2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
9 * This library is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Library General Public
11 * License as published by the Free Software Foundation; either
12 * version 2 of the License, or (at your option) any later version.
14 * This library is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Library General Public License for more details.
19 * You should have received a copy of the GNU Library General Public
20 * License along with this library; if not, write to the
21 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
22 * Boston, MA 02110-1301, USA.
27 * SECTION:element-rtpjitterbuffer
29 * This element reorders and removes duplicate RTP packets as they are received
30 * from a network source.
32 * The element needs the clock-rate of the RTP payload in order to estimate the
33 * delay. This information is obtained either from the caps on the sink pad or,
34 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
35 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
37 * The rtpjitterbuffer will wait for missing packets up to a configurable time
38 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
39 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
40 * property is set, lost packets will result in a custom serialized downstream
41 * event of name GstRTPPacketLost. The lost packet events are usually used by a
42 * depayloader or other element to create concealment data or some other logic
43 * to gracefully handle the missing packets.
45 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
46 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
49 * The jitterbuffer can also be configured to send early retransmission events
50 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
51 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
52 * sends a custom upstream event named GstRTPRetransmissionRequest when the
53 * packet is considered late. The initial expected packet arrival time is
54 * calculated as follows:
56 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
57 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
58 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
59 * packets with different rtptime.
61 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
62 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
63 * previously scheduled timeout is overwritten.
65 * - If seqnum N arrived, all seqnum older than
66 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
67 * immediately. This is to request fast feedback for abonormally reorder
68 * packets before any of the previous timeouts is triggered.
70 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
71 * event. After the initial timeout expires and the retransmission event is
72 * sent, the timeout is scheduled for
73 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
74 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
75 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
76 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
77 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
78 * retransmission requests are sent and the regular logic is performed to
79 * schedule a lost packet as discussed above.
81 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
84 * This element will automatically be used inside rtpbin.
87 * <title>Example pipelines</title>
89 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
90 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
91 * inserted into the pipeline to smooth out network jitter and to reorder the
92 * out-of-order RTP packets.
102 #include <gst/rtp/gstrtpbuffer.h>
104 #include "gstrtpjitterbuffer.h"
105 #include "rtpjitterbuffer.h"
106 #include "rtpstats.h"
108 #include <gst/glib-compat-private.h>
110 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
111 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
113 /* RTPJitterBuffer signals and args */
116 SIGNAL_REQUEST_PT_MAP,
124 #define DEFAULT_LATENCY_MS 200
125 #define DEFAULT_DROP_ON_LATENCY FALSE
126 #define DEFAULT_TS_OFFSET 0
127 #define DEFAULT_DO_LOST FALSE
128 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
129 #define DEFAULT_PERCENT 0
130 #define DEFAULT_DO_RETRANSMISSION FALSE
131 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
132 #define DEFAULT_RTX_DELAY -1
133 #define DEFAULT_RTX_MIN_DELAY 0
134 #define DEFAULT_RTX_DELAY_REORDER 3
135 #define DEFAULT_RTX_RETRY_TIMEOUT -1
136 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
137 #define DEFAULT_RTX_RETRY_PERIOD -1
138 #define DEFAULT_RTX_MAX_RETRIES -1
139 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
140 #define DEFAULT_MAX_DROPOUT_TIME 60000
141 #define DEFAULT_MAX_MISORDER_TIME 2000
143 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
144 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
150 PROP_DROP_ON_LATENCY,
155 PROP_DO_RETRANSMISSION,
156 PROP_RTX_NEXT_SEQNUM,
159 PROP_RTX_DELAY_REORDER,
160 PROP_RTX_RETRY_TIMEOUT,
161 PROP_RTX_MIN_RETRY_TIMEOUT,
162 PROP_RTX_RETRY_PERIOD,
163 PROP_RTX_MAX_RETRIES,
165 PROP_MAX_RTCP_RTP_TIME_DIFF,
166 PROP_MAX_DROPOUT_TIME,
167 PROP_MAX_MISORDER_TIME
170 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
172 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
174 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
177 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
179 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
180 GST_DEBUG ("waiting timer"); \
181 (priv)->waiting_timer = TRUE; \
182 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
183 (priv)->waiting_timer = FALSE; \
184 GST_DEBUG ("waiting timer done"); \
186 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
187 if (G_UNLIKELY ((priv)->waiting_timer)) { \
188 GST_DEBUG ("signal timer"); \
189 g_cond_signal (&(priv)->jbuf_timer); \
193 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
194 GST_DEBUG ("waiting event"); \
195 (priv)->waiting_event = TRUE; \
196 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
197 (priv)->waiting_event = FALSE; \
198 GST_DEBUG ("waiting event done"); \
199 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
202 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
203 if (G_UNLIKELY ((priv)->waiting_event)) { \
204 GST_DEBUG ("signal event"); \
205 g_cond_signal (&(priv)->jbuf_event); \
209 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
210 GST_DEBUG ("waiting query"); \
211 (priv)->waiting_query = TRUE; \
212 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
213 (priv)->waiting_query = FALSE; \
214 GST_DEBUG ("waiting query done"); \
215 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
218 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
219 (priv)->last_query = res; \
220 if (G_UNLIKELY ((priv)->waiting_query)) { \
221 GST_DEBUG ("signal query"); \
222 g_cond_signal (&(priv)->jbuf_query); \
227 struct _GstRtpJitterBufferPrivate
229 GstPad *sinkpad, *srcpad;
232 RTPJitterBuffer *jbuf;
234 gboolean waiting_timer;
236 gboolean waiting_event;
238 gboolean waiting_query;
246 gboolean timer_running;
247 GThread *timer_thread;
252 gboolean drop_on_latency;
255 gboolean do_retransmission;
256 gboolean rtx_next_seqnum;
259 gint rtx_delay_reorder;
260 gint rtx_retry_timeout;
261 gint rtx_min_retry_timeout;
262 gint rtx_retry_period;
263 gint rtx_max_retries;
264 gint max_rtcp_rtp_time_diff;
265 guint32 max_dropout_time;
266 guint32 max_misorder_time;
268 /* the last seqnum we pushed out */
269 guint32 last_popped_seqnum;
270 /* the next expected seqnum we push */
272 /* seqnum-base, if known */
274 /* last output time */
275 GstClockTime last_out_time;
276 /* last valid input timestamp and rtptime pair */
277 GstClockTime ips_dts;
279 GstClockTime packet_spacing;
283 /* the next expected seqnum we receive */
284 GstClockTime last_in_dts;
285 guint32 next_in_seqnum;
289 /* start and stop ranges */
290 GstClockTime npt_start;
291 GstClockTime npt_stop;
292 guint64 ext_timestamp;
293 guint64 last_elapsed;
294 guint64 estimated_eos;
301 /* clock rate and rtp timestamp offset */
305 gint64 prev_ts_offset;
307 /* when we are shutting down */
308 GstFlowReturn srcresult;
314 GstClockTime timer_timeout;
315 guint16 timer_seqnum;
316 /* the latency of the upstream peer, we have to take this into account when
317 * synchronizing the buffers. */
318 GstClockTime peer_latency;
322 /* some accounting */
324 guint64 num_duplicates;
325 guint64 num_rtx_requests;
326 guint64 num_rtx_success;
327 guint64 num_rtx_failed;
332 GstClockTime last_dts;
333 guint64 last_rtptime;
334 GstClockTime avg_jitter;
351 GstClockTime timeout;
352 GstClockTime duration;
353 GstClockTime rtx_base;
354 GstClockTime rtx_delay;
355 GstClockTime rtx_retry;
356 GstClockTime rtx_last;
360 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
361 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
362 GstRtpJitterBufferPrivate))
364 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
365 GST_STATIC_PAD_TEMPLATE ("sink",
368 GST_STATIC_CAPS ("application/x-rtp"
369 /* "clock-rate = (int) [ 1, 2147483647 ], "
370 * "payload = (int) , "
371 * "encoding-name = (string) "
375 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
376 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
379 GST_STATIC_CAPS ("application/x-rtcp")
382 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
383 GST_STATIC_PAD_TEMPLATE ("src",
386 GST_STATIC_CAPS ("application/x-rtp"
387 /* "payload = (int) , "
388 * "clock-rate = (int) , "
389 * "encoding-name = (string) "
393 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
395 #define gst_rtp_jitter_buffer_parent_class parent_class
396 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
398 /* object overrides */
399 static void gst_rtp_jitter_buffer_set_property (GObject * object,
400 guint prop_id, const GValue * value, GParamSpec * pspec);
401 static void gst_rtp_jitter_buffer_get_property (GObject * object,
402 guint prop_id, GValue * value, GParamSpec * pspec);
403 static void gst_rtp_jitter_buffer_finalize (GObject * object);
405 /* element overrides */
406 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
407 * element, GstStateChange transition);
408 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
409 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
410 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
412 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
415 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
416 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
419 /* sinkpad overrides */
420 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
421 GstObject * parent, GstEvent * event);
422 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
423 GstObject * parent, GstBuffer * buffer);
425 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
426 GstObject * parent, GstEvent * event);
427 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
428 GstObject * parent, GstBuffer * buffer);
430 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
431 GstObject * parent, GstQuery * query);
433 /* srcpad overrides */
434 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
435 GstObject * parent, GstEvent * event);
436 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
437 GstObject * parent, GstPadMode mode, gboolean active);
438 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
439 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
440 GstObject * parent, GstQuery * query);
443 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
445 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
446 gboolean active, guint64 base_time);
447 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
449 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
450 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
452 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
454 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
458 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
460 GObjectClass *gobject_class;
461 GstElementClass *gstelement_class;
463 gobject_class = (GObjectClass *) klass;
464 gstelement_class = (GstElementClass *) klass;
466 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
468 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
470 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
471 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
474 * GstRtpJitterBuffer:latency:
476 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
477 * for at most this time.
479 g_object_class_install_property (gobject_class, PROP_LATENCY,
480 g_param_spec_uint ("latency", "Buffer latency in ms",
481 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
482 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
484 * GstRtpJitterBuffer:drop-on-latency:
486 * Drop oldest buffers when the queue is completely filled.
488 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
489 g_param_spec_boolean ("drop-on-latency",
490 "Drop buffers when maximum latency is reached",
491 "Tells the jitterbuffer to never exceed the given latency in size",
492 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
494 * GstRtpJitterBuffer:ts-offset:
496 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
497 * This is mainly used to ensure interstream synchronisation.
499 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
500 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
501 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
502 G_MAXINT64, DEFAULT_TS_OFFSET,
503 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
506 * GstRtpJitterBuffer:do-lost:
508 * Send out a GstRTPPacketLost event downstream when a packet is considered
511 g_object_class_install_property (gobject_class, PROP_DO_LOST,
512 g_param_spec_boolean ("do-lost", "Do Lost",
513 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
514 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
517 * GstRtpJitterBuffer:mode:
519 * Control the buffering and timestamping mode used by the jitterbuffer.
521 g_object_class_install_property (gobject_class, PROP_MODE,
522 g_param_spec_enum ("mode", "Mode",
523 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
524 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
526 * GstRtpJitterBuffer:percent:
528 * The percent of the jitterbuffer that is filled.
530 g_object_class_install_property (gobject_class, PROP_PERCENT,
531 g_param_spec_int ("percent", "percent",
532 "The buffer filled percent", 0, 100,
533 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
535 * GstRtpJitterBuffer:do-retransmission:
537 * Send out a GstRTPRetransmission event upstream when a packet is considered
538 * late and should be retransmitted.
542 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
543 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
544 "Send retransmission events upstream when a packet is late",
545 DEFAULT_DO_RETRANSMISSION,
546 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
549 * GstRtpJitterBuffer:rtx-next-seqnum
551 * Estimate when the next packet should arrive and schedule a retransmission
553 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
554 * for packet N+1. So it will be requested if it does not arrive at the expected time.
555 * The expected time is calculated using the dts of N and the packet spacing.
559 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
560 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
561 "Estimate when the next packet should arrive and schedule a "
562 "retransmission request for it.",
563 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
566 * GstRtpJitterBuffer:rtx-delay:
568 * When a packet did not arrive at the expected time, wait this extra amount
569 * of time before sending a retransmission event.
571 * When -1 is used, the max jitter will be used as extra delay.
575 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
576 g_param_spec_int ("rtx-delay", "RTX Delay",
577 "Extra time in ms to wait before sending retransmission "
578 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
579 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
582 * GstRtpJitterBuffer:rtx-min-delay:
584 * When a packet did not arrive at the expected time, wait at least this extra amount
585 * of time before sending a retransmission event.
589 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
590 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
591 "Minimum time in ms to wait before sending retransmission "
592 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
593 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
595 * GstRtpJitterBuffer:rtx-delay-reorder:
597 * Assume that a retransmission event should be sent when we see
598 * this much packet reordering.
600 * When -1 is used, the value will be estimated based on observed packet
605 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
606 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
607 "Sending retransmission event when this much reordering (-1 automatic)",
608 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
609 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
611 * GstRtpJitterBuffer::rtx-retry-timeout:
613 * When no packet has been received after sending a retransmission event
614 * for this time, retry sending a retransmission event.
616 * When -1 is used, the value will be estimated based on observed round
621 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
622 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
623 "Retry sending a transmission event after this timeout in "
624 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
625 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
627 * GstRtpJitterBuffer::rtx-min-retry-timeout:
629 * The minimum amount of time between retry timeouts. When
630 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
631 * minimum interval between retry timeouts.
633 * When -1 is used, the value will be estimated based on the
638 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
639 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
640 "Minimum timeout between sending a transmission event in "
641 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
642 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
644 * GstRtpJitterBuffer:rtx-retry-period:
646 * The amount of time to try to get a retransmission.
648 * When -1 is used, the value will be estimated based on the jitterbuffer
649 * latency and the observed round trip time.
653 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
654 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
655 "Try to get a retransmission for this many ms "
656 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
657 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
659 * GstRtpJitterBuffer:rtx-max-retries:
661 * The maximum number of retries to request a retransmission.
663 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
664 * When -1 is used, the number of retransmission request will not be limited.
668 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
669 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
670 "The maximum number of retries to request a retransmission. "
671 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
672 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
674 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
675 g_param_spec_uint ("max-dropout-time", "Max dropout time",
676 "The maximum time (milliseconds) of missing packets tolerated.",
677 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
678 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
680 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
681 g_param_spec_uint ("max-misorder-time", "Max misorder time",
682 "The maximum time (milliseconds) of misordered packets tolerated.",
683 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
684 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
686 * GstRtpJitterBuffer:stats:
688 * Various jitterbuffer statistics. This property returns a GstStructure
689 * with name application/x-rtp-jitterbuffer-stats with the following fields:
695 * <classname>"rtx-count"</classname>:
696 * the number of retransmissions requested.
702 * <classname>"rtx-success-count"</classname>:
703 * the number of successful retransmissions.
709 * <classname>"rtx-per-packet"</classname>:
710 * average number of RTX per packet.
716 * <classname>"rtx-rtt"</classname>:
717 * average round trip time per RTX.
724 g_object_class_install_property (gobject_class, PROP_STATS,
725 g_param_spec_boxed ("stats", "Statistics",
726 "Various statistics", GST_TYPE_STRUCTURE,
727 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
730 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
732 * The maximum amount of time in ms that the RTP time in the RTCP SRs
733 * is allowed to be ahead of the last RTP packet we received. Use
734 * -1 to disable ignoring of RTCP packets.
738 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
739 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
740 "Maximum amount of time in ms that the RTP time in RTCP SRs "
741 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
742 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
743 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
746 * GstRtpJitterBuffer::request-pt-map:
747 * @buffer: the object which received the signal
750 * Request the payload type as #GstCaps for @pt.
752 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
753 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
754 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
755 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
756 GST_TYPE_CAPS, 1, G_TYPE_UINT);
758 * GstRtpJitterBuffer::handle-sync:
759 * @buffer: the object which received the signal
760 * @struct: a GstStructure containing sync values.
762 * Be notified of new sync values.
764 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
765 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
766 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
767 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
768 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
771 * GstRtpJitterBuffer::on-npt-stop:
772 * @buffer: the object which received the signal
774 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
775 * the npt-stop position.
777 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
778 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
779 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
780 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
781 G_TYPE_NONE, 0, G_TYPE_NONE);
784 * GstRtpJitterBuffer::clear-pt-map:
785 * @buffer: the object which received the signal
787 * Invalidate the clock-rate as obtained with the
788 * #GstRtpJitterBuffer::request-pt-map signal.
790 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
791 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
792 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
793 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
794 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
797 * GstRtpJitterBuffer::set-active:
798 * @buffer: the object which received the signal
800 * Start pushing out packets with the given base time. This signal is only
801 * useful in buffering mode.
803 * Returns: the time of the last pushed packet.
805 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
806 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
807 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
808 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
809 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
812 gstelement_class->change_state =
813 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
814 gstelement_class->request_new_pad =
815 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
816 gstelement_class->release_pad =
817 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
818 gstelement_class->provide_clock =
819 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
821 gst_element_class_add_pad_template (gstelement_class,
822 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
823 gst_element_class_add_pad_template (gstelement_class,
824 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
825 gst_element_class_add_pad_template (gstelement_class,
826 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
828 gst_element_class_set_static_metadata (gstelement_class,
829 "RTP packet jitter-buffer", "Filter/Network/RTP",
830 "A buffer that deals with network jitter and other transmission faults",
831 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
832 "Wim Taymans <wim.taymans@gmail.com>");
834 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
835 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
837 GST_DEBUG_CATEGORY_INIT
838 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
842 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
844 GstRtpJitterBufferPrivate *priv;
846 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
847 jitterbuffer->priv = priv;
849 priv->latency_ms = DEFAULT_LATENCY_MS;
850 priv->latency_ns = priv->latency_ms * GST_MSECOND;
851 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
852 priv->do_lost = DEFAULT_DO_LOST;
853 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
854 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
855 priv->rtx_delay = DEFAULT_RTX_DELAY;
856 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
857 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
858 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
859 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
860 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
861 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
862 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
863 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
864 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
867 priv->last_rtptime = -1;
868 priv->avg_jitter = 0;
869 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
870 priv->jbuf = rtp_jitter_buffer_new ();
871 g_mutex_init (&priv->jbuf_lock);
872 g_cond_init (&priv->jbuf_timer);
873 g_cond_init (&priv->jbuf_event);
874 g_cond_init (&priv->jbuf_query);
875 g_queue_init (&priv->gap_packets);
877 /* reset skew detection initialy */
878 rtp_jitter_buffer_reset_skew (priv->jbuf);
879 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
880 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
884 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
887 gst_pad_set_activatemode_function (priv->srcpad,
888 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
889 gst_pad_set_query_function (priv->srcpad,
890 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
891 gst_pad_set_event_function (priv->srcpad,
892 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
895 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
898 gst_pad_set_chain_function (priv->sinkpad,
899 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
900 gst_pad_set_event_function (priv->sinkpad,
901 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
902 gst_pad_set_query_function (priv->sinkpad,
903 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
905 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
906 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
908 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
911 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
913 #define ITEM_TYPE_BUFFER 0
914 #define ITEM_TYPE_LOST 1
915 #define ITEM_TYPE_EVENT 2
916 #define ITEM_TYPE_QUERY 3
918 static RTPJitterBufferItem *
919 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
920 guint seqnum, guint count, guint rtptime)
922 RTPJitterBufferItem *item;
924 item = g_slice_new (RTPJitterBufferItem);
931 item->seqnum = seqnum;
933 item->rtptime = rtptime;
939 free_item (RTPJitterBufferItem * item)
941 g_return_if_fail (item != NULL);
943 if (item->data && item->type != ITEM_TYPE_QUERY)
944 gst_mini_object_unref (item->data);
945 g_slice_free (RTPJitterBufferItem, item);
949 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
951 GList **l = user_data;
953 if (item->data && item->type == ITEM_TYPE_EVENT
954 && GST_EVENT_IS_STICKY (item->data)) {
955 *l = g_list_prepend (*l, item->data);
956 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
957 gst_mini_object_unref (item->data);
959 g_slice_free (RTPJitterBufferItem, item);
963 gst_rtp_jitter_buffer_finalize (GObject * object)
965 GstRtpJitterBuffer *jitterbuffer;
966 GstRtpJitterBufferPrivate *priv;
968 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
969 priv = jitterbuffer->priv;
971 g_array_free (priv->timers, TRUE);
972 g_mutex_clear (&priv->jbuf_lock);
973 g_cond_clear (&priv->jbuf_timer);
974 g_cond_clear (&priv->jbuf_event);
975 g_cond_clear (&priv->jbuf_query);
977 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
978 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
979 g_queue_clear (&priv->gap_packets);
980 g_object_unref (priv->jbuf);
982 G_OBJECT_CLASS (parent_class)->finalize (object);
986 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
988 GstRtpJitterBuffer *jitterbuffer;
989 GstPad *otherpad = NULL;
990 GstIterator *it = NULL;
993 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
995 if (pad == jitterbuffer->priv->sinkpad) {
996 otherpad = jitterbuffer->priv->srcpad;
997 } else if (pad == jitterbuffer->priv->srcpad) {
998 otherpad = jitterbuffer->priv->sinkpad;
999 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1000 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1004 g_value_init (&val, GST_TYPE_PAD);
1005 g_value_set_object (&val, otherpad);
1006 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1007 g_value_unset (&val);
1014 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1016 GstRtpJitterBufferPrivate *priv;
1018 priv = jitterbuffer->priv;
1020 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1023 gst_pad_new_from_static_template
1024 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1025 gst_pad_set_chain_function (priv->rtcpsinkpad,
1026 gst_rtp_jitter_buffer_chain_rtcp);
1027 gst_pad_set_event_function (priv->rtcpsinkpad,
1028 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1029 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1030 gst_rtp_jitter_buffer_iterate_internal_links);
1031 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1032 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1034 return priv->rtcpsinkpad;
1038 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1040 GstRtpJitterBufferPrivate *priv;
1042 priv = jitterbuffer->priv;
1044 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1046 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1048 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1049 priv->rtcpsinkpad = NULL;
1053 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1054 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1056 GstRtpJitterBuffer *jitterbuffer;
1057 GstElementClass *klass;
1059 GstRtpJitterBufferPrivate *priv;
1061 g_return_val_if_fail (templ != NULL, NULL);
1062 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1064 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1065 priv = jitterbuffer->priv;
1066 klass = GST_ELEMENT_GET_CLASS (element);
1068 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1070 /* figure out the template */
1071 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1072 if (priv->rtcpsinkpad != NULL)
1075 result = create_rtcp_sink (jitterbuffer);
1077 goto wrong_template;
1084 g_warning ("rtpjitterbuffer: this is not our template");
1089 g_warning ("rtpjitterbuffer: pad already requested");
1095 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1097 GstRtpJitterBuffer *jitterbuffer;
1098 GstRtpJitterBufferPrivate *priv;
1100 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1101 g_return_if_fail (GST_IS_PAD (pad));
1103 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1104 priv = jitterbuffer->priv;
1106 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1108 if (priv->rtcpsinkpad == pad) {
1109 remove_rtcp_sink (jitterbuffer);
1118 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1124 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1126 return gst_system_clock_obtain ();
1130 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1132 GstRtpJitterBufferPrivate *priv;
1134 priv = jitterbuffer->priv;
1136 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1139 priv->clock_rate = -1;
1140 /* do not clear current content, but refresh state for new arrival */
1141 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1142 rtp_jitter_buffer_reset_skew (priv->jbuf);
1147 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1150 GstRtpJitterBufferPrivate *priv;
1151 GstClockTime last_out;
1152 RTPJitterBufferItem *item;
1157 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1158 active, GST_TIME_ARGS (offset));
1160 if (active != priv->active) {
1161 /* add the amount of time spent in paused to the output offset. All
1162 * outgoing buffers will have this offset applied to their timestamps in
1163 * order to make them arrive in time in the sink. */
1164 priv->out_offset = offset;
1165 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1166 GST_TIME_ARGS (priv->out_offset));
1167 priv->active = active;
1168 JBUF_SIGNAL_EVENT (priv);
1171 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1173 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1174 /* head buffer timestamp and offset gives our output time */
1175 last_out = item->dts + priv->ts_offset;
1177 /* use last known time when the buffer is empty */
1178 last_out = priv->last_out_time;
1186 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1188 GstRtpJitterBuffer *jitterbuffer;
1189 GstRtpJitterBufferPrivate *priv;
1194 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1195 priv = jitterbuffer->priv;
1197 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1199 caps = gst_pad_peer_query_caps (other, filter);
1201 templ = gst_pad_get_pad_template_caps (pad);
1203 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1208 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1210 intersect = gst_caps_intersect (caps, templ);
1211 gst_caps_unref (caps);
1212 gst_caps_unref (templ);
1216 gst_object_unref (jitterbuffer);
1222 * Must be called with JBUF_LOCK held
1226 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1229 GstRtpJitterBufferPrivate *priv;
1230 GstStructure *caps_struct;
1234 priv = jitterbuffer->priv;
1236 /* first parse the caps */
1237 caps_struct = gst_caps_get_structure (caps, 0);
1239 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1241 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1242 * measure the amount of data in the buffer */
1243 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1246 if (priv->clock_rate <= 0)
1249 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1251 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1253 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1254 * can use this to track the amount of time elapsed on the sender. */
1255 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1256 priv->clock_base = val;
1258 priv->clock_base = -1;
1260 priv->ext_timestamp = priv->clock_base;
1262 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1265 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1266 /* first expected seqnum, only update when we didn't have a previous base. */
1267 if (priv->next_in_seqnum == -1)
1268 priv->next_in_seqnum = val;
1269 if (priv->next_seqnum == -1) {
1270 priv->next_seqnum = val;
1271 JBUF_SIGNAL_EVENT (priv);
1273 priv->seqnum_base = val;
1275 priv->seqnum_base = -1;
1278 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1280 /* the start and stop times. The seqnum-base corresponds to the start time. We
1281 * will keep track of the seqnums on the output and when we reach the one
1282 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1283 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1284 priv->npt_start = tval;
1286 priv->npt_start = 0;
1288 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1289 priv->npt_stop = tval;
1291 priv->npt_stop = -1;
1293 GST_DEBUG_OBJECT (jitterbuffer,
1294 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1295 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1302 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1307 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1313 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1315 GstRtpJitterBufferPrivate *priv;
1317 priv = jitterbuffer->priv;
1320 /* mark ourselves as flushing */
1321 priv->srcresult = GST_FLOW_FLUSHING;
1322 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1323 /* this unblocks any waiting pops on the src pad task */
1324 JBUF_SIGNAL_EVENT (priv);
1325 JBUF_SIGNAL_QUERY (priv, FALSE);
1330 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1332 GstRtpJitterBufferPrivate *priv;
1334 priv = jitterbuffer->priv;
1337 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1338 /* Mark as non flushing */
1339 priv->srcresult = GST_FLOW_OK;
1340 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1341 priv->last_popped_seqnum = -1;
1342 priv->last_out_time = -1;
1343 priv->next_seqnum = -1;
1344 priv->seqnum_base = -1;
1345 priv->ips_rtptime = -1;
1346 priv->ips_dts = GST_CLOCK_TIME_NONE;
1347 priv->packet_spacing = 0;
1348 priv->next_in_seqnum = -1;
1349 priv->clock_rate = -1;
1352 priv->estimated_eos = -1;
1353 priv->last_elapsed = 0;
1354 priv->ext_timestamp = -1;
1355 priv->avg_jitter = 0;
1356 priv->last_dts = -1;
1357 priv->last_rtptime = -1;
1358 priv->last_in_dts = 0;
1359 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1360 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1361 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1362 rtp_jitter_buffer_reset_skew (priv->jbuf);
1363 remove_all_timers (jitterbuffer);
1364 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1365 g_queue_clear (&priv->gap_packets);
1370 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1371 GstPadMode mode, gboolean active)
1374 GstRtpJitterBuffer *jitterbuffer = NULL;
1376 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1379 case GST_PAD_MODE_PUSH:
1381 /* allow data processing */
1382 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1384 /* start pushing out buffers */
1385 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1386 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1387 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1389 /* make sure all data processing stops ASAP */
1390 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1392 /* NOTE this will hardlock if the state change is called from the src pad
1393 * task thread because we will _join() the thread. */
1394 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1395 result = gst_pad_stop_task (pad);
1405 static GstStateChangeReturn
1406 gst_rtp_jitter_buffer_change_state (GstElement * element,
1407 GstStateChange transition)
1409 GstRtpJitterBuffer *jitterbuffer;
1410 GstRtpJitterBufferPrivate *priv;
1411 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1413 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1414 priv = jitterbuffer->priv;
1416 switch (transition) {
1417 case GST_STATE_CHANGE_NULL_TO_READY:
1419 case GST_STATE_CHANGE_READY_TO_PAUSED:
1421 /* reset negotiated values */
1422 priv->clock_rate = -1;
1423 priv->clock_base = -1;
1424 priv->peer_latency = 0;
1426 /* block until we go to PLAYING */
1427 priv->blocked = TRUE;
1428 priv->timer_running = TRUE;
1429 priv->timer_thread =
1430 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1433 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1435 /* unblock to allow streaming in PLAYING */
1436 priv->blocked = FALSE;
1437 JBUF_SIGNAL_EVENT (priv);
1438 JBUF_SIGNAL_TIMER (priv);
1445 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1447 switch (transition) {
1448 case GST_STATE_CHANGE_READY_TO_PAUSED:
1449 /* we are a live element because we sync to the clock, which we can only
1450 * do in the PLAYING state */
1451 if (ret != GST_STATE_CHANGE_FAILURE)
1452 ret = GST_STATE_CHANGE_NO_PREROLL;
1454 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1456 /* block to stop streaming when PAUSED */
1457 priv->blocked = TRUE;
1458 unschedule_current_timer (jitterbuffer);
1460 if (ret != GST_STATE_CHANGE_FAILURE)
1461 ret = GST_STATE_CHANGE_NO_PREROLL;
1463 case GST_STATE_CHANGE_PAUSED_TO_READY:
1465 gst_buffer_replace (&priv->last_sr, NULL);
1466 priv->timer_running = FALSE;
1467 unschedule_current_timer (jitterbuffer);
1468 JBUF_SIGNAL_TIMER (priv);
1469 JBUF_SIGNAL_QUERY (priv, FALSE);
1471 g_thread_join (priv->timer_thread);
1472 priv->timer_thread = NULL;
1474 case GST_STATE_CHANGE_READY_TO_NULL:
1484 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1487 gboolean ret = TRUE;
1488 GstRtpJitterBuffer *jitterbuffer;
1489 GstRtpJitterBufferPrivate *priv;
1491 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1492 priv = jitterbuffer->priv;
1494 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1496 switch (GST_EVENT_TYPE (event)) {
1497 case GST_EVENT_LATENCY:
1499 GstClockTime latency;
1501 gst_event_parse_latency (event, &latency);
1503 GST_DEBUG_OBJECT (jitterbuffer,
1504 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1507 /* adjust the overall buffer delay to the total pipeline latency in
1508 * buffering mode because if downstream consumes too fast (because of
1509 * large latency or queues, we would start rebuffering again. */
1510 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1511 RTP_JITTER_BUFFER_MODE_BUFFER) {
1512 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1516 ret = gst_pad_push_event (priv->sinkpad, event);
1520 ret = gst_pad_push_event (priv->sinkpad, event);
1527 /* handles and stores the event in the jitterbuffer, must be called with
1530 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1532 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1533 RTPJitterBufferItem *item;
1536 switch (GST_EVENT_TYPE (event)) {
1537 case GST_EVENT_CAPS:
1541 gst_event_parse_caps (event, &caps);
1542 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1545 case GST_EVENT_SEGMENT:
1546 gst_event_copy_segment (event, &priv->segment);
1548 /* we need time for now */
1549 if (priv->segment.format != GST_FORMAT_TIME)
1550 goto newseg_wrong_format;
1552 GST_DEBUG_OBJECT (jitterbuffer,
1553 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1557 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1564 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1565 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1566 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1568 JBUF_SIGNAL_EVENT (priv);
1573 newseg_wrong_format:
1575 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1576 gst_event_unref (event);
1582 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1585 gboolean ret = TRUE;
1586 GstRtpJitterBuffer *jitterbuffer;
1587 GstRtpJitterBufferPrivate *priv;
1589 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1590 priv = jitterbuffer->priv;
1592 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1594 switch (GST_EVENT_TYPE (event)) {
1595 case GST_EVENT_FLUSH_START:
1596 ret = gst_pad_push_event (priv->srcpad, event);
1597 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1598 /* wait for the loop to go into PAUSED */
1599 gst_pad_pause_task (priv->srcpad);
1601 case GST_EVENT_FLUSH_STOP:
1602 ret = gst_pad_push_event (priv->srcpad, event);
1604 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1605 GST_PAD_MODE_PUSH, TRUE);
1608 if (GST_EVENT_IS_SERIALIZED (event)) {
1609 /* serialized events go in the queue */
1611 if (priv->srcresult != GST_FLOW_OK) {
1612 /* Errors in sticky event pushing are no problem and ignored here
1613 * as they will cause more meaningful errors during data flow.
1614 * For EOS events, that are not followed by data flow, we still
1615 * return FALSE here though.
1617 if (!GST_EVENT_IS_STICKY (event) ||
1618 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1619 goto out_flow_error;
1621 /* refuse more events on EOS */
1624 ret = queue_event (jitterbuffer, event);
1627 /* non-serialized events are forwarded downstream immediately */
1628 ret = gst_pad_push_event (priv->srcpad, event);
1637 GST_DEBUG_OBJECT (jitterbuffer,
1638 "refusing event, we have a downstream flow error: %s",
1639 gst_flow_get_name (priv->srcresult));
1641 gst_event_unref (event);
1646 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1648 gst_event_unref (event);
1654 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1657 gboolean ret = TRUE;
1658 GstRtpJitterBuffer *jitterbuffer;
1660 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1662 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1664 switch (GST_EVENT_TYPE (event)) {
1665 case GST_EVENT_FLUSH_START:
1666 gst_event_unref (event);
1668 case GST_EVENT_FLUSH_STOP:
1669 gst_event_unref (event);
1672 ret = gst_pad_event_default (pad, parent, event);
1680 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1681 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1682 * GST_FLOW_FLUSHING when the element is shutting down. On success
1683 * GST_FLOW_OK is returned.
1685 static GstFlowReturn
1686 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1690 GValue args[2] = { {0}, {0} };
1694 g_value_init (&args[0], GST_TYPE_ELEMENT);
1695 g_value_set_object (&args[0], jitterbuffer);
1696 g_value_init (&args[1], G_TYPE_UINT);
1697 g_value_set_uint (&args[1], pt);
1699 g_value_init (&ret, GST_TYPE_CAPS);
1700 g_value_set_boxed (&ret, NULL);
1702 JBUF_UNLOCK (jitterbuffer->priv);
1703 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1705 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1707 g_value_unset (&args[0]);
1708 g_value_unset (&args[1]);
1709 caps = (GstCaps *) g_value_dup_boxed (&ret);
1710 g_value_unset (&ret);
1714 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1715 gst_caps_unref (caps);
1717 if (G_UNLIKELY (!res))
1725 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1726 return GST_FLOW_ERROR;
1730 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1731 return GST_FLOW_FLUSHING;
1735 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1736 return GST_FLOW_ERROR;
1740 /* call with jbuf lock held */
1742 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1744 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1745 GstMessage *message = NULL;
1750 /* Post a buffering message */
1751 if (priv->last_percent != percent) {
1752 priv->last_percent = percent;
1754 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1755 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1762 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1764 GstRtpJitterBufferPrivate *priv;
1766 priv = jitterbuffer->priv;
1768 if (timestamp == -1)
1771 /* apply the timestamp offset, this is used for inter stream sync */
1772 timestamp += priv->ts_offset;
1773 /* add the offset, this is used when buffering */
1774 timestamp += priv->out_offset;
1780 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1782 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1783 TimerData *timer = NULL;
1786 len = priv->timers->len;
1787 for (i = 0; i < len; i++) {
1788 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1789 if (test->seqnum == seqnum && test->type == type) {
1798 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1800 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1802 if (priv->clock_id) {
1803 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1804 gst_clock_id_unschedule (priv->clock_id);
1805 priv->clock_id = NULL;
1810 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1812 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1813 GstClockTime test_timeout;
1815 if ((test_timeout = timer->timeout) == -1)
1818 if (timer->type != TIMER_TYPE_EXPECTED) {
1819 /* add our latency and offset to get output times. */
1820 test_timeout = apply_offset (jitterbuffer, test_timeout);
1821 test_timeout += priv->latency_ns;
1823 return test_timeout;
1827 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1829 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1831 if (priv->clock_id) {
1832 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1834 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1835 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1837 if (timeout == -1 || timeout < priv->timer_timeout)
1838 unschedule_current_timer (jitterbuffer);
1843 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1844 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1845 GstClockTime duration)
1847 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1851 GST_DEBUG_OBJECT (jitterbuffer,
1852 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1853 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1854 GST_TIME_ARGS (delay));
1856 len = priv->timers->len;
1857 g_array_set_size (priv->timers, len + 1);
1858 timer = &g_array_index (priv->timers, TimerData, len);
1861 timer->seqnum = seqnum;
1863 timer->timeout = timeout + delay;
1864 timer->duration = duration;
1865 if (type == TIMER_TYPE_EXPECTED) {
1866 timer->rtx_base = timeout;
1867 timer->rtx_delay = delay;
1868 timer->rtx_retry = 0;
1870 timer->num_rtx_retry = 0;
1871 recalculate_timer (jitterbuffer, timer);
1872 JBUF_SIGNAL_TIMER (priv);
1878 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1879 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1881 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1882 gboolean seqchange, timechange;
1885 seqchange = timer->seqnum != seqnum;
1886 timechange = timer->timeout != timeout;
1888 if (!seqchange && !timechange)
1891 oldseq = timer->seqnum;
1893 GST_DEBUG_OBJECT (jitterbuffer,
1894 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1895 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1897 timer->timeout = timeout + delay;
1898 timer->seqnum = seqnum;
1900 timer->rtx_base = timeout;
1901 timer->rtx_delay = delay;
1902 timer->rtx_retry = 0;
1905 timer->num_rtx_retry = 0;
1907 if (priv->clock_id) {
1908 /* we changed the seqnum and there is a timer currently waiting with this
1909 * seqnum, unschedule it */
1910 if (seqchange && priv->timer_seqnum == oldseq)
1911 unschedule_current_timer (jitterbuffer);
1912 /* we changed the time, check if it is earlier than what we are waiting
1913 * for and unschedule if so */
1914 else if (timechange)
1915 recalculate_timer (jitterbuffer, timer);
1920 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1921 guint16 seqnum, GstClockTime timeout)
1925 /* find the seqnum timer */
1926 timer = find_timer (jitterbuffer, type, seqnum);
1927 if (timer == NULL) {
1928 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1930 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1936 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1938 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1941 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1942 unschedule_current_timer (jitterbuffer);
1945 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1946 g_array_remove_index_fast (priv->timers, idx);
1951 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1953 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1954 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1955 g_array_set_size (priv->timers, 0);
1956 unschedule_current_timer (jitterbuffer);
1959 /* get the extra delay to wait before sending RTX */
1961 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1965 if (priv->rtx_delay == -1) {
1966 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
1967 delay = DEFAULT_AUTO_RTX_DELAY;
1969 /* jitter is in nanoseconds, maximum of 2x jitter and half the
1970 * packet spacing is a good margin */
1971 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
1974 delay = priv->rtx_delay * GST_MSECOND;
1976 if (priv->rtx_min_delay > 0)
1977 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1982 /* we just received a packet with seqnum and dts.
1984 * First check for old seqnum that we are still expecting. If the gap with the
1985 * current seqnum is too big, unschedule the timeouts.
1987 * If we have a valid packet spacing estimate we can set a timer for when we
1988 * should receive the next packet.
1989 * If we don't have a valid estimate, we remove any timer we might have
1990 * had for this packet.
1993 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1994 GstClockTime dts, gboolean do_next_seqnum)
1996 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1997 TimerData *timer = NULL;
2000 /* go through all timers and unschedule the ones with a large gap, also find
2001 * the timer for the seqnum */
2002 len = priv->timers->len;
2003 for (i = 0; i < len; i++) {
2004 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2007 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2009 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2010 test->type, test->seqnum, seqnum, gap);
2013 GST_DEBUG ("found timer for current seqnum");
2014 /* the timer for the current seqnum */
2016 /* when no retransmission, we can stop now, we only need to find the
2017 * timer for the current seqnum */
2018 if (!priv->do_retransmission)
2020 } else if (gap > priv->rtx_delay_reorder) {
2021 /* max gap, we exceeded the max reorder distance and we don't expect the
2022 * missing packet to be this reordered */
2023 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2024 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2028 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2029 && priv->do_retransmission && priv->rtx_next_seqnum;
2031 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2032 if (timer->num_rtx_retry > 0) {
2033 GstClockTime rtx_last, delay;
2035 /* we scheduled a retry for this packet and now we have it */
2036 priv->num_rtx_success++;
2037 /* all the previous retry attempts failed */
2038 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2039 /* number of retries before receiving the packet */
2040 if (priv->avg_rtx_num == 0.0)
2041 priv->avg_rtx_num = timer->num_rtx_retry;
2043 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2044 /* calculate the delay between retransmission request and receiving this
2045 * packet, start with when we scheduled this timeout last */
2046 rtx_last = timer->rtx_last;
2047 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2048 /* we have a valid delay if this packet arrived after we scheduled the
2050 delay = dts - rtx_last;
2051 if (priv->avg_rtx_rtt == 0)
2052 priv->avg_rtx_rtt = delay;
2054 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2058 GST_LOG_OBJECT (jitterbuffer,
2059 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2060 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2061 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2062 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2063 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2064 GST_TIME_ARGS (priv->avg_rtx_rtt));
2066 /* don't try to estimate the next seqnum because this is a retransmitted
2067 * packet and it probably did not arrive with the expected packet
2069 do_next_seqnum = FALSE;
2073 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2074 GstClockTime expected, delay;
2076 /* calculate expected arrival time of the next seqnum */
2077 expected = dts + priv->packet_spacing;
2079 delay = get_rtx_delay (priv);
2081 /* and update/install timer for next seqnum */
2083 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2086 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2087 expected, delay, priv->packet_spacing);
2089 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2090 /* if we had a timer, remove it, we don't know when to expect the next
2092 remove_timer (jitterbuffer, timer);
2097 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2100 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2102 /* we need consecutive seqnums with a different
2103 * rtptime to estimate the packet spacing. */
2104 if (priv->ips_rtptime != rtptime) {
2105 /* rtptime changed, check dts diff */
2106 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2107 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2108 GstClockTime old_packet_spacing = priv->packet_spacing;
2110 /* Biased towards bigger packet spacings to prevent
2111 * too many unneeded retransmission requests for next
2112 * packets that just arrive a little later than we would
2114 if (old_packet_spacing > new_packet_spacing)
2115 priv->packet_spacing =
2116 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2117 else if (old_packet_spacing > 0)
2118 priv->packet_spacing =
2119 (3 * new_packet_spacing + old_packet_spacing) / 4;
2121 priv->packet_spacing = new_packet_spacing;
2123 GST_DEBUG_OBJECT (jitterbuffer,
2124 "new packet spacing %" GST_TIME_FORMAT
2125 " old packet spacing %" GST_TIME_FORMAT
2126 " combined to %" GST_TIME_FORMAT,
2127 GST_TIME_ARGS (new_packet_spacing),
2128 GST_TIME_ARGS (old_packet_spacing),
2129 GST_TIME_ARGS (priv->packet_spacing));
2131 priv->ips_rtptime = rtptime;
2132 priv->ips_dts = dts;
2137 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2138 guint16 seqnum, GstClockTime dts, gint gap)
2140 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2141 GstClockTime total_duration, duration, expected_dts;
2144 GST_DEBUG_OBJECT (jitterbuffer,
2145 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2146 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2148 if (dts == GST_CLOCK_TIME_NONE) {
2149 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2153 /* the total duration spanned by the missing packets */
2154 if (dts >= priv->last_in_dts)
2155 total_duration = dts - priv->last_in_dts;
2159 /* interpolate between the current time and the last time based on
2160 * number of packets we are missing, this is the estimated duration
2161 * for the missing packet based on equidistant packet spacing. */
2162 duration = total_duration / (gap + 1);
2164 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2165 GST_TIME_ARGS (duration));
2167 if (total_duration > priv->latency_ns) {
2168 GstClockTime gap_time;
2172 GstClockTime gap_dur = gap * duration;
2173 if (gap_dur > priv->latency_ns)
2174 gap_time = gap_dur - priv->latency_ns;
2177 lost_packets = gap_time / duration;
2179 gap_time = total_duration - priv->latency_ns;
2183 /* too many lost packets, some of the missing packets are already
2184 * too late and we can generate lost packet events for them. */
2185 GST_DEBUG_OBJECT (jitterbuffer,
2186 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2187 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2188 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2189 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2190 GST_TIME_ARGS (gap_time));
2192 /* this timer will fire immediately and the lost event will be pushed from
2193 * the timer thread */
2194 if (lost_packets > 0) {
2195 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2196 priv->last_in_dts + duration, 0, gap_time);
2197 expected += lost_packets;
2198 priv->last_in_dts += gap_time;
2202 expected_dts = priv->last_in_dts + duration;
2204 if (priv->do_retransmission) {
2207 type = TIMER_TYPE_EXPECTED;
2208 /* if we had a timer for the first missing packet, update it. */
2209 if ((timer = find_timer (jitterbuffer, type, expected))) {
2210 GstClockTime timeout = timer->timeout;
2212 timer->duration = duration;
2213 if (timeout > (expected_dts + timer->rtx_retry)) {
2214 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2215 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2219 expected_dts += duration;
2222 type = TIMER_TYPE_LOST;
2225 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2226 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2227 expected_dts += duration;
2233 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2237 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2238 GstRtpJitterBufferPrivate *priv;
2240 priv = jitterbuffer->priv;
2242 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2245 if (priv->last_dts != -1)
2246 dtsdiff = dts - priv->last_dts;
2250 if (priv->last_rtptime != -1)
2251 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2255 priv->last_dts = dts;
2256 priv->last_rtptime = rtptime;
2260 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2263 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2265 diff = ABS (dtsdiff - rtpdiffns);
2267 /* jitter is stored in nanoseconds */
2268 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2270 GST_LOG_OBJECT (jitterbuffer,
2271 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2272 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2273 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2274 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2281 GST_DEBUG_OBJECT (jitterbuffer,
2282 "no dts or no clock-rate, can't calculate jitter");
2288 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2290 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2291 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2294 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2295 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2296 gst_rtp_buffer_unmap (&rtp_a);
2298 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2299 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2300 gst_rtp_buffer_unmap (&rtp_b);
2302 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2306 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2307 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap)
2309 GstRtpJitterBufferPrivate *priv;
2310 guint gap_packets_length;
2311 gboolean reset = FALSE;
2313 priv = jitterbuffer->priv;
2315 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2317 guint32 prev_gap_seq = -1;
2318 gboolean all_consecutive = TRUE;
2320 g_queue_insert_sorted (&priv->gap_packets, buffer,
2321 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2323 for (l = priv->gap_packets.head; l; l = l->next) {
2324 GstBuffer *gap_buffer = l->data;
2325 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2328 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2330 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2332 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2333 if (prev_gap_seq == -1)
2334 prev_gap_seq = gap_seq;
2335 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2336 all_consecutive = FALSE;
2338 prev_gap_seq = gap_seq;
2340 gst_rtp_buffer_unmap (&gap_rtp);
2341 if (!all_consecutive)
2345 if (all_consecutive && gap_packets_length > 3) {
2346 GST_DEBUG_OBJECT (jitterbuffer,
2347 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2348 (future ? "new" : "old"), gap,
2349 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2351 } else if (!all_consecutive) {
2352 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2353 g_queue_clear (&priv->gap_packets);
2354 GST_DEBUG_OBJECT (jitterbuffer,
2355 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2356 (future ? "new" : "old"), gap,
2357 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2360 GST_DEBUG_OBJECT (jitterbuffer,
2361 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2362 (future ? "new" : "old"), gap,
2363 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER),
2364 gap_packets_length + 1);
2368 GST_DEBUG_OBJECT (jitterbuffer,
2369 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2370 gap, -RTP_MAX_MISORDER);
2371 g_queue_push_tail (&priv->gap_packets, buffer);
2379 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2381 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2382 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2385 GstClockTime base_time =
2386 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2387 GstClockTime clock_time = gst_clock_get_time (clock);
2389 if (clock_time > base_time)
2390 running_time = clock_time - base_time;
2394 gst_object_unref (clock);
2397 return running_time;
2400 static GstFlowReturn
2401 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2404 GstRtpJitterBuffer *jitterbuffer;
2405 GstRtpJitterBufferPrivate *priv;
2407 guint32 expected, rtptime;
2408 GstFlowReturn ret = GST_FLOW_OK;
2409 GstClockTime dts, pts;
2414 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2415 gboolean do_next_seqnum = FALSE;
2416 RTPJitterBufferItem *item;
2417 GstMessage *msg = NULL;
2418 gboolean estimated_dts = FALSE;
2420 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2422 priv = jitterbuffer->priv;
2424 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2425 goto invalid_buffer;
2427 pt = gst_rtp_buffer_get_payload_type (&rtp);
2428 seqnum = gst_rtp_buffer_get_seq (&rtp);
2429 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2430 gst_rtp_buffer_unmap (&rtp);
2432 /* make sure we have PTS and DTS set */
2433 pts = GST_BUFFER_PTS (buffer);
2434 dts = GST_BUFFER_DTS (buffer);
2441 /* If we have no DTS here, i.e. no capture time, get one from the
2442 * clock now to have something to calculate with in the future. */
2443 dts = get_current_running_time (jitterbuffer);
2446 /* Remember that we estimated the DTS if we are running already
2447 * and this is not our first packet (or first packet after a reset).
2448 * If it's the first packet, we somehow must generate a timestamp for
2449 * everything, otherwise we can't calculate any times
2451 estimated_dts = (priv->next_in_seqnum != -1);
2453 /* take the DTS of the buffer. This is the time when the packet was
2454 * received and is used to calculate jitter and clock skew. We will adjust
2455 * this DTS with the smoothed value after processing it in the
2456 * jitterbuffer and assign it as the PTS. */
2457 /* bring to running time */
2458 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2461 GST_DEBUG_OBJECT (jitterbuffer,
2462 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2463 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2465 JBUF_LOCK_CHECK (priv, out_flushing);
2467 if (G_UNLIKELY (priv->last_pt != pt)) {
2470 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2474 /* reset clock-rate so that we get a new one */
2475 priv->clock_rate = -1;
2477 /* Try to get the clock-rate from the caps first if we can. If there are no
2478 * caps we must fire the signal to get the clock-rate. */
2479 if ((caps = gst_pad_get_current_caps (pad))) {
2480 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2481 gst_caps_unref (caps);
2485 if (G_UNLIKELY (priv->clock_rate == -1)) {
2486 /* no clock rate given on the caps, try to get one with the signal */
2487 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2488 pt) == GST_FLOW_FLUSHING)
2491 if (G_UNLIKELY (priv->clock_rate == -1))
2495 /* don't accept more data on EOS */
2496 if (G_UNLIKELY (priv->eos))
2499 calculate_jitter (jitterbuffer, dts, rtptime);
2501 if (priv->seqnum_base != -1) {
2504 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2507 GST_DEBUG_OBJECT (jitterbuffer,
2508 "packet seqnum #%d before seqnum-base #%d", seqnum,
2510 gst_buffer_unref (buffer);
2513 } else if (gap > 16384) {
2514 /* From now on don't compare against the seqnum base anymore as
2515 * at some point in the future we will wrap around and also that
2516 * much reordering is very unlikely */
2517 priv->seqnum_base = -1;
2521 expected = priv->next_in_seqnum;
2523 /* now check against our expected seqnum */
2524 if (G_LIKELY (expected != -1)) {
2527 /* now calculate gap */
2528 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2530 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2531 expected, seqnum, gap);
2533 if (G_LIKELY (gap == 0)) {
2534 /* packet is expected */
2535 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2536 do_next_seqnum = TRUE;
2538 gboolean reset = FALSE;
2541 /* we received an old packet */
2542 if (G_UNLIKELY (gap != -1 && gap < -RTP_MAX_MISORDER)) {
2544 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2548 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2551 /* new packet, we are missing some packets */
2552 if (G_UNLIKELY (priv->timers->len >= RTP_MAX_DROPOUT)) {
2553 /* If we have timers for more than RTP_MAX_DROPOUT packets
2554 * pending this means that we have a huge gap overall. We can
2555 * reset the jitterbuffer at this point because there's
2556 * just too much data missing to be able to do anything
2557 * sensible with the past data. Just try again from the
2559 GST_WARNING_OBJECT (jitterbuffer,
2560 "%d pending timers > %d - resetting", priv->timers->len,
2563 gst_buffer_unref (buffer);
2565 } else if (G_UNLIKELY (gap >= RTP_MAX_DROPOUT)) {
2567 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2571 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2572 /* fill in the gap with EXPECTED timers */
2573 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2575 do_next_seqnum = TRUE;
2578 if (G_UNLIKELY (reset)) {
2579 GList *events = NULL, *l;
2582 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2583 rtp_jitter_buffer_flush (priv->jbuf,
2584 (GFunc) free_item_and_retain_events, &events);
2585 rtp_jitter_buffer_reset_skew (priv->jbuf);
2586 remove_all_timers (jitterbuffer);
2587 priv->discont = TRUE;
2588 priv->last_popped_seqnum = -1;
2589 priv->next_seqnum = seqnum;
2591 priv->last_in_dts = -1;
2592 priv->next_in_seqnum = -1;
2594 /* Insert all sticky events again in order, otherwise we would
2595 * potentially loose STREAM_START, CAPS or SEGMENT events
2597 events = g_list_reverse (events);
2598 for (l = events; l; l = l->next) {
2599 RTPJitterBufferItem *item;
2601 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2602 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2604 g_list_free (events);
2606 JBUF_SIGNAL_EVENT (priv);
2608 /* reset spacing estimation when gap */
2609 priv->ips_rtptime = -1;
2610 priv->ips_dts = GST_CLOCK_TIME_NONE;
2612 buffers = g_list_copy (priv->gap_packets.head);
2613 g_queue_clear (&priv->gap_packets);
2615 priv->ips_rtptime = -1;
2616 priv->ips_dts = GST_CLOCK_TIME_NONE;
2617 JBUF_UNLOCK (jitterbuffer->priv);
2619 for (l = buffers; l; l = l->next) {
2620 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2622 if (ret != GST_FLOW_OK)
2625 for (; l; l = l->next)
2626 gst_buffer_unref (l->data);
2627 g_list_free (buffers);
2631 /* reset spacing estimation when gap */
2632 priv->ips_rtptime = -1;
2633 priv->ips_dts = GST_CLOCK_TIME_NONE;
2636 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2638 /* we don't know what the next_in_seqnum should be, wait for the last
2639 * possible moment to push this buffer, maybe we get an earlier seqnum
2641 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2642 do_next_seqnum = TRUE;
2643 /* take rtptime and dts to calculate packet spacing */
2644 priv->ips_rtptime = rtptime;
2645 priv->ips_dts = dts;
2648 /* We had no huge gap, let's drop all the gap packets */
2649 if (buffer != NULL) {
2650 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2651 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2652 g_queue_clear (&priv->gap_packets);
2654 GST_DEBUG_OBJECT (jitterbuffer,
2655 "Had big gap, waiting for more consecutive packets");
2656 JBUF_UNLOCK (jitterbuffer->priv);
2660 if (do_next_seqnum) {
2661 priv->last_in_dts = dts;
2662 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2665 /* let's check if this buffer is too late, we can only accept packets with
2666 * bigger seqnum than the one we last pushed. */
2667 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2670 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2672 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2673 if (G_UNLIKELY (gap <= 0))
2677 /* let's drop oldest packet if the queue is already full and drop-on-latency
2678 * is set. We can only do this when there actually is a latency. When no
2679 * latency is set, we just pump it in the queue and let the other end push it
2680 * out as fast as possible. */
2681 if (priv->latency_ms && priv->drop_on_latency) {
2683 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2685 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2686 RTPJitterBufferItem *old_item;
2688 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2690 if (IS_DROPABLE (old_item)) {
2691 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2692 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2694 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2695 free_item (old_item);
2697 /* we might have removed some head buffers, signal the pushing thread to
2698 * see if it can push now */
2699 JBUF_SIGNAL_EVENT (priv);
2703 /* If we estimated the DTS, don't consider it in the clock skew calculations
2704 * later. The code above always sets dts to pts or the other way around if
2705 * any of those is valid in the buffer, so we know that if we estimated the
2706 * dts that both are unknown */
2709 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2710 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2712 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2714 /* now insert the packet into the queue in sorted order. This function returns
2715 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2716 * have a duplicate. */
2717 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2722 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2724 /* we had an unhandled SR, handle it now */
2726 do_handle_sync (jitterbuffer);
2728 if (G_UNLIKELY (head)) {
2729 /* signal addition of new buffer when the _loop is waiting. */
2730 if (G_LIKELY (priv->active))
2731 JBUF_SIGNAL_EVENT (priv);
2733 /* let's unschedule and unblock any waiting buffers. We only want to do this
2734 * when the head buffer changed */
2735 if (G_UNLIKELY (priv->clock_id)) {
2736 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2737 unschedule_current_timer (jitterbuffer);
2741 GST_DEBUG_OBJECT (jitterbuffer,
2742 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2743 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2745 msg = check_buffering_percent (jitterbuffer, percent);
2751 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2758 /* this is not fatal but should be filtered earlier */
2759 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2760 ("Received invalid RTP payload, dropping"));
2761 gst_buffer_unref (buffer);
2766 GST_WARNING_OBJECT (jitterbuffer,
2767 "No clock-rate in caps!, dropping buffer");
2768 gst_buffer_unref (buffer);
2773 ret = priv->srcresult;
2774 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2775 gst_buffer_unref (buffer);
2781 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2782 gst_buffer_unref (buffer);
2787 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2788 " popped, dropping", seqnum, priv->last_popped_seqnum);
2790 gst_buffer_unref (buffer);
2795 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2797 priv->num_duplicates++;
2804 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2806 guint64 ext_time, elapsed;
2808 GstRtpJitterBufferPrivate *priv;
2810 priv = jitterbuffer->priv;
2811 rtp_time = item->rtptime;
2813 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2814 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2816 ext_time = priv->ext_timestamp;
2817 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2818 if (ext_time < priv->ext_timestamp) {
2819 ext_time = priv->ext_timestamp;
2821 priv->ext_timestamp = ext_time;
2824 if (ext_time > priv->clock_base)
2825 elapsed = ext_time - priv->clock_base;
2829 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2834 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2835 RTPJitterBufferItem * item)
2837 guint64 total, elapsed, left, estimated;
2838 GstClockTime out_time;
2839 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2841 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2842 || priv->clock_base == -1 || priv->clock_rate <= 0)
2845 /* compute the elapsed time */
2846 elapsed = compute_elapsed (jitterbuffer, item);
2848 /* do nothing if elapsed time doesn't increment */
2849 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2852 priv->last_elapsed = elapsed;
2854 /* this is the total time we need to play */
2855 total = priv->npt_stop - priv->npt_start;
2856 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2857 GST_TIME_ARGS (total));
2859 /* this is how much time there is left */
2860 if (total > elapsed)
2861 left = total - elapsed;
2865 /* if we have less time left that the size of the buffer, we will not
2866 * be able to keep it filled, disabled buffering then */
2867 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2868 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2869 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2870 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2873 /* this is the current time as running-time */
2874 out_time = item->dts;
2877 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2879 /* if there is almost nothing left,
2880 * we may never advance enough to end up in the above case */
2881 if (total < GST_SECOND)
2882 estimated = GST_SECOND;
2886 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2887 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2889 if (estimated != -1 && priv->estimated_eos != estimated) {
2890 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2891 priv->estimated_eos = estimated;
2895 /* take a buffer from the queue and push it */
2896 static GstFlowReturn
2897 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2899 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2900 GstFlowReturn result = GST_FLOW_OK;
2901 RTPJitterBufferItem *item;
2902 GstBuffer *outbuf = NULL;
2903 GstEvent *outevent = NULL;
2904 GstQuery *outquery = NULL;
2905 GstClockTime dts, pts;
2907 gboolean do_push = TRUE;
2911 /* when we get here we are ready to pop and push the buffer */
2912 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2916 case ITEM_TYPE_BUFFER:
2918 /* we need to make writable to change the flags and timestamps */
2919 outbuf = gst_buffer_make_writable (item->data);
2921 if (G_UNLIKELY (priv->discont)) {
2922 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2923 * into the jitterbuffer so we can modify now. */
2924 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2925 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2926 priv->discont = FALSE;
2928 if (G_UNLIKELY (priv->ts_discont)) {
2929 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2930 priv->ts_discont = FALSE;
2934 gst_segment_position_from_running_time (&priv->segment,
2935 GST_FORMAT_TIME, item->dts);
2937 gst_segment_position_from_running_time (&priv->segment,
2938 GST_FORMAT_TIME, item->pts);
2940 /* apply timestamp with offset to buffer now */
2941 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2942 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2944 /* update the elapsed time when we need to check against the npt stop time. */
2945 update_estimated_eos (jitterbuffer, item);
2947 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2949 case ITEM_TYPE_LOST:
2950 priv->discont = TRUE;
2954 case ITEM_TYPE_EVENT:
2955 outevent = item->data;
2957 case ITEM_TYPE_QUERY:
2958 outquery = item->data;
2962 /* now we are ready to push the buffer. Save the seqnum and release the lock
2963 * so the other end can push stuff in the queue again. */
2965 priv->last_popped_seqnum = seqnum;
2966 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2968 msg = check_buffering_percent (jitterbuffer, percent);
2975 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2978 case ITEM_TYPE_BUFFER:
2980 GST_DEBUG_OBJECT (jitterbuffer,
2981 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2982 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2983 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2984 result = gst_pad_push (priv->srcpad, outbuf);
2986 JBUF_LOCK_CHECK (priv, out_flushing);
2988 case ITEM_TYPE_LOST:
2989 case ITEM_TYPE_EVENT:
2990 /* We got not enough consecutive packets with a huge gap, we can
2991 * as well just drop them here now on EOS */
2992 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
2993 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
2994 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2995 g_queue_clear (&priv->gap_packets);
2998 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2999 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3002 gst_pad_push_event (priv->srcpad, outevent);
3004 gst_event_unref (outevent);
3006 result = GST_FLOW_OK;
3008 JBUF_LOCK_CHECK (priv, out_flushing);
3010 case ITEM_TYPE_QUERY:
3014 res = gst_pad_peer_query (priv->srcpad, outquery);
3016 JBUF_LOCK_CHECK (priv, out_flushing);
3017 result = GST_FLOW_OK;
3018 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3019 JBUF_SIGNAL_QUERY (priv, res);
3028 return priv->srcresult;
3032 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3034 /* Peek a buffer and compare the seqnum to the expected seqnum.
3035 * If all is fine, the buffer is pushed.
3036 * If something is wrong, we wait for some event
3038 static GstFlowReturn
3039 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3041 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3042 GstFlowReturn result;
3043 RTPJitterBufferItem *item;
3045 guint32 next_seqnum;
3047 /* only push buffers when PLAYING and active and not buffering */
3048 if (priv->blocked || !priv->active ||
3049 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3050 return GST_FLOW_WAIT;
3053 /* peek a buffer, we're just looking at the sequence number.
3054 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3055 * wait for a timeout or something to change.
3056 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3057 item = rtp_jitter_buffer_peek (priv->jbuf);
3062 /* get the seqnum and the next expected seqnum */
3063 seqnum = item->seqnum;
3065 return pop_and_push_next (jitterbuffer, seqnum);
3068 next_seqnum = priv->next_seqnum;
3070 /* get the gap between this and the previous packet. If we don't know the
3071 * previous packet seqnum assume no gap. */
3072 if (G_UNLIKELY (next_seqnum == -1)) {
3073 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3074 /* we don't know what the next_seqnum should be, the chain function should
3075 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3076 * fires, so wait for that */
3077 result = GST_FLOW_WAIT;
3079 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3081 if (G_LIKELY (gap == 0)) {
3082 /* no missing packet, pop and push */
3083 result = pop_and_push_next (jitterbuffer, seqnum);
3084 } else if (G_UNLIKELY (gap < 0)) {
3085 /* if we have a packet that we already pushed or considered dropped, pop it
3086 * off and get the next packet */
3087 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3088 seqnum, next_seqnum);
3089 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3091 result = GST_FLOW_OK;
3093 /* the chain function has scheduled timers to request retransmission or
3094 * when to consider the packet lost, wait for that */
3095 GST_DEBUG_OBJECT (jitterbuffer,
3096 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3097 next_seqnum, seqnum, gap);
3098 result = GST_FLOW_WAIT;
3106 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3108 return GST_FLOW_EOS;
3110 return GST_FLOW_WAIT;
3116 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3118 GstClockTime rtx_retry_timeout;
3119 GstClockTime rtx_min_retry_timeout;
3121 if (priv->rtx_retry_timeout == -1) {
3122 if (priv->avg_rtx_rtt == 0)
3123 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3125 /* we want to ask for a retransmission after we waited for a
3126 * complete RTT and the additional jitter */
3127 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3129 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3131 /* make sure we don't retry too often. On very low latency networks,
3132 * the RTT and jitter can be very low. */
3133 if (priv->rtx_min_retry_timeout == -1) {
3134 rtx_min_retry_timeout = priv->packet_spacing;
3136 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3138 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3140 return rtx_retry_timeout;
3144 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3145 GstClockTime rtx_retry_timeout)
3147 GstClockTime rtx_retry_period;
3149 if (priv->rtx_retry_period == -1) {
3150 /* we retry up to the configured jitterbuffer size but leaving some
3151 * room for the retransmission to arrive in time */
3152 if (rtx_retry_timeout > priv->latency_ns) {
3153 rtx_retry_period = 0;
3155 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3158 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3160 return rtx_retry_period;
3163 /* the timeout for when we expected a packet expired */
3165 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3168 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3170 guint delay, delay_ms, avg_rtx_rtt_ms;
3171 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3172 GstClockTime rtx_retry_period;
3173 GstClockTime rtx_retry_timeout;
3176 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3177 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3179 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3180 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3182 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3183 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3184 GST_TIME_ARGS (rtx_retry_period));
3186 delay = timer->rtx_delay + timer->rtx_retry;
3188 delay_ms = GST_TIME_AS_MSECONDS (delay);
3189 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3190 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3191 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3193 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3194 gst_structure_new ("GstRTPRetransmissionRequest",
3195 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3196 "running-time", G_TYPE_UINT64, timer->rtx_base,
3197 "delay", G_TYPE_UINT, delay_ms,
3198 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3199 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3200 "period", G_TYPE_UINT, rtx_retry_period_ms,
3201 "deadline", G_TYPE_UINT, priv->latency_ms,
3202 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3203 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3205 priv->num_rtx_requests++;
3206 timer->num_rtx_retry++;
3208 GST_OBJECT_LOCK (jitterbuffer);
3209 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3210 timer->rtx_last = gst_clock_get_time (clock);
3211 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3213 timer->rtx_last = now;
3215 GST_OBJECT_UNLOCK (jitterbuffer);
3217 /* calculate the timeout for the next retransmission attempt */
3218 timer->rtx_retry += rtx_retry_timeout;
3219 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3220 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3221 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3222 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3223 if ((priv->rtx_max_retries != -1
3224 && timer->num_rtx_retry >= priv->rtx_max_retries)
3225 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3226 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3227 /* too many retransmission request, we now convert the timer
3228 * to a lost timer, leave the num_rtx_retry as it is for stats */
3229 timer->type = TIMER_TYPE_LOST;
3230 timer->rtx_delay = 0;
3231 timer->rtx_retry = 0;
3233 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3234 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3237 gst_pad_push_event (priv->sinkpad, event);
3243 /* a packet is lost */
3245 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3248 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3249 GstClockTime duration, timestamp;
3250 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3253 RTPJitterBufferItem *item;
3255 seqnum = timer->seqnum;
3256 timestamp = apply_offset (jitterbuffer, timer->timeout);
3257 duration = timer->duration;
3258 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3259 duration = priv->packet_spacing;
3260 lost_packets = MAX (timer->num, 1);
3261 num_rtx_retry = timer->num_rtx_retry;
3263 /* we had a gap and thus we lost some packets. Create an event for this. */
3264 if (lost_packets > 1)
3265 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3266 seqnum + lost_packets - 1);
3268 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3270 priv->num_late += lost_packets;
3271 priv->num_rtx_failed += num_rtx_retry;
3273 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3275 /* we now only accept seqnum bigger than this */
3276 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3277 priv->next_in_seqnum = next_in_seqnum;
3279 /* create paket lost event */
3280 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3281 gst_structure_new ("GstRTPPacketLost",
3282 "seqnum", G_TYPE_UINT, (guint) seqnum,
3283 "timestamp", G_TYPE_UINT64, timestamp,
3284 "duration", G_TYPE_UINT64, duration,
3285 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3287 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3288 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3290 /* remove timer now */
3291 remove_timer (jitterbuffer, timer);
3293 JBUF_SIGNAL_EVENT (priv);
3299 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3302 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3304 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3305 remove_timer (jitterbuffer, timer);
3307 /* there was no EOS in the buffer, put one in there now */
3308 queue_event (jitterbuffer, gst_event_new_eos ());
3310 JBUF_SIGNAL_EVENT (priv);
3316 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3319 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3321 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3323 /* timer seqnum might have been obsoleted by caps seqnum-base,
3324 * only mess with current ongoing seqnum if still unknown */
3325 if (priv->next_seqnum == -1)
3326 priv->next_seqnum = timer->seqnum;
3327 remove_timer (jitterbuffer, timer);
3328 JBUF_SIGNAL_EVENT (priv);
3334 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3337 gboolean removed = FALSE;
3339 switch (timer->type) {
3340 case TIMER_TYPE_EXPECTED:
3341 removed = do_expected_timeout (jitterbuffer, timer, now);
3343 case TIMER_TYPE_LOST:
3344 removed = do_lost_timeout (jitterbuffer, timer, now);
3346 case TIMER_TYPE_DEADLINE:
3347 removed = do_deadline_timeout (jitterbuffer, timer, now);
3349 case TIMER_TYPE_EOS:
3350 removed = do_eos_timeout (jitterbuffer, timer, now);
3356 /* called when we need to wait for the next timeout.
3358 * We loop over the array of recorded timeouts and wait for the earliest one.
3359 * When it timed out, do the logic associated with the timer.
3361 * If there are no timers, we wait on a gcond until something new happens.
3364 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3366 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3367 GstClockTime now = 0;
3370 while (priv->timer_running) {
3371 TimerData *timer = NULL;
3372 GstClockTime timer_timeout = -1;
3375 /* If we have a clock, update "now" now with the very latest running time
3376 * we have. It is used below when timeouts are triggered to calculate
3377 * any next possible timeout. If we only update it after waiting for the
3378 * clock, we would give a too old time to the timeout functions.
3380 GST_OBJECT_LOCK (jitterbuffer);
3381 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3383 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3384 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3386 GST_OBJECT_UNLOCK (jitterbuffer);
3388 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3389 GST_TIME_ARGS (now));
3391 len = priv->timers->len;
3392 for (i = 0; i < len; i++) {
3393 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3394 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3395 gboolean save_best = FALSE;
3397 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3398 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3400 /* find the smallest timeout */
3401 if (timer == NULL) {
3403 } else if (timer_timeout == -1) {
3404 /* we already have an immediate timeout, the new timer must be an
3405 * immediate timer with smaller seqnum to become the best */
3406 if (test_timeout == -1
3407 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3408 timer->seqnum) > 0))
3410 } else if (test_timeout == -1) {
3411 /* first immediate timer */
3413 } else if (test_timeout < timer_timeout) {
3416 } else if (test_timeout == timer_timeout
3417 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3418 timer->seqnum) > 0)) {
3419 /* same timer, smaller seqnum */
3423 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3425 timer_timeout = test_timeout;
3428 if (timer && !priv->blocked) {
3430 GstClockTime sync_time;
3433 GstClockTimeDiff clock_jitter;
3435 if (timer_timeout == -1 || timer_timeout <= now) {
3436 do_timeout (jitterbuffer, timer, now);
3437 /* check here, do_timeout could have released the lock */
3438 if (!priv->timer_running)
3443 GST_OBJECT_LOCK (jitterbuffer);
3444 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3446 GST_OBJECT_UNLOCK (jitterbuffer);
3447 /* let's just push if there is no clock */
3448 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3449 now = timer_timeout;
3453 /* prepare for sync against clock */
3454 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3455 /* add latency of peer to get input time */
3456 sync_time += priv->peer_latency;
3458 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3459 " with sync time %" GST_TIME_FORMAT,
3460 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3462 /* create an entry for the clock */
3463 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3464 priv->timer_timeout = timer_timeout;
3465 priv->timer_seqnum = timer->seqnum;
3466 GST_OBJECT_UNLOCK (jitterbuffer);
3468 /* release the lock so that the other end can push stuff or unlock */
3471 ret = gst_clock_id_wait (id, &clock_jitter);
3474 if (!priv->timer_running) {
3475 gst_clock_id_unref (id);
3476 priv->clock_id = NULL;
3480 if (ret != GST_CLOCK_UNSCHEDULED) {
3481 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3482 ret, priv->timer_seqnum, clock_jitter);
3484 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3486 /* and free the entry */
3487 gst_clock_id_unref (id);
3488 priv->clock_id = NULL;
3490 /* no timers, wait for activity */
3491 JBUF_WAIT_TIMER (priv);
3496 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3501 * This funcion implements the main pushing loop on the source pad.
3503 * It first tries to push as many buffers as possible. If there is a seqnum
3504 * mismatch, we wait for the next timeouts.
3507 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3509 GstRtpJitterBufferPrivate *priv;
3510 GstFlowReturn result = GST_FLOW_OK;
3512 priv = jitterbuffer->priv;
3514 JBUF_LOCK_CHECK (priv, flushing);
3516 result = handle_next_buffer (jitterbuffer);
3517 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3518 /* now wait for the next event */
3519 JBUF_WAIT_EVENT (priv, flushing);
3520 result = GST_FLOW_OK;
3522 } while (result == GST_FLOW_OK);
3523 /* store result for upstream */
3524 priv->srcresult = result;
3525 /* if we get here we need to pause */
3531 result = priv->srcresult;
3538 JBUF_SIGNAL_QUERY (priv, FALSE);
3541 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3542 gst_flow_get_name (result));
3543 gst_pad_pause_task (priv->srcpad);
3544 if (result == GST_FLOW_EOS) {
3545 event = gst_event_new_eos ();
3546 gst_pad_push_event (priv->srcpad, event);
3552 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3553 * some sanity checks and then emit the handle-sync signal with the parameters.
3554 * This function must be called with the LOCK */
3556 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3558 GstRtpJitterBufferPrivate *priv;
3559 guint64 base_rtptime, base_time;
3561 guint64 last_rtptime;
3563 guint64 ext_rtptime, diff;
3564 gboolean valid = TRUE, keep = FALSE;
3566 priv = jitterbuffer->priv;
3568 /* get the last values from the jitterbuffer */
3569 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3570 &clock_rate, &last_rtptime);
3572 clock_base = priv->clock_base;
3573 ext_rtptime = priv->ext_rtptime;
3575 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3576 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3577 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3578 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3580 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3581 /* we keep this SR packet for later. When we get a valid RTP packet the
3582 * above values will be set and we can try to use the SR packet */
3583 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3586 /* we can't accept anything that happened before we did the last resync */
3587 if (base_rtptime > ext_rtptime) {
3588 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3591 /* the SR RTP timestamp must be something close to what we last observed
3592 * in the jitterbuffer */
3593 if (ext_rtptime > last_rtptime) {
3594 /* check how far ahead it is to our RTP timestamps */
3595 diff = ext_rtptime - last_rtptime;
3596 /* if bigger than 1 second, we drop it */
3597 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3599 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3600 clock_rate, 1000)) {
3601 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3602 /* should drop this, but some RTSP servers end up with bogus
3603 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3604 * so still trigger rptbin sync but invalidate RTCP data
3605 * (sync might use other methods) */
3608 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3609 G_GUINT64_FORMAT, last_rtptime, diff);
3615 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3619 s = gst_structure_new ("application/x-rtp-sync",
3620 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3621 "base-time", G_TYPE_UINT64, base_time,
3622 "clock-rate", G_TYPE_UINT, clock_rate,
3623 "clock-base", G_TYPE_UINT64, clock_base,
3624 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3625 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3627 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3628 gst_buffer_replace (&priv->last_sr, NULL);
3630 g_signal_emit (jitterbuffer,
3631 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3633 gst_structure_free (s);
3635 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3636 gst_buffer_replace (&priv->last_sr, NULL);
3640 static GstFlowReturn
3641 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3644 GstRtpJitterBuffer *jitterbuffer;
3645 GstRtpJitterBufferPrivate *priv;
3646 GstFlowReturn ret = GST_FLOW_OK;
3648 GstRTCPPacket packet;
3649 guint64 ext_rtptime;
3651 GstRTCPBuffer rtcp = { NULL, };
3653 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3655 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3656 goto invalid_buffer;
3658 priv = jitterbuffer->priv;
3660 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3662 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3665 /* first packet must be SR or RR or else the validate would have failed */
3666 switch (gst_rtcp_packet_get_type (&packet)) {
3667 case GST_RTCP_TYPE_SR:
3668 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3674 gst_rtcp_buffer_unmap (&rtcp);
3676 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3679 /* convert the RTP timestamp to our extended timestamp, using the same offset
3680 * we used in the jitterbuffer */
3681 ext_rtptime = priv->jbuf->ext_rtptime;
3682 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3684 priv->ext_rtptime = ext_rtptime;
3685 gst_buffer_replace (&priv->last_sr, buffer);
3687 do_handle_sync (jitterbuffer);
3691 gst_buffer_unref (buffer);
3697 /* this is not fatal but should be filtered earlier */
3698 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3699 ("Received invalid RTCP payload, dropping"));
3705 /* this is not fatal but should be filtered earlier */
3706 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3707 ("Received empty RTCP payload, dropping"));
3708 gst_rtcp_buffer_unmap (&rtcp);
3714 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3715 gst_rtcp_buffer_unmap (&rtcp);
3722 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3725 gboolean res = FALSE;
3726 GstRtpJitterBuffer *jitterbuffer;
3727 GstRtpJitterBufferPrivate *priv;
3729 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3730 priv = jitterbuffer->priv;
3732 switch (GST_QUERY_TYPE (query)) {
3733 case GST_QUERY_CAPS:
3735 GstCaps *filter, *caps;
3737 gst_query_parse_caps (query, &filter);
3738 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3739 gst_query_set_caps_result (query, caps);
3740 gst_caps_unref (caps);
3745 if (GST_QUERY_IS_SERIALIZED (query)) {
3746 RTPJitterBufferItem *item;
3749 JBUF_LOCK_CHECK (priv, out_flushing);
3750 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3751 RTP_JITTER_BUFFER_MODE_BUFFER) {
3752 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3753 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3754 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3756 JBUF_SIGNAL_EVENT (priv);
3757 JBUF_WAIT_QUERY (priv, out_flushing);
3758 res = priv->last_query;
3760 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3765 res = gst_pad_query_default (pad, parent, query);
3773 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3781 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3784 GstRtpJitterBuffer *jitterbuffer;
3785 GstRtpJitterBufferPrivate *priv;
3786 gboolean res = FALSE;
3788 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3789 priv = jitterbuffer->priv;
3791 switch (GST_QUERY_TYPE (query)) {
3792 case GST_QUERY_LATENCY:
3794 /* We need to send the query upstream and add the returned latency to our
3796 GstClockTime min_latency, max_latency;
3798 GstClockTime our_latency;
3800 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3801 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3803 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3804 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3805 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3807 /* store this so that we can safely sync on the peer buffers. */
3809 priv->peer_latency = min_latency;
3810 our_latency = priv->latency_ns;
3813 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3814 GST_TIME_ARGS (our_latency));
3816 /* we add some latency but can buffer an infinite amount of time */
3817 min_latency += our_latency;
3820 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3821 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3822 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3824 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3828 case GST_QUERY_POSITION:
3830 GstClockTime start, last_out;
3833 gst_query_parse_position (query, &fmt, NULL);
3834 if (fmt != GST_FORMAT_TIME) {
3835 res = gst_pad_query_default (pad, parent, query);
3840 start = priv->npt_start;
3841 last_out = priv->last_out_time;
3844 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3845 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3846 GST_TIME_ARGS (last_out));
3848 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3849 /* bring 0-based outgoing time to stream time */
3850 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3853 res = gst_pad_query_default (pad, parent, query);
3857 case GST_QUERY_CAPS:
3859 GstCaps *filter, *caps;
3861 gst_query_parse_caps (query, &filter);
3862 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3863 gst_query_set_caps_result (query, caps);
3864 gst_caps_unref (caps);
3869 res = gst_pad_query_default (pad, parent, query);
3877 gst_rtp_jitter_buffer_set_property (GObject * object,
3878 guint prop_id, const GValue * value, GParamSpec * pspec)
3880 GstRtpJitterBuffer *jitterbuffer;
3881 GstRtpJitterBufferPrivate *priv;
3883 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3884 priv = jitterbuffer->priv;
3889 guint new_latency, old_latency;
3891 new_latency = g_value_get_uint (value);
3894 old_latency = priv->latency_ms;
3895 priv->latency_ms = new_latency;
3896 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3897 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3900 /* post message if latency changed, this will inform the parent pipeline
3901 * that a latency reconfiguration is possible/needed. */
3902 if (new_latency != old_latency) {
3903 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3904 GST_TIME_ARGS (new_latency * GST_MSECOND));
3906 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3907 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3911 case PROP_DROP_ON_LATENCY:
3913 priv->drop_on_latency = g_value_get_boolean (value);
3916 case PROP_TS_OFFSET:
3918 priv->ts_offset = g_value_get_int64 (value);
3919 priv->ts_discont = TRUE;
3924 priv->do_lost = g_value_get_boolean (value);
3929 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3932 case PROP_DO_RETRANSMISSION:
3934 priv->do_retransmission = g_value_get_boolean (value);
3937 case PROP_RTX_NEXT_SEQNUM:
3939 priv->rtx_next_seqnum = g_value_get_boolean (value);
3942 case PROP_RTX_DELAY:
3944 priv->rtx_delay = g_value_get_int (value);
3947 case PROP_RTX_MIN_DELAY:
3949 priv->rtx_min_delay = g_value_get_uint (value);
3952 case PROP_RTX_DELAY_REORDER:
3954 priv->rtx_delay_reorder = g_value_get_int (value);
3957 case PROP_RTX_RETRY_TIMEOUT:
3959 priv->rtx_retry_timeout = g_value_get_int (value);
3962 case PROP_RTX_MIN_RETRY_TIMEOUT:
3964 priv->rtx_min_retry_timeout = g_value_get_int (value);
3967 case PROP_RTX_RETRY_PERIOD:
3969 priv->rtx_retry_period = g_value_get_int (value);
3972 case PROP_RTX_MAX_RETRIES:
3974 priv->rtx_max_retries = g_value_get_int (value);
3977 case PROP_MAX_RTCP_RTP_TIME_DIFF:
3979 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
3982 case PROP_MAX_DROPOUT_TIME:
3984 priv->max_dropout_time = g_value_get_uint (value);
3987 case PROP_MAX_MISORDER_TIME:
3989 priv->max_misorder_time = g_value_get_uint (value);
3993 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3999 gst_rtp_jitter_buffer_get_property (GObject * object,
4000 guint prop_id, GValue * value, GParamSpec * pspec)
4002 GstRtpJitterBuffer *jitterbuffer;
4003 GstRtpJitterBufferPrivate *priv;
4005 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4006 priv = jitterbuffer->priv;
4011 g_value_set_uint (value, priv->latency_ms);
4014 case PROP_DROP_ON_LATENCY:
4016 g_value_set_boolean (value, priv->drop_on_latency);
4019 case PROP_TS_OFFSET:
4021 g_value_set_int64 (value, priv->ts_offset);
4026 g_value_set_boolean (value, priv->do_lost);
4031 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4039 if (priv->srcresult != GST_FLOW_OK)
4042 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4044 g_value_set_int (value, percent);
4048 case PROP_DO_RETRANSMISSION:
4050 g_value_set_boolean (value, priv->do_retransmission);
4053 case PROP_RTX_NEXT_SEQNUM:
4055 g_value_set_boolean (value, priv->rtx_next_seqnum);
4058 case PROP_RTX_DELAY:
4060 g_value_set_int (value, priv->rtx_delay);
4063 case PROP_RTX_MIN_DELAY:
4065 g_value_set_uint (value, priv->rtx_min_delay);
4068 case PROP_RTX_DELAY_REORDER:
4070 g_value_set_int (value, priv->rtx_delay_reorder);
4073 case PROP_RTX_RETRY_TIMEOUT:
4075 g_value_set_int (value, priv->rtx_retry_timeout);
4078 case PROP_RTX_MIN_RETRY_TIMEOUT:
4080 g_value_set_int (value, priv->rtx_min_retry_timeout);
4083 case PROP_RTX_RETRY_PERIOD:
4085 g_value_set_int (value, priv->rtx_retry_period);
4088 case PROP_RTX_MAX_RETRIES:
4090 g_value_set_int (value, priv->rtx_max_retries);
4094 g_value_take_boxed (value,
4095 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4097 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4099 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4102 case PROP_MAX_DROPOUT_TIME:
4104 g_value_set_uint (value, priv->max_dropout_time);
4107 case PROP_MAX_MISORDER_TIME:
4109 g_value_set_uint (value, priv->max_misorder_time);
4113 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4118 static GstStructure *
4119 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4123 JBUF_LOCK (jbuf->priv);
4124 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4125 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4126 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4127 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4128 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4129 JBUF_UNLOCK (jbuf->priv);