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_DELAY -1
132 #define DEFAULT_RTX_MIN_DELAY 0
133 #define DEFAULT_RTX_DELAY_REORDER 3
134 #define DEFAULT_RTX_RETRY_TIMEOUT -1
135 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
136 #define DEFAULT_RTX_RETRY_PERIOD -1
138 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
139 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
145 PROP_DROP_ON_LATENCY,
150 PROP_DO_RETRANSMISSION,
153 PROP_RTX_DELAY_REORDER,
154 PROP_RTX_RETRY_TIMEOUT,
155 PROP_RTX_MIN_RETRY_TIMEOUT,
156 PROP_RTX_RETRY_PERIOD,
161 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
163 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
165 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
168 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
170 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
171 GST_DEBUG ("waiting timer"); \
172 (priv)->waiting_timer = TRUE; \
173 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
174 (priv)->waiting_timer = FALSE; \
175 GST_DEBUG ("waiting timer done"); \
177 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
178 if (G_UNLIKELY ((priv)->waiting_timer)) { \
179 GST_DEBUG ("signal timer"); \
180 g_cond_signal (&(priv)->jbuf_timer); \
184 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
185 GST_DEBUG ("waiting event"); \
186 (priv)->waiting_event = TRUE; \
187 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
188 (priv)->waiting_event = FALSE; \
189 GST_DEBUG ("waiting event done"); \
190 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
193 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
194 if (G_UNLIKELY ((priv)->waiting_event)) { \
195 GST_DEBUG ("signal event"); \
196 g_cond_signal (&(priv)->jbuf_event); \
200 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
201 GST_DEBUG ("waiting query"); \
202 (priv)->waiting_query = TRUE; \
203 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
204 (priv)->waiting_query = FALSE; \
205 GST_DEBUG ("waiting query done"); \
206 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
209 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
210 (priv)->last_query = res; \
211 if (G_UNLIKELY ((priv)->waiting_query)) { \
212 GST_DEBUG ("signal query"); \
213 g_cond_signal (&(priv)->jbuf_query); \
218 struct _GstRtpJitterBufferPrivate
220 GstPad *sinkpad, *srcpad;
223 RTPJitterBuffer *jbuf;
225 gboolean waiting_timer;
227 gboolean waiting_event;
229 gboolean waiting_query;
237 gboolean timer_running;
238 GThread *timer_thread;
243 gboolean drop_on_latency;
246 gboolean do_retransmission;
249 gint rtx_delay_reorder;
250 gint rtx_retry_timeout;
251 gint rtx_min_retry_timeout;
252 gint rtx_retry_period;
254 /* the last seqnum we pushed out */
255 guint32 last_popped_seqnum;
256 /* the next expected seqnum we push */
258 /* seqnum-base, if known */
260 /* last output time */
261 GstClockTime last_out_time;
262 /* last valid input timestamp and rtptime pair */
263 GstClockTime ips_dts;
265 GstClockTime packet_spacing;
267 /* the next expected seqnum we receive */
268 GstClockTime last_in_dts;
269 guint32 last_in_seqnum;
270 guint32 next_in_seqnum;
274 /* start and stop ranges */
275 GstClockTime npt_start;
276 GstClockTime npt_stop;
277 guint64 ext_timestamp;
278 guint64 last_elapsed;
279 guint64 estimated_eos;
286 /* clock rate and rtp timestamp offset */
290 gint64 prev_ts_offset;
292 /* when we are shutting down */
293 GstFlowReturn srcresult;
299 GstClockTime timer_timeout;
300 guint16 timer_seqnum;
301 /* the latency of the upstream peer, we have to take this into account when
302 * synchronizing the buffers. */
303 GstClockTime peer_latency;
307 /* some accounting */
309 guint64 num_duplicates;
310 guint64 num_rtx_requests;
311 guint64 num_rtx_success;
312 guint64 num_rtx_failed;
317 GstClockTime last_dts;
318 guint64 last_rtptime;
319 GstClockTime avg_jitter;
336 GstClockTime timeout;
337 GstClockTime duration;
338 GstClockTime rtx_base;
339 GstClockTime rtx_delay;
340 GstClockTime rtx_retry;
341 GstClockTime rtx_last;
345 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
346 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
347 GstRtpJitterBufferPrivate))
349 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
350 GST_STATIC_PAD_TEMPLATE ("sink",
353 GST_STATIC_CAPS ("application/x-rtp"
354 /* "clock-rate = (int) [ 1, 2147483647 ], "
355 * "payload = (int) , "
356 * "encoding-name = (string) "
360 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
361 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
364 GST_STATIC_CAPS ("application/x-rtcp")
367 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
368 GST_STATIC_PAD_TEMPLATE ("src",
371 GST_STATIC_CAPS ("application/x-rtp"
372 /* "payload = (int) , "
373 * "clock-rate = (int) , "
374 * "encoding-name = (string) "
378 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
380 #define gst_rtp_jitter_buffer_parent_class parent_class
381 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
383 /* object overrides */
384 static void gst_rtp_jitter_buffer_set_property (GObject * object,
385 guint prop_id, const GValue * value, GParamSpec * pspec);
386 static void gst_rtp_jitter_buffer_get_property (GObject * object,
387 guint prop_id, GValue * value, GParamSpec * pspec);
388 static void gst_rtp_jitter_buffer_finalize (GObject * object);
390 /* element overrides */
391 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
392 * element, GstStateChange transition);
393 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
394 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
395 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
397 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
400 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
401 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
404 /* sinkpad overrides */
405 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
406 GstObject * parent, GstEvent * event);
407 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
408 GstObject * parent, GstBuffer * buffer);
410 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
411 GstObject * parent, GstEvent * event);
412 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
413 GstObject * parent, GstBuffer * buffer);
415 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
416 GstObject * parent, GstQuery * query);
418 /* srcpad overrides */
419 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
420 GstObject * parent, GstEvent * event);
421 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
422 GstObject * parent, GstPadMode mode, gboolean active);
423 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
424 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
425 GstObject * parent, GstQuery * query);
428 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
430 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
431 gboolean active, guint64 base_time);
432 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
434 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
435 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
437 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
439 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
443 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
445 GObjectClass *gobject_class;
446 GstElementClass *gstelement_class;
448 gobject_class = (GObjectClass *) klass;
449 gstelement_class = (GstElementClass *) klass;
451 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
453 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
455 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
456 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
459 * GstRtpJitterBuffer:latency:
461 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
462 * for at most this time.
464 g_object_class_install_property (gobject_class, PROP_LATENCY,
465 g_param_spec_uint ("latency", "Buffer latency in ms",
466 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
467 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
469 * GstRtpJitterBuffer:drop-on-latency:
471 * Drop oldest buffers when the queue is completely filled.
473 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
474 g_param_spec_boolean ("drop-on-latency",
475 "Drop buffers when maximum latency is reached",
476 "Tells the jitterbuffer to never exceed the given latency in size",
477 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
479 * GstRtpJitterBuffer:ts-offset:
481 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
482 * This is mainly used to ensure interstream synchronisation.
484 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
485 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
486 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
487 G_MAXINT64, DEFAULT_TS_OFFSET,
488 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
491 * GstRtpJitterBuffer:do-lost:
493 * Send out a GstRTPPacketLost event downstream when a packet is considered
496 g_object_class_install_property (gobject_class, PROP_DO_LOST,
497 g_param_spec_boolean ("do-lost", "Do Lost",
498 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
499 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
502 * GstRtpJitterBuffer:mode:
504 * Control the buffering and timestamping mode used by the jitterbuffer.
506 g_object_class_install_property (gobject_class, PROP_MODE,
507 g_param_spec_enum ("mode", "Mode",
508 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
509 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
511 * GstRtpJitterBuffer:percent:
513 * The percent of the jitterbuffer that is filled.
515 g_object_class_install_property (gobject_class, PROP_PERCENT,
516 g_param_spec_int ("percent", "percent",
517 "The buffer filled percent", 0, 100,
518 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
520 * GstRtpJitterBuffer:do-retransmission:
522 * Send out a GstRTPRetransmission event upstream when a packet is considered
523 * late and should be retransmitted.
527 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
528 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
529 "Send retransmission events upstream when a packet is late",
530 DEFAULT_DO_RETRANSMISSION,
531 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
534 * GstRtpJitterBuffer:rtx-delay:
536 * When a packet did not arrive at the expected time, wait this extra amount
537 * of time before sending a retransmission event.
539 * When -1 is used, the max jitter will be used as extra delay.
543 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
544 g_param_spec_int ("rtx-delay", "RTX Delay",
545 "Extra time in ms to wait before sending retransmission "
546 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
547 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
550 * GstRtpJitterBuffer:rtx-min-delay:
552 * When a packet did not arrive at the expected time, wait at least this extra amount
553 * of time before sending a retransmission event.
557 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
558 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
559 "Minimum time in ms to wait before sending retransmission "
560 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
561 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
563 * GstRtpJitterBuffer:rtx-delay-reorder:
565 * Assume that a retransmission event should be sent when we see
566 * this much packet reordering.
568 * When -1 is used, the value will be estimated based on observed packet
573 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
574 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
575 "Sending retransmission event when this much reordering (-1 automatic)",
576 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
577 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
579 * GstRtpJitterBuffer::rtx-retry-timeout:
581 * When no packet has been received after sending a retransmission event
582 * for this time, retry sending a retransmission event.
584 * When -1 is used, the value will be estimated based on observed round
589 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
590 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
591 "Retry sending a transmission event after this timeout in "
592 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
593 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
595 * GstRtpJitterBuffer::rtx-min-retry-timeout:
597 * The minimum amount of time between retry timeouts. When
598 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
599 * minimum interval between retry timeouts.
601 * When -1 is used, the value will be estimated based on the
606 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
607 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
608 "Minimum timeout between sending a transmission event in "
609 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
610 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
612 * GstRtpJitterBuffer:rtx-retry-period:
614 * The amount of time to try to get a retransmission.
616 * When -1 is used, the value will be estimated based on the jitterbuffer
617 * latency and the observed round trip time.
621 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
622 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
623 "Try to get a retransmission for this many ms "
624 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
625 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
627 * GstRtpJitterBuffer:stats:
629 * Various jitterbuffer statistics. This property returns a GstStructure
630 * with name application/x-rtp-jitterbuffer-stats with the following fields:
632 * "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
633 * "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
634 * "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
635 * "rtx-rtt" G_TYPE_UINT64 Average round trip time per RTX
639 g_object_class_install_property (gobject_class, PROP_STATS,
640 g_param_spec_boxed ("stats", "Statistics",
641 "Various statistics", GST_TYPE_STRUCTURE,
642 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
645 * GstRtpJitterBuffer::request-pt-map:
646 * @buffer: the object which received the signal
649 * Request the payload type as #GstCaps for @pt.
651 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
652 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
653 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
654 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
655 GST_TYPE_CAPS, 1, G_TYPE_UINT);
657 * GstRtpJitterBuffer::handle-sync:
658 * @buffer: the object which received the signal
659 * @struct: a GstStructure containing sync values.
661 * Be notified of new sync values.
663 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
664 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
665 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
666 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
667 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
670 * GstRtpJitterBuffer::on-npt-stop:
671 * @buffer: the object which received the signal
673 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
674 * the npt-stop position.
676 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
677 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
678 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
679 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
680 G_TYPE_NONE, 0, G_TYPE_NONE);
683 * GstRtpJitterBuffer::clear-pt-map:
684 * @buffer: the object which received the signal
686 * Invalidate the clock-rate as obtained with the
687 * #GstRtpJitterBuffer::request-pt-map signal.
689 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
690 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
691 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
692 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
693 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
696 * GstRtpJitterBuffer::set-active:
697 * @buffer: the object which received the signal
699 * Start pushing out packets with the given base time. This signal is only
700 * useful in buffering mode.
702 * Returns: the time of the last pushed packet.
704 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
705 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
706 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
707 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
708 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
711 gstelement_class->change_state =
712 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
713 gstelement_class->request_new_pad =
714 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
715 gstelement_class->release_pad =
716 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
717 gstelement_class->provide_clock =
718 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
720 gst_element_class_add_pad_template (gstelement_class,
721 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
722 gst_element_class_add_pad_template (gstelement_class,
723 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
724 gst_element_class_add_pad_template (gstelement_class,
725 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
727 gst_element_class_set_static_metadata (gstelement_class,
728 "RTP packet jitter-buffer", "Filter/Network/RTP",
729 "A buffer that deals with network jitter and other transmission faults",
730 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
731 "Wim Taymans <wim.taymans@gmail.com>");
733 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
734 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
736 GST_DEBUG_CATEGORY_INIT
737 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
741 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
743 GstRtpJitterBufferPrivate *priv;
745 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
746 jitterbuffer->priv = priv;
748 priv->latency_ms = DEFAULT_LATENCY_MS;
749 priv->latency_ns = priv->latency_ms * GST_MSECOND;
750 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
751 priv->do_lost = DEFAULT_DO_LOST;
752 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
753 priv->rtx_delay = DEFAULT_RTX_DELAY;
754 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
755 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
756 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
757 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
758 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
761 priv->last_rtptime = -1;
762 priv->avg_jitter = 0;
763 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
764 priv->jbuf = rtp_jitter_buffer_new ();
765 g_mutex_init (&priv->jbuf_lock);
766 g_cond_init (&priv->jbuf_timer);
767 g_cond_init (&priv->jbuf_event);
768 g_cond_init (&priv->jbuf_query);
770 /* reset skew detection initialy */
771 rtp_jitter_buffer_reset_skew (priv->jbuf);
772 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
773 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
777 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
780 gst_pad_set_activatemode_function (priv->srcpad,
781 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
782 gst_pad_set_query_function (priv->srcpad,
783 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
784 gst_pad_set_event_function (priv->srcpad,
785 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
788 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
791 gst_pad_set_chain_function (priv->sinkpad,
792 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
793 gst_pad_set_event_function (priv->sinkpad,
794 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
795 gst_pad_set_query_function (priv->sinkpad,
796 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
798 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
799 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
801 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
804 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
806 #define ITEM_TYPE_BUFFER 0
807 #define ITEM_TYPE_LOST 1
808 #define ITEM_TYPE_EVENT 2
809 #define ITEM_TYPE_QUERY 3
811 static RTPJitterBufferItem *
812 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
813 guint seqnum, guint count, guint rtptime)
815 RTPJitterBufferItem *item;
817 item = g_slice_new (RTPJitterBufferItem);
824 item->seqnum = seqnum;
826 item->rtptime = rtptime;
832 free_item (RTPJitterBufferItem * item)
834 if (item->data && item->type != ITEM_TYPE_QUERY)
835 gst_mini_object_unref (item->data);
836 g_slice_free (RTPJitterBufferItem, item);
840 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
842 GList **l = user_data;
844 if (item->data && item->type == ITEM_TYPE_EVENT
845 && GST_EVENT_IS_STICKY (item->data)) {
846 *l = g_list_prepend (*l, item->data);
847 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
848 gst_mini_object_unref (item->data);
850 g_slice_free (RTPJitterBufferItem, item);
854 gst_rtp_jitter_buffer_finalize (GObject * object)
856 GstRtpJitterBuffer *jitterbuffer;
857 GstRtpJitterBufferPrivate *priv;
859 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
860 priv = jitterbuffer->priv;
862 g_array_free (priv->timers, TRUE);
863 g_mutex_clear (&priv->jbuf_lock);
864 g_cond_clear (&priv->jbuf_timer);
865 g_cond_clear (&priv->jbuf_event);
866 g_cond_clear (&priv->jbuf_query);
868 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
869 g_object_unref (priv->jbuf);
871 G_OBJECT_CLASS (parent_class)->finalize (object);
875 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
877 GstRtpJitterBuffer *jitterbuffer;
878 GstPad *otherpad = NULL;
879 GstIterator *it = NULL;
882 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
884 if (pad == jitterbuffer->priv->sinkpad) {
885 otherpad = jitterbuffer->priv->srcpad;
886 } else if (pad == jitterbuffer->priv->srcpad) {
887 otherpad = jitterbuffer->priv->sinkpad;
888 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
889 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
893 g_value_init (&val, GST_TYPE_PAD);
894 g_value_set_object (&val, otherpad);
895 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
896 g_value_unset (&val);
903 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
905 GstRtpJitterBufferPrivate *priv;
907 priv = jitterbuffer->priv;
909 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
912 gst_pad_new_from_static_template
913 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
914 gst_pad_set_chain_function (priv->rtcpsinkpad,
915 gst_rtp_jitter_buffer_chain_rtcp);
916 gst_pad_set_event_function (priv->rtcpsinkpad,
917 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
918 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
919 gst_rtp_jitter_buffer_iterate_internal_links);
920 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
921 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
923 return priv->rtcpsinkpad;
927 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
929 GstRtpJitterBufferPrivate *priv;
931 priv = jitterbuffer->priv;
933 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
935 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
937 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
938 priv->rtcpsinkpad = NULL;
942 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
943 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
945 GstRtpJitterBuffer *jitterbuffer;
946 GstElementClass *klass;
948 GstRtpJitterBufferPrivate *priv;
950 g_return_val_if_fail (templ != NULL, NULL);
951 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
953 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
954 priv = jitterbuffer->priv;
955 klass = GST_ELEMENT_GET_CLASS (element);
957 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
959 /* figure out the template */
960 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
961 if (priv->rtcpsinkpad != NULL)
964 result = create_rtcp_sink (jitterbuffer);
973 g_warning ("rtpjitterbuffer: this is not our template");
978 g_warning ("rtpjitterbuffer: pad already requested");
984 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
986 GstRtpJitterBuffer *jitterbuffer;
987 GstRtpJitterBufferPrivate *priv;
989 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
990 g_return_if_fail (GST_IS_PAD (pad));
992 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
993 priv = jitterbuffer->priv;
995 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
997 if (priv->rtcpsinkpad == pad) {
998 remove_rtcp_sink (jitterbuffer);
1007 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1013 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1015 return gst_system_clock_obtain ();
1019 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1021 GstRtpJitterBufferPrivate *priv;
1023 priv = jitterbuffer->priv;
1025 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1028 priv->clock_rate = -1;
1029 /* do not clear current content, but refresh state for new arrival */
1030 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1031 rtp_jitter_buffer_reset_skew (priv->jbuf);
1036 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1039 GstRtpJitterBufferPrivate *priv;
1040 GstClockTime last_out;
1041 RTPJitterBufferItem *item;
1046 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1047 active, GST_TIME_ARGS (offset));
1049 if (active != priv->active) {
1050 /* add the amount of time spent in paused to the output offset. All
1051 * outgoing buffers will have this offset applied to their timestamps in
1052 * order to make them arrive in time in the sink. */
1053 priv->out_offset = offset;
1054 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1055 GST_TIME_ARGS (priv->out_offset));
1056 priv->active = active;
1057 JBUF_SIGNAL_EVENT (priv);
1060 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1062 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1063 /* head buffer timestamp and offset gives our output time */
1064 last_out = item->dts + priv->ts_offset;
1066 /* use last known time when the buffer is empty */
1067 last_out = priv->last_out_time;
1075 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1077 GstRtpJitterBuffer *jitterbuffer;
1078 GstRtpJitterBufferPrivate *priv;
1083 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1084 priv = jitterbuffer->priv;
1086 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1088 caps = gst_pad_peer_query_caps (other, filter);
1090 templ = gst_pad_get_pad_template_caps (pad);
1092 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1097 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1099 intersect = gst_caps_intersect (caps, templ);
1100 gst_caps_unref (caps);
1101 gst_caps_unref (templ);
1105 gst_object_unref (jitterbuffer);
1111 * Must be called with JBUF_LOCK held
1115 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1118 GstRtpJitterBufferPrivate *priv;
1119 GstStructure *caps_struct;
1123 priv = jitterbuffer->priv;
1125 /* first parse the caps */
1126 caps_struct = gst_caps_get_structure (caps, 0);
1128 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1130 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1131 * measure the amount of data in the buffer */
1132 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1135 if (priv->clock_rate <= 0)
1138 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1140 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1142 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1143 * can use this to track the amount of time elapsed on the sender. */
1144 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1145 priv->clock_base = val;
1147 priv->clock_base = -1;
1149 priv->ext_timestamp = priv->clock_base;
1151 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1154 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1155 /* first expected seqnum, only update when we didn't have a previous base. */
1156 if (priv->next_in_seqnum == -1)
1157 priv->next_in_seqnum = val;
1158 if (priv->next_seqnum == -1) {
1159 priv->next_seqnum = val;
1160 JBUF_SIGNAL_EVENT (priv);
1162 priv->seqnum_base = val;
1164 priv->seqnum_base = -1;
1167 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1169 /* the start and stop times. The seqnum-base corresponds to the start time. We
1170 * will keep track of the seqnums on the output and when we reach the one
1171 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1172 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1173 priv->npt_start = tval;
1175 priv->npt_start = 0;
1177 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1178 priv->npt_stop = tval;
1180 priv->npt_stop = -1;
1182 GST_DEBUG_OBJECT (jitterbuffer,
1183 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1184 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1191 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1196 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1202 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1204 GstRtpJitterBufferPrivate *priv;
1206 priv = jitterbuffer->priv;
1209 /* mark ourselves as flushing */
1210 priv->srcresult = GST_FLOW_FLUSHING;
1211 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1212 /* this unblocks any waiting pops on the src pad task */
1213 JBUF_SIGNAL_EVENT (priv);
1214 JBUF_SIGNAL_QUERY (priv, FALSE);
1219 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1221 GstRtpJitterBufferPrivate *priv;
1223 priv = jitterbuffer->priv;
1226 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1227 /* Mark as non flushing */
1228 priv->srcresult = GST_FLOW_OK;
1229 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1230 priv->last_popped_seqnum = -1;
1231 priv->last_out_time = -1;
1232 priv->next_seqnum = -1;
1233 priv->seqnum_base = -1;
1234 priv->ips_rtptime = -1;
1235 priv->ips_dts = GST_CLOCK_TIME_NONE;
1236 priv->packet_spacing = 0;
1237 priv->next_in_seqnum = -1;
1238 priv->clock_rate = -1;
1241 priv->estimated_eos = -1;
1242 priv->last_elapsed = 0;
1243 priv->ext_timestamp = -1;
1244 priv->avg_jitter = 0;
1245 priv->last_dts = -1;
1246 priv->last_rtptime = -1;
1247 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1248 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1249 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1250 rtp_jitter_buffer_reset_skew (priv->jbuf);
1251 remove_all_timers (jitterbuffer);
1256 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1257 GstPadMode mode, gboolean active)
1260 GstRtpJitterBuffer *jitterbuffer = NULL;
1262 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1265 case GST_PAD_MODE_PUSH:
1267 /* allow data processing */
1268 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1270 /* start pushing out buffers */
1271 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1272 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1273 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1275 /* make sure all data processing stops ASAP */
1276 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1278 /* NOTE this will hardlock if the state change is called from the src pad
1279 * task thread because we will _join() the thread. */
1280 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1281 result = gst_pad_stop_task (pad);
1291 static GstStateChangeReturn
1292 gst_rtp_jitter_buffer_change_state (GstElement * element,
1293 GstStateChange transition)
1295 GstRtpJitterBuffer *jitterbuffer;
1296 GstRtpJitterBufferPrivate *priv;
1297 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1299 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1300 priv = jitterbuffer->priv;
1302 switch (transition) {
1303 case GST_STATE_CHANGE_NULL_TO_READY:
1305 case GST_STATE_CHANGE_READY_TO_PAUSED:
1307 /* reset negotiated values */
1308 priv->clock_rate = -1;
1309 priv->clock_base = -1;
1310 priv->peer_latency = 0;
1312 /* block until we go to PLAYING */
1313 priv->blocked = TRUE;
1314 priv->timer_running = TRUE;
1315 priv->timer_thread =
1316 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1319 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1321 /* unblock to allow streaming in PLAYING */
1322 priv->blocked = FALSE;
1323 JBUF_SIGNAL_EVENT (priv);
1324 JBUF_SIGNAL_TIMER (priv);
1331 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1333 switch (transition) {
1334 case GST_STATE_CHANGE_READY_TO_PAUSED:
1335 /* we are a live element because we sync to the clock, which we can only
1336 * do in the PLAYING state */
1337 if (ret != GST_STATE_CHANGE_FAILURE)
1338 ret = GST_STATE_CHANGE_NO_PREROLL;
1340 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1342 /* block to stop streaming when PAUSED */
1343 priv->blocked = TRUE;
1344 unschedule_current_timer (jitterbuffer);
1346 if (ret != GST_STATE_CHANGE_FAILURE)
1347 ret = GST_STATE_CHANGE_NO_PREROLL;
1349 case GST_STATE_CHANGE_PAUSED_TO_READY:
1351 gst_buffer_replace (&priv->last_sr, NULL);
1352 priv->timer_running = FALSE;
1353 unschedule_current_timer (jitterbuffer);
1354 JBUF_SIGNAL_TIMER (priv);
1355 JBUF_SIGNAL_QUERY (priv, FALSE);
1357 g_thread_join (priv->timer_thread);
1358 priv->timer_thread = NULL;
1360 case GST_STATE_CHANGE_READY_TO_NULL:
1370 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1373 gboolean ret = TRUE;
1374 GstRtpJitterBuffer *jitterbuffer;
1375 GstRtpJitterBufferPrivate *priv;
1377 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1378 priv = jitterbuffer->priv;
1380 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1382 switch (GST_EVENT_TYPE (event)) {
1383 case GST_EVENT_LATENCY:
1385 GstClockTime latency;
1387 gst_event_parse_latency (event, &latency);
1389 GST_DEBUG_OBJECT (jitterbuffer,
1390 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1393 /* adjust the overall buffer delay to the total pipeline latency in
1394 * buffering mode because if downstream consumes too fast (because of
1395 * large latency or queues, we would start rebuffering again. */
1396 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1397 RTP_JITTER_BUFFER_MODE_BUFFER) {
1398 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1402 ret = gst_pad_push_event (priv->sinkpad, event);
1406 ret = gst_pad_push_event (priv->sinkpad, event);
1413 /* handles and stores the event in the jitterbuffer, must be called with
1416 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1418 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1419 RTPJitterBufferItem *item;
1422 switch (GST_EVENT_TYPE (event)) {
1423 case GST_EVENT_CAPS:
1427 gst_event_parse_caps (event, &caps);
1428 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1431 case GST_EVENT_SEGMENT:
1432 gst_event_copy_segment (event, &priv->segment);
1434 /* we need time for now */
1435 if (priv->segment.format != GST_FORMAT_TIME)
1436 goto newseg_wrong_format;
1438 GST_DEBUG_OBJECT (jitterbuffer,
1439 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1443 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1450 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1451 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1452 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1454 JBUF_SIGNAL_EVENT (priv);
1459 newseg_wrong_format:
1461 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1462 gst_event_unref (event);
1468 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1471 gboolean ret = TRUE;
1472 GstRtpJitterBuffer *jitterbuffer;
1473 GstRtpJitterBufferPrivate *priv;
1475 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1476 priv = jitterbuffer->priv;
1478 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1480 switch (GST_EVENT_TYPE (event)) {
1481 case GST_EVENT_FLUSH_START:
1482 ret = gst_pad_push_event (priv->srcpad, event);
1483 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1484 /* wait for the loop to go into PAUSED */
1485 gst_pad_pause_task (priv->srcpad);
1487 case GST_EVENT_FLUSH_STOP:
1488 ret = gst_pad_push_event (priv->srcpad, event);
1490 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1491 GST_PAD_MODE_PUSH, TRUE);
1494 if (GST_EVENT_IS_SERIALIZED (event)) {
1495 /* serialized events go in the queue */
1497 if (priv->srcresult != GST_FLOW_OK) {
1498 /* Errors in sticky event pushing are no problem and ignored here
1499 * as they will cause more meaningful errors during data flow.
1500 * For EOS events, that are not followed by data flow, we still
1501 * return FALSE here though.
1503 if (!GST_EVENT_IS_STICKY (event) ||
1504 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1505 goto out_flow_error;
1507 /* refuse more events on EOS */
1510 ret = queue_event (jitterbuffer, event);
1513 /* non-serialized events are forwarded downstream immediately */
1514 ret = gst_pad_push_event (priv->srcpad, event);
1523 GST_DEBUG_OBJECT (jitterbuffer,
1524 "refusing event, we have a downstream flow error: %s",
1525 gst_flow_get_name (priv->srcresult));
1527 gst_event_unref (event);
1532 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1534 gst_event_unref (event);
1540 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1543 gboolean ret = TRUE;
1544 GstRtpJitterBuffer *jitterbuffer;
1546 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1548 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1550 switch (GST_EVENT_TYPE (event)) {
1551 case GST_EVENT_FLUSH_START:
1552 gst_event_unref (event);
1554 case GST_EVENT_FLUSH_STOP:
1555 gst_event_unref (event);
1558 ret = gst_pad_event_default (pad, parent, event);
1566 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1567 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1568 * GST_FLOW_FLUSHING when the element is shutting down. On success
1569 * GST_FLOW_OK is returned.
1571 static GstFlowReturn
1572 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1576 GValue args[2] = { {0}, {0} };
1580 g_value_init (&args[0], GST_TYPE_ELEMENT);
1581 g_value_set_object (&args[0], jitterbuffer);
1582 g_value_init (&args[1], G_TYPE_UINT);
1583 g_value_set_uint (&args[1], pt);
1585 g_value_init (&ret, GST_TYPE_CAPS);
1586 g_value_set_boxed (&ret, NULL);
1588 JBUF_UNLOCK (jitterbuffer->priv);
1589 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1591 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1593 g_value_unset (&args[0]);
1594 g_value_unset (&args[1]);
1595 caps = (GstCaps *) g_value_dup_boxed (&ret);
1596 g_value_unset (&ret);
1600 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1601 gst_caps_unref (caps);
1603 if (G_UNLIKELY (!res))
1611 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1612 return GST_FLOW_ERROR;
1616 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1617 return GST_FLOW_FLUSHING;
1621 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1622 return GST_FLOW_ERROR;
1626 /* call with jbuf lock held */
1628 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1630 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1631 GstMessage *message = NULL;
1636 /* Post a buffering message */
1637 if (priv->last_percent != percent) {
1638 priv->last_percent = percent;
1640 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1641 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1648 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1650 GstRtpJitterBufferPrivate *priv;
1652 priv = jitterbuffer->priv;
1654 if (timestamp == -1)
1657 /* apply the timestamp offset, this is used for inter stream sync */
1658 timestamp += priv->ts_offset;
1659 /* add the offset, this is used when buffering */
1660 timestamp += priv->out_offset;
1666 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1668 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1669 TimerData *timer = NULL;
1672 len = priv->timers->len;
1673 for (i = 0; i < len; i++) {
1674 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1675 if (test->seqnum == seqnum && test->type == type) {
1684 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1686 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1688 if (priv->clock_id) {
1689 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1690 gst_clock_id_unschedule (priv->clock_id);
1691 priv->clock_id = NULL;
1696 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1698 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1699 GstClockTime test_timeout;
1701 if ((test_timeout = timer->timeout) == -1)
1704 if (timer->type != TIMER_TYPE_EXPECTED) {
1705 /* add our latency and offset to get output times. */
1706 test_timeout = apply_offset (jitterbuffer, test_timeout);
1707 test_timeout += priv->latency_ns;
1709 return test_timeout;
1713 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1715 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1717 if (priv->clock_id) {
1718 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1720 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1721 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1723 if (timeout == -1 || timeout < priv->timer_timeout)
1724 unschedule_current_timer (jitterbuffer);
1729 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1730 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1731 GstClockTime duration)
1733 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1737 GST_DEBUG_OBJECT (jitterbuffer,
1738 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1739 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1740 GST_TIME_ARGS (delay));
1742 len = priv->timers->len;
1743 g_array_set_size (priv->timers, len + 1);
1744 timer = &g_array_index (priv->timers, TimerData, len);
1747 timer->seqnum = seqnum;
1749 timer->timeout = timeout + delay;
1750 timer->duration = duration;
1751 if (type == TIMER_TYPE_EXPECTED) {
1752 timer->rtx_base = timeout;
1753 timer->rtx_delay = delay;
1754 timer->rtx_retry = 0;
1756 timer->num_rtx_retry = 0;
1757 recalculate_timer (jitterbuffer, timer);
1758 JBUF_SIGNAL_TIMER (priv);
1764 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1765 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1767 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1768 gboolean seqchange, timechange;
1771 seqchange = timer->seqnum != seqnum;
1772 timechange = timer->timeout != timeout;
1774 if (!seqchange && !timechange)
1777 oldseq = timer->seqnum;
1779 GST_DEBUG_OBJECT (jitterbuffer,
1780 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1781 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1783 timer->timeout = timeout + delay;
1784 timer->seqnum = seqnum;
1786 timer->rtx_base = timeout;
1787 timer->rtx_delay = delay;
1788 timer->rtx_retry = 0;
1791 timer->num_rtx_retry = 0;
1793 if (priv->clock_id) {
1794 /* we changed the seqnum and there is a timer currently waiting with this
1795 * seqnum, unschedule it */
1796 if (seqchange && priv->timer_seqnum == oldseq)
1797 unschedule_current_timer (jitterbuffer);
1798 /* we changed the time, check if it is earlier than what we are waiting
1799 * for and unschedule if so */
1800 else if (timechange)
1801 recalculate_timer (jitterbuffer, timer);
1806 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1807 guint16 seqnum, GstClockTime timeout)
1811 /* find the seqnum timer */
1812 timer = find_timer (jitterbuffer, type, seqnum);
1813 if (timer == NULL) {
1814 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1816 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1822 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1824 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1827 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1828 unschedule_current_timer (jitterbuffer);
1831 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1832 g_array_remove_index_fast (priv->timers, idx);
1837 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1839 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1840 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1841 g_array_set_size (priv->timers, 0);
1842 unschedule_current_timer (jitterbuffer);
1845 /* get the extra delay to wait before sending RTX */
1847 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1851 if (priv->rtx_delay == -1) {
1852 if (priv->avg_jitter == 0)
1853 delay = DEFAULT_AUTO_RTX_DELAY;
1855 /* jitter is in nanoseconds, 2x jitter is a good margin */
1856 delay = priv->avg_jitter * 2;
1858 delay = priv->rtx_delay * GST_MSECOND;
1860 if (priv->rtx_min_delay > 0)
1861 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1866 /* we just received a packet with seqnum and dts.
1868 * First check for old seqnum that we are still expecting. If the gap with the
1869 * current seqnum is too big, unschedule the timeouts.
1871 * If we have a valid packet spacing estimate we can set a timer for when we
1872 * should receive the next packet.
1873 * If we don't have a valid estimate, we remove any timer we might have
1874 * had for this packet.
1877 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1878 GstClockTime dts, gboolean do_next_seqnum)
1880 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1881 TimerData *timer = NULL;
1884 /* go through all timers and unschedule the ones with a large gap, also find
1885 * the timer for the seqnum */
1886 len = priv->timers->len;
1887 for (i = 0; i < len; i++) {
1888 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1891 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1893 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
1894 test->type, test->seqnum, seqnum, gap);
1897 GST_DEBUG ("found timer for current seqnum");
1898 /* the timer for the current seqnum */
1900 /* when no retransmission, we can stop now, we only need to find the
1901 * timer for the current seqnum */
1902 if (!priv->do_retransmission)
1904 } else if (gap > priv->rtx_delay_reorder) {
1905 /* max gap, we exceeded the max reorder distance and we don't expect the
1906 * missing packet to be this reordered */
1907 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1908 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
1912 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
1913 && priv->do_retransmission;
1915 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1916 if (timer->num_rtx_retry > 0) {
1917 GstClockTime rtx_last, delay;
1919 /* we scheduled a retry for this packet and now we have it */
1920 priv->num_rtx_success++;
1921 /* all the previous retry attempts failed */
1922 priv->num_rtx_failed += timer->num_rtx_retry - 1;
1923 /* number of retries before receiving the packet */
1924 if (priv->avg_rtx_num == 0.0)
1925 priv->avg_rtx_num = timer->num_rtx_retry;
1927 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
1928 /* calculate the delay between retransmission request and receiving this
1929 * packet, start with when we scheduled this timeout last */
1930 rtx_last = timer->rtx_last;
1931 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
1932 /* we have a valid delay if this packet arrived after we scheduled the
1934 delay = dts - rtx_last;
1935 if (priv->avg_rtx_rtt == 0)
1936 priv->avg_rtx_rtt = delay;
1938 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
1942 GST_LOG_OBJECT (jitterbuffer,
1943 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
1944 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
1945 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
1946 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
1947 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
1948 GST_TIME_ARGS (priv->avg_rtx_rtt));
1950 /* don't try to estimate the next seqnum because this is a retransmitted
1951 * packet and it probably did not arrive with the expected packet
1953 do_next_seqnum = FALSE;
1957 if (do_next_seqnum) {
1958 GstClockTime expected, delay;
1960 /* calculate expected arrival time of the next seqnum */
1961 expected = dts + priv->packet_spacing;
1963 delay = get_rtx_delay (priv);
1965 /* and update/install timer for next seqnum */
1967 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
1970 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
1971 expected, delay, priv->packet_spacing);
1972 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1973 /* if we had a timer, remove it, we don't know when to expect the next
1975 remove_timer (jitterbuffer, timer);
1980 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
1983 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1985 /* we need consecutive seqnums with a different
1986 * rtptime to estimate the packet spacing. */
1987 if (priv->ips_rtptime != rtptime) {
1988 /* rtptime changed, check dts diff */
1989 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
1990 priv->packet_spacing = dts - priv->ips_dts;
1991 GST_DEBUG_OBJECT (jitterbuffer,
1992 "new packet spacing %" GST_TIME_FORMAT,
1993 GST_TIME_ARGS (priv->packet_spacing));
1995 priv->ips_rtptime = rtptime;
1996 priv->ips_dts = dts;
2001 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2002 guint16 seqnum, GstClockTime dts, gint gap)
2004 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2005 GstClockTime total_duration, duration, expected_dts;
2008 GST_DEBUG_OBJECT (jitterbuffer,
2009 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2010 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2012 /* the total duration spanned by the missing packets */
2013 if (dts >= priv->last_in_dts)
2014 total_duration = dts - priv->last_in_dts;
2018 /* interpolate between the current time and the last time based on
2019 * number of packets we are missing, this is the estimated duration
2020 * for the missing packet based on equidistant packet spacing. */
2021 duration = total_duration / (gap + 1);
2023 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2024 GST_TIME_ARGS (duration));
2026 if (total_duration > priv->latency_ns) {
2027 GstClockTime gap_time;
2030 gap_time = total_duration - priv->latency_ns;
2033 lost_packets = gap_time / duration;
2034 gap_time = lost_packets * duration;
2039 /* too many lost packets, some of the missing packets are already
2040 * too late and we can generate lost packet events for them. */
2041 GST_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
2042 " > %" GST_TIME_FORMAT ", consider %u lost",
2043 GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
2046 /* this timer will fire immediately and the lost event will be pushed from
2047 * the timer thread */
2048 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2049 priv->last_in_dts + duration, 0, gap_time);
2051 expected += lost_packets;
2052 priv->last_in_dts += gap_time;
2055 expected_dts = priv->last_in_dts + duration;
2057 if (priv->do_retransmission) {
2060 type = TIMER_TYPE_EXPECTED;
2061 /* if we had a timer for the first missing packet, update it. */
2062 if ((timer = find_timer (jitterbuffer, type, expected))) {
2063 GstClockTime timeout = timer->timeout;
2065 timer->duration = duration;
2066 if (timeout > expected_dts) {
2067 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2068 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2072 expected_dts += duration;
2075 type = TIMER_TYPE_LOST;
2078 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2079 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2080 expected_dts += duration;
2086 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2090 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2091 GstRtpJitterBufferPrivate *priv;
2093 priv = jitterbuffer->priv;
2095 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2098 if (priv->last_dts != -1)
2099 dtsdiff = dts - priv->last_dts;
2103 if (priv->last_rtptime != -1)
2104 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2108 priv->last_dts = dts;
2109 priv->last_rtptime = rtptime;
2113 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2116 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2118 diff = ABS (dtsdiff - rtpdiffns);
2120 /* jitter is stored in nanoseconds */
2121 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2123 GST_LOG_OBJECT (jitterbuffer,
2124 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2125 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2126 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2127 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2134 GST_DEBUG_OBJECT (jitterbuffer,
2135 "no dts or no clock-rate, can't calculate jitter");
2140 static GstFlowReturn
2141 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2144 GstRtpJitterBuffer *jitterbuffer;
2145 GstRtpJitterBufferPrivate *priv;
2147 guint32 expected, rtptime;
2148 GstFlowReturn ret = GST_FLOW_OK;
2149 GstClockTime dts, pts;
2154 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2155 gboolean do_next_seqnum = FALSE;
2156 RTPJitterBufferItem *item;
2157 GstMessage *msg = NULL;
2159 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2161 priv = jitterbuffer->priv;
2163 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2164 goto invalid_buffer;
2166 pt = gst_rtp_buffer_get_payload_type (&rtp);
2167 seqnum = gst_rtp_buffer_get_seq (&rtp);
2168 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2169 gst_rtp_buffer_unmap (&rtp);
2171 /* make sure we have PTS and DTS set */
2172 pts = GST_BUFFER_PTS (buffer);
2173 dts = GST_BUFFER_DTS (buffer);
2179 /* take the DTS of the buffer. This is the time when the packet was
2180 * received and is used to calculate jitter and clock skew. We will adjust
2181 * this DTS with the smoothed value after processing it in the
2182 * jitterbuffer and assign it as the PTS. */
2183 /* bring to running time */
2184 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2186 GST_DEBUG_OBJECT (jitterbuffer,
2187 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2188 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2190 JBUF_LOCK_CHECK (priv, out_flushing);
2192 if (G_UNLIKELY (priv->last_pt != pt)) {
2195 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2199 /* reset clock-rate so that we get a new one */
2200 priv->clock_rate = -1;
2202 /* Try to get the clock-rate from the caps first if we can. If there are no
2203 * caps we must fire the signal to get the clock-rate. */
2204 if ((caps = gst_pad_get_current_caps (pad))) {
2205 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2206 gst_caps_unref (caps);
2210 if (G_UNLIKELY (priv->clock_rate == -1)) {
2211 /* no clock rate given on the caps, try to get one with the signal */
2212 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2213 pt) == GST_FLOW_FLUSHING)
2216 if (G_UNLIKELY (priv->clock_rate == -1))
2220 /* don't accept more data on EOS */
2221 if (G_UNLIKELY (priv->eos))
2224 calculate_jitter (jitterbuffer, dts, rtptime);
2226 if (priv->seqnum_base != -1) {
2229 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2232 GST_DEBUG_OBJECT (jitterbuffer,
2233 "packet seqnum #%d before seqnum-base #%d", seqnum,
2235 gst_buffer_unref (buffer);
2238 } else if (gap > 16384) {
2239 /* From now on don't compare against the seqnum base anymore as
2240 * at some point in the future we will wrap around and also that
2241 * much reordering is very unlikely */
2242 priv->seqnum_base = -1;
2246 expected = priv->next_in_seqnum;
2248 /* now check against our expected seqnum */
2249 if (G_LIKELY (expected != -1)) {
2252 /* now calculate gap */
2253 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2255 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2256 expected, seqnum, gap);
2258 if (G_LIKELY (gap == 0)) {
2259 /* packet is expected */
2260 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2261 do_next_seqnum = TRUE;
2263 gboolean reset = FALSE;
2265 if (!GST_CLOCK_TIME_IS_VALID (dts)) {
2266 /* We would run into calculations with GST_CLOCK_TIME_NONE below
2267 * and can't compensate for anything without DTS on RTP packets
2269 goto gap_but_no_dts;
2270 } else if (gap < 0) {
2271 /* we received an old packet */
2272 if (G_UNLIKELY (gap < -RTP_MAX_MISORDER)) {
2273 /* too old packet, reset */
2274 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d < %d", gap,
2278 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2281 /* new packet, we are missing some packets */
2282 if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
2283 /* packet too far in future, reset */
2284 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too new %d > %d", gap,
2288 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2289 /* fill in the gap with EXPECTED timers */
2290 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2292 do_next_seqnum = TRUE;
2295 if (G_UNLIKELY (reset)) {
2296 GList *events = NULL, *l;
2298 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2299 rtp_jitter_buffer_flush (priv->jbuf,
2300 (GFunc) free_item_and_retain_events, &events);
2301 rtp_jitter_buffer_reset_skew (priv->jbuf);
2302 remove_all_timers (jitterbuffer);
2303 priv->last_popped_seqnum = -1;
2304 priv->next_seqnum = seqnum;
2305 do_next_seqnum = TRUE;
2307 /* Insert all sticky events again in order, otherwise we would
2308 * potentially loose STREAM_START, CAPS or SEGMENT events
2310 events = g_list_reverse (events);
2311 for (l = events; l; l = l->next) {
2312 RTPJitterBufferItem *item;
2314 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2315 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2317 g_list_free (events);
2319 JBUF_SIGNAL_EVENT (priv);
2321 /* reset spacing estimation when gap */
2322 priv->ips_rtptime = -1;
2323 priv->ips_dts = GST_CLOCK_TIME_NONE;
2326 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2327 /* we don't know what the next_in_seqnum should be, wait for the last
2328 * possible moment to push this buffer, maybe we get an earlier seqnum
2330 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2331 do_next_seqnum = TRUE;
2332 /* take rtptime and dts to calculate packet spacing */
2333 priv->ips_rtptime = rtptime;
2334 priv->ips_dts = dts;
2336 if (do_next_seqnum) {
2337 priv->last_in_seqnum = seqnum;
2338 priv->last_in_dts = dts;
2339 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2342 /* let's check if this buffer is too late, we can only accept packets with
2343 * bigger seqnum than the one we last pushed. */
2344 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2347 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2349 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2350 if (G_UNLIKELY (gap <= 0))
2354 /* let's drop oldest packet if the queue is already full and drop-on-latency
2355 * is set. We can only do this when there actually is a latency. When no
2356 * latency is set, we just pump it in the queue and let the other end push it
2357 * out as fast as possible. */
2358 if (priv->latency_ms && priv->drop_on_latency) {
2360 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2362 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2363 RTPJitterBufferItem *old_item;
2365 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2367 if (IS_DROPABLE (old_item)) {
2368 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2369 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2371 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2372 free_item (old_item);
2374 /* we might have removed some head buffers, signal the pushing thread to
2375 * see if it can push now */
2376 JBUF_SIGNAL_EVENT (priv);
2380 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2382 /* now insert the packet into the queue in sorted order. This function returns
2383 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2384 * have a duplicate. */
2385 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2390 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2392 /* we had an unhandled SR, handle it now */
2394 do_handle_sync (jitterbuffer);
2396 if (G_UNLIKELY (head)) {
2397 /* signal addition of new buffer when the _loop is waiting. */
2398 if (G_LIKELY (priv->active))
2399 JBUF_SIGNAL_EVENT (priv);
2401 /* let's unschedule and unblock any waiting buffers. We only want to do this
2402 * when the head buffer changed */
2403 if (G_UNLIKELY (priv->clock_id)) {
2404 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2405 unschedule_current_timer (jitterbuffer);
2409 GST_DEBUG_OBJECT (jitterbuffer,
2410 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2411 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2413 msg = check_buffering_percent (jitterbuffer, percent);
2419 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2426 /* this is not fatal but should be filtered earlier */
2427 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2428 ("Received invalid RTP payload, dropping"));
2429 gst_buffer_unref (buffer);
2434 GST_WARNING_OBJECT (jitterbuffer,
2435 "No clock-rate in caps!, dropping buffer");
2436 gst_buffer_unref (buffer);
2441 ret = priv->srcresult;
2442 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2443 gst_buffer_unref (buffer);
2449 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2450 gst_buffer_unref (buffer);
2455 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2456 " popped, dropping", seqnum, priv->last_popped_seqnum);
2458 gst_buffer_unref (buffer);
2463 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2465 priv->num_duplicates++;
2471 /* this is fatal as we can't compensate for gaps without DTS */
2472 GST_ELEMENT_ERROR (jitterbuffer, STREAM, DECODE, (NULL),
2473 ("Received packet without DTS after a gap"));
2474 gst_buffer_unref (buffer);
2475 ret = GST_FLOW_ERROR;
2481 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2483 guint64 ext_time, elapsed;
2485 GstRtpJitterBufferPrivate *priv;
2487 priv = jitterbuffer->priv;
2488 rtp_time = item->rtptime;
2490 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2491 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2493 if (rtp_time < priv->ext_timestamp) {
2494 ext_time = priv->ext_timestamp;
2496 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
2499 if (ext_time > priv->clock_base)
2500 elapsed = ext_time - priv->clock_base;
2504 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2509 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2510 RTPJitterBufferItem * item)
2512 guint64 total, elapsed, left, estimated;
2513 GstClockTime out_time;
2514 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2516 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2517 || priv->clock_base == -1 || priv->clock_rate <= 0)
2520 /* compute the elapsed time */
2521 elapsed = compute_elapsed (jitterbuffer, item);
2523 /* do nothing if elapsed time doesn't increment */
2524 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2527 priv->last_elapsed = elapsed;
2529 /* this is the total time we need to play */
2530 total = priv->npt_stop - priv->npt_start;
2531 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2532 GST_TIME_ARGS (total));
2534 /* this is how much time there is left */
2535 if (total > elapsed)
2536 left = total - elapsed;
2540 /* if we have less time left that the size of the buffer, we will not
2541 * be able to keep it filled, disabled buffering then */
2542 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2543 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2544 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2545 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2548 /* this is the current time as running-time */
2549 out_time = item->dts;
2552 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2554 /* if there is almost nothing left,
2555 * we may never advance enough to end up in the above case */
2556 if (total < GST_SECOND)
2557 estimated = GST_SECOND;
2561 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2562 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2564 if (estimated != -1 && priv->estimated_eos != estimated) {
2565 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2566 priv->estimated_eos = estimated;
2570 /* take a buffer from the queue and push it */
2571 static GstFlowReturn
2572 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2574 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2575 GstFlowReturn result = GST_FLOW_OK;
2576 RTPJitterBufferItem *item;
2577 GstBuffer *outbuf = NULL;
2578 GstEvent *outevent = NULL;
2579 GstQuery *outquery = NULL;
2580 GstClockTime dts, pts;
2582 gboolean do_push = TRUE;
2586 /* when we get here we are ready to pop and push the buffer */
2587 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2591 case ITEM_TYPE_BUFFER:
2593 /* we need to make writable to change the flags and timestamps */
2594 outbuf = gst_buffer_make_writable (item->data);
2596 if (G_UNLIKELY (priv->discont)) {
2597 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2598 * into the jitterbuffer so we can modify now. */
2599 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2600 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2601 priv->discont = FALSE;
2603 if (G_UNLIKELY (priv->ts_discont)) {
2604 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2605 priv->ts_discont = FALSE;
2609 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2611 gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2613 /* apply timestamp with offset to buffer now */
2614 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2615 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2617 /* update the elapsed time when we need to check against the npt stop time. */
2618 update_estimated_eos (jitterbuffer, item);
2620 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2622 case ITEM_TYPE_LOST:
2623 priv->discont = TRUE;
2627 case ITEM_TYPE_EVENT:
2628 outevent = item->data;
2630 case ITEM_TYPE_QUERY:
2631 outquery = item->data;
2635 /* now we are ready to push the buffer. Save the seqnum and release the lock
2636 * so the other end can push stuff in the queue again. */
2638 priv->last_popped_seqnum = seqnum;
2639 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2641 msg = check_buffering_percent (jitterbuffer, percent);
2648 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2651 case ITEM_TYPE_BUFFER:
2653 GST_DEBUG_OBJECT (jitterbuffer,
2654 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2655 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2656 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2657 result = gst_pad_push (priv->srcpad, outbuf);
2659 JBUF_LOCK_CHECK (priv, out_flushing);
2661 case ITEM_TYPE_LOST:
2662 case ITEM_TYPE_EVENT:
2663 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
2664 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
2667 gst_pad_push_event (priv->srcpad, outevent);
2669 gst_event_unref (outevent);
2671 result = GST_FLOW_OK;
2673 JBUF_LOCK_CHECK (priv, out_flushing);
2675 case ITEM_TYPE_QUERY:
2679 res = gst_pad_peer_query (priv->srcpad, outquery);
2681 JBUF_LOCK_CHECK (priv, out_flushing);
2682 result = GST_FLOW_OK;
2683 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
2684 JBUF_SIGNAL_QUERY (priv, res);
2693 return priv->srcresult;
2697 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2699 /* Peek a buffer and compare the seqnum to the expected seqnum.
2700 * If all is fine, the buffer is pushed.
2701 * If something is wrong, we wait for some event
2703 static GstFlowReturn
2704 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2706 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2707 GstFlowReturn result = GST_FLOW_OK;
2708 RTPJitterBufferItem *item;
2710 guint32 next_seqnum;
2713 /* only push buffers when PLAYING and active and not buffering */
2714 if (priv->blocked || !priv->active ||
2715 rtp_jitter_buffer_is_buffering (priv->jbuf))
2716 return GST_FLOW_WAIT;
2719 /* peek a buffer, we're just looking at the sequence number.
2720 * If all is fine, we'll pop and push it. If the sequence number is wrong we
2721 * wait for a timeout or something to change.
2722 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
2723 item = rtp_jitter_buffer_peek (priv->jbuf);
2727 /* get the seqnum and the next expected seqnum */
2728 seqnum = item->seqnum;
2732 next_seqnum = priv->next_seqnum;
2734 /* get the gap between this and the previous packet. If we don't know the
2735 * previous packet seqnum assume no gap. */
2736 if (G_UNLIKELY (next_seqnum == -1)) {
2737 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2738 /* we don't know what the next_seqnum should be, the chain function should
2739 * have scheduled a DEADLINE timer that will increment next_seqnum when it
2740 * fires, so wait for that */
2741 result = GST_FLOW_WAIT;
2743 /* else calculate GAP */
2744 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2746 if (G_LIKELY (gap == 0)) {
2748 /* no missing packet, pop and push */
2749 result = pop_and_push_next (jitterbuffer, seqnum);
2750 } else if (G_UNLIKELY (gap < 0)) {
2751 RTPJitterBufferItem *item;
2752 /* if we have a packet that we already pushed or considered dropped, pop it
2753 * off and get the next packet */
2754 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2755 seqnum, next_seqnum);
2756 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2760 /* the chain function has scheduled timers to request retransmission or
2761 * when to consider the packet lost, wait for that */
2762 GST_DEBUG_OBJECT (jitterbuffer,
2763 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2764 next_seqnum, seqnum, gap);
2765 result = GST_FLOW_WAIT;
2772 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
2774 result = GST_FLOW_EOS;
2776 result = GST_FLOW_WAIT;
2782 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
2784 GstClockTime rtx_retry_timeout;
2785 GstClockTime rtx_min_retry_timeout;
2787 if (priv->rtx_retry_timeout == -1) {
2788 if (priv->avg_rtx_rtt == 0)
2789 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
2791 /* we want to ask for a retransmission after we waited for a
2792 * complete RTT and the additional jitter */
2793 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
2795 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
2797 /* make sure we don't retry too often. On very low latency networks,
2798 * the RTT and jitter can be very low. */
2799 if (priv->rtx_min_retry_timeout == -1) {
2800 rtx_min_retry_timeout = priv->packet_spacing;
2802 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
2804 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
2806 return rtx_retry_timeout;
2810 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
2811 GstClockTime rtx_retry_timeout)
2813 GstClockTime rtx_retry_period;
2815 if (priv->rtx_retry_period == -1) {
2816 /* we retry up to the configured jitterbuffer size but leaving some
2817 * room for the retransmission to arrive in time */
2818 if (rtx_retry_timeout > priv->latency_ns) {
2819 rtx_retry_period = 0;
2821 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
2824 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
2826 return rtx_retry_period;
2829 /* the timeout for when we expected a packet expired */
2831 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2834 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2836 guint delay, delay_ms, avg_rtx_rtt_ms;
2837 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
2838 GstClockTime rtx_retry_period;
2839 GstClockTime rtx_retry_timeout;
2842 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
2843 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
2845 rtx_retry_timeout = get_rtx_retry_timeout (priv);
2846 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
2848 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
2849 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
2850 GST_TIME_ARGS (rtx_retry_period));
2852 delay = timer->rtx_delay + timer->rtx_retry;
2854 delay_ms = GST_TIME_AS_MSECONDS (delay);
2855 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
2856 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
2857 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
2859 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2860 gst_structure_new ("GstRTPRetransmissionRequest",
2861 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
2862 "running-time", G_TYPE_UINT64, timer->rtx_base,
2863 "delay", G_TYPE_UINT, delay_ms,
2864 "retry", G_TYPE_UINT, timer->num_rtx_retry,
2865 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
2866 "period", G_TYPE_UINT, rtx_retry_period_ms,
2867 "deadline", G_TYPE_UINT, priv->latency_ms,
2868 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
2869 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
2871 priv->num_rtx_requests++;
2872 timer->num_rtx_retry++;
2874 GST_OBJECT_LOCK (jitterbuffer);
2875 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
2876 timer->rtx_last = gst_clock_get_time (clock);
2877 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
2879 timer->rtx_last = now;
2881 GST_OBJECT_UNLOCK (jitterbuffer);
2883 /* calculate the timeout for the next retransmission attempt */
2884 timer->rtx_retry += rtx_retry_timeout;
2885 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
2886 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
2887 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
2888 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
2890 if (timer->rtx_retry + timer->rtx_delay > rtx_retry_period) {
2891 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
2892 /* too many retransmission request, we now convert the timer
2893 * to a lost timer, leave the num_rtx_retry as it is for stats */
2894 timer->type = TIMER_TYPE_LOST;
2895 timer->rtx_delay = 0;
2896 timer->rtx_retry = 0;
2898 reschedule_timer (jitterbuffer, timer, timer->seqnum,
2899 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
2902 gst_pad_push_event (priv->sinkpad, event);
2908 /* a packet is lost */
2910 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2913 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2914 GstClockTime duration, timestamp;
2915 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
2916 gboolean late, head;
2918 RTPJitterBufferItem *item;
2920 seqnum = timer->seqnum;
2921 timestamp = apply_offset (jitterbuffer, timer->timeout);
2922 duration = timer->duration;
2923 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
2924 duration = priv->packet_spacing;
2925 lost_packets = MAX (timer->num, 1);
2926 late = timer->num > 0;
2927 num_rtx_retry = timer->num_rtx_retry;
2929 /* we had a gap and thus we lost some packets. Create an event for this. */
2930 if (lost_packets > 1)
2931 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
2932 seqnum + lost_packets - 1);
2934 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
2936 priv->num_late += lost_packets;
2937 priv->num_rtx_failed += num_rtx_retry;
2939 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
2941 /* we now only accept seqnum bigger than this */
2942 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
2943 priv->next_in_seqnum = next_in_seqnum;
2945 /* create paket lost event */
2946 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
2947 gst_structure_new ("GstRTPPacketLost",
2948 "seqnum", G_TYPE_UINT, (guint) seqnum,
2949 "timestamp", G_TYPE_UINT64, timestamp,
2950 "duration", G_TYPE_UINT64, duration,
2951 "late", G_TYPE_BOOLEAN, late,
2952 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
2954 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
2955 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2957 /* remove timer now */
2958 remove_timer (jitterbuffer, timer);
2960 JBUF_SIGNAL_EVENT (priv);
2966 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2969 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2971 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
2972 remove_timer (jitterbuffer, timer);
2974 /* there was no EOS in the buffer, put one in there now */
2975 queue_event (jitterbuffer, gst_event_new_eos ());
2977 JBUF_SIGNAL_EVENT (priv);
2983 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2986 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2988 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
2990 /* timer seqnum might have been obsoleted by caps seqnum-base,
2991 * only mess with current ongoing seqnum if still unknown */
2992 if (priv->next_seqnum == -1)
2993 priv->next_seqnum = timer->seqnum;
2994 remove_timer (jitterbuffer, timer);
2995 JBUF_SIGNAL_EVENT (priv);
3001 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3004 gboolean removed = FALSE;
3006 switch (timer->type) {
3007 case TIMER_TYPE_EXPECTED:
3008 removed = do_expected_timeout (jitterbuffer, timer, now);
3010 case TIMER_TYPE_LOST:
3011 removed = do_lost_timeout (jitterbuffer, timer, now);
3013 case TIMER_TYPE_DEADLINE:
3014 removed = do_deadline_timeout (jitterbuffer, timer, now);
3016 case TIMER_TYPE_EOS:
3017 removed = do_eos_timeout (jitterbuffer, timer, now);
3023 /* called when we need to wait for the next timeout.
3025 * We loop over the array of recorded timeouts and wait for the earliest one.
3026 * When it timed out, do the logic associated with the timer.
3028 * If there are no timers, we wait on a gcond until something new happens.
3031 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3033 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3034 GstClockTime now = 0;
3037 while (priv->timer_running) {
3038 TimerData *timer = NULL;
3039 GstClockTime timer_timeout = -1;
3042 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3043 GST_TIME_ARGS (now));
3045 len = priv->timers->len;
3046 for (i = 0; i < len; i++) {
3047 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3048 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3049 gboolean save_best = FALSE;
3051 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3052 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3054 /* find the smallest timeout */
3055 if (timer == NULL) {
3057 } else if (timer_timeout == -1) {
3058 /* we already have an immediate timeout, the new timer must be an
3059 * immediate timer with smaller seqnum to become the best */
3060 if (test_timeout == -1
3061 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3062 timer->seqnum) > 0))
3064 } else if (test_timeout == -1) {
3065 /* first immediate timer */
3067 } else if (test_timeout < timer_timeout) {
3070 } else if (test_timeout == timer_timeout
3071 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3072 timer->seqnum) > 0)) {
3073 /* same timer, smaller seqnum */
3077 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3079 timer_timeout = test_timeout;
3082 if (timer && !priv->blocked) {
3084 GstClockTime sync_time;
3087 GstClockTimeDiff clock_jitter;
3089 if (timer_timeout == -1 || timer_timeout <= now) {
3090 do_timeout (jitterbuffer, timer, now);
3091 /* check here, do_timeout could have released the lock */
3092 if (!priv->timer_running)
3097 GST_OBJECT_LOCK (jitterbuffer);
3098 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3100 GST_OBJECT_UNLOCK (jitterbuffer);
3101 /* let's just push if there is no clock */
3102 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3103 now = timer_timeout;
3107 /* prepare for sync against clock */
3108 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3109 /* add latency of peer to get input time */
3110 sync_time += priv->peer_latency;
3112 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3113 " with sync time %" GST_TIME_FORMAT,
3114 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3116 /* create an entry for the clock */
3117 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3118 priv->timer_timeout = timer_timeout;
3119 priv->timer_seqnum = timer->seqnum;
3120 GST_OBJECT_UNLOCK (jitterbuffer);
3122 /* release the lock so that the other end can push stuff or unlock */
3125 ret = gst_clock_id_wait (id, &clock_jitter);
3128 if (!priv->timer_running) {
3129 gst_clock_id_unref (id);
3130 priv->clock_id = NULL;
3134 if (ret != GST_CLOCK_UNSCHEDULED) {
3135 now = timer_timeout + MAX (clock_jitter, 0);
3136 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
3137 ret, priv->timer_seqnum, clock_jitter);
3139 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3141 /* and free the entry */
3142 gst_clock_id_unref (id);
3143 priv->clock_id = NULL;
3145 /* no timers, wait for activity */
3146 JBUF_WAIT_TIMER (priv);
3151 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3156 * This funcion implements the main pushing loop on the source pad.
3158 * It first tries to push as many buffers as possible. If there is a seqnum
3159 * mismatch, we wait for the next timeouts.
3162 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3164 GstRtpJitterBufferPrivate *priv;
3165 GstFlowReturn result = GST_FLOW_OK;
3167 priv = jitterbuffer->priv;
3169 JBUF_LOCK_CHECK (priv, flushing);
3171 result = handle_next_buffer (jitterbuffer);
3172 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3173 /* now wait for the next event */
3174 JBUF_WAIT_EVENT (priv, flushing);
3175 result = GST_FLOW_OK;
3178 while (result == GST_FLOW_OK);
3179 /* store result for upstream */
3180 priv->srcresult = result;
3181 /* if we get here we need to pause */
3187 result = priv->srcresult;
3194 JBUF_SIGNAL_QUERY (priv, FALSE);
3197 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3198 gst_flow_get_name (result));
3199 gst_pad_pause_task (priv->srcpad);
3200 if (result == GST_FLOW_EOS) {
3201 event = gst_event_new_eos ();
3202 gst_pad_push_event (priv->srcpad, event);
3208 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3209 * some sanity checks and then emit the handle-sync signal with the parameters.
3210 * This function must be called with the LOCK */
3212 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3214 GstRtpJitterBufferPrivate *priv;
3215 guint64 base_rtptime, base_time;
3217 guint64 last_rtptime;
3219 guint64 ext_rtptime, diff;
3220 gboolean valid = TRUE, keep = FALSE;
3222 priv = jitterbuffer->priv;
3224 /* get the last values from the jitterbuffer */
3225 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3226 &clock_rate, &last_rtptime);
3228 clock_base = priv->clock_base;
3229 ext_rtptime = priv->ext_rtptime;
3231 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3232 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3233 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3234 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3236 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3237 /* we keep this SR packet for later. When we get a valid RTP packet the
3238 * above values will be set and we can try to use the SR packet */
3239 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3242 /* we can't accept anything that happened before we did the last resync */
3243 if (base_rtptime > ext_rtptime) {
3244 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3247 /* the SR RTP timestamp must be something close to what we last observed
3248 * in the jitterbuffer */
3249 if (ext_rtptime > last_rtptime) {
3250 /* check how far ahead it is to our RTP timestamps */
3251 diff = ext_rtptime - last_rtptime;
3252 /* if bigger than 1 second, we drop it */
3253 if (diff > clock_rate) {
3254 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3255 /* should drop this, but some RTSP servers end up with bogus
3256 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3257 * so still trigger rptbin sync but invalidate RTCP data
3258 * (sync might use other methods) */
3261 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3262 G_GUINT64_FORMAT, last_rtptime, diff);
3268 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3272 s = gst_structure_new ("application/x-rtp-sync",
3273 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3274 "base-time", G_TYPE_UINT64, base_time,
3275 "clock-rate", G_TYPE_UINT, clock_rate,
3276 "clock-base", G_TYPE_UINT64, clock_base,
3277 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3278 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3280 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3281 gst_buffer_replace (&priv->last_sr, NULL);
3283 g_signal_emit (jitterbuffer,
3284 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3286 gst_structure_free (s);
3288 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3289 gst_buffer_replace (&priv->last_sr, NULL);
3293 static GstFlowReturn
3294 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3297 GstRtpJitterBuffer *jitterbuffer;
3298 GstRtpJitterBufferPrivate *priv;
3299 GstFlowReturn ret = GST_FLOW_OK;
3301 GstRTCPPacket packet;
3302 guint64 ext_rtptime;
3304 GstRTCPBuffer rtcp = { NULL, };
3306 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3308 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
3309 goto invalid_buffer;
3311 priv = jitterbuffer->priv;
3313 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3315 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3318 /* first packet must be SR or RR or else the validate would have failed */
3319 switch (gst_rtcp_packet_get_type (&packet)) {
3320 case GST_RTCP_TYPE_SR:
3321 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3327 gst_rtcp_buffer_unmap (&rtcp);
3329 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3332 /* convert the RTP timestamp to our extended timestamp, using the same offset
3333 * we used in the jitterbuffer */
3334 ext_rtptime = priv->jbuf->ext_rtptime;
3335 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3337 priv->ext_rtptime = ext_rtptime;
3338 gst_buffer_replace (&priv->last_sr, buffer);
3340 do_handle_sync (jitterbuffer);
3344 gst_buffer_unref (buffer);
3350 /* this is not fatal but should be filtered earlier */
3351 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3352 ("Received invalid RTCP payload, dropping"));
3358 /* this is not fatal but should be filtered earlier */
3359 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3360 ("Received empty RTCP payload, dropping"));
3361 gst_rtcp_buffer_unmap (&rtcp);
3367 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3368 gst_rtcp_buffer_unmap (&rtcp);
3375 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3378 gboolean res = FALSE;
3379 GstRtpJitterBuffer *jitterbuffer;
3380 GstRtpJitterBufferPrivate *priv;
3382 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3383 priv = jitterbuffer->priv;
3385 switch (GST_QUERY_TYPE (query)) {
3386 case GST_QUERY_CAPS:
3388 GstCaps *filter, *caps;
3390 gst_query_parse_caps (query, &filter);
3391 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3392 gst_query_set_caps_result (query, caps);
3393 gst_caps_unref (caps);
3398 if (GST_QUERY_IS_SERIALIZED (query)) {
3399 RTPJitterBufferItem *item;
3402 JBUF_LOCK_CHECK (priv, out_flushing);
3403 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3404 RTP_JITTER_BUFFER_MODE_BUFFER) {
3405 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3406 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3407 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3409 JBUF_SIGNAL_EVENT (priv);
3410 JBUF_WAIT_QUERY (priv, out_flushing);
3411 res = priv->last_query;
3413 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3418 res = gst_pad_query_default (pad, parent, query);
3426 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3434 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3437 GstRtpJitterBuffer *jitterbuffer;
3438 GstRtpJitterBufferPrivate *priv;
3439 gboolean res = FALSE;
3441 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3442 priv = jitterbuffer->priv;
3444 switch (GST_QUERY_TYPE (query)) {
3445 case GST_QUERY_LATENCY:
3447 /* We need to send the query upstream and add the returned latency to our
3449 GstClockTime min_latency, max_latency;
3451 GstClockTime our_latency;
3453 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3454 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3456 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3457 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3458 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3460 /* store this so that we can safely sync on the peer buffers. */
3462 priv->peer_latency = min_latency;
3463 our_latency = priv->latency_ns;
3466 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3467 GST_TIME_ARGS (our_latency));
3469 /* we add some latency but can buffer an infinite amount of time */
3470 min_latency += our_latency;
3473 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3474 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3475 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3477 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3481 case GST_QUERY_POSITION:
3483 GstClockTime start, last_out;
3486 gst_query_parse_position (query, &fmt, NULL);
3487 if (fmt != GST_FORMAT_TIME) {
3488 res = gst_pad_query_default (pad, parent, query);
3493 start = priv->npt_start;
3494 last_out = priv->last_out_time;
3497 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3498 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3499 GST_TIME_ARGS (last_out));
3501 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3502 /* bring 0-based outgoing time to stream time */
3503 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3506 res = gst_pad_query_default (pad, parent, query);
3510 case GST_QUERY_CAPS:
3512 GstCaps *filter, *caps;
3514 gst_query_parse_caps (query, &filter);
3515 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3516 gst_query_set_caps_result (query, caps);
3517 gst_caps_unref (caps);
3522 res = gst_pad_query_default (pad, parent, query);
3530 gst_rtp_jitter_buffer_set_property (GObject * object,
3531 guint prop_id, const GValue * value, GParamSpec * pspec)
3533 GstRtpJitterBuffer *jitterbuffer;
3534 GstRtpJitterBufferPrivate *priv;
3536 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3537 priv = jitterbuffer->priv;
3542 guint new_latency, old_latency;
3544 new_latency = g_value_get_uint (value);
3547 old_latency = priv->latency_ms;
3548 priv->latency_ms = new_latency;
3549 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3550 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3553 /* post message if latency changed, this will inform the parent pipeline
3554 * that a latency reconfiguration is possible/needed. */
3555 if (new_latency != old_latency) {
3556 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3557 GST_TIME_ARGS (new_latency * GST_MSECOND));
3559 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3560 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3564 case PROP_DROP_ON_LATENCY:
3566 priv->drop_on_latency = g_value_get_boolean (value);
3569 case PROP_TS_OFFSET:
3571 priv->ts_offset = g_value_get_int64 (value);
3572 priv->ts_discont = TRUE;
3577 priv->do_lost = g_value_get_boolean (value);
3582 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3585 case PROP_DO_RETRANSMISSION:
3587 priv->do_retransmission = g_value_get_boolean (value);
3590 case PROP_RTX_DELAY:
3592 priv->rtx_delay = g_value_get_int (value);
3595 case PROP_RTX_MIN_DELAY:
3597 priv->rtx_min_delay = g_value_get_uint (value);
3600 case PROP_RTX_DELAY_REORDER:
3602 priv->rtx_delay_reorder = g_value_get_int (value);
3605 case PROP_RTX_RETRY_TIMEOUT:
3607 priv->rtx_retry_timeout = g_value_get_int (value);
3610 case PROP_RTX_MIN_RETRY_TIMEOUT:
3612 priv->rtx_min_retry_timeout = g_value_get_int (value);
3615 case PROP_RTX_RETRY_PERIOD:
3617 priv->rtx_retry_period = g_value_get_int (value);
3621 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3627 gst_rtp_jitter_buffer_get_property (GObject * object,
3628 guint prop_id, GValue * value, GParamSpec * pspec)
3630 GstRtpJitterBuffer *jitterbuffer;
3631 GstRtpJitterBufferPrivate *priv;
3633 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3634 priv = jitterbuffer->priv;
3639 g_value_set_uint (value, priv->latency_ms);
3642 case PROP_DROP_ON_LATENCY:
3644 g_value_set_boolean (value, priv->drop_on_latency);
3647 case PROP_TS_OFFSET:
3649 g_value_set_int64 (value, priv->ts_offset);
3654 g_value_set_boolean (value, priv->do_lost);
3659 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3667 if (priv->srcresult != GST_FLOW_OK)
3670 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3672 g_value_set_int (value, percent);
3676 case PROP_DO_RETRANSMISSION:
3678 g_value_set_boolean (value, priv->do_retransmission);
3681 case PROP_RTX_DELAY:
3683 g_value_set_int (value, priv->rtx_delay);
3686 case PROP_RTX_MIN_DELAY:
3688 g_value_set_uint (value, priv->rtx_min_delay);
3691 case PROP_RTX_DELAY_REORDER:
3693 g_value_set_int (value, priv->rtx_delay_reorder);
3696 case PROP_RTX_RETRY_TIMEOUT:
3698 g_value_set_int (value, priv->rtx_retry_timeout);
3701 case PROP_RTX_MIN_RETRY_TIMEOUT:
3703 g_value_set_int (value, priv->rtx_min_retry_timeout);
3706 case PROP_RTX_RETRY_PERIOD:
3708 g_value_set_int (value, priv->rtx_retry_period);
3712 g_value_take_boxed (value,
3713 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
3716 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3721 static GstStructure *
3722 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
3726 JBUF_LOCK (jbuf->priv);
3727 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
3728 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
3729 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
3730 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
3731 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
3732 JBUF_UNLOCK (jbuf->priv);