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);
900 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
902 /* reset skew detection initialy */
903 rtp_jitter_buffer_reset_skew (priv->jbuf);
904 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
905 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
909 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
912 gst_pad_set_activatemode_function (priv->srcpad,
913 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
914 gst_pad_set_query_function (priv->srcpad,
915 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
916 gst_pad_set_event_function (priv->srcpad,
917 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
920 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
923 gst_pad_set_chain_function (priv->sinkpad,
924 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
925 gst_pad_set_event_function (priv->sinkpad,
926 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
927 gst_pad_set_query_function (priv->sinkpad,
928 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
930 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
931 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
933 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
936 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
938 #define ITEM_TYPE_BUFFER 0
939 #define ITEM_TYPE_LOST 1
940 #define ITEM_TYPE_EVENT 2
941 #define ITEM_TYPE_QUERY 3
943 static RTPJitterBufferItem *
944 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
945 guint seqnum, guint count, guint rtptime)
947 RTPJitterBufferItem *item;
949 item = g_slice_new (RTPJitterBufferItem);
956 item->seqnum = seqnum;
958 item->rtptime = rtptime;
964 free_item (RTPJitterBufferItem * item)
966 g_return_if_fail (item != NULL);
968 if (item->data && item->type != ITEM_TYPE_QUERY)
969 gst_mini_object_unref (item->data);
970 g_slice_free (RTPJitterBufferItem, item);
974 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
976 GList **l = user_data;
978 if (item->data && item->type == ITEM_TYPE_EVENT
979 && GST_EVENT_IS_STICKY (item->data)) {
980 *l = g_list_prepend (*l, item->data);
981 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
982 gst_mini_object_unref (item->data);
984 g_slice_free (RTPJitterBufferItem, item);
988 gst_rtp_jitter_buffer_finalize (GObject * object)
990 GstRtpJitterBuffer *jitterbuffer;
991 GstRtpJitterBufferPrivate *priv;
993 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
994 priv = jitterbuffer->priv;
996 g_array_free (priv->timers, TRUE);
997 g_mutex_clear (&priv->jbuf_lock);
998 g_cond_clear (&priv->jbuf_timer);
999 g_cond_clear (&priv->jbuf_event);
1000 g_cond_clear (&priv->jbuf_query);
1002 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1003 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1004 g_queue_clear (&priv->gap_packets);
1005 g_object_unref (priv->jbuf);
1007 G_OBJECT_CLASS (parent_class)->finalize (object);
1010 static GstIterator *
1011 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1013 GstRtpJitterBuffer *jitterbuffer;
1014 GstPad *otherpad = NULL;
1015 GstIterator *it = NULL;
1016 GValue val = { 0, };
1018 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1020 if (pad == jitterbuffer->priv->sinkpad) {
1021 otherpad = jitterbuffer->priv->srcpad;
1022 } else if (pad == jitterbuffer->priv->srcpad) {
1023 otherpad = jitterbuffer->priv->sinkpad;
1024 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1025 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1029 g_value_init (&val, GST_TYPE_PAD);
1030 g_value_set_object (&val, otherpad);
1031 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1032 g_value_unset (&val);
1039 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1041 GstRtpJitterBufferPrivate *priv;
1043 priv = jitterbuffer->priv;
1045 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1048 gst_pad_new_from_static_template
1049 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1050 gst_pad_set_chain_function (priv->rtcpsinkpad,
1051 gst_rtp_jitter_buffer_chain_rtcp);
1052 gst_pad_set_event_function (priv->rtcpsinkpad,
1053 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1054 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1055 gst_rtp_jitter_buffer_iterate_internal_links);
1056 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1057 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1059 return priv->rtcpsinkpad;
1063 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1065 GstRtpJitterBufferPrivate *priv;
1067 priv = jitterbuffer->priv;
1069 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1071 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1073 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1074 priv->rtcpsinkpad = NULL;
1078 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1079 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1081 GstRtpJitterBuffer *jitterbuffer;
1082 GstElementClass *klass;
1084 GstRtpJitterBufferPrivate *priv;
1086 g_return_val_if_fail (templ != NULL, NULL);
1087 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1089 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1090 priv = jitterbuffer->priv;
1091 klass = GST_ELEMENT_GET_CLASS (element);
1093 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1095 /* figure out the template */
1096 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1097 if (priv->rtcpsinkpad != NULL)
1100 result = create_rtcp_sink (jitterbuffer);
1102 goto wrong_template;
1109 g_warning ("rtpjitterbuffer: this is not our template");
1114 g_warning ("rtpjitterbuffer: pad already requested");
1120 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1122 GstRtpJitterBuffer *jitterbuffer;
1123 GstRtpJitterBufferPrivate *priv;
1125 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1126 g_return_if_fail (GST_IS_PAD (pad));
1128 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1129 priv = jitterbuffer->priv;
1131 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1133 if (priv->rtcpsinkpad == pad) {
1134 remove_rtcp_sink (jitterbuffer);
1143 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1149 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1151 return gst_system_clock_obtain ();
1155 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1157 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1159 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1161 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1165 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1167 GstRtpJitterBufferPrivate *priv;
1169 priv = jitterbuffer->priv;
1171 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1174 priv->clock_rate = -1;
1175 /* do not clear current content, but refresh state for new arrival */
1176 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1177 rtp_jitter_buffer_reset_skew (priv->jbuf);
1182 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1185 GstRtpJitterBufferPrivate *priv;
1186 GstClockTime last_out;
1187 RTPJitterBufferItem *item;
1192 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1193 active, GST_TIME_ARGS (offset));
1195 if (active != priv->active) {
1196 /* add the amount of time spent in paused to the output offset. All
1197 * outgoing buffers will have this offset applied to their timestamps in
1198 * order to make them arrive in time in the sink. */
1199 priv->out_offset = offset;
1200 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1201 GST_TIME_ARGS (priv->out_offset));
1202 priv->active = active;
1203 JBUF_SIGNAL_EVENT (priv);
1206 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1208 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1209 /* head buffer timestamp and offset gives our output time */
1210 last_out = item->dts + priv->ts_offset;
1212 /* use last known time when the buffer is empty */
1213 last_out = priv->last_out_time;
1221 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1223 GstRtpJitterBuffer *jitterbuffer;
1224 GstRtpJitterBufferPrivate *priv;
1229 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1230 priv = jitterbuffer->priv;
1232 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1234 caps = gst_pad_peer_query_caps (other, filter);
1236 templ = gst_pad_get_pad_template_caps (pad);
1238 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1243 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1245 intersect = gst_caps_intersect (caps, templ);
1246 gst_caps_unref (caps);
1247 gst_caps_unref (templ);
1251 gst_object_unref (jitterbuffer);
1257 * Must be called with JBUF_LOCK held
1261 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1262 GstCaps * caps, gint pt)
1264 GstRtpJitterBufferPrivate *priv;
1265 GstStructure *caps_struct;
1269 const gchar *ts_refclk, *mediaclk;
1271 priv = jitterbuffer->priv;
1273 /* first parse the caps */
1274 caps_struct = gst_caps_get_structure (caps, 0);
1276 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1278 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1280 GST_ERROR_OBJECT (jitterbuffer,
1281 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1285 if (payload != -1) {
1286 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1287 priv->last_pt = payload;
1290 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1291 * measure the amount of data in the buffer */
1292 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1295 if (priv->clock_rate <= 0)
1298 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1300 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1302 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1303 * can use this to track the amount of time elapsed on the sender. */
1304 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1305 priv->clock_base = val;
1307 priv->clock_base = -1;
1309 priv->ext_timestamp = priv->clock_base;
1311 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1314 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1315 /* first expected seqnum, only update when we didn't have a previous base. */
1316 if (priv->next_in_seqnum == -1)
1317 priv->next_in_seqnum = val;
1318 if (priv->next_seqnum == -1) {
1319 priv->next_seqnum = val;
1320 JBUF_SIGNAL_EVENT (priv);
1322 priv->seqnum_base = val;
1324 priv->seqnum_base = -1;
1327 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1329 /* the start and stop times. The seqnum-base corresponds to the start time. We
1330 * will keep track of the seqnums on the output and when we reach the one
1331 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1332 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1333 priv->npt_start = tval;
1335 priv->npt_start = 0;
1337 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1338 priv->npt_stop = tval;
1340 priv->npt_stop = -1;
1342 GST_DEBUG_OBJECT (jitterbuffer,
1343 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1344 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1346 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1347 GstClock *clock = NULL;
1348 guint64 clock_offset = -1;
1350 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1353 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1354 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1355 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1357 const gchar *host, *portstr;
1361 host = ts_refclk + sizeof ("ntp=") - 1;
1362 if (host[0] == '[') {
1364 portstr = strchr (host, ']');
1365 if (portstr && portstr[1] == ':')
1366 portstr = portstr + 1;
1370 portstr = strrchr (host, ':');
1374 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1378 hostname = g_strndup (host, (portstr - host));
1380 hostname = g_strdup (host);
1382 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1385 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1386 const gchar *domainstr =
1387 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1390 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1393 clock = gst_ptp_clock_new (NULL, domain);
1395 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1398 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1399 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1401 if (!g_str_has_prefix (mediaclk, "direct=")
1402 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1403 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1404 if (strstr (mediaclk, "rate=") != NULL) {
1405 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1410 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1412 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1420 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1425 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1431 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1433 GstRtpJitterBufferPrivate *priv;
1435 priv = jitterbuffer->priv;
1438 /* mark ourselves as flushing */
1439 priv->srcresult = GST_FLOW_FLUSHING;
1440 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1441 /* this unblocks any waiting pops on the src pad task */
1442 JBUF_SIGNAL_EVENT (priv);
1443 JBUF_SIGNAL_QUERY (priv, FALSE);
1448 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1450 GstRtpJitterBufferPrivate *priv;
1452 priv = jitterbuffer->priv;
1455 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1456 /* Mark as non flushing */
1457 priv->srcresult = GST_FLOW_OK;
1458 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1459 priv->last_popped_seqnum = -1;
1460 priv->last_out_time = -1;
1461 priv->next_seqnum = -1;
1462 priv->seqnum_base = -1;
1463 priv->ips_rtptime = -1;
1464 priv->ips_dts = GST_CLOCK_TIME_NONE;
1465 priv->packet_spacing = 0;
1466 priv->next_in_seqnum = -1;
1467 priv->clock_rate = -1;
1470 priv->estimated_eos = -1;
1471 priv->last_elapsed = 0;
1472 priv->ext_timestamp = -1;
1473 priv->avg_jitter = 0;
1474 priv->last_dts = -1;
1475 priv->last_rtptime = -1;
1476 priv->last_in_dts = 0;
1477 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1478 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1479 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1480 rtp_jitter_buffer_reset_skew (priv->jbuf);
1481 remove_all_timers (jitterbuffer);
1482 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1483 g_queue_clear (&priv->gap_packets);
1488 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1489 GstPadMode mode, gboolean active)
1492 GstRtpJitterBuffer *jitterbuffer = NULL;
1494 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1497 case GST_PAD_MODE_PUSH:
1499 /* allow data processing */
1500 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1502 /* start pushing out buffers */
1503 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1504 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1505 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1507 /* make sure all data processing stops ASAP */
1508 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1510 /* NOTE this will hardlock if the state change is called from the src pad
1511 * task thread because we will _join() the thread. */
1512 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1513 result = gst_pad_stop_task (pad);
1523 static GstStateChangeReturn
1524 gst_rtp_jitter_buffer_change_state (GstElement * element,
1525 GstStateChange transition)
1527 GstRtpJitterBuffer *jitterbuffer;
1528 GstRtpJitterBufferPrivate *priv;
1529 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1531 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1532 priv = jitterbuffer->priv;
1534 switch (transition) {
1535 case GST_STATE_CHANGE_NULL_TO_READY:
1537 case GST_STATE_CHANGE_READY_TO_PAUSED:
1539 /* reset negotiated values */
1540 priv->clock_rate = -1;
1541 priv->clock_base = -1;
1542 priv->peer_latency = 0;
1544 /* block until we go to PLAYING */
1545 priv->blocked = TRUE;
1546 priv->timer_running = TRUE;
1547 priv->timer_thread =
1548 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1551 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1553 /* unblock to allow streaming in PLAYING */
1554 priv->blocked = FALSE;
1555 JBUF_SIGNAL_EVENT (priv);
1556 JBUF_SIGNAL_TIMER (priv);
1563 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1565 switch (transition) {
1566 case GST_STATE_CHANGE_READY_TO_PAUSED:
1567 /* we are a live element because we sync to the clock, which we can only
1568 * do in the PLAYING state */
1569 if (ret != GST_STATE_CHANGE_FAILURE)
1570 ret = GST_STATE_CHANGE_NO_PREROLL;
1572 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1574 /* block to stop streaming when PAUSED */
1575 priv->blocked = TRUE;
1576 unschedule_current_timer (jitterbuffer);
1578 if (ret != GST_STATE_CHANGE_FAILURE)
1579 ret = GST_STATE_CHANGE_NO_PREROLL;
1581 case GST_STATE_CHANGE_PAUSED_TO_READY:
1583 gst_buffer_replace (&priv->last_sr, NULL);
1584 priv->timer_running = FALSE;
1585 unschedule_current_timer (jitterbuffer);
1586 JBUF_SIGNAL_TIMER (priv);
1587 JBUF_SIGNAL_QUERY (priv, FALSE);
1589 g_thread_join (priv->timer_thread);
1590 priv->timer_thread = NULL;
1592 case GST_STATE_CHANGE_READY_TO_NULL:
1602 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1605 gboolean ret = TRUE;
1606 GstRtpJitterBuffer *jitterbuffer;
1607 GstRtpJitterBufferPrivate *priv;
1609 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1610 priv = jitterbuffer->priv;
1612 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1614 switch (GST_EVENT_TYPE (event)) {
1615 case GST_EVENT_LATENCY:
1617 GstClockTime latency;
1619 gst_event_parse_latency (event, &latency);
1621 GST_DEBUG_OBJECT (jitterbuffer,
1622 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1625 /* adjust the overall buffer delay to the total pipeline latency in
1626 * buffering mode because if downstream consumes too fast (because of
1627 * large latency or queues, we would start rebuffering again. */
1628 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1629 RTP_JITTER_BUFFER_MODE_BUFFER) {
1630 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1634 ret = gst_pad_push_event (priv->sinkpad, event);
1638 ret = gst_pad_push_event (priv->sinkpad, event);
1645 /* handles and stores the event in the jitterbuffer, must be called with
1648 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1650 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1651 RTPJitterBufferItem *item;
1654 switch (GST_EVENT_TYPE (event)) {
1655 case GST_EVENT_CAPS:
1659 gst_event_parse_caps (event, &caps);
1660 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1663 case GST_EVENT_SEGMENT:
1666 gst_event_copy_segment (event, &segment);
1668 /* we need time for now */
1669 if (segment.format != GST_FORMAT_TIME) {
1670 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1671 gst_event_unref (event);
1673 gst_segment_init (&segment, GST_FORMAT_TIME);
1674 event = gst_event_new_segment (&segment);
1677 priv->segment = segment;
1682 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1689 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1690 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1691 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1693 JBUF_SIGNAL_EVENT (priv);
1699 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1702 gboolean ret = TRUE;
1703 GstRtpJitterBuffer *jitterbuffer;
1704 GstRtpJitterBufferPrivate *priv;
1706 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1707 priv = jitterbuffer->priv;
1709 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1711 switch (GST_EVENT_TYPE (event)) {
1712 case GST_EVENT_FLUSH_START:
1713 ret = gst_pad_push_event (priv->srcpad, event);
1714 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1715 /* wait for the loop to go into PAUSED */
1716 gst_pad_pause_task (priv->srcpad);
1718 case GST_EVENT_FLUSH_STOP:
1719 ret = gst_pad_push_event (priv->srcpad, event);
1721 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1722 GST_PAD_MODE_PUSH, TRUE);
1725 if (GST_EVENT_IS_SERIALIZED (event)) {
1726 /* serialized events go in the queue */
1728 if (priv->srcresult != GST_FLOW_OK) {
1729 /* Errors in sticky event pushing are no problem and ignored here
1730 * as they will cause more meaningful errors during data flow.
1731 * For EOS events, that are not followed by data flow, we still
1732 * return FALSE here though.
1734 if (!GST_EVENT_IS_STICKY (event) ||
1735 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1736 goto out_flow_error;
1738 /* refuse more events on EOS */
1741 ret = queue_event (jitterbuffer, event);
1744 /* non-serialized events are forwarded downstream immediately */
1745 ret = gst_pad_push_event (priv->srcpad, event);
1754 GST_DEBUG_OBJECT (jitterbuffer,
1755 "refusing event, we have a downstream flow error: %s",
1756 gst_flow_get_name (priv->srcresult));
1758 gst_event_unref (event);
1763 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1765 gst_event_unref (event);
1771 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1774 gboolean ret = TRUE;
1775 GstRtpJitterBuffer *jitterbuffer;
1777 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1779 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1781 switch (GST_EVENT_TYPE (event)) {
1782 case GST_EVENT_FLUSH_START:
1783 gst_event_unref (event);
1785 case GST_EVENT_FLUSH_STOP:
1786 gst_event_unref (event);
1789 ret = gst_pad_event_default (pad, parent, event);
1797 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1798 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1799 * GST_FLOW_FLUSHING when the element is shutting down. On success
1800 * GST_FLOW_OK is returned.
1802 static GstFlowReturn
1803 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1807 GValue args[2] = { {0}, {0} };
1811 g_value_init (&args[0], GST_TYPE_ELEMENT);
1812 g_value_set_object (&args[0], jitterbuffer);
1813 g_value_init (&args[1], G_TYPE_UINT);
1814 g_value_set_uint (&args[1], pt);
1816 g_value_init (&ret, GST_TYPE_CAPS);
1817 g_value_set_boxed (&ret, NULL);
1819 JBUF_UNLOCK (jitterbuffer->priv);
1820 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1822 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1824 g_value_unset (&args[0]);
1825 g_value_unset (&args[1]);
1826 caps = (GstCaps *) g_value_dup_boxed (&ret);
1827 g_value_unset (&ret);
1831 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1832 gst_caps_unref (caps);
1834 if (G_UNLIKELY (!res))
1842 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1843 return GST_FLOW_ERROR;
1847 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1848 return GST_FLOW_FLUSHING;
1852 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1853 return GST_FLOW_ERROR;
1857 /* call with jbuf lock held */
1859 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1861 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1862 GstMessage *message = NULL;
1867 /* Post a buffering message */
1868 if (priv->last_percent != percent) {
1869 priv->last_percent = percent;
1871 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1872 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1879 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1881 GstRtpJitterBufferPrivate *priv;
1883 priv = jitterbuffer->priv;
1885 if (timestamp == -1)
1888 /* apply the timestamp offset, this is used for inter stream sync */
1889 timestamp += priv->ts_offset;
1890 /* add the offset, this is used when buffering */
1891 timestamp += priv->out_offset;
1897 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1899 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1900 TimerData *timer = NULL;
1903 len = priv->timers->len;
1904 for (i = 0; i < len; i++) {
1905 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1906 if (test->seqnum == seqnum && test->type == type) {
1915 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1917 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1919 if (priv->clock_id) {
1920 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1921 gst_clock_id_unschedule (priv->clock_id);
1922 priv->clock_id = NULL;
1927 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1929 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1930 GstClockTime test_timeout;
1932 if ((test_timeout = timer->timeout) == -1)
1935 if (timer->type != TIMER_TYPE_EXPECTED) {
1936 /* add our latency and offset to get output times. */
1937 test_timeout = apply_offset (jitterbuffer, test_timeout);
1938 test_timeout += priv->latency_ns;
1940 return test_timeout;
1944 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1946 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1948 if (priv->clock_id) {
1949 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1951 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1952 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1954 if (timeout == -1 || timeout < priv->timer_timeout)
1955 unschedule_current_timer (jitterbuffer);
1960 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1961 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1962 GstClockTime duration)
1964 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1968 GST_DEBUG_OBJECT (jitterbuffer,
1969 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1970 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1971 GST_TIME_ARGS (delay));
1973 len = priv->timers->len;
1974 g_array_set_size (priv->timers, len + 1);
1975 timer = &g_array_index (priv->timers, TimerData, len);
1978 timer->seqnum = seqnum;
1980 timer->timeout = timeout + delay;
1981 timer->duration = duration;
1982 if (type == TIMER_TYPE_EXPECTED) {
1983 timer->rtx_base = timeout;
1984 timer->rtx_delay = delay;
1985 timer->rtx_retry = 0;
1987 timer->num_rtx_retry = 0;
1988 recalculate_timer (jitterbuffer, timer);
1989 JBUF_SIGNAL_TIMER (priv);
1995 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1996 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1998 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1999 gboolean seqchange, timechange;
2002 seqchange = timer->seqnum != seqnum;
2003 timechange = timer->timeout != timeout;
2005 if (!seqchange && !timechange)
2008 oldseq = timer->seqnum;
2010 GST_DEBUG_OBJECT (jitterbuffer,
2011 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
2012 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
2014 timer->timeout = timeout + delay;
2015 timer->seqnum = seqnum;
2017 timer->rtx_base = timeout;
2018 timer->rtx_delay = delay;
2019 timer->rtx_retry = 0;
2022 timer->num_rtx_retry = 0;
2024 if (priv->clock_id) {
2025 /* we changed the seqnum and there is a timer currently waiting with this
2026 * seqnum, unschedule it */
2027 if (seqchange && priv->timer_seqnum == oldseq)
2028 unschedule_current_timer (jitterbuffer);
2029 /* we changed the time, check if it is earlier than what we are waiting
2030 * for and unschedule if so */
2031 else if (timechange)
2032 recalculate_timer (jitterbuffer, timer);
2037 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2038 guint16 seqnum, GstClockTime timeout)
2042 /* find the seqnum timer */
2043 timer = find_timer (jitterbuffer, type, seqnum);
2044 if (timer == NULL) {
2045 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2047 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2053 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2055 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2058 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2059 unschedule_current_timer (jitterbuffer);
2062 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2063 g_array_remove_index_fast (priv->timers, idx);
2068 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2070 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2071 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2072 g_array_set_size (priv->timers, 0);
2073 unschedule_current_timer (jitterbuffer);
2076 /* get the extra delay to wait before sending RTX */
2078 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2082 if (priv->rtx_delay == -1) {
2083 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2084 delay = DEFAULT_AUTO_RTX_DELAY;
2086 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2087 * packet spacing is a good margin */
2088 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2091 delay = priv->rtx_delay * GST_MSECOND;
2093 if (priv->rtx_min_delay > 0)
2094 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2099 /* Check if packet with seqnum is already considered definitely lost by being
2100 * part of a "lost timer" for multiple packets */
2102 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2104 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2107 len = priv->timers->len;
2108 for (i = 0; i < len; i++) {
2109 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2110 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2112 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2114 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2115 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2123 /* we just received a packet with seqnum and dts.
2125 * First check for old seqnum that we are still expecting. If the gap with the
2126 * current seqnum is too big, unschedule the timeouts.
2128 * If we have a valid packet spacing estimate we can set a timer for when we
2129 * should receive the next packet.
2130 * If we don't have a valid estimate, we remove any timer we might have
2131 * had for this packet.
2134 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2135 GstClockTime dts, gboolean do_next_seqnum)
2137 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2138 TimerData *timer = NULL;
2141 /* go through all timers and unschedule the ones with a large gap, also find
2142 * the timer for the seqnum */
2143 len = priv->timers->len;
2144 for (i = 0; i < len; i++) {
2145 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2148 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2150 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2151 test->type, test->seqnum, seqnum, gap);
2154 GST_DEBUG ("found timer for current seqnum");
2155 /* the timer for the current seqnum */
2157 /* when no retransmission, we can stop now, we only need to find the
2158 * timer for the current seqnum */
2159 if (!priv->do_retransmission)
2161 } else if (gap > priv->rtx_delay_reorder) {
2162 /* max gap, we exceeded the max reorder distance and we don't expect the
2163 * missing packet to be this reordered */
2164 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2165 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2169 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2170 && priv->do_retransmission && priv->rtx_next_seqnum;
2172 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2173 if (timer->num_rtx_retry > 0) {
2174 GstClockTime rtx_last, delay;
2176 /* we scheduled a retry for this packet and now we have it */
2177 priv->num_rtx_success++;
2178 /* all the previous retry attempts failed */
2179 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2180 /* number of retries before receiving the packet */
2181 if (priv->avg_rtx_num == 0.0)
2182 priv->avg_rtx_num = timer->num_rtx_retry;
2184 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2185 /* calculate the delay between retransmission request and receiving this
2186 * packet, start with when we scheduled this timeout last */
2187 rtx_last = timer->rtx_last;
2188 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2189 /* we have a valid delay if this packet arrived after we scheduled the
2191 delay = dts - rtx_last;
2192 if (priv->avg_rtx_rtt == 0)
2193 priv->avg_rtx_rtt = delay;
2195 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2199 GST_LOG_OBJECT (jitterbuffer,
2200 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2201 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2202 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2203 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2204 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2205 GST_TIME_ARGS (priv->avg_rtx_rtt));
2207 /* don't try to estimate the next seqnum because this is a retransmitted
2208 * packet and it probably did not arrive with the expected packet
2210 do_next_seqnum = FALSE;
2214 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2215 GstClockTime expected, delay;
2217 /* calculate expected arrival time of the next seqnum */
2218 expected = dts + priv->packet_spacing;
2220 delay = get_rtx_delay (priv);
2222 /* and update/install timer for next seqnum */
2224 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2227 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2228 expected, delay, priv->packet_spacing);
2230 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2231 /* if we had a timer, remove it, we don't know when to expect the next
2233 remove_timer (jitterbuffer, timer);
2238 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2241 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2243 /* we need consecutive seqnums with a different
2244 * rtptime to estimate the packet spacing. */
2245 if (priv->ips_rtptime != rtptime) {
2246 /* rtptime changed, check dts diff */
2247 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2248 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2249 GstClockTime old_packet_spacing = priv->packet_spacing;
2251 /* Biased towards bigger packet spacings to prevent
2252 * too many unneeded retransmission requests for next
2253 * packets that just arrive a little later than we would
2255 if (old_packet_spacing > new_packet_spacing)
2256 priv->packet_spacing =
2257 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2258 else if (old_packet_spacing > 0)
2259 priv->packet_spacing =
2260 (3 * new_packet_spacing + old_packet_spacing) / 4;
2262 priv->packet_spacing = new_packet_spacing;
2264 GST_DEBUG_OBJECT (jitterbuffer,
2265 "new packet spacing %" GST_TIME_FORMAT
2266 " old packet spacing %" GST_TIME_FORMAT
2267 " combined to %" GST_TIME_FORMAT,
2268 GST_TIME_ARGS (new_packet_spacing),
2269 GST_TIME_ARGS (old_packet_spacing),
2270 GST_TIME_ARGS (priv->packet_spacing));
2272 priv->ips_rtptime = rtptime;
2273 priv->ips_dts = dts;
2278 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2279 guint16 seqnum, GstClockTime dts, gint gap)
2281 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2282 GstClockTime total_duration, duration, expected_dts;
2285 GST_DEBUG_OBJECT (jitterbuffer,
2286 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2287 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2289 if (dts == GST_CLOCK_TIME_NONE) {
2290 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2294 /* the total duration spanned by the missing packets */
2295 if (dts >= priv->last_in_dts)
2296 total_duration = dts - priv->last_in_dts;
2300 /* interpolate between the current time and the last time based on
2301 * number of packets we are missing, this is the estimated duration
2302 * for the missing packet based on equidistant packet spacing. */
2303 duration = total_duration / (gap + 1);
2305 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2306 GST_TIME_ARGS (duration));
2308 if (total_duration > priv->latency_ns) {
2309 GstClockTime gap_time;
2313 GstClockTime gap_dur = gap * duration;
2314 if (gap_dur > priv->latency_ns)
2315 gap_time = gap_dur - priv->latency_ns;
2318 lost_packets = gap_time / duration;
2320 gap_time = total_duration - priv->latency_ns;
2324 /* too many lost packets, some of the missing packets are already
2325 * too late and we can generate lost packet events for them. */
2326 GST_DEBUG_OBJECT (jitterbuffer,
2327 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2328 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2329 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2330 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2331 GST_TIME_ARGS (gap_time));
2333 /* this timer will fire immediately and the lost event will be pushed from
2334 * the timer thread */
2335 if (lost_packets > 0) {
2336 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2337 priv->last_in_dts + duration, 0, gap_time);
2338 expected += lost_packets;
2339 priv->last_in_dts += gap_time;
2343 expected_dts = priv->last_in_dts + duration;
2345 if (priv->do_retransmission) {
2348 type = TIMER_TYPE_EXPECTED;
2349 /* if we had a timer for the first missing packet, update it. */
2350 if ((timer = find_timer (jitterbuffer, type, expected))) {
2351 GstClockTime timeout = timer->timeout;
2353 timer->duration = duration;
2354 if (timeout > (expected_dts + timer->rtx_retry)) {
2355 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2356 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2360 expected_dts += duration;
2363 type = TIMER_TYPE_LOST;
2366 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2367 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2368 expected_dts += duration;
2374 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2378 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2379 GstRtpJitterBufferPrivate *priv;
2381 priv = jitterbuffer->priv;
2383 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2386 if (priv->last_dts != -1)
2387 dtsdiff = dts - priv->last_dts;
2391 if (priv->last_rtptime != -1)
2392 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2396 priv->last_dts = dts;
2397 priv->last_rtptime = rtptime;
2401 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2404 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2406 diff = ABS (dtsdiff - rtpdiffns);
2408 /* jitter is stored in nanoseconds */
2409 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2411 GST_LOG_OBJECT (jitterbuffer,
2412 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2413 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2414 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2415 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2422 GST_DEBUG_OBJECT (jitterbuffer,
2423 "no dts or no clock-rate, can't calculate jitter");
2429 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2431 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2432 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2435 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2436 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2437 gst_rtp_buffer_unmap (&rtp_a);
2439 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2440 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2441 gst_rtp_buffer_unmap (&rtp_b);
2443 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2447 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2448 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2451 GstRtpJitterBufferPrivate *priv;
2452 guint gap_packets_length;
2453 gboolean reset = FALSE;
2455 priv = jitterbuffer->priv;
2457 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2459 guint32 prev_gap_seq = -1;
2460 gboolean all_consecutive = TRUE;
2462 g_queue_insert_sorted (&priv->gap_packets, buffer,
2463 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2465 for (l = priv->gap_packets.head; l; l = l->next) {
2466 GstBuffer *gap_buffer = l->data;
2467 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2470 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2472 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2474 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2475 if (prev_gap_seq == -1)
2476 prev_gap_seq = gap_seq;
2477 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2478 all_consecutive = FALSE;
2480 prev_gap_seq = gap_seq;
2482 gst_rtp_buffer_unmap (&gap_rtp);
2483 if (!all_consecutive)
2487 if (all_consecutive && gap_packets_length > 3) {
2488 GST_DEBUG_OBJECT (jitterbuffer,
2489 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2490 (future ? "new" : "old"), gap,
2491 (future ? max_dropout : -max_misorder));
2493 } else if (!all_consecutive) {
2494 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2495 g_queue_clear (&priv->gap_packets);
2496 GST_DEBUG_OBJECT (jitterbuffer,
2497 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2498 (future ? "new" : "old"), gap,
2499 (future ? max_dropout : -max_misorder));
2502 GST_DEBUG_OBJECT (jitterbuffer,
2503 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2504 (future ? "new" : "old"), gap,
2505 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2509 GST_DEBUG_OBJECT (jitterbuffer,
2510 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2511 gap, -max_misorder);
2512 g_queue_push_tail (&priv->gap_packets, buffer);
2520 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2522 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2523 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2526 GstClockTime base_time =
2527 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2528 GstClockTime clock_time = gst_clock_get_time (clock);
2530 if (clock_time > base_time)
2531 running_time = clock_time - base_time;
2535 gst_object_unref (clock);
2538 return running_time;
2541 static GstFlowReturn
2542 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2545 GstRtpJitterBuffer *jitterbuffer;
2546 GstRtpJitterBufferPrivate *priv;
2548 guint32 expected, rtptime;
2549 GstFlowReturn ret = GST_FLOW_OK;
2550 GstClockTime dts, pts;
2555 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2556 gboolean do_next_seqnum = FALSE;
2557 RTPJitterBufferItem *item;
2558 GstMessage *msg = NULL;
2559 gboolean estimated_dts = FALSE;
2560 guint32 packet_rate, max_dropout, max_misorder;
2562 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2564 priv = jitterbuffer->priv;
2566 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2567 goto invalid_buffer;
2569 pt = gst_rtp_buffer_get_payload_type (&rtp);
2570 seqnum = gst_rtp_buffer_get_seq (&rtp);
2571 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2572 gst_rtp_buffer_unmap (&rtp);
2574 /* make sure we have PTS and DTS set */
2575 pts = GST_BUFFER_PTS (buffer);
2576 dts = GST_BUFFER_DTS (buffer);
2583 /* If we have no DTS here, i.e. no capture time, get one from the
2584 * clock now to have something to calculate with in the future. */
2585 dts = get_current_running_time (jitterbuffer);
2588 /* Remember that we estimated the DTS if we are running already
2589 * and this is not our first packet (or first packet after a reset).
2590 * If it's the first packet, we somehow must generate a timestamp for
2591 * everything, otherwise we can't calculate any times
2593 estimated_dts = (priv->next_in_seqnum != -1);
2595 /* take the DTS of the buffer. This is the time when the packet was
2596 * received and is used to calculate jitter and clock skew. We will adjust
2597 * this DTS with the smoothed value after processing it in the
2598 * jitterbuffer and assign it as the PTS. */
2599 /* bring to running time */
2600 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2603 GST_DEBUG_OBJECT (jitterbuffer,
2604 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2605 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2607 JBUF_LOCK_CHECK (priv, out_flushing);
2609 if (G_UNLIKELY (priv->last_pt != pt)) {
2612 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2616 /* reset clock-rate so that we get a new one */
2617 priv->clock_rate = -1;
2619 /* Try to get the clock-rate from the caps first if we can. If there are no
2620 * caps we must fire the signal to get the clock-rate. */
2621 if ((caps = gst_pad_get_current_caps (pad))) {
2622 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2623 gst_caps_unref (caps);
2627 if (G_UNLIKELY (priv->clock_rate == -1)) {
2628 /* no clock rate given on the caps, try to get one with the signal */
2629 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2630 pt) == GST_FLOW_FLUSHING)
2633 if (G_UNLIKELY (priv->clock_rate == -1))
2637 /* don't accept more data on EOS */
2638 if (G_UNLIKELY (priv->eos))
2641 calculate_jitter (jitterbuffer, dts, rtptime);
2643 if (priv->seqnum_base != -1) {
2646 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2649 GST_DEBUG_OBJECT (jitterbuffer,
2650 "packet seqnum #%d before seqnum-base #%d", seqnum,
2652 gst_buffer_unref (buffer);
2655 } else if (gap > 16384) {
2656 /* From now on don't compare against the seqnum base anymore as
2657 * at some point in the future we will wrap around and also that
2658 * much reordering is very unlikely */
2659 priv->seqnum_base = -1;
2663 expected = priv->next_in_seqnum;
2666 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2668 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2669 priv->max_dropout_time);
2671 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2672 priv->max_misorder_time);
2673 GST_TRACE_OBJECT (jitterbuffer,
2674 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2675 max_dropout, max_misorder);
2677 /* now check against our expected seqnum */
2678 if (G_LIKELY (expected != -1)) {
2681 /* now calculate gap */
2682 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2684 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2685 expected, seqnum, gap);
2687 if (G_LIKELY (gap == 0)) {
2688 /* packet is expected */
2689 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2690 do_next_seqnum = TRUE;
2692 gboolean reset = FALSE;
2695 /* we received an old packet */
2696 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2698 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2699 gap, max_dropout, max_misorder);
2702 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2705 /* new packet, we are missing some packets */
2706 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2707 /* If we have timers for more than RTP_MAX_DROPOUT packets
2708 * pending this means that we have a huge gap overall. We can
2709 * reset the jitterbuffer at this point because there's
2710 * just too much data missing to be able to do anything
2711 * sensible with the past data. Just try again from the
2713 GST_WARNING_OBJECT (jitterbuffer,
2714 "%d pending timers > %d - resetting", priv->timers->len,
2717 gst_buffer_unref (buffer);
2719 } else if (G_UNLIKELY (gap >= max_dropout)) {
2721 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2722 gap, max_dropout, max_misorder);
2725 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2726 /* fill in the gap with EXPECTED timers */
2727 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2729 do_next_seqnum = TRUE;
2732 if (G_UNLIKELY (reset)) {
2733 GList *events = NULL, *l;
2736 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2737 rtp_jitter_buffer_flush (priv->jbuf,
2738 (GFunc) free_item_and_retain_events, &events);
2739 rtp_jitter_buffer_reset_skew (priv->jbuf);
2740 remove_all_timers (jitterbuffer);
2741 priv->discont = TRUE;
2742 priv->last_popped_seqnum = -1;
2744 if (priv->gap_packets.head) {
2745 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2746 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2748 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2749 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2750 gst_rtp_buffer_unmap (&gap_rtp);
2752 priv->next_seqnum = seqnum;
2755 priv->last_in_dts = -1;
2756 priv->next_in_seqnum = -1;
2758 /* Insert all sticky events again in order, otherwise we would
2759 * potentially loose STREAM_START, CAPS or SEGMENT events
2761 events = g_list_reverse (events);
2762 for (l = events; l; l = l->next) {
2763 RTPJitterBufferItem *item;
2765 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2766 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2768 g_list_free (events);
2770 JBUF_SIGNAL_EVENT (priv);
2772 /* reset spacing estimation when gap */
2773 priv->ips_rtptime = -1;
2774 priv->ips_dts = GST_CLOCK_TIME_NONE;
2776 buffers = g_list_copy (priv->gap_packets.head);
2777 g_queue_clear (&priv->gap_packets);
2779 priv->ips_rtptime = -1;
2780 priv->ips_dts = GST_CLOCK_TIME_NONE;
2781 JBUF_UNLOCK (jitterbuffer->priv);
2783 for (l = buffers; l; l = l->next) {
2784 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2786 if (ret != GST_FLOW_OK)
2789 for (; l; l = l->next)
2790 gst_buffer_unref (l->data);
2791 g_list_free (buffers);
2795 /* reset spacing estimation when gap */
2796 priv->ips_rtptime = -1;
2797 priv->ips_dts = GST_CLOCK_TIME_NONE;
2800 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2802 /* we don't know what the next_in_seqnum should be, wait for the last
2803 * possible moment to push this buffer, maybe we get an earlier seqnum
2805 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2806 do_next_seqnum = TRUE;
2807 /* take rtptime and dts to calculate packet spacing */
2808 priv->ips_rtptime = rtptime;
2809 priv->ips_dts = dts;
2812 /* We had no huge gap, let's drop all the gap packets */
2813 if (buffer != NULL) {
2814 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2815 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2816 g_queue_clear (&priv->gap_packets);
2818 GST_DEBUG_OBJECT (jitterbuffer,
2819 "Had big gap, waiting for more consecutive packets");
2820 JBUF_UNLOCK (jitterbuffer->priv);
2824 if (do_next_seqnum) {
2825 priv->last_in_dts = dts;
2826 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2829 /* let's check if this buffer is too late, we can only accept packets with
2830 * bigger seqnum than the one we last pushed. */
2831 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2834 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2836 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2837 if (G_UNLIKELY (gap <= 0))
2841 if (already_lost (jitterbuffer, seqnum))
2844 /* let's drop oldest packet if the queue is already full and drop-on-latency
2845 * is set. We can only do this when there actually is a latency. When no
2846 * latency is set, we just pump it in the queue and let the other end push it
2847 * out as fast as possible. */
2848 if (priv->latency_ms && priv->drop_on_latency) {
2850 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2852 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2853 RTPJitterBufferItem *old_item;
2855 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2857 if (IS_DROPABLE (old_item)) {
2858 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2859 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2861 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2862 free_item (old_item);
2864 /* we might have removed some head buffers, signal the pushing thread to
2865 * see if it can push now */
2866 JBUF_SIGNAL_EVENT (priv);
2870 /* If we estimated the DTS, don't consider it in the clock skew calculations
2871 * later. The code above always sets dts to pts or the other way around if
2872 * any of those is valid in the buffer, so we know that if we estimated the
2873 * dts that both are unknown */
2876 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2877 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2879 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2881 /* now insert the packet into the queue in sorted order. This function returns
2882 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2883 * have a duplicate. */
2884 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2886 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)))))
2890 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2892 /* we had an unhandled SR, handle it now */
2894 do_handle_sync (jitterbuffer);
2896 if (G_UNLIKELY (head)) {
2897 /* signal addition of new buffer when the _loop is waiting. */
2898 if (G_LIKELY (priv->active))
2899 JBUF_SIGNAL_EVENT (priv);
2901 /* let's unschedule and unblock any waiting buffers. We only want to do this
2902 * when the head buffer changed */
2903 if (G_UNLIKELY (priv->clock_id)) {
2904 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2905 unschedule_current_timer (jitterbuffer);
2909 GST_DEBUG_OBJECT (jitterbuffer,
2910 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2911 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2913 msg = check_buffering_percent (jitterbuffer, percent);
2919 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2926 /* this is not fatal but should be filtered earlier */
2927 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2928 ("Received invalid RTP payload, dropping"));
2929 gst_buffer_unref (buffer);
2934 GST_WARNING_OBJECT (jitterbuffer,
2935 "No clock-rate in caps!, dropping buffer");
2936 gst_buffer_unref (buffer);
2941 ret = priv->srcresult;
2942 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2943 gst_buffer_unref (buffer);
2949 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2950 gst_buffer_unref (buffer);
2955 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2956 " popped, dropping", seqnum, priv->last_popped_seqnum);
2958 gst_buffer_unref (buffer);
2963 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
2964 "considered lost", seqnum);
2966 gst_buffer_unref (buffer);
2971 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2973 priv->num_duplicates++;
2980 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2982 guint64 ext_time, elapsed;
2984 GstRtpJitterBufferPrivate *priv;
2986 priv = jitterbuffer->priv;
2987 rtp_time = item->rtptime;
2989 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2990 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2992 ext_time = priv->ext_timestamp;
2993 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2994 if (ext_time < priv->ext_timestamp) {
2995 ext_time = priv->ext_timestamp;
2997 priv->ext_timestamp = ext_time;
3000 if (ext_time > priv->clock_base)
3001 elapsed = ext_time - priv->clock_base;
3005 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3010 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3011 RTPJitterBufferItem * item)
3013 guint64 total, elapsed, left, estimated;
3014 GstClockTime out_time;
3015 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3017 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3018 || priv->clock_base == -1 || priv->clock_rate <= 0)
3021 /* compute the elapsed time */
3022 elapsed = compute_elapsed (jitterbuffer, item);
3024 /* do nothing if elapsed time doesn't increment */
3025 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3028 priv->last_elapsed = elapsed;
3030 /* this is the total time we need to play */
3031 total = priv->npt_stop - priv->npt_start;
3032 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3033 GST_TIME_ARGS (total));
3035 /* this is how much time there is left */
3036 if (total > elapsed)
3037 left = total - elapsed;
3041 /* if we have less time left that the size of the buffer, we will not
3042 * be able to keep it filled, disabled buffering then */
3043 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3044 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3045 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3046 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3049 /* this is the current time as running-time */
3050 out_time = item->dts;
3053 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3055 /* if there is almost nothing left,
3056 * we may never advance enough to end up in the above case */
3057 if (total < GST_SECOND)
3058 estimated = GST_SECOND;
3062 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3063 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3065 if (estimated != -1 && priv->estimated_eos != estimated) {
3066 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3067 priv->estimated_eos = estimated;
3071 /* take a buffer from the queue and push it */
3072 static GstFlowReturn
3073 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3075 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3076 GstFlowReturn result = GST_FLOW_OK;
3077 RTPJitterBufferItem *item;
3078 GstBuffer *outbuf = NULL;
3079 GstEvent *outevent = NULL;
3080 GstQuery *outquery = NULL;
3081 GstClockTime dts, pts;
3083 gboolean do_push = TRUE;
3087 /* when we get here we are ready to pop and push the buffer */
3088 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3092 case ITEM_TYPE_BUFFER:
3094 /* we need to make writable to change the flags and timestamps */
3095 outbuf = gst_buffer_make_writable (item->data);
3097 if (G_UNLIKELY (priv->discont)) {
3098 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3099 * into the jitterbuffer so we can modify now. */
3100 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3101 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3102 priv->discont = FALSE;
3104 if (G_UNLIKELY (priv->ts_discont)) {
3105 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3106 priv->ts_discont = FALSE;
3110 gst_segment_position_from_running_time (&priv->segment,
3111 GST_FORMAT_TIME, item->dts);
3113 gst_segment_position_from_running_time (&priv->segment,
3114 GST_FORMAT_TIME, item->pts);
3116 /* apply timestamp with offset to buffer now */
3117 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3118 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3120 /* update the elapsed time when we need to check against the npt stop time. */
3121 update_estimated_eos (jitterbuffer, item);
3123 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3125 case ITEM_TYPE_LOST:
3126 priv->discont = TRUE;
3130 case ITEM_TYPE_EVENT:
3131 outevent = item->data;
3133 case ITEM_TYPE_QUERY:
3134 outquery = item->data;
3138 /* now we are ready to push the buffer. Save the seqnum and release the lock
3139 * so the other end can push stuff in the queue again. */
3141 priv->last_popped_seqnum = seqnum;
3142 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3144 msg = check_buffering_percent (jitterbuffer, percent);
3151 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3154 case ITEM_TYPE_BUFFER:
3156 GST_DEBUG_OBJECT (jitterbuffer,
3157 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3158 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3159 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3160 result = gst_pad_push (priv->srcpad, outbuf);
3162 JBUF_LOCK_CHECK (priv, out_flushing);
3164 case ITEM_TYPE_LOST:
3165 case ITEM_TYPE_EVENT:
3166 /* We got not enough consecutive packets with a huge gap, we can
3167 * as well just drop them here now on EOS */
3168 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3169 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3170 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3171 g_queue_clear (&priv->gap_packets);
3174 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3175 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3178 gst_pad_push_event (priv->srcpad, outevent);
3180 gst_event_unref (outevent);
3182 result = GST_FLOW_OK;
3184 JBUF_LOCK_CHECK (priv, out_flushing);
3186 case ITEM_TYPE_QUERY:
3190 res = gst_pad_peer_query (priv->srcpad, outquery);
3192 JBUF_LOCK_CHECK (priv, out_flushing);
3193 result = GST_FLOW_OK;
3194 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3195 JBUF_SIGNAL_QUERY (priv, res);
3204 return priv->srcresult;
3208 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3210 /* Peek a buffer and compare the seqnum to the expected seqnum.
3211 * If all is fine, the buffer is pushed.
3212 * If something is wrong, we wait for some event
3214 static GstFlowReturn
3215 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3217 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3218 GstFlowReturn result;
3219 RTPJitterBufferItem *item;
3221 guint32 next_seqnum;
3223 /* only push buffers when PLAYING and active and not buffering */
3224 if (priv->blocked || !priv->active ||
3225 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3226 return GST_FLOW_WAIT;
3229 /* peek a buffer, we're just looking at the sequence number.
3230 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3231 * wait for a timeout or something to change.
3232 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3233 item = rtp_jitter_buffer_peek (priv->jbuf);
3238 /* get the seqnum and the next expected seqnum */
3239 seqnum = item->seqnum;
3241 return pop_and_push_next (jitterbuffer, seqnum);
3244 next_seqnum = priv->next_seqnum;
3246 /* get the gap between this and the previous packet. If we don't know the
3247 * previous packet seqnum assume no gap. */
3248 if (G_UNLIKELY (next_seqnum == -1)) {
3249 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3250 /* we don't know what the next_seqnum should be, the chain function should
3251 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3252 * fires, so wait for that */
3253 result = GST_FLOW_WAIT;
3255 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3257 if (G_LIKELY (gap == 0)) {
3258 /* no missing packet, pop and push */
3259 result = pop_and_push_next (jitterbuffer, seqnum);
3260 } else if (G_UNLIKELY (gap < 0)) {
3261 /* if we have a packet that we already pushed or considered dropped, pop it
3262 * off and get the next packet */
3263 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3264 seqnum, next_seqnum);
3265 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3267 result = GST_FLOW_OK;
3269 /* the chain function has scheduled timers to request retransmission or
3270 * when to consider the packet lost, wait for that */
3271 GST_DEBUG_OBJECT (jitterbuffer,
3272 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3273 next_seqnum, seqnum, gap);
3274 result = GST_FLOW_WAIT;
3282 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3284 return GST_FLOW_EOS;
3286 return GST_FLOW_WAIT;
3292 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3294 GstClockTime rtx_retry_timeout;
3295 GstClockTime rtx_min_retry_timeout;
3297 if (priv->rtx_retry_timeout == -1) {
3298 if (priv->avg_rtx_rtt == 0)
3299 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3301 /* we want to ask for a retransmission after we waited for a
3302 * complete RTT and the additional jitter */
3303 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3305 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3307 /* make sure we don't retry too often. On very low latency networks,
3308 * the RTT and jitter can be very low. */
3309 if (priv->rtx_min_retry_timeout == -1) {
3310 rtx_min_retry_timeout = priv->packet_spacing;
3312 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3314 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3316 return rtx_retry_timeout;
3320 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3321 GstClockTime rtx_retry_timeout)
3323 GstClockTime rtx_retry_period;
3325 if (priv->rtx_retry_period == -1) {
3326 /* we retry up to the configured jitterbuffer size but leaving some
3327 * room for the retransmission to arrive in time */
3328 if (rtx_retry_timeout > priv->latency_ns) {
3329 rtx_retry_period = 0;
3331 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3334 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3336 return rtx_retry_period;
3339 /* the timeout for when we expected a packet expired */
3341 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3344 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3346 guint delay, delay_ms, avg_rtx_rtt_ms;
3347 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3348 GstClockTime rtx_retry_period;
3349 GstClockTime rtx_retry_timeout;
3352 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3353 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3355 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3356 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3358 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3359 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3360 GST_TIME_ARGS (rtx_retry_period));
3362 delay = timer->rtx_delay + timer->rtx_retry;
3364 delay_ms = GST_TIME_AS_MSECONDS (delay);
3365 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3366 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3367 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3369 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3370 gst_structure_new ("GstRTPRetransmissionRequest",
3371 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3372 "running-time", G_TYPE_UINT64, timer->rtx_base,
3373 "delay", G_TYPE_UINT, delay_ms,
3374 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3375 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3376 "period", G_TYPE_UINT, rtx_retry_period_ms,
3377 "deadline", G_TYPE_UINT, priv->latency_ms,
3378 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3379 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3381 priv->num_rtx_requests++;
3382 timer->num_rtx_retry++;
3384 GST_OBJECT_LOCK (jitterbuffer);
3385 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3386 timer->rtx_last = gst_clock_get_time (clock);
3387 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3389 timer->rtx_last = now;
3391 GST_OBJECT_UNLOCK (jitterbuffer);
3393 /* calculate the timeout for the next retransmission attempt */
3394 timer->rtx_retry += rtx_retry_timeout;
3395 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3396 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3397 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3398 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3399 if ((priv->rtx_max_retries != -1
3400 && timer->num_rtx_retry >= priv->rtx_max_retries)
3401 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3402 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3403 /* too many retransmission request, we now convert the timer
3404 * to a lost timer, leave the num_rtx_retry as it is for stats */
3405 timer->type = TIMER_TYPE_LOST;
3406 timer->rtx_delay = 0;
3407 timer->rtx_retry = 0;
3409 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3410 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3413 gst_pad_push_event (priv->sinkpad, event);
3419 /* a packet is lost */
3421 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3424 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3425 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3427 GstEvent *event = NULL;
3428 RTPJitterBufferItem *item;
3430 seqnum = timer->seqnum;
3431 lost_packets = MAX (timer->num, 1);
3432 num_rtx_retry = timer->num_rtx_retry;
3434 /* we had a gap and thus we lost some packets. Create an event for this. */
3435 if (lost_packets > 1)
3436 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3437 seqnum + lost_packets - 1);
3439 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3441 priv->num_late += lost_packets;
3442 priv->num_rtx_failed += num_rtx_retry;
3444 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3446 /* we now only accept seqnum bigger than this */
3447 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3448 priv->next_in_seqnum = next_in_seqnum;
3450 /* Avoid creating events if we don't need it. Note that we still need to create
3451 * the lost *ITEM* since it will be used to notify the outgoing thread of
3452 * lost items (so that we can set discont flags and such) */
3453 if (priv->do_lost) {
3454 GstClockTime duration, timestamp;
3455 /* create paket lost event */
3456 timestamp = apply_offset (jitterbuffer, timer->timeout);
3457 duration = timer->duration;
3458 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3459 duration = priv->packet_spacing;
3460 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3461 gst_structure_new ("GstRTPPacketLost",
3462 "seqnum", G_TYPE_UINT, (guint) seqnum,
3463 "timestamp", G_TYPE_UINT64, timestamp,
3464 "duration", G_TYPE_UINT64, duration,
3465 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3467 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3468 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3470 /* remove timer now */
3471 remove_timer (jitterbuffer, timer);
3473 JBUF_SIGNAL_EVENT (priv);
3479 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3482 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3484 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3485 remove_timer (jitterbuffer, timer);
3487 /* there was no EOS in the buffer, put one in there now */
3488 queue_event (jitterbuffer, gst_event_new_eos ());
3490 JBUF_SIGNAL_EVENT (priv);
3496 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3499 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3501 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3503 /* timer seqnum might have been obsoleted by caps seqnum-base,
3504 * only mess with current ongoing seqnum if still unknown */
3505 if (priv->next_seqnum == -1)
3506 priv->next_seqnum = timer->seqnum;
3507 remove_timer (jitterbuffer, timer);
3508 JBUF_SIGNAL_EVENT (priv);
3514 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3517 gboolean removed = FALSE;
3519 switch (timer->type) {
3520 case TIMER_TYPE_EXPECTED:
3521 removed = do_expected_timeout (jitterbuffer, timer, now);
3523 case TIMER_TYPE_LOST:
3524 removed = do_lost_timeout (jitterbuffer, timer, now);
3526 case TIMER_TYPE_DEADLINE:
3527 removed = do_deadline_timeout (jitterbuffer, timer, now);
3529 case TIMER_TYPE_EOS:
3530 removed = do_eos_timeout (jitterbuffer, timer, now);
3536 /* called when we need to wait for the next timeout.
3538 * We loop over the array of recorded timeouts and wait for the earliest one.
3539 * When it timed out, do the logic associated with the timer.
3541 * If there are no timers, we wait on a gcond until something new happens.
3544 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3546 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3547 GstClockTime now = 0;
3550 while (priv->timer_running) {
3551 TimerData *timer = NULL;
3552 GstClockTime timer_timeout = -1;
3555 /* If we have a clock, update "now" now with the very
3556 * latest running time we have. If timers are unscheduled below we
3557 * otherwise wouldn't update now (it's only updated when timers
3558 * expire), and also for the very first loop iteration now would
3559 * otherwise always be 0
3561 GST_OBJECT_LOCK (jitterbuffer);
3562 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3564 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3565 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3567 GST_OBJECT_UNLOCK (jitterbuffer);
3569 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3570 GST_TIME_ARGS (now));
3572 len = priv->timers->len;
3573 for (i = 0; i < len;) {
3574 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3575 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3576 gboolean save_best = FALSE;
3578 GST_DEBUG_OBJECT (jitterbuffer,
3579 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3580 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3581 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3583 /* Weed out anything too late */
3584 if (test->type == TIMER_TYPE_LOST &&
3585 (test_timeout == -1 || test_timeout <= now)) {
3586 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3587 do_lost_timeout (jitterbuffer, test, now);
3588 if (!priv->timer_running)
3590 /* We don't move the iterator forward since we just removed the current entry,
3591 * but we update the termination condition */
3592 len = priv->timers->len;
3594 /* find the smallest timeout */
3595 if (timer == NULL) {
3597 } else if (timer_timeout == -1) {
3598 /* we already have an immediate timeout, the new timer must be an
3599 * immediate timer with smaller seqnum to become the best */
3600 if (test_timeout == -1
3601 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3602 timer->seqnum) > 0))
3604 } else if (test_timeout == -1) {
3605 /* first immediate timer */
3607 } else if (test_timeout < timer_timeout) {
3610 } else if (test_timeout == timer_timeout
3611 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3612 timer->seqnum) > 0)) {
3613 /* same timer, smaller seqnum */
3618 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3620 timer_timeout = test_timeout;
3625 if (timer && !priv->blocked) {
3627 GstClockTime sync_time;
3630 GstClockTimeDiff clock_jitter;
3632 if (timer_timeout == -1 || timer_timeout <= now) {
3633 /* We have normally removed all lost timers in the loop above */
3634 g_assert (timer->type != TIMER_TYPE_LOST);
3636 do_timeout (jitterbuffer, timer, now);
3637 /* check here, do_timeout could have released the lock */
3638 if (!priv->timer_running)
3643 GST_OBJECT_LOCK (jitterbuffer);
3644 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3646 GST_OBJECT_UNLOCK (jitterbuffer);
3647 /* let's just push if there is no clock */
3648 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3649 now = timer_timeout;
3653 /* prepare for sync against clock */
3654 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3655 /* add latency of peer to get input time */
3656 sync_time += priv->peer_latency;
3658 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3659 " with sync time %" GST_TIME_FORMAT,
3660 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3662 /* create an entry for the clock */
3663 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3664 priv->timer_timeout = timer_timeout;
3665 priv->timer_seqnum = timer->seqnum;
3666 GST_OBJECT_UNLOCK (jitterbuffer);
3668 /* release the lock so that the other end can push stuff or unlock */
3671 ret = gst_clock_id_wait (id, &clock_jitter);
3674 if (!priv->timer_running) {
3675 gst_clock_id_unref (id);
3676 priv->clock_id = NULL;
3680 if (ret != GST_CLOCK_UNSCHEDULED) {
3681 now = timer_timeout + MAX (clock_jitter, 0);
3682 GST_DEBUG_OBJECT (jitterbuffer,
3683 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3684 GST_STIME_ARGS (clock_jitter));
3686 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3688 /* and free the entry */
3689 gst_clock_id_unref (id);
3690 priv->clock_id = NULL;
3692 /* no timers, wait for activity */
3693 JBUF_WAIT_TIMER (priv);
3698 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3703 * This funcion implements the main pushing loop on the source pad.
3705 * It first tries to push as many buffers as possible. If there is a seqnum
3706 * mismatch, we wait for the next timeouts.
3709 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3711 GstRtpJitterBufferPrivate *priv;
3712 GstFlowReturn result = GST_FLOW_OK;
3714 priv = jitterbuffer->priv;
3716 JBUF_LOCK_CHECK (priv, flushing);
3718 result = handle_next_buffer (jitterbuffer);
3719 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3720 /* now wait for the next event */
3721 JBUF_WAIT_EVENT (priv, flushing);
3722 result = GST_FLOW_OK;
3724 } while (result == GST_FLOW_OK);
3725 /* store result for upstream */
3726 priv->srcresult = result;
3727 /* if we get here we need to pause */
3733 result = priv->srcresult;
3740 JBUF_SIGNAL_QUERY (priv, FALSE);
3743 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3744 gst_flow_get_name (result));
3745 gst_pad_pause_task (priv->srcpad);
3746 if (result == GST_FLOW_EOS) {
3747 event = gst_event_new_eos ();
3748 gst_pad_push_event (priv->srcpad, event);
3754 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3755 * some sanity checks and then emit the handle-sync signal with the parameters.
3756 * This function must be called with the LOCK */
3758 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3760 GstRtpJitterBufferPrivate *priv;
3761 guint64 base_rtptime, base_time;
3763 guint64 last_rtptime;
3765 guint64 ext_rtptime, diff;
3766 gboolean valid = TRUE, keep = FALSE;
3768 priv = jitterbuffer->priv;
3770 /* get the last values from the jitterbuffer */
3771 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3772 &clock_rate, &last_rtptime);
3774 clock_base = priv->clock_base;
3775 ext_rtptime = priv->ext_rtptime;
3777 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3778 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3779 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3780 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3782 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3783 /* we keep this SR packet for later. When we get a valid RTP packet the
3784 * above values will be set and we can try to use the SR packet */
3785 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3788 /* we can't accept anything that happened before we did the last resync */
3789 if (base_rtptime > ext_rtptime) {
3790 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3793 /* the SR RTP timestamp must be something close to what we last observed
3794 * in the jitterbuffer */
3795 if (ext_rtptime > last_rtptime) {
3796 /* check how far ahead it is to our RTP timestamps */
3797 diff = ext_rtptime - last_rtptime;
3798 /* if bigger than 1 second, we drop it */
3799 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3801 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3802 clock_rate, 1000)) {
3803 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3804 /* should drop this, but some RTSP servers end up with bogus
3805 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3806 * so still trigger rptbin sync but invalidate RTCP data
3807 * (sync might use other methods) */
3810 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3811 G_GUINT64_FORMAT, last_rtptime, diff);
3817 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3821 s = gst_structure_new ("application/x-rtp-sync",
3822 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3823 "base-time", G_TYPE_UINT64, base_time,
3824 "clock-rate", G_TYPE_UINT, clock_rate,
3825 "clock-base", G_TYPE_UINT64, clock_base,
3826 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3827 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3829 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3830 gst_buffer_replace (&priv->last_sr, NULL);
3832 g_signal_emit (jitterbuffer,
3833 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3835 gst_structure_free (s);
3837 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3838 gst_buffer_replace (&priv->last_sr, NULL);
3842 static GstFlowReturn
3843 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3846 GstRtpJitterBuffer *jitterbuffer;
3847 GstRtpJitterBufferPrivate *priv;
3848 GstFlowReturn ret = GST_FLOW_OK;
3850 GstRTCPPacket packet;
3851 guint64 ext_rtptime;
3853 GstRTCPBuffer rtcp = { NULL, };
3855 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3857 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3858 goto invalid_buffer;
3860 priv = jitterbuffer->priv;
3862 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3864 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3867 /* first packet must be SR or RR or else the validate would have failed */
3868 switch (gst_rtcp_packet_get_type (&packet)) {
3869 case GST_RTCP_TYPE_SR:
3870 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3876 gst_rtcp_buffer_unmap (&rtcp);
3878 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3881 /* convert the RTP timestamp to our extended timestamp, using the same offset
3882 * we used in the jitterbuffer */
3883 ext_rtptime = priv->jbuf->ext_rtptime;
3884 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3886 priv->ext_rtptime = ext_rtptime;
3887 gst_buffer_replace (&priv->last_sr, buffer);
3889 do_handle_sync (jitterbuffer);
3893 gst_buffer_unref (buffer);
3899 /* this is not fatal but should be filtered earlier */
3900 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3901 ("Received invalid RTCP payload, dropping"));
3907 /* this is not fatal but should be filtered earlier */
3908 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3909 ("Received empty RTCP payload, dropping"));
3910 gst_rtcp_buffer_unmap (&rtcp);
3916 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3917 gst_rtcp_buffer_unmap (&rtcp);
3924 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3927 gboolean res = FALSE;
3928 GstRtpJitterBuffer *jitterbuffer;
3929 GstRtpJitterBufferPrivate *priv;
3931 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3932 priv = jitterbuffer->priv;
3934 switch (GST_QUERY_TYPE (query)) {
3935 case GST_QUERY_CAPS:
3937 GstCaps *filter, *caps;
3939 gst_query_parse_caps (query, &filter);
3940 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3941 gst_query_set_caps_result (query, caps);
3942 gst_caps_unref (caps);
3947 if (GST_QUERY_IS_SERIALIZED (query)) {
3948 RTPJitterBufferItem *item;
3951 JBUF_LOCK_CHECK (priv, out_flushing);
3952 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3953 RTP_JITTER_BUFFER_MODE_BUFFER) {
3954 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3955 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3956 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3958 JBUF_SIGNAL_EVENT (priv);
3959 JBUF_WAIT_QUERY (priv, out_flushing);
3960 res = priv->last_query;
3962 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3967 res = gst_pad_query_default (pad, parent, query);
3975 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3983 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3986 GstRtpJitterBuffer *jitterbuffer;
3987 GstRtpJitterBufferPrivate *priv;
3988 gboolean res = FALSE;
3990 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3991 priv = jitterbuffer->priv;
3993 switch (GST_QUERY_TYPE (query)) {
3994 case GST_QUERY_LATENCY:
3996 /* We need to send the query upstream and add the returned latency to our
3998 GstClockTime min_latency, max_latency;
4000 GstClockTime our_latency;
4002 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4003 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4005 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4006 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4007 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4009 /* store this so that we can safely sync on the peer buffers. */
4011 priv->peer_latency = min_latency;
4012 our_latency = priv->latency_ns;
4015 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4016 GST_TIME_ARGS (our_latency));
4018 /* we add some latency but can buffer an infinite amount of time */
4019 min_latency += our_latency;
4022 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4023 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4024 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4026 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4030 case GST_QUERY_POSITION:
4032 GstClockTime start, last_out;
4035 gst_query_parse_position (query, &fmt, NULL);
4036 if (fmt != GST_FORMAT_TIME) {
4037 res = gst_pad_query_default (pad, parent, query);
4042 start = priv->npt_start;
4043 last_out = priv->last_out_time;
4046 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4047 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4048 GST_TIME_ARGS (last_out));
4050 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4051 /* bring 0-based outgoing time to stream time */
4052 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4055 res = gst_pad_query_default (pad, parent, query);
4059 case GST_QUERY_CAPS:
4061 GstCaps *filter, *caps;
4063 gst_query_parse_caps (query, &filter);
4064 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4065 gst_query_set_caps_result (query, caps);
4066 gst_caps_unref (caps);
4071 res = gst_pad_query_default (pad, parent, query);
4079 gst_rtp_jitter_buffer_set_property (GObject * object,
4080 guint prop_id, const GValue * value, GParamSpec * pspec)
4082 GstRtpJitterBuffer *jitterbuffer;
4083 GstRtpJitterBufferPrivate *priv;
4085 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4086 priv = jitterbuffer->priv;
4091 guint new_latency, old_latency;
4093 new_latency = g_value_get_uint (value);
4096 old_latency = priv->latency_ms;
4097 priv->latency_ms = new_latency;
4098 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4099 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4102 /* post message if latency changed, this will inform the parent pipeline
4103 * that a latency reconfiguration is possible/needed. */
4104 if (new_latency != old_latency) {
4105 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4106 GST_TIME_ARGS (new_latency * GST_MSECOND));
4108 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4109 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4113 case PROP_DROP_ON_LATENCY:
4115 priv->drop_on_latency = g_value_get_boolean (value);
4118 case PROP_TS_OFFSET:
4120 priv->ts_offset = g_value_get_int64 (value);
4121 priv->ts_discont = TRUE;
4126 priv->do_lost = g_value_get_boolean (value);
4131 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4134 case PROP_DO_RETRANSMISSION:
4136 priv->do_retransmission = g_value_get_boolean (value);
4139 case PROP_RTX_NEXT_SEQNUM:
4141 priv->rtx_next_seqnum = g_value_get_boolean (value);
4144 case PROP_RTX_DELAY:
4146 priv->rtx_delay = g_value_get_int (value);
4149 case PROP_RTX_MIN_DELAY:
4151 priv->rtx_min_delay = g_value_get_uint (value);
4154 case PROP_RTX_DELAY_REORDER:
4156 priv->rtx_delay_reorder = g_value_get_int (value);
4159 case PROP_RTX_RETRY_TIMEOUT:
4161 priv->rtx_retry_timeout = g_value_get_int (value);
4164 case PROP_RTX_MIN_RETRY_TIMEOUT:
4166 priv->rtx_min_retry_timeout = g_value_get_int (value);
4169 case PROP_RTX_RETRY_PERIOD:
4171 priv->rtx_retry_period = g_value_get_int (value);
4174 case PROP_RTX_MAX_RETRIES:
4176 priv->rtx_max_retries = g_value_get_int (value);
4179 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4181 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4184 case PROP_MAX_DROPOUT_TIME:
4186 priv->max_dropout_time = g_value_get_uint (value);
4189 case PROP_MAX_MISORDER_TIME:
4191 priv->max_misorder_time = g_value_get_uint (value);
4194 case PROP_RFC7273_SYNC:
4196 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4197 g_value_get_boolean (value));
4201 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4207 gst_rtp_jitter_buffer_get_property (GObject * object,
4208 guint prop_id, GValue * value, GParamSpec * pspec)
4210 GstRtpJitterBuffer *jitterbuffer;
4211 GstRtpJitterBufferPrivate *priv;
4213 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4214 priv = jitterbuffer->priv;
4219 g_value_set_uint (value, priv->latency_ms);
4222 case PROP_DROP_ON_LATENCY:
4224 g_value_set_boolean (value, priv->drop_on_latency);
4227 case PROP_TS_OFFSET:
4229 g_value_set_int64 (value, priv->ts_offset);
4234 g_value_set_boolean (value, priv->do_lost);
4239 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4247 if (priv->srcresult != GST_FLOW_OK)
4250 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4252 g_value_set_int (value, percent);
4256 case PROP_DO_RETRANSMISSION:
4258 g_value_set_boolean (value, priv->do_retransmission);
4261 case PROP_RTX_NEXT_SEQNUM:
4263 g_value_set_boolean (value, priv->rtx_next_seqnum);
4266 case PROP_RTX_DELAY:
4268 g_value_set_int (value, priv->rtx_delay);
4271 case PROP_RTX_MIN_DELAY:
4273 g_value_set_uint (value, priv->rtx_min_delay);
4276 case PROP_RTX_DELAY_REORDER:
4278 g_value_set_int (value, priv->rtx_delay_reorder);
4281 case PROP_RTX_RETRY_TIMEOUT:
4283 g_value_set_int (value, priv->rtx_retry_timeout);
4286 case PROP_RTX_MIN_RETRY_TIMEOUT:
4288 g_value_set_int (value, priv->rtx_min_retry_timeout);
4291 case PROP_RTX_RETRY_PERIOD:
4293 g_value_set_int (value, priv->rtx_retry_period);
4296 case PROP_RTX_MAX_RETRIES:
4298 g_value_set_int (value, priv->rtx_max_retries);
4302 g_value_take_boxed (value,
4303 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4305 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4307 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4310 case PROP_MAX_DROPOUT_TIME:
4312 g_value_set_uint (value, priv->max_dropout_time);
4315 case PROP_MAX_MISORDER_TIME:
4317 g_value_set_uint (value, priv->max_misorder_time);
4320 case PROP_RFC7273_SYNC:
4322 g_value_set_boolean (value,
4323 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4327 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4332 static GstStructure *
4333 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4337 JBUF_LOCK (jbuf->priv);
4338 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4339 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4340 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4341 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4342 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4343 JBUF_UNLOCK (jbuf->priv);