2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
10 * Copyright 2016 Pexip AS
11 * @author: Havard Graff <havard@pexip.com>
12 * @author: Stian Selnes <stian@pexip.com>
14 * This library is free software; you can redistribute it and/or
15 * modify it under the terms of the GNU Library General Public
16 * License as published by the Free Software Foundation; either
17 * version 2 of the License, or (at your option) any later version.
19 * This library is distributed in the hope that it will be useful,
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
22 * Library General Public License for more details.
24 * You should have received a copy of the GNU Library General Public
25 * License along with this library; if not, write to the
26 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
27 * Boston, MA 02110-1301, USA.
32 * SECTION:element-rtpjitterbuffer
34 * This element reorders and removes duplicate RTP packets as they are received
35 * from a network source.
37 * The element needs the clock-rate of the RTP payload in order to estimate the
38 * delay. This information is obtained either from the caps on the sink pad or,
39 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
40 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
42 * The rtpjitterbuffer will wait for missing packets up to a configurable time
43 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
44 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
45 * property is set, lost packets will result in a custom serialized downstream
46 * event of name GstRTPPacketLost. The lost packet events are usually used by a
47 * depayloader or other element to create concealment data or some other logic
48 * to gracefully handle the missing packets.
50 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
51 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
54 * The jitterbuffer can also be configured to send early retransmission events
55 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
56 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
57 * sends a custom upstream event named GstRTPRetransmissionRequest when the
58 * packet is considered late. The initial expected packet arrival time is
59 * calculated as follows:
61 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
62 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
63 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
64 * packets with different rtptime.
66 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
67 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
68 * previously scheduled timeout is overwritten.
70 * - If seqnum N arrived, all seqnum older than
71 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
72 * immediately. This is to request fast feedback for abonormally reorder
73 * packets before any of the previous timeouts is triggered.
75 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
76 * event. After the initial timeout expires and the retransmission event is
77 * sent, the timeout is scheduled for
78 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
79 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
80 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
81 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
82 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
83 * retransmission requests are sent and the regular logic is performed to
84 * schedule a lost packet as discussed above.
86 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
89 * This element will automatically be used inside rtpbin.
92 * <title>Example pipelines</title>
94 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
95 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
96 * inserted into the pipeline to smooth out network jitter and to reorder the
97 * out-of-order RTP packets.
108 #include <gst/rtp/gstrtpbuffer.h>
109 #include <gst/net/net.h>
111 #include "gstrtpjitterbuffer.h"
112 #include "rtpjitterbuffer.h"
113 #include "rtpstats.h"
115 #include <gst/glib-compat-private.h>
117 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
118 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
120 /* RTPJitterBuffer signals and args */
123 SIGNAL_REQUEST_PT_MAP,
131 #define DEFAULT_LATENCY_MS 200
132 #define DEFAULT_DROP_ON_LATENCY FALSE
133 #define DEFAULT_TS_OFFSET 0
134 #define DEFAULT_DO_LOST FALSE
135 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
136 #define DEFAULT_PERCENT 0
137 #define DEFAULT_DO_RETRANSMISSION FALSE
138 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
139 #define DEFAULT_RTX_DELAY -1
140 #define DEFAULT_RTX_MIN_DELAY 0
141 #define DEFAULT_RTX_DELAY_REORDER 3
142 #define DEFAULT_RTX_RETRY_TIMEOUT -1
143 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
144 #define DEFAULT_RTX_RETRY_PERIOD -1
145 #define DEFAULT_RTX_MAX_RETRIES -1
146 #define DEFAULT_RTX_DEADLINE -1
147 #define DEFAULT_RTX_STATS_TIMEOUT 1000
148 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
149 #define DEFAULT_MAX_DROPOUT_TIME 60000
150 #define DEFAULT_MAX_MISORDER_TIME 2000
151 #define DEFAULT_RFC7273_SYNC FALSE
152 #define DEFAULT_FASTSTART_MIN_PACKETS 0
154 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
155 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
161 PROP_DROP_ON_LATENCY,
166 PROP_DO_RETRANSMISSION,
167 PROP_RTX_NEXT_SEQNUM,
170 PROP_RTX_DELAY_REORDER,
171 PROP_RTX_RETRY_TIMEOUT,
172 PROP_RTX_MIN_RETRY_TIMEOUT,
173 PROP_RTX_RETRY_PERIOD,
174 PROP_RTX_MAX_RETRIES,
176 PROP_RTX_STATS_TIMEOUT,
178 PROP_MAX_RTCP_RTP_TIME_DIFF,
179 PROP_MAX_DROPOUT_TIME,
180 PROP_MAX_MISORDER_TIME,
182 PROP_FASTSTART_MIN_PACKETS
185 #define JBUF_LOCK(priv) G_STMT_START { \
186 GST_TRACE("Locking from thread %p", g_thread_self()); \
187 (g_mutex_lock (&(priv)->jbuf_lock)); \
188 GST_TRACE("Locked from thread %p", g_thread_self()); \
191 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
193 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
196 #define JBUF_UNLOCK(priv) G_STMT_START { \
197 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
198 (g_mutex_unlock (&(priv)->jbuf_lock)); \
201 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
202 GST_DEBUG ("waiting timer"); \
203 (priv)->waiting_timer = TRUE; \
204 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
205 (priv)->waiting_timer = FALSE; \
206 GST_DEBUG ("waiting timer done"); \
208 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
209 if (G_UNLIKELY ((priv)->waiting_timer)) { \
210 GST_DEBUG ("signal timer"); \
211 g_cond_signal (&(priv)->jbuf_timer); \
215 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
216 GST_DEBUG ("waiting event"); \
217 (priv)->waiting_event = TRUE; \
218 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
219 (priv)->waiting_event = FALSE; \
220 GST_DEBUG ("waiting event done"); \
221 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
224 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
225 if (G_UNLIKELY ((priv)->waiting_event)) { \
226 GST_DEBUG ("signal event"); \
227 g_cond_signal (&(priv)->jbuf_event); \
231 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
232 GST_DEBUG ("waiting query"); \
233 (priv)->waiting_query = TRUE; \
234 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
235 (priv)->waiting_query = FALSE; \
236 GST_DEBUG ("waiting query done"); \
237 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
240 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
241 (priv)->last_query = res; \
242 if (G_UNLIKELY ((priv)->waiting_query)) { \
243 GST_DEBUG ("signal query"); \
244 g_cond_signal (&(priv)->jbuf_query); \
248 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
249 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
251 typedef struct TimerQueue
254 GHashTable *hashtable;
257 struct _GstRtpJitterBufferPrivate
259 GstPad *sinkpad, *srcpad;
262 RTPJitterBuffer *jbuf;
264 gboolean waiting_timer;
266 gboolean waiting_event;
268 gboolean waiting_query;
276 gboolean timer_running;
277 GThread *timer_thread;
282 gboolean drop_on_latency;
285 gboolean do_retransmission;
286 gboolean rtx_next_seqnum;
289 gint rtx_delay_reorder;
290 gint rtx_retry_timeout;
291 gint rtx_min_retry_timeout;
292 gint rtx_retry_period;
293 gint rtx_max_retries;
294 guint rtx_stats_timeout;
295 gint rtx_deadline_ms;
296 gint max_rtcp_rtp_time_diff;
297 guint32 max_dropout_time;
298 guint32 max_misorder_time;
299 guint faststart_min_packets;
301 /* the last seqnum we pushed out */
302 guint32 last_popped_seqnum;
303 /* the next expected seqnum we push */
305 /* seqnum-base, if known */
307 /* last output time */
308 GstClockTime last_out_time;
309 /* last valid input timestamp and rtptime pair */
310 GstClockTime ips_pts;
312 GstClockTime packet_spacing;
317 /* the next expected seqnum we receive */
318 GstClockTime last_in_pts;
319 guint32 next_in_seqnum;
322 TimerQueue *rtx_stats_timers;
324 /* start and stop ranges */
325 GstClockTime npt_start;
326 GstClockTime npt_stop;
327 guint64 ext_timestamp;
328 guint64 last_elapsed;
329 guint64 estimated_eos;
336 /* clock rate and rtp timestamp offset */
340 gint64 prev_ts_offset;
342 /* when we are shutting down */
343 GstFlowReturn srcresult;
349 GstClockTime timer_timeout;
350 guint16 timer_seqnum;
351 /* the latency of the upstream peer, we have to take this into account when
352 * synchronizing the buffers. */
353 GstClockTime peer_latency;
357 /* some accounting */
361 guint64 num_duplicates;
362 guint64 num_rtx_requests;
363 guint64 num_rtx_success;
364 guint64 num_rtx_failed;
367 RTPPacketRateCtx packet_rate_ctx;
370 GstClockTime last_dts;
371 guint64 last_rtptime;
372 GstClockTime avg_jitter;
389 GstClockTime timeout;
390 GstClockTime duration;
391 GstClockTime rtx_base;
392 GstClockTime rtx_delay;
393 GstClockTime rtx_retry;
394 GstClockTime rtx_last;
396 guint num_rtx_received;
399 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
400 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
401 GstRtpJitterBufferPrivate))
403 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
404 GST_STATIC_PAD_TEMPLATE ("sink",
407 GST_STATIC_CAPS ("application/x-rtp"
408 /* "clock-rate = (int) [ 1, 2147483647 ], "
409 * "payload = (int) , "
410 * "encoding-name = (string) "
414 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
415 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
418 GST_STATIC_CAPS ("application/x-rtcp")
421 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
422 GST_STATIC_PAD_TEMPLATE ("src",
425 GST_STATIC_CAPS ("application/x-rtp"
426 /* "payload = (int) , "
427 * "clock-rate = (int) , "
428 * "encoding-name = (string) "
432 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
434 #define gst_rtp_jitter_buffer_parent_class parent_class
435 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
437 /* object overrides */
438 static void gst_rtp_jitter_buffer_set_property (GObject * object,
439 guint prop_id, const GValue * value, GParamSpec * pspec);
440 static void gst_rtp_jitter_buffer_get_property (GObject * object,
441 guint prop_id, GValue * value, GParamSpec * pspec);
442 static void gst_rtp_jitter_buffer_finalize (GObject * object);
444 /* element overrides */
445 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
446 * element, GstStateChange transition);
447 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
448 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
449 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
451 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
452 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
456 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
457 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
460 /* sinkpad overrides */
461 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
462 GstObject * parent, GstEvent * event);
463 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
464 GstObject * parent, GstBuffer * buffer);
466 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
467 GstObject * parent, GstEvent * event);
468 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
469 GstObject * parent, GstBuffer * buffer);
471 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
472 GstObject * parent, GstQuery * query);
474 /* srcpad overrides */
475 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
476 GstObject * parent, GstEvent * event);
477 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
478 GstObject * parent, GstPadMode mode, gboolean active);
479 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
480 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
481 GstObject * parent, GstQuery * query);
484 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
486 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
487 gboolean active, guint64 base_time);
488 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
490 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
491 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
493 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
495 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
498 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
499 TimerData * timer, GstClockTime dts, gboolean success);
501 static TimerQueue *timer_queue_new (void);
502 static void timer_queue_free (TimerQueue * queue);
505 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
507 GObjectClass *gobject_class;
508 GstElementClass *gstelement_class;
510 gobject_class = (GObjectClass *) klass;
511 gstelement_class = (GstElementClass *) klass;
513 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
515 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
517 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
518 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
521 * GstRtpJitterBuffer:latency:
523 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
524 * for at most this time.
526 g_object_class_install_property (gobject_class, PROP_LATENCY,
527 g_param_spec_uint ("latency", "Buffer latency in ms",
528 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
529 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
531 * GstRtpJitterBuffer:drop-on-latency:
533 * Drop oldest buffers when the queue is completely filled.
535 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
536 g_param_spec_boolean ("drop-on-latency",
537 "Drop buffers when maximum latency is reached",
538 "Tells the jitterbuffer to never exceed the given latency in size",
539 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
541 * GstRtpJitterBuffer:ts-offset:
543 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
544 * This is mainly used to ensure interstream synchronisation.
546 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
547 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
548 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
549 G_MAXINT64, DEFAULT_TS_OFFSET,
550 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
553 * GstRtpJitterBuffer:do-lost:
555 * Send out a GstRTPPacketLost event downstream when a packet is considered
558 g_object_class_install_property (gobject_class, PROP_DO_LOST,
559 g_param_spec_boolean ("do-lost", "Do Lost",
560 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
561 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
564 * GstRtpJitterBuffer:mode:
566 * Control the buffering and timestamping mode used by the jitterbuffer.
568 g_object_class_install_property (gobject_class, PROP_MODE,
569 g_param_spec_enum ("mode", "Mode",
570 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
571 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 * GstRtpJitterBuffer:percent:
575 * The percent of the jitterbuffer that is filled.
577 g_object_class_install_property (gobject_class, PROP_PERCENT,
578 g_param_spec_int ("percent", "percent",
579 "The buffer filled percent", 0, 100,
580 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
582 * GstRtpJitterBuffer:do-retransmission:
584 * Send out a GstRTPRetransmission event upstream when a packet is considered
585 * late and should be retransmitted.
589 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
590 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
591 "Send retransmission events upstream when a packet is late",
592 DEFAULT_DO_RETRANSMISSION,
593 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
596 * GstRtpJitterBuffer:rtx-next-seqnum
598 * Estimate when the next packet should arrive and schedule a retransmission
600 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
601 * for packet N+1. So it will be requested if it does not arrive at the expected time.
602 * The expected time is calculated using the dts of N and the packet spacing.
606 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
607 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
608 "Estimate when the next packet should arrive and schedule a "
609 "retransmission request for it.",
610 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
613 * GstRtpJitterBuffer:rtx-delay:
615 * When a packet did not arrive at the expected time, wait this extra amount
616 * of time before sending a retransmission event.
618 * When -1 is used, the max jitter will be used as extra delay.
622 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
623 g_param_spec_int ("rtx-delay", "RTX Delay",
624 "Extra time in ms to wait before sending retransmission "
625 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
626 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
629 * GstRtpJitterBuffer:rtx-min-delay:
631 * When a packet did not arrive at the expected time, wait at least this extra amount
632 * of time before sending a retransmission event.
636 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
637 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
638 "Minimum time in ms to wait before sending retransmission "
639 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
640 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
642 * GstRtpJitterBuffer:rtx-delay-reorder:
644 * Assume that a retransmission event should be sent when we see
645 * this much packet reordering.
647 * When -1 is used, the value will be estimated based on observed packet
648 * reordering. When 0 is used packet reordering alone will not cause a
649 * retransmission event (Since 1.10).
653 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
654 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
655 "Sending retransmission event when this much reordering "
656 "(0 disable, -1 automatic)",
657 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
658 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
660 * GstRtpJitterBuffer::rtx-retry-timeout:
662 * When no packet has been received after sending a retransmission event
663 * for this time, retry sending a retransmission event.
665 * When -1 is used, the value will be estimated based on observed round
670 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
671 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
672 "Retry sending a transmission event after this timeout in "
673 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
674 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
676 * GstRtpJitterBuffer::rtx-min-retry-timeout:
678 * The minimum amount of time between retry timeouts. When
679 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
680 * minimum interval between retry timeouts.
682 * When -1 is used, the value will be estimated based on the
687 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
688 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
689 "Minimum timeout between sending a transmission event in "
690 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
691 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
693 * GstRtpJitterBuffer:rtx-retry-period:
695 * The amount of time to try to get a retransmission.
697 * When -1 is used, the value will be estimated based on the jitterbuffer
698 * latency and the observed round trip time.
702 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
703 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
704 "Try to get a retransmission for this many ms "
705 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
706 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
708 * GstRtpJitterBuffer:rtx-max-retries:
710 * The maximum number of retries to request a retransmission.
712 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
713 * When -1 is used, the number of retransmission request will not be limited.
717 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
718 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
719 "The maximum number of retries to request a retransmission. "
720 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
721 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
723 * GstRtpJitterBuffer:rtx-deadline:
725 * The deadline for a valid RTX request in ms.
727 * How long the RTX RTCP will be valid for.
728 * When -1 is used, the size of the jitterbuffer will be used.
732 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
733 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
734 "The deadline for a valid RTX request in milliseconds. "
735 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
736 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
738 * GstRtpJitterBuffer::rtx-stats-timeout:
740 * The time to wait for a retransmitted packet after it has been
741 * considered lost in order to collect RTX statistics.
745 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
746 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
747 "The time to wait for a retransmitted packet after it has been "
748 "considered lost in order to collect statistics (ms)",
749 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
750 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
752 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
753 g_param_spec_uint ("max-dropout-time", "Max dropout time",
754 "The maximum time (milliseconds) of missing packets tolerated.",
755 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
756 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
758 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
759 g_param_spec_uint ("max-misorder-time", "Max misorder time",
760 "The maximum time (milliseconds) of misordered packets tolerated.",
761 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
762 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
764 * GstRtpJitterBuffer:stats:
766 * Various jitterbuffer statistics. This property returns a GstStructure
767 * with name application/x-rtp-jitterbuffer-stats with the following fields:
773 * <classname>"num-pushed"</classname>:
774 * the number of packets pushed out.
780 * <classname>"num-lost"</classname>:
781 * the number of packets considered lost.
787 * <classname>"num-late"</classname>:
788 * the number of packets arriving too late.
794 * <classname>"num-duplicates"</classname>:
795 * the number of duplicate packets.
801 * <classname>"rtx-count"</classname>:
802 * the number of retransmissions requested.
808 * <classname>"rtx-success-count"</classname>:
809 * the number of successful retransmissions.
815 * <classname>"rtx-per-packet"</classname>:
816 * average number of RTX per packet.
822 * <classname>"rtx-rtt"</classname>:
823 * average round trip time per RTX.
830 g_object_class_install_property (gobject_class, PROP_STATS,
831 g_param_spec_boxed ("stats", "Statistics",
832 "Various statistics", GST_TYPE_STRUCTURE,
833 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
836 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
838 * The maximum amount of time in ms that the RTP time in the RTCP SRs
839 * is allowed to be ahead of the last RTP packet we received. Use
840 * -1 to disable ignoring of RTCP packets.
844 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
845 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
846 "Maximum amount of time in ms that the RTP time in RTCP SRs "
847 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
848 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
849 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
851 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
852 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
853 "Synchronize received streams to the RFC7273 clock "
854 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
855 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
858 * GstRtpJitterBuffer:faststart-min-packets
860 * The number of consecutive packets needed to start (set to 0 to
861 * disable faststart. The jitterbuffer will by default start after the
862 * latency has elapsed)
866 g_object_class_install_property (gobject_class, PROP_FASTSTART_MIN_PACKETS,
867 g_param_spec_uint ("faststart-min-packets", "Faststart minimum packets",
868 "The number of consecutive packets needed to start (set to 0 to "
869 "disable faststart. The jitterbuffer will by default start after "
870 "the latency has elapsed)",
871 0, G_MAXUINT, DEFAULT_FASTSTART_MIN_PACKETS,
872 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
875 * GstRtpJitterBuffer::request-pt-map:
876 * @buffer: the object which received the signal
879 * Request the payload type as #GstCaps for @pt.
881 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
882 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
883 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
884 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
885 GST_TYPE_CAPS, 1, G_TYPE_UINT);
887 * GstRtpJitterBuffer::handle-sync:
888 * @buffer: the object which received the signal
889 * @struct: a GstStructure containing sync values.
891 * Be notified of new sync values.
893 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
894 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
895 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
896 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
897 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
900 * GstRtpJitterBuffer::on-npt-stop:
901 * @buffer: the object which received the signal
903 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
904 * the npt-stop position.
906 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
907 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
908 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
909 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
910 G_TYPE_NONE, 0, G_TYPE_NONE);
913 * GstRtpJitterBuffer::clear-pt-map:
914 * @buffer: the object which received the signal
916 * Invalidate the clock-rate as obtained with the
917 * #GstRtpJitterBuffer::request-pt-map signal.
919 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
920 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
921 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
922 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
923 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
926 * GstRtpJitterBuffer::set-active:
927 * @buffer: the object which received the signal
929 * Start pushing out packets with the given base time. This signal is only
930 * useful in buffering mode.
932 * Returns: the time of the last pushed packet.
934 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
935 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
936 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
937 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
938 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
941 gstelement_class->change_state =
942 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
943 gstelement_class->request_new_pad =
944 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
945 gstelement_class->release_pad =
946 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
947 gstelement_class->provide_clock =
948 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
949 gstelement_class->set_clock =
950 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
952 gst_element_class_add_static_pad_template (gstelement_class,
953 &gst_rtp_jitter_buffer_src_template);
954 gst_element_class_add_static_pad_template (gstelement_class,
955 &gst_rtp_jitter_buffer_sink_template);
956 gst_element_class_add_static_pad_template (gstelement_class,
957 &gst_rtp_jitter_buffer_sink_rtcp_template);
959 gst_element_class_set_static_metadata (gstelement_class,
960 "RTP packet jitter-buffer", "Filter/Network/RTP",
961 "A buffer that deals with network jitter and other transmission faults",
962 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
963 "Wim Taymans <wim.taymans@gmail.com>");
965 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
966 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
968 GST_DEBUG_CATEGORY_INIT
969 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
973 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
975 GstRtpJitterBufferPrivate *priv;
977 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
978 jitterbuffer->priv = priv;
980 priv->latency_ms = DEFAULT_LATENCY_MS;
981 priv->latency_ns = priv->latency_ms * GST_MSECOND;
982 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
983 priv->do_lost = DEFAULT_DO_LOST;
984 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
985 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
986 priv->rtx_delay = DEFAULT_RTX_DELAY;
987 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
988 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
989 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
990 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
991 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
992 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
993 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
994 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
995 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
996 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
997 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
998 priv->faststart_min_packets = DEFAULT_FASTSTART_MIN_PACKETS;
1000 priv->last_dts = -1;
1001 priv->last_rtptime = -1;
1002 priv->avg_jitter = 0;
1003 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
1004 priv->rtx_stats_timers = timer_queue_new ();
1005 priv->jbuf = rtp_jitter_buffer_new ();
1006 g_mutex_init (&priv->jbuf_lock);
1007 g_cond_init (&priv->jbuf_timer);
1008 g_cond_init (&priv->jbuf_event);
1009 g_cond_init (&priv->jbuf_query);
1010 g_queue_init (&priv->gap_packets);
1011 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1013 /* reset skew detection initialy */
1014 rtp_jitter_buffer_reset_skew (priv->jbuf);
1015 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
1016 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1017 priv->active = TRUE;
1020 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1023 gst_pad_set_activatemode_function (priv->srcpad,
1024 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1025 gst_pad_set_query_function (priv->srcpad,
1026 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1027 gst_pad_set_event_function (priv->srcpad,
1028 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1031 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1034 gst_pad_set_chain_function (priv->sinkpad,
1035 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1036 gst_pad_set_event_function (priv->sinkpad,
1037 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1038 gst_pad_set_query_function (priv->sinkpad,
1039 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1041 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1042 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1044 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1047 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1049 #define ITEM_TYPE_BUFFER 0
1050 #define ITEM_TYPE_LOST 1
1051 #define ITEM_TYPE_EVENT 2
1052 #define ITEM_TYPE_QUERY 3
1054 static RTPJitterBufferItem *
1055 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1056 guint seqnum, guint count, guint rtptime)
1058 RTPJitterBufferItem *item;
1060 item = g_slice_new (RTPJitterBufferItem);
1067 item->seqnum = seqnum;
1068 item->count = count;
1069 item->rtptime = rtptime;
1075 free_item (RTPJitterBufferItem * item)
1077 g_return_if_fail (item != NULL);
1079 if (item->data && item->type != ITEM_TYPE_QUERY)
1080 gst_mini_object_unref (item->data);
1081 g_slice_free (RTPJitterBufferItem, item);
1085 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1087 GList **l = user_data;
1089 if (item->data && item->type == ITEM_TYPE_EVENT
1090 && GST_EVENT_IS_STICKY (item->data)) {
1091 *l = g_list_prepend (*l, item->data);
1092 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1093 gst_mini_object_unref (item->data);
1095 g_slice_free (RTPJitterBufferItem, item);
1099 gst_rtp_jitter_buffer_finalize (GObject * object)
1101 GstRtpJitterBuffer *jitterbuffer;
1102 GstRtpJitterBufferPrivate *priv;
1104 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1105 priv = jitterbuffer->priv;
1107 g_array_free (priv->timers, TRUE);
1108 timer_queue_free (priv->rtx_stats_timers);
1109 g_mutex_clear (&priv->jbuf_lock);
1110 g_cond_clear (&priv->jbuf_timer);
1111 g_cond_clear (&priv->jbuf_event);
1112 g_cond_clear (&priv->jbuf_query);
1114 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1115 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1116 g_queue_clear (&priv->gap_packets);
1117 g_object_unref (priv->jbuf);
1119 G_OBJECT_CLASS (parent_class)->finalize (object);
1122 static GstIterator *
1123 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1125 GstRtpJitterBuffer *jitterbuffer;
1126 GstPad *otherpad = NULL;
1127 GstIterator *it = NULL;
1128 GValue val = { 0, };
1130 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1132 if (pad == jitterbuffer->priv->sinkpad) {
1133 otherpad = jitterbuffer->priv->srcpad;
1134 } else if (pad == jitterbuffer->priv->srcpad) {
1135 otherpad = jitterbuffer->priv->sinkpad;
1136 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1137 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1141 g_value_init (&val, GST_TYPE_PAD);
1142 g_value_set_object (&val, otherpad);
1143 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1144 g_value_unset (&val);
1151 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1153 GstRtpJitterBufferPrivate *priv;
1155 priv = jitterbuffer->priv;
1157 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1160 gst_pad_new_from_static_template
1161 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1162 gst_pad_set_chain_function (priv->rtcpsinkpad,
1163 gst_rtp_jitter_buffer_chain_rtcp);
1164 gst_pad_set_event_function (priv->rtcpsinkpad,
1165 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1166 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1167 gst_rtp_jitter_buffer_iterate_internal_links);
1168 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1169 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1171 return priv->rtcpsinkpad;
1175 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1177 GstRtpJitterBufferPrivate *priv;
1179 priv = jitterbuffer->priv;
1181 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1183 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1185 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1186 priv->rtcpsinkpad = NULL;
1190 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1191 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1193 GstRtpJitterBuffer *jitterbuffer;
1194 GstElementClass *klass;
1196 GstRtpJitterBufferPrivate *priv;
1198 g_return_val_if_fail (templ != NULL, NULL);
1199 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1201 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1202 priv = jitterbuffer->priv;
1203 klass = GST_ELEMENT_GET_CLASS (element);
1205 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1207 /* figure out the template */
1208 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1209 if (priv->rtcpsinkpad != NULL)
1212 result = create_rtcp_sink (jitterbuffer);
1214 goto wrong_template;
1221 g_warning ("rtpjitterbuffer: this is not our template");
1226 g_warning ("rtpjitterbuffer: pad already requested");
1232 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1234 GstRtpJitterBuffer *jitterbuffer;
1235 GstRtpJitterBufferPrivate *priv;
1237 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1238 g_return_if_fail (GST_IS_PAD (pad));
1240 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1241 priv = jitterbuffer->priv;
1243 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1245 if (priv->rtcpsinkpad == pad) {
1246 remove_rtcp_sink (jitterbuffer);
1255 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1261 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1263 return gst_system_clock_obtain ();
1267 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1269 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1271 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1273 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1277 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1279 GstRtpJitterBufferPrivate *priv;
1281 priv = jitterbuffer->priv;
1283 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1286 priv->clock_rate = -1;
1287 /* do not clear current content, but refresh state for new arrival */
1288 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1289 rtp_jitter_buffer_reset_skew (priv->jbuf);
1294 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1297 GstRtpJitterBufferPrivate *priv;
1298 GstClockTime last_out;
1299 RTPJitterBufferItem *item;
1304 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1305 active, GST_TIME_ARGS (offset));
1307 if (active != priv->active) {
1308 /* add the amount of time spent in paused to the output offset. All
1309 * outgoing buffers will have this offset applied to their timestamps in
1310 * order to make them arrive in time in the sink. */
1311 priv->out_offset = offset;
1312 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1313 GST_TIME_ARGS (priv->out_offset));
1314 priv->active = active;
1315 JBUF_SIGNAL_EVENT (priv);
1318 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1320 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1321 /* head buffer timestamp and offset gives our output time */
1322 last_out = item->pts + priv->ts_offset;
1324 /* use last known time when the buffer is empty */
1325 last_out = priv->last_out_time;
1333 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1335 GstRtpJitterBuffer *jitterbuffer;
1336 GstRtpJitterBufferPrivate *priv;
1341 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1342 priv = jitterbuffer->priv;
1344 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1346 caps = gst_pad_peer_query_caps (other, filter);
1348 templ = gst_pad_get_pad_template_caps (pad);
1350 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1355 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1357 intersect = gst_caps_intersect (caps, templ);
1358 gst_caps_unref (caps);
1359 gst_caps_unref (templ);
1363 gst_object_unref (jitterbuffer);
1369 * Must be called with JBUF_LOCK held
1373 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1374 GstCaps * caps, gint pt)
1376 GstRtpJitterBufferPrivate *priv;
1377 GstStructure *caps_struct;
1381 const gchar *ts_refclk, *mediaclk;
1383 priv = jitterbuffer->priv;
1385 /* first parse the caps */
1386 caps_struct = gst_caps_get_structure (caps, 0);
1388 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1390 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1392 GST_ERROR_OBJECT (jitterbuffer,
1393 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1397 if (payload != -1) {
1398 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1399 priv->last_pt = payload;
1402 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1403 * measure the amount of data in the buffer */
1404 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1407 if (priv->clock_rate <= 0)
1410 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1412 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1414 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1416 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1417 * can use this to track the amount of time elapsed on the sender. */
1418 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1419 priv->clock_base = val;
1421 priv->clock_base = -1;
1423 priv->ext_timestamp = priv->clock_base;
1425 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1428 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1429 /* first expected seqnum, only update when we didn't have a previous base. */
1430 if (priv->next_in_seqnum == -1)
1431 priv->next_in_seqnum = val;
1432 if (priv->next_seqnum == -1) {
1433 priv->next_seqnum = val;
1434 JBUF_SIGNAL_EVENT (priv);
1436 priv->seqnum_base = val;
1438 priv->seqnum_base = -1;
1441 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1443 /* the start and stop times. The seqnum-base corresponds to the start time. We
1444 * will keep track of the seqnums on the output and when we reach the one
1445 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1446 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1447 priv->npt_start = tval;
1449 priv->npt_start = 0;
1451 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1452 priv->npt_stop = tval;
1454 priv->npt_stop = -1;
1456 GST_DEBUG_OBJECT (jitterbuffer,
1457 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1458 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1460 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1461 GstClock *clock = NULL;
1462 guint64 clock_offset = -1;
1464 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1467 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1468 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1469 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1471 const gchar *host, *portstr;
1475 host = ts_refclk + sizeof ("ntp=") - 1;
1476 if (host[0] == '[') {
1478 portstr = strchr (host, ']');
1479 if (portstr && portstr[1] == ':')
1480 portstr = portstr + 1;
1484 portstr = strrchr (host, ':');
1488 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1492 hostname = g_strndup (host, (portstr - host));
1494 hostname = g_strdup (host);
1496 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1499 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1500 const gchar *domainstr =
1501 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1504 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1507 clock = gst_ptp_clock_new (NULL, domain);
1509 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1512 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1513 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1515 if (!g_str_has_prefix (mediaclk, "direct=")
1516 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1517 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1518 if (strstr (mediaclk, "rate=") != NULL) {
1519 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1524 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1526 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1534 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1539 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1545 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1547 GstRtpJitterBufferPrivate *priv;
1549 priv = jitterbuffer->priv;
1552 /* mark ourselves as flushing */
1553 priv->srcresult = GST_FLOW_FLUSHING;
1554 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1555 /* this unblocks any waiting pops on the src pad task */
1556 JBUF_SIGNAL_EVENT (priv);
1557 JBUF_SIGNAL_QUERY (priv, FALSE);
1562 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1564 GstRtpJitterBufferPrivate *priv;
1566 priv = jitterbuffer->priv;
1569 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1570 /* Mark as non flushing */
1571 priv->srcresult = GST_FLOW_OK;
1572 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1573 priv->last_popped_seqnum = -1;
1574 priv->last_out_time = -1;
1575 priv->next_seqnum = -1;
1576 priv->seqnum_base = -1;
1577 priv->ips_rtptime = -1;
1578 priv->ips_pts = GST_CLOCK_TIME_NONE;
1579 priv->packet_spacing = 0;
1580 priv->next_in_seqnum = -1;
1581 priv->clock_rate = -1;
1584 priv->estimated_eos = -1;
1585 priv->last_elapsed = 0;
1586 priv->ext_timestamp = -1;
1587 priv->avg_jitter = 0;
1588 priv->last_dts = -1;
1589 priv->last_rtptime = -1;
1590 priv->last_in_pts = 0;
1591 priv->equidistant = 0;
1592 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1593 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1594 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1595 rtp_jitter_buffer_reset_skew (priv->jbuf);
1596 remove_all_timers (jitterbuffer);
1597 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1598 g_queue_clear (&priv->gap_packets);
1603 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1604 GstPadMode mode, gboolean active)
1607 GstRtpJitterBuffer *jitterbuffer = NULL;
1609 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1612 case GST_PAD_MODE_PUSH:
1614 /* allow data processing */
1615 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1617 /* start pushing out buffers */
1618 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1619 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1620 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1622 /* make sure all data processing stops ASAP */
1623 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1625 /* NOTE this will hardlock if the state change is called from the src pad
1626 * task thread because we will _join() the thread. */
1627 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1628 result = gst_pad_stop_task (pad);
1638 static GstStateChangeReturn
1639 gst_rtp_jitter_buffer_change_state (GstElement * element,
1640 GstStateChange transition)
1642 GstRtpJitterBuffer *jitterbuffer;
1643 GstRtpJitterBufferPrivate *priv;
1644 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1646 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1647 priv = jitterbuffer->priv;
1649 switch (transition) {
1650 case GST_STATE_CHANGE_NULL_TO_READY:
1652 case GST_STATE_CHANGE_READY_TO_PAUSED:
1654 /* reset negotiated values */
1655 priv->clock_rate = -1;
1656 priv->clock_base = -1;
1657 priv->peer_latency = 0;
1659 /* block until we go to PLAYING */
1660 priv->blocked = TRUE;
1661 priv->timer_running = TRUE;
1662 priv->timer_thread =
1663 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1666 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1668 /* unblock to allow streaming in PLAYING */
1669 priv->blocked = FALSE;
1670 JBUF_SIGNAL_EVENT (priv);
1671 JBUF_SIGNAL_TIMER (priv);
1678 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1680 switch (transition) {
1681 case GST_STATE_CHANGE_READY_TO_PAUSED:
1682 /* we are a live element because we sync to the clock, which we can only
1683 * do in the PLAYING state */
1684 if (ret != GST_STATE_CHANGE_FAILURE)
1685 ret = GST_STATE_CHANGE_NO_PREROLL;
1687 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1689 /* block to stop streaming when PAUSED */
1690 priv->blocked = TRUE;
1691 unschedule_current_timer (jitterbuffer);
1693 if (ret != GST_STATE_CHANGE_FAILURE)
1694 ret = GST_STATE_CHANGE_NO_PREROLL;
1696 case GST_STATE_CHANGE_PAUSED_TO_READY:
1698 gst_buffer_replace (&priv->last_sr, NULL);
1699 priv->timer_running = FALSE;
1700 unschedule_current_timer (jitterbuffer);
1701 JBUF_SIGNAL_TIMER (priv);
1702 JBUF_SIGNAL_QUERY (priv, FALSE);
1704 g_thread_join (priv->timer_thread);
1705 priv->timer_thread = NULL;
1707 case GST_STATE_CHANGE_READY_TO_NULL:
1717 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1720 gboolean ret = TRUE;
1721 GstRtpJitterBuffer *jitterbuffer;
1722 GstRtpJitterBufferPrivate *priv;
1724 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1725 priv = jitterbuffer->priv;
1727 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1729 switch (GST_EVENT_TYPE (event)) {
1730 case GST_EVENT_LATENCY:
1732 GstClockTime latency;
1734 gst_event_parse_latency (event, &latency);
1736 GST_DEBUG_OBJECT (jitterbuffer,
1737 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1740 /* adjust the overall buffer delay to the total pipeline latency in
1741 * buffering mode because if downstream consumes too fast (because of
1742 * large latency or queues, we would start rebuffering again. */
1743 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1744 RTP_JITTER_BUFFER_MODE_BUFFER) {
1745 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1749 ret = gst_pad_push_event (priv->sinkpad, event);
1753 ret = gst_pad_push_event (priv->sinkpad, event);
1760 /* handles and stores the event in the jitterbuffer, must be called with
1763 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1765 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1766 RTPJitterBufferItem *item;
1769 switch (GST_EVENT_TYPE (event)) {
1770 case GST_EVENT_CAPS:
1774 gst_event_parse_caps (event, &caps);
1775 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1778 case GST_EVENT_SEGMENT:
1781 gst_event_copy_segment (event, &segment);
1783 /* we need time for now */
1784 if (segment.format != GST_FORMAT_TIME) {
1785 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1786 gst_event_unref (event);
1788 gst_segment_init (&segment, GST_FORMAT_TIME);
1789 event = gst_event_new_segment (&segment);
1792 priv->segment = segment;
1797 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1804 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1805 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1806 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1808 JBUF_SIGNAL_EVENT (priv);
1814 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1817 gboolean ret = TRUE;
1818 GstRtpJitterBuffer *jitterbuffer;
1819 GstRtpJitterBufferPrivate *priv;
1821 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1822 priv = jitterbuffer->priv;
1824 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1826 switch (GST_EVENT_TYPE (event)) {
1827 case GST_EVENT_FLUSH_START:
1828 ret = gst_pad_push_event (priv->srcpad, event);
1829 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1830 /* wait for the loop to go into PAUSED */
1831 gst_pad_pause_task (priv->srcpad);
1833 case GST_EVENT_FLUSH_STOP:
1834 ret = gst_pad_push_event (priv->srcpad, event);
1836 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1837 GST_PAD_MODE_PUSH, TRUE);
1840 if (GST_EVENT_IS_SERIALIZED (event)) {
1841 /* serialized events go in the queue */
1843 if (priv->srcresult != GST_FLOW_OK) {
1844 /* Errors in sticky event pushing are no problem and ignored here
1845 * as they will cause more meaningful errors during data flow.
1846 * For EOS events, that are not followed by data flow, we still
1847 * return FALSE here though.
1849 if (!GST_EVENT_IS_STICKY (event) ||
1850 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1851 goto out_flow_error;
1853 /* refuse more events on EOS */
1856 ret = queue_event (jitterbuffer, event);
1859 /* non-serialized events are forwarded downstream immediately */
1860 ret = gst_pad_push_event (priv->srcpad, event);
1869 GST_DEBUG_OBJECT (jitterbuffer,
1870 "refusing event, we have a downstream flow error: %s",
1871 gst_flow_get_name (priv->srcresult));
1873 gst_event_unref (event);
1878 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1880 gst_event_unref (event);
1886 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1889 gboolean ret = TRUE;
1890 GstRtpJitterBuffer *jitterbuffer;
1892 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1894 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1896 switch (GST_EVENT_TYPE (event)) {
1897 case GST_EVENT_FLUSH_START:
1898 gst_event_unref (event);
1900 case GST_EVENT_FLUSH_STOP:
1901 gst_event_unref (event);
1904 ret = gst_pad_event_default (pad, parent, event);
1912 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1913 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1914 * GST_FLOW_FLUSHING when the element is shutting down. On success
1915 * GST_FLOW_OK is returned.
1917 static GstFlowReturn
1918 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1922 GValue args[2] = { {0}, {0} };
1926 g_value_init (&args[0], GST_TYPE_ELEMENT);
1927 g_value_set_object (&args[0], jitterbuffer);
1928 g_value_init (&args[1], G_TYPE_UINT);
1929 g_value_set_uint (&args[1], pt);
1931 g_value_init (&ret, GST_TYPE_CAPS);
1932 g_value_set_boxed (&ret, NULL);
1934 JBUF_UNLOCK (jitterbuffer->priv);
1935 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1937 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1939 g_value_unset (&args[0]);
1940 g_value_unset (&args[1]);
1941 caps = (GstCaps *) g_value_dup_boxed (&ret);
1942 g_value_unset (&ret);
1946 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1947 gst_caps_unref (caps);
1949 if (G_UNLIKELY (!res))
1957 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1958 return GST_FLOW_ERROR;
1962 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1963 return GST_FLOW_FLUSHING;
1967 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1968 return GST_FLOW_ERROR;
1972 /* call with jbuf lock held */
1974 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1976 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1977 GstMessage *message = NULL;
1982 /* Post a buffering message */
1983 if (priv->last_percent != percent) {
1984 priv->last_percent = percent;
1986 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1987 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1994 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1996 GstRtpJitterBufferPrivate *priv;
1998 priv = jitterbuffer->priv;
2000 if (timestamp == -1)
2003 /* apply the timestamp offset, this is used for inter stream sync */
2004 timestamp += priv->ts_offset;
2005 /* add the offset, this is used when buffering */
2006 timestamp += priv->out_offset;
2012 timer_queue_new (void)
2016 queue = g_slice_new (TimerQueue);
2017 queue->timers = g_queue_new ();
2018 queue->hashtable = g_hash_table_new (NULL, NULL);
2024 timer_queue_free (TimerQueue * queue)
2029 g_hash_table_destroy (queue->hashtable);
2030 g_queue_free_full (queue->timers, g_free);
2031 g_slice_free (TimerQueue, queue);
2035 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2036 GstClockTime timeout, gboolean lost)
2040 copy = g_memdup (timer, sizeof (*timer));
2041 copy->timeout = timeout;
2042 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2045 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2046 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2047 g_queue_push_tail (queue->timers, copy);
2048 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2052 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2056 test = g_queue_peek_head (queue->timers);
2057 while (test && test->timeout < timeout) {
2058 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2059 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2060 GST_TIME_ARGS (timeout));
2061 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2062 g_free (g_queue_pop_head (queue->timers));
2063 test = g_queue_peek_head (queue->timers);
2068 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2070 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2074 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2076 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2077 TimerData *timer = NULL;
2080 len = priv->timers->len;
2081 for (i = 0; i < len; i++) {
2082 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2083 if (test->seqnum == seqnum) {
2092 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2094 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2096 if (priv->clock_id) {
2097 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2098 gst_clock_id_unschedule (priv->clock_id);
2099 priv->clock_id = NULL;
2104 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2106 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2107 GstClockTime test_timeout;
2109 if ((test_timeout = timer->timeout) == -1)
2112 if (timer->type != TIMER_TYPE_EXPECTED) {
2113 /* add our latency and offset to get output times. */
2114 test_timeout = apply_offset (jitterbuffer, test_timeout);
2115 test_timeout += priv->latency_ns;
2117 return test_timeout;
2121 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2123 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2125 if (priv->clock_id) {
2126 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2128 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2129 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2131 if (timeout == -1 || timeout < priv->timer_timeout)
2132 unschedule_current_timer (jitterbuffer);
2137 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2138 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2139 GstClockTime duration)
2141 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2145 GST_DEBUG_OBJECT (jitterbuffer,
2146 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2147 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2148 GST_TIME_ARGS (delay));
2150 len = priv->timers->len;
2151 g_array_set_size (priv->timers, len + 1);
2152 timer = &g_array_index (priv->timers, TimerData, len);
2155 timer->seqnum = seqnum;
2157 timer->timeout = timeout + delay;
2158 timer->duration = duration;
2159 if (type == TIMER_TYPE_EXPECTED) {
2160 timer->rtx_base = timeout;
2161 timer->rtx_delay = delay;
2162 timer->rtx_retry = 0;
2164 timer->rtx_last = GST_CLOCK_TIME_NONE;
2165 timer->num_rtx_retry = 0;
2166 timer->num_rtx_received = 0;
2167 recalculate_timer (jitterbuffer, timer);
2168 JBUF_SIGNAL_TIMER (priv);
2174 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2175 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2177 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2178 gboolean seqchange, timechange;
2180 GstClockTime new_timeout;
2182 oldseq = timer->seqnum;
2183 new_timeout = timeout + delay;
2184 seqchange = oldseq != seqnum;
2185 timechange = timer->timeout != new_timeout;
2187 if (!seqchange && !timechange) {
2188 GST_DEBUG_OBJECT (jitterbuffer,
2189 "No changes in seqnum (%d) and timeout (%" GST_TIME_FORMAT
2190 "), skipping", oldseq, GST_TIME_ARGS (timer->timeout));
2194 GST_DEBUG_OBJECT (jitterbuffer,
2195 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2196 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2197 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (new_timeout));
2199 timer->timeout = new_timeout;
2200 timer->seqnum = seqnum;
2202 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2203 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2204 GST_TIME_ARGS (delay));
2205 timer->rtx_base = timeout;
2206 timer->rtx_delay = delay;
2207 timer->rtx_retry = 0;
2210 timer->num_rtx_retry = 0;
2211 timer->num_rtx_received = 0;
2214 if (priv->clock_id) {
2215 /* we changed the seqnum and there is a timer currently waiting with this
2216 * seqnum, unschedule it */
2217 if (seqchange && priv->timer_seqnum == oldseq)
2218 unschedule_current_timer (jitterbuffer);
2219 /* we changed the time, check if it is earlier than what we are waiting
2220 * for and unschedule if so */
2221 else if (timechange)
2222 recalculate_timer (jitterbuffer, timer);
2227 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2228 guint16 seqnum, GstClockTime timeout)
2232 /* find the seqnum timer */
2233 timer = find_timer (jitterbuffer, seqnum);
2234 if (timer == NULL) {
2235 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2237 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2243 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2245 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2248 if (timer->idx == -1)
2251 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2252 unschedule_current_timer (jitterbuffer);
2255 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2256 g_array_remove_index_fast (priv->timers, idx);
2261 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2263 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2264 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2265 g_array_set_size (priv->timers, 0);
2266 unschedule_current_timer (jitterbuffer);
2269 /* get the extra delay to wait before sending RTX */
2271 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2275 if (priv->rtx_delay == -1) {
2276 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2277 delay = DEFAULT_AUTO_RTX_DELAY;
2279 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2280 * packet spacing is a good margin */
2281 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2284 delay = priv->rtx_delay * GST_MSECOND;
2286 if (priv->rtx_min_delay > 0)
2287 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2292 /* Check if packet with seqnum is already considered definitely lost by being
2293 * part of a "lost timer" for multiple packets */
2295 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2297 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2300 len = priv->timers->len;
2301 for (i = 0; i < len; i++) {
2302 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2303 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2305 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2307 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2308 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2316 /* we just received a packet with seqnum and dts.
2318 * First check for old seqnum that we are still expecting. If the gap with the
2319 * current seqnum is too big, unschedule the timeouts.
2321 * If we have a valid packet spacing estimate we can set a timer for when we
2322 * should receive the next packet.
2323 * If we don't have a valid estimate, we remove any timer we might have
2324 * had for this packet.
2327 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2328 GstClockTime dts, GstClockTime pts, gboolean do_next_seqnum,
2329 gboolean is_rtx, TimerData * timer)
2331 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2333 /* go through all timers and unschedule the ones with a large gap */
2334 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2336 len = priv->timers->len;
2337 for (i = 0; i < len; i++) {
2338 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2341 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2343 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2344 test->type, test->seqnum, seqnum, gap);
2346 if (gap > priv->rtx_delay_reorder) {
2347 /* max gap, we exceeded the max reorder distance and we don't expect the
2348 * missing packet to be this reordered */
2349 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2350 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2355 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2356 && priv->do_retransmission && priv->rtx_next_seqnum;
2358 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2359 if (timer->num_rtx_retry > 0) {
2361 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2362 /* don't try to estimate the next seqnum because this is a retransmitted
2363 * packet and it probably did not arrive with the expected packet
2365 do_next_seqnum = FALSE;
2368 if (!is_rtx || timer->num_rtx_retry > 1) {
2369 /* Store timer in order to record stats when/if the retransmitted
2370 * packet arrives. We should also store timer information if we've
2371 * requested retransmission more than once since we may receive
2372 * several retransmitted packets. For accuracy we should update the
2373 * stats also when the redundant retransmitted packets arrives. */
2374 timer_queue_append (priv->rtx_stats_timers, timer,
2375 pts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2380 if (do_next_seqnum && pts != GST_CLOCK_TIME_NONE) {
2381 GstClockTime expected, delay;
2383 /* calculate expected arrival time of the next seqnum */
2384 expected = pts + priv->packet_spacing;
2386 delay = get_rtx_delay (priv);
2388 /* and update/install timer for next seqnum */
2389 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2390 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2391 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2392 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2393 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2396 timer->type = TIMER_TYPE_EXPECTED;
2397 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2400 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2401 expected, delay, priv->packet_spacing);
2403 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2404 /* if we had a timer, remove it, we don't know when to expect the next
2406 remove_timer (jitterbuffer, timer);
2411 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2414 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2416 /* we need consecutive seqnums with a different
2417 * rtptime to estimate the packet spacing. */
2418 if (priv->ips_rtptime != rtptime) {
2419 /* rtptime changed, check pts diff */
2420 if (priv->ips_pts != -1 && pts != -1 && pts > priv->ips_pts) {
2421 GstClockTime new_packet_spacing = pts - priv->ips_pts;
2422 GstClockTime old_packet_spacing = priv->packet_spacing;
2424 /* Biased towards bigger packet spacings to prevent
2425 * too many unneeded retransmission requests for next
2426 * packets that just arrive a little later than we would
2428 if (old_packet_spacing > new_packet_spacing)
2429 priv->packet_spacing =
2430 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2431 else if (old_packet_spacing > 0)
2432 priv->packet_spacing =
2433 (3 * new_packet_spacing + old_packet_spacing) / 4;
2435 priv->packet_spacing = new_packet_spacing;
2437 GST_DEBUG_OBJECT (jitterbuffer,
2438 "new packet spacing %" GST_TIME_FORMAT
2439 " old packet spacing %" GST_TIME_FORMAT
2440 " combined to %" GST_TIME_FORMAT,
2441 GST_TIME_ARGS (new_packet_spacing),
2442 GST_TIME_ARGS (old_packet_spacing),
2443 GST_TIME_ARGS (priv->packet_spacing));
2445 priv->ips_rtptime = rtptime;
2446 priv->ips_pts = pts;
2451 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2452 guint16 seqnum, GstClockTime pts, gint gap)
2454 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2455 GstClockTime duration, expected_pts, delay;
2457 gboolean equidistant = priv->equidistant > 0;
2459 GST_DEBUG_OBJECT (jitterbuffer,
2460 "pts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2461 GST_TIME_ARGS (pts), GST_TIME_ARGS (priv->last_in_pts));
2463 if (pts == GST_CLOCK_TIME_NONE) {
2464 GST_WARNING_OBJECT (jitterbuffer, "Have no PTS");
2469 GstClockTime total_duration;
2470 /* the total duration spanned by the missing packets */
2471 if (pts >= priv->last_in_pts)
2472 total_duration = pts - priv->last_in_pts;
2476 /* interpolate between the current time and the last time based on
2477 * number of packets we are missing, this is the estimated duration
2478 * for the missing packet based on equidistant packet spacing. */
2479 duration = total_duration / (gap + 1);
2481 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2482 GST_TIME_ARGS (duration));
2484 if (total_duration > priv->latency_ns) {
2485 GstClockTime gap_time;
2489 GstClockTime gap_dur = gap * duration;
2490 if (gap_dur > priv->latency_ns)
2491 gap_time = gap_dur - priv->latency_ns;
2494 lost_packets = gap_time / duration;
2496 gap_time = total_duration - priv->latency_ns;
2500 /* too many lost packets, some of the missing packets are already
2501 * too late and we can generate lost packet events for them. */
2502 GST_INFO_OBJECT (jitterbuffer,
2503 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2504 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2505 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2506 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2507 GST_TIME_ARGS (gap_time));
2509 /* this timer will fire immediately and the lost event will be pushed from
2510 * the timer thread */
2511 if (lost_packets > 0) {
2512 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2513 priv->last_in_pts + duration, 0, gap_time);
2514 expected += lost_packets;
2515 priv->last_in_pts += gap_time;
2519 expected_pts = priv->last_in_pts + duration;
2521 /* If we cannot assume equidistant packet spacing, the only thing we now
2522 * for sure is that the missing packets have expected pts not later than
2523 * the last received pts. */
2530 if (priv->do_retransmission) {
2531 TimerData *timer = find_timer (jitterbuffer, expected);
2533 type = TIMER_TYPE_EXPECTED;
2534 delay = get_rtx_delay (priv);
2536 /* if we had a timer for the first missing packet, update it. */
2537 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2538 GstClockTime timeout = timer->timeout;
2540 timer->duration = duration;
2541 if (timeout > (expected_pts + delay) && timer->num_rtx_retry == 0) {
2542 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_pts,
2546 expected_pts += duration;
2549 type = TIMER_TYPE_LOST;
2552 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2553 add_timer (jitterbuffer, type, expected, 0, expected_pts, delay, duration);
2554 expected_pts += duration;
2560 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2564 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2565 GstRtpJitterBufferPrivate *priv;
2567 priv = jitterbuffer->priv;
2569 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2572 if (priv->last_dts != -1)
2573 dtsdiff = dts - priv->last_dts;
2577 if (priv->last_rtptime != -1)
2578 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2582 /* Guess whether stream currently uses equidistant packet spacing. If we
2583 * often see identical timestamps it means the packets are not
2585 if (rtptime == priv->last_rtptime)
2586 priv->equidistant -= 2;
2588 priv->equidistant += 1;
2589 priv->equidistant = CLAMP (priv->equidistant, -7, 7);
2591 priv->last_dts = dts;
2592 priv->last_rtptime = rtptime;
2596 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2599 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2601 diff = ABS (dtsdiff - rtpdiffns);
2603 /* jitter is stored in nanoseconds */
2604 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2606 GST_LOG_OBJECT (jitterbuffer,
2607 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2608 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2609 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2610 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2617 GST_DEBUG_OBJECT (jitterbuffer,
2618 "no dts or no clock-rate, can't calculate jitter");
2624 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2626 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2627 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2630 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2631 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2632 gst_rtp_buffer_unmap (&rtp_a);
2634 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2635 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2636 gst_rtp_buffer_unmap (&rtp_b);
2638 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2642 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, GstBuffer * buffer,
2643 guint8 pt, guint16 seqnum, gint gap, guint max_dropout, guint max_misorder)
2645 GstRtpJitterBufferPrivate *priv;
2646 guint gap_packets_length;
2647 gboolean reset = FALSE;
2648 gboolean future = gap > 0;
2650 priv = jitterbuffer->priv;
2652 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2654 guint32 prev_gap_seq = -1;
2655 gboolean all_consecutive = TRUE;
2657 g_queue_insert_sorted (&priv->gap_packets, buffer,
2658 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2660 for (l = priv->gap_packets.head; l; l = l->next) {
2661 GstBuffer *gap_buffer = l->data;
2662 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2665 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2667 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2669 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2670 if (prev_gap_seq == -1)
2671 prev_gap_seq = gap_seq;
2672 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2673 all_consecutive = FALSE;
2675 prev_gap_seq = gap_seq;
2677 gst_rtp_buffer_unmap (&gap_rtp);
2678 if (!all_consecutive)
2682 if (all_consecutive && gap_packets_length > 3) {
2683 GST_DEBUG_OBJECT (jitterbuffer,
2684 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2685 (future ? "new" : "old"), gap,
2686 (future ? max_dropout : -max_misorder));
2688 } else if (!all_consecutive) {
2689 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2690 g_queue_clear (&priv->gap_packets);
2691 GST_DEBUG_OBJECT (jitterbuffer,
2692 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2693 (future ? "new" : "old"), gap,
2694 (future ? max_dropout : -max_misorder));
2697 GST_DEBUG_OBJECT (jitterbuffer,
2698 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2699 (future ? "new" : "old"), gap,
2700 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2704 GST_DEBUG_OBJECT (jitterbuffer,
2705 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2706 gap, -max_misorder);
2707 g_queue_push_tail (&priv->gap_packets, buffer);
2715 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2717 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2718 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2721 GstClockTime base_time =
2722 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2723 GstClockTime clock_time = gst_clock_get_time (clock);
2725 if (clock_time > base_time)
2726 running_time = clock_time - base_time;
2730 gst_object_unref (clock);
2733 return running_time;
2736 static GstFlowReturn
2737 gst_rtp_jitter_buffer_reset (GstRtpJitterBuffer * jitterbuffer,
2738 GstPad * pad, GstObject * parent, guint16 seqnum)
2740 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2741 GstFlowReturn ret = GST_FLOW_OK;
2742 GList *events = NULL, *l;
2746 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2747 rtp_jitter_buffer_flush (priv->jbuf,
2748 (GFunc) free_item_and_retain_events, &events);
2749 rtp_jitter_buffer_reset_skew (priv->jbuf);
2750 remove_all_timers (jitterbuffer);
2751 priv->discont = TRUE;
2752 priv->last_popped_seqnum = -1;
2754 if (priv->gap_packets.head) {
2755 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2756 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2758 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2759 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2760 gst_rtp_buffer_unmap (&gap_rtp);
2762 priv->next_seqnum = seqnum;
2765 priv->last_in_pts = -1;
2766 priv->next_in_seqnum = -1;
2768 /* Insert all sticky events again in order, otherwise we would
2769 * potentially loose STREAM_START, CAPS or SEGMENT events
2771 events = g_list_reverse (events);
2772 for (l = events; l; l = l->next) {
2773 RTPJitterBufferItem *item;
2775 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2776 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2778 g_list_free (events);
2780 JBUF_SIGNAL_EVENT (priv);
2782 /* reset spacing estimation when gap */
2783 priv->ips_rtptime = -1;
2784 priv->ips_pts = GST_CLOCK_TIME_NONE;
2786 buffers = g_list_copy (priv->gap_packets.head);
2787 g_queue_clear (&priv->gap_packets);
2789 priv->ips_rtptime = -1;
2790 priv->ips_pts = GST_CLOCK_TIME_NONE;
2791 JBUF_UNLOCK (jitterbuffer->priv);
2793 for (l = buffers; l; l = l->next) {
2794 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2796 if (ret != GST_FLOW_OK) {
2801 for (; l; l = l->next)
2802 gst_buffer_unref (l->data);
2803 g_list_free (buffers);
2809 gst_rtp_jitter_buffer_fast_start (GstRtpJitterBuffer * jitterbuffer)
2811 GstRtpJitterBufferPrivate *priv;
2812 RTPJitterBufferItem *item;
2815 priv = jitterbuffer->priv;
2817 if (priv->faststart_min_packets == 0)
2820 item = rtp_jitter_buffer_peek (priv->jbuf);
2824 timer = find_timer (jitterbuffer, item->seqnum);
2825 if (!timer || timer->type != TIMER_TYPE_DEADLINE)
2828 if (rtp_jitter_buffer_can_fast_start (priv->jbuf,
2829 priv->faststart_min_packets)) {
2830 GST_INFO_OBJECT (jitterbuffer, "We found %i consecutive packet, start now",
2831 priv->faststart_min_packets);
2832 timer->timeout = -1;
2839 static GstFlowReturn
2840 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2843 GstRtpJitterBuffer *jitterbuffer;
2844 GstRtpJitterBufferPrivate *priv;
2846 guint32 expected, rtptime;
2847 GstFlowReturn ret = GST_FLOW_OK;
2848 GstClockTime dts, pts;
2853 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2854 gboolean do_next_seqnum = FALSE;
2855 RTPJitterBufferItem *item;
2856 GstMessage *msg = NULL;
2857 gboolean estimated_dts = FALSE;
2858 gint32 packet_rate, max_dropout, max_misorder;
2859 TimerData *timer = NULL;
2861 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2863 priv = jitterbuffer->priv;
2865 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2866 goto invalid_buffer;
2868 pt = gst_rtp_buffer_get_payload_type (&rtp);
2869 seqnum = gst_rtp_buffer_get_seq (&rtp);
2870 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2871 gst_rtp_buffer_unmap (&rtp);
2873 /* make sure we have PTS and DTS set */
2874 pts = GST_BUFFER_PTS (buffer);
2875 dts = GST_BUFFER_DTS (buffer);
2882 /* If we have no DTS here, i.e. no capture time, get one from the
2883 * clock now to have something to calculate with in the future. */
2884 dts = get_current_running_time (jitterbuffer);
2887 /* Remember that we estimated the DTS if we are running already
2888 * and this is not our first packet (or first packet after a reset).
2889 * If it's the first packet, we somehow must generate a timestamp for
2890 * everything, otherwise we can't calculate any times
2892 estimated_dts = (priv->next_in_seqnum != -1);
2894 /* take the DTS of the buffer. This is the time when the packet was
2895 * received and is used to calculate jitter and clock skew. We will adjust
2896 * this DTS with the smoothed value after processing it in the
2897 * jitterbuffer and assign it as the PTS. */
2898 /* bring to running time */
2899 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2902 GST_DEBUG_OBJECT (jitterbuffer,
2903 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2904 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2905 GST_BUFFER_IS_RETRANSMISSION (buffer));
2907 JBUF_LOCK_CHECK (priv, out_flushing);
2909 if (G_UNLIKELY (priv->last_pt != pt)) {
2912 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2916 /* reset clock-rate so that we get a new one */
2917 priv->clock_rate = -1;
2919 /* Try to get the clock-rate from the caps first if we can. If there are no
2920 * caps we must fire the signal to get the clock-rate. */
2921 if ((caps = gst_pad_get_current_caps (pad))) {
2922 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2923 gst_caps_unref (caps);
2927 if (G_UNLIKELY (priv->clock_rate == -1)) {
2928 /* no clock rate given on the caps, try to get one with the signal */
2929 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2930 pt) == GST_FLOW_FLUSHING)
2933 if (G_UNLIKELY (priv->clock_rate == -1))
2936 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2939 /* don't accept more data on EOS */
2940 if (G_UNLIKELY (priv->eos))
2943 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2944 calculate_jitter (jitterbuffer, dts, rtptime);
2946 if (priv->seqnum_base != -1) {
2949 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2952 GST_DEBUG_OBJECT (jitterbuffer,
2953 "packet seqnum #%d before seqnum-base #%d", seqnum,
2955 gst_buffer_unref (buffer);
2957 } else if (gap > 16384) {
2958 /* From now on don't compare against the seqnum base anymore as
2959 * at some point in the future we will wrap around and also that
2960 * much reordering is very unlikely */
2961 priv->seqnum_base = -1;
2965 expected = priv->next_in_seqnum;
2968 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2970 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2971 priv->max_dropout_time);
2973 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2974 priv->max_misorder_time);
2975 GST_TRACE_OBJECT (jitterbuffer,
2976 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2977 max_dropout, max_misorder);
2979 /* now check against our expected seqnum */
2980 if (G_UNLIKELY (expected == -1)) {
2981 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2983 /* calculate a pts based on rtptime and arrival time (dts) */
2985 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
2986 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
2988 /* we don't know what the next_in_seqnum should be, wait for the last
2989 * possible moment to push this buffer, maybe we get an earlier seqnum
2991 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, pts);
2993 do_next_seqnum = TRUE;
2994 /* take rtptime and pts to calculate packet spacing */
2995 priv->ips_rtptime = rtptime;
2996 priv->ips_pts = pts;
3000 /* now calculate gap */
3001 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
3002 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
3003 expected, seqnum, gap);
3005 if (G_UNLIKELY (gap > 0 && priv->timers->len >= max_dropout)) {
3006 /* If we have timers for more than RTP_MAX_DROPOUT packets
3007 * pending this means that we have a huge gap overall. We can
3008 * reset the jitterbuffer at this point because there's
3009 * just too much data missing to be able to do anything
3010 * sensible with the past data. Just try again from the
3012 GST_WARNING_OBJECT (jitterbuffer, "%d pending timers > %d - resetting",
3013 priv->timers->len, max_dropout);
3014 gst_buffer_unref (buffer);
3015 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3018 /* Special handling of large gaps */
3019 if ((gap != -1 && gap < -max_misorder) || (gap >= max_dropout)) {
3020 gboolean reset = handle_big_gap_buffer (jitterbuffer, buffer, pt, seqnum,
3021 gap, max_dropout, max_misorder);
3023 return gst_rtp_jitter_buffer_reset (jitterbuffer, pad, parent, seqnum);
3025 GST_DEBUG_OBJECT (jitterbuffer,
3026 "Had big gap, waiting for more consecutive packets");
3031 /* We had no huge gap, let's drop all the gap packets */
3032 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
3033 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3034 g_queue_clear (&priv->gap_packets);
3036 /* calculate a pts based on rtptime and arrival time (dts) */
3037 /* If we estimated the DTS, don't consider it in the clock skew calculations */
3039 rtp_jitter_buffer_calculate_pts (priv->jbuf, dts, estimated_dts,
3040 rtptime, gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer)));
3042 if (G_LIKELY (gap == 0)) {
3043 /* packet is expected */
3044 calculate_packet_spacing (jitterbuffer, rtptime, pts);
3045 do_next_seqnum = TRUE;
3050 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
3051 /* fill in the gap with EXPECTED timers */
3052 calculate_expected (jitterbuffer, expected, seqnum, pts, gap);
3053 do_next_seqnum = TRUE;
3055 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
3056 do_next_seqnum = FALSE;
3059 /* reset spacing estimation when gap */
3060 priv->ips_rtptime = -1;
3061 priv->ips_pts = GST_CLOCK_TIME_NONE;
3065 if (do_next_seqnum) {
3066 priv->last_in_pts = pts;
3067 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
3070 timer = find_timer (jitterbuffer, seqnum);
3071 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
3073 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
3075 timer->num_rtx_received++;
3078 /* let's check if this buffer is too late, we can only accept packets with
3079 * bigger seqnum than the one we last pushed. */
3080 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
3083 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
3085 /* priv->last_popped_seqnum >= seqnum, we're too late. */
3086 if (G_UNLIKELY (gap <= 0)) {
3087 if (priv->do_retransmission) {
3088 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3089 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3090 /* Only count the retranmitted packet too late if it has been
3091 * considered lost. If the original packet arrived before the
3092 * retransmitted we just count it as a duplicate. */
3093 if (timer->type != TIMER_TYPE_LOST)
3101 if (already_lost (jitterbuffer, seqnum))
3104 /* let's drop oldest packet if the queue is already full and drop-on-latency
3105 * is set. We can only do this when there actually is a latency. When no
3106 * latency is set, we just pump it in the queue and let the other end push it
3107 * out as fast as possible. */
3108 if (priv->latency_ms && priv->drop_on_latency) {
3110 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3112 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3113 RTPJitterBufferItem *old_item;
3115 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3117 if (IS_DROPABLE (old_item)) {
3118 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3119 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3121 priv->next_seqnum = (old_item->seqnum + old_item->count) & 0xffff;
3122 free_item (old_item);
3124 /* we might have removed some head buffers, signal the pushing thread to
3125 * see if it can push now */
3126 JBUF_SIGNAL_EVENT (priv);
3130 /* If we estimated the DTS, don't consider it in the clock skew calculations
3131 * later. The code above always sets dts to pts or the other way around if
3132 * any of those is valid in the buffer, so we know that if we estimated the
3133 * dts that both are unknown */
3136 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3137 pts, seqnum, 1, rtptime);
3139 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3141 /* now insert the packet into the queue in sorted order. This function returns
3142 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3143 * have a duplicate. */
3144 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item, &head,
3146 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3147 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3151 /* Trigger fast start if needed */
3152 if (gst_rtp_jitter_buffer_fast_start (jitterbuffer))
3156 update_timers (jitterbuffer, seqnum, dts, pts, do_next_seqnum,
3157 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3159 /* we had an unhandled SR, handle it now */
3161 do_handle_sync (jitterbuffer);
3163 if (G_UNLIKELY (head)) {
3164 /* signal addition of new buffer when the _loop is waiting. */
3165 if (G_LIKELY (priv->active))
3166 JBUF_SIGNAL_EVENT (priv);
3168 /* let's unschedule and unblock any waiting buffers. We only want to do this
3169 * when the head buffer changed */
3170 if (G_UNLIKELY (priv->clock_id)) {
3171 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3172 unschedule_current_timer (jitterbuffer);
3176 GST_DEBUG_OBJECT (jitterbuffer,
3177 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3178 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3180 msg = check_buffering_percent (jitterbuffer, percent);
3186 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3193 /* this is not fatal but should be filtered earlier */
3194 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3195 ("Received invalid RTP payload, dropping"));
3196 gst_buffer_unref (buffer);
3201 GST_WARNING_OBJECT (jitterbuffer,
3202 "No clock-rate in caps!, dropping buffer");
3203 gst_buffer_unref (buffer);
3208 ret = priv->srcresult;
3209 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3210 gst_buffer_unref (buffer);
3216 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3217 gst_buffer_unref (buffer);
3222 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3223 " popped, dropping", seqnum, priv->last_popped_seqnum);
3225 gst_buffer_unref (buffer);
3230 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3231 "considered lost", seqnum);
3233 gst_buffer_unref (buffer);
3238 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3240 priv->num_duplicates++;
3246 GST_DEBUG_OBJECT (jitterbuffer,
3247 "Duplicate RTX packet #%d detected, dropping", seqnum);
3248 priv->num_duplicates++;
3249 gst_buffer_unref (buffer);
3255 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3257 guint64 ext_time, elapsed;
3259 GstRtpJitterBufferPrivate *priv;
3261 priv = jitterbuffer->priv;
3262 rtp_time = item->rtptime;
3264 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3265 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3267 ext_time = priv->ext_timestamp;
3268 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3269 if (ext_time < priv->ext_timestamp) {
3270 ext_time = priv->ext_timestamp;
3272 priv->ext_timestamp = ext_time;
3275 if (ext_time > priv->clock_base)
3276 elapsed = ext_time - priv->clock_base;
3280 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3285 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3286 RTPJitterBufferItem * item)
3288 guint64 total, elapsed, left, estimated;
3289 GstClockTime out_time;
3290 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3292 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3293 || priv->clock_base == -1 || priv->clock_rate <= 0)
3296 /* compute the elapsed time */
3297 elapsed = compute_elapsed (jitterbuffer, item);
3299 /* do nothing if elapsed time doesn't increment */
3300 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3303 priv->last_elapsed = elapsed;
3305 /* this is the total time we need to play */
3306 total = priv->npt_stop - priv->npt_start;
3307 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3308 GST_TIME_ARGS (total));
3310 /* this is how much time there is left */
3311 if (total > elapsed)
3312 left = total - elapsed;
3316 /* if we have less time left that the size of the buffer, we will not
3317 * be able to keep it filled, disabled buffering then */
3318 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3319 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3320 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3321 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3324 /* this is the current time as running-time */
3325 out_time = item->pts;
3328 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3330 /* if there is almost nothing left,
3331 * we may never advance enough to end up in the above case */
3332 if (total < GST_SECOND)
3333 estimated = GST_SECOND;
3337 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3338 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3340 if (estimated != -1 && priv->estimated_eos != estimated) {
3341 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3342 priv->estimated_eos = estimated;
3346 /* take a buffer from the queue and push it */
3347 static GstFlowReturn
3348 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3350 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3351 GstFlowReturn result = GST_FLOW_OK;
3352 RTPJitterBufferItem *item;
3353 GstBuffer *outbuf = NULL;
3354 GstEvent *outevent = NULL;
3355 GstQuery *outquery = NULL;
3356 GstClockTime dts, pts;
3358 gboolean do_push = TRUE;
3362 /* when we get here we are ready to pop and push the buffer */
3363 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3367 case ITEM_TYPE_BUFFER:
3369 /* we need to make writable to change the flags and timestamps */
3370 outbuf = gst_buffer_make_writable (item->data);
3372 if (G_UNLIKELY (priv->discont)) {
3373 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3374 * into the jitterbuffer so we can modify now. */
3375 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3376 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3377 priv->discont = FALSE;
3379 if (G_UNLIKELY (priv->ts_discont)) {
3380 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3381 priv->ts_discont = FALSE;
3385 gst_segment_position_from_running_time (&priv->segment,
3386 GST_FORMAT_TIME, item->dts);
3388 gst_segment_position_from_running_time (&priv->segment,
3389 GST_FORMAT_TIME, item->pts);
3391 /* apply timestamp with offset to buffer now */
3392 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3393 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3395 /* update the elapsed time when we need to check against the npt stop time. */
3396 update_estimated_eos (jitterbuffer, item);
3398 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3400 case ITEM_TYPE_LOST:
3401 priv->discont = TRUE;
3405 case ITEM_TYPE_EVENT:
3406 outevent = item->data;
3408 case ITEM_TYPE_QUERY:
3409 outquery = item->data;
3413 /* now we are ready to push the buffer. Save the seqnum and release the lock
3414 * so the other end can push stuff in the queue again. */
3416 priv->last_popped_seqnum = seqnum;
3417 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3419 msg = check_buffering_percent (jitterbuffer, percent);
3426 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3429 case ITEM_TYPE_BUFFER:
3431 GST_DEBUG_OBJECT (jitterbuffer,
3432 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3433 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3434 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3436 result = gst_pad_push (priv->srcpad, outbuf);
3438 JBUF_LOCK_CHECK (priv, out_flushing);
3440 case ITEM_TYPE_LOST:
3441 case ITEM_TYPE_EVENT:
3442 /* We got not enough consecutive packets with a huge gap, we can
3443 * as well just drop them here now on EOS */
3444 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3445 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3446 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3447 g_queue_clear (&priv->gap_packets);
3450 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3451 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3454 gst_pad_push_event (priv->srcpad, outevent);
3456 gst_event_unref (outevent);
3458 result = GST_FLOW_OK;
3460 JBUF_LOCK_CHECK (priv, out_flushing);
3462 case ITEM_TYPE_QUERY:
3466 res = gst_pad_peer_query (priv->srcpad, outquery);
3468 JBUF_LOCK_CHECK (priv, out_flushing);
3469 result = GST_FLOW_OK;
3470 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3471 JBUF_SIGNAL_QUERY (priv, res);
3480 return priv->srcresult;
3484 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3486 /* Peek a buffer and compare the seqnum to the expected seqnum.
3487 * If all is fine, the buffer is pushed.
3488 * If something is wrong, we wait for some event
3490 static GstFlowReturn
3491 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3493 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3494 GstFlowReturn result;
3495 RTPJitterBufferItem *item;
3497 guint32 next_seqnum;
3499 /* only push buffers when PLAYING and active and not buffering */
3500 if (priv->blocked || !priv->active ||
3501 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3502 return GST_FLOW_WAIT;
3505 /* peek a buffer, we're just looking at the sequence number.
3506 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3507 * wait for a timeout or something to change.
3508 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3509 item = rtp_jitter_buffer_peek (priv->jbuf);
3514 /* get the seqnum and the next expected seqnum */
3515 seqnum = item->seqnum;
3517 return pop_and_push_next (jitterbuffer, seqnum);
3520 next_seqnum = priv->next_seqnum;
3522 /* get the gap between this and the previous packet. If we don't know the
3523 * previous packet seqnum assume no gap. */
3524 if (G_UNLIKELY (next_seqnum == -1)) {
3525 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3526 /* we don't know what the next_seqnum should be, the chain function should
3527 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3528 * fires, so wait for that */
3529 result = GST_FLOW_WAIT;
3531 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3533 if (G_LIKELY (gap == 0)) {
3534 /* no missing packet, pop and push */
3535 result = pop_and_push_next (jitterbuffer, seqnum);
3536 } else if (G_UNLIKELY (gap < 0)) {
3537 /* if we have a packet that we already pushed or considered dropped, pop it
3538 * off and get the next packet */
3539 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3540 seqnum, next_seqnum);
3541 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3543 result = GST_FLOW_OK;
3545 /* the chain function has scheduled timers to request retransmission or
3546 * when to consider the packet lost, wait for that */
3547 GST_DEBUG_OBJECT (jitterbuffer,
3548 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3549 next_seqnum, seqnum, gap);
3550 result = GST_FLOW_WAIT;
3558 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3560 return GST_FLOW_EOS;
3562 return GST_FLOW_WAIT;
3568 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3570 GstClockTime rtx_retry_timeout;
3571 GstClockTime rtx_min_retry_timeout;
3573 if (priv->rtx_retry_timeout == -1) {
3574 if (priv->avg_rtx_rtt == 0)
3575 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3577 /* we want to ask for a retransmission after we waited for a
3578 * complete RTT and the additional jitter */
3579 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3581 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3583 /* make sure we don't retry too often. On very low latency networks,
3584 * the RTT and jitter can be very low. */
3585 if (priv->rtx_min_retry_timeout == -1) {
3586 rtx_min_retry_timeout = priv->packet_spacing;
3588 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3590 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3592 return rtx_retry_timeout;
3596 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3597 GstClockTime rtx_retry_timeout)
3599 GstClockTime rtx_retry_period;
3601 if (priv->rtx_retry_period == -1) {
3602 /* we retry up to the configured jitterbuffer size but leaving some
3603 * room for the retransmission to arrive in time */
3604 if (rtx_retry_timeout > priv->latency_ns) {
3605 rtx_retry_period = 0;
3607 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3610 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3612 return rtx_retry_period;
3616 1. For *larger* rtx-rtt, weigh a new measurement as before (1/8th)
3617 2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less (1/16th)
3618 3. For very large measurements (> avg * 2), consider them "outliers"
3619 and count them a lot less (1/48th)
3622 update_avg_rtx_rtt (GstRtpJitterBufferPrivate * priv, GstClockTime rtt)
3626 if (priv->avg_rtx_rtt == 0) {
3627 priv->avg_rtx_rtt = rtt;
3631 if (rtt > 2 * priv->avg_rtx_rtt)
3633 else if (rtt > priv->avg_rtx_rtt)
3638 priv->avg_rtx_rtt = (rtt + (weight - 1) * priv->avg_rtx_rtt) / weight;
3642 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3643 GstClockTime dts, gboolean success)
3645 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3649 /* we scheduled a retry for this packet and now we have it */
3650 priv->num_rtx_success++;
3651 /* all the previous retry attempts failed */
3652 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3654 /* All retries failed or was too late */
3655 priv->num_rtx_failed += timer->num_rtx_retry;
3658 /* number of retries before (hopefully) receiving the packet */
3659 if (priv->avg_rtx_num == 0.0)
3660 priv->avg_rtx_num = timer->num_rtx_retry;
3662 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3664 /* Calculate the delay between retransmission request and receiving this
3665 * packet. We have a valid delay if and only if this packet is a response to
3666 * our last request. If not we don't know if this is a response to an
3667 * earlier request and delay could be way off. For RTT is more important
3668 * with correct values than to update for every packet. */
3669 if (timer->num_rtx_retry == timer->num_rtx_received &&
3670 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3671 delay = dts - timer->rtx_last;
3672 update_avg_rtx_rtt (priv, delay);
3677 GST_LOG_OBJECT (jitterbuffer,
3678 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3679 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3680 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3681 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3682 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3683 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3684 GST_TIME_ARGS (priv->avg_rtx_rtt));
3687 /* the timeout for when we expected a packet expired */
3689 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3692 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3694 guint delay, delay_ms, avg_rtx_rtt_ms;
3695 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3696 guint rtx_deadline_ms;
3697 GstClockTime rtx_retry_period;
3698 GstClockTime rtx_retry_timeout;
3701 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3702 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3704 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3705 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3707 delay = timer->rtx_delay + timer->rtx_retry;
3709 delay_ms = GST_TIME_AS_MSECONDS (delay);
3710 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3711 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3712 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3714 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3716 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3717 gst_structure_new ("GstRTPRetransmissionRequest",
3718 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3719 "running-time", G_TYPE_UINT64, timer->rtx_base,
3720 "delay", G_TYPE_UINT, delay_ms,
3721 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3722 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3723 "period", G_TYPE_UINT, rtx_retry_period_ms,
3724 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3725 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3726 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3727 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3729 priv->num_rtx_requests++;
3730 timer->num_rtx_retry++;
3732 GST_OBJECT_LOCK (jitterbuffer);
3733 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3734 timer->rtx_last = gst_clock_get_time (clock);
3735 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3737 timer->rtx_last = now;
3739 GST_OBJECT_UNLOCK (jitterbuffer);
3741 /* calculate the timeout for the next retransmission attempt */
3742 timer->rtx_retry += rtx_retry_timeout;
3743 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3744 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3745 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3746 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3747 if ((priv->rtx_max_retries != -1
3748 && timer->num_rtx_retry >= priv->rtx_max_retries)
3749 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)
3750 || (timer->rtx_base + rtx_retry_period < now)) {
3751 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3752 /* too many retransmission request, we now convert the timer
3753 * to a lost timer, leave the num_rtx_retry as it is for stats */
3754 timer->type = TIMER_TYPE_LOST;
3755 timer->rtx_delay = 0;
3756 timer->rtx_retry = 0;
3758 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3759 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3762 gst_pad_push_event (priv->sinkpad, event);
3768 /* a packet is lost */
3770 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3773 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3774 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3776 GstEvent *event = NULL;
3777 RTPJitterBufferItem *item;
3779 seqnum = timer->seqnum;
3780 lost_packets = MAX (timer->num, 1);
3781 num_rtx_retry = timer->num_rtx_retry;
3783 /* we had a gap and thus we lost some packets. Create an event for this. */
3784 if (lost_packets > 1)
3785 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3786 seqnum + lost_packets - 1);
3788 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3790 priv->num_lost += lost_packets;
3791 priv->num_rtx_failed += num_rtx_retry;
3793 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3795 /* we now only accept seqnum bigger than this */
3796 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0) {
3797 priv->next_in_seqnum = next_in_seqnum;
3798 priv->last_in_pts = apply_offset (jitterbuffer, timer->timeout);
3801 /* Avoid creating events if we don't need it. Note that we still need to create
3802 * the lost *ITEM* since it will be used to notify the outgoing thread of
3803 * lost items (so that we can set discont flags and such) */
3804 if (priv->do_lost) {
3805 GstClockTime duration, timestamp;
3806 /* create paket lost event */
3807 timestamp = apply_offset (jitterbuffer, timer->timeout);
3808 duration = timer->duration;
3809 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3810 duration = priv->packet_spacing;
3811 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3812 gst_structure_new ("GstRTPPacketLost",
3813 "seqnum", G_TYPE_UINT, (guint) seqnum,
3814 "timestamp", G_TYPE_UINT64, timestamp,
3815 "duration", G_TYPE_UINT64, duration,
3816 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3818 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3819 if (!rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL))
3823 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3824 /* Store info to update stats if the packet arrives too late */
3825 timer_queue_append (priv->rtx_stats_timers, timer,
3826 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3828 remove_timer (jitterbuffer, timer);
3831 JBUF_SIGNAL_EVENT (priv);
3837 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3840 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3842 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3843 remove_timer (jitterbuffer, timer);
3845 /* there was no EOS in the buffer, put one in there now */
3846 queue_event (jitterbuffer, gst_event_new_eos ());
3848 JBUF_SIGNAL_EVENT (priv);
3854 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3857 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3859 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3861 /* timer seqnum might have been obsoleted by caps seqnum-base,
3862 * only mess with current ongoing seqnum if still unknown */
3863 if (priv->next_seqnum == -1)
3864 priv->next_seqnum = timer->seqnum;
3865 remove_timer (jitterbuffer, timer);
3866 JBUF_SIGNAL_EVENT (priv);
3872 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3875 gboolean removed = FALSE;
3877 switch (timer->type) {
3878 case TIMER_TYPE_EXPECTED:
3879 removed = do_expected_timeout (jitterbuffer, timer, now);
3881 case TIMER_TYPE_LOST:
3882 removed = do_lost_timeout (jitterbuffer, timer, now);
3884 case TIMER_TYPE_DEADLINE:
3885 removed = do_deadline_timeout (jitterbuffer, timer, now);
3887 case TIMER_TYPE_EOS:
3888 removed = do_eos_timeout (jitterbuffer, timer, now);
3894 /* called when we need to wait for the next timeout.
3896 * We loop over the array of recorded timeouts and wait for the earliest one.
3897 * When it timed out, do the logic associated with the timer.
3899 * If there are no timers, we wait on a gcond until something new happens.
3902 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3904 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3905 GstClockTime now = 0;
3908 while (priv->timer_running) {
3909 TimerData *timer = NULL;
3910 GstClockTime timer_timeout = -1;
3913 /* If we have a clock, update "now" now with the very
3914 * latest running time we have. If timers are unscheduled below we
3915 * otherwise wouldn't update now (it's only updated when timers
3916 * expire), and also for the very first loop iteration now would
3917 * otherwise always be 0
3919 GST_OBJECT_LOCK (jitterbuffer);
3920 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3922 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3923 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3925 GST_OBJECT_UNLOCK (jitterbuffer);
3927 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3928 GST_TIME_ARGS (now));
3930 /* Clear expired rtx-stats timers */
3931 if (priv->do_retransmission)
3932 timer_queue_clear_until (priv->rtx_stats_timers, now);
3934 /* Iterate "normal" timers */
3935 len = priv->timers->len;
3936 for (i = 0; i < len;) {
3937 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3938 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3939 gboolean save_best = FALSE;
3941 GST_DEBUG_OBJECT (jitterbuffer,
3942 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3943 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3944 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3946 /* Weed out anything too late */
3947 if (test->type == TIMER_TYPE_LOST &&
3948 (test_timeout == -1 || test_timeout <= now)) {
3949 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3950 do_lost_timeout (jitterbuffer, test, now);
3951 if (!priv->timer_running)
3953 /* We don't move the iterator forward since we just removed the current entry,
3954 * but we update the termination condition */
3955 len = priv->timers->len;
3957 /* find the smallest timeout */
3958 if (timer == NULL) {
3960 } else if (timer_timeout == -1) {
3961 /* we already have an immediate timeout, the new timer must be an
3962 * immediate timer with smaller seqnum to become the best */
3963 if (test_timeout == -1
3964 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3965 timer->seqnum) > 0))
3967 } else if (test_timeout == -1) {
3968 /* first immediate timer */
3970 } else if (test_timeout < timer_timeout) {
3973 } else if (test_timeout == timer_timeout
3974 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3975 timer->seqnum) > 0)) {
3976 /* same timer, smaller seqnum */
3981 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3983 timer_timeout = test_timeout;
3988 if (timer && !priv->blocked) {
3990 GstClockTime sync_time;
3993 GstClockTimeDiff clock_jitter;
3995 if (timer_timeout == -1 || timer_timeout <= now) {
3996 /* We have normally removed all lost timers in the loop above */
3997 g_assert (timer->type != TIMER_TYPE_LOST);
3999 do_timeout (jitterbuffer, timer, now);
4000 /* check here, do_timeout could have released the lock */
4001 if (!priv->timer_running)
4006 GST_OBJECT_LOCK (jitterbuffer);
4007 clock = GST_ELEMENT_CLOCK (jitterbuffer);
4009 GST_OBJECT_UNLOCK (jitterbuffer);
4010 /* let's just push if there is no clock */
4011 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
4012 now = timer_timeout;
4016 /* prepare for sync against clock */
4017 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
4018 /* add latency of peer to get input time */
4019 sync_time += priv->peer_latency;
4021 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
4022 " with sync time %" GST_TIME_FORMAT,
4023 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
4025 /* create an entry for the clock */
4026 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
4027 priv->timer_timeout = timer_timeout;
4028 priv->timer_seqnum = timer->seqnum;
4029 GST_OBJECT_UNLOCK (jitterbuffer);
4031 /* release the lock so that the other end can push stuff or unlock */
4034 ret = gst_clock_id_wait (id, &clock_jitter);
4037 if (!priv->timer_running) {
4038 gst_clock_id_unref (id);
4039 priv->clock_id = NULL;
4043 if (ret != GST_CLOCK_UNSCHEDULED) {
4044 now = timer_timeout + MAX (clock_jitter, 0);
4045 GST_DEBUG_OBJECT (jitterbuffer,
4046 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
4047 GST_STIME_ARGS (clock_jitter));
4049 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
4051 /* and free the entry */
4052 gst_clock_id_unref (id);
4053 priv->clock_id = NULL;
4055 /* no timers, wait for activity */
4056 JBUF_WAIT_TIMER (priv);
4061 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
4066 * This funcion implements the main pushing loop on the source pad.
4068 * It first tries to push as many buffers as possible. If there is a seqnum
4069 * mismatch, we wait for the next timeouts.
4072 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
4074 GstRtpJitterBufferPrivate *priv;
4075 GstFlowReturn result = GST_FLOW_OK;
4077 priv = jitterbuffer->priv;
4079 JBUF_LOCK_CHECK (priv, flushing);
4081 result = handle_next_buffer (jitterbuffer);
4082 if (G_LIKELY (result == GST_FLOW_WAIT)) {
4083 /* now wait for the next event */
4084 JBUF_WAIT_EVENT (priv, flushing);
4085 result = GST_FLOW_OK;
4087 } while (result == GST_FLOW_OK);
4088 /* store result for upstream */
4089 priv->srcresult = result;
4090 /* if we get here we need to pause */
4096 result = priv->srcresult;
4103 JBUF_SIGNAL_QUERY (priv, FALSE);
4106 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
4107 gst_flow_get_name (result));
4108 gst_pad_pause_task (priv->srcpad);
4109 if (result == GST_FLOW_EOS) {
4110 event = gst_event_new_eos ();
4111 gst_pad_push_event (priv->srcpad, event);
4117 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
4118 * some sanity checks and then emit the handle-sync signal with the parameters.
4119 * This function must be called with the LOCK */
4121 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4123 GstRtpJitterBufferPrivate *priv;
4124 guint64 base_rtptime, base_time;
4126 guint64 last_rtptime;
4128 guint64 ext_rtptime, diff;
4129 gboolean valid = TRUE, keep = FALSE;
4131 priv = jitterbuffer->priv;
4133 /* get the last values from the jitterbuffer */
4134 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4135 &clock_rate, &last_rtptime);
4137 clock_base = priv->clock_base;
4138 ext_rtptime = priv->ext_rtptime;
4140 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4141 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4142 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4143 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4145 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4146 /* we keep this SR packet for later. When we get a valid RTP packet the
4147 * above values will be set and we can try to use the SR packet */
4148 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4151 /* we can't accept anything that happened before we did the last resync */
4152 if (base_rtptime > ext_rtptime) {
4153 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4156 /* the SR RTP timestamp must be something close to what we last observed
4157 * in the jitterbuffer */
4158 if (ext_rtptime > last_rtptime) {
4159 /* check how far ahead it is to our RTP timestamps */
4160 diff = ext_rtptime - last_rtptime;
4161 /* if bigger than 1 second, we drop it */
4162 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4164 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4165 clock_rate, 1000)) {
4166 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4167 /* should drop this, but some RTSP servers end up with bogus
4168 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4169 * so still trigger rptbin sync but invalidate RTCP data
4170 * (sync might use other methods) */
4173 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4174 G_GUINT64_FORMAT, last_rtptime, diff);
4180 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4184 s = gst_structure_new ("application/x-rtp-sync",
4185 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4186 "base-time", G_TYPE_UINT64, base_time,
4187 "clock-rate", G_TYPE_UINT, clock_rate,
4188 "clock-base", G_TYPE_UINT64, clock_base,
4189 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4190 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4192 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4193 gst_buffer_replace (&priv->last_sr, NULL);
4195 g_signal_emit (jitterbuffer,
4196 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4198 gst_structure_free (s);
4200 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4201 gst_buffer_replace (&priv->last_sr, NULL);
4205 static GstFlowReturn
4206 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4209 GstRtpJitterBuffer *jitterbuffer;
4210 GstRtpJitterBufferPrivate *priv;
4211 GstFlowReturn ret = GST_FLOW_OK;
4213 GstRTCPPacket packet;
4214 guint64 ext_rtptime;
4216 GstRTCPBuffer rtcp = { NULL, };
4218 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4220 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4221 goto invalid_buffer;
4223 priv = jitterbuffer->priv;
4225 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4227 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4230 /* first packet must be SR or RR or else the validate would have failed */
4231 switch (gst_rtcp_packet_get_type (&packet)) {
4232 case GST_RTCP_TYPE_SR:
4233 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4239 gst_rtcp_buffer_unmap (&rtcp);
4241 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4244 /* convert the RTP timestamp to our extended timestamp, using the same offset
4245 * we used in the jitterbuffer */
4246 ext_rtptime = priv->jbuf->ext_rtptime;
4247 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4249 priv->ext_rtptime = ext_rtptime;
4250 gst_buffer_replace (&priv->last_sr, buffer);
4252 do_handle_sync (jitterbuffer);
4256 gst_buffer_unref (buffer);
4262 /* this is not fatal but should be filtered earlier */
4263 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4264 ("Received invalid RTCP payload, dropping"));
4270 /* this is not fatal but should be filtered earlier */
4271 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4272 ("Received empty RTCP payload, dropping"));
4273 gst_rtcp_buffer_unmap (&rtcp);
4279 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4280 gst_rtcp_buffer_unmap (&rtcp);
4287 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4290 gboolean res = FALSE;
4291 GstRtpJitterBuffer *jitterbuffer;
4292 GstRtpJitterBufferPrivate *priv;
4294 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4295 priv = jitterbuffer->priv;
4297 switch (GST_QUERY_TYPE (query)) {
4298 case GST_QUERY_CAPS:
4300 GstCaps *filter, *caps;
4302 gst_query_parse_caps (query, &filter);
4303 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4304 gst_query_set_caps_result (query, caps);
4305 gst_caps_unref (caps);
4310 if (GST_QUERY_IS_SERIALIZED (query)) {
4311 RTPJitterBufferItem *item;
4314 JBUF_LOCK_CHECK (priv, out_flushing);
4315 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4316 RTP_JITTER_BUFFER_MODE_BUFFER) {
4317 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4318 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4319 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
4321 JBUF_SIGNAL_EVENT (priv);
4322 JBUF_WAIT_QUERY (priv, out_flushing);
4323 res = priv->last_query;
4325 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4330 res = gst_pad_query_default (pad, parent, query);
4338 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4346 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4349 GstRtpJitterBuffer *jitterbuffer;
4350 GstRtpJitterBufferPrivate *priv;
4351 gboolean res = FALSE;
4353 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4354 priv = jitterbuffer->priv;
4356 switch (GST_QUERY_TYPE (query)) {
4357 case GST_QUERY_LATENCY:
4359 /* We need to send the query upstream and add the returned latency to our
4361 GstClockTime min_latency, max_latency;
4363 GstClockTime our_latency;
4365 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4366 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4368 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4369 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4370 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4372 /* store this so that we can safely sync on the peer buffers. */
4374 priv->peer_latency = min_latency;
4375 our_latency = priv->latency_ns;
4378 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4379 GST_TIME_ARGS (our_latency));
4381 /* we add some latency but can buffer an infinite amount of time */
4382 min_latency += our_latency;
4385 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4386 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4387 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4389 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4393 case GST_QUERY_POSITION:
4395 GstClockTime start, last_out;
4398 gst_query_parse_position (query, &fmt, NULL);
4399 if (fmt != GST_FORMAT_TIME) {
4400 res = gst_pad_query_default (pad, parent, query);
4405 start = priv->npt_start;
4406 last_out = priv->last_out_time;
4409 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4410 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4411 GST_TIME_ARGS (last_out));
4413 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4414 /* bring 0-based outgoing time to stream time */
4415 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4418 res = gst_pad_query_default (pad, parent, query);
4422 case GST_QUERY_CAPS:
4424 GstCaps *filter, *caps;
4426 gst_query_parse_caps (query, &filter);
4427 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4428 gst_query_set_caps_result (query, caps);
4429 gst_caps_unref (caps);
4434 res = gst_pad_query_default (pad, parent, query);
4442 gst_rtp_jitter_buffer_set_property (GObject * object,
4443 guint prop_id, const GValue * value, GParamSpec * pspec)
4445 GstRtpJitterBuffer *jitterbuffer;
4446 GstRtpJitterBufferPrivate *priv;
4448 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4449 priv = jitterbuffer->priv;
4454 guint new_latency, old_latency;
4456 new_latency = g_value_get_uint (value);
4459 old_latency = priv->latency_ms;
4460 priv->latency_ms = new_latency;
4461 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4462 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4465 /* post message if latency changed, this will inform the parent pipeline
4466 * that a latency reconfiguration is possible/needed. */
4467 if (new_latency != old_latency) {
4468 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4469 GST_TIME_ARGS (new_latency * GST_MSECOND));
4471 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4472 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4476 case PROP_DROP_ON_LATENCY:
4478 priv->drop_on_latency = g_value_get_boolean (value);
4481 case PROP_TS_OFFSET:
4483 priv->ts_offset = g_value_get_int64 (value);
4484 priv->ts_discont = TRUE;
4489 priv->do_lost = g_value_get_boolean (value);
4494 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4497 case PROP_DO_RETRANSMISSION:
4499 priv->do_retransmission = g_value_get_boolean (value);
4502 case PROP_RTX_NEXT_SEQNUM:
4504 priv->rtx_next_seqnum = g_value_get_boolean (value);
4507 case PROP_RTX_DELAY:
4509 priv->rtx_delay = g_value_get_int (value);
4512 case PROP_RTX_MIN_DELAY:
4514 priv->rtx_min_delay = g_value_get_uint (value);
4517 case PROP_RTX_DELAY_REORDER:
4519 priv->rtx_delay_reorder = g_value_get_int (value);
4522 case PROP_RTX_RETRY_TIMEOUT:
4524 priv->rtx_retry_timeout = g_value_get_int (value);
4527 case PROP_RTX_MIN_RETRY_TIMEOUT:
4529 priv->rtx_min_retry_timeout = g_value_get_int (value);
4532 case PROP_RTX_RETRY_PERIOD:
4534 priv->rtx_retry_period = g_value_get_int (value);
4537 case PROP_RTX_MAX_RETRIES:
4539 priv->rtx_max_retries = g_value_get_int (value);
4542 case PROP_RTX_DEADLINE:
4544 priv->rtx_deadline_ms = g_value_get_int (value);
4547 case PROP_RTX_STATS_TIMEOUT:
4549 priv->rtx_stats_timeout = g_value_get_uint (value);
4552 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4554 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4557 case PROP_MAX_DROPOUT_TIME:
4559 priv->max_dropout_time = g_value_get_uint (value);
4562 case PROP_MAX_MISORDER_TIME:
4564 priv->max_misorder_time = g_value_get_uint (value);
4567 case PROP_RFC7273_SYNC:
4569 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4570 g_value_get_boolean (value));
4573 case PROP_FASTSTART_MIN_PACKETS:
4575 priv->faststart_min_packets = g_value_get_uint (value);
4579 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4585 gst_rtp_jitter_buffer_get_property (GObject * object,
4586 guint prop_id, GValue * value, GParamSpec * pspec)
4588 GstRtpJitterBuffer *jitterbuffer;
4589 GstRtpJitterBufferPrivate *priv;
4591 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4592 priv = jitterbuffer->priv;
4597 g_value_set_uint (value, priv->latency_ms);
4600 case PROP_DROP_ON_LATENCY:
4602 g_value_set_boolean (value, priv->drop_on_latency);
4605 case PROP_TS_OFFSET:
4607 g_value_set_int64 (value, priv->ts_offset);
4612 g_value_set_boolean (value, priv->do_lost);
4617 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4625 if (priv->srcresult != GST_FLOW_OK)
4628 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4630 g_value_set_int (value, percent);
4634 case PROP_DO_RETRANSMISSION:
4636 g_value_set_boolean (value, priv->do_retransmission);
4639 case PROP_RTX_NEXT_SEQNUM:
4641 g_value_set_boolean (value, priv->rtx_next_seqnum);
4644 case PROP_RTX_DELAY:
4646 g_value_set_int (value, priv->rtx_delay);
4649 case PROP_RTX_MIN_DELAY:
4651 g_value_set_uint (value, priv->rtx_min_delay);
4654 case PROP_RTX_DELAY_REORDER:
4656 g_value_set_int (value, priv->rtx_delay_reorder);
4659 case PROP_RTX_RETRY_TIMEOUT:
4661 g_value_set_int (value, priv->rtx_retry_timeout);
4664 case PROP_RTX_MIN_RETRY_TIMEOUT:
4666 g_value_set_int (value, priv->rtx_min_retry_timeout);
4669 case PROP_RTX_RETRY_PERIOD:
4671 g_value_set_int (value, priv->rtx_retry_period);
4674 case PROP_RTX_MAX_RETRIES:
4676 g_value_set_int (value, priv->rtx_max_retries);
4679 case PROP_RTX_DEADLINE:
4681 g_value_set_int (value, priv->rtx_deadline_ms);
4684 case PROP_RTX_STATS_TIMEOUT:
4686 g_value_set_uint (value, priv->rtx_stats_timeout);
4690 g_value_take_boxed (value,
4691 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4693 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4695 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4698 case PROP_MAX_DROPOUT_TIME:
4700 g_value_set_uint (value, priv->max_dropout_time);
4703 case PROP_MAX_MISORDER_TIME:
4705 g_value_set_uint (value, priv->max_misorder_time);
4708 case PROP_RFC7273_SYNC:
4710 g_value_set_boolean (value,
4711 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4714 case PROP_FASTSTART_MIN_PACKETS:
4716 g_value_set_uint (value, priv->faststart_min_packets);
4720 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4725 static GstStructure *
4726 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4728 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4732 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4733 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4734 "num-lost", G_TYPE_UINT64, priv->num_lost,
4735 "num-late", G_TYPE_UINT64, priv->num_late,
4736 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4737 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4738 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4739 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4740 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4741 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);