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>
9 * This library is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Library General Public
11 * License as published by the Free Software Foundation; either
12 * version 2 of the License, or (at your option) any later version.
14 * This library is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Library General Public License for more details.
19 * You should have received a copy of the GNU Library General Public
20 * License along with this library; if not, write to the
21 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
22 * Boston, MA 02110-1301, USA.
27 * SECTION:element-rtpjitterbuffer
29 * This element reorders and removes duplicate RTP packets as they are received
30 * from a network source.
32 * The element needs the clock-rate of the RTP payload in order to estimate the
33 * delay. This information is obtained either from the caps on the sink pad or,
34 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
35 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
37 * The rtpjitterbuffer will wait for missing packets up to a configurable time
38 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
39 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
40 * property is set, lost packets will result in a custom serialized downstream
41 * event of name GstRTPPacketLost. The lost packet events are usually used by a
42 * depayloader or other element to create concealment data or some other logic
43 * to gracefully handle the missing packets.
45 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
46 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
49 * The jitterbuffer can also be configured to send early retransmission events
50 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
51 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
52 * sends a custom upstream event named GstRTPRetransmissionRequest when the
53 * packet is considered late. The initial expected packet arrival time is
54 * calculated as follows:
56 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
57 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
58 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
59 * packets with different rtptime.
61 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
62 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
63 * previously scheduled timeout is overwritten.
65 * - If seqnum N arrived, all seqnum older than
66 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
67 * immediately. This is to request fast feedback for abonormally reorder
68 * packets before any of the previous timeouts is triggered.
70 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
71 * event. After the initial timeout expires and the retransmission event is
72 * sent, the timeout is scheduled for
73 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
74 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
75 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
76 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
77 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
78 * retransmission requests are sent and the regular logic is performed to
79 * schedule a lost packet as discussed above.
81 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
84 * This element will automatically be used inside rtpbin.
87 * <title>Example pipelines</title>
89 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
90 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
91 * inserted into the pipeline to smooth out network jitter and to reorder the
92 * out-of-order RTP packets.
102 #include <gst/rtp/gstrtpbuffer.h>
104 #include "gstrtpjitterbuffer.h"
105 #include "rtpjitterbuffer.h"
106 #include "rtpstats.h"
108 #include <gst/glib-compat-private.h>
110 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
111 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
113 /* RTPJitterBuffer signals and args */
116 SIGNAL_REQUEST_PT_MAP,
124 #define DEFAULT_LATENCY_MS 200
125 #define DEFAULT_DROP_ON_LATENCY FALSE
126 #define DEFAULT_TS_OFFSET 0
127 #define DEFAULT_DO_LOST FALSE
128 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
129 #define DEFAULT_PERCENT 0
130 #define DEFAULT_DO_RETRANSMISSION FALSE
131 #define DEFAULT_RTX_DELAY -1
132 #define DEFAULT_RTX_MIN_DELAY 0
133 #define DEFAULT_RTX_DELAY_REORDER 3
134 #define DEFAULT_RTX_RETRY_TIMEOUT -1
135 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
136 #define DEFAULT_RTX_RETRY_PERIOD -1
138 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
139 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
145 PROP_DROP_ON_LATENCY,
150 PROP_DO_RETRANSMISSION,
153 PROP_RTX_DELAY_REORDER,
154 PROP_RTX_RETRY_TIMEOUT,
155 PROP_RTX_MIN_RETRY_TIMEOUT,
156 PROP_RTX_RETRY_PERIOD,
161 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
163 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
165 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
168 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
170 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
171 GST_DEBUG ("waiting timer"); \
172 (priv)->waiting_timer = TRUE; \
173 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
174 (priv)->waiting_timer = FALSE; \
175 GST_DEBUG ("waiting timer done"); \
177 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
178 if (G_UNLIKELY ((priv)->waiting_timer)) { \
179 GST_DEBUG ("signal timer"); \
180 g_cond_signal (&(priv)->jbuf_timer); \
184 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
185 GST_DEBUG ("waiting event"); \
186 (priv)->waiting_event = TRUE; \
187 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
188 (priv)->waiting_event = FALSE; \
189 GST_DEBUG ("waiting event done"); \
190 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
193 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
194 if (G_UNLIKELY ((priv)->waiting_event)) { \
195 GST_DEBUG ("signal event"); \
196 g_cond_signal (&(priv)->jbuf_event); \
200 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
201 GST_DEBUG ("waiting query"); \
202 (priv)->waiting_query = TRUE; \
203 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
204 (priv)->waiting_query = FALSE; \
205 GST_DEBUG ("waiting query done"); \
206 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
209 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
210 (priv)->last_query = res; \
211 if (G_UNLIKELY ((priv)->waiting_query)) { \
212 GST_DEBUG ("signal query"); \
213 g_cond_signal (&(priv)->jbuf_query); \
218 struct _GstRtpJitterBufferPrivate
220 GstPad *sinkpad, *srcpad;
223 RTPJitterBuffer *jbuf;
225 gboolean waiting_timer;
227 gboolean waiting_event;
229 gboolean waiting_query;
237 gboolean timer_running;
238 GThread *timer_thread;
243 gboolean drop_on_latency;
246 gboolean do_retransmission;
249 gint rtx_delay_reorder;
250 gint rtx_retry_timeout;
251 gint rtx_min_retry_timeout;
252 gint rtx_retry_period;
254 /* the last seqnum we pushed out */
255 guint32 last_popped_seqnum;
256 /* the next expected seqnum we push */
258 /* last output time */
259 GstClockTime last_out_time;
260 /* last valid input timestamp and rtptime pair */
261 GstClockTime ips_dts;
263 GstClockTime packet_spacing;
265 /* the next expected seqnum we receive */
266 GstClockTime last_in_dts;
267 guint32 last_in_seqnum;
268 guint32 next_in_seqnum;
272 /* start and stop ranges */
273 GstClockTime npt_start;
274 GstClockTime npt_stop;
275 guint64 ext_timestamp;
276 guint64 last_elapsed;
277 guint64 estimated_eos;
284 /* clock rate and rtp timestamp offset */
288 gint64 prev_ts_offset;
290 /* when we are shutting down */
291 GstFlowReturn srcresult;
297 GstClockTime timer_timeout;
298 guint16 timer_seqnum;
299 /* the latency of the upstream peer, we have to take this into account when
300 * synchronizing the buffers. */
301 GstClockTime peer_latency;
305 /* some accounting */
307 guint64 num_duplicates;
308 guint64 num_rtx_requests;
309 guint64 num_rtx_success;
310 guint64 num_rtx_failed;
315 GstClockTime last_dts;
316 guint64 last_rtptime;
317 GstClockTime avg_jitter;
334 GstClockTime timeout;
335 GstClockTime duration;
336 GstClockTime rtx_base;
337 GstClockTime rtx_delay;
338 GstClockTime rtx_retry;
339 GstClockTime rtx_last;
343 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
344 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
345 GstRtpJitterBufferPrivate))
347 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
348 GST_STATIC_PAD_TEMPLATE ("sink",
351 GST_STATIC_CAPS ("application/x-rtp"
352 /* "clock-rate = (int) [ 1, 2147483647 ], "
353 * "payload = (int) , "
354 * "encoding-name = (string) "
358 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
359 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
362 GST_STATIC_CAPS ("application/x-rtcp")
365 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
366 GST_STATIC_PAD_TEMPLATE ("src",
369 GST_STATIC_CAPS ("application/x-rtp"
370 /* "payload = (int) , "
371 * "clock-rate = (int) , "
372 * "encoding-name = (string) "
376 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
378 #define gst_rtp_jitter_buffer_parent_class parent_class
379 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
381 /* object overrides */
382 static void gst_rtp_jitter_buffer_set_property (GObject * object,
383 guint prop_id, const GValue * value, GParamSpec * pspec);
384 static void gst_rtp_jitter_buffer_get_property (GObject * object,
385 guint prop_id, GValue * value, GParamSpec * pspec);
386 static void gst_rtp_jitter_buffer_finalize (GObject * object);
388 /* element overrides */
389 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
390 * element, GstStateChange transition);
391 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
392 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
393 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
395 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
398 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
399 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
402 /* sinkpad overrides */
403 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
404 GstObject * parent, GstEvent * event);
405 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
406 GstObject * parent, GstBuffer * buffer);
408 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
409 GstObject * parent, GstEvent * event);
410 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
411 GstObject * parent, GstBuffer * buffer);
413 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
414 GstObject * parent, GstQuery * query);
416 /* srcpad overrides */
417 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
418 GstObject * parent, GstEvent * event);
419 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
420 GstObject * parent, GstPadMode mode, gboolean active);
421 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
422 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
423 GstObject * parent, GstQuery * query);
426 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
428 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
429 gboolean active, guint64 base_time);
430 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
432 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
433 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
435 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
437 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
441 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
443 GObjectClass *gobject_class;
444 GstElementClass *gstelement_class;
446 gobject_class = (GObjectClass *) klass;
447 gstelement_class = (GstElementClass *) klass;
449 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
451 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
453 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
454 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
457 * GstRtpJitterBuffer:latency:
459 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
460 * for at most this time.
462 g_object_class_install_property (gobject_class, PROP_LATENCY,
463 g_param_spec_uint ("latency", "Buffer latency in ms",
464 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
465 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
467 * GstRtpJitterBuffer:drop-on-latency:
469 * Drop oldest buffers when the queue is completely filled.
471 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
472 g_param_spec_boolean ("drop-on-latency",
473 "Drop buffers when maximum latency is reached",
474 "Tells the jitterbuffer to never exceed the given latency in size",
475 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
477 * GstRtpJitterBuffer:ts-offset:
479 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
480 * This is mainly used to ensure interstream synchronisation.
482 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
483 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
484 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
485 G_MAXINT64, DEFAULT_TS_OFFSET,
486 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
489 * GstRtpJitterBuffer:do-lost:
491 * Send out a GstRTPPacketLost event downstream when a packet is considered
494 g_object_class_install_property (gobject_class, PROP_DO_LOST,
495 g_param_spec_boolean ("do-lost", "Do Lost",
496 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
497 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
500 * GstRtpJitterBuffer:mode:
502 * Control the buffering and timestamping mode used by the jitterbuffer.
504 g_object_class_install_property (gobject_class, PROP_MODE,
505 g_param_spec_enum ("mode", "Mode",
506 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
507 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
509 * GstRtpJitterBuffer:percent:
511 * The percent of the jitterbuffer that is filled.
513 g_object_class_install_property (gobject_class, PROP_PERCENT,
514 g_param_spec_int ("percent", "percent",
515 "The buffer filled percent", 0, 100,
516 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
518 * GstRtpJitterBuffer:do-retransmission:
520 * Send out a GstRTPRetransmission event upstream when a packet is considered
521 * late and should be retransmitted.
525 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
526 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
527 "Send retransmission events upstream when a packet is late",
528 DEFAULT_DO_RETRANSMISSION,
529 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
532 * GstRtpJitterBuffer:rtx-delay:
534 * When a packet did not arrive at the expected time, wait this extra amount
535 * of time before sending a retransmission event.
537 * When -1 is used, the max jitter will be used as extra delay.
541 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
542 g_param_spec_int ("rtx-delay", "RTX Delay",
543 "Extra time in ms to wait before sending retransmission "
544 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
545 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
548 * GstRtpJitterBuffer:rtx-min-delay:
550 * When a packet did not arrive at the expected time, wait at least this extra amount
551 * of time before sending a retransmission event.
555 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
556 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
557 "Minimum time in ms to wait before sending retransmission "
558 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
559 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
561 * GstRtpJitterBuffer:rtx-delay-reorder:
563 * Assume that a retransmission event should be sent when we see
564 * this much packet reordering.
566 * When -1 is used, the value will be estimated based on observed packet
571 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
572 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
573 "Sending retransmission event when this much reordering (-1 automatic)",
574 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
575 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
577 * GstRtpJitterBuffer::rtx-retry-timeout:
579 * When no packet has been received after sending a retransmission event
580 * for this time, retry sending a retransmission event.
582 * When -1 is used, the value will be estimated based on observed round
587 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
588 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
589 "Retry sending a transmission event after this timeout in "
590 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
591 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
593 * GstRtpJitterBuffer::rtx-min-retry-timeout:
595 * The minimum amount of time between retry timeouts. When
596 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
597 * minimum interval between retry timeouts.
599 * When -1 is used, the value will be estimated based on the
604 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
605 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
606 "Minimum timeout between sending a transmission event in "
607 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
608 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
610 * GstRtpJitterBuffer:rtx-retry-period:
612 * The amount of time to try to get a retransmission.
614 * When -1 is used, the value will be estimated based on the jitterbuffer
615 * latency and the observed round trip time.
619 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
620 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
621 "Try to get a retransmission for this many ms "
622 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
623 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
625 * GstRtpJitterBuffer:stats:
627 * Various jitterbuffer statistics. This property returns a GstStructure
628 * with name application/x-rtp-jitterbuffer-stats with the following fields:
630 * "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
631 * "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
632 * "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
633 * "rtx-rtt" G_TYPE_UINT64 Average round trip time per RTX
637 g_object_class_install_property (gobject_class, PROP_STATS,
638 g_param_spec_boxed ("stats", "Statistics",
639 "Various statistics", GST_TYPE_STRUCTURE,
640 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
643 * GstRtpJitterBuffer::request-pt-map:
644 * @buffer: the object which received the signal
647 * Request the payload type as #GstCaps for @pt.
649 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
650 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
651 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
652 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
653 GST_TYPE_CAPS, 1, G_TYPE_UINT);
655 * GstRtpJitterBuffer::handle-sync:
656 * @buffer: the object which received the signal
657 * @struct: a GstStructure containing sync values.
659 * Be notified of new sync values.
661 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
662 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
663 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
664 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
665 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
668 * GstRtpJitterBuffer::on-npt-stop:
669 * @buffer: the object which received the signal
671 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
672 * the npt-stop position.
674 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
675 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
676 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
677 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
678 G_TYPE_NONE, 0, G_TYPE_NONE);
681 * GstRtpJitterBuffer::clear-pt-map:
682 * @buffer: the object which received the signal
684 * Invalidate the clock-rate as obtained with the
685 * #GstRtpJitterBuffer::request-pt-map signal.
687 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
688 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
689 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
690 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
691 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
694 * GstRtpJitterBuffer::set-active:
695 * @buffer: the object which received the signal
697 * Start pushing out packets with the given base time. This signal is only
698 * useful in buffering mode.
700 * Returns: the time of the last pushed packet.
702 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
703 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
704 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
705 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
706 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
709 gstelement_class->change_state =
710 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
711 gstelement_class->request_new_pad =
712 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
713 gstelement_class->release_pad =
714 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
715 gstelement_class->provide_clock =
716 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
718 gst_element_class_add_pad_template (gstelement_class,
719 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
720 gst_element_class_add_pad_template (gstelement_class,
721 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
722 gst_element_class_add_pad_template (gstelement_class,
723 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
725 gst_element_class_set_static_metadata (gstelement_class,
726 "RTP packet jitter-buffer", "Filter/Network/RTP",
727 "A buffer that deals with network jitter and other transmission faults",
728 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
729 "Wim Taymans <wim.taymans@gmail.com>");
731 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
732 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
734 GST_DEBUG_CATEGORY_INIT
735 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
739 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
741 GstRtpJitterBufferPrivate *priv;
743 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
744 jitterbuffer->priv = priv;
746 priv->latency_ms = DEFAULT_LATENCY_MS;
747 priv->latency_ns = priv->latency_ms * GST_MSECOND;
748 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
749 priv->do_lost = DEFAULT_DO_LOST;
750 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
751 priv->rtx_delay = DEFAULT_RTX_DELAY;
752 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
753 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
754 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
755 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
756 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
759 priv->last_rtptime = -1;
760 priv->avg_jitter = 0;
761 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
762 priv->jbuf = rtp_jitter_buffer_new ();
763 g_mutex_init (&priv->jbuf_lock);
764 g_cond_init (&priv->jbuf_timer);
765 g_cond_init (&priv->jbuf_event);
766 g_cond_init (&priv->jbuf_query);
768 /* reset skew detection initialy */
769 rtp_jitter_buffer_reset_skew (priv->jbuf);
770 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
771 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
775 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
778 gst_pad_set_activatemode_function (priv->srcpad,
779 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
780 gst_pad_set_query_function (priv->srcpad,
781 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
782 gst_pad_set_event_function (priv->srcpad,
783 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
786 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
789 gst_pad_set_chain_function (priv->sinkpad,
790 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
791 gst_pad_set_event_function (priv->sinkpad,
792 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
793 gst_pad_set_query_function (priv->sinkpad,
794 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
796 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
797 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
799 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
802 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
804 #define ITEM_TYPE_BUFFER 0
805 #define ITEM_TYPE_LOST 1
806 #define ITEM_TYPE_EVENT 2
807 #define ITEM_TYPE_QUERY 3
809 static RTPJitterBufferItem *
810 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
811 guint seqnum, guint count, guint rtptime)
813 RTPJitterBufferItem *item;
815 item = g_slice_new (RTPJitterBufferItem);
822 item->seqnum = seqnum;
824 item->rtptime = rtptime;
830 free_item (RTPJitterBufferItem * item)
832 if (item->data && item->type != ITEM_TYPE_QUERY)
833 gst_mini_object_unref (item->data);
834 g_slice_free (RTPJitterBufferItem, item);
838 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
840 GList **l = user_data;
842 if (item->data && item->type == ITEM_TYPE_EVENT
843 && GST_EVENT_IS_STICKY (item->data)) {
844 *l = g_list_prepend (*l, item->data);
845 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
846 gst_mini_object_unref (item->data);
848 g_slice_free (RTPJitterBufferItem, item);
852 gst_rtp_jitter_buffer_finalize (GObject * object)
854 GstRtpJitterBuffer *jitterbuffer;
855 GstRtpJitterBufferPrivate *priv;
857 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
858 priv = jitterbuffer->priv;
860 g_array_free (priv->timers, TRUE);
861 g_mutex_clear (&priv->jbuf_lock);
862 g_cond_clear (&priv->jbuf_timer);
863 g_cond_clear (&priv->jbuf_event);
864 g_cond_clear (&priv->jbuf_query);
866 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
867 g_object_unref (priv->jbuf);
869 G_OBJECT_CLASS (parent_class)->finalize (object);
873 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
875 GstRtpJitterBuffer *jitterbuffer;
876 GstPad *otherpad = NULL;
877 GstIterator *it = NULL;
880 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
882 if (pad == jitterbuffer->priv->sinkpad) {
883 otherpad = jitterbuffer->priv->srcpad;
884 } else if (pad == jitterbuffer->priv->srcpad) {
885 otherpad = jitterbuffer->priv->sinkpad;
886 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
887 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
891 g_value_init (&val, GST_TYPE_PAD);
892 g_value_set_object (&val, otherpad);
893 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
894 g_value_unset (&val);
901 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
903 GstRtpJitterBufferPrivate *priv;
905 priv = jitterbuffer->priv;
907 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
910 gst_pad_new_from_static_template
911 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
912 gst_pad_set_chain_function (priv->rtcpsinkpad,
913 gst_rtp_jitter_buffer_chain_rtcp);
914 gst_pad_set_event_function (priv->rtcpsinkpad,
915 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
916 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
917 gst_rtp_jitter_buffer_iterate_internal_links);
918 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
919 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
921 return priv->rtcpsinkpad;
925 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
927 GstRtpJitterBufferPrivate *priv;
929 priv = jitterbuffer->priv;
931 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
933 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
935 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
936 priv->rtcpsinkpad = NULL;
940 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
941 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
943 GstRtpJitterBuffer *jitterbuffer;
944 GstElementClass *klass;
946 GstRtpJitterBufferPrivate *priv;
948 g_return_val_if_fail (templ != NULL, NULL);
949 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
951 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
952 priv = jitterbuffer->priv;
953 klass = GST_ELEMENT_GET_CLASS (element);
955 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
957 /* figure out the template */
958 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
959 if (priv->rtcpsinkpad != NULL)
962 result = create_rtcp_sink (jitterbuffer);
971 g_warning ("rtpjitterbuffer: this is not our template");
976 g_warning ("rtpjitterbuffer: pad already requested");
982 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
984 GstRtpJitterBuffer *jitterbuffer;
985 GstRtpJitterBufferPrivate *priv;
987 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
988 g_return_if_fail (GST_IS_PAD (pad));
990 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
991 priv = jitterbuffer->priv;
993 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
995 if (priv->rtcpsinkpad == pad) {
996 remove_rtcp_sink (jitterbuffer);
1005 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1011 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1013 return gst_system_clock_obtain ();
1017 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1019 GstRtpJitterBufferPrivate *priv;
1021 priv = jitterbuffer->priv;
1023 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1026 priv->clock_rate = -1;
1027 /* do not clear current content, but refresh state for new arrival */
1028 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1029 rtp_jitter_buffer_reset_skew (priv->jbuf);
1034 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1037 GstRtpJitterBufferPrivate *priv;
1038 GstClockTime last_out;
1039 RTPJitterBufferItem *item;
1044 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1045 active, GST_TIME_ARGS (offset));
1047 if (active != priv->active) {
1048 /* add the amount of time spent in paused to the output offset. All
1049 * outgoing buffers will have this offset applied to their timestamps in
1050 * order to make them arrive in time in the sink. */
1051 priv->out_offset = offset;
1052 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1053 GST_TIME_ARGS (priv->out_offset));
1054 priv->active = active;
1055 JBUF_SIGNAL_EVENT (priv);
1058 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1060 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1061 /* head buffer timestamp and offset gives our output time */
1062 last_out = item->dts + priv->ts_offset;
1064 /* use last known time when the buffer is empty */
1065 last_out = priv->last_out_time;
1073 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1075 GstRtpJitterBuffer *jitterbuffer;
1076 GstRtpJitterBufferPrivate *priv;
1081 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1082 priv = jitterbuffer->priv;
1084 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1086 caps = gst_pad_peer_query_caps (other, filter);
1088 templ = gst_pad_get_pad_template_caps (pad);
1090 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1095 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1097 intersect = gst_caps_intersect (caps, templ);
1098 gst_caps_unref (caps);
1099 gst_caps_unref (templ);
1103 gst_object_unref (jitterbuffer);
1109 * Must be called with JBUF_LOCK held
1113 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1116 GstRtpJitterBufferPrivate *priv;
1117 GstStructure *caps_struct;
1121 priv = jitterbuffer->priv;
1123 /* first parse the caps */
1124 caps_struct = gst_caps_get_structure (caps, 0);
1126 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1128 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1129 * measure the amount of data in the buffer */
1130 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1133 if (priv->clock_rate <= 0)
1136 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1138 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1140 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1141 * can use this to track the amount of time elapsed on the sender. */
1142 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1143 priv->clock_base = val;
1145 priv->clock_base = -1;
1147 priv->ext_timestamp = priv->clock_base;
1149 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1152 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1153 /* first expected seqnum, only update when we didn't have a previous base. */
1154 if (priv->next_in_seqnum == -1)
1155 priv->next_in_seqnum = val;
1156 if (priv->next_seqnum == -1) {
1157 priv->next_seqnum = val;
1158 JBUF_SIGNAL_EVENT (priv);
1162 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1164 /* the start and stop times. The seqnum-base corresponds to the start time. We
1165 * will keep track of the seqnums on the output and when we reach the one
1166 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1167 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1168 priv->npt_start = tval;
1170 priv->npt_start = 0;
1172 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1173 priv->npt_stop = tval;
1175 priv->npt_stop = -1;
1177 GST_DEBUG_OBJECT (jitterbuffer,
1178 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1179 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1186 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1191 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1197 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1199 GstRtpJitterBufferPrivate *priv;
1201 priv = jitterbuffer->priv;
1204 /* mark ourselves as flushing */
1205 priv->srcresult = GST_FLOW_FLUSHING;
1206 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1207 /* this unblocks any waiting pops on the src pad task */
1208 JBUF_SIGNAL_EVENT (priv);
1209 JBUF_SIGNAL_QUERY (priv, FALSE);
1214 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1216 GstRtpJitterBufferPrivate *priv;
1218 priv = jitterbuffer->priv;
1221 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1222 /* Mark as non flushing */
1223 priv->srcresult = GST_FLOW_OK;
1224 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1225 priv->last_popped_seqnum = -1;
1226 priv->last_out_time = -1;
1227 priv->next_seqnum = -1;
1228 priv->ips_rtptime = -1;
1229 priv->ips_dts = GST_CLOCK_TIME_NONE;
1230 priv->packet_spacing = 0;
1231 priv->next_in_seqnum = -1;
1232 priv->clock_rate = -1;
1235 priv->estimated_eos = -1;
1236 priv->last_elapsed = 0;
1237 priv->ext_timestamp = -1;
1238 priv->avg_jitter = 0;
1239 priv->last_dts = -1;
1240 priv->last_rtptime = -1;
1241 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1242 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1243 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1244 rtp_jitter_buffer_reset_skew (priv->jbuf);
1245 remove_all_timers (jitterbuffer);
1250 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1251 GstPadMode mode, gboolean active)
1254 GstRtpJitterBuffer *jitterbuffer = NULL;
1256 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1259 case GST_PAD_MODE_PUSH:
1261 /* allow data processing */
1262 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1264 /* start pushing out buffers */
1265 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1266 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1267 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1269 /* make sure all data processing stops ASAP */
1270 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1272 /* NOTE this will hardlock if the state change is called from the src pad
1273 * task thread because we will _join() the thread. */
1274 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1275 result = gst_pad_stop_task (pad);
1285 static GstStateChangeReturn
1286 gst_rtp_jitter_buffer_change_state (GstElement * element,
1287 GstStateChange transition)
1289 GstRtpJitterBuffer *jitterbuffer;
1290 GstRtpJitterBufferPrivate *priv;
1291 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1293 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1294 priv = jitterbuffer->priv;
1296 switch (transition) {
1297 case GST_STATE_CHANGE_NULL_TO_READY:
1299 case GST_STATE_CHANGE_READY_TO_PAUSED:
1301 /* reset negotiated values */
1302 priv->clock_rate = -1;
1303 priv->clock_base = -1;
1304 priv->peer_latency = 0;
1306 /* block until we go to PLAYING */
1307 priv->blocked = TRUE;
1308 priv->timer_running = TRUE;
1309 priv->timer_thread =
1310 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1313 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1315 /* unblock to allow streaming in PLAYING */
1316 priv->blocked = FALSE;
1317 JBUF_SIGNAL_EVENT (priv);
1318 JBUF_SIGNAL_TIMER (priv);
1325 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1327 switch (transition) {
1328 case GST_STATE_CHANGE_READY_TO_PAUSED:
1329 /* we are a live element because we sync to the clock, which we can only
1330 * do in the PLAYING state */
1331 if (ret != GST_STATE_CHANGE_FAILURE)
1332 ret = GST_STATE_CHANGE_NO_PREROLL;
1334 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1336 /* block to stop streaming when PAUSED */
1337 priv->blocked = TRUE;
1338 unschedule_current_timer (jitterbuffer);
1340 if (ret != GST_STATE_CHANGE_FAILURE)
1341 ret = GST_STATE_CHANGE_NO_PREROLL;
1343 case GST_STATE_CHANGE_PAUSED_TO_READY:
1345 gst_buffer_replace (&priv->last_sr, NULL);
1346 priv->timer_running = FALSE;
1347 unschedule_current_timer (jitterbuffer);
1348 JBUF_SIGNAL_TIMER (priv);
1349 JBUF_SIGNAL_QUERY (priv, FALSE);
1351 g_thread_join (priv->timer_thread);
1352 priv->timer_thread = NULL;
1354 case GST_STATE_CHANGE_READY_TO_NULL:
1364 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1367 gboolean ret = TRUE;
1368 GstRtpJitterBuffer *jitterbuffer;
1369 GstRtpJitterBufferPrivate *priv;
1371 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1372 priv = jitterbuffer->priv;
1374 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1376 switch (GST_EVENT_TYPE (event)) {
1377 case GST_EVENT_LATENCY:
1379 GstClockTime latency;
1381 gst_event_parse_latency (event, &latency);
1383 GST_DEBUG_OBJECT (jitterbuffer,
1384 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1387 /* adjust the overall buffer delay to the total pipeline latency in
1388 * buffering mode because if downstream consumes too fast (because of
1389 * large latency or queues, we would start rebuffering again. */
1390 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1391 RTP_JITTER_BUFFER_MODE_BUFFER) {
1392 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1396 ret = gst_pad_push_event (priv->sinkpad, event);
1400 ret = gst_pad_push_event (priv->sinkpad, event);
1407 /* handles and stores the event in the jitterbuffer, must be called with
1410 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1412 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1413 RTPJitterBufferItem *item;
1416 switch (GST_EVENT_TYPE (event)) {
1417 case GST_EVENT_CAPS:
1421 gst_event_parse_caps (event, &caps);
1422 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1425 case GST_EVENT_SEGMENT:
1426 gst_event_copy_segment (event, &priv->segment);
1428 /* we need time for now */
1429 if (priv->segment.format != GST_FORMAT_TIME)
1430 goto newseg_wrong_format;
1432 GST_DEBUG_OBJECT (jitterbuffer,
1433 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1437 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1444 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1445 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1446 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1448 JBUF_SIGNAL_EVENT (priv);
1453 newseg_wrong_format:
1455 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1456 gst_event_unref (event);
1462 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1465 gboolean ret = TRUE;
1466 GstRtpJitterBuffer *jitterbuffer;
1467 GstRtpJitterBufferPrivate *priv;
1469 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1470 priv = jitterbuffer->priv;
1472 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1474 switch (GST_EVENT_TYPE (event)) {
1475 case GST_EVENT_FLUSH_START:
1476 ret = gst_pad_push_event (priv->srcpad, event);
1477 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1478 /* wait for the loop to go into PAUSED */
1479 gst_pad_pause_task (priv->srcpad);
1481 case GST_EVENT_FLUSH_STOP:
1482 ret = gst_pad_push_event (priv->srcpad, event);
1484 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1485 GST_PAD_MODE_PUSH, TRUE);
1488 if (GST_EVENT_IS_SERIALIZED (event)) {
1489 /* serialized events go in the queue */
1491 if (priv->srcresult != GST_FLOW_OK) {
1492 /* Errors in sticky event pushing are no problem and ignored here
1493 * as they will cause more meaningful errors during data flow.
1494 * For EOS events, that are not followed by data flow, we still
1495 * return FALSE here though.
1497 if (!GST_EVENT_IS_STICKY (event) ||
1498 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1499 goto out_flow_error;
1501 /* refuse more events on EOS */
1504 ret = queue_event (jitterbuffer, event);
1507 /* non-serialized events are forwarded downstream immediately */
1508 ret = gst_pad_push_event (priv->srcpad, event);
1517 GST_DEBUG_OBJECT (jitterbuffer,
1518 "refusing event, we have a downstream flow error: %s",
1519 gst_flow_get_name (priv->srcresult));
1521 gst_event_unref (event);
1526 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1528 gst_event_unref (event);
1534 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1537 gboolean ret = TRUE;
1538 GstRtpJitterBuffer *jitterbuffer;
1540 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1542 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1544 switch (GST_EVENT_TYPE (event)) {
1545 case GST_EVENT_FLUSH_START:
1546 gst_event_unref (event);
1548 case GST_EVENT_FLUSH_STOP:
1549 gst_event_unref (event);
1552 ret = gst_pad_event_default (pad, parent, event);
1560 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1561 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1562 * GST_FLOW_FLUSHING when the element is shutting down. On success
1563 * GST_FLOW_OK is returned.
1565 static GstFlowReturn
1566 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1570 GValue args[2] = { {0}, {0} };
1574 g_value_init (&args[0], GST_TYPE_ELEMENT);
1575 g_value_set_object (&args[0], jitterbuffer);
1576 g_value_init (&args[1], G_TYPE_UINT);
1577 g_value_set_uint (&args[1], pt);
1579 g_value_init (&ret, GST_TYPE_CAPS);
1580 g_value_set_boxed (&ret, NULL);
1582 JBUF_UNLOCK (jitterbuffer->priv);
1583 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1585 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1587 g_value_unset (&args[0]);
1588 g_value_unset (&args[1]);
1589 caps = (GstCaps *) g_value_dup_boxed (&ret);
1590 g_value_unset (&ret);
1594 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1595 gst_caps_unref (caps);
1597 if (G_UNLIKELY (!res))
1605 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1606 return GST_FLOW_ERROR;
1610 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1611 return GST_FLOW_FLUSHING;
1615 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1616 return GST_FLOW_ERROR;
1620 /* call with jbuf lock held */
1622 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1624 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1625 GstMessage *message = NULL;
1630 /* Post a buffering message */
1631 if (priv->last_percent != percent) {
1632 priv->last_percent = percent;
1634 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1635 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1642 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1644 GstRtpJitterBufferPrivate *priv;
1646 priv = jitterbuffer->priv;
1648 if (timestamp == -1)
1651 /* apply the timestamp offset, this is used for inter stream sync */
1652 timestamp += priv->ts_offset;
1653 /* add the offset, this is used when buffering */
1654 timestamp += priv->out_offset;
1660 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1662 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1663 TimerData *timer = NULL;
1666 len = priv->timers->len;
1667 for (i = 0; i < len; i++) {
1668 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1669 if (test->seqnum == seqnum && test->type == type) {
1678 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1680 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1682 if (priv->clock_id) {
1683 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1684 gst_clock_id_unschedule (priv->clock_id);
1685 priv->clock_id = NULL;
1690 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1692 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1693 GstClockTime test_timeout;
1695 if ((test_timeout = timer->timeout) == -1)
1698 if (timer->type != TIMER_TYPE_EXPECTED) {
1699 /* add our latency and offset to get output times. */
1700 test_timeout = apply_offset (jitterbuffer, test_timeout);
1701 test_timeout += priv->latency_ns;
1703 return test_timeout;
1707 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1709 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1711 if (priv->clock_id) {
1712 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1714 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1715 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1717 if (timeout == -1 || timeout < priv->timer_timeout)
1718 unschedule_current_timer (jitterbuffer);
1723 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1724 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1725 GstClockTime duration)
1727 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1731 GST_DEBUG_OBJECT (jitterbuffer,
1732 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1733 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1734 GST_TIME_ARGS (delay));
1736 len = priv->timers->len;
1737 g_array_set_size (priv->timers, len + 1);
1738 timer = &g_array_index (priv->timers, TimerData, len);
1741 timer->seqnum = seqnum;
1743 timer->timeout = timeout + delay;
1744 timer->duration = duration;
1745 if (type == TIMER_TYPE_EXPECTED) {
1746 timer->rtx_base = timeout;
1747 timer->rtx_delay = delay;
1748 timer->rtx_retry = 0;
1750 timer->num_rtx_retry = 0;
1751 recalculate_timer (jitterbuffer, timer);
1752 JBUF_SIGNAL_TIMER (priv);
1758 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1759 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1761 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1762 gboolean seqchange, timechange;
1765 seqchange = timer->seqnum != seqnum;
1766 timechange = timer->timeout != timeout;
1768 if (!seqchange && !timechange)
1771 oldseq = timer->seqnum;
1773 GST_DEBUG_OBJECT (jitterbuffer,
1774 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1775 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1777 timer->timeout = timeout + delay;
1778 timer->seqnum = seqnum;
1780 timer->rtx_base = timeout;
1781 timer->rtx_delay = delay;
1782 timer->rtx_retry = 0;
1785 timer->num_rtx_retry = 0;
1787 if (priv->clock_id) {
1788 /* we changed the seqnum and there is a timer currently waiting with this
1789 * seqnum, unschedule it */
1790 if (seqchange && priv->timer_seqnum == oldseq)
1791 unschedule_current_timer (jitterbuffer);
1792 /* we changed the time, check if it is earlier than what we are waiting
1793 * for and unschedule if so */
1794 else if (timechange)
1795 recalculate_timer (jitterbuffer, timer);
1800 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1801 guint16 seqnum, GstClockTime timeout)
1805 /* find the seqnum timer */
1806 timer = find_timer (jitterbuffer, type, seqnum);
1807 if (timer == NULL) {
1808 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1810 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1816 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1818 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1821 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1822 unschedule_current_timer (jitterbuffer);
1825 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1826 g_array_remove_index_fast (priv->timers, idx);
1831 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1833 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1834 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1835 g_array_set_size (priv->timers, 0);
1836 unschedule_current_timer (jitterbuffer);
1839 /* get the extra delay to wait before sending RTX */
1841 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1845 if (priv->rtx_delay == -1) {
1846 if (priv->avg_jitter == 0)
1847 delay = DEFAULT_AUTO_RTX_DELAY;
1849 /* jitter is in nanoseconds, 2x jitter is a good margin */
1850 delay = priv->avg_jitter * 2;
1852 delay = priv->rtx_delay * GST_MSECOND;
1854 if (priv->rtx_min_delay > 0)
1855 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1860 /* we just received a packet with seqnum and dts.
1862 * First check for old seqnum that we are still expecting. If the gap with the
1863 * current seqnum is too big, unschedule the timeouts.
1865 * If we have a valid packet spacing estimate we can set a timer for when we
1866 * should receive the next packet.
1867 * If we don't have a valid estimate, we remove any timer we might have
1868 * had for this packet.
1871 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1872 GstClockTime dts, gboolean do_next_seqnum)
1874 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1875 TimerData *timer = NULL;
1878 /* go through all timers and unschedule the ones with a large gap, also find
1879 * the timer for the seqnum */
1880 len = priv->timers->len;
1881 for (i = 0; i < len; i++) {
1882 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1885 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1887 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
1888 test->type, test->seqnum, seqnum, gap);
1891 GST_DEBUG ("found timer for current seqnum");
1892 /* the timer for the current seqnum */
1894 /* when no retransmission, we can stop now, we only need to find the
1895 * timer for the current seqnum */
1896 if (!priv->do_retransmission)
1898 } else if (gap > priv->rtx_delay_reorder) {
1899 /* max gap, we exceeded the max reorder distance and we don't expect the
1900 * missing packet to be this reordered */
1901 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1902 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
1906 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
1907 && priv->do_retransmission;
1909 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1910 if (timer->num_rtx_retry > 0) {
1911 GstClockTime rtx_last, delay;
1913 /* we scheduled a retry for this packet and now we have it */
1914 priv->num_rtx_success++;
1915 /* all the previous retry attempts failed */
1916 priv->num_rtx_failed += timer->num_rtx_retry - 1;
1917 /* number of retries before receiving the packet */
1918 if (priv->avg_rtx_num == 0.0)
1919 priv->avg_rtx_num = timer->num_rtx_retry;
1921 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
1922 /* calculate the delay between retransmission request and receiving this
1923 * packet, start with when we scheduled this timeout last */
1924 rtx_last = timer->rtx_last;
1925 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
1926 /* we have a valid delay if this packet arrived after we scheduled the
1928 delay = dts - rtx_last;
1929 if (priv->avg_rtx_rtt == 0)
1930 priv->avg_rtx_rtt = delay;
1932 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
1936 GST_LOG_OBJECT (jitterbuffer,
1937 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
1938 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
1939 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
1940 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
1941 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
1942 GST_TIME_ARGS (priv->avg_rtx_rtt));
1944 /* don't try to estimate the next seqnum because this is a retransmitted
1945 * packet and it probably did not arrive with the expected packet
1947 do_next_seqnum = FALSE;
1951 if (do_next_seqnum) {
1952 GstClockTime expected, delay;
1954 /* calculate expected arrival time of the next seqnum */
1955 expected = dts + priv->packet_spacing;
1957 delay = get_rtx_delay (priv);
1959 /* and update/install timer for next seqnum */
1961 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
1964 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
1965 expected, delay, priv->packet_spacing);
1966 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1967 /* if we had a timer, remove it, we don't know when to expect the next
1969 remove_timer (jitterbuffer, timer);
1974 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
1977 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1979 /* we need consecutive seqnums with a different
1980 * rtptime to estimate the packet spacing. */
1981 if (priv->ips_rtptime != rtptime) {
1982 /* rtptime changed, check dts diff */
1983 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
1984 priv->packet_spacing = dts - priv->ips_dts;
1985 GST_DEBUG_OBJECT (jitterbuffer,
1986 "new packet spacing %" GST_TIME_FORMAT,
1987 GST_TIME_ARGS (priv->packet_spacing));
1989 priv->ips_rtptime = rtptime;
1990 priv->ips_dts = dts;
1995 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
1996 guint16 seqnum, GstClockTime dts, gint gap)
1998 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1999 GstClockTime total_duration, duration, expected_dts;
2002 GST_DEBUG_OBJECT (jitterbuffer,
2003 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2004 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2006 /* the total duration spanned by the missing packets */
2007 if (dts >= priv->last_in_dts)
2008 total_duration = dts - priv->last_in_dts;
2012 /* interpolate between the current time and the last time based on
2013 * number of packets we are missing, this is the estimated duration
2014 * for the missing packet based on equidistant packet spacing. */
2015 duration = total_duration / (gap + 1);
2017 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2018 GST_TIME_ARGS (duration));
2020 if (total_duration > priv->latency_ns) {
2021 GstClockTime gap_time;
2024 gap_time = total_duration - priv->latency_ns;
2027 lost_packets = gap_time / duration;
2028 gap_time = lost_packets * duration;
2033 /* too many lost packets, some of the missing packets are already
2034 * too late and we can generate lost packet events for them. */
2035 GST_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
2036 " > %" GST_TIME_FORMAT ", consider %u lost",
2037 GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
2040 /* this timer will fire immediately and the lost event will be pushed from
2041 * the timer thread */
2042 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2043 priv->last_in_dts + duration, 0, gap_time);
2045 expected += lost_packets;
2046 priv->last_in_dts += gap_time;
2049 expected_dts = priv->last_in_dts + duration;
2051 if (priv->do_retransmission) {
2054 type = TIMER_TYPE_EXPECTED;
2055 /* if we had a timer for the first missing packet, update it. */
2056 if ((timer = find_timer (jitterbuffer, type, expected))) {
2057 GstClockTime timeout = timer->timeout;
2059 timer->duration = duration;
2060 if (timeout > expected_dts) {
2061 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2062 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2066 expected_dts += duration;
2069 type = TIMER_TYPE_LOST;
2072 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2073 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2074 expected_dts += duration;
2080 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2084 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2085 GstRtpJitterBufferPrivate *priv;
2087 priv = jitterbuffer->priv;
2089 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2092 if (priv->last_dts != -1)
2093 dtsdiff = dts - priv->last_dts;
2097 if (priv->last_rtptime != -1)
2098 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2102 priv->last_dts = dts;
2103 priv->last_rtptime = rtptime;
2107 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2110 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2112 diff = ABS (dtsdiff - rtpdiffns);
2114 /* jitter is stored in nanoseconds */
2115 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2117 GST_LOG_OBJECT (jitterbuffer,
2118 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2119 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2120 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2121 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2128 GST_DEBUG_OBJECT (jitterbuffer,
2129 "no dts or no clock-rate, can't calculate jitter");
2134 static GstFlowReturn
2135 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2138 GstRtpJitterBuffer *jitterbuffer;
2139 GstRtpJitterBufferPrivate *priv;
2141 guint32 expected, rtptime;
2142 GstFlowReturn ret = GST_FLOW_OK;
2143 GstClockTime dts, pts;
2148 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2149 gboolean do_next_seqnum = FALSE;
2150 RTPJitterBufferItem *item;
2151 GstMessage *msg = NULL;
2153 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2155 priv = jitterbuffer->priv;
2157 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2158 goto invalid_buffer;
2160 pt = gst_rtp_buffer_get_payload_type (&rtp);
2161 seqnum = gst_rtp_buffer_get_seq (&rtp);
2162 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2163 gst_rtp_buffer_unmap (&rtp);
2165 /* make sure we have PTS and DTS set */
2166 pts = GST_BUFFER_PTS (buffer);
2167 dts = GST_BUFFER_DTS (buffer);
2173 /* take the DTS of the buffer. This is the time when the packet was
2174 * received and is used to calculate jitter and clock skew. We will adjust
2175 * this DTS with the smoothed value after processing it in the
2176 * jitterbuffer and assign it as the PTS. */
2177 /* bring to running time */
2178 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2180 GST_DEBUG_OBJECT (jitterbuffer,
2181 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2182 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2184 JBUF_LOCK_CHECK (priv, out_flushing);
2186 if (G_UNLIKELY (priv->last_pt != pt)) {
2189 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2193 /* reset clock-rate so that we get a new one */
2194 priv->clock_rate = -1;
2196 /* Try to get the clock-rate from the caps first if we can. If there are no
2197 * caps we must fire the signal to get the clock-rate. */
2198 if ((caps = gst_pad_get_current_caps (pad))) {
2199 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2200 gst_caps_unref (caps);
2204 if (G_UNLIKELY (priv->clock_rate == -1)) {
2205 /* no clock rate given on the caps, try to get one with the signal */
2206 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2207 pt) == GST_FLOW_FLUSHING)
2210 if (G_UNLIKELY (priv->clock_rate == -1))
2214 /* don't accept more data on EOS */
2215 if (G_UNLIKELY (priv->eos))
2218 calculate_jitter (jitterbuffer, dts, rtptime);
2220 expected = priv->next_in_seqnum;
2222 /* now check against our expected seqnum */
2223 if (G_LIKELY (expected != -1)) {
2226 /* now calculate gap */
2227 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2229 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2230 expected, seqnum, gap);
2232 if (G_LIKELY (gap == 0)) {
2233 /* packet is expected */
2234 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2235 do_next_seqnum = TRUE;
2237 gboolean reset = FALSE;
2239 if (!GST_CLOCK_TIME_IS_VALID (dts)) {
2240 /* We would run into calculations with GST_CLOCK_TIME_NONE below
2241 * and can't compensate for anything without DTS on RTP packets
2243 goto gap_but_no_dts;
2244 } else if (gap < 0) {
2245 /* we received an old packet */
2246 if (G_UNLIKELY (gap < -RTP_MAX_MISORDER)) {
2247 /* too old packet, reset */
2248 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d < %d", gap,
2252 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2255 /* new packet, we are missing some packets */
2256 if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
2257 /* packet too far in future, reset */
2258 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too new %d > %d", gap,
2262 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2263 /* fill in the gap with EXPECTED timers */
2264 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2266 do_next_seqnum = TRUE;
2269 if (G_UNLIKELY (reset)) {
2270 GList *events = NULL, *l;
2272 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2273 rtp_jitter_buffer_flush (priv->jbuf,
2274 (GFunc) free_item_and_retain_events, &events);
2275 rtp_jitter_buffer_reset_skew (priv->jbuf);
2276 remove_all_timers (jitterbuffer);
2277 priv->last_popped_seqnum = -1;
2278 priv->next_seqnum = seqnum;
2279 do_next_seqnum = TRUE;
2281 /* Insert all sticky events again in order, otherwise we would
2282 * potentially loose STREAM_START, CAPS or SEGMENT events
2284 events = g_list_reverse (events);
2285 for (l = events; l; l = l->next) {
2286 RTPJitterBufferItem *item;
2288 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2289 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2291 g_list_free (events);
2293 JBUF_SIGNAL_EVENT (priv);
2295 /* reset spacing estimation when gap */
2296 priv->ips_rtptime = -1;
2297 priv->ips_dts = GST_CLOCK_TIME_NONE;
2300 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2301 /* we don't know what the next_in_seqnum should be, wait for the last
2302 * possible moment to push this buffer, maybe we get an earlier seqnum
2304 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2305 do_next_seqnum = TRUE;
2306 /* take rtptime and dts to calculate packet spacing */
2307 priv->ips_rtptime = rtptime;
2308 priv->ips_dts = dts;
2310 if (do_next_seqnum) {
2311 priv->last_in_seqnum = seqnum;
2312 priv->last_in_dts = dts;
2313 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2316 /* let's check if this buffer is too late, we can only accept packets with
2317 * bigger seqnum than the one we last pushed. */
2318 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2321 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2323 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2324 if (G_UNLIKELY (gap <= 0))
2328 /* let's drop oldest packet if the queue is already full and drop-on-latency
2329 * is set. We can only do this when there actually is a latency. When no
2330 * latency is set, we just pump it in the queue and let the other end push it
2331 * out as fast as possible. */
2332 if (priv->latency_ms && priv->drop_on_latency) {
2334 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2336 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2337 RTPJitterBufferItem *old_item;
2339 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2341 if (IS_DROPABLE (old_item)) {
2342 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2343 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2345 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2346 free_item (old_item);
2348 /* we might have removed some head buffers, signal the pushing thread to
2349 * see if it can push now */
2350 JBUF_SIGNAL_EVENT (priv);
2354 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2356 /* now insert the packet into the queue in sorted order. This function returns
2357 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2358 * have a duplicate. */
2359 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2364 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2366 /* we had an unhandled SR, handle it now */
2368 do_handle_sync (jitterbuffer);
2370 if (G_UNLIKELY (head)) {
2371 /* signal addition of new buffer when the _loop is waiting. */
2372 if (G_LIKELY (priv->active))
2373 JBUF_SIGNAL_EVENT (priv);
2375 /* let's unschedule and unblock any waiting buffers. We only want to do this
2376 * when the head buffer changed */
2377 if (G_UNLIKELY (priv->clock_id)) {
2378 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2379 unschedule_current_timer (jitterbuffer);
2383 GST_DEBUG_OBJECT (jitterbuffer,
2384 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2385 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2387 msg = check_buffering_percent (jitterbuffer, percent);
2393 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2400 /* this is not fatal but should be filtered earlier */
2401 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2402 ("Received invalid RTP payload, dropping"));
2403 gst_buffer_unref (buffer);
2408 GST_WARNING_OBJECT (jitterbuffer,
2409 "No clock-rate in caps!, dropping buffer");
2410 gst_buffer_unref (buffer);
2415 ret = priv->srcresult;
2416 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2417 gst_buffer_unref (buffer);
2423 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2424 gst_buffer_unref (buffer);
2429 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2430 " popped, dropping", seqnum, priv->last_popped_seqnum);
2432 gst_buffer_unref (buffer);
2437 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2439 priv->num_duplicates++;
2445 /* this is fatal as we can't compensate for gaps without DTS */
2446 GST_ELEMENT_ERROR (jitterbuffer, STREAM, DECODE, (NULL),
2447 ("Received packet without DTS after a gap"));
2448 gst_buffer_unref (buffer);
2449 ret = GST_FLOW_ERROR;
2455 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2457 guint64 ext_time, elapsed;
2459 GstRtpJitterBufferPrivate *priv;
2461 priv = jitterbuffer->priv;
2462 rtp_time = item->rtptime;
2464 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2465 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2467 if (rtp_time < priv->ext_timestamp) {
2468 ext_time = priv->ext_timestamp;
2470 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
2473 if (ext_time > priv->clock_base)
2474 elapsed = ext_time - priv->clock_base;
2478 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2483 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2484 RTPJitterBufferItem * item)
2486 guint64 total, elapsed, left, estimated;
2487 GstClockTime out_time;
2488 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2490 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2491 || priv->clock_base == -1 || priv->clock_rate <= 0)
2494 /* compute the elapsed time */
2495 elapsed = compute_elapsed (jitterbuffer, item);
2497 /* do nothing if elapsed time doesn't increment */
2498 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2501 priv->last_elapsed = elapsed;
2503 /* this is the total time we need to play */
2504 total = priv->npt_stop - priv->npt_start;
2505 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2506 GST_TIME_ARGS (total));
2508 /* this is how much time there is left */
2509 if (total > elapsed)
2510 left = total - elapsed;
2514 /* if we have less time left that the size of the buffer, we will not
2515 * be able to keep it filled, disabled buffering then */
2516 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2517 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2518 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2519 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2522 /* this is the current time as running-time */
2523 out_time = item->dts;
2526 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2528 /* if there is almost nothing left,
2529 * we may never advance enough to end up in the above case */
2530 if (total < GST_SECOND)
2531 estimated = GST_SECOND;
2535 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2536 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2538 if (estimated != -1 && priv->estimated_eos != estimated) {
2539 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2540 priv->estimated_eos = estimated;
2544 /* take a buffer from the queue and push it */
2545 static GstFlowReturn
2546 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2548 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2549 GstFlowReturn result = GST_FLOW_OK;
2550 RTPJitterBufferItem *item;
2551 GstBuffer *outbuf = NULL;
2552 GstEvent *outevent = NULL;
2553 GstQuery *outquery = NULL;
2554 GstClockTime dts, pts;
2556 gboolean do_push = TRUE;
2560 /* when we get here we are ready to pop and push the buffer */
2561 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2565 case ITEM_TYPE_BUFFER:
2567 /* we need to make writable to change the flags and timestamps */
2568 outbuf = gst_buffer_make_writable (item->data);
2570 if (G_UNLIKELY (priv->discont)) {
2571 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2572 * into the jitterbuffer so we can modify now. */
2573 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2574 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2575 priv->discont = FALSE;
2577 if (G_UNLIKELY (priv->ts_discont)) {
2578 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2579 priv->ts_discont = FALSE;
2583 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2585 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2587 /* apply timestamp with offset to buffer now */
2588 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2589 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2591 /* update the elapsed time when we need to check against the npt stop time. */
2592 update_estimated_eos (jitterbuffer, item);
2594 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2596 case ITEM_TYPE_LOST:
2597 priv->discont = TRUE;
2601 case ITEM_TYPE_EVENT:
2602 outevent = item->data;
2604 case ITEM_TYPE_QUERY:
2605 outquery = item->data;
2609 /* now we are ready to push the buffer. Save the seqnum and release the lock
2610 * so the other end can push stuff in the queue again. */
2612 priv->last_popped_seqnum = seqnum;
2613 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2615 msg = check_buffering_percent (jitterbuffer, percent);
2622 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2625 case ITEM_TYPE_BUFFER:
2627 GST_DEBUG_OBJECT (jitterbuffer,
2628 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2629 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2630 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2631 result = gst_pad_push (priv->srcpad, outbuf);
2633 JBUF_LOCK_CHECK (priv, out_flushing);
2635 case ITEM_TYPE_LOST:
2636 case ITEM_TYPE_EVENT:
2637 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2638 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
2641 gst_pad_push_event (priv->srcpad, outevent);
2643 gst_event_unref (outevent);
2645 result = GST_FLOW_OK;
2647 JBUF_LOCK_CHECK (priv, out_flushing);
2649 case ITEM_TYPE_QUERY:
2653 res = gst_pad_peer_query (priv->srcpad, outquery);
2655 JBUF_LOCK_CHECK (priv, out_flushing);
2656 result = GST_FLOW_OK;
2657 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
2658 JBUF_SIGNAL_QUERY (priv, res);
2667 return priv->srcresult;
2671 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2673 /* Peek a buffer and compare the seqnum to the expected seqnum.
2674 * If all is fine, the buffer is pushed.
2675 * If something is wrong, we wait for some event
2677 static GstFlowReturn
2678 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2680 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2681 GstFlowReturn result = GST_FLOW_OK;
2682 RTPJitterBufferItem *item;
2684 guint32 next_seqnum;
2687 /* only push buffers when PLAYING and active and not buffering */
2688 if (priv->blocked || !priv->active ||
2689 rtp_jitter_buffer_is_buffering (priv->jbuf))
2690 return GST_FLOW_WAIT;
2693 /* peek a buffer, we're just looking at the sequence number.
2694 * If all is fine, we'll pop and push it. If the sequence number is wrong we
2695 * wait for a timeout or something to change.
2696 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
2697 item = rtp_jitter_buffer_peek (priv->jbuf);
2701 /* get the seqnum and the next expected seqnum */
2702 seqnum = item->seqnum;
2706 next_seqnum = priv->next_seqnum;
2708 /* get the gap between this and the previous packet. If we don't know the
2709 * previous packet seqnum assume no gap. */
2710 if (G_UNLIKELY (next_seqnum == -1)) {
2711 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2712 /* we don't know what the next_seqnum should be, the chain function should
2713 * have scheduled a DEADLINE timer that will increment next_seqnum when it
2714 * fires, so wait for that */
2715 result = GST_FLOW_WAIT;
2717 /* else calculate GAP */
2718 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2720 if (G_LIKELY (gap == 0)) {
2722 /* no missing packet, pop and push */
2723 result = pop_and_push_next (jitterbuffer, seqnum);
2724 } else if (G_UNLIKELY (gap < 0)) {
2725 RTPJitterBufferItem *item;
2726 /* if we have a packet that we already pushed or considered dropped, pop it
2727 * off and get the next packet */
2728 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2729 seqnum, next_seqnum);
2730 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2734 /* the chain function has scheduled timers to request retransmission or
2735 * when to consider the packet lost, wait for that */
2736 GST_DEBUG_OBJECT (jitterbuffer,
2737 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2738 next_seqnum, seqnum, gap);
2739 result = GST_FLOW_WAIT;
2746 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
2748 result = GST_FLOW_EOS;
2750 result = GST_FLOW_WAIT;
2756 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
2758 GstClockTime rtx_retry_timeout;
2759 GstClockTime rtx_min_retry_timeout;
2761 if (priv->rtx_retry_timeout == -1) {
2762 if (priv->avg_rtx_rtt == 0)
2763 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
2765 /* we want to ask for a retransmission after we waited for a
2766 * complete RTT and the additional jitter */
2767 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
2769 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
2771 /* make sure we don't retry too often. On very low latency networks,
2772 * the RTT and jitter can be very low. */
2773 if (priv->rtx_min_retry_timeout == -1) {
2774 rtx_min_retry_timeout = priv->packet_spacing;
2776 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
2778 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
2780 return rtx_retry_timeout;
2784 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
2785 GstClockTime rtx_retry_timeout)
2787 GstClockTime rtx_retry_period;
2789 if (priv->rtx_retry_period == -1) {
2790 /* we retry up to the configured jitterbuffer size but leaving some
2791 * room for the retransmission to arrive in time */
2792 if (rtx_retry_timeout > priv->latency_ns) {
2793 rtx_retry_period = 0;
2795 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
2798 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
2800 return rtx_retry_period;
2803 /* the timeout for when we expected a packet expired */
2805 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2808 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2810 guint delay, delay_ms, avg_rtx_rtt_ms;
2811 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
2812 GstClockTime rtx_retry_period;
2813 GstClockTime rtx_retry_timeout;
2816 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
2817 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
2819 rtx_retry_timeout = get_rtx_retry_timeout (priv);
2820 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
2822 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
2823 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
2824 GST_TIME_ARGS (rtx_retry_period));
2826 delay = timer->rtx_delay + timer->rtx_retry;
2828 delay_ms = GST_TIME_AS_MSECONDS (delay);
2829 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
2830 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
2831 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
2833 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2834 gst_structure_new ("GstRTPRetransmissionRequest",
2835 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
2836 "running-time", G_TYPE_UINT64, timer->rtx_base,
2837 "delay", G_TYPE_UINT, delay_ms,
2838 "retry", G_TYPE_UINT, timer->num_rtx_retry,
2839 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
2840 "period", G_TYPE_UINT, rtx_retry_period_ms,
2841 "deadline", G_TYPE_UINT, priv->latency_ms,
2842 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
2843 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
2845 priv->num_rtx_requests++;
2846 timer->num_rtx_retry++;
2848 GST_OBJECT_LOCK (jitterbuffer);
2849 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
2850 timer->rtx_last = gst_clock_get_time (clock);
2851 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
2853 timer->rtx_last = now;
2855 GST_OBJECT_UNLOCK (jitterbuffer);
2857 /* calculate the timeout for the next retransmission attempt */
2858 timer->rtx_retry += rtx_retry_timeout;
2859 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
2860 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
2861 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
2862 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
2864 if (timer->rtx_retry + timer->rtx_delay > rtx_retry_period) {
2865 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
2866 /* too many retransmission request, we now convert the timer
2867 * to a lost timer, leave the num_rtx_retry as it is for stats */
2868 timer->type = TIMER_TYPE_LOST;
2869 timer->rtx_delay = 0;
2870 timer->rtx_retry = 0;
2872 reschedule_timer (jitterbuffer, timer, timer->seqnum,
2873 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
2876 gst_pad_push_event (priv->sinkpad, event);
2882 /* a packet is lost */
2884 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2887 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2888 GstClockTime duration, timestamp;
2889 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
2890 gboolean late, head;
2892 RTPJitterBufferItem *item;
2894 seqnum = timer->seqnum;
2895 timestamp = apply_offset (jitterbuffer, timer->timeout);
2896 duration = timer->duration;
2897 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
2898 duration = priv->packet_spacing;
2899 lost_packets = MAX (timer->num, 1);
2900 late = timer->num > 0;
2901 num_rtx_retry = timer->num_rtx_retry;
2903 /* we had a gap and thus we lost some packets. Create an event for this. */
2904 if (lost_packets > 1)
2905 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
2906 seqnum + lost_packets - 1);
2908 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
2910 priv->num_late += lost_packets;
2911 priv->num_rtx_failed += num_rtx_retry;
2913 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
2915 /* we now only accept seqnum bigger than this */
2916 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
2917 priv->next_in_seqnum = next_in_seqnum;
2919 /* create paket lost event */
2920 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
2921 gst_structure_new ("GstRTPPacketLost",
2922 "seqnum", G_TYPE_UINT, (guint) seqnum,
2923 "timestamp", G_TYPE_UINT64, timestamp,
2924 "duration", G_TYPE_UINT64, duration,
2925 "late", G_TYPE_BOOLEAN, late,
2926 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
2928 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
2929 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2931 /* remove timer now */
2932 remove_timer (jitterbuffer, timer);
2934 JBUF_SIGNAL_EVENT (priv);
2940 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2943 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2945 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
2946 remove_timer (jitterbuffer, timer);
2948 /* there was no EOS in the buffer, put one in there now */
2949 queue_event (jitterbuffer, gst_event_new_eos ());
2951 JBUF_SIGNAL_EVENT (priv);
2957 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2960 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2962 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
2964 /* timer seqnum might have been obsoleted by caps seqnum-base,
2965 * only mess with current ongoing seqnum if still unknown */
2966 if (priv->next_seqnum == -1)
2967 priv->next_seqnum = timer->seqnum;
2968 remove_timer (jitterbuffer, timer);
2969 JBUF_SIGNAL_EVENT (priv);
2975 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2978 gboolean removed = FALSE;
2980 switch (timer->type) {
2981 case TIMER_TYPE_EXPECTED:
2982 removed = do_expected_timeout (jitterbuffer, timer, now);
2984 case TIMER_TYPE_LOST:
2985 removed = do_lost_timeout (jitterbuffer, timer, now);
2987 case TIMER_TYPE_DEADLINE:
2988 removed = do_deadline_timeout (jitterbuffer, timer, now);
2990 case TIMER_TYPE_EOS:
2991 removed = do_eos_timeout (jitterbuffer, timer, now);
2997 /* called when we need to wait for the next timeout.
2999 * We loop over the array of recorded timeouts and wait for the earliest one.
3000 * When it timed out, do the logic associated with the timer.
3002 * If there are no timers, we wait on a gcond until something new happens.
3005 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3007 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3008 GstClockTime now = 0;
3011 while (priv->timer_running) {
3012 TimerData *timer = NULL;
3013 GstClockTime timer_timeout = -1;
3016 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3017 GST_TIME_ARGS (now));
3019 len = priv->timers->len;
3020 for (i = 0; i < len; i++) {
3021 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3022 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3023 gboolean save_best = FALSE;
3025 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3026 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3028 /* find the smallest timeout */
3029 if (timer == NULL) {
3031 } else if (timer_timeout == -1) {
3032 /* we already have an immediate timeout, the new timer must be an
3033 * immediate timer with smaller seqnum to become the best */
3034 if (test_timeout == -1
3035 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3036 timer->seqnum) > 0))
3038 } else if (test_timeout == -1) {
3039 /* first immediate timer */
3041 } else if (test_timeout < timer_timeout) {
3044 } else if (test_timeout == timer_timeout
3045 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3046 timer->seqnum) > 0)) {
3047 /* same timer, smaller seqnum */
3051 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3053 timer_timeout = test_timeout;
3056 if (timer && !priv->blocked) {
3058 GstClockTime sync_time;
3061 GstClockTimeDiff clock_jitter;
3063 if (timer_timeout == -1 || timer_timeout <= now) {
3064 do_timeout (jitterbuffer, timer, now);
3065 /* check here, do_timeout could have released the lock */
3066 if (!priv->timer_running)
3071 GST_OBJECT_LOCK (jitterbuffer);
3072 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3074 GST_OBJECT_UNLOCK (jitterbuffer);
3075 /* let's just push if there is no clock */
3076 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3077 now = timer_timeout;
3081 /* prepare for sync against clock */
3082 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3083 /* add latency of peer to get input time */
3084 sync_time += priv->peer_latency;
3086 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3087 " with sync time %" GST_TIME_FORMAT,
3088 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3090 /* create an entry for the clock */
3091 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3092 priv->timer_timeout = timer_timeout;
3093 priv->timer_seqnum = timer->seqnum;
3094 GST_OBJECT_UNLOCK (jitterbuffer);
3096 /* release the lock so that the other end can push stuff or unlock */
3099 ret = gst_clock_id_wait (id, &clock_jitter);
3102 if (!priv->timer_running) {
3103 gst_clock_id_unref (id);
3104 priv->clock_id = NULL;
3108 if (ret != GST_CLOCK_UNSCHEDULED) {
3109 now = timer_timeout + MAX (clock_jitter, 0);
3110 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3111 ret, priv->timer_seqnum, clock_jitter);
3113 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3115 /* and free the entry */
3116 gst_clock_id_unref (id);
3117 priv->clock_id = NULL;
3119 /* no timers, wait for activity */
3120 JBUF_WAIT_TIMER (priv);
3125 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3130 * This funcion implements the main pushing loop on the source pad.
3132 * It first tries to push as many buffers as possible. If there is a seqnum
3133 * mismatch, we wait for the next timeouts.
3136 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3138 GstRtpJitterBufferPrivate *priv;
3139 GstFlowReturn result = GST_FLOW_OK;
3141 priv = jitterbuffer->priv;
3143 JBUF_LOCK_CHECK (priv, flushing);
3145 result = handle_next_buffer (jitterbuffer);
3146 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3147 /* now wait for the next event */
3148 JBUF_WAIT_EVENT (priv, flushing);
3149 result = GST_FLOW_OK;
3152 while (result == GST_FLOW_OK);
3153 /* store result for upstream */
3154 priv->srcresult = result;
3155 /* if we get here we need to pause */
3161 result = priv->srcresult;
3168 JBUF_SIGNAL_QUERY (priv, FALSE);
3171 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3172 gst_flow_get_name (result));
3173 gst_pad_pause_task (priv->srcpad);
3174 if (result == GST_FLOW_EOS) {
3175 event = gst_event_new_eos ();
3176 gst_pad_push_event (priv->srcpad, event);
3182 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3183 * some sanity checks and then emit the handle-sync signal with the parameters.
3184 * This function must be called with the LOCK */
3186 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3188 GstRtpJitterBufferPrivate *priv;
3189 guint64 base_rtptime, base_time;
3191 guint64 last_rtptime;
3193 guint64 ext_rtptime, diff;
3194 gboolean valid = TRUE, keep = FALSE;
3196 priv = jitterbuffer->priv;
3198 /* get the last values from the jitterbuffer */
3199 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3200 &clock_rate, &last_rtptime);
3202 clock_base = priv->clock_base;
3203 ext_rtptime = priv->ext_rtptime;
3205 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3206 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3207 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3208 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3210 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3211 /* we keep this SR packet for later. When we get a valid RTP packet the
3212 * above values will be set and we can try to use the SR packet */
3213 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3216 /* we can't accept anything that happened before we did the last resync */
3217 if (base_rtptime > ext_rtptime) {
3218 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3221 /* the SR RTP timestamp must be something close to what we last observed
3222 * in the jitterbuffer */
3223 if (ext_rtptime > last_rtptime) {
3224 /* check how far ahead it is to our RTP timestamps */
3225 diff = ext_rtptime - last_rtptime;
3226 /* if bigger than 1 second, we drop it */
3227 if (diff > clock_rate) {
3228 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3229 /* should drop this, but some RTSP servers end up with bogus
3230 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3231 * so still trigger rptbin sync but invalidate RTCP data
3232 * (sync might use other methods) */
3235 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3236 G_GUINT64_FORMAT, last_rtptime, diff);
3242 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3246 s = gst_structure_new ("application/x-rtp-sync",
3247 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3248 "base-time", G_TYPE_UINT64, base_time,
3249 "clock-rate", G_TYPE_UINT, clock_rate,
3250 "clock-base", G_TYPE_UINT64, clock_base,
3251 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3252 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3254 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3255 gst_buffer_replace (&priv->last_sr, NULL);
3257 g_signal_emit (jitterbuffer,
3258 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3260 gst_structure_free (s);
3262 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3263 gst_buffer_replace (&priv->last_sr, NULL);
3267 static GstFlowReturn
3268 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3271 GstRtpJitterBuffer *jitterbuffer;
3272 GstRtpJitterBufferPrivate *priv;
3273 GstFlowReturn ret = GST_FLOW_OK;
3275 GstRTCPPacket packet;
3276 guint64 ext_rtptime;
3278 GstRTCPBuffer rtcp = { NULL, };
3280 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3282 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
3283 goto invalid_buffer;
3285 priv = jitterbuffer->priv;
3287 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3289 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3292 /* first packet must be SR or RR or else the validate would have failed */
3293 switch (gst_rtcp_packet_get_type (&packet)) {
3294 case GST_RTCP_TYPE_SR:
3295 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3301 gst_rtcp_buffer_unmap (&rtcp);
3303 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3306 /* convert the RTP timestamp to our extended timestamp, using the same offset
3307 * we used in the jitterbuffer */
3308 ext_rtptime = priv->jbuf->ext_rtptime;
3309 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3311 priv->ext_rtptime = ext_rtptime;
3312 gst_buffer_replace (&priv->last_sr, buffer);
3314 do_handle_sync (jitterbuffer);
3318 gst_buffer_unref (buffer);
3324 /* this is not fatal but should be filtered earlier */
3325 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3326 ("Received invalid RTCP payload, dropping"));
3332 /* this is not fatal but should be filtered earlier */
3333 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3334 ("Received empty RTCP payload, dropping"));
3335 gst_rtcp_buffer_unmap (&rtcp);
3341 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3342 gst_rtcp_buffer_unmap (&rtcp);
3349 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3352 gboolean res = FALSE;
3353 GstRtpJitterBuffer *jitterbuffer;
3354 GstRtpJitterBufferPrivate *priv;
3356 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3357 priv = jitterbuffer->priv;
3359 switch (GST_QUERY_TYPE (query)) {
3360 case GST_QUERY_CAPS:
3362 GstCaps *filter, *caps;
3364 gst_query_parse_caps (query, &filter);
3365 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3366 gst_query_set_caps_result (query, caps);
3367 gst_caps_unref (caps);
3372 if (GST_QUERY_IS_SERIALIZED (query)) {
3373 RTPJitterBufferItem *item;
3376 JBUF_LOCK_CHECK (priv, out_flushing);
3377 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3378 RTP_JITTER_BUFFER_MODE_BUFFER) {
3379 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3380 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3381 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3383 JBUF_SIGNAL_EVENT (priv);
3384 JBUF_WAIT_QUERY (priv, out_flushing);
3385 res = priv->last_query;
3387 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3392 res = gst_pad_query_default (pad, parent, query);
3400 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3408 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3411 GstRtpJitterBuffer *jitterbuffer;
3412 GstRtpJitterBufferPrivate *priv;
3413 gboolean res = FALSE;
3415 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3416 priv = jitterbuffer->priv;
3418 switch (GST_QUERY_TYPE (query)) {
3419 case GST_QUERY_LATENCY:
3421 /* We need to send the query upstream and add the returned latency to our
3423 GstClockTime min_latency, max_latency;
3425 GstClockTime our_latency;
3427 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3428 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3430 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3431 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3432 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3434 /* store this so that we can safely sync on the peer buffers. */
3436 priv->peer_latency = min_latency;
3437 our_latency = priv->latency_ns;
3440 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3441 GST_TIME_ARGS (our_latency));
3443 /* we add some latency but can buffer an infinite amount of time */
3444 min_latency += our_latency;
3447 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3448 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3449 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3451 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3455 case GST_QUERY_POSITION:
3457 GstClockTime start, last_out;
3460 gst_query_parse_position (query, &fmt, NULL);
3461 if (fmt != GST_FORMAT_TIME) {
3462 res = gst_pad_query_default (pad, parent, query);
3467 start = priv->npt_start;
3468 last_out = priv->last_out_time;
3471 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3472 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3473 GST_TIME_ARGS (last_out));
3475 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3476 /* bring 0-based outgoing time to stream time */
3477 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3480 res = gst_pad_query_default (pad, parent, query);
3484 case GST_QUERY_CAPS:
3486 GstCaps *filter, *caps;
3488 gst_query_parse_caps (query, &filter);
3489 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3490 gst_query_set_caps_result (query, caps);
3491 gst_caps_unref (caps);
3496 res = gst_pad_query_default (pad, parent, query);
3504 gst_rtp_jitter_buffer_set_property (GObject * object,
3505 guint prop_id, const GValue * value, GParamSpec * pspec)
3507 GstRtpJitterBuffer *jitterbuffer;
3508 GstRtpJitterBufferPrivate *priv;
3510 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3511 priv = jitterbuffer->priv;
3516 guint new_latency, old_latency;
3518 new_latency = g_value_get_uint (value);
3521 old_latency = priv->latency_ms;
3522 priv->latency_ms = new_latency;
3523 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3524 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3527 /* post message if latency changed, this will inform the parent pipeline
3528 * that a latency reconfiguration is possible/needed. */
3529 if (new_latency != old_latency) {
3530 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3531 GST_TIME_ARGS (new_latency * GST_MSECOND));
3533 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3534 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3538 case PROP_DROP_ON_LATENCY:
3540 priv->drop_on_latency = g_value_get_boolean (value);
3543 case PROP_TS_OFFSET:
3545 priv->ts_offset = g_value_get_int64 (value);
3546 priv->ts_discont = TRUE;
3551 priv->do_lost = g_value_get_boolean (value);
3556 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3559 case PROP_DO_RETRANSMISSION:
3561 priv->do_retransmission = g_value_get_boolean (value);
3564 case PROP_RTX_DELAY:
3566 priv->rtx_delay = g_value_get_int (value);
3569 case PROP_RTX_MIN_DELAY:
3571 priv->rtx_min_delay = g_value_get_uint (value);
3574 case PROP_RTX_DELAY_REORDER:
3576 priv->rtx_delay_reorder = g_value_get_int (value);
3579 case PROP_RTX_RETRY_TIMEOUT:
3581 priv->rtx_retry_timeout = g_value_get_int (value);
3584 case PROP_RTX_MIN_RETRY_TIMEOUT:
3586 priv->rtx_min_retry_timeout = g_value_get_int (value);
3589 case PROP_RTX_RETRY_PERIOD:
3591 priv->rtx_retry_period = g_value_get_int (value);
3595 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3601 gst_rtp_jitter_buffer_get_property (GObject * object,
3602 guint prop_id, GValue * value, GParamSpec * pspec)
3604 GstRtpJitterBuffer *jitterbuffer;
3605 GstRtpJitterBufferPrivate *priv;
3607 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3608 priv = jitterbuffer->priv;
3613 g_value_set_uint (value, priv->latency_ms);
3616 case PROP_DROP_ON_LATENCY:
3618 g_value_set_boolean (value, priv->drop_on_latency);
3621 case PROP_TS_OFFSET:
3623 g_value_set_int64 (value, priv->ts_offset);
3628 g_value_set_boolean (value, priv->do_lost);
3633 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3641 if (priv->srcresult != GST_FLOW_OK)
3644 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3646 g_value_set_int (value, percent);
3650 case PROP_DO_RETRANSMISSION:
3652 g_value_set_boolean (value, priv->do_retransmission);
3655 case PROP_RTX_DELAY:
3657 g_value_set_int (value, priv->rtx_delay);
3660 case PROP_RTX_MIN_DELAY:
3662 g_value_set_uint (value, priv->rtx_min_delay);
3665 case PROP_RTX_DELAY_REORDER:
3667 g_value_set_int (value, priv->rtx_delay_reorder);
3670 case PROP_RTX_RETRY_TIMEOUT:
3672 g_value_set_int (value, priv->rtx_retry_timeout);
3675 case PROP_RTX_MIN_RETRY_TIMEOUT:
3677 g_value_set_int (value, priv->rtx_min_retry_timeout);
3680 case PROP_RTX_RETRY_PERIOD:
3682 g_value_set_int (value, priv->rtx_retry_period);
3686 g_value_take_boxed (value,
3687 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
3690 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3695 static GstStructure *
3696 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
3700 JBUF_LOCK (jbuf->priv);
3701 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
3702 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
3703 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
3704 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
3705 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
3706 JBUF_UNLOCK (jbuf->priv);