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
140 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
141 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
147 PROP_DROP_ON_LATENCY,
152 PROP_DO_RETRANSMISSION,
153 PROP_RTX_NEXT_SEQNUM,
156 PROP_RTX_DELAY_REORDER,
157 PROP_RTX_RETRY_TIMEOUT,
158 PROP_RTX_MIN_RETRY_TIMEOUT,
159 PROP_RTX_RETRY_PERIOD,
160 PROP_RTX_MAX_RETRIES,
164 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
166 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
168 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
171 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
173 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
174 GST_DEBUG ("waiting timer"); \
175 (priv)->waiting_timer = TRUE; \
176 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
177 (priv)->waiting_timer = FALSE; \
178 GST_DEBUG ("waiting timer done"); \
180 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
181 if (G_UNLIKELY ((priv)->waiting_timer)) { \
182 GST_DEBUG ("signal timer"); \
183 g_cond_signal (&(priv)->jbuf_timer); \
187 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
188 GST_DEBUG ("waiting event"); \
189 (priv)->waiting_event = TRUE; \
190 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
191 (priv)->waiting_event = FALSE; \
192 GST_DEBUG ("waiting event done"); \
193 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
196 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
197 if (G_UNLIKELY ((priv)->waiting_event)) { \
198 GST_DEBUG ("signal event"); \
199 g_cond_signal (&(priv)->jbuf_event); \
203 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
204 GST_DEBUG ("waiting query"); \
205 (priv)->waiting_query = TRUE; \
206 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
207 (priv)->waiting_query = FALSE; \
208 GST_DEBUG ("waiting query done"); \
209 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
212 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
213 (priv)->last_query = res; \
214 if (G_UNLIKELY ((priv)->waiting_query)) { \
215 GST_DEBUG ("signal query"); \
216 g_cond_signal (&(priv)->jbuf_query); \
221 struct _GstRtpJitterBufferPrivate
223 GstPad *sinkpad, *srcpad;
226 RTPJitterBuffer *jbuf;
228 gboolean waiting_timer;
230 gboolean waiting_event;
232 gboolean waiting_query;
240 gboolean timer_running;
241 GThread *timer_thread;
246 gboolean drop_on_latency;
249 gboolean do_retransmission;
250 gboolean rtx_next_seqnum;
253 gint rtx_delay_reorder;
254 gint rtx_retry_timeout;
255 gint rtx_min_retry_timeout;
256 gint rtx_retry_period;
257 gint rtx_max_retries;
259 /* the last seqnum we pushed out */
260 guint32 last_popped_seqnum;
261 /* the next expected seqnum we push */
263 /* seqnum-base, if known */
265 /* last output time */
266 GstClockTime last_out_time;
267 /* last valid input timestamp and rtptime pair */
268 GstClockTime ips_dts;
270 GstClockTime packet_spacing;
274 /* the next expected seqnum we receive */
275 GstClockTime last_in_dts;
276 guint32 next_in_seqnum;
280 /* start and stop ranges */
281 GstClockTime npt_start;
282 GstClockTime npt_stop;
283 guint64 ext_timestamp;
284 guint64 last_elapsed;
285 guint64 estimated_eos;
292 /* clock rate and rtp timestamp offset */
296 gint64 prev_ts_offset;
298 /* when we are shutting down */
299 GstFlowReturn srcresult;
305 GstClockTime timer_timeout;
306 guint16 timer_seqnum;
307 /* the latency of the upstream peer, we have to take this into account when
308 * synchronizing the buffers. */
309 GstClockTime peer_latency;
313 /* some accounting */
315 guint64 num_duplicates;
316 guint64 num_rtx_requests;
317 guint64 num_rtx_success;
318 guint64 num_rtx_failed;
323 GstClockTime last_dts;
324 guint64 last_rtptime;
325 GstClockTime avg_jitter;
342 GstClockTime timeout;
343 GstClockTime duration;
344 GstClockTime rtx_base;
345 GstClockTime rtx_delay;
346 GstClockTime rtx_retry;
347 GstClockTime rtx_last;
351 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
352 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
353 GstRtpJitterBufferPrivate))
355 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
356 GST_STATIC_PAD_TEMPLATE ("sink",
359 GST_STATIC_CAPS ("application/x-rtp"
360 /* "clock-rate = (int) [ 1, 2147483647 ], "
361 * "payload = (int) , "
362 * "encoding-name = (string) "
366 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
367 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
370 GST_STATIC_CAPS ("application/x-rtcp")
373 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
374 GST_STATIC_PAD_TEMPLATE ("src",
377 GST_STATIC_CAPS ("application/x-rtp"
378 /* "payload = (int) , "
379 * "clock-rate = (int) , "
380 * "encoding-name = (string) "
384 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
386 #define gst_rtp_jitter_buffer_parent_class parent_class
387 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
389 /* object overrides */
390 static void gst_rtp_jitter_buffer_set_property (GObject * object,
391 guint prop_id, const GValue * value, GParamSpec * pspec);
392 static void gst_rtp_jitter_buffer_get_property (GObject * object,
393 guint prop_id, GValue * value, GParamSpec * pspec);
394 static void gst_rtp_jitter_buffer_finalize (GObject * object);
396 /* element overrides */
397 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
398 * element, GstStateChange transition);
399 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
400 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
401 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
403 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
406 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
407 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
410 /* sinkpad overrides */
411 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
412 GstObject * parent, GstEvent * event);
413 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
414 GstObject * parent, GstBuffer * buffer);
416 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
417 GstObject * parent, GstEvent * event);
418 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
419 GstObject * parent, GstBuffer * buffer);
421 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
422 GstObject * parent, GstQuery * query);
424 /* srcpad overrides */
425 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
426 GstObject * parent, GstEvent * event);
427 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
428 GstObject * parent, GstPadMode mode, gboolean active);
429 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
430 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
431 GstObject * parent, GstQuery * query);
434 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
436 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
437 gboolean active, guint64 base_time);
438 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
440 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
441 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
443 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
445 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
449 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
451 GObjectClass *gobject_class;
452 GstElementClass *gstelement_class;
454 gobject_class = (GObjectClass *) klass;
455 gstelement_class = (GstElementClass *) klass;
457 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
459 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
461 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
462 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
465 * GstRtpJitterBuffer:latency:
467 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
468 * for at most this time.
470 g_object_class_install_property (gobject_class, PROP_LATENCY,
471 g_param_spec_uint ("latency", "Buffer latency in ms",
472 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
473 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
475 * GstRtpJitterBuffer:drop-on-latency:
477 * Drop oldest buffers when the queue is completely filled.
479 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
480 g_param_spec_boolean ("drop-on-latency",
481 "Drop buffers when maximum latency is reached",
482 "Tells the jitterbuffer to never exceed the given latency in size",
483 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
485 * GstRtpJitterBuffer:ts-offset:
487 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
488 * This is mainly used to ensure interstream synchronisation.
490 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
491 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
492 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
493 G_MAXINT64, DEFAULT_TS_OFFSET,
494 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
497 * GstRtpJitterBuffer:do-lost:
499 * Send out a GstRTPPacketLost event downstream when a packet is considered
502 g_object_class_install_property (gobject_class, PROP_DO_LOST,
503 g_param_spec_boolean ("do-lost", "Do Lost",
504 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
505 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
508 * GstRtpJitterBuffer:mode:
510 * Control the buffering and timestamping mode used by the jitterbuffer.
512 g_object_class_install_property (gobject_class, PROP_MODE,
513 g_param_spec_enum ("mode", "Mode",
514 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
515 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
517 * GstRtpJitterBuffer:percent:
519 * The percent of the jitterbuffer that is filled.
521 g_object_class_install_property (gobject_class, PROP_PERCENT,
522 g_param_spec_int ("percent", "percent",
523 "The buffer filled percent", 0, 100,
524 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
526 * GstRtpJitterBuffer:do-retransmission:
528 * Send out a GstRTPRetransmission event upstream when a packet is considered
529 * late and should be retransmitted.
533 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
534 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
535 "Send retransmission events upstream when a packet is late",
536 DEFAULT_DO_RETRANSMISSION,
537 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
540 * GstRtpJitterBuffer:rtx-next-seqnum
542 * Estimate when the next packet should arrive and schedule a retransmission
544 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
545 * for packet N+1. So it will be requested if it does not arrive at the expected time.
546 * The expected time is calculated using the dts of N and the packet spacing.
550 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
551 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
552 "Estimate when the next packet should arrive and schedule a "
553 "retransmission request for it.",
554 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
557 * GstRtpJitterBuffer:rtx-delay:
559 * When a packet did not arrive at the expected time, wait this extra amount
560 * of time before sending a retransmission event.
562 * When -1 is used, the max jitter will be used as extra delay.
566 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
567 g_param_spec_int ("rtx-delay", "RTX Delay",
568 "Extra time in ms to wait before sending retransmission "
569 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
570 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 * GstRtpJitterBuffer:rtx-min-delay:
575 * When a packet did not arrive at the expected time, wait at least this extra amount
576 * of time before sending a retransmission event.
580 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
581 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
582 "Minimum time in ms to wait before sending retransmission "
583 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
584 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
586 * GstRtpJitterBuffer:rtx-delay-reorder:
588 * Assume that a retransmission event should be sent when we see
589 * this much packet reordering.
591 * When -1 is used, the value will be estimated based on observed packet
596 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
597 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
598 "Sending retransmission event when this much reordering (-1 automatic)",
599 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
600 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
602 * GstRtpJitterBuffer::rtx-retry-timeout:
604 * When no packet has been received after sending a retransmission event
605 * for this time, retry sending a retransmission event.
607 * When -1 is used, the value will be estimated based on observed round
612 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
613 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
614 "Retry sending a transmission event after this timeout in "
615 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
616 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
618 * GstRtpJitterBuffer::rtx-min-retry-timeout:
620 * The minimum amount of time between retry timeouts. When
621 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
622 * minimum interval between retry timeouts.
624 * When -1 is used, the value will be estimated based on the
629 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
630 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
631 "Minimum timeout between sending a transmission event in "
632 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
633 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
635 * GstRtpJitterBuffer:rtx-retry-period:
637 * The amount of time to try to get a retransmission.
639 * When -1 is used, the value will be estimated based on the jitterbuffer
640 * latency and the observed round trip time.
644 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
645 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
646 "Try to get a retransmission for this many ms "
647 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
648 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
650 * GstRtpJitterBuffer:rtx-max-retries:
652 * The maximum number of retries to request a retransmission.
654 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
655 * When -1 is used, the number of retransmission request will not be limited.
659 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
660 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
661 "The maximum number of retries to request a retransmission. "
662 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
663 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
665 * GstRtpJitterBuffer:stats:
667 * Various jitterbuffer statistics. This property returns a GstStructure
668 * with name application/x-rtp-jitterbuffer-stats with the following fields:
674 * <classname>"rtx-count"</classname>:
675 * the number of retransmissions requested.
681 * <classname>"rtx-success-count"</classname>:
682 * the number of successful retransmissions.
688 * <classname>"rtx-per-packet"</classname>:
689 * average number of RTX per packet.
695 * <classname>"rtx-rtt"</classname>:
696 * average round trip time per RTX.
703 g_object_class_install_property (gobject_class, PROP_STATS,
704 g_param_spec_boxed ("stats", "Statistics",
705 "Various statistics", GST_TYPE_STRUCTURE,
706 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
709 * GstRtpJitterBuffer::request-pt-map:
710 * @buffer: the object which received the signal
713 * Request the payload type as #GstCaps for @pt.
715 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
716 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
717 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
718 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
719 GST_TYPE_CAPS, 1, G_TYPE_UINT);
721 * GstRtpJitterBuffer::handle-sync:
722 * @buffer: the object which received the signal
723 * @struct: a GstStructure containing sync values.
725 * Be notified of new sync values.
727 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
728 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
729 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
730 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
731 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
734 * GstRtpJitterBuffer::on-npt-stop:
735 * @buffer: the object which received the signal
737 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
738 * the npt-stop position.
740 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
741 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
742 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
743 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
744 G_TYPE_NONE, 0, G_TYPE_NONE);
747 * GstRtpJitterBuffer::clear-pt-map:
748 * @buffer: the object which received the signal
750 * Invalidate the clock-rate as obtained with the
751 * #GstRtpJitterBuffer::request-pt-map signal.
753 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
754 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
755 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
756 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
757 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
760 * GstRtpJitterBuffer::set-active:
761 * @buffer: the object which received the signal
763 * Start pushing out packets with the given base time. This signal is only
764 * useful in buffering mode.
766 * Returns: the time of the last pushed packet.
768 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
769 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
770 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
771 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
772 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
775 gstelement_class->change_state =
776 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
777 gstelement_class->request_new_pad =
778 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
779 gstelement_class->release_pad =
780 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
781 gstelement_class->provide_clock =
782 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
784 gst_element_class_add_pad_template (gstelement_class,
785 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
786 gst_element_class_add_pad_template (gstelement_class,
787 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
788 gst_element_class_add_pad_template (gstelement_class,
789 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
791 gst_element_class_set_static_metadata (gstelement_class,
792 "RTP packet jitter-buffer", "Filter/Network/RTP",
793 "A buffer that deals with network jitter and other transmission faults",
794 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
795 "Wim Taymans <wim.taymans@gmail.com>");
797 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
798 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
800 GST_DEBUG_CATEGORY_INIT
801 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
805 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
807 GstRtpJitterBufferPrivate *priv;
809 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
810 jitterbuffer->priv = priv;
812 priv->latency_ms = DEFAULT_LATENCY_MS;
813 priv->latency_ns = priv->latency_ms * GST_MSECOND;
814 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
815 priv->do_lost = DEFAULT_DO_LOST;
816 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
817 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
818 priv->rtx_delay = DEFAULT_RTX_DELAY;
819 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
820 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
821 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
822 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
823 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
824 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
827 priv->last_rtptime = -1;
828 priv->avg_jitter = 0;
829 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
830 priv->jbuf = rtp_jitter_buffer_new ();
831 g_mutex_init (&priv->jbuf_lock);
832 g_cond_init (&priv->jbuf_timer);
833 g_cond_init (&priv->jbuf_event);
834 g_cond_init (&priv->jbuf_query);
835 g_queue_init (&priv->gap_packets);
837 /* reset skew detection initialy */
838 rtp_jitter_buffer_reset_skew (priv->jbuf);
839 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
840 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
844 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
847 gst_pad_set_activatemode_function (priv->srcpad,
848 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
849 gst_pad_set_query_function (priv->srcpad,
850 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
851 gst_pad_set_event_function (priv->srcpad,
852 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
855 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
858 gst_pad_set_chain_function (priv->sinkpad,
859 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
860 gst_pad_set_event_function (priv->sinkpad,
861 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
862 gst_pad_set_query_function (priv->sinkpad,
863 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
865 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
866 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
868 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
871 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
873 #define ITEM_TYPE_BUFFER 0
874 #define ITEM_TYPE_LOST 1
875 #define ITEM_TYPE_EVENT 2
876 #define ITEM_TYPE_QUERY 3
878 static RTPJitterBufferItem *
879 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
880 guint seqnum, guint count, guint rtptime)
882 RTPJitterBufferItem *item;
884 item = g_slice_new (RTPJitterBufferItem);
891 item->seqnum = seqnum;
893 item->rtptime = rtptime;
899 free_item (RTPJitterBufferItem * item)
901 g_return_if_fail (item != NULL);
903 if (item->data && item->type != ITEM_TYPE_QUERY)
904 gst_mini_object_unref (item->data);
905 g_slice_free (RTPJitterBufferItem, item);
909 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
911 GList **l = user_data;
913 if (item->data && item->type == ITEM_TYPE_EVENT
914 && GST_EVENT_IS_STICKY (item->data)) {
915 *l = g_list_prepend (*l, item->data);
916 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
917 gst_mini_object_unref (item->data);
919 g_slice_free (RTPJitterBufferItem, item);
923 gst_rtp_jitter_buffer_finalize (GObject * object)
925 GstRtpJitterBuffer *jitterbuffer;
926 GstRtpJitterBufferPrivate *priv;
928 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
929 priv = jitterbuffer->priv;
931 g_array_free (priv->timers, TRUE);
932 g_mutex_clear (&priv->jbuf_lock);
933 g_cond_clear (&priv->jbuf_timer);
934 g_cond_clear (&priv->jbuf_event);
935 g_cond_clear (&priv->jbuf_query);
937 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
938 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
939 g_queue_clear (&priv->gap_packets);
940 g_object_unref (priv->jbuf);
942 G_OBJECT_CLASS (parent_class)->finalize (object);
946 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
948 GstRtpJitterBuffer *jitterbuffer;
949 GstPad *otherpad = NULL;
950 GstIterator *it = NULL;
953 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
955 if (pad == jitterbuffer->priv->sinkpad) {
956 otherpad = jitterbuffer->priv->srcpad;
957 } else if (pad == jitterbuffer->priv->srcpad) {
958 otherpad = jitterbuffer->priv->sinkpad;
959 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
960 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
964 g_value_init (&val, GST_TYPE_PAD);
965 g_value_set_object (&val, otherpad);
966 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
967 g_value_unset (&val);
974 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
976 GstRtpJitterBufferPrivate *priv;
978 priv = jitterbuffer->priv;
980 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
983 gst_pad_new_from_static_template
984 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
985 gst_pad_set_chain_function (priv->rtcpsinkpad,
986 gst_rtp_jitter_buffer_chain_rtcp);
987 gst_pad_set_event_function (priv->rtcpsinkpad,
988 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
989 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
990 gst_rtp_jitter_buffer_iterate_internal_links);
991 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
992 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
994 return priv->rtcpsinkpad;
998 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1000 GstRtpJitterBufferPrivate *priv;
1002 priv = jitterbuffer->priv;
1004 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1006 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1008 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1009 priv->rtcpsinkpad = NULL;
1013 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1014 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1016 GstRtpJitterBuffer *jitterbuffer;
1017 GstElementClass *klass;
1019 GstRtpJitterBufferPrivate *priv;
1021 g_return_val_if_fail (templ != NULL, NULL);
1022 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1024 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1025 priv = jitterbuffer->priv;
1026 klass = GST_ELEMENT_GET_CLASS (element);
1028 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1030 /* figure out the template */
1031 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1032 if (priv->rtcpsinkpad != NULL)
1035 result = create_rtcp_sink (jitterbuffer);
1037 goto wrong_template;
1044 g_warning ("rtpjitterbuffer: this is not our template");
1049 g_warning ("rtpjitterbuffer: pad already requested");
1055 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1057 GstRtpJitterBuffer *jitterbuffer;
1058 GstRtpJitterBufferPrivate *priv;
1060 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1061 g_return_if_fail (GST_IS_PAD (pad));
1063 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1064 priv = jitterbuffer->priv;
1066 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1068 if (priv->rtcpsinkpad == pad) {
1069 remove_rtcp_sink (jitterbuffer);
1078 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1084 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1086 return gst_system_clock_obtain ();
1090 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1092 GstRtpJitterBufferPrivate *priv;
1094 priv = jitterbuffer->priv;
1096 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1099 priv->clock_rate = -1;
1100 /* do not clear current content, but refresh state for new arrival */
1101 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1102 rtp_jitter_buffer_reset_skew (priv->jbuf);
1107 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1110 GstRtpJitterBufferPrivate *priv;
1111 GstClockTime last_out;
1112 RTPJitterBufferItem *item;
1117 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1118 active, GST_TIME_ARGS (offset));
1120 if (active != priv->active) {
1121 /* add the amount of time spent in paused to the output offset. All
1122 * outgoing buffers will have this offset applied to their timestamps in
1123 * order to make them arrive in time in the sink. */
1124 priv->out_offset = offset;
1125 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1126 GST_TIME_ARGS (priv->out_offset));
1127 priv->active = active;
1128 JBUF_SIGNAL_EVENT (priv);
1131 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1133 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1134 /* head buffer timestamp and offset gives our output time */
1135 last_out = item->dts + priv->ts_offset;
1137 /* use last known time when the buffer is empty */
1138 last_out = priv->last_out_time;
1146 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1148 GstRtpJitterBuffer *jitterbuffer;
1149 GstRtpJitterBufferPrivate *priv;
1154 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1155 priv = jitterbuffer->priv;
1157 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1159 caps = gst_pad_peer_query_caps (other, filter);
1161 templ = gst_pad_get_pad_template_caps (pad);
1163 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1168 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1170 intersect = gst_caps_intersect (caps, templ);
1171 gst_caps_unref (caps);
1172 gst_caps_unref (templ);
1176 gst_object_unref (jitterbuffer);
1182 * Must be called with JBUF_LOCK held
1186 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1189 GstRtpJitterBufferPrivate *priv;
1190 GstStructure *caps_struct;
1194 priv = jitterbuffer->priv;
1196 /* first parse the caps */
1197 caps_struct = gst_caps_get_structure (caps, 0);
1199 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1201 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1202 * measure the amount of data in the buffer */
1203 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1206 if (priv->clock_rate <= 0)
1209 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1211 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1213 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1214 * can use this to track the amount of time elapsed on the sender. */
1215 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1216 priv->clock_base = val;
1218 priv->clock_base = -1;
1220 priv->ext_timestamp = priv->clock_base;
1222 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1225 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1226 /* first expected seqnum, only update when we didn't have a previous base. */
1227 if (priv->next_in_seqnum == -1)
1228 priv->next_in_seqnum = val;
1229 if (priv->next_seqnum == -1) {
1230 priv->next_seqnum = val;
1231 JBUF_SIGNAL_EVENT (priv);
1233 priv->seqnum_base = val;
1235 priv->seqnum_base = -1;
1238 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1240 /* the start and stop times. The seqnum-base corresponds to the start time. We
1241 * will keep track of the seqnums on the output and when we reach the one
1242 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1243 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1244 priv->npt_start = tval;
1246 priv->npt_start = 0;
1248 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1249 priv->npt_stop = tval;
1251 priv->npt_stop = -1;
1253 GST_DEBUG_OBJECT (jitterbuffer,
1254 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1255 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1262 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1267 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1273 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1275 GstRtpJitterBufferPrivate *priv;
1277 priv = jitterbuffer->priv;
1280 /* mark ourselves as flushing */
1281 priv->srcresult = GST_FLOW_FLUSHING;
1282 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1283 /* this unblocks any waiting pops on the src pad task */
1284 JBUF_SIGNAL_EVENT (priv);
1285 JBUF_SIGNAL_QUERY (priv, FALSE);
1290 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1292 GstRtpJitterBufferPrivate *priv;
1294 priv = jitterbuffer->priv;
1297 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1298 /* Mark as non flushing */
1299 priv->srcresult = GST_FLOW_OK;
1300 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1301 priv->last_popped_seqnum = -1;
1302 priv->last_out_time = -1;
1303 priv->next_seqnum = -1;
1304 priv->seqnum_base = -1;
1305 priv->ips_rtptime = -1;
1306 priv->ips_dts = GST_CLOCK_TIME_NONE;
1307 priv->packet_spacing = 0;
1308 priv->next_in_seqnum = -1;
1309 priv->clock_rate = -1;
1312 priv->estimated_eos = -1;
1313 priv->last_elapsed = 0;
1314 priv->ext_timestamp = -1;
1315 priv->avg_jitter = 0;
1316 priv->last_dts = -1;
1317 priv->last_rtptime = -1;
1318 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1319 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1320 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1321 rtp_jitter_buffer_reset_skew (priv->jbuf);
1322 remove_all_timers (jitterbuffer);
1323 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1324 g_queue_clear (&priv->gap_packets);
1329 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1330 GstPadMode mode, gboolean active)
1333 GstRtpJitterBuffer *jitterbuffer = NULL;
1335 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1338 case GST_PAD_MODE_PUSH:
1340 /* allow data processing */
1341 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1343 /* start pushing out buffers */
1344 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1345 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1346 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1348 /* make sure all data processing stops ASAP */
1349 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1351 /* NOTE this will hardlock if the state change is called from the src pad
1352 * task thread because we will _join() the thread. */
1353 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1354 result = gst_pad_stop_task (pad);
1364 static GstStateChangeReturn
1365 gst_rtp_jitter_buffer_change_state (GstElement * element,
1366 GstStateChange transition)
1368 GstRtpJitterBuffer *jitterbuffer;
1369 GstRtpJitterBufferPrivate *priv;
1370 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1372 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1373 priv = jitterbuffer->priv;
1375 switch (transition) {
1376 case GST_STATE_CHANGE_NULL_TO_READY:
1378 case GST_STATE_CHANGE_READY_TO_PAUSED:
1380 /* reset negotiated values */
1381 priv->clock_rate = -1;
1382 priv->clock_base = -1;
1383 priv->peer_latency = 0;
1385 /* block until we go to PLAYING */
1386 priv->blocked = TRUE;
1387 priv->timer_running = TRUE;
1388 priv->timer_thread =
1389 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1392 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1394 /* unblock to allow streaming in PLAYING */
1395 priv->blocked = FALSE;
1396 JBUF_SIGNAL_EVENT (priv);
1397 JBUF_SIGNAL_TIMER (priv);
1404 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1406 switch (transition) {
1407 case GST_STATE_CHANGE_READY_TO_PAUSED:
1408 /* we are a live element because we sync to the clock, which we can only
1409 * do in the PLAYING state */
1410 if (ret != GST_STATE_CHANGE_FAILURE)
1411 ret = GST_STATE_CHANGE_NO_PREROLL;
1413 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1415 /* block to stop streaming when PAUSED */
1416 priv->blocked = TRUE;
1417 unschedule_current_timer (jitterbuffer);
1419 if (ret != GST_STATE_CHANGE_FAILURE)
1420 ret = GST_STATE_CHANGE_NO_PREROLL;
1422 case GST_STATE_CHANGE_PAUSED_TO_READY:
1424 gst_buffer_replace (&priv->last_sr, NULL);
1425 priv->timer_running = FALSE;
1426 unschedule_current_timer (jitterbuffer);
1427 JBUF_SIGNAL_TIMER (priv);
1428 JBUF_SIGNAL_QUERY (priv, FALSE);
1430 g_thread_join (priv->timer_thread);
1431 priv->timer_thread = NULL;
1433 case GST_STATE_CHANGE_READY_TO_NULL:
1443 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1446 gboolean ret = TRUE;
1447 GstRtpJitterBuffer *jitterbuffer;
1448 GstRtpJitterBufferPrivate *priv;
1450 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1451 priv = jitterbuffer->priv;
1453 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1455 switch (GST_EVENT_TYPE (event)) {
1456 case GST_EVENT_LATENCY:
1458 GstClockTime latency;
1460 gst_event_parse_latency (event, &latency);
1462 GST_DEBUG_OBJECT (jitterbuffer,
1463 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1466 /* adjust the overall buffer delay to the total pipeline latency in
1467 * buffering mode because if downstream consumes too fast (because of
1468 * large latency or queues, we would start rebuffering again. */
1469 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1470 RTP_JITTER_BUFFER_MODE_BUFFER) {
1471 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1475 ret = gst_pad_push_event (priv->sinkpad, event);
1479 ret = gst_pad_push_event (priv->sinkpad, event);
1486 /* handles and stores the event in the jitterbuffer, must be called with
1489 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1491 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1492 RTPJitterBufferItem *item;
1495 switch (GST_EVENT_TYPE (event)) {
1496 case GST_EVENT_CAPS:
1500 gst_event_parse_caps (event, &caps);
1501 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1504 case GST_EVENT_SEGMENT:
1505 gst_event_copy_segment (event, &priv->segment);
1507 /* we need time for now */
1508 if (priv->segment.format != GST_FORMAT_TIME)
1509 goto newseg_wrong_format;
1511 GST_DEBUG_OBJECT (jitterbuffer,
1512 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1516 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1523 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1524 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1525 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1527 JBUF_SIGNAL_EVENT (priv);
1532 newseg_wrong_format:
1534 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1535 gst_event_unref (event);
1541 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1544 gboolean ret = TRUE;
1545 GstRtpJitterBuffer *jitterbuffer;
1546 GstRtpJitterBufferPrivate *priv;
1548 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1549 priv = jitterbuffer->priv;
1551 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1553 switch (GST_EVENT_TYPE (event)) {
1554 case GST_EVENT_FLUSH_START:
1555 ret = gst_pad_push_event (priv->srcpad, event);
1556 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1557 /* wait for the loop to go into PAUSED */
1558 gst_pad_pause_task (priv->srcpad);
1560 case GST_EVENT_FLUSH_STOP:
1561 ret = gst_pad_push_event (priv->srcpad, event);
1563 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1564 GST_PAD_MODE_PUSH, TRUE);
1567 if (GST_EVENT_IS_SERIALIZED (event)) {
1568 /* serialized events go in the queue */
1570 if (priv->srcresult != GST_FLOW_OK) {
1571 /* Errors in sticky event pushing are no problem and ignored here
1572 * as they will cause more meaningful errors during data flow.
1573 * For EOS events, that are not followed by data flow, we still
1574 * return FALSE here though.
1576 if (!GST_EVENT_IS_STICKY (event) ||
1577 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1578 goto out_flow_error;
1580 /* refuse more events on EOS */
1583 ret = queue_event (jitterbuffer, event);
1586 /* non-serialized events are forwarded downstream immediately */
1587 ret = gst_pad_push_event (priv->srcpad, event);
1596 GST_DEBUG_OBJECT (jitterbuffer,
1597 "refusing event, we have a downstream flow error: %s",
1598 gst_flow_get_name (priv->srcresult));
1600 gst_event_unref (event);
1605 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1607 gst_event_unref (event);
1613 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1616 gboolean ret = TRUE;
1617 GstRtpJitterBuffer *jitterbuffer;
1619 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1621 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1623 switch (GST_EVENT_TYPE (event)) {
1624 case GST_EVENT_FLUSH_START:
1625 gst_event_unref (event);
1627 case GST_EVENT_FLUSH_STOP:
1628 gst_event_unref (event);
1631 ret = gst_pad_event_default (pad, parent, event);
1639 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1640 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1641 * GST_FLOW_FLUSHING when the element is shutting down. On success
1642 * GST_FLOW_OK is returned.
1644 static GstFlowReturn
1645 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1649 GValue args[2] = { {0}, {0} };
1653 g_value_init (&args[0], GST_TYPE_ELEMENT);
1654 g_value_set_object (&args[0], jitterbuffer);
1655 g_value_init (&args[1], G_TYPE_UINT);
1656 g_value_set_uint (&args[1], pt);
1658 g_value_init (&ret, GST_TYPE_CAPS);
1659 g_value_set_boxed (&ret, NULL);
1661 JBUF_UNLOCK (jitterbuffer->priv);
1662 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1664 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1666 g_value_unset (&args[0]);
1667 g_value_unset (&args[1]);
1668 caps = (GstCaps *) g_value_dup_boxed (&ret);
1669 g_value_unset (&ret);
1673 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1674 gst_caps_unref (caps);
1676 if (G_UNLIKELY (!res))
1684 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1685 return GST_FLOW_ERROR;
1689 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1690 return GST_FLOW_FLUSHING;
1694 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1695 return GST_FLOW_ERROR;
1699 /* call with jbuf lock held */
1701 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1703 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1704 GstMessage *message = NULL;
1709 /* Post a buffering message */
1710 if (priv->last_percent != percent) {
1711 priv->last_percent = percent;
1713 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1714 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1721 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1723 GstRtpJitterBufferPrivate *priv;
1725 priv = jitterbuffer->priv;
1727 if (timestamp == -1)
1730 /* apply the timestamp offset, this is used for inter stream sync */
1731 timestamp += priv->ts_offset;
1732 /* add the offset, this is used when buffering */
1733 timestamp += priv->out_offset;
1739 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1741 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1742 TimerData *timer = NULL;
1745 len = priv->timers->len;
1746 for (i = 0; i < len; i++) {
1747 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1748 if (test->seqnum == seqnum && test->type == type) {
1757 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1759 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1761 if (priv->clock_id) {
1762 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1763 gst_clock_id_unschedule (priv->clock_id);
1764 priv->clock_id = NULL;
1769 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1771 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1772 GstClockTime test_timeout;
1774 if ((test_timeout = timer->timeout) == -1)
1777 if (timer->type != TIMER_TYPE_EXPECTED) {
1778 /* add our latency and offset to get output times. */
1779 test_timeout = apply_offset (jitterbuffer, test_timeout);
1780 test_timeout += priv->latency_ns;
1782 return test_timeout;
1786 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1788 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1790 if (priv->clock_id) {
1791 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1793 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1794 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1796 if (timeout == -1 || timeout < priv->timer_timeout)
1797 unschedule_current_timer (jitterbuffer);
1802 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1803 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1804 GstClockTime duration)
1806 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1810 GST_DEBUG_OBJECT (jitterbuffer,
1811 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1812 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1813 GST_TIME_ARGS (delay));
1815 len = priv->timers->len;
1816 g_array_set_size (priv->timers, len + 1);
1817 timer = &g_array_index (priv->timers, TimerData, len);
1820 timer->seqnum = seqnum;
1822 timer->timeout = timeout + delay;
1823 timer->duration = duration;
1824 if (type == TIMER_TYPE_EXPECTED) {
1825 timer->rtx_base = timeout;
1826 timer->rtx_delay = delay;
1827 timer->rtx_retry = 0;
1829 timer->num_rtx_retry = 0;
1830 recalculate_timer (jitterbuffer, timer);
1831 JBUF_SIGNAL_TIMER (priv);
1837 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1838 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1840 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1841 gboolean seqchange, timechange;
1844 seqchange = timer->seqnum != seqnum;
1845 timechange = timer->timeout != timeout;
1847 if (!seqchange && !timechange)
1850 oldseq = timer->seqnum;
1852 GST_DEBUG_OBJECT (jitterbuffer,
1853 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1854 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1856 timer->timeout = timeout + delay;
1857 timer->seqnum = seqnum;
1859 timer->rtx_base = timeout;
1860 timer->rtx_delay = delay;
1861 timer->rtx_retry = 0;
1864 timer->num_rtx_retry = 0;
1866 if (priv->clock_id) {
1867 /* we changed the seqnum and there is a timer currently waiting with this
1868 * seqnum, unschedule it */
1869 if (seqchange && priv->timer_seqnum == oldseq)
1870 unschedule_current_timer (jitterbuffer);
1871 /* we changed the time, check if it is earlier than what we are waiting
1872 * for and unschedule if so */
1873 else if (timechange)
1874 recalculate_timer (jitterbuffer, timer);
1879 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1880 guint16 seqnum, GstClockTime timeout)
1884 /* find the seqnum timer */
1885 timer = find_timer (jitterbuffer, type, seqnum);
1886 if (timer == NULL) {
1887 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1889 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1895 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1897 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1900 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1901 unschedule_current_timer (jitterbuffer);
1904 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1905 g_array_remove_index_fast (priv->timers, idx);
1910 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1912 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1913 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1914 g_array_set_size (priv->timers, 0);
1915 unschedule_current_timer (jitterbuffer);
1918 /* get the extra delay to wait before sending RTX */
1920 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1924 if (priv->rtx_delay == -1) {
1925 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
1926 delay = DEFAULT_AUTO_RTX_DELAY;
1928 /* jitter is in nanoseconds, maximum of 2x jitter and half the
1929 * packet spacing is a good margin */
1930 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
1933 delay = priv->rtx_delay * GST_MSECOND;
1935 if (priv->rtx_min_delay > 0)
1936 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1941 /* we just received a packet with seqnum and dts.
1943 * First check for old seqnum that we are still expecting. If the gap with the
1944 * current seqnum is too big, unschedule the timeouts.
1946 * If we have a valid packet spacing estimate we can set a timer for when we
1947 * should receive the next packet.
1948 * If we don't have a valid estimate, we remove any timer we might have
1949 * had for this packet.
1952 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1953 GstClockTime dts, gboolean do_next_seqnum)
1955 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1956 TimerData *timer = NULL;
1959 /* go through all timers and unschedule the ones with a large gap, also find
1960 * the timer for the seqnum */
1961 len = priv->timers->len;
1962 for (i = 0; i < len; i++) {
1963 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1966 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1968 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
1969 test->type, test->seqnum, seqnum, gap);
1972 GST_DEBUG ("found timer for current seqnum");
1973 /* the timer for the current seqnum */
1975 /* when no retransmission, we can stop now, we only need to find the
1976 * timer for the current seqnum */
1977 if (!priv->do_retransmission)
1979 } else if (gap > priv->rtx_delay_reorder) {
1980 /* max gap, we exceeded the max reorder distance and we don't expect the
1981 * missing packet to be this reordered */
1982 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1983 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
1987 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
1988 && priv->do_retransmission && priv->rtx_next_seqnum;
1990 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1991 if (timer->num_rtx_retry > 0) {
1992 GstClockTime rtx_last, delay;
1994 /* we scheduled a retry for this packet and now we have it */
1995 priv->num_rtx_success++;
1996 /* all the previous retry attempts failed */
1997 priv->num_rtx_failed += timer->num_rtx_retry - 1;
1998 /* number of retries before receiving the packet */
1999 if (priv->avg_rtx_num == 0.0)
2000 priv->avg_rtx_num = timer->num_rtx_retry;
2002 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2003 /* calculate the delay between retransmission request and receiving this
2004 * packet, start with when we scheduled this timeout last */
2005 rtx_last = timer->rtx_last;
2006 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2007 /* we have a valid delay if this packet arrived after we scheduled the
2009 delay = dts - rtx_last;
2010 if (priv->avg_rtx_rtt == 0)
2011 priv->avg_rtx_rtt = delay;
2013 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2017 GST_LOG_OBJECT (jitterbuffer,
2018 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2019 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2020 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2021 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2022 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2023 GST_TIME_ARGS (priv->avg_rtx_rtt));
2025 /* don't try to estimate the next seqnum because this is a retransmitted
2026 * packet and it probably did not arrive with the expected packet
2028 do_next_seqnum = FALSE;
2032 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2033 GstClockTime expected, delay;
2035 /* calculate expected arrival time of the next seqnum */
2036 expected = dts + priv->packet_spacing;
2038 delay = get_rtx_delay (priv);
2040 /* and update/install timer for next seqnum */
2042 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2045 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2046 expected, delay, priv->packet_spacing);
2048 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2049 /* if we had a timer, remove it, we don't know when to expect the next
2051 remove_timer (jitterbuffer, timer);
2056 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2059 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2061 /* we need consecutive seqnums with a different
2062 * rtptime to estimate the packet spacing. */
2063 if (priv->ips_rtptime != rtptime) {
2064 /* rtptime changed, check dts diff */
2065 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2066 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2067 GstClockTime old_packet_spacing = priv->packet_spacing;
2069 /* Biased towards bigger packet spacings to prevent
2070 * too many unneeded retransmission requests for next
2071 * packets that just arrive a little later than we would
2073 if (old_packet_spacing > new_packet_spacing)
2074 priv->packet_spacing =
2075 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2076 else if (old_packet_spacing > 0)
2077 priv->packet_spacing =
2078 (3 * new_packet_spacing + old_packet_spacing) / 4;
2080 priv->packet_spacing = new_packet_spacing;
2082 GST_DEBUG_OBJECT (jitterbuffer,
2083 "new packet spacing %" GST_TIME_FORMAT
2084 " old packet spacing %" GST_TIME_FORMAT
2085 " combined to %" GST_TIME_FORMAT,
2086 GST_TIME_ARGS (new_packet_spacing),
2087 GST_TIME_ARGS (old_packet_spacing),
2088 GST_TIME_ARGS (priv->packet_spacing));
2090 priv->ips_rtptime = rtptime;
2091 priv->ips_dts = dts;
2096 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2097 guint16 seqnum, GstClockTime dts, gint gap)
2099 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2100 GstClockTime total_duration, duration, expected_dts;
2103 GST_DEBUG_OBJECT (jitterbuffer,
2104 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2105 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2107 if (dts == GST_CLOCK_TIME_NONE) {
2108 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2112 /* the total duration spanned by the missing packets */
2113 if (dts >= priv->last_in_dts)
2114 total_duration = dts - priv->last_in_dts;
2118 /* interpolate between the current time and the last time based on
2119 * number of packets we are missing, this is the estimated duration
2120 * for the missing packet based on equidistant packet spacing. */
2121 duration = total_duration / (gap + 1);
2123 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2124 GST_TIME_ARGS (duration));
2126 if (total_duration > priv->latency_ns) {
2127 GstClockTime gap_time;
2131 GstClockTime gap_dur = gap * duration;
2132 if (gap_dur > priv->latency_ns)
2133 gap_time = gap_dur - priv->latency_ns;
2136 lost_packets = gap_time / duration;
2138 gap_time = total_duration - priv->latency_ns;
2142 /* too many lost packets, some of the missing packets are already
2143 * too late and we can generate lost packet events for them. */
2144 GST_DEBUG_OBJECT (jitterbuffer,
2145 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2146 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2147 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2148 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2149 GST_TIME_ARGS (gap_time));
2151 /* this timer will fire immediately and the lost event will be pushed from
2152 * the timer thread */
2153 if (lost_packets > 0) {
2154 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2155 priv->last_in_dts + duration, 0, gap_time);
2156 expected += lost_packets;
2157 priv->last_in_dts += gap_time;
2161 expected_dts = priv->last_in_dts + duration;
2163 if (priv->do_retransmission) {
2166 type = TIMER_TYPE_EXPECTED;
2167 /* if we had a timer for the first missing packet, update it. */
2168 if ((timer = find_timer (jitterbuffer, type, expected))) {
2169 GstClockTime timeout = timer->timeout;
2171 timer->duration = duration;
2172 if (timeout > (expected_dts + timer->rtx_retry)) {
2173 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2174 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2178 expected_dts += duration;
2181 type = TIMER_TYPE_LOST;
2184 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2185 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2186 expected_dts += duration;
2192 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2196 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2197 GstRtpJitterBufferPrivate *priv;
2199 priv = jitterbuffer->priv;
2201 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2204 if (priv->last_dts != -1)
2205 dtsdiff = dts - priv->last_dts;
2209 if (priv->last_rtptime != -1)
2210 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2214 priv->last_dts = dts;
2215 priv->last_rtptime = rtptime;
2219 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2222 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2224 diff = ABS (dtsdiff - rtpdiffns);
2226 /* jitter is stored in nanoseconds */
2227 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2229 GST_LOG_OBJECT (jitterbuffer,
2230 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2231 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2232 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2233 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2240 GST_DEBUG_OBJECT (jitterbuffer,
2241 "no dts or no clock-rate, can't calculate jitter");
2247 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2249 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2250 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2253 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2254 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2255 gst_rtp_buffer_unmap (&rtp_a);
2257 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2258 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2259 gst_rtp_buffer_unmap (&rtp_b);
2261 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2265 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2266 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap)
2268 GstRtpJitterBufferPrivate *priv;
2269 guint gap_packets_length;
2270 gboolean reset = FALSE;
2272 priv = jitterbuffer->priv;
2274 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2276 guint32 prev_gap_seq = -1;
2277 gboolean all_consecutive = TRUE;
2279 g_queue_insert_sorted (&priv->gap_packets, buffer,
2280 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2282 for (l = priv->gap_packets.head; l; l = l->next) {
2283 GstBuffer *gap_buffer = l->data;
2284 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2287 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2289 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2291 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2292 if (prev_gap_seq == -1)
2293 prev_gap_seq = gap_seq;
2294 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2295 all_consecutive = FALSE;
2297 prev_gap_seq = gap_seq;
2299 gst_rtp_buffer_unmap (&gap_rtp);
2300 if (!all_consecutive)
2304 if (all_consecutive && gap_packets_length > 3) {
2305 GST_DEBUG_OBJECT (jitterbuffer,
2306 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2307 (future ? "new" : "old"), gap,
2308 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2310 } else if (!all_consecutive) {
2311 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2312 g_queue_clear (&priv->gap_packets);
2313 GST_DEBUG_OBJECT (jitterbuffer,
2314 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2315 (future ? "new" : "old"), gap,
2316 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2319 GST_DEBUG_OBJECT (jitterbuffer,
2320 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2321 (future ? "new" : "old"), gap,
2322 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER),
2323 gap_packets_length + 1);
2327 GST_DEBUG_OBJECT (jitterbuffer,
2328 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2329 gap, -RTP_MAX_MISORDER);
2330 g_queue_push_tail (&priv->gap_packets, buffer);
2338 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2340 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2341 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2344 GstClockTime base_time =
2345 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2346 GstClockTime clock_time = gst_clock_get_time (clock);
2348 if (clock_time > base_time)
2349 running_time = clock_time - base_time;
2353 gst_object_unref (clock);
2356 return running_time;
2359 static GstFlowReturn
2360 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2363 GstRtpJitterBuffer *jitterbuffer;
2364 GstRtpJitterBufferPrivate *priv;
2366 guint32 expected, rtptime;
2367 GstFlowReturn ret = GST_FLOW_OK;
2368 GstClockTime dts, pts;
2373 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2374 gboolean do_next_seqnum = FALSE;
2375 RTPJitterBufferItem *item;
2376 GstMessage *msg = NULL;
2377 gboolean estimated_dts = FALSE;
2379 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2381 priv = jitterbuffer->priv;
2383 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2384 goto invalid_buffer;
2386 pt = gst_rtp_buffer_get_payload_type (&rtp);
2387 seqnum = gst_rtp_buffer_get_seq (&rtp);
2388 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2389 gst_rtp_buffer_unmap (&rtp);
2391 /* make sure we have PTS and DTS set */
2392 pts = GST_BUFFER_PTS (buffer);
2393 dts = GST_BUFFER_DTS (buffer);
2400 /* If we have no DTS here, i.e. no capture time, get one from the
2401 * clock now to have something to calculate with in the future. */
2402 dts = get_current_running_time (jitterbuffer);
2405 /* Remember that we estimated the DTS if we are running already
2406 * and this is not our first packet (or first packet after a reset).
2407 * If it's the first packet, we somehow must generate a timestamp for
2408 * everything, otherwise we can't calculate any times
2410 estimated_dts = (priv->next_in_seqnum != -1);
2412 /* take the DTS of the buffer. This is the time when the packet was
2413 * received and is used to calculate jitter and clock skew. We will adjust
2414 * this DTS with the smoothed value after processing it in the
2415 * jitterbuffer and assign it as the PTS. */
2416 /* bring to running time */
2417 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2420 GST_DEBUG_OBJECT (jitterbuffer,
2421 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2422 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2424 JBUF_LOCK_CHECK (priv, out_flushing);
2426 if (G_UNLIKELY (priv->last_pt != pt)) {
2429 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2433 /* reset clock-rate so that we get a new one */
2434 priv->clock_rate = -1;
2436 /* Try to get the clock-rate from the caps first if we can. If there are no
2437 * caps we must fire the signal to get the clock-rate. */
2438 if ((caps = gst_pad_get_current_caps (pad))) {
2439 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2440 gst_caps_unref (caps);
2444 if (G_UNLIKELY (priv->clock_rate == -1)) {
2445 /* no clock rate given on the caps, try to get one with the signal */
2446 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2447 pt) == GST_FLOW_FLUSHING)
2450 if (G_UNLIKELY (priv->clock_rate == -1))
2454 /* don't accept more data on EOS */
2455 if (G_UNLIKELY (priv->eos))
2458 calculate_jitter (jitterbuffer, dts, rtptime);
2460 if (priv->seqnum_base != -1) {
2463 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2466 GST_DEBUG_OBJECT (jitterbuffer,
2467 "packet seqnum #%d before seqnum-base #%d", seqnum,
2469 gst_buffer_unref (buffer);
2472 } else if (gap > 16384) {
2473 /* From now on don't compare against the seqnum base anymore as
2474 * at some point in the future we will wrap around and also that
2475 * much reordering is very unlikely */
2476 priv->seqnum_base = -1;
2480 expected = priv->next_in_seqnum;
2482 /* now check against our expected seqnum */
2483 if (G_LIKELY (expected != -1)) {
2486 /* now calculate gap */
2487 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2489 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2490 expected, seqnum, gap);
2492 if (G_LIKELY (gap == 0)) {
2493 /* packet is expected */
2494 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2495 do_next_seqnum = TRUE;
2497 gboolean reset = FALSE;
2500 /* we received an old packet */
2501 if (G_UNLIKELY (gap != -1 && gap < -RTP_MAX_MISORDER)) {
2503 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2507 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2510 /* new packet, we are missing some packets */
2511 if (G_UNLIKELY (priv->timers->len >= RTP_MAX_DROPOUT)) {
2512 /* If we have timers for more than RTP_MAX_DROPOUT packets
2513 * pending this means that we have a huge gap overall. We can
2514 * reset the jitterbuffer at this point because there's
2515 * just too much data missing to be able to do anything
2516 * sensible with the past data. Just try again from the
2518 GST_WARNING_OBJECT (jitterbuffer,
2519 "%d pending timers > %d - resetting", priv->timers->len,
2522 gst_buffer_unref (buffer);
2524 } else if (G_UNLIKELY (gap >= RTP_MAX_DROPOUT)) {
2526 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2530 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2531 /* fill in the gap with EXPECTED timers */
2532 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2534 do_next_seqnum = TRUE;
2537 if (G_UNLIKELY (reset)) {
2538 GList *events = NULL, *l;
2541 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2542 rtp_jitter_buffer_flush (priv->jbuf,
2543 (GFunc) free_item_and_retain_events, &events);
2544 rtp_jitter_buffer_reset_skew (priv->jbuf);
2545 remove_all_timers (jitterbuffer);
2546 priv->discont = TRUE;
2547 priv->last_popped_seqnum = -1;
2548 priv->next_seqnum = seqnum;
2550 priv->last_in_dts = -1;
2551 priv->next_in_seqnum = -1;
2553 /* Insert all sticky events again in order, otherwise we would
2554 * potentially loose STREAM_START, CAPS or SEGMENT events
2556 events = g_list_reverse (events);
2557 for (l = events; l; l = l->next) {
2558 RTPJitterBufferItem *item;
2560 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2561 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2563 g_list_free (events);
2565 JBUF_SIGNAL_EVENT (priv);
2567 /* reset spacing estimation when gap */
2568 priv->ips_rtptime = -1;
2569 priv->ips_dts = GST_CLOCK_TIME_NONE;
2571 buffers = g_list_copy (priv->gap_packets.head);
2572 g_queue_clear (&priv->gap_packets);
2574 priv->ips_rtptime = -1;
2575 priv->ips_dts = GST_CLOCK_TIME_NONE;
2576 JBUF_UNLOCK (jitterbuffer->priv);
2578 for (l = buffers; l; l = l->next) {
2579 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2581 if (ret != GST_FLOW_OK)
2584 for (; l; l = l->next)
2585 gst_buffer_unref (l->data);
2586 g_list_free (buffers);
2590 /* reset spacing estimation when gap */
2591 priv->ips_rtptime = -1;
2592 priv->ips_dts = GST_CLOCK_TIME_NONE;
2595 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2597 /* we don't know what the next_in_seqnum should be, wait for the last
2598 * possible moment to push this buffer, maybe we get an earlier seqnum
2600 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2601 do_next_seqnum = TRUE;
2602 /* take rtptime and dts to calculate packet spacing */
2603 priv->ips_rtptime = rtptime;
2604 priv->ips_dts = dts;
2607 /* We had no huge gap, let's drop all the gap packets */
2608 if (buffer != NULL) {
2609 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2610 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2611 g_queue_clear (&priv->gap_packets);
2613 GST_DEBUG_OBJECT (jitterbuffer,
2614 "Had big gap, waiting for more consecutive packets");
2615 JBUF_UNLOCK (jitterbuffer->priv);
2619 if (do_next_seqnum) {
2620 priv->last_in_dts = dts;
2621 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2624 /* let's check if this buffer is too late, we can only accept packets with
2625 * bigger seqnum than the one we last pushed. */
2626 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2629 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2631 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2632 if (G_UNLIKELY (gap <= 0))
2636 /* let's drop oldest packet if the queue is already full and drop-on-latency
2637 * is set. We can only do this when there actually is a latency. When no
2638 * latency is set, we just pump it in the queue and let the other end push it
2639 * out as fast as possible. */
2640 if (priv->latency_ms && priv->drop_on_latency) {
2642 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2644 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2645 RTPJitterBufferItem *old_item;
2647 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2649 if (IS_DROPABLE (old_item)) {
2650 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2651 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2653 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2654 free_item (old_item);
2656 /* we might have removed some head buffers, signal the pushing thread to
2657 * see if it can push now */
2658 JBUF_SIGNAL_EVENT (priv);
2662 /* If we estimated the DTS, don't consider it in the clock skew calculations
2663 * later. The code above always sets dts to pts or the other way around if
2664 * any of those is valid in the buffer, so we know that if we estimated the
2665 * dts that both are unknown */
2668 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2669 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2671 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2673 /* now insert the packet into the queue in sorted order. This function returns
2674 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2675 * have a duplicate. */
2676 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2681 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2683 /* we had an unhandled SR, handle it now */
2685 do_handle_sync (jitterbuffer);
2687 if (G_UNLIKELY (head)) {
2688 /* signal addition of new buffer when the _loop is waiting. */
2689 if (G_LIKELY (priv->active))
2690 JBUF_SIGNAL_EVENT (priv);
2692 /* let's unschedule and unblock any waiting buffers. We only want to do this
2693 * when the head buffer changed */
2694 if (G_UNLIKELY (priv->clock_id)) {
2695 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2696 unschedule_current_timer (jitterbuffer);
2700 GST_DEBUG_OBJECT (jitterbuffer,
2701 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2702 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2704 msg = check_buffering_percent (jitterbuffer, percent);
2710 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2717 /* this is not fatal but should be filtered earlier */
2718 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2719 ("Received invalid RTP payload, dropping"));
2720 gst_buffer_unref (buffer);
2725 GST_WARNING_OBJECT (jitterbuffer,
2726 "No clock-rate in caps!, dropping buffer");
2727 gst_buffer_unref (buffer);
2732 ret = priv->srcresult;
2733 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2734 gst_buffer_unref (buffer);
2740 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2741 gst_buffer_unref (buffer);
2746 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2747 " popped, dropping", seqnum, priv->last_popped_seqnum);
2749 gst_buffer_unref (buffer);
2754 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2756 priv->num_duplicates++;
2763 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2765 guint64 ext_time, elapsed;
2767 GstRtpJitterBufferPrivate *priv;
2769 priv = jitterbuffer->priv;
2770 rtp_time = item->rtptime;
2772 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2773 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2775 ext_time = priv->ext_timestamp;
2776 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2777 if (ext_time < priv->ext_timestamp) {
2778 ext_time = priv->ext_timestamp;
2780 priv->ext_timestamp = ext_time;
2783 if (ext_time > priv->clock_base)
2784 elapsed = ext_time - priv->clock_base;
2788 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2793 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2794 RTPJitterBufferItem * item)
2796 guint64 total, elapsed, left, estimated;
2797 GstClockTime out_time;
2798 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2800 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2801 || priv->clock_base == -1 || priv->clock_rate <= 0)
2804 /* compute the elapsed time */
2805 elapsed = compute_elapsed (jitterbuffer, item);
2807 /* do nothing if elapsed time doesn't increment */
2808 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2811 priv->last_elapsed = elapsed;
2813 /* this is the total time we need to play */
2814 total = priv->npt_stop - priv->npt_start;
2815 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2816 GST_TIME_ARGS (total));
2818 /* this is how much time there is left */
2819 if (total > elapsed)
2820 left = total - elapsed;
2824 /* if we have less time left that the size of the buffer, we will not
2825 * be able to keep it filled, disabled buffering then */
2826 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2827 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2828 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2829 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2832 /* this is the current time as running-time */
2833 out_time = item->dts;
2836 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2838 /* if there is almost nothing left,
2839 * we may never advance enough to end up in the above case */
2840 if (total < GST_SECOND)
2841 estimated = GST_SECOND;
2845 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2846 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2848 if (estimated != -1 && priv->estimated_eos != estimated) {
2849 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2850 priv->estimated_eos = estimated;
2854 /* take a buffer from the queue and push it */
2855 static GstFlowReturn
2856 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2858 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2859 GstFlowReturn result = GST_FLOW_OK;
2860 RTPJitterBufferItem *item;
2861 GstBuffer *outbuf = NULL;
2862 GstEvent *outevent = NULL;
2863 GstQuery *outquery = NULL;
2864 GstClockTime dts, pts;
2866 gboolean do_push = TRUE;
2870 /* when we get here we are ready to pop and push the buffer */
2871 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2875 case ITEM_TYPE_BUFFER:
2877 /* we need to make writable to change the flags and timestamps */
2878 outbuf = gst_buffer_make_writable (item->data);
2880 if (G_UNLIKELY (priv->discont)) {
2881 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2882 * into the jitterbuffer so we can modify now. */
2883 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2884 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2885 priv->discont = FALSE;
2887 if (G_UNLIKELY (priv->ts_discont)) {
2888 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2889 priv->ts_discont = FALSE;
2893 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2895 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2897 /* apply timestamp with offset to buffer now */
2898 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2899 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2901 /* update the elapsed time when we need to check against the npt stop time. */
2902 update_estimated_eos (jitterbuffer, item);
2904 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2906 case ITEM_TYPE_LOST:
2907 priv->discont = TRUE;
2911 case ITEM_TYPE_EVENT:
2912 outevent = item->data;
2914 case ITEM_TYPE_QUERY:
2915 outquery = item->data;
2919 /* now we are ready to push the buffer. Save the seqnum and release the lock
2920 * so the other end can push stuff in the queue again. */
2922 priv->last_popped_seqnum = seqnum;
2923 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2925 msg = check_buffering_percent (jitterbuffer, percent);
2932 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2935 case ITEM_TYPE_BUFFER:
2937 GST_DEBUG_OBJECT (jitterbuffer,
2938 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2939 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2940 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2941 result = gst_pad_push (priv->srcpad, outbuf);
2943 JBUF_LOCK_CHECK (priv, out_flushing);
2945 case ITEM_TYPE_LOST:
2946 case ITEM_TYPE_EVENT:
2947 /* We got not enough consecutive packets with a huge gap, we can
2948 * as well just drop them here now on EOS */
2949 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
2950 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
2951 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2952 g_queue_clear (&priv->gap_packets);
2955 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2956 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
2959 gst_pad_push_event (priv->srcpad, outevent);
2961 gst_event_unref (outevent);
2963 result = GST_FLOW_OK;
2965 JBUF_LOCK_CHECK (priv, out_flushing);
2967 case ITEM_TYPE_QUERY:
2971 res = gst_pad_peer_query (priv->srcpad, outquery);
2973 JBUF_LOCK_CHECK (priv, out_flushing);
2974 result = GST_FLOW_OK;
2975 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
2976 JBUF_SIGNAL_QUERY (priv, res);
2985 return priv->srcresult;
2989 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2991 /* Peek a buffer and compare the seqnum to the expected seqnum.
2992 * If all is fine, the buffer is pushed.
2993 * If something is wrong, we wait for some event
2995 static GstFlowReturn
2996 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2998 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2999 GstFlowReturn result;
3000 RTPJitterBufferItem *item;
3002 guint32 next_seqnum;
3004 /* only push buffers when PLAYING and active and not buffering */
3005 if (priv->blocked || !priv->active ||
3006 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3007 return GST_FLOW_WAIT;
3010 /* peek a buffer, we're just looking at the sequence number.
3011 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3012 * wait for a timeout or something to change.
3013 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3014 item = rtp_jitter_buffer_peek (priv->jbuf);
3019 /* get the seqnum and the next expected seqnum */
3020 seqnum = item->seqnum;
3022 return pop_and_push_next (jitterbuffer, seqnum);
3025 next_seqnum = priv->next_seqnum;
3027 /* get the gap between this and the previous packet. If we don't know the
3028 * previous packet seqnum assume no gap. */
3029 if (G_UNLIKELY (next_seqnum == -1)) {
3030 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3031 /* we don't know what the next_seqnum should be, the chain function should
3032 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3033 * fires, so wait for that */
3034 result = GST_FLOW_WAIT;
3036 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3038 if (G_LIKELY (gap == 0)) {
3039 /* no missing packet, pop and push */
3040 result = pop_and_push_next (jitterbuffer, seqnum);
3041 } else if (G_UNLIKELY (gap < 0)) {
3042 /* if we have a packet that we already pushed or considered dropped, pop it
3043 * off and get the next packet */
3044 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3045 seqnum, next_seqnum);
3046 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3048 result = GST_FLOW_OK;
3050 /* the chain function has scheduled timers to request retransmission or
3051 * when to consider the packet lost, wait for that */
3052 GST_DEBUG_OBJECT (jitterbuffer,
3053 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3054 next_seqnum, seqnum, gap);
3055 result = GST_FLOW_WAIT;
3063 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3065 return GST_FLOW_EOS;
3067 return GST_FLOW_WAIT;
3073 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3075 GstClockTime rtx_retry_timeout;
3076 GstClockTime rtx_min_retry_timeout;
3078 if (priv->rtx_retry_timeout == -1) {
3079 if (priv->avg_rtx_rtt == 0)
3080 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3082 /* we want to ask for a retransmission after we waited for a
3083 * complete RTT and the additional jitter */
3084 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3086 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3088 /* make sure we don't retry too often. On very low latency networks,
3089 * the RTT and jitter can be very low. */
3090 if (priv->rtx_min_retry_timeout == -1) {
3091 rtx_min_retry_timeout = priv->packet_spacing;
3093 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3095 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3097 return rtx_retry_timeout;
3101 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3102 GstClockTime rtx_retry_timeout)
3104 GstClockTime rtx_retry_period;
3106 if (priv->rtx_retry_period == -1) {
3107 /* we retry up to the configured jitterbuffer size but leaving some
3108 * room for the retransmission to arrive in time */
3109 if (rtx_retry_timeout > priv->latency_ns) {
3110 rtx_retry_period = 0;
3112 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3115 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3117 return rtx_retry_period;
3120 /* the timeout for when we expected a packet expired */
3122 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3125 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3127 guint delay, delay_ms, avg_rtx_rtt_ms;
3128 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3129 GstClockTime rtx_retry_period;
3130 GstClockTime rtx_retry_timeout;
3133 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3134 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3136 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3137 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3139 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3140 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3141 GST_TIME_ARGS (rtx_retry_period));
3143 delay = timer->rtx_delay + timer->rtx_retry;
3145 delay_ms = GST_TIME_AS_MSECONDS (delay);
3146 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3147 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3148 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3150 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3151 gst_structure_new ("GstRTPRetransmissionRequest",
3152 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3153 "running-time", G_TYPE_UINT64, timer->rtx_base,
3154 "delay", G_TYPE_UINT, delay_ms,
3155 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3156 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3157 "period", G_TYPE_UINT, rtx_retry_period_ms,
3158 "deadline", G_TYPE_UINT, priv->latency_ms,
3159 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3160 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3162 priv->num_rtx_requests++;
3163 timer->num_rtx_retry++;
3165 GST_OBJECT_LOCK (jitterbuffer);
3166 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3167 timer->rtx_last = gst_clock_get_time (clock);
3168 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3170 timer->rtx_last = now;
3172 GST_OBJECT_UNLOCK (jitterbuffer);
3174 /* calculate the timeout for the next retransmission attempt */
3175 timer->rtx_retry += rtx_retry_timeout;
3176 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3177 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3178 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3179 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3180 if ((priv->rtx_max_retries != -1
3181 && timer->num_rtx_retry >= priv->rtx_max_retries)
3182 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3183 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3184 /* too many retransmission request, we now convert the timer
3185 * to a lost timer, leave the num_rtx_retry as it is for stats */
3186 timer->type = TIMER_TYPE_LOST;
3187 timer->rtx_delay = 0;
3188 timer->rtx_retry = 0;
3190 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3191 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3194 gst_pad_push_event (priv->sinkpad, event);
3200 /* a packet is lost */
3202 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3205 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3206 GstClockTime duration, timestamp;
3207 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3210 RTPJitterBufferItem *item;
3212 seqnum = timer->seqnum;
3213 timestamp = apply_offset (jitterbuffer, timer->timeout);
3214 duration = timer->duration;
3215 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3216 duration = priv->packet_spacing;
3217 lost_packets = MAX (timer->num, 1);
3218 num_rtx_retry = timer->num_rtx_retry;
3220 /* we had a gap and thus we lost some packets. Create an event for this. */
3221 if (lost_packets > 1)
3222 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3223 seqnum + lost_packets - 1);
3225 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3227 priv->num_late += lost_packets;
3228 priv->num_rtx_failed += num_rtx_retry;
3230 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3232 /* we now only accept seqnum bigger than this */
3233 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3234 priv->next_in_seqnum = next_in_seqnum;
3236 /* create paket lost event */
3237 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3238 gst_structure_new ("GstRTPPacketLost",
3239 "seqnum", G_TYPE_UINT, (guint) seqnum,
3240 "timestamp", G_TYPE_UINT64, timestamp,
3241 "duration", G_TYPE_UINT64, duration,
3242 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3244 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3245 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3247 /* remove timer now */
3248 remove_timer (jitterbuffer, timer);
3250 JBUF_SIGNAL_EVENT (priv);
3256 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3259 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3261 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3262 remove_timer (jitterbuffer, timer);
3264 /* there was no EOS in the buffer, put one in there now */
3265 queue_event (jitterbuffer, gst_event_new_eos ());
3267 JBUF_SIGNAL_EVENT (priv);
3273 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3276 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3278 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3280 /* timer seqnum might have been obsoleted by caps seqnum-base,
3281 * only mess with current ongoing seqnum if still unknown */
3282 if (priv->next_seqnum == -1)
3283 priv->next_seqnum = timer->seqnum;
3284 remove_timer (jitterbuffer, timer);
3285 JBUF_SIGNAL_EVENT (priv);
3291 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3294 gboolean removed = FALSE;
3296 switch (timer->type) {
3297 case TIMER_TYPE_EXPECTED:
3298 removed = do_expected_timeout (jitterbuffer, timer, now);
3300 case TIMER_TYPE_LOST:
3301 removed = do_lost_timeout (jitterbuffer, timer, now);
3303 case TIMER_TYPE_DEADLINE:
3304 removed = do_deadline_timeout (jitterbuffer, timer, now);
3306 case TIMER_TYPE_EOS:
3307 removed = do_eos_timeout (jitterbuffer, timer, now);
3313 /* called when we need to wait for the next timeout.
3315 * We loop over the array of recorded timeouts and wait for the earliest one.
3316 * When it timed out, do the logic associated with the timer.
3318 * If there are no timers, we wait on a gcond until something new happens.
3321 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3323 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3324 GstClockTime now = 0;
3327 while (priv->timer_running) {
3328 TimerData *timer = NULL;
3329 GstClockTime timer_timeout = -1;
3332 /* If we have a clock, update "now" now with the very latest running time
3333 * we have. It is used below when timeouts are triggered to calculate
3334 * any next possible timeout. If we only update it after waiting for the
3335 * clock, we would give a too old time to the timeout functions.
3337 GST_OBJECT_LOCK (jitterbuffer);
3338 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3340 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3341 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3343 GST_OBJECT_UNLOCK (jitterbuffer);
3345 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3346 GST_TIME_ARGS (now));
3348 len = priv->timers->len;
3349 for (i = 0; i < len; i++) {
3350 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3351 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3352 gboolean save_best = FALSE;
3354 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3355 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3357 /* find the smallest timeout */
3358 if (timer == NULL) {
3360 } else if (timer_timeout == -1) {
3361 /* we already have an immediate timeout, the new timer must be an
3362 * immediate timer with smaller seqnum to become the best */
3363 if (test_timeout == -1
3364 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3365 timer->seqnum) > 0))
3367 } else if (test_timeout == -1) {
3368 /* first immediate timer */
3370 } else if (test_timeout < timer_timeout) {
3373 } else if (test_timeout == timer_timeout
3374 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3375 timer->seqnum) > 0)) {
3376 /* same timer, smaller seqnum */
3380 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3382 timer_timeout = test_timeout;
3385 if (timer && !priv->blocked) {
3387 GstClockTime sync_time;
3390 GstClockTimeDiff clock_jitter;
3392 if (timer_timeout == -1 || timer_timeout <= now) {
3393 do_timeout (jitterbuffer, timer, now);
3394 /* check here, do_timeout could have released the lock */
3395 if (!priv->timer_running)
3400 GST_OBJECT_LOCK (jitterbuffer);
3401 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3403 GST_OBJECT_UNLOCK (jitterbuffer);
3404 /* let's just push if there is no clock */
3405 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3406 now = timer_timeout;
3410 /* prepare for sync against clock */
3411 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3412 /* add latency of peer to get input time */
3413 sync_time += priv->peer_latency;
3415 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3416 " with sync time %" GST_TIME_FORMAT,
3417 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3419 /* create an entry for the clock */
3420 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3421 priv->timer_timeout = timer_timeout;
3422 priv->timer_seqnum = timer->seqnum;
3423 GST_OBJECT_UNLOCK (jitterbuffer);
3425 /* release the lock so that the other end can push stuff or unlock */
3428 ret = gst_clock_id_wait (id, &clock_jitter);
3431 if (!priv->timer_running) {
3432 gst_clock_id_unref (id);
3433 priv->clock_id = NULL;
3437 if (ret != GST_CLOCK_UNSCHEDULED) {
3438 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3439 ret, priv->timer_seqnum, clock_jitter);
3441 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3443 /* and free the entry */
3444 gst_clock_id_unref (id);
3445 priv->clock_id = NULL;
3447 /* no timers, wait for activity */
3448 JBUF_WAIT_TIMER (priv);
3453 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3458 * This funcion implements the main pushing loop on the source pad.
3460 * It first tries to push as many buffers as possible. If there is a seqnum
3461 * mismatch, we wait for the next timeouts.
3464 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3466 GstRtpJitterBufferPrivate *priv;
3467 GstFlowReturn result = GST_FLOW_OK;
3469 priv = jitterbuffer->priv;
3471 JBUF_LOCK_CHECK (priv, flushing);
3473 result = handle_next_buffer (jitterbuffer);
3474 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3475 /* now wait for the next event */
3476 JBUF_WAIT_EVENT (priv, flushing);
3477 result = GST_FLOW_OK;
3479 } while (result == GST_FLOW_OK);
3480 /* store result for upstream */
3481 priv->srcresult = result;
3482 /* if we get here we need to pause */
3488 result = priv->srcresult;
3495 JBUF_SIGNAL_QUERY (priv, FALSE);
3498 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3499 gst_flow_get_name (result));
3500 gst_pad_pause_task (priv->srcpad);
3501 if (result == GST_FLOW_EOS) {
3502 event = gst_event_new_eos ();
3503 gst_pad_push_event (priv->srcpad, event);
3509 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3510 * some sanity checks and then emit the handle-sync signal with the parameters.
3511 * This function must be called with the LOCK */
3513 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3515 GstRtpJitterBufferPrivate *priv;
3516 guint64 base_rtptime, base_time;
3518 guint64 last_rtptime;
3520 guint64 ext_rtptime, diff;
3521 gboolean valid = TRUE, keep = FALSE;
3523 priv = jitterbuffer->priv;
3525 /* get the last values from the jitterbuffer */
3526 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3527 &clock_rate, &last_rtptime);
3529 clock_base = priv->clock_base;
3530 ext_rtptime = priv->ext_rtptime;
3532 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3533 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3534 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3535 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3537 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3538 /* we keep this SR packet for later. When we get a valid RTP packet the
3539 * above values will be set and we can try to use the SR packet */
3540 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3543 /* we can't accept anything that happened before we did the last resync */
3544 if (base_rtptime > ext_rtptime) {
3545 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3548 /* the SR RTP timestamp must be something close to what we last observed
3549 * in the jitterbuffer */
3550 if (ext_rtptime > last_rtptime) {
3551 /* check how far ahead it is to our RTP timestamps */
3552 diff = ext_rtptime - last_rtptime;
3553 /* if bigger than 1 second, we drop it */
3554 if (diff > clock_rate) {
3555 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3556 /* should drop this, but some RTSP servers end up with bogus
3557 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3558 * so still trigger rptbin sync but invalidate RTCP data
3559 * (sync might use other methods) */
3562 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3563 G_GUINT64_FORMAT, last_rtptime, diff);
3569 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3573 s = gst_structure_new ("application/x-rtp-sync",
3574 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3575 "base-time", G_TYPE_UINT64, base_time,
3576 "clock-rate", G_TYPE_UINT, clock_rate,
3577 "clock-base", G_TYPE_UINT64, clock_base,
3578 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3579 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3581 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3582 gst_buffer_replace (&priv->last_sr, NULL);
3584 g_signal_emit (jitterbuffer,
3585 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3587 gst_structure_free (s);
3589 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3590 gst_buffer_replace (&priv->last_sr, NULL);
3594 static GstFlowReturn
3595 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3598 GstRtpJitterBuffer *jitterbuffer;
3599 GstRtpJitterBufferPrivate *priv;
3600 GstFlowReturn ret = GST_FLOW_OK;
3602 GstRTCPPacket packet;
3603 guint64 ext_rtptime;
3605 GstRTCPBuffer rtcp = { NULL, };
3607 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3609 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3610 goto invalid_buffer;
3612 priv = jitterbuffer->priv;
3614 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3616 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3619 /* first packet must be SR or RR or else the validate would have failed */
3620 switch (gst_rtcp_packet_get_type (&packet)) {
3621 case GST_RTCP_TYPE_SR:
3622 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3628 gst_rtcp_buffer_unmap (&rtcp);
3630 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3633 /* convert the RTP timestamp to our extended timestamp, using the same offset
3634 * we used in the jitterbuffer */
3635 ext_rtptime = priv->jbuf->ext_rtptime;
3636 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3638 priv->ext_rtptime = ext_rtptime;
3639 gst_buffer_replace (&priv->last_sr, buffer);
3641 do_handle_sync (jitterbuffer);
3645 gst_buffer_unref (buffer);
3651 /* this is not fatal but should be filtered earlier */
3652 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3653 ("Received invalid RTCP payload, dropping"));
3659 /* this is not fatal but should be filtered earlier */
3660 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3661 ("Received empty RTCP payload, dropping"));
3662 gst_rtcp_buffer_unmap (&rtcp);
3668 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3669 gst_rtcp_buffer_unmap (&rtcp);
3676 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3679 gboolean res = FALSE;
3680 GstRtpJitterBuffer *jitterbuffer;
3681 GstRtpJitterBufferPrivate *priv;
3683 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3684 priv = jitterbuffer->priv;
3686 switch (GST_QUERY_TYPE (query)) {
3687 case GST_QUERY_CAPS:
3689 GstCaps *filter, *caps;
3691 gst_query_parse_caps (query, &filter);
3692 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3693 gst_query_set_caps_result (query, caps);
3694 gst_caps_unref (caps);
3699 if (GST_QUERY_IS_SERIALIZED (query)) {
3700 RTPJitterBufferItem *item;
3703 JBUF_LOCK_CHECK (priv, out_flushing);
3704 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3705 RTP_JITTER_BUFFER_MODE_BUFFER) {
3706 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3707 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3708 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3710 JBUF_SIGNAL_EVENT (priv);
3711 JBUF_WAIT_QUERY (priv, out_flushing);
3712 res = priv->last_query;
3714 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3719 res = gst_pad_query_default (pad, parent, query);
3727 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3735 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3738 GstRtpJitterBuffer *jitterbuffer;
3739 GstRtpJitterBufferPrivate *priv;
3740 gboolean res = FALSE;
3742 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3743 priv = jitterbuffer->priv;
3745 switch (GST_QUERY_TYPE (query)) {
3746 case GST_QUERY_LATENCY:
3748 /* We need to send the query upstream and add the returned latency to our
3750 GstClockTime min_latency, max_latency;
3752 GstClockTime our_latency;
3754 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3755 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3757 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3758 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3759 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3761 /* store this so that we can safely sync on the peer buffers. */
3763 priv->peer_latency = min_latency;
3764 our_latency = priv->latency_ns;
3767 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3768 GST_TIME_ARGS (our_latency));
3770 /* we add some latency but can buffer an infinite amount of time */
3771 min_latency += our_latency;
3774 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3775 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3776 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3778 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3782 case GST_QUERY_POSITION:
3784 GstClockTime start, last_out;
3787 gst_query_parse_position (query, &fmt, NULL);
3788 if (fmt != GST_FORMAT_TIME) {
3789 res = gst_pad_query_default (pad, parent, query);
3794 start = priv->npt_start;
3795 last_out = priv->last_out_time;
3798 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3799 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3800 GST_TIME_ARGS (last_out));
3802 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3803 /* bring 0-based outgoing time to stream time */
3804 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3807 res = gst_pad_query_default (pad, parent, query);
3811 case GST_QUERY_CAPS:
3813 GstCaps *filter, *caps;
3815 gst_query_parse_caps (query, &filter);
3816 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3817 gst_query_set_caps_result (query, caps);
3818 gst_caps_unref (caps);
3823 res = gst_pad_query_default (pad, parent, query);
3831 gst_rtp_jitter_buffer_set_property (GObject * object,
3832 guint prop_id, const GValue * value, GParamSpec * pspec)
3834 GstRtpJitterBuffer *jitterbuffer;
3835 GstRtpJitterBufferPrivate *priv;
3837 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3838 priv = jitterbuffer->priv;
3843 guint new_latency, old_latency;
3845 new_latency = g_value_get_uint (value);
3848 old_latency = priv->latency_ms;
3849 priv->latency_ms = new_latency;
3850 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3851 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3854 /* post message if latency changed, this will inform the parent pipeline
3855 * that a latency reconfiguration is possible/needed. */
3856 if (new_latency != old_latency) {
3857 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3858 GST_TIME_ARGS (new_latency * GST_MSECOND));
3860 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3861 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3865 case PROP_DROP_ON_LATENCY:
3867 priv->drop_on_latency = g_value_get_boolean (value);
3870 case PROP_TS_OFFSET:
3872 priv->ts_offset = g_value_get_int64 (value);
3873 priv->ts_discont = TRUE;
3878 priv->do_lost = g_value_get_boolean (value);
3883 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3886 case PROP_DO_RETRANSMISSION:
3888 priv->do_retransmission = g_value_get_boolean (value);
3891 case PROP_RTX_NEXT_SEQNUM:
3893 priv->rtx_next_seqnum = g_value_get_boolean (value);
3896 case PROP_RTX_DELAY:
3898 priv->rtx_delay = g_value_get_int (value);
3901 case PROP_RTX_MIN_DELAY:
3903 priv->rtx_min_delay = g_value_get_uint (value);
3906 case PROP_RTX_DELAY_REORDER:
3908 priv->rtx_delay_reorder = g_value_get_int (value);
3911 case PROP_RTX_RETRY_TIMEOUT:
3913 priv->rtx_retry_timeout = g_value_get_int (value);
3916 case PROP_RTX_MIN_RETRY_TIMEOUT:
3918 priv->rtx_min_retry_timeout = g_value_get_int (value);
3921 case PROP_RTX_RETRY_PERIOD:
3923 priv->rtx_retry_period = g_value_get_int (value);
3926 case PROP_RTX_MAX_RETRIES:
3928 priv->rtx_max_retries = g_value_get_int (value);
3932 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3938 gst_rtp_jitter_buffer_get_property (GObject * object,
3939 guint prop_id, GValue * value, GParamSpec * pspec)
3941 GstRtpJitterBuffer *jitterbuffer;
3942 GstRtpJitterBufferPrivate *priv;
3944 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3945 priv = jitterbuffer->priv;
3950 g_value_set_uint (value, priv->latency_ms);
3953 case PROP_DROP_ON_LATENCY:
3955 g_value_set_boolean (value, priv->drop_on_latency);
3958 case PROP_TS_OFFSET:
3960 g_value_set_int64 (value, priv->ts_offset);
3965 g_value_set_boolean (value, priv->do_lost);
3970 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3978 if (priv->srcresult != GST_FLOW_OK)
3981 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3983 g_value_set_int (value, percent);
3987 case PROP_DO_RETRANSMISSION:
3989 g_value_set_boolean (value, priv->do_retransmission);
3992 case PROP_RTX_NEXT_SEQNUM:
3994 g_value_set_boolean (value, priv->rtx_next_seqnum);
3997 case PROP_RTX_DELAY:
3999 g_value_set_int (value, priv->rtx_delay);
4002 case PROP_RTX_MIN_DELAY:
4004 g_value_set_uint (value, priv->rtx_min_delay);
4007 case PROP_RTX_DELAY_REORDER:
4009 g_value_set_int (value, priv->rtx_delay_reorder);
4012 case PROP_RTX_RETRY_TIMEOUT:
4014 g_value_set_int (value, priv->rtx_retry_timeout);
4017 case PROP_RTX_MIN_RETRY_TIMEOUT:
4019 g_value_set_int (value, priv->rtx_min_retry_timeout);
4022 case PROP_RTX_RETRY_PERIOD:
4024 g_value_set_int (value, priv->rtx_retry_period);
4027 case PROP_RTX_MAX_RETRIES:
4029 g_value_set_int (value, priv->rtx_max_retries);
4033 g_value_take_boxed (value,
4034 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4037 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4042 static GstStructure *
4043 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4047 JBUF_LOCK (jbuf->priv);
4048 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4049 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4050 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4051 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4052 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4053 JBUF_UNLOCK (jbuf->priv);