2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
11 * This library is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Library General Public
13 * License as published by the Free Software Foundation; either
14 * version 2 of the License, or (at your option) any later version.
16 * This library is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Library General Public License for more details.
21 * You should have received a copy of the GNU Library General Public
22 * License along with this library; if not, write to the
23 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
24 * Boston, MA 02110-1301, USA.
29 * SECTION:element-rtpjitterbuffer
31 * This element reorders and removes duplicate RTP packets as they are received
32 * from a network source.
34 * The element needs the clock-rate of the RTP payload in order to estimate the
35 * delay. This information is obtained either from the caps on the sink pad or,
36 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
37 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
39 * The rtpjitterbuffer will wait for missing packets up to a configurable time
40 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
41 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
42 * property is set, lost packets will result in a custom serialized downstream
43 * event of name GstRTPPacketLost. The lost packet events are usually used by a
44 * depayloader or other element to create concealment data or some other logic
45 * to gracefully handle the missing packets.
47 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
48 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
51 * The jitterbuffer can also be configured to send early retransmission events
52 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
53 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
54 * sends a custom upstream event named GstRTPRetransmissionRequest when the
55 * packet is considered late. The initial expected packet arrival time is
56 * calculated as follows:
58 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
59 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
60 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
61 * packets with different rtptime.
63 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
64 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
65 * previously scheduled timeout is overwritten.
67 * - If seqnum N arrived, all seqnum older than
68 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
69 * immediately. This is to request fast feedback for abonormally reorder
70 * packets before any of the previous timeouts is triggered.
72 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
73 * event. After the initial timeout expires and the retransmission event is
74 * sent, the timeout is scheduled for
75 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
76 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
77 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
78 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
79 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
80 * retransmission requests are sent and the regular logic is performed to
81 * schedule a lost packet as discussed above.
83 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
86 * This element will automatically be used inside rtpbin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
92 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
93 * inserted into the pipeline to smooth out network jitter and to reorder the
94 * out-of-order RTP packets.
105 #include <gst/rtp/gstrtpbuffer.h>
106 #include <gst/net/net.h>
108 #include "gstrtpjitterbuffer.h"
109 #include "rtpjitterbuffer.h"
110 #include "rtpstats.h"
112 #include <gst/glib-compat-private.h>
114 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
115 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
117 /* RTPJitterBuffer signals and args */
120 SIGNAL_REQUEST_PT_MAP,
128 #define DEFAULT_LATENCY_MS 200
129 #define DEFAULT_DROP_ON_LATENCY FALSE
130 #define DEFAULT_TS_OFFSET 0
131 #define DEFAULT_DO_LOST FALSE
132 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
133 #define DEFAULT_PERCENT 0
134 #define DEFAULT_DO_RETRANSMISSION FALSE
135 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
136 #define DEFAULT_RTX_DELAY -1
137 #define DEFAULT_RTX_MIN_DELAY 0
138 #define DEFAULT_RTX_DELAY_REORDER 3
139 #define DEFAULT_RTX_RETRY_TIMEOUT -1
140 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
141 #define DEFAULT_RTX_RETRY_PERIOD -1
142 #define DEFAULT_RTX_MAX_RETRIES -1
143 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
144 #define DEFAULT_MAX_DROPOUT_TIME 60000
145 #define DEFAULT_MAX_MISORDER_TIME 2000
147 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
148 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
154 PROP_DROP_ON_LATENCY,
159 PROP_DO_RETRANSMISSION,
160 PROP_RTX_NEXT_SEQNUM,
163 PROP_RTX_DELAY_REORDER,
164 PROP_RTX_RETRY_TIMEOUT,
165 PROP_RTX_MIN_RETRY_TIMEOUT,
166 PROP_RTX_RETRY_PERIOD,
167 PROP_RTX_MAX_RETRIES,
169 PROP_MAX_RTCP_RTP_TIME_DIFF,
170 PROP_MAX_DROPOUT_TIME,
171 PROP_MAX_MISORDER_TIME
174 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
176 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
178 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
181 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
183 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
184 GST_DEBUG ("waiting timer"); \
185 (priv)->waiting_timer = TRUE; \
186 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
187 (priv)->waiting_timer = FALSE; \
188 GST_DEBUG ("waiting timer done"); \
190 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
191 if (G_UNLIKELY ((priv)->waiting_timer)) { \
192 GST_DEBUG ("signal timer"); \
193 g_cond_signal (&(priv)->jbuf_timer); \
197 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
198 GST_DEBUG ("waiting event"); \
199 (priv)->waiting_event = TRUE; \
200 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
201 (priv)->waiting_event = FALSE; \
202 GST_DEBUG ("waiting event done"); \
203 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
206 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
207 if (G_UNLIKELY ((priv)->waiting_event)) { \
208 GST_DEBUG ("signal event"); \
209 g_cond_signal (&(priv)->jbuf_event); \
213 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
214 GST_DEBUG ("waiting query"); \
215 (priv)->waiting_query = TRUE; \
216 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
217 (priv)->waiting_query = FALSE; \
218 GST_DEBUG ("waiting query done"); \
219 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
222 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
223 (priv)->last_query = res; \
224 if (G_UNLIKELY ((priv)->waiting_query)) { \
225 GST_DEBUG ("signal query"); \
226 g_cond_signal (&(priv)->jbuf_query); \
231 struct _GstRtpJitterBufferPrivate
233 GstPad *sinkpad, *srcpad;
236 RTPJitterBuffer *jbuf;
238 gboolean waiting_timer;
240 gboolean waiting_event;
242 gboolean waiting_query;
250 gboolean timer_running;
251 GThread *timer_thread;
256 gboolean drop_on_latency;
259 gboolean do_retransmission;
260 gboolean rtx_next_seqnum;
263 gint rtx_delay_reorder;
264 gint rtx_retry_timeout;
265 gint rtx_min_retry_timeout;
266 gint rtx_retry_period;
267 gint rtx_max_retries;
268 gint max_rtcp_rtp_time_diff;
269 guint32 max_dropout_time;
270 guint32 max_misorder_time;
272 /* the last seqnum we pushed out */
273 guint32 last_popped_seqnum;
274 /* the next expected seqnum we push */
276 /* seqnum-base, if known */
278 /* last output time */
279 GstClockTime last_out_time;
280 /* last valid input timestamp and rtptime pair */
281 GstClockTime ips_dts;
283 GstClockTime packet_spacing;
287 /* the next expected seqnum we receive */
288 GstClockTime last_in_dts;
289 guint32 next_in_seqnum;
293 /* start and stop ranges */
294 GstClockTime npt_start;
295 GstClockTime npt_stop;
296 guint64 ext_timestamp;
297 guint64 last_elapsed;
298 guint64 estimated_eos;
305 /* clock rate and rtp timestamp offset */
309 gint64 prev_ts_offset;
311 /* when we are shutting down */
312 GstFlowReturn srcresult;
318 GstClockTime timer_timeout;
319 guint16 timer_seqnum;
320 /* the latency of the upstream peer, we have to take this into account when
321 * synchronizing the buffers. */
322 GstClockTime peer_latency;
326 /* some accounting */
328 guint64 num_duplicates;
329 guint64 num_rtx_requests;
330 guint64 num_rtx_success;
331 guint64 num_rtx_failed;
334 RTPPacketRateCtx packet_rate_ctx;
337 GstClockTime last_dts;
338 guint64 last_rtptime;
339 GstClockTime avg_jitter;
356 GstClockTime timeout;
357 GstClockTime duration;
358 GstClockTime rtx_base;
359 GstClockTime rtx_delay;
360 GstClockTime rtx_retry;
361 GstClockTime rtx_last;
365 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
366 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
367 GstRtpJitterBufferPrivate))
369 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
370 GST_STATIC_PAD_TEMPLATE ("sink",
373 GST_STATIC_CAPS ("application/x-rtp"
374 /* "clock-rate = (int) [ 1, 2147483647 ], "
375 * "payload = (int) , "
376 * "encoding-name = (string) "
380 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
381 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
384 GST_STATIC_CAPS ("application/x-rtcp")
387 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
388 GST_STATIC_PAD_TEMPLATE ("src",
391 GST_STATIC_CAPS ("application/x-rtp"
392 /* "payload = (int) , "
393 * "clock-rate = (int) , "
394 * "encoding-name = (string) "
398 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
400 #define gst_rtp_jitter_buffer_parent_class parent_class
401 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
403 /* object overrides */
404 static void gst_rtp_jitter_buffer_set_property (GObject * object,
405 guint prop_id, const GValue * value, GParamSpec * pspec);
406 static void gst_rtp_jitter_buffer_get_property (GObject * object,
407 guint prop_id, GValue * value, GParamSpec * pspec);
408 static void gst_rtp_jitter_buffer_finalize (GObject * object);
410 /* element overrides */
411 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
412 * element, GstStateChange transition);
413 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
414 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
415 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
417 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
418 static gboolean gst_rtp_jitter_buffer_set_clock (GstElement * element,
422 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
423 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
426 /* sinkpad overrides */
427 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
428 GstObject * parent, GstEvent * event);
429 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
430 GstObject * parent, GstBuffer * buffer);
432 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
433 GstObject * parent, GstEvent * event);
434 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
435 GstObject * parent, GstBuffer * buffer);
437 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
438 GstObject * parent, GstQuery * query);
440 /* srcpad overrides */
441 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
442 GstObject * parent, GstEvent * event);
443 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
444 GstObject * parent, GstPadMode mode, gboolean active);
445 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
446 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
447 GstObject * parent, GstQuery * query);
450 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
452 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
453 gboolean active, guint64 base_time);
454 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
456 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
457 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
459 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
461 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
465 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
467 GObjectClass *gobject_class;
468 GstElementClass *gstelement_class;
470 gobject_class = (GObjectClass *) klass;
471 gstelement_class = (GstElementClass *) klass;
473 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
475 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
477 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
478 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
481 * GstRtpJitterBuffer:latency:
483 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
484 * for at most this time.
486 g_object_class_install_property (gobject_class, PROP_LATENCY,
487 g_param_spec_uint ("latency", "Buffer latency in ms",
488 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
489 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
491 * GstRtpJitterBuffer:drop-on-latency:
493 * Drop oldest buffers when the queue is completely filled.
495 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
496 g_param_spec_boolean ("drop-on-latency",
497 "Drop buffers when maximum latency is reached",
498 "Tells the jitterbuffer to never exceed the given latency in size",
499 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
501 * GstRtpJitterBuffer:ts-offset:
503 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
504 * This is mainly used to ensure interstream synchronisation.
506 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
507 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
508 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
509 G_MAXINT64, DEFAULT_TS_OFFSET,
510 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
513 * GstRtpJitterBuffer:do-lost:
515 * Send out a GstRTPPacketLost event downstream when a packet is considered
518 g_object_class_install_property (gobject_class, PROP_DO_LOST,
519 g_param_spec_boolean ("do-lost", "Do Lost",
520 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
521 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
524 * GstRtpJitterBuffer:mode:
526 * Control the buffering and timestamping mode used by the jitterbuffer.
528 g_object_class_install_property (gobject_class, PROP_MODE,
529 g_param_spec_enum ("mode", "Mode",
530 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
531 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
533 * GstRtpJitterBuffer:percent:
535 * The percent of the jitterbuffer that is filled.
537 g_object_class_install_property (gobject_class, PROP_PERCENT,
538 g_param_spec_int ("percent", "percent",
539 "The buffer filled percent", 0, 100,
540 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
542 * GstRtpJitterBuffer:do-retransmission:
544 * Send out a GstRTPRetransmission event upstream when a packet is considered
545 * late and should be retransmitted.
549 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
550 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
551 "Send retransmission events upstream when a packet is late",
552 DEFAULT_DO_RETRANSMISSION,
553 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
556 * GstRtpJitterBuffer:rtx-next-seqnum
558 * Estimate when the next packet should arrive and schedule a retransmission
560 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
561 * for packet N+1. So it will be requested if it does not arrive at the expected time.
562 * The expected time is calculated using the dts of N and the packet spacing.
566 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
567 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
568 "Estimate when the next packet should arrive and schedule a "
569 "retransmission request for it.",
570 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
573 * GstRtpJitterBuffer:rtx-delay:
575 * When a packet did not arrive at the expected time, wait this extra amount
576 * of time before sending a retransmission event.
578 * When -1 is used, the max jitter will be used as extra delay.
582 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
583 g_param_spec_int ("rtx-delay", "RTX Delay",
584 "Extra time in ms to wait before sending retransmission "
585 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
586 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
589 * GstRtpJitterBuffer:rtx-min-delay:
591 * When a packet did not arrive at the expected time, wait at least this extra amount
592 * of time before sending a retransmission event.
596 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
597 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
598 "Minimum time in ms to wait before sending retransmission "
599 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
600 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
602 * GstRtpJitterBuffer:rtx-delay-reorder:
604 * Assume that a retransmission event should be sent when we see
605 * this much packet reordering.
607 * When -1 is used, the value will be estimated based on observed packet
612 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
613 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
614 "Sending retransmission event when this much reordering (-1 automatic)",
615 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
616 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
618 * GstRtpJitterBuffer::rtx-retry-timeout:
620 * When no packet has been received after sending a retransmission event
621 * for this time, retry sending a retransmission event.
623 * When -1 is used, the value will be estimated based on observed round
628 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
629 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
630 "Retry sending a transmission event after this timeout in "
631 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
632 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
634 * GstRtpJitterBuffer::rtx-min-retry-timeout:
636 * The minimum amount of time between retry timeouts. When
637 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
638 * minimum interval between retry timeouts.
640 * When -1 is used, the value will be estimated based on the
645 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
646 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
647 "Minimum timeout between sending a transmission event in "
648 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
649 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
651 * GstRtpJitterBuffer:rtx-retry-period:
653 * The amount of time to try to get a retransmission.
655 * When -1 is used, the value will be estimated based on the jitterbuffer
656 * latency and the observed round trip time.
660 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
661 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
662 "Try to get a retransmission for this many ms "
663 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
664 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
666 * GstRtpJitterBuffer:rtx-max-retries:
668 * The maximum number of retries to request a retransmission.
670 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
671 * When -1 is used, the number of retransmission request will not be limited.
675 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
676 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
677 "The maximum number of retries to request a retransmission. "
678 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
679 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
681 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
682 g_param_spec_uint ("max-dropout-time", "Max dropout time",
683 "The maximum time (milliseconds) of missing packets tolerated.",
684 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
685 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
687 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
688 g_param_spec_uint ("max-misorder-time", "Max misorder time",
689 "The maximum time (milliseconds) of misordered packets tolerated.",
690 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
691 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
693 * GstRtpJitterBuffer:stats:
695 * Various jitterbuffer statistics. This property returns a GstStructure
696 * with name application/x-rtp-jitterbuffer-stats with the following fields:
702 * <classname>"rtx-count"</classname>:
703 * the number of retransmissions requested.
709 * <classname>"rtx-success-count"</classname>:
710 * the number of successful retransmissions.
716 * <classname>"rtx-per-packet"</classname>:
717 * average number of RTX per packet.
723 * <classname>"rtx-rtt"</classname>:
724 * average round trip time per RTX.
731 g_object_class_install_property (gobject_class, PROP_STATS,
732 g_param_spec_boxed ("stats", "Statistics",
733 "Various statistics", GST_TYPE_STRUCTURE,
734 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
737 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
739 * The maximum amount of time in ms that the RTP time in the RTCP SRs
740 * is allowed to be ahead of the last RTP packet we received. Use
741 * -1 to disable ignoring of RTCP packets.
745 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
746 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
747 "Maximum amount of time in ms that the RTP time in RTCP SRs "
748 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
749 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
750 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
753 * GstRtpJitterBuffer::request-pt-map:
754 * @buffer: the object which received the signal
757 * Request the payload type as #GstCaps for @pt.
759 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
760 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
761 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
762 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
763 GST_TYPE_CAPS, 1, G_TYPE_UINT);
765 * GstRtpJitterBuffer::handle-sync:
766 * @buffer: the object which received the signal
767 * @struct: a GstStructure containing sync values.
769 * Be notified of new sync values.
771 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
772 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
773 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
774 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
775 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
778 * GstRtpJitterBuffer::on-npt-stop:
779 * @buffer: the object which received the signal
781 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
782 * the npt-stop position.
784 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
785 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
786 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
787 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
788 G_TYPE_NONE, 0, G_TYPE_NONE);
791 * GstRtpJitterBuffer::clear-pt-map:
792 * @buffer: the object which received the signal
794 * Invalidate the clock-rate as obtained with the
795 * #GstRtpJitterBuffer::request-pt-map signal.
797 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
798 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
799 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
800 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
801 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
804 * GstRtpJitterBuffer::set-active:
805 * @buffer: the object which received the signal
807 * Start pushing out packets with the given base time. This signal is only
808 * useful in buffering mode.
810 * Returns: the time of the last pushed packet.
812 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
813 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
814 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
815 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
816 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
819 gstelement_class->change_state =
820 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
821 gstelement_class->request_new_pad =
822 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
823 gstelement_class->release_pad =
824 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
825 gstelement_class->provide_clock =
826 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
827 gstelement_class->set_clock =
828 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_clock);
830 gst_element_class_add_pad_template (gstelement_class,
831 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
832 gst_element_class_add_pad_template (gstelement_class,
833 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
834 gst_element_class_add_pad_template (gstelement_class,
835 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
837 gst_element_class_set_static_metadata (gstelement_class,
838 "RTP packet jitter-buffer", "Filter/Network/RTP",
839 "A buffer that deals with network jitter and other transmission faults",
840 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
841 "Wim Taymans <wim.taymans@gmail.com>");
843 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
844 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
846 GST_DEBUG_CATEGORY_INIT
847 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
851 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
853 GstRtpJitterBufferPrivate *priv;
855 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
856 jitterbuffer->priv = priv;
858 priv->latency_ms = DEFAULT_LATENCY_MS;
859 priv->latency_ns = priv->latency_ms * GST_MSECOND;
860 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
861 priv->do_lost = DEFAULT_DO_LOST;
862 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
863 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
864 priv->rtx_delay = DEFAULT_RTX_DELAY;
865 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
866 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
867 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
868 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
869 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
870 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
871 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
872 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
873 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
876 priv->last_rtptime = -1;
877 priv->avg_jitter = 0;
878 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
879 priv->jbuf = rtp_jitter_buffer_new ();
880 g_mutex_init (&priv->jbuf_lock);
881 g_cond_init (&priv->jbuf_timer);
882 g_cond_init (&priv->jbuf_event);
883 g_cond_init (&priv->jbuf_query);
884 g_queue_init (&priv->gap_packets);
886 /* reset skew detection initialy */
887 rtp_jitter_buffer_reset_skew (priv->jbuf);
888 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
889 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
893 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
896 gst_pad_set_activatemode_function (priv->srcpad,
897 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
898 gst_pad_set_query_function (priv->srcpad,
899 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
900 gst_pad_set_event_function (priv->srcpad,
901 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
904 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
907 gst_pad_set_chain_function (priv->sinkpad,
908 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
909 gst_pad_set_event_function (priv->sinkpad,
910 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
911 gst_pad_set_query_function (priv->sinkpad,
912 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
914 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
915 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
917 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
920 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
922 #define ITEM_TYPE_BUFFER 0
923 #define ITEM_TYPE_LOST 1
924 #define ITEM_TYPE_EVENT 2
925 #define ITEM_TYPE_QUERY 3
927 static RTPJitterBufferItem *
928 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
929 guint seqnum, guint count, guint rtptime)
931 RTPJitterBufferItem *item;
933 item = g_slice_new (RTPJitterBufferItem);
940 item->seqnum = seqnum;
942 item->rtptime = rtptime;
948 free_item (RTPJitterBufferItem * item)
950 g_return_if_fail (item != NULL);
952 if (item->data && item->type != ITEM_TYPE_QUERY)
953 gst_mini_object_unref (item->data);
954 g_slice_free (RTPJitterBufferItem, item);
958 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
960 GList **l = user_data;
962 if (item->data && item->type == ITEM_TYPE_EVENT
963 && GST_EVENT_IS_STICKY (item->data)) {
964 *l = g_list_prepend (*l, item->data);
965 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
966 gst_mini_object_unref (item->data);
968 g_slice_free (RTPJitterBufferItem, item);
972 gst_rtp_jitter_buffer_finalize (GObject * object)
974 GstRtpJitterBuffer *jitterbuffer;
975 GstRtpJitterBufferPrivate *priv;
977 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
978 priv = jitterbuffer->priv;
980 g_array_free (priv->timers, TRUE);
981 g_mutex_clear (&priv->jbuf_lock);
982 g_cond_clear (&priv->jbuf_timer);
983 g_cond_clear (&priv->jbuf_event);
984 g_cond_clear (&priv->jbuf_query);
986 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
987 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
988 g_queue_clear (&priv->gap_packets);
989 g_object_unref (priv->jbuf);
991 G_OBJECT_CLASS (parent_class)->finalize (object);
995 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
997 GstRtpJitterBuffer *jitterbuffer;
998 GstPad *otherpad = NULL;
999 GstIterator *it = NULL;
1000 GValue val = { 0, };
1002 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1004 if (pad == jitterbuffer->priv->sinkpad) {
1005 otherpad = jitterbuffer->priv->srcpad;
1006 } else if (pad == jitterbuffer->priv->srcpad) {
1007 otherpad = jitterbuffer->priv->sinkpad;
1008 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1009 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1013 g_value_init (&val, GST_TYPE_PAD);
1014 g_value_set_object (&val, otherpad);
1015 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1016 g_value_unset (&val);
1023 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1025 GstRtpJitterBufferPrivate *priv;
1027 priv = jitterbuffer->priv;
1029 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1032 gst_pad_new_from_static_template
1033 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1034 gst_pad_set_chain_function (priv->rtcpsinkpad,
1035 gst_rtp_jitter_buffer_chain_rtcp);
1036 gst_pad_set_event_function (priv->rtcpsinkpad,
1037 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1038 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1039 gst_rtp_jitter_buffer_iterate_internal_links);
1040 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1041 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1043 return priv->rtcpsinkpad;
1047 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1049 GstRtpJitterBufferPrivate *priv;
1051 priv = jitterbuffer->priv;
1053 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1055 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1057 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1058 priv->rtcpsinkpad = NULL;
1062 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1063 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1065 GstRtpJitterBuffer *jitterbuffer;
1066 GstElementClass *klass;
1068 GstRtpJitterBufferPrivate *priv;
1070 g_return_val_if_fail (templ != NULL, NULL);
1071 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1073 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1074 priv = jitterbuffer->priv;
1075 klass = GST_ELEMENT_GET_CLASS (element);
1077 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1079 /* figure out the template */
1080 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1081 if (priv->rtcpsinkpad != NULL)
1084 result = create_rtcp_sink (jitterbuffer);
1086 goto wrong_template;
1093 g_warning ("rtpjitterbuffer: this is not our template");
1098 g_warning ("rtpjitterbuffer: pad already requested");
1104 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1106 GstRtpJitterBuffer *jitterbuffer;
1107 GstRtpJitterBufferPrivate *priv;
1109 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1110 g_return_if_fail (GST_IS_PAD (pad));
1112 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1113 priv = jitterbuffer->priv;
1115 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1117 if (priv->rtcpsinkpad == pad) {
1118 remove_rtcp_sink (jitterbuffer);
1127 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1133 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1135 return gst_system_clock_obtain ();
1139 gst_rtp_jitter_buffer_set_clock (GstElement * element, GstClock * clock)
1141 GstRtpJitterBuffer *jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1143 rtp_jitter_buffer_set_pipeline_clock (jitterbuffer->priv->jbuf, clock);
1149 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1151 GstRtpJitterBufferPrivate *priv;
1153 priv = jitterbuffer->priv;
1155 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1158 priv->clock_rate = -1;
1159 /* do not clear current content, but refresh state for new arrival */
1160 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1161 rtp_jitter_buffer_reset_skew (priv->jbuf);
1166 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1169 GstRtpJitterBufferPrivate *priv;
1170 GstClockTime last_out;
1171 RTPJitterBufferItem *item;
1176 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1177 active, GST_TIME_ARGS (offset));
1179 if (active != priv->active) {
1180 /* add the amount of time spent in paused to the output offset. All
1181 * outgoing buffers will have this offset applied to their timestamps in
1182 * order to make them arrive in time in the sink. */
1183 priv->out_offset = offset;
1184 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1185 GST_TIME_ARGS (priv->out_offset));
1186 priv->active = active;
1187 JBUF_SIGNAL_EVENT (priv);
1190 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1192 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1193 /* head buffer timestamp and offset gives our output time */
1194 last_out = item->dts + priv->ts_offset;
1196 /* use last known time when the buffer is empty */
1197 last_out = priv->last_out_time;
1205 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1207 GstRtpJitterBuffer *jitterbuffer;
1208 GstRtpJitterBufferPrivate *priv;
1213 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1214 priv = jitterbuffer->priv;
1216 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1218 caps = gst_pad_peer_query_caps (other, filter);
1220 templ = gst_pad_get_pad_template_caps (pad);
1222 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1227 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1229 intersect = gst_caps_intersect (caps, templ);
1230 gst_caps_unref (caps);
1231 gst_caps_unref (templ);
1235 gst_object_unref (jitterbuffer);
1241 * Must be called with JBUF_LOCK held
1245 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1248 GstRtpJitterBufferPrivate *priv;
1249 GstStructure *caps_struct;
1252 const gchar *ts_refclk, *mediaclk;
1254 priv = jitterbuffer->priv;
1256 /* first parse the caps */
1257 caps_struct = gst_caps_get_structure (caps, 0);
1259 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1261 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1262 * measure the amount of data in the buffer */
1263 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1266 if (priv->clock_rate <= 0)
1269 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1271 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1273 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1274 * can use this to track the amount of time elapsed on the sender. */
1275 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1276 priv->clock_base = val;
1278 priv->clock_base = -1;
1280 priv->ext_timestamp = priv->clock_base;
1282 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1285 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1286 /* first expected seqnum, only update when we didn't have a previous base. */
1287 if (priv->next_in_seqnum == -1)
1288 priv->next_in_seqnum = val;
1289 if (priv->next_seqnum == -1) {
1290 priv->next_seqnum = val;
1291 JBUF_SIGNAL_EVENT (priv);
1293 priv->seqnum_base = val;
1295 priv->seqnum_base = -1;
1298 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1300 /* the start and stop times. The seqnum-base corresponds to the start time. We
1301 * will keep track of the seqnums on the output and when we reach the one
1302 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1303 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1304 priv->npt_start = tval;
1306 priv->npt_start = 0;
1308 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1309 priv->npt_stop = tval;
1311 priv->npt_stop = -1;
1313 GST_DEBUG_OBJECT (jitterbuffer,
1314 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1315 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1317 if ((ts_refclk = gst_structure_get_string (caps_struct, "a-ts-refclk"))) {
1318 GstClock *clock = NULL;
1319 guint64 clock_offset = -1;
1321 GST_DEBUG_OBJECT (jitterbuffer, "Have timestamp reference clock %s",
1324 if (g_str_has_prefix (ts_refclk, "ntp=")) {
1325 if (g_str_has_prefix (ts_refclk, "ntp=/traceable/")) {
1326 GST_FIXME_OBJECT (jitterbuffer, "Can't handle traceable NTP clocks");
1328 const gchar *host, *portstr;
1332 host = ts_refclk + sizeof ("ntp=") - 1;
1333 if (host[0] == '[') {
1335 portstr = strchr (host, ']');
1336 if (portstr && portstr[1] == ':')
1337 portstr = portstr + 1;
1341 portstr = strrchr (host, ':');
1345 if (!portstr || sscanf (portstr, ":%u", &port) != 1)
1349 hostname = g_strndup (host, (portstr - host));
1351 hostname = g_strdup (host);
1353 clock = gst_ntp_clock_new (NULL, hostname, port, 0);
1356 } else if (g_str_has_prefix (ts_refclk, "ptp=IEEE1588-2008:")) {
1357 const gchar *domainstr =
1358 ts_refclk + sizeof ("ptp=IEEE1588-2008:XX-XX-XX-XX-XX-XX-XX-XX") - 1;
1361 if (domainstr[0] != ':' || sscanf (domainstr, ":%u", &domain) != 1)
1364 clock = gst_ptp_clock_new (NULL, domain);
1366 GST_FIXME_OBJECT (jitterbuffer, "Unsupported timestamp reference clock");
1369 if ((mediaclk = gst_structure_get_string (caps_struct, "a-mediaclk"))) {
1370 GST_DEBUG_OBJECT (jitterbuffer, "Got media clock %s", mediaclk);
1372 if (!g_str_has_prefix (mediaclk, "direct=")
1373 || sscanf (mediaclk, "direct=%" G_GUINT64_FORMAT, &clock_offset) != 1)
1374 GST_FIXME_OBJECT (jitterbuffer, "Unsupported media clock");
1375 if (strstr (mediaclk, "rate=") != NULL) {
1376 GST_FIXME_OBJECT (jitterbuffer, "Rate property not supported");
1381 rtp_jitter_buffer_set_media_clock (priv->jbuf, clock, clock_offset);
1383 rtp_jitter_buffer_set_media_clock (priv->jbuf, NULL, -1);
1391 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1396 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1402 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1404 GstRtpJitterBufferPrivate *priv;
1406 priv = jitterbuffer->priv;
1409 /* mark ourselves as flushing */
1410 priv->srcresult = GST_FLOW_FLUSHING;
1411 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1412 /* this unblocks any waiting pops on the src pad task */
1413 JBUF_SIGNAL_EVENT (priv);
1414 JBUF_SIGNAL_QUERY (priv, FALSE);
1419 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1421 GstRtpJitterBufferPrivate *priv;
1423 priv = jitterbuffer->priv;
1426 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1427 /* Mark as non flushing */
1428 priv->srcresult = GST_FLOW_OK;
1429 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1430 priv->last_popped_seqnum = -1;
1431 priv->last_out_time = -1;
1432 priv->next_seqnum = -1;
1433 priv->seqnum_base = -1;
1434 priv->ips_rtptime = -1;
1435 priv->ips_dts = GST_CLOCK_TIME_NONE;
1436 priv->packet_spacing = 0;
1437 priv->next_in_seqnum = -1;
1438 priv->clock_rate = -1;
1441 priv->estimated_eos = -1;
1442 priv->last_elapsed = 0;
1443 priv->ext_timestamp = -1;
1444 priv->avg_jitter = 0;
1445 priv->last_dts = -1;
1446 priv->last_rtptime = -1;
1447 priv->last_in_dts = 0;
1448 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1449 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1450 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1451 rtp_jitter_buffer_reset_skew (priv->jbuf);
1452 remove_all_timers (jitterbuffer);
1453 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1454 g_queue_clear (&priv->gap_packets);
1459 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1460 GstPadMode mode, gboolean active)
1463 GstRtpJitterBuffer *jitterbuffer = NULL;
1465 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1468 case GST_PAD_MODE_PUSH:
1470 /* allow data processing */
1471 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1473 /* start pushing out buffers */
1474 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1475 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1476 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1478 /* make sure all data processing stops ASAP */
1479 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1481 /* NOTE this will hardlock if the state change is called from the src pad
1482 * task thread because we will _join() the thread. */
1483 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1484 result = gst_pad_stop_task (pad);
1494 static GstStateChangeReturn
1495 gst_rtp_jitter_buffer_change_state (GstElement * element,
1496 GstStateChange transition)
1498 GstRtpJitterBuffer *jitterbuffer;
1499 GstRtpJitterBufferPrivate *priv;
1500 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1502 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1503 priv = jitterbuffer->priv;
1505 switch (transition) {
1506 case GST_STATE_CHANGE_NULL_TO_READY:
1508 case GST_STATE_CHANGE_READY_TO_PAUSED:
1510 /* reset negotiated values */
1511 priv->clock_rate = -1;
1512 priv->clock_base = -1;
1513 priv->peer_latency = 0;
1515 /* block until we go to PLAYING */
1516 priv->blocked = TRUE;
1517 priv->timer_running = TRUE;
1518 priv->timer_thread =
1519 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1522 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1524 /* unblock to allow streaming in PLAYING */
1525 priv->blocked = FALSE;
1526 JBUF_SIGNAL_EVENT (priv);
1527 JBUF_SIGNAL_TIMER (priv);
1534 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1536 switch (transition) {
1537 case GST_STATE_CHANGE_READY_TO_PAUSED:
1538 /* we are a live element because we sync to the clock, which we can only
1539 * do in the PLAYING state */
1540 if (ret != GST_STATE_CHANGE_FAILURE)
1541 ret = GST_STATE_CHANGE_NO_PREROLL;
1543 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1545 /* block to stop streaming when PAUSED */
1546 priv->blocked = TRUE;
1547 unschedule_current_timer (jitterbuffer);
1549 if (ret != GST_STATE_CHANGE_FAILURE)
1550 ret = GST_STATE_CHANGE_NO_PREROLL;
1552 case GST_STATE_CHANGE_PAUSED_TO_READY:
1554 gst_buffer_replace (&priv->last_sr, NULL);
1555 priv->timer_running = FALSE;
1556 unschedule_current_timer (jitterbuffer);
1557 JBUF_SIGNAL_TIMER (priv);
1558 JBUF_SIGNAL_QUERY (priv, FALSE);
1560 g_thread_join (priv->timer_thread);
1561 priv->timer_thread = NULL;
1563 case GST_STATE_CHANGE_READY_TO_NULL:
1573 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1576 gboolean ret = TRUE;
1577 GstRtpJitterBuffer *jitterbuffer;
1578 GstRtpJitterBufferPrivate *priv;
1580 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1581 priv = jitterbuffer->priv;
1583 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1585 switch (GST_EVENT_TYPE (event)) {
1586 case GST_EVENT_LATENCY:
1588 GstClockTime latency;
1590 gst_event_parse_latency (event, &latency);
1592 GST_DEBUG_OBJECT (jitterbuffer,
1593 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1596 /* adjust the overall buffer delay to the total pipeline latency in
1597 * buffering mode because if downstream consumes too fast (because of
1598 * large latency or queues, we would start rebuffering again. */
1599 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1600 RTP_JITTER_BUFFER_MODE_BUFFER) {
1601 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1605 ret = gst_pad_push_event (priv->sinkpad, event);
1609 ret = gst_pad_push_event (priv->sinkpad, event);
1616 /* handles and stores the event in the jitterbuffer, must be called with
1619 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1621 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1622 RTPJitterBufferItem *item;
1625 switch (GST_EVENT_TYPE (event)) {
1626 case GST_EVENT_CAPS:
1630 gst_event_parse_caps (event, &caps);
1631 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1634 case GST_EVENT_SEGMENT:
1635 gst_event_copy_segment (event, &priv->segment);
1637 /* we need time for now */
1638 if (priv->segment.format != GST_FORMAT_TIME)
1639 goto newseg_wrong_format;
1641 GST_DEBUG_OBJECT (jitterbuffer,
1642 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1646 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1653 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1654 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1655 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1657 JBUF_SIGNAL_EVENT (priv);
1662 newseg_wrong_format:
1664 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1665 gst_event_unref (event);
1671 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1674 gboolean ret = TRUE;
1675 GstRtpJitterBuffer *jitterbuffer;
1676 GstRtpJitterBufferPrivate *priv;
1678 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1679 priv = jitterbuffer->priv;
1681 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1683 switch (GST_EVENT_TYPE (event)) {
1684 case GST_EVENT_FLUSH_START:
1685 ret = gst_pad_push_event (priv->srcpad, event);
1686 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1687 /* wait for the loop to go into PAUSED */
1688 gst_pad_pause_task (priv->srcpad);
1690 case GST_EVENT_FLUSH_STOP:
1691 ret = gst_pad_push_event (priv->srcpad, event);
1693 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1694 GST_PAD_MODE_PUSH, TRUE);
1697 if (GST_EVENT_IS_SERIALIZED (event)) {
1698 /* serialized events go in the queue */
1700 if (priv->srcresult != GST_FLOW_OK) {
1701 /* Errors in sticky event pushing are no problem and ignored here
1702 * as they will cause more meaningful errors during data flow.
1703 * For EOS events, that are not followed by data flow, we still
1704 * return FALSE here though.
1706 if (!GST_EVENT_IS_STICKY (event) ||
1707 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1708 goto out_flow_error;
1710 /* refuse more events on EOS */
1713 ret = queue_event (jitterbuffer, event);
1716 /* non-serialized events are forwarded downstream immediately */
1717 ret = gst_pad_push_event (priv->srcpad, event);
1726 GST_DEBUG_OBJECT (jitterbuffer,
1727 "refusing event, we have a downstream flow error: %s",
1728 gst_flow_get_name (priv->srcresult));
1730 gst_event_unref (event);
1735 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1737 gst_event_unref (event);
1743 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1746 gboolean ret = TRUE;
1747 GstRtpJitterBuffer *jitterbuffer;
1749 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1751 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1753 switch (GST_EVENT_TYPE (event)) {
1754 case GST_EVENT_FLUSH_START:
1755 gst_event_unref (event);
1757 case GST_EVENT_FLUSH_STOP:
1758 gst_event_unref (event);
1761 ret = gst_pad_event_default (pad, parent, event);
1769 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1770 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1771 * GST_FLOW_FLUSHING when the element is shutting down. On success
1772 * GST_FLOW_OK is returned.
1774 static GstFlowReturn
1775 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1779 GValue args[2] = { {0}, {0} };
1783 g_value_init (&args[0], GST_TYPE_ELEMENT);
1784 g_value_set_object (&args[0], jitterbuffer);
1785 g_value_init (&args[1], G_TYPE_UINT);
1786 g_value_set_uint (&args[1], pt);
1788 g_value_init (&ret, GST_TYPE_CAPS);
1789 g_value_set_boxed (&ret, NULL);
1791 JBUF_UNLOCK (jitterbuffer->priv);
1792 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1794 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1796 g_value_unset (&args[0]);
1797 g_value_unset (&args[1]);
1798 caps = (GstCaps *) g_value_dup_boxed (&ret);
1799 g_value_unset (&ret);
1803 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1804 gst_caps_unref (caps);
1806 if (G_UNLIKELY (!res))
1814 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1815 return GST_FLOW_ERROR;
1819 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1820 return GST_FLOW_FLUSHING;
1824 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1825 return GST_FLOW_ERROR;
1829 /* call with jbuf lock held */
1831 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1833 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1834 GstMessage *message = NULL;
1839 /* Post a buffering message */
1840 if (priv->last_percent != percent) {
1841 priv->last_percent = percent;
1843 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1844 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1851 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1853 GstRtpJitterBufferPrivate *priv;
1855 priv = jitterbuffer->priv;
1857 if (timestamp == -1)
1860 /* apply the timestamp offset, this is used for inter stream sync */
1861 timestamp += priv->ts_offset;
1862 /* add the offset, this is used when buffering */
1863 timestamp += priv->out_offset;
1869 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1871 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1872 TimerData *timer = NULL;
1875 len = priv->timers->len;
1876 for (i = 0; i < len; i++) {
1877 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1878 if (test->seqnum == seqnum && test->type == type) {
1887 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1889 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1891 if (priv->clock_id) {
1892 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1893 gst_clock_id_unschedule (priv->clock_id);
1894 priv->clock_id = NULL;
1899 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1901 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1902 GstClockTime test_timeout;
1904 if ((test_timeout = timer->timeout) == -1)
1907 if (timer->type != TIMER_TYPE_EXPECTED) {
1908 /* add our latency and offset to get output times. */
1909 test_timeout = apply_offset (jitterbuffer, test_timeout);
1910 test_timeout += priv->latency_ns;
1912 return test_timeout;
1916 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1918 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1920 if (priv->clock_id) {
1921 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1923 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1924 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1926 if (timeout == -1 || timeout < priv->timer_timeout)
1927 unschedule_current_timer (jitterbuffer);
1932 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1933 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1934 GstClockTime duration)
1936 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1940 GST_DEBUG_OBJECT (jitterbuffer,
1941 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1942 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1943 GST_TIME_ARGS (delay));
1945 len = priv->timers->len;
1946 g_array_set_size (priv->timers, len + 1);
1947 timer = &g_array_index (priv->timers, TimerData, len);
1950 timer->seqnum = seqnum;
1952 timer->timeout = timeout + delay;
1953 timer->duration = duration;
1954 if (type == TIMER_TYPE_EXPECTED) {
1955 timer->rtx_base = timeout;
1956 timer->rtx_delay = delay;
1957 timer->rtx_retry = 0;
1959 timer->num_rtx_retry = 0;
1960 recalculate_timer (jitterbuffer, timer);
1961 JBUF_SIGNAL_TIMER (priv);
1967 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1968 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1970 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1971 gboolean seqchange, timechange;
1974 seqchange = timer->seqnum != seqnum;
1975 timechange = timer->timeout != timeout;
1977 if (!seqchange && !timechange)
1980 oldseq = timer->seqnum;
1982 GST_DEBUG_OBJECT (jitterbuffer,
1983 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1984 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1986 timer->timeout = timeout + delay;
1987 timer->seqnum = seqnum;
1989 timer->rtx_base = timeout;
1990 timer->rtx_delay = delay;
1991 timer->rtx_retry = 0;
1994 timer->num_rtx_retry = 0;
1996 if (priv->clock_id) {
1997 /* we changed the seqnum and there is a timer currently waiting with this
1998 * seqnum, unschedule it */
1999 if (seqchange && priv->timer_seqnum == oldseq)
2000 unschedule_current_timer (jitterbuffer);
2001 /* we changed the time, check if it is earlier than what we are waiting
2002 * for and unschedule if so */
2003 else if (timechange)
2004 recalculate_timer (jitterbuffer, timer);
2009 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
2010 guint16 seqnum, GstClockTime timeout)
2014 /* find the seqnum timer */
2015 timer = find_timer (jitterbuffer, type, seqnum);
2016 if (timer == NULL) {
2017 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
2019 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
2025 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
2027 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2030 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
2031 unschedule_current_timer (jitterbuffer);
2034 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
2035 g_array_remove_index_fast (priv->timers, idx);
2040 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
2042 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2043 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
2044 g_array_set_size (priv->timers, 0);
2045 unschedule_current_timer (jitterbuffer);
2048 /* get the extra delay to wait before sending RTX */
2050 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
2054 if (priv->rtx_delay == -1) {
2055 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
2056 delay = DEFAULT_AUTO_RTX_DELAY;
2058 /* jitter is in nanoseconds, maximum of 2x jitter and half the
2059 * packet spacing is a good margin */
2060 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
2063 delay = priv->rtx_delay * GST_MSECOND;
2065 if (priv->rtx_min_delay > 0)
2066 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
2071 /* we just received a packet with seqnum and dts.
2073 * First check for old seqnum that we are still expecting. If the gap with the
2074 * current seqnum is too big, unschedule the timeouts.
2076 * If we have a valid packet spacing estimate we can set a timer for when we
2077 * should receive the next packet.
2078 * If we don't have a valid estimate, we remove any timer we might have
2079 * had for this packet.
2082 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
2083 GstClockTime dts, gboolean do_next_seqnum)
2085 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2086 TimerData *timer = NULL;
2089 /* go through all timers and unschedule the ones with a large gap, also find
2090 * the timer for the seqnum */
2091 len = priv->timers->len;
2092 for (i = 0; i < len; i++) {
2093 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2096 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2098 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2099 test->type, test->seqnum, seqnum, gap);
2102 GST_DEBUG ("found timer for current seqnum");
2103 /* the timer for the current seqnum */
2105 /* when no retransmission, we can stop now, we only need to find the
2106 * timer for the current seqnum */
2107 if (!priv->do_retransmission)
2109 } else if (gap > priv->rtx_delay_reorder) {
2110 /* max gap, we exceeded the max reorder distance and we don't expect the
2111 * missing packet to be this reordered */
2112 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2113 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2117 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2118 && priv->do_retransmission && priv->rtx_next_seqnum;
2120 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2121 if (timer->num_rtx_retry > 0) {
2122 GstClockTime rtx_last, delay;
2124 /* we scheduled a retry for this packet and now we have it */
2125 priv->num_rtx_success++;
2126 /* all the previous retry attempts failed */
2127 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2128 /* number of retries before receiving the packet */
2129 if (priv->avg_rtx_num == 0.0)
2130 priv->avg_rtx_num = timer->num_rtx_retry;
2132 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2133 /* calculate the delay between retransmission request and receiving this
2134 * packet, start with when we scheduled this timeout last */
2135 rtx_last = timer->rtx_last;
2136 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2137 /* we have a valid delay if this packet arrived after we scheduled the
2139 delay = dts - rtx_last;
2140 if (priv->avg_rtx_rtt == 0)
2141 priv->avg_rtx_rtt = delay;
2143 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2147 GST_LOG_OBJECT (jitterbuffer,
2148 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2149 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2150 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2151 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2152 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2153 GST_TIME_ARGS (priv->avg_rtx_rtt));
2155 /* don't try to estimate the next seqnum because this is a retransmitted
2156 * packet and it probably did not arrive with the expected packet
2158 do_next_seqnum = FALSE;
2162 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2163 GstClockTime expected, delay;
2165 /* calculate expected arrival time of the next seqnum */
2166 expected = dts + priv->packet_spacing;
2168 delay = get_rtx_delay (priv);
2170 /* and update/install timer for next seqnum */
2172 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2175 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2176 expected, delay, priv->packet_spacing);
2178 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2179 /* if we had a timer, remove it, we don't know when to expect the next
2181 remove_timer (jitterbuffer, timer);
2186 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2189 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2191 /* we need consecutive seqnums with a different
2192 * rtptime to estimate the packet spacing. */
2193 if (priv->ips_rtptime != rtptime) {
2194 /* rtptime changed, check dts diff */
2195 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2196 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2197 GstClockTime old_packet_spacing = priv->packet_spacing;
2199 /* Biased towards bigger packet spacings to prevent
2200 * too many unneeded retransmission requests for next
2201 * packets that just arrive a little later than we would
2203 if (old_packet_spacing > new_packet_spacing)
2204 priv->packet_spacing =
2205 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2206 else if (old_packet_spacing > 0)
2207 priv->packet_spacing =
2208 (3 * new_packet_spacing + old_packet_spacing) / 4;
2210 priv->packet_spacing = new_packet_spacing;
2212 GST_DEBUG_OBJECT (jitterbuffer,
2213 "new packet spacing %" GST_TIME_FORMAT
2214 " old packet spacing %" GST_TIME_FORMAT
2215 " combined to %" GST_TIME_FORMAT,
2216 GST_TIME_ARGS (new_packet_spacing),
2217 GST_TIME_ARGS (old_packet_spacing),
2218 GST_TIME_ARGS (priv->packet_spacing));
2220 priv->ips_rtptime = rtptime;
2221 priv->ips_dts = dts;
2226 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2227 guint16 seqnum, GstClockTime dts, gint gap)
2229 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2230 GstClockTime total_duration, duration, expected_dts;
2233 GST_DEBUG_OBJECT (jitterbuffer,
2234 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2235 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2237 if (dts == GST_CLOCK_TIME_NONE) {
2238 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2242 /* the total duration spanned by the missing packets */
2243 if (dts >= priv->last_in_dts)
2244 total_duration = dts - priv->last_in_dts;
2248 /* interpolate between the current time and the last time based on
2249 * number of packets we are missing, this is the estimated duration
2250 * for the missing packet based on equidistant packet spacing. */
2251 duration = total_duration / (gap + 1);
2253 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2254 GST_TIME_ARGS (duration));
2256 if (total_duration > priv->latency_ns) {
2257 GstClockTime gap_time;
2261 GstClockTime gap_dur = gap * duration;
2262 if (gap_dur > priv->latency_ns)
2263 gap_time = gap_dur - priv->latency_ns;
2266 lost_packets = gap_time / duration;
2268 gap_time = total_duration - priv->latency_ns;
2272 /* too many lost packets, some of the missing packets are already
2273 * too late and we can generate lost packet events for them. */
2274 GST_DEBUG_OBJECT (jitterbuffer,
2275 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2276 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2277 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2278 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2279 GST_TIME_ARGS (gap_time));
2281 /* this timer will fire immediately and the lost event will be pushed from
2282 * the timer thread */
2283 if (lost_packets > 0) {
2284 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2285 priv->last_in_dts + duration, 0, gap_time);
2286 expected += lost_packets;
2287 priv->last_in_dts += gap_time;
2291 expected_dts = priv->last_in_dts + duration;
2293 if (priv->do_retransmission) {
2296 type = TIMER_TYPE_EXPECTED;
2297 /* if we had a timer for the first missing packet, update it. */
2298 if ((timer = find_timer (jitterbuffer, type, expected))) {
2299 GstClockTime timeout = timer->timeout;
2301 timer->duration = duration;
2302 if (timeout > (expected_dts + timer->rtx_retry)) {
2303 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2304 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2308 expected_dts += duration;
2311 type = TIMER_TYPE_LOST;
2314 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2315 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2316 expected_dts += duration;
2322 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2326 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2327 GstRtpJitterBufferPrivate *priv;
2329 priv = jitterbuffer->priv;
2331 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2334 if (priv->last_dts != -1)
2335 dtsdiff = dts - priv->last_dts;
2339 if (priv->last_rtptime != -1)
2340 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2344 priv->last_dts = dts;
2345 priv->last_rtptime = rtptime;
2349 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2352 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2354 diff = ABS (dtsdiff - rtpdiffns);
2356 /* jitter is stored in nanoseconds */
2357 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2359 GST_LOG_OBJECT (jitterbuffer,
2360 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2361 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2362 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2363 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2370 GST_DEBUG_OBJECT (jitterbuffer,
2371 "no dts or no clock-rate, can't calculate jitter");
2377 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2379 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2380 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2383 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2384 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2385 gst_rtp_buffer_unmap (&rtp_a);
2387 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2388 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2389 gst_rtp_buffer_unmap (&rtp_b);
2391 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2395 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2396 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2399 GstRtpJitterBufferPrivate *priv;
2400 guint gap_packets_length;
2401 gboolean reset = FALSE;
2403 priv = jitterbuffer->priv;
2405 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2407 guint32 prev_gap_seq = -1;
2408 gboolean all_consecutive = TRUE;
2410 g_queue_insert_sorted (&priv->gap_packets, buffer,
2411 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2413 for (l = priv->gap_packets.head; l; l = l->next) {
2414 GstBuffer *gap_buffer = l->data;
2415 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2418 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2420 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2422 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2423 if (prev_gap_seq == -1)
2424 prev_gap_seq = gap_seq;
2425 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2426 all_consecutive = FALSE;
2428 prev_gap_seq = gap_seq;
2430 gst_rtp_buffer_unmap (&gap_rtp);
2431 if (!all_consecutive)
2435 if (all_consecutive && gap_packets_length > 3) {
2436 GST_DEBUG_OBJECT (jitterbuffer,
2437 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2438 (future ? "new" : "old"), gap,
2439 (future ? max_dropout : -max_misorder));
2441 } else if (!all_consecutive) {
2442 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2443 g_queue_clear (&priv->gap_packets);
2444 GST_DEBUG_OBJECT (jitterbuffer,
2445 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2446 (future ? "new" : "old"), gap,
2447 (future ? max_dropout : -max_misorder));
2450 GST_DEBUG_OBJECT (jitterbuffer,
2451 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2452 (future ? "new" : "old"), gap,
2453 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2457 GST_DEBUG_OBJECT (jitterbuffer,
2458 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2459 gap, -max_misorder);
2460 g_queue_push_tail (&priv->gap_packets, buffer);
2468 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2470 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2471 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2474 GstClockTime base_time =
2475 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2476 GstClockTime clock_time = gst_clock_get_time (clock);
2478 if (clock_time > base_time)
2479 running_time = clock_time - base_time;
2483 gst_object_unref (clock);
2486 return running_time;
2489 static GstFlowReturn
2490 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2493 GstRtpJitterBuffer *jitterbuffer;
2494 GstRtpJitterBufferPrivate *priv;
2496 guint32 expected, rtptime;
2497 GstFlowReturn ret = GST_FLOW_OK;
2498 GstClockTime dts, pts;
2503 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2504 gboolean do_next_seqnum = FALSE;
2505 RTPJitterBufferItem *item;
2506 GstMessage *msg = NULL;
2507 gboolean estimated_dts = FALSE;
2508 guint32 packet_rate, max_dropout, max_misorder;
2510 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2512 priv = jitterbuffer->priv;
2514 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2515 goto invalid_buffer;
2517 pt = gst_rtp_buffer_get_payload_type (&rtp);
2518 seqnum = gst_rtp_buffer_get_seq (&rtp);
2519 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2520 gst_rtp_buffer_unmap (&rtp);
2522 /* make sure we have PTS and DTS set */
2523 pts = GST_BUFFER_PTS (buffer);
2524 dts = GST_BUFFER_DTS (buffer);
2531 /* If we have no DTS here, i.e. no capture time, get one from the
2532 * clock now to have something to calculate with in the future. */
2533 dts = get_current_running_time (jitterbuffer);
2536 /* Remember that we estimated the DTS if we are running already
2537 * and this is not our first packet (or first packet after a reset).
2538 * If it's the first packet, we somehow must generate a timestamp for
2539 * everything, otherwise we can't calculate any times
2541 estimated_dts = (priv->next_in_seqnum != -1);
2543 /* take the DTS of the buffer. This is the time when the packet was
2544 * received and is used to calculate jitter and clock skew. We will adjust
2545 * this DTS with the smoothed value after processing it in the
2546 * jitterbuffer and assign it as the PTS. */
2547 /* bring to running time */
2548 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2551 GST_DEBUG_OBJECT (jitterbuffer,
2552 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2553 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2555 JBUF_LOCK_CHECK (priv, out_flushing);
2557 if (G_UNLIKELY (priv->last_pt != pt)) {
2560 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2564 /* reset clock-rate so that we get a new one */
2565 priv->clock_rate = -1;
2567 /* Try to get the clock-rate from the caps first if we can. If there are no
2568 * caps we must fire the signal to get the clock-rate. */
2569 if ((caps = gst_pad_get_current_caps (pad))) {
2570 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2571 gst_caps_unref (caps);
2575 if (G_UNLIKELY (priv->clock_rate == -1)) {
2576 /* no clock rate given on the caps, try to get one with the signal */
2577 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2578 pt) == GST_FLOW_FLUSHING)
2581 if (G_UNLIKELY (priv->clock_rate == -1))
2585 /* don't accept more data on EOS */
2586 if (G_UNLIKELY (priv->eos))
2589 calculate_jitter (jitterbuffer, dts, rtptime);
2591 if (priv->seqnum_base != -1) {
2594 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2597 GST_DEBUG_OBJECT (jitterbuffer,
2598 "packet seqnum #%d before seqnum-base #%d", seqnum,
2600 gst_buffer_unref (buffer);
2603 } else if (gap > 16384) {
2604 /* From now on don't compare against the seqnum base anymore as
2605 * at some point in the future we will wrap around and also that
2606 * much reordering is very unlikely */
2607 priv->seqnum_base = -1;
2611 expected = priv->next_in_seqnum;
2614 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2616 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2617 priv->max_dropout_time);
2619 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2620 priv->max_misorder_time);
2621 GST_TRACE_OBJECT (jitterbuffer,
2622 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2623 max_dropout, max_misorder);
2625 /* now check against our expected seqnum */
2626 if (G_LIKELY (expected != -1)) {
2629 /* now calculate gap */
2630 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2632 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2633 expected, seqnum, gap);
2635 if (G_LIKELY (gap == 0)) {
2636 /* packet is expected */
2637 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2638 do_next_seqnum = TRUE;
2640 gboolean reset = FALSE;
2643 /* we received an old packet */
2644 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2646 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2647 gap, max_dropout, max_misorder);
2650 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2653 /* new packet, we are missing some packets */
2654 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2655 /* If we have timers for more than RTP_MAX_DROPOUT packets
2656 * pending this means that we have a huge gap overall. We can
2657 * reset the jitterbuffer at this point because there's
2658 * just too much data missing to be able to do anything
2659 * sensible with the past data. Just try again from the
2661 GST_WARNING_OBJECT (jitterbuffer,
2662 "%d pending timers > %d - resetting", priv->timers->len,
2665 gst_buffer_unref (buffer);
2667 } else if (G_UNLIKELY (gap >= max_dropout)) {
2669 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2670 gap, max_dropout, max_misorder);
2673 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2674 /* fill in the gap with EXPECTED timers */
2675 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2677 do_next_seqnum = TRUE;
2680 if (G_UNLIKELY (reset)) {
2681 GList *events = NULL, *l;
2684 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2685 rtp_jitter_buffer_flush (priv->jbuf,
2686 (GFunc) free_item_and_retain_events, &events);
2687 rtp_jitter_buffer_reset_skew (priv->jbuf);
2688 remove_all_timers (jitterbuffer);
2689 priv->discont = TRUE;
2690 priv->last_popped_seqnum = -1;
2692 if (priv->gap_packets.head) {
2693 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2694 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2696 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2697 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2698 gst_rtp_buffer_unmap (&gap_rtp);
2700 priv->next_seqnum = seqnum;
2703 priv->last_in_dts = -1;
2704 priv->next_in_seqnum = -1;
2706 /* Insert all sticky events again in order, otherwise we would
2707 * potentially loose STREAM_START, CAPS or SEGMENT events
2709 events = g_list_reverse (events);
2710 for (l = events; l; l = l->next) {
2711 RTPJitterBufferItem *item;
2713 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2714 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2716 g_list_free (events);
2718 JBUF_SIGNAL_EVENT (priv);
2720 /* reset spacing estimation when gap */
2721 priv->ips_rtptime = -1;
2722 priv->ips_dts = GST_CLOCK_TIME_NONE;
2724 buffers = g_list_copy (priv->gap_packets.head);
2725 g_queue_clear (&priv->gap_packets);
2727 priv->ips_rtptime = -1;
2728 priv->ips_dts = GST_CLOCK_TIME_NONE;
2729 JBUF_UNLOCK (jitterbuffer->priv);
2731 for (l = buffers; l; l = l->next) {
2732 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2734 if (ret != GST_FLOW_OK)
2737 for (; l; l = l->next)
2738 gst_buffer_unref (l->data);
2739 g_list_free (buffers);
2743 /* reset spacing estimation when gap */
2744 priv->ips_rtptime = -1;
2745 priv->ips_dts = GST_CLOCK_TIME_NONE;
2748 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2750 /* we don't know what the next_in_seqnum should be, wait for the last
2751 * possible moment to push this buffer, maybe we get an earlier seqnum
2753 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2754 do_next_seqnum = TRUE;
2755 /* take rtptime and dts to calculate packet spacing */
2756 priv->ips_rtptime = rtptime;
2757 priv->ips_dts = dts;
2760 /* We had no huge gap, let's drop all the gap packets */
2761 if (buffer != NULL) {
2762 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2763 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2764 g_queue_clear (&priv->gap_packets);
2766 GST_DEBUG_OBJECT (jitterbuffer,
2767 "Had big gap, waiting for more consecutive packets");
2768 JBUF_UNLOCK (jitterbuffer->priv);
2772 if (do_next_seqnum) {
2773 priv->last_in_dts = dts;
2774 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2777 /* let's check if this buffer is too late, we can only accept packets with
2778 * bigger seqnum than the one we last pushed. */
2779 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2782 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2784 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2785 if (G_UNLIKELY (gap <= 0))
2789 /* let's drop oldest packet if the queue is already full and drop-on-latency
2790 * is set. We can only do this when there actually is a latency. When no
2791 * latency is set, we just pump it in the queue and let the other end push it
2792 * out as fast as possible. */
2793 if (priv->latency_ms && priv->drop_on_latency) {
2795 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2797 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2798 RTPJitterBufferItem *old_item;
2800 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2802 if (IS_DROPABLE (old_item)) {
2803 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2804 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2806 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2807 free_item (old_item);
2809 /* we might have removed some head buffers, signal the pushing thread to
2810 * see if it can push now */
2811 JBUF_SIGNAL_EVENT (priv);
2815 /* If we estimated the DTS, don't consider it in the clock skew calculations
2816 * later. The code above always sets dts to pts or the other way around if
2817 * any of those is valid in the buffer, so we know that if we estimated the
2818 * dts that both are unknown */
2821 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2822 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2824 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2826 /* now insert the packet into the queue in sorted order. This function returns
2827 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2828 * have a duplicate. */
2829 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2834 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2836 /* we had an unhandled SR, handle it now */
2838 do_handle_sync (jitterbuffer);
2840 if (G_UNLIKELY (head)) {
2841 /* signal addition of new buffer when the _loop is waiting. */
2842 if (G_LIKELY (priv->active))
2843 JBUF_SIGNAL_EVENT (priv);
2845 /* let's unschedule and unblock any waiting buffers. We only want to do this
2846 * when the head buffer changed */
2847 if (G_UNLIKELY (priv->clock_id)) {
2848 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2849 unschedule_current_timer (jitterbuffer);
2853 GST_DEBUG_OBJECT (jitterbuffer,
2854 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2855 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2857 msg = check_buffering_percent (jitterbuffer, percent);
2863 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2870 /* this is not fatal but should be filtered earlier */
2871 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2872 ("Received invalid RTP payload, dropping"));
2873 gst_buffer_unref (buffer);
2878 GST_WARNING_OBJECT (jitterbuffer,
2879 "No clock-rate in caps!, dropping buffer");
2880 gst_buffer_unref (buffer);
2885 ret = priv->srcresult;
2886 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2887 gst_buffer_unref (buffer);
2893 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2894 gst_buffer_unref (buffer);
2899 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2900 " popped, dropping", seqnum, priv->last_popped_seqnum);
2902 gst_buffer_unref (buffer);
2907 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2909 priv->num_duplicates++;
2916 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2918 guint64 ext_time, elapsed;
2920 GstRtpJitterBufferPrivate *priv;
2922 priv = jitterbuffer->priv;
2923 rtp_time = item->rtptime;
2925 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2926 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2928 ext_time = priv->ext_timestamp;
2929 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2930 if (ext_time < priv->ext_timestamp) {
2931 ext_time = priv->ext_timestamp;
2933 priv->ext_timestamp = ext_time;
2936 if (ext_time > priv->clock_base)
2937 elapsed = ext_time - priv->clock_base;
2941 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2946 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2947 RTPJitterBufferItem * item)
2949 guint64 total, elapsed, left, estimated;
2950 GstClockTime out_time;
2951 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2953 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2954 || priv->clock_base == -1 || priv->clock_rate <= 0)
2957 /* compute the elapsed time */
2958 elapsed = compute_elapsed (jitterbuffer, item);
2960 /* do nothing if elapsed time doesn't increment */
2961 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2964 priv->last_elapsed = elapsed;
2966 /* this is the total time we need to play */
2967 total = priv->npt_stop - priv->npt_start;
2968 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2969 GST_TIME_ARGS (total));
2971 /* this is how much time there is left */
2972 if (total > elapsed)
2973 left = total - elapsed;
2977 /* if we have less time left that the size of the buffer, we will not
2978 * be able to keep it filled, disabled buffering then */
2979 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2980 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2981 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2982 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2985 /* this is the current time as running-time */
2986 out_time = item->dts;
2989 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2991 /* if there is almost nothing left,
2992 * we may never advance enough to end up in the above case */
2993 if (total < GST_SECOND)
2994 estimated = GST_SECOND;
2998 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2999 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
3001 if (estimated != -1 && priv->estimated_eos != estimated) {
3002 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
3003 priv->estimated_eos = estimated;
3007 /* take a buffer from the queue and push it */
3008 static GstFlowReturn
3009 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
3011 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3012 GstFlowReturn result = GST_FLOW_OK;
3013 RTPJitterBufferItem *item;
3014 GstBuffer *outbuf = NULL;
3015 GstEvent *outevent = NULL;
3016 GstQuery *outquery = NULL;
3017 GstClockTime dts, pts;
3019 gboolean do_push = TRUE;
3023 /* when we get here we are ready to pop and push the buffer */
3024 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
3028 case ITEM_TYPE_BUFFER:
3030 /* we need to make writable to change the flags and timestamps */
3031 outbuf = gst_buffer_make_writable (item->data);
3033 if (G_UNLIKELY (priv->discont)) {
3034 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
3035 * into the jitterbuffer so we can modify now. */
3036 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
3037 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
3038 priv->discont = FALSE;
3040 if (G_UNLIKELY (priv->ts_discont)) {
3041 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
3042 priv->ts_discont = FALSE;
3046 gst_segment_position_from_running_time (&priv->segment,
3047 GST_FORMAT_TIME, item->dts);
3049 gst_segment_position_from_running_time (&priv->segment,
3050 GST_FORMAT_TIME, item->pts);
3052 /* apply timestamp with offset to buffer now */
3053 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
3054 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
3056 /* update the elapsed time when we need to check against the npt stop time. */
3057 update_estimated_eos (jitterbuffer, item);
3059 priv->last_out_time = GST_BUFFER_PTS (outbuf);
3061 case ITEM_TYPE_LOST:
3062 priv->discont = TRUE;
3066 case ITEM_TYPE_EVENT:
3067 outevent = item->data;
3069 case ITEM_TYPE_QUERY:
3070 outquery = item->data;
3074 /* now we are ready to push the buffer. Save the seqnum and release the lock
3075 * so the other end can push stuff in the queue again. */
3077 priv->last_popped_seqnum = seqnum;
3078 priv->next_seqnum = (seqnum + item->count) & 0xffff;
3080 msg = check_buffering_percent (jitterbuffer, percent);
3087 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3090 case ITEM_TYPE_BUFFER:
3092 GST_DEBUG_OBJECT (jitterbuffer,
3093 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3094 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3095 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3096 result = gst_pad_push (priv->srcpad, outbuf);
3098 JBUF_LOCK_CHECK (priv, out_flushing);
3100 case ITEM_TYPE_LOST:
3101 case ITEM_TYPE_EVENT:
3102 /* We got not enough consecutive packets with a huge gap, we can
3103 * as well just drop them here now on EOS */
3104 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3105 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3106 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3107 g_queue_clear (&priv->gap_packets);
3110 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3111 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3114 gst_pad_push_event (priv->srcpad, outevent);
3116 gst_event_unref (outevent);
3118 result = GST_FLOW_OK;
3120 JBUF_LOCK_CHECK (priv, out_flushing);
3122 case ITEM_TYPE_QUERY:
3126 res = gst_pad_peer_query (priv->srcpad, outquery);
3128 JBUF_LOCK_CHECK (priv, out_flushing);
3129 result = GST_FLOW_OK;
3130 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3131 JBUF_SIGNAL_QUERY (priv, res);
3140 return priv->srcresult;
3144 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3146 /* Peek a buffer and compare the seqnum to the expected seqnum.
3147 * If all is fine, the buffer is pushed.
3148 * If something is wrong, we wait for some event
3150 static GstFlowReturn
3151 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3153 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3154 GstFlowReturn result;
3155 RTPJitterBufferItem *item;
3157 guint32 next_seqnum;
3159 /* only push buffers when PLAYING and active and not buffering */
3160 if (priv->blocked || !priv->active ||
3161 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3162 return GST_FLOW_WAIT;
3165 /* peek a buffer, we're just looking at the sequence number.
3166 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3167 * wait for a timeout or something to change.
3168 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3169 item = rtp_jitter_buffer_peek (priv->jbuf);
3174 /* get the seqnum and the next expected seqnum */
3175 seqnum = item->seqnum;
3177 return pop_and_push_next (jitterbuffer, seqnum);
3180 next_seqnum = priv->next_seqnum;
3182 /* get the gap between this and the previous packet. If we don't know the
3183 * previous packet seqnum assume no gap. */
3184 if (G_UNLIKELY (next_seqnum == -1)) {
3185 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3186 /* we don't know what the next_seqnum should be, the chain function should
3187 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3188 * fires, so wait for that */
3189 result = GST_FLOW_WAIT;
3191 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3193 if (G_LIKELY (gap == 0)) {
3194 /* no missing packet, pop and push */
3195 result = pop_and_push_next (jitterbuffer, seqnum);
3196 } else if (G_UNLIKELY (gap < 0)) {
3197 /* if we have a packet that we already pushed or considered dropped, pop it
3198 * off and get the next packet */
3199 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3200 seqnum, next_seqnum);
3201 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3203 result = GST_FLOW_OK;
3205 /* the chain function has scheduled timers to request retransmission or
3206 * when to consider the packet lost, wait for that */
3207 GST_DEBUG_OBJECT (jitterbuffer,
3208 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3209 next_seqnum, seqnum, gap);
3210 result = GST_FLOW_WAIT;
3218 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3220 return GST_FLOW_EOS;
3222 return GST_FLOW_WAIT;
3228 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3230 GstClockTime rtx_retry_timeout;
3231 GstClockTime rtx_min_retry_timeout;
3233 if (priv->rtx_retry_timeout == -1) {
3234 if (priv->avg_rtx_rtt == 0)
3235 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3237 /* we want to ask for a retransmission after we waited for a
3238 * complete RTT and the additional jitter */
3239 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3241 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3243 /* make sure we don't retry too often. On very low latency networks,
3244 * the RTT and jitter can be very low. */
3245 if (priv->rtx_min_retry_timeout == -1) {
3246 rtx_min_retry_timeout = priv->packet_spacing;
3248 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3250 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3252 return rtx_retry_timeout;
3256 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3257 GstClockTime rtx_retry_timeout)
3259 GstClockTime rtx_retry_period;
3261 if (priv->rtx_retry_period == -1) {
3262 /* we retry up to the configured jitterbuffer size but leaving some
3263 * room for the retransmission to arrive in time */
3264 if (rtx_retry_timeout > priv->latency_ns) {
3265 rtx_retry_period = 0;
3267 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3270 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3272 return rtx_retry_period;
3275 /* the timeout for when we expected a packet expired */
3277 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3280 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3282 guint delay, delay_ms, avg_rtx_rtt_ms;
3283 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3284 GstClockTime rtx_retry_period;
3285 GstClockTime rtx_retry_timeout;
3288 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3289 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3291 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3292 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3294 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3295 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3296 GST_TIME_ARGS (rtx_retry_period));
3298 delay = timer->rtx_delay + timer->rtx_retry;
3300 delay_ms = GST_TIME_AS_MSECONDS (delay);
3301 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3302 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3303 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3305 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3306 gst_structure_new ("GstRTPRetransmissionRequest",
3307 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3308 "running-time", G_TYPE_UINT64, timer->rtx_base,
3309 "delay", G_TYPE_UINT, delay_ms,
3310 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3311 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3312 "period", G_TYPE_UINT, rtx_retry_period_ms,
3313 "deadline", G_TYPE_UINT, priv->latency_ms,
3314 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3315 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3317 priv->num_rtx_requests++;
3318 timer->num_rtx_retry++;
3320 GST_OBJECT_LOCK (jitterbuffer);
3321 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3322 timer->rtx_last = gst_clock_get_time (clock);
3323 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3325 timer->rtx_last = now;
3327 GST_OBJECT_UNLOCK (jitterbuffer);
3329 /* calculate the timeout for the next retransmission attempt */
3330 timer->rtx_retry += rtx_retry_timeout;
3331 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3332 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3333 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3334 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3335 if ((priv->rtx_max_retries != -1
3336 && timer->num_rtx_retry >= priv->rtx_max_retries)
3337 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3338 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3339 /* too many retransmission request, we now convert the timer
3340 * to a lost timer, leave the num_rtx_retry as it is for stats */
3341 timer->type = TIMER_TYPE_LOST;
3342 timer->rtx_delay = 0;
3343 timer->rtx_retry = 0;
3345 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3346 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3349 gst_pad_push_event (priv->sinkpad, event);
3355 /* a packet is lost */
3357 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3360 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3361 GstClockTime duration, timestamp;
3362 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3365 RTPJitterBufferItem *item;
3367 seqnum = timer->seqnum;
3368 timestamp = apply_offset (jitterbuffer, timer->timeout);
3369 duration = timer->duration;
3370 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3371 duration = priv->packet_spacing;
3372 lost_packets = MAX (timer->num, 1);
3373 num_rtx_retry = timer->num_rtx_retry;
3375 /* we had a gap and thus we lost some packets. Create an event for this. */
3376 if (lost_packets > 1)
3377 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3378 seqnum + lost_packets - 1);
3380 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3382 priv->num_late += lost_packets;
3383 priv->num_rtx_failed += num_rtx_retry;
3385 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3387 /* we now only accept seqnum bigger than this */
3388 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3389 priv->next_in_seqnum = next_in_seqnum;
3391 /* create paket lost event */
3392 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3393 gst_structure_new ("GstRTPPacketLost",
3394 "seqnum", G_TYPE_UINT, (guint) seqnum,
3395 "timestamp", G_TYPE_UINT64, timestamp,
3396 "duration", G_TYPE_UINT64, duration,
3397 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3399 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3400 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3402 /* remove timer now */
3403 remove_timer (jitterbuffer, timer);
3405 JBUF_SIGNAL_EVENT (priv);
3411 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3414 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3416 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3417 remove_timer (jitterbuffer, timer);
3419 /* there was no EOS in the buffer, put one in there now */
3420 queue_event (jitterbuffer, gst_event_new_eos ());
3422 JBUF_SIGNAL_EVENT (priv);
3428 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3431 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3433 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3435 /* timer seqnum might have been obsoleted by caps seqnum-base,
3436 * only mess with current ongoing seqnum if still unknown */
3437 if (priv->next_seqnum == -1)
3438 priv->next_seqnum = timer->seqnum;
3439 remove_timer (jitterbuffer, timer);
3440 JBUF_SIGNAL_EVENT (priv);
3446 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3449 gboolean removed = FALSE;
3451 switch (timer->type) {
3452 case TIMER_TYPE_EXPECTED:
3453 removed = do_expected_timeout (jitterbuffer, timer, now);
3455 case TIMER_TYPE_LOST:
3456 removed = do_lost_timeout (jitterbuffer, timer, now);
3458 case TIMER_TYPE_DEADLINE:
3459 removed = do_deadline_timeout (jitterbuffer, timer, now);
3461 case TIMER_TYPE_EOS:
3462 removed = do_eos_timeout (jitterbuffer, timer, now);
3468 /* called when we need to wait for the next timeout.
3470 * We loop over the array of recorded timeouts and wait for the earliest one.
3471 * When it timed out, do the logic associated with the timer.
3473 * If there are no timers, we wait on a gcond until something new happens.
3476 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3478 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3479 GstClockTime now = 0;
3482 while (priv->timer_running) {
3483 TimerData *timer = NULL;
3484 GstClockTime timer_timeout = -1;
3487 /* If we have a clock, update "now" now with the very
3488 * latest running time we have. If timers are unscheduled below we
3489 * otherwise wouldn't update now (it's only updated when timers
3490 * expire), and also for the very first loop iteration now would
3491 * otherwise always be 0
3493 GST_OBJECT_LOCK (jitterbuffer);
3494 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3496 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3497 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3499 GST_OBJECT_UNLOCK (jitterbuffer);
3501 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3502 GST_TIME_ARGS (now));
3504 len = priv->timers->len;
3505 for (i = 0; i < len; i++) {
3506 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3507 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3508 gboolean save_best = FALSE;
3510 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3511 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3513 /* find the smallest timeout */
3514 if (timer == NULL) {
3516 } else if (timer_timeout == -1) {
3517 /* we already have an immediate timeout, the new timer must be an
3518 * immediate timer with smaller seqnum to become the best */
3519 if (test_timeout == -1
3520 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3521 timer->seqnum) > 0))
3523 } else if (test_timeout == -1) {
3524 /* first immediate timer */
3526 } else if (test_timeout < timer_timeout) {
3529 } else if (test_timeout == timer_timeout
3530 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3531 timer->seqnum) > 0)) {
3532 /* same timer, smaller seqnum */
3536 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3538 timer_timeout = test_timeout;
3541 if (timer && !priv->blocked) {
3543 GstClockTime sync_time;
3546 GstClockTimeDiff clock_jitter;
3548 if (timer_timeout == -1 || timer_timeout <= now) {
3549 do_timeout (jitterbuffer, timer, now);
3550 /* check here, do_timeout could have released the lock */
3551 if (!priv->timer_running)
3556 GST_OBJECT_LOCK (jitterbuffer);
3557 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3559 GST_OBJECT_UNLOCK (jitterbuffer);
3560 /* let's just push if there is no clock */
3561 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3562 now = timer_timeout;
3566 /* prepare for sync against clock */
3567 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3568 /* add latency of peer to get input time */
3569 sync_time += priv->peer_latency;
3571 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3572 " with sync time %" GST_TIME_FORMAT,
3573 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3575 /* create an entry for the clock */
3576 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3577 priv->timer_timeout = timer_timeout;
3578 priv->timer_seqnum = timer->seqnum;
3579 GST_OBJECT_UNLOCK (jitterbuffer);
3581 /* release the lock so that the other end can push stuff or unlock */
3584 ret = gst_clock_id_wait (id, &clock_jitter);
3587 if (!priv->timer_running) {
3588 gst_clock_id_unref (id);
3589 priv->clock_id = NULL;
3593 if (ret != GST_CLOCK_UNSCHEDULED) {
3594 now = timer_timeout + MAX (clock_jitter, 0);
3595 GST_DEBUG_OBJECT (jitterbuffer,
3596 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3597 GST_STIME_ARGS (clock_jitter));
3599 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3601 /* and free the entry */
3602 gst_clock_id_unref (id);
3603 priv->clock_id = NULL;
3605 /* no timers, wait for activity */
3606 JBUF_WAIT_TIMER (priv);
3611 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3616 * This funcion implements the main pushing loop on the source pad.
3618 * It first tries to push as many buffers as possible. If there is a seqnum
3619 * mismatch, we wait for the next timeouts.
3622 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3624 GstRtpJitterBufferPrivate *priv;
3625 GstFlowReturn result = GST_FLOW_OK;
3627 priv = jitterbuffer->priv;
3629 JBUF_LOCK_CHECK (priv, flushing);
3631 result = handle_next_buffer (jitterbuffer);
3632 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3633 /* now wait for the next event */
3634 JBUF_WAIT_EVENT (priv, flushing);
3635 result = GST_FLOW_OK;
3637 } while (result == GST_FLOW_OK);
3638 /* store result for upstream */
3639 priv->srcresult = result;
3640 /* if we get here we need to pause */
3646 result = priv->srcresult;
3653 JBUF_SIGNAL_QUERY (priv, FALSE);
3656 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3657 gst_flow_get_name (result));
3658 gst_pad_pause_task (priv->srcpad);
3659 if (result == GST_FLOW_EOS) {
3660 event = gst_event_new_eos ();
3661 gst_pad_push_event (priv->srcpad, event);
3667 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3668 * some sanity checks and then emit the handle-sync signal with the parameters.
3669 * This function must be called with the LOCK */
3671 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3673 GstRtpJitterBufferPrivate *priv;
3674 guint64 base_rtptime, base_time;
3676 guint64 last_rtptime;
3678 guint64 ext_rtptime, diff;
3679 gboolean valid = TRUE, keep = FALSE;
3681 priv = jitterbuffer->priv;
3683 /* get the last values from the jitterbuffer */
3684 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3685 &clock_rate, &last_rtptime);
3687 clock_base = priv->clock_base;
3688 ext_rtptime = priv->ext_rtptime;
3690 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3691 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3692 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3693 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3695 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3696 /* we keep this SR packet for later. When we get a valid RTP packet the
3697 * above values will be set and we can try to use the SR packet */
3698 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3701 /* we can't accept anything that happened before we did the last resync */
3702 if (base_rtptime > ext_rtptime) {
3703 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3706 /* the SR RTP timestamp must be something close to what we last observed
3707 * in the jitterbuffer */
3708 if (ext_rtptime > last_rtptime) {
3709 /* check how far ahead it is to our RTP timestamps */
3710 diff = ext_rtptime - last_rtptime;
3711 /* if bigger than 1 second, we drop it */
3712 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3714 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3715 clock_rate, 1000)) {
3716 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3717 /* should drop this, but some RTSP servers end up with bogus
3718 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3719 * so still trigger rptbin sync but invalidate RTCP data
3720 * (sync might use other methods) */
3723 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3724 G_GUINT64_FORMAT, last_rtptime, diff);
3730 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3734 s = gst_structure_new ("application/x-rtp-sync",
3735 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3736 "base-time", G_TYPE_UINT64, base_time,
3737 "clock-rate", G_TYPE_UINT, clock_rate,
3738 "clock-base", G_TYPE_UINT64, clock_base,
3739 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3740 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3742 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3743 gst_buffer_replace (&priv->last_sr, NULL);
3745 g_signal_emit (jitterbuffer,
3746 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3748 gst_structure_free (s);
3750 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3751 gst_buffer_replace (&priv->last_sr, NULL);
3755 static GstFlowReturn
3756 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3759 GstRtpJitterBuffer *jitterbuffer;
3760 GstRtpJitterBufferPrivate *priv;
3761 GstFlowReturn ret = GST_FLOW_OK;
3763 GstRTCPPacket packet;
3764 guint64 ext_rtptime;
3766 GstRTCPBuffer rtcp = { NULL, };
3768 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3770 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3771 goto invalid_buffer;
3773 priv = jitterbuffer->priv;
3775 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3777 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3780 /* first packet must be SR or RR or else the validate would have failed */
3781 switch (gst_rtcp_packet_get_type (&packet)) {
3782 case GST_RTCP_TYPE_SR:
3783 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3789 gst_rtcp_buffer_unmap (&rtcp);
3791 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3794 /* convert the RTP timestamp to our extended timestamp, using the same offset
3795 * we used in the jitterbuffer */
3796 ext_rtptime = priv->jbuf->ext_rtptime;
3797 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3799 priv->ext_rtptime = ext_rtptime;
3800 gst_buffer_replace (&priv->last_sr, buffer);
3802 do_handle_sync (jitterbuffer);
3806 gst_buffer_unref (buffer);
3812 /* this is not fatal but should be filtered earlier */
3813 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3814 ("Received invalid RTCP payload, dropping"));
3820 /* this is not fatal but should be filtered earlier */
3821 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3822 ("Received empty RTCP payload, dropping"));
3823 gst_rtcp_buffer_unmap (&rtcp);
3829 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3830 gst_rtcp_buffer_unmap (&rtcp);
3837 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3840 gboolean res = FALSE;
3841 GstRtpJitterBuffer *jitterbuffer;
3842 GstRtpJitterBufferPrivate *priv;
3844 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3845 priv = jitterbuffer->priv;
3847 switch (GST_QUERY_TYPE (query)) {
3848 case GST_QUERY_CAPS:
3850 GstCaps *filter, *caps;
3852 gst_query_parse_caps (query, &filter);
3853 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3854 gst_query_set_caps_result (query, caps);
3855 gst_caps_unref (caps);
3860 if (GST_QUERY_IS_SERIALIZED (query)) {
3861 RTPJitterBufferItem *item;
3864 JBUF_LOCK_CHECK (priv, out_flushing);
3865 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3866 RTP_JITTER_BUFFER_MODE_BUFFER) {
3867 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3868 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3869 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3871 JBUF_SIGNAL_EVENT (priv);
3872 JBUF_WAIT_QUERY (priv, out_flushing);
3873 res = priv->last_query;
3875 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3880 res = gst_pad_query_default (pad, parent, query);
3888 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3896 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3899 GstRtpJitterBuffer *jitterbuffer;
3900 GstRtpJitterBufferPrivate *priv;
3901 gboolean res = FALSE;
3903 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3904 priv = jitterbuffer->priv;
3906 switch (GST_QUERY_TYPE (query)) {
3907 case GST_QUERY_LATENCY:
3909 /* We need to send the query upstream and add the returned latency to our
3911 GstClockTime min_latency, max_latency;
3913 GstClockTime our_latency;
3915 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3916 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3918 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3919 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3920 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3922 /* store this so that we can safely sync on the peer buffers. */
3924 priv->peer_latency = min_latency;
3925 our_latency = priv->latency_ns;
3928 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3929 GST_TIME_ARGS (our_latency));
3931 /* we add some latency but can buffer an infinite amount of time */
3932 min_latency += our_latency;
3935 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3936 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3937 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3939 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3943 case GST_QUERY_POSITION:
3945 GstClockTime start, last_out;
3948 gst_query_parse_position (query, &fmt, NULL);
3949 if (fmt != GST_FORMAT_TIME) {
3950 res = gst_pad_query_default (pad, parent, query);
3955 start = priv->npt_start;
3956 last_out = priv->last_out_time;
3959 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3960 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3961 GST_TIME_ARGS (last_out));
3963 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3964 /* bring 0-based outgoing time to stream time */
3965 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3968 res = gst_pad_query_default (pad, parent, query);
3972 case GST_QUERY_CAPS:
3974 GstCaps *filter, *caps;
3976 gst_query_parse_caps (query, &filter);
3977 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3978 gst_query_set_caps_result (query, caps);
3979 gst_caps_unref (caps);
3984 res = gst_pad_query_default (pad, parent, query);
3992 gst_rtp_jitter_buffer_set_property (GObject * object,
3993 guint prop_id, const GValue * value, GParamSpec * pspec)
3995 GstRtpJitterBuffer *jitterbuffer;
3996 GstRtpJitterBufferPrivate *priv;
3998 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3999 priv = jitterbuffer->priv;
4004 guint new_latency, old_latency;
4006 new_latency = g_value_get_uint (value);
4009 old_latency = priv->latency_ms;
4010 priv->latency_ms = new_latency;
4011 priv->latency_ns = priv->latency_ms * GST_MSECOND;
4012 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
4015 /* post message if latency changed, this will inform the parent pipeline
4016 * that a latency reconfiguration is possible/needed. */
4017 if (new_latency != old_latency) {
4018 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
4019 GST_TIME_ARGS (new_latency * GST_MSECOND));
4021 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
4022 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
4026 case PROP_DROP_ON_LATENCY:
4028 priv->drop_on_latency = g_value_get_boolean (value);
4031 case PROP_TS_OFFSET:
4033 priv->ts_offset = g_value_get_int64 (value);
4034 priv->ts_discont = TRUE;
4039 priv->do_lost = g_value_get_boolean (value);
4044 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
4047 case PROP_DO_RETRANSMISSION:
4049 priv->do_retransmission = g_value_get_boolean (value);
4052 case PROP_RTX_NEXT_SEQNUM:
4054 priv->rtx_next_seqnum = g_value_get_boolean (value);
4057 case PROP_RTX_DELAY:
4059 priv->rtx_delay = g_value_get_int (value);
4062 case PROP_RTX_MIN_DELAY:
4064 priv->rtx_min_delay = g_value_get_uint (value);
4067 case PROP_RTX_DELAY_REORDER:
4069 priv->rtx_delay_reorder = g_value_get_int (value);
4072 case PROP_RTX_RETRY_TIMEOUT:
4074 priv->rtx_retry_timeout = g_value_get_int (value);
4077 case PROP_RTX_MIN_RETRY_TIMEOUT:
4079 priv->rtx_min_retry_timeout = g_value_get_int (value);
4082 case PROP_RTX_RETRY_PERIOD:
4084 priv->rtx_retry_period = g_value_get_int (value);
4087 case PROP_RTX_MAX_RETRIES:
4089 priv->rtx_max_retries = g_value_get_int (value);
4092 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4094 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4097 case PROP_MAX_DROPOUT_TIME:
4099 priv->max_dropout_time = g_value_get_uint (value);
4102 case PROP_MAX_MISORDER_TIME:
4104 priv->max_misorder_time = g_value_get_uint (value);
4108 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4114 gst_rtp_jitter_buffer_get_property (GObject * object,
4115 guint prop_id, GValue * value, GParamSpec * pspec)
4117 GstRtpJitterBuffer *jitterbuffer;
4118 GstRtpJitterBufferPrivate *priv;
4120 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4121 priv = jitterbuffer->priv;
4126 g_value_set_uint (value, priv->latency_ms);
4129 case PROP_DROP_ON_LATENCY:
4131 g_value_set_boolean (value, priv->drop_on_latency);
4134 case PROP_TS_OFFSET:
4136 g_value_set_int64 (value, priv->ts_offset);
4141 g_value_set_boolean (value, priv->do_lost);
4146 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4154 if (priv->srcresult != GST_FLOW_OK)
4157 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4159 g_value_set_int (value, percent);
4163 case PROP_DO_RETRANSMISSION:
4165 g_value_set_boolean (value, priv->do_retransmission);
4168 case PROP_RTX_NEXT_SEQNUM:
4170 g_value_set_boolean (value, priv->rtx_next_seqnum);
4173 case PROP_RTX_DELAY:
4175 g_value_set_int (value, priv->rtx_delay);
4178 case PROP_RTX_MIN_DELAY:
4180 g_value_set_uint (value, priv->rtx_min_delay);
4183 case PROP_RTX_DELAY_REORDER:
4185 g_value_set_int (value, priv->rtx_delay_reorder);
4188 case PROP_RTX_RETRY_TIMEOUT:
4190 g_value_set_int (value, priv->rtx_retry_timeout);
4193 case PROP_RTX_MIN_RETRY_TIMEOUT:
4195 g_value_set_int (value, priv->rtx_min_retry_timeout);
4198 case PROP_RTX_RETRY_PERIOD:
4200 g_value_set_int (value, priv->rtx_retry_period);
4203 case PROP_RTX_MAX_RETRIES:
4205 g_value_set_int (value, priv->rtx_max_retries);
4209 g_value_take_boxed (value,
4210 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4212 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4214 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4217 case PROP_MAX_DROPOUT_TIME:
4219 g_value_set_uint (value, priv->max_dropout_time);
4222 case PROP_MAX_MISORDER_TIME:
4224 g_value_set_uint (value, priv->max_misorder_time);
4228 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4233 static GstStructure *
4234 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4238 JBUF_LOCK (jbuf->priv);
4239 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4240 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4241 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4242 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4243 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4244 JBUF_UNLOCK (jbuf->priv);