2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
10 * Copyright 2016 Pexip AS
11 * @author: Havard Graff <havard@pexip.com>
12 * @author: Stian Selnes <stian@pexip.com>
14 * This library is free software; you can redistribute it and/or
15 * modify it under the terms of the GNU Library General Public
16 * License as published by the Free Software Foundation; either
17 * version 2 of the License, or (at your option) any later version.
19 * This library is distributed in the hope that it will be useful,
20 * but WITHOUT ANY WARRANTY; without even the implied warranty of
21 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
22 * Library General Public License for more details.
24 * You should have received a copy of the GNU Library General Public
25 * License along with this library; if not, write to the
26 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
27 * Boston, MA 02110-1301, USA.
32 * SECTION:element-rtpjitterbuffer
34 * This element reorders and removes duplicate RTP packets as they are received
35 * from a network source.
37 * The element needs the clock-rate of the RTP payload in order to estimate the
38 * delay. This information is obtained either from the caps on the sink pad or,
39 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
40 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
42 * The rtpjitterbuffer will wait for missing packets up to a configurable time
43 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
44 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
45 * property is set, lost packets will result in a custom serialized downstream
46 * event of name GstRTPPacketLost. The lost packet events are usually used by a
47 * depayloader or other element to create concealment data or some other logic
48 * to gracefully handle the missing packets.
50 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
51 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
54 * The jitterbuffer can also be configured to send early retransmission events
55 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
56 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
57 * sends a custom upstream event named GstRTPRetransmissionRequest when the
58 * packet is considered late. The initial expected packet arrival time is
59 * calculated as follows:
61 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
62 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
63 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
64 * packets with different rtptime.
66 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
67 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
68 * previously scheduled timeout is overwritten.
70 * - If seqnum N arrived, all seqnum older than
71 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
72 * immediately. This is to request fast feedback for abonormally reorder
73 * packets before any of the previous timeouts is triggered.
75 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
76 * event. After the initial timeout expires and the retransmission event is
77 * sent, the timeout is scheduled for
78 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
79 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
80 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
81 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
82 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
83 * retransmission requests are sent and the regular logic is performed to
84 * schedule a lost packet as discussed above.
86 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
89 * This element will automatically be used inside rtpbin.
92 * <title>Example pipelines</title>
94 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
95 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
96 * inserted into the pipeline to smooth out network jitter and to reorder the
97 * out-of-order RTP packets.
108 #include <gst/rtp/gstrtpbuffer.h>
109 #include <gst/net/net.h>
111 #include "gstrtpjitterbuffer.h"
112 #include "rtpjitterbuffer.h"
113 #include "rtpstats.h"
115 #include <gst/glib-compat-private.h>
117 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
118 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
120 /* RTPJitterBuffer signals and args */
123 SIGNAL_REQUEST_PT_MAP,
131 #define DEFAULT_LATENCY_MS 200
132 #define DEFAULT_DROP_ON_LATENCY FALSE
133 #define DEFAULT_TS_OFFSET 0
134 #define DEFAULT_DO_LOST FALSE
135 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
136 #define DEFAULT_PERCENT 0
137 #define DEFAULT_DO_RETRANSMISSION FALSE
138 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
139 #define DEFAULT_RTX_DELAY -1
140 #define DEFAULT_RTX_MIN_DELAY 0
141 #define DEFAULT_RTX_DELAY_REORDER 3
142 #define DEFAULT_RTX_RETRY_TIMEOUT -1
143 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
144 #define DEFAULT_RTX_RETRY_PERIOD -1
145 #define DEFAULT_RTX_MAX_RETRIES -1
146 #define DEFAULT_RTX_DEADLINE -1
147 #define DEFAULT_RTX_STATS_TIMEOUT 1000
148 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
149 #define DEFAULT_MAX_DROPOUT_TIME 60000
150 #define DEFAULT_MAX_MISORDER_TIME 2000
151 #define DEFAULT_RFC7273_SYNC FALSE
153 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
154 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
160 PROP_DROP_ON_LATENCY,
165 PROP_DO_RETRANSMISSION,
166 PROP_RTX_NEXT_SEQNUM,
169 PROP_RTX_DELAY_REORDER,
170 PROP_RTX_RETRY_TIMEOUT,
171 PROP_RTX_MIN_RETRY_TIMEOUT,
172 PROP_RTX_RETRY_PERIOD,
173 PROP_RTX_MAX_RETRIES,
175 PROP_RTX_STATS_TIMEOUT,
177 PROP_MAX_RTCP_RTP_TIME_DIFF,
178 PROP_MAX_DROPOUT_TIME,
179 PROP_MAX_MISORDER_TIME,
183 #define JBUF_LOCK(priv) G_STMT_START { \
184 GST_TRACE("Locking from thread %p", g_thread_self()); \
185 (g_mutex_lock (&(priv)->jbuf_lock)); \
186 GST_TRACE("Locked from thread %p", g_thread_self()); \
189 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
191 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
194 #define JBUF_UNLOCK(priv) G_STMT_START { \
195 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
196 (g_mutex_unlock (&(priv)->jbuf_lock)); \
199 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
200 GST_DEBUG ("waiting timer"); \
201 (priv)->waiting_timer = TRUE; \
202 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
203 (priv)->waiting_timer = FALSE; \
204 GST_DEBUG ("waiting timer done"); \
206 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
207 if (G_UNLIKELY ((priv)->waiting_timer)) { \
208 GST_DEBUG ("signal timer"); \
209 g_cond_signal (&(priv)->jbuf_timer); \
213 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
214 GST_DEBUG ("waiting event"); \
215 (priv)->waiting_event = TRUE; \
216 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
217 (priv)->waiting_event = FALSE; \
218 GST_DEBUG ("waiting event done"); \
219 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
222 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
223 if (G_UNLIKELY ((priv)->waiting_event)) { \
224 GST_DEBUG ("signal event"); \
225 g_cond_signal (&(priv)->jbuf_event); \
229 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
230 GST_DEBUG ("waiting query"); \
231 (priv)->waiting_query = TRUE; \
232 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
233 (priv)->waiting_query = FALSE; \
234 GST_DEBUG ("waiting query done"); \
235 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
238 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
239 (priv)->last_query = res; \
240 if (G_UNLIKELY ((priv)->waiting_query)) { \
241 GST_DEBUG ("signal query"); \
242 g_cond_signal (&(priv)->jbuf_query); \
246 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
247 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
249 typedef struct TimerQueue
252 GHashTable *hashtable;
255 struct _GstRtpJitterBufferPrivate
257 GstPad *sinkpad, *srcpad;
260 RTPJitterBuffer *jbuf;
262 gboolean waiting_timer;
264 gboolean waiting_event;
266 gboolean waiting_query;
274 gboolean timer_running;
275 GThread *timer_thread;
280 gboolean drop_on_latency;
283 gboolean do_retransmission;
284 gboolean rtx_next_seqnum;
287 gint rtx_delay_reorder;
288 gint rtx_retry_timeout;
289 gint rtx_min_retry_timeout;
290 gint rtx_retry_period;
291 gint rtx_max_retries;
292 guint rtx_stats_timeout;
293 gint rtx_deadline_ms;
294 gint max_rtcp_rtp_time_diff;
295 guint32 max_dropout_time;
296 guint32 max_misorder_time;
298 /* the last seqnum we pushed out */
299 guint32 last_popped_seqnum;
300 /* the next expected seqnum we push */
302 /* seqnum-base, if known */
304 /* last output time */
305 GstClockTime last_out_time;
306 /* last valid input timestamp and rtptime pair */
307 GstClockTime ips_dts;
309 GstClockTime packet_spacing;
313 /* the next expected seqnum we receive */
314 GstClockTime last_in_dts;
315 guint32 next_in_seqnum;
318 TimerQueue *rtx_stats_timers;
320 /* start and stop ranges */
321 GstClockTime npt_start;
322 GstClockTime npt_stop;
323 guint64 ext_timestamp;
324 guint64 last_elapsed;
325 guint64 estimated_eos;
332 /* clock rate and rtp timestamp offset */
336 gint64 prev_ts_offset;
338 /* when we are shutting down */
339 GstFlowReturn srcresult;
345 GstClockTime timer_timeout;
346 guint16 timer_seqnum;
347 /* the latency of the upstream peer, we have to take this into account when
348 * synchronizing the buffers. */
349 GstClockTime peer_latency;
353 /* some accounting */
357 guint64 num_duplicates;
358 guint64 num_rtx_requests;
359 guint64 num_rtx_success;
360 guint64 num_rtx_failed;
363 RTPPacketRateCtx packet_rate_ctx;
366 GstClockTime last_dts;
367 guint64 last_rtptime;
368 GstClockTime avg_jitter;
385 GstClockTime timeout;
386 GstClockTime duration;
387 GstClockTime rtx_base;
388 GstClockTime rtx_delay;
389 GstClockTime rtx_retry;
390 GstClockTime rtx_last;
392 guint num_rtx_received;
395 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
396 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
397 GstRtpJitterBufferPrivate))
399 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
400 GST_STATIC_PAD_TEMPLATE ("sink",
403 GST_STATIC_CAPS ("application/x-rtp"
404 /* "clock-rate = (int) [ 1, 2147483647 ], "
405 * "payload = (int) , "
406 * "encoding-name = (string) "
410 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
411 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
414 GST_STATIC_CAPS ("application/x-rtcp")
417 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
418 GST_STATIC_PAD_TEMPLATE ("src",
421 GST_STATIC_CAPS ("application/x-rtp"
422 /* "payload = (int) , "
423 * "clock-rate = (int) , "
424 * "encoding-name = (string) "
428 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
430 #define gst_rtp_jitter_buffer_parent_class parent_class
431 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
433 /* object overrides */
434 static void gst_rtp_jitter_buffer_set_property (GObject * object,
435 guint prop_id, const GValue * value, GParamSpec * pspec);
436 static void gst_rtp_jitter_buffer_get_property (GObject * object,
437 guint prop_id, GValue * value, GParamSpec * pspec);
438 static void gst_rtp_jitter_buffer_finalize (GObject * object);
440 /* element overrides */
441 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
442 * element, GstStateChange transition);
443 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
444 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
445 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
447 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
448 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
452 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
453 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
456 /* sinkpad overrides */
457 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
458 GstObject * parent, GstEvent * event);
459 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
460 GstObject * parent, GstBuffer * buffer);
462 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
463 GstObject * parent, GstEvent * event);
464 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
465 GstObject * parent, GstBuffer * buffer);
467 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
468 GstObject * parent, GstQuery * query);
470 /* srcpad overrides */
471 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
472 GstObject * parent, GstEvent * event);
473 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
474 GstObject * parent, GstPadMode mode, gboolean active);
475 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
476 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
477 GstObject * parent, GstQuery * query);
480 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
482 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
483 gboolean active, guint64 base_time);
484 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
486 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
487 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
489 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
491 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
494 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
495 TimerData * timer, GstClockTime dts, gboolean success);
497 static TimerQueue *timer_queue_new (void);
498 static void timer_queue_free (TimerQueue * queue);
501 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
503 GObjectClass *gobject_class;
504 GstElementClass *gstelement_class;
506 gobject_class = (GObjectClass *) klass;
507 gstelement_class = (GstElementClass *) klass;
509 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
511 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
513 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
514 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
517 * GstRtpJitterBuffer:latency:
519 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
520 * for at most this time.
522 g_object_class_install_property (gobject_class, PROP_LATENCY,
523 g_param_spec_uint ("latency", "Buffer latency in ms",
524 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
525 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
527 * GstRtpJitterBuffer:drop-on-latency:
529 * Drop oldest buffers when the queue is completely filled.
531 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
532 g_param_spec_boolean ("drop-on-latency",
533 "Drop buffers when maximum latency is reached",
534 "Tells the jitterbuffer to never exceed the given latency in size",
535 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
537 * GstRtpJitterBuffer:ts-offset:
539 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
540 * This is mainly used to ensure interstream synchronisation.
542 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
543 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
544 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
545 G_MAXINT64, DEFAULT_TS_OFFSET,
546 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
549 * GstRtpJitterBuffer:do-lost:
551 * Send out a GstRTPPacketLost event downstream when a packet is considered
554 g_object_class_install_property (gobject_class, PROP_DO_LOST,
555 g_param_spec_boolean ("do-lost", "Do Lost",
556 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
557 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
560 * GstRtpJitterBuffer:mode:
562 * Control the buffering and timestamping mode used by the jitterbuffer.
564 g_object_class_install_property (gobject_class, PROP_MODE,
565 g_param_spec_enum ("mode", "Mode",
566 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
567 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
569 * GstRtpJitterBuffer:percent:
571 * The percent of the jitterbuffer that is filled.
573 g_object_class_install_property (gobject_class, PROP_PERCENT,
574 g_param_spec_int ("percent", "percent",
575 "The buffer filled percent", 0, 100,
576 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
578 * GstRtpJitterBuffer:do-retransmission:
580 * Send out a GstRTPRetransmission event upstream when a packet is considered
581 * late and should be retransmitted.
585 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
586 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
587 "Send retransmission events upstream when a packet is late",
588 DEFAULT_DO_RETRANSMISSION,
589 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
592 * GstRtpJitterBuffer:rtx-next-seqnum
594 * Estimate when the next packet should arrive and schedule a retransmission
596 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
597 * for packet N+1. So it will be requested if it does not arrive at the expected time.
598 * The expected time is calculated using the dts of N and the packet spacing.
602 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
603 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
604 "Estimate when the next packet should arrive and schedule a "
605 "retransmission request for it.",
606 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
609 * GstRtpJitterBuffer:rtx-delay:
611 * When a packet did not arrive at the expected time, wait this extra amount
612 * of time before sending a retransmission event.
614 * When -1 is used, the max jitter will be used as extra delay.
618 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
619 g_param_spec_int ("rtx-delay", "RTX Delay",
620 "Extra time in ms to wait before sending retransmission "
621 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
622 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
625 * GstRtpJitterBuffer:rtx-min-delay:
627 * When a packet did not arrive at the expected time, wait at least this extra amount
628 * of time before sending a retransmission event.
632 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
633 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
634 "Minimum time in ms to wait before sending retransmission "
635 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
636 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
638 * GstRtpJitterBuffer:rtx-delay-reorder:
640 * Assume that a retransmission event should be sent when we see
641 * this much packet reordering.
643 * When -1 is used, the value will be estimated based on observed packet
644 * reordering. When 0 is used packet reordering alone will not cause a
645 * retransmission event (Since 1.10).
649 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
650 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
651 "Sending retransmission event when this much reordering "
652 "(0 disable, -1 automatic)",
653 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
654 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
656 * GstRtpJitterBuffer::rtx-retry-timeout:
658 * When no packet has been received after sending a retransmission event
659 * for this time, retry sending a retransmission event.
661 * When -1 is used, the value will be estimated based on observed round
666 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
667 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
668 "Retry sending a transmission event after this timeout in "
669 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
670 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
672 * GstRtpJitterBuffer::rtx-min-retry-timeout:
674 * The minimum amount of time between retry timeouts. When
675 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
676 * minimum interval between retry timeouts.
678 * When -1 is used, the value will be estimated based on the
683 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
684 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
685 "Minimum timeout between sending a transmission event in "
686 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
687 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
689 * GstRtpJitterBuffer:rtx-retry-period:
691 * The amount of time to try to get a retransmission.
693 * When -1 is used, the value will be estimated based on the jitterbuffer
694 * latency and the observed round trip time.
698 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
699 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
700 "Try to get a retransmission for this many ms "
701 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
702 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
704 * GstRtpJitterBuffer:rtx-max-retries:
706 * The maximum number of retries to request a retransmission.
708 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
709 * When -1 is used, the number of retransmission request will not be limited.
713 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
714 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
715 "The maximum number of retries to request a retransmission. "
716 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
717 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
719 * GstRtpJitterBuffer:rtx-deadline:
721 * The deadline for a valid RTX request in ms.
723 * How long the RTX RTCP will be valid for.
724 * When -1 is used, the size of the jitterbuffer will be used.
728 g_object_class_install_property (gobject_class, PROP_RTX_DEADLINE,
729 g_param_spec_int ("rtx-deadline", "RTX Deadline (ms)",
730 "The deadline for a valid RTX request in milliseconds. "
731 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DEADLINE,
732 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
734 * GstRtpJitterBuffer::rtx-stats-timeout:
736 * The time to wait for a retransmitted packet after it has been
737 * considered lost in order to collect RTX statistics.
741 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
742 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
743 "The time to wait for a retransmitted packet after it has been "
744 "considered lost in order to collect statistics (ms)",
745 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
746 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
748 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
749 g_param_spec_uint ("max-dropout-time", "Max dropout time",
750 "The maximum time (milliseconds) of missing packets tolerated.",
751 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
752 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
754 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
755 g_param_spec_uint ("max-misorder-time", "Max misorder time",
756 "The maximum time (milliseconds) of misordered packets tolerated.",
757 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
758 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
760 * GstRtpJitterBuffer:stats:
762 * Various jitterbuffer statistics. This property returns a GstStructure
763 * with name application/x-rtp-jitterbuffer-stats with the following fields:
769 * <classname>"num-pushed"</classname>:
770 * the number of packets pushed out.
776 * <classname>"num-lost"</classname>:
777 * the number of packets considered lost.
783 * <classname>"num-late"</classname>:
784 * the number of packets arriving too late.
790 * <classname>"num-duplicates"</classname>:
791 * the number of duplicate packets.
797 * <classname>"rtx-count"</classname>:
798 * the number of retransmissions requested.
804 * <classname>"rtx-success-count"</classname>:
805 * the number of successful retransmissions.
811 * <classname>"rtx-per-packet"</classname>:
812 * average number of RTX per packet.
818 * <classname>"rtx-rtt"</classname>:
819 * average round trip time per RTX.
826 g_object_class_install_property (gobject_class, PROP_STATS,
827 g_param_spec_boxed ("stats", "Statistics",
828 "Various statistics", GST_TYPE_STRUCTURE,
829 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
832 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
834 * The maximum amount of time in ms that the RTP time in the RTCP SRs
835 * is allowed to be ahead of the last RTP packet we received. Use
836 * -1 to disable ignoring of RTCP packets.
840 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
841 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
842 "Maximum amount of time in ms that the RTP time in RTCP SRs "
843 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
844 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
845 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
847 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
848 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
849 "Synchronize received streams to the RFC7273 clock "
850 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
851 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
854 * GstRtpJitterBuffer::request-pt-map:
855 * @buffer: the object which received the signal
858 * Request the payload type as #GstCaps for @pt.
860 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
861 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
862 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
863 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
864 GST_TYPE_CAPS, 1, G_TYPE_UINT);
866 * GstRtpJitterBuffer::handle-sync:
867 * @buffer: the object which received the signal
868 * @struct: a GstStructure containing sync values.
870 * Be notified of new sync values.
872 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
873 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
874 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
875 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
876 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
879 * GstRtpJitterBuffer::on-npt-stop:
880 * @buffer: the object which received the signal
882 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
883 * the npt-stop position.
885 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
886 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
887 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
888 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
889 G_TYPE_NONE, 0, G_TYPE_NONE);
892 * GstRtpJitterBuffer::clear-pt-map:
893 * @buffer: the object which received the signal
895 * Invalidate the clock-rate as obtained with the
896 * #GstRtpJitterBuffer::request-pt-map signal.
898 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
899 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
900 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
901 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
902 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
905 * GstRtpJitterBuffer::set-active:
906 * @buffer: the object which received the signal
908 * Start pushing out packets with the given base time. This signal is only
909 * useful in buffering mode.
911 * Returns: the time of the last pushed packet.
913 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
914 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
915 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
916 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
917 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
920 gstelement_class->change_state =
921 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
922 gstelement_class->request_new_pad =
923 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
924 gstelement_class->release_pad =
925 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
926 gstelement_class->provide_clock =
927 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
928 gstelement_class->set_clock =
929 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
931 gst_element_class_add_static_pad_template (gstelement_class,
932 &gst_rtp_jitter_buffer_src_template);
933 gst_element_class_add_static_pad_template (gstelement_class,
934 &gst_rtp_jitter_buffer_sink_template);
935 gst_element_class_add_static_pad_template (gstelement_class,
936 &gst_rtp_jitter_buffer_sink_rtcp_template);
938 gst_element_class_set_static_metadata (gstelement_class,
939 "RTP packet jitter-buffer", "Filter/Network/RTP",
940 "A buffer that deals with network jitter and other transmission faults",
941 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
942 "Wim Taymans <wim.taymans@gmail.com>");
944 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
945 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
947 GST_DEBUG_CATEGORY_INIT
948 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
952 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
954 GstRtpJitterBufferPrivate *priv;
956 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
957 jitterbuffer->priv = priv;
959 priv->latency_ms = DEFAULT_LATENCY_MS;
960 priv->latency_ns = priv->latency_ms * GST_MSECOND;
961 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
962 priv->do_lost = DEFAULT_DO_LOST;
963 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
964 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
965 priv->rtx_delay = DEFAULT_RTX_DELAY;
966 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
967 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
968 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
969 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
970 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
971 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
972 priv->rtx_deadline_ms = DEFAULT_RTX_DEADLINE;
973 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
974 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
975 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
976 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
979 priv->last_rtptime = -1;
980 priv->avg_jitter = 0;
981 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
982 priv->rtx_stats_timers = timer_queue_new ();
983 priv->jbuf = rtp_jitter_buffer_new ();
984 g_mutex_init (&priv->jbuf_lock);
985 g_cond_init (&priv->jbuf_timer);
986 g_cond_init (&priv->jbuf_event);
987 g_cond_init (&priv->jbuf_query);
988 g_queue_init (&priv->gap_packets);
989 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
991 /* reset skew detection initialy */
992 rtp_jitter_buffer_reset_skew (priv->jbuf);
993 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
994 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
998 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
1001 gst_pad_set_activatemode_function (priv->srcpad,
1002 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
1003 gst_pad_set_query_function (priv->srcpad,
1004 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
1005 gst_pad_set_event_function (priv->srcpad,
1006 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
1009 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
1012 gst_pad_set_chain_function (priv->sinkpad,
1013 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
1014 gst_pad_set_event_function (priv->sinkpad,
1015 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
1016 gst_pad_set_query_function (priv->sinkpad,
1017 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1019 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1020 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1022 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1025 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1027 #define ITEM_TYPE_BUFFER 0
1028 #define ITEM_TYPE_LOST 1
1029 #define ITEM_TYPE_EVENT 2
1030 #define ITEM_TYPE_QUERY 3
1032 static RTPJitterBufferItem *
1033 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1034 guint seqnum, guint count, guint rtptime)
1036 RTPJitterBufferItem *item;
1038 item = g_slice_new (RTPJitterBufferItem);
1045 item->seqnum = seqnum;
1046 item->count = count;
1047 item->rtptime = rtptime;
1053 free_item (RTPJitterBufferItem * item)
1055 g_return_if_fail (item != NULL);
1057 if (item->data && item->type != ITEM_TYPE_QUERY)
1058 gst_mini_object_unref (item->data);
1059 g_slice_free (RTPJitterBufferItem, item);
1063 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1065 GList **l = user_data;
1067 if (item->data && item->type == ITEM_TYPE_EVENT
1068 && GST_EVENT_IS_STICKY (item->data)) {
1069 *l = g_list_prepend (*l, item->data);
1070 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1071 gst_mini_object_unref (item->data);
1073 g_slice_free (RTPJitterBufferItem, item);
1077 gst_rtp_jitter_buffer_finalize (GObject * object)
1079 GstRtpJitterBuffer *jitterbuffer;
1080 GstRtpJitterBufferPrivate *priv;
1082 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1083 priv = jitterbuffer->priv;
1085 g_array_free (priv->timers, TRUE);
1086 timer_queue_free (priv->rtx_stats_timers);
1087 g_mutex_clear (&priv->jbuf_lock);
1088 g_cond_clear (&priv->jbuf_timer);
1089 g_cond_clear (&priv->jbuf_event);
1090 g_cond_clear (&priv->jbuf_query);
1092 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1093 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1094 g_queue_clear (&priv->gap_packets);
1095 g_object_unref (priv->jbuf);
1097 G_OBJECT_CLASS (parent_class)->finalize (object);
1100 static GstIterator *
1101 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1103 GstRtpJitterBuffer *jitterbuffer;
1104 GstPad *otherpad = NULL;
1105 GstIterator *it = NULL;
1106 GValue val = { 0, };
1108 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1110 if (pad == jitterbuffer->priv->sinkpad) {
1111 otherpad = jitterbuffer->priv->srcpad;
1112 } else if (pad == jitterbuffer->priv->srcpad) {
1113 otherpad = jitterbuffer->priv->sinkpad;
1114 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1115 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1119 g_value_init (&val, GST_TYPE_PAD);
1120 g_value_set_object (&val, otherpad);
1121 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1122 g_value_unset (&val);
1129 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1131 GstRtpJitterBufferPrivate *priv;
1133 priv = jitterbuffer->priv;
1135 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1138 gst_pad_new_from_static_template
1139 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1140 gst_pad_set_chain_function (priv->rtcpsinkpad,
1141 gst_rtp_jitter_buffer_chain_rtcp);
1142 gst_pad_set_event_function (priv->rtcpsinkpad,
1143 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1144 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1145 gst_rtp_jitter_buffer_iterate_internal_links);
1146 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1147 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1149 return priv->rtcpsinkpad;
1153 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1155 GstRtpJitterBufferPrivate *priv;
1157 priv = jitterbuffer->priv;
1159 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1161 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1163 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1164 priv->rtcpsinkpad = NULL;
1168 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1169 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1171 GstRtpJitterBuffer *jitterbuffer;
1172 GstElementClass *klass;
1174 GstRtpJitterBufferPrivate *priv;
1176 g_return_val_if_fail (templ != NULL, NULL);
1177 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1179 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1180 priv = jitterbuffer->priv;
1181 klass = GST_ELEMENT_GET_CLASS (element);
1183 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1185 /* figure out the template */
1186 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1187 if (priv->rtcpsinkpad != NULL)
1190 result = create_rtcp_sink (jitterbuffer);
1192 goto wrong_template;
1199 g_warning ("rtpjitterbuffer: this is not our template");
1204 g_warning ("rtpjitterbuffer: pad already requested");
1210 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1212 GstRtpJitterBuffer *jitterbuffer;
1213 GstRtpJitterBufferPrivate *priv;
1215 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1216 g_return_if_fail (GST_IS_PAD (pad));
1218 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1219 priv = jitterbuffer->priv;
1221 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1223 if (priv->rtcpsinkpad == pad) {
1224 remove_rtcp_sink (jitterbuffer);
1233 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1239 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1241 return gst_system_clock_obtain ();
1245 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1247 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1249 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1251 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1255 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1257 GstRtpJitterBufferPrivate *priv;
1259 priv = jitterbuffer->priv;
1261 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1264 priv->clock_rate = -1;
1265 /* do not clear current content, but refresh state for new arrival */
1266 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1267 rtp_jitter_buffer_reset_skew (priv->jbuf);
1272 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1275 GstRtpJitterBufferPrivate *priv;
1276 GstClockTime last_out;
1277 RTPJitterBufferItem *item;
1282 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1283 active, GST_TIME_ARGS (offset));
1285 if (active != priv->active) {
1286 /* add the amount of time spent in paused to the output offset. All
1287 * outgoing buffers will have this offset applied to their timestamps in
1288 * order to make them arrive in time in the sink. */
1289 priv->out_offset = offset;
1290 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1291 GST_TIME_ARGS (priv->out_offset));
1292 priv->active = active;
1293 JBUF_SIGNAL_EVENT (priv);
1296 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1298 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1299 /* head buffer timestamp and offset gives our output time */
1300 last_out = item->dts + priv->ts_offset;
1302 /* use last known time when the buffer is empty */
1303 last_out = priv->last_out_time;
1311 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1313 GstRtpJitterBuffer *jitterbuffer;
1314 GstRtpJitterBufferPrivate *priv;
1319 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1320 priv = jitterbuffer->priv;
1322 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1324 caps = gst_pad_peer_query_caps (other, filter);
1326 templ = gst_pad_get_pad_template_caps (pad);
1328 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1333 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1335 intersect = gst_caps_intersect (caps, templ);
1336 gst_caps_unref (caps);
1337 gst_caps_unref (templ);
1341 gst_object_unref (jitterbuffer);
1347 * Must be called with JBUF_LOCK held
1351 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1352 GstCaps * caps, gint pt)
1354 GstRtpJitterBufferPrivate *priv;
1355 GstStructure *caps_struct;
1359 const gchar *ts_refclk, *mediaclk;
1361 priv = jitterbuffer->priv;
1363 /* first parse the caps */
1364 caps_struct = gst_caps_get_structure (caps, 0);
1366 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1368 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1370 GST_ERROR_OBJECT (jitterbuffer,
1371 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1375 if (payload != -1) {
1376 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1377 priv->last_pt = payload;
1380 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1381 * measure the amount of data in the buffer */
1382 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1385 if (priv->clock_rate <= 0)
1388 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1390 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1392 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1394 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1395 * can use this to track the amount of time elapsed on the sender. */
1396 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1397 priv->clock_base = val;
1399 priv->clock_base = -1;
1401 priv->ext_timestamp = priv->clock_base;
1403 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1406 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1407 /* first expected seqnum, only update when we didn't have a previous base. */
1408 if (priv->next_in_seqnum == -1)
1409 priv->next_in_seqnum = val;
1410 if (priv->next_seqnum == -1) {
1411 priv->next_seqnum = val;
1412 JBUF_SIGNAL_EVENT (priv);
1414 priv->seqnum_base = val;
1416 priv->seqnum_base = -1;
1419 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1421 /* the start and stop times. The seqnum-base corresponds to the start time. We
1422 * will keep track of the seqnums on the output and when we reach the one
1423 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1424 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1425 priv->npt_start = tval;
1427 priv->npt_start = 0;
1429 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1430 priv->npt_stop = tval;
1432 priv->npt_stop = -1;
1434 GST_DEBUG_OBJECT (jitterbuffer,
1435 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1436 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1438 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1439 GstClock *clock = NULL;
1440 guint64 clock_offset = -1;
1442 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1445 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1446 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1447 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1449 const gchar *host, *portstr;
1453 host = ts_refclk + sizeof ("ntp=") - 1;
1454 if (host[0] == '[') {
1456 portstr = strchr (host, ']');
1457 if (portstr && portstr[1] == ':')
1458 portstr = portstr + 1;
1462 portstr = strrchr (host, ':');
1466 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1470 hostname = g_strndup (host, (portstr - host));
1472 hostname = g_strdup (host);
1474 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1477 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1478 const gchar *domainstr =
1479 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1482 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1485 clock = gst_ptp_clock_new (NULL, domain);
1487 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1490 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1491 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1493 if (!g_str_has_prefix (mediaclk, "direct=")
1494 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1495 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1496 if (strstr (mediaclk, "rate=") != NULL) {
1497 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1502 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1504 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1512 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1517 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1523 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1525 GstRtpJitterBufferPrivate *priv;
1527 priv = jitterbuffer->priv;
1530 /* mark ourselves as flushing */
1531 priv->srcresult = GST_FLOW_FLUSHING;
1532 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1533 /* this unblocks any waiting pops on the src pad task */
1534 JBUF_SIGNAL_EVENT (priv);
1535 JBUF_SIGNAL_QUERY (priv, FALSE);
1540 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1542 GstRtpJitterBufferPrivate *priv;
1544 priv = jitterbuffer->priv;
1547 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1548 /* Mark as non flushing */
1549 priv->srcresult = GST_FLOW_OK;
1550 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1551 priv->last_popped_seqnum = -1;
1552 priv->last_out_time = -1;
1553 priv->next_seqnum = -1;
1554 priv->seqnum_base = -1;
1555 priv->ips_rtptime = -1;
1556 priv->ips_dts = GST_CLOCK_TIME_NONE;
1557 priv->packet_spacing = 0;
1558 priv->next_in_seqnum = -1;
1559 priv->clock_rate = -1;
1562 priv->estimated_eos = -1;
1563 priv->last_elapsed = 0;
1564 priv->ext_timestamp = -1;
1565 priv->avg_jitter = 0;
1566 priv->last_dts = -1;
1567 priv->last_rtptime = -1;
1568 priv->last_in_dts = 0;
1569 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1570 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1571 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1572 rtp_jitter_buffer_reset_skew (priv->jbuf);
1573 remove_all_timers (jitterbuffer);
1574 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1575 g_queue_clear (&priv->gap_packets);
1580 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1581 GstPadMode mode, gboolean active)
1584 GstRtpJitterBuffer *jitterbuffer = NULL;
1586 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1589 case GST_PAD_MODE_PUSH:
1591 /* allow data processing */
1592 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1594 /* start pushing out buffers */
1595 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1596 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1597 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1599 /* make sure all data processing stops ASAP */
1600 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1602 /* NOTE this will hardlock if the state change is called from the src pad
1603 * task thread because we will _join() the thread. */
1604 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1605 result = gst_pad_stop_task (pad);
1615 static GstStateChangeReturn
1616 gst_rtp_jitter_buffer_change_state (GstElement * element,
1617 GstStateChange transition)
1619 GstRtpJitterBuffer *jitterbuffer;
1620 GstRtpJitterBufferPrivate *priv;
1621 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1623 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1624 priv = jitterbuffer->priv;
1626 switch (transition) {
1627 case GST_STATE_CHANGE_NULL_TO_READY:
1629 case GST_STATE_CHANGE_READY_TO_PAUSED:
1631 /* reset negotiated values */
1632 priv->clock_rate = -1;
1633 priv->clock_base = -1;
1634 priv->peer_latency = 0;
1636 /* block until we go to PLAYING */
1637 priv->blocked = TRUE;
1638 priv->timer_running = TRUE;
1639 priv->timer_thread =
1640 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1643 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1645 /* unblock to allow streaming in PLAYING */
1646 priv->blocked = FALSE;
1647 JBUF_SIGNAL_EVENT (priv);
1648 JBUF_SIGNAL_TIMER (priv);
1655 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1657 switch (transition) {
1658 case GST_STATE_CHANGE_READY_TO_PAUSED:
1659 /* we are a live element because we sync to the clock, which we can only
1660 * do in the PLAYING state */
1661 if (ret != GST_STATE_CHANGE_FAILURE)
1662 ret = GST_STATE_CHANGE_NO_PREROLL;
1664 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1666 /* block to stop streaming when PAUSED */
1667 priv->blocked = TRUE;
1668 unschedule_current_timer (jitterbuffer);
1670 if (ret != GST_STATE_CHANGE_FAILURE)
1671 ret = GST_STATE_CHANGE_NO_PREROLL;
1673 case GST_STATE_CHANGE_PAUSED_TO_READY:
1675 gst_buffer_replace (&priv->last_sr, NULL);
1676 priv->timer_running = FALSE;
1677 unschedule_current_timer (jitterbuffer);
1678 JBUF_SIGNAL_TIMER (priv);
1679 JBUF_SIGNAL_QUERY (priv, FALSE);
1681 g_thread_join (priv->timer_thread);
1682 priv->timer_thread = NULL;
1684 case GST_STATE_CHANGE_READY_TO_NULL:
1694 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1697 gboolean ret = TRUE;
1698 GstRtpJitterBuffer *jitterbuffer;
1699 GstRtpJitterBufferPrivate *priv;
1701 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1702 priv = jitterbuffer->priv;
1704 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1706 switch (GST_EVENT_TYPE (event)) {
1707 case GST_EVENT_LATENCY:
1709 GstClockTime latency;
1711 gst_event_parse_latency (event, &latency);
1713 GST_DEBUG_OBJECT (jitterbuffer,
1714 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1717 /* adjust the overall buffer delay to the total pipeline latency in
1718 * buffering mode because if downstream consumes too fast (because of
1719 * large latency or queues, we would start rebuffering again. */
1720 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1721 RTP_JITTER_BUFFER_MODE_BUFFER) {
1722 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1726 ret = gst_pad_push_event (priv->sinkpad, event);
1730 ret = gst_pad_push_event (priv->sinkpad, event);
1737 /* handles and stores the event in the jitterbuffer, must be called with
1740 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1742 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1743 RTPJitterBufferItem *item;
1746 switch (GST_EVENT_TYPE (event)) {
1747 case GST_EVENT_CAPS:
1751 gst_event_parse_caps (event, &caps);
1752 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1755 case GST_EVENT_SEGMENT:
1758 gst_event_copy_segment (event, &segment);
1760 /* we need time for now */
1761 if (segment.format != GST_FORMAT_TIME) {
1762 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1763 gst_event_unref (event);
1765 gst_segment_init (&segment, GST_FORMAT_TIME);
1766 event = gst_event_new_segment (&segment);
1769 priv->segment = segment;
1774 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1781 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1782 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1783 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1785 JBUF_SIGNAL_EVENT (priv);
1791 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1794 gboolean ret = TRUE;
1795 GstRtpJitterBuffer *jitterbuffer;
1796 GstRtpJitterBufferPrivate *priv;
1798 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1799 priv = jitterbuffer->priv;
1801 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1803 switch (GST_EVENT_TYPE (event)) {
1804 case GST_EVENT_FLUSH_START:
1805 ret = gst_pad_push_event (priv->srcpad, event);
1806 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1807 /* wait for the loop to go into PAUSED */
1808 gst_pad_pause_task (priv->srcpad);
1810 case GST_EVENT_FLUSH_STOP:
1811 ret = gst_pad_push_event (priv->srcpad, event);
1813 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1814 GST_PAD_MODE_PUSH, TRUE);
1817 if (GST_EVENT_IS_SERIALIZED (event)) {
1818 /* serialized events go in the queue */
1820 if (priv->srcresult != GST_FLOW_OK) {
1821 /* Errors in sticky event pushing are no problem and ignored here
1822 * as they will cause more meaningful errors during data flow.
1823 * For EOS events, that are not followed by data flow, we still
1824 * return FALSE here though.
1826 if (!GST_EVENT_IS_STICKY (event) ||
1827 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1828 goto out_flow_error;
1830 /* refuse more events on EOS */
1833 ret = queue_event (jitterbuffer, event);
1836 /* non-serialized events are forwarded downstream immediately */
1837 ret = gst_pad_push_event (priv->srcpad, event);
1846 GST_DEBUG_OBJECT (jitterbuffer,
1847 "refusing event, we have a downstream flow error: %s",
1848 gst_flow_get_name (priv->srcresult));
1850 gst_event_unref (event);
1855 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1857 gst_event_unref (event);
1863 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1866 gboolean ret = TRUE;
1867 GstRtpJitterBuffer *jitterbuffer;
1869 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1871 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1873 switch (GST_EVENT_TYPE (event)) {
1874 case GST_EVENT_FLUSH_START:
1875 gst_event_unref (event);
1877 case GST_EVENT_FLUSH_STOP:
1878 gst_event_unref (event);
1881 ret = gst_pad_event_default (pad, parent, event);
1889 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1890 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1891 * GST_FLOW_FLUSHING when the element is shutting down. On success
1892 * GST_FLOW_OK is returned.
1894 static GstFlowReturn
1895 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1899 GValue args[2] = { {0}, {0} };
1903 g_value_init (&args[0], GST_TYPE_ELEMENT);
1904 g_value_set_object (&args[0], jitterbuffer);
1905 g_value_init (&args[1], G_TYPE_UINT);
1906 g_value_set_uint (&args[1], pt);
1908 g_value_init (&ret, GST_TYPE_CAPS);
1909 g_value_set_boxed (&ret, NULL);
1911 JBUF_UNLOCK (jitterbuffer->priv);
1912 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1914 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1916 g_value_unset (&args[0]);
1917 g_value_unset (&args[1]);
1918 caps = (GstCaps *) g_value_dup_boxed (&ret);
1919 g_value_unset (&ret);
1923 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1924 gst_caps_unref (caps);
1926 if (G_UNLIKELY (!res))
1934 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1935 return GST_FLOW_ERROR;
1939 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1940 return GST_FLOW_FLUSHING;
1944 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1945 return GST_FLOW_ERROR;
1949 /* call with jbuf lock held */
1951 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1953 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1954 GstMessage *message = NULL;
1959 /* Post a buffering message */
1960 if (priv->last_percent != percent) {
1961 priv->last_percent = percent;
1963 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1964 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1971 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1973 GstRtpJitterBufferPrivate *priv;
1975 priv = jitterbuffer->priv;
1977 if (timestamp == -1)
1980 /* apply the timestamp offset, this is used for inter stream sync */
1981 timestamp += priv->ts_offset;
1982 /* add the offset, this is used when buffering */
1983 timestamp += priv->out_offset;
1989 timer_queue_new (void)
1993 queue = g_slice_new (TimerQueue);
1994 queue->timers = g_queue_new ();
1995 queue->hashtable = g_hash_table_new (NULL, NULL);
2001 timer_queue_free (TimerQueue * queue)
2006 g_hash_table_destroy (queue->hashtable);
2007 g_queue_free_full (queue->timers, g_free);
2008 g_slice_free (TimerQueue, queue);
2012 timer_queue_append (TimerQueue * queue, const TimerData * timer,
2013 GstClockTime timeout, gboolean lost)
2017 copy = g_memdup (timer, sizeof (*timer));
2018 copy->timeout = timeout;
2019 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2022 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2023 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2024 g_queue_push_tail (queue->timers, copy);
2025 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2029 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2033 test = g_queue_peek_head (queue->timers);
2034 while (test && test->timeout < timeout) {
2035 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2036 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2037 GST_TIME_ARGS (timeout));
2038 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2039 g_free (g_queue_pop_head (queue->timers));
2040 test = g_queue_peek_head (queue->timers);
2045 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2047 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2051 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2053 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2054 TimerData *timer = NULL;
2057 len = priv->timers->len;
2058 for (i = 0; i < len; i++) {
2059 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2060 if (test->seqnum == seqnum) {
2069 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2071 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2073 if (priv->clock_id) {
2074 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2075 gst_clock_id_unschedule (priv->clock_id);
2076 priv->clock_id = NULL;
2081 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2083 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2084 GstClockTime test_timeout;
2086 if ((test_timeout = timer->timeout) == -1)
2089 if (timer->type != TIMER_TYPE_EXPECTED) {
2090 /* add our latency and offset to get output times. */
2091 test_timeout = apply_offset (jitterbuffer, test_timeout);
2092 test_timeout += priv->latency_ns;
2094 return test_timeout;
2098 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2100 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2102 if (priv->clock_id) {
2103 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2105 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2106 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2108 if (timeout == -1 || timeout < priv->timer_timeout)
2109 unschedule_current_timer (jitterbuffer);
2114 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2115 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2116 GstClockTime duration)
2118 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2122 GST_DEBUG_OBJECT (jitterbuffer,
2123 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2124 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2125 GST_TIME_ARGS (delay));
2127 len = priv->timers->len;
2128 g_array_set_size (priv->timers, len + 1);
2129 timer = &g_array_index (priv->timers, TimerData, len);
2132 timer->seqnum = seqnum;
2134 timer->timeout = timeout + delay;
2135 timer->duration = duration;
2136 if (type == TIMER_TYPE_EXPECTED) {
2137 timer->rtx_base = timeout;
2138 timer->rtx_delay = delay;
2139 timer->rtx_retry = 0;
2141 timer->rtx_last = GST_CLOCK_TIME_NONE;
2142 timer->num_rtx_retry = 0;
2143 timer->num_rtx_received = 0;
2144 recalculate_timer (jitterbuffer, timer);
2145 JBUF_SIGNAL_TIMER (priv);
2151 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2152 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2154 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2155 gboolean seqchange, timechange;
2158 seqchange = timer->seqnum != seqnum;
2159 timechange = timer->timeout != timeout;
2161 if (!seqchange && !timechange)
2164 oldseq = timer->seqnum;
2166 GST_DEBUG_OBJECT (jitterbuffer,
2167 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2168 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2169 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (timeout + delay));
2171 timer->timeout = timeout + delay;
2172 timer->seqnum = seqnum;
2174 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2175 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2176 GST_TIME_ARGS (delay));
2177 timer->rtx_base = timeout;
2178 timer->rtx_delay = delay;
2179 timer->rtx_retry = 0;
2182 timer->num_rtx_retry = 0;
2183 timer->num_rtx_received = 0;
2186 if (priv->clock_id) {
2187 /* we changed the seqnum and there is a timer currently waiting with this
2188 * seqnum, unschedule it */
2189 if (seqchange && priv->timer_seqnum == oldseq)
2190 unschedule_current_timer (jitterbuffer);
2191 /* we changed the time, check if it is earlier than what we are waiting
2192 * for and unschedule if so */
2193 else if (timechange)
2194 recalculate_timer (jitterbuffer, timer);
2199 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2200 guint16 seqnum, GstClockTime timeout)
2204 /* find the seqnum timer */
2205 timer = find_timer (jitterbuffer, seqnum);
2206 if (timer == NULL) {
2207 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2209 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2215 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2217 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2220 if (timer->idx == -1)
2223 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2224 unschedule_current_timer (jitterbuffer);
2227 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2228 g_array_remove_index_fast (priv->timers, idx);
2233 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2235 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2236 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2237 g_array_set_size (priv->timers, 0);
2238 unschedule_current_timer (jitterbuffer);
2241 /* get the extra delay to wait before sending RTX */
2243 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2247 if (priv->rtx_delay == -1) {
2248 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2249 delay = DEFAULT_AUTO_RTX_DELAY;
2251 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2252 * packet spacing is a good margin */
2253 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2256 delay = priv->rtx_delay * GST_MSECOND;
2258 if (priv->rtx_min_delay > 0)
2259 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2264 /* Check if packet with seqnum is already considered definitely lost by being
2265 * part of a "lost timer" for multiple packets */
2267 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2269 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2272 len = priv->timers->len;
2273 for (i = 0; i < len; i++) {
2274 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2275 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2277 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2279 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2280 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2288 /* we just received a packet with seqnum and dts.
2290 * First check for old seqnum that we are still expecting. If the gap with the
2291 * current seqnum is too big, unschedule the timeouts.
2293 * If we have a valid packet spacing estimate we can set a timer for when we
2294 * should receive the next packet.
2295 * If we don't have a valid estimate, we remove any timer we might have
2296 * had for this packet.
2299 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2300 GstClockTime dts, gboolean do_next_seqnum, gboolean is_rtx,
2303 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2305 /* go through all timers and unschedule the ones with a large gap */
2306 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2308 len = priv->timers->len;
2309 for (i = 0; i < len; i++) {
2310 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2313 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2315 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2316 test->type, test->seqnum, seqnum, gap);
2318 if (gap > priv->rtx_delay_reorder) {
2319 /* max gap, we exceeded the max reorder distance and we don't expect the
2320 * missing packet to be this reordered */
2321 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2322 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2327 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2328 && priv->do_retransmission && priv->rtx_next_seqnum;
2330 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2331 if (timer->num_rtx_retry > 0) {
2333 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2334 /* don't try to estimate the next seqnum because this is a retransmitted
2335 * packet and it probably did not arrive with the expected packet
2337 do_next_seqnum = FALSE;
2340 if (!is_rtx || timer->num_rtx_retry > 1) {
2341 /* Store timer in order to record stats when/if the retransmitted
2342 * packet arrives. We should also store timer information if we've
2343 * requested retransmission more than once since we may receive
2344 * several retransmitted packets. For accuracy we should update the
2345 * stats also when the redundant retransmitted packets arrives. */
2346 timer_queue_append (priv->rtx_stats_timers, timer,
2347 dts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2352 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2353 GstClockTime expected, delay;
2355 /* calculate expected arrival time of the next seqnum */
2356 expected = dts + priv->packet_spacing;
2358 delay = get_rtx_delay (priv);
2360 /* and update/install timer for next seqnum */
2361 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2362 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2363 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2364 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2365 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2368 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2371 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2372 expected, delay, priv->packet_spacing);
2374 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2375 /* if we had a timer, remove it, we don't know when to expect the next
2377 remove_timer (jitterbuffer, timer);
2382 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2385 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2387 /* we need consecutive seqnums with a different
2388 * rtptime to estimate the packet spacing. */
2389 if (priv->ips_rtptime != rtptime) {
2390 /* rtptime changed, check dts diff */
2391 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2392 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2393 GstClockTime old_packet_spacing = priv->packet_spacing;
2395 /* Biased towards bigger packet spacings to prevent
2396 * too many unneeded retransmission requests for next
2397 * packets that just arrive a little later than we would
2399 if (old_packet_spacing > new_packet_spacing)
2400 priv->packet_spacing =
2401 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2402 else if (old_packet_spacing > 0)
2403 priv->packet_spacing =
2404 (3 * new_packet_spacing + old_packet_spacing) / 4;
2406 priv->packet_spacing = new_packet_spacing;
2408 GST_DEBUG_OBJECT (jitterbuffer,
2409 "new packet spacing %" GST_TIME_FORMAT
2410 " old packet spacing %" GST_TIME_FORMAT
2411 " combined to %" GST_TIME_FORMAT,
2412 GST_TIME_ARGS (new_packet_spacing),
2413 GST_TIME_ARGS (old_packet_spacing),
2414 GST_TIME_ARGS (priv->packet_spacing));
2416 priv->ips_rtptime = rtptime;
2417 priv->ips_dts = dts;
2422 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2423 guint16 seqnum, GstClockTime dts, gint gap)
2425 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2426 GstClockTime total_duration, duration, expected_dts, delay;
2429 GST_DEBUG_OBJECT (jitterbuffer,
2430 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2431 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2433 if (dts == GST_CLOCK_TIME_NONE) {
2434 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2438 /* the total duration spanned by the missing packets */
2439 if (dts >= priv->last_in_dts)
2440 total_duration = dts - priv->last_in_dts;
2444 /* interpolate between the current time and the last time based on
2445 * number of packets we are missing, this is the estimated duration
2446 * for the missing packet based on equidistant packet spacing. */
2447 duration = total_duration / (gap + 1);
2449 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2450 GST_TIME_ARGS (duration));
2452 if (total_duration > priv->latency_ns) {
2453 GstClockTime gap_time;
2457 GstClockTime gap_dur = gap * duration;
2458 if (gap_dur > priv->latency_ns)
2459 gap_time = gap_dur - priv->latency_ns;
2462 lost_packets = gap_time / duration;
2464 gap_time = total_duration - priv->latency_ns;
2468 /* too many lost packets, some of the missing packets are already
2469 * too late and we can generate lost packet events for them. */
2470 GST_DEBUG_OBJECT (jitterbuffer,
2471 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2472 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2473 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2474 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2475 GST_TIME_ARGS (gap_time));
2477 /* this timer will fire immediately and the lost event will be pushed from
2478 * the timer thread */
2479 if (lost_packets > 0) {
2480 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2481 priv->last_in_dts + duration, 0, gap_time);
2482 expected += lost_packets;
2483 priv->last_in_dts += gap_time;
2487 expected_dts = priv->last_in_dts + duration;
2490 if (priv->do_retransmission) {
2491 TimerData *timer = find_timer (jitterbuffer, expected);
2493 type = TIMER_TYPE_EXPECTED;
2494 delay = get_rtx_delay (priv);
2496 /* if we had a timer for the first missing packet, update it. */
2497 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2498 GstClockTime timeout = timer->timeout;
2500 timer->duration = duration;
2501 if (timeout > (expected_dts + delay) && timer->num_rtx_retry == 0) {
2502 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2506 expected_dts += duration;
2509 type = TIMER_TYPE_LOST;
2512 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2513 add_timer (jitterbuffer, type, expected, 0, expected_dts, delay, duration);
2514 expected_dts += duration;
2520 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2524 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2525 GstRtpJitterBufferPrivate *priv;
2527 priv = jitterbuffer->priv;
2529 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2532 if (priv->last_dts != -1)
2533 dtsdiff = dts - priv->last_dts;
2537 if (priv->last_rtptime != -1)
2538 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2542 priv->last_dts = dts;
2543 priv->last_rtptime = rtptime;
2547 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2550 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2552 diff = ABS (dtsdiff - rtpdiffns);
2554 /* jitter is stored in nanoseconds */
2555 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2557 GST_LOG_OBJECT (jitterbuffer,
2558 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2559 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2560 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2561 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2568 GST_DEBUG_OBJECT (jitterbuffer,
2569 "no dts or no clock-rate, can't calculate jitter");
2575 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2577 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2578 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2581 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2582 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2583 gst_rtp_buffer_unmap (&rtp_a);
2585 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2586 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2587 gst_rtp_buffer_unmap (&rtp_b);
2589 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2593 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2594 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2597 GstRtpJitterBufferPrivate *priv;
2598 guint gap_packets_length;
2599 gboolean reset = FALSE;
2601 priv = jitterbuffer->priv;
2603 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2605 guint32 prev_gap_seq = -1;
2606 gboolean all_consecutive = TRUE;
2608 g_queue_insert_sorted (&priv->gap_packets, buffer,
2609 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2611 for (l = priv->gap_packets.head; l; l = l->next) {
2612 GstBuffer *gap_buffer = l->data;
2613 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2616 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2618 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2620 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2621 if (prev_gap_seq == -1)
2622 prev_gap_seq = gap_seq;
2623 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2624 all_consecutive = FALSE;
2626 prev_gap_seq = gap_seq;
2628 gst_rtp_buffer_unmap (&gap_rtp);
2629 if (!all_consecutive)
2633 if (all_consecutive && gap_packets_length > 3) {
2634 GST_DEBUG_OBJECT (jitterbuffer,
2635 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2636 (future ? "new" : "old"), gap,
2637 (future ? max_dropout : -max_misorder));
2639 } else if (!all_consecutive) {
2640 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2641 g_queue_clear (&priv->gap_packets);
2642 GST_DEBUG_OBJECT (jitterbuffer,
2643 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2644 (future ? "new" : "old"), gap,
2645 (future ? max_dropout : -max_misorder));
2648 GST_DEBUG_OBJECT (jitterbuffer,
2649 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2650 (future ? "new" : "old"), gap,
2651 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2655 GST_DEBUG_OBJECT (jitterbuffer,
2656 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2657 gap, -max_misorder);
2658 g_queue_push_tail (&priv->gap_packets, buffer);
2666 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2668 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2669 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2672 GstClockTime base_time =
2673 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2674 GstClockTime clock_time = gst_clock_get_time (clock);
2676 if (clock_time > base_time)
2677 running_time = clock_time - base_time;
2681 gst_object_unref (clock);
2684 return running_time;
2687 static GstFlowReturn
2688 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2691 GstRtpJitterBuffer *jitterbuffer;
2692 GstRtpJitterBufferPrivate *priv;
2694 guint32 expected, rtptime;
2695 GstFlowReturn ret = GST_FLOW_OK;
2696 GstClockTime dts, pts;
2701 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2702 gboolean do_next_seqnum = FALSE;
2703 RTPJitterBufferItem *item;
2704 GstMessage *msg = NULL;
2705 gboolean estimated_dts = FALSE;
2706 guint32 packet_rate, max_dropout, max_misorder;
2707 TimerData *timer = NULL;
2709 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2711 priv = jitterbuffer->priv;
2713 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2714 goto invalid_buffer;
2716 pt = gst_rtp_buffer_get_payload_type (&rtp);
2717 seqnum = gst_rtp_buffer_get_seq (&rtp);
2718 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2719 gst_rtp_buffer_unmap (&rtp);
2721 /* make sure we have PTS and DTS set */
2722 pts = GST_BUFFER_PTS (buffer);
2723 dts = GST_BUFFER_DTS (buffer);
2730 /* If we have no DTS here, i.e. no capture time, get one from the
2731 * clock now to have something to calculate with in the future. */
2732 dts = get_current_running_time (jitterbuffer);
2735 /* Remember that we estimated the DTS if we are running already
2736 * and this is not our first packet (or first packet after a reset).
2737 * If it's the first packet, we somehow must generate a timestamp for
2738 * everything, otherwise we can't calculate any times
2740 estimated_dts = (priv->next_in_seqnum != -1);
2742 /* take the DTS of the buffer. This is the time when the packet was
2743 * received and is used to calculate jitter and clock skew. We will adjust
2744 * this DTS with the smoothed value after processing it in the
2745 * jitterbuffer and assign it as the PTS. */
2746 /* bring to running time */
2747 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2750 GST_DEBUG_OBJECT (jitterbuffer,
2751 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2752 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2753 GST_BUFFER_IS_RETRANSMISSION (buffer));
2755 JBUF_LOCK_CHECK (priv, out_flushing);
2757 if (G_UNLIKELY (priv->last_pt != pt)) {
2760 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2764 /* reset clock-rate so that we get a new one */
2765 priv->clock_rate = -1;
2767 /* Try to get the clock-rate from the caps first if we can. If there are no
2768 * caps we must fire the signal to get the clock-rate. */
2769 if ((caps = gst_pad_get_current_caps (pad))) {
2770 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2771 gst_caps_unref (caps);
2775 if (G_UNLIKELY (priv->clock_rate == -1)) {
2776 /* no clock rate given on the caps, try to get one with the signal */
2777 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2778 pt) == GST_FLOW_FLUSHING)
2781 if (G_UNLIKELY (priv->clock_rate == -1))
2784 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2787 /* don't accept more data on EOS */
2788 if (G_UNLIKELY (priv->eos))
2791 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2792 calculate_jitter (jitterbuffer, dts, rtptime);
2794 if (priv->seqnum_base != -1) {
2797 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2800 GST_DEBUG_OBJECT (jitterbuffer,
2801 "packet seqnum #%d before seqnum-base #%d", seqnum,
2803 gst_buffer_unref (buffer);
2806 } else if (gap > 16384) {
2807 /* From now on don't compare against the seqnum base anymore as
2808 * at some point in the future we will wrap around and also that
2809 * much reordering is very unlikely */
2810 priv->seqnum_base = -1;
2814 expected = priv->next_in_seqnum;
2817 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2819 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2820 priv->max_dropout_time);
2822 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2823 priv->max_misorder_time);
2824 GST_TRACE_OBJECT (jitterbuffer,
2825 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2826 max_dropout, max_misorder);
2828 /* now check against our expected seqnum */
2829 if (G_LIKELY (expected != -1)) {
2832 /* now calculate gap */
2833 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2835 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2836 expected, seqnum, gap);
2838 if (G_LIKELY (gap == 0)) {
2839 /* packet is expected */
2840 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2841 do_next_seqnum = TRUE;
2843 gboolean reset = FALSE;
2846 /* we received an old packet */
2847 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2849 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2850 gap, max_dropout, max_misorder);
2853 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2856 /* new packet, we are missing some packets */
2857 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2858 /* If we have timers for more than RTP_MAX_DROPOUT packets
2859 * pending this means that we have a huge gap overall. We can
2860 * reset the jitterbuffer at this point because there's
2861 * just too much data missing to be able to do anything
2862 * sensible with the past data. Just try again from the
2864 GST_WARNING_OBJECT (jitterbuffer,
2865 "%d pending timers > %d - resetting", priv->timers->len,
2868 gst_buffer_unref (buffer);
2870 } else if (G_UNLIKELY (gap >= max_dropout)) {
2872 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2873 gap, max_dropout, max_misorder);
2876 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2877 /* fill in the gap with EXPECTED timers */
2878 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2880 do_next_seqnum = TRUE;
2883 if (G_UNLIKELY (reset)) {
2884 GList *events = NULL, *l;
2887 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2888 rtp_jitter_buffer_flush (priv->jbuf,
2889 (GFunc) free_item_and_retain_events, &events);
2890 rtp_jitter_buffer_reset_skew (priv->jbuf);
2891 remove_all_timers (jitterbuffer);
2892 priv->discont = TRUE;
2893 priv->last_popped_seqnum = -1;
2895 if (priv->gap_packets.head) {
2896 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2897 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2899 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2900 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2901 gst_rtp_buffer_unmap (&gap_rtp);
2903 priv->next_seqnum = seqnum;
2906 priv->last_in_dts = -1;
2907 priv->next_in_seqnum = -1;
2909 /* Insert all sticky events again in order, otherwise we would
2910 * potentially loose STREAM_START, CAPS or SEGMENT events
2912 events = g_list_reverse (events);
2913 for (l = events; l; l = l->next) {
2914 RTPJitterBufferItem *item;
2916 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2917 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2919 g_list_free (events);
2921 JBUF_SIGNAL_EVENT (priv);
2923 /* reset spacing estimation when gap */
2924 priv->ips_rtptime = -1;
2925 priv->ips_dts = GST_CLOCK_TIME_NONE;
2927 buffers = g_list_copy (priv->gap_packets.head);
2928 g_queue_clear (&priv->gap_packets);
2930 priv->ips_rtptime = -1;
2931 priv->ips_dts = GST_CLOCK_TIME_NONE;
2932 JBUF_UNLOCK (jitterbuffer->priv);
2934 for (l = buffers; l; l = l->next) {
2935 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2937 if (ret != GST_FLOW_OK) {
2942 for (; l; l = l->next)
2943 gst_buffer_unref (l->data);
2944 g_list_free (buffers);
2948 /* reset spacing estimation when gap */
2949 priv->ips_rtptime = -1;
2950 priv->ips_dts = GST_CLOCK_TIME_NONE;
2953 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2955 /* we don't know what the next_in_seqnum should be, wait for the last
2956 * possible moment to push this buffer, maybe we get an earlier seqnum
2958 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2959 do_next_seqnum = TRUE;
2960 /* take rtptime and dts to calculate packet spacing */
2961 priv->ips_rtptime = rtptime;
2962 priv->ips_dts = dts;
2965 /* We had no huge gap, let's drop all the gap packets */
2966 if (buffer != NULL) {
2967 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2968 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2969 g_queue_clear (&priv->gap_packets);
2971 GST_DEBUG_OBJECT (jitterbuffer,
2972 "Had big gap, waiting for more consecutive packets");
2973 JBUF_UNLOCK (jitterbuffer->priv);
2977 if (do_next_seqnum) {
2978 priv->last_in_dts = dts;
2979 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2982 timer = find_timer (jitterbuffer, seqnum);
2983 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
2985 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
2987 timer->num_rtx_received++;
2990 /* let's check if this buffer is too late, we can only accept packets with
2991 * bigger seqnum than the one we last pushed. */
2992 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2995 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2997 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2998 if (G_UNLIKELY (gap <= 0)) {
2999 if (priv->do_retransmission) {
3000 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
3001 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3002 /* Only count the retranmitted packet too late if it has been
3003 * considered lost. If the original packet arrived before the
3004 * retransmitted we just count it as a duplicate. */
3005 if (timer->type != TIMER_TYPE_LOST)
3013 if (already_lost (jitterbuffer, seqnum))
3016 /* let's drop oldest packet if the queue is already full and drop-on-latency
3017 * is set. We can only do this when there actually is a latency. When no
3018 * latency is set, we just pump it in the queue and let the other end push it
3019 * out as fast as possible. */
3020 if (priv->latency_ms && priv->drop_on_latency) {
3022 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3024 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3025 RTPJitterBufferItem *old_item;
3027 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3029 if (IS_DROPABLE (old_item)) {
3030 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3031 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3033 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
3034 free_item (old_item);
3036 /* we might have removed some head buffers, signal the pushing thread to
3037 * see if it can push now */
3038 JBUF_SIGNAL_EVENT (priv);
3042 /* If we estimated the DTS, don't consider it in the clock skew calculations
3043 * later. The code above always sets dts to pts or the other way around if
3044 * any of those is valid in the buffer, so we know that if we estimated the
3045 * dts that both are unknown */
3048 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3049 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
3051 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3053 /* now insert the packet into the queue in sorted order. This function returns
3054 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3055 * have a duplicate. */
3056 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
3058 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer))))) {
3059 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3060 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3065 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum,
3066 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3068 /* we had an unhandled SR, handle it now */
3070 do_handle_sync (jitterbuffer);
3072 if (G_UNLIKELY (head)) {
3073 /* signal addition of new buffer when the _loop is waiting. */
3074 if (G_LIKELY (priv->active))
3075 JBUF_SIGNAL_EVENT (priv);
3077 /* let's unschedule and unblock any waiting buffers. We only want to do this
3078 * when the head buffer changed */
3079 if (G_UNLIKELY (priv->clock_id)) {
3080 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3081 unschedule_current_timer (jitterbuffer);
3085 GST_DEBUG_OBJECT (jitterbuffer,
3086 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3087 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3089 msg = check_buffering_percent (jitterbuffer, percent);
3095 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3102 /* this is not fatal but should be filtered earlier */
3103 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3104 ("Received invalid RTP payload, dropping"));
3105 gst_buffer_unref (buffer);
3110 GST_WARNING_OBJECT (jitterbuffer,
3111 "No clock-rate in caps!, dropping buffer");
3112 gst_buffer_unref (buffer);
3117 ret = priv->srcresult;
3118 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3119 gst_buffer_unref (buffer);
3125 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3126 gst_buffer_unref (buffer);
3131 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3132 " popped, dropping", seqnum, priv->last_popped_seqnum);
3134 gst_buffer_unref (buffer);
3139 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3140 "considered lost", seqnum);
3142 gst_buffer_unref (buffer);
3147 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3149 priv->num_duplicates++;
3155 GST_DEBUG_OBJECT (jitterbuffer,
3156 "Duplicate RTX packet #%d detected, dropping", seqnum);
3157 priv->num_duplicates++;
3158 gst_buffer_unref (buffer);
3164 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3166 guint64 ext_time, elapsed;
3168 GstRtpJitterBufferPrivate *priv;
3170 priv = jitterbuffer->priv;
3171 rtp_time = item->rtptime;
3173 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3174 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3176 ext_time = priv->ext_timestamp;
3177 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3178 if (ext_time < priv->ext_timestamp) {
3179 ext_time = priv->ext_timestamp;
3181 priv->ext_timestamp = ext_time;
3184 if (ext_time > priv->clock_base)
3185 elapsed = ext_time - priv->clock_base;
3189 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3194 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3195 RTPJitterBufferItem * item)
3197 guint64 total, elapsed, left, estimated;
3198 GstClockTime out_time;
3199 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3201 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3202 || priv->clock_base == -1 || priv->clock_rate <= 0)
3205 /* compute the elapsed time */
3206 elapsed = compute_elapsed (jitterbuffer, item);
3208 /* do nothing if elapsed time doesn't increment */
3209 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3212 priv->last_elapsed = elapsed;
3214 /* this is the total time we need to play */
3215 total = priv->npt_stop - priv->npt_start;
3216 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3217 GST_TIME_ARGS (total));
3219 /* this is how much time there is left */
3220 if (total > elapsed)
3221 left = total - elapsed;
3225 /* if we have less time left that the size of the buffer, we will not
3226 * be able to keep it filled, disabled buffering then */
3227 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3228 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3229 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3230 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3233 /* this is the current time as running-time */
3234 out_time = item->dts;
3237 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3239 /* if there is almost nothing left,
3240 * we may never advance enough to end up in the above case */
3241 if (total < GST_SECOND)
3242 estimated = GST_SECOND;
3246 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3247 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3249 if (estimated != -1 && priv->estimated_eos != estimated) {
3250 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3251 priv->estimated_eos = estimated;
3255 /* take a buffer from the queue and push it */
3256 static GstFlowReturn
3257 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3259 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3260 GstFlowReturn result = GST_FLOW_OK;
3261 RTPJitterBufferItem *item;
3262 GstBuffer *outbuf = NULL;
3263 GstEvent *outevent = NULL;
3264 GstQuery *outquery = NULL;
3265 GstClockTime dts, pts;
3267 gboolean do_push = TRUE;
3271 /* when we get here we are ready to pop and push the buffer */
3272 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3276 case ITEM_TYPE_BUFFER:
3278 /* we need to make writable to change the flags and timestamps */
3279 outbuf = gst_buffer_make_writable (item->data);
3281 if (G_UNLIKELY (priv->discont)) {
3282 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3283 * into the jitterbuffer so we can modify now. */
3284 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3285 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3286 priv->discont = FALSE;
3288 if (G_UNLIKELY (priv->ts_discont)) {
3289 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3290 priv->ts_discont = FALSE;
3294 gst_segment_position_from_running_time (&priv->segment,
3295 GST_FORMAT_TIME, item->dts);
3297 gst_segment_position_from_running_time (&priv->segment,
3298 GST_FORMAT_TIME, item->pts);
3300 /* apply timestamp with offset to buffer now */
3301 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3302 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3304 /* update the elapsed time when we need to check against the npt stop time. */
3305 update_estimated_eos (jitterbuffer, item);
3307 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3309 case ITEM_TYPE_LOST:
3310 priv->discont = TRUE;
3314 case ITEM_TYPE_EVENT:
3315 outevent = item->data;
3317 case ITEM_TYPE_QUERY:
3318 outquery = item->data;
3322 /* now we are ready to push the buffer. Save the seqnum and release the lock
3323 * so the other end can push stuff in the queue again. */
3325 priv->last_popped_seqnum = seqnum;
3326 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3328 msg = check_buffering_percent (jitterbuffer, percent);
3335 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3338 case ITEM_TYPE_BUFFER:
3340 GST_DEBUG_OBJECT (jitterbuffer,
3341 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3342 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3343 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3345 result = gst_pad_push (priv->srcpad, outbuf);
3347 JBUF_LOCK_CHECK (priv, out_flushing);
3349 case ITEM_TYPE_LOST:
3350 case ITEM_TYPE_EVENT:
3351 /* We got not enough consecutive packets with a huge gap, we can
3352 * as well just drop them here now on EOS */
3353 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3354 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3355 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3356 g_queue_clear (&priv->gap_packets);
3359 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3360 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3363 gst_pad_push_event (priv->srcpad, outevent);
3365 gst_event_unref (outevent);
3367 result = GST_FLOW_OK;
3369 JBUF_LOCK_CHECK (priv, out_flushing);
3371 case ITEM_TYPE_QUERY:
3375 res = gst_pad_peer_query (priv->srcpad, outquery);
3377 JBUF_LOCK_CHECK (priv, out_flushing);
3378 result = GST_FLOW_OK;
3379 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3380 JBUF_SIGNAL_QUERY (priv, res);
3389 return priv->srcresult;
3393 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3395 /* Peek a buffer and compare the seqnum to the expected seqnum.
3396 * If all is fine, the buffer is pushed.
3397 * If something is wrong, we wait for some event
3399 static GstFlowReturn
3400 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3402 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3403 GstFlowReturn result;
3404 RTPJitterBufferItem *item;
3406 guint32 next_seqnum;
3408 /* only push buffers when PLAYING and active and not buffering */
3409 if (priv->blocked || !priv->active ||
3410 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3411 return GST_FLOW_WAIT;
3414 /* peek a buffer, we're just looking at the sequence number.
3415 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3416 * wait for a timeout or something to change.
3417 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3418 item = rtp_jitter_buffer_peek (priv->jbuf);
3423 /* get the seqnum and the next expected seqnum */
3424 seqnum = item->seqnum;
3426 return pop_and_push_next (jitterbuffer, seqnum);
3429 next_seqnum = priv->next_seqnum;
3431 /* get the gap between this and the previous packet. If we don't know the
3432 * previous packet seqnum assume no gap. */
3433 if (G_UNLIKELY (next_seqnum == -1)) {
3434 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3435 /* we don't know what the next_seqnum should be, the chain function should
3436 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3437 * fires, so wait for that */
3438 result = GST_FLOW_WAIT;
3440 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3442 if (G_LIKELY (gap == 0)) {
3443 /* no missing packet, pop and push */
3444 result = pop_and_push_next (jitterbuffer, seqnum);
3445 } else if (G_UNLIKELY (gap < 0)) {
3446 /* if we have a packet that we already pushed or considered dropped, pop it
3447 * off and get the next packet */
3448 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3449 seqnum, next_seqnum);
3450 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3452 result = GST_FLOW_OK;
3454 /* the chain function has scheduled timers to request retransmission or
3455 * when to consider the packet lost, wait for that */
3456 GST_DEBUG_OBJECT (jitterbuffer,
3457 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3458 next_seqnum, seqnum, gap);
3459 result = GST_FLOW_WAIT;
3467 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3469 return GST_FLOW_EOS;
3471 return GST_FLOW_WAIT;
3477 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3479 GstClockTime rtx_retry_timeout;
3480 GstClockTime rtx_min_retry_timeout;
3482 if (priv->rtx_retry_timeout == -1) {
3483 if (priv->avg_rtx_rtt == 0)
3484 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3486 /* we want to ask for a retransmission after we waited for a
3487 * complete RTT and the additional jitter */
3488 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3490 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3492 /* make sure we don't retry too often. On very low latency networks,
3493 * the RTT and jitter can be very low. */
3494 if (priv->rtx_min_retry_timeout == -1) {
3495 rtx_min_retry_timeout = priv->packet_spacing;
3497 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3499 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3501 return rtx_retry_timeout;
3505 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3506 GstClockTime rtx_retry_timeout)
3508 GstClockTime rtx_retry_period;
3510 if (priv->rtx_retry_period == -1) {
3511 /* we retry up to the configured jitterbuffer size but leaving some
3512 * room for the retransmission to arrive in time */
3513 if (rtx_retry_timeout > priv->latency_ns) {
3514 rtx_retry_period = 0;
3516 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3519 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3521 return rtx_retry_period;
3525 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3526 GstClockTime dts, gboolean success)
3528 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3532 /* we scheduled a retry for this packet and now we have it */
3533 priv->num_rtx_success++;
3534 /* all the previous retry attempts failed */
3535 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3537 /* All retries failed or was too late */
3538 priv->num_rtx_failed += timer->num_rtx_retry;
3541 /* number of retries before (hopefully) receiving the packet */
3542 if (priv->avg_rtx_num == 0.0)
3543 priv->avg_rtx_num = timer->num_rtx_retry;
3545 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3547 /* Calculate the delay between retransmission request and receiving this
3548 * packet. We have a valid delay if and only if this packet is a response to
3549 * our last request. If not we don't know if this is a response to an
3550 * earlier request and delay could be way off. For RTT is more important
3551 * with correct values than to update for every packet. */
3552 if (timer->num_rtx_retry == timer->num_rtx_received &&
3553 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3554 delay = dts - timer->rtx_last;
3555 if (priv->avg_rtx_rtt == 0)
3556 priv->avg_rtx_rtt = delay;
3558 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
3563 GST_LOG_OBJECT (jitterbuffer,
3564 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3565 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3566 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3567 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3568 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3569 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3570 GST_TIME_ARGS (priv->avg_rtx_rtt));
3573 /* the timeout for when we expected a packet expired */
3575 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3578 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3580 guint delay, delay_ms, avg_rtx_rtt_ms;
3581 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3582 guint rtx_deadline_ms;
3583 GstClockTime rtx_retry_period;
3584 GstClockTime rtx_retry_timeout;
3587 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3588 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3590 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3591 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3593 delay = timer->rtx_delay + timer->rtx_retry;
3595 delay_ms = GST_TIME_AS_MSECONDS (delay);
3596 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3597 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3598 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3600 priv->rtx_deadline_ms != -1 ? priv->rtx_deadline_ms : priv->latency_ms;
3602 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3603 gst_structure_new ("GstRTPRetransmissionRequest",
3604 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3605 "running-time", G_TYPE_UINT64, timer->rtx_base,
3606 "delay", G_TYPE_UINT, delay_ms,
3607 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3608 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3609 "period", G_TYPE_UINT, rtx_retry_period_ms,
3610 "deadline", G_TYPE_UINT, rtx_deadline_ms,
3611 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3612 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3613 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3615 priv->num_rtx_requests++;
3616 timer->num_rtx_retry++;
3618 GST_OBJECT_LOCK (jitterbuffer);
3619 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3620 timer->rtx_last = gst_clock_get_time (clock);
3621 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3623 timer->rtx_last = now;
3625 GST_OBJECT_UNLOCK (jitterbuffer);
3627 /* calculate the timeout for the next retransmission attempt */
3628 timer->rtx_retry += rtx_retry_timeout;
3629 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3630 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3631 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3632 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3633 if ((priv->rtx_max_retries != -1
3634 && timer->num_rtx_retry >= priv->rtx_max_retries)
3635 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3636 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3637 /* too many retransmission request, we now convert the timer
3638 * to a lost timer, leave the num_rtx_retry as it is for stats */
3639 timer->type = TIMER_TYPE_LOST;
3640 timer->rtx_delay = 0;
3641 timer->rtx_retry = 0;
3643 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3644 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3647 gst_pad_push_event (priv->sinkpad, event);
3653 /* a packet is lost */
3655 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3658 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3659 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3661 GstEvent *event = NULL;
3662 RTPJitterBufferItem *item;
3664 seqnum = timer->seqnum;
3665 lost_packets = MAX (timer->num, 1);
3666 num_rtx_retry = timer->num_rtx_retry;
3668 /* we had a gap and thus we lost some packets. Create an event for this. */
3669 if (lost_packets > 1)
3670 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3671 seqnum + lost_packets - 1);
3673 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3675 priv->num_lost += lost_packets;
3676 priv->num_rtx_failed += num_rtx_retry;
3678 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3680 /* we now only accept seqnum bigger than this */
3681 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3682 priv->next_in_seqnum = next_in_seqnum;
3684 /* Avoid creating events if we don't need it. Note that we still need to create
3685 * the lost *ITEM* since it will be used to notify the outgoing thread of
3686 * lost items (so that we can set discont flags and such) */
3687 if (priv->do_lost) {
3688 GstClockTime duration, timestamp;
3689 /* create paket lost event */
3690 timestamp = apply_offset (jitterbuffer, timer->timeout);
3691 duration = timer->duration;
3692 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3693 duration = priv->packet_spacing;
3694 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3695 gst_structure_new ("GstRTPPacketLost",
3696 "seqnum", G_TYPE_UINT, (guint) seqnum,
3697 "timestamp", G_TYPE_UINT64, timestamp,
3698 "duration", G_TYPE_UINT64, duration,
3699 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3701 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3702 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3704 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3705 /* Store info to update stats if the packet arrives too late */
3706 timer_queue_append (priv->rtx_stats_timers, timer,
3707 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3709 remove_timer (jitterbuffer, timer);
3712 JBUF_SIGNAL_EVENT (priv);
3718 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3721 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3723 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3724 remove_timer (jitterbuffer, timer);
3726 /* there was no EOS in the buffer, put one in there now */
3727 queue_event (jitterbuffer, gst_event_new_eos ());
3729 JBUF_SIGNAL_EVENT (priv);
3735 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3738 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3740 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3742 /* timer seqnum might have been obsoleted by caps seqnum-base,
3743 * only mess with current ongoing seqnum if still unknown */
3744 if (priv->next_seqnum == -1)
3745 priv->next_seqnum = timer->seqnum;
3746 remove_timer (jitterbuffer, timer);
3747 JBUF_SIGNAL_EVENT (priv);
3753 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3756 gboolean removed = FALSE;
3758 switch (timer->type) {
3759 case TIMER_TYPE_EXPECTED:
3760 removed = do_expected_timeout (jitterbuffer, timer, now);
3762 case TIMER_TYPE_LOST:
3763 removed = do_lost_timeout (jitterbuffer, timer, now);
3765 case TIMER_TYPE_DEADLINE:
3766 removed = do_deadline_timeout (jitterbuffer, timer, now);
3768 case TIMER_TYPE_EOS:
3769 removed = do_eos_timeout (jitterbuffer, timer, now);
3775 /* called when we need to wait for the next timeout.
3777 * We loop over the array of recorded timeouts and wait for the earliest one.
3778 * When it timed out, do the logic associated with the timer.
3780 * If there are no timers, we wait on a gcond until something new happens.
3783 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3785 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3786 GstClockTime now = 0;
3789 while (priv->timer_running) {
3790 TimerData *timer = NULL;
3791 GstClockTime timer_timeout = -1;
3794 /* If we have a clock, update "now" now with the very
3795 * latest running time we have. If timers are unscheduled below we
3796 * otherwise wouldn't update now (it's only updated when timers
3797 * expire), and also for the very first loop iteration now would
3798 * otherwise always be 0
3800 GST_OBJECT_LOCK (jitterbuffer);
3801 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3803 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3804 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3806 GST_OBJECT_UNLOCK (jitterbuffer);
3808 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3809 GST_TIME_ARGS (now));
3811 /* Clear expired rtx-stats timers */
3812 if (priv->do_retransmission)
3813 timer_queue_clear_until (priv->rtx_stats_timers, now);
3815 /* Iterate "normal" timers */
3816 len = priv->timers->len;
3817 for (i = 0; i < len;) {
3818 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3819 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3820 gboolean save_best = FALSE;
3822 GST_DEBUG_OBJECT (jitterbuffer,
3823 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3824 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3825 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3827 /* Weed out anything too late */
3828 if (test->type == TIMER_TYPE_LOST &&
3829 (test_timeout == -1 || test_timeout <= now)) {
3830 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3831 do_lost_timeout (jitterbuffer, test, now);
3832 if (!priv->timer_running)
3834 /* We don't move the iterator forward since we just removed the current entry,
3835 * but we update the termination condition */
3836 len = priv->timers->len;
3838 /* find the smallest timeout */
3839 if (timer == NULL) {
3841 } else if (timer_timeout == -1) {
3842 /* we already have an immediate timeout, the new timer must be an
3843 * immediate timer with smaller seqnum to become the best */
3844 if (test_timeout == -1
3845 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3846 timer->seqnum) > 0))
3848 } else if (test_timeout == -1) {
3849 /* first immediate timer */
3851 } else if (test_timeout < timer_timeout) {
3854 } else if (test_timeout == timer_timeout
3855 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3856 timer->seqnum) > 0)) {
3857 /* same timer, smaller seqnum */
3862 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3864 timer_timeout = test_timeout;
3869 if (timer && !priv->blocked) {
3871 GstClockTime sync_time;
3874 GstClockTimeDiff clock_jitter;
3876 if (timer_timeout == -1 || timer_timeout <= now) {
3877 /* We have normally removed all lost timers in the loop above */
3878 g_assert (timer->type != TIMER_TYPE_LOST);
3880 do_timeout (jitterbuffer, timer, now);
3881 /* check here, do_timeout could have released the lock */
3882 if (!priv->timer_running)
3887 GST_OBJECT_LOCK (jitterbuffer);
3888 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3890 GST_OBJECT_UNLOCK (jitterbuffer);
3891 /* let's just push if there is no clock */
3892 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3893 now = timer_timeout;
3897 /* prepare for sync against clock */
3898 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3899 /* add latency of peer to get input time */
3900 sync_time += priv->peer_latency;
3902 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3903 " with sync time %" GST_TIME_FORMAT,
3904 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3906 /* create an entry for the clock */
3907 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3908 priv->timer_timeout = timer_timeout;
3909 priv->timer_seqnum = timer->seqnum;
3910 GST_OBJECT_UNLOCK (jitterbuffer);
3912 /* release the lock so that the other end can push stuff or unlock */
3915 ret = gst_clock_id_wait (id, &clock_jitter);
3918 if (!priv->timer_running) {
3919 gst_clock_id_unref (id);
3920 priv->clock_id = NULL;
3924 if (ret != GST_CLOCK_UNSCHEDULED) {
3925 now = timer_timeout + MAX (clock_jitter, 0);
3926 GST_DEBUG_OBJECT (jitterbuffer,
3927 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3928 GST_STIME_ARGS (clock_jitter));
3930 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3932 /* and free the entry */
3933 gst_clock_id_unref (id);
3934 priv->clock_id = NULL;
3936 /* no timers, wait for activity */
3937 JBUF_WAIT_TIMER (priv);
3942 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3947 * This funcion implements the main pushing loop on the source pad.
3949 * It first tries to push as many buffers as possible. If there is a seqnum
3950 * mismatch, we wait for the next timeouts.
3953 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3955 GstRtpJitterBufferPrivate *priv;
3956 GstFlowReturn result = GST_FLOW_OK;
3958 priv = jitterbuffer->priv;
3960 JBUF_LOCK_CHECK (priv, flushing);
3962 result = handle_next_buffer (jitterbuffer);
3963 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3964 /* now wait for the next event */
3965 JBUF_WAIT_EVENT (priv, flushing);
3966 result = GST_FLOW_OK;
3968 } while (result == GST_FLOW_OK);
3969 /* store result for upstream */
3970 priv->srcresult = result;
3971 /* if we get here we need to pause */
3977 result = priv->srcresult;
3984 JBUF_SIGNAL_QUERY (priv, FALSE);
3987 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3988 gst_flow_get_name (result));
3989 gst_pad_pause_task (priv->srcpad);
3990 if (result == GST_FLOW_EOS) {
3991 event = gst_event_new_eos ();
3992 gst_pad_push_event (priv->srcpad, event);
3998 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3999 * some sanity checks and then emit the handle-sync signal with the parameters.
4000 * This function must be called with the LOCK */
4002 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
4004 GstRtpJitterBufferPrivate *priv;
4005 guint64 base_rtptime, base_time;
4007 guint64 last_rtptime;
4009 guint64 ext_rtptime, diff;
4010 gboolean valid = TRUE, keep = FALSE;
4012 priv = jitterbuffer->priv;
4014 /* get the last values from the jitterbuffer */
4015 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
4016 &clock_rate, &last_rtptime);
4018 clock_base = priv->clock_base;
4019 ext_rtptime = priv->ext_rtptime;
4021 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4022 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4023 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4024 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4026 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4027 /* we keep this SR packet for later. When we get a valid RTP packet the
4028 * above values will be set and we can try to use the SR packet */
4029 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4032 /* we can't accept anything that happened before we did the last resync */
4033 if (base_rtptime > ext_rtptime) {
4034 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4037 /* the SR RTP timestamp must be something close to what we last observed
4038 * in the jitterbuffer */
4039 if (ext_rtptime > last_rtptime) {
4040 /* check how far ahead it is to our RTP timestamps */
4041 diff = ext_rtptime - last_rtptime;
4042 /* if bigger than 1 second, we drop it */
4043 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4045 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4046 clock_rate, 1000)) {
4047 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4048 /* should drop this, but some RTSP servers end up with bogus
4049 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4050 * so still trigger rptbin sync but invalidate RTCP data
4051 * (sync might use other methods) */
4054 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4055 G_GUINT64_FORMAT, last_rtptime, diff);
4061 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4065 s = gst_structure_new ("application/x-rtp-sync",
4066 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4067 "base-time", G_TYPE_UINT64, base_time,
4068 "clock-rate", G_TYPE_UINT, clock_rate,
4069 "clock-base", G_TYPE_UINT64, clock_base,
4070 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4071 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4073 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4074 gst_buffer_replace (&priv->last_sr, NULL);
4076 g_signal_emit (jitterbuffer,
4077 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4079 gst_structure_free (s);
4081 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4082 gst_buffer_replace (&priv->last_sr, NULL);
4086 static GstFlowReturn
4087 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4090 GstRtpJitterBuffer *jitterbuffer;
4091 GstRtpJitterBufferPrivate *priv;
4092 GstFlowReturn ret = GST_FLOW_OK;
4094 GstRTCPPacket packet;
4095 guint64 ext_rtptime;
4097 GstRTCPBuffer rtcp = { NULL, };
4099 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4101 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4102 goto invalid_buffer;
4104 priv = jitterbuffer->priv;
4106 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4108 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4111 /* first packet must be SR or RR or else the validate would have failed */
4112 switch (gst_rtcp_packet_get_type (&packet)) {
4113 case GST_RTCP_TYPE_SR:
4114 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4120 gst_rtcp_buffer_unmap (&rtcp);
4122 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4125 /* convert the RTP timestamp to our extended timestamp, using the same offset
4126 * we used in the jitterbuffer */
4127 ext_rtptime = priv->jbuf->ext_rtptime;
4128 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4130 priv->ext_rtptime = ext_rtptime;
4131 gst_buffer_replace (&priv->last_sr, buffer);
4133 do_handle_sync (jitterbuffer);
4137 gst_buffer_unref (buffer);
4143 /* this is not fatal but should be filtered earlier */
4144 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4145 ("Received invalid RTCP payload, dropping"));
4151 /* this is not fatal but should be filtered earlier */
4152 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4153 ("Received empty RTCP payload, dropping"));
4154 gst_rtcp_buffer_unmap (&rtcp);
4160 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4161 gst_rtcp_buffer_unmap (&rtcp);
4168 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4171 gboolean res = FALSE;
4172 GstRtpJitterBuffer *jitterbuffer;
4173 GstRtpJitterBufferPrivate *priv;
4175 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4176 priv = jitterbuffer->priv;
4178 switch (GST_QUERY_TYPE (query)) {
4179 case GST_QUERY_CAPS:
4181 GstCaps *filter, *caps;
4183 gst_query_parse_caps (query, &filter);
4184 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4185 gst_query_set_caps_result (query, caps);
4186 gst_caps_unref (caps);
4191 if (GST_QUERY_IS_SERIALIZED (query)) {
4192 RTPJitterBufferItem *item;
4195 JBUF_LOCK_CHECK (priv, out_flushing);
4196 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4197 RTP_JITTER_BUFFER_MODE_BUFFER) {
4198 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4199 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4200 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
4202 JBUF_SIGNAL_EVENT (priv);
4203 JBUF_WAIT_QUERY (priv, out_flushing);
4204 res = priv->last_query;
4206 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4211 res = gst_pad_query_default (pad, parent, query);
4219 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4227 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4230 GstRtpJitterBuffer *jitterbuffer;
4231 GstRtpJitterBufferPrivate *priv;
4232 gboolean res = FALSE;
4234 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4235 priv = jitterbuffer->priv;
4237 switch (GST_QUERY_TYPE (query)) {
4238 case GST_QUERY_LATENCY:
4240 /* We need to send the query upstream and add the returned latency to our
4242 GstClockTime min_latency, max_latency;
4244 GstClockTime our_latency;
4246 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4247 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4249 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4250 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4251 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4253 /* store this so that we can safely sync on the peer buffers. */
4255 priv->peer_latency = min_latency;
4256 our_latency = priv->latency_ns;
4259 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4260 GST_TIME_ARGS (our_latency));
4262 /* we add some latency but can buffer an infinite amount of time */
4263 min_latency += our_latency;
4266 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4267 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4268 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4270 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4274 case GST_QUERY_POSITION:
4276 GstClockTime start, last_out;
4279 gst_query_parse_position (query, &fmt, NULL);
4280 if (fmt != GST_FORMAT_TIME) {
4281 res = gst_pad_query_default (pad, parent, query);
4286 start = priv->npt_start;
4287 last_out = priv->last_out_time;
4290 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4291 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4292 GST_TIME_ARGS (last_out));
4294 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4295 /* bring 0-based outgoing time to stream time */
4296 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4299 res = gst_pad_query_default (pad, parent, query);
4303 case GST_QUERY_CAPS:
4305 GstCaps *filter, *caps;
4307 gst_query_parse_caps (query, &filter);
4308 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4309 gst_query_set_caps_result (query, caps);
4310 gst_caps_unref (caps);
4315 res = gst_pad_query_default (pad, parent, query);
4323 gst_rtp_jitter_buffer_set_property (GObject * object,
4324 guint prop_id, const GValue * value, GParamSpec * pspec)
4326 GstRtpJitterBuffer *jitterbuffer;
4327 GstRtpJitterBufferPrivate *priv;
4329 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4330 priv = jitterbuffer->priv;
4335 guint new_latency, old_latency;
4337 new_latency = g_value_get_uint (value);
4340 old_latency = priv->latency_ms;
4341 priv->latency_ms = new_latency;
4342 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4343 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4346 /* post message if latency changed, this will inform the parent pipeline
4347 * that a latency reconfiguration is possible/needed. */
4348 if (new_latency != old_latency) {
4349 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4350 GST_TIME_ARGS (new_latency * GST_MSECOND));
4352 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4353 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4357 case PROP_DROP_ON_LATENCY:
4359 priv->drop_on_latency = g_value_get_boolean (value);
4362 case PROP_TS_OFFSET:
4364 priv->ts_offset = g_value_get_int64 (value);
4365 priv->ts_discont = TRUE;
4370 priv->do_lost = g_value_get_boolean (value);
4375 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4378 case PROP_DO_RETRANSMISSION:
4380 priv->do_retransmission = g_value_get_boolean (value);
4383 case PROP_RTX_NEXT_SEQNUM:
4385 priv->rtx_next_seqnum = g_value_get_boolean (value);
4388 case PROP_RTX_DELAY:
4390 priv->rtx_delay = g_value_get_int (value);
4393 case PROP_RTX_MIN_DELAY:
4395 priv->rtx_min_delay = g_value_get_uint (value);
4398 case PROP_RTX_DELAY_REORDER:
4400 priv->rtx_delay_reorder = g_value_get_int (value);
4403 case PROP_RTX_RETRY_TIMEOUT:
4405 priv->rtx_retry_timeout = g_value_get_int (value);
4408 case PROP_RTX_MIN_RETRY_TIMEOUT:
4410 priv->rtx_min_retry_timeout = g_value_get_int (value);
4413 case PROP_RTX_RETRY_PERIOD:
4415 priv->rtx_retry_period = g_value_get_int (value);
4418 case PROP_RTX_MAX_RETRIES:
4420 priv->rtx_max_retries = g_value_get_int (value);
4423 case PROP_RTX_DEADLINE:
4425 priv->rtx_deadline_ms = g_value_get_int (value);
4428 case PROP_RTX_STATS_TIMEOUT:
4430 priv->rtx_stats_timeout = g_value_get_uint (value);
4433 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4435 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4438 case PROP_MAX_DROPOUT_TIME:
4440 priv->max_dropout_time = g_value_get_uint (value);
4443 case PROP_MAX_MISORDER_TIME:
4445 priv->max_misorder_time = g_value_get_uint (value);
4448 case PROP_RFC7273_SYNC:
4450 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4451 g_value_get_boolean (value));
4455 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4461 gst_rtp_jitter_buffer_get_property (GObject * object,
4462 guint prop_id, GValue * value, GParamSpec * pspec)
4464 GstRtpJitterBuffer *jitterbuffer;
4465 GstRtpJitterBufferPrivate *priv;
4467 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4468 priv = jitterbuffer->priv;
4473 g_value_set_uint (value, priv->latency_ms);
4476 case PROP_DROP_ON_LATENCY:
4478 g_value_set_boolean (value, priv->drop_on_latency);
4481 case PROP_TS_OFFSET:
4483 g_value_set_int64 (value, priv->ts_offset);
4488 g_value_set_boolean (value, priv->do_lost);
4493 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4501 if (priv->srcresult != GST_FLOW_OK)
4504 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4506 g_value_set_int (value, percent);
4510 case PROP_DO_RETRANSMISSION:
4512 g_value_set_boolean (value, priv->do_retransmission);
4515 case PROP_RTX_NEXT_SEQNUM:
4517 g_value_set_boolean (value, priv->rtx_next_seqnum);
4520 case PROP_RTX_DELAY:
4522 g_value_set_int (value, priv->rtx_delay);
4525 case PROP_RTX_MIN_DELAY:
4527 g_value_set_uint (value, priv->rtx_min_delay);
4530 case PROP_RTX_DELAY_REORDER:
4532 g_value_set_int (value, priv->rtx_delay_reorder);
4535 case PROP_RTX_RETRY_TIMEOUT:
4537 g_value_set_int (value, priv->rtx_retry_timeout);
4540 case PROP_RTX_MIN_RETRY_TIMEOUT:
4542 g_value_set_int (value, priv->rtx_min_retry_timeout);
4545 case PROP_RTX_RETRY_PERIOD:
4547 g_value_set_int (value, priv->rtx_retry_period);
4550 case PROP_RTX_MAX_RETRIES:
4552 g_value_set_int (value, priv->rtx_max_retries);
4555 case PROP_RTX_DEADLINE:
4557 g_value_set_int (value, priv->rtx_deadline_ms);
4560 case PROP_RTX_STATS_TIMEOUT:
4562 g_value_set_uint (value, priv->rtx_stats_timeout);
4566 g_value_take_boxed (value,
4567 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4569 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4571 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4574 case PROP_MAX_DROPOUT_TIME:
4576 g_value_set_uint (value, priv->max_dropout_time);
4579 case PROP_MAX_MISORDER_TIME:
4581 g_value_set_uint (value, priv->max_misorder_time);
4584 case PROP_RFC7273_SYNC:
4586 g_value_set_boolean (value,
4587 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4591 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4596 static GstStructure *
4597 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4599 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4603 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4604 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4605 "num-lost", G_TYPE_UINT64, priv->num_lost,
4606 "num-late", G_TYPE_UINT64, priv->num_late,
4607 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4608 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4609 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4610 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4611 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4612 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);