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>
11 * This library is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Library General Public
13 * License as published by the Free Software Foundation; either
14 * version 2 of the License, or (at your option) any later version.
16 * This library is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Library General Public License for more details.
21 * You should have received a copy of the GNU Library General Public
22 * License along with this library; if not, write to the
23 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
24 * Boston, MA 02110-1301, USA.
29 * SECTION:element-rtpjitterbuffer
31 * This element reorders and removes duplicate RTP packets as they are received
32 * from a network source.
34 * The element needs the clock-rate of the RTP payload in order to estimate the
35 * delay. This information is obtained either from the caps on the sink pad or,
36 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
37 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
39 * The rtpjitterbuffer will wait for missing packets up to a configurable time
40 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
41 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
42 * property is set, lost packets will result in a custom serialized downstream
43 * event of name GstRTPPacketLost. The lost packet events are usually used by a
44 * depayloader or other element to create concealment data or some other logic
45 * to gracefully handle the missing packets.
47 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
48 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
51 * The jitterbuffer can also be configured to send early retransmission events
52 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
53 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
54 * sends a custom upstream event named GstRTPRetransmissionRequest when the
55 * packet is considered late. The initial expected packet arrival time is
56 * calculated as follows:
58 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
59 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
60 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
61 * packets with different rtptime.
63 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
64 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
65 * previously scheduled timeout is overwritten.
67 * - If seqnum N arrived, all seqnum older than
68 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
69 * immediately. This is to request fast feedback for abonormally reorder
70 * packets before any of the previous timeouts is triggered.
72 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
73 * event. After the initial timeout expires and the retransmission event is
74 * sent, the timeout is scheduled for
75 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
76 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
77 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
78 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
79 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
80 * retransmission requests are sent and the regular logic is performed to
81 * schedule a lost packet as discussed above.
83 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
86 * This element will automatically be used inside rtpbin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
92 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
93 * inserted into the pipeline to smooth out network jitter and to reorder the
94 * out-of-order RTP packets.
105 #include <gst/rtp/gstrtpbuffer.h>
106 #include <gst/net/net.h>
108 #include "gstrtpjitterbuffer.h"
109 #include "rtpjitterbuffer.h"
110 #include "rtpstats.h"
112 #include <gst/glib-compat-private.h>
114 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
115 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
117 /* RTPJitterBuffer signals and args */
120 SIGNAL_REQUEST_PT_MAP,
128 #define DEFAULT_LATENCY_MS 200
129 #define DEFAULT_DROP_ON_LATENCY FALSE
130 #define DEFAULT_TS_OFFSET 0
131 #define DEFAULT_DO_LOST FALSE
132 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
133 #define DEFAULT_PERCENT 0
134 #define DEFAULT_DO_RETRANSMISSION FALSE
135 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
136 #define DEFAULT_RTX_DELAY -1
137 #define DEFAULT_RTX_MIN_DELAY 0
138 #define DEFAULT_RTX_DELAY_REORDER 3
139 #define DEFAULT_RTX_RETRY_TIMEOUT -1
140 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
141 #define DEFAULT_RTX_RETRY_PERIOD -1
142 #define DEFAULT_RTX_MAX_RETRIES -1
143 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
144 #define DEFAULT_MAX_DROPOUT_TIME 60000
145 #define DEFAULT_MAX_MISORDER_TIME 2000
146 #define DEFAULT_RFC7273_SYNC FALSE
148 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
149 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
155 PROP_DROP_ON_LATENCY,
160 PROP_DO_RETRANSMISSION,
161 PROP_RTX_NEXT_SEQNUM,
164 PROP_RTX_DELAY_REORDER,
165 PROP_RTX_RETRY_TIMEOUT,
166 PROP_RTX_MIN_RETRY_TIMEOUT,
167 PROP_RTX_RETRY_PERIOD,
168 PROP_RTX_MAX_RETRIES,
170 PROP_MAX_RTCP_RTP_TIME_DIFF,
171 PROP_MAX_DROPOUT_TIME,
172 PROP_MAX_MISORDER_TIME,
176 #define JBUF_LOCK(priv) G_STMT_START { \
177 GST_TRACE("Locking from thread %p", g_thread_self()); \
178 (g_mutex_lock (&(priv)->jbuf_lock)); \
179 GST_TRACE("Locked from thread %p", g_thread_self()); \
182 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
184 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
187 #define JBUF_UNLOCK(priv) G_STMT_START { \
188 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
189 (g_mutex_unlock (&(priv)->jbuf_lock)); \
192 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
193 GST_DEBUG ("waiting timer"); \
194 (priv)->waiting_timer = TRUE; \
195 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
196 (priv)->waiting_timer = FALSE; \
197 GST_DEBUG ("waiting timer done"); \
199 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
200 if (G_UNLIKELY ((priv)->waiting_timer)) { \
201 GST_DEBUG ("signal timer"); \
202 g_cond_signal (&(priv)->jbuf_timer); \
206 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
207 GST_DEBUG ("waiting event"); \
208 (priv)->waiting_event = TRUE; \
209 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
210 (priv)->waiting_event = FALSE; \
211 GST_DEBUG ("waiting event done"); \
212 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
215 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
216 if (G_UNLIKELY ((priv)->waiting_event)) { \
217 GST_DEBUG ("signal event"); \
218 g_cond_signal (&(priv)->jbuf_event); \
222 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
223 GST_DEBUG ("waiting query"); \
224 (priv)->waiting_query = TRUE; \
225 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
226 (priv)->waiting_query = FALSE; \
227 GST_DEBUG ("waiting query done"); \
228 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
231 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
232 (priv)->last_query = res; \
233 if (G_UNLIKELY ((priv)->waiting_query)) { \
234 GST_DEBUG ("signal query"); \
235 g_cond_signal (&(priv)->jbuf_query); \
240 struct _GstRtpJitterBufferPrivate
242 GstPad *sinkpad, *srcpad;
245 RTPJitterBuffer *jbuf;
247 gboolean waiting_timer;
249 gboolean waiting_event;
251 gboolean waiting_query;
259 gboolean timer_running;
260 GThread *timer_thread;
265 gboolean drop_on_latency;
268 gboolean do_retransmission;
269 gboolean rtx_next_seqnum;
272 gint rtx_delay_reorder;
273 gint rtx_retry_timeout;
274 gint rtx_min_retry_timeout;
275 gint rtx_retry_period;
276 gint rtx_max_retries;
277 gint max_rtcp_rtp_time_diff;
278 guint32 max_dropout_time;
279 guint32 max_misorder_time;
281 /* the last seqnum we pushed out */
282 guint32 last_popped_seqnum;
283 /* the next expected seqnum we push */
285 /* seqnum-base, if known */
287 /* last output time */
288 GstClockTime last_out_time;
289 /* last valid input timestamp and rtptime pair */
290 GstClockTime ips_dts;
292 GstClockTime packet_spacing;
296 /* the next expected seqnum we receive */
297 GstClockTime last_in_dts;
298 guint32 next_in_seqnum;
302 /* start and stop ranges */
303 GstClockTime npt_start;
304 GstClockTime npt_stop;
305 guint64 ext_timestamp;
306 guint64 last_elapsed;
307 guint64 estimated_eos;
314 /* clock rate and rtp timestamp offset */
318 gint64 prev_ts_offset;
320 /* when we are shutting down */
321 GstFlowReturn srcresult;
327 GstClockTime timer_timeout;
328 guint16 timer_seqnum;
329 /* the latency of the upstream peer, we have to take this into account when
330 * synchronizing the buffers. */
331 GstClockTime peer_latency;
335 /* some accounting */
337 guint64 num_duplicates;
338 guint64 num_rtx_requests;
339 guint64 num_rtx_success;
340 guint64 num_rtx_failed;
343 RTPPacketRateCtx packet_rate_ctx;
346 GstClockTime last_dts;
347 guint64 last_rtptime;
348 GstClockTime avg_jitter;
365 GstClockTime timeout;
366 GstClockTime duration;
367 GstClockTime rtx_base;
368 GstClockTime rtx_delay;
369 GstClockTime rtx_retry;
370 GstClockTime rtx_last;
374 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
375 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
376 GstRtpJitterBufferPrivate))
378 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
379 GST_STATIC_PAD_TEMPLATE ("sink",
382 GST_STATIC_CAPS ("application/x-rtp"
383 /* "clock-rate = (int) [ 1, 2147483647 ], "
384 * "payload = (int) , "
385 * "encoding-name = (string) "
389 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
390 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
393 GST_STATIC_CAPS ("application/x-rtcp")
396 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
397 GST_STATIC_PAD_TEMPLATE ("src",
400 GST_STATIC_CAPS ("application/x-rtp"
401 /* "payload = (int) , "
402 * "clock-rate = (int) , "
403 * "encoding-name = (string) "
407 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
409 #define gst_rtp_jitter_buffer_parent_class parent_class
410 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
412 /* object overrides */
413 static void gst_rtp_jitter_buffer_set_property (GObject * object,
414 guint prop_id, const GValue * value, GParamSpec * pspec);
415 static void gst_rtp_jitter_buffer_get_property (GObject * object,
416 guint prop_id, GValue * value, GParamSpec * pspec);
417 static void gst_rtp_jitter_buffer_finalize (GObject * object);
419 /* element overrides */
420 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
421 * element, GstStateChange transition);
422 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
423 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
424 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
426 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
427 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
431 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
432 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
435 /* sinkpad overrides */
436 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
437 GstObject * parent, GstEvent * event);
438 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
439 GstObject * parent, GstBuffer * buffer);
441 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
442 GstObject * parent, GstEvent * event);
443 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
444 GstObject * parent, GstBuffer * buffer);
446 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
447 GstObject * parent, GstQuery * query);
449 /* srcpad overrides */
450 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
451 GstObject * parent, GstEvent * event);
452 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
453 GstObject * parent, GstPadMode mode, gboolean active);
454 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
455 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
456 GstObject * parent, GstQuery * query);
459 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
461 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
462 gboolean active, guint64 base_time);
463 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
465 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
466 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
468 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
470 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
474 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
476 GObjectClass *gobject_class;
477 GstElementClass *gstelement_class;
479 gobject_class = (GObjectClass *) klass;
480 gstelement_class = (GstElementClass *) klass;
482 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
484 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
486 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
487 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
490 * GstRtpJitterBuffer:latency:
492 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
493 * for at most this time.
495 g_object_class_install_property (gobject_class, PROP_LATENCY,
496 g_param_spec_uint ("latency", "Buffer latency in ms",
497 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
498 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
500 * GstRtpJitterBuffer:drop-on-latency:
502 * Drop oldest buffers when the queue is completely filled.
504 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
505 g_param_spec_boolean ("drop-on-latency",
506 "Drop buffers when maximum latency is reached",
507 "Tells the jitterbuffer to never exceed the given latency in size",
508 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
510 * GstRtpJitterBuffer:ts-offset:
512 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
513 * This is mainly used to ensure interstream synchronisation.
515 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
516 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
517 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
518 G_MAXINT64, DEFAULT_TS_OFFSET,
519 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
522 * GstRtpJitterBuffer:do-lost:
524 * Send out a GstRTPPacketLost event downstream when a packet is considered
527 g_object_class_install_property (gobject_class, PROP_DO_LOST,
528 g_param_spec_boolean ("do-lost", "Do Lost",
529 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
530 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
533 * GstRtpJitterBuffer:mode:
535 * Control the buffering and timestamping mode used by the jitterbuffer.
537 g_object_class_install_property (gobject_class, PROP_MODE,
538 g_param_spec_enum ("mode", "Mode",
539 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
540 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
542 * GstRtpJitterBuffer:percent:
544 * The percent of the jitterbuffer that is filled.
546 g_object_class_install_property (gobject_class, PROP_PERCENT,
547 g_param_spec_int ("percent", "percent",
548 "The buffer filled percent", 0, 100,
549 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
551 * GstRtpJitterBuffer:do-retransmission:
553 * Send out a GstRTPRetransmission event upstream when a packet is considered
554 * late and should be retransmitted.
558 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
559 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
560 "Send retransmission events upstream when a packet is late",
561 DEFAULT_DO_RETRANSMISSION,
562 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
565 * GstRtpJitterBuffer:rtx-next-seqnum
567 * Estimate when the next packet should arrive and schedule a retransmission
569 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
570 * for packet N+1. So it will be requested if it does not arrive at the expected time.
571 * The expected time is calculated using the dts of N and the packet spacing.
575 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
576 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
577 "Estimate when the next packet should arrive and schedule a "
578 "retransmission request for it.",
579 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
582 * GstRtpJitterBuffer:rtx-delay:
584 * When a packet did not arrive at the expected time, wait this extra amount
585 * of time before sending a retransmission event.
587 * When -1 is used, the max jitter will be used as extra delay.
591 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
592 g_param_spec_int ("rtx-delay", "RTX Delay",
593 "Extra time in ms to wait before sending retransmission "
594 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
595 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
598 * GstRtpJitterBuffer:rtx-min-delay:
600 * When a packet did not arrive at the expected time, wait at least this extra amount
601 * of time before sending a retransmission event.
605 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
606 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
607 "Minimum time in ms to wait before sending retransmission "
608 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
609 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRtpJitterBuffer:rtx-delay-reorder:
613 * Assume that a retransmission event should be sent when we see
614 * this much packet reordering.
616 * When -1 is used, the value will be estimated based on observed packet
621 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
622 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
623 "Sending retransmission event when this much reordering (-1 automatic)",
624 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
625 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
627 * GstRtpJitterBuffer::rtx-retry-timeout:
629 * When no packet has been received after sending a retransmission event
630 * for this time, retry sending a retransmission event.
632 * When -1 is used, the value will be estimated based on observed round
637 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
638 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
639 "Retry sending a transmission event after this timeout in "
640 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
641 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
643 * GstRtpJitterBuffer::rtx-min-retry-timeout:
645 * The minimum amount of time between retry timeouts. When
646 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
647 * minimum interval between retry timeouts.
649 * When -1 is used, the value will be estimated based on the
654 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
655 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
656 "Minimum timeout between sending a transmission event in "
657 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
658 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
660 * GstRtpJitterBuffer:rtx-retry-period:
662 * The amount of time to try to get a retransmission.
664 * When -1 is used, the value will be estimated based on the jitterbuffer
665 * latency and the observed round trip time.
669 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
670 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
671 "Try to get a retransmission for this many ms "
672 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
673 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
675 * GstRtpJitterBuffer:rtx-max-retries:
677 * The maximum number of retries to request a retransmission.
679 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
680 * When -1 is used, the number of retransmission request will not be limited.
684 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
685 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
686 "The maximum number of retries to request a retransmission. "
687 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
688 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
690 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
691 g_param_spec_uint ("max-dropout-time", "Max dropout time",
692 "The maximum time (milliseconds) of missing packets tolerated.",
693 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
694 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
696 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
697 g_param_spec_uint ("max-misorder-time", "Max misorder time",
698 "The maximum time (milliseconds) of misordered packets tolerated.",
699 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
700 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
702 * GstRtpJitterBuffer:stats:
704 * Various jitterbuffer statistics. This property returns a GstStructure
705 * with name application/x-rtp-jitterbuffer-stats with the following fields:
711 * <classname>"rtx-count"</classname>:
712 * the number of retransmissions requested.
718 * <classname>"rtx-success-count"</classname>:
719 * the number of successful retransmissions.
725 * <classname>"rtx-per-packet"</classname>:
726 * average number of RTX per packet.
732 * <classname>"rtx-rtt"</classname>:
733 * average round trip time per RTX.
740 g_object_class_install_property (gobject_class, PROP_STATS,
741 g_param_spec_boxed ("stats", "Statistics",
742 "Various statistics", GST_TYPE_STRUCTURE,
743 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
746 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
748 * The maximum amount of time in ms that the RTP time in the RTCP SRs
749 * is allowed to be ahead of the last RTP packet we received. Use
750 * -1 to disable ignoring of RTCP packets.
754 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
755 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
756 "Maximum amount of time in ms that the RTP time in RTCP SRs "
757 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
758 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
759 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
761 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
762 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
763 "Synchronize received streams to the RFC7273 clock "
764 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
765 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
768 * GstRtpJitterBuffer::request-pt-map:
769 * @buffer: the object which received the signal
772 * Request the payload type as #GstCaps for @pt.
774 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
775 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
776 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
777 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
778 GST_TYPE_CAPS, 1, G_TYPE_UINT);
780 * GstRtpJitterBuffer::handle-sync:
781 * @buffer: the object which received the signal
782 * @struct: a GstStructure containing sync values.
784 * Be notified of new sync values.
786 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
787 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
788 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
789 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
790 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
793 * GstRtpJitterBuffer::on-npt-stop:
794 * @buffer: the object which received the signal
796 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
797 * the npt-stop position.
799 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
800 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
801 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
802 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
803 G_TYPE_NONE, 0, G_TYPE_NONE);
806 * GstRtpJitterBuffer::clear-pt-map:
807 * @buffer: the object which received the signal
809 * Invalidate the clock-rate as obtained with the
810 * #GstRtpJitterBuffer::request-pt-map signal.
812 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
813 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
814 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
815 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
816 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
819 * GstRtpJitterBuffer::set-active:
820 * @buffer: the object which received the signal
822 * Start pushing out packets with the given base time. This signal is only
823 * useful in buffering mode.
825 * Returns: the time of the last pushed packet.
827 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
828 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
829 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
830 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
831 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
834 gstelement_class->change_state =
835 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
836 gstelement_class->request_new_pad =
837 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
838 gstelement_class->release_pad =
839 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
840 gstelement_class->provide_clock =
841 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
842 gstelement_class->set_clock =
843 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
845 gst_element_class_add_static_pad_template (gstelement_class,
846 &gst_rtp_jitter_buffer_src_template);
847 gst_element_class_add_static_pad_template (gstelement_class,
848 &gst_rtp_jitter_buffer_sink_template);
849 gst_element_class_add_static_pad_template (gstelement_class,
850 &gst_rtp_jitter_buffer_sink_rtcp_template);
852 gst_element_class_set_static_metadata (gstelement_class,
853 "RTP packet jitter-buffer", "Filter/Network/RTP",
854 "A buffer that deals with network jitter and other transmission faults",
855 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
856 "Wim Taymans <wim.taymans@gmail.com>");
858 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
859 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
861 GST_DEBUG_CATEGORY_INIT
862 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
866 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
868 GstRtpJitterBufferPrivate *priv;
870 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
871 jitterbuffer->priv = priv;
873 priv->latency_ms = DEFAULT_LATENCY_MS;
874 priv->latency_ns = priv->latency_ms * GST_MSECOND;
875 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
876 priv->do_lost = DEFAULT_DO_LOST;
877 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
878 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
879 priv->rtx_delay = DEFAULT_RTX_DELAY;
880 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
881 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
882 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
883 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
884 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
885 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
886 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
887 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
888 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
891 priv->last_rtptime = -1;
892 priv->avg_jitter = 0;
893 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
894 priv->jbuf = rtp_jitter_buffer_new ();
895 g_mutex_init (&priv->jbuf_lock);
896 g_cond_init (&priv->jbuf_timer);
897 g_cond_init (&priv->jbuf_event);
898 g_cond_init (&priv->jbuf_query);
899 g_queue_init (&priv->gap_packets);
901 /* reset skew detection initialy */
902 rtp_jitter_buffer_reset_skew (priv->jbuf);
903 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
904 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
908 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
911 gst_pad_set_activatemode_function (priv->srcpad,
912 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
913 gst_pad_set_query_function (priv->srcpad,
914 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
915 gst_pad_set_event_function (priv->srcpad,
916 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
919 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
922 gst_pad_set_chain_function (priv->sinkpad,
923 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
924 gst_pad_set_event_function (priv->sinkpad,
925 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
926 gst_pad_set_query_function (priv->sinkpad,
927 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
929 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
930 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
932 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
935 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
937 #define ITEM_TYPE_BUFFER 0
938 #define ITEM_TYPE_LOST 1
939 #define ITEM_TYPE_EVENT 2
940 #define ITEM_TYPE_QUERY 3
942 static RTPJitterBufferItem *
943 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
944 guint seqnum, guint count, guint rtptime)
946 RTPJitterBufferItem *item;
948 item = g_slice_new (RTPJitterBufferItem);
955 item->seqnum = seqnum;
957 item->rtptime = rtptime;
963 free_item (RTPJitterBufferItem * item)
965 g_return_if_fail (item != NULL);
967 if (item->data && item->type != ITEM_TYPE_QUERY)
968 gst_mini_object_unref (item->data);
969 g_slice_free (RTPJitterBufferItem, item);
973 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
975 GList **l = user_data;
977 if (item->data && item->type == ITEM_TYPE_EVENT
978 && GST_EVENT_IS_STICKY (item->data)) {
979 *l = g_list_prepend (*l, item->data);
980 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
981 gst_mini_object_unref (item->data);
983 g_slice_free (RTPJitterBufferItem, item);
987 gst_rtp_jitter_buffer_finalize (GObject * object)
989 GstRtpJitterBuffer *jitterbuffer;
990 GstRtpJitterBufferPrivate *priv;
992 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
993 priv = jitterbuffer->priv;
995 g_array_free (priv->timers, TRUE);
996 g_mutex_clear (&priv->jbuf_lock);
997 g_cond_clear (&priv->jbuf_timer);
998 g_cond_clear (&priv->jbuf_event);
999 g_cond_clear (&priv->jbuf_query);
1001 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1002 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1003 g_queue_clear (&priv->gap_packets);
1004 g_object_unref (priv->jbuf);
1006 G_OBJECT_CLASS (parent_class)->finalize (object);
1009 static GstIterator *
1010 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1012 GstRtpJitterBuffer *jitterbuffer;
1013 GstPad *otherpad = NULL;
1014 GstIterator *it = NULL;
1015 GValue val = { 0, };
1017 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1019 if (pad == jitterbuffer->priv->sinkpad) {
1020 otherpad = jitterbuffer->priv->srcpad;
1021 } else if (pad == jitterbuffer->priv->srcpad) {
1022 otherpad = jitterbuffer->priv->sinkpad;
1023 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1024 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1028 g_value_init (&val, GST_TYPE_PAD);
1029 g_value_set_object (&val, otherpad);
1030 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1031 g_value_unset (&val);
1038 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1040 GstRtpJitterBufferPrivate *priv;
1042 priv = jitterbuffer->priv;
1044 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1047 gst_pad_new_from_static_template
1048 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1049 gst_pad_set_chain_function (priv->rtcpsinkpad,
1050 gst_rtp_jitter_buffer_chain_rtcp);
1051 gst_pad_set_event_function (priv->rtcpsinkpad,
1052 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1053 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1054 gst_rtp_jitter_buffer_iterate_internal_links);
1055 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1056 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1058 return priv->rtcpsinkpad;
1062 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1064 GstRtpJitterBufferPrivate *priv;
1066 priv = jitterbuffer->priv;
1068 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1070 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1072 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1073 priv->rtcpsinkpad = NULL;
1077 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1078 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1080 GstRtpJitterBuffer *jitterbuffer;
1081 GstElementClass *klass;
1083 GstRtpJitterBufferPrivate *priv;
1085 g_return_val_if_fail (templ != NULL, NULL);
1086 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1088 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1089 priv = jitterbuffer->priv;
1090 klass = GST_ELEMENT_GET_CLASS (element);
1092 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1094 /* figure out the template */
1095 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1096 if (priv->rtcpsinkpad != NULL)
1099 result = create_rtcp_sink (jitterbuffer);
1101 goto wrong_template;
1108 g_warning ("rtpjitterbuffer: this is not our template");
1113 g_warning ("rtpjitterbuffer: pad already requested");
1119 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1121 GstRtpJitterBuffer *jitterbuffer;
1122 GstRtpJitterBufferPrivate *priv;
1124 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1125 g_return_if_fail (GST_IS_PAD (pad));
1127 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1128 priv = jitterbuffer->priv;
1130 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1132 if (priv->rtcpsinkpad == pad) {
1133 remove_rtcp_sink (jitterbuffer);
1142 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1148 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1150 return gst_system_clock_obtain ();
1154 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1156 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1158 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1160 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1164 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1166 GstRtpJitterBufferPrivate *priv;
1168 priv = jitterbuffer->priv;
1170 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1173 priv->clock_rate = -1;
1174 /* do not clear current content, but refresh state for new arrival */
1175 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1176 rtp_jitter_buffer_reset_skew (priv->jbuf);
1181 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1184 GstRtpJitterBufferPrivate *priv;
1185 GstClockTime last_out;
1186 RTPJitterBufferItem *item;
1191 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1192 active, GST_TIME_ARGS (offset));
1194 if (active != priv->active) {
1195 /* add the amount of time spent in paused to the output offset. All
1196 * outgoing buffers will have this offset applied to their timestamps in
1197 * order to make them arrive in time in the sink. */
1198 priv->out_offset = offset;
1199 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1200 GST_TIME_ARGS (priv->out_offset));
1201 priv->active = active;
1202 JBUF_SIGNAL_EVENT (priv);
1205 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1207 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1208 /* head buffer timestamp and offset gives our output time */
1209 last_out = item->dts + priv->ts_offset;
1211 /* use last known time when the buffer is empty */
1212 last_out = priv->last_out_time;
1220 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1222 GstRtpJitterBuffer *jitterbuffer;
1223 GstRtpJitterBufferPrivate *priv;
1228 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1229 priv = jitterbuffer->priv;
1231 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1233 caps = gst_pad_peer_query_caps (other, filter);
1235 templ = gst_pad_get_pad_template_caps (pad);
1237 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1242 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1244 intersect = gst_caps_intersect (caps, templ);
1245 gst_caps_unref (caps);
1246 gst_caps_unref (templ);
1250 gst_object_unref (jitterbuffer);
1256 * Must be called with JBUF_LOCK held
1260 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1263 GstRtpJitterBufferPrivate *priv;
1264 GstStructure *caps_struct;
1267 const gchar *ts_refclk, *mediaclk;
1269 priv = jitterbuffer->priv;
1271 /* first parse the caps */
1272 caps_struct = gst_caps_get_structure (caps, 0);
1274 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1276 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1277 * measure the amount of data in the buffer */
1278 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1281 if (priv->clock_rate <= 0)
1284 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1286 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1288 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1289 * can use this to track the amount of time elapsed on the sender. */
1290 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1291 priv->clock_base = val;
1293 priv->clock_base = -1;
1295 priv->ext_timestamp = priv->clock_base;
1297 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1300 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1301 /* first expected seqnum, only update when we didn't have a previous base. */
1302 if (priv->next_in_seqnum == -1)
1303 priv->next_in_seqnum = val;
1304 if (priv->next_seqnum == -1) {
1305 priv->next_seqnum = val;
1306 JBUF_SIGNAL_EVENT (priv);
1308 priv->seqnum_base = val;
1310 priv->seqnum_base = -1;
1313 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1315 /* the start and stop times. The seqnum-base corresponds to the start time. We
1316 * will keep track of the seqnums on the output and when we reach the one
1317 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1318 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1319 priv->npt_start = tval;
1321 priv->npt_start = 0;
1323 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1324 priv->npt_stop = tval;
1326 priv->npt_stop = -1;
1328 GST_DEBUG_OBJECT (jitterbuffer,
1329 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1330 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1332 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1333 GstClock *clock = NULL;
1334 guint64 clock_offset = -1;
1336 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1339 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1340 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1341 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1343 const gchar *host, *portstr;
1347 host = ts_refclk + sizeof ("ntp=") - 1;
1348 if (host[0] == '[') {
1350 portstr = strchr (host, ']');
1351 if (portstr && portstr[1] == ':')
1352 portstr = portstr + 1;
1356 portstr = strrchr (host, ':');
1360 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1364 hostname = g_strndup (host, (portstr - host));
1366 hostname = g_strdup (host);
1368 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1371 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1372 const gchar *domainstr =
1373 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1376 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1379 clock = gst_ptp_clock_new (NULL, domain);
1381 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1384 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1385 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1387 if (!g_str_has_prefix (mediaclk, "direct=")
1388 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1389 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1390 if (strstr (mediaclk, "rate=") != NULL) {
1391 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1396 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1398 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1406 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1411 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1417 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1419 GstRtpJitterBufferPrivate *priv;
1421 priv = jitterbuffer->priv;
1424 /* mark ourselves as flushing */
1425 priv->srcresult = GST_FLOW_FLUSHING;
1426 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1427 /* this unblocks any waiting pops on the src pad task */
1428 JBUF_SIGNAL_EVENT (priv);
1429 JBUF_SIGNAL_QUERY (priv, FALSE);
1434 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1436 GstRtpJitterBufferPrivate *priv;
1438 priv = jitterbuffer->priv;
1441 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1442 /* Mark as non flushing */
1443 priv->srcresult = GST_FLOW_OK;
1444 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1445 priv->last_popped_seqnum = -1;
1446 priv->last_out_time = -1;
1447 priv->next_seqnum = -1;
1448 priv->seqnum_base = -1;
1449 priv->ips_rtptime = -1;
1450 priv->ips_dts = GST_CLOCK_TIME_NONE;
1451 priv->packet_spacing = 0;
1452 priv->next_in_seqnum = -1;
1453 priv->clock_rate = -1;
1456 priv->estimated_eos = -1;
1457 priv->last_elapsed = 0;
1458 priv->ext_timestamp = -1;
1459 priv->avg_jitter = 0;
1460 priv->last_dts = -1;
1461 priv->last_rtptime = -1;
1462 priv->last_in_dts = 0;
1463 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1464 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1465 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1466 rtp_jitter_buffer_reset_skew (priv->jbuf);
1467 remove_all_timers (jitterbuffer);
1468 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1469 g_queue_clear (&priv->gap_packets);
1474 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1475 GstPadMode mode, gboolean active)
1478 GstRtpJitterBuffer *jitterbuffer = NULL;
1480 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1483 case GST_PAD_MODE_PUSH:
1485 /* allow data processing */
1486 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1488 /* start pushing out buffers */
1489 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1490 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1491 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1493 /* make sure all data processing stops ASAP */
1494 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1496 /* NOTE this will hardlock if the state change is called from the src pad
1497 * task thread because we will _join() the thread. */
1498 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1499 result = gst_pad_stop_task (pad);
1509 static GstStateChangeReturn
1510 gst_rtp_jitter_buffer_change_state (GstElement * element,
1511 GstStateChange transition)
1513 GstRtpJitterBuffer *jitterbuffer;
1514 GstRtpJitterBufferPrivate *priv;
1515 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1517 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1518 priv = jitterbuffer->priv;
1520 switch (transition) {
1521 case GST_STATE_CHANGE_NULL_TO_READY:
1523 case GST_STATE_CHANGE_READY_TO_PAUSED:
1525 /* reset negotiated values */
1526 priv->clock_rate = -1;
1527 priv->clock_base = -1;
1528 priv->peer_latency = 0;
1530 /* block until we go to PLAYING */
1531 priv->blocked = TRUE;
1532 priv->timer_running = TRUE;
1533 priv->timer_thread =
1534 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1537 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1539 /* unblock to allow streaming in PLAYING */
1540 priv->blocked = FALSE;
1541 JBUF_SIGNAL_EVENT (priv);
1542 JBUF_SIGNAL_TIMER (priv);
1549 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1551 switch (transition) {
1552 case GST_STATE_CHANGE_READY_TO_PAUSED:
1553 /* we are a live element because we sync to the clock, which we can only
1554 * do in the PLAYING state */
1555 if (ret != GST_STATE_CHANGE_FAILURE)
1556 ret = GST_STATE_CHANGE_NO_PREROLL;
1558 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1560 /* block to stop streaming when PAUSED */
1561 priv->blocked = TRUE;
1562 unschedule_current_timer (jitterbuffer);
1564 if (ret != GST_STATE_CHANGE_FAILURE)
1565 ret = GST_STATE_CHANGE_NO_PREROLL;
1567 case GST_STATE_CHANGE_PAUSED_TO_READY:
1569 gst_buffer_replace (&priv->last_sr, NULL);
1570 priv->timer_running = FALSE;
1571 unschedule_current_timer (jitterbuffer);
1572 JBUF_SIGNAL_TIMER (priv);
1573 JBUF_SIGNAL_QUERY (priv, FALSE);
1575 g_thread_join (priv->timer_thread);
1576 priv->timer_thread = NULL;
1578 case GST_STATE_CHANGE_READY_TO_NULL:
1588 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1591 gboolean ret = TRUE;
1592 GstRtpJitterBuffer *jitterbuffer;
1593 GstRtpJitterBufferPrivate *priv;
1595 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1596 priv = jitterbuffer->priv;
1598 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1600 switch (GST_EVENT_TYPE (event)) {
1601 case GST_EVENT_LATENCY:
1603 GstClockTime latency;
1605 gst_event_parse_latency (event, &latency);
1607 GST_DEBUG_OBJECT (jitterbuffer,
1608 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1611 /* adjust the overall buffer delay to the total pipeline latency in
1612 * buffering mode because if downstream consumes too fast (because of
1613 * large latency or queues, we would start rebuffering again. */
1614 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1615 RTP_JITTER_BUFFER_MODE_BUFFER) {
1616 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1620 ret = gst_pad_push_event (priv->sinkpad, event);
1624 ret = gst_pad_push_event (priv->sinkpad, event);
1631 /* handles and stores the event in the jitterbuffer, must be called with
1634 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1636 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1637 RTPJitterBufferItem *item;
1640 switch (GST_EVENT_TYPE (event)) {
1641 case GST_EVENT_CAPS:
1645 gst_event_parse_caps (event, &caps);
1646 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1649 case GST_EVENT_SEGMENT:
1650 gst_event_copy_segment (event, &priv->segment);
1652 /* we need time for now */
1653 if (priv->segment.format != GST_FORMAT_TIME)
1654 goto newseg_wrong_format;
1656 GST_DEBUG_OBJECT (jitterbuffer,
1657 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1661 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1668 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1669 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1670 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1672 JBUF_SIGNAL_EVENT (priv);
1677 newseg_wrong_format:
1679 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1680 gst_event_unref (event);
1686 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1689 gboolean ret = TRUE;
1690 GstRtpJitterBuffer *jitterbuffer;
1691 GstRtpJitterBufferPrivate *priv;
1693 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1694 priv = jitterbuffer->priv;
1696 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1698 switch (GST_EVENT_TYPE (event)) {
1699 case GST_EVENT_FLUSH_START:
1700 ret = gst_pad_push_event (priv->srcpad, event);
1701 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1702 /* wait for the loop to go into PAUSED */
1703 gst_pad_pause_task (priv->srcpad);
1705 case GST_EVENT_FLUSH_STOP:
1706 ret = gst_pad_push_event (priv->srcpad, event);
1708 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1709 GST_PAD_MODE_PUSH, TRUE);
1712 if (GST_EVENT_IS_SERIALIZED (event)) {
1713 /* serialized events go in the queue */
1715 if (priv->srcresult != GST_FLOW_OK) {
1716 /* Errors in sticky event pushing are no problem and ignored here
1717 * as they will cause more meaningful errors during data flow.
1718 * For EOS events, that are not followed by data flow, we still
1719 * return FALSE here though.
1721 if (!GST_EVENT_IS_STICKY (event) ||
1722 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1723 goto out_flow_error;
1725 /* refuse more events on EOS */
1728 ret = queue_event (jitterbuffer, event);
1731 /* non-serialized events are forwarded downstream immediately */
1732 ret = gst_pad_push_event (priv->srcpad, event);
1741 GST_DEBUG_OBJECT (jitterbuffer,
1742 "refusing event, we have a downstream flow error: %s",
1743 gst_flow_get_name (priv->srcresult));
1745 gst_event_unref (event);
1750 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1752 gst_event_unref (event);
1758 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1761 gboolean ret = TRUE;
1762 GstRtpJitterBuffer *jitterbuffer;
1764 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1766 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1768 switch (GST_EVENT_TYPE (event)) {
1769 case GST_EVENT_FLUSH_START:
1770 gst_event_unref (event);
1772 case GST_EVENT_FLUSH_STOP:
1773 gst_event_unref (event);
1776 ret = gst_pad_event_default (pad, parent, event);
1784 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1785 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1786 * GST_FLOW_FLUSHING when the element is shutting down. On success
1787 * GST_FLOW_OK is returned.
1789 static GstFlowReturn
1790 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1794 GValue args[2] = { {0}, {0} };
1798 g_value_init (&args[0], GST_TYPE_ELEMENT);
1799 g_value_set_object (&args[0], jitterbuffer);
1800 g_value_init (&args[1], G_TYPE_UINT);
1801 g_value_set_uint (&args[1], pt);
1803 g_value_init (&ret, GST_TYPE_CAPS);
1804 g_value_set_boxed (&ret, NULL);
1806 JBUF_UNLOCK (jitterbuffer->priv);
1807 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1809 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1811 g_value_unset (&args[0]);
1812 g_value_unset (&args[1]);
1813 caps = (GstCaps *) g_value_dup_boxed (&ret);
1814 g_value_unset (&ret);
1818 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1819 gst_caps_unref (caps);
1821 if (G_UNLIKELY (!res))
1829 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1830 return GST_FLOW_ERROR;
1834 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1835 return GST_FLOW_FLUSHING;
1839 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1840 return GST_FLOW_ERROR;
1844 /* call with jbuf lock held */
1846 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1848 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1849 GstMessage *message = NULL;
1854 /* Post a buffering message */
1855 if (priv->last_percent != percent) {
1856 priv->last_percent = percent;
1858 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1859 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1866 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1868 GstRtpJitterBufferPrivate *priv;
1870 priv = jitterbuffer->priv;
1872 if (timestamp == -1)
1875 /* apply the timestamp offset, this is used for inter stream sync */
1876 timestamp += priv->ts_offset;
1877 /* add the offset, this is used when buffering */
1878 timestamp += priv->out_offset;
1884 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1886 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1887 TimerData *timer = NULL;
1890 len = priv->timers->len;
1891 for (i = 0; i < len; i++) {
1892 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1893 if (test->seqnum == seqnum && test->type == type) {
1902 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1904 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1906 if (priv->clock_id) {
1907 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1908 gst_clock_id_unschedule (priv->clock_id);
1909 priv->clock_id = NULL;
1914 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1916 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1917 GstClockTime test_timeout;
1919 if ((test_timeout = timer->timeout) == -1)
1922 if (timer->type != TIMER_TYPE_EXPECTED) {
1923 /* add our latency and offset to get output times. */
1924 test_timeout = apply_offset (jitterbuffer, test_timeout);
1925 test_timeout += priv->latency_ns;
1927 return test_timeout;
1931 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1933 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1935 if (priv->clock_id) {
1936 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1938 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1939 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1941 if (timeout == -1 || timeout < priv->timer_timeout)
1942 unschedule_current_timer (jitterbuffer);
1947 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1948 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1949 GstClockTime duration)
1951 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1955 GST_DEBUG_OBJECT (jitterbuffer,
1956 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1957 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1958 GST_TIME_ARGS (delay));
1960 len = priv->timers->len;
1961 g_array_set_size (priv->timers, len + 1);
1962 timer = &g_array_index (priv->timers, TimerData, len);
1965 timer->seqnum = seqnum;
1967 timer->timeout = timeout + delay;
1968 timer->duration = duration;
1969 if (type == TIMER_TYPE_EXPECTED) {
1970 timer->rtx_base = timeout;
1971 timer->rtx_delay = delay;
1972 timer->rtx_retry = 0;
1974 timer->num_rtx_retry = 0;
1975 recalculate_timer (jitterbuffer, timer);
1976 JBUF_SIGNAL_TIMER (priv);
1982 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1983 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1985 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1986 gboolean seqchange, timechange;
1989 seqchange = timer->seqnum != seqnum;
1990 timechange = timer->timeout != timeout;
1992 if (!seqchange && !timechange)
1995 oldseq = timer->seqnum;
1997 GST_DEBUG_OBJECT (jitterbuffer,
1998 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1999 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
2001 timer->timeout = timeout + delay;
2002 timer->seqnum = seqnum;
2004 timer->rtx_base = timeout;
2005 timer->rtx_delay = delay;
2006 timer->rtx_retry = 0;
2009 timer->num_rtx_retry = 0;
2011 if (priv->clock_id) {
2012 /* we changed the seqnum and there is a timer currently waiting with this
2013 * seqnum, unschedule it */
2014 if (seqchange && priv->timer_seqnum == oldseq)
2015 unschedule_current_timer (jitterbuffer);
2016 /* we changed the time, check if it is earlier than what we are waiting
2017 * for and unschedule if so */
2018 else if (timechange)
2019 recalculate_timer (jitterbuffer, timer);
2024 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2025 guint16 seqnum, GstClockTime timeout)
2029 /* find the seqnum timer */
2030 timer = find_timer (jitterbuffer, type, seqnum);
2031 if (timer == NULL) {
2032 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2034 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2040 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2042 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2045 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2046 unschedule_current_timer (jitterbuffer);
2049 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2050 g_array_remove_index_fast (priv->timers, idx);
2055 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2057 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2058 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2059 g_array_set_size (priv->timers, 0);
2060 unschedule_current_timer (jitterbuffer);
2063 /* get the extra delay to wait before sending RTX */
2065 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2069 if (priv->rtx_delay == -1) {
2070 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2071 delay = DEFAULT_AUTO_RTX_DELAY;
2073 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2074 * packet spacing is a good margin */
2075 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2078 delay = priv->rtx_delay * GST_MSECOND;
2080 if (priv->rtx_min_delay > 0)
2081 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2086 /* we just received a packet with seqnum and dts.
2088 * First check for old seqnum that we are still expecting. If the gap with the
2089 * current seqnum is too big, unschedule the timeouts.
2091 * If we have a valid packet spacing estimate we can set a timer for when we
2092 * should receive the next packet.
2093 * If we don't have a valid estimate, we remove any timer we might have
2094 * had for this packet.
2097 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2098 GstClockTime dts, gboolean do_next_seqnum)
2100 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2101 TimerData *timer = NULL;
2104 /* go through all timers and unschedule the ones with a large gap, also find
2105 * the timer for the seqnum */
2106 len = priv->timers->len;
2107 for (i = 0; i < len; i++) {
2108 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2111 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2113 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2114 test->type, test->seqnum, seqnum, gap);
2117 GST_DEBUG ("found timer for current seqnum");
2118 /* the timer for the current seqnum */
2120 /* when no retransmission, we can stop now, we only need to find the
2121 * timer for the current seqnum */
2122 if (!priv->do_retransmission)
2124 } else if (gap > priv->rtx_delay_reorder) {
2125 /* max gap, we exceeded the max reorder distance and we don't expect the
2126 * missing packet to be this reordered */
2127 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2128 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2132 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2133 && priv->do_retransmission && priv->rtx_next_seqnum;
2135 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2136 if (timer->num_rtx_retry > 0) {
2137 GstClockTime rtx_last, delay;
2139 /* we scheduled a retry for this packet and now we have it */
2140 priv->num_rtx_success++;
2141 /* all the previous retry attempts failed */
2142 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2143 /* number of retries before receiving the packet */
2144 if (priv->avg_rtx_num == 0.0)
2145 priv->avg_rtx_num = timer->num_rtx_retry;
2147 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2148 /* calculate the delay between retransmission request and receiving this
2149 * packet, start with when we scheduled this timeout last */
2150 rtx_last = timer->rtx_last;
2151 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2152 /* we have a valid delay if this packet arrived after we scheduled the
2154 delay = dts - rtx_last;
2155 if (priv->avg_rtx_rtt == 0)
2156 priv->avg_rtx_rtt = delay;
2158 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2162 GST_LOG_OBJECT (jitterbuffer,
2163 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2164 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2165 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2166 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2167 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2168 GST_TIME_ARGS (priv->avg_rtx_rtt));
2170 /* don't try to estimate the next seqnum because this is a retransmitted
2171 * packet and it probably did not arrive with the expected packet
2173 do_next_seqnum = FALSE;
2177 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2178 GstClockTime expected, delay;
2180 /* calculate expected arrival time of the next seqnum */
2181 expected = dts + priv->packet_spacing;
2183 delay = get_rtx_delay (priv);
2185 /* and update/install timer for next seqnum */
2187 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2190 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2191 expected, delay, priv->packet_spacing);
2193 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2194 /* if we had a timer, remove it, we don't know when to expect the next
2196 remove_timer (jitterbuffer, timer);
2201 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2204 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2206 /* we need consecutive seqnums with a different
2207 * rtptime to estimate the packet spacing. */
2208 if (priv->ips_rtptime != rtptime) {
2209 /* rtptime changed, check dts diff */
2210 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2211 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2212 GstClockTime old_packet_spacing = priv->packet_spacing;
2214 /* Biased towards bigger packet spacings to prevent
2215 * too many unneeded retransmission requests for next
2216 * packets that just arrive a little later than we would
2218 if (old_packet_spacing > new_packet_spacing)
2219 priv->packet_spacing =
2220 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2221 else if (old_packet_spacing > 0)
2222 priv->packet_spacing =
2223 (3 * new_packet_spacing + old_packet_spacing) / 4;
2225 priv->packet_spacing = new_packet_spacing;
2227 GST_DEBUG_OBJECT (jitterbuffer,
2228 "new packet spacing %" GST_TIME_FORMAT
2229 " old packet spacing %" GST_TIME_FORMAT
2230 " combined to %" GST_TIME_FORMAT,
2231 GST_TIME_ARGS (new_packet_spacing),
2232 GST_TIME_ARGS (old_packet_spacing),
2233 GST_TIME_ARGS (priv->packet_spacing));
2235 priv->ips_rtptime = rtptime;
2236 priv->ips_dts = dts;
2241 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2242 guint16 seqnum, GstClockTime dts, gint gap)
2244 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2245 GstClockTime total_duration, duration, expected_dts;
2248 GST_DEBUG_OBJECT (jitterbuffer,
2249 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2250 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2252 if (dts == GST_CLOCK_TIME_NONE) {
2253 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2257 /* the total duration spanned by the missing packets */
2258 if (dts >= priv->last_in_dts)
2259 total_duration = dts - priv->last_in_dts;
2263 /* interpolate between the current time and the last time based on
2264 * number of packets we are missing, this is the estimated duration
2265 * for the missing packet based on equidistant packet spacing. */
2266 duration = total_duration / (gap + 1);
2268 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2269 GST_TIME_ARGS (duration));
2271 if (total_duration > priv->latency_ns) {
2272 GstClockTime gap_time;
2276 GstClockTime gap_dur = gap * duration;
2277 if (gap_dur > priv->latency_ns)
2278 gap_time = gap_dur - priv->latency_ns;
2281 lost_packets = gap_time / duration;
2283 gap_time = total_duration - priv->latency_ns;
2287 /* too many lost packets, some of the missing packets are already
2288 * too late and we can generate lost packet events for them. */
2289 GST_DEBUG_OBJECT (jitterbuffer,
2290 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2291 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2292 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2293 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2294 GST_TIME_ARGS (gap_time));
2296 /* this timer will fire immediately and the lost event will be pushed from
2297 * the timer thread */
2298 if (lost_packets > 0) {
2299 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2300 priv->last_in_dts + duration, 0, gap_time);
2301 expected += lost_packets;
2302 priv->last_in_dts += gap_time;
2306 expected_dts = priv->last_in_dts + duration;
2308 if (priv->do_retransmission) {
2311 type = TIMER_TYPE_EXPECTED;
2312 /* if we had a timer for the first missing packet, update it. */
2313 if ((timer = find_timer (jitterbuffer, type, expected))) {
2314 GstClockTime timeout = timer->timeout;
2316 timer->duration = duration;
2317 if (timeout > (expected_dts + timer->rtx_retry)) {
2318 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2319 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2323 expected_dts += duration;
2326 type = TIMER_TYPE_LOST;
2329 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2330 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2331 expected_dts += duration;
2337 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2341 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2342 GstRtpJitterBufferPrivate *priv;
2344 priv = jitterbuffer->priv;
2346 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2349 if (priv->last_dts != -1)
2350 dtsdiff = dts - priv->last_dts;
2354 if (priv->last_rtptime != -1)
2355 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2359 priv->last_dts = dts;
2360 priv->last_rtptime = rtptime;
2364 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2367 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2369 diff = ABS (dtsdiff - rtpdiffns);
2371 /* jitter is stored in nanoseconds */
2372 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2374 GST_LOG_OBJECT (jitterbuffer,
2375 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2376 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2377 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2378 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2385 GST_DEBUG_OBJECT (jitterbuffer,
2386 "no dts or no clock-rate, can't calculate jitter");
2392 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2394 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2395 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2398 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2399 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2400 gst_rtp_buffer_unmap (&rtp_a);
2402 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2403 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2404 gst_rtp_buffer_unmap (&rtp_b);
2406 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2410 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2411 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2414 GstRtpJitterBufferPrivate *priv;
2415 guint gap_packets_length;
2416 gboolean reset = FALSE;
2418 priv = jitterbuffer->priv;
2420 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2422 guint32 prev_gap_seq = -1;
2423 gboolean all_consecutive = TRUE;
2425 g_queue_insert_sorted (&priv->gap_packets, buffer,
2426 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2428 for (l = priv->gap_packets.head; l; l = l->next) {
2429 GstBuffer *gap_buffer = l->data;
2430 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2433 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2435 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2437 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2438 if (prev_gap_seq == -1)
2439 prev_gap_seq = gap_seq;
2440 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2441 all_consecutive = FALSE;
2443 prev_gap_seq = gap_seq;
2445 gst_rtp_buffer_unmap (&gap_rtp);
2446 if (!all_consecutive)
2450 if (all_consecutive && gap_packets_length > 3) {
2451 GST_DEBUG_OBJECT (jitterbuffer,
2452 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2453 (future ? "new" : "old"), gap,
2454 (future ? max_dropout : -max_misorder));
2456 } else if (!all_consecutive) {
2457 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2458 g_queue_clear (&priv->gap_packets);
2459 GST_DEBUG_OBJECT (jitterbuffer,
2460 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2461 (future ? "new" : "old"), gap,
2462 (future ? max_dropout : -max_misorder));
2465 GST_DEBUG_OBJECT (jitterbuffer,
2466 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2467 (future ? "new" : "old"), gap,
2468 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2472 GST_DEBUG_OBJECT (jitterbuffer,
2473 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2474 gap, -max_misorder);
2475 g_queue_push_tail (&priv->gap_packets, buffer);
2483 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2485 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2486 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2489 GstClockTime base_time =
2490 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2491 GstClockTime clock_time = gst_clock_get_time (clock);
2493 if (clock_time > base_time)
2494 running_time = clock_time - base_time;
2498 gst_object_unref (clock);
2501 return running_time;
2504 static GstFlowReturn
2505 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2508 GstRtpJitterBuffer *jitterbuffer;
2509 GstRtpJitterBufferPrivate *priv;
2511 guint32 expected, rtptime;
2512 GstFlowReturn ret = GST_FLOW_OK;
2513 GstClockTime dts, pts;
2518 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2519 gboolean do_next_seqnum = FALSE;
2520 RTPJitterBufferItem *item;
2521 GstMessage *msg = NULL;
2522 gboolean estimated_dts = FALSE;
2523 guint32 packet_rate, max_dropout, max_misorder;
2525 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2527 priv = jitterbuffer->priv;
2529 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2530 goto invalid_buffer;
2532 pt = gst_rtp_buffer_get_payload_type (&rtp);
2533 seqnum = gst_rtp_buffer_get_seq (&rtp);
2534 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2535 gst_rtp_buffer_unmap (&rtp);
2537 /* make sure we have PTS and DTS set */
2538 pts = GST_BUFFER_PTS (buffer);
2539 dts = GST_BUFFER_DTS (buffer);
2546 /* If we have no DTS here, i.e. no capture time, get one from the
2547 * clock now to have something to calculate with in the future. */
2548 dts = get_current_running_time (jitterbuffer);
2551 /* Remember that we estimated the DTS if we are running already
2552 * and this is not our first packet (or first packet after a reset).
2553 * If it's the first packet, we somehow must generate a timestamp for
2554 * everything, otherwise we can't calculate any times
2556 estimated_dts = (priv->next_in_seqnum != -1);
2558 /* take the DTS of the buffer. This is the time when the packet was
2559 * received and is used to calculate jitter and clock skew. We will adjust
2560 * this DTS with the smoothed value after processing it in the
2561 * jitterbuffer and assign it as the PTS. */
2562 /* bring to running time */
2563 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2566 GST_DEBUG_OBJECT (jitterbuffer,
2567 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2568 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2570 JBUF_LOCK_CHECK (priv, out_flushing);
2572 if (G_UNLIKELY (priv->last_pt != pt)) {
2575 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2579 /* reset clock-rate so that we get a new one */
2580 priv->clock_rate = -1;
2582 /* Try to get the clock-rate from the caps first if we can. If there are no
2583 * caps we must fire the signal to get the clock-rate. */
2584 if ((caps = gst_pad_get_current_caps (pad))) {
2585 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2586 gst_caps_unref (caps);
2590 if (G_UNLIKELY (priv->clock_rate == -1)) {
2591 /* no clock rate given on the caps, try to get one with the signal */
2592 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2593 pt) == GST_FLOW_FLUSHING)
2596 if (G_UNLIKELY (priv->clock_rate == -1))
2600 /* don't accept more data on EOS */
2601 if (G_UNLIKELY (priv->eos))
2604 calculate_jitter (jitterbuffer, dts, rtptime);
2606 if (priv->seqnum_base != -1) {
2609 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2612 GST_DEBUG_OBJECT (jitterbuffer,
2613 "packet seqnum #%d before seqnum-base #%d", seqnum,
2615 gst_buffer_unref (buffer);
2618 } else if (gap > 16384) {
2619 /* From now on don't compare against the seqnum base anymore as
2620 * at some point in the future we will wrap around and also that
2621 * much reordering is very unlikely */
2622 priv->seqnum_base = -1;
2626 expected = priv->next_in_seqnum;
2629 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2631 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2632 priv->max_dropout_time);
2634 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2635 priv->max_misorder_time);
2636 GST_TRACE_OBJECT (jitterbuffer,
2637 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2638 max_dropout, max_misorder);
2640 /* now check against our expected seqnum */
2641 if (G_LIKELY (expected != -1)) {
2644 /* now calculate gap */
2645 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2647 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2648 expected, seqnum, gap);
2650 if (G_LIKELY (gap == 0)) {
2651 /* packet is expected */
2652 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2653 do_next_seqnum = TRUE;
2655 gboolean reset = FALSE;
2658 /* we received an old packet */
2659 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2661 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2662 gap, max_dropout, max_misorder);
2665 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2668 /* new packet, we are missing some packets */
2669 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2670 /* If we have timers for more than RTP_MAX_DROPOUT packets
2671 * pending this means that we have a huge gap overall. We can
2672 * reset the jitterbuffer at this point because there's
2673 * just too much data missing to be able to do anything
2674 * sensible with the past data. Just try again from the
2676 GST_WARNING_OBJECT (jitterbuffer,
2677 "%d pending timers > %d - resetting", priv->timers->len,
2680 gst_buffer_unref (buffer);
2682 } else if (G_UNLIKELY (gap >= max_dropout)) {
2684 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2685 gap, max_dropout, max_misorder);
2688 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2689 /* fill in the gap with EXPECTED timers */
2690 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2692 do_next_seqnum = TRUE;
2695 if (G_UNLIKELY (reset)) {
2696 GList *events = NULL, *l;
2699 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2700 rtp_jitter_buffer_flush (priv->jbuf,
2701 (GFunc) free_item_and_retain_events, &events);
2702 rtp_jitter_buffer_reset_skew (priv->jbuf);
2703 remove_all_timers (jitterbuffer);
2704 priv->discont = TRUE;
2705 priv->last_popped_seqnum = -1;
2707 if (priv->gap_packets.head) {
2708 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2709 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2711 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2712 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2713 gst_rtp_buffer_unmap (&gap_rtp);
2715 priv->next_seqnum = seqnum;
2718 priv->last_in_dts = -1;
2719 priv->next_in_seqnum = -1;
2721 /* Insert all sticky events again in order, otherwise we would
2722 * potentially loose STREAM_START, CAPS or SEGMENT events
2724 events = g_list_reverse (events);
2725 for (l = events; l; l = l->next) {
2726 RTPJitterBufferItem *item;
2728 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2729 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2731 g_list_free (events);
2733 JBUF_SIGNAL_EVENT (priv);
2735 /* reset spacing estimation when gap */
2736 priv->ips_rtptime = -1;
2737 priv->ips_dts = GST_CLOCK_TIME_NONE;
2739 buffers = g_list_copy (priv->gap_packets.head);
2740 g_queue_clear (&priv->gap_packets);
2742 priv->ips_rtptime = -1;
2743 priv->ips_dts = GST_CLOCK_TIME_NONE;
2744 JBUF_UNLOCK (jitterbuffer->priv);
2746 for (l = buffers; l; l = l->next) {
2747 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2749 if (ret != GST_FLOW_OK)
2752 for (; l; l = l->next)
2753 gst_buffer_unref (l->data);
2754 g_list_free (buffers);
2758 /* reset spacing estimation when gap */
2759 priv->ips_rtptime = -1;
2760 priv->ips_dts = GST_CLOCK_TIME_NONE;
2763 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2765 /* we don't know what the next_in_seqnum should be, wait for the last
2766 * possible moment to push this buffer, maybe we get an earlier seqnum
2768 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2769 do_next_seqnum = TRUE;
2770 /* take rtptime and dts to calculate packet spacing */
2771 priv->ips_rtptime = rtptime;
2772 priv->ips_dts = dts;
2775 /* We had no huge gap, let's drop all the gap packets */
2776 if (buffer != NULL) {
2777 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2778 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2779 g_queue_clear (&priv->gap_packets);
2781 GST_DEBUG_OBJECT (jitterbuffer,
2782 "Had big gap, waiting for more consecutive packets");
2783 JBUF_UNLOCK (jitterbuffer->priv);
2787 if (do_next_seqnum) {
2788 priv->last_in_dts = dts;
2789 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2792 /* let's check if this buffer is too late, we can only accept packets with
2793 * bigger seqnum than the one we last pushed. */
2794 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2797 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2799 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2800 if (G_UNLIKELY (gap <= 0))
2804 /* let's drop oldest packet if the queue is already full and drop-on-latency
2805 * is set. We can only do this when there actually is a latency. When no
2806 * latency is set, we just pump it in the queue and let the other end push it
2807 * out as fast as possible. */
2808 if (priv->latency_ms && priv->drop_on_latency) {
2810 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2812 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2813 RTPJitterBufferItem *old_item;
2815 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2817 if (IS_DROPABLE (old_item)) {
2818 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2819 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2821 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2822 free_item (old_item);
2824 /* we might have removed some head buffers, signal the pushing thread to
2825 * see if it can push now */
2826 JBUF_SIGNAL_EVENT (priv);
2830 /* If we estimated the DTS, don't consider it in the clock skew calculations
2831 * later. The code above always sets dts to pts or the other way around if
2832 * any of those is valid in the buffer, so we know that if we estimated the
2833 * dts that both are unknown */
2836 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2837 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2839 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2841 /* now insert the packet into the queue in sorted order. This function returns
2842 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2843 * have a duplicate. */
2844 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2846 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)))))
2850 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2852 /* we had an unhandled SR, handle it now */
2854 do_handle_sync (jitterbuffer);
2856 if (G_UNLIKELY (head)) {
2857 /* signal addition of new buffer when the _loop is waiting. */
2858 if (G_LIKELY (priv->active))
2859 JBUF_SIGNAL_EVENT (priv);
2861 /* let's unschedule and unblock any waiting buffers. We only want to do this
2862 * when the head buffer changed */
2863 if (G_UNLIKELY (priv->clock_id)) {
2864 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2865 unschedule_current_timer (jitterbuffer);
2869 GST_DEBUG_OBJECT (jitterbuffer,
2870 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2871 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2873 msg = check_buffering_percent (jitterbuffer, percent);
2879 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2886 /* this is not fatal but should be filtered earlier */
2887 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2888 ("Received invalid RTP payload, dropping"));
2889 gst_buffer_unref (buffer);
2894 GST_WARNING_OBJECT (jitterbuffer,
2895 "No clock-rate in caps!, dropping buffer");
2896 gst_buffer_unref (buffer);
2901 ret = priv->srcresult;
2902 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2903 gst_buffer_unref (buffer);
2909 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2910 gst_buffer_unref (buffer);
2915 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2916 " popped, dropping", seqnum, priv->last_popped_seqnum);
2918 gst_buffer_unref (buffer);
2923 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2925 priv->num_duplicates++;
2932 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2934 guint64 ext_time, elapsed;
2936 GstRtpJitterBufferPrivate *priv;
2938 priv = jitterbuffer->priv;
2939 rtp_time = item->rtptime;
2941 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2942 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2944 ext_time = priv->ext_timestamp;
2945 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2946 if (ext_time < priv->ext_timestamp) {
2947 ext_time = priv->ext_timestamp;
2949 priv->ext_timestamp = ext_time;
2952 if (ext_time > priv->clock_base)
2953 elapsed = ext_time - priv->clock_base;
2957 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2962 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2963 RTPJitterBufferItem * item)
2965 guint64 total, elapsed, left, estimated;
2966 GstClockTime out_time;
2967 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2969 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2970 || priv->clock_base == -1 || priv->clock_rate <= 0)
2973 /* compute the elapsed time */
2974 elapsed = compute_elapsed (jitterbuffer, item);
2976 /* do nothing if elapsed time doesn't increment */
2977 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2980 priv->last_elapsed = elapsed;
2982 /* this is the total time we need to play */
2983 total = priv->npt_stop - priv->npt_start;
2984 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2985 GST_TIME_ARGS (total));
2987 /* this is how much time there is left */
2988 if (total > elapsed)
2989 left = total - elapsed;
2993 /* if we have less time left that the size of the buffer, we will not
2994 * be able to keep it filled, disabled buffering then */
2995 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2996 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2997 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2998 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3001 /* this is the current time as running-time */
3002 out_time = item->dts;
3005 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3007 /* if there is almost nothing left,
3008 * we may never advance enough to end up in the above case */
3009 if (total < GST_SECOND)
3010 estimated = GST_SECOND;
3014 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3015 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3017 if (estimated != -1 && priv->estimated_eos != estimated) {
3018 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3019 priv->estimated_eos = estimated;
3023 /* take a buffer from the queue and push it */
3024 static GstFlowReturn
3025 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3027 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3028 GstFlowReturn result = GST_FLOW_OK;
3029 RTPJitterBufferItem *item;
3030 GstBuffer *outbuf = NULL;
3031 GstEvent *outevent = NULL;
3032 GstQuery *outquery = NULL;
3033 GstClockTime dts, pts;
3035 gboolean do_push = TRUE;
3039 /* when we get here we are ready to pop and push the buffer */
3040 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3044 case ITEM_TYPE_BUFFER:
3046 /* we need to make writable to change the flags and timestamps */
3047 outbuf = gst_buffer_make_writable (item->data);
3049 if (G_UNLIKELY (priv->discont)) {
3050 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3051 * into the jitterbuffer so we can modify now. */
3052 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3053 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3054 priv->discont = FALSE;
3056 if (G_UNLIKELY (priv->ts_discont)) {
3057 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3058 priv->ts_discont = FALSE;
3062 gst_segment_position_from_running_time (&priv->segment,
3063 GST_FORMAT_TIME, item->dts);
3065 gst_segment_position_from_running_time (&priv->segment,
3066 GST_FORMAT_TIME, item->pts);
3068 /* apply timestamp with offset to buffer now */
3069 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3070 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3072 /* update the elapsed time when we need to check against the npt stop time. */
3073 update_estimated_eos (jitterbuffer, item);
3075 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3077 case ITEM_TYPE_LOST:
3078 priv->discont = TRUE;
3082 case ITEM_TYPE_EVENT:
3083 outevent = item->data;
3085 case ITEM_TYPE_QUERY:
3086 outquery = item->data;
3090 /* now we are ready to push the buffer. Save the seqnum and release the lock
3091 * so the other end can push stuff in the queue again. */
3093 priv->last_popped_seqnum = seqnum;
3094 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3096 msg = check_buffering_percent (jitterbuffer, percent);
3103 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3106 case ITEM_TYPE_BUFFER:
3108 GST_DEBUG_OBJECT (jitterbuffer,
3109 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3110 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3111 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3112 result = gst_pad_push (priv->srcpad, outbuf);
3114 JBUF_LOCK_CHECK (priv, out_flushing);
3116 case ITEM_TYPE_LOST:
3117 case ITEM_TYPE_EVENT:
3118 /* We got not enough consecutive packets with a huge gap, we can
3119 * as well just drop them here now on EOS */
3120 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3121 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3122 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3123 g_queue_clear (&priv->gap_packets);
3126 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3127 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3130 gst_pad_push_event (priv->srcpad, outevent);
3132 gst_event_unref (outevent);
3134 result = GST_FLOW_OK;
3136 JBUF_LOCK_CHECK (priv, out_flushing);
3138 case ITEM_TYPE_QUERY:
3142 res = gst_pad_peer_query (priv->srcpad, outquery);
3144 JBUF_LOCK_CHECK (priv, out_flushing);
3145 result = GST_FLOW_OK;
3146 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3147 JBUF_SIGNAL_QUERY (priv, res);
3156 return priv->srcresult;
3160 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3162 /* Peek a buffer and compare the seqnum to the expected seqnum.
3163 * If all is fine, the buffer is pushed.
3164 * If something is wrong, we wait for some event
3166 static GstFlowReturn
3167 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3169 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3170 GstFlowReturn result;
3171 RTPJitterBufferItem *item;
3173 guint32 next_seqnum;
3175 /* only push buffers when PLAYING and active and not buffering */
3176 if (priv->blocked || !priv->active ||
3177 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3178 return GST_FLOW_WAIT;
3181 /* peek a buffer, we're just looking at the sequence number.
3182 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3183 * wait for a timeout or something to change.
3184 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3185 item = rtp_jitter_buffer_peek (priv->jbuf);
3190 /* get the seqnum and the next expected seqnum */
3191 seqnum = item->seqnum;
3193 return pop_and_push_next (jitterbuffer, seqnum);
3196 next_seqnum = priv->next_seqnum;
3198 /* get the gap between this and the previous packet. If we don't know the
3199 * previous packet seqnum assume no gap. */
3200 if (G_UNLIKELY (next_seqnum == -1)) {
3201 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3202 /* we don't know what the next_seqnum should be, the chain function should
3203 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3204 * fires, so wait for that */
3205 result = GST_FLOW_WAIT;
3207 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3209 if (G_LIKELY (gap == 0)) {
3210 /* no missing packet, pop and push */
3211 result = pop_and_push_next (jitterbuffer, seqnum);
3212 } else if (G_UNLIKELY (gap < 0)) {
3213 /* if we have a packet that we already pushed or considered dropped, pop it
3214 * off and get the next packet */
3215 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3216 seqnum, next_seqnum);
3217 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3219 result = GST_FLOW_OK;
3221 /* the chain function has scheduled timers to request retransmission or
3222 * when to consider the packet lost, wait for that */
3223 GST_DEBUG_OBJECT (jitterbuffer,
3224 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3225 next_seqnum, seqnum, gap);
3226 result = GST_FLOW_WAIT;
3234 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3236 return GST_FLOW_EOS;
3238 return GST_FLOW_WAIT;
3244 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3246 GstClockTime rtx_retry_timeout;
3247 GstClockTime rtx_min_retry_timeout;
3249 if (priv->rtx_retry_timeout == -1) {
3250 if (priv->avg_rtx_rtt == 0)
3251 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3253 /* we want to ask for a retransmission after we waited for a
3254 * complete RTT and the additional jitter */
3255 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3257 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3259 /* make sure we don't retry too often. On very low latency networks,
3260 * the RTT and jitter can be very low. */
3261 if (priv->rtx_min_retry_timeout == -1) {
3262 rtx_min_retry_timeout = priv->packet_spacing;
3264 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3266 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3268 return rtx_retry_timeout;
3272 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3273 GstClockTime rtx_retry_timeout)
3275 GstClockTime rtx_retry_period;
3277 if (priv->rtx_retry_period == -1) {
3278 /* we retry up to the configured jitterbuffer size but leaving some
3279 * room for the retransmission to arrive in time */
3280 if (rtx_retry_timeout > priv->latency_ns) {
3281 rtx_retry_period = 0;
3283 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3286 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3288 return rtx_retry_period;
3291 /* the timeout for when we expected a packet expired */
3293 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3296 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3298 guint delay, delay_ms, avg_rtx_rtt_ms;
3299 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3300 GstClockTime rtx_retry_period;
3301 GstClockTime rtx_retry_timeout;
3304 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3305 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3307 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3308 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3310 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3311 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3312 GST_TIME_ARGS (rtx_retry_period));
3314 delay = timer->rtx_delay + timer->rtx_retry;
3316 delay_ms = GST_TIME_AS_MSECONDS (delay);
3317 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3318 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3319 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3321 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3322 gst_structure_new ("GstRTPRetransmissionRequest",
3323 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3324 "running-time", G_TYPE_UINT64, timer->rtx_base,
3325 "delay", G_TYPE_UINT, delay_ms,
3326 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3327 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3328 "period", G_TYPE_UINT, rtx_retry_period_ms,
3329 "deadline", G_TYPE_UINT, priv->latency_ms,
3330 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3331 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3333 priv->num_rtx_requests++;
3334 timer->num_rtx_retry++;
3336 GST_OBJECT_LOCK (jitterbuffer);
3337 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3338 timer->rtx_last = gst_clock_get_time (clock);
3339 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3341 timer->rtx_last = now;
3343 GST_OBJECT_UNLOCK (jitterbuffer);
3345 /* calculate the timeout for the next retransmission attempt */
3346 timer->rtx_retry += rtx_retry_timeout;
3347 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3348 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3349 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3350 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3351 if ((priv->rtx_max_retries != -1
3352 && timer->num_rtx_retry >= priv->rtx_max_retries)
3353 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3354 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3355 /* too many retransmission request, we now convert the timer
3356 * to a lost timer, leave the num_rtx_retry as it is for stats */
3357 timer->type = TIMER_TYPE_LOST;
3358 timer->rtx_delay = 0;
3359 timer->rtx_retry = 0;
3361 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3362 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3365 gst_pad_push_event (priv->sinkpad, event);
3371 /* a packet is lost */
3373 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3376 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3377 GstClockTime duration, timestamp;
3378 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3381 RTPJitterBufferItem *item;
3383 seqnum = timer->seqnum;
3384 timestamp = apply_offset (jitterbuffer, timer->timeout);
3385 duration = timer->duration;
3386 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3387 duration = priv->packet_spacing;
3388 lost_packets = MAX (timer->num, 1);
3389 num_rtx_retry = timer->num_rtx_retry;
3391 /* we had a gap and thus we lost some packets. Create an event for this. */
3392 if (lost_packets > 1)
3393 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3394 seqnum + lost_packets - 1);
3396 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3398 priv->num_late += lost_packets;
3399 priv->num_rtx_failed += num_rtx_retry;
3401 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3403 /* we now only accept seqnum bigger than this */
3404 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3405 priv->next_in_seqnum = next_in_seqnum;
3407 /* create paket lost event */
3408 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3409 gst_structure_new ("GstRTPPacketLost",
3410 "seqnum", G_TYPE_UINT, (guint) seqnum,
3411 "timestamp", G_TYPE_UINT64, timestamp,
3412 "duration", G_TYPE_UINT64, duration,
3413 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3415 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3416 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3418 /* remove timer now */
3419 remove_timer (jitterbuffer, timer);
3421 JBUF_SIGNAL_EVENT (priv);
3427 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3430 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3432 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3433 remove_timer (jitterbuffer, timer);
3435 /* there was no EOS in the buffer, put one in there now */
3436 queue_event (jitterbuffer, gst_event_new_eos ());
3438 JBUF_SIGNAL_EVENT (priv);
3444 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3447 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3449 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3451 /* timer seqnum might have been obsoleted by caps seqnum-base,
3452 * only mess with current ongoing seqnum if still unknown */
3453 if (priv->next_seqnum == -1)
3454 priv->next_seqnum = timer->seqnum;
3455 remove_timer (jitterbuffer, timer);
3456 JBUF_SIGNAL_EVENT (priv);
3462 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3465 gboolean removed = FALSE;
3467 switch (timer->type) {
3468 case TIMER_TYPE_EXPECTED:
3469 removed = do_expected_timeout (jitterbuffer, timer, now);
3471 case TIMER_TYPE_LOST:
3472 removed = do_lost_timeout (jitterbuffer, timer, now);
3474 case TIMER_TYPE_DEADLINE:
3475 removed = do_deadline_timeout (jitterbuffer, timer, now);
3477 case TIMER_TYPE_EOS:
3478 removed = do_eos_timeout (jitterbuffer, timer, now);
3484 /* called when we need to wait for the next timeout.
3486 * We loop over the array of recorded timeouts and wait for the earliest one.
3487 * When it timed out, do the logic associated with the timer.
3489 * If there are no timers, we wait on a gcond until something new happens.
3492 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3494 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3495 GstClockTime now = 0;
3498 while (priv->timer_running) {
3499 TimerData *timer = NULL;
3500 GstClockTime timer_timeout = -1;
3503 /* If we have a clock, update "now" now with the very
3504 * latest running time we have. If timers are unscheduled below we
3505 * otherwise wouldn't update now (it's only updated when timers
3506 * expire), and also for the very first loop iteration now would
3507 * otherwise always be 0
3509 GST_OBJECT_LOCK (jitterbuffer);
3510 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3512 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3513 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3515 GST_OBJECT_UNLOCK (jitterbuffer);
3517 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3518 GST_TIME_ARGS (now));
3520 len = priv->timers->len;
3521 for (i = 0; i < len; i++) {
3522 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3523 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3524 gboolean save_best = FALSE;
3526 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3527 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3529 /* find the smallest timeout */
3530 if (timer == NULL) {
3532 } else if (timer_timeout == -1) {
3533 /* we already have an immediate timeout, the new timer must be an
3534 * immediate timer with smaller seqnum to become the best */
3535 if (test_timeout == -1
3536 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3537 timer->seqnum) > 0))
3539 } else if (test_timeout == -1) {
3540 /* first immediate timer */
3542 } else if (test_timeout < timer_timeout) {
3545 } else if (test_timeout == timer_timeout
3546 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3547 timer->seqnum) > 0)) {
3548 /* same timer, smaller seqnum */
3552 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3554 timer_timeout = test_timeout;
3557 if (timer && !priv->blocked) {
3559 GstClockTime sync_time;
3562 GstClockTimeDiff clock_jitter;
3564 if (timer_timeout == -1 || timer_timeout <= now) {
3565 do_timeout (jitterbuffer, timer, now);
3566 /* check here, do_timeout could have released the lock */
3567 if (!priv->timer_running)
3572 GST_OBJECT_LOCK (jitterbuffer);
3573 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3575 GST_OBJECT_UNLOCK (jitterbuffer);
3576 /* let's just push if there is no clock */
3577 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3578 now = timer_timeout;
3582 /* prepare for sync against clock */
3583 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3584 /* add latency of peer to get input time */
3585 sync_time += priv->peer_latency;
3587 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3588 " with sync time %" GST_TIME_FORMAT,
3589 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3591 /* create an entry for the clock */
3592 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3593 priv->timer_timeout = timer_timeout;
3594 priv->timer_seqnum = timer->seqnum;
3595 GST_OBJECT_UNLOCK (jitterbuffer);
3597 /* release the lock so that the other end can push stuff or unlock */
3600 ret = gst_clock_id_wait (id, &clock_jitter);
3603 if (!priv->timer_running) {
3604 gst_clock_id_unref (id);
3605 priv->clock_id = NULL;
3609 if (ret != GST_CLOCK_UNSCHEDULED) {
3610 now = timer_timeout + MAX (clock_jitter, 0);
3611 GST_DEBUG_OBJECT (jitterbuffer,
3612 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3613 GST_STIME_ARGS (clock_jitter));
3615 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3617 /* and free the entry */
3618 gst_clock_id_unref (id);
3619 priv->clock_id = NULL;
3621 /* no timers, wait for activity */
3622 JBUF_WAIT_TIMER (priv);
3627 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3632 * This funcion implements the main pushing loop on the source pad.
3634 * It first tries to push as many buffers as possible. If there is a seqnum
3635 * mismatch, we wait for the next timeouts.
3638 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3640 GstRtpJitterBufferPrivate *priv;
3641 GstFlowReturn result = GST_FLOW_OK;
3643 priv = jitterbuffer->priv;
3645 JBUF_LOCK_CHECK (priv, flushing);
3647 result = handle_next_buffer (jitterbuffer);
3648 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3649 /* now wait for the next event */
3650 JBUF_WAIT_EVENT (priv, flushing);
3651 result = GST_FLOW_OK;
3653 } while (result == GST_FLOW_OK);
3654 /* store result for upstream */
3655 priv->srcresult = result;
3656 /* if we get here we need to pause */
3662 result = priv->srcresult;
3669 JBUF_SIGNAL_QUERY (priv, FALSE);
3672 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3673 gst_flow_get_name (result));
3674 gst_pad_pause_task (priv->srcpad);
3675 if (result == GST_FLOW_EOS) {
3676 event = gst_event_new_eos ();
3677 gst_pad_push_event (priv->srcpad, event);
3683 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3684 * some sanity checks and then emit the handle-sync signal with the parameters.
3685 * This function must be called with the LOCK */
3687 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3689 GstRtpJitterBufferPrivate *priv;
3690 guint64 base_rtptime, base_time;
3692 guint64 last_rtptime;
3694 guint64 ext_rtptime, diff;
3695 gboolean valid = TRUE, keep = FALSE;
3697 priv = jitterbuffer->priv;
3699 /* get the last values from the jitterbuffer */
3700 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3701 &clock_rate, &last_rtptime);
3703 clock_base = priv->clock_base;
3704 ext_rtptime = priv->ext_rtptime;
3706 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3707 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3708 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3709 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3711 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3712 /* we keep this SR packet for later. When we get a valid RTP packet the
3713 * above values will be set and we can try to use the SR packet */
3714 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3717 /* we can't accept anything that happened before we did the last resync */
3718 if (base_rtptime > ext_rtptime) {
3719 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3722 /* the SR RTP timestamp must be something close to what we last observed
3723 * in the jitterbuffer */
3724 if (ext_rtptime > last_rtptime) {
3725 /* check how far ahead it is to our RTP timestamps */
3726 diff = ext_rtptime - last_rtptime;
3727 /* if bigger than 1 second, we drop it */
3728 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3730 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3731 clock_rate, 1000)) {
3732 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3733 /* should drop this, but some RTSP servers end up with bogus
3734 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3735 * so still trigger rptbin sync but invalidate RTCP data
3736 * (sync might use other methods) */
3739 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3740 G_GUINT64_FORMAT, last_rtptime, diff);
3746 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3750 s = gst_structure_new ("application/x-rtp-sync",
3751 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3752 "base-time", G_TYPE_UINT64, base_time,
3753 "clock-rate", G_TYPE_UINT, clock_rate,
3754 "clock-base", G_TYPE_UINT64, clock_base,
3755 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3756 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3758 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3759 gst_buffer_replace (&priv->last_sr, NULL);
3761 g_signal_emit (jitterbuffer,
3762 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3764 gst_structure_free (s);
3766 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3767 gst_buffer_replace (&priv->last_sr, NULL);
3771 static GstFlowReturn
3772 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3775 GstRtpJitterBuffer *jitterbuffer;
3776 GstRtpJitterBufferPrivate *priv;
3777 GstFlowReturn ret = GST_FLOW_OK;
3779 GstRTCPPacket packet;
3780 guint64 ext_rtptime;
3782 GstRTCPBuffer rtcp = { NULL, };
3784 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3786 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3787 goto invalid_buffer;
3789 priv = jitterbuffer->priv;
3791 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3793 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3796 /* first packet must be SR or RR or else the validate would have failed */
3797 switch (gst_rtcp_packet_get_type (&packet)) {
3798 case GST_RTCP_TYPE_SR:
3799 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3805 gst_rtcp_buffer_unmap (&rtcp);
3807 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3810 /* convert the RTP timestamp to our extended timestamp, using the same offset
3811 * we used in the jitterbuffer */
3812 ext_rtptime = priv->jbuf->ext_rtptime;
3813 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3815 priv->ext_rtptime = ext_rtptime;
3816 gst_buffer_replace (&priv->last_sr, buffer);
3818 do_handle_sync (jitterbuffer);
3822 gst_buffer_unref (buffer);
3828 /* this is not fatal but should be filtered earlier */
3829 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3830 ("Received invalid RTCP payload, dropping"));
3836 /* this is not fatal but should be filtered earlier */
3837 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3838 ("Received empty RTCP payload, dropping"));
3839 gst_rtcp_buffer_unmap (&rtcp);
3845 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3846 gst_rtcp_buffer_unmap (&rtcp);
3853 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3856 gboolean res = FALSE;
3857 GstRtpJitterBuffer *jitterbuffer;
3858 GstRtpJitterBufferPrivate *priv;
3860 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3861 priv = jitterbuffer->priv;
3863 switch (GST_QUERY_TYPE (query)) {
3864 case GST_QUERY_CAPS:
3866 GstCaps *filter, *caps;
3868 gst_query_parse_caps (query, &filter);
3869 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3870 gst_query_set_caps_result (query, caps);
3871 gst_caps_unref (caps);
3876 if (GST_QUERY_IS_SERIALIZED (query)) {
3877 RTPJitterBufferItem *item;
3880 JBUF_LOCK_CHECK (priv, out_flushing);
3881 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3882 RTP_JITTER_BUFFER_MODE_BUFFER) {
3883 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3884 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3885 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3887 JBUF_SIGNAL_EVENT (priv);
3888 JBUF_WAIT_QUERY (priv, out_flushing);
3889 res = priv->last_query;
3891 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3896 res = gst_pad_query_default (pad, parent, query);
3904 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3912 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3915 GstRtpJitterBuffer *jitterbuffer;
3916 GstRtpJitterBufferPrivate *priv;
3917 gboolean res = FALSE;
3919 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3920 priv = jitterbuffer->priv;
3922 switch (GST_QUERY_TYPE (query)) {
3923 case GST_QUERY_LATENCY:
3925 /* We need to send the query upstream and add the returned latency to our
3927 GstClockTime min_latency, max_latency;
3929 GstClockTime our_latency;
3931 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3932 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3934 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3935 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3936 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3938 /* store this so that we can safely sync on the peer buffers. */
3940 priv->peer_latency = min_latency;
3941 our_latency = priv->latency_ns;
3944 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3945 GST_TIME_ARGS (our_latency));
3947 /* we add some latency but can buffer an infinite amount of time */
3948 min_latency += our_latency;
3951 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3952 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3953 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3955 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3959 case GST_QUERY_POSITION:
3961 GstClockTime start, last_out;
3964 gst_query_parse_position (query, &fmt, NULL);
3965 if (fmt != GST_FORMAT_TIME) {
3966 res = gst_pad_query_default (pad, parent, query);
3971 start = priv->npt_start;
3972 last_out = priv->last_out_time;
3975 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3976 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3977 GST_TIME_ARGS (last_out));
3979 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3980 /* bring 0-based outgoing time to stream time */
3981 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3984 res = gst_pad_query_default (pad, parent, query);
3988 case GST_QUERY_CAPS:
3990 GstCaps *filter, *caps;
3992 gst_query_parse_caps (query, &filter);
3993 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3994 gst_query_set_caps_result (query, caps);
3995 gst_caps_unref (caps);
4000 res = gst_pad_query_default (pad, parent, query);
4008 gst_rtp_jitter_buffer_set_property (GObject * object,
4009 guint prop_id, const GValue * value, GParamSpec * pspec)
4011 GstRtpJitterBuffer *jitterbuffer;
4012 GstRtpJitterBufferPrivate *priv;
4014 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4015 priv = jitterbuffer->priv;
4020 guint new_latency, old_latency;
4022 new_latency = g_value_get_uint (value);
4025 old_latency = priv->latency_ms;
4026 priv->latency_ms = new_latency;
4027 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4028 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4031 /* post message if latency changed, this will inform the parent pipeline
4032 * that a latency reconfiguration is possible/needed. */
4033 if (new_latency != old_latency) {
4034 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4035 GST_TIME_ARGS (new_latency * GST_MSECOND));
4037 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4038 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4042 case PROP_DROP_ON_LATENCY:
4044 priv->drop_on_latency = g_value_get_boolean (value);
4047 case PROP_TS_OFFSET:
4049 priv->ts_offset = g_value_get_int64 (value);
4050 priv->ts_discont = TRUE;
4055 priv->do_lost = g_value_get_boolean (value);
4060 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4063 case PROP_DO_RETRANSMISSION:
4065 priv->do_retransmission = g_value_get_boolean (value);
4068 case PROP_RTX_NEXT_SEQNUM:
4070 priv->rtx_next_seqnum = g_value_get_boolean (value);
4073 case PROP_RTX_DELAY:
4075 priv->rtx_delay = g_value_get_int (value);
4078 case PROP_RTX_MIN_DELAY:
4080 priv->rtx_min_delay = g_value_get_uint (value);
4083 case PROP_RTX_DELAY_REORDER:
4085 priv->rtx_delay_reorder = g_value_get_int (value);
4088 case PROP_RTX_RETRY_TIMEOUT:
4090 priv->rtx_retry_timeout = g_value_get_int (value);
4093 case PROP_RTX_MIN_RETRY_TIMEOUT:
4095 priv->rtx_min_retry_timeout = g_value_get_int (value);
4098 case PROP_RTX_RETRY_PERIOD:
4100 priv->rtx_retry_period = g_value_get_int (value);
4103 case PROP_RTX_MAX_RETRIES:
4105 priv->rtx_max_retries = g_value_get_int (value);
4108 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4110 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4113 case PROP_MAX_DROPOUT_TIME:
4115 priv->max_dropout_time = g_value_get_uint (value);
4118 case PROP_MAX_MISORDER_TIME:
4120 priv->max_misorder_time = g_value_get_uint (value);
4123 case PROP_RFC7273_SYNC:
4125 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4126 g_value_get_boolean (value));
4130 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4136 gst_rtp_jitter_buffer_get_property (GObject * object,
4137 guint prop_id, GValue * value, GParamSpec * pspec)
4139 GstRtpJitterBuffer *jitterbuffer;
4140 GstRtpJitterBufferPrivate *priv;
4142 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4143 priv = jitterbuffer->priv;
4148 g_value_set_uint (value, priv->latency_ms);
4151 case PROP_DROP_ON_LATENCY:
4153 g_value_set_boolean (value, priv->drop_on_latency);
4156 case PROP_TS_OFFSET:
4158 g_value_set_int64 (value, priv->ts_offset);
4163 g_value_set_boolean (value, priv->do_lost);
4168 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4176 if (priv->srcresult != GST_FLOW_OK)
4179 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4181 g_value_set_int (value, percent);
4185 case PROP_DO_RETRANSMISSION:
4187 g_value_set_boolean (value, priv->do_retransmission);
4190 case PROP_RTX_NEXT_SEQNUM:
4192 g_value_set_boolean (value, priv->rtx_next_seqnum);
4195 case PROP_RTX_DELAY:
4197 g_value_set_int (value, priv->rtx_delay);
4200 case PROP_RTX_MIN_DELAY:
4202 g_value_set_uint (value, priv->rtx_min_delay);
4205 case PROP_RTX_DELAY_REORDER:
4207 g_value_set_int (value, priv->rtx_delay_reorder);
4210 case PROP_RTX_RETRY_TIMEOUT:
4212 g_value_set_int (value, priv->rtx_retry_timeout);
4215 case PROP_RTX_MIN_RETRY_TIMEOUT:
4217 g_value_set_int (value, priv->rtx_min_retry_timeout);
4220 case PROP_RTX_RETRY_PERIOD:
4222 g_value_set_int (value, priv->rtx_retry_period);
4225 case PROP_RTX_MAX_RETRIES:
4227 g_value_set_int (value, priv->rtx_max_retries);
4231 g_value_take_boxed (value,
4232 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4234 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4236 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4239 case PROP_MAX_DROPOUT_TIME:
4241 g_value_set_uint (value, priv->max_dropout_time);
4244 case PROP_MAX_MISORDER_TIME:
4246 g_value_set_uint (value, priv->max_misorder_time);
4249 case PROP_RFC7273_SYNC:
4251 g_value_set_boolean (value,
4252 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4256 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4261 static GstStructure *
4262 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4266 JBUF_LOCK (jbuf->priv);
4267 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4268 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4269 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4270 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4271 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4272 JBUF_UNLOCK (jbuf->priv);