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_NEXT_SEQNUM TRUE
132 #define DEFAULT_RTX_DELAY -1
133 #define DEFAULT_RTX_MIN_DELAY 0
134 #define DEFAULT_RTX_DELAY_REORDER 3
135 #define DEFAULT_RTX_RETRY_TIMEOUT -1
136 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
137 #define DEFAULT_RTX_RETRY_PERIOD -1
138 #define DEFAULT_RTX_MAX_RETRIES -1
139 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
141 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
142 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
148 PROP_DROP_ON_LATENCY,
153 PROP_DO_RETRANSMISSION,
154 PROP_RTX_NEXT_SEQNUM,
157 PROP_RTX_DELAY_REORDER,
158 PROP_RTX_RETRY_TIMEOUT,
159 PROP_RTX_MIN_RETRY_TIMEOUT,
160 PROP_RTX_RETRY_PERIOD,
161 PROP_RTX_MAX_RETRIES,
163 PROP_MAX_RTCP_RTP_TIME_DIFF
166 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
168 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
170 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
173 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
175 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
176 GST_DEBUG ("waiting timer"); \
177 (priv)->waiting_timer = TRUE; \
178 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
179 (priv)->waiting_timer = FALSE; \
180 GST_DEBUG ("waiting timer done"); \
182 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
183 if (G_UNLIKELY ((priv)->waiting_timer)) { \
184 GST_DEBUG ("signal timer"); \
185 g_cond_signal (&(priv)->jbuf_timer); \
189 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
190 GST_DEBUG ("waiting event"); \
191 (priv)->waiting_event = TRUE; \
192 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
193 (priv)->waiting_event = FALSE; \
194 GST_DEBUG ("waiting event done"); \
195 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
198 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
199 if (G_UNLIKELY ((priv)->waiting_event)) { \
200 GST_DEBUG ("signal event"); \
201 g_cond_signal (&(priv)->jbuf_event); \
205 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
206 GST_DEBUG ("waiting query"); \
207 (priv)->waiting_query = TRUE; \
208 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
209 (priv)->waiting_query = FALSE; \
210 GST_DEBUG ("waiting query done"); \
211 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
214 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
215 (priv)->last_query = res; \
216 if (G_UNLIKELY ((priv)->waiting_query)) { \
217 GST_DEBUG ("signal query"); \
218 g_cond_signal (&(priv)->jbuf_query); \
223 struct _GstRtpJitterBufferPrivate
225 GstPad *sinkpad, *srcpad;
228 RTPJitterBuffer *jbuf;
230 gboolean waiting_timer;
232 gboolean waiting_event;
234 gboolean waiting_query;
242 gboolean timer_running;
243 GThread *timer_thread;
248 gboolean drop_on_latency;
251 gboolean do_retransmission;
252 gboolean rtx_next_seqnum;
255 gint rtx_delay_reorder;
256 gint rtx_retry_timeout;
257 gint rtx_min_retry_timeout;
258 gint rtx_retry_period;
259 gint rtx_max_retries;
260 gint max_rtcp_rtp_time_diff;
262 /* the last seqnum we pushed out */
263 guint32 last_popped_seqnum;
264 /* the next expected seqnum we push */
266 /* seqnum-base, if known */
268 /* last output time */
269 GstClockTime last_out_time;
270 /* last valid input timestamp and rtptime pair */
271 GstClockTime ips_dts;
273 GstClockTime packet_spacing;
277 /* the next expected seqnum we receive */
278 GstClockTime last_in_dts;
279 guint32 next_in_seqnum;
283 /* start and stop ranges */
284 GstClockTime npt_start;
285 GstClockTime npt_stop;
286 guint64 ext_timestamp;
287 guint64 last_elapsed;
288 guint64 estimated_eos;
295 /* clock rate and rtp timestamp offset */
299 gint64 prev_ts_offset;
301 /* when we are shutting down */
302 GstFlowReturn srcresult;
308 GstClockTime timer_timeout;
309 guint16 timer_seqnum;
310 /* the latency of the upstream peer, we have to take this into account when
311 * synchronizing the buffers. */
312 GstClockTime peer_latency;
316 /* some accounting */
318 guint64 num_duplicates;
319 guint64 num_rtx_requests;
320 guint64 num_rtx_success;
321 guint64 num_rtx_failed;
326 GstClockTime last_dts;
327 guint64 last_rtptime;
328 GstClockTime avg_jitter;
345 GstClockTime timeout;
346 GstClockTime duration;
347 GstClockTime rtx_base;
348 GstClockTime rtx_delay;
349 GstClockTime rtx_retry;
350 GstClockTime rtx_last;
354 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
355 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
356 GstRtpJitterBufferPrivate))
358 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
359 GST_STATIC_PAD_TEMPLATE ("sink",
362 GST_STATIC_CAPS ("application/x-rtp"
363 /* "clock-rate = (int) [ 1, 2147483647 ], "
364 * "payload = (int) , "
365 * "encoding-name = (string) "
369 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
370 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
373 GST_STATIC_CAPS ("application/x-rtcp")
376 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
377 GST_STATIC_PAD_TEMPLATE ("src",
380 GST_STATIC_CAPS ("application/x-rtp"
381 /* "payload = (int) , "
382 * "clock-rate = (int) , "
383 * "encoding-name = (string) "
387 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
389 #define gst_rtp_jitter_buffer_parent_class parent_class
390 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
392 /* object overrides */
393 static void gst_rtp_jitter_buffer_set_property (GObject * object,
394 guint prop_id, const GValue * value, GParamSpec * pspec);
395 static void gst_rtp_jitter_buffer_get_property (GObject * object,
396 guint prop_id, GValue * value, GParamSpec * pspec);
397 static void gst_rtp_jitter_buffer_finalize (GObject * object);
399 /* element overrides */
400 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
401 * element, GstStateChange transition);
402 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
403 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
404 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
406 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
409 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
410 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
413 /* sinkpad overrides */
414 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
415 GstObject * parent, GstEvent * event);
416 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
417 GstObject * parent, GstBuffer * buffer);
419 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
420 GstObject * parent, GstEvent * event);
421 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
422 GstObject * parent, GstBuffer * buffer);
424 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
425 GstObject * parent, GstQuery * query);
427 /* srcpad overrides */
428 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
429 GstObject * parent, GstEvent * event);
430 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
431 GstObject * parent, GstPadMode mode, gboolean active);
432 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
433 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
434 GstObject * parent, GstQuery * query);
437 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
439 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
440 gboolean active, guint64 base_time);
441 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
443 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
444 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
446 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
448 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
452 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
454 GObjectClass *gobject_class;
455 GstElementClass *gstelement_class;
457 gobject_class = (GObjectClass *) klass;
458 gstelement_class = (GstElementClass *) klass;
460 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
462 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
464 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
465 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
468 * GstRtpJitterBuffer:latency:
470 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
471 * for at most this time.
473 g_object_class_install_property (gobject_class, PROP_LATENCY,
474 g_param_spec_uint ("latency", "Buffer latency in ms",
475 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
476 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
478 * GstRtpJitterBuffer:drop-on-latency:
480 * Drop oldest buffers when the queue is completely filled.
482 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
483 g_param_spec_boolean ("drop-on-latency",
484 "Drop buffers when maximum latency is reached",
485 "Tells the jitterbuffer to never exceed the given latency in size",
486 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
488 * GstRtpJitterBuffer:ts-offset:
490 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
491 * This is mainly used to ensure interstream synchronisation.
493 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
494 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
495 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
496 G_MAXINT64, DEFAULT_TS_OFFSET,
497 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
500 * GstRtpJitterBuffer:do-lost:
502 * Send out a GstRTPPacketLost event downstream when a packet is considered
505 g_object_class_install_property (gobject_class, PROP_DO_LOST,
506 g_param_spec_boolean ("do-lost", "Do Lost",
507 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
508 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
511 * GstRtpJitterBuffer:mode:
513 * Control the buffering and timestamping mode used by the jitterbuffer.
515 g_object_class_install_property (gobject_class, PROP_MODE,
516 g_param_spec_enum ("mode", "Mode",
517 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
518 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
520 * GstRtpJitterBuffer:percent:
522 * The percent of the jitterbuffer that is filled.
524 g_object_class_install_property (gobject_class, PROP_PERCENT,
525 g_param_spec_int ("percent", "percent",
526 "The buffer filled percent", 0, 100,
527 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
529 * GstRtpJitterBuffer:do-retransmission:
531 * Send out a GstRTPRetransmission event upstream when a packet is considered
532 * late and should be retransmitted.
536 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
537 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
538 "Send retransmission events upstream when a packet is late",
539 DEFAULT_DO_RETRANSMISSION,
540 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
543 * GstRtpJitterBuffer:rtx-next-seqnum
545 * Estimate when the next packet should arrive and schedule a retransmission
547 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
548 * for packet N+1. So it will be requested if it does not arrive at the expected time.
549 * The expected time is calculated using the dts of N and the packet spacing.
553 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
554 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
555 "Estimate when the next packet should arrive and schedule a "
556 "retransmission request for it.",
557 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
560 * GstRtpJitterBuffer:rtx-delay:
562 * When a packet did not arrive at the expected time, wait this extra amount
563 * of time before sending a retransmission event.
565 * When -1 is used, the max jitter will be used as extra delay.
569 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
570 g_param_spec_int ("rtx-delay", "RTX Delay",
571 "Extra time in ms to wait before sending retransmission "
572 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
573 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
576 * GstRtpJitterBuffer:rtx-min-delay:
578 * When a packet did not arrive at the expected time, wait at least this extra amount
579 * of time before sending a retransmission event.
583 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
584 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
585 "Minimum time in ms to wait before sending retransmission "
586 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
587 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
589 * GstRtpJitterBuffer:rtx-delay-reorder:
591 * Assume that a retransmission event should be sent when we see
592 * this much packet reordering.
594 * When -1 is used, the value will be estimated based on observed packet
599 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
600 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
601 "Sending retransmission event when this much reordering (-1 automatic)",
602 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
603 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
605 * GstRtpJitterBuffer::rtx-retry-timeout:
607 * When no packet has been received after sending a retransmission event
608 * for this time, retry sending a retransmission event.
610 * When -1 is used, the value will be estimated based on observed round
615 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
616 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
617 "Retry sending a transmission event after this timeout in "
618 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
619 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
621 * GstRtpJitterBuffer::rtx-min-retry-timeout:
623 * The minimum amount of time between retry timeouts. When
624 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
625 * minimum interval between retry timeouts.
627 * When -1 is used, the value will be estimated based on the
632 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
633 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
634 "Minimum timeout between sending a transmission event in "
635 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
636 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
638 * GstRtpJitterBuffer:rtx-retry-period:
640 * The amount of time to try to get a retransmission.
642 * When -1 is used, the value will be estimated based on the jitterbuffer
643 * latency and the observed round trip time.
647 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
648 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
649 "Try to get a retransmission for this many ms "
650 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
651 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
653 * GstRtpJitterBuffer:rtx-max-retries:
655 * The maximum number of retries to request a retransmission.
657 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
658 * When -1 is used, the number of retransmission request will not be limited.
662 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
663 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
664 "The maximum number of retries to request a retransmission. "
665 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
666 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
668 * GstRtpJitterBuffer:stats:
670 * Various jitterbuffer statistics. This property returns a GstStructure
671 * with name application/x-rtp-jitterbuffer-stats with the following fields:
677 * <classname>"rtx-count"</classname>:
678 * the number of retransmissions requested.
684 * <classname>"rtx-success-count"</classname>:
685 * the number of successful retransmissions.
691 * <classname>"rtx-per-packet"</classname>:
692 * average number of RTX per packet.
698 * <classname>"rtx-rtt"</classname>:
699 * average round trip time per RTX.
706 g_object_class_install_property (gobject_class, PROP_STATS,
707 g_param_spec_boxed ("stats", "Statistics",
708 "Various statistics", GST_TYPE_STRUCTURE,
709 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
712 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
714 * The maximum amount of time in ms that the RTP time in the RTCP SRs
715 * is allowed to be ahead of the last RTP packet we received. Use
716 * -1 to disable ignoring of RTCP packets.
720 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
721 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
722 "Maximum amount of time in ms that the RTP time in RTCP SRs "
723 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
724 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
725 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
728 * GstRtpJitterBuffer::request-pt-map:
729 * @buffer: the object which received the signal
732 * Request the payload type as #GstCaps for @pt.
734 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
735 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
736 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
737 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
738 GST_TYPE_CAPS, 1, G_TYPE_UINT);
740 * GstRtpJitterBuffer::handle-sync:
741 * @buffer: the object which received the signal
742 * @struct: a GstStructure containing sync values.
744 * Be notified of new sync values.
746 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
747 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
748 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
749 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
750 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
753 * GstRtpJitterBuffer::on-npt-stop:
754 * @buffer: the object which received the signal
756 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
757 * the npt-stop position.
759 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
760 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
761 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
762 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
763 G_TYPE_NONE, 0, G_TYPE_NONE);
766 * GstRtpJitterBuffer::clear-pt-map:
767 * @buffer: the object which received the signal
769 * Invalidate the clock-rate as obtained with the
770 * #GstRtpJitterBuffer::request-pt-map signal.
772 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
773 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
774 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
775 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
776 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
779 * GstRtpJitterBuffer::set-active:
780 * @buffer: the object which received the signal
782 * Start pushing out packets with the given base time. This signal is only
783 * useful in buffering mode.
785 * Returns: the time of the last pushed packet.
787 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
788 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
789 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
790 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
791 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
794 gstelement_class->change_state =
795 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
796 gstelement_class->request_new_pad =
797 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
798 gstelement_class->release_pad =
799 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
800 gstelement_class->provide_clock =
801 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
803 gst_element_class_add_pad_template (gstelement_class,
804 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
805 gst_element_class_add_pad_template (gstelement_class,
806 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
807 gst_element_class_add_pad_template (gstelement_class,
808 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
810 gst_element_class_set_static_metadata (gstelement_class,
811 "RTP packet jitter-buffer", "Filter/Network/RTP",
812 "A buffer that deals with network jitter and other transmission faults",
813 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
814 "Wim Taymans <wim.taymans@gmail.com>");
816 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
817 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
819 GST_DEBUG_CATEGORY_INIT
820 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
824 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
826 GstRtpJitterBufferPrivate *priv;
828 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
829 jitterbuffer->priv = priv;
831 priv->latency_ms = DEFAULT_LATENCY_MS;
832 priv->latency_ns = priv->latency_ms * GST_MSECOND;
833 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
834 priv->do_lost = DEFAULT_DO_LOST;
835 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
836 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
837 priv->rtx_delay = DEFAULT_RTX_DELAY;
838 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
839 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
840 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
841 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
842 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
843 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
844 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
847 priv->last_rtptime = -1;
848 priv->avg_jitter = 0;
849 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
850 priv->jbuf = rtp_jitter_buffer_new ();
851 g_mutex_init (&priv->jbuf_lock);
852 g_cond_init (&priv->jbuf_timer);
853 g_cond_init (&priv->jbuf_event);
854 g_cond_init (&priv->jbuf_query);
855 g_queue_init (&priv->gap_packets);
857 /* reset skew detection initialy */
858 rtp_jitter_buffer_reset_skew (priv->jbuf);
859 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
860 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
864 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
867 gst_pad_set_activatemode_function (priv->srcpad,
868 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
869 gst_pad_set_query_function (priv->srcpad,
870 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
871 gst_pad_set_event_function (priv->srcpad,
872 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
875 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
878 gst_pad_set_chain_function (priv->sinkpad,
879 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
880 gst_pad_set_event_function (priv->sinkpad,
881 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
882 gst_pad_set_query_function (priv->sinkpad,
883 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
885 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
886 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
888 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
891 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
893 #define ITEM_TYPE_BUFFER 0
894 #define ITEM_TYPE_LOST 1
895 #define ITEM_TYPE_EVENT 2
896 #define ITEM_TYPE_QUERY 3
898 static RTPJitterBufferItem *
899 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
900 guint seqnum, guint count, guint rtptime)
902 RTPJitterBufferItem *item;
904 item = g_slice_new (RTPJitterBufferItem);
911 item->seqnum = seqnum;
913 item->rtptime = rtptime;
919 free_item (RTPJitterBufferItem * item)
921 g_return_if_fail (item != NULL);
923 if (item->data && item->type != ITEM_TYPE_QUERY)
924 gst_mini_object_unref (item->data);
925 g_slice_free (RTPJitterBufferItem, item);
929 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
931 GList **l = user_data;
933 if (item->data && item->type == ITEM_TYPE_EVENT
934 && GST_EVENT_IS_STICKY (item->data)) {
935 *l = g_list_prepend (*l, item->data);
936 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
937 gst_mini_object_unref (item->data);
939 g_slice_free (RTPJitterBufferItem, item);
943 gst_rtp_jitter_buffer_finalize (GObject * object)
945 GstRtpJitterBuffer *jitterbuffer;
946 GstRtpJitterBufferPrivate *priv;
948 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
949 priv = jitterbuffer->priv;
951 g_array_free (priv->timers, TRUE);
952 g_mutex_clear (&priv->jbuf_lock);
953 g_cond_clear (&priv->jbuf_timer);
954 g_cond_clear (&priv->jbuf_event);
955 g_cond_clear (&priv->jbuf_query);
957 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
958 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
959 g_queue_clear (&priv->gap_packets);
960 g_object_unref (priv->jbuf);
962 G_OBJECT_CLASS (parent_class)->finalize (object);
966 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
968 GstRtpJitterBuffer *jitterbuffer;
969 GstPad *otherpad = NULL;
970 GstIterator *it = NULL;
973 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
975 if (pad == jitterbuffer->priv->sinkpad) {
976 otherpad = jitterbuffer->priv->srcpad;
977 } else if (pad == jitterbuffer->priv->srcpad) {
978 otherpad = jitterbuffer->priv->sinkpad;
979 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
980 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
984 g_value_init (&val, GST_TYPE_PAD);
985 g_value_set_object (&val, otherpad);
986 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
987 g_value_unset (&val);
994 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
996 GstRtpJitterBufferPrivate *priv;
998 priv = jitterbuffer->priv;
1000 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1003 gst_pad_new_from_static_template
1004 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1005 gst_pad_set_chain_function (priv->rtcpsinkpad,
1006 gst_rtp_jitter_buffer_chain_rtcp);
1007 gst_pad_set_event_function (priv->rtcpsinkpad,
1008 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1009 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1010 gst_rtp_jitter_buffer_iterate_internal_links);
1011 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1012 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1014 return priv->rtcpsinkpad;
1018 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1020 GstRtpJitterBufferPrivate *priv;
1022 priv = jitterbuffer->priv;
1024 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1026 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1028 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1029 priv->rtcpsinkpad = NULL;
1033 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1034 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1036 GstRtpJitterBuffer *jitterbuffer;
1037 GstElementClass *klass;
1039 GstRtpJitterBufferPrivate *priv;
1041 g_return_val_if_fail (templ != NULL, NULL);
1042 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1044 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1045 priv = jitterbuffer->priv;
1046 klass = GST_ELEMENT_GET_CLASS (element);
1048 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1050 /* figure out the template */
1051 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1052 if (priv->rtcpsinkpad != NULL)
1055 result = create_rtcp_sink (jitterbuffer);
1057 goto wrong_template;
1064 g_warning ("rtpjitterbuffer: this is not our template");
1069 g_warning ("rtpjitterbuffer: pad already requested");
1075 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1077 GstRtpJitterBuffer *jitterbuffer;
1078 GstRtpJitterBufferPrivate *priv;
1080 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1081 g_return_if_fail (GST_IS_PAD (pad));
1083 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1084 priv = jitterbuffer->priv;
1086 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1088 if (priv->rtcpsinkpad == pad) {
1089 remove_rtcp_sink (jitterbuffer);
1098 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1104 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1106 return gst_system_clock_obtain ();
1110 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1112 GstRtpJitterBufferPrivate *priv;
1114 priv = jitterbuffer->priv;
1116 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1119 priv->clock_rate = -1;
1120 /* do not clear current content, but refresh state for new arrival */
1121 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1122 rtp_jitter_buffer_reset_skew (priv->jbuf);
1127 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1130 GstRtpJitterBufferPrivate *priv;
1131 GstClockTime last_out;
1132 RTPJitterBufferItem *item;
1137 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1138 active, GST_TIME_ARGS (offset));
1140 if (active != priv->active) {
1141 /* add the amount of time spent in paused to the output offset. All
1142 * outgoing buffers will have this offset applied to their timestamps in
1143 * order to make them arrive in time in the sink. */
1144 priv->out_offset = offset;
1145 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1146 GST_TIME_ARGS (priv->out_offset));
1147 priv->active = active;
1148 JBUF_SIGNAL_EVENT (priv);
1151 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1153 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1154 /* head buffer timestamp and offset gives our output time */
1155 last_out = item->dts + priv->ts_offset;
1157 /* use last known time when the buffer is empty */
1158 last_out = priv->last_out_time;
1166 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1168 GstRtpJitterBuffer *jitterbuffer;
1169 GstRtpJitterBufferPrivate *priv;
1174 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1175 priv = jitterbuffer->priv;
1177 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1179 caps = gst_pad_peer_query_caps (other, filter);
1181 templ = gst_pad_get_pad_template_caps (pad);
1183 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1188 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1190 intersect = gst_caps_intersect (caps, templ);
1191 gst_caps_unref (caps);
1192 gst_caps_unref (templ);
1196 gst_object_unref (jitterbuffer);
1202 * Must be called with JBUF_LOCK held
1206 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1209 GstRtpJitterBufferPrivate *priv;
1210 GstStructure *caps_struct;
1214 priv = jitterbuffer->priv;
1216 /* first parse the caps */
1217 caps_struct = gst_caps_get_structure (caps, 0);
1219 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1221 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1222 * measure the amount of data in the buffer */
1223 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1226 if (priv->clock_rate <= 0)
1229 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1231 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1233 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1234 * can use this to track the amount of time elapsed on the sender. */
1235 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1236 priv->clock_base = val;
1238 priv->clock_base = -1;
1240 priv->ext_timestamp = priv->clock_base;
1242 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1245 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1246 /* first expected seqnum, only update when we didn't have a previous base. */
1247 if (priv->next_in_seqnum == -1)
1248 priv->next_in_seqnum = val;
1249 if (priv->next_seqnum == -1) {
1250 priv->next_seqnum = val;
1251 JBUF_SIGNAL_EVENT (priv);
1253 priv->seqnum_base = val;
1255 priv->seqnum_base = -1;
1258 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1260 /* the start and stop times. The seqnum-base corresponds to the start time. We
1261 * will keep track of the seqnums on the output and when we reach the one
1262 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1263 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1264 priv->npt_start = tval;
1266 priv->npt_start = 0;
1268 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1269 priv->npt_stop = tval;
1271 priv->npt_stop = -1;
1273 GST_DEBUG_OBJECT (jitterbuffer,
1274 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1275 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1282 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1287 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1293 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1295 GstRtpJitterBufferPrivate *priv;
1297 priv = jitterbuffer->priv;
1300 /* mark ourselves as flushing */
1301 priv->srcresult = GST_FLOW_FLUSHING;
1302 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1303 /* this unblocks any waiting pops on the src pad task */
1304 JBUF_SIGNAL_EVENT (priv);
1305 JBUF_SIGNAL_QUERY (priv, FALSE);
1310 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1312 GstRtpJitterBufferPrivate *priv;
1314 priv = jitterbuffer->priv;
1317 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1318 /* Mark as non flushing */
1319 priv->srcresult = GST_FLOW_OK;
1320 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1321 priv->last_popped_seqnum = -1;
1322 priv->last_out_time = -1;
1323 priv->next_seqnum = -1;
1324 priv->seqnum_base = -1;
1325 priv->ips_rtptime = -1;
1326 priv->ips_dts = GST_CLOCK_TIME_NONE;
1327 priv->packet_spacing = 0;
1328 priv->next_in_seqnum = -1;
1329 priv->clock_rate = -1;
1332 priv->estimated_eos = -1;
1333 priv->last_elapsed = 0;
1334 priv->ext_timestamp = -1;
1335 priv->avg_jitter = 0;
1336 priv->last_dts = -1;
1337 priv->last_rtptime = -1;
1338 priv->last_in_dts = 0;
1339 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1340 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1341 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1342 rtp_jitter_buffer_reset_skew (priv->jbuf);
1343 remove_all_timers (jitterbuffer);
1344 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1345 g_queue_clear (&priv->gap_packets);
1350 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1351 GstPadMode mode, gboolean active)
1354 GstRtpJitterBuffer *jitterbuffer = NULL;
1356 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1359 case GST_PAD_MODE_PUSH:
1361 /* allow data processing */
1362 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1364 /* start pushing out buffers */
1365 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1366 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1367 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1369 /* make sure all data processing stops ASAP */
1370 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1372 /* NOTE this will hardlock if the state change is called from the src pad
1373 * task thread because we will _join() the thread. */
1374 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1375 result = gst_pad_stop_task (pad);
1385 static GstStateChangeReturn
1386 gst_rtp_jitter_buffer_change_state (GstElement * element,
1387 GstStateChange transition)
1389 GstRtpJitterBuffer *jitterbuffer;
1390 GstRtpJitterBufferPrivate *priv;
1391 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1393 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1394 priv = jitterbuffer->priv;
1396 switch (transition) {
1397 case GST_STATE_CHANGE_NULL_TO_READY:
1399 case GST_STATE_CHANGE_READY_TO_PAUSED:
1401 /* reset negotiated values */
1402 priv->clock_rate = -1;
1403 priv->clock_base = -1;
1404 priv->peer_latency = 0;
1406 /* block until we go to PLAYING */
1407 priv->blocked = TRUE;
1408 priv->timer_running = TRUE;
1409 priv->timer_thread =
1410 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1413 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1415 /* unblock to allow streaming in PLAYING */
1416 priv->blocked = FALSE;
1417 JBUF_SIGNAL_EVENT (priv);
1418 JBUF_SIGNAL_TIMER (priv);
1425 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1427 switch (transition) {
1428 case GST_STATE_CHANGE_READY_TO_PAUSED:
1429 /* we are a live element because we sync to the clock, which we can only
1430 * do in the PLAYING state */
1431 if (ret != GST_STATE_CHANGE_FAILURE)
1432 ret = GST_STATE_CHANGE_NO_PREROLL;
1434 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1436 /* block to stop streaming when PAUSED */
1437 priv->blocked = TRUE;
1438 unschedule_current_timer (jitterbuffer);
1440 if (ret != GST_STATE_CHANGE_FAILURE)
1441 ret = GST_STATE_CHANGE_NO_PREROLL;
1443 case GST_STATE_CHANGE_PAUSED_TO_READY:
1445 gst_buffer_replace (&priv->last_sr, NULL);
1446 priv->timer_running = FALSE;
1447 unschedule_current_timer (jitterbuffer);
1448 JBUF_SIGNAL_TIMER (priv);
1449 JBUF_SIGNAL_QUERY (priv, FALSE);
1451 g_thread_join (priv->timer_thread);
1452 priv->timer_thread = NULL;
1454 case GST_STATE_CHANGE_READY_TO_NULL:
1464 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1467 gboolean ret = TRUE;
1468 GstRtpJitterBuffer *jitterbuffer;
1469 GstRtpJitterBufferPrivate *priv;
1471 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1472 priv = jitterbuffer->priv;
1474 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1476 switch (GST_EVENT_TYPE (event)) {
1477 case GST_EVENT_LATENCY:
1479 GstClockTime latency;
1481 gst_event_parse_latency (event, &latency);
1483 GST_DEBUG_OBJECT (jitterbuffer,
1484 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1487 /* adjust the overall buffer delay to the total pipeline latency in
1488 * buffering mode because if downstream consumes too fast (because of
1489 * large latency or queues, we would start rebuffering again. */
1490 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1491 RTP_JITTER_BUFFER_MODE_BUFFER) {
1492 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1496 ret = gst_pad_push_event (priv->sinkpad, event);
1500 ret = gst_pad_push_event (priv->sinkpad, event);
1507 /* handles and stores the event in the jitterbuffer, must be called with
1510 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1512 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1513 RTPJitterBufferItem *item;
1516 switch (GST_EVENT_TYPE (event)) {
1517 case GST_EVENT_CAPS:
1521 gst_event_parse_caps (event, &caps);
1522 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1525 case GST_EVENT_SEGMENT:
1526 gst_event_copy_segment (event, &priv->segment);
1528 /* we need time for now */
1529 if (priv->segment.format != GST_FORMAT_TIME)
1530 goto newseg_wrong_format;
1532 GST_DEBUG_OBJECT (jitterbuffer,
1533 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1537 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1544 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1545 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1546 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1548 JBUF_SIGNAL_EVENT (priv);
1553 newseg_wrong_format:
1555 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1556 gst_event_unref (event);
1562 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1565 gboolean ret = TRUE;
1566 GstRtpJitterBuffer *jitterbuffer;
1567 GstRtpJitterBufferPrivate *priv;
1569 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1570 priv = jitterbuffer->priv;
1572 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1574 switch (GST_EVENT_TYPE (event)) {
1575 case GST_EVENT_FLUSH_START:
1576 ret = gst_pad_push_event (priv->srcpad, event);
1577 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1578 /* wait for the loop to go into PAUSED */
1579 gst_pad_pause_task (priv->srcpad);
1581 case GST_EVENT_FLUSH_STOP:
1582 ret = gst_pad_push_event (priv->srcpad, event);
1584 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1585 GST_PAD_MODE_PUSH, TRUE);
1588 if (GST_EVENT_IS_SERIALIZED (event)) {
1589 /* serialized events go in the queue */
1591 if (priv->srcresult != GST_FLOW_OK) {
1592 /* Errors in sticky event pushing are no problem and ignored here
1593 * as they will cause more meaningful errors during data flow.
1594 * For EOS events, that are not followed by data flow, we still
1595 * return FALSE here though.
1597 if (!GST_EVENT_IS_STICKY (event) ||
1598 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1599 goto out_flow_error;
1601 /* refuse more events on EOS */
1604 ret = queue_event (jitterbuffer, event);
1607 /* non-serialized events are forwarded downstream immediately */
1608 ret = gst_pad_push_event (priv->srcpad, event);
1617 GST_DEBUG_OBJECT (jitterbuffer,
1618 "refusing event, we have a downstream flow error: %s",
1619 gst_flow_get_name (priv->srcresult));
1621 gst_event_unref (event);
1626 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1628 gst_event_unref (event);
1634 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1637 gboolean ret = TRUE;
1638 GstRtpJitterBuffer *jitterbuffer;
1640 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1642 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1644 switch (GST_EVENT_TYPE (event)) {
1645 case GST_EVENT_FLUSH_START:
1646 gst_event_unref (event);
1648 case GST_EVENT_FLUSH_STOP:
1649 gst_event_unref (event);
1652 ret = gst_pad_event_default (pad, parent, event);
1660 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1661 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1662 * GST_FLOW_FLUSHING when the element is shutting down. On success
1663 * GST_FLOW_OK is returned.
1665 static GstFlowReturn
1666 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1670 GValue args[2] = { {0}, {0} };
1674 g_value_init (&args[0], GST_TYPE_ELEMENT);
1675 g_value_set_object (&args[0], jitterbuffer);
1676 g_value_init (&args[1], G_TYPE_UINT);
1677 g_value_set_uint (&args[1], pt);
1679 g_value_init (&ret, GST_TYPE_CAPS);
1680 g_value_set_boxed (&ret, NULL);
1682 JBUF_UNLOCK (jitterbuffer->priv);
1683 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1685 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1687 g_value_unset (&args[0]);
1688 g_value_unset (&args[1]);
1689 caps = (GstCaps *) g_value_dup_boxed (&ret);
1690 g_value_unset (&ret);
1694 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1695 gst_caps_unref (caps);
1697 if (G_UNLIKELY (!res))
1705 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1706 return GST_FLOW_ERROR;
1710 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1711 return GST_FLOW_FLUSHING;
1715 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1716 return GST_FLOW_ERROR;
1720 /* call with jbuf lock held */
1722 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1724 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1725 GstMessage *message = NULL;
1730 /* Post a buffering message */
1731 if (priv->last_percent != percent) {
1732 priv->last_percent = percent;
1734 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1735 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1742 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1744 GstRtpJitterBufferPrivate *priv;
1746 priv = jitterbuffer->priv;
1748 if (timestamp == -1)
1751 /* apply the timestamp offset, this is used for inter stream sync */
1752 timestamp += priv->ts_offset;
1753 /* add the offset, this is used when buffering */
1754 timestamp += priv->out_offset;
1760 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1762 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1763 TimerData *timer = NULL;
1766 len = priv->timers->len;
1767 for (i = 0; i < len; i++) {
1768 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1769 if (test->seqnum == seqnum && test->type == type) {
1778 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1780 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1782 if (priv->clock_id) {
1783 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1784 gst_clock_id_unschedule (priv->clock_id);
1785 priv->clock_id = NULL;
1790 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1792 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1793 GstClockTime test_timeout;
1795 if ((test_timeout = timer->timeout) == -1)
1798 if (timer->type != TIMER_TYPE_EXPECTED) {
1799 /* add our latency and offset to get output times. */
1800 test_timeout = apply_offset (jitterbuffer, test_timeout);
1801 test_timeout += priv->latency_ns;
1803 return test_timeout;
1807 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1809 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1811 if (priv->clock_id) {
1812 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1814 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1815 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1817 if (timeout == -1 || timeout < priv->timer_timeout)
1818 unschedule_current_timer (jitterbuffer);
1823 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1824 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1825 GstClockTime duration)
1827 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1831 GST_DEBUG_OBJECT (jitterbuffer,
1832 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1833 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1834 GST_TIME_ARGS (delay));
1836 len = priv->timers->len;
1837 g_array_set_size (priv->timers, len + 1);
1838 timer = &g_array_index (priv->timers, TimerData, len);
1841 timer->seqnum = seqnum;
1843 timer->timeout = timeout + delay;
1844 timer->duration = duration;
1845 if (type == TIMER_TYPE_EXPECTED) {
1846 timer->rtx_base = timeout;
1847 timer->rtx_delay = delay;
1848 timer->rtx_retry = 0;
1850 timer->num_rtx_retry = 0;
1851 recalculate_timer (jitterbuffer, timer);
1852 JBUF_SIGNAL_TIMER (priv);
1858 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1859 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1861 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1862 gboolean seqchange, timechange;
1865 seqchange = timer->seqnum != seqnum;
1866 timechange = timer->timeout != timeout;
1868 if (!seqchange && !timechange)
1871 oldseq = timer->seqnum;
1873 GST_DEBUG_OBJECT (jitterbuffer,
1874 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1875 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1877 timer->timeout = timeout + delay;
1878 timer->seqnum = seqnum;
1880 timer->rtx_base = timeout;
1881 timer->rtx_delay = delay;
1882 timer->rtx_retry = 0;
1885 timer->num_rtx_retry = 0;
1887 if (priv->clock_id) {
1888 /* we changed the seqnum and there is a timer currently waiting with this
1889 * seqnum, unschedule it */
1890 if (seqchange && priv->timer_seqnum == oldseq)
1891 unschedule_current_timer (jitterbuffer);
1892 /* we changed the time, check if it is earlier than what we are waiting
1893 * for and unschedule if so */
1894 else if (timechange)
1895 recalculate_timer (jitterbuffer, timer);
1900 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1901 guint16 seqnum, GstClockTime timeout)
1905 /* find the seqnum timer */
1906 timer = find_timer (jitterbuffer, type, seqnum);
1907 if (timer == NULL) {
1908 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1910 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1916 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1918 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1921 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1922 unschedule_current_timer (jitterbuffer);
1925 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1926 g_array_remove_index_fast (priv->timers, idx);
1931 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1933 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1934 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1935 g_array_set_size (priv->timers, 0);
1936 unschedule_current_timer (jitterbuffer);
1939 /* get the extra delay to wait before sending RTX */
1941 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1945 if (priv->rtx_delay == -1) {
1946 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
1947 delay = DEFAULT_AUTO_RTX_DELAY;
1949 /* jitter is in nanoseconds, maximum of 2x jitter and half the
1950 * packet spacing is a good margin */
1951 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
1954 delay = priv->rtx_delay * GST_MSECOND;
1956 if (priv->rtx_min_delay > 0)
1957 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1962 /* we just received a packet with seqnum and dts.
1964 * First check for old seqnum that we are still expecting. If the gap with the
1965 * current seqnum is too big, unschedule the timeouts.
1967 * If we have a valid packet spacing estimate we can set a timer for when we
1968 * should receive the next packet.
1969 * If we don't have a valid estimate, we remove any timer we might have
1970 * had for this packet.
1973 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1974 GstClockTime dts, gboolean do_next_seqnum)
1976 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1977 TimerData *timer = NULL;
1980 /* go through all timers and unschedule the ones with a large gap, also find
1981 * the timer for the seqnum */
1982 len = priv->timers->len;
1983 for (i = 0; i < len; i++) {
1984 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1987 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1989 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
1990 test->type, test->seqnum, seqnum, gap);
1993 GST_DEBUG ("found timer for current seqnum");
1994 /* the timer for the current seqnum */
1996 /* when no retransmission, we can stop now, we only need to find the
1997 * timer for the current seqnum */
1998 if (!priv->do_retransmission)
2000 } else if (gap > priv->rtx_delay_reorder) {
2001 /* max gap, we exceeded the max reorder distance and we don't expect the
2002 * missing packet to be this reordered */
2003 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2004 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2008 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2009 && priv->do_retransmission && priv->rtx_next_seqnum;
2011 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2012 if (timer->num_rtx_retry > 0) {
2013 GstClockTime rtx_last, delay;
2015 /* we scheduled a retry for this packet and now we have it */
2016 priv->num_rtx_success++;
2017 /* all the previous retry attempts failed */
2018 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2019 /* number of retries before receiving the packet */
2020 if (priv->avg_rtx_num == 0.0)
2021 priv->avg_rtx_num = timer->num_rtx_retry;
2023 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2024 /* calculate the delay between retransmission request and receiving this
2025 * packet, start with when we scheduled this timeout last */
2026 rtx_last = timer->rtx_last;
2027 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2028 /* we have a valid delay if this packet arrived after we scheduled the
2030 delay = dts - rtx_last;
2031 if (priv->avg_rtx_rtt == 0)
2032 priv->avg_rtx_rtt = delay;
2034 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2038 GST_LOG_OBJECT (jitterbuffer,
2039 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2040 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2041 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2042 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2043 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2044 GST_TIME_ARGS (priv->avg_rtx_rtt));
2046 /* don't try to estimate the next seqnum because this is a retransmitted
2047 * packet and it probably did not arrive with the expected packet
2049 do_next_seqnum = FALSE;
2053 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2054 GstClockTime expected, delay;
2056 /* calculate expected arrival time of the next seqnum */
2057 expected = dts + priv->packet_spacing;
2059 delay = get_rtx_delay (priv);
2061 /* and update/install timer for next seqnum */
2063 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2066 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2067 expected, delay, priv->packet_spacing);
2069 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2070 /* if we had a timer, remove it, we don't know when to expect the next
2072 remove_timer (jitterbuffer, timer);
2077 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2080 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2082 /* we need consecutive seqnums with a different
2083 * rtptime to estimate the packet spacing. */
2084 if (priv->ips_rtptime != rtptime) {
2085 /* rtptime changed, check dts diff */
2086 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2087 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2088 GstClockTime old_packet_spacing = priv->packet_spacing;
2090 /* Biased towards bigger packet spacings to prevent
2091 * too many unneeded retransmission requests for next
2092 * packets that just arrive a little later than we would
2094 if (old_packet_spacing > new_packet_spacing)
2095 priv->packet_spacing =
2096 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2097 else if (old_packet_spacing > 0)
2098 priv->packet_spacing =
2099 (3 * new_packet_spacing + old_packet_spacing) / 4;
2101 priv->packet_spacing = new_packet_spacing;
2103 GST_DEBUG_OBJECT (jitterbuffer,
2104 "new packet spacing %" GST_TIME_FORMAT
2105 " old packet spacing %" GST_TIME_FORMAT
2106 " combined to %" GST_TIME_FORMAT,
2107 GST_TIME_ARGS (new_packet_spacing),
2108 GST_TIME_ARGS (old_packet_spacing),
2109 GST_TIME_ARGS (priv->packet_spacing));
2111 priv->ips_rtptime = rtptime;
2112 priv->ips_dts = dts;
2117 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2118 guint16 seqnum, GstClockTime dts, gint gap)
2120 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2121 GstClockTime total_duration, duration, expected_dts;
2124 GST_DEBUG_OBJECT (jitterbuffer,
2125 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2126 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2128 if (dts == GST_CLOCK_TIME_NONE) {
2129 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2133 /* the total duration spanned by the missing packets */
2134 if (dts >= priv->last_in_dts)
2135 total_duration = dts - priv->last_in_dts;
2139 /* interpolate between the current time and the last time based on
2140 * number of packets we are missing, this is the estimated duration
2141 * for the missing packet based on equidistant packet spacing. */
2142 duration = total_duration / (gap + 1);
2144 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2145 GST_TIME_ARGS (duration));
2147 if (total_duration > priv->latency_ns) {
2148 GstClockTime gap_time;
2152 GstClockTime gap_dur = gap * duration;
2153 if (gap_dur > priv->latency_ns)
2154 gap_time = gap_dur - priv->latency_ns;
2157 lost_packets = gap_time / duration;
2159 gap_time = total_duration - priv->latency_ns;
2163 /* too many lost packets, some of the missing packets are already
2164 * too late and we can generate lost packet events for them. */
2165 GST_DEBUG_OBJECT (jitterbuffer,
2166 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2167 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2168 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2169 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2170 GST_TIME_ARGS (gap_time));
2172 /* this timer will fire immediately and the lost event will be pushed from
2173 * the timer thread */
2174 if (lost_packets > 0) {
2175 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2176 priv->last_in_dts + duration, 0, gap_time);
2177 expected += lost_packets;
2178 priv->last_in_dts += gap_time;
2182 expected_dts = priv->last_in_dts + duration;
2184 if (priv->do_retransmission) {
2187 type = TIMER_TYPE_EXPECTED;
2188 /* if we had a timer for the first missing packet, update it. */
2189 if ((timer = find_timer (jitterbuffer, type, expected))) {
2190 GstClockTime timeout = timer->timeout;
2192 timer->duration = duration;
2193 if (timeout > (expected_dts + timer->rtx_retry)) {
2194 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2195 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2199 expected_dts += duration;
2202 type = TIMER_TYPE_LOST;
2205 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2206 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2207 expected_dts += duration;
2213 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2217 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2218 GstRtpJitterBufferPrivate *priv;
2220 priv = jitterbuffer->priv;
2222 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2225 if (priv->last_dts != -1)
2226 dtsdiff = dts - priv->last_dts;
2230 if (priv->last_rtptime != -1)
2231 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2235 priv->last_dts = dts;
2236 priv->last_rtptime = rtptime;
2240 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2243 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2245 diff = ABS (dtsdiff - rtpdiffns);
2247 /* jitter is stored in nanoseconds */
2248 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2250 GST_LOG_OBJECT (jitterbuffer,
2251 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2252 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2253 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2254 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2261 GST_DEBUG_OBJECT (jitterbuffer,
2262 "no dts or no clock-rate, can't calculate jitter");
2268 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2270 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2271 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2274 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2275 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2276 gst_rtp_buffer_unmap (&rtp_a);
2278 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2279 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2280 gst_rtp_buffer_unmap (&rtp_b);
2282 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2286 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2287 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap)
2289 GstRtpJitterBufferPrivate *priv;
2290 guint gap_packets_length;
2291 gboolean reset = FALSE;
2293 priv = jitterbuffer->priv;
2295 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2297 guint32 prev_gap_seq = -1;
2298 gboolean all_consecutive = TRUE;
2300 g_queue_insert_sorted (&priv->gap_packets, buffer,
2301 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2303 for (l = priv->gap_packets.head; l; l = l->next) {
2304 GstBuffer *gap_buffer = l->data;
2305 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2308 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2310 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2312 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2313 if (prev_gap_seq == -1)
2314 prev_gap_seq = gap_seq;
2315 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2316 all_consecutive = FALSE;
2318 prev_gap_seq = gap_seq;
2320 gst_rtp_buffer_unmap (&gap_rtp);
2321 if (!all_consecutive)
2325 if (all_consecutive && gap_packets_length > 3) {
2326 GST_DEBUG_OBJECT (jitterbuffer,
2327 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2328 (future ? "new" : "old"), gap,
2329 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2331 } else if (!all_consecutive) {
2332 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2333 g_queue_clear (&priv->gap_packets);
2334 GST_DEBUG_OBJECT (jitterbuffer,
2335 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2336 (future ? "new" : "old"), gap,
2337 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2340 GST_DEBUG_OBJECT (jitterbuffer,
2341 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2342 (future ? "new" : "old"), gap,
2343 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER),
2344 gap_packets_length + 1);
2348 GST_DEBUG_OBJECT (jitterbuffer,
2349 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2350 gap, -RTP_MAX_MISORDER);
2351 g_queue_push_tail (&priv->gap_packets, buffer);
2359 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2361 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2362 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2365 GstClockTime base_time =
2366 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2367 GstClockTime clock_time = gst_clock_get_time (clock);
2369 if (clock_time > base_time)
2370 running_time = clock_time - base_time;
2374 gst_object_unref (clock);
2377 return running_time;
2380 static GstFlowReturn
2381 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2384 GstRtpJitterBuffer *jitterbuffer;
2385 GstRtpJitterBufferPrivate *priv;
2387 guint32 expected, rtptime;
2388 GstFlowReturn ret = GST_FLOW_OK;
2389 GstClockTime dts, pts;
2394 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2395 gboolean do_next_seqnum = FALSE;
2396 RTPJitterBufferItem *item;
2397 GstMessage *msg = NULL;
2398 gboolean estimated_dts = FALSE;
2400 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2402 priv = jitterbuffer->priv;
2404 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2405 goto invalid_buffer;
2407 pt = gst_rtp_buffer_get_payload_type (&rtp);
2408 seqnum = gst_rtp_buffer_get_seq (&rtp);
2409 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2410 gst_rtp_buffer_unmap (&rtp);
2412 /* make sure we have PTS and DTS set */
2413 pts = GST_BUFFER_PTS (buffer);
2414 dts = GST_BUFFER_DTS (buffer);
2421 /* If we have no DTS here, i.e. no capture time, get one from the
2422 * clock now to have something to calculate with in the future. */
2423 dts = get_current_running_time (jitterbuffer);
2426 /* Remember that we estimated the DTS if we are running already
2427 * and this is not our first packet (or first packet after a reset).
2428 * If it's the first packet, we somehow must generate a timestamp for
2429 * everything, otherwise we can't calculate any times
2431 estimated_dts = (priv->next_in_seqnum != -1);
2433 /* take the DTS of the buffer. This is the time when the packet was
2434 * received and is used to calculate jitter and clock skew. We will adjust
2435 * this DTS with the smoothed value after processing it in the
2436 * jitterbuffer and assign it as the PTS. */
2437 /* bring to running time */
2438 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2441 GST_DEBUG_OBJECT (jitterbuffer,
2442 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2443 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2445 JBUF_LOCK_CHECK (priv, out_flushing);
2447 if (G_UNLIKELY (priv->last_pt != pt)) {
2450 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2454 /* reset clock-rate so that we get a new one */
2455 priv->clock_rate = -1;
2457 /* Try to get the clock-rate from the caps first if we can. If there are no
2458 * caps we must fire the signal to get the clock-rate. */
2459 if ((caps = gst_pad_get_current_caps (pad))) {
2460 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2461 gst_caps_unref (caps);
2465 if (G_UNLIKELY (priv->clock_rate == -1)) {
2466 /* no clock rate given on the caps, try to get one with the signal */
2467 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2468 pt) == GST_FLOW_FLUSHING)
2471 if (G_UNLIKELY (priv->clock_rate == -1))
2475 /* don't accept more data on EOS */
2476 if (G_UNLIKELY (priv->eos))
2479 calculate_jitter (jitterbuffer, dts, rtptime);
2481 if (priv->seqnum_base != -1) {
2484 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2487 GST_DEBUG_OBJECT (jitterbuffer,
2488 "packet seqnum #%d before seqnum-base #%d", seqnum,
2490 gst_buffer_unref (buffer);
2493 } else if (gap > 16384) {
2494 /* From now on don't compare against the seqnum base anymore as
2495 * at some point in the future we will wrap around and also that
2496 * much reordering is very unlikely */
2497 priv->seqnum_base = -1;
2501 expected = priv->next_in_seqnum;
2503 /* now check against our expected seqnum */
2504 if (G_LIKELY (expected != -1)) {
2507 /* now calculate gap */
2508 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2510 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2511 expected, seqnum, gap);
2513 if (G_LIKELY (gap == 0)) {
2514 /* packet is expected */
2515 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2516 do_next_seqnum = TRUE;
2518 gboolean reset = FALSE;
2521 /* we received an old packet */
2522 if (G_UNLIKELY (gap != -1 && gap < -RTP_MAX_MISORDER)) {
2524 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2528 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2531 /* new packet, we are missing some packets */
2532 if (G_UNLIKELY (priv->timers->len >= RTP_MAX_DROPOUT)) {
2533 /* If we have timers for more than RTP_MAX_DROPOUT packets
2534 * pending this means that we have a huge gap overall. We can
2535 * reset the jitterbuffer at this point because there's
2536 * just too much data missing to be able to do anything
2537 * sensible with the past data. Just try again from the
2539 GST_WARNING_OBJECT (jitterbuffer,
2540 "%d pending timers > %d - resetting", priv->timers->len,
2543 gst_buffer_unref (buffer);
2545 } else if (G_UNLIKELY (gap >= RTP_MAX_DROPOUT)) {
2547 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2551 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2552 /* fill in the gap with EXPECTED timers */
2553 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2555 do_next_seqnum = TRUE;
2558 if (G_UNLIKELY (reset)) {
2559 GList *events = NULL, *l;
2562 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2563 rtp_jitter_buffer_flush (priv->jbuf,
2564 (GFunc) free_item_and_retain_events, &events);
2565 rtp_jitter_buffer_reset_skew (priv->jbuf);
2566 remove_all_timers (jitterbuffer);
2567 priv->discont = TRUE;
2568 priv->last_popped_seqnum = -1;
2569 priv->next_seqnum = seqnum;
2571 priv->last_in_dts = -1;
2572 priv->next_in_seqnum = -1;
2574 /* Insert all sticky events again in order, otherwise we would
2575 * potentially loose STREAM_START, CAPS or SEGMENT events
2577 events = g_list_reverse (events);
2578 for (l = events; l; l = l->next) {
2579 RTPJitterBufferItem *item;
2581 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2582 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2584 g_list_free (events);
2586 JBUF_SIGNAL_EVENT (priv);
2588 /* reset spacing estimation when gap */
2589 priv->ips_rtptime = -1;
2590 priv->ips_dts = GST_CLOCK_TIME_NONE;
2592 buffers = g_list_copy (priv->gap_packets.head);
2593 g_queue_clear (&priv->gap_packets);
2595 priv->ips_rtptime = -1;
2596 priv->ips_dts = GST_CLOCK_TIME_NONE;
2597 JBUF_UNLOCK (jitterbuffer->priv);
2599 for (l = buffers; l; l = l->next) {
2600 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2602 if (ret != GST_FLOW_OK)
2605 for (; l; l = l->next)
2606 gst_buffer_unref (l->data);
2607 g_list_free (buffers);
2611 /* reset spacing estimation when gap */
2612 priv->ips_rtptime = -1;
2613 priv->ips_dts = GST_CLOCK_TIME_NONE;
2616 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2618 /* we don't know what the next_in_seqnum should be, wait for the last
2619 * possible moment to push this buffer, maybe we get an earlier seqnum
2621 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2622 do_next_seqnum = TRUE;
2623 /* take rtptime and dts to calculate packet spacing */
2624 priv->ips_rtptime = rtptime;
2625 priv->ips_dts = dts;
2628 /* We had no huge gap, let's drop all the gap packets */
2629 if (buffer != NULL) {
2630 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2631 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2632 g_queue_clear (&priv->gap_packets);
2634 GST_DEBUG_OBJECT (jitterbuffer,
2635 "Had big gap, waiting for more consecutive packets");
2636 JBUF_UNLOCK (jitterbuffer->priv);
2640 if (do_next_seqnum) {
2641 priv->last_in_dts = dts;
2642 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2645 /* let's check if this buffer is too late, we can only accept packets with
2646 * bigger seqnum than the one we last pushed. */
2647 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2650 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2652 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2653 if (G_UNLIKELY (gap <= 0))
2657 /* let's drop oldest packet if the queue is already full and drop-on-latency
2658 * is set. We can only do this when there actually is a latency. When no
2659 * latency is set, we just pump it in the queue and let the other end push it
2660 * out as fast as possible. */
2661 if (priv->latency_ms && priv->drop_on_latency) {
2663 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2665 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2666 RTPJitterBufferItem *old_item;
2668 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2670 if (IS_DROPABLE (old_item)) {
2671 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2672 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2674 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2675 free_item (old_item);
2677 /* we might have removed some head buffers, signal the pushing thread to
2678 * see if it can push now */
2679 JBUF_SIGNAL_EVENT (priv);
2683 /* If we estimated the DTS, don't consider it in the clock skew calculations
2684 * later. The code above always sets dts to pts or the other way around if
2685 * any of those is valid in the buffer, so we know that if we estimated the
2686 * dts that both are unknown */
2689 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2690 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2692 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2694 /* now insert the packet into the queue in sorted order. This function returns
2695 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2696 * have a duplicate. */
2697 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2702 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2704 /* we had an unhandled SR, handle it now */
2706 do_handle_sync (jitterbuffer);
2708 if (G_UNLIKELY (head)) {
2709 /* signal addition of new buffer when the _loop is waiting. */
2710 if (G_LIKELY (priv->active))
2711 JBUF_SIGNAL_EVENT (priv);
2713 /* let's unschedule and unblock any waiting buffers. We only want to do this
2714 * when the head buffer changed */
2715 if (G_UNLIKELY (priv->clock_id)) {
2716 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2717 unschedule_current_timer (jitterbuffer);
2721 GST_DEBUG_OBJECT (jitterbuffer,
2722 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2723 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2725 msg = check_buffering_percent (jitterbuffer, percent);
2731 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2738 /* this is not fatal but should be filtered earlier */
2739 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2740 ("Received invalid RTP payload, dropping"));
2741 gst_buffer_unref (buffer);
2746 GST_WARNING_OBJECT (jitterbuffer,
2747 "No clock-rate in caps!, dropping buffer");
2748 gst_buffer_unref (buffer);
2753 ret = priv->srcresult;
2754 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2755 gst_buffer_unref (buffer);
2761 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2762 gst_buffer_unref (buffer);
2767 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2768 " popped, dropping", seqnum, priv->last_popped_seqnum);
2770 gst_buffer_unref (buffer);
2775 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2777 priv->num_duplicates++;
2784 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2786 guint64 ext_time, elapsed;
2788 GstRtpJitterBufferPrivate *priv;
2790 priv = jitterbuffer->priv;
2791 rtp_time = item->rtptime;
2793 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2794 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2796 ext_time = priv->ext_timestamp;
2797 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2798 if (ext_time < priv->ext_timestamp) {
2799 ext_time = priv->ext_timestamp;
2801 priv->ext_timestamp = ext_time;
2804 if (ext_time > priv->clock_base)
2805 elapsed = ext_time - priv->clock_base;
2809 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2814 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2815 RTPJitterBufferItem * item)
2817 guint64 total, elapsed, left, estimated;
2818 GstClockTime out_time;
2819 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2821 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2822 || priv->clock_base == -1 || priv->clock_rate <= 0)
2825 /* compute the elapsed time */
2826 elapsed = compute_elapsed (jitterbuffer, item);
2828 /* do nothing if elapsed time doesn't increment */
2829 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2832 priv->last_elapsed = elapsed;
2834 /* this is the total time we need to play */
2835 total = priv->npt_stop - priv->npt_start;
2836 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2837 GST_TIME_ARGS (total));
2839 /* this is how much time there is left */
2840 if (total > elapsed)
2841 left = total - elapsed;
2845 /* if we have less time left that the size of the buffer, we will not
2846 * be able to keep it filled, disabled buffering then */
2847 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2848 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2849 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2850 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2853 /* this is the current time as running-time */
2854 out_time = item->dts;
2857 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2859 /* if there is almost nothing left,
2860 * we may never advance enough to end up in the above case */
2861 if (total < GST_SECOND)
2862 estimated = GST_SECOND;
2866 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2867 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2869 if (estimated != -1 && priv->estimated_eos != estimated) {
2870 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2871 priv->estimated_eos = estimated;
2875 /* take a buffer from the queue and push it */
2876 static GstFlowReturn
2877 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2879 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2880 GstFlowReturn result = GST_FLOW_OK;
2881 RTPJitterBufferItem *item;
2882 GstBuffer *outbuf = NULL;
2883 GstEvent *outevent = NULL;
2884 GstQuery *outquery = NULL;
2885 GstClockTime dts, pts;
2887 gboolean do_push = TRUE;
2891 /* when we get here we are ready to pop and push the buffer */
2892 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2896 case ITEM_TYPE_BUFFER:
2898 /* we need to make writable to change the flags and timestamps */
2899 outbuf = gst_buffer_make_writable (item->data);
2901 if (G_UNLIKELY (priv->discont)) {
2902 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2903 * into the jitterbuffer so we can modify now. */
2904 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2905 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2906 priv->discont = FALSE;
2908 if (G_UNLIKELY (priv->ts_discont)) {
2909 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2910 priv->ts_discont = FALSE;
2914 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2916 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2918 /* apply timestamp with offset to buffer now */
2919 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2920 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2922 /* update the elapsed time when we need to check against the npt stop time. */
2923 update_estimated_eos (jitterbuffer, item);
2925 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2927 case ITEM_TYPE_LOST:
2928 priv->discont = TRUE;
2932 case ITEM_TYPE_EVENT:
2933 outevent = item->data;
2935 case ITEM_TYPE_QUERY:
2936 outquery = item->data;
2940 /* now we are ready to push the buffer. Save the seqnum and release the lock
2941 * so the other end can push stuff in the queue again. */
2943 priv->last_popped_seqnum = seqnum;
2944 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2946 msg = check_buffering_percent (jitterbuffer, percent);
2953 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2956 case ITEM_TYPE_BUFFER:
2958 GST_DEBUG_OBJECT (jitterbuffer,
2959 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2960 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2961 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2962 result = gst_pad_push (priv->srcpad, outbuf);
2964 JBUF_LOCK_CHECK (priv, out_flushing);
2966 case ITEM_TYPE_LOST:
2967 case ITEM_TYPE_EVENT:
2968 /* We got not enough consecutive packets with a huge gap, we can
2969 * as well just drop them here now on EOS */
2970 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
2971 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
2972 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2973 g_queue_clear (&priv->gap_packets);
2976 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2977 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
2980 gst_pad_push_event (priv->srcpad, outevent);
2982 gst_event_unref (outevent);
2984 result = GST_FLOW_OK;
2986 JBUF_LOCK_CHECK (priv, out_flushing);
2988 case ITEM_TYPE_QUERY:
2992 res = gst_pad_peer_query (priv->srcpad, outquery);
2994 JBUF_LOCK_CHECK (priv, out_flushing);
2995 result = GST_FLOW_OK;
2996 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
2997 JBUF_SIGNAL_QUERY (priv, res);
3006 return priv->srcresult;
3010 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3012 /* Peek a buffer and compare the seqnum to the expected seqnum.
3013 * If all is fine, the buffer is pushed.
3014 * If something is wrong, we wait for some event
3016 static GstFlowReturn
3017 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3019 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3020 GstFlowReturn result;
3021 RTPJitterBufferItem *item;
3023 guint32 next_seqnum;
3025 /* only push buffers when PLAYING and active and not buffering */
3026 if (priv->blocked || !priv->active ||
3027 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3028 return GST_FLOW_WAIT;
3031 /* peek a buffer, we're just looking at the sequence number.
3032 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3033 * wait for a timeout or something to change.
3034 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3035 item = rtp_jitter_buffer_peek (priv->jbuf);
3040 /* get the seqnum and the next expected seqnum */
3041 seqnum = item->seqnum;
3043 return pop_and_push_next (jitterbuffer, seqnum);
3046 next_seqnum = priv->next_seqnum;
3048 /* get the gap between this and the previous packet. If we don't know the
3049 * previous packet seqnum assume no gap. */
3050 if (G_UNLIKELY (next_seqnum == -1)) {
3051 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3052 /* we don't know what the next_seqnum should be, the chain function should
3053 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3054 * fires, so wait for that */
3055 result = GST_FLOW_WAIT;
3057 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3059 if (G_LIKELY (gap == 0)) {
3060 /* no missing packet, pop and push */
3061 result = pop_and_push_next (jitterbuffer, seqnum);
3062 } else if (G_UNLIKELY (gap < 0)) {
3063 /* if we have a packet that we already pushed or considered dropped, pop it
3064 * off and get the next packet */
3065 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3066 seqnum, next_seqnum);
3067 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3069 result = GST_FLOW_OK;
3071 /* the chain function has scheduled timers to request retransmission or
3072 * when to consider the packet lost, wait for that */
3073 GST_DEBUG_OBJECT (jitterbuffer,
3074 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3075 next_seqnum, seqnum, gap);
3076 result = GST_FLOW_WAIT;
3084 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3086 return GST_FLOW_EOS;
3088 return GST_FLOW_WAIT;
3094 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3096 GstClockTime rtx_retry_timeout;
3097 GstClockTime rtx_min_retry_timeout;
3099 if (priv->rtx_retry_timeout == -1) {
3100 if (priv->avg_rtx_rtt == 0)
3101 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3103 /* we want to ask for a retransmission after we waited for a
3104 * complete RTT and the additional jitter */
3105 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3107 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3109 /* make sure we don't retry too often. On very low latency networks,
3110 * the RTT and jitter can be very low. */
3111 if (priv->rtx_min_retry_timeout == -1) {
3112 rtx_min_retry_timeout = priv->packet_spacing;
3114 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3116 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3118 return rtx_retry_timeout;
3122 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3123 GstClockTime rtx_retry_timeout)
3125 GstClockTime rtx_retry_period;
3127 if (priv->rtx_retry_period == -1) {
3128 /* we retry up to the configured jitterbuffer size but leaving some
3129 * room for the retransmission to arrive in time */
3130 if (rtx_retry_timeout > priv->latency_ns) {
3131 rtx_retry_period = 0;
3133 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3136 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3138 return rtx_retry_period;
3141 /* the timeout for when we expected a packet expired */
3143 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3146 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3148 guint delay, delay_ms, avg_rtx_rtt_ms;
3149 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3150 GstClockTime rtx_retry_period;
3151 GstClockTime rtx_retry_timeout;
3154 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3155 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3157 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3158 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3160 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3161 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3162 GST_TIME_ARGS (rtx_retry_period));
3164 delay = timer->rtx_delay + timer->rtx_retry;
3166 delay_ms = GST_TIME_AS_MSECONDS (delay);
3167 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3168 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3169 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3171 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3172 gst_structure_new ("GstRTPRetransmissionRequest",
3173 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3174 "running-time", G_TYPE_UINT64, timer->rtx_base,
3175 "delay", G_TYPE_UINT, delay_ms,
3176 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3177 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3178 "period", G_TYPE_UINT, rtx_retry_period_ms,
3179 "deadline", G_TYPE_UINT, priv->latency_ms,
3180 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3181 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3183 priv->num_rtx_requests++;
3184 timer->num_rtx_retry++;
3186 GST_OBJECT_LOCK (jitterbuffer);
3187 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3188 timer->rtx_last = gst_clock_get_time (clock);
3189 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3191 timer->rtx_last = now;
3193 GST_OBJECT_UNLOCK (jitterbuffer);
3195 /* calculate the timeout for the next retransmission attempt */
3196 timer->rtx_retry += rtx_retry_timeout;
3197 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3198 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3199 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3200 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3201 if ((priv->rtx_max_retries != -1
3202 && timer->num_rtx_retry >= priv->rtx_max_retries)
3203 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3204 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3205 /* too many retransmission request, we now convert the timer
3206 * to a lost timer, leave the num_rtx_retry as it is for stats */
3207 timer->type = TIMER_TYPE_LOST;
3208 timer->rtx_delay = 0;
3209 timer->rtx_retry = 0;
3211 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3212 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3215 gst_pad_push_event (priv->sinkpad, event);
3221 /* a packet is lost */
3223 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3226 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3227 GstClockTime duration, timestamp;
3228 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3231 RTPJitterBufferItem *item;
3233 seqnum = timer->seqnum;
3234 timestamp = apply_offset (jitterbuffer, timer->timeout);
3235 duration = timer->duration;
3236 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3237 duration = priv->packet_spacing;
3238 lost_packets = MAX (timer->num, 1);
3239 num_rtx_retry = timer->num_rtx_retry;
3241 /* we had a gap and thus we lost some packets. Create an event for this. */
3242 if (lost_packets > 1)
3243 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3244 seqnum + lost_packets - 1);
3246 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3248 priv->num_late += lost_packets;
3249 priv->num_rtx_failed += num_rtx_retry;
3251 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3253 /* we now only accept seqnum bigger than this */
3254 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3255 priv->next_in_seqnum = next_in_seqnum;
3257 /* create paket lost event */
3258 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3259 gst_structure_new ("GstRTPPacketLost",
3260 "seqnum", G_TYPE_UINT, (guint) seqnum,
3261 "timestamp", G_TYPE_UINT64, timestamp,
3262 "duration", G_TYPE_UINT64, duration,
3263 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3265 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3266 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3268 /* remove timer now */
3269 remove_timer (jitterbuffer, timer);
3271 JBUF_SIGNAL_EVENT (priv);
3277 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3280 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3282 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3283 remove_timer (jitterbuffer, timer);
3285 /* there was no EOS in the buffer, put one in there now */
3286 queue_event (jitterbuffer, gst_event_new_eos ());
3288 JBUF_SIGNAL_EVENT (priv);
3294 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3297 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3299 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3301 /* timer seqnum might have been obsoleted by caps seqnum-base,
3302 * only mess with current ongoing seqnum if still unknown */
3303 if (priv->next_seqnum == -1)
3304 priv->next_seqnum = timer->seqnum;
3305 remove_timer (jitterbuffer, timer);
3306 JBUF_SIGNAL_EVENT (priv);
3312 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3315 gboolean removed = FALSE;
3317 switch (timer->type) {
3318 case TIMER_TYPE_EXPECTED:
3319 removed = do_expected_timeout (jitterbuffer, timer, now);
3321 case TIMER_TYPE_LOST:
3322 removed = do_lost_timeout (jitterbuffer, timer, now);
3324 case TIMER_TYPE_DEADLINE:
3325 removed = do_deadline_timeout (jitterbuffer, timer, now);
3327 case TIMER_TYPE_EOS:
3328 removed = do_eos_timeout (jitterbuffer, timer, now);
3334 /* called when we need to wait for the next timeout.
3336 * We loop over the array of recorded timeouts and wait for the earliest one.
3337 * When it timed out, do the logic associated with the timer.
3339 * If there are no timers, we wait on a gcond until something new happens.
3342 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3344 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3345 GstClockTime now = 0;
3348 while (priv->timer_running) {
3349 TimerData *timer = NULL;
3350 GstClockTime timer_timeout = -1;
3353 /* If we have a clock, update "now" now with the very latest running time
3354 * we have. It is used below when timeouts are triggered to calculate
3355 * any next possible timeout. If we only update it after waiting for the
3356 * clock, we would give a too old time to the timeout functions.
3358 GST_OBJECT_LOCK (jitterbuffer);
3359 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3361 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3362 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3364 GST_OBJECT_UNLOCK (jitterbuffer);
3366 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3367 GST_TIME_ARGS (now));
3369 len = priv->timers->len;
3370 for (i = 0; i < len; i++) {
3371 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3372 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3373 gboolean save_best = FALSE;
3375 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3376 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3378 /* find the smallest timeout */
3379 if (timer == NULL) {
3381 } else if (timer_timeout == -1) {
3382 /* we already have an immediate timeout, the new timer must be an
3383 * immediate timer with smaller seqnum to become the best */
3384 if (test_timeout == -1
3385 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3386 timer->seqnum) > 0))
3388 } else if (test_timeout == -1) {
3389 /* first immediate timer */
3391 } else if (test_timeout < timer_timeout) {
3394 } else if (test_timeout == timer_timeout
3395 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3396 timer->seqnum) > 0)) {
3397 /* same timer, smaller seqnum */
3401 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3403 timer_timeout = test_timeout;
3406 if (timer && !priv->blocked) {
3408 GstClockTime sync_time;
3411 GstClockTimeDiff clock_jitter;
3413 if (timer_timeout == -1 || timer_timeout <= now) {
3414 do_timeout (jitterbuffer, timer, now);
3415 /* check here, do_timeout could have released the lock */
3416 if (!priv->timer_running)
3421 GST_OBJECT_LOCK (jitterbuffer);
3422 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3424 GST_OBJECT_UNLOCK (jitterbuffer);
3425 /* let's just push if there is no clock */
3426 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3427 now = timer_timeout;
3431 /* prepare for sync against clock */
3432 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3433 /* add latency of peer to get input time */
3434 sync_time += priv->peer_latency;
3436 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3437 " with sync time %" GST_TIME_FORMAT,
3438 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3440 /* create an entry for the clock */
3441 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3442 priv->timer_timeout = timer_timeout;
3443 priv->timer_seqnum = timer->seqnum;
3444 GST_OBJECT_UNLOCK (jitterbuffer);
3446 /* release the lock so that the other end can push stuff or unlock */
3449 ret = gst_clock_id_wait (id, &clock_jitter);
3452 if (!priv->timer_running) {
3453 gst_clock_id_unref (id);
3454 priv->clock_id = NULL;
3458 if (ret != GST_CLOCK_UNSCHEDULED) {
3459 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3460 ret, priv->timer_seqnum, clock_jitter);
3462 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3464 /* and free the entry */
3465 gst_clock_id_unref (id);
3466 priv->clock_id = NULL;
3468 /* no timers, wait for activity */
3469 JBUF_WAIT_TIMER (priv);
3474 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3479 * This funcion implements the main pushing loop on the source pad.
3481 * It first tries to push as many buffers as possible. If there is a seqnum
3482 * mismatch, we wait for the next timeouts.
3485 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3487 GstRtpJitterBufferPrivate *priv;
3488 GstFlowReturn result = GST_FLOW_OK;
3490 priv = jitterbuffer->priv;
3492 JBUF_LOCK_CHECK (priv, flushing);
3494 result = handle_next_buffer (jitterbuffer);
3495 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3496 /* now wait for the next event */
3497 JBUF_WAIT_EVENT (priv, flushing);
3498 result = GST_FLOW_OK;
3500 } while (result == GST_FLOW_OK);
3501 /* store result for upstream */
3502 priv->srcresult = result;
3503 /* if we get here we need to pause */
3509 result = priv->srcresult;
3516 JBUF_SIGNAL_QUERY (priv, FALSE);
3519 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3520 gst_flow_get_name (result));
3521 gst_pad_pause_task (priv->srcpad);
3522 if (result == GST_FLOW_EOS) {
3523 event = gst_event_new_eos ();
3524 gst_pad_push_event (priv->srcpad, event);
3530 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3531 * some sanity checks and then emit the handle-sync signal with the parameters.
3532 * This function must be called with the LOCK */
3534 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3536 GstRtpJitterBufferPrivate *priv;
3537 guint64 base_rtptime, base_time;
3539 guint64 last_rtptime;
3541 guint64 ext_rtptime, diff;
3542 gboolean valid = TRUE, keep = FALSE;
3544 priv = jitterbuffer->priv;
3546 /* get the last values from the jitterbuffer */
3547 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3548 &clock_rate, &last_rtptime);
3550 clock_base = priv->clock_base;
3551 ext_rtptime = priv->ext_rtptime;
3553 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3554 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3555 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3556 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3558 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3559 /* we keep this SR packet for later. When we get a valid RTP packet the
3560 * above values will be set and we can try to use the SR packet */
3561 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3564 /* we can't accept anything that happened before we did the last resync */
3565 if (base_rtptime > ext_rtptime) {
3566 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3569 /* the SR RTP timestamp must be something close to what we last observed
3570 * in the jitterbuffer */
3571 if (ext_rtptime > last_rtptime) {
3572 /* check how far ahead it is to our RTP timestamps */
3573 diff = ext_rtptime - last_rtptime;
3574 /* if bigger than 1 second, we drop it */
3575 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3577 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3578 clock_rate, 1000)) {
3579 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3580 /* should drop this, but some RTSP servers end up with bogus
3581 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3582 * so still trigger rptbin sync but invalidate RTCP data
3583 * (sync might use other methods) */
3586 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3587 G_GUINT64_FORMAT, last_rtptime, diff);
3593 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3597 s = gst_structure_new ("application/x-rtp-sync",
3598 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3599 "base-time", G_TYPE_UINT64, base_time,
3600 "clock-rate", G_TYPE_UINT, clock_rate,
3601 "clock-base", G_TYPE_UINT64, clock_base,
3602 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3603 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3605 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3606 gst_buffer_replace (&priv->last_sr, NULL);
3608 g_signal_emit (jitterbuffer,
3609 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3611 gst_structure_free (s);
3613 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3614 gst_buffer_replace (&priv->last_sr, NULL);
3618 static GstFlowReturn
3619 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3622 GstRtpJitterBuffer *jitterbuffer;
3623 GstRtpJitterBufferPrivate *priv;
3624 GstFlowReturn ret = GST_FLOW_OK;
3626 GstRTCPPacket packet;
3627 guint64 ext_rtptime;
3629 GstRTCPBuffer rtcp = { NULL, };
3631 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3633 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3634 goto invalid_buffer;
3636 priv = jitterbuffer->priv;
3638 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3640 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3643 /* first packet must be SR or RR or else the validate would have failed */
3644 switch (gst_rtcp_packet_get_type (&packet)) {
3645 case GST_RTCP_TYPE_SR:
3646 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3652 gst_rtcp_buffer_unmap (&rtcp);
3654 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3657 /* convert the RTP timestamp to our extended timestamp, using the same offset
3658 * we used in the jitterbuffer */
3659 ext_rtptime = priv->jbuf->ext_rtptime;
3660 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3662 priv->ext_rtptime = ext_rtptime;
3663 gst_buffer_replace (&priv->last_sr, buffer);
3665 do_handle_sync (jitterbuffer);
3669 gst_buffer_unref (buffer);
3675 /* this is not fatal but should be filtered earlier */
3676 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3677 ("Received invalid RTCP payload, dropping"));
3683 /* this is not fatal but should be filtered earlier */
3684 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3685 ("Received empty RTCP payload, dropping"));
3686 gst_rtcp_buffer_unmap (&rtcp);
3692 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3693 gst_rtcp_buffer_unmap (&rtcp);
3700 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3703 gboolean res = FALSE;
3704 GstRtpJitterBuffer *jitterbuffer;
3705 GstRtpJitterBufferPrivate *priv;
3707 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3708 priv = jitterbuffer->priv;
3710 switch (GST_QUERY_TYPE (query)) {
3711 case GST_QUERY_CAPS:
3713 GstCaps *filter, *caps;
3715 gst_query_parse_caps (query, &filter);
3716 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3717 gst_query_set_caps_result (query, caps);
3718 gst_caps_unref (caps);
3723 if (GST_QUERY_IS_SERIALIZED (query)) {
3724 RTPJitterBufferItem *item;
3727 JBUF_LOCK_CHECK (priv, out_flushing);
3728 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3729 RTP_JITTER_BUFFER_MODE_BUFFER) {
3730 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3731 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3732 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3734 JBUF_SIGNAL_EVENT (priv);
3735 JBUF_WAIT_QUERY (priv, out_flushing);
3736 res = priv->last_query;
3738 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3743 res = gst_pad_query_default (pad, parent, query);
3751 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3759 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3762 GstRtpJitterBuffer *jitterbuffer;
3763 GstRtpJitterBufferPrivate *priv;
3764 gboolean res = FALSE;
3766 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3767 priv = jitterbuffer->priv;
3769 switch (GST_QUERY_TYPE (query)) {
3770 case GST_QUERY_LATENCY:
3772 /* We need to send the query upstream and add the returned latency to our
3774 GstClockTime min_latency, max_latency;
3776 GstClockTime our_latency;
3778 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3779 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3781 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3782 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3783 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3785 /* store this so that we can safely sync on the peer buffers. */
3787 priv->peer_latency = min_latency;
3788 our_latency = priv->latency_ns;
3791 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3792 GST_TIME_ARGS (our_latency));
3794 /* we add some latency but can buffer an infinite amount of time */
3795 min_latency += our_latency;
3798 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3799 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3800 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3802 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3806 case GST_QUERY_POSITION:
3808 GstClockTime start, last_out;
3811 gst_query_parse_position (query, &fmt, NULL);
3812 if (fmt != GST_FORMAT_TIME) {
3813 res = gst_pad_query_default (pad, parent, query);
3818 start = priv->npt_start;
3819 last_out = priv->last_out_time;
3822 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3823 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3824 GST_TIME_ARGS (last_out));
3826 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3827 /* bring 0-based outgoing time to stream time */
3828 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3831 res = gst_pad_query_default (pad, parent, query);
3835 case GST_QUERY_CAPS:
3837 GstCaps *filter, *caps;
3839 gst_query_parse_caps (query, &filter);
3840 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3841 gst_query_set_caps_result (query, caps);
3842 gst_caps_unref (caps);
3847 res = gst_pad_query_default (pad, parent, query);
3855 gst_rtp_jitter_buffer_set_property (GObject * object,
3856 guint prop_id, const GValue * value, GParamSpec * pspec)
3858 GstRtpJitterBuffer *jitterbuffer;
3859 GstRtpJitterBufferPrivate *priv;
3861 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3862 priv = jitterbuffer->priv;
3867 guint new_latency, old_latency;
3869 new_latency = g_value_get_uint (value);
3872 old_latency = priv->latency_ms;
3873 priv->latency_ms = new_latency;
3874 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3875 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3878 /* post message if latency changed, this will inform the parent pipeline
3879 * that a latency reconfiguration is possible/needed. */
3880 if (new_latency != old_latency) {
3881 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3882 GST_TIME_ARGS (new_latency * GST_MSECOND));
3884 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3885 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3889 case PROP_DROP_ON_LATENCY:
3891 priv->drop_on_latency = g_value_get_boolean (value);
3894 case PROP_TS_OFFSET:
3896 priv->ts_offset = g_value_get_int64 (value);
3897 priv->ts_discont = TRUE;
3902 priv->do_lost = g_value_get_boolean (value);
3907 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3910 case PROP_DO_RETRANSMISSION:
3912 priv->do_retransmission = g_value_get_boolean (value);
3915 case PROP_RTX_NEXT_SEQNUM:
3917 priv->rtx_next_seqnum = g_value_get_boolean (value);
3920 case PROP_RTX_DELAY:
3922 priv->rtx_delay = g_value_get_int (value);
3925 case PROP_RTX_MIN_DELAY:
3927 priv->rtx_min_delay = g_value_get_uint (value);
3930 case PROP_RTX_DELAY_REORDER:
3932 priv->rtx_delay_reorder = g_value_get_int (value);
3935 case PROP_RTX_RETRY_TIMEOUT:
3937 priv->rtx_retry_timeout = g_value_get_int (value);
3940 case PROP_RTX_MIN_RETRY_TIMEOUT:
3942 priv->rtx_min_retry_timeout = g_value_get_int (value);
3945 case PROP_RTX_RETRY_PERIOD:
3947 priv->rtx_retry_period = g_value_get_int (value);
3950 case PROP_RTX_MAX_RETRIES:
3952 priv->rtx_max_retries = g_value_get_int (value);
3955 case PROP_MAX_RTCP_RTP_TIME_DIFF:
3957 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
3961 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3967 gst_rtp_jitter_buffer_get_property (GObject * object,
3968 guint prop_id, GValue * value, GParamSpec * pspec)
3970 GstRtpJitterBuffer *jitterbuffer;
3971 GstRtpJitterBufferPrivate *priv;
3973 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3974 priv = jitterbuffer->priv;
3979 g_value_set_uint (value, priv->latency_ms);
3982 case PROP_DROP_ON_LATENCY:
3984 g_value_set_boolean (value, priv->drop_on_latency);
3987 case PROP_TS_OFFSET:
3989 g_value_set_int64 (value, priv->ts_offset);
3994 g_value_set_boolean (value, priv->do_lost);
3999 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4007 if (priv->srcresult != GST_FLOW_OK)
4010 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4012 g_value_set_int (value, percent);
4016 case PROP_DO_RETRANSMISSION:
4018 g_value_set_boolean (value, priv->do_retransmission);
4021 case PROP_RTX_NEXT_SEQNUM:
4023 g_value_set_boolean (value, priv->rtx_next_seqnum);
4026 case PROP_RTX_DELAY:
4028 g_value_set_int (value, priv->rtx_delay);
4031 case PROP_RTX_MIN_DELAY:
4033 g_value_set_uint (value, priv->rtx_min_delay);
4036 case PROP_RTX_DELAY_REORDER:
4038 g_value_set_int (value, priv->rtx_delay_reorder);
4041 case PROP_RTX_RETRY_TIMEOUT:
4043 g_value_set_int (value, priv->rtx_retry_timeout);
4046 case PROP_RTX_MIN_RETRY_TIMEOUT:
4048 g_value_set_int (value, priv->rtx_min_retry_timeout);
4051 case PROP_RTX_RETRY_PERIOD:
4053 g_value_set_int (value, priv->rtx_retry_period);
4056 case PROP_RTX_MAX_RETRIES:
4058 g_value_set_int (value, priv->rtx_max_retries);
4062 g_value_take_boxed (value,
4063 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4065 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4067 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4071 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4076 static GstStructure *
4077 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4081 JBUF_LOCK (jbuf->priv);
4082 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4083 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4084 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4085 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4086 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4087 JBUF_UNLOCK (jbuf->priv);