->priv;
1086 g_array_free (priv->timers, TRUE);
1087 timer_queue_free (priv->rtx_stats_timers);
1088 g_mutex_clear (&priv->jbuf_lock);
1089 g_cond_clear (&priv->jbuf_timer);
1090 g_cond_clear (&priv->jbuf_event);
1091 g_cond_clear (&priv->jbuf_query);
1093 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1094 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1095 g_queue_clear (&priv->gap_packets);
1096 g_object_unref (priv->jbuf);
1098 G_OBJECT_CLASS (parent_class)->finalize (object);
1101 static GstIterator *
1102 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1104 GstRtpJitterBuffer *jitterbuffer;
1105 GstPad *otherpad = NULL;
1106 GstIterator *it = NULL;
1107 GValue val = { 0, };
1109 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1111 if (pad == jitterbuffer->priv->sinkpad) {
1112 otherpad = jitterbuffer->priv->srcpad;
1113 } else if (pad == jitterbuffer->priv->srcpad) {
1114 otherpad = jitterbuffer->priv->sinkpad;
1115 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1116 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1120 g_value_init (&val, GST_TYPE_PAD);
1121 g_value_set_object (&val, otherpad);
1122 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1123 g_value_unset (&val);
1130 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1132 GstRtpJitterBufferPrivate *priv;
1134 priv = jitterbuffer->priv;
1136 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1139 gst_pad_new_from_static_template
1140 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1141 gst_pad_set_chain_function (priv->rtcpsinkpad,
1142 gst_rtp_jitter_buffer_chain_rtcp);
1143 gst_pad_set_event_function (priv->rtcpsinkpad,
1144 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1145 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1146 gst_rtp_jitter_buffer_iterate_internal_links);
1147 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1148 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1150 return priv->rtcpsinkpad;
1154 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1156 GstRtpJitterBufferPrivate *priv;
1158 priv = jitterbuffer->priv;
1160 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1162 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1164 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1165 priv->rtcpsinkpad = NULL;
1169 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1170 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1172 GstRtpJitterBuffer *jitterbuffer;
1173 GstElementClass *klass;
1175 GstRtpJitterBufferPrivate *priv;
1177 g_return_val_if_fail (templ != NULL, NULL);
1178 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1180 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1181 priv = jitterbuffer->priv;
1182 klass = GST_ELEMENT_GET_CLASS (element);
1184 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1186 /* figure out the template */
1187 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1188 if (priv->rtcpsinkpad != NULL)
1191 result = create_rtcp_sink (jitterbuffer);
1193 goto wrong_template;
1200 g_warning ("rtpjitterbuffer: this is not our template");
1205 g_warning ("rtpjitterbuffer: pad already requested");
1211 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1213 GstRtpJitterBuffer *jitterbuffer;
1214 GstRtpJitterBufferPrivate *priv;
1216 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1217 g_return_if_fail (GST_IS_PAD (pad));
1219 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1220 priv = jitterbuffer->priv;
1222 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1224 if (priv->rtcpsinkpad == pad) {
1225 remove_rtcp_sink (jitterbuffer);
1234 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1240 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1242 return gst_system_clock_obtain ();
1246 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1248 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1250 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1252 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1256 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1258 GstRtpJitterBufferPrivate *priv;
1260 priv = jitterbuffer->priv;
1262 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1265 priv->clock_rate = -1;
1266 /* do not clear current content, but refresh state for new arrival */
1267 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1268 rtp_jitter_buffer_reset_skew (priv->jbuf);
1273 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1276 GstRtpJitterBufferPrivate *priv;
1277 GstClockTime last_out;
1278 RTPJitterBufferItem *item;
1283 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1284 active, GST_TIME_ARGS (offset));
1286 if (active != priv->active) {
1287 /* add the amount of time spent in paused to the output offset. All
1288 * outgoing buffers will have this offset applied to their timestamps in
1289 * order to make them arrive in time in the sink. */
1290 priv->out_offset = offset;
1291 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1292 GST_TIME_ARGS (priv->out_offset));
1293 priv->active = active;
1294 JBUF_SIGNAL_EVENT (priv);
1297 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1299 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1300 /* head buffer timestamp and offset gives our output time */
1301 last_out = item->pts + priv->ts_offset;
1303 /* use last known time when the buffer is empty */
1304 last_out = priv->last_out_time;
1312 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1314 GstRtpJitterBuffer *jitterbuffer;
1315 GstRtpJitterBufferPrivate *priv;
1320 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1321 priv = jitterbuffer->priv;
1323 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1325 caps = gst_pad_peer_query_caps (other, filter);
1327 templ = gst_pad_get_pad_template_caps (pad);
1329 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1334 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1336 intersect = gst_caps_intersect (caps, templ);
1337 gst_caps_unref (caps);
1338 gst_caps_unref (templ);
1342 gst_object_unref (jitterbuffer);
1348 * Must be called with JBUF_LOCK held
1352 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1353 GstCaps * caps, gint pt)
1355 GstRtpJitterBufferPrivate *priv;
1356 GstStructure *caps_struct;
1360 const gchar *ts_refclk, *mediaclk;
1362 priv = jitterbuffer->priv;
1364 /* first parse the caps */
1365 caps_struct = gst_caps_get_structure (caps, 0);
1367 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1369 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1371 GST_ERROR_OBJECT (jitterbuffer,
1372 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1376 if (payload != -1) {
1377 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1378 priv->last_pt = payload;
1381 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1382 * measure the amount of data in the buffer */
1383 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1386 if (priv->clock_rate <= 0)
1389 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1391 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1393 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1395 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1396 * can use this to track the amount of time elapsed on the sender. */
1397 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1398 priv->clock_base = val;
1400 priv->clock_base = -1;
1402 priv->ext_timestamp = priv->clock_base;
1404 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1407 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1408 /* first expected seqnum, only update when we didn't have a previous base. */
1409 if (priv->next_in_seqnum == -1)
1410 priv->next_in_seqnum = val;
1411 if (priv->next_seqnum == -1) {
1412 priv->next_seqnum = val;
1413 JBUF_SIGNAL_EVENT (priv);
1415 priv->seqnum_base = val;
1417 priv->seqnum_base = -1;
1420 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1422 /* the start and stop times. The seqnum-base corresponds to the start time. We
1423 * will keep track of the seqnums on the output and when we reach the one
1424 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1425 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1426 priv->npt_start = tval;
1428 priv->npt_start = 0;
1430 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1431 priv->npt_stop = tval;
1433 priv->npt_stop = -1;
1435 GST_DEBUG_OBJECT (jitterbuffer,
1436 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1437 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1439 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1440 GstClock *clock = NULL;
1441 guint64 clock_offset = -1;
1443 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1446 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1447 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1448 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1450 const gchar *host, *portstr;
1454 host = ts_refclk + sizeof ("ntp=") - 1;
1455 if (host[0] == '[') {
1457 portstr = strchr (host, ']');
1458 if (portstr && portstr[1] == ':')
1459 portstr = portstr + 1;
1463 portstr = strrchr (host, ':');
1467 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1471 hostname = g_strndup (host, (portstr - host));
1473 hostname = g_strdup (host);
1475 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1478 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1479 const gchar *domainstr =
1480 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1483 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1486 clock = gst_ptp_clock_new (NULL, domain);
1488 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1491 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1492 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1494 if (!g_str_has_prefix (mediaclk, "direct=")
1495 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1496 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1497 if (strstr (mediaclk, "rate=") != NULL) {
1498 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1503 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1505 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1513 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1518 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1524 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1526 GstRtpJitterBufferPrivate *priv;
1528 priv = jitterbuffer->priv;
1531 /* mark ourselves as flushing */
1532 priv->srcresult = GST_FLOW_FLUSHING;
1533 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1534 /* this unblocks any waiting pops on the src pad task */
1535 JBUF_SIGNAL_EVENT (priv);
1536 JBUF_SIGNAL_QUERY (priv, FALSE);
1541 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1543 GstRtpJitterBufferPrivate *priv;
1545 priv = jitterbuffer->priv;
1548 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1549 /* Mark as non flushing */
1550 priv->srcresult = GST_FLOW_OK;
1551 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1552 priv->last_popped_seqnum = -1;
1553 priv->last_out_time = -1;
1554 priv->next_seqnum = -1;
1555 priv->seqnum_base = -1;
1556 priv->ips_rtptime = -1;
1557 priv->ips_pts = GST_CLOCK_TIME_NONE;
1558 priv->packet_spacing = 0;
1559 priv->next_in_seqnum = -1;
1560 priv->clock_rate = -1;
1563 priv->estimated_eos = -1;
1564 priv->last_elapsed = 0;
1565 priv->ext_timestamp = -1;
1566 priv->avg_jitter = 0;
1567 priv->last_dts = -1;
1568 priv->last_rtptime = -1;
1569 priv->last_in_pts = 0;
1570 priv->equidistant = 0;
1571 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1572 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1573 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1574 rtp_jitter_buffer_reset_skew (priv->jbuf);
1575 remove_all_timers (jitterbuffer);
1576 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1577 g_queue_clear (&priv->gap_packets);
1582 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1583 GstPadMode mode, gboolean active)
1586 GstRtpJitterBuffer *jitterbuffer = NULL;
1588 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1591 case GST_PAD_MODE_PUSH:
1593 /* allow data processing */
1594 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1596 /* start pushing out buffers */
1597 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1598 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1599 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1601 /* make sure all data processing stops ASAP */
1602 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1604 /* NOTE this will hardlock if the state change is called from the src pad
1605 * task thread because we will _join() the thread. */
1606 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1607 result = gst_pad_stop_task (pad);
1617 static GstStateChangeReturn
1618 gst_rtp_jitter_buffer_change_state (GstElement * element,
1619 GstStateChange transition)
1621 GstRtpJitterBuffer *jitterbuffer;
1622 GstRtpJitterBufferPrivate *priv;
1623 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1625 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1626 priv = jitterbuffer->priv;
1628 switch (transition) {
1629 case GST_STATE_CHANGE_NULL_TO_READY:
1631 case GST_STATE_CHANGE_READY_TO_PAUSED:
1633 /* reset negotiated values */
1634 priv->clock_rate = -1;
1635 priv->clock_base = -1;
1636 priv->peer_latency = 0;
1638 /* block until we go to PLAYING */
1639 priv->blocked = TRUE;
1640 priv->timer_running = TRUE;
1641 priv->timer_thread =
1642 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1645 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1647 /* unblock to allow streaming in PLAYING */
1648 priv->blocked = FALSE;
1649 JBUF_SIGNAL_EVENT (priv);
1650 JBUF_SIGNAL_TIMER (priv);
1657 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1659 switch (transition) {
1660 case GST_STATE_CHANGE_READY_TO_PAUSED:
1661 /* we are a live element because we sync to the clock, which we can only
1662 * do in the PLAYING state */
1663 if (ret != GST_STATE_CHANGE_FAILURE)
1664 ret = GST_STATE_CHANGE_NO_PREROLL;
1666 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1668 /* block to stop streaming when PAUSED */
1669 priv->blocked = TRUE;
1670 unschedule_current_timer (jitterbuffer);
1672 if (ret != GST_STATE_CHANGE_FAILURE)
1673 ret = GST_STATE_CHANGE_NO_PREROLL;
1675 case GST_STATE_CHANGE_PAUSED_TO_READY:
1677 gst_buffer_replace (&priv->last_sr, NULL);
1678 priv->timer_running = FALSE;
1679 unschedule_current_timer (jitterbuffer);
1680 JBUF_SIGNAL_TIMER (priv);
1681 JBUF_SIGNAL_QUERY (priv, FALSE);
1683 g_thread_join (priv->timer_thread);
1684 priv->timer_thread = NULL;
1686 case GST_STATE_CHANGE_READY_TO_NULL:
1696 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1699 gboolean ret = TRUE;
1700 GstRtpJitterBuffer *jitterbuffer;
1701 GstRtpJitterBufferPrivate *priv;
1703 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1704 priv = jitterbuffer->priv;
1706 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1708 switch (GST_EVENT_TYPE (event)) {
1709 case GST_EVENT_LATENCY:
1711 GstClockTime latency;
1713 gst_event_parse_latency (event, &latency);
1715 GST_DEBUG_OBJECT (jitterbuffer,
1716 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1719 /* adjust the overall buffer delay to the total pipeline latency in
1720 * buffering mode because if downstream consumes too fast (because of
1721 * large latency or queues, we would start rebuffering again. */
1722 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1723 RTP_JITTER_BUFFER_MODE_BUFFER) {
1724 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1728 ret = gst_pad_push_event (priv->sinkpad, event);
1732 ret = gst_pad_push_event (priv->sinkpad, event);
1739 /* handles and stores the event in the jitterbuffer, must be called with
1742 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1744 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1745 RTPJitterBufferItem *item;
1748 switch (GST_EVENT_TYPE (event)) {
1749 case GST_EVENT_CAPS:
1753 gst_event_parse_caps (event, &caps);
1754 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1757 case GST_EVENT_SEGMENT:
1760 gst_event_copy_segment (event, &segment);
1762 /* we need time for now */
1763 if (segment.format != GST_FORMAT_TIME) {
1764 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1765 gst_event_unref (event);
1767 gst_segment_init (&segment, GST_FORMAT_TIME);
1768 event = gst_event_new_segment (&segment);
1771 priv->segment = segment;
1776 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1783 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1784 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1785 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1787 JBUF_SIGNAL_EVENT (priv);
1793 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1796 gboolean ret = TRUE;
1797 GstRtpJitterBuffer *jitterbuffer;
1798 GstRtpJitterBufferPrivate *priv;
1800 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1801 priv = jitterbuffer->priv;
1803 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1805 switch (GST_EVENT_TYPE (event)) {
1806 case GST_EVENT_FLUSH_START:
1807 ret = gst_pad_push_event (priv->srcpad, event);
1808 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1809 /* wait for the loop to go into PAUSED */
1810 gst_pad_pause_task (priv->srcpad);
1812 case GST_EVENT_FLUSH_STOP:
1813 ret = gst_pad_push_event (priv->srcpad, event);
1815 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1816 GST_PAD_MODE_PUSH, TRUE);
1819 if (GST_EVENT_IS_SERIALIZED (event)) {
1820 /* serialized events go in the queue */
1822 if (priv->srcresult != GST_FLOW_OK) {
1823 /* Errors in sticky event pushing are no problem and ignored here
1824 * as they will cause more meaningful errors during data flow.
1825 * For EOS events, that are not followed by data flow, we still
1826 * return FALSE here though.
1828 if (!GST_EVENT_IS_STICKY (event) ||
1829 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1830 goto out_flow_error;
1832 /* refuse more events on EOS */
1835 ret = queue_event (jitterbuffer, event);
1838 /* non-serialized events are forwarded downstream immediately */
1839 ret = gst_pad_push_event (priv->srcpad, event);
1848 GST_DEBUG_OBJECT (jitterbuffer,
1849 "refusing event, we have a downstream flow error: %s",
1850 gst_flow_get_name (priv->srcresult));
1852 gst_event_unref (event);
1857 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1859 gst_event_unref (event);
1865 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1868 gboolean ret = TRUE;
1869 GstRtpJitterBuffer *jitterbuffer;
1871 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1873 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1875 switch (GST_EVENT_TYPE (event)) {
1876 case GST_EVENT_FLUSH_START:
1877 gst_event_unref (event);
1879 case GST_EVENT_FLUSH_STOP:
1880 gst_event_unref (event);
1883 ret = gst_pad_event_default (pad, parent, event);
1891 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1892 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1893 * GST_FLOW_FLUSHING when the element is shutting down. On success
1894 * GST_FLOW_OK is returned.
1896 static GstFlowReturn
1897 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1901 GValue args[2] = { {0}, {0} };
1905 g_value_init (&args[0], GST_TYPE_ELEMENT);
1906 g_value_set_object (&args[0], jitterbuffer);
1907 g_value_init (&args[1], G_TYPE_UINT);
1908 g_value_set_uint (&args[1], pt);
1910 g_value_init (&ret, GST_TYPE_CAPS);
1911 g_value_set_boxed (&ret, NULL);
1913 JBUF_UNLOCK (jitterbuffer->priv);
1914 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1916 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1918 g_value_unset (&args[0]);
1919 g_value_unset (&args[1]);
1920 caps = (GstCaps *) g_value_dup_boxed (&ret);
1921 g_value_unset (&ret);
1925 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1926 gst_caps_unref (caps);
1928 if (G_UNLIKELY (!res))
1936 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1937 return GST_FLOW_ERROR;
1941 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1942 return GST_FLOW_FLUSHING;
1946 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1947 return GST_FLOW_ERROR;
1951 /* call with jbuf lock held */
1953 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1955 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1956 GstMessage *message = NULL;
1961 /* Post a buffering message */
1962 if (priv->last_percent != percent) {
1963 priv->last_percent = percent;
1965 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1966 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1973 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1975 GstRtpJitterBufferPrivate *priv;
1977 priv = jitterbuffer->priv;
1979 if (timestamp == -1)
1982 /* apply the timestamp offset, this is used for inter stream sync */
1983 timestamp += priv->ts_offset;
1984 /* add the offset, this is used when buffering */
1985 timestamp += priv->out_offset;
1991 timer_queue_new (void)
1995 queue = g_slice_new (TimerQueue);
1996 queue->timers = g_queue_new ();
1997 queue->hashtable = g_hash_table_new (NULL, NULL);
2003 timer_queue_free (TimerQueue * queue)
2008 g_hash_table_destroy (queue->hashtable);
2009 g_queue_free_full (queue->timers, g_free);
2010 g_slice_free (TimerQueue, queue);
2014 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2015 GstClockTime timeout, gboolean lost)
2019 copy = g_memdup (timer, sizeof (*timer));
2020 copy->timeout = timeout;
2021 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2024 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2025 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2026 g_queue_push_tail (queue->timers, copy);
2027 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2031 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2035 test = g_queue_peek_head (queue->timers);
2036 while (test && test->timeout < timeout) {
2037 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2038 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2039 GST_TIME_ARGS (timeout));
2040 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2041 g_free (g_queue_pop_head (queue->timers));
2042 test = g_queue_peek_head (queue->timers);
2047 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2049 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2053 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2055 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2056 TimerData *timer = NULL;
2059 len = priv->timers->len;
2060 for (i = 0; i < len; i++) {
2061 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2062 if (test->seqnum == seqnum) {
2071 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2073 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2075 if (priv->clock_id) {
2076 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2077 gst_clock_id_unschedule (priv->clock_id);
2078 priv->clock_id = NULL;
2083 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2085 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2086 GstClockTime test_timeout;
2088 if ((test_timeout = timer->timeout) == -1)
2091 if (timer->type != TIMER_TYPE_EXPECTED) {
2092 /* add our latency and offset to get output times. */
2093 test_timeout = apply_offset (jitterbuffer, test_timeout);
2094 test_timeout += priv->latency_ns;
2096 return test_timeout;
2100 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2102 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2104 if (priv->clock_id) {
2105 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2107 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2108 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2110 if (timeout == -1 || timeout < priv->timer_timeout)
2111 unschedule_current_timer (jitterbuffer);
2116 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2117 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2118 GstClockTime duration)
2120 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2124 GST_DEBUG_OBJECT (jitterbuffer,
2125 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2126 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2127 GST_TIME_ARGS (delay));
2129 len = priv->timers->len;
2130 g_array_set_size (priv->timers, len + 1);
2131 timer = &g_array_index (priv->timers, TimerData, len);
2134 timer->seqnum = seqnum;
2136 timer->timeout = timeout + delay;
2137 timer->duration = duration;
2138 if (type == TIMER_TYPE_EXPECTED) {
2139 timer->rtx_base = timeout;
2140 timer->rtx_delay = delay;
2141 timer->rtx_retry = 0;
2143 timer->rtx_last = GST_CLOCK_TIME_NONE;
2144 timer->num_rtx_retry = 0;
2145 timer->num_rtx_received = 0;
2146 recalculate_timer (jitterbuffer, timer);
2147 JBUF_SIGNAL_TIMER (priv);
2153 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2154 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2156 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2157 gboolean seqchange, timechange;
2159 GstClockTime new_timeout;
2161 oldseq = timer->seqnum;
2162 new_timeout = timeout + delay;
2163 seqchange = oldseq != seqnum;
2164 timechange = timer->timeout != new_timeout;
2166 if (!seqchange && !timechange) {
2167 GST_DEBUG_OBJECT (jitterbuffer,
2168 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2169 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2173 GST_DEBUG_OBJECT (jitterbuffer,
2174 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2175 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2176 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2178 timer->timeout = new_timeout;
2179 timer->seqnum = seqnum;
2181 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2182 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2183 GST_TIME_ARGS (delay));
2184 timer->rtx_base = timeout;
2185 timer->rtx_delay = delay;
2186 timer->rtx_retry = 0;
2189 timer->num_rtx_retry = 0;
2190 timer->num_rtx_received = 0;
2193 if (priv->clock_id) {
2194 /* we changed the seqnum and there is a timer currently waiting with this
2195 * seqnum, unschedule it */
2196 if (seqchange && priv->timer_seqnum == oldseq)
2197 unschedule_current_timer (jitterbuffer);
2198 /* we changed the time, check if it is earlier than what we are waiting
2199 * for and unschedule if so */
2200 else if (timechange)
2201 recalculate_timer (jitterbuffer, timer);
2206 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2207 guint16 seqnum, GstClockTime timeout)
2211 /* find the seqnum timer */
2212 timer = find_timer (jitterbuffer, seqnum);
2213 if (timer == NULL) {
2214 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2216 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2222 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2224 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2227 if (timer->idx == -1)
2230 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2231 unschedule_current_timer (jitterbuffer);
2234 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2235 g_array_remove_index_fast (priv->timers, idx);
2240 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2242 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2243 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2244 g_array_set_size (priv->timers, 0);
2245 unschedule_current_timer (jitterbuffer);
2248 /* get the extra delay to wait before sending RTX */
2250 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2254 if (priv->rtx_delay == -1) {
2255 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2256 delay = DEFAULT_AUTO_RTX_DELAY;
2258 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2259 * packet spacing is a good margin */
2260 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2263 delay = priv->rtx_delay * GST_MSECOND;
2265 if (priv->rtx_min_delay > 0)
2266 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2271 /* Check if packet with seqnum is already considered definitely lost by being
2272 * part of a "lost timer" for multiple packets */
2274 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2276 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2279 len = priv->timers->len;
2280 for (i = 0; i < len; i++) {
2281 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2282 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2284 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2286 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2287 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2295 /* we just received a packet with seqnum and dts.
2297 * First check for old seqnum that we are still expecting. If the gap with the
2298 * current seqnum is too big, unschedule the timeouts.
2300 * If we have a valid packet spacing estimate we can set a timer for when we
2301 * should receive the next packet.
2302 * If we don't have a valid estimate, we remove any timer we might have
2303 * had for this packet.
2306 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2307 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2308 gboolean is_rtx, TimerData * timer)
2310 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2312 /* go through all timers and unschedule the ones with a large gap */
2313 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2315 len = priv->timers->len;
2316 for (i = 0; i < len; i++) {
2317 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2320 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2322 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2323 test->type, test->seqnum, seqnum, gap);
2325 if (gap > priv->rtx_delay_reorder) {
2326 /* max gap, we exceeded the max reorder distance and we don't expect the
2327 * missing packet to be this reordered */
2328 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2329 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2334 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2335 && priv->do_retransmission && priv->rtx_next_seqnum;
2337 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2338 if (timer->num_rtx_retry > 0) {
2340 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2341 /* don't try to estimate the next seqnum because this is a retransmitted
2342 * packet and it probably did not arrive with the expected packet
2344 do_next_seqnum = FALSE;
2347 if (!is_rtx || timer->num_rtx_retry > 1) {
2348 /* Store timer in order to record stats when/if the retransmitted
2349 * packet arrives. We should also store timer information if we've
2350 * requested retransmission more than once since we may receive
2351 * several retransmitted packets. For accuracy we should update the
2352 * stats also when the redundant retransmitted packets arrives. */
2353 timer_queue_append (priv->rtx_stats_timers, timer,
2354 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2359 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2360 GstClockTime expected, delay;
2362 /* calculate expected arrival time of the next seqnum */
2363 expected = pts + priv->packet_spacing;
2365 delay = get_rtx_delay (priv);
2367 /* and update/install timer for next seqnum */
2368 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2369 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2370 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2371 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2372 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2375 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2378 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2379 expected, delay, priv->packet_spacing);
2381 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2382 /* if we had a timer, remove it, we don't know when to expect the next
2384 remove_timer (jitterbuffer, timer);
2389 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2392 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2394 /* we need consecutive seqnums with a different
2395 * rtptime to estimate the packet spacing. */
2396 if (priv->ips_rtptime != rtptime) {
2397 /* rtptime changed, check pts diff */
2398 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2399 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2400 GstClockTime old_packet_spacing = priv->packet_spacing;
2402 /* Biased towards bigger packet spacings to prevent
2403 * too many unneeded retransmission requests for next
2404 * packets that just arrive a little later than we would
2406 if (old_packet_spacing > new_packet_spacing)
2407 priv->packet_spacing =
2408 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2409 else if (old_packet_spacing > 0)
2410 priv->packet_spacing =
2411 (3 * new_packet_spacing + old_packet_spacing) / 4;
2413 priv->packet_spacing = new_packet_spacing;
2415 GST_DEBUG_OBJECT (jitterbuffer,
2416 "new packet spacing %" GST_TIME_FORMAT
2417 " old packet spacing %" GST_TIME_FORMAT
2418 " combined to %" GST_TIME_FORMAT,
2419 GST_TIME_ARGS (new_packet_spacing),
2420 GST_TIME_ARGS (old_packet_spacing),
2421 GST_TIME_ARGS (priv->packet_spacing));
2423 priv->ips_rtptime = rtptime;
2424 priv->ips_pts = pts;
2429 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2430 guint16 seqnum, GstClockTime pts, gint gap)
2432 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2433 GstClockTime duration, expected_pts, delay;
2435 gboolean equidistant = priv->equidistant > 0;
2437 GST_DEBUG_OBJECT (jitterbuffer,
2438 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2439 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2441 if (pts == GST_CLOCK_TIME_NONE) {
2442 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2447 GstClockTime total_duration;
2448 /* the total duration spanned by the missing packets */
2449 if (pts >= priv->last_in_pts)
2450 total_duration = pts - priv->last_in_pts;
2454 /* interpolate between the current time and the last time based on
2455 * number of packets we are missing, this is the estimated duration
2456 * for the missing packet based on equidistant packet spacing. */
2457 duration = total_duration / (gap + 1);
2459 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2460 GST_TIME_ARGS (duration));
2462 if (total_duration > priv->latency_ns) {
2463 GstClockTime gap_time;
2467 GstClockTime gap_dur = gap * duration;
2468 if (gap_dur > priv->latency_ns)
2469 gap_time = gap_dur - priv->latency_ns;
2472 lost_packets = gap_time / duration;
2474 gap_time = total_duration - priv->latency_ns;
2478 /* too many lost packets, some of the missing packets are already
2479 * too late and we can generate lost packet events for them. */
2480 GST_INFO_OBJECT (jitterbuffer,
2481 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2482 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2483 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2484 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2485 GST_TIME_ARGS (gap_time));
2487 /* this timer will fire immediately and the lost event will be pushed from
2488 * the timer thread */
2489 if (lost_packets > 0) {
2490 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2491 priv->last_in_pts + duration, 0, gap_time);
2492 expected += lost_packets;
2493 priv->last_in_pts += gap_time;
2497 expected_pts = priv->last_in_pts + duration;
2499 /* If we cannot assume equidistant packet spacing, the only thing we now
2500 * for sure is that the missing packets have expected pts not later than
2501 * the last received pts. */
2508 if (priv->do_retransmission) {
2509 TimerData *timer = find_timer (jitterbuffer, expected);
2511 type = TIMER_TYPE_EXPECTED;
2512 delay = get_rtx_delay (priv);
2514 /* if we had a timer for the first missing packet, update it. */
2515 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2516 GstClockTime timeout = timer->timeout;
2518 timer->duration = duration;
2519 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2520 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2524 expected_pts += duration;
2527 type = TIMER_TYPE_LOST;
2530 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2531 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2532 expected_pts += duration;
2538 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2542 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2543 GstRtpJitterBufferPrivate *priv;
2545 priv = jitterbuffer->priv;
2547 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2550 if (priv->last_dts != -1)
2551 dtsdiff = dts - priv->last_dts;
2555 if (priv->last_rtptime != -1)
2556 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2560 /* Guess whether stream currently uses equidistant packet spacing. If we
2561 * often see identical timestamps it means the packets are not
2563 if (rtptime == priv->last_rtptime)
2564 priv->equidistant -= 2;
2566 priv->equidistant += 1;
2567 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2569 priv->last_dts = dts;
2570 priv->last_rtptime = rtptime;
2574 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2577 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2579 diff = ABS (dtsdiff - rtpdiffns);
2581 /* jitter is stored in nanoseconds */
2582 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2584 GST_LOG_OBJECT (jitterbuffer,
2585 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2586 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2587 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2588 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2595 GST_DEBUG_OBJECT (jitterbuffer,
2596 "no dts or no clock-rate, can't calculate jitter");
2602 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2604 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2605 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2608 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2609 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2610 gst_rtp_buffer_unmap (&rtp_a);
2612 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2613 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2614 gst_rtp_buffer_unmap (&rtp_b);
2616 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2620 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2621 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2623 GstRtpJitterBufferPrivate *priv;
2624 guint gap_packets_length;
2625 gboolean reset = FALSE;
2626 gboolean future = gap > 0;
2628 priv = jitterbuffer->priv;
2630 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2632 guint32 prev_gap_seq = -1;
2633 gboolean all_consecutive = TRUE;
2635 g_queue_insert_sorted (&priv->gap_packets, buffer,
2636 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2638 for (l = priv->gap_packets.head; l; l = l->next) {
2639 GstBuffer *gap_buffer = l->data;
2640 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2643 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2645 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2647 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2648 if (prev_gap_seq == -1)
2649 prev_gap_seq = gap_seq;
2650 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2651 all_consecutive = FALSE;
2653 prev_gap_seq = gap_seq;
2655 gst_rtp_buffer_unmap (&gap_rtp);
2656 if (!all_consecutive)
2660 if (all_consecutive && gap_packets_length > 3) {
2661 GST_DEBUG_OBJECT (jitterbuffer,
2662 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2663 (future ? "new" : "old"), gap,
2664 (future ? max_dropout : -max_misorder));
2666 } else if (!all_consecutive) {
2667 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2668 g_queue_clear (&priv->gap_packets);
2669 GST_DEBUG_OBJECT (jitterbuffer,
2670 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2671 (future ? "new" : "old"), gap,
2672 (future ? max_dropout : -max_misorder));
2675 GST_DEBUG_OBJECT (jitterbuffer,
2676 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2677 (future ? "new" : "old"), gap,
2678 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2682 GST_DEBUG_OBJECT (jitterbuffer,
2683 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2684 gap, -max_misorder);
2685 g_queue_push_tail (&priv->gap_packets, buffer);
2693 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2695 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2696 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2699 GstClockTime base_time =
2700 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2701 GstClockTime clock_time = gst_clock_get_time (clock);
2703 if (clock_time > base_time)
2704 running_time = clock_time - base_time;
2708 gst_object_unref (clock);
2711 return running_time;
2714 static GstFlowReturn
2715 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2716 GstPad * pad, GstObject * parent, guint16 seqnum)
2718 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2719 GstFlowReturn ret = GST_FLOW_OK;
2720 GList *events = NULL, *l;
2724 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2725 rtp_jitter_buffer_flush (priv->jbuf,
2726 (GFunc) free_item_and_retain_events, &events);
2727 rtp_jitter_buffer_reset_skew (priv->jbuf);
2728 remove_all_timers (jitterbuffer);
2729 priv->discont = TRUE;
2730 priv->last_popped_seqnum = -1;
2732 if (priv->gap_packets.head) {
2733 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2734 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2736 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2737 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2738 gst_rtp_buffer_unmap (&gap_rtp);
2740 priv->next_seqnum = seqnum;
2743 priv->last_in_pts = -1;
2744 priv->next_in_seqnum = -1;
2746 /* Insert all sticky events again in order, otherwise we would
2747 * potentially loose STREAM_START, CAPS or SEGMENT events
2749 events = g_list_reverse (events);
2750 for (l = events; l; l = l->next) {
2751 RTPJitterBufferItem *item;
2753 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2754 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2756 g_list_free (events);
2758 JBUF_SIGNAL_EVENT (priv);
2760 /* reset spacing estimation when gap */
2761 priv->ips_rtptime = -1;
2762 priv->ips_pts = GST_CLOCK_TIME_NONE;
2764 buffers = g_list_copy (priv->gap_packets.head);
2765 g_queue_clear (&priv->gap_packets);
2767 priv->ips_rtptime = -1;
2768 priv->ips_pts = GST_CLOCK_TIME_NONE;
2769 JBUF_UNLOCK (jitterbuffer->priv);
2771 for (l = buffers; l; l = l->next) {
2772 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2774 if (ret != GST_FLOW_OK) {
2779 for (; l; l = l->next)
2780 gst_buffer_unref (l->data);
2781 g_list_free (buffers);
2786 static GstFlowReturn
2787 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2790 GstRtpJitterBuffer *jitterbuffer;
2791 GstRtpJitterBufferPrivate *priv;
2793 guint32 expected, rtptime;
2794 GstFlowReturn ret = GST_FLOW_OK;
2795 GstClockTime dts, pts;
2800 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2801 gboolean do_next_seqnum = FALSE;
2802 RTPJitterBufferItem *item;
2803 GstMessage *msg = NULL;
2804 gboolean estimated_dts = FALSE;
2805 gint32 packet_rate, max_dropout, max_misorder;
2806 TimerData *timer = NULL;
2808 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2810 priv = jitterbuffer->priv;
2812 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2813 goto invalid_buffer;
2815 pt = gst_rtp_buffer_get_payload_type (&rtp);
2816 seqnum = gst_rtp_buffer_get_seq (&rtp);
2817 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2818 gst_rtp_buffer_unmap (&rtp);
2820 /* make sure we have PTS and DTS set */
2821 pts = GST_BUFFER_PTS (buffer);
2822 dts = GST_BUFFER_DTS (buffer);
2829 /* If we have no DTS here, i.e. no capture time, get one from the
2830 * clock now to have something to calculate with in the future. */
2831 dts = get_current_running_time (jitterbuffer);
2834 /* Remember that we estimated the DTS if we are running already
2835 * and this is not our first packet (or first packet after a reset).
2836 * If it's the first packet, we somehow must generate a timestamp for
2837 * everything, otherwise we can't calculate any times
2839 estimated_dts = (priv->next_in_seqnum != -1);
2841 /* take the DTS of the buffer. This is the time when the packet was
2842 * received and is used to calculate jitter and clock skew. We will adjust
2843 * this DTS with the smoothed value after processing it in the
2844 * jitterbuffer and assign it as the PTS. */
2845 /* bring to running time */
2846 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2849 GST_DEBUG_OBJECT (jitterbuffer,
2850 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2851 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2852 GST_BUFFER_IS_RETRANSMISSION (buffer));
2854 JBUF_LOCK_CHECK (priv, out_flushing);
2856 if (G_UNLIKELY (priv->last_pt != pt)) {
2859 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2863 /* reset clock-rate so that we get a new one */
2864 priv->clock_rate = -1;
2866 /* Try to get the clock-rate from the caps first if we can. If there are no
2867 * caps we must fire the signal to get the clock-rate. */
2868 if ((caps = gst_pad_get_current_caps (pad))) {
2869 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2870 gst_caps_unref (caps);
2874 if (G_UNLIKELY (priv->clock_rate == -1)) {
2875 /* no clock rate given on the caps, try to get one with the signal */
2876 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2877 pt) == GST_FLOW_FLUSHING)
2880 if (G_UNLIKELY (priv->clock_rate == -1))
2883 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2886 /* don't accept more data on EOS */
2887 if (G_UNLIKELY (priv->eos))
2890 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2891 calculate_jitter (jitterbuffer, dts, rtptime);
2893 if (priv->seqnum_base != -1) {
2896 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2899 GST_DEBUG_OBJECT (jitterbuffer,
2900 "packet seqnum #%d before seqnum-base #%d", seqnum,
2902 gst_buffer_unref (buffer);
2904 } else if (gap > 16384) {
2905 /* From now on don't compare against the seqnum base anymore as
2906 * at some point in the future we will wrap around and also that
2907 * much reordering is very unlikely */
2908 priv->seqnum_base = -1;
2912 expected = priv->next_in_seqnum;
2915 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2917 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2918 priv->max_dropout_time);
2920 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2921 priv->max_misorder_time);
2922 GST_TRACE_OBJECT (jitterbuffer,
2923 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2924 max_dropout, max_misorder);
2926 /* now check against our expected seqnum */
2927 if (G_UNLIKELY (expected == -1)) {
2928 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2930 /* calculate a pts based on rtptime and arrival time (dts) */
2931 pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, rtptime,
2932 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
2934 /* we don't know what the next_in_seqnum should be, wait for the last
2935 * possible moment to push this buffer, maybe we get an earlier seqnum
2937 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
2939 do_next_seqnum = TRUE;
2940 /* take rtptime and pts to calculate packet spacing */
2941 priv->ips_rtptime = rtptime;
2942 priv->ips_pts = pts;
2946 /* now calculate gap */
2947 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2948 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2949 expected, seqnum, gap);
2951 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
2952 /* If we have timers for more than RTP_MAX_DROPOUT packets
2953 * pending this means that we have a huge gap overall. We can
2954 * reset the jitterbuffer at this point because there's
2955 * just too much data missing to be able to do anything
2956 * sensible with the past data. Just try again from the
2958 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
2959 priv->timers->len, max_dropout);
2960 gst_buffer_unref (buffer);
2961 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
2964 /* Special handling of large gaps */
2965 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
2966 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
2967 gap, max_dropout, max_misorder);
2969 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
2971 GST_DEBUG_OBJECT (jitterbuffer,
2972 "Had big gap, waiting for more consecutive packets");
2977 /* We had no huge gap, let's drop all the gap packets */
2978 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2979 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2980 g_queue_clear (&priv->gap_packets);
2982 /* calculate a pts based on rtptime and arrival time (dts) */
2983 pts = rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, rtptime,
2984 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
2986 if (G_LIKELY (gap == 0)) {
2987 /* packet is expected */
2988 calculate_packet_spacing (jitterbuffer, rtptime, pts);
2989 do_next_seqnum = TRUE;
2994 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2995 /* fill in the gap with EXPECTED timers */
2996 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
2997 do_next_seqnum = TRUE;
2999 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3000 do_next_seqnum = FALSE;
3003 /* reset spacing estimation when gap */
3004 priv->ips_rtptime = -1;
3005 priv->ips_pts = GST_CLOCK_TIME_NONE;
3009 if (do_next_seqnum) {
3010 priv->last_in_pts = pts;
3011 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3014 timer = find_timer (jitterbuffer, seqnum);
3015 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3017 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3019 timer->num_rtx_received++;
3022 /* let's check if this buffer is too late, we can only accept packets with
3023 * bigger seqnum than the one we last pushed. */
3024 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3027 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3029 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3030 if (G_UNLIKELY (gap <= 0)) {
3031 if (priv->do_retransmission) {
3032 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3033 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3034 /* Only count the retranmitted packet too late if it has been
3035 * considered lost. If the original packet arrived before the
3036 * retransmitted we just count it as a duplicate. */
3037 if (timer->type != TIMER_TYPE_LOST)
3045 if (already_lost (jitterbuffer, seqnum))
3048 /* let's drop oldest packet if the queue is already full and drop-on-latency
3049 * is set. We can only do this when there actually is a latency. When no
3050 * latency is set, we just pump it in the queue and let the other end push it
3051 * out as fast as possible. */
3052 if (priv->latency_ms && priv->drop_on_latency) {
3054 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3056 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3057 RTPJitterBufferItem *old_item;
3059 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3061 if (IS_DROPABLE (old_item)) {
3062 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3063 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3065 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3066 free_item (old_item);
3068 /* we might have removed some head buffers, signal the pushing thread to
3069 * see if it can push now */
3070 JBUF_SIGNAL_EVENT (priv);
3074 /* If we estimated the DTS, don't consider it in the clock skew calculations
3075 * later. The code above always sets dts to pts or the other way around if
3076 * any of those is valid in the buffer, so we know that if we estimated the
3077 * dts that both are unknown */
3080 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3081 pts, seqnum, 1, rtptime);
3083 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3085 /* now insert the packet into the queue in sorted order. This function returns
3086 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3087 * have a duplicate. */
3088 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3090 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3091 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3096 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3097 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3099 /* we had an unhandled SR, handle it now */
3101 do_handle_sync (jitterbuffer);
3103 if (G_UNLIKELY (head)) {
3104 /* signal addition of new buffer when the _loop is waiting. */
3105 if (G_LIKELY (priv->active))
3106 JBUF_SIGNAL_EVENT (priv);
3108 /* let's unschedule and unblock any waiting buffers. We only want to do this
3109 * when the head buffer changed */
3110 if (G_UNLIKELY (priv->clock_id)) {
3111 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3112 unschedule_current_timer (jitterbuffer);
3116 GST_DEBUG_OBJECT (jitterbuffer,
3117 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3118 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3120 msg = check_buffering_percent (jitterbuffer, percent);
3126 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3133 /* this is not fatal but should be filtered earlier */
3134 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3135 ("Received invalid RTP payload, dropping"));
3136 gst_buffer_unref (buffer);
3141 GST_WARNING_OBJECT (jitterbuffer,
3142 "No clock-rate in caps!, dropping buffer");
3143 gst_buffer_unref (buffer);
3148 ret = priv->srcresult;
3149 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3150 gst_buffer_unref (buffer);
3156 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3157 gst_buffer_unref (buffer);
3162 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3163 " popped, dropping", seqnum, priv->last_popped_seqnum);
3165 gst_buffer_unref (buffer);
3170 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3171 "considered lost", seqnum);
3173 gst_buffer_unref (buffer);
3178 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3180 priv->num_duplicates++;
3186 GST_DEBUG_OBJECT (jitterbuffer,
3187 "Duplicate RTX packet #%d detected, dropping", seqnum);
3188 priv->num_duplicates++;
3189 gst_buffer_unref (buffer);
3195 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3197 guint64 ext_time, elapsed;
3199 GstRtpJitterBufferPrivate *priv;
3201 priv = jitterbuffer->priv;
3202 rtp_time = item->rtptime;
3204 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3205 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3207 ext_time = priv->ext_timestamp;
3208 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3209 if (ext_time < priv->ext_timestamp) {
3210 ext_time = priv->ext_timestamp;
3212 priv->ext_timestamp = ext_time;
3215 if (ext_time > priv->clock_base)
3216 elapsed = ext_time - priv->clock_base;
3220 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3225 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3226 RTPJitterBufferItem * item)
3228 guint64 total, elapsed, left, estimated;
3229 GstClockTime out_time;
3230 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3232 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3233 || priv->clock_base == -1 || priv->clock_rate <= 0)
3236 /* compute the elapsed time */
3237 elapsed = compute_elapsed (jitterbuffer, item);
3239 /* do nothing if elapsed time doesn't increment */
3240 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3243 priv->last_elapsed = elapsed;
3245 /* this is the total time we need to play */
3246 total = priv->npt_stop - priv->npt_start;
3247 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3248 GST_TIME_ARGS (total));
3250 /* this is how much time there is left */
3251 if (total > elapsed)
3252 left = total - elapsed;
3256 /* if we have less time left that the size of the buffer, we will not
3257 * be able to keep it filled, disabled buffering then */
3258 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3259 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3260 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3261 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3264 /* this is the current time as running-time */
3265 out_time = item->pts;
3268 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3270 /* if there is almost nothing left,
3271 * we may never advance enough to end up in the above case */
3272 if (total < GST_SECOND)
3273 estimated = GST_SECOND;
3277 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3278 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3280 if (estimated != -1 && priv->estimated_eos != estimated) {
3281 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3282 priv->estimated_eos = estimated;
3286 /* take a buffer from the queue and push it */
3287 static GstFlowReturn
3288 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3290 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3291 GstFlowReturn result = GST_FLOW_OK;
3292 RTPJitterBufferItem *item;
3293 GstBuffer *outbuf = NULL;
3294 GstEvent *outevent = NULL;
3295 GstQuery *outquery = NULL;
3296 GstClockTime dts, pts;
3298 gboolean do_push = TRUE;
3302 /* when we get here we are ready to pop and push the buffer */
3303 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3307 case ITEM_TYPE_BUFFER:
3309 /* we need to make writable to change the flags and timestamps */
3310 outbuf = gst_buffer_make_writable (item->data);
3312 if (G_UNLIKELY (priv->discont)) {
3313 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3314 * into the jitterbuffer so we can modify now. */
3315 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3316 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3317 priv->discont = FALSE;
3319 if (G_UNLIKELY (priv->ts_discont)) {
3320 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3321 priv->ts_discont = FALSE;
3325 gst_segment_position_from_running_time (&priv->segment,
3326 GST_FORMAT_TIME, item->dts);
3328 gst_segment_position_from_running_time (&priv->segment,
3329 GST_FORMAT_TIME, item->pts);
3331 /* apply timestamp with offset to buffer now */
3332 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3333 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3335 /* update the elapsed time when we need to check against the npt stop time. */
3336 update_estimated_eos (jitterbuffer, item);
3338 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3340 case ITEM_TYPE_LOST:
3341 priv->discont = TRUE;
3345 case ITEM_TYPE_EVENT:
3346 outevent = item->data;
3348 case ITEM_TYPE_QUERY:
3349 outquery = item->data;
3353 /* now we are ready to push the buffer. Save the seqnum and release the lock
3354 * so the other end can push stuff in the queue again. */
3356 priv->last_popped_seqnum = seqnum;
3357 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3359 msg = check_buffering_percent (jitterbuffer, percent);
3366 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3369 case ITEM_TYPE_BUFFER:
3371 GST_DEBUG_OBJECT (jitterbuffer,
3372 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3373 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3374 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3376 result = gst_pad_push (priv->srcpad, outbuf);
3378 JBUF_LOCK_CHECK (priv, out_flushing);
3380 case ITEM_TYPE_LOST:
3381 case ITEM_TYPE_EVENT:
3382 /* We got not enough consecutive packets with a huge gap, we can
3383 * as well just drop them here now on EOS */
3384 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3385 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3386 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3387 g_queue_clear (&priv->gap_packets);
3390 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3391 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3394 gst_pad_push_event (priv->srcpad, outevent);
3396 gst_event_unref (outevent);
3398 result = GST_FLOW_OK;
3400 JBUF_LOCK_CHECK (priv, out_flushing);
3402 case ITEM_TYPE_QUERY:
3406 res = gst_pad_peer_query (priv->srcpad, outquery);
3408 JBUF_LOCK_CHECK (priv, out_flushing);
3409 result = GST_FLOW_OK;
3410 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3411 JBUF_SIGNAL_QUERY (priv, res);
3420 return priv->srcresult;
3424 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3426 /* Peek a buffer and compare the seqnum to the expected seqnum.
3427 * If all is fine, the buffer is pushed.
3428 * If something is wrong, we wait for some event
3430 static GstFlowReturn
3431 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3433 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3434 GstFlowReturn result;
3435 RTPJitterBufferItem *item;
3437 guint32 next_seqnum;
3439 /* only push buffers when PLAYING and active and not buffering */
3440 if (priv->blocked || !priv->active ||
3441 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3442 return GST_FLOW_WAIT;
3445 /* peek a buffer, we're just looking at the sequence number.
3446 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3447 * wait for a timeout or something to change.
3448 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3449 item = rtp_jitter_buffer_peek (priv->jbuf);
3454 /* get the seqnum and the next expected seqnum */
3455 seqnum = item->seqnum;
3457 return pop_and_push_next (jitterbuffer, seqnum);
3460 next_seqnum = priv->next_seqnum;
3462 /* get the gap between this and the previous packet. If we don't know the
3463 * previous packet seqnum assume no gap. */
3464 if (G_UNLIKELY (next_seqnum == -1)) {
3465 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3466 /* we don't know what the next_seqnum should be, the chain function should
3467 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3468 * fires, so wait for that */
3469 result = GST_FLOW_WAIT;
3471 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3473 if (G_LIKELY (gap == 0)) {
3474 /* no missing packet, pop and push */
3475 result = pop_and_push_next (jitterbuffer, seqnum);
3476 } else if (G_UNLIKELY (gap < 0)) {
3477 /* if we have a packet that we already pushed or considered dropped, pop it
3478 * off and get the next packet */
3479 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3480 seqnum, next_seqnum);
3481 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3483 result = GST_FLOW_OK;
3485 /* the chain function has scheduled timers to request retransmission or
3486 * when to consider the packet lost, wait for that */
3487 GST_DEBUG_OBJECT (jitterbuffer,
3488 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3489 next_seqnum, seqnum, gap);
3490 result = GST_FLOW_WAIT;
3498 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3500 return GST_FLOW_EOS;
3502 return GST_FLOW_WAIT;
3508 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3510 GstClockTime rtx_retry_timeout;
3511 GstClockTime rtx_min_retry_timeout;
3513 if (priv->rtx_retry_timeout == -1) {
3514 if (priv->avg_rtx_rtt == 0)
3515 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3517 /* we want to ask for a retransmission after we waited for a
3518 * complete RTT and the additional jitter */
3519 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3521 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3523 /* make sure we don't retry too often. On very low latency networks,
3524 * the RTT and jitter can be very low. */
3525 if (priv->rtx_min_retry_timeout == -1) {
3526 rtx_min_retry_timeout = priv->packet_spacing;
3528 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3530 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3532 return rtx_retry_timeout;
3536 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3537 GstClockTime rtx_retry_timeout)
3539 GstClockTime rtx_retry_period;
3541 if (priv->rtx_retry_period == -1) {
3542 /* we retry up to the configured jitterbuffer size but leaving some
3543 * room for the retransmission to arrive in time */
3544 if (rtx_retry_timeout > priv->latency_ns) {
3545 rtx_retry_period = 0;
3547 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3550 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3552 return rtx_retry_period;
3556 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3557 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3558 3. For very large measurements (> avg * 2), consider them "outliers"
3559 and count them a lot less (1/48th)
3562 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3566 if (priv->avg_rtx_rtt == 0) {
3567 priv->avg_rtx_rtt = rtt;
3571 if (rtt > 2 * priv->avg_rtx_rtt)
3573 else if (rtt > priv->avg_rtx_rtt)
3578 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3582 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3583 GstClockTime dts, gboolean success)
3585 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3589 /* we scheduled a retry for this packet and now we have it */
3590 priv->num_rtx_success++;
3591 /* all the previous retry attempts failed */
3592 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3594 /* All retries failed or was too late */
3595 priv->num_rtx_failed += timer->num_rtx_retry;
3598 /* number of retries before (hopefully) receiving the packet */
3599 if (priv->avg_rtx_num == 0.0)
3600 priv->avg_rtx_num = timer->num_rtx_retry;
3602 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3604 /* Calculate the delay between retransmission request and receiving this
3605 * packet. We have a valid delay if and only if this packet is a response to
3606 * our last request. If not we don't know if this is a response to an
3607 * earlier request and delay could be way off. For RTT is more important
3608 * with correct values than to update for every packet. */
3609 if (timer->num_rtx_retry == timer->num_rtx_received &&
3610 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3611 delay = dts - timer->rtx_last;
3612 update_avg_rtx_rtt (priv, delay);
3617 GST_LOG_OBJECT (jitterbuffer,
3618 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3619 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3620 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3621 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3622 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3623 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3624 GST_TIME_ARGS (priv->avg_rtx_rtt));
3627 /* the timeout for when we expected a packet expired */
3629 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3632 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3634 guint delay, delay_ms, avg_rtx_rtt_ms;
3635 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3636 guint rtx_deadline_ms;
3637 GstClockTime rtx_retry_period;
3638 GstClockTime rtx_retry_timeout;
3641 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3642 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3644 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3645 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3647 delay = timer->rtx_delay + timer->rtx_retry;
3649 delay_ms = GST_TIME_AS_MSECONDS (delay);
3650 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3651 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3652 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3654 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3656 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3657 gst_structure_new ("GstRTPRetransmissionRequest",
3658 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3659 "running-time", G_TYPE_UINT64, timer->rtx_base,
3660 "delay", G_TYPE_UINT, delay_ms,
3661 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3662 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3663 "period", G_TYPE_UINT, rtx_retry_period_ms,
3664 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3665 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3666 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3667 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3669 priv->num_rtx_requests++;
3670 timer->num_rtx_retry++;
3672 GST_OBJECT_LOCK (jitterbuffer);
3673 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3674 timer->rtx_last = gst_clock_get_time (clock);
3675 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3677 timer->rtx_last = now;
3679 GST_OBJECT_UNLOCK (jitterbuffer);
3681 /* calculate the timeout for the next retransmission attempt */
3682 timer->rtx_retry += rtx_retry_timeout;
3683 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3684 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3685 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3686 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3687 if ((priv->rtx_max_retries != -1
3688 && timer->num_rtx_retry >= priv->rtx_max_retries)
3689 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3690 || (timer->rtx_base + rtx_retry_period < now)) {
3691 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3692 /* too many retransmission request, we now convert the timer
3693 * to a lost timer, leave the num_rtx_retry as it is for stats */
3694 timer->type = TIMER_TYPE_LOST;
3695 timer->rtx_delay = 0;
3696 timer->rtx_retry = 0;
3698 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3699 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3702 gst_pad_push_event (priv->sinkpad, event);
3708 /* a packet is lost */
3710 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3713 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3714 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3716 GstEvent *event = NULL;
3717 RTPJitterBufferItem *item;
3719 seqnum = timer->seqnum;
3720 lost_packets = MAX (timer->num, 1);
3721 num_rtx_retry = timer->num_rtx_retry;
3723 /* we had a gap and thus we lost some packets. Create an event for this. */
3724 if (lost_packets > 1)
3725 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3726 seqnum + lost_packets - 1);
3728 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3730 priv->num_lost += lost_packets;
3731 priv->num_rtx_failed += num_rtx_retry;
3733 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3735 /* we now only accept seqnum bigger than this */
3736 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3737 priv->next_in_seqnum = next_in_seqnum;
3738 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3741 /* Avoid creating events if we don't need it. Note that we still need to create
3742 * the lost *ITEM* since it will be used to notify the outgoing thread of
3743 * lost items (so that we can set discont flags and such) */
3744 if (priv->do_lost) {
3745 GstClockTime duration, timestamp;
3746 /* create paket lost event */
3747 timestamp = apply_offset (jitterbuffer, timer->timeout);
3748 duration = timer->duration;
3749 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3750 duration = priv->packet_spacing;
3751 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3752 gst_structure_new ("GstRTPPacketLost",
3753 "seqnum", G_TYPE_UINT, (guint) seqnum,
3754 "timestamp", G_TYPE_UINT64, timestamp,
3755 "duration", G_TYPE_UINT64, duration,
3756 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3758 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3759 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3761 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3762 /* Store info to update stats if the packet arrives too late */
3763 timer_queue_append (priv->rtx_stats_timers, timer,
3764 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3766 remove_timer (jitterbuffer, timer);
3769 JBUF_SIGNAL_EVENT (priv);
3775 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3778 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3780 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3781 remove_timer (jitterbuffer, timer);
3783 /* there was no EOS in the buffer, put one in there now */
3784 queue_event (jitterbuffer, gst_event_new_eos ());
3786 JBUF_SIGNAL_EVENT (priv);
3792 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3795 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3797 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3799 /* timer seqnum might have been obsoleted by caps seqnum-base,
3800 * only mess with current ongoing seqnum if still unknown */
3801 if (priv->next_seqnum == -1)
3802 priv->next_seqnum = timer->seqnum;
3803 remove_timer (jitterbuffer, timer);
3804 JBUF_SIGNAL_EVENT (priv);
3810 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3813 gboolean removed = FALSE;
3815 switch (timer->type) {
3816 case TIMER_TYPE_EXPECTED:
3817 removed = do_expected_timeout (jitterbuffer, timer, now);
3819 case TIMER_TYPE_LOST:
3820 removed = do_lost_timeout (jitterbuffer, timer, now);
3822 case TIMER_TYPE_DEADLINE:
3823 removed = do_deadline_timeout (jitterbuffer, timer, now);
3825 case TIMER_TYPE_EOS:
3826 removed = do_eos_timeout (jitterbuffer, timer, now);
3832 /* called when we need to wait for the next timeout.
3834 * We loop over the array of recorded timeouts and wait for the earliest one.
3835 * When it timed out, do the logic associated with the timer.
3837 * If there are no timers, we wait on a gcond until something new happens.
3840 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3842 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3843 GstClockTime now = 0;
3846 while (priv->timer_running) {
3847 TimerData *timer = NULL;
3848 GstClockTime timer_timeout = -1;
3851 /* If we have a clock, update "now" now with the very
3852 * latest running time we have. If timers are unscheduled below we
3853 * otherwise wouldn't update now (it's only updated when timers
3854 * expire), and also for the very first loop iteration now would
3855 * otherwise always be 0
3857 GST_OBJECT_LOCK (jitterbuffer);
3858 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3860 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3861 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3863 GST_OBJECT_UNLOCK (jitterbuffer);
3865 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3866 GST_TIME_ARGS (now));
3868 /* Clear expired rtx-stats timers */
3869 if (priv->do_retransmission)
3870 timer_queue_clear_until (priv->rtx_stats_timers, now);
3872 /* Iterate "normal" timers */
3873 len = priv->timers->len;
3874 for (i = 0; i < len;) {
3875 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3876 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3877 gboolean save_best = FALSE;
3879 GST_DEBUG_OBJECT (jitterbuffer,
3880 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3881 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3882 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3884 /* Weed out anything too late */
3885 if (test->type == TIMER_TYPE_LOST &&
3886 (test_timeout == -1 || test_timeout <= now)) {
3887 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3888 do_lost_timeout (jitterbuffer, test, now);
3889 if (!priv->timer_running)
3891 /* We don't move the iterator forward since we just removed the current entry,
3892 * but we update the termination condition */
3893 len = priv->timers->len;
3895 /* find the smallest timeout */
3896 if (timer == NULL) {
3898 } else if (timer_timeout == -1) {
3899 /* we already have an immediate timeout, the new timer must be an
3900 * immediate timer with smaller seqnum to become the best */
3901 if (test_timeout == -1
3902 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3903 timer->seqnum) > 0))
3905 } else if (test_timeout == -1) {
3906 /* first immediate timer */
3908 } else if (test_timeout < timer_timeout) {
3911 } else if (test_timeout == timer_timeout
3912 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3913 timer->seqnum) > 0)) {
3914 /* same timer, smaller seqnum */
3919 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3921 timer_timeout = test_timeout;
3926 if (timer && !priv->blocked) {
3928 GstClockTime sync_time;
3931 GstClockTimeDiff clock_jitter;
3933 if (timer_timeout == -1 || timer_timeout <= now) {
3934 /* We have normally removed all lost timers in the loop above */
3935 g_assert (timer->type != TIMER_TYPE_LOST);
3937 do_timeout (jitterbuffer, timer, now);
3938 /* check here, do_timeout could have released the lock */
3939 if (!priv->timer_running)
3944 GST_OBJECT_LOCK (jitterbuffer);
3945 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3947 GST_OBJECT_UNLOCK (jitterbuffer);
3948 /* let's just push if there is no clock */
3949 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3950 now = timer_timeout;
3954 /* prepare for sync against clock */
3955 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3956 /* add latency of peer to get input time */
3957 sync_time += priv->peer_latency;
3959 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3960 " with sync time %" GST_TIME_FORMAT,
3961 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3963 /* create an entry for the clock */
3964 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3965 priv->timer_timeout = timer_timeout;
3966 priv->timer_seqnum = timer->seqnum;
3967 GST_OBJECT_UNLOCK (jitterbuffer);
3969 /* release the lock so that the other end can push stuff or unlock */
3972 ret = gst_clock_id_wait (id, &clock_jitter);
3975 if (!priv->timer_running) {
3976 gst_clock_id_unref (id);
3977 priv->clock_id = NULL;
3981 if (ret != GST_CLOCK_UNSCHEDULED) {
3982 now = timer_timeout + MAX (clock_jitter, 0);
3983 GST_DEBUG_OBJECT (jitterbuffer,
3984 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3985 GST_STIME_ARGS (clock_jitter));
3987 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3989 /* and free the entry */
3990 gst_clock_id_unref (id);
3991 priv->clock_id = NULL;
3993 /* no timers, wait for activity */
3994 JBUF_WAIT_TIMER (priv);
3999 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4004 * This funcion implements the main pushing loop on the source pad.
4006 * It first tries to push as many buffers as possible. If there is a seqnum
4007 * mismatch, we wait for the next timeouts.
4010 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4012 GstRtpJitterBufferPrivate *priv;
4013 GstFlowReturn result = GST_FLOW_OK;
4015 priv = jitterbuffer->priv;
4017 JBUF_LOCK_CHECK (priv, flushing);
4019 result = handle_next_buffer (jitterbuffer);
4020 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4021 /* now wait for the next event */
4022 JBUF_WAIT_EVENT (priv, flushing);
4023 result = GST_FLOW_OK;
4025 } while (result == GST_FLOW_OK);
4026 /* store result for upstream */
4027 priv->srcresult = result;
4028 /* if we get here we need to pause */
4034 result = priv->srcresult;
4041 JBUF_SIGNAL_QUERY (priv, FALSE);
4044 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4045 gst_flow_get_name (result));
4046 gst_pad_pause_task (priv->srcpad);
4047 if (result == GST_FLOW_EOS) {
4048 event = gst_event_new_eos ();
4049 gst_pad_push_event (priv->srcpad, event);
4055 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4056 * some sanity checks and then emit the handle-sync signal with the parameters.
4057 * This function must be called with the LOCK */
4059 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4061 GstRtpJitterBufferPrivate *priv;
4062 guint64 base_rtptime, base_time;
4064 guint64 last_rtptime;
4066 guint64 ext_rtptime, diff;
4067 gboolean valid = TRUE, keep = FALSE;
4069 priv = jitterbuffer->priv;
4071 /* get the last values from the jitterbuffer */
4072 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4073 &clock_rate, &last_rtptime);
4075 clock_base = priv->clock_base;
4076 ext_rtptime = priv->ext_rtptime;
4078 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4079 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4080 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4081 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4083 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4084 /* we keep this SR packet for later. When we get a valid RTP packet the
4085 * above values will be set and we can try to use the SR packet */
4086 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4089 /* we can't accept anything that happened before we did the last resync */
4090 if (base_rtptime > ext_rtptime) {
4091 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4094 /* the SR RTP timestamp must be something close to what we last observed
4095 * in the jitterbuffer */
4096 if (ext_rtptime > last_rtptime) {
4097 /* check how far ahead it is to our RTP timestamps */
4098 diff = ext_rtptime - last_rtptime;
4099 /* if bigger than 1 second, we drop it */
4100 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4102 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4103 clock_rate, 1000)) {
4104 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4105 /* should drop this, but some RTSP servers end up with bogus
4106 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4107 * so still trigger rptbin sync but invalidate RTCP data
4108 * (sync might use other methods) */
4111 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4112 G_GUINT64_FORMAT, last_rtptime, diff);
4118 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4122 s = gst_structure_new ("application/x-rtp-sync",
4123 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4124 "base-time", G_TYPE_UINT64, base_time,
4125 "clock-rate", G_TYPE_UINT, clock_rate,
4126 "clock-base", G_TYPE_UINT64, clock_base,
4127 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4128 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4130 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4131 gst_buffer_replace (&priv->last_sr, NULL);
4133 g_signal_emit (jitterbuffer,
4134 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4136 gst_structure_free (s);
4138 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4139 gst_buffer_replace (&priv->last_sr, NULL);
4143 static GstFlowReturn
4144 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4147 GstRtpJitterBuffer *jitterbuffer;
4148 GstRtpJitterBufferPrivate *priv;
4149 GstFlowReturn ret = GST_FLOW_OK;
4151 GstRTCPPacket packet;
4152 guint64 ext_rtptime;
4154 GstRTCPBuffer rtcp = { NULL, };
4156 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4158 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4159 goto invalid_buffer;
4161 priv = jitterbuffer->priv;
4163 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4165 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4168 /* first packet must be SR or RR or else the validate would have failed */
4169 switch (gst_rtcp_packet_get_type (&packet)) {
4170 case GST_RTCP_TYPE_SR:
4171 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4177 gst_rtcp_buffer_unmap (&rtcp);
4179 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4182 /* convert the RTP timestamp to our extended timestamp, using the same offset
4183 * we used in the jitterbuffer */
4184 ext_rtptime = priv->jbuf->ext_rtptime;
4185 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4187 priv->ext_rtptime = ext_rtptime;
4188 gst_buffer_replace (&priv->last_sr, buffer);
4190 do_handle_sync (jitterbuffer);
4194 gst_buffer_unref (buffer);
4200 /* this is not fatal but should be filtered earlier */
4201 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4202 ("Received invalid RTCP payload, dropping"));
4208 /* this is not fatal but should be filtered earlier */
4209 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4210 ("Received empty RTCP payload, dropping"));
4211 gst_rtcp_buffer_unmap (&rtcp);
4217 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4218 gst_rtcp_buffer_unmap (&rtcp);
4225 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4228 gboolean res = FALSE;
4229 GstRtpJitterBuffer *jitterbuffer;
4230 GstRtpJitterBufferPrivate *priv;
4232 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4233 priv = jitterbuffer->priv;
4235 switch (GST_QUERY_TYPE (query)) {
4236 case GST_QUERY_CAPS:
4238 GstCaps *filter, *caps;
4240 gst_query_parse_caps (query, &filter);
4241 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4242 gst_query_set_caps_result (query, caps);
4243 gst_caps_unref (caps);
4248 if (GST_QUERY_IS_SERIALIZED (query)) {
4249 RTPJitterBufferItem *item;
4252 JBUF_LOCK_CHECK (priv, out_flushing);
4253 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4254 RTP_JITTER_BUFFER_MODE_BUFFER) {
4255 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4256 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4257 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4259 JBUF_SIGNAL_EVENT (priv);
4260 JBUF_WAIT_QUERY (priv, out_flushing);
4261 res = priv->last_query;
4263 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4268 res = gst_pad_query_default (pad, parent, query);
4276 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4284 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4287 GstRtpJitterBuffer *jitterbuffer;
4288 GstRtpJitterBufferPrivate *priv;
4289 gboolean res = FALSE;
4291 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4292 priv = jitterbuffer->priv;
4294 switch (GST_QUERY_TYPE (query)) {
4295 case GST_QUERY_LATENCY:
4297 /* We need to send the query upstream and add the returned latency to our
4299 GstClockTime min_latency, max_latency;
4301 GstClockTime our_latency;
4303 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4304 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4306 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4307 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4308 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4310 /* store this so that we can safely sync on the peer buffers. */
4312 priv->peer_latency = min_latency;
4313 our_latency = priv->latency_ns;
4316 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4317 GST_TIME_ARGS (our_latency));
4319 /* we add some latency but can buffer an infinite amount of time */
4320 min_latency += our_latency;
4323 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4324 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4325 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4327 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4331 case GST_QUERY_POSITION:
4333 GstClockTime start, last_out;
4336 gst_query_parse_position (query, &fmt, NULL);
4337 if (fmt != GST_FORMAT_TIME) {
4338 res = gst_pad_query_default (pad, parent, query);
4343 start = priv->npt_start;
4344 last_out = priv->last_out_time;
4347 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4348 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4349 GST_TIME_ARGS (last_out));
4351 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4352 /* bring 0-based outgoing time to stream time */
4353 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4356 res = gst_pad_query_default (pad, parent, query);
4360 case GST_QUERY_CAPS:
4362 GstCaps *filter, *caps;
4364 gst_query_parse_caps (query, &filter);
4365 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4366 gst_query_set_caps_result (query, caps);
4367 gst_caps_unref (caps);
4372 res = gst_pad_query_default (pad, parent, query);
4380 gst_rtp_jitter_buffer_set_property (GObject * object,
4381 guint prop_id, const GValue * value, GParamSpec * pspec)
4383 GstRtpJitterBuffer *jitterbuffer;
4384 GstRtpJitterBufferPrivate *priv;
4386 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4387 priv = jitterbuffer->priv;
4392 guint new_latency, old_latency;
4394 new_latency = g_value_get_uint (value);
4397 old_latency = priv->latency_ms;
4398 priv->latency_ms = new_latency;
4399 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4400 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4403 /* post message if latency changed, this will inform the parent pipeline
4404 * that a latency reconfiguration is possible/needed. */
4405 if (new_latency != old_latency) {
4406 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4407 GST_TIME_ARGS (new_latency * GST_MSECOND));
4409 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4410 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4414 case PROP_DROP_ON_LATENCY:
4416 priv->drop_on_latency = g_value_get_boolean (value);
4419 case PROP_TS_OFFSET:
4421 priv->ts_offset = g_value_get_int64 (value);
4422 priv->ts_discont = TRUE;
4427 priv->do_lost = g_value_get_boolean (value);
4432 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4435 case PROP_DO_RETRANSMISSION:
4437 priv->do_retransmission = g_value_get_boolean (value);
4440 case PROP_RTX_NEXT_SEQNUM:
4442 priv->rtx_next_seqnum = g_value_get_boolean (value);
4445 case PROP_RTX_DELAY:
4447 priv->rtx_delay = g_value_get_int (value);
4450 case PROP_RTX_MIN_DELAY:
4452 priv->rtx_min_delay = g_value_get_uint (value);
4455 case PROP_RTX_DELAY_REORDER:
4457 priv->rtx_delay_reorder = g_value_get_int (value);
4460 case PROP_RTX_RETRY_TIMEOUT:
4462 priv->rtx_retry_timeout = g_value_get_int (value);
4465 case PROP_RTX_MIN_RETRY_TIMEOUT:
4467 priv->rtx_min_retry_timeout = g_value_get_int (value);
4470 case PROP_RTX_RETRY_PERIOD:
4472 priv->rtx_retry_period = g_value_get_int (value);
4475 case PROP_RTX_MAX_RETRIES:
4477 priv->rtx_max_retries = g_value_get_int (value);
4480 case PROP_RTX_DEADLINE:
4482 priv->rtx_deadline_ms = g_value_get_int (value);
4485 case PROP_RTX_STATS_TIMEOUT:
4487 priv->rtx_stats_timeout = g_value_get_uint (value);
4490 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4492 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4495 case PROP_MAX_DROPOUT_TIME:
4497 priv->max_dropout_time = g_value_get_uint (value);
4500 case PROP_MAX_MISORDER_TIME:
4502 priv->max_misorder_time = g_value_get_uint (value);
4505 case PROP_RFC7273_SYNC:
4507 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4508 g_value_get_boolean (value));
4512 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4518 gst_rtp_jitter_buffer_get_property (GObject * object,
4519 guint prop_id, GValue * value, GParamSpec * pspec)
4521 GstRtpJitterBuffer *jitterbuffer;
4522 GstRtpJitterBufferPrivate *priv;
4524 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4525 priv = jitterbuffer->priv;
4530 g_value_set_uint (value, priv->latency_ms);
4533 case PROP_DROP_ON_LATENCY:
4535 g_value_set_boolean (value, priv->drop_on_latency);
4538 case PROP_TS_OFFSET:
4540 g_value_set_int64 (value, priv->ts_offset);
4545 g_value_set_boolean (value, priv->do_lost);
4550 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4558 if (priv->srcresult != GST_FLOW_OK)
4561 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4563 g_value_set_int (value, percent);
4567 case PROP_DO_RETRANSMISSION:
4569 g_value_set_boolean (value, priv->do_retransmission);
4572 case PROP_RTX_NEXT_SEQNUM:
4574 g_value_set_boolean (value, priv->rtx_next_seqnum);
4577 case PROP_RTX_DELAY:
4579 g_value_set_int (value, priv->rtx_delay);
4582 case PROP_RTX_MIN_DELAY:
4584 g_value_set_uint (value, priv->rtx_min_delay);
4587 case PROP_RTX_DELAY_REORDER:
4589 g_value_set_int (value, priv->rtx_delay_reorder);
4592 case PROP_RTX_RETRY_TIMEOUT:
4594 g_value_set_int (value, priv->rtx_retry_timeout);
4597 case PROP_RTX_MIN_RETRY_TIMEOUT:
4599 g_value_set_int (value, priv->rtx_min_retry_timeout);
4602 case PROP_RTX_RETRY_PERIOD:
4604 g_value_set_int (value, priv->rtx_retry_period);
4607 case PROP_RTX_MAX_RETRIES:
4609 g_value_set_int (value, priv->rtx_max_retries);
4612 case PROP_RTX_DEADLINE:
4614 g_value_set_int (value, priv->rtx_deadline_ms);
4617 case PROP_RTX_STATS_TIMEOUT:
4619 g_value_set_uint (value, priv->rtx_stats_timeout);
4623 g_value_take_boxed (value,
4624 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4626 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4628 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4631 case PROP_MAX_DROPOUT_TIME:
4633 g_value_set_uint (value, priv->max_dropout_time);
4636 case PROP_MAX_MISORDER_TIME:
4638 g_value_set_uint (value, priv->max_misorder_time);
4641 case PROP_RFC7273_SYNC:
4643 g_value_set_boolean (value,
4644 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4648 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4653 static GstStructure *
4654 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4656 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4660 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4661 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4662 "num-lost", G_TYPE_UINT64, priv->num_lost,
4663 "num-late", G_TYPE_UINT64, priv->num_late,
4664 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4665 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4666 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4667 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4668 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4669 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);