2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
9 * This library is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Library General Public
11 * License as published by the Free Software Foundation; either
12 * version 2 of the License, or (at your option) any later version.
14 * This library is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Library General Public License for more details.
19 * You should have received a copy of the GNU Library General Public
20 * License along with this library; if not, write to the
21 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
22 * Boston, MA 02110-1301, USA.
27 * SECTION:element-rtpjitterbuffer
29 * This element reorders and removes duplicate RTP packets as they are received
30 * from a network source.
32 * The element needs the clock-rate of the RTP payload in order to estimate the
33 * delay. This information is obtained either from the caps on the sink pad or,
34 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
35 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
37 * The rtpjitterbuffer will wait for missing packets up to a configurable time
38 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
39 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
40 * property is set, lost packets will result in a custom serialized downstream
41 * event of name GstRTPPacketLost. The lost packet events are usually used by a
42 * depayloader or other element to create concealment data or some other logic
43 * to gracefully handle the missing packets.
45 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
46 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
49 * The jitterbuffer can also be configured to send early retransmission events
50 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
51 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
52 * sends a custom upstream event named GstRTPRetransmissionRequest when the
53 * packet is considered late. The initial expected packet arrival time is
54 * calculated as follows:
56 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
57 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
58 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
59 * packets with different rtptime.
61 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
62 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
63 * previously scheduled timeout is overwritten.
65 * - If seqnum N arrived, all seqnum older than
66 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
67 * immediately. This is to request fast feedback for abonormally reorder
68 * packets before any of the previous timeouts is triggered.
70 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
71 * event. After the initial timeout expires and the retransmission event is
72 * sent, the timeout is scheduled for
73 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
74 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
75 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
76 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
77 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
78 * retransmission requests are sent and the regular logic is performed to
79 * schedule a lost packet as discussed above.
81 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
84 * This element will automatically be used inside rtpbin.
87 * <title>Example pipelines</title>
89 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
90 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
91 * inserted into the pipeline to smooth out network jitter and to reorder the
92 * out-of-order RTP packets.
102 #include <gst/rtp/gstrtpbuffer.h>
104 #include "gstrtpjitterbuffer.h"
105 #include "rtpjitterbuffer.h"
106 #include "rtpstats.h"
108 #include <gst/glib-compat-private.h>
110 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
111 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
113 /* RTPJitterBuffer signals and args */
116 SIGNAL_REQUEST_PT_MAP,
124 #define DEFAULT_LATENCY_MS 200
125 #define DEFAULT_DROP_ON_LATENCY FALSE
126 #define DEFAULT_TS_OFFSET 0
127 #define DEFAULT_DO_LOST FALSE
128 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
129 #define DEFAULT_PERCENT 0
130 #define DEFAULT_DO_RETRANSMISSION FALSE
131 #define DEFAULT_RTX_NEXT_SEQNUM TRUE
132 #define DEFAULT_RTX_DELAY -1
133 #define DEFAULT_RTX_MIN_DELAY 0
134 #define DEFAULT_RTX_DELAY_REORDER 3
135 #define DEFAULT_RTX_RETRY_TIMEOUT -1
136 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
137 #define DEFAULT_RTX_RETRY_PERIOD -1
138 #define DEFAULT_RTX_MAX_RETRIES -1
140 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
141 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
147 PROP_DROP_ON_LATENCY,
152 PROP_DO_RETRANSMISSION,
153 PROP_RTX_NEXT_SEQNUM,
156 PROP_RTX_DELAY_REORDER,
157 PROP_RTX_RETRY_TIMEOUT,
158 PROP_RTX_MIN_RETRY_TIMEOUT,
159 PROP_RTX_RETRY_PERIOD,
160 PROP_RTX_MAX_RETRIES,
164 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
166 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
168 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
171 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
173 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
174 GST_DEBUG ("waiting timer"); \
175 (priv)->waiting_timer = TRUE; \
176 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
177 (priv)->waiting_timer = FALSE; \
178 GST_DEBUG ("waiting timer done"); \
180 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
181 if (G_UNLIKELY ((priv)->waiting_timer)) { \
182 GST_DEBUG ("signal timer"); \
183 g_cond_signal (&(priv)->jbuf_timer); \
187 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
188 GST_DEBUG ("waiting event"); \
189 (priv)->waiting_event = TRUE; \
190 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
191 (priv)->waiting_event = FALSE; \
192 GST_DEBUG ("waiting event done"); \
193 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
196 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
197 if (G_UNLIKELY ((priv)->waiting_event)) { \
198 GST_DEBUG ("signal event"); \
199 g_cond_signal (&(priv)->jbuf_event); \
203 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
204 GST_DEBUG ("waiting query"); \
205 (priv)->waiting_query = TRUE; \
206 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
207 (priv)->waiting_query = FALSE; \
208 GST_DEBUG ("waiting query done"); \
209 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
212 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
213 (priv)->last_query = res; \
214 if (G_UNLIKELY ((priv)->waiting_query)) { \
215 GST_DEBUG ("signal query"); \
216 g_cond_signal (&(priv)->jbuf_query); \
221 struct _GstRtpJitterBufferPrivate
223 GstPad *sinkpad, *srcpad;
226 RTPJitterBuffer *jbuf;
228 gboolean waiting_timer;
230 gboolean waiting_event;
232 gboolean waiting_query;
240 gboolean timer_running;
241 GThread *timer_thread;
246 gboolean drop_on_latency;
249 gboolean do_retransmission;
250 gboolean rtx_next_seqnum;
253 gint rtx_delay_reorder;
254 gint rtx_retry_timeout;
255 gint rtx_min_retry_timeout;
256 gint rtx_retry_period;
257 gint rtx_max_retries;
259 /* the last seqnum we pushed out */
260 guint32 last_popped_seqnum;
261 /* the next expected seqnum we push */
263 /* seqnum-base, if known */
265 /* last output time */
266 GstClockTime last_out_time;
267 /* last valid input timestamp and rtptime pair */
268 GstClockTime ips_dts;
270 GstClockTime packet_spacing;
274 /* the next expected seqnum we receive */
275 GstClockTime last_in_dts;
276 guint32 last_in_seqnum;
277 guint32 next_in_seqnum;
281 /* start and stop ranges */
282 GstClockTime npt_start;
283 GstClockTime npt_stop;
284 guint64 ext_timestamp;
285 guint64 last_elapsed;
286 guint64 estimated_eos;
293 /* clock rate and rtp timestamp offset */
297 gint64 prev_ts_offset;
299 /* when we are shutting down */
300 GstFlowReturn srcresult;
306 GstClockTime timer_timeout;
307 guint16 timer_seqnum;
308 /* the latency of the upstream peer, we have to take this into account when
309 * synchronizing the buffers. */
310 GstClockTime peer_latency;
314 /* some accounting */
316 guint64 num_duplicates;
317 guint64 num_rtx_requests;
318 guint64 num_rtx_success;
319 guint64 num_rtx_failed;
324 GstClockTime last_dts;
325 guint64 last_rtptime;
326 GstClockTime avg_jitter;
343 GstClockTime timeout;
344 GstClockTime duration;
345 GstClockTime rtx_base;
346 GstClockTime rtx_delay;
347 GstClockTime rtx_retry;
348 GstClockTime rtx_last;
352 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
353 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
354 GstRtpJitterBufferPrivate))
356 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
357 GST_STATIC_PAD_TEMPLATE ("sink",
360 GST_STATIC_CAPS ("application/x-rtp"
361 /* "clock-rate = (int) [ 1, 2147483647 ], "
362 * "payload = (int) , "
363 * "encoding-name = (string) "
367 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
368 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
371 GST_STATIC_CAPS ("application/x-rtcp")
374 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
375 GST_STATIC_PAD_TEMPLATE ("src",
378 GST_STATIC_CAPS ("application/x-rtp"
379 /* "payload = (int) , "
380 * "clock-rate = (int) , "
381 * "encoding-name = (string) "
385 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
387 #define gst_rtp_jitter_buffer_parent_class parent_class
388 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
390 /* object overrides */
391 static void gst_rtp_jitter_buffer_set_property (GObject * object,
392 guint prop_id, const GValue * value, GParamSpec * pspec);
393 static void gst_rtp_jitter_buffer_get_property (GObject * object,
394 guint prop_id, GValue * value, GParamSpec * pspec);
395 static void gst_rtp_jitter_buffer_finalize (GObject * object);
397 /* element overrides */
398 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
399 * element, GstStateChange transition);
400 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
401 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
402 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
404 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
407 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
408 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
411 /* sinkpad overrides */
412 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
413 GstObject * parent, GstEvent * event);
414 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
415 GstObject * parent, GstBuffer * buffer);
417 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
418 GstObject * parent, GstEvent * event);
419 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
420 GstObject * parent, GstBuffer * buffer);
422 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
423 GstObject * parent, GstQuery * query);
425 /* srcpad overrides */
426 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
427 GstObject * parent, GstEvent * event);
428 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
429 GstObject * parent, GstPadMode mode, gboolean active);
430 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
431 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
432 GstObject * parent, GstQuery * query);
435 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
437 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
438 gboolean active, guint64 base_time);
439 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
441 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
442 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
444 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
446 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
450 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
452 GObjectClass *gobject_class;
453 GstElementClass *gstelement_class;
455 gobject_class = (GObjectClass *) klass;
456 gstelement_class = (GstElementClass *) klass;
458 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
460 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
462 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
463 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
466 * GstRtpJitterBuffer:latency:
468 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
469 * for at most this time.
471 g_object_class_install_property (gobject_class, PROP_LATENCY,
472 g_param_spec_uint ("latency", "Buffer latency in ms",
473 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
474 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
476 * GstRtpJitterBuffer:drop-on-latency:
478 * Drop oldest buffers when the queue is completely filled.
480 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
481 g_param_spec_boolean ("drop-on-latency",
482 "Drop buffers when maximum latency is reached",
483 "Tells the jitterbuffer to never exceed the given latency in size",
484 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
486 * GstRtpJitterBuffer:ts-offset:
488 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
489 * This is mainly used to ensure interstream synchronisation.
491 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
492 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
493 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
494 G_MAXINT64, DEFAULT_TS_OFFSET,
495 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
498 * GstRtpJitterBuffer:do-lost:
500 * Send out a GstRTPPacketLost event downstream when a packet is considered
503 g_object_class_install_property (gobject_class, PROP_DO_LOST,
504 g_param_spec_boolean ("do-lost", "Do Lost",
505 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
506 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
509 * GstRtpJitterBuffer:mode:
511 * Control the buffering and timestamping mode used by the jitterbuffer.
513 g_object_class_install_property (gobject_class, PROP_MODE,
514 g_param_spec_enum ("mode", "Mode",
515 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
516 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
518 * GstRtpJitterBuffer:percent:
520 * The percent of the jitterbuffer that is filled.
522 g_object_class_install_property (gobject_class, PROP_PERCENT,
523 g_param_spec_int ("percent", "percent",
524 "The buffer filled percent", 0, 100,
525 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
527 * GstRtpJitterBuffer:do-retransmission:
529 * Send out a GstRTPRetransmission event upstream when a packet is considered
530 * late and should be retransmitted.
534 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
535 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
536 "Send retransmission events upstream when a packet is late",
537 DEFAULT_DO_RETRANSMISSION,
538 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
541 * GstRtpJitterBuffer:rtx-next-seqnum
543 * Estimate when the next packet should arrive and schedule a retransmission
545 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
546 * for packet N+1. So it will be requested if it does not arrive at the expected time.
547 * The expected time is calculated using the dts of N and the packet spacing.
551 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
552 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
553 "Estimate when the next packet should arrive and schedule a "
554 "retransmission request for it.",
555 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
558 * GstRtpJitterBuffer:rtx-delay:
560 * When a packet did not arrive at the expected time, wait this extra amount
561 * of time before sending a retransmission event.
563 * When -1 is used, the max jitter will be used as extra delay.
567 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
568 g_param_spec_int ("rtx-delay", "RTX Delay",
569 "Extra time in ms to wait before sending retransmission "
570 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
571 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
574 * GstRtpJitterBuffer:rtx-min-delay:
576 * When a packet did not arrive at the expected time, wait at least this extra amount
577 * of time before sending a retransmission event.
581 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
582 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
583 "Minimum time in ms to wait before sending retransmission "
584 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
585 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
587 * GstRtpJitterBuffer:rtx-delay-reorder:
589 * Assume that a retransmission event should be sent when we see
590 * this much packet reordering.
592 * When -1 is used, the value will be estimated based on observed packet
597 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
598 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
599 "Sending retransmission event when this much reordering (-1 automatic)",
600 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
601 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
603 * GstRtpJitterBuffer::rtx-retry-timeout:
605 * When no packet has been received after sending a retransmission event
606 * for this time, retry sending a retransmission event.
608 * When -1 is used, the value will be estimated based on observed round
613 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
614 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
615 "Retry sending a transmission event after this timeout in "
616 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
617 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
619 * GstRtpJitterBuffer::rtx-min-retry-timeout:
621 * The minimum amount of time between retry timeouts. When
622 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
623 * minimum interval between retry timeouts.
625 * When -1 is used, the value will be estimated based on the
630 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
631 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
632 "Minimum timeout between sending a transmission event in "
633 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
634 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
636 * GstRtpJitterBuffer:rtx-retry-period:
638 * The amount of time to try to get a retransmission.
640 * When -1 is used, the value will be estimated based on the jitterbuffer
641 * latency and the observed round trip time.
645 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
646 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
647 "Try to get a retransmission for this many ms "
648 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
649 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
651 * GstRtpJitterBuffer:rtx-max-retries:
653 * The maximum number of retries to request a retransmission.
655 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
656 * When -1 is used, the number of retransmission request will not be limited.
660 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
661 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
662 "The maximum number of retries to request a retransmission. "
663 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
664 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
666 * GstRtpJitterBuffer:stats:
668 * Various jitterbuffer statistics. This property returns a GstStructure
669 * with name application/x-rtp-jitterbuffer-stats with the following fields:
675 * <classname>"rtx-count"</classname>:
676 * the number of retransmissions requested.
682 * <classname>"rtx-success-count"</classname>:
683 * the number of successful retransmissions.
689 * <classname>"rtx-per-packet"</classname>:
690 * average number of RTX per packet.
696 * <classname>"rtx-rtt"</classname>:
697 * average round trip time per RTX.
704 g_object_class_install_property (gobject_class, PROP_STATS,
705 g_param_spec_boxed ("stats", "Statistics",
706 "Various statistics", GST_TYPE_STRUCTURE,
707 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
710 * GstRtpJitterBuffer::request-pt-map:
711 * @buffer: the object which received the signal
714 * Request the payload type as #GstCaps for @pt.
716 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
717 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
718 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
719 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
720 GST_TYPE_CAPS, 1, G_TYPE_UINT);
722 * GstRtpJitterBuffer::handle-sync:
723 * @buffer: the object which received the signal
724 * @struct: a GstStructure containing sync values.
726 * Be notified of new sync values.
728 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
729 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
730 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
731 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
732 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
735 * GstRtpJitterBuffer::on-npt-stop:
736 * @buffer: the object which received the signal
738 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
739 * the npt-stop position.
741 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
742 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
743 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
744 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
745 G_TYPE_NONE, 0, G_TYPE_NONE);
748 * GstRtpJitterBuffer::clear-pt-map:
749 * @buffer: the object which received the signal
751 * Invalidate the clock-rate as obtained with the
752 * #GstRtpJitterBuffer::request-pt-map signal.
754 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
755 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
756 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
757 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
758 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
761 * GstRtpJitterBuffer::set-active:
762 * @buffer: the object which received the signal
764 * Start pushing out packets with the given base time. This signal is only
765 * useful in buffering mode.
767 * Returns: the time of the last pushed packet.
769 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
770 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
771 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
772 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
773 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
776 gstelement_class->change_state =
777 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
778 gstelement_class->request_new_pad =
779 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
780 gstelement_class->release_pad =
781 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
782 gstelement_class->provide_clock =
783 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
785 gst_element_class_add_pad_template (gstelement_class,
786 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
787 gst_element_class_add_pad_template (gstelement_class,
788 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
789 gst_element_class_add_pad_template (gstelement_class,
790 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
792 gst_element_class_set_static_metadata (gstelement_class,
793 "RTP packet jitter-buffer", "Filter/Network/RTP",
794 "A buffer that deals with network jitter and other transmission faults",
795 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
796 "Wim Taymans <wim.taymans@gmail.com>");
798 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
799 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
801 GST_DEBUG_CATEGORY_INIT
802 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
806 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
808 GstRtpJitterBufferPrivate *priv;
810 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
811 jitterbuffer->priv = priv;
813 priv->latency_ms = DEFAULT_LATENCY_MS;
814 priv->latency_ns = priv->latency_ms * GST_MSECOND;
815 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
816 priv->do_lost = DEFAULT_DO_LOST;
817 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
818 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
819 priv->rtx_delay = DEFAULT_RTX_DELAY;
820 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
821 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
822 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
823 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
824 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
825 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
828 priv->last_rtptime = -1;
829 priv->avg_jitter = 0;
830 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
831 priv->jbuf = rtp_jitter_buffer_new ();
832 g_mutex_init (&priv->jbuf_lock);
833 g_cond_init (&priv->jbuf_timer);
834 g_cond_init (&priv->jbuf_event);
835 g_cond_init (&priv->jbuf_query);
836 g_queue_init (&priv->gap_packets);
838 /* reset skew detection initialy */
839 rtp_jitter_buffer_reset_skew (priv->jbuf);
840 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
841 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
845 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
848 gst_pad_set_activatemode_function (priv->srcpad,
849 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
850 gst_pad_set_query_function (priv->srcpad,
851 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
852 gst_pad_set_event_function (priv->srcpad,
853 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
856 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
859 gst_pad_set_chain_function (priv->sinkpad,
860 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
861 gst_pad_set_event_function (priv->sinkpad,
862 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
863 gst_pad_set_query_function (priv->sinkpad,
864 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
866 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
867 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
869 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
872 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
874 #define ITEM_TYPE_BUFFER 0
875 #define ITEM_TYPE_LOST 1
876 #define ITEM_TYPE_EVENT 2
877 #define ITEM_TYPE_QUERY 3
879 static RTPJitterBufferItem *
880 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
881 guint seqnum, guint count, guint rtptime)
883 RTPJitterBufferItem *item;
885 item = g_slice_new (RTPJitterBufferItem);
892 item->seqnum = seqnum;
894 item->rtptime = rtptime;
900 free_item (RTPJitterBufferItem * item)
902 if (item->data && item->type != ITEM_TYPE_QUERY)
903 gst_mini_object_unref (item->data);
904 g_slice_free (RTPJitterBufferItem, item);
908 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
910 GList **l = user_data;
912 if (item->data && item->type == ITEM_TYPE_EVENT
913 && GST_EVENT_IS_STICKY (item->data)) {
914 *l = g_list_prepend (*l, item->data);
915 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
916 gst_mini_object_unref (item->data);
918 g_slice_free (RTPJitterBufferItem, item);
922 gst_rtp_jitter_buffer_finalize (GObject * object)
924 GstRtpJitterBuffer *jitterbuffer;
925 GstRtpJitterBufferPrivate *priv;
927 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
928 priv = jitterbuffer->priv;
930 g_array_free (priv->timers, TRUE);
931 g_mutex_clear (&priv->jbuf_lock);
932 g_cond_clear (&priv->jbuf_timer);
933 g_cond_clear (&priv->jbuf_event);
934 g_cond_clear (&priv->jbuf_query);
936 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
937 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
938 g_queue_clear (&priv->gap_packets);
939 g_object_unref (priv->jbuf);
941 G_OBJECT_CLASS (parent_class)->finalize (object);
945 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
947 GstRtpJitterBuffer *jitterbuffer;
948 GstPad *otherpad = NULL;
949 GstIterator *it = NULL;
952 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
954 if (pad == jitterbuffer->priv->sinkpad) {
955 otherpad = jitterbuffer->priv->srcpad;
956 } else if (pad == jitterbuffer->priv->srcpad) {
957 otherpad = jitterbuffer->priv->sinkpad;
958 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
959 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
963 g_value_init (&val, GST_TYPE_PAD);
964 g_value_set_object (&val, otherpad);
965 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
966 g_value_unset (&val);
973 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
975 GstRtpJitterBufferPrivate *priv;
977 priv = jitterbuffer->priv;
979 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
982 gst_pad_new_from_static_template
983 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
984 gst_pad_set_chain_function (priv->rtcpsinkpad,
985 gst_rtp_jitter_buffer_chain_rtcp);
986 gst_pad_set_event_function (priv->rtcpsinkpad,
987 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
988 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
989 gst_rtp_jitter_buffer_iterate_internal_links);
990 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
991 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
993 return priv->rtcpsinkpad;
997 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
999 GstRtpJitterBufferPrivate *priv;
1001 priv = jitterbuffer->priv;
1003 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1005 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1007 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1008 priv->rtcpsinkpad = NULL;
1012 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1013 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1015 GstRtpJitterBuffer *jitterbuffer;
1016 GstElementClass *klass;
1018 GstRtpJitterBufferPrivate *priv;
1020 g_return_val_if_fail (templ != NULL, NULL);
1021 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1023 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1024 priv = jitterbuffer->priv;
1025 klass = GST_ELEMENT_GET_CLASS (element);
1027 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1029 /* figure out the template */
1030 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1031 if (priv->rtcpsinkpad != NULL)
1034 result = create_rtcp_sink (jitterbuffer);
1036 goto wrong_template;
1043 g_warning ("rtpjitterbuffer: this is not our template");
1048 g_warning ("rtpjitterbuffer: pad already requested");
1054 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1056 GstRtpJitterBuffer *jitterbuffer;
1057 GstRtpJitterBufferPrivate *priv;
1059 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1060 g_return_if_fail (GST_IS_PAD (pad));
1062 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1063 priv = jitterbuffer->priv;
1065 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1067 if (priv->rtcpsinkpad == pad) {
1068 remove_rtcp_sink (jitterbuffer);
1077 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1083 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1085 return gst_system_clock_obtain ();
1089 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1091 GstRtpJitterBufferPrivate *priv;
1093 priv = jitterbuffer->priv;
1095 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1098 priv->clock_rate = -1;
1099 /* do not clear current content, but refresh state for new arrival */
1100 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1101 rtp_jitter_buffer_reset_skew (priv->jbuf);
1106 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1109 GstRtpJitterBufferPrivate *priv;
1110 GstClockTime last_out;
1111 RTPJitterBufferItem *item;
1116 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1117 active, GST_TIME_ARGS (offset));
1119 if (active != priv->active) {
1120 /* add the amount of time spent in paused to the output offset. All
1121 * outgoing buffers will have this offset applied to their timestamps in
1122 * order to make them arrive in time in the sink. */
1123 priv->out_offset = offset;
1124 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1125 GST_TIME_ARGS (priv->out_offset));
1126 priv->active = active;
1127 JBUF_SIGNAL_EVENT (priv);
1130 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1132 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1133 /* head buffer timestamp and offset gives our output time */
1134 last_out = item->dts + priv->ts_offset;
1136 /* use last known time when the buffer is empty */
1137 last_out = priv->last_out_time;
1145 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1147 GstRtpJitterBuffer *jitterbuffer;
1148 GstRtpJitterBufferPrivate *priv;
1153 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1154 priv = jitterbuffer->priv;
1156 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1158 caps = gst_pad_peer_query_caps (other, filter);
1160 templ = gst_pad_get_pad_template_caps (pad);
1162 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1167 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1169 intersect = gst_caps_intersect (caps, templ);
1170 gst_caps_unref (caps);
1171 gst_caps_unref (templ);
1175 gst_object_unref (jitterbuffer);
1181 * Must be called with JBUF_LOCK held
1185 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1188 GstRtpJitterBufferPrivate *priv;
1189 GstStructure *caps_struct;
1193 priv = jitterbuffer->priv;
1195 /* first parse the caps */
1196 caps_struct = gst_caps_get_structure (caps, 0);
1198 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1200 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1201 * measure the amount of data in the buffer */
1202 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1205 if (priv->clock_rate <= 0)
1208 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1210 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1212 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1213 * can use this to track the amount of time elapsed on the sender. */
1214 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1215 priv->clock_base = val;
1217 priv->clock_base = -1;
1219 priv->ext_timestamp = priv->clock_base;
1221 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1224 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1225 /* first expected seqnum, only update when we didn't have a previous base. */
1226 if (priv->next_in_seqnum == -1)
1227 priv->next_in_seqnum = val;
1228 if (priv->next_seqnum == -1) {
1229 priv->next_seqnum = val;
1230 JBUF_SIGNAL_EVENT (priv);
1232 priv->seqnum_base = val;
1234 priv->seqnum_base = -1;
1237 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1239 /* the start and stop times. The seqnum-base corresponds to the start time. We
1240 * will keep track of the seqnums on the output and when we reach the one
1241 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1242 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1243 priv->npt_start = tval;
1245 priv->npt_start = 0;
1247 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1248 priv->npt_stop = tval;
1250 priv->npt_stop = -1;
1252 GST_DEBUG_OBJECT (jitterbuffer,
1253 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1254 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1261 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1266 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1272 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1274 GstRtpJitterBufferPrivate *priv;
1276 priv = jitterbuffer->priv;
1279 /* mark ourselves as flushing */
1280 priv->srcresult = GST_FLOW_FLUSHING;
1281 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1282 /* this unblocks any waiting pops on the src pad task */
1283 JBUF_SIGNAL_EVENT (priv);
1284 JBUF_SIGNAL_QUERY (priv, FALSE);
1289 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1291 GstRtpJitterBufferPrivate *priv;
1293 priv = jitterbuffer->priv;
1296 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1297 /* Mark as non flushing */
1298 priv->srcresult = GST_FLOW_OK;
1299 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1300 priv->last_popped_seqnum = -1;
1301 priv->last_out_time = -1;
1302 priv->next_seqnum = -1;
1303 priv->seqnum_base = -1;
1304 priv->ips_rtptime = -1;
1305 priv->ips_dts = GST_CLOCK_TIME_NONE;
1306 priv->packet_spacing = 0;
1307 priv->next_in_seqnum = -1;
1308 priv->clock_rate = -1;
1311 priv->estimated_eos = -1;
1312 priv->last_elapsed = 0;
1313 priv->ext_timestamp = -1;
1314 priv->avg_jitter = 0;
1315 priv->last_dts = -1;
1316 priv->last_rtptime = -1;
1317 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1318 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1319 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1320 rtp_jitter_buffer_reset_skew (priv->jbuf);
1321 remove_all_timers (jitterbuffer);
1322 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1323 g_queue_clear (&priv->gap_packets);
1328 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1329 GstPadMode mode, gboolean active)
1332 GstRtpJitterBuffer *jitterbuffer = NULL;
1334 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1337 case GST_PAD_MODE_PUSH:
1339 /* allow data processing */
1340 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1342 /* start pushing out buffers */
1343 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1344 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1345 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1347 /* make sure all data processing stops ASAP */
1348 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1350 /* NOTE this will hardlock if the state change is called from the src pad
1351 * task thread because we will _join() the thread. */
1352 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1353 result = gst_pad_stop_task (pad);
1363 static GstStateChangeReturn
1364 gst_rtp_jitter_buffer_change_state (GstElement * element,
1365 GstStateChange transition)
1367 GstRtpJitterBuffer *jitterbuffer;
1368 GstRtpJitterBufferPrivate *priv;
1369 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1371 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1372 priv = jitterbuffer->priv;
1374 switch (transition) {
1375 case GST_STATE_CHANGE_NULL_TO_READY:
1377 case GST_STATE_CHANGE_READY_TO_PAUSED:
1379 /* reset negotiated values */
1380 priv->clock_rate = -1;
1381 priv->clock_base = -1;
1382 priv->peer_latency = 0;
1384 /* block until we go to PLAYING */
1385 priv->blocked = TRUE;
1386 priv->timer_running = TRUE;
1387 priv->timer_thread =
1388 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1391 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1393 /* unblock to allow streaming in PLAYING */
1394 priv->blocked = FALSE;
1395 JBUF_SIGNAL_EVENT (priv);
1396 JBUF_SIGNAL_TIMER (priv);
1403 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1405 switch (transition) {
1406 case GST_STATE_CHANGE_READY_TO_PAUSED:
1407 /* we are a live element because we sync to the clock, which we can only
1408 * do in the PLAYING state */
1409 if (ret != GST_STATE_CHANGE_FAILURE)
1410 ret = GST_STATE_CHANGE_NO_PREROLL;
1412 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1414 /* block to stop streaming when PAUSED */
1415 priv->blocked = TRUE;
1416 unschedule_current_timer (jitterbuffer);
1418 if (ret != GST_STATE_CHANGE_FAILURE)
1419 ret = GST_STATE_CHANGE_NO_PREROLL;
1421 case GST_STATE_CHANGE_PAUSED_TO_READY:
1423 gst_buffer_replace (&priv->last_sr, NULL);
1424 priv->timer_running = FALSE;
1425 unschedule_current_timer (jitterbuffer);
1426 JBUF_SIGNAL_TIMER (priv);
1427 JBUF_SIGNAL_QUERY (priv, FALSE);
1429 g_thread_join (priv->timer_thread);
1430 priv->timer_thread = NULL;
1432 case GST_STATE_CHANGE_READY_TO_NULL:
1442 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1445 gboolean ret = TRUE;
1446 GstRtpJitterBuffer *jitterbuffer;
1447 GstRtpJitterBufferPrivate *priv;
1449 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1450 priv = jitterbuffer->priv;
1452 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1454 switch (GST_EVENT_TYPE (event)) {
1455 case GST_EVENT_LATENCY:
1457 GstClockTime latency;
1459 gst_event_parse_latency (event, &latency);
1461 GST_DEBUG_OBJECT (jitterbuffer,
1462 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1465 /* adjust the overall buffer delay to the total pipeline latency in
1466 * buffering mode because if downstream consumes too fast (because of
1467 * large latency or queues, we would start rebuffering again. */
1468 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1469 RTP_JITTER_BUFFER_MODE_BUFFER) {
1470 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1474 ret = gst_pad_push_event (priv->sinkpad, event);
1478 ret = gst_pad_push_event (priv->sinkpad, event);
1485 /* handles and stores the event in the jitterbuffer, must be called with
1488 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1490 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1491 RTPJitterBufferItem *item;
1494 switch (GST_EVENT_TYPE (event)) {
1495 case GST_EVENT_CAPS:
1499 gst_event_parse_caps (event, &caps);
1500 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1503 case GST_EVENT_SEGMENT:
1504 gst_event_copy_segment (event, &priv->segment);
1506 /* we need time for now */
1507 if (priv->segment.format != GST_FORMAT_TIME)
1508 goto newseg_wrong_format;
1510 GST_DEBUG_OBJECT (jitterbuffer,
1511 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1515 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1522 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1523 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1524 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1526 JBUF_SIGNAL_EVENT (priv);
1531 newseg_wrong_format:
1533 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1534 gst_event_unref (event);
1540 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1543 gboolean ret = TRUE;
1544 GstRtpJitterBuffer *jitterbuffer;
1545 GstRtpJitterBufferPrivate *priv;
1547 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1548 priv = jitterbuffer->priv;
1550 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1552 switch (GST_EVENT_TYPE (event)) {
1553 case GST_EVENT_FLUSH_START:
1554 ret = gst_pad_push_event (priv->srcpad, event);
1555 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1556 /* wait for the loop to go into PAUSED */
1557 gst_pad_pause_task (priv->srcpad);
1559 case GST_EVENT_FLUSH_STOP:
1560 ret = gst_pad_push_event (priv->srcpad, event);
1562 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1563 GST_PAD_MODE_PUSH, TRUE);
1566 if (GST_EVENT_IS_SERIALIZED (event)) {
1567 /* serialized events go in the queue */
1569 if (priv->srcresult != GST_FLOW_OK) {
1570 /* Errors in sticky event pushing are no problem and ignored here
1571 * as they will cause more meaningful errors during data flow.
1572 * For EOS events, that are not followed by data flow, we still
1573 * return FALSE here though.
1575 if (!GST_EVENT_IS_STICKY (event) ||
1576 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1577 goto out_flow_error;
1579 /* refuse more events on EOS */
1582 ret = queue_event (jitterbuffer, event);
1585 /* non-serialized events are forwarded downstream immediately */
1586 ret = gst_pad_push_event (priv->srcpad, event);
1595 GST_DEBUG_OBJECT (jitterbuffer,
1596 "refusing event, we have a downstream flow error: %s",
1597 gst_flow_get_name (priv->srcresult));
1599 gst_event_unref (event);
1604 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1606 gst_event_unref (event);
1612 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1615 gboolean ret = TRUE;
1616 GstRtpJitterBuffer *jitterbuffer;
1618 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1620 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1622 switch (GST_EVENT_TYPE (event)) {
1623 case GST_EVENT_FLUSH_START:
1624 gst_event_unref (event);
1626 case GST_EVENT_FLUSH_STOP:
1627 gst_event_unref (event);
1630 ret = gst_pad_event_default (pad, parent, event);
1638 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1639 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1640 * GST_FLOW_FLUSHING when the element is shutting down. On success
1641 * GST_FLOW_OK is returned.
1643 static GstFlowReturn
1644 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1648 GValue args[2] = { {0}, {0} };
1652 g_value_init (&args[0], GST_TYPE_ELEMENT);
1653 g_value_set_object (&args[0], jitterbuffer);
1654 g_value_init (&args[1], G_TYPE_UINT);
1655 g_value_set_uint (&args[1], pt);
1657 g_value_init (&ret, GST_TYPE_CAPS);
1658 g_value_set_boxed (&ret, NULL);
1660 JBUF_UNLOCK (jitterbuffer->priv);
1661 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1663 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1665 g_value_unset (&args[0]);
1666 g_value_unset (&args[1]);
1667 caps = (GstCaps *) g_value_dup_boxed (&ret);
1668 g_value_unset (&ret);
1672 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1673 gst_caps_unref (caps);
1675 if (G_UNLIKELY (!res))
1683 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1684 return GST_FLOW_ERROR;
1688 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1689 return GST_FLOW_FLUSHING;
1693 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1694 return GST_FLOW_ERROR;
1698 /* call with jbuf lock held */
1700 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1702 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1703 GstMessage *message = NULL;
1708 /* Post a buffering message */
1709 if (priv->last_percent != percent) {
1710 priv->last_percent = percent;
1712 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1713 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1720 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1722 GstRtpJitterBufferPrivate *priv;
1724 priv = jitterbuffer->priv;
1726 if (timestamp == -1)
1729 /* apply the timestamp offset, this is used for inter stream sync */
1730 timestamp += priv->ts_offset;
1731 /* add the offset, this is used when buffering */
1732 timestamp += priv->out_offset;
1738 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1740 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1741 TimerData *timer = NULL;
1744 len = priv->timers->len;
1745 for (i = 0; i < len; i++) {
1746 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1747 if (test->seqnum == seqnum && test->type == type) {
1756 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1758 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1760 if (priv->clock_id) {
1761 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1762 gst_clock_id_unschedule (priv->clock_id);
1763 priv->clock_id = NULL;
1768 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1770 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1771 GstClockTime test_timeout;
1773 if ((test_timeout = timer->timeout) == -1)
1776 if (timer->type != TIMER_TYPE_EXPECTED) {
1777 /* add our latency and offset to get output times. */
1778 test_timeout = apply_offset (jitterbuffer, test_timeout);
1779 test_timeout += priv->latency_ns;
1781 return test_timeout;
1785 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1787 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1789 if (priv->clock_id) {
1790 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1792 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1793 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1795 if (timeout == -1 || timeout < priv->timer_timeout)
1796 unschedule_current_timer (jitterbuffer);
1801 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1802 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1803 GstClockTime duration)
1805 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1809 GST_DEBUG_OBJECT (jitterbuffer,
1810 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1811 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1812 GST_TIME_ARGS (delay));
1814 len = priv->timers->len;
1815 g_array_set_size (priv->timers, len + 1);
1816 timer = &g_array_index (priv->timers, TimerData, len);
1819 timer->seqnum = seqnum;
1821 timer->timeout = timeout + delay;
1822 timer->duration = duration;
1823 if (type == TIMER_TYPE_EXPECTED) {
1824 timer->rtx_base = timeout;
1825 timer->rtx_delay = delay;
1826 timer->rtx_retry = 0;
1828 timer->num_rtx_retry = 0;
1829 recalculate_timer (jitterbuffer, timer);
1830 JBUF_SIGNAL_TIMER (priv);
1836 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1837 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1839 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1840 gboolean seqchange, timechange;
1843 seqchange = timer->seqnum != seqnum;
1844 timechange = timer->timeout != timeout;
1846 if (!seqchange && !timechange)
1849 oldseq = timer->seqnum;
1851 GST_DEBUG_OBJECT (jitterbuffer,
1852 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1853 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1855 timer->timeout = timeout + delay;
1856 timer->seqnum = seqnum;
1858 timer->rtx_base = timeout;
1859 timer->rtx_delay = delay;
1860 timer->rtx_retry = 0;
1863 timer->num_rtx_retry = 0;
1865 if (priv->clock_id) {
1866 /* we changed the seqnum and there is a timer currently waiting with this
1867 * seqnum, unschedule it */
1868 if (seqchange && priv->timer_seqnum == oldseq)
1869 unschedule_current_timer (jitterbuffer);
1870 /* we changed the time, check if it is earlier than what we are waiting
1871 * for and unschedule if so */
1872 else if (timechange)
1873 recalculate_timer (jitterbuffer, timer);
1878 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1879 guint16 seqnum, GstClockTime timeout)
1883 /* find the seqnum timer */
1884 timer = find_timer (jitterbuffer, type, seqnum);
1885 if (timer == NULL) {
1886 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1888 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1894 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1896 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1899 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1900 unschedule_current_timer (jitterbuffer);
1903 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1904 g_array_remove_index_fast (priv->timers, idx);
1909 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1911 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1912 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1913 g_array_set_size (priv->timers, 0);
1914 unschedule_current_timer (jitterbuffer);
1917 /* get the extra delay to wait before sending RTX */
1919 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1923 if (priv->rtx_delay == -1) {
1924 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
1925 delay = DEFAULT_AUTO_RTX_DELAY;
1927 /* jitter is in nanoseconds, maximum of 2x jitter and half the
1928 * packet spacing is a good margin */
1929 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
1932 delay = priv->rtx_delay * GST_MSECOND;
1934 if (priv->rtx_min_delay > 0)
1935 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1940 /* we just received a packet with seqnum and dts.
1942 * First check for old seqnum that we are still expecting. If the gap with the
1943 * current seqnum is too big, unschedule the timeouts.
1945 * If we have a valid packet spacing estimate we can set a timer for when we
1946 * should receive the next packet.
1947 * If we don't have a valid estimate, we remove any timer we might have
1948 * had for this packet.
1951 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1952 GstClockTime dts, gboolean do_next_seqnum)
1954 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1955 TimerData *timer = NULL;
1958 /* go through all timers and unschedule the ones with a large gap, also find
1959 * the timer for the seqnum */
1960 len = priv->timers->len;
1961 for (i = 0; i < len; i++) {
1962 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1965 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1967 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
1968 test->type, test->seqnum, seqnum, gap);
1971 GST_DEBUG ("found timer for current seqnum");
1972 /* the timer for the current seqnum */
1974 /* when no retransmission, we can stop now, we only need to find the
1975 * timer for the current seqnum */
1976 if (!priv->do_retransmission)
1978 } else if (gap > priv->rtx_delay_reorder) {
1979 /* max gap, we exceeded the max reorder distance and we don't expect the
1980 * missing packet to be this reordered */
1981 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1982 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
1986 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
1987 && priv->do_retransmission && priv->rtx_next_seqnum;
1989 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1990 if (timer->num_rtx_retry > 0) {
1991 GstClockTime rtx_last, delay;
1993 /* we scheduled a retry for this packet and now we have it */
1994 priv->num_rtx_success++;
1995 /* all the previous retry attempts failed */
1996 priv->num_rtx_failed += timer->num_rtx_retry - 1;
1997 /* number of retries before receiving the packet */
1998 if (priv->avg_rtx_num == 0.0)
1999 priv->avg_rtx_num = timer->num_rtx_retry;
2001 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2002 /* calculate the delay between retransmission request and receiving this
2003 * packet, start with when we scheduled this timeout last */
2004 rtx_last = timer->rtx_last;
2005 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2006 /* we have a valid delay if this packet arrived after we scheduled the
2008 delay = dts - rtx_last;
2009 if (priv->avg_rtx_rtt == 0)
2010 priv->avg_rtx_rtt = delay;
2012 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2016 GST_LOG_OBJECT (jitterbuffer,
2017 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2018 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2019 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2020 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2021 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2022 GST_TIME_ARGS (priv->avg_rtx_rtt));
2024 /* don't try to estimate the next seqnum because this is a retransmitted
2025 * packet and it probably did not arrive with the expected packet
2027 do_next_seqnum = FALSE;
2031 if (do_next_seqnum) {
2032 GstClockTime expected, delay;
2034 /* calculate expected arrival time of the next seqnum */
2035 expected = dts + priv->packet_spacing;
2037 delay = get_rtx_delay (priv);
2039 /* and update/install timer for next seqnum */
2041 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2044 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2045 expected, delay, priv->packet_spacing);
2046 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2047 /* if we had a timer, remove it, we don't know when to expect the next
2049 remove_timer (jitterbuffer, timer);
2054 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2057 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2059 /* we need consecutive seqnums with a different
2060 * rtptime to estimate the packet spacing. */
2061 if (priv->ips_rtptime != rtptime) {
2062 /* rtptime changed, check dts diff */
2063 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2064 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2065 GstClockTime old_packet_spacing = priv->packet_spacing;
2067 /* Biased towards bigger packet spacings to prevent
2068 * too many unneeded retransmission requests for next
2069 * packets that just arrive a little later than we would
2071 if (old_packet_spacing > new_packet_spacing)
2072 priv->packet_spacing =
2073 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2074 else if (old_packet_spacing > 0)
2075 priv->packet_spacing =
2076 (3 * new_packet_spacing + old_packet_spacing) / 4;
2078 priv->packet_spacing = new_packet_spacing;
2080 GST_DEBUG_OBJECT (jitterbuffer,
2081 "new packet spacing %" GST_TIME_FORMAT
2082 " old packet spacing %" GST_TIME_FORMAT
2083 " combined to %" GST_TIME_FORMAT,
2084 GST_TIME_ARGS (new_packet_spacing),
2085 GST_TIME_ARGS (old_packet_spacing),
2086 GST_TIME_ARGS (priv->packet_spacing));
2088 priv->ips_rtptime = rtptime;
2089 priv->ips_dts = dts;
2094 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2095 guint16 seqnum, GstClockTime dts, gint gap)
2097 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2098 GstClockTime total_duration, duration, expected_dts;
2100 guint lost_packets = 0;
2102 GST_DEBUG_OBJECT (jitterbuffer,
2103 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2104 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2106 /* the total duration spanned by the missing packets */
2107 if (dts >= priv->last_in_dts)
2108 total_duration = dts - priv->last_in_dts;
2112 /* interpolate between the current time and the last time based on
2113 * number of packets we are missing, this is the estimated duration
2114 * for the missing packet based on equidistant packet spacing. */
2115 duration = total_duration / (gap + 1);
2117 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2118 GST_TIME_ARGS (duration));
2120 if (total_duration > priv->latency_ns) {
2121 GstClockTime gap_time;
2123 gap_time = total_duration - priv->latency_ns;
2126 lost_packets = gap_time / duration;
2127 gap_time = lost_packets * duration;
2132 /* too many lost packets, some of the missing packets are already
2133 * too late and we can generate lost packet events for them. */
2134 GST_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
2135 " > %" GST_TIME_FORMAT ", consider %u lost",
2136 GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
2139 /* this timer will fire immediately and the lost event will be pushed from
2140 * the timer thread */
2141 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2142 priv->last_in_dts + duration, 0, gap_time);
2144 expected += lost_packets;
2145 priv->last_in_dts += gap_time;
2148 expected_dts = priv->last_in_dts + (lost_packets + 1) * duration;
2150 if (priv->do_retransmission) {
2153 type = TIMER_TYPE_EXPECTED;
2154 /* if we had a timer for the first missing packet, update it. */
2155 if ((timer = find_timer (jitterbuffer, type, expected))) {
2156 GstClockTime timeout = timer->timeout;
2158 timer->duration = duration;
2159 if (timeout > (expected_dts + timer->rtx_retry)) {
2160 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2161 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2165 expected_dts += duration;
2168 type = TIMER_TYPE_LOST;
2171 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2172 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2173 expected_dts += duration;
2179 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2183 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2184 GstRtpJitterBufferPrivate *priv;
2186 priv = jitterbuffer->priv;
2188 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2191 if (priv->last_dts != -1)
2192 dtsdiff = dts - priv->last_dts;
2196 if (priv->last_rtptime != -1)
2197 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2201 priv->last_dts = dts;
2202 priv->last_rtptime = rtptime;
2206 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2209 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2211 diff = ABS (dtsdiff - rtpdiffns);
2213 /* jitter is stored in nanoseconds */
2214 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2216 GST_LOG_OBJECT (jitterbuffer,
2217 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2218 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2219 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2220 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2227 GST_DEBUG_OBJECT (jitterbuffer,
2228 "no dts or no clock-rate, can't calculate jitter");
2234 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2236 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2237 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2240 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2241 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2242 gst_rtp_buffer_unmap (&rtp_a);
2244 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2245 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2246 gst_rtp_buffer_unmap (&rtp_b);
2248 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2252 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2253 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap)
2255 GstRtpJitterBufferPrivate *priv;
2256 guint gap_packets_length;
2257 gboolean reset = FALSE;
2259 priv = jitterbuffer->priv;
2261 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2263 guint32 prev_gap_seq = -1;
2264 gboolean all_consecutive = TRUE;
2266 g_queue_insert_sorted (&priv->gap_packets, buffer,
2267 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2269 for (l = priv->gap_packets.head; l; l = l->next) {
2270 GstBuffer *gap_buffer = l->data;
2271 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2274 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2276 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2278 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2279 if (prev_gap_seq == -1)
2280 prev_gap_seq = gap_seq;
2281 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2282 all_consecutive = FALSE;
2284 prev_gap_seq = gap_seq;
2286 gst_rtp_buffer_unmap (&gap_rtp);
2287 if (!all_consecutive)
2291 if (all_consecutive && gap_packets_length > 3) {
2292 GST_DEBUG_OBJECT (jitterbuffer,
2293 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2294 (future ? "new" : "old"), gap,
2295 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2297 } else if (!all_consecutive) {
2298 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2299 g_queue_clear (&priv->gap_packets);
2300 GST_DEBUG_OBJECT (jitterbuffer,
2301 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2302 (future ? "new" : "old"), gap,
2303 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER));
2306 GST_DEBUG_OBJECT (jitterbuffer,
2307 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2308 (future ? "new" : "old"), gap,
2309 (future ? RTP_MAX_DROPOUT : -RTP_MAX_MISORDER),
2310 gap_packets_length + 1);
2314 GST_DEBUG_OBJECT (jitterbuffer,
2315 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2316 gap, -RTP_MAX_MISORDER);
2317 g_queue_push_tail (&priv->gap_packets, buffer);
2324 static GstFlowReturn
2325 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2328 GstRtpJitterBuffer *jitterbuffer;
2329 GstRtpJitterBufferPrivate *priv;
2331 guint32 expected, rtptime;
2332 GstFlowReturn ret = GST_FLOW_OK;
2333 GstClockTime dts, pts;
2338 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2339 gboolean do_next_seqnum = FALSE;
2340 RTPJitterBufferItem *item;
2341 GstMessage *msg = NULL;
2343 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2345 priv = jitterbuffer->priv;
2347 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2348 goto invalid_buffer;
2350 pt = gst_rtp_buffer_get_payload_type (&rtp);
2351 seqnum = gst_rtp_buffer_get_seq (&rtp);
2352 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2353 gst_rtp_buffer_unmap (&rtp);
2355 /* make sure we have PTS and DTS set */
2356 pts = GST_BUFFER_PTS (buffer);
2357 dts = GST_BUFFER_DTS (buffer);
2363 /* take the DTS of the buffer. This is the time when the packet was
2364 * received and is used to calculate jitter and clock skew. We will adjust
2365 * this DTS with the smoothed value after processing it in the
2366 * jitterbuffer and assign it as the PTS. */
2367 /* bring to running time */
2368 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2370 GST_DEBUG_OBJECT (jitterbuffer,
2371 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2372 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2374 JBUF_LOCK_CHECK (priv, out_flushing);
2376 if (G_UNLIKELY (priv->last_pt != pt)) {
2379 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2383 /* reset clock-rate so that we get a new one */
2384 priv->clock_rate = -1;
2386 /* Try to get the clock-rate from the caps first if we can. If there are no
2387 * caps we must fire the signal to get the clock-rate. */
2388 if ((caps = gst_pad_get_current_caps (pad))) {
2389 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2390 gst_caps_unref (caps);
2394 if (G_UNLIKELY (priv->clock_rate == -1)) {
2395 /* no clock rate given on the caps, try to get one with the signal */
2396 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2397 pt) == GST_FLOW_FLUSHING)
2400 if (G_UNLIKELY (priv->clock_rate == -1))
2404 /* don't accept more data on EOS */
2405 if (G_UNLIKELY (priv->eos))
2408 calculate_jitter (jitterbuffer, dts, rtptime);
2410 if (priv->seqnum_base != -1) {
2413 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2416 GST_DEBUG_OBJECT (jitterbuffer,
2417 "packet seqnum #%d before seqnum-base #%d", seqnum,
2419 gst_buffer_unref (buffer);
2422 } else if (gap > 16384) {
2423 /* From now on don't compare against the seqnum base anymore as
2424 * at some point in the future we will wrap around and also that
2425 * much reordering is very unlikely */
2426 priv->seqnum_base = -1;
2430 expected = priv->next_in_seqnum;
2432 /* now check against our expected seqnum */
2433 if (G_LIKELY (expected != -1)) {
2436 /* now calculate gap */
2437 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2439 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2440 expected, seqnum, gap);
2442 if (G_LIKELY (gap == 0)) {
2443 /* packet is expected */
2444 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2445 do_next_seqnum = TRUE;
2447 gboolean reset = FALSE;
2449 if (!GST_CLOCK_TIME_IS_VALID (dts)) {
2450 /* We would run into calculations with GST_CLOCK_TIME_NONE below
2451 * and can't compensate for anything without DTS on RTP packets
2453 goto gap_but_no_dts;
2454 } else if (gap < 0) {
2455 /* we received an old packet */
2456 if (G_UNLIKELY (gap != -1 && gap < -RTP_MAX_MISORDER)) {
2458 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2462 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2465 /* new packet, we are missing some packets */
2466 if (G_UNLIKELY (gap >= RTP_MAX_DROPOUT)) {
2468 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2472 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2473 /* fill in the gap with EXPECTED timers */
2474 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2476 do_next_seqnum = TRUE;
2479 if (G_UNLIKELY (reset)) {
2480 GList *events = NULL, *l;
2483 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2484 rtp_jitter_buffer_flush (priv->jbuf,
2485 (GFunc) free_item_and_retain_events, &events);
2486 rtp_jitter_buffer_reset_skew (priv->jbuf);
2487 remove_all_timers (jitterbuffer);
2488 priv->discont = TRUE;
2489 priv->last_popped_seqnum = -1;
2490 priv->next_seqnum = seqnum;
2492 priv->last_in_seqnum = -1;
2493 priv->last_in_dts = -1;
2494 priv->next_in_seqnum = -1;
2496 /* Insert all sticky events again in order, otherwise we would
2497 * potentially loose STREAM_START, CAPS or SEGMENT events
2499 events = g_list_reverse (events);
2500 for (l = events; l; l = l->next) {
2501 RTPJitterBufferItem *item;
2503 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2504 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2506 g_list_free (events);
2508 JBUF_SIGNAL_EVENT (priv);
2510 /* reset spacing estimation when gap */
2511 priv->ips_rtptime = -1;
2512 priv->ips_dts = GST_CLOCK_TIME_NONE;
2514 buffers = g_list_copy (priv->gap_packets.head);
2515 g_queue_clear (&priv->gap_packets);
2517 priv->ips_rtptime = -1;
2518 priv->ips_dts = GST_CLOCK_TIME_NONE;
2519 JBUF_UNLOCK (jitterbuffer->priv);
2521 for (l = buffers; l; l = l->next) {
2522 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2524 if (ret != GST_FLOW_OK)
2527 for (; l; l = l->next)
2528 gst_buffer_unref (l->data);
2529 g_list_free (buffers);
2533 /* reset spacing estimation when gap */
2534 priv->ips_rtptime = -1;
2535 priv->ips_dts = GST_CLOCK_TIME_NONE;
2538 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2539 /* we don't know what the next_in_seqnum should be, wait for the last
2540 * possible moment to push this buffer, maybe we get an earlier seqnum
2542 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2543 do_next_seqnum = TRUE;
2544 /* take rtptime and dts to calculate packet spacing */
2545 priv->ips_rtptime = rtptime;
2546 priv->ips_dts = dts;
2549 /* We had no huge gap, let's drop all the gap packets */
2550 if (buffer != NULL) {
2551 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2552 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2553 g_queue_clear (&priv->gap_packets);
2555 GST_DEBUG_OBJECT (jitterbuffer,
2556 "Had big gap, waiting for more consecutive packets");
2557 JBUF_UNLOCK (jitterbuffer->priv);
2561 if (do_next_seqnum) {
2562 priv->last_in_seqnum = seqnum;
2563 priv->last_in_dts = dts;
2564 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2567 /* let's check if this buffer is too late, we can only accept packets with
2568 * bigger seqnum than the one we last pushed. */
2569 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2572 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2574 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2575 if (G_UNLIKELY (gap <= 0))
2579 /* let's drop oldest packet if the queue is already full and drop-on-latency
2580 * is set. We can only do this when there actually is a latency. When no
2581 * latency is set, we just pump it in the queue and let the other end push it
2582 * out as fast as possible. */
2583 if (priv->latency_ms && priv->drop_on_latency) {
2585 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2587 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2588 RTPJitterBufferItem *old_item;
2590 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2592 if (IS_DROPABLE (old_item)) {
2593 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2594 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2596 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2597 free_item (old_item);
2599 /* we might have removed some head buffers, signal the pushing thread to
2600 * see if it can push now */
2601 JBUF_SIGNAL_EVENT (priv);
2605 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2607 /* now insert the packet into the queue in sorted order. This function returns
2608 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2609 * have a duplicate. */
2610 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2615 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2617 /* we had an unhandled SR, handle it now */
2619 do_handle_sync (jitterbuffer);
2621 if (G_UNLIKELY (head)) {
2622 /* signal addition of new buffer when the _loop is waiting. */
2623 if (G_LIKELY (priv->active))
2624 JBUF_SIGNAL_EVENT (priv);
2626 /* let's unschedule and unblock any waiting buffers. We only want to do this
2627 * when the head buffer changed */
2628 if (G_UNLIKELY (priv->clock_id)) {
2629 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2630 unschedule_current_timer (jitterbuffer);
2634 GST_DEBUG_OBJECT (jitterbuffer,
2635 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2636 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2638 msg = check_buffering_percent (jitterbuffer, percent);
2644 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2651 /* this is not fatal but should be filtered earlier */
2652 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2653 ("Received invalid RTP payload, dropping"));
2654 gst_buffer_unref (buffer);
2659 GST_WARNING_OBJECT (jitterbuffer,
2660 "No clock-rate in caps!, dropping buffer");
2661 gst_buffer_unref (buffer);
2666 ret = priv->srcresult;
2667 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2668 gst_buffer_unref (buffer);
2674 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2675 gst_buffer_unref (buffer);
2680 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2681 " popped, dropping", seqnum, priv->last_popped_seqnum);
2683 gst_buffer_unref (buffer);
2688 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2690 priv->num_duplicates++;
2696 /* this is fatal as we can't compensate for gaps without DTS */
2697 GST_ELEMENT_ERROR (jitterbuffer, STREAM, DECODE, (NULL),
2698 ("Received packet without DTS after a gap"));
2699 gst_buffer_unref (buffer);
2700 ret = GST_FLOW_ERROR;
2706 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2708 guint64 ext_time, elapsed;
2710 GstRtpJitterBufferPrivate *priv;
2712 priv = jitterbuffer->priv;
2713 rtp_time = item->rtptime;
2715 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2716 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2718 if (rtp_time < priv->ext_timestamp) {
2719 ext_time = priv->ext_timestamp;
2721 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
2724 if (ext_time > priv->clock_base)
2725 elapsed = ext_time - priv->clock_base;
2729 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2734 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2735 RTPJitterBufferItem * item)
2737 guint64 total, elapsed, left, estimated;
2738 GstClockTime out_time;
2739 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2741 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2742 || priv->clock_base == -1 || priv->clock_rate <= 0)
2745 /* compute the elapsed time */
2746 elapsed = compute_elapsed (jitterbuffer, item);
2748 /* do nothing if elapsed time doesn't increment */
2749 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2752 priv->last_elapsed = elapsed;
2754 /* this is the total time we need to play */
2755 total = priv->npt_stop - priv->npt_start;
2756 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2757 GST_TIME_ARGS (total));
2759 /* this is how much time there is left */
2760 if (total > elapsed)
2761 left = total - elapsed;
2765 /* if we have less time left that the size of the buffer, we will not
2766 * be able to keep it filled, disabled buffering then */
2767 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2768 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2769 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2770 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2773 /* this is the current time as running-time */
2774 out_time = item->dts;
2777 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2779 /* if there is almost nothing left,
2780 * we may never advance enough to end up in the above case */
2781 if (total < GST_SECOND)
2782 estimated = GST_SECOND;
2786 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2787 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2789 if (estimated != -1 && priv->estimated_eos != estimated) {
2790 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2791 priv->estimated_eos = estimated;
2795 /* take a buffer from the queue and push it */
2796 static GstFlowReturn
2797 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2799 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2800 GstFlowReturn result = GST_FLOW_OK;
2801 RTPJitterBufferItem *item;
2802 GstBuffer *outbuf = NULL;
2803 GstEvent *outevent = NULL;
2804 GstQuery *outquery = NULL;
2805 GstClockTime dts, pts;
2807 gboolean do_push = TRUE;
2811 /* when we get here we are ready to pop and push the buffer */
2812 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2816 case ITEM_TYPE_BUFFER:
2818 /* we need to make writable to change the flags and timestamps */
2819 outbuf = gst_buffer_make_writable (item->data);
2821 if (G_UNLIKELY (priv->discont)) {
2822 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2823 * into the jitterbuffer so we can modify now. */
2824 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2825 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2826 priv->discont = FALSE;
2828 if (G_UNLIKELY (priv->ts_discont)) {
2829 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2830 priv->ts_discont = FALSE;
2834 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2836 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2838 /* apply timestamp with offset to buffer now */
2839 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2840 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2842 /* update the elapsed time when we need to check against the npt stop time. */
2843 update_estimated_eos (jitterbuffer, item);
2845 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2847 case ITEM_TYPE_LOST:
2848 priv->discont = TRUE;
2852 case ITEM_TYPE_EVENT:
2853 outevent = item->data;
2855 case ITEM_TYPE_QUERY:
2856 outquery = item->data;
2860 /* now we are ready to push the buffer. Save the seqnum and release the lock
2861 * so the other end can push stuff in the queue again. */
2863 priv->last_popped_seqnum = seqnum;
2864 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2866 msg = check_buffering_percent (jitterbuffer, percent);
2873 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2876 case ITEM_TYPE_BUFFER:
2878 GST_DEBUG_OBJECT (jitterbuffer,
2879 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2880 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2881 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2882 result = gst_pad_push (priv->srcpad, outbuf);
2884 JBUF_LOCK_CHECK (priv, out_flushing);
2886 case ITEM_TYPE_LOST:
2887 case ITEM_TYPE_EVENT:
2888 /* We got not enough consecutive packets with a huge gap, we can
2889 * as well just drop them here now on EOS */
2890 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
2891 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
2892 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2893 g_queue_clear (&priv->gap_packets);
2896 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2897 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
2900 gst_pad_push_event (priv->srcpad, outevent);
2902 gst_event_unref (outevent);
2904 result = GST_FLOW_OK;
2906 JBUF_LOCK_CHECK (priv, out_flushing);
2908 case ITEM_TYPE_QUERY:
2912 res = gst_pad_peer_query (priv->srcpad, outquery);
2914 JBUF_LOCK_CHECK (priv, out_flushing);
2915 result = GST_FLOW_OK;
2916 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
2917 JBUF_SIGNAL_QUERY (priv, res);
2926 return priv->srcresult;
2930 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2932 /* Peek a buffer and compare the seqnum to the expected seqnum.
2933 * If all is fine, the buffer is pushed.
2934 * If something is wrong, we wait for some event
2936 static GstFlowReturn
2937 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2939 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2940 GstFlowReturn result = GST_FLOW_OK;
2941 RTPJitterBufferItem *item;
2943 guint32 next_seqnum;
2946 /* only push buffers when PLAYING and active and not buffering */
2947 if (priv->blocked || !priv->active ||
2948 rtp_jitter_buffer_is_buffering (priv->jbuf))
2949 return GST_FLOW_WAIT;
2952 /* peek a buffer, we're just looking at the sequence number.
2953 * If all is fine, we'll pop and push it. If the sequence number is wrong we
2954 * wait for a timeout or something to change.
2955 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
2956 item = rtp_jitter_buffer_peek (priv->jbuf);
2960 /* get the seqnum and the next expected seqnum */
2961 seqnum = item->seqnum;
2965 next_seqnum = priv->next_seqnum;
2967 /* get the gap between this and the previous packet. If we don't know the
2968 * previous packet seqnum assume no gap. */
2969 if (G_UNLIKELY (next_seqnum == -1)) {
2970 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2971 /* we don't know what the next_seqnum should be, the chain function should
2972 * have scheduled a DEADLINE timer that will increment next_seqnum when it
2973 * fires, so wait for that */
2974 result = GST_FLOW_WAIT;
2976 /* else calculate GAP */
2977 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2979 if (G_LIKELY (gap == 0)) {
2981 /* no missing packet, pop and push */
2982 result = pop_and_push_next (jitterbuffer, seqnum);
2983 } else if (G_UNLIKELY (gap < 0)) {
2984 RTPJitterBufferItem *item;
2985 /* if we have a packet that we already pushed or considered dropped, pop it
2986 * off and get the next packet */
2987 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2988 seqnum, next_seqnum);
2989 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2993 /* the chain function has scheduled timers to request retransmission or
2994 * when to consider the packet lost, wait for that */
2995 GST_DEBUG_OBJECT (jitterbuffer,
2996 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2997 next_seqnum, seqnum, gap);
2998 result = GST_FLOW_WAIT;
3005 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3007 result = GST_FLOW_EOS;
3009 result = GST_FLOW_WAIT;
3015 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3017 GstClockTime rtx_retry_timeout;
3018 GstClockTime rtx_min_retry_timeout;
3020 if (priv->rtx_retry_timeout == -1) {
3021 if (priv->avg_rtx_rtt == 0)
3022 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3024 /* we want to ask for a retransmission after we waited for a
3025 * complete RTT and the additional jitter */
3026 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3028 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3030 /* make sure we don't retry too often. On very low latency networks,
3031 * the RTT and jitter can be very low. */
3032 if (priv->rtx_min_retry_timeout == -1) {
3033 rtx_min_retry_timeout = priv->packet_spacing;
3035 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3037 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3039 return rtx_retry_timeout;
3043 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3044 GstClockTime rtx_retry_timeout)
3046 GstClockTime rtx_retry_period;
3048 if (priv->rtx_retry_period == -1) {
3049 /* we retry up to the configured jitterbuffer size but leaving some
3050 * room for the retransmission to arrive in time */
3051 if (rtx_retry_timeout > priv->latency_ns) {
3052 rtx_retry_period = 0;
3054 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3057 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3059 return rtx_retry_period;
3062 /* the timeout for when we expected a packet expired */
3064 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3067 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3069 guint delay, delay_ms, avg_rtx_rtt_ms;
3070 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3071 GstClockTime rtx_retry_period;
3072 GstClockTime rtx_retry_timeout;
3075 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3076 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3078 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3079 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3081 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3082 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3083 GST_TIME_ARGS (rtx_retry_period));
3085 delay = timer->rtx_delay + timer->rtx_retry;
3087 delay_ms = GST_TIME_AS_MSECONDS (delay);
3088 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3089 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3090 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3092 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3093 gst_structure_new ("GstRTPRetransmissionRequest",
3094 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3095 "running-time", G_TYPE_UINT64, timer->rtx_base,
3096 "delay", G_TYPE_UINT, delay_ms,
3097 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3098 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3099 "period", G_TYPE_UINT, rtx_retry_period_ms,
3100 "deadline", G_TYPE_UINT, priv->latency_ms,
3101 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3102 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3104 priv->num_rtx_requests++;
3105 timer->num_rtx_retry++;
3107 GST_OBJECT_LOCK (jitterbuffer);
3108 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3109 timer->rtx_last = gst_clock_get_time (clock);
3110 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3112 timer->rtx_last = now;
3114 GST_OBJECT_UNLOCK (jitterbuffer);
3116 /* calculate the timeout for the next retransmission attempt */
3117 timer->rtx_retry += rtx_retry_timeout;
3118 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3119 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3120 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3121 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3122 if ((priv->rtx_max_retries != -1
3123 && timer->num_rtx_retry >= priv->rtx_max_retries)
3124 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3125 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3126 /* too many retransmission request, we now convert the timer
3127 * to a lost timer, leave the num_rtx_retry as it is for stats */
3128 timer->type = TIMER_TYPE_LOST;
3129 timer->rtx_delay = 0;
3130 timer->rtx_retry = 0;
3132 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3133 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3136 gst_pad_push_event (priv->sinkpad, event);
3142 /* a packet is lost */
3144 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3147 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3148 GstClockTime duration, timestamp;
3149 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3150 gboolean late, head;
3152 RTPJitterBufferItem *item;
3154 seqnum = timer->seqnum;
3155 timestamp = apply_offset (jitterbuffer, timer->timeout);
3156 duration = timer->duration;
3157 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3158 duration = priv->packet_spacing;
3159 lost_packets = MAX (timer->num, 1);
3160 late = timer->num > 0;
3161 num_rtx_retry = timer->num_rtx_retry;
3163 /* we had a gap and thus we lost some packets. Create an event for this. */
3164 if (lost_packets > 1)
3165 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3166 seqnum + lost_packets - 1);
3168 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3170 priv->num_late += lost_packets;
3171 priv->num_rtx_failed += num_rtx_retry;
3173 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3175 /* we now only accept seqnum bigger than this */
3176 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3177 priv->next_in_seqnum = next_in_seqnum;
3179 /* create paket lost event */
3180 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3181 gst_structure_new ("GstRTPPacketLost",
3182 "seqnum", G_TYPE_UINT, (guint) seqnum,
3183 "timestamp", G_TYPE_UINT64, timestamp,
3184 "duration", G_TYPE_UINT64, duration,
3185 "late", G_TYPE_BOOLEAN, late,
3186 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3188 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3189 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3191 /* remove timer now */
3192 remove_timer (jitterbuffer, timer);
3194 JBUF_SIGNAL_EVENT (priv);
3200 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3203 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3205 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3206 remove_timer (jitterbuffer, timer);
3208 /* there was no EOS in the buffer, put one in there now */
3209 queue_event (jitterbuffer, gst_event_new_eos ());
3211 JBUF_SIGNAL_EVENT (priv);
3217 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3220 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3222 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3224 /* timer seqnum might have been obsoleted by caps seqnum-base,
3225 * only mess with current ongoing seqnum if still unknown */
3226 if (priv->next_seqnum == -1)
3227 priv->next_seqnum = timer->seqnum;
3228 remove_timer (jitterbuffer, timer);
3229 JBUF_SIGNAL_EVENT (priv);
3235 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3238 gboolean removed = FALSE;
3240 switch (timer->type) {
3241 case TIMER_TYPE_EXPECTED:
3242 removed = do_expected_timeout (jitterbuffer, timer, now);
3244 case TIMER_TYPE_LOST:
3245 removed = do_lost_timeout (jitterbuffer, timer, now);
3247 case TIMER_TYPE_DEADLINE:
3248 removed = do_deadline_timeout (jitterbuffer, timer, now);
3250 case TIMER_TYPE_EOS:
3251 removed = do_eos_timeout (jitterbuffer, timer, now);
3257 /* called when we need to wait for the next timeout.
3259 * We loop over the array of recorded timeouts and wait for the earliest one.
3260 * When it timed out, do the logic associated with the timer.
3262 * If there are no timers, we wait on a gcond until something new happens.
3265 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3267 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3268 GstClockTime now = 0;
3271 while (priv->timer_running) {
3272 TimerData *timer = NULL;
3273 GstClockTime timer_timeout = -1;
3276 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3277 GST_TIME_ARGS (now));
3279 len = priv->timers->len;
3280 for (i = 0; i < len; i++) {
3281 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3282 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3283 gboolean save_best = FALSE;
3285 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3286 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3288 /* find the smallest timeout */
3289 if (timer == NULL) {
3291 } else if (timer_timeout == -1) {
3292 /* we already have an immediate timeout, the new timer must be an
3293 * immediate timer with smaller seqnum to become the best */
3294 if (test_timeout == -1
3295 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3296 timer->seqnum) > 0))
3298 } else if (test_timeout == -1) {
3299 /* first immediate timer */
3301 } else if (test_timeout < timer_timeout) {
3304 } else if (test_timeout == timer_timeout
3305 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3306 timer->seqnum) > 0)) {
3307 /* same timer, smaller seqnum */
3311 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3313 timer_timeout = test_timeout;
3316 if (timer && !priv->blocked) {
3318 GstClockTime sync_time;
3321 GstClockTimeDiff clock_jitter;
3323 if (timer_timeout == -1 || timer_timeout <= now) {
3324 do_timeout (jitterbuffer, timer, now);
3325 /* check here, do_timeout could have released the lock */
3326 if (!priv->timer_running)
3331 GST_OBJECT_LOCK (jitterbuffer);
3332 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3334 GST_OBJECT_UNLOCK (jitterbuffer);
3335 /* let's just push if there is no clock */
3336 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3337 now = timer_timeout;
3341 /* prepare for sync against clock */
3342 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3343 /* add latency of peer to get input time */
3344 sync_time += priv->peer_latency;
3346 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3347 " with sync time %" GST_TIME_FORMAT,
3348 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3350 /* create an entry for the clock */
3351 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3352 priv->timer_timeout = timer_timeout;
3353 priv->timer_seqnum = timer->seqnum;
3354 GST_OBJECT_UNLOCK (jitterbuffer);
3356 /* release the lock so that the other end can push stuff or unlock */
3359 ret = gst_clock_id_wait (id, &clock_jitter);
3362 if (!priv->timer_running) {
3363 gst_clock_id_unref (id);
3364 priv->clock_id = NULL;
3368 if (ret != GST_CLOCK_UNSCHEDULED) {
3369 now = timer_timeout + MAX (clock_jitter, 0);
3370 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3371 ret, priv->timer_seqnum, clock_jitter);
3373 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3375 /* and free the entry */
3376 gst_clock_id_unref (id);
3377 priv->clock_id = NULL;
3379 /* no timers, wait for activity */
3380 JBUF_WAIT_TIMER (priv);
3385 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3390 * This funcion implements the main pushing loop on the source pad.
3392 * It first tries to push as many buffers as possible. If there is a seqnum
3393 * mismatch, we wait for the next timeouts.
3396 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3398 GstRtpJitterBufferPrivate *priv;
3399 GstFlowReturn result = GST_FLOW_OK;
3401 priv = jitterbuffer->priv;
3403 JBUF_LOCK_CHECK (priv, flushing);
3405 result = handle_next_buffer (jitterbuffer);
3406 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3407 /* now wait for the next event */
3408 JBUF_WAIT_EVENT (priv, flushing);
3409 result = GST_FLOW_OK;
3412 while (result == GST_FLOW_OK);
3413 /* store result for upstream */
3414 priv->srcresult = result;
3415 /* if we get here we need to pause */
3421 result = priv->srcresult;
3428 JBUF_SIGNAL_QUERY (priv, FALSE);
3431 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3432 gst_flow_get_name (result));
3433 gst_pad_pause_task (priv->srcpad);
3434 if (result == GST_FLOW_EOS) {
3435 event = gst_event_new_eos ();
3436 gst_pad_push_event (priv->srcpad, event);
3442 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3443 * some sanity checks and then emit the handle-sync signal with the parameters.
3444 * This function must be called with the LOCK */
3446 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3448 GstRtpJitterBufferPrivate *priv;
3449 guint64 base_rtptime, base_time;
3451 guint64 last_rtptime;
3453 guint64 ext_rtptime, diff;
3454 gboolean valid = TRUE, keep = FALSE;
3456 priv = jitterbuffer->priv;
3458 /* get the last values from the jitterbuffer */
3459 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3460 &clock_rate, &last_rtptime);
3462 clock_base = priv->clock_base;
3463 ext_rtptime = priv->ext_rtptime;
3465 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3466 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3467 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3468 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3470 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3471 /* we keep this SR packet for later. When we get a valid RTP packet the
3472 * above values will be set and we can try to use the SR packet */
3473 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3476 /* we can't accept anything that happened before we did the last resync */
3477 if (base_rtptime > ext_rtptime) {
3478 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3481 /* the SR RTP timestamp must be something close to what we last observed
3482 * in the jitterbuffer */
3483 if (ext_rtptime > last_rtptime) {
3484 /* check how far ahead it is to our RTP timestamps */
3485 diff = ext_rtptime - last_rtptime;
3486 /* if bigger than 1 second, we drop it */
3487 if (diff > clock_rate) {
3488 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3489 /* should drop this, but some RTSP servers end up with bogus
3490 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3491 * so still trigger rptbin sync but invalidate RTCP data
3492 * (sync might use other methods) */
3495 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3496 G_GUINT64_FORMAT, last_rtptime, diff);
3502 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3506 s = gst_structure_new ("application/x-rtp-sync",
3507 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3508 "base-time", G_TYPE_UINT64, base_time,
3509 "clock-rate", G_TYPE_UINT, clock_rate,
3510 "clock-base", G_TYPE_UINT64, clock_base,
3511 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3512 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3514 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3515 gst_buffer_replace (&priv->last_sr, NULL);
3517 g_signal_emit (jitterbuffer,
3518 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3520 gst_structure_free (s);
3522 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3523 gst_buffer_replace (&priv->last_sr, NULL);
3527 static GstFlowReturn
3528 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3531 GstRtpJitterBuffer *jitterbuffer;
3532 GstRtpJitterBufferPrivate *priv;
3533 GstFlowReturn ret = GST_FLOW_OK;
3535 GstRTCPPacket packet;
3536 guint64 ext_rtptime;
3538 GstRTCPBuffer rtcp = { NULL, };
3540 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3542 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3543 goto invalid_buffer;
3545 priv = jitterbuffer->priv;
3547 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3549 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3552 /* first packet must be SR or RR or else the validate would have failed */
3553 switch (gst_rtcp_packet_get_type (&packet)) {
3554 case GST_RTCP_TYPE_SR:
3555 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3561 gst_rtcp_buffer_unmap (&rtcp);
3563 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3566 /* convert the RTP timestamp to our extended timestamp, using the same offset
3567 * we used in the jitterbuffer */
3568 ext_rtptime = priv->jbuf->ext_rtptime;
3569 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3571 priv->ext_rtptime = ext_rtptime;
3572 gst_buffer_replace (&priv->last_sr, buffer);
3574 do_handle_sync (jitterbuffer);
3578 gst_buffer_unref (buffer);
3584 /* this is not fatal but should be filtered earlier */
3585 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3586 ("Received invalid RTCP payload, dropping"));
3592 /* this is not fatal but should be filtered earlier */
3593 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3594 ("Received empty RTCP payload, dropping"));
3595 gst_rtcp_buffer_unmap (&rtcp);
3601 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3602 gst_rtcp_buffer_unmap (&rtcp);
3609 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3612 gboolean res = FALSE;
3613 GstRtpJitterBuffer *jitterbuffer;
3614 GstRtpJitterBufferPrivate *priv;
3616 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3617 priv = jitterbuffer->priv;
3619 switch (GST_QUERY_TYPE (query)) {
3620 case GST_QUERY_CAPS:
3622 GstCaps *filter, *caps;
3624 gst_query_parse_caps (query, &filter);
3625 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3626 gst_query_set_caps_result (query, caps);
3627 gst_caps_unref (caps);
3632 if (GST_QUERY_IS_SERIALIZED (query)) {
3633 RTPJitterBufferItem *item;
3636 JBUF_LOCK_CHECK (priv, out_flushing);
3637 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3638 RTP_JITTER_BUFFER_MODE_BUFFER) {
3639 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3640 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3641 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3643 JBUF_SIGNAL_EVENT (priv);
3644 JBUF_WAIT_QUERY (priv, out_flushing);
3645 res = priv->last_query;
3647 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3652 res = gst_pad_query_default (pad, parent, query);
3660 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3668 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3671 GstRtpJitterBuffer *jitterbuffer;
3672 GstRtpJitterBufferPrivate *priv;
3673 gboolean res = FALSE;
3675 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3676 priv = jitterbuffer->priv;
3678 switch (GST_QUERY_TYPE (query)) {
3679 case GST_QUERY_LATENCY:
3681 /* We need to send the query upstream and add the returned latency to our
3683 GstClockTime min_latency, max_latency;
3685 GstClockTime our_latency;
3687 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3688 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3690 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3691 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3692 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3694 /* store this so that we can safely sync on the peer buffers. */
3696 priv->peer_latency = min_latency;
3697 our_latency = priv->latency_ns;
3700 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3701 GST_TIME_ARGS (our_latency));
3703 /* we add some latency but can buffer an infinite amount of time */
3704 min_latency += our_latency;
3707 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3708 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3709 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3711 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3715 case GST_QUERY_POSITION:
3717 GstClockTime start, last_out;
3720 gst_query_parse_position (query, &fmt, NULL);
3721 if (fmt != GST_FORMAT_TIME) {
3722 res = gst_pad_query_default (pad, parent, query);
3727 start = priv->npt_start;
3728 last_out = priv->last_out_time;
3731 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3732 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3733 GST_TIME_ARGS (last_out));
3735 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3736 /* bring 0-based outgoing time to stream time */
3737 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3740 res = gst_pad_query_default (pad, parent, query);
3744 case GST_QUERY_CAPS:
3746 GstCaps *filter, *caps;
3748 gst_query_parse_caps (query, &filter);
3749 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3750 gst_query_set_caps_result (query, caps);
3751 gst_caps_unref (caps);
3756 res = gst_pad_query_default (pad, parent, query);
3764 gst_rtp_jitter_buffer_set_property (GObject * object,
3765 guint prop_id, const GValue * value, GParamSpec * pspec)
3767 GstRtpJitterBuffer *jitterbuffer;
3768 GstRtpJitterBufferPrivate *priv;
3770 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3771 priv = jitterbuffer->priv;
3776 guint new_latency, old_latency;
3778 new_latency = g_value_get_uint (value);
3781 old_latency = priv->latency_ms;
3782 priv->latency_ms = new_latency;
3783 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3784 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3787 /* post message if latency changed, this will inform the parent pipeline
3788 * that a latency reconfiguration is possible/needed. */
3789 if (new_latency != old_latency) {
3790 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3791 GST_TIME_ARGS (new_latency * GST_MSECOND));
3793 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3794 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3798 case PROP_DROP_ON_LATENCY:
3800 priv->drop_on_latency = g_value_get_boolean (value);
3803 case PROP_TS_OFFSET:
3805 priv->ts_offset = g_value_get_int64 (value);
3806 priv->ts_discont = TRUE;
3811 priv->do_lost = g_value_get_boolean (value);
3816 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3819 case PROP_DO_RETRANSMISSION:
3821 priv->do_retransmission = g_value_get_boolean (value);
3824 case PROP_RTX_NEXT_SEQNUM:
3826 priv->rtx_next_seqnum = g_value_get_boolean (value);
3829 case PROP_RTX_DELAY:
3831 priv->rtx_delay = g_value_get_int (value);
3834 case PROP_RTX_MIN_DELAY:
3836 priv->rtx_min_delay = g_value_get_uint (value);
3839 case PROP_RTX_DELAY_REORDER:
3841 priv->rtx_delay_reorder = g_value_get_int (value);
3844 case PROP_RTX_RETRY_TIMEOUT:
3846 priv->rtx_retry_timeout = g_value_get_int (value);
3849 case PROP_RTX_MIN_RETRY_TIMEOUT:
3851 priv->rtx_min_retry_timeout = g_value_get_int (value);
3854 case PROP_RTX_RETRY_PERIOD:
3856 priv->rtx_retry_period = g_value_get_int (value);
3859 case PROP_RTX_MAX_RETRIES:
3861 priv->rtx_max_retries = g_value_get_int (value);
3865 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3871 gst_rtp_jitter_buffer_get_property (GObject * object,
3872 guint prop_id, GValue * value, GParamSpec * pspec)
3874 GstRtpJitterBuffer *jitterbuffer;
3875 GstRtpJitterBufferPrivate *priv;
3877 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3878 priv = jitterbuffer->priv;
3883 g_value_set_uint (value, priv->latency_ms);
3886 case PROP_DROP_ON_LATENCY:
3888 g_value_set_boolean (value, priv->drop_on_latency);
3891 case PROP_TS_OFFSET:
3893 g_value_set_int64 (value, priv->ts_offset);
3898 g_value_set_boolean (value, priv->do_lost);
3903 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3911 if (priv->srcresult != GST_FLOW_OK)
3914 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3916 g_value_set_int (value, percent);
3920 case PROP_DO_RETRANSMISSION:
3922 g_value_set_boolean (value, priv->do_retransmission);
3925 case PROP_RTX_NEXT_SEQNUM:
3927 g_value_set_boolean (value, priv->rtx_next_seqnum);
3930 case PROP_RTX_DELAY:
3932 g_value_set_int (value, priv->rtx_delay);
3935 case PROP_RTX_MIN_DELAY:
3937 g_value_set_uint (value, priv->rtx_min_delay);
3940 case PROP_RTX_DELAY_REORDER:
3942 g_value_set_int (value, priv->rtx_delay_reorder);
3945 case PROP_RTX_RETRY_TIMEOUT:
3947 g_value_set_int (value, priv->rtx_retry_timeout);
3950 case PROP_RTX_MIN_RETRY_TIMEOUT:
3952 g_value_set_int (value, priv->rtx_min_retry_timeout);
3955 case PROP_RTX_RETRY_PERIOD:
3957 g_value_set_int (value, priv->rtx_retry_period);
3960 case PROP_RTX_MAX_RETRIES:
3962 g_value_set_int (value, priv->rtx_max_retries);
3966 g_value_take_boxed (value,
3967 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
3970 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3975 static GstStructure *
3976 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
3980 JBUF_LOCK (jbuf->priv);
3981 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
3982 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
3983 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
3984 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
3985 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
3986 JBUF_UNLOCK (jbuf->priv);