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_STATS_TIMEOUT 1000
147 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
148 #define DEFAULT_MAX_DROPOUT_TIME 60000
149 #define DEFAULT_MAX_MISORDER_TIME 2000
150 #define DEFAULT_RFC7273_SYNC FALSE
152 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
153 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
159 PROP_DROP_ON_LATENCY,
164 PROP_DO_RETRANSMISSION,
165 PROP_RTX_NEXT_SEQNUM,
168 PROP_RTX_DELAY_REORDER,
169 PROP_RTX_RETRY_TIMEOUT,
170 PROP_RTX_MIN_RETRY_TIMEOUT,
171 PROP_RTX_RETRY_PERIOD,
172 PROP_RTX_MAX_RETRIES,
173 PROP_RTX_STATS_TIMEOUT,
175 PROP_MAX_RTCP_RTP_TIME_DIFF,
176 PROP_MAX_DROPOUT_TIME,
177 PROP_MAX_MISORDER_TIME,
181 #define JBUF_LOCK(priv) G_STMT_START { \
182 GST_TRACE("Locking from thread %p", g_thread_self()); \
183 (g_mutex_lock (&(priv)->jbuf_lock)); \
184 GST_TRACE("Locked from thread %p", g_thread_self()); \
187 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
189 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
192 #define JBUF_UNLOCK(priv) G_STMT_START { \
193 GST_TRACE ("Unlocking from thread %p", g_thread_self ()); \
194 (g_mutex_unlock (&(priv)->jbuf_lock)); \
197 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
198 GST_DEBUG ("waiting timer"); \
199 (priv)->waiting_timer = TRUE; \
200 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
201 (priv)->waiting_timer = FALSE; \
202 GST_DEBUG ("waiting timer done"); \
204 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
205 if (G_UNLIKELY ((priv)->waiting_timer)) { \
206 GST_DEBUG ("signal timer"); \
207 g_cond_signal (&(priv)->jbuf_timer); \
211 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
212 GST_DEBUG ("waiting event"); \
213 (priv)->waiting_event = TRUE; \
214 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
215 (priv)->waiting_event = FALSE; \
216 GST_DEBUG ("waiting event done"); \
217 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
220 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
221 if (G_UNLIKELY ((priv)->waiting_event)) { \
222 GST_DEBUG ("signal event"); \
223 g_cond_signal (&(priv)->jbuf_event); \
227 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
228 GST_DEBUG ("waiting query"); \
229 (priv)->waiting_query = TRUE; \
230 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
231 (priv)->waiting_query = FALSE; \
232 GST_DEBUG ("waiting query done"); \
233 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
236 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
237 (priv)->last_query = res; \
238 if (G_UNLIKELY ((priv)->waiting_query)) { \
239 GST_DEBUG ("signal query"); \
240 g_cond_signal (&(priv)->jbuf_query); \
244 #define GST_BUFFER_IS_RETRANSMISSION(buffer) \
245 GST_BUFFER_FLAG_IS_SET (buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION)
247 typedef struct TimerQueue
250 GHashTable *hashtable;
253 struct _GstRtpJitterBufferPrivate
255 GstPad *sinkpad, *srcpad;
258 RTPJitterBuffer *jbuf;
260 gboolean waiting_timer;
262 gboolean waiting_event;
264 gboolean waiting_query;
272 gboolean timer_running;
273 GThread *timer_thread;
278 gboolean drop_on_latency;
281 gboolean do_retransmission;
282 gboolean rtx_next_seqnum;
285 gint rtx_delay_reorder;
286 gint rtx_retry_timeout;
287 gint rtx_min_retry_timeout;
288 gint rtx_retry_period;
289 gint rtx_max_retries;
290 guint rtx_stats_timeout;
291 gint max_rtcp_rtp_time_diff;
292 guint32 max_dropout_time;
293 guint32 max_misorder_time;
295 /* the last seqnum we pushed out */
296 guint32 last_popped_seqnum;
297 /* the next expected seqnum we push */
299 /* seqnum-base, if known */
301 /* last output time */
302 GstClockTime last_out_time;
303 /* last valid input timestamp and rtptime pair */
304 GstClockTime ips_dts;
306 GstClockTime packet_spacing;
310 /* the next expected seqnum we receive */
311 GstClockTime last_in_dts;
312 guint32 next_in_seqnum;
315 TimerQueue *rtx_stats_timers;
317 /* start and stop ranges */
318 GstClockTime npt_start;
319 GstClockTime npt_stop;
320 guint64 ext_timestamp;
321 guint64 last_elapsed;
322 guint64 estimated_eos;
329 /* clock rate and rtp timestamp offset */
333 gint64 prev_ts_offset;
335 /* when we are shutting down */
336 GstFlowReturn srcresult;
342 GstClockTime timer_timeout;
343 guint16 timer_seqnum;
344 /* the latency of the upstream peer, we have to take this into account when
345 * synchronizing the buffers. */
346 GstClockTime peer_latency;
350 /* some accounting */
354 guint64 num_duplicates;
355 guint64 num_rtx_requests;
356 guint64 num_rtx_success;
357 guint64 num_rtx_failed;
360 RTPPacketRateCtx packet_rate_ctx;
363 GstClockTime last_dts;
364 guint64 last_rtptime;
365 GstClockTime avg_jitter;
382 GstClockTime timeout;
383 GstClockTime duration;
384 GstClockTime rtx_base;
385 GstClockTime rtx_delay;
386 GstClockTime rtx_retry;
387 GstClockTime rtx_last;
389 guint num_rtx_received;
392 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
393 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
394 GstRtpJitterBufferPrivate))
396 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
397 GST_STATIC_PAD_TEMPLATE ("sink",
400 GST_STATIC_CAPS ("application/x-rtp"
401 /* "clock-rate = (int) [ 1, 2147483647 ], "
402 * "payload = (int) , "
403 * "encoding-name = (string) "
407 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
408 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
411 GST_STATIC_CAPS ("application/x-rtcp")
414 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
415 GST_STATIC_PAD_TEMPLATE ("src",
418 GST_STATIC_CAPS ("application/x-rtp"
419 /* "payload = (int) , "
420 * "clock-rate = (int) , "
421 * "encoding-name = (string) "
425 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
427 #define gst_rtp_jitter_buffer_parent_class parent_class
428 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
430 /* object overrides */
431 static void gst_rtp_jitter_buffer_set_property (GObject * object,
432 guint prop_id, const GValue * value, GParamSpec * pspec);
433 static void gst_rtp_jitter_buffer_get_property (GObject * object,
434 guint prop_id, GValue * value, GParamSpec * pspec);
435 static void gst_rtp_jitter_buffer_finalize (GObject * object);
437 /* element overrides */
438 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
439 * element, GstStateChange transition);
440 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
441 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
442 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
444 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
445 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
449 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
450 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
453 /* sinkpad overrides */
454 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
455 GstObject * parent, GstEvent * event);
456 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
457 GstObject * parent, GstBuffer * buffer);
459 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
460 GstObject * parent, GstEvent * event);
461 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
462 GstObject * parent, GstBuffer * buffer);
464 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
465 GstObject * parent, GstQuery * query);
467 /* srcpad overrides */
468 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
469 GstObject * parent, GstEvent * event);
470 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
471 GstObject * parent, GstPadMode mode, gboolean active);
472 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
473 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
474 GstObject * parent, GstQuery * query);
477 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
479 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
480 gboolean active, guint64 base_time);
481 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
483 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
484 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
486 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
488 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
491 static void update_rtx_stats (GstRtpJitterBuffer * jitterbuffer,
492 TimerData * timer, GstClockTime dts, gboolean success);
494 static TimerQueue *timer_queue_new (void);
495 static void timer_queue_free (TimerQueue * queue);
498 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
500 GObjectClass *gobject_class;
501 GstElementClass *gstelement_class;
503 gobject_class = (GObjectClass *) klass;
504 gstelement_class = (GstElementClass *) klass;
506 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
508 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
510 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
511 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
514 * GstRtpJitterBuffer:latency:
516 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
517 * for at most this time.
519 g_object_class_install_property (gobject_class, PROP_LATENCY,
520 g_param_spec_uint ("latency", "Buffer latency in ms",
521 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
522 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
524 * GstRtpJitterBuffer:drop-on-latency:
526 * Drop oldest buffers when the queue is completely filled.
528 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
529 g_param_spec_boolean ("drop-on-latency",
530 "Drop buffers when maximum latency is reached",
531 "Tells the jitterbuffer to never exceed the given latency in size",
532 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
534 * GstRtpJitterBuffer:ts-offset:
536 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
537 * This is mainly used to ensure interstream synchronisation.
539 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
540 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
541 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
542 G_MAXINT64, DEFAULT_TS_OFFSET,
543 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
546 * GstRtpJitterBuffer:do-lost:
548 * Send out a GstRTPPacketLost event downstream when a packet is considered
551 g_object_class_install_property (gobject_class, PROP_DO_LOST,
552 g_param_spec_boolean ("do-lost", "Do Lost",
553 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
554 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
557 * GstRtpJitterBuffer:mode:
559 * Control the buffering and timestamping mode used by the jitterbuffer.
561 g_object_class_install_property (gobject_class, PROP_MODE,
562 g_param_spec_enum ("mode", "Mode",
563 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
564 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
566 * GstRtpJitterBuffer:percent:
568 * The percent of the jitterbuffer that is filled.
570 g_object_class_install_property (gobject_class, PROP_PERCENT,
571 g_param_spec_int ("percent", "percent",
572 "The buffer filled percent", 0, 100,
573 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
575 * GstRtpJitterBuffer:do-retransmission:
577 * Send out a GstRTPRetransmission event upstream when a packet is considered
578 * late and should be retransmitted.
582 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
583 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
584 "Send retransmission events upstream when a packet is late",
585 DEFAULT_DO_RETRANSMISSION,
586 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
589 * GstRtpJitterBuffer:rtx-next-seqnum
591 * Estimate when the next packet should arrive and schedule a retransmission
593 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
594 * for packet N+1. So it will be requested if it does not arrive at the expected time.
595 * The expected time is calculated using the dts of N and the packet spacing.
599 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
600 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
601 "Estimate when the next packet should arrive and schedule a "
602 "retransmission request for it.",
603 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
606 * GstRtpJitterBuffer:rtx-delay:
608 * When a packet did not arrive at the expected time, wait this extra amount
609 * of time before sending a retransmission event.
611 * When -1 is used, the max jitter will be used as extra delay.
615 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
616 g_param_spec_int ("rtx-delay", "RTX Delay",
617 "Extra time in ms to wait before sending retransmission "
618 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
619 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
622 * GstRtpJitterBuffer:rtx-min-delay:
624 * When a packet did not arrive at the expected time, wait at least this extra amount
625 * of time before sending a retransmission event.
629 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
630 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
631 "Minimum time in ms to wait before sending retransmission "
632 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
633 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
635 * GstRtpJitterBuffer:rtx-delay-reorder:
637 * Assume that a retransmission event should be sent when we see
638 * this much packet reordering.
640 * When -1 is used, the value will be estimated based on observed packet
641 * reordering. When 0 is used packet reordering alone will not cause a
642 * retransmission event (Since 1.10).
646 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
647 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
648 "Sending retransmission event when this much reordering "
649 "(0 disable, -1 automatic)",
650 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
651 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
653 * GstRtpJitterBuffer::rtx-retry-timeout:
655 * When no packet has been received after sending a retransmission event
656 * for this time, retry sending a retransmission event.
658 * When -1 is used, the value will be estimated based on observed round
663 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
664 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
665 "Retry sending a transmission event after this timeout in "
666 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
667 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
669 * GstRtpJitterBuffer::rtx-min-retry-timeout:
671 * The minimum amount of time between retry timeouts. When
672 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
673 * minimum interval between retry timeouts.
675 * When -1 is used, the value will be estimated based on the
680 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
681 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
682 "Minimum timeout between sending a transmission event in "
683 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
684 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
686 * GstRtpJitterBuffer:rtx-retry-period:
688 * The amount of time to try to get a retransmission.
690 * When -1 is used, the value will be estimated based on the jitterbuffer
691 * latency and the observed round trip time.
695 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
696 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
697 "Try to get a retransmission for this many ms "
698 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
699 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
701 * GstRtpJitterBuffer:rtx-max-retries:
703 * The maximum number of retries to request a retransmission.
705 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
706 * When -1 is used, the number of retransmission request will not be limited.
710 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
711 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
712 "The maximum number of retries to request a retransmission. "
713 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
714 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
716 * GstRtpJitterBuffer::rtx-stats-timeout:
718 * The time to wait for a retransmitted packet after it has been
719 * considered lost in order to collect RTX statistics.
723 g_object_class_install_property (gobject_class, PROP_RTX_STATS_TIMEOUT,
724 g_param_spec_uint ("rtx-stats-timeout", "RTX Statistics Timeout",
725 "The time to wait for a retransmitted packet after it has been "
726 "considered lost in order to collect statistics (ms)",
727 0, G_MAXUINT, DEFAULT_RTX_STATS_TIMEOUT,
728 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
730 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
731 g_param_spec_uint ("max-dropout-time", "Max dropout time",
732 "The maximum time (milliseconds) of missing packets tolerated.",
733 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
734 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
736 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
737 g_param_spec_uint ("max-misorder-time", "Max misorder time",
738 "The maximum time (milliseconds) of misordered packets tolerated.",
739 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
740 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
742 * GstRtpJitterBuffer:stats:
744 * Various jitterbuffer statistics. This property returns a GstStructure
745 * with name application/x-rtp-jitterbuffer-stats with the following fields:
751 * <classname>"num-pushed"</classname>:
752 * the number of packets pushed out.
758 * <classname>"num-lost"</classname>:
759 * the number of packets considered lost.
765 * <classname>"num-late"</classname>:
766 * the number of packets arriving too late.
772 * <classname>"num-duplicates"</classname>:
773 * the number of duplicate packets.
779 * <classname>"rtx-count"</classname>:
780 * the number of retransmissions requested.
786 * <classname>"rtx-success-count"</classname>:
787 * the number of successful retransmissions.
793 * <classname>"rtx-per-packet"</classname>:
794 * average number of RTX per packet.
800 * <classname>"rtx-rtt"</classname>:
801 * average round trip time per RTX.
808 g_object_class_install_property (gobject_class, PROP_STATS,
809 g_param_spec_boxed ("stats", "Statistics",
810 "Various statistics", GST_TYPE_STRUCTURE,
811 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
814 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
816 * The maximum amount of time in ms that the RTP time in the RTCP SRs
817 * is allowed to be ahead of the last RTP packet we received. Use
818 * -1 to disable ignoring of RTCP packets.
822 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
823 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
824 "Maximum amount of time in ms that the RTP time in RTCP SRs "
825 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
826 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
827 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
829 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
830 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
831 "Synchronize received streams to the RFC7273 clock "
832 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
833 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
836 * GstRtpJitterBuffer::request-pt-map:
837 * @buffer: the object which received the signal
840 * Request the payload type as #GstCaps for @pt.
842 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
843 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
844 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
845 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
846 GST_TYPE_CAPS, 1, G_TYPE_UINT);
848 * GstRtpJitterBuffer::handle-sync:
849 * @buffer: the object which received the signal
850 * @struct: a GstStructure containing sync values.
852 * Be notified of new sync values.
854 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
855 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
856 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
857 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
858 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
861 * GstRtpJitterBuffer::on-npt-stop:
862 * @buffer: the object which received the signal
864 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
865 * the npt-stop position.
867 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
868 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
869 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
870 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
871 G_TYPE_NONE, 0, G_TYPE_NONE);
874 * GstRtpJitterBuffer::clear-pt-map:
875 * @buffer: the object which received the signal
877 * Invalidate the clock-rate as obtained with the
878 * #GstRtpJitterBuffer::request-pt-map signal.
880 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
881 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
882 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
883 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
884 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
887 * GstRtpJitterBuffer::set-active:
888 * @buffer: the object which received the signal
890 * Start pushing out packets with the given base time. This signal is only
891 * useful in buffering mode.
893 * Returns: the time of the last pushed packet.
895 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
896 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
897 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
898 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
899 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
902 gstelement_class->change_state =
903 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
904 gstelement_class->request_new_pad =
905 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
906 gstelement_class->release_pad =
907 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
908 gstelement_class->provide_clock =
909 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
910 gstelement_class->set_clock =
911 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
913 gst_element_class_add_static_pad_template (gstelement_class,
914 &gst_rtp_jitter_buffer_src_template);
915 gst_element_class_add_static_pad_template (gstelement_class,
916 &gst_rtp_jitter_buffer_sink_template);
917 gst_element_class_add_static_pad_template (gstelement_class,
918 &gst_rtp_jitter_buffer_sink_rtcp_template);
920 gst_element_class_set_static_metadata (gstelement_class,
921 "RTP packet jitter-buffer", "Filter/Network/RTP",
922 "A buffer that deals with network jitter and other transmission faults",
923 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
924 "Wim Taymans <wim.taymans@gmail.com>");
926 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
927 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
929 GST_DEBUG_CATEGORY_INIT
930 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
934 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
936 GstRtpJitterBufferPrivate *priv;
938 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
939 jitterbuffer->priv = priv;
941 priv->latency_ms = DEFAULT_LATENCY_MS;
942 priv->latency_ns = priv->latency_ms * GST_MSECOND;
943 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
944 priv->do_lost = DEFAULT_DO_LOST;
945 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
946 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
947 priv->rtx_delay = DEFAULT_RTX_DELAY;
948 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
949 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
950 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
951 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
952 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
953 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
954 priv->rtx_stats_timeout = DEFAULT_RTX_STATS_TIMEOUT;
955 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
956 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
957 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
960 priv->last_rtptime = -1;
961 priv->avg_jitter = 0;
962 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
963 priv->rtx_stats_timers = timer_queue_new ();
964 priv->jbuf = rtp_jitter_buffer_new ();
965 g_mutex_init (&priv->jbuf_lock);
966 g_cond_init (&priv->jbuf_timer);
967 g_cond_init (&priv->jbuf_event);
968 g_cond_init (&priv->jbuf_query);
969 g_queue_init (&priv->gap_packets);
970 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
972 /* reset skew detection initialy */
973 rtp_jitter_buffer_reset_skew (priv->jbuf);
974 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
975 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
979 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
982 gst_pad_set_activatemode_function (priv->srcpad,
983 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
984 gst_pad_set_query_function (priv->srcpad,
985 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
986 gst_pad_set_event_function (priv->srcpad,
987 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
990 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
993 gst_pad_set_chain_function (priv->sinkpad,
994 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
995 gst_pad_set_event_function (priv->sinkpad,
996 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
997 gst_pad_set_query_function (priv->sinkpad,
998 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
1000 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
1001 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
1003 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
1006 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
1008 #define ITEM_TYPE_BUFFER 0
1009 #define ITEM_TYPE_LOST 1
1010 #define ITEM_TYPE_EVENT 2
1011 #define ITEM_TYPE_QUERY 3
1013 static RTPJitterBufferItem *
1014 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
1015 guint seqnum, guint count, guint rtptime)
1017 RTPJitterBufferItem *item;
1019 item = g_slice_new (RTPJitterBufferItem);
1026 item->seqnum = seqnum;
1027 item->count = count;
1028 item->rtptime = rtptime;
1034 free_item (RTPJitterBufferItem * item)
1036 g_return_if_fail (item != NULL);
1038 if (item->data && item->type != ITEM_TYPE_QUERY)
1039 gst_mini_object_unref (item->data);
1040 g_slice_free (RTPJitterBufferItem, item);
1044 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
1046 GList **l = user_data;
1048 if (item->data && item->type == ITEM_TYPE_EVENT
1049 && GST_EVENT_IS_STICKY (item->data)) {
1050 *l = g_list_prepend (*l, item->data);
1051 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
1052 gst_mini_object_unref (item->data);
1054 g_slice_free (RTPJitterBufferItem, item);
1058 gst_rtp_jitter_buffer_finalize (GObject * object)
1060 GstRtpJitterBuffer *jitterbuffer;
1061 GstRtpJitterBufferPrivate *priv;
1063 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
1064 priv = jitterbuffer->priv;
1066 g_array_free (priv->timers, TRUE);
1067 timer_queue_free (priv->rtx_stats_timers);
1068 g_mutex_clear (&priv->jbuf_lock);
1069 g_cond_clear (&priv->jbuf_timer);
1070 g_cond_clear (&priv->jbuf_event);
1071 g_cond_clear (&priv->jbuf_query);
1073 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1074 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1075 g_queue_clear (&priv->gap_packets);
1076 g_object_unref (priv->jbuf);
1078 G_OBJECT_CLASS (parent_class)->finalize (object);
1081 static GstIterator *
1082 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
1084 GstRtpJitterBuffer *jitterbuffer;
1085 GstPad *otherpad = NULL;
1086 GstIterator *it = NULL;
1087 GValue val = { 0, };
1089 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1091 if (pad == jitterbuffer->priv->sinkpad) {
1092 otherpad = jitterbuffer->priv->srcpad;
1093 } else if (pad == jitterbuffer->priv->srcpad) {
1094 otherpad = jitterbuffer->priv->sinkpad;
1095 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1096 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1100 g_value_init (&val, GST_TYPE_PAD);
1101 g_value_set_object (&val, otherpad);
1102 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1103 g_value_unset (&val);
1110 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1112 GstRtpJitterBufferPrivate *priv;
1114 priv = jitterbuffer->priv;
1116 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1119 gst_pad_new_from_static_template
1120 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1121 gst_pad_set_chain_function (priv->rtcpsinkpad,
1122 gst_rtp_jitter_buffer_chain_rtcp);
1123 gst_pad_set_event_function (priv->rtcpsinkpad,
1124 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1125 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1126 gst_rtp_jitter_buffer_iterate_internal_links);
1127 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1128 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1130 return priv->rtcpsinkpad;
1134 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1136 GstRtpJitterBufferPrivate *priv;
1138 priv = jitterbuffer->priv;
1140 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1142 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1144 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1145 priv->rtcpsinkpad = NULL;
1149 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1150 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1152 GstRtpJitterBuffer *jitterbuffer;
1153 GstElementClass *klass;
1155 GstRtpJitterBufferPrivate *priv;
1157 g_return_val_if_fail (templ != NULL, NULL);
1158 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1160 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1161 priv = jitterbuffer->priv;
1162 klass = GST_ELEMENT_GET_CLASS (element);
1164 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1166 /* figure out the template */
1167 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1168 if (priv->rtcpsinkpad != NULL)
1171 result = create_rtcp_sink (jitterbuffer);
1173 goto wrong_template;
1180 g_warning ("rtpjitterbuffer: this is not our template");
1185 g_warning ("rtpjitterbuffer: pad already requested");
1191 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1193 GstRtpJitterBuffer *jitterbuffer;
1194 GstRtpJitterBufferPrivate *priv;
1196 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1197 g_return_if_fail (GST_IS_PAD (pad));
1199 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1200 priv = jitterbuffer->priv;
1202 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1204 if (priv->rtcpsinkpad == pad) {
1205 remove_rtcp_sink (jitterbuffer);
1214 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1220 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1222 return gst_system_clock_obtain ();
1226 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1228 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1230 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1232 return GST_ELEMENT_CLASS (parent_class)->set_clock (element, clock);
1236 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1238 GstRtpJitterBufferPrivate *priv;
1240 priv = jitterbuffer->priv;
1242 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1245 priv->clock_rate = -1;
1246 /* do not clear current content, but refresh state for new arrival */
1247 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1248 rtp_jitter_buffer_reset_skew (priv->jbuf);
1253 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1256 GstRtpJitterBufferPrivate *priv;
1257 GstClockTime last_out;
1258 RTPJitterBufferItem *item;
1263 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1264 active, GST_TIME_ARGS (offset));
1266 if (active != priv->active) {
1267 /* add the amount of time spent in paused to the output offset. All
1268 * outgoing buffers will have this offset applied to their timestamps in
1269 * order to make them arrive in time in the sink. */
1270 priv->out_offset = offset;
1271 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1272 GST_TIME_ARGS (priv->out_offset));
1273 priv->active = active;
1274 JBUF_SIGNAL_EVENT (priv);
1277 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1279 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1280 /* head buffer timestamp and offset gives our output time */
1281 last_out = item->dts + priv->ts_offset;
1283 /* use last known time when the buffer is empty */
1284 last_out = priv->last_out_time;
1292 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1294 GstRtpJitterBuffer *jitterbuffer;
1295 GstRtpJitterBufferPrivate *priv;
1300 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1301 priv = jitterbuffer->priv;
1303 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1305 caps = gst_pad_peer_query_caps (other, filter);
1307 templ = gst_pad_get_pad_template_caps (pad);
1309 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1314 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1316 intersect = gst_caps_intersect (caps, templ);
1317 gst_caps_unref (caps);
1318 gst_caps_unref (templ);
1322 gst_object_unref (jitterbuffer);
1328 * Must be called with JBUF_LOCK held
1332 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1333 GstCaps * caps, gint pt)
1335 GstRtpJitterBufferPrivate *priv;
1336 GstStructure *caps_struct;
1340 const gchar *ts_refclk, *mediaclk;
1342 priv = jitterbuffer->priv;
1344 /* first parse the caps */
1345 caps_struct = gst_caps_get_structure (caps, 0);
1347 GST_DEBUG_OBJECT (jitterbuffer, "got caps %" GST_PTR_FORMAT, caps);
1349 if (gst_structure_get_int (caps_struct, "payload", &payload) && pt != -1
1351 GST_ERROR_OBJECT (jitterbuffer,
1352 "Got caps with wrong payload type (got %d, expected %d)", payload, pt);
1356 if (payload != -1) {
1357 GST_DEBUG_OBJECT (jitterbuffer, "Got payload type %d", payload);
1358 priv->last_pt = payload;
1361 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1362 * measure the amount of data in the buffer */
1363 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1366 if (priv->clock_rate <= 0)
1369 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1371 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1373 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
1375 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1376 * can use this to track the amount of time elapsed on the sender. */
1377 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1378 priv->clock_base = val;
1380 priv->clock_base = -1;
1382 priv->ext_timestamp = priv->clock_base;
1384 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1387 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1388 /* first expected seqnum, only update when we didn't have a previous base. */
1389 if (priv->next_in_seqnum == -1)
1390 priv->next_in_seqnum = val;
1391 if (priv->next_seqnum == -1) {
1392 priv->next_seqnum = val;
1393 JBUF_SIGNAL_EVENT (priv);
1395 priv->seqnum_base = val;
1397 priv->seqnum_base = -1;
1400 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1402 /* the start and stop times. The seqnum-base corresponds to the start time. We
1403 * will keep track of the seqnums on the output and when we reach the one
1404 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1405 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1406 priv->npt_start = tval;
1408 priv->npt_start = 0;
1410 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1411 priv->npt_stop = tval;
1413 priv->npt_stop = -1;
1415 GST_DEBUG_OBJECT (jitterbuffer,
1416 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1417 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1419 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1420 GstClock *clock = NULL;
1421 guint64 clock_offset = -1;
1423 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1426 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1427 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1428 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1430 const gchar *host, *portstr;
1434 host = ts_refclk + sizeof ("ntp=") - 1;
1435 if (host[0] == '[') {
1437 portstr = strchr (host, ']');
1438 if (portstr && portstr[1] == ':')
1439 portstr = portstr + 1;
1443 portstr = strrchr (host, ':');
1447 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1451 hostname = g_strndup (host, (portstr - host));
1453 hostname = g_strdup (host);
1455 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1458 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1459 const gchar *domainstr =
1460 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1463 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1466 clock = gst_ptp_clock_new (NULL, domain);
1468 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1471 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1472 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1474 if (!g_str_has_prefix (mediaclk, "direct=")
1475 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1476 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1477 if (strstr (mediaclk, "rate=") != NULL) {
1478 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1483 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1485 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1493 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1498 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1504 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1506 GstRtpJitterBufferPrivate *priv;
1508 priv = jitterbuffer->priv;
1511 /* mark ourselves as flushing */
1512 priv->srcresult = GST_FLOW_FLUSHING;
1513 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1514 /* this unblocks any waiting pops on the src pad task */
1515 JBUF_SIGNAL_EVENT (priv);
1516 JBUF_SIGNAL_QUERY (priv, FALSE);
1521 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1523 GstRtpJitterBufferPrivate *priv;
1525 priv = jitterbuffer->priv;
1528 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1529 /* Mark as non flushing */
1530 priv->srcresult = GST_FLOW_OK;
1531 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1532 priv->last_popped_seqnum = -1;
1533 priv->last_out_time = -1;
1534 priv->next_seqnum = -1;
1535 priv->seqnum_base = -1;
1536 priv->ips_rtptime = -1;
1537 priv->ips_dts = GST_CLOCK_TIME_NONE;
1538 priv->packet_spacing = 0;
1539 priv->next_in_seqnum = -1;
1540 priv->clock_rate = -1;
1543 priv->estimated_eos = -1;
1544 priv->last_elapsed = 0;
1545 priv->ext_timestamp = -1;
1546 priv->avg_jitter = 0;
1547 priv->last_dts = -1;
1548 priv->last_rtptime = -1;
1549 priv->last_in_dts = 0;
1550 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1551 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1552 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1553 rtp_jitter_buffer_reset_skew (priv->jbuf);
1554 remove_all_timers (jitterbuffer);
1555 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1556 g_queue_clear (&priv->gap_packets);
1561 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1562 GstPadMode mode, gboolean active)
1565 GstRtpJitterBuffer *jitterbuffer = NULL;
1567 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1570 case GST_PAD_MODE_PUSH:
1572 /* allow data processing */
1573 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1575 /* start pushing out buffers */
1576 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1577 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1578 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1580 /* make sure all data processing stops ASAP */
1581 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1583 /* NOTE this will hardlock if the state change is called from the src pad
1584 * task thread because we will _join() the thread. */
1585 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1586 result = gst_pad_stop_task (pad);
1596 static GstStateChangeReturn
1597 gst_rtp_jitter_buffer_change_state (GstElement * element,
1598 GstStateChange transition)
1600 GstRtpJitterBuffer *jitterbuffer;
1601 GstRtpJitterBufferPrivate *priv;
1602 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1604 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1605 priv = jitterbuffer->priv;
1607 switch (transition) {
1608 case GST_STATE_CHANGE_NULL_TO_READY:
1610 case GST_STATE_CHANGE_READY_TO_PAUSED:
1612 /* reset negotiated values */
1613 priv->clock_rate = -1;
1614 priv->clock_base = -1;
1615 priv->peer_latency = 0;
1617 /* block until we go to PLAYING */
1618 priv->blocked = TRUE;
1619 priv->timer_running = TRUE;
1620 priv->timer_thread =
1621 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1624 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1626 /* unblock to allow streaming in PLAYING */
1627 priv->blocked = FALSE;
1628 JBUF_SIGNAL_EVENT (priv);
1629 JBUF_SIGNAL_TIMER (priv);
1636 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1638 switch (transition) {
1639 case GST_STATE_CHANGE_READY_TO_PAUSED:
1640 /* we are a live element because we sync to the clock, which we can only
1641 * do in the PLAYING state */
1642 if (ret != GST_STATE_CHANGE_FAILURE)
1643 ret = GST_STATE_CHANGE_NO_PREROLL;
1645 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1647 /* block to stop streaming when PAUSED */
1648 priv->blocked = TRUE;
1649 unschedule_current_timer (jitterbuffer);
1651 if (ret != GST_STATE_CHANGE_FAILURE)
1652 ret = GST_STATE_CHANGE_NO_PREROLL;
1654 case GST_STATE_CHANGE_PAUSED_TO_READY:
1656 gst_buffer_replace (&priv->last_sr, NULL);
1657 priv->timer_running = FALSE;
1658 unschedule_current_timer (jitterbuffer);
1659 JBUF_SIGNAL_TIMER (priv);
1660 JBUF_SIGNAL_QUERY (priv, FALSE);
1662 g_thread_join (priv->timer_thread);
1663 priv->timer_thread = NULL;
1665 case GST_STATE_CHANGE_READY_TO_NULL:
1675 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1678 gboolean ret = TRUE;
1679 GstRtpJitterBuffer *jitterbuffer;
1680 GstRtpJitterBufferPrivate *priv;
1682 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1683 priv = jitterbuffer->priv;
1685 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1687 switch (GST_EVENT_TYPE (event)) {
1688 case GST_EVENT_LATENCY:
1690 GstClockTime latency;
1692 gst_event_parse_latency (event, &latency);
1694 GST_DEBUG_OBJECT (jitterbuffer,
1695 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1698 /* adjust the overall buffer delay to the total pipeline latency in
1699 * buffering mode because if downstream consumes too fast (because of
1700 * large latency or queues, we would start rebuffering again. */
1701 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1702 RTP_JITTER_BUFFER_MODE_BUFFER) {
1703 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1707 ret = gst_pad_push_event (priv->sinkpad, event);
1711 ret = gst_pad_push_event (priv->sinkpad, event);
1718 /* handles and stores the event in the jitterbuffer, must be called with
1721 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1723 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1724 RTPJitterBufferItem *item;
1727 switch (GST_EVENT_TYPE (event)) {
1728 case GST_EVENT_CAPS:
1732 gst_event_parse_caps (event, &caps);
1733 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, -1);
1736 case GST_EVENT_SEGMENT:
1739 gst_event_copy_segment (event, &segment);
1741 /* we need time for now */
1742 if (segment.format != GST_FORMAT_TIME) {
1743 GST_DEBUG_OBJECT (jitterbuffer, "ignoring non-TIME newsegment");
1744 gst_event_unref (event);
1746 gst_segment_init (&segment, GST_FORMAT_TIME);
1747 event = gst_event_new_segment (&segment);
1750 priv->segment = segment;
1755 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1762 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1763 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1764 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
1766 JBUF_SIGNAL_EVENT (priv);
1772 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1775 gboolean ret = TRUE;
1776 GstRtpJitterBuffer *jitterbuffer;
1777 GstRtpJitterBufferPrivate *priv;
1779 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1780 priv = jitterbuffer->priv;
1782 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1784 switch (GST_EVENT_TYPE (event)) {
1785 case GST_EVENT_FLUSH_START:
1786 ret = gst_pad_push_event (priv->srcpad, event);
1787 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1788 /* wait for the loop to go into PAUSED */
1789 gst_pad_pause_task (priv->srcpad);
1791 case GST_EVENT_FLUSH_STOP:
1792 ret = gst_pad_push_event (priv->srcpad, event);
1794 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1795 GST_PAD_MODE_PUSH, TRUE);
1798 if (GST_EVENT_IS_SERIALIZED (event)) {
1799 /* serialized events go in the queue */
1801 if (priv->srcresult != GST_FLOW_OK) {
1802 /* Errors in sticky event pushing are no problem and ignored here
1803 * as they will cause more meaningful errors during data flow.
1804 * For EOS events, that are not followed by data flow, we still
1805 * return FALSE here though.
1807 if (!GST_EVENT_IS_STICKY (event) ||
1808 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1809 goto out_flow_error;
1811 /* refuse more events on EOS */
1814 ret = queue_event (jitterbuffer, event);
1817 /* non-serialized events are forwarded downstream immediately */
1818 ret = gst_pad_push_event (priv->srcpad, event);
1827 GST_DEBUG_OBJECT (jitterbuffer,
1828 "refusing event, we have a downstream flow error: %s",
1829 gst_flow_get_name (priv->srcresult));
1831 gst_event_unref (event);
1836 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1838 gst_event_unref (event);
1844 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1847 gboolean ret = TRUE;
1848 GstRtpJitterBuffer *jitterbuffer;
1850 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1852 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1854 switch (GST_EVENT_TYPE (event)) {
1855 case GST_EVENT_FLUSH_START:
1856 gst_event_unref (event);
1858 case GST_EVENT_FLUSH_STOP:
1859 gst_event_unref (event);
1862 ret = gst_pad_event_default (pad, parent, event);
1870 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1871 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1872 * GST_FLOW_FLUSHING when the element is shutting down. On success
1873 * GST_FLOW_OK is returned.
1875 static GstFlowReturn
1876 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1880 GValue args[2] = { {0}, {0} };
1884 g_value_init (&args[0], GST_TYPE_ELEMENT);
1885 g_value_set_object (&args[0], jitterbuffer);
1886 g_value_init (&args[1], G_TYPE_UINT);
1887 g_value_set_uint (&args[1], pt);
1889 g_value_init (&ret, GST_TYPE_CAPS);
1890 g_value_set_boxed (&ret, NULL);
1892 JBUF_UNLOCK (jitterbuffer->priv);
1893 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1895 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1897 g_value_unset (&args[0]);
1898 g_value_unset (&args[1]);
1899 caps = (GstCaps *) g_value_dup_boxed (&ret);
1900 g_value_unset (&ret);
1904 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
1905 gst_caps_unref (caps);
1907 if (G_UNLIKELY (!res))
1915 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1916 return GST_FLOW_ERROR;
1920 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1921 return GST_FLOW_FLUSHING;
1925 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1926 return GST_FLOW_ERROR;
1930 /* call with jbuf lock held */
1932 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1934 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1935 GstMessage *message = NULL;
1940 /* Post a buffering message */
1941 if (priv->last_percent != percent) {
1942 priv->last_percent = percent;
1944 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1945 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1952 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1954 GstRtpJitterBufferPrivate *priv;
1956 priv = jitterbuffer->priv;
1958 if (timestamp == -1)
1961 /* apply the timestamp offset, this is used for inter stream sync */
1962 timestamp += priv->ts_offset;
1963 /* add the offset, this is used when buffering */
1964 timestamp += priv->out_offset;
1970 timer_queue_new (void)
1974 queue = g_slice_new (TimerQueue);
1975 queue->timers = g_queue_new ();
1976 queue->hashtable = g_hash_table_new (NULL, NULL);
1982 timer_queue_free (TimerQueue * queue)
1987 g_hash_table_destroy (queue->hashtable);
1988 g_queue_free_full (queue->timers, g_free);
1989 g_slice_free (TimerQueue, queue);
1993 timer_queue_append (TimerQueue * queue, const TimerData * timer,
1994 GstClockTime timeout, gboolean lost)
1998 copy = g_memdup (timer, sizeof (*timer));
1999 copy->timeout = timeout;
2000 copy->type = lost ? TIMER_TYPE_LOST : TIMER_TYPE_EXPECTED;
2003 GST_LOG ("Append rtx-stats timer #%d, %" GST_TIME_FORMAT,
2004 copy->seqnum, GST_TIME_ARGS (copy->timeout));
2005 g_queue_push_tail (queue->timers, copy);
2006 g_hash_table_insert (queue->hashtable, GINT_TO_POINTER (copy->seqnum), copy);
2010 timer_queue_clear_until (TimerQueue * queue, GstClockTime timeout)
2014 test = g_queue_peek_head (queue->timers);
2015 while (test && test->timeout < timeout) {
2016 GST_LOG ("Pop rtx-stats timer #%d, %" GST_TIME_FORMAT " < %"
2017 GST_TIME_FORMAT, test->seqnum, GST_TIME_ARGS (test->timeout),
2018 GST_TIME_ARGS (timeout));
2019 g_hash_table_remove (queue->hashtable, GINT_TO_POINTER (test->seqnum));
2020 g_free (g_queue_pop_head (queue->timers));
2021 test = g_queue_peek_head (queue->timers);
2026 timer_queue_find (TimerQueue * queue, guint16 seqnum)
2028 return g_hash_table_lookup (queue->hashtable, GINT_TO_POINTER (seqnum));
2032 find_timer (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2034 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2035 TimerData *timer = NULL;
2038 len = priv->timers->len;
2039 for (i = 0; i < len; i++) {
2040 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2041 if (test->seqnum == seqnum) {
2050 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
2052 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2054 if (priv->clock_id) {
2055 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
2056 gst_clock_id_unschedule (priv->clock_id);
2057 priv->clock_id = NULL;
2062 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2064 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2065 GstClockTime test_timeout;
2067 if ((test_timeout = timer->timeout) == -1)
2070 if (timer->type != TIMER_TYPE_EXPECTED) {
2071 /* add our latency and offset to get output times. */
2072 test_timeout = apply_offset (jitterbuffer, test_timeout);
2073 test_timeout += priv->latency_ns;
2075 return test_timeout;
2079 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2081 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2083 if (priv->clock_id) {
2084 GstClockTime timeout = get_timeout (jitterbuffer, timer);
2086 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
2087 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
2089 if (timeout == -1 || timeout < priv->timer_timeout)
2090 unschedule_current_timer (jitterbuffer);
2095 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2096 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
2097 GstClockTime duration)
2099 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2103 GST_DEBUG_OBJECT (jitterbuffer,
2104 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
2105 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
2106 GST_TIME_ARGS (delay));
2108 len = priv->timers->len;
2109 g_array_set_size (priv->timers, len + 1);
2110 timer = &g_array_index (priv->timers, TimerData, len);
2113 timer->seqnum = seqnum;
2115 timer->timeout = timeout + delay;
2116 timer->duration = duration;
2117 if (type == TIMER_TYPE_EXPECTED) {
2118 timer->rtx_base = timeout;
2119 timer->rtx_delay = delay;
2120 timer->rtx_retry = 0;
2122 timer->rtx_last = GST_CLOCK_TIME_NONE;
2123 timer->num_rtx_retry = 0;
2124 timer->num_rtx_received = 0;
2125 recalculate_timer (jitterbuffer, timer);
2126 JBUF_SIGNAL_TIMER (priv);
2132 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2133 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
2135 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2136 gboolean seqchange, timechange;
2139 seqchange = timer->seqnum != seqnum;
2140 timechange = timer->timeout != timeout;
2142 if (!seqchange && !timechange)
2145 oldseq = timer->seqnum;
2147 GST_DEBUG_OBJECT (jitterbuffer,
2148 "replace timer %d for seqnum %d->%d timeout %" GST_TIME_FORMAT
2149 "->%" GST_TIME_FORMAT, timer->type, oldseq, seqnum,
2150 GST_TIME_ARGS (timer->timeout), GST_TIME_ARGS (timeout + delay));
2152 timer->timeout = timeout + delay;
2153 timer->seqnum = seqnum;
2155 GST_DEBUG_OBJECT (jitterbuffer, "reset rtx delay %" GST_TIME_FORMAT
2156 "->%" GST_TIME_FORMAT, GST_TIME_ARGS (timer->rtx_delay),
2157 GST_TIME_ARGS (delay));
2158 timer->rtx_base = timeout;
2159 timer->rtx_delay = delay;
2160 timer->rtx_retry = 0;
2163 timer->num_rtx_retry = 0;
2164 timer->num_rtx_received = 0;
2167 if (priv->clock_id) {
2168 /* we changed the seqnum and there is a timer currently waiting with this
2169 * seqnum, unschedule it */
2170 if (seqchange && priv->timer_seqnum == oldseq)
2171 unschedule_current_timer (jitterbuffer);
2172 /* we changed the time, check if it is earlier than what we are waiting
2173 * for and unschedule if so */
2174 else if (timechange)
2175 recalculate_timer (jitterbuffer, timer);
2180 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2181 guint16 seqnum, GstClockTime timeout)
2185 /* find the seqnum timer */
2186 timer = find_timer (jitterbuffer, seqnum);
2187 if (timer == NULL) {
2188 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2190 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2196 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2198 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2201 if (timer->idx == -1)
2204 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2205 unschedule_current_timer (jitterbuffer);
2208 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2209 g_array_remove_index_fast (priv->timers, idx);
2214 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2216 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2217 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2218 g_array_set_size (priv->timers, 0);
2219 unschedule_current_timer (jitterbuffer);
2222 /* get the extra delay to wait before sending RTX */
2224 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2228 if (priv->rtx_delay == -1) {
2229 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2230 delay = DEFAULT_AUTO_RTX_DELAY;
2232 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2233 * packet spacing is a good margin */
2234 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2237 delay = priv->rtx_delay * GST_MSECOND;
2239 if (priv->rtx_min_delay > 0)
2240 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2245 /* Check if packet with seqnum is already considered definitely lost by being
2246 * part of a "lost timer" for multiple packets */
2248 already_lost (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum)
2250 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2253 len = priv->timers->len;
2254 for (i = 0; i < len; i++) {
2255 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2256 gint gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2258 if (test->num > 1 && test->type == TIMER_TYPE_LOST && gap >= 0 &&
2260 GST_DEBUG ("seqnum #%d already considered definitely lost (#%d->#%d)",
2261 seqnum, test->seqnum, (test->seqnum + test->num - 1) & 0xffff);
2269 /* we just received a packet with seqnum and dts.
2271 * First check for old seqnum that we are still expecting. If the gap with the
2272 * current seqnum is too big, unschedule the timeouts.
2274 * If we have a valid packet spacing estimate we can set a timer for when we
2275 * should receive the next packet.
2276 * If we don't have a valid estimate, we remove any timer we might have
2277 * had for this packet.
2280 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2281 GstClockTime dts, gboolean do_next_seqnum, gboolean is_rtx,
2284 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2286 /* go through all timers and unschedule the ones with a large gap */
2287 if (priv->do_retransmission && priv->rtx_delay_reorder > 0) {
2289 len = priv->timers->len;
2290 for (i = 0; i < len; i++) {
2291 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2294 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2296 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d",
2297 test->type, test->seqnum, seqnum, gap);
2299 if (gap > priv->rtx_delay_reorder) {
2300 /* max gap, we exceeded the max reorder distance and we don't expect the
2301 * missing packet to be this reordered */
2302 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2303 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2308 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2309 && priv->do_retransmission && priv->rtx_next_seqnum;
2311 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2312 if (timer->num_rtx_retry > 0) {
2314 update_rtx_stats (jitterbuffer, timer, dts, TRUE);
2315 /* don't try to estimate the next seqnum because this is a retransmitted
2316 * packet and it probably did not arrive with the expected packet
2318 do_next_seqnum = FALSE;
2321 if (!is_rtx || timer->num_rtx_retry > 1) {
2322 /* Store timer in order to record stats when/if the retransmitted
2323 * packet arrives. We should also store timer information if we've
2324 * requested retransmission more than once since we may receive
2325 * several retransmitted packets. For accuracy we should update the
2326 * stats also when the redundant retransmitted packets arrives. */
2327 timer_queue_append (priv->rtx_stats_timers, timer,
2328 dts + priv->rtx_stats_timeout * GST_MSECOND, FALSE);
2333 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2334 GstClockTime expected, delay;
2336 /* calculate expected arrival time of the next seqnum */
2337 expected = dts + priv->packet_spacing;
2339 delay = get_rtx_delay (priv);
2341 /* and update/install timer for next seqnum */
2342 GST_DEBUG_OBJECT (jitterbuffer, "Add RTX timer #%d, expected %"
2343 GST_TIME_FORMAT ", delay %" GST_TIME_FORMAT ", packet-spacing %"
2344 GST_TIME_FORMAT ", jitter %" GST_TIME_FORMAT, priv->next_in_seqnum,
2345 GST_TIME_ARGS (expected), GST_TIME_ARGS (delay),
2346 GST_TIME_ARGS (priv->packet_spacing), GST_TIME_ARGS (priv->avg_jitter));
2349 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2352 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2353 expected, delay, priv->packet_spacing);
2355 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2356 /* if we had a timer, remove it, we don't know when to expect the next
2358 remove_timer (jitterbuffer, timer);
2363 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2366 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2368 /* we need consecutive seqnums with a different
2369 * rtptime to estimate the packet spacing. */
2370 if (priv->ips_rtptime != rtptime) {
2371 /* rtptime changed, check dts diff */
2372 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2373 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2374 GstClockTime old_packet_spacing = priv->packet_spacing;
2376 /* Biased towards bigger packet spacings to prevent
2377 * too many unneeded retransmission requests for next
2378 * packets that just arrive a little later than we would
2380 if (old_packet_spacing > new_packet_spacing)
2381 priv->packet_spacing =
2382 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2383 else if (old_packet_spacing > 0)
2384 priv->packet_spacing =
2385 (3 * new_packet_spacing + old_packet_spacing) / 4;
2387 priv->packet_spacing = new_packet_spacing;
2389 GST_DEBUG_OBJECT (jitterbuffer,
2390 "new packet spacing %" GST_TIME_FORMAT
2391 " old packet spacing %" GST_TIME_FORMAT
2392 " combined to %" GST_TIME_FORMAT,
2393 GST_TIME_ARGS (new_packet_spacing),
2394 GST_TIME_ARGS (old_packet_spacing),
2395 GST_TIME_ARGS (priv->packet_spacing));
2397 priv->ips_rtptime = rtptime;
2398 priv->ips_dts = dts;
2403 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2404 guint16 seqnum, GstClockTime dts, gint gap)
2406 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2407 GstClockTime total_duration, duration, expected_dts, delay;
2410 GST_DEBUG_OBJECT (jitterbuffer,
2411 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2412 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2414 if (dts == GST_CLOCK_TIME_NONE) {
2415 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2419 /* the total duration spanned by the missing packets */
2420 if (dts >= priv->last_in_dts)
2421 total_duration = dts - priv->last_in_dts;
2425 /* interpolate between the current time and the last time based on
2426 * number of packets we are missing, this is the estimated duration
2427 * for the missing packet based on equidistant packet spacing. */
2428 duration = total_duration / (gap + 1);
2430 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2431 GST_TIME_ARGS (duration));
2433 if (total_duration > priv->latency_ns) {
2434 GstClockTime gap_time;
2438 GstClockTime gap_dur = gap * duration;
2439 if (gap_dur > priv->latency_ns)
2440 gap_time = gap_dur - priv->latency_ns;
2443 lost_packets = gap_time / duration;
2445 gap_time = total_duration - priv->latency_ns;
2449 /* too many lost packets, some of the missing packets are already
2450 * too late and we can generate lost packet events for them. */
2451 GST_DEBUG_OBJECT (jitterbuffer,
2452 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2453 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2454 gap, expected, seqnum - 1, GST_TIME_ARGS (total_duration),
2455 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2456 GST_TIME_ARGS (gap_time));
2458 /* this timer will fire immediately and the lost event will be pushed from
2459 * the timer thread */
2460 if (lost_packets > 0) {
2461 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2462 priv->last_in_dts + duration, 0, gap_time);
2463 expected += lost_packets;
2464 priv->last_in_dts += gap_time;
2468 expected_dts = priv->last_in_dts + duration;
2471 if (priv->do_retransmission) {
2472 TimerData *timer = find_timer (jitterbuffer, expected);
2474 type = TIMER_TYPE_EXPECTED;
2475 delay = get_rtx_delay (priv);
2477 /* if we had a timer for the first missing packet, update it. */
2478 if (timer && timer->type == TIMER_TYPE_EXPECTED) {
2479 GstClockTime timeout = timer->timeout;
2481 timer->duration = duration;
2482 if (timeout > (expected_dts + delay) && timer->num_rtx_retry == 0) {
2483 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2487 expected_dts += duration;
2490 type = TIMER_TYPE_LOST;
2493 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2494 add_timer (jitterbuffer, type, expected, 0, expected_dts, delay, duration);
2495 expected_dts += duration;
2501 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2505 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2506 GstRtpJitterBufferPrivate *priv;
2508 priv = jitterbuffer->priv;
2510 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2513 if (priv->last_dts != -1)
2514 dtsdiff = dts - priv->last_dts;
2518 if (priv->last_rtptime != -1)
2519 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2523 priv->last_dts = dts;
2524 priv->last_rtptime = rtptime;
2528 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2531 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2533 diff = ABS (dtsdiff - rtpdiffns);
2535 /* jitter is stored in nanoseconds */
2536 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2538 GST_LOG_OBJECT (jitterbuffer,
2539 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2540 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2541 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2542 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2549 GST_DEBUG_OBJECT (jitterbuffer,
2550 "no dts or no clock-rate, can't calculate jitter");
2556 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2558 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2559 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2562 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2563 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2564 gst_rtp_buffer_unmap (&rtp_a);
2566 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2567 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2568 gst_rtp_buffer_unmap (&rtp_b);
2570 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2574 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2575 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2578 GstRtpJitterBufferPrivate *priv;
2579 guint gap_packets_length;
2580 gboolean reset = FALSE;
2582 priv = jitterbuffer->priv;
2584 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2586 guint32 prev_gap_seq = -1;
2587 gboolean all_consecutive = TRUE;
2589 g_queue_insert_sorted (&priv->gap_packets, buffer,
2590 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2592 for (l = priv->gap_packets.head; l; l = l->next) {
2593 GstBuffer *gap_buffer = l->data;
2594 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2597 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2599 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2601 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2602 if (prev_gap_seq == -1)
2603 prev_gap_seq = gap_seq;
2604 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2605 all_consecutive = FALSE;
2607 prev_gap_seq = gap_seq;
2609 gst_rtp_buffer_unmap (&gap_rtp);
2610 if (!all_consecutive)
2614 if (all_consecutive && gap_packets_length > 3) {
2615 GST_DEBUG_OBJECT (jitterbuffer,
2616 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2617 (future ? "new" : "old"), gap,
2618 (future ? max_dropout : -max_misorder));
2620 } else if (!all_consecutive) {
2621 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2622 g_queue_clear (&priv->gap_packets);
2623 GST_DEBUG_OBJECT (jitterbuffer,
2624 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2625 (future ? "new" : "old"), gap,
2626 (future ? max_dropout : -max_misorder));
2629 GST_DEBUG_OBJECT (jitterbuffer,
2630 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2631 (future ? "new" : "old"), gap,
2632 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2636 GST_DEBUG_OBJECT (jitterbuffer,
2637 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2638 gap, -max_misorder);
2639 g_queue_push_tail (&priv->gap_packets, buffer);
2647 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2649 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2650 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2653 GstClockTime base_time =
2654 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2655 GstClockTime clock_time = gst_clock_get_time (clock);
2657 if (clock_time > base_time)
2658 running_time = clock_time - base_time;
2662 gst_object_unref (clock);
2665 return running_time;
2668 static GstFlowReturn
2669 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2672 GstRtpJitterBuffer *jitterbuffer;
2673 GstRtpJitterBufferPrivate *priv;
2675 guint32 expected, rtptime;
2676 GstFlowReturn ret = GST_FLOW_OK;
2677 GstClockTime dts, pts;
2682 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2683 gboolean do_next_seqnum = FALSE;
2684 RTPJitterBufferItem *item;
2685 GstMessage *msg = NULL;
2686 gboolean estimated_dts = FALSE;
2687 guint32 packet_rate, max_dropout, max_misorder;
2688 TimerData *timer = NULL;
2690 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2692 priv = jitterbuffer->priv;
2694 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2695 goto invalid_buffer;
2697 pt = gst_rtp_buffer_get_payload_type (&rtp);
2698 seqnum = gst_rtp_buffer_get_seq (&rtp);
2699 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2700 gst_rtp_buffer_unmap (&rtp);
2702 /* make sure we have PTS and DTS set */
2703 pts = GST_BUFFER_PTS (buffer);
2704 dts = GST_BUFFER_DTS (buffer);
2711 /* If we have no DTS here, i.e. no capture time, get one from the
2712 * clock now to have something to calculate with in the future. */
2713 dts = get_current_running_time (jitterbuffer);
2716 /* Remember that we estimated the DTS if we are running already
2717 * and this is not our first packet (or first packet after a reset).
2718 * If it's the first packet, we somehow must generate a timestamp for
2719 * everything, otherwise we can't calculate any times
2721 estimated_dts = (priv->next_in_seqnum != -1);
2723 /* take the DTS of the buffer. This is the time when the packet was
2724 * received and is used to calculate jitter and clock skew. We will adjust
2725 * this DTS with the smoothed value after processing it in the
2726 * jitterbuffer and assign it as the PTS. */
2727 /* bring to running time */
2728 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2731 GST_DEBUG_OBJECT (jitterbuffer,
2732 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d, rtx %d",
2733 seqnum, GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer),
2734 GST_BUFFER_IS_RETRANSMISSION (buffer));
2736 JBUF_LOCK_CHECK (priv, out_flushing);
2738 if (G_UNLIKELY (priv->last_pt != pt)) {
2741 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2745 /* reset clock-rate so that we get a new one */
2746 priv->clock_rate = -1;
2748 /* Try to get the clock-rate from the caps first if we can. If there are no
2749 * caps we must fire the signal to get the clock-rate. */
2750 if ((caps = gst_pad_get_current_caps (pad))) {
2751 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps, pt);
2752 gst_caps_unref (caps);
2756 if (G_UNLIKELY (priv->clock_rate == -1)) {
2757 /* no clock rate given on the caps, try to get one with the signal */
2758 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2759 pt) == GST_FLOW_FLUSHING)
2762 if (G_UNLIKELY (priv->clock_rate == -1))
2765 gst_rtp_packet_rate_ctx_reset (&priv->packet_rate_ctx, priv->clock_rate);
2768 /* don't accept more data on EOS */
2769 if (G_UNLIKELY (priv->eos))
2772 if (!GST_BUFFER_IS_RETRANSMISSION (buffer))
2773 calculate_jitter (jitterbuffer, dts, rtptime);
2775 if (priv->seqnum_base != -1) {
2778 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2781 GST_DEBUG_OBJECT (jitterbuffer,
2782 "packet seqnum #%d before seqnum-base #%d", seqnum,
2784 gst_buffer_unref (buffer);
2787 } else if (gap > 16384) {
2788 /* From now on don't compare against the seqnum base anymore as
2789 * at some point in the future we will wrap around and also that
2790 * much reordering is very unlikely */
2791 priv->seqnum_base = -1;
2795 expected = priv->next_in_seqnum;
2798 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2800 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2801 priv->max_dropout_time);
2803 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2804 priv->max_misorder_time);
2805 GST_TRACE_OBJECT (jitterbuffer,
2806 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2807 max_dropout, max_misorder);
2809 /* now check against our expected seqnum */
2810 if (G_LIKELY (expected != -1)) {
2813 /* now calculate gap */
2814 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2816 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2817 expected, seqnum, gap);
2819 if (G_LIKELY (gap == 0)) {
2820 /* packet is expected */
2821 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2822 do_next_seqnum = TRUE;
2824 gboolean reset = FALSE;
2827 /* we received an old packet */
2828 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2830 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2831 gap, max_dropout, max_misorder);
2834 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2837 /* new packet, we are missing some packets */
2838 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2839 /* If we have timers for more than RTP_MAX_DROPOUT packets
2840 * pending this means that we have a huge gap overall. We can
2841 * reset the jitterbuffer at this point because there's
2842 * just too much data missing to be able to do anything
2843 * sensible with the past data. Just try again from the
2845 GST_WARNING_OBJECT (jitterbuffer,
2846 "%d pending timers > %d - resetting", priv->timers->len,
2849 gst_buffer_unref (buffer);
2851 } else if (G_UNLIKELY (gap >= max_dropout)) {
2853 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2854 gap, max_dropout, max_misorder);
2857 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2858 /* fill in the gap with EXPECTED timers */
2859 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2861 do_next_seqnum = TRUE;
2864 if (G_UNLIKELY (reset)) {
2865 GList *events = NULL, *l;
2868 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2869 rtp_jitter_buffer_flush (priv->jbuf,
2870 (GFunc) free_item_and_retain_events, &events);
2871 rtp_jitter_buffer_reset_skew (priv->jbuf);
2872 remove_all_timers (jitterbuffer);
2873 priv->discont = TRUE;
2874 priv->last_popped_seqnum = -1;
2876 if (priv->gap_packets.head) {
2877 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2878 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2880 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2881 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2882 gst_rtp_buffer_unmap (&gap_rtp);
2884 priv->next_seqnum = seqnum;
2887 priv->last_in_dts = -1;
2888 priv->next_in_seqnum = -1;
2890 /* Insert all sticky events again in order, otherwise we would
2891 * potentially loose STREAM_START, CAPS or SEGMENT events
2893 events = g_list_reverse (events);
2894 for (l = events; l; l = l->next) {
2895 RTPJitterBufferItem *item;
2897 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2898 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
2900 g_list_free (events);
2902 JBUF_SIGNAL_EVENT (priv);
2904 /* reset spacing estimation when gap */
2905 priv->ips_rtptime = -1;
2906 priv->ips_dts = GST_CLOCK_TIME_NONE;
2908 buffers = g_list_copy (priv->gap_packets.head);
2909 g_queue_clear (&priv->gap_packets);
2911 priv->ips_rtptime = -1;
2912 priv->ips_dts = GST_CLOCK_TIME_NONE;
2913 JBUF_UNLOCK (jitterbuffer->priv);
2915 for (l = buffers; l; l = l->next) {
2916 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2918 if (ret != GST_FLOW_OK) {
2923 for (; l; l = l->next)
2924 gst_buffer_unref (l->data);
2925 g_list_free (buffers);
2929 /* reset spacing estimation when gap */
2930 priv->ips_rtptime = -1;
2931 priv->ips_dts = GST_CLOCK_TIME_NONE;
2934 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2936 /* we don't know what the next_in_seqnum should be, wait for the last
2937 * possible moment to push this buffer, maybe we get an earlier seqnum
2939 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2940 do_next_seqnum = TRUE;
2941 /* take rtptime and dts to calculate packet spacing */
2942 priv->ips_rtptime = rtptime;
2943 priv->ips_dts = dts;
2946 /* We had no huge gap, let's drop all the gap packets */
2947 if (buffer != NULL) {
2948 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2949 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2950 g_queue_clear (&priv->gap_packets);
2952 GST_DEBUG_OBJECT (jitterbuffer,
2953 "Had big gap, waiting for more consecutive packets");
2954 JBUF_UNLOCK (jitterbuffer->priv);
2958 if (do_next_seqnum) {
2959 priv->last_in_dts = dts;
2960 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2963 timer = find_timer (jitterbuffer, seqnum);
2964 if (GST_BUFFER_IS_RETRANSMISSION (buffer)) {
2966 timer = timer_queue_find (priv->rtx_stats_timers, seqnum);
2968 timer->num_rtx_received++;
2971 /* let's check if this buffer is too late, we can only accept packets with
2972 * bigger seqnum than the one we last pushed. */
2973 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2976 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2978 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2979 if (G_UNLIKELY (gap <= 0)) {
2980 if (priv->do_retransmission) {
2981 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer) {
2982 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
2983 /* Only count the retranmitted packet too late if it has been
2984 * considered lost. If the original packet arrived before the
2985 * retransmitted we just count it as a duplicate. */
2986 if (timer->type != TIMER_TYPE_LOST)
2994 if (already_lost (jitterbuffer, seqnum))
2997 /* let's drop oldest packet if the queue is already full and drop-on-latency
2998 * is set. We can only do this when there actually is a latency. When no
2999 * latency is set, we just pump it in the queue and let the other end push it
3000 * out as fast as possible. */
3001 if (priv->latency_ms && priv->drop_on_latency) {
3003 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
3005 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
3006 RTPJitterBufferItem *old_item;
3008 old_item = rtp_jitter_buffer_peek (priv->jbuf);
3010 if (IS_DROPABLE (old_item)) {
3011 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3012 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
3014 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
3015 free_item (old_item);
3017 /* we might have removed some head buffers, signal the pushing thread to
3018 * see if it can push now */
3019 JBUF_SIGNAL_EVENT (priv);
3023 /* If we estimated the DTS, don't consider it in the clock skew calculations
3024 * later. The code above always sets dts to pts or the other way around if
3025 * any of those is valid in the buffer, so we know that if we estimated the
3026 * dts that both are unknown */
3029 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
3030 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
3032 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
3034 /* now insert the packet into the queue in sorted order. This function returns
3035 * FALSE if a packet with the same seqnum was already in the queue, meaning we
3036 * have a duplicate. */
3037 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
3039 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer))))) {
3040 if (GST_BUFFER_IS_RETRANSMISSION (buffer) && timer)
3041 update_rtx_stats (jitterbuffer, timer, dts, FALSE);
3046 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum,
3047 GST_BUFFER_IS_RETRANSMISSION (buffer), timer);
3049 /* we had an unhandled SR, handle it now */
3051 do_handle_sync (jitterbuffer);
3053 if (G_UNLIKELY (head)) {
3054 /* signal addition of new buffer when the _loop is waiting. */
3055 if (G_LIKELY (priv->active))
3056 JBUF_SIGNAL_EVENT (priv);
3058 /* let's unschedule and unblock any waiting buffers. We only want to do this
3059 * when the head buffer changed */
3060 if (G_UNLIKELY (priv->clock_id)) {
3061 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
3062 unschedule_current_timer (jitterbuffer);
3066 GST_DEBUG_OBJECT (jitterbuffer,
3067 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
3068 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
3070 msg = check_buffering_percent (jitterbuffer, percent);
3076 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3083 /* this is not fatal but should be filtered earlier */
3084 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3085 ("Received invalid RTP payload, dropping"));
3086 gst_buffer_unref (buffer);
3091 GST_WARNING_OBJECT (jitterbuffer,
3092 "No clock-rate in caps!, dropping buffer");
3093 gst_buffer_unref (buffer);
3098 ret = priv->srcresult;
3099 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
3100 gst_buffer_unref (buffer);
3106 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
3107 gst_buffer_unref (buffer);
3112 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
3113 " popped, dropping", seqnum, priv->last_popped_seqnum);
3115 gst_buffer_unref (buffer);
3120 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as it was already "
3121 "considered lost", seqnum);
3123 gst_buffer_unref (buffer);
3128 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
3130 priv->num_duplicates++;
3136 GST_DEBUG_OBJECT (jitterbuffer,
3137 "Duplicate RTX packet #%d detected, dropping", seqnum);
3138 priv->num_duplicates++;
3139 gst_buffer_unref (buffer);
3145 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
3147 guint64 ext_time, elapsed;
3149 GstRtpJitterBufferPrivate *priv;
3151 priv = jitterbuffer->priv;
3152 rtp_time = item->rtptime;
3154 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
3155 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
3157 ext_time = priv->ext_timestamp;
3158 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
3159 if (ext_time < priv->ext_timestamp) {
3160 ext_time = priv->ext_timestamp;
3162 priv->ext_timestamp = ext_time;
3165 if (ext_time > priv->clock_base)
3166 elapsed = ext_time - priv->clock_base;
3170 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
3175 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
3176 RTPJitterBufferItem * item)
3178 guint64 total, elapsed, left, estimated;
3179 GstClockTime out_time;
3180 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3182 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
3183 || priv->clock_base == -1 || priv->clock_rate <= 0)
3186 /* compute the elapsed time */
3187 elapsed = compute_elapsed (jitterbuffer, item);
3189 /* do nothing if elapsed time doesn't increment */
3190 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
3193 priv->last_elapsed = elapsed;
3195 /* this is the total time we need to play */
3196 total = priv->npt_stop - priv->npt_start;
3197 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
3198 GST_TIME_ARGS (total));
3200 /* this is how much time there is left */
3201 if (total > elapsed)
3202 left = total - elapsed;
3206 /* if we have less time left that the size of the buffer, we will not
3207 * be able to keep it filled, disabled buffering then */
3208 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
3209 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
3210 ", disable buffering close to EOS", GST_TIME_ARGS (left));
3211 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
3214 /* this is the current time as running-time */
3215 out_time = item->dts;
3218 estimated = gst_util_uint64_scale (out_time, total, elapsed);
3220 /* if there is almost nothing left,
3221 * we may never advance enough to end up in the above case */
3222 if (total < GST_SECOND)
3223 estimated = GST_SECOND;
3227 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
3228 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3230 if (estimated != -1 && priv->estimated_eos != estimated) {
3231 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3232 priv->estimated_eos = estimated;
3236 /* take a buffer from the queue and push it */
3237 static GstFlowReturn
3238 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3240 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3241 GstFlowReturn result = GST_FLOW_OK;
3242 RTPJitterBufferItem *item;
3243 GstBuffer *outbuf = NULL;
3244 GstEvent *outevent = NULL;
3245 GstQuery *outquery = NULL;
3246 GstClockTime dts, pts;
3248 gboolean do_push = TRUE;
3252 /* when we get here we are ready to pop and push the buffer */
3253 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3257 case ITEM_TYPE_BUFFER:
3259 /* we need to make writable to change the flags and timestamps */
3260 outbuf = gst_buffer_make_writable (item->data);
3262 if (G_UNLIKELY (priv->discont)) {
3263 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3264 * into the jitterbuffer so we can modify now. */
3265 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3266 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3267 priv->discont = FALSE;
3269 if (G_UNLIKELY (priv->ts_discont)) {
3270 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3271 priv->ts_discont = FALSE;
3275 gst_segment_position_from_running_time (&priv->segment,
3276 GST_FORMAT_TIME, item->dts);
3278 gst_segment_position_from_running_time (&priv->segment,
3279 GST_FORMAT_TIME, item->pts);
3281 /* apply timestamp with offset to buffer now */
3282 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3283 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3285 /* update the elapsed time when we need to check against the npt stop time. */
3286 update_estimated_eos (jitterbuffer, item);
3288 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3290 case ITEM_TYPE_LOST:
3291 priv->discont = TRUE;
3295 case ITEM_TYPE_EVENT:
3296 outevent = item->data;
3298 case ITEM_TYPE_QUERY:
3299 outquery = item->data;
3303 /* now we are ready to push the buffer. Save the seqnum and release the lock
3304 * so the other end can push stuff in the queue again. */
3306 priv->last_popped_seqnum = seqnum;
3307 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3309 msg = check_buffering_percent (jitterbuffer, percent);
3316 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3319 case ITEM_TYPE_BUFFER:
3321 GST_DEBUG_OBJECT (jitterbuffer,
3322 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3323 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3324 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3326 result = gst_pad_push (priv->srcpad, outbuf);
3328 JBUF_LOCK_CHECK (priv, out_flushing);
3330 case ITEM_TYPE_LOST:
3331 case ITEM_TYPE_EVENT:
3332 /* We got not enough consecutive packets with a huge gap, we can
3333 * as well just drop them here now on EOS */
3334 if (outevent && GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3335 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3336 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3337 g_queue_clear (&priv->gap_packets);
3340 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3341 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3344 gst_pad_push_event (priv->srcpad, outevent);
3346 gst_event_unref (outevent);
3348 result = GST_FLOW_OK;
3350 JBUF_LOCK_CHECK (priv, out_flushing);
3352 case ITEM_TYPE_QUERY:
3356 res = gst_pad_peer_query (priv->srcpad, outquery);
3358 JBUF_LOCK_CHECK (priv, out_flushing);
3359 result = GST_FLOW_OK;
3360 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3361 JBUF_SIGNAL_QUERY (priv, res);
3370 return priv->srcresult;
3374 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3376 /* Peek a buffer and compare the seqnum to the expected seqnum.
3377 * If all is fine, the buffer is pushed.
3378 * If something is wrong, we wait for some event
3380 static GstFlowReturn
3381 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3383 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3384 GstFlowReturn result;
3385 RTPJitterBufferItem *item;
3387 guint32 next_seqnum;
3389 /* only push buffers when PLAYING and active and not buffering */
3390 if (priv->blocked || !priv->active ||
3391 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3392 return GST_FLOW_WAIT;
3395 /* peek a buffer, we're just looking at the sequence number.
3396 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3397 * wait for a timeout or something to change.
3398 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3399 item = rtp_jitter_buffer_peek (priv->jbuf);
3404 /* get the seqnum and the next expected seqnum */
3405 seqnum = item->seqnum;
3407 return pop_and_push_next (jitterbuffer, seqnum);
3410 next_seqnum = priv->next_seqnum;
3412 /* get the gap between this and the previous packet. If we don't know the
3413 * previous packet seqnum assume no gap. */
3414 if (G_UNLIKELY (next_seqnum == -1)) {
3415 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3416 /* we don't know what the next_seqnum should be, the chain function should
3417 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3418 * fires, so wait for that */
3419 result = GST_FLOW_WAIT;
3421 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3423 if (G_LIKELY (gap == 0)) {
3424 /* no missing packet, pop and push */
3425 result = pop_and_push_next (jitterbuffer, seqnum);
3426 } else if (G_UNLIKELY (gap < 0)) {
3427 /* if we have a packet that we already pushed or considered dropped, pop it
3428 * off and get the next packet */
3429 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3430 seqnum, next_seqnum);
3431 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3433 result = GST_FLOW_OK;
3435 /* the chain function has scheduled timers to request retransmission or
3436 * when to consider the packet lost, wait for that */
3437 GST_DEBUG_OBJECT (jitterbuffer,
3438 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3439 next_seqnum, seqnum, gap);
3440 result = GST_FLOW_WAIT;
3448 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3450 return GST_FLOW_EOS;
3452 return GST_FLOW_WAIT;
3458 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3460 GstClockTime rtx_retry_timeout;
3461 GstClockTime rtx_min_retry_timeout;
3463 if (priv->rtx_retry_timeout == -1) {
3464 if (priv->avg_rtx_rtt == 0)
3465 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3467 /* we want to ask for a retransmission after we waited for a
3468 * complete RTT and the additional jitter */
3469 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3471 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3473 /* make sure we don't retry too often. On very low latency networks,
3474 * the RTT and jitter can be very low. */
3475 if (priv->rtx_min_retry_timeout == -1) {
3476 rtx_min_retry_timeout = priv->packet_spacing;
3478 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3480 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3482 return rtx_retry_timeout;
3486 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3487 GstClockTime rtx_retry_timeout)
3489 GstClockTime rtx_retry_period;
3491 if (priv->rtx_retry_period == -1) {
3492 /* we retry up to the configured jitterbuffer size but leaving some
3493 * room for the retransmission to arrive in time */
3494 if (rtx_retry_timeout > priv->latency_ns) {
3495 rtx_retry_period = 0;
3497 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3500 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3502 return rtx_retry_period;
3506 update_rtx_stats (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3507 GstClockTime dts, gboolean success)
3509 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3513 /* we scheduled a retry for this packet and now we have it */
3514 priv->num_rtx_success++;
3515 /* all the previous retry attempts failed */
3516 priv->num_rtx_failed += timer->num_rtx_retry - 1;
3518 /* All retries failed or was too late */
3519 priv->num_rtx_failed += timer->num_rtx_retry;
3522 /* number of retries before (hopefully) receiving the packet */
3523 if (priv->avg_rtx_num == 0.0)
3524 priv->avg_rtx_num = timer->num_rtx_retry;
3526 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
3528 /* Calculate the delay between retransmission request and receiving this
3529 * packet. We have a valid delay if and only if this packet is a response to
3530 * our last request. If not we don't know if this is a response to an
3531 * earlier request and delay could be way off. For RTT is more important
3532 * with correct values than to update for every packet. */
3533 if (timer->num_rtx_retry == timer->num_rtx_received &&
3534 dts != GST_CLOCK_TIME_NONE && dts > timer->rtx_last) {
3535 delay = dts - timer->rtx_last;
3536 if (priv->avg_rtx_rtt == 0)
3537 priv->avg_rtx_rtt = delay;
3539 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
3544 GST_LOG_OBJECT (jitterbuffer,
3545 "RTX #%d, result %d, success %" G_GUINT64_FORMAT ", failed %"
3546 G_GUINT64_FORMAT ", requests %" G_GUINT64_FORMAT ", dups %"
3547 G_GUINT64_FORMAT ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %"
3548 GST_TIME_FORMAT, timer->seqnum, success, priv->num_rtx_success,
3549 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
3550 priv->avg_rtx_num, GST_TIME_ARGS (delay),
3551 GST_TIME_ARGS (priv->avg_rtx_rtt));
3554 /* the timeout for when we expected a packet expired */
3556 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3559 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3561 guint delay, delay_ms, avg_rtx_rtt_ms;
3562 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3563 GstClockTime rtx_retry_period;
3564 GstClockTime rtx_retry_timeout;
3567 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3568 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3570 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3571 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3573 delay = timer->rtx_delay + timer->rtx_retry;
3575 delay_ms = GST_TIME_AS_MSECONDS (delay);
3576 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3577 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3578 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3580 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3581 gst_structure_new ("GstRTPRetransmissionRequest",
3582 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3583 "running-time", G_TYPE_UINT64, timer->rtx_base,
3584 "delay", G_TYPE_UINT, delay_ms,
3585 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3586 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3587 "period", G_TYPE_UINT, rtx_retry_period_ms,
3588 "deadline", G_TYPE_UINT, priv->latency_ms,
3589 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3590 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3591 GST_DEBUG_OBJECT (jitterbuffer, "Request RTX: %" GST_PTR_FORMAT, event);
3593 priv->num_rtx_requests++;
3594 timer->num_rtx_retry++;
3596 GST_OBJECT_LOCK (jitterbuffer);
3597 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3598 timer->rtx_last = gst_clock_get_time (clock);
3599 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3601 timer->rtx_last = now;
3603 GST_OBJECT_UNLOCK (jitterbuffer);
3605 /* calculate the timeout for the next retransmission attempt */
3606 timer->rtx_retry += rtx_retry_timeout;
3607 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3608 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3609 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3610 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3611 if ((priv->rtx_max_retries != -1
3612 && timer->num_rtx_retry >= priv->rtx_max_retries)
3613 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3614 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3615 /* too many retransmission request, we now convert the timer
3616 * to a lost timer, leave the num_rtx_retry as it is for stats */
3617 timer->type = TIMER_TYPE_LOST;
3618 timer->rtx_delay = 0;
3619 timer->rtx_retry = 0;
3621 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3622 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3625 gst_pad_push_event (priv->sinkpad, event);
3631 /* a packet is lost */
3633 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3636 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3637 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3639 GstEvent *event = NULL;
3640 RTPJitterBufferItem *item;
3642 seqnum = timer->seqnum;
3643 lost_packets = MAX (timer->num, 1);
3644 num_rtx_retry = timer->num_rtx_retry;
3646 /* we had a gap and thus we lost some packets. Create an event for this. */
3647 if (lost_packets > 1)
3648 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3649 seqnum + lost_packets - 1);
3651 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3653 priv->num_lost += lost_packets;
3654 priv->num_rtx_failed += num_rtx_retry;
3656 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3658 /* we now only accept seqnum bigger than this */
3659 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3660 priv->next_in_seqnum = next_in_seqnum;
3662 /* Avoid creating events if we don't need it. Note that we still need to create
3663 * the lost *ITEM* since it will be used to notify the outgoing thread of
3664 * lost items (so that we can set discont flags and such) */
3665 if (priv->do_lost) {
3666 GstClockTime duration, timestamp;
3667 /* create paket lost event */
3668 timestamp = apply_offset (jitterbuffer, timer->timeout);
3669 duration = timer->duration;
3670 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3671 duration = priv->packet_spacing;
3672 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3673 gst_structure_new ("GstRTPPacketLost",
3674 "seqnum", G_TYPE_UINT, (guint) seqnum,
3675 "timestamp", G_TYPE_UINT64, timestamp,
3676 "duration", G_TYPE_UINT64, duration,
3677 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3679 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3680 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
3682 if (GST_CLOCK_TIME_IS_VALID (timer->rtx_last)) {
3683 /* Store info to update stats if the packet arrives too late */
3684 timer_queue_append (priv->rtx_stats_timers, timer,
3685 now + priv->rtx_stats_timeout * GST_MSECOND, TRUE);
3687 remove_timer (jitterbuffer, timer);
3690 JBUF_SIGNAL_EVENT (priv);
3696 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3699 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3701 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3702 remove_timer (jitterbuffer, timer);
3704 /* there was no EOS in the buffer, put one in there now */
3705 queue_event (jitterbuffer, gst_event_new_eos ());
3707 JBUF_SIGNAL_EVENT (priv);
3713 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3716 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3718 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3720 /* timer seqnum might have been obsoleted by caps seqnum-base,
3721 * only mess with current ongoing seqnum if still unknown */
3722 if (priv->next_seqnum == -1)
3723 priv->next_seqnum = timer->seqnum;
3724 remove_timer (jitterbuffer, timer);
3725 JBUF_SIGNAL_EVENT (priv);
3731 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3734 gboolean removed = FALSE;
3736 switch (timer->type) {
3737 case TIMER_TYPE_EXPECTED:
3738 removed = do_expected_timeout (jitterbuffer, timer, now);
3740 case TIMER_TYPE_LOST:
3741 removed = do_lost_timeout (jitterbuffer, timer, now);
3743 case TIMER_TYPE_DEADLINE:
3744 removed = do_deadline_timeout (jitterbuffer, timer, now);
3746 case TIMER_TYPE_EOS:
3747 removed = do_eos_timeout (jitterbuffer, timer, now);
3753 /* called when we need to wait for the next timeout.
3755 * We loop over the array of recorded timeouts and wait for the earliest one.
3756 * When it timed out, do the logic associated with the timer.
3758 * If there are no timers, we wait on a gcond until something new happens.
3761 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3763 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3764 GstClockTime now = 0;
3767 while (priv->timer_running) {
3768 TimerData *timer = NULL;
3769 GstClockTime timer_timeout = -1;
3772 /* If we have a clock, update "now" now with the very
3773 * latest running time we have. If timers are unscheduled below we
3774 * otherwise wouldn't update now (it's only updated when timers
3775 * expire), and also for the very first loop iteration now would
3776 * otherwise always be 0
3778 GST_OBJECT_LOCK (jitterbuffer);
3779 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3781 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3782 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3784 GST_OBJECT_UNLOCK (jitterbuffer);
3786 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3787 GST_TIME_ARGS (now));
3789 /* Clear expired rtx-stats timers */
3790 if (priv->do_retransmission)
3791 timer_queue_clear_until (priv->rtx_stats_timers, now);
3793 /* Iterate "normal" timers */
3794 len = priv->timers->len;
3795 for (i = 0; i < len;) {
3796 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3797 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3798 gboolean save_best = FALSE;
3800 GST_DEBUG_OBJECT (jitterbuffer,
3801 "%d, %d, %d, %" GST_TIME_FORMAT " diff:%" GST_STIME_FORMAT, i,
3802 test->type, test->seqnum, GST_TIME_ARGS (test_timeout),
3803 GST_STIME_ARGS ((gint64) (test_timeout - now)));
3805 /* Weed out anything too late */
3806 if (test->type == TIMER_TYPE_LOST &&
3807 (test_timeout == -1 || test_timeout <= now)) {
3808 GST_DEBUG_OBJECT (jitterbuffer, "Weeding out late entry");
3809 do_lost_timeout (jitterbuffer, test, now);
3810 if (!priv->timer_running)
3812 /* We don't move the iterator forward since we just removed the current entry,
3813 * but we update the termination condition */
3814 len = priv->timers->len;
3816 /* find the smallest timeout */
3817 if (timer == NULL) {
3819 } else if (timer_timeout == -1) {
3820 /* we already have an immediate timeout, the new timer must be an
3821 * immediate timer with smaller seqnum to become the best */
3822 if (test_timeout == -1
3823 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3824 timer->seqnum) > 0))
3826 } else if (test_timeout == -1) {
3827 /* first immediate timer */
3829 } else if (test_timeout < timer_timeout) {
3832 } else if (test_timeout == timer_timeout
3833 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3834 timer->seqnum) > 0)) {
3835 /* same timer, smaller seqnum */
3840 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3842 timer_timeout = test_timeout;
3847 if (timer && !priv->blocked) {
3849 GstClockTime sync_time;
3852 GstClockTimeDiff clock_jitter;
3854 if (timer_timeout == -1 || timer_timeout <= now) {
3855 /* We have normally removed all lost timers in the loop above */
3856 g_assert (timer->type != TIMER_TYPE_LOST);
3858 do_timeout (jitterbuffer, timer, now);
3859 /* check here, do_timeout could have released the lock */
3860 if (!priv->timer_running)
3865 GST_OBJECT_LOCK (jitterbuffer);
3866 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3868 GST_OBJECT_UNLOCK (jitterbuffer);
3869 /* let's just push if there is no clock */
3870 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3871 now = timer_timeout;
3875 /* prepare for sync against clock */
3876 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3877 /* add latency of peer to get input time */
3878 sync_time += priv->peer_latency;
3880 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3881 " with sync time %" GST_TIME_FORMAT,
3882 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3884 /* create an entry for the clock */
3885 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3886 priv->timer_timeout = timer_timeout;
3887 priv->timer_seqnum = timer->seqnum;
3888 GST_OBJECT_UNLOCK (jitterbuffer);
3890 /* release the lock so that the other end can push stuff or unlock */
3893 ret = gst_clock_id_wait (id, &clock_jitter);
3896 if (!priv->timer_running) {
3897 gst_clock_id_unref (id);
3898 priv->clock_id = NULL;
3902 if (ret != GST_CLOCK_UNSCHEDULED) {
3903 now = timer_timeout + MAX (clock_jitter, 0);
3904 GST_DEBUG_OBJECT (jitterbuffer,
3905 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3906 GST_STIME_ARGS (clock_jitter));
3908 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3910 /* and free the entry */
3911 gst_clock_id_unref (id);
3912 priv->clock_id = NULL;
3914 /* no timers, wait for activity */
3915 JBUF_WAIT_TIMER (priv);
3920 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3925 * This funcion implements the main pushing loop on the source pad.
3927 * It first tries to push as many buffers as possible. If there is a seqnum
3928 * mismatch, we wait for the next timeouts.
3931 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3933 GstRtpJitterBufferPrivate *priv;
3934 GstFlowReturn result = GST_FLOW_OK;
3936 priv = jitterbuffer->priv;
3938 JBUF_LOCK_CHECK (priv, flushing);
3940 result = handle_next_buffer (jitterbuffer);
3941 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3942 /* now wait for the next event */
3943 JBUF_WAIT_EVENT (priv, flushing);
3944 result = GST_FLOW_OK;
3946 } while (result == GST_FLOW_OK);
3947 /* store result for upstream */
3948 priv->srcresult = result;
3949 /* if we get here we need to pause */
3955 result = priv->srcresult;
3962 JBUF_SIGNAL_QUERY (priv, FALSE);
3965 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3966 gst_flow_get_name (result));
3967 gst_pad_pause_task (priv->srcpad);
3968 if (result == GST_FLOW_EOS) {
3969 event = gst_event_new_eos ();
3970 gst_pad_push_event (priv->srcpad, event);
3976 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3977 * some sanity checks and then emit the handle-sync signal with the parameters.
3978 * This function must be called with the LOCK */
3980 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3982 GstRtpJitterBufferPrivate *priv;
3983 guint64 base_rtptime, base_time;
3985 guint64 last_rtptime;
3987 guint64 ext_rtptime, diff;
3988 gboolean valid = TRUE, keep = FALSE;
3990 priv = jitterbuffer->priv;
3992 /* get the last values from the jitterbuffer */
3993 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3994 &clock_rate, &last_rtptime);
3996 clock_base = priv->clock_base;
3997 ext_rtptime = priv->ext_rtptime;
3999 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
4000 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
4001 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
4002 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
4004 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
4005 /* we keep this SR packet for later. When we get a valid RTP packet the
4006 * above values will be set and we can try to use the SR packet */
4007 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
4010 /* we can't accept anything that happened before we did the last resync */
4011 if (base_rtptime > ext_rtptime) {
4012 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
4015 /* the SR RTP timestamp must be something close to what we last observed
4016 * in the jitterbuffer */
4017 if (ext_rtptime > last_rtptime) {
4018 /* check how far ahead it is to our RTP timestamps */
4019 diff = ext_rtptime - last_rtptime;
4020 /* if bigger than 1 second, we drop it */
4021 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
4023 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
4024 clock_rate, 1000)) {
4025 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
4026 /* should drop this, but some RTSP servers end up with bogus
4027 * way too ahead RTCP packet when repeated PAUSE/PLAY,
4028 * so still trigger rptbin sync but invalidate RTCP data
4029 * (sync might use other methods) */
4032 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
4033 G_GUINT64_FORMAT, last_rtptime, diff);
4039 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
4043 s = gst_structure_new ("application/x-rtp-sync",
4044 "base-rtptime", G_TYPE_UINT64, base_rtptime,
4045 "base-time", G_TYPE_UINT64, base_time,
4046 "clock-rate", G_TYPE_UINT, clock_rate,
4047 "clock-base", G_TYPE_UINT64, clock_base,
4048 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
4049 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
4051 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
4052 gst_buffer_replace (&priv->last_sr, NULL);
4054 g_signal_emit (jitterbuffer,
4055 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
4057 gst_structure_free (s);
4059 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
4060 gst_buffer_replace (&priv->last_sr, NULL);
4064 static GstFlowReturn
4065 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
4068 GstRtpJitterBuffer *jitterbuffer;
4069 GstRtpJitterBufferPrivate *priv;
4070 GstFlowReturn ret = GST_FLOW_OK;
4072 GstRTCPPacket packet;
4073 guint64 ext_rtptime;
4075 GstRTCPBuffer rtcp = { NULL, };
4077 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4079 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
4080 goto invalid_buffer;
4082 priv = jitterbuffer->priv;
4084 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
4086 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
4089 /* first packet must be SR or RR or else the validate would have failed */
4090 switch (gst_rtcp_packet_get_type (&packet)) {
4091 case GST_RTCP_TYPE_SR:
4092 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
4098 gst_rtcp_buffer_unmap (&rtcp);
4100 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
4103 /* convert the RTP timestamp to our extended timestamp, using the same offset
4104 * we used in the jitterbuffer */
4105 ext_rtptime = priv->jbuf->ext_rtptime;
4106 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
4108 priv->ext_rtptime = ext_rtptime;
4109 gst_buffer_replace (&priv->last_sr, buffer);
4111 do_handle_sync (jitterbuffer);
4115 gst_buffer_unref (buffer);
4121 /* this is not fatal but should be filtered earlier */
4122 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4123 ("Received invalid RTCP payload, dropping"));
4129 /* this is not fatal but should be filtered earlier */
4130 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
4131 ("Received empty RTCP payload, dropping"));
4132 gst_rtcp_buffer_unmap (&rtcp);
4138 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
4139 gst_rtcp_buffer_unmap (&rtcp);
4146 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
4149 gboolean res = FALSE;
4150 GstRtpJitterBuffer *jitterbuffer;
4151 GstRtpJitterBufferPrivate *priv;
4153 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4154 priv = jitterbuffer->priv;
4156 switch (GST_QUERY_TYPE (query)) {
4157 case GST_QUERY_CAPS:
4159 GstCaps *filter, *caps;
4161 gst_query_parse_caps (query, &filter);
4162 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4163 gst_query_set_caps_result (query, caps);
4164 gst_caps_unref (caps);
4169 if (GST_QUERY_IS_SERIALIZED (query)) {
4170 RTPJitterBufferItem *item;
4173 JBUF_LOCK_CHECK (priv, out_flushing);
4174 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
4175 RTP_JITTER_BUFFER_MODE_BUFFER) {
4176 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
4177 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
4178 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL, -1);
4180 JBUF_SIGNAL_EVENT (priv);
4181 JBUF_WAIT_QUERY (priv, out_flushing);
4182 res = priv->last_query;
4184 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
4189 res = gst_pad_query_default (pad, parent, query);
4197 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
4205 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
4208 GstRtpJitterBuffer *jitterbuffer;
4209 GstRtpJitterBufferPrivate *priv;
4210 gboolean res = FALSE;
4212 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
4213 priv = jitterbuffer->priv;
4215 switch (GST_QUERY_TYPE (query)) {
4216 case GST_QUERY_LATENCY:
4218 /* We need to send the query upstream and add the returned latency to our
4220 GstClockTime min_latency, max_latency;
4222 GstClockTime our_latency;
4224 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
4225 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
4227 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
4228 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4229 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4231 /* store this so that we can safely sync on the peer buffers. */
4233 priv->peer_latency = min_latency;
4234 our_latency = priv->latency_ns;
4237 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
4238 GST_TIME_ARGS (our_latency));
4240 /* we add some latency but can buffer an infinite amount of time */
4241 min_latency += our_latency;
4244 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
4245 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
4246 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
4248 gst_query_set_latency (query, TRUE, min_latency, max_latency);
4252 case GST_QUERY_POSITION:
4254 GstClockTime start, last_out;
4257 gst_query_parse_position (query, &fmt, NULL);
4258 if (fmt != GST_FORMAT_TIME) {
4259 res = gst_pad_query_default (pad, parent, query);
4264 start = priv->npt_start;
4265 last_out = priv->last_out_time;
4268 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
4269 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
4270 GST_TIME_ARGS (last_out));
4272 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
4273 /* bring 0-based outgoing time to stream time */
4274 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
4277 res = gst_pad_query_default (pad, parent, query);
4281 case GST_QUERY_CAPS:
4283 GstCaps *filter, *caps;
4285 gst_query_parse_caps (query, &filter);
4286 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
4287 gst_query_set_caps_result (query, caps);
4288 gst_caps_unref (caps);
4293 res = gst_pad_query_default (pad, parent, query);
4301 gst_rtp_jitter_buffer_set_property (GObject * object,
4302 guint prop_id, const GValue * value, GParamSpec * pspec)
4304 GstRtpJitterBuffer *jitterbuffer;
4305 GstRtpJitterBufferPrivate *priv;
4307 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4308 priv = jitterbuffer->priv;
4313 guint new_latency, old_latency;
4315 new_latency = g_value_get_uint (value);
4318 old_latency = priv->latency_ms;
4319 priv->latency_ms = new_latency;
4320 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4321 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4324 /* post message if latency changed, this will inform the parent pipeline
4325 * that a latency reconfiguration is possible/needed. */
4326 if (new_latency != old_latency) {
4327 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4328 GST_TIME_ARGS (new_latency * GST_MSECOND));
4330 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4331 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4335 case PROP_DROP_ON_LATENCY:
4337 priv->drop_on_latency = g_value_get_boolean (value);
4340 case PROP_TS_OFFSET:
4342 priv->ts_offset = g_value_get_int64 (value);
4343 priv->ts_discont = TRUE;
4348 priv->do_lost = g_value_get_boolean (value);
4353 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4356 case PROP_DO_RETRANSMISSION:
4358 priv->do_retransmission = g_value_get_boolean (value);
4361 case PROP_RTX_NEXT_SEQNUM:
4363 priv->rtx_next_seqnum = g_value_get_boolean (value);
4366 case PROP_RTX_DELAY:
4368 priv->rtx_delay = g_value_get_int (value);
4371 case PROP_RTX_MIN_DELAY:
4373 priv->rtx_min_delay = g_value_get_uint (value);
4376 case PROP_RTX_DELAY_REORDER:
4378 priv->rtx_delay_reorder = g_value_get_int (value);
4381 case PROP_RTX_RETRY_TIMEOUT:
4383 priv->rtx_retry_timeout = g_value_get_int (value);
4386 case PROP_RTX_MIN_RETRY_TIMEOUT:
4388 priv->rtx_min_retry_timeout = g_value_get_int (value);
4391 case PROP_RTX_RETRY_PERIOD:
4393 priv->rtx_retry_period = g_value_get_int (value);
4396 case PROP_RTX_MAX_RETRIES:
4398 priv->rtx_max_retries = g_value_get_int (value);
4401 case PROP_RTX_STATS_TIMEOUT:
4403 priv->rtx_stats_timeout = g_value_get_uint (value);
4406 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4408 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4411 case PROP_MAX_DROPOUT_TIME:
4413 priv->max_dropout_time = g_value_get_uint (value);
4416 case PROP_MAX_MISORDER_TIME:
4418 priv->max_misorder_time = g_value_get_uint (value);
4421 case PROP_RFC7273_SYNC:
4423 rtp_jitter_buffer_set_rfc7273_sync (priv->jbuf,
4424 g_value_get_boolean (value));
4428 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4434 gst_rtp_jitter_buffer_get_property (GObject * object,
4435 guint prop_id, GValue * value, GParamSpec * pspec)
4437 GstRtpJitterBuffer *jitterbuffer;
4438 GstRtpJitterBufferPrivate *priv;
4440 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4441 priv = jitterbuffer->priv;
4446 g_value_set_uint (value, priv->latency_ms);
4449 case PROP_DROP_ON_LATENCY:
4451 g_value_set_boolean (value, priv->drop_on_latency);
4454 case PROP_TS_OFFSET:
4456 g_value_set_int64 (value, priv->ts_offset);
4461 g_value_set_boolean (value, priv->do_lost);
4466 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4474 if (priv->srcresult != GST_FLOW_OK)
4477 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4479 g_value_set_int (value, percent);
4483 case PROP_DO_RETRANSMISSION:
4485 g_value_set_boolean (value, priv->do_retransmission);
4488 case PROP_RTX_NEXT_SEQNUM:
4490 g_value_set_boolean (value, priv->rtx_next_seqnum);
4493 case PROP_RTX_DELAY:
4495 g_value_set_int (value, priv->rtx_delay);
4498 case PROP_RTX_MIN_DELAY:
4500 g_value_set_uint (value, priv->rtx_min_delay);
4503 case PROP_RTX_DELAY_REORDER:
4505 g_value_set_int (value, priv->rtx_delay_reorder);
4508 case PROP_RTX_RETRY_TIMEOUT:
4510 g_value_set_int (value, priv->rtx_retry_timeout);
4513 case PROP_RTX_MIN_RETRY_TIMEOUT:
4515 g_value_set_int (value, priv->rtx_min_retry_timeout);
4518 case PROP_RTX_RETRY_PERIOD:
4520 g_value_set_int (value, priv->rtx_retry_period);
4523 case PROP_RTX_MAX_RETRIES:
4525 g_value_set_int (value, priv->rtx_max_retries);
4528 case PROP_RTX_STATS_TIMEOUT:
4530 g_value_set_uint (value, priv->rtx_stats_timeout);
4534 g_value_take_boxed (value,
4535 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4537 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4539 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4542 case PROP_MAX_DROPOUT_TIME:
4544 g_value_set_uint (value, priv->max_dropout_time);
4547 case PROP_MAX_MISORDER_TIME:
4549 g_value_set_uint (value, priv->max_misorder_time);
4552 case PROP_RFC7273_SYNC:
4554 g_value_set_boolean (value,
4555 rtp_jitter_buffer_get_rfc7273_sync (priv->jbuf));
4559 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4564 static GstStructure *
4565 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4567 GstRtpJitterBufferPrivate *priv = jbuf->priv;
4571 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4572 "num-pushed", G_TYPE_UINT64, priv->num_pushed,
4573 "num-lost", G_TYPE_UINT64, priv->num_lost,
4574 "num-late", G_TYPE_UINT64, priv->num_late,
4575 "num-duplicates", G_TYPE_UINT64, priv->num_duplicates,
4576 "avg-jitter", G_TYPE_UINT64, priv->avg_jitter,
4577 "rtx-count", G_TYPE_UINT64, priv->num_rtx_requests,
4578 "rtx-success-count", G_TYPE_UINT64, priv->num_rtx_success,
4579 "rtx-per-packet", G_TYPE_DOUBLE, priv->avg_rtx_num,
4580 "rtx-rtt", G_TYPE_UINT64, priv->avg_rtx_rtt, NULL);