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.
95 * Last reviewed on 2007-05-28 (0.10.5)
104 #include <gst/rtp/gstrtpbuffer.h>
106 #include "gstrtpjitterbuffer.h"
107 #include "rtpjitterbuffer.h"
108 #include "rtpstats.h"
110 #include <gst/glib-compat-private.h>
112 GST_DEBUG_CATEGORY (rtpjitterbuffer_debug);
113 #define GST_CAT_DEFAULT (rtpjitterbuffer_debug)
115 /* RTPJitterBuffer signals and args */
118 SIGNAL_REQUEST_PT_MAP,
126 #define DEFAULT_LATENCY_MS 200
127 #define DEFAULT_DROP_ON_LATENCY FALSE
128 #define DEFAULT_TS_OFFSET 0
129 #define DEFAULT_DO_LOST FALSE
130 #define DEFAULT_MODE RTP_JITTER_BUFFER_MODE_SLAVE
131 #define DEFAULT_PERCENT 0
132 #define DEFAULT_DO_RETRANSMISSION FALSE
133 #define DEFAULT_RTX_DELAY 20
134 #define DEFAULT_RTX_DELAY_REORDER 3
135 #define DEFAULT_RTX_RETRY_TIMEOUT 40
136 #define DEFAULT_RTX_RETRY_PERIOD 160
142 PROP_DROP_ON_LATENCY,
147 PROP_DO_RETRANSMISSION,
149 PROP_RTX_DELAY_REORDER,
150 PROP_RTX_RETRY_TIMEOUT,
151 PROP_RTX_RETRY_PERIOD,
156 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
158 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
160 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
163 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
165 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
166 GST_DEBUG ("waiting timer"); \
167 (priv)->waiting_timer = TRUE; \
168 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
169 (priv)->waiting_timer = FALSE; \
170 GST_DEBUG ("waiting timer done"); \
172 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
173 if (G_UNLIKELY ((priv)->waiting_timer)) { \
174 GST_DEBUG ("signal timer"); \
175 g_cond_signal (&(priv)->jbuf_timer); \
179 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
180 GST_DEBUG ("waiting event"); \
181 (priv)->waiting_event = TRUE; \
182 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
183 (priv)->waiting_event = FALSE; \
184 GST_DEBUG ("waiting event done"); \
185 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
188 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
189 if (G_UNLIKELY ((priv)->waiting_event)) { \
190 GST_DEBUG ("signal event"); \
191 g_cond_signal (&(priv)->jbuf_event); \
195 struct _GstRtpJitterBufferPrivate
197 GstPad *sinkpad, *srcpad;
200 RTPJitterBuffer *jbuf;
202 gboolean waiting_timer;
204 gboolean waiting_event;
211 gboolean timer_running;
212 GThread *timer_thread;
217 gboolean drop_on_latency;
220 gboolean do_retransmission;
222 gint rtx_delay_reorder;
223 gint rtx_retry_timeout;
224 gint rtx_retry_period;
226 /* the last seqnum we pushed out */
227 guint32 last_popped_seqnum;
228 /* the next expected seqnum we push */
230 /* last output time */
231 GstClockTime last_out_time;
232 /* last valid input timestamp and rtptime pair */
233 GstClockTime ips_dts;
235 GstClockTime packet_spacing;
237 /* the next expected seqnum we receive */
238 GstClockTime last_in_dts;
239 guint32 last_in_seqnum;
240 guint32 next_in_seqnum;
244 /* start and stop ranges */
245 GstClockTime npt_start;
246 GstClockTime npt_stop;
247 guint64 ext_timestamp;
248 guint64 last_elapsed;
249 guint64 estimated_eos;
255 /* clock rate and rtp timestamp offset */
259 gint64 prev_ts_offset;
261 /* when we are shutting down */
262 GstFlowReturn srcresult;
268 GstClockTime timer_timeout;
269 guint16 timer_seqnum;
270 /* the latency of the upstream peer, we have to take this into account when
271 * synchronizing the buffers. */
272 GstClockTime peer_latency;
276 /* some accounting */
278 guint64 num_duplicates;
279 guint64 num_rtx_requests;
280 guint64 num_rtx_success;
281 guint64 num_rtx_failed;
300 GstClockTime timeout;
301 GstClockTime duration;
302 GstClockTime rtx_base;
303 GstClockTime rtx_delay;
304 GstClockTime rtx_retry;
305 GstClockTime rtx_last;
309 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
310 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
311 GstRtpJitterBufferPrivate))
313 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
314 GST_STATIC_PAD_TEMPLATE ("sink",
317 GST_STATIC_CAPS ("application/x-rtp, "
318 "clock-rate = (int) [ 1, 2147483647 ]"
319 /* "payload = (int) , "
320 * "encoding-name = (string) "
324 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
325 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
328 GST_STATIC_CAPS ("application/x-rtcp")
331 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
332 GST_STATIC_PAD_TEMPLATE ("src",
335 GST_STATIC_CAPS ("application/x-rtp"
336 /* "payload = (int) , "
337 * "clock-rate = (int) , "
338 * "encoding-name = (string) "
342 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
344 #define gst_rtp_jitter_buffer_parent_class parent_class
345 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
347 /* object overrides */
348 static void gst_rtp_jitter_buffer_set_property (GObject * object,
349 guint prop_id, const GValue * value, GParamSpec * pspec);
350 static void gst_rtp_jitter_buffer_get_property (GObject * object,
351 guint prop_id, GValue * value, GParamSpec * pspec);
352 static void gst_rtp_jitter_buffer_finalize (GObject * object);
354 /* element overrides */
355 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
356 * element, GstStateChange transition);
357 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
358 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
359 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
361 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
364 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
365 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
368 /* sinkpad overrides */
369 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
370 GstObject * parent, GstEvent * event);
371 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
372 GstObject * parent, GstBuffer * buffer);
374 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
375 GstObject * parent, GstEvent * event);
376 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
377 GstObject * parent, GstBuffer * buffer);
379 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
380 GstObject * parent, GstQuery * query);
382 /* srcpad overrides */
383 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
384 GstObject * parent, GstEvent * event);
385 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
386 GstObject * parent, GstPadMode mode, gboolean active);
387 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
388 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
389 GstObject * parent, GstQuery * query);
392 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
394 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
395 gboolean active, guint64 base_time);
396 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
398 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
399 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
401 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
403 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
407 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
409 GObjectClass *gobject_class;
410 GstElementClass *gstelement_class;
412 gobject_class = (GObjectClass *) klass;
413 gstelement_class = (GstElementClass *) klass;
415 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
417 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
419 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
420 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
423 * GstRtpJitterBuffer:latency:
425 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
426 * for at most this time.
428 g_object_class_install_property (gobject_class, PROP_LATENCY,
429 g_param_spec_uint ("latency", "Buffer latency in ms",
430 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
431 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
433 * GstRtpJitterBuffer:drop-on-latency:
435 * Drop oldest buffers when the queue is completely filled.
437 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
438 g_param_spec_boolean ("drop-on-latency",
439 "Drop buffers when maximum latency is reached",
440 "Tells the jitterbuffer to never exceed the given latency in size",
441 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
443 * GstRtpJitterBuffer:ts-offset:
445 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
446 * This is mainly used to ensure interstream synchronisation.
448 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
449 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
450 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
451 G_MAXINT64, DEFAULT_TS_OFFSET,
452 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
455 * GstRtpJitterBuffer:do-lost:
457 * Send out a GstRTPPacketLost event downstream when a packet is considered
460 g_object_class_install_property (gobject_class, PROP_DO_LOST,
461 g_param_spec_boolean ("do-lost", "Do Lost",
462 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
463 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
466 * GstRtpJitterBuffer:mode:
468 * Control the buffering and timestamping mode used by the jitterbuffer.
470 g_object_class_install_property (gobject_class, PROP_MODE,
471 g_param_spec_enum ("mode", "Mode",
472 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
473 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
475 * GstRtpJitterBuffer:percent:
477 * The percent of the jitterbuffer that is filled.
479 g_object_class_install_property (gobject_class, PROP_PERCENT,
480 g_param_spec_int ("percent", "percent",
481 "The buffer filled percent", 0, 100,
482 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
484 * GstRtpJitterBuffer:do-retransmission:
486 * Send out a GstRTPRetransmission event upstream when a packet is considered
487 * late and should be retransmitted.
491 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
492 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
493 "Send retransmission events upstream when a packet is late",
494 DEFAULT_DO_RETRANSMISSION,
495 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
498 * GstRtpJitterBuffer:rtx-delay:
500 * When a packet did not arrive at the expected time, wait this extra amount
501 * of time before sending a retransmission event.
503 * When -1 is used, the max jitter will be used as extra delay.
507 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
508 g_param_spec_int ("rtx-delay", "RTX Delay",
509 "Extra time in ms to wait before sending retransmission "
510 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
511 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
513 * GstRtpJitterBuffer:rtx-delay-reorder:
515 * Assume that a retransmission event should be sent when we see
516 * this much packet reordering.
518 * When -1 is used, the value will be estimated based on observed packet
523 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
524 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
525 "Sending retransmission event when this much reordering (-1 automatic)",
526 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
527 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
529 * GstRtpJitterBuffer::rtx-retry-timeout:
531 * When no packet has been received after sending a retransmission event
532 * for this time, retry sending a retransmission event.
534 * When -1 is used, the value will be estimated based on observed round
539 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
540 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
541 "Retry sending a transmission event after this timeout in "
542 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
543 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
545 * GstRtpJitterBuffer:rtx-retry-period:
547 * The amount of time to try to get a retransmission.
549 * When -1 is used, the value will be estimated based on the jitterbuffer
550 * latency and the observed round trip time.
554 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
555 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
556 "Try to get a retransmission for this many ms "
557 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
558 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
560 * GstRtpJitterBuffer:stats:
562 * Various jitterbuffer statistics. This property returns a GstStructure
563 * with name application/x-rtp-jitterbuffer-stats with the following fields:
565 * "rtx-count" G_TYPE_UINT64 The number of retransmissions requested
566 * "rtx-success-count" G_TYPE_UINT64 The number of successful retransmissions
567 * "rtx-per-packet" G_TYPE_DOUBLE Average number of RTX per packet
568 * "rtx-rtt" G_TYPE_UINT64 Average round trip time per RTX
572 g_object_class_install_property (gobject_class, PROP_STATS,
573 g_param_spec_boxed ("stats", "Statistics",
574 "Various statistics", GST_TYPE_STRUCTURE,
575 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
578 * GstRtpJitterBuffer::request-pt-map:
579 * @buffer: the object which received the signal
582 * Request the payload type as #GstCaps for @pt.
584 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
585 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
586 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
587 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
588 GST_TYPE_CAPS, 1, G_TYPE_UINT);
590 * GstRtpJitterBuffer::handle-sync:
591 * @buffer: the object which received the signal
592 * @struct: a GstStructure containing sync values.
594 * Be notified of new sync values.
596 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
597 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
598 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
599 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
600 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
603 * GstRtpJitterBuffer::on-npt-stop:
604 * @buffer: the object which received the signal
606 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
607 * the npt-stop position.
609 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
610 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
611 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
612 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
613 G_TYPE_NONE, 0, G_TYPE_NONE);
616 * GstRtpJitterBuffer::clear-pt-map:
617 * @buffer: the object which received the signal
619 * Invalidate the clock-rate as obtained with the
620 * #GstRtpJitterBuffer::request-pt-map signal.
622 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
623 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
624 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
625 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
626 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
629 * GstRtpJitterBuffer::set-active:
630 * @buffer: the object which received the signal
632 * Start pushing out packets with the given base time. This signal is only
633 * useful in buffering mode.
635 * Returns: the time of the last pushed packet.
637 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
638 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
639 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
640 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
641 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
644 gstelement_class->change_state =
645 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
646 gstelement_class->request_new_pad =
647 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
648 gstelement_class->release_pad =
649 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
650 gstelement_class->provide_clock =
651 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
653 gst_element_class_add_pad_template (gstelement_class,
654 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
655 gst_element_class_add_pad_template (gstelement_class,
656 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
657 gst_element_class_add_pad_template (gstelement_class,
658 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
660 gst_element_class_set_static_metadata (gstelement_class,
661 "RTP packet jitter-buffer", "Filter/Network/RTP",
662 "A buffer that deals with network jitter and other transmission faults",
663 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
664 "Wim Taymans <wim.taymans@gmail.com>");
666 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
667 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
669 GST_DEBUG_CATEGORY_INIT
670 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
674 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
676 GstRtpJitterBufferPrivate *priv;
678 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
679 jitterbuffer->priv = priv;
681 priv->latency_ms = DEFAULT_LATENCY_MS;
682 priv->latency_ns = priv->latency_ms * GST_MSECOND;
683 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
684 priv->do_lost = DEFAULT_DO_LOST;
685 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
686 priv->rtx_delay = DEFAULT_RTX_DELAY;
687 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
688 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
689 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
691 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
692 priv->jbuf = rtp_jitter_buffer_new ();
693 g_mutex_init (&priv->jbuf_lock);
694 g_cond_init (&priv->jbuf_timer);
695 g_cond_init (&priv->jbuf_event);
697 /* reset skew detection initialy */
698 rtp_jitter_buffer_reset_skew (priv->jbuf);
699 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
700 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
704 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
707 gst_pad_set_activatemode_function (priv->srcpad,
708 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
709 gst_pad_set_query_function (priv->srcpad,
710 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
711 gst_pad_set_event_function (priv->srcpad,
712 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
715 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
718 gst_pad_set_chain_function (priv->sinkpad,
719 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
720 gst_pad_set_event_function (priv->sinkpad,
721 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
722 gst_pad_set_query_function (priv->sinkpad,
723 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
725 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
726 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
728 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
731 #define ITEM_TYPE_BUFFER 0
732 #define ITEM_TYPE_LOST 1
733 #define ITEM_TYPE_EVENT 2
735 static RTPJitterBufferItem *
736 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
737 guint seqnum, guint count, guint rtptime)
739 RTPJitterBufferItem *item;
741 item = g_slice_new (RTPJitterBufferItem);
748 item->seqnum = seqnum;
750 item->rtptime = rtptime;
756 free_item (RTPJitterBufferItem * item)
759 gst_mini_object_unref (item->data);
760 g_slice_free (RTPJitterBufferItem, item);
764 gst_rtp_jitter_buffer_finalize (GObject * object)
766 GstRtpJitterBuffer *jitterbuffer;
767 GstRtpJitterBufferPrivate *priv;
769 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
770 priv = jitterbuffer->priv;
772 g_array_free (priv->timers, TRUE);
773 g_mutex_clear (&priv->jbuf_lock);
774 g_cond_clear (&priv->jbuf_timer);
775 g_cond_clear (&priv->jbuf_event);
777 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
778 g_object_unref (priv->jbuf);
780 G_OBJECT_CLASS (parent_class)->finalize (object);
784 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
786 GstRtpJitterBuffer *jitterbuffer;
787 GstPad *otherpad = NULL;
791 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
793 if (pad == jitterbuffer->priv->sinkpad) {
794 otherpad = jitterbuffer->priv->srcpad;
795 } else if (pad == jitterbuffer->priv->srcpad) {
796 otherpad = jitterbuffer->priv->sinkpad;
797 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
801 g_value_init (&val, GST_TYPE_PAD);
802 g_value_set_object (&val, otherpad);
803 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
804 g_value_unset (&val);
810 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
812 GstRtpJitterBufferPrivate *priv;
814 priv = jitterbuffer->priv;
816 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
819 gst_pad_new_from_static_template
820 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
821 gst_pad_set_chain_function (priv->rtcpsinkpad,
822 gst_rtp_jitter_buffer_chain_rtcp);
823 gst_pad_set_event_function (priv->rtcpsinkpad,
824 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
825 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
826 gst_rtp_jitter_buffer_iterate_internal_links);
827 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
828 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
830 return priv->rtcpsinkpad;
834 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
836 GstRtpJitterBufferPrivate *priv;
838 priv = jitterbuffer->priv;
840 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
842 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
844 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
845 priv->rtcpsinkpad = NULL;
849 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
850 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
852 GstRtpJitterBuffer *jitterbuffer;
853 GstElementClass *klass;
855 GstRtpJitterBufferPrivate *priv;
857 g_return_val_if_fail (templ != NULL, NULL);
858 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
860 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
861 priv = jitterbuffer->priv;
862 klass = GST_ELEMENT_GET_CLASS (element);
864 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
866 /* figure out the template */
867 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
868 if (priv->rtcpsinkpad != NULL)
871 result = create_rtcp_sink (jitterbuffer);
880 g_warning ("rtpjitterbuffer: this is not our template");
885 g_warning ("rtpjitterbuffer: pad already requested");
891 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
893 GstRtpJitterBuffer *jitterbuffer;
894 GstRtpJitterBufferPrivate *priv;
896 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
897 g_return_if_fail (GST_IS_PAD (pad));
899 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
900 priv = jitterbuffer->priv;
902 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
904 if (priv->rtcpsinkpad == pad) {
905 remove_rtcp_sink (jitterbuffer);
914 g_warning ("gstjitterbuffer: asked to release an unknown pad");
920 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
922 return gst_system_clock_obtain ();
926 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
928 GstRtpJitterBufferPrivate *priv;
930 priv = jitterbuffer->priv;
932 /* this will trigger a new pt-map request signal, FIXME, do something better. */
935 priv->clock_rate = -1;
936 /* do not clear current content, but refresh state for new arrival */
937 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
938 rtp_jitter_buffer_reset_skew (priv->jbuf);
943 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
946 GstRtpJitterBufferPrivate *priv;
947 GstClockTime last_out;
948 RTPJitterBufferItem *item;
953 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
954 active, GST_TIME_ARGS (offset));
956 if (active != priv->active) {
957 /* add the amount of time spent in paused to the output offset. All
958 * outgoing buffers will have this offset applied to their timestamps in
959 * order to make them arrive in time in the sink. */
960 priv->out_offset = offset;
961 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
962 GST_TIME_ARGS (priv->out_offset));
963 priv->active = active;
964 JBUF_SIGNAL_EVENT (priv);
967 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
969 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
970 /* head buffer timestamp and offset gives our output time */
971 last_out = item->dts + priv->ts_offset;
973 /* use last known time when the buffer is empty */
974 last_out = priv->last_out_time;
982 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
984 GstRtpJitterBuffer *jitterbuffer;
985 GstRtpJitterBufferPrivate *priv;
990 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
991 priv = jitterbuffer->priv;
993 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
995 caps = gst_pad_peer_query_caps (other, filter);
997 templ = gst_pad_get_pad_template_caps (pad);
999 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1004 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1006 intersect = gst_caps_intersect (caps, templ);
1007 gst_caps_unref (caps);
1008 gst_caps_unref (templ);
1012 gst_object_unref (jitterbuffer);
1018 * Must be called with JBUF_LOCK held
1022 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1025 GstRtpJitterBufferPrivate *priv;
1026 GstStructure *caps_struct;
1030 priv = jitterbuffer->priv;
1032 /* first parse the caps */
1033 caps_struct = gst_caps_get_structure (caps, 0);
1035 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1037 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1038 * measure the amount of data in the buffer */
1039 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1042 if (priv->clock_rate <= 0)
1045 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1047 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1049 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1050 * can use this to track the amount of time elapsed on the sender. */
1051 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1052 priv->clock_base = val;
1054 priv->clock_base = -1;
1056 priv->ext_timestamp = priv->clock_base;
1058 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1061 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1062 /* first expected seqnum, only update when we didn't have a previous base. */
1063 if (priv->next_in_seqnum == -1)
1064 priv->next_in_seqnum = val;
1065 if (priv->next_seqnum == -1)
1066 priv->next_seqnum = val;
1069 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1071 /* the start and stop times. The seqnum-base corresponds to the start time. We
1072 * will keep track of the seqnums on the output and when we reach the one
1073 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1074 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1075 priv->npt_start = tval;
1077 priv->npt_start = 0;
1079 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1080 priv->npt_stop = tval;
1082 priv->npt_stop = -1;
1084 GST_DEBUG_OBJECT (jitterbuffer,
1085 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1086 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1093 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1098 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1104 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1106 GstRtpJitterBufferPrivate *priv;
1108 priv = jitterbuffer->priv;
1111 /* mark ourselves as flushing */
1112 priv->srcresult = GST_FLOW_FLUSHING;
1113 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1114 /* this unblocks any waiting pops on the src pad task */
1115 JBUF_SIGNAL_EVENT (priv);
1120 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1122 GstRtpJitterBufferPrivate *priv;
1124 priv = jitterbuffer->priv;
1127 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1128 /* Mark as non flushing */
1129 priv->srcresult = GST_FLOW_OK;
1130 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1131 priv->last_popped_seqnum = -1;
1132 priv->last_out_time = -1;
1133 priv->next_seqnum = -1;
1134 priv->ips_rtptime = -1;
1135 priv->ips_dts = GST_CLOCK_TIME_NONE;
1136 priv->packet_spacing = 0;
1137 priv->next_in_seqnum = -1;
1138 priv->clock_rate = -1;
1140 priv->estimated_eos = -1;
1141 priv->last_elapsed = 0;
1142 priv->ext_timestamp = -1;
1143 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1144 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1145 rtp_jitter_buffer_reset_skew (priv->jbuf);
1146 remove_all_timers (jitterbuffer);
1151 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1152 GstPadMode mode, gboolean active)
1155 GstRtpJitterBuffer *jitterbuffer = NULL;
1157 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1160 case GST_PAD_MODE_PUSH:
1162 /* allow data processing */
1163 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1165 /* start pushing out buffers */
1166 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1167 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1168 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1170 /* make sure all data processing stops ASAP */
1171 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1173 /* NOTE this will hardlock if the state change is called from the src pad
1174 * task thread because we will _join() the thread. */
1175 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1176 result = gst_pad_stop_task (pad);
1186 static GstStateChangeReturn
1187 gst_rtp_jitter_buffer_change_state (GstElement * element,
1188 GstStateChange transition)
1190 GstRtpJitterBuffer *jitterbuffer;
1191 GstRtpJitterBufferPrivate *priv;
1192 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1194 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1195 priv = jitterbuffer->priv;
1197 switch (transition) {
1198 case GST_STATE_CHANGE_NULL_TO_READY:
1200 case GST_STATE_CHANGE_READY_TO_PAUSED:
1202 /* reset negotiated values */
1203 priv->clock_rate = -1;
1204 priv->clock_base = -1;
1205 priv->peer_latency = 0;
1207 /* block until we go to PLAYING */
1208 priv->blocked = TRUE;
1209 priv->timer_running = TRUE;
1210 priv->timer_thread =
1211 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1214 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1216 /* unblock to allow streaming in PLAYING */
1217 priv->blocked = FALSE;
1218 JBUF_SIGNAL_EVENT (priv);
1219 JBUF_SIGNAL_TIMER (priv);
1226 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1228 switch (transition) {
1229 case GST_STATE_CHANGE_READY_TO_PAUSED:
1230 /* we are a live element because we sync to the clock, which we can only
1231 * do in the PLAYING state */
1232 if (ret != GST_STATE_CHANGE_FAILURE)
1233 ret = GST_STATE_CHANGE_NO_PREROLL;
1235 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1237 /* block to stop streaming when PAUSED */
1238 priv->blocked = TRUE;
1239 unschedule_current_timer (jitterbuffer);
1241 if (ret != GST_STATE_CHANGE_FAILURE)
1242 ret = GST_STATE_CHANGE_NO_PREROLL;
1244 case GST_STATE_CHANGE_PAUSED_TO_READY:
1246 gst_buffer_replace (&priv->last_sr, NULL);
1247 priv->timer_running = FALSE;
1248 unschedule_current_timer (jitterbuffer);
1249 JBUF_SIGNAL_TIMER (priv);
1251 g_thread_join (priv->timer_thread);
1252 priv->timer_thread = NULL;
1254 case GST_STATE_CHANGE_READY_TO_NULL:
1264 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1267 gboolean ret = TRUE;
1268 GstRtpJitterBuffer *jitterbuffer;
1269 GstRtpJitterBufferPrivate *priv;
1271 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1272 priv = jitterbuffer->priv;
1274 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1276 switch (GST_EVENT_TYPE (event)) {
1277 case GST_EVENT_LATENCY:
1279 GstClockTime latency;
1281 gst_event_parse_latency (event, &latency);
1283 GST_DEBUG_OBJECT (jitterbuffer,
1284 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1287 /* adjust the overall buffer delay to the total pipeline latency in
1288 * buffering mode because if downstream consumes too fast (because of
1289 * large latency or queues, we would start rebuffering again. */
1290 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1291 RTP_JITTER_BUFFER_MODE_BUFFER) {
1292 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1296 ret = gst_pad_push_event (priv->sinkpad, event);
1300 ret = gst_pad_push_event (priv->sinkpad, event);
1307 /* handles and stores the event in the jitterbuffer, must be called with
1310 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1312 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1313 RTPJitterBufferItem *item;
1315 switch (GST_EVENT_TYPE (event)) {
1316 case GST_EVENT_CAPS:
1320 gst_event_parse_caps (event, &caps);
1321 if (!gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps))
1326 case GST_EVENT_SEGMENT:
1327 gst_event_copy_segment (event, &priv->segment);
1329 /* we need time for now */
1330 if (priv->segment.format != GST_FORMAT_TIME)
1331 goto newseg_wrong_format;
1333 GST_DEBUG_OBJECT (jitterbuffer,
1334 "newsegment: %" GST_SEGMENT_FORMAT, &priv->segment);
1344 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1345 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1346 rtp_jitter_buffer_insert (priv->jbuf, item, NULL, NULL);
1347 JBUF_SIGNAL_EVENT (priv);
1354 GST_DEBUG_OBJECT (jitterbuffer, "received invalid caps");
1355 gst_event_unref (event);
1358 newseg_wrong_format:
1360 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1361 gst_event_unref (event);
1367 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1370 gboolean ret = TRUE;
1371 GstRtpJitterBuffer *jitterbuffer;
1372 GstRtpJitterBufferPrivate *priv;
1374 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1375 priv = jitterbuffer->priv;
1377 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1379 switch (GST_EVENT_TYPE (event)) {
1380 case GST_EVENT_FLUSH_START:
1381 ret = gst_pad_push_event (priv->srcpad, event);
1382 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1383 /* wait for the loop to go into PAUSED */
1384 gst_pad_pause_task (priv->srcpad);
1386 case GST_EVENT_FLUSH_STOP:
1387 ret = gst_pad_push_event (priv->srcpad, event);
1389 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1390 GST_PAD_MODE_PUSH, TRUE);
1393 if (GST_EVENT_IS_SERIALIZED (event)) {
1394 /* serialized events go in the queue */
1396 if (priv->srcresult != GST_FLOW_OK) {
1397 /* Errors in sticky event pushing are no problem and ignored here
1398 * as they will cause more meaningful errors during data flow.
1399 * For EOS events, that are not followed by data flow, we still
1400 * return FALSE here though.
1402 if (!GST_EVENT_IS_STICKY (event) ||
1403 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1404 goto out_flow_error;
1406 /* refuse more events on EOS */
1409 ret = queue_event (jitterbuffer, event);
1412 /* non-serialized events are forwarded downstream immediately */
1413 ret = gst_pad_push_event (priv->srcpad, event);
1422 GST_DEBUG_OBJECT (jitterbuffer,
1423 "refusing event, we have a downstream flow error: %s",
1424 gst_flow_get_name (priv->srcresult));
1426 gst_event_unref (event);
1431 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1433 gst_event_unref (event);
1439 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1442 gboolean ret = TRUE;
1443 GstRtpJitterBuffer *jitterbuffer;
1445 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1447 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1449 switch (GST_EVENT_TYPE (event)) {
1450 case GST_EVENT_FLUSH_START:
1451 gst_event_unref (event);
1453 case GST_EVENT_FLUSH_STOP:
1454 gst_event_unref (event);
1457 ret = gst_pad_event_default (pad, parent, event);
1465 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1466 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1467 * GST_FLOW_FLUSHING when the element is shutting down. On success
1468 * GST_FLOW_OK is returned.
1470 static GstFlowReturn
1471 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1475 GValue args[2] = { {0}, {0} };
1479 g_value_init (&args[0], GST_TYPE_ELEMENT);
1480 g_value_set_object (&args[0], jitterbuffer);
1481 g_value_init (&args[1], G_TYPE_UINT);
1482 g_value_set_uint (&args[1], pt);
1484 g_value_init (&ret, GST_TYPE_CAPS);
1485 g_value_set_boxed (&ret, NULL);
1487 JBUF_UNLOCK (jitterbuffer->priv);
1488 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1490 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1492 g_value_unset (&args[0]);
1493 g_value_unset (&args[1]);
1494 caps = (GstCaps *) g_value_dup_boxed (&ret);
1495 g_value_unset (&ret);
1499 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1500 gst_caps_unref (caps);
1502 if (G_UNLIKELY (!res))
1510 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1511 return GST_FLOW_ERROR;
1515 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1516 return GST_FLOW_FLUSHING;
1520 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1521 return GST_FLOW_ERROR;
1525 /* call with jbuf lock held */
1527 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint * percent)
1529 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1531 /* too short a stream, or too close to EOS will never really fill buffer */
1532 if (*percent != -1 && priv->npt_stop != -1 &&
1533 priv->npt_stop - priv->npt_start <=
1534 rtp_jitter_buffer_get_delay (priv->jbuf)) {
1535 GST_DEBUG_OBJECT (jitterbuffer, "short stream; faking full buffer");
1536 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
1542 post_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1544 GstMessage *message;
1546 /* Post a buffering message */
1547 message = gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1548 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1550 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), message);
1554 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1556 GstRtpJitterBufferPrivate *priv;
1558 priv = jitterbuffer->priv;
1560 if (timestamp == -1)
1563 /* apply the timestamp offset, this is used for inter stream sync */
1564 timestamp += priv->ts_offset;
1565 /* add the offset, this is used when buffering */
1566 timestamp += priv->out_offset;
1572 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1574 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1575 TimerData *timer = NULL;
1578 len = priv->timers->len;
1579 for (i = 0; i < len; i++) {
1580 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1581 if (test->seqnum == seqnum && test->type == type) {
1590 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1592 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1594 if (priv->clock_id) {
1595 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1596 gst_clock_id_unschedule (priv->clock_id);
1597 priv->clock_id = NULL;
1602 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1604 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1605 GstClockTime test_timeout;
1607 if ((test_timeout = timer->timeout) == -1)
1610 if (timer->type != TIMER_TYPE_EXPECTED) {
1611 /* add our latency and offset to get output times. */
1612 test_timeout = apply_offset (jitterbuffer, test_timeout);
1613 test_timeout += priv->latency_ns;
1615 return test_timeout;
1619 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1621 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1623 if (priv->clock_id) {
1624 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1626 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1627 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1629 if (timeout == -1 || timeout < priv->timer_timeout)
1630 unschedule_current_timer (jitterbuffer);
1635 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1636 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1637 GstClockTime duration)
1639 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1643 GST_DEBUG_OBJECT (jitterbuffer,
1644 "add timer for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1645 GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (timeout), GST_TIME_ARGS (delay));
1647 len = priv->timers->len;
1648 g_array_set_size (priv->timers, len + 1);
1649 timer = &g_array_index (priv->timers, TimerData, len);
1652 timer->seqnum = seqnum;
1654 timer->timeout = timeout + delay;
1655 timer->duration = duration;
1656 if (type == TIMER_TYPE_EXPECTED) {
1657 timer->rtx_base = timeout;
1658 timer->rtx_delay = delay;
1659 timer->rtx_retry = 0;
1661 timer->num_rtx_retry = 0;
1662 recalculate_timer (jitterbuffer, timer);
1663 JBUF_SIGNAL_TIMER (priv);
1669 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1670 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1672 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1673 gboolean seqchange, timechange;
1676 seqchange = timer->seqnum != seqnum;
1677 timechange = timer->timeout != timeout;
1679 if (!seqchange && !timechange)
1682 oldseq = timer->seqnum;
1684 GST_DEBUG_OBJECT (jitterbuffer,
1685 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1686 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1688 timer->timeout = timeout + delay;
1689 timer->seqnum = seqnum;
1691 timer->rtx_base = timeout;
1692 timer->rtx_delay = delay;
1693 timer->rtx_retry = 0;
1696 if (priv->clock_id) {
1697 /* we changed the seqnum and there is a timer currently waiting with this
1698 * seqnum, unschedule it */
1699 if (seqchange && priv->timer_seqnum == oldseq)
1700 unschedule_current_timer (jitterbuffer);
1701 /* we changed the time, check if it is earlier than what we are waiting
1702 * for and unschedule if so */
1703 else if (timechange)
1704 recalculate_timer (jitterbuffer, timer);
1709 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1710 guint16 seqnum, GstClockTime timeout)
1714 /* find the seqnum timer */
1715 timer = find_timer (jitterbuffer, type, seqnum);
1716 if (timer == NULL) {
1717 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1719 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1725 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1727 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1730 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1731 unschedule_current_timer (jitterbuffer);
1734 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1735 g_array_remove_index_fast (priv->timers, idx);
1740 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1742 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1743 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1744 g_array_set_size (priv->timers, 0);
1745 unschedule_current_timer (jitterbuffer);
1748 /* we just received a packet with seqnum and dts.
1750 * First check for old seqnum that we are still expecting. If the gap with the
1751 * current seqnum is too big, unschedule the timeouts.
1753 * If we have a valid packet spacing estimate we can set a timer for when we
1754 * should receive the next packet.
1755 * If we don't have a valid estimate, we remove any timer we might have
1756 * had for this packet.
1759 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1760 GstClockTime dts, gboolean do_next_seqnum)
1762 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1763 TimerData *timer = NULL;
1766 /* go through all timers and unschedule the ones with a large gap, also find
1767 * the timer for the seqnum */
1768 len = priv->timers->len;
1769 for (i = 0; i < len; i++) {
1770 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1773 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
1775 GST_DEBUG_OBJECT (jitterbuffer, "%d, #%d<->#%d gap %d", i,
1776 test->seqnum, seqnum, gap);
1779 GST_DEBUG ("found timer for current seqnum");
1780 /* the timer for the current seqnum */
1782 } else if (gap > priv->rtx_delay_reorder) {
1783 /* max gap, we exceeded the max reorder distance and we don't expect the
1784 * missing packet to be this reordered */
1785 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
1786 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
1790 if (priv->packet_spacing > 0 && do_next_seqnum && priv->do_retransmission) {
1791 GstClockTime expected, delay;
1793 /* calculate expected arrival time of the next seqnum */
1794 expected = dts + priv->packet_spacing;
1795 delay = priv->rtx_delay * GST_MSECOND;
1797 /* and update/install timer for next seqnum */
1799 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
1802 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
1803 expected, delay, priv->packet_spacing);
1804 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
1806 if (timer->num_rtx_retry > 0) {
1807 GstClockTime rtx_last;
1809 /* we scheduled a retry for this packet and now we have it */
1810 priv->num_rtx_success++;
1811 /* all the previous retry attempts failed */
1812 priv->num_rtx_failed += timer->num_rtx_retry - 1;
1813 /* number of retries before receiving the packet */
1814 if (priv->avg_rtx_num == 0.0)
1815 priv->avg_rtx_num = timer->num_rtx_retry;
1817 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
1818 /* calculate the delay between retransmission request and receiving this
1819 * packet, start with when we scheduled this timeout last */
1820 rtx_last = timer->rtx_last;
1821 if (dts > rtx_last) {
1823 /* we have a valid delay if this packet arrived after we scheduled the
1825 delay = dts - rtx_last;
1826 if (priv->avg_rtx_rtt == 0)
1827 priv->avg_rtx_rtt = delay;
1829 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
1831 GST_LOG_OBJECT (jitterbuffer,
1832 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
1833 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
1834 ", avg-num %g, avg-rtt %" G_GUINT64_FORMAT, priv->num_rtx_success,
1835 priv->num_rtx_failed, priv->num_rtx_requests, priv->num_duplicates,
1836 priv->avg_rtx_num, priv->avg_rtx_rtt);
1838 /* if we had a timer, remove it, we don't know when to expect the next
1840 remove_timer (jitterbuffer, timer);
1841 /* we signal the _loop function because this new packet could be the one
1842 * it was waiting for */
1843 JBUF_SIGNAL_EVENT (priv);
1848 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
1851 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1853 /* we need consecutive seqnums with a different
1854 * rtptime to estimate the packet spacing. */
1855 if (priv->ips_rtptime != rtptime) {
1856 /* rtptime changed, check dts diff */
1857 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
1858 priv->packet_spacing = dts - priv->ips_dts;
1859 GST_DEBUG_OBJECT (jitterbuffer,
1860 "new packet spacing %" GST_TIME_FORMAT,
1861 GST_TIME_ARGS (priv->packet_spacing));
1863 priv->ips_rtptime = rtptime;
1864 priv->ips_dts = dts;
1869 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
1870 guint16 seqnum, GstClockTime dts, gint gap)
1872 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1873 GstClockTime total_duration, duration, expected_dts;
1876 GST_DEBUG_OBJECT (jitterbuffer,
1877 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
1878 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
1880 /* the total duration spanned by the missing packets */
1881 if (dts >= priv->last_in_dts)
1882 total_duration = dts - priv->last_in_dts;
1886 /* interpolate between the current time and the last time based on
1887 * number of packets we are missing, this is the estimated duration
1888 * for the missing packet based on equidistant packet spacing. */
1889 duration = total_duration / (gap + 1);
1891 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
1892 GST_TIME_ARGS (duration));
1894 if (total_duration > priv->latency_ns) {
1895 GstClockTime gap_time;
1898 gap_time = total_duration - priv->latency_ns;
1901 lost_packets = gap_time / duration;
1902 gap_time = lost_packets * duration;
1907 /* too many lost packets, some of the missing packets are already
1908 * too late and we can generate lost packet events for them. */
1909 GST_DEBUG_OBJECT (jitterbuffer, "too many lost packets %" GST_TIME_FORMAT
1910 " > %" GST_TIME_FORMAT ", consider %u lost",
1911 GST_TIME_ARGS (total_duration), GST_TIME_ARGS (priv->latency_ns),
1914 /* this timer will fire immediately and the lost event will be pushed from
1915 * the timer thread */
1916 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
1917 priv->last_in_dts + duration, 0, gap_time);
1919 expected += lost_packets;
1920 priv->last_in_dts += gap_time;
1923 expected_dts = priv->last_in_dts + duration;
1925 if (priv->do_retransmission) {
1928 type = TIMER_TYPE_EXPECTED;
1929 /* if we had a timer for the first missing packet, update it. */
1930 if ((timer = find_timer (jitterbuffer, type, expected))) {
1931 GstClockTime timeout = timer->timeout;
1933 timer->duration = duration;
1934 if (timeout > expected_dts) {
1935 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
1936 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
1940 expected_dts += duration;
1943 type = TIMER_TYPE_LOST;
1946 while (expected < seqnum) {
1947 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
1948 expected_dts += duration;
1953 static GstFlowReturn
1954 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
1957 GstRtpJitterBuffer *jitterbuffer;
1958 GstRtpJitterBufferPrivate *priv;
1960 guint32 expected, rtptime;
1961 GstFlowReturn ret = GST_FLOW_OK;
1962 GstClockTime dts, pts;
1967 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
1968 gboolean do_next_seqnum = FALSE;
1969 RTPJitterBufferItem *item;
1971 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1973 priv = jitterbuffer->priv;
1975 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
1976 goto invalid_buffer;
1978 pt = gst_rtp_buffer_get_payload_type (&rtp);
1979 seqnum = gst_rtp_buffer_get_seq (&rtp);
1980 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1981 gst_rtp_buffer_unmap (&rtp);
1983 /* make sure we have PTS and DTS set */
1984 pts = GST_BUFFER_PTS (buffer);
1985 dts = GST_BUFFER_DTS (buffer);
1991 /* take the DTS of the buffer. This is the time when the packet was
1992 * received and is used to calculate jitter and clock skew. We will adjust
1993 * this DTS with the smoothed value after processing it in the
1994 * jitterbuffer and assign it as the PTS. */
1995 /* bring to running time */
1996 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
1998 GST_DEBUG_OBJECT (jitterbuffer,
1999 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2000 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2002 JBUF_LOCK_CHECK (priv, out_flushing);
2004 if (G_UNLIKELY (priv->last_pt != pt)) {
2007 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2011 /* reset clock-rate so that we get a new one */
2012 priv->clock_rate = -1;
2014 /* Try to get the clock-rate from the caps first if we can. If there are no
2015 * caps we must fire the signal to get the clock-rate. */
2016 if ((caps = gst_pad_get_current_caps (pad))) {
2017 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2018 gst_caps_unref (caps);
2022 if (G_UNLIKELY (priv->clock_rate == -1)) {
2023 /* no clock rate given on the caps, try to get one with the signal */
2024 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2025 pt) == GST_FLOW_FLUSHING)
2028 if (G_UNLIKELY (priv->clock_rate == -1))
2032 /* don't accept more data on EOS */
2033 if (G_UNLIKELY (priv->eos))
2036 expected = priv->next_in_seqnum;
2038 /* now check against our expected seqnum */
2039 if (G_LIKELY (expected != -1)) {
2042 /* now calculate gap */
2043 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2045 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2046 expected, seqnum, gap);
2048 if (G_LIKELY (gap == 0)) {
2049 /* packet is expected */
2050 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2051 do_next_seqnum = TRUE;
2053 gboolean reset = FALSE;
2056 /* we received an old packet */
2057 if (G_UNLIKELY (gap < -RTP_MAX_MISORDER)) {
2058 /* too old packet, reset */
2059 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too old %d < %d", gap,
2063 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2066 /* new packet, we are missing some packets */
2067 if (G_UNLIKELY (gap > RTP_MAX_DROPOUT)) {
2068 /* packet too far in future, reset */
2069 GST_DEBUG_OBJECT (jitterbuffer, "reset: buffer too new %d > %d", gap,
2073 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2074 /* fill in the gap with EXPECTED timers */
2075 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2077 do_next_seqnum = TRUE;
2080 if (G_UNLIKELY (reset)) {
2081 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2082 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
2083 rtp_jitter_buffer_reset_skew (priv->jbuf);
2084 remove_all_timers (jitterbuffer);
2085 priv->last_popped_seqnum = -1;
2086 priv->next_seqnum = seqnum;
2087 do_next_seqnum = TRUE;
2089 /* reset spacing estimation when gap */
2090 priv->ips_rtptime = -1;
2091 priv->ips_dts = GST_CLOCK_TIME_NONE;
2094 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2095 /* we don't know what the next_in_seqnum should be, wait for the last
2096 * possible moment to push this buffer, maybe we get an earlier seqnum
2098 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2099 do_next_seqnum = TRUE;
2100 /* take rtptime and dts to calculate packet spacing */
2101 priv->ips_rtptime = rtptime;
2102 priv->ips_dts = dts;
2104 if (do_next_seqnum) {
2105 priv->last_in_seqnum = seqnum;
2106 priv->last_in_dts = dts;
2107 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2110 /* let's check if this buffer is too late, we can only accept packets with
2111 * bigger seqnum than the one we last pushed. */
2112 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2115 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2117 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2118 if (G_UNLIKELY (gap <= 0))
2122 /* let's drop oldest packet if the queue is already full and drop-on-latency
2123 * is set. We can only do this when there actually is a latency. When no
2124 * latency is set, we just pump it in the queue and let the other end push it
2125 * out as fast as possible. */
2126 if (priv->latency_ms && priv->drop_on_latency) {
2128 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2130 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2131 RTPJitterBufferItem *old_item;
2133 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2134 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2136 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2137 free_item (old_item);
2141 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2143 /* now insert the packet into the queue in sorted order. This function returns
2144 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2145 * have a duplicate. */
2146 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2151 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2153 /* we had an unhandled SR, handle it now */
2155 do_handle_sync (jitterbuffer);
2157 /* signal addition of new buffer when the _loop is waiting. */
2159 JBUF_SIGNAL_EVENT (priv);
2161 /* let's unschedule and unblock any waiting buffers. We only want to do this
2162 * when the tail buffer changed */
2163 if (G_UNLIKELY (priv->clock_id && tail)) {
2164 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2165 unschedule_current_timer (jitterbuffer);
2168 GST_DEBUG_OBJECT (jitterbuffer, "Pushed packet #%d, now %d packets, tail: %d",
2169 seqnum, rtp_jitter_buffer_num_packets (priv->jbuf), tail);
2171 check_buffering_percent (jitterbuffer, &percent);
2177 post_buffering_percent (jitterbuffer, percent);
2184 /* this is not fatal but should be filtered earlier */
2185 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2186 ("Received invalid RTP payload, dropping"));
2187 gst_buffer_unref (buffer);
2192 GST_WARNING_OBJECT (jitterbuffer,
2193 "No clock-rate in caps!, dropping buffer");
2194 gst_buffer_unref (buffer);
2199 ret = priv->srcresult;
2200 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2201 gst_buffer_unref (buffer);
2207 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2208 gst_buffer_unref (buffer);
2213 GST_WARNING_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2214 " popped, dropping", seqnum, priv->last_popped_seqnum);
2216 gst_buffer_unref (buffer);
2221 GST_WARNING_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2223 priv->num_duplicates++;
2230 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2232 guint64 ext_time, elapsed;
2234 GstRtpJitterBufferPrivate *priv;
2236 priv = jitterbuffer->priv;
2237 rtp_time = item->rtptime;
2239 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2240 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2242 if (rtp_time < priv->ext_timestamp) {
2243 ext_time = priv->ext_timestamp;
2245 ext_time = gst_rtp_buffer_ext_timestamp (&priv->ext_timestamp, rtp_time);
2248 if (ext_time > priv->clock_base)
2249 elapsed = ext_time - priv->clock_base;
2253 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2258 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2259 RTPJitterBufferItem * item)
2261 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2263 if (priv->npt_stop != -1 && priv->ext_timestamp != -1
2264 && priv->clock_base != -1 && priv->clock_rate > 0) {
2265 guint64 elapsed, estimated;
2267 elapsed = compute_elapsed (jitterbuffer, item);
2269 if (elapsed > priv->last_elapsed || !priv->last_elapsed) {
2271 GstClockTime out_time;
2273 priv->last_elapsed = elapsed;
2275 left = priv->npt_stop - priv->npt_start;
2276 GST_LOG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT,
2277 GST_TIME_ARGS (left));
2279 out_time = item->dts;
2282 estimated = gst_util_uint64_scale (out_time, left, elapsed);
2284 /* if there is almost nothing left,
2285 * we may never advance enough to end up in the above case */
2286 if (left < GST_SECOND)
2287 estimated = GST_SECOND;
2292 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2293 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2295 if (estimated != -1 && priv->estimated_eos != estimated) {
2296 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2297 priv->estimated_eos = estimated;
2303 /* take a buffer from the queue and push it */
2304 static GstFlowReturn
2305 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2307 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2308 GstFlowReturn result;
2309 RTPJitterBufferItem *item;
2312 GstClockTime dts, pts;
2314 gboolean is_buffer, do_push = TRUE;
2316 /* when we get here we are ready to pop and push the buffer */
2317 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2319 is_buffer = GST_IS_BUFFER (item->data);
2322 check_buffering_percent (jitterbuffer, &percent);
2324 /* we need to make writable to change the flags and timestamps */
2325 outbuf = gst_buffer_make_writable (item->data);
2327 if (G_UNLIKELY (priv->discont)) {
2328 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2329 * into the jitterbuffer so we can modify now. */
2330 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2331 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2332 priv->discont = FALSE;
2334 if (G_UNLIKELY (priv->ts_discont)) {
2335 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2336 priv->ts_discont = FALSE;
2339 dts = gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->dts);
2340 pts = gst_segment_to_position (&priv->segment, GST_FORMAT_TIME, item->pts);
2342 /* apply timestamp with offset to buffer now */
2343 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2344 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2346 /* update the elapsed time when we need to check against the npt stop time. */
2347 update_estimated_eos (jitterbuffer, item);
2349 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2351 outevent = item->data;
2352 if (item->type == ITEM_TYPE_LOST) {
2353 priv->discont = TRUE;
2359 /* now we are ready to push the buffer. Save the seqnum and release the lock
2360 * so the other end can push stuff in the queue again. */
2362 priv->last_popped_seqnum = seqnum;
2363 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2373 post_buffering_percent (jitterbuffer, percent);
2375 GST_DEBUG_OBJECT (jitterbuffer,
2376 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
2377 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
2378 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
2379 result = gst_pad_push (priv->srcpad, outbuf);
2381 GST_DEBUG_OBJECT (jitterbuffer, "Pushing event %d", seqnum);
2384 gst_pad_push_event (priv->srcpad, outevent);
2386 gst_event_unref (outevent);
2388 result = GST_FLOW_OK;
2390 JBUF_LOCK_CHECK (priv, out_flushing);
2397 return priv->srcresult;
2401 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
2403 /* Peek a buffer and compare the seqnum to the expected seqnum.
2404 * If all is fine, the buffer is pushed.
2405 * If something is wrong, we wait for some event
2407 static GstFlowReturn
2408 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
2410 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2411 GstFlowReturn result = GST_FLOW_OK;
2412 RTPJitterBufferItem *item;
2414 guint32 next_seqnum;
2417 /* only push buffers when PLAYING and active and not buffering */
2418 if (priv->blocked || !priv->active ||
2419 rtp_jitter_buffer_is_buffering (priv->jbuf))
2420 return GST_FLOW_WAIT;
2423 /* peek a buffer, we're just looking at the sequence number.
2424 * If all is fine, we'll pop and push it. If the sequence number is wrong we
2425 * wait for a timeout or something to change.
2426 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
2427 item = rtp_jitter_buffer_peek (priv->jbuf);
2431 /* get the seqnum and the next expected seqnum */
2432 seqnum = item->seqnum;
2436 next_seqnum = priv->next_seqnum;
2438 /* get the gap between this and the previous packet. If we don't know the
2439 * previous packet seqnum assume no gap. */
2440 if (G_UNLIKELY (next_seqnum == -1)) {
2441 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2442 /* we don't know what the next_seqnum should be, the chain function should
2443 * have scheduled a DEADLINE timer that will increment next_seqnum when it
2444 * fires, so wait for that */
2445 result = GST_FLOW_WAIT;
2447 /* else calculate GAP */
2448 gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
2450 if (G_LIKELY (gap == 0)) {
2452 /* no missing packet, pop and push */
2453 result = pop_and_push_next (jitterbuffer, seqnum);
2454 } else if (G_UNLIKELY (gap < 0)) {
2455 RTPJitterBufferItem *item;
2456 /* if we have a packet that we already pushed or considered dropped, pop it
2457 * off and get the next packet */
2458 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
2459 seqnum, next_seqnum);
2460 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
2464 /* the chain function has scheduled timers to request retransmission or
2465 * when to consider the packet lost, wait for that */
2466 GST_DEBUG_OBJECT (jitterbuffer,
2467 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
2468 next_seqnum, seqnum, gap);
2469 result = GST_FLOW_WAIT;
2476 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
2478 result = GST_FLOW_EOS;
2480 result = GST_FLOW_WAIT;
2485 /* the timeout for when we expected a packet expired */
2487 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2490 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2494 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive", timer->seqnum);
2496 delay = timer->rtx_delay + timer->rtx_retry;
2497 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2498 gst_structure_new ("GstRTPRetransmissionRequest",
2499 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
2500 "running-time", G_TYPE_UINT64, timer->rtx_base,
2501 "delay", G_TYPE_UINT, GST_TIME_AS_MSECONDS (delay),
2502 "retry", G_TYPE_UINT, timer->num_rtx_retry,
2503 "frequency", G_TYPE_UINT, priv->rtx_retry_timeout,
2504 "period", G_TYPE_UINT, priv->rtx_retry_period,
2505 "deadline", G_TYPE_UINT, priv->latency_ms,
2506 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing, NULL));
2508 priv->num_rtx_requests++;
2509 timer->num_rtx_retry++;
2510 timer->rtx_last = now;
2512 /* calculate the timeout for the next retransmission attempt */
2513 timer->rtx_retry += (priv->rtx_retry_timeout * GST_MSECOND);
2514 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
2515 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT,
2516 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
2517 GST_TIME_ARGS (timer->rtx_retry));
2519 if (timer->rtx_retry + timer->rtx_delay >
2520 (priv->rtx_retry_period * GST_MSECOND)) {
2521 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
2522 /* too many retransmission request, we now convert the timer
2523 * to a lost timer, leave the num_rtx_retry as it is for stats */
2524 timer->type = TIMER_TYPE_LOST;
2525 timer->rtx_delay = 0;
2526 timer->rtx_retry = 0;
2528 reschedule_timer (jitterbuffer, timer, timer->seqnum,
2529 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
2532 gst_pad_push_event (priv->sinkpad, event);
2538 /* a packet is lost */
2540 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2543 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2544 GstClockTime duration, timestamp;
2545 guint seqnum, lost_packets, num_rtx_retry;
2548 RTPJitterBufferItem *item;
2550 seqnum = timer->seqnum;
2551 timestamp = apply_offset (jitterbuffer, timer->timeout);
2552 duration = timer->duration;
2553 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
2554 duration = priv->packet_spacing;
2555 lost_packets = MAX (timer->num, 1);
2556 late = timer->num > 0;
2557 num_rtx_retry = timer->num_rtx_retry;
2559 /* we had a gap and thus we lost some packets. Create an event for this. */
2560 if (lost_packets > 1)
2561 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
2562 seqnum + lost_packets - 1);
2564 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
2566 priv->num_late += lost_packets;
2567 priv->num_rtx_failed += num_rtx_retry;
2569 /* create paket lost event */
2570 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
2571 gst_structure_new ("GstRTPPacketLost",
2572 "seqnum", G_TYPE_UINT, (guint) seqnum,
2573 "timestamp", G_TYPE_UINT64, timestamp,
2574 "duration", G_TYPE_UINT64, duration,
2575 "late", G_TYPE_BOOLEAN, late,
2576 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
2578 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
2579 rtp_jitter_buffer_insert (priv->jbuf, item, NULL, NULL);
2581 /* remove timer now */
2582 remove_timer (jitterbuffer, timer);
2583 JBUF_SIGNAL_EVENT (priv);
2589 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2592 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2594 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
2595 remove_timer (jitterbuffer, timer);
2596 JBUF_SIGNAL_EVENT (priv);
2602 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2605 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2607 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
2609 priv->next_seqnum = timer->seqnum;
2610 remove_timer (jitterbuffer, timer);
2611 JBUF_SIGNAL_EVENT (priv);
2617 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
2620 gboolean removed = FALSE;
2622 switch (timer->type) {
2623 case TIMER_TYPE_EXPECTED:
2624 removed = do_expected_timeout (jitterbuffer, timer, now);
2626 case TIMER_TYPE_LOST:
2627 removed = do_lost_timeout (jitterbuffer, timer, now);
2629 case TIMER_TYPE_DEADLINE:
2630 removed = do_deadline_timeout (jitterbuffer, timer, now);
2632 case TIMER_TYPE_EOS:
2633 removed = do_eos_timeout (jitterbuffer, timer, now);
2639 /* called when we need to wait for the next timeout.
2641 * We loop over the array of recorded timeouts and wait for the earliest one.
2642 * When it timed out, do the logic associated with the timer.
2644 * If there are no timers, we wait on a gcond until something new happens.
2647 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
2649 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2650 GstClockTime now = 0;
2653 while (priv->timer_running) {
2654 TimerData *timer = NULL;
2655 GstClockTime timer_timeout = -1;
2658 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
2659 GST_TIME_ARGS (now));
2661 len = priv->timers->len;
2662 for (i = 0; i < len; i++) {
2663 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2664 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
2665 gboolean save_best = FALSE;
2667 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
2668 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
2670 /* find the smallest timeout */
2671 if (timer == NULL) {
2673 } else if (timer_timeout == -1) {
2674 /* we already have an immediate timeout, the new timer must be an
2675 * immediate timer with smaller seqnum to become the best */
2676 if (test_timeout == -1 && test->seqnum < timer->seqnum)
2678 } else if (test_timeout == -1) {
2679 /* first immediate timer */
2681 } else if (test_timeout < timer_timeout) {
2684 } else if (test_timeout == timer_timeout && test->seqnum < timer->seqnum) {
2685 /* same timer, smaller seqnum */
2689 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
2691 timer_timeout = test_timeout;
2694 if (timer && !priv->blocked) {
2696 GstClockTime sync_time;
2699 GstClockTimeDiff clock_jitter;
2701 if (timer_timeout == -1 || timer_timeout <= now) {
2702 do_timeout (jitterbuffer, timer, now);
2703 /* check here, do_timeout could have released the lock */
2704 if (!priv->timer_running)
2709 GST_OBJECT_LOCK (jitterbuffer);
2710 clock = GST_ELEMENT_CLOCK (jitterbuffer);
2712 GST_OBJECT_UNLOCK (jitterbuffer);
2713 /* let's just push if there is no clock */
2714 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
2715 now = timer_timeout;
2719 /* prepare for sync against clock */
2720 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
2721 /* add latency of peer to get input time */
2722 sync_time += priv->peer_latency;
2724 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
2725 " with sync time %" GST_TIME_FORMAT,
2726 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
2728 /* create an entry for the clock */
2729 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
2730 priv->timer_timeout = timer_timeout;
2731 priv->timer_seqnum = timer->seqnum;
2732 GST_OBJECT_UNLOCK (jitterbuffer);
2734 /* release the lock so that the other end can push stuff or unlock */
2737 ret = gst_clock_id_wait (id, &clock_jitter);
2740 if (!priv->timer_running)
2743 if (ret != GST_CLOCK_UNSCHEDULED) {
2744 now = timer_timeout + MAX (clock_jitter, 0);
2745 GST_DEBUG_OBJECT (jitterbuffer, "sync done, %d, #%d, %" G_GINT64_FORMAT,
2746 ret, priv->timer_seqnum, clock_jitter);
2748 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
2750 /* and free the entry */
2751 gst_clock_id_unref (id);
2752 priv->clock_id = NULL;
2754 /* no timers, wait for activity */
2755 JBUF_WAIT_TIMER (priv);
2760 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
2765 * This funcion implements the main pushing loop on the source pad.
2767 * It first tries to push as many buffers as possible. If there is a seqnum
2768 * mismatch, we wait for the next timeouts.
2771 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
2773 GstRtpJitterBufferPrivate *priv;
2774 GstFlowReturn result;
2776 priv = jitterbuffer->priv;
2778 JBUF_LOCK_CHECK (priv, flushing);
2780 result = handle_next_buffer (jitterbuffer);
2781 if (G_LIKELY (result == GST_FLOW_WAIT)) {
2782 /* now wait for the next event */
2783 JBUF_WAIT_EVENT (priv, flushing);
2784 result = GST_FLOW_OK;
2787 while (result == GST_FLOW_OK);
2788 /* store result for upstream */
2789 priv->srcresult = result;
2792 /* if we get here we need to pause */
2798 result = priv->srcresult;
2804 const gchar *reason = gst_flow_get_name (result);
2807 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s", reason);
2808 gst_pad_pause_task (priv->srcpad);
2809 if (result == GST_FLOW_EOS) {
2810 event = gst_event_new_eos ();
2811 gst_pad_push_event (priv->srcpad, event);
2817 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
2818 * some sanity checks and then emit the handle-sync signal with the parameters.
2819 * This function must be called with the LOCK */
2821 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
2823 GstRtpJitterBufferPrivate *priv;
2824 guint64 base_rtptime, base_time;
2826 guint64 last_rtptime;
2828 guint64 ext_rtptime, diff;
2829 gboolean drop = FALSE;
2831 priv = jitterbuffer->priv;
2833 /* get the last values from the jitterbuffer */
2834 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
2835 &clock_rate, &last_rtptime);
2837 clock_base = priv->clock_base;
2838 ext_rtptime = priv->ext_rtptime;
2840 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
2841 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
2842 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
2843 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
2845 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
2846 GST_DEBUG_OBJECT (jitterbuffer, "dropping, no RTP values");
2849 /* we can't accept anything that happened before we did the last resync */
2850 if (base_rtptime > ext_rtptime) {
2851 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
2854 /* the SR RTP timestamp must be something close to what we last observed
2855 * in the jitterbuffer */
2856 if (ext_rtptime > last_rtptime) {
2857 /* check how far ahead it is to our RTP timestamps */
2858 diff = ext_rtptime - last_rtptime;
2859 /* if bigger than 1 second, we drop it */
2860 if (diff > clock_rate) {
2861 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
2862 /* should drop this, but some RTSP servers end up with bogus
2863 * way too ahead RTCP packet when repeated PAUSE/PLAY,
2864 * so still trigger rptbin sync but invalidate RTCP data
2865 * (sync might use other methods) */
2868 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
2869 G_GUINT64_FORMAT, last_rtptime, diff);
2877 s = gst_structure_new ("application/x-rtp-sync",
2878 "base-rtptime", G_TYPE_UINT64, base_rtptime,
2879 "base-time", G_TYPE_UINT64, base_time,
2880 "clock-rate", G_TYPE_UINT, clock_rate,
2881 "clock-base", G_TYPE_UINT64, clock_base,
2882 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
2883 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
2885 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
2886 gst_buffer_replace (&priv->last_sr, NULL);
2888 g_signal_emit (jitterbuffer,
2889 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
2891 gst_structure_free (s);
2893 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
2897 static GstFlowReturn
2898 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
2901 GstRtpJitterBuffer *jitterbuffer;
2902 GstRtpJitterBufferPrivate *priv;
2903 GstFlowReturn ret = GST_FLOW_OK;
2905 GstRTCPPacket packet;
2906 guint64 ext_rtptime;
2908 GstRTCPBuffer rtcp = { NULL, };
2910 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
2912 if (G_UNLIKELY (!gst_rtcp_buffer_validate (buffer)))
2913 goto invalid_buffer;
2915 priv = jitterbuffer->priv;
2917 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2919 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
2922 /* first packet must be SR or RR or else the validate would have failed */
2923 switch (gst_rtcp_packet_get_type (&packet)) {
2924 case GST_RTCP_TYPE_SR:
2925 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
2931 gst_rtcp_buffer_unmap (&rtcp);
2933 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
2936 /* convert the RTP timestamp to our extended timestamp, using the same offset
2937 * we used in the jitterbuffer */
2938 ext_rtptime = priv->jbuf->ext_rtptime;
2939 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
2941 priv->ext_rtptime = ext_rtptime;
2942 gst_buffer_replace (&priv->last_sr, buffer);
2944 do_handle_sync (jitterbuffer);
2948 gst_buffer_unref (buffer);
2954 /* this is not fatal but should be filtered earlier */
2955 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2956 ("Received invalid RTCP payload, dropping"));
2962 /* this is not fatal but should be filtered earlier */
2963 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2964 ("Received empty RTCP payload, dropping"));
2965 gst_rtcp_buffer_unmap (&rtcp);
2971 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
2972 gst_rtcp_buffer_unmap (&rtcp);
2979 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
2982 gboolean res = FALSE;
2984 switch (GST_QUERY_TYPE (query)) {
2985 case GST_QUERY_CAPS:
2987 GstCaps *filter, *caps;
2989 gst_query_parse_caps (query, &filter);
2990 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
2991 gst_query_set_caps_result (query, caps);
2992 gst_caps_unref (caps);
2997 if (GST_QUERY_IS_SERIALIZED (query)) {
2998 GST_WARNING_OBJECT (pad, "unhandled serialized query");
3001 res = gst_pad_query_default (pad, parent, query);
3009 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3012 GstRtpJitterBuffer *jitterbuffer;
3013 GstRtpJitterBufferPrivate *priv;
3014 gboolean res = FALSE;
3016 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3017 priv = jitterbuffer->priv;
3019 switch (GST_QUERY_TYPE (query)) {
3020 case GST_QUERY_LATENCY:
3022 /* We need to send the query upstream and add the returned latency to our
3024 GstClockTime min_latency, max_latency;
3026 GstClockTime our_latency;
3028 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3029 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3031 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3032 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3033 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3035 /* store this so that we can safely sync on the peer buffers. */
3037 priv->peer_latency = min_latency;
3038 our_latency = priv->latency_ns;
3041 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3042 GST_TIME_ARGS (our_latency));
3044 /* we add some latency but can buffer an infinite amount of time */
3045 min_latency += our_latency;
3048 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3049 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3050 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3052 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3056 case GST_QUERY_POSITION:
3058 GstClockTime start, last_out;
3061 gst_query_parse_position (query, &fmt, NULL);
3062 if (fmt != GST_FORMAT_TIME) {
3063 res = gst_pad_query_default (pad, parent, query);
3068 start = priv->npt_start;
3069 last_out = priv->last_out_time;
3072 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3073 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3074 GST_TIME_ARGS (last_out));
3076 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3077 /* bring 0-based outgoing time to stream time */
3078 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3081 res = gst_pad_query_default (pad, parent, query);
3085 case GST_QUERY_CAPS:
3087 GstCaps *filter, *caps;
3089 gst_query_parse_caps (query, &filter);
3090 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3091 gst_query_set_caps_result (query, caps);
3092 gst_caps_unref (caps);
3097 res = gst_pad_query_default (pad, parent, query);
3105 gst_rtp_jitter_buffer_set_property (GObject * object,
3106 guint prop_id, const GValue * value, GParamSpec * pspec)
3108 GstRtpJitterBuffer *jitterbuffer;
3109 GstRtpJitterBufferPrivate *priv;
3111 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3112 priv = jitterbuffer->priv;
3117 guint new_latency, old_latency;
3119 new_latency = g_value_get_uint (value);
3122 old_latency = priv->latency_ms;
3123 priv->latency_ms = new_latency;
3124 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3125 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3128 /* post message if latency changed, this will inform the parent pipeline
3129 * that a latency reconfiguration is possible/needed. */
3130 if (new_latency != old_latency) {
3131 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3132 GST_TIME_ARGS (new_latency * GST_MSECOND));
3134 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3135 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3139 case PROP_DROP_ON_LATENCY:
3141 priv->drop_on_latency = g_value_get_boolean (value);
3144 case PROP_TS_OFFSET:
3146 priv->ts_offset = g_value_get_int64 (value);
3147 priv->ts_discont = TRUE;
3152 priv->do_lost = g_value_get_boolean (value);
3157 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3160 case PROP_DO_RETRANSMISSION:
3162 priv->do_retransmission = g_value_get_boolean (value);
3165 case PROP_RTX_DELAY:
3167 priv->rtx_delay = g_value_get_int (value);
3170 case PROP_RTX_DELAY_REORDER:
3172 priv->rtx_delay_reorder = g_value_get_int (value);
3175 case PROP_RTX_RETRY_TIMEOUT:
3177 priv->rtx_retry_timeout = g_value_get_int (value);
3180 case PROP_RTX_RETRY_PERIOD:
3182 priv->rtx_retry_period = g_value_get_int (value);
3186 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3192 gst_rtp_jitter_buffer_get_property (GObject * object,
3193 guint prop_id, GValue * value, GParamSpec * pspec)
3195 GstRtpJitterBuffer *jitterbuffer;
3196 GstRtpJitterBufferPrivate *priv;
3198 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3199 priv = jitterbuffer->priv;
3204 g_value_set_uint (value, priv->latency_ms);
3207 case PROP_DROP_ON_LATENCY:
3209 g_value_set_boolean (value, priv->drop_on_latency);
3212 case PROP_TS_OFFSET:
3214 g_value_set_int64 (value, priv->ts_offset);
3219 g_value_set_boolean (value, priv->do_lost);
3224 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
3232 if (priv->srcresult != GST_FLOW_OK)
3235 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
3237 g_value_set_int (value, percent);
3241 case PROP_DO_RETRANSMISSION:
3243 g_value_set_boolean (value, priv->do_retransmission);
3246 case PROP_RTX_DELAY:
3248 g_value_set_int (value, priv->rtx_delay);
3251 case PROP_RTX_DELAY_REORDER:
3253 g_value_set_int (value, priv->rtx_delay_reorder);
3256 case PROP_RTX_RETRY_TIMEOUT:
3258 g_value_set_int (value, priv->rtx_retry_timeout);
3261 case PROP_RTX_RETRY_PERIOD:
3263 g_value_set_int (value, priv->rtx_retry_period);
3267 g_value_take_boxed (value,
3268 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
3271 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3276 static GstStructure *
3277 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
3281 JBUF_LOCK (jbuf->priv);
3282 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
3283 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
3284 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
3285 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
3286 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
3287 JBUF_UNLOCK (jbuf->priv);