2 * Farsight Voice+Video library
4 * Copyright 2007 Collabora Ltd,
5 * Copyright 2007 Nokia Corporation
6 * @author: Philippe Kalaf <philippe.kalaf@collabora.co.uk>.
7 * Copyright 2007 Wim Taymans <wim.taymans@gmail.com>
8 * Copyright 2015 Kurento (http://kurento.org/)
9 * @author: Miguel ParĂs <mparisdiaz@gmail.com>
11 * This library is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Library General Public
13 * License as published by the Free Software Foundation; either
14 * version 2 of the License, or (at your option) any later version.
16 * This library is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Library General Public License for more details.
21 * You should have received a copy of the GNU Library General Public
22 * License along with this library; if not, write to the
23 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
24 * Boston, MA 02110-1301, USA.
29 * SECTION:element-rtpjitterbuffer
31 * This element reorders and removes duplicate RTP packets as they are received
32 * from a network source.
34 * The element needs the clock-rate of the RTP payload in order to estimate the
35 * delay. This information is obtained either from the caps on the sink pad or,
36 * when no caps are present, from the #GstRtpJitterBuffer::request-pt-map signal.
37 * To clear the previous pt-map use the #GstRtpJitterBuffer::clear-pt-map signal.
39 * The rtpjitterbuffer will wait for missing packets up to a configurable time
40 * limit using the #GstRtpJitterBuffer:latency property. Packets arriving too
41 * late are considered to be lost packets. If the #GstRtpJitterBuffer:do-lost
42 * property is set, lost packets will result in a custom serialized downstream
43 * event of name GstRTPPacketLost. The lost packet events are usually used by a
44 * depayloader or other element to create concealment data or some other logic
45 * to gracefully handle the missing packets.
47 * The jitterbuffer will use the DTS (or PTS if no DTS is set) of the incomming
48 * buffer and the rtptime inside the RTP packet to create a PTS on the outgoing
51 * The jitterbuffer can also be configured to send early retransmission events
52 * upstream by setting the #GstRtpJitterBuffer:do-retransmission property. In
53 * this mode, the jitterbuffer tries to estimate when a packet should arrive and
54 * sends a custom upstream event named GstRTPRetransmissionRequest when the
55 * packet is considered late. The initial expected packet arrival time is
56 * calculated as follows:
58 * - If seqnum N arrived at time T, seqnum N+1 is expected to arrive at
59 * T + packet-spacing + #GstRtpJitterBuffer:rtx-delay. The packet spacing is
60 * calculated from the DTS (or PTS is no DTS) of two consecutive RTP
61 * packets with different rtptime.
63 * - If seqnum N0 arrived at time T0 and seqnum Nm arrived at time Tm,
64 * seqnum Ni is expected at time Ti = T0 + i*(Tm - T0)/(Nm - N0). Any
65 * previously scheduled timeout is overwritten.
67 * - If seqnum N arrived, all seqnum older than
68 * N - #GstRtpJitterBuffer:rtx-delay-reorder are considered late
69 * immediately. This is to request fast feedback for abonormally reorder
70 * packets before any of the previous timeouts is triggered.
72 * A late packet triggers the GstRTPRetransmissionRequest custom upstream
73 * event. After the initial timeout expires and the retransmission event is
74 * sent, the timeout is scheduled for
75 * T + #GstRtpJitterBuffer:rtx-retry-timeout. If the missing packet did not
76 * arrive after #GstRtpJitterBuffer:rtx-retry-timeout, a new
77 * GstRTPRetransmissionRequest is sent upstream and the timeout is rescheduled
78 * again for T + #GstRtpJitterBuffer:rtx-retry-timeout. This repeats until
79 * #GstRtpJitterBuffer:rtx-retry-period elapsed, at which point no further
80 * retransmission requests are sent and the regular logic is performed to
81 * schedule a lost packet as discussed above.
83 * This element acts as a live element and so adds #GstRtpJitterBuffer:latency
86 * This element will automatically be used inside rtpbin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 rtspsrc location=rtsp://192.168.1.133:8554/mpeg1or2AudioVideoTest ! rtpjitterbuffer ! rtpmpvdepay ! mpeg2dec ! xvimagesink
92 * ]| Connect to a streaming server and decode the MPEG video. The jitterbuffer is
93 * inserted into the pipeline to smooth out network jitter and to reorder the
94 * out-of-order RTP packets.
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_NEXT_SEQNUM TRUE
134 #define DEFAULT_RTX_DELAY -1
135 #define DEFAULT_RTX_MIN_DELAY 0
136 #define DEFAULT_RTX_DELAY_REORDER 3
137 #define DEFAULT_RTX_RETRY_TIMEOUT -1
138 #define DEFAULT_RTX_MIN_RETRY_TIMEOUT -1
139 #define DEFAULT_RTX_RETRY_PERIOD -1
140 #define DEFAULT_RTX_MAX_RETRIES -1
141 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
142 #define DEFAULT_MAX_DROPOUT_TIME 60000
143 #define DEFAULT_MAX_MISORDER_TIME 2000
145 #define DEFAULT_AUTO_RTX_DELAY (20 * GST_MSECOND)
146 #define DEFAULT_AUTO_RTX_TIMEOUT (40 * GST_MSECOND)
152 PROP_DROP_ON_LATENCY,
157 PROP_DO_RETRANSMISSION,
158 PROP_RTX_NEXT_SEQNUM,
161 PROP_RTX_DELAY_REORDER,
162 PROP_RTX_RETRY_TIMEOUT,
163 PROP_RTX_MIN_RETRY_TIMEOUT,
164 PROP_RTX_RETRY_PERIOD,
165 PROP_RTX_MAX_RETRIES,
167 PROP_MAX_RTCP_RTP_TIME_DIFF,
168 PROP_MAX_DROPOUT_TIME,
169 PROP_MAX_MISORDER_TIME
172 #define JBUF_LOCK(priv) (g_mutex_lock (&(priv)->jbuf_lock))
174 #define JBUF_LOCK_CHECK(priv,label) G_STMT_START { \
176 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
179 #define JBUF_UNLOCK(priv) (g_mutex_unlock (&(priv)->jbuf_lock))
181 #define JBUF_WAIT_TIMER(priv) G_STMT_START { \
182 GST_DEBUG ("waiting timer"); \
183 (priv)->waiting_timer = TRUE; \
184 g_cond_wait (&(priv)->jbuf_timer, &(priv)->jbuf_lock); \
185 (priv)->waiting_timer = FALSE; \
186 GST_DEBUG ("waiting timer done"); \
188 #define JBUF_SIGNAL_TIMER(priv) G_STMT_START { \
189 if (G_UNLIKELY ((priv)->waiting_timer)) { \
190 GST_DEBUG ("signal timer"); \
191 g_cond_signal (&(priv)->jbuf_timer); \
195 #define JBUF_WAIT_EVENT(priv,label) G_STMT_START { \
196 GST_DEBUG ("waiting event"); \
197 (priv)->waiting_event = TRUE; \
198 g_cond_wait (&(priv)->jbuf_event, &(priv)->jbuf_lock); \
199 (priv)->waiting_event = FALSE; \
200 GST_DEBUG ("waiting event done"); \
201 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
204 #define JBUF_SIGNAL_EVENT(priv) G_STMT_START { \
205 if (G_UNLIKELY ((priv)->waiting_event)) { \
206 GST_DEBUG ("signal event"); \
207 g_cond_signal (&(priv)->jbuf_event); \
211 #define JBUF_WAIT_QUERY(priv,label) G_STMT_START { \
212 GST_DEBUG ("waiting query"); \
213 (priv)->waiting_query = TRUE; \
214 g_cond_wait (&(priv)->jbuf_query, &(priv)->jbuf_lock); \
215 (priv)->waiting_query = FALSE; \
216 GST_DEBUG ("waiting query done"); \
217 if (G_UNLIKELY (priv->srcresult != GST_FLOW_OK)) \
220 #define JBUF_SIGNAL_QUERY(priv,res) G_STMT_START { \
221 (priv)->last_query = res; \
222 if (G_UNLIKELY ((priv)->waiting_query)) { \
223 GST_DEBUG ("signal query"); \
224 g_cond_signal (&(priv)->jbuf_query); \
229 struct _GstRtpJitterBufferPrivate
231 GstPad *sinkpad, *srcpad;
234 RTPJitterBuffer *jbuf;
236 gboolean waiting_timer;
238 gboolean waiting_event;
240 gboolean waiting_query;
248 gboolean timer_running;
249 GThread *timer_thread;
254 gboolean drop_on_latency;
257 gboolean do_retransmission;
258 gboolean rtx_next_seqnum;
261 gint rtx_delay_reorder;
262 gint rtx_retry_timeout;
263 gint rtx_min_retry_timeout;
264 gint rtx_retry_period;
265 gint rtx_max_retries;
266 gint max_rtcp_rtp_time_diff;
267 guint32 max_dropout_time;
268 guint32 max_misorder_time;
270 /* the last seqnum we pushed out */
271 guint32 last_popped_seqnum;
272 /* the next expected seqnum we push */
274 /* seqnum-base, if known */
276 /* last output time */
277 GstClockTime last_out_time;
278 /* last valid input timestamp and rtptime pair */
279 GstClockTime ips_dts;
281 GstClockTime packet_spacing;
285 /* the next expected seqnum we receive */
286 GstClockTime last_in_dts;
287 guint32 next_in_seqnum;
291 /* start and stop ranges */
292 GstClockTime npt_start;
293 GstClockTime npt_stop;
294 guint64 ext_timestamp;
295 guint64 last_elapsed;
296 guint64 estimated_eos;
303 /* clock rate and rtp timestamp offset */
307 gint64 prev_ts_offset;
309 /* when we are shutting down */
310 GstFlowReturn srcresult;
316 GstClockTime timer_timeout;
317 guint16 timer_seqnum;
318 /* the latency of the upstream peer, we have to take this into account when
319 * synchronizing the buffers. */
320 GstClockTime peer_latency;
324 /* some accounting */
326 guint64 num_duplicates;
327 guint64 num_rtx_requests;
328 guint64 num_rtx_success;
329 guint64 num_rtx_failed;
332 RTPPacketRateCtx packet_rate_ctx;
335 GstClockTime last_dts;
336 guint64 last_rtptime;
337 GstClockTime avg_jitter;
354 GstClockTime timeout;
355 GstClockTime duration;
356 GstClockTime rtx_base;
357 GstClockTime rtx_delay;
358 GstClockTime rtx_retry;
359 GstClockTime rtx_last;
363 #define GST_RTP_JITTER_BUFFER_GET_PRIVATE(o) \
364 (G_TYPE_INSTANCE_GET_PRIVATE ((o), GST_TYPE_RTP_JITTER_BUFFER, \
365 GstRtpJitterBufferPrivate))
367 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_template =
368 GST_STATIC_PAD_TEMPLATE ("sink",
371 GST_STATIC_CAPS ("application/x-rtp"
372 /* "clock-rate = (int) [ 1, 2147483647 ], "
373 * "payload = (int) , "
374 * "encoding-name = (string) "
378 static GstStaticPadTemplate gst_rtp_jitter_buffer_sink_rtcp_template =
379 GST_STATIC_PAD_TEMPLATE ("sink_rtcp",
382 GST_STATIC_CAPS ("application/x-rtcp")
385 static GstStaticPadTemplate gst_rtp_jitter_buffer_src_template =
386 GST_STATIC_PAD_TEMPLATE ("src",
389 GST_STATIC_CAPS ("application/x-rtp"
390 /* "payload = (int) , "
391 * "clock-rate = (int) , "
392 * "encoding-name = (string) "
396 static guint gst_rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 };
398 #define gst_rtp_jitter_buffer_parent_class parent_class
399 G_DEFINE_TYPE (GstRtpJitterBuffer, gst_rtp_jitter_buffer, GST_TYPE_ELEMENT);
401 /* object overrides */
402 static void gst_rtp_jitter_buffer_set_property (GObject * object,
403 guint prop_id, const GValue * value, GParamSpec * pspec);
404 static void gst_rtp_jitter_buffer_get_property (GObject * object,
405 guint prop_id, GValue * value, GParamSpec * pspec);
406 static void gst_rtp_jitter_buffer_finalize (GObject * object);
408 /* element overrides */
409 static GstStateChangeReturn gst_rtp_jitter_buffer_change_state (GstElement
410 * element, GstStateChange transition);
411 static GstPad *gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
412 GstPadTemplate * templ, const gchar * name, const GstCaps * filter);
413 static void gst_rtp_jitter_buffer_release_pad (GstElement * element,
415 static GstClock *gst_rtp_jitter_buffer_provide_clock (GstElement * element);
418 static GstCaps *gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter);
419 static GstIterator *gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad,
422 /* sinkpad overrides */
423 static gboolean gst_rtp_jitter_buffer_sink_event (GstPad * pad,
424 GstObject * parent, GstEvent * event);
425 static GstFlowReturn gst_rtp_jitter_buffer_chain (GstPad * pad,
426 GstObject * parent, GstBuffer * buffer);
428 static gboolean gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad,
429 GstObject * parent, GstEvent * event);
430 static GstFlowReturn gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad,
431 GstObject * parent, GstBuffer * buffer);
433 static gboolean gst_rtp_jitter_buffer_sink_query (GstPad * pad,
434 GstObject * parent, GstQuery * query);
436 /* srcpad overrides */
437 static gboolean gst_rtp_jitter_buffer_src_event (GstPad * pad,
438 GstObject * parent, GstEvent * event);
439 static gboolean gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad,
440 GstObject * parent, GstPadMode mode, gboolean active);
441 static void gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer);
442 static gboolean gst_rtp_jitter_buffer_src_query (GstPad * pad,
443 GstObject * parent, GstQuery * query);
446 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer);
448 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jitterbuffer,
449 gboolean active, guint64 base_time);
450 static void do_handle_sync (GstRtpJitterBuffer * jitterbuffer);
452 static void unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer);
453 static void remove_all_timers (GstRtpJitterBuffer * jitterbuffer);
455 static void wait_next_timeout (GstRtpJitterBuffer * jitterbuffer);
457 static GstStructure *gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer *
461 gst_rtp_jitter_buffer_class_init (GstRtpJitterBufferClass * klass)
463 GObjectClass *gobject_class;
464 GstElementClass *gstelement_class;
466 gobject_class = (GObjectClass *) klass;
467 gstelement_class = (GstElementClass *) klass;
469 g_type_class_add_private (klass, sizeof (GstRtpJitterBufferPrivate));
471 gobject_class->finalize = gst_rtp_jitter_buffer_finalize;
473 gobject_class->set_property = gst_rtp_jitter_buffer_set_property;
474 gobject_class->get_property = gst_rtp_jitter_buffer_get_property;
477 * GstRtpJitterBuffer:latency:
479 * The maximum latency of the jitterbuffer. Packets will be kept in the buffer
480 * for at most this time.
482 g_object_class_install_property (gobject_class, PROP_LATENCY,
483 g_param_spec_uint ("latency", "Buffer latency in ms",
484 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
485 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
487 * GstRtpJitterBuffer:drop-on-latency:
489 * Drop oldest buffers when the queue is completely filled.
491 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
492 g_param_spec_boolean ("drop-on-latency",
493 "Drop buffers when maximum latency is reached",
494 "Tells the jitterbuffer to never exceed the given latency in size",
495 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
497 * GstRtpJitterBuffer:ts-offset:
499 * Adjust GStreamer output buffer timestamps in the jitterbuffer with offset.
500 * This is mainly used to ensure interstream synchronisation.
502 g_object_class_install_property (gobject_class, PROP_TS_OFFSET,
503 g_param_spec_int64 ("ts-offset", "Timestamp Offset",
504 "Adjust buffer timestamps with offset in nanoseconds", G_MININT64,
505 G_MAXINT64, DEFAULT_TS_OFFSET,
506 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
509 * GstRtpJitterBuffer:do-lost:
511 * Send out a GstRTPPacketLost event downstream when a packet is considered
514 g_object_class_install_property (gobject_class, PROP_DO_LOST,
515 g_param_spec_boolean ("do-lost", "Do Lost",
516 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
517 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
520 * GstRtpJitterBuffer:mode:
522 * Control the buffering and timestamping mode used by the jitterbuffer.
524 g_object_class_install_property (gobject_class, PROP_MODE,
525 g_param_spec_enum ("mode", "Mode",
526 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
527 DEFAULT_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
529 * GstRtpJitterBuffer:percent:
531 * The percent of the jitterbuffer that is filled.
533 g_object_class_install_property (gobject_class, PROP_PERCENT,
534 g_param_spec_int ("percent", "percent",
535 "The buffer filled percent", 0, 100,
536 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
538 * GstRtpJitterBuffer:do-retransmission:
540 * Send out a GstRTPRetransmission event upstream when a packet is considered
541 * late and should be retransmitted.
545 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
546 g_param_spec_boolean ("do-retransmission", "Do Retransmission",
547 "Send retransmission events upstream when a packet is late",
548 DEFAULT_DO_RETRANSMISSION,
549 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
552 * GstRtpJitterBuffer:rtx-next-seqnum
554 * Estimate when the next packet should arrive and schedule a retransmission
556 * This is, when packet N arrives, a GstRTPRetransmission event is schedule
557 * for packet N+1. So it will be requested if it does not arrive at the expected time.
558 * The expected time is calculated using the dts of N and the packet spacing.
562 g_object_class_install_property (gobject_class, PROP_RTX_NEXT_SEQNUM,
563 g_param_spec_boolean ("rtx-next-seqnum", "RTX next seqnum",
564 "Estimate when the next packet should arrive and schedule a "
565 "retransmission request for it.",
566 DEFAULT_RTX_NEXT_SEQNUM, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
569 * GstRtpJitterBuffer:rtx-delay:
571 * When a packet did not arrive at the expected time, wait this extra amount
572 * of time before sending a retransmission event.
574 * When -1 is used, the max jitter will be used as extra delay.
578 g_object_class_install_property (gobject_class, PROP_RTX_DELAY,
579 g_param_spec_int ("rtx-delay", "RTX Delay",
580 "Extra time in ms to wait before sending retransmission "
581 "event (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_DELAY,
582 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
585 * GstRtpJitterBuffer:rtx-min-delay:
587 * When a packet did not arrive at the expected time, wait at least this extra amount
588 * of time before sending a retransmission event.
592 g_object_class_install_property (gobject_class, PROP_RTX_MIN_DELAY,
593 g_param_spec_uint ("rtx-min-delay", "Minimum RTX Delay",
594 "Minimum time in ms to wait before sending retransmission "
595 "event", 0, G_MAXUINT, DEFAULT_RTX_MIN_DELAY,
596 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
598 * GstRtpJitterBuffer:rtx-delay-reorder:
600 * Assume that a retransmission event should be sent when we see
601 * this much packet reordering.
603 * When -1 is used, the value will be estimated based on observed packet
608 g_object_class_install_property (gobject_class, PROP_RTX_DELAY_REORDER,
609 g_param_spec_int ("rtx-delay-reorder", "RTX Delay Reorder",
610 "Sending retransmission event when this much reordering (-1 automatic)",
611 -1, G_MAXINT, DEFAULT_RTX_DELAY_REORDER,
612 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
614 * GstRtpJitterBuffer::rtx-retry-timeout:
616 * When no packet has been received after sending a retransmission event
617 * for this time, retry sending a retransmission event.
619 * When -1 is used, the value will be estimated based on observed round
624 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_TIMEOUT,
625 g_param_spec_int ("rtx-retry-timeout", "RTX Retry Timeout",
626 "Retry sending a transmission event after this timeout in "
627 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_TIMEOUT,
628 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
630 * GstRtpJitterBuffer::rtx-min-retry-timeout:
632 * The minimum amount of time between retry timeouts. When
633 * GstRtpJitterBuffer::rtx-retry-timeout is -1, this value ensures a
634 * minimum interval between retry timeouts.
636 * When -1 is used, the value will be estimated based on the
641 g_object_class_install_property (gobject_class, PROP_RTX_MIN_RETRY_TIMEOUT,
642 g_param_spec_int ("rtx-min-retry-timeout", "RTX Min Retry Timeout",
643 "Minimum timeout between sending a transmission event in "
644 "ms (-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_MIN_RETRY_TIMEOUT,
645 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
647 * GstRtpJitterBuffer:rtx-retry-period:
649 * The amount of time to try to get a retransmission.
651 * When -1 is used, the value will be estimated based on the jitterbuffer
652 * latency and the observed round trip time.
656 g_object_class_install_property (gobject_class, PROP_RTX_RETRY_PERIOD,
657 g_param_spec_int ("rtx-retry-period", "RTX Retry Period",
658 "Try to get a retransmission for this many ms "
659 "(-1 automatic)", -1, G_MAXINT, DEFAULT_RTX_RETRY_PERIOD,
660 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
662 * GstRtpJitterBuffer:rtx-max-retries:
664 * The maximum number of retries to request a retransmission.
666 * This implies that as maximum (rtx-max-retries + 1) retransmissions will be requested.
667 * When -1 is used, the number of retransmission request will not be limited.
671 g_object_class_install_property (gobject_class, PROP_RTX_MAX_RETRIES,
672 g_param_spec_int ("rtx-max-retries", "RTX Max Retries",
673 "The maximum number of retries to request a retransmission. "
674 "(-1 not limited)", -1, G_MAXINT, DEFAULT_RTX_MAX_RETRIES,
675 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
677 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
678 g_param_spec_uint ("max-dropout-time", "Max dropout time",
679 "The maximum time (milliseconds) of missing packets tolerated.",
680 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
681 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
683 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
684 g_param_spec_uint ("max-misorder-time", "Max misorder time",
685 "The maximum time (milliseconds) of misordered packets tolerated.",
686 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
687 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
689 * GstRtpJitterBuffer:stats:
691 * Various jitterbuffer statistics. This property returns a GstStructure
692 * with name application/x-rtp-jitterbuffer-stats with the following fields:
698 * <classname>"rtx-count"</classname>:
699 * the number of retransmissions requested.
705 * <classname>"rtx-success-count"</classname>:
706 * the number of successful retransmissions.
712 * <classname>"rtx-per-packet"</classname>:
713 * average number of RTX per packet.
719 * <classname>"rtx-rtt"</classname>:
720 * average round trip time per RTX.
727 g_object_class_install_property (gobject_class, PROP_STATS,
728 g_param_spec_boxed ("stats", "Statistics",
729 "Various statistics", GST_TYPE_STRUCTURE,
730 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
733 * GstRtpJitterBuffer:max-rtcp-rtp-time-diff
735 * The maximum amount of time in ms that the RTP time in the RTCP SRs
736 * is allowed to be ahead of the last RTP packet we received. Use
737 * -1 to disable ignoring of RTCP packets.
741 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
742 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
743 "Maximum amount of time in ms that the RTP time in RTCP SRs "
744 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
745 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
746 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
749 * GstRtpJitterBuffer::request-pt-map:
750 * @buffer: the object which received the signal
753 * Request the payload type as #GstCaps for @pt.
755 gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP] =
756 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
757 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
758 request_pt_map), NULL, NULL, g_cclosure_marshal_generic,
759 GST_TYPE_CAPS, 1, G_TYPE_UINT);
761 * GstRtpJitterBuffer::handle-sync:
762 * @buffer: the object which received the signal
763 * @struct: a GstStructure containing sync values.
765 * Be notified of new sync values.
767 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC] =
768 g_signal_new ("handle-sync", G_TYPE_FROM_CLASS (klass),
769 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
770 handle_sync), NULL, NULL, g_cclosure_marshal_VOID__BOXED,
771 G_TYPE_NONE, 1, GST_TYPE_STRUCTURE | G_SIGNAL_TYPE_STATIC_SCOPE);
774 * GstRtpJitterBuffer::on-npt-stop:
775 * @buffer: the object which received the signal
777 * Signal that the jitterbufer has pushed the RTP packet that corresponds to
778 * the npt-stop position.
780 gst_rtp_jitter_buffer_signals[SIGNAL_ON_NPT_STOP] =
781 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
782 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpJitterBufferClass,
783 on_npt_stop), NULL, NULL, g_cclosure_marshal_VOID__VOID,
784 G_TYPE_NONE, 0, G_TYPE_NONE);
787 * GstRtpJitterBuffer::clear-pt-map:
788 * @buffer: the object which received the signal
790 * Invalidate the clock-rate as obtained with the
791 * #GstRtpJitterBuffer::request-pt-map signal.
793 gst_rtp_jitter_buffer_signals[SIGNAL_CLEAR_PT_MAP] =
794 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
795 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
796 G_STRUCT_OFFSET (GstRtpJitterBufferClass, clear_pt_map), NULL, NULL,
797 g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
800 * GstRtpJitterBuffer::set-active:
801 * @buffer: the object which received the signal
803 * Start pushing out packets with the given base time. This signal is only
804 * useful in buffering mode.
806 * Returns: the time of the last pushed packet.
808 gst_rtp_jitter_buffer_signals[SIGNAL_SET_ACTIVE] =
809 g_signal_new ("set-active", G_TYPE_FROM_CLASS (klass),
810 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
811 G_STRUCT_OFFSET (GstRtpJitterBufferClass, set_active), NULL, NULL,
812 g_cclosure_marshal_generic, G_TYPE_UINT64, 2, G_TYPE_BOOLEAN,
815 gstelement_class->change_state =
816 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_change_state);
817 gstelement_class->request_new_pad =
818 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_request_new_pad);
819 gstelement_class->release_pad =
820 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_release_pad);
821 gstelement_class->provide_clock =
822 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_provide_clock);
824 gst_element_class_add_pad_template (gstelement_class,
825 gst_static_pad_template_get (&gst_rtp_jitter_buffer_src_template));
826 gst_element_class_add_pad_template (gstelement_class,
827 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_template));
828 gst_element_class_add_pad_template (gstelement_class,
829 gst_static_pad_template_get (&gst_rtp_jitter_buffer_sink_rtcp_template));
831 gst_element_class_set_static_metadata (gstelement_class,
832 "RTP packet jitter-buffer", "Filter/Network/RTP",
833 "A buffer that deals with network jitter and other transmission faults",
834 "Philippe Kalaf <philippe.kalaf@collabora.co.uk>, "
835 "Wim Taymans <wim.taymans@gmail.com>");
837 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_clear_pt_map);
838 klass->set_active = GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_set_active);
840 GST_DEBUG_CATEGORY_INIT
841 (rtpjitterbuffer_debug, "rtpjitterbuffer", 0, "RTP Jitter Buffer");
845 gst_rtp_jitter_buffer_init (GstRtpJitterBuffer * jitterbuffer)
847 GstRtpJitterBufferPrivate *priv;
849 priv = GST_RTP_JITTER_BUFFER_GET_PRIVATE (jitterbuffer);
850 jitterbuffer->priv = priv;
852 priv->latency_ms = DEFAULT_LATENCY_MS;
853 priv->latency_ns = priv->latency_ms * GST_MSECOND;
854 priv->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
855 priv->do_lost = DEFAULT_DO_LOST;
856 priv->do_retransmission = DEFAULT_DO_RETRANSMISSION;
857 priv->rtx_next_seqnum = DEFAULT_RTX_NEXT_SEQNUM;
858 priv->rtx_delay = DEFAULT_RTX_DELAY;
859 priv->rtx_min_delay = DEFAULT_RTX_MIN_DELAY;
860 priv->rtx_delay_reorder = DEFAULT_RTX_DELAY_REORDER;
861 priv->rtx_retry_timeout = DEFAULT_RTX_RETRY_TIMEOUT;
862 priv->rtx_min_retry_timeout = DEFAULT_RTX_MIN_RETRY_TIMEOUT;
863 priv->rtx_retry_period = DEFAULT_RTX_RETRY_PERIOD;
864 priv->rtx_max_retries = DEFAULT_RTX_MAX_RETRIES;
865 priv->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
866 priv->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
867 priv->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
870 priv->last_rtptime = -1;
871 priv->avg_jitter = 0;
872 priv->timers = g_array_new (FALSE, TRUE, sizeof (TimerData));
873 priv->jbuf = rtp_jitter_buffer_new ();
874 g_mutex_init (&priv->jbuf_lock);
875 g_cond_init (&priv->jbuf_timer);
876 g_cond_init (&priv->jbuf_event);
877 g_cond_init (&priv->jbuf_query);
878 g_queue_init (&priv->gap_packets);
880 /* reset skew detection initialy */
881 rtp_jitter_buffer_reset_skew (priv->jbuf);
882 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
883 rtp_jitter_buffer_set_buffering (priv->jbuf, FALSE);
887 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_src_template,
890 gst_pad_set_activatemode_function (priv->srcpad,
891 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_activate_mode));
892 gst_pad_set_query_function (priv->srcpad,
893 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_query));
894 gst_pad_set_event_function (priv->srcpad,
895 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_src_event));
898 gst_pad_new_from_static_template (&gst_rtp_jitter_buffer_sink_template,
901 gst_pad_set_chain_function (priv->sinkpad,
902 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_chain));
903 gst_pad_set_event_function (priv->sinkpad,
904 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_event));
905 gst_pad_set_query_function (priv->sinkpad,
906 GST_DEBUG_FUNCPTR (gst_rtp_jitter_buffer_sink_query));
908 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->srcpad);
909 gst_element_add_pad (GST_ELEMENT (jitterbuffer), priv->sinkpad);
911 GST_OBJECT_FLAG_SET (jitterbuffer, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
914 #define IS_DROPABLE(it) (((it)->type == ITEM_TYPE_BUFFER) || ((it)->type == ITEM_TYPE_LOST))
916 #define ITEM_TYPE_BUFFER 0
917 #define ITEM_TYPE_LOST 1
918 #define ITEM_TYPE_EVENT 2
919 #define ITEM_TYPE_QUERY 3
921 static RTPJitterBufferItem *
922 alloc_item (gpointer data, guint type, GstClockTime dts, GstClockTime pts,
923 guint seqnum, guint count, guint rtptime)
925 RTPJitterBufferItem *item;
927 item = g_slice_new (RTPJitterBufferItem);
934 item->seqnum = seqnum;
936 item->rtptime = rtptime;
942 free_item (RTPJitterBufferItem * item)
944 g_return_if_fail (item != NULL);
946 if (item->data && item->type != ITEM_TYPE_QUERY)
947 gst_mini_object_unref (item->data);
948 g_slice_free (RTPJitterBufferItem, item);
952 free_item_and_retain_events (RTPJitterBufferItem * item, gpointer user_data)
954 GList **l = user_data;
956 if (item->data && item->type == ITEM_TYPE_EVENT
957 && GST_EVENT_IS_STICKY (item->data)) {
958 *l = g_list_prepend (*l, item->data);
959 } else if (item->data && item->type != ITEM_TYPE_QUERY) {
960 gst_mini_object_unref (item->data);
962 g_slice_free (RTPJitterBufferItem, item);
966 gst_rtp_jitter_buffer_finalize (GObject * object)
968 GstRtpJitterBuffer *jitterbuffer;
969 GstRtpJitterBufferPrivate *priv;
971 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
972 priv = jitterbuffer->priv;
974 g_array_free (priv->timers, TRUE);
975 g_mutex_clear (&priv->jbuf_lock);
976 g_cond_clear (&priv->jbuf_timer);
977 g_cond_clear (&priv->jbuf_event);
978 g_cond_clear (&priv->jbuf_query);
980 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
981 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
982 g_queue_clear (&priv->gap_packets);
983 g_object_unref (priv->jbuf);
985 G_OBJECT_CLASS (parent_class)->finalize (object);
989 gst_rtp_jitter_buffer_iterate_internal_links (GstPad * pad, GstObject * parent)
991 GstRtpJitterBuffer *jitterbuffer;
992 GstPad *otherpad = NULL;
993 GstIterator *it = NULL;
996 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
998 if (pad == jitterbuffer->priv->sinkpad) {
999 otherpad = jitterbuffer->priv->srcpad;
1000 } else if (pad == jitterbuffer->priv->srcpad) {
1001 otherpad = jitterbuffer->priv->sinkpad;
1002 } else if (pad == jitterbuffer->priv->rtcpsinkpad) {
1003 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1007 g_value_init (&val, GST_TYPE_PAD);
1008 g_value_set_object (&val, otherpad);
1009 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1010 g_value_unset (&val);
1017 create_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1019 GstRtpJitterBufferPrivate *priv;
1021 priv = jitterbuffer->priv;
1023 GST_DEBUG_OBJECT (jitterbuffer, "creating RTCP sink pad");
1026 gst_pad_new_from_static_template
1027 (&gst_rtp_jitter_buffer_sink_rtcp_template, "sink_rtcp");
1028 gst_pad_set_chain_function (priv->rtcpsinkpad,
1029 gst_rtp_jitter_buffer_chain_rtcp);
1030 gst_pad_set_event_function (priv->rtcpsinkpad,
1031 (GstPadEventFunction) gst_rtp_jitter_buffer_sink_rtcp_event);
1032 gst_pad_set_iterate_internal_links_function (priv->rtcpsinkpad,
1033 gst_rtp_jitter_buffer_iterate_internal_links);
1034 gst_pad_set_active (priv->rtcpsinkpad, TRUE);
1035 gst_element_add_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1037 return priv->rtcpsinkpad;
1041 remove_rtcp_sink (GstRtpJitterBuffer * jitterbuffer)
1043 GstRtpJitterBufferPrivate *priv;
1045 priv = jitterbuffer->priv;
1047 GST_DEBUG_OBJECT (jitterbuffer, "removing RTCP sink pad");
1049 gst_pad_set_active (priv->rtcpsinkpad, FALSE);
1051 gst_element_remove_pad (GST_ELEMENT_CAST (jitterbuffer), priv->rtcpsinkpad);
1052 priv->rtcpsinkpad = NULL;
1056 gst_rtp_jitter_buffer_request_new_pad (GstElement * element,
1057 GstPadTemplate * templ, const gchar * name, const GstCaps * filter)
1059 GstRtpJitterBuffer *jitterbuffer;
1060 GstElementClass *klass;
1062 GstRtpJitterBufferPrivate *priv;
1064 g_return_val_if_fail (templ != NULL, NULL);
1065 g_return_val_if_fail (GST_IS_RTP_JITTER_BUFFER (element), NULL);
1067 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1068 priv = jitterbuffer->priv;
1069 klass = GST_ELEMENT_GET_CLASS (element);
1071 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
1073 /* figure out the template */
1074 if (templ == gst_element_class_get_pad_template (klass, "sink_rtcp")) {
1075 if (priv->rtcpsinkpad != NULL)
1078 result = create_rtcp_sink (jitterbuffer);
1080 goto wrong_template;
1087 g_warning ("rtpjitterbuffer: this is not our template");
1092 g_warning ("rtpjitterbuffer: pad already requested");
1098 gst_rtp_jitter_buffer_release_pad (GstElement * element, GstPad * pad)
1100 GstRtpJitterBuffer *jitterbuffer;
1101 GstRtpJitterBufferPrivate *priv;
1103 g_return_if_fail (GST_IS_RTP_JITTER_BUFFER (element));
1104 g_return_if_fail (GST_IS_PAD (pad));
1106 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (element);
1107 priv = jitterbuffer->priv;
1109 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
1111 if (priv->rtcpsinkpad == pad) {
1112 remove_rtcp_sink (jitterbuffer);
1121 g_warning ("gstjitterbuffer: asked to release an unknown pad");
1127 gst_rtp_jitter_buffer_provide_clock (GstElement * element)
1129 return gst_system_clock_obtain ();
1133 gst_rtp_jitter_buffer_clear_pt_map (GstRtpJitterBuffer * jitterbuffer)
1135 GstRtpJitterBufferPrivate *priv;
1137 priv = jitterbuffer->priv;
1139 /* this will trigger a new pt-map request signal, FIXME, do something better. */
1142 priv->clock_rate = -1;
1143 /* do not clear current content, but refresh state for new arrival */
1144 GST_DEBUG_OBJECT (jitterbuffer, "reset jitterbuffer");
1145 rtp_jitter_buffer_reset_skew (priv->jbuf);
1150 gst_rtp_jitter_buffer_set_active (GstRtpJitterBuffer * jbuf, gboolean active,
1153 GstRtpJitterBufferPrivate *priv;
1154 GstClockTime last_out;
1155 RTPJitterBufferItem *item;
1160 GST_DEBUG_OBJECT (jbuf, "setting active %d with offset %" GST_TIME_FORMAT,
1161 active, GST_TIME_ARGS (offset));
1163 if (active != priv->active) {
1164 /* add the amount of time spent in paused to the output offset. All
1165 * outgoing buffers will have this offset applied to their timestamps in
1166 * order to make them arrive in time in the sink. */
1167 priv->out_offset = offset;
1168 GST_DEBUG_OBJECT (jbuf, "out offset %" GST_TIME_FORMAT,
1169 GST_TIME_ARGS (priv->out_offset));
1170 priv->active = active;
1171 JBUF_SIGNAL_EVENT (priv);
1174 rtp_jitter_buffer_set_buffering (priv->jbuf, TRUE);
1176 if ((item = rtp_jitter_buffer_peek (priv->jbuf))) {
1177 /* head buffer timestamp and offset gives our output time */
1178 last_out = item->dts + priv->ts_offset;
1180 /* use last known time when the buffer is empty */
1181 last_out = priv->last_out_time;
1189 gst_rtp_jitter_buffer_getcaps (GstPad * pad, GstCaps * filter)
1191 GstRtpJitterBuffer *jitterbuffer;
1192 GstRtpJitterBufferPrivate *priv;
1197 jitterbuffer = GST_RTP_JITTER_BUFFER (gst_pad_get_parent (pad));
1198 priv = jitterbuffer->priv;
1200 other = (pad == priv->srcpad ? priv->sinkpad : priv->srcpad);
1202 caps = gst_pad_peer_query_caps (other, filter);
1204 templ = gst_pad_get_pad_template_caps (pad);
1206 GST_DEBUG_OBJECT (jitterbuffer, "use template");
1211 GST_DEBUG_OBJECT (jitterbuffer, "intersect with template");
1213 intersect = gst_caps_intersect (caps, templ);
1214 gst_caps_unref (caps);
1215 gst_caps_unref (templ);
1219 gst_object_unref (jitterbuffer);
1225 * Must be called with JBUF_LOCK held
1229 gst_jitter_buffer_sink_parse_caps (GstRtpJitterBuffer * jitterbuffer,
1232 GstRtpJitterBufferPrivate *priv;
1233 GstStructure *caps_struct;
1237 priv = jitterbuffer->priv;
1239 /* first parse the caps */
1240 caps_struct = gst_caps_get_structure (caps, 0);
1242 GST_DEBUG_OBJECT (jitterbuffer, "got caps");
1244 /* we need a clock-rate to convert the rtp timestamps to GStreamer time and to
1245 * measure the amount of data in the buffer */
1246 if (!gst_structure_get_int (caps_struct, "clock-rate", &priv->clock_rate))
1249 if (priv->clock_rate <= 0)
1252 GST_DEBUG_OBJECT (jitterbuffer, "got clock-rate %d", priv->clock_rate);
1254 rtp_jitter_buffer_set_clock_rate (priv->jbuf, priv->clock_rate);
1256 /* The clock base is the RTP timestamp corrsponding to the npt-start value. We
1257 * can use this to track the amount of time elapsed on the sender. */
1258 if (gst_structure_get_uint (caps_struct, "clock-base", &val))
1259 priv->clock_base = val;
1261 priv->clock_base = -1;
1263 priv->ext_timestamp = priv->clock_base;
1265 GST_DEBUG_OBJECT (jitterbuffer, "got clock-base %" G_GINT64_FORMAT,
1268 if (gst_structure_get_uint (caps_struct, "seqnum-base", &val)) {
1269 /* first expected seqnum, only update when we didn't have a previous base. */
1270 if (priv->next_in_seqnum == -1)
1271 priv->next_in_seqnum = val;
1272 if (priv->next_seqnum == -1) {
1273 priv->next_seqnum = val;
1274 JBUF_SIGNAL_EVENT (priv);
1276 priv->seqnum_base = val;
1278 priv->seqnum_base = -1;
1281 GST_DEBUG_OBJECT (jitterbuffer, "got seqnum-base %d", priv->next_in_seqnum);
1283 /* the start and stop times. The seqnum-base corresponds to the start time. We
1284 * will keep track of the seqnums on the output and when we reach the one
1285 * corresponding to npt-stop, we emit the npt-stop-reached signal */
1286 if (gst_structure_get_clock_time (caps_struct, "npt-start", &tval))
1287 priv->npt_start = tval;
1289 priv->npt_start = 0;
1291 if (gst_structure_get_clock_time (caps_struct, "npt-stop", &tval))
1292 priv->npt_stop = tval;
1294 priv->npt_stop = -1;
1296 GST_DEBUG_OBJECT (jitterbuffer,
1297 "npt start/stop: %" GST_TIME_FORMAT "-%" GST_TIME_FORMAT,
1298 GST_TIME_ARGS (priv->npt_start), GST_TIME_ARGS (priv->npt_stop));
1305 GST_DEBUG_OBJECT (jitterbuffer, "No clock-rate in caps!");
1310 GST_DEBUG_OBJECT (jitterbuffer, "Invalid clock-rate %d", priv->clock_rate);
1316 gst_rtp_jitter_buffer_flush_start (GstRtpJitterBuffer * jitterbuffer)
1318 GstRtpJitterBufferPrivate *priv;
1320 priv = jitterbuffer->priv;
1323 /* mark ourselves as flushing */
1324 priv->srcresult = GST_FLOW_FLUSHING;
1325 GST_DEBUG_OBJECT (jitterbuffer, "Disabling pop on queue");
1326 /* this unblocks any waiting pops on the src pad task */
1327 JBUF_SIGNAL_EVENT (priv);
1328 JBUF_SIGNAL_QUERY (priv, FALSE);
1333 gst_rtp_jitter_buffer_flush_stop (GstRtpJitterBuffer * jitterbuffer)
1335 GstRtpJitterBufferPrivate *priv;
1337 priv = jitterbuffer->priv;
1340 GST_DEBUG_OBJECT (jitterbuffer, "Enabling pop on queue");
1341 /* Mark as non flushing */
1342 priv->srcresult = GST_FLOW_OK;
1343 gst_segment_init (&priv->segment, GST_FORMAT_TIME);
1344 priv->last_popped_seqnum = -1;
1345 priv->last_out_time = -1;
1346 priv->next_seqnum = -1;
1347 priv->seqnum_base = -1;
1348 priv->ips_rtptime = -1;
1349 priv->ips_dts = GST_CLOCK_TIME_NONE;
1350 priv->packet_spacing = 0;
1351 priv->next_in_seqnum = -1;
1352 priv->clock_rate = -1;
1355 priv->estimated_eos = -1;
1356 priv->last_elapsed = 0;
1357 priv->ext_timestamp = -1;
1358 priv->avg_jitter = 0;
1359 priv->last_dts = -1;
1360 priv->last_rtptime = -1;
1361 priv->last_in_dts = 0;
1362 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
1363 rtp_jitter_buffer_flush (priv->jbuf, (GFunc) free_item, NULL);
1364 rtp_jitter_buffer_disable_buffering (priv->jbuf, FALSE);
1365 rtp_jitter_buffer_reset_skew (priv->jbuf);
1366 remove_all_timers (jitterbuffer);
1367 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
1368 g_queue_clear (&priv->gap_packets);
1373 gst_rtp_jitter_buffer_src_activate_mode (GstPad * pad, GstObject * parent,
1374 GstPadMode mode, gboolean active)
1377 GstRtpJitterBuffer *jitterbuffer = NULL;
1379 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1382 case GST_PAD_MODE_PUSH:
1384 /* allow data processing */
1385 gst_rtp_jitter_buffer_flush_stop (jitterbuffer);
1387 /* start pushing out buffers */
1388 GST_DEBUG_OBJECT (jitterbuffer, "Starting task on srcpad");
1389 result = gst_pad_start_task (jitterbuffer->priv->srcpad,
1390 (GstTaskFunction) gst_rtp_jitter_buffer_loop, jitterbuffer, NULL);
1392 /* make sure all data processing stops ASAP */
1393 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1395 /* NOTE this will hardlock if the state change is called from the src pad
1396 * task thread because we will _join() the thread. */
1397 GST_DEBUG_OBJECT (jitterbuffer, "Stopping task on srcpad");
1398 result = gst_pad_stop_task (pad);
1408 static GstStateChangeReturn
1409 gst_rtp_jitter_buffer_change_state (GstElement * element,
1410 GstStateChange transition)
1412 GstRtpJitterBuffer *jitterbuffer;
1413 GstRtpJitterBufferPrivate *priv;
1414 GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
1416 jitterbuffer = GST_RTP_JITTER_BUFFER (element);
1417 priv = jitterbuffer->priv;
1419 switch (transition) {
1420 case GST_STATE_CHANGE_NULL_TO_READY:
1422 case GST_STATE_CHANGE_READY_TO_PAUSED:
1424 /* reset negotiated values */
1425 priv->clock_rate = -1;
1426 priv->clock_base = -1;
1427 priv->peer_latency = 0;
1429 /* block until we go to PLAYING */
1430 priv->blocked = TRUE;
1431 priv->timer_running = TRUE;
1432 priv->timer_thread =
1433 g_thread_new ("timer", (GThreadFunc) wait_next_timeout, jitterbuffer);
1436 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1438 /* unblock to allow streaming in PLAYING */
1439 priv->blocked = FALSE;
1440 JBUF_SIGNAL_EVENT (priv);
1441 JBUF_SIGNAL_TIMER (priv);
1448 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1450 switch (transition) {
1451 case GST_STATE_CHANGE_READY_TO_PAUSED:
1452 /* we are a live element because we sync to the clock, which we can only
1453 * do in the PLAYING state */
1454 if (ret != GST_STATE_CHANGE_FAILURE)
1455 ret = GST_STATE_CHANGE_NO_PREROLL;
1457 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1459 /* block to stop streaming when PAUSED */
1460 priv->blocked = TRUE;
1461 unschedule_current_timer (jitterbuffer);
1463 if (ret != GST_STATE_CHANGE_FAILURE)
1464 ret = GST_STATE_CHANGE_NO_PREROLL;
1466 case GST_STATE_CHANGE_PAUSED_TO_READY:
1468 gst_buffer_replace (&priv->last_sr, NULL);
1469 priv->timer_running = FALSE;
1470 unschedule_current_timer (jitterbuffer);
1471 JBUF_SIGNAL_TIMER (priv);
1472 JBUF_SIGNAL_QUERY (priv, FALSE);
1474 g_thread_join (priv->timer_thread);
1475 priv->timer_thread = NULL;
1477 case GST_STATE_CHANGE_READY_TO_NULL:
1487 gst_rtp_jitter_buffer_src_event (GstPad * pad, GstObject * parent,
1490 gboolean ret = TRUE;
1491 GstRtpJitterBuffer *jitterbuffer;
1492 GstRtpJitterBufferPrivate *priv;
1494 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
1495 priv = jitterbuffer->priv;
1497 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1499 switch (GST_EVENT_TYPE (event)) {
1500 case GST_EVENT_LATENCY:
1502 GstClockTime latency;
1504 gst_event_parse_latency (event, &latency);
1506 GST_DEBUG_OBJECT (jitterbuffer,
1507 "configuring latency of %" GST_TIME_FORMAT, GST_TIME_ARGS (latency));
1510 /* adjust the overall buffer delay to the total pipeline latency in
1511 * buffering mode because if downstream consumes too fast (because of
1512 * large latency or queues, we would start rebuffering again. */
1513 if (rtp_jitter_buffer_get_mode (priv->jbuf) ==
1514 RTP_JITTER_BUFFER_MODE_BUFFER) {
1515 rtp_jitter_buffer_set_delay (priv->jbuf, latency);
1519 ret = gst_pad_push_event (priv->sinkpad, event);
1523 ret = gst_pad_push_event (priv->sinkpad, event);
1530 /* handles and stores the event in the jitterbuffer, must be called with
1533 queue_event (GstRtpJitterBuffer * jitterbuffer, GstEvent * event)
1535 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1536 RTPJitterBufferItem *item;
1539 switch (GST_EVENT_TYPE (event)) {
1540 case GST_EVENT_CAPS:
1544 gst_event_parse_caps (event, &caps);
1545 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1548 case GST_EVENT_SEGMENT:
1549 gst_event_copy_segment (event, &priv->segment);
1551 /* we need time for now */
1552 if (priv->segment.format != GST_FORMAT_TIME)
1553 goto newseg_wrong_format;
1555 GST_DEBUG_OBJECT (jitterbuffer,
1556 "segment: %" GST_SEGMENT_FORMAT, &priv->segment);
1560 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
1567 GST_DEBUG_OBJECT (jitterbuffer, "adding event");
1568 item = alloc_item (event, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
1569 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
1571 JBUF_SIGNAL_EVENT (priv);
1576 newseg_wrong_format:
1578 GST_DEBUG_OBJECT (jitterbuffer, "received non TIME newsegment");
1579 gst_event_unref (event);
1585 gst_rtp_jitter_buffer_sink_event (GstPad * pad, GstObject * parent,
1588 gboolean ret = TRUE;
1589 GstRtpJitterBuffer *jitterbuffer;
1590 GstRtpJitterBufferPrivate *priv;
1592 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1593 priv = jitterbuffer->priv;
1595 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1597 switch (GST_EVENT_TYPE (event)) {
1598 case GST_EVENT_FLUSH_START:
1599 ret = gst_pad_push_event (priv->srcpad, event);
1600 gst_rtp_jitter_buffer_flush_start (jitterbuffer);
1601 /* wait for the loop to go into PAUSED */
1602 gst_pad_pause_task (priv->srcpad);
1604 case GST_EVENT_FLUSH_STOP:
1605 ret = gst_pad_push_event (priv->srcpad, event);
1607 gst_rtp_jitter_buffer_src_activate_mode (priv->srcpad, parent,
1608 GST_PAD_MODE_PUSH, TRUE);
1611 if (GST_EVENT_IS_SERIALIZED (event)) {
1612 /* serialized events go in the queue */
1614 if (priv->srcresult != GST_FLOW_OK) {
1615 /* Errors in sticky event pushing are no problem and ignored here
1616 * as they will cause more meaningful errors during data flow.
1617 * For EOS events, that are not followed by data flow, we still
1618 * return FALSE here though.
1620 if (!GST_EVENT_IS_STICKY (event) ||
1621 GST_EVENT_TYPE (event) == GST_EVENT_EOS)
1622 goto out_flow_error;
1624 /* refuse more events on EOS */
1627 ret = queue_event (jitterbuffer, event);
1630 /* non-serialized events are forwarded downstream immediately */
1631 ret = gst_pad_push_event (priv->srcpad, event);
1640 GST_DEBUG_OBJECT (jitterbuffer,
1641 "refusing event, we have a downstream flow error: %s",
1642 gst_flow_get_name (priv->srcresult));
1644 gst_event_unref (event);
1649 GST_DEBUG_OBJECT (jitterbuffer, "refusing event, we are EOS");
1651 gst_event_unref (event);
1657 gst_rtp_jitter_buffer_sink_rtcp_event (GstPad * pad, GstObject * parent,
1660 gboolean ret = TRUE;
1661 GstRtpJitterBuffer *jitterbuffer;
1663 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
1665 GST_DEBUG_OBJECT (jitterbuffer, "received %s", GST_EVENT_TYPE_NAME (event));
1667 switch (GST_EVENT_TYPE (event)) {
1668 case GST_EVENT_FLUSH_START:
1669 gst_event_unref (event);
1671 case GST_EVENT_FLUSH_STOP:
1672 gst_event_unref (event);
1675 ret = gst_pad_event_default (pad, parent, event);
1683 * Must be called with JBUF_LOCK held, will release the LOCK when emiting the
1684 * signal. The function returns GST_FLOW_ERROR when a parsing error happened and
1685 * GST_FLOW_FLUSHING when the element is shutting down. On success
1686 * GST_FLOW_OK is returned.
1688 static GstFlowReturn
1689 gst_rtp_jitter_buffer_get_clock_rate (GstRtpJitterBuffer * jitterbuffer,
1693 GValue args[2] = { {0}, {0} };
1697 g_value_init (&args[0], GST_TYPE_ELEMENT);
1698 g_value_set_object (&args[0], jitterbuffer);
1699 g_value_init (&args[1], G_TYPE_UINT);
1700 g_value_set_uint (&args[1], pt);
1702 g_value_init (&ret, GST_TYPE_CAPS);
1703 g_value_set_boxed (&ret, NULL);
1705 JBUF_UNLOCK (jitterbuffer->priv);
1706 g_signal_emitv (args, gst_rtp_jitter_buffer_signals[SIGNAL_REQUEST_PT_MAP], 0,
1708 JBUF_LOCK_CHECK (jitterbuffer->priv, out_flushing);
1710 g_value_unset (&args[0]);
1711 g_value_unset (&args[1]);
1712 caps = (GstCaps *) g_value_dup_boxed (&ret);
1713 g_value_unset (&ret);
1717 res = gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
1718 gst_caps_unref (caps);
1720 if (G_UNLIKELY (!res))
1728 GST_DEBUG_OBJECT (jitterbuffer, "could not get caps");
1729 return GST_FLOW_ERROR;
1733 JBUF_UNLOCK (jitterbuffer->priv);
1734 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
1735 return GST_FLOW_FLUSHING;
1739 GST_DEBUG_OBJECT (jitterbuffer, "parse failed");
1740 return GST_FLOW_ERROR;
1744 /* call with jbuf lock held */
1746 check_buffering_percent (GstRtpJitterBuffer * jitterbuffer, gint percent)
1748 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1749 GstMessage *message = NULL;
1754 /* Post a buffering message */
1755 if (priv->last_percent != percent) {
1756 priv->last_percent = percent;
1758 gst_message_new_buffering (GST_OBJECT_CAST (jitterbuffer), percent);
1759 gst_message_set_buffering_stats (message, GST_BUFFERING_LIVE, -1, -1, -1);
1766 apply_offset (GstRtpJitterBuffer * jitterbuffer, GstClockTime timestamp)
1768 GstRtpJitterBufferPrivate *priv;
1770 priv = jitterbuffer->priv;
1772 if (timestamp == -1)
1775 /* apply the timestamp offset, this is used for inter stream sync */
1776 timestamp += priv->ts_offset;
1777 /* add the offset, this is used when buffering */
1778 timestamp += priv->out_offset;
1784 find_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type, guint16 seqnum)
1786 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1787 TimerData *timer = NULL;
1790 len = priv->timers->len;
1791 for (i = 0; i < len; i++) {
1792 TimerData *test = &g_array_index (priv->timers, TimerData, i);
1793 if (test->seqnum == seqnum && test->type == type) {
1802 unschedule_current_timer (GstRtpJitterBuffer * jitterbuffer)
1804 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1806 if (priv->clock_id) {
1807 GST_DEBUG_OBJECT (jitterbuffer, "unschedule current timer");
1808 gst_clock_id_unschedule (priv->clock_id);
1809 priv->clock_id = NULL;
1814 get_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1816 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1817 GstClockTime test_timeout;
1819 if ((test_timeout = timer->timeout) == -1)
1822 if (timer->type != TIMER_TYPE_EXPECTED) {
1823 /* add our latency and offset to get output times. */
1824 test_timeout = apply_offset (jitterbuffer, test_timeout);
1825 test_timeout += priv->latency_ns;
1827 return test_timeout;
1831 recalculate_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1833 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1835 if (priv->clock_id) {
1836 GstClockTime timeout = get_timeout (jitterbuffer, timer);
1838 GST_DEBUG ("%" GST_TIME_FORMAT " <> %" GST_TIME_FORMAT,
1839 GST_TIME_ARGS (timeout), GST_TIME_ARGS (priv->timer_timeout));
1841 if (timeout == -1 || timeout < priv->timer_timeout)
1842 unschedule_current_timer (jitterbuffer);
1847 add_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1848 guint16 seqnum, guint num, GstClockTime timeout, GstClockTime delay,
1849 GstClockTime duration)
1851 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1855 GST_DEBUG_OBJECT (jitterbuffer,
1856 "add timer %d for seqnum %d to %" GST_TIME_FORMAT ", delay %"
1857 GST_TIME_FORMAT, type, seqnum, GST_TIME_ARGS (timeout),
1858 GST_TIME_ARGS (delay));
1860 len = priv->timers->len;
1861 g_array_set_size (priv->timers, len + 1);
1862 timer = &g_array_index (priv->timers, TimerData, len);
1865 timer->seqnum = seqnum;
1867 timer->timeout = timeout + delay;
1868 timer->duration = duration;
1869 if (type == TIMER_TYPE_EXPECTED) {
1870 timer->rtx_base = timeout;
1871 timer->rtx_delay = delay;
1872 timer->rtx_retry = 0;
1874 timer->num_rtx_retry = 0;
1875 recalculate_timer (jitterbuffer, timer);
1876 JBUF_SIGNAL_TIMER (priv);
1882 reschedule_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
1883 guint16 seqnum, GstClockTime timeout, GstClockTime delay, gboolean reset)
1885 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1886 gboolean seqchange, timechange;
1889 seqchange = timer->seqnum != seqnum;
1890 timechange = timer->timeout != timeout;
1892 if (!seqchange && !timechange)
1895 oldseq = timer->seqnum;
1897 GST_DEBUG_OBJECT (jitterbuffer,
1898 "replace timer for seqnum %d->%d to %" GST_TIME_FORMAT,
1899 oldseq, seqnum, GST_TIME_ARGS (timeout + delay));
1901 timer->timeout = timeout + delay;
1902 timer->seqnum = seqnum;
1904 timer->rtx_base = timeout;
1905 timer->rtx_delay = delay;
1906 timer->rtx_retry = 0;
1909 timer->num_rtx_retry = 0;
1911 if (priv->clock_id) {
1912 /* we changed the seqnum and there is a timer currently waiting with this
1913 * seqnum, unschedule it */
1914 if (seqchange && priv->timer_seqnum == oldseq)
1915 unschedule_current_timer (jitterbuffer);
1916 /* we changed the time, check if it is earlier than what we are waiting
1917 * for and unschedule if so */
1918 else if (timechange)
1919 recalculate_timer (jitterbuffer, timer);
1924 set_timer (GstRtpJitterBuffer * jitterbuffer, TimerType type,
1925 guint16 seqnum, GstClockTime timeout)
1929 /* find the seqnum timer */
1930 timer = find_timer (jitterbuffer, type, seqnum);
1931 if (timer == NULL) {
1932 timer = add_timer (jitterbuffer, type, seqnum, 0, timeout, 0, -1);
1934 reschedule_timer (jitterbuffer, timer, seqnum, timeout, 0, FALSE);
1940 remove_timer (GstRtpJitterBuffer * jitterbuffer, TimerData * timer)
1942 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1945 if (priv->clock_id && priv->timer_seqnum == timer->seqnum)
1946 unschedule_current_timer (jitterbuffer);
1949 GST_DEBUG_OBJECT (jitterbuffer, "removed index %d", idx);
1950 g_array_remove_index_fast (priv->timers, idx);
1955 remove_all_timers (GstRtpJitterBuffer * jitterbuffer)
1957 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
1958 GST_DEBUG_OBJECT (jitterbuffer, "removed all timers");
1959 g_array_set_size (priv->timers, 0);
1960 unschedule_current_timer (jitterbuffer);
1963 /* get the extra delay to wait before sending RTX */
1965 get_rtx_delay (GstRtpJitterBufferPrivate * priv)
1969 if (priv->rtx_delay == -1) {
1970 if (priv->avg_jitter == 0 && priv->packet_spacing == 0) {
1971 delay = DEFAULT_AUTO_RTX_DELAY;
1973 /* jitter is in nanoseconds, maximum of 2x jitter and half the
1974 * packet spacing is a good margin */
1975 delay = MAX (priv->avg_jitter * 2, priv->packet_spacing / 2);
1978 delay = priv->rtx_delay * GST_MSECOND;
1980 if (priv->rtx_min_delay > 0)
1981 delay = MAX (delay, priv->rtx_min_delay * GST_MSECOND);
1986 /* we just received a packet with seqnum and dts.
1988 * First check for old seqnum that we are still expecting. If the gap with the
1989 * current seqnum is too big, unschedule the timeouts.
1991 * If we have a valid packet spacing estimate we can set a timer for when we
1992 * should receive the next packet.
1993 * If we don't have a valid estimate, we remove any timer we might have
1994 * had for this packet.
1997 update_timers (GstRtpJitterBuffer * jitterbuffer, guint16 seqnum,
1998 GstClockTime dts, gboolean do_next_seqnum)
2000 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2001 TimerData *timer = NULL;
2004 /* go through all timers and unschedule the ones with a large gap, also find
2005 * the timer for the seqnum */
2006 len = priv->timers->len;
2007 for (i = 0; i < len; i++) {
2008 TimerData *test = &g_array_index (priv->timers, TimerData, i);
2011 gap = gst_rtp_buffer_compare_seqnum (test->seqnum, seqnum);
2013 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, #%d<->#%d gap %d", i,
2014 test->type, test->seqnum, seqnum, gap);
2017 GST_DEBUG ("found timer for current seqnum");
2018 /* the timer for the current seqnum */
2020 /* when no retransmission, we can stop now, we only need to find the
2021 * timer for the current seqnum */
2022 if (!priv->do_retransmission)
2024 } else if (gap > priv->rtx_delay_reorder) {
2025 /* max gap, we exceeded the max reorder distance and we don't expect the
2026 * missing packet to be this reordered */
2027 if (test->num_rtx_retry == 0 && test->type == TIMER_TYPE_EXPECTED)
2028 reschedule_timer (jitterbuffer, test, test->seqnum, -1, 0, FALSE);
2032 do_next_seqnum = do_next_seqnum && priv->packet_spacing > 0
2033 && priv->do_retransmission && priv->rtx_next_seqnum;
2035 if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2036 if (timer->num_rtx_retry > 0) {
2037 GstClockTime rtx_last, delay;
2039 /* we scheduled a retry for this packet and now we have it */
2040 priv->num_rtx_success++;
2041 /* all the previous retry attempts failed */
2042 priv->num_rtx_failed += timer->num_rtx_retry - 1;
2043 /* number of retries before receiving the packet */
2044 if (priv->avg_rtx_num == 0.0)
2045 priv->avg_rtx_num = timer->num_rtx_retry;
2047 priv->avg_rtx_num = (timer->num_rtx_retry + 7 * priv->avg_rtx_num) / 8;
2048 /* calculate the delay between retransmission request and receiving this
2049 * packet, start with when we scheduled this timeout last */
2050 rtx_last = timer->rtx_last;
2051 if (dts != GST_CLOCK_TIME_NONE && dts > rtx_last) {
2052 /* we have a valid delay if this packet arrived after we scheduled the
2054 delay = dts - rtx_last;
2055 if (priv->avg_rtx_rtt == 0)
2056 priv->avg_rtx_rtt = delay;
2058 priv->avg_rtx_rtt = (delay + 7 * priv->avg_rtx_rtt) / 8;
2062 GST_LOG_OBJECT (jitterbuffer,
2063 "RTX success %" G_GUINT64_FORMAT ", failed %" G_GUINT64_FORMAT
2064 ", requests %" G_GUINT64_FORMAT ", dups %" G_GUINT64_FORMAT
2065 ", avg-num %g, delay %" GST_TIME_FORMAT ", avg-rtt %" GST_TIME_FORMAT,
2066 priv->num_rtx_success, priv->num_rtx_failed, priv->num_rtx_requests,
2067 priv->num_duplicates, priv->avg_rtx_num, GST_TIME_ARGS (delay),
2068 GST_TIME_ARGS (priv->avg_rtx_rtt));
2070 /* don't try to estimate the next seqnum because this is a retransmitted
2071 * packet and it probably did not arrive with the expected packet
2073 do_next_seqnum = FALSE;
2077 if (do_next_seqnum && dts != GST_CLOCK_TIME_NONE) {
2078 GstClockTime expected, delay;
2080 /* calculate expected arrival time of the next seqnum */
2081 expected = dts + priv->packet_spacing;
2083 delay = get_rtx_delay (priv);
2085 /* and update/install timer for next seqnum */
2087 reschedule_timer (jitterbuffer, timer, priv->next_in_seqnum, expected,
2090 add_timer (jitterbuffer, TIMER_TYPE_EXPECTED, priv->next_in_seqnum, 0,
2091 expected, delay, priv->packet_spacing);
2093 } else if (timer && timer->type != TIMER_TYPE_DEADLINE) {
2094 /* if we had a timer, remove it, we don't know when to expect the next
2096 remove_timer (jitterbuffer, timer);
2101 calculate_packet_spacing (GstRtpJitterBuffer * jitterbuffer, guint32 rtptime,
2104 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2106 /* we need consecutive seqnums with a different
2107 * rtptime to estimate the packet spacing. */
2108 if (priv->ips_rtptime != rtptime) {
2109 /* rtptime changed, check dts diff */
2110 if (priv->ips_dts != -1 && dts != -1 && dts > priv->ips_dts) {
2111 GstClockTime new_packet_spacing = dts - priv->ips_dts;
2112 GstClockTime old_packet_spacing = priv->packet_spacing;
2114 /* Biased towards bigger packet spacings to prevent
2115 * too many unneeded retransmission requests for next
2116 * packets that just arrive a little later than we would
2118 if (old_packet_spacing > new_packet_spacing)
2119 priv->packet_spacing =
2120 (new_packet_spacing + 3 * old_packet_spacing) / 4;
2121 else if (old_packet_spacing > 0)
2122 priv->packet_spacing =
2123 (3 * new_packet_spacing + old_packet_spacing) / 4;
2125 priv->packet_spacing = new_packet_spacing;
2127 GST_DEBUG_OBJECT (jitterbuffer,
2128 "new packet spacing %" GST_TIME_FORMAT
2129 " old packet spacing %" GST_TIME_FORMAT
2130 " combined to %" GST_TIME_FORMAT,
2131 GST_TIME_ARGS (new_packet_spacing),
2132 GST_TIME_ARGS (old_packet_spacing),
2133 GST_TIME_ARGS (priv->packet_spacing));
2135 priv->ips_rtptime = rtptime;
2136 priv->ips_dts = dts;
2141 calculate_expected (GstRtpJitterBuffer * jitterbuffer, guint32 expected,
2142 guint16 seqnum, GstClockTime dts, gint gap)
2144 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2145 GstClockTime total_duration, duration, expected_dts;
2148 GST_DEBUG_OBJECT (jitterbuffer,
2149 "dts %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT,
2150 GST_TIME_ARGS (dts), GST_TIME_ARGS (priv->last_in_dts));
2152 if (dts == GST_CLOCK_TIME_NONE) {
2153 GST_WARNING_OBJECT (jitterbuffer, "Have no DTS");
2157 /* the total duration spanned by the missing packets */
2158 if (dts >= priv->last_in_dts)
2159 total_duration = dts - priv->last_in_dts;
2163 /* interpolate between the current time and the last time based on
2164 * number of packets we are missing, this is the estimated duration
2165 * for the missing packet based on equidistant packet spacing. */
2166 duration = total_duration / (gap + 1);
2168 GST_DEBUG_OBJECT (jitterbuffer, "duration %" GST_TIME_FORMAT,
2169 GST_TIME_ARGS (duration));
2171 if (total_duration > priv->latency_ns) {
2172 GstClockTime gap_time;
2176 GstClockTime gap_dur = gap * duration;
2177 if (gap_dur > priv->latency_ns)
2178 gap_time = gap_dur - priv->latency_ns;
2181 lost_packets = gap_time / duration;
2183 gap_time = total_duration - priv->latency_ns;
2187 /* too many lost packets, some of the missing packets are already
2188 * too late and we can generate lost packet events for them. */
2189 GST_DEBUG_OBJECT (jitterbuffer,
2190 "lost packets (%d, #%d->#%d) duration too large %" GST_TIME_FORMAT
2191 " > %" GST_TIME_FORMAT ", consider %u lost (%" GST_TIME_FORMAT ")",
2192 gap, expected, seqnum, GST_TIME_ARGS (total_duration),
2193 GST_TIME_ARGS (priv->latency_ns), lost_packets,
2194 GST_TIME_ARGS (gap_time));
2196 /* this timer will fire immediately and the lost event will be pushed from
2197 * the timer thread */
2198 if (lost_packets > 0) {
2199 add_timer (jitterbuffer, TIMER_TYPE_LOST, expected, lost_packets,
2200 priv->last_in_dts + duration, 0, gap_time);
2201 expected += lost_packets;
2202 priv->last_in_dts += gap_time;
2206 expected_dts = priv->last_in_dts + duration;
2208 if (priv->do_retransmission) {
2211 type = TIMER_TYPE_EXPECTED;
2212 /* if we had a timer for the first missing packet, update it. */
2213 if ((timer = find_timer (jitterbuffer, type, expected))) {
2214 GstClockTime timeout = timer->timeout;
2216 timer->duration = duration;
2217 if (timeout > (expected_dts + timer->rtx_retry)) {
2218 GstClockTime delay = timeout - expected_dts - timer->rtx_retry;
2219 reschedule_timer (jitterbuffer, timer, timer->seqnum, expected_dts,
2223 expected_dts += duration;
2226 type = TIMER_TYPE_LOST;
2229 while (gst_rtp_buffer_compare_seqnum (expected, seqnum) > 0) {
2230 add_timer (jitterbuffer, type, expected, 0, expected_dts, 0, duration);
2231 expected_dts += duration;
2237 calculate_jitter (GstRtpJitterBuffer * jitterbuffer, GstClockTime dts,
2241 GstClockTimeDiff dtsdiff, rtpdiffns, diff;
2242 GstRtpJitterBufferPrivate *priv;
2244 priv = jitterbuffer->priv;
2246 if (G_UNLIKELY (dts == GST_CLOCK_TIME_NONE) || priv->clock_rate <= 0)
2249 if (priv->last_dts != -1)
2250 dtsdiff = dts - priv->last_dts;
2254 if (priv->last_rtptime != -1)
2255 rtpdiff = rtptime - (guint32) priv->last_rtptime;
2259 priv->last_dts = dts;
2260 priv->last_rtptime = rtptime;
2264 gst_util_uint64_scale_int (rtpdiff, GST_SECOND, priv->clock_rate);
2267 -gst_util_uint64_scale_int (-rtpdiff, GST_SECOND, priv->clock_rate);
2269 diff = ABS (dtsdiff - rtpdiffns);
2271 /* jitter is stored in nanoseconds */
2272 priv->avg_jitter = (diff + (15 * priv->avg_jitter)) >> 4;
2274 GST_LOG_OBJECT (jitterbuffer,
2275 "dtsdiff %" GST_TIME_FORMAT " rtptime %" GST_TIME_FORMAT
2276 ", clock-rate %d, diff %" GST_TIME_FORMAT ", jitter: %" GST_TIME_FORMAT,
2277 GST_TIME_ARGS (dtsdiff), GST_TIME_ARGS (rtpdiffns), priv->clock_rate,
2278 GST_TIME_ARGS (diff), GST_TIME_ARGS (priv->avg_jitter));
2285 GST_DEBUG_OBJECT (jitterbuffer,
2286 "no dts or no clock-rate, can't calculate jitter");
2292 compare_buffer_seqnum (GstBuffer * a, GstBuffer * b, gpointer user_data)
2294 GstRTPBuffer rtp_a = GST_RTP_BUFFER_INIT;
2295 GstRTPBuffer rtp_b = GST_RTP_BUFFER_INIT;
2298 gst_rtp_buffer_map (a, GST_MAP_READ, &rtp_a);
2299 seq_a = gst_rtp_buffer_get_seq (&rtp_a);
2300 gst_rtp_buffer_unmap (&rtp_a);
2302 gst_rtp_buffer_map (b, GST_MAP_READ, &rtp_b);
2303 seq_b = gst_rtp_buffer_get_seq (&rtp_b);
2304 gst_rtp_buffer_unmap (&rtp_b);
2306 return gst_rtp_buffer_compare_seqnum (seq_b, seq_a);
2310 handle_big_gap_buffer (GstRtpJitterBuffer * jitterbuffer, gboolean future,
2311 GstBuffer * buffer, guint8 pt, guint16 seqnum, gint gap, guint max_dropout,
2314 GstRtpJitterBufferPrivate *priv;
2315 guint gap_packets_length;
2316 gboolean reset = FALSE;
2318 priv = jitterbuffer->priv;
2320 if ((gap_packets_length = g_queue_get_length (&priv->gap_packets)) > 0) {
2322 guint32 prev_gap_seq = -1;
2323 gboolean all_consecutive = TRUE;
2325 g_queue_insert_sorted (&priv->gap_packets, buffer,
2326 (GCompareDataFunc) compare_buffer_seqnum, NULL);
2328 for (l = priv->gap_packets.head; l; l = l->next) {
2329 GstBuffer *gap_buffer = l->data;
2330 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2333 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2335 all_consecutive = (gst_rtp_buffer_get_payload_type (&gap_rtp) == pt);
2337 gap_seq = gst_rtp_buffer_get_seq (&gap_rtp);
2338 if (prev_gap_seq == -1)
2339 prev_gap_seq = gap_seq;
2340 else if (gst_rtp_buffer_compare_seqnum (gap_seq, prev_gap_seq) != -1)
2341 all_consecutive = FALSE;
2343 prev_gap_seq = gap_seq;
2345 gst_rtp_buffer_unmap (&gap_rtp);
2346 if (!all_consecutive)
2350 if (all_consecutive && gap_packets_length > 3) {
2351 GST_DEBUG_OBJECT (jitterbuffer,
2352 "buffer too %s %d < %d, got 5 consecutive ones - reset",
2353 (future ? "new" : "old"), gap,
2354 (future ? max_dropout : -max_misorder));
2356 } else if (!all_consecutive) {
2357 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2358 g_queue_clear (&priv->gap_packets);
2359 GST_DEBUG_OBJECT (jitterbuffer,
2360 "buffer too %s %d < %d, got no 5 consecutive ones - dropping",
2361 (future ? "new" : "old"), gap,
2362 (future ? max_dropout : -max_misorder));
2365 GST_DEBUG_OBJECT (jitterbuffer,
2366 "buffer too %s %d < %d, got %u consecutive ones - waiting",
2367 (future ? "new" : "old"), gap,
2368 (future ? max_dropout : -max_misorder), gap_packets_length + 1);
2372 GST_DEBUG_OBJECT (jitterbuffer,
2373 "buffer too %s %d < %d, first one - waiting", (future ? "new" : "old"),
2374 gap, -max_misorder);
2375 g_queue_push_tail (&priv->gap_packets, buffer);
2383 get_current_running_time (GstRtpJitterBuffer * jitterbuffer)
2385 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (jitterbuffer));
2386 GstClockTime running_time = GST_CLOCK_TIME_NONE;
2389 GstClockTime base_time =
2390 gst_element_get_base_time (GST_ELEMENT_CAST (jitterbuffer));
2391 GstClockTime clock_time = gst_clock_get_time (clock);
2393 if (clock_time > base_time)
2394 running_time = clock_time - base_time;
2398 gst_object_unref (clock);
2401 return running_time;
2404 static GstFlowReturn
2405 gst_rtp_jitter_buffer_chain (GstPad * pad, GstObject * parent,
2408 GstRtpJitterBuffer *jitterbuffer;
2409 GstRtpJitterBufferPrivate *priv;
2411 guint32 expected, rtptime;
2412 GstFlowReturn ret = GST_FLOW_OK;
2413 GstClockTime dts, pts;
2418 GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
2419 gboolean do_next_seqnum = FALSE;
2420 RTPJitterBufferItem *item;
2421 GstMessage *msg = NULL;
2422 gboolean estimated_dts = FALSE;
2423 guint32 packet_rate, max_dropout, max_misorder;
2425 jitterbuffer = GST_RTP_JITTER_BUFFER_CAST (parent);
2427 priv = jitterbuffer->priv;
2429 if (G_UNLIKELY (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)))
2430 goto invalid_buffer;
2432 pt = gst_rtp_buffer_get_payload_type (&rtp);
2433 seqnum = gst_rtp_buffer_get_seq (&rtp);
2434 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2435 gst_rtp_buffer_unmap (&rtp);
2437 /* make sure we have PTS and DTS set */
2438 pts = GST_BUFFER_PTS (buffer);
2439 dts = GST_BUFFER_DTS (buffer);
2446 /* If we have no DTS here, i.e. no capture time, get one from the
2447 * clock now to have something to calculate with in the future. */
2448 dts = get_current_running_time (jitterbuffer);
2451 /* Remember that we estimated the DTS if we are running already
2452 * and this is not our first packet (or first packet after a reset).
2453 * If it's the first packet, we somehow must generate a timestamp for
2454 * everything, otherwise we can't calculate any times
2456 estimated_dts = (priv->next_in_seqnum != -1);
2458 /* take the DTS of the buffer. This is the time when the packet was
2459 * received and is used to calculate jitter and clock skew. We will adjust
2460 * this DTS with the smoothed value after processing it in the
2461 * jitterbuffer and assign it as the PTS. */
2462 /* bring to running time */
2463 dts = gst_segment_to_running_time (&priv->segment, GST_FORMAT_TIME, dts);
2466 GST_DEBUG_OBJECT (jitterbuffer,
2467 "Received packet #%d at time %" GST_TIME_FORMAT ", discont %d", seqnum,
2468 GST_TIME_ARGS (dts), GST_BUFFER_IS_DISCONT (buffer));
2470 JBUF_LOCK_CHECK (priv, out_flushing);
2472 if (G_UNLIKELY (priv->last_pt != pt)) {
2475 GST_DEBUG_OBJECT (jitterbuffer, "pt changed from %u to %u", priv->last_pt,
2479 /* reset clock-rate so that we get a new one */
2480 priv->clock_rate = -1;
2482 /* Try to get the clock-rate from the caps first if we can. If there are no
2483 * caps we must fire the signal to get the clock-rate. */
2484 if ((caps = gst_pad_get_current_caps (pad))) {
2485 gst_jitter_buffer_sink_parse_caps (jitterbuffer, caps);
2486 gst_caps_unref (caps);
2490 if (G_UNLIKELY (priv->clock_rate == -1)) {
2491 /* no clock rate given on the caps, try to get one with the signal */
2492 if (gst_rtp_jitter_buffer_get_clock_rate (jitterbuffer,
2493 pt) == GST_FLOW_FLUSHING)
2496 if (G_UNLIKELY (priv->clock_rate == -1))
2500 /* don't accept more data on EOS */
2501 if (G_UNLIKELY (priv->eos))
2504 calculate_jitter (jitterbuffer, dts, rtptime);
2506 if (priv->seqnum_base != -1) {
2509 gap = gst_rtp_buffer_compare_seqnum (priv->seqnum_base, seqnum);
2512 GST_DEBUG_OBJECT (jitterbuffer,
2513 "packet seqnum #%d before seqnum-base #%d", seqnum,
2515 gst_buffer_unref (buffer);
2518 } else if (gap > 16384) {
2519 /* From now on don't compare against the seqnum base anymore as
2520 * at some point in the future we will wrap around and also that
2521 * much reordering is very unlikely */
2522 priv->seqnum_base = -1;
2526 expected = priv->next_in_seqnum;
2529 gst_rtp_packet_rate_ctx_update (&priv->packet_rate_ctx, seqnum, rtptime);
2531 gst_rtp_packet_rate_ctx_get_max_dropout (&priv->packet_rate_ctx,
2532 priv->max_dropout_time);
2534 gst_rtp_packet_rate_ctx_get_max_misorder (&priv->packet_rate_ctx,
2535 priv->max_misorder_time);
2536 GST_TRACE_OBJECT (jitterbuffer,
2537 "packet_rate: %d, max_dropout: %d, max_misorder: %d", packet_rate,
2538 max_dropout, max_misorder);
2540 /* now check against our expected seqnum */
2541 if (G_LIKELY (expected != -1)) {
2544 /* now calculate gap */
2545 gap = gst_rtp_buffer_compare_seqnum (expected, seqnum);
2547 GST_DEBUG_OBJECT (jitterbuffer, "expected #%d, got #%d, gap of %d",
2548 expected, seqnum, gap);
2550 if (G_LIKELY (gap == 0)) {
2551 /* packet is expected */
2552 calculate_packet_spacing (jitterbuffer, rtptime, dts);
2553 do_next_seqnum = TRUE;
2555 gboolean reset = FALSE;
2558 /* we received an old packet */
2559 if (G_UNLIKELY (gap != -1 && gap < -max_misorder)) {
2561 handle_big_gap_buffer (jitterbuffer, FALSE, buffer, pt, seqnum,
2562 gap, max_dropout, max_misorder);
2565 GST_DEBUG_OBJECT (jitterbuffer, "old packet received");
2568 /* new packet, we are missing some packets */
2569 if (G_UNLIKELY (priv->timers->len >= max_dropout)) {
2570 /* If we have timers for more than RTP_MAX_DROPOUT packets
2571 * pending this means that we have a huge gap overall. We can
2572 * reset the jitterbuffer at this point because there's
2573 * just too much data missing to be able to do anything
2574 * sensible with the past data. Just try again from the
2576 GST_WARNING_OBJECT (jitterbuffer,
2577 "%d pending timers > %d - resetting", priv->timers->len,
2580 gst_buffer_unref (buffer);
2582 } else if (G_UNLIKELY (gap >= max_dropout)) {
2584 handle_big_gap_buffer (jitterbuffer, TRUE, buffer, pt, seqnum,
2585 gap, max_dropout, max_misorder);
2588 GST_DEBUG_OBJECT (jitterbuffer, "%d missing packets", gap);
2589 /* fill in the gap with EXPECTED timers */
2590 calculate_expected (jitterbuffer, expected, seqnum, dts, gap);
2592 do_next_seqnum = TRUE;
2595 if (G_UNLIKELY (reset)) {
2596 GList *events = NULL, *l;
2599 GST_DEBUG_OBJECT (jitterbuffer, "flush and reset jitterbuffer");
2600 rtp_jitter_buffer_flush (priv->jbuf,
2601 (GFunc) free_item_and_retain_events, &events);
2602 rtp_jitter_buffer_reset_skew (priv->jbuf);
2603 remove_all_timers (jitterbuffer);
2604 priv->discont = TRUE;
2605 priv->last_popped_seqnum = -1;
2607 if (priv->gap_packets.head) {
2608 GstBuffer *gap_buffer = priv->gap_packets.head->data;
2609 GstRTPBuffer gap_rtp = GST_RTP_BUFFER_INIT;
2611 gst_rtp_buffer_map (gap_buffer, GST_MAP_READ, &gap_rtp);
2612 priv->next_seqnum = gst_rtp_buffer_get_seq (&gap_rtp);
2613 gst_rtp_buffer_unmap (&gap_rtp);
2615 priv->next_seqnum = seqnum;
2618 priv->last_in_dts = -1;
2619 priv->next_in_seqnum = -1;
2621 /* Insert all sticky events again in order, otherwise we would
2622 * potentially loose STREAM_START, CAPS or SEGMENT events
2624 events = g_list_reverse (events);
2625 for (l = events; l; l = l->next) {
2626 RTPJitterBufferItem *item;
2628 item = alloc_item (l->data, ITEM_TYPE_EVENT, -1, -1, -1, 0, -1);
2629 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
2631 g_list_free (events);
2633 JBUF_SIGNAL_EVENT (priv);
2635 /* reset spacing estimation when gap */
2636 priv->ips_rtptime = -1;
2637 priv->ips_dts = GST_CLOCK_TIME_NONE;
2639 buffers = g_list_copy (priv->gap_packets.head);
2640 g_queue_clear (&priv->gap_packets);
2642 priv->ips_rtptime = -1;
2643 priv->ips_dts = GST_CLOCK_TIME_NONE;
2644 JBUF_UNLOCK (jitterbuffer->priv);
2646 for (l = buffers; l; l = l->next) {
2647 ret = gst_rtp_jitter_buffer_chain (pad, parent, l->data);
2649 if (ret != GST_FLOW_OK)
2652 for (; l; l = l->next)
2653 gst_buffer_unref (l->data);
2654 g_list_free (buffers);
2658 /* reset spacing estimation when gap */
2659 priv->ips_rtptime = -1;
2660 priv->ips_dts = GST_CLOCK_TIME_NONE;
2663 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
2665 /* we don't know what the next_in_seqnum should be, wait for the last
2666 * possible moment to push this buffer, maybe we get an earlier seqnum
2668 set_timer (jitterbuffer, TIMER_TYPE_DEADLINE, seqnum, dts);
2669 do_next_seqnum = TRUE;
2670 /* take rtptime and dts to calculate packet spacing */
2671 priv->ips_rtptime = rtptime;
2672 priv->ips_dts = dts;
2675 /* We had no huge gap, let's drop all the gap packets */
2676 if (buffer != NULL) {
2677 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets");
2678 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
2679 g_queue_clear (&priv->gap_packets);
2681 GST_DEBUG_OBJECT (jitterbuffer,
2682 "Had big gap, waiting for more consecutive packets");
2683 JBUF_UNLOCK (jitterbuffer->priv);
2687 if (do_next_seqnum) {
2688 priv->last_in_dts = dts;
2689 priv->next_in_seqnum = (seqnum + 1) & 0xffff;
2692 /* let's check if this buffer is too late, we can only accept packets with
2693 * bigger seqnum than the one we last pushed. */
2694 if (G_LIKELY (priv->last_popped_seqnum != -1)) {
2697 gap = gst_rtp_buffer_compare_seqnum (priv->last_popped_seqnum, seqnum);
2699 /* priv->last_popped_seqnum >= seqnum, we're too late. */
2700 if (G_UNLIKELY (gap <= 0))
2704 /* let's drop oldest packet if the queue is already full and drop-on-latency
2705 * is set. We can only do this when there actually is a latency. When no
2706 * latency is set, we just pump it in the queue and let the other end push it
2707 * out as fast as possible. */
2708 if (priv->latency_ms && priv->drop_on_latency) {
2710 gst_util_uint64_scale_int (priv->latency_ms, priv->clock_rate, 1000);
2712 if (G_UNLIKELY (rtp_jitter_buffer_get_ts_diff (priv->jbuf) >= latency_ts)) {
2713 RTPJitterBufferItem *old_item;
2715 old_item = rtp_jitter_buffer_peek (priv->jbuf);
2717 if (IS_DROPABLE (old_item)) {
2718 old_item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2719 GST_DEBUG_OBJECT (jitterbuffer, "Queue full, dropping old packet %p",
2721 priv->next_seqnum = (old_item->seqnum + 1) & 0xffff;
2722 free_item (old_item);
2724 /* we might have removed some head buffers, signal the pushing thread to
2725 * see if it can push now */
2726 JBUF_SIGNAL_EVENT (priv);
2730 /* If we estimated the DTS, don't consider it in the clock skew calculations
2731 * later. The code above always sets dts to pts or the other way around if
2732 * any of those is valid in the buffer, so we know that if we estimated the
2733 * dts that both are unknown */
2736 alloc_item (buffer, ITEM_TYPE_BUFFER, GST_CLOCK_TIME_NONE,
2737 GST_CLOCK_TIME_NONE, seqnum, 1, rtptime);
2739 item = alloc_item (buffer, ITEM_TYPE_BUFFER, dts, pts, seqnum, 1, rtptime);
2741 /* now insert the packet into the queue in sorted order. This function returns
2742 * FALSE if a packet with the same seqnum was already in the queue, meaning we
2743 * have a duplicate. */
2744 if (G_UNLIKELY (!rtp_jitter_buffer_insert (priv->jbuf, item,
2749 update_timers (jitterbuffer, seqnum, dts, do_next_seqnum);
2751 /* we had an unhandled SR, handle it now */
2753 do_handle_sync (jitterbuffer);
2755 if (G_UNLIKELY (head)) {
2756 /* signal addition of new buffer when the _loop is waiting. */
2757 if (G_LIKELY (priv->active))
2758 JBUF_SIGNAL_EVENT (priv);
2760 /* let's unschedule and unblock any waiting buffers. We only want to do this
2761 * when the head buffer changed */
2762 if (G_UNLIKELY (priv->clock_id)) {
2763 GST_DEBUG_OBJECT (jitterbuffer, "Unscheduling waiting new buffer");
2764 unschedule_current_timer (jitterbuffer);
2768 GST_DEBUG_OBJECT (jitterbuffer,
2769 "Pushed packet #%d, now %d packets, head: %d, " "percent %d", seqnum,
2770 rtp_jitter_buffer_num_packets (priv->jbuf), head, percent);
2772 msg = check_buffering_percent (jitterbuffer, percent);
2778 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
2785 /* this is not fatal but should be filtered earlier */
2786 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
2787 ("Received invalid RTP payload, dropping"));
2788 gst_buffer_unref (buffer);
2793 GST_WARNING_OBJECT (jitterbuffer,
2794 "No clock-rate in caps!, dropping buffer");
2795 gst_buffer_unref (buffer);
2800 ret = priv->srcresult;
2801 GST_DEBUG_OBJECT (jitterbuffer, "flushing %s", gst_flow_get_name (ret));
2802 gst_buffer_unref (buffer);
2808 GST_WARNING_OBJECT (jitterbuffer, "we are EOS, refusing buffer");
2809 gst_buffer_unref (buffer);
2814 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d too late as #%d was already"
2815 " popped, dropping", seqnum, priv->last_popped_seqnum);
2817 gst_buffer_unref (buffer);
2822 GST_DEBUG_OBJECT (jitterbuffer, "Duplicate packet #%d detected, dropping",
2824 priv->num_duplicates++;
2831 compute_elapsed (GstRtpJitterBuffer * jitterbuffer, RTPJitterBufferItem * item)
2833 guint64 ext_time, elapsed;
2835 GstRtpJitterBufferPrivate *priv;
2837 priv = jitterbuffer->priv;
2838 rtp_time = item->rtptime;
2840 GST_LOG_OBJECT (jitterbuffer, "rtp %" G_GUINT32_FORMAT ", ext %"
2841 G_GUINT64_FORMAT, rtp_time, priv->ext_timestamp);
2843 ext_time = priv->ext_timestamp;
2844 ext_time = gst_rtp_buffer_ext_timestamp (&ext_time, rtp_time);
2845 if (ext_time < priv->ext_timestamp) {
2846 ext_time = priv->ext_timestamp;
2848 priv->ext_timestamp = ext_time;
2851 if (ext_time > priv->clock_base)
2852 elapsed = ext_time - priv->clock_base;
2856 elapsed = gst_util_uint64_scale_int (elapsed, GST_SECOND, priv->clock_rate);
2861 update_estimated_eos (GstRtpJitterBuffer * jitterbuffer,
2862 RTPJitterBufferItem * item)
2864 guint64 total, elapsed, left, estimated;
2865 GstClockTime out_time;
2866 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2868 if (priv->npt_stop == -1 || priv->ext_timestamp == -1
2869 || priv->clock_base == -1 || priv->clock_rate <= 0)
2872 /* compute the elapsed time */
2873 elapsed = compute_elapsed (jitterbuffer, item);
2875 /* do nothing if elapsed time doesn't increment */
2876 if (priv->last_elapsed && elapsed <= priv->last_elapsed)
2879 priv->last_elapsed = elapsed;
2881 /* this is the total time we need to play */
2882 total = priv->npt_stop - priv->npt_start;
2883 GST_LOG_OBJECT (jitterbuffer, "total %" GST_TIME_FORMAT,
2884 GST_TIME_ARGS (total));
2886 /* this is how much time there is left */
2887 if (total > elapsed)
2888 left = total - elapsed;
2892 /* if we have less time left that the size of the buffer, we will not
2893 * be able to keep it filled, disabled buffering then */
2894 if (left < rtp_jitter_buffer_get_delay (priv->jbuf)) {
2895 GST_DEBUG_OBJECT (jitterbuffer, "left %" GST_TIME_FORMAT
2896 ", disable buffering close to EOS", GST_TIME_ARGS (left));
2897 rtp_jitter_buffer_disable_buffering (priv->jbuf, TRUE);
2900 /* this is the current time as running-time */
2901 out_time = item->dts;
2904 estimated = gst_util_uint64_scale (out_time, total, elapsed);
2906 /* if there is almost nothing left,
2907 * we may never advance enough to end up in the above case */
2908 if (total < GST_SECOND)
2909 estimated = GST_SECOND;
2913 GST_LOG_OBJECT (jitterbuffer, "elapsed %" GST_TIME_FORMAT ", estimated %"
2914 GST_TIME_FORMAT, GST_TIME_ARGS (elapsed), GST_TIME_ARGS (estimated));
2916 if (estimated != -1 && priv->estimated_eos != estimated) {
2917 set_timer (jitterbuffer, TIMER_TYPE_EOS, -1, estimated);
2918 priv->estimated_eos = estimated;
2922 /* take a buffer from the queue and push it */
2923 static GstFlowReturn
2924 pop_and_push_next (GstRtpJitterBuffer * jitterbuffer, guint seqnum)
2926 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
2927 GstFlowReturn result = GST_FLOW_OK;
2928 RTPJitterBufferItem *item;
2929 GstBuffer *outbuf = NULL;
2930 GstEvent *outevent = NULL;
2931 GstQuery *outquery = NULL;
2932 GstClockTime dts, pts;
2934 gboolean do_push = TRUE;
2938 /* when we get here we are ready to pop and push the buffer */
2939 item = rtp_jitter_buffer_pop (priv->jbuf, &percent);
2943 case ITEM_TYPE_BUFFER:
2945 /* we need to make writable to change the flags and timestamps */
2946 outbuf = gst_buffer_make_writable (item->data);
2948 if (G_UNLIKELY (priv->discont)) {
2949 /* set DISCONT flag when we missed a packet. We pushed the buffer writable
2950 * into the jitterbuffer so we can modify now. */
2951 GST_DEBUG_OBJECT (jitterbuffer, "mark output buffer discont");
2952 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
2953 priv->discont = FALSE;
2955 if (G_UNLIKELY (priv->ts_discont)) {
2956 GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_RESYNC);
2957 priv->ts_discont = FALSE;
2961 gst_segment_position_from_running_time (&priv->segment,
2962 GST_FORMAT_TIME, item->dts);
2964 gst_segment_position_from_running_time (&priv->segment,
2965 GST_FORMAT_TIME, item->pts);
2967 /* apply timestamp with offset to buffer now */
2968 GST_BUFFER_DTS (outbuf) = apply_offset (jitterbuffer, dts);
2969 GST_BUFFER_PTS (outbuf) = apply_offset (jitterbuffer, pts);
2971 /* update the elapsed time when we need to check against the npt stop time. */
2972 update_estimated_eos (jitterbuffer, item);
2974 priv->last_out_time = GST_BUFFER_PTS (outbuf);
2976 case ITEM_TYPE_LOST:
2977 priv->discont = TRUE;
2981 case ITEM_TYPE_EVENT:
2982 outevent = item->data;
2984 case ITEM_TYPE_QUERY:
2985 outquery = item->data;
2989 /* now we are ready to push the buffer. Save the seqnum and release the lock
2990 * so the other end can push stuff in the queue again. */
2992 priv->last_popped_seqnum = seqnum;
2993 priv->next_seqnum = (seqnum + item->count) & 0xffff;
2995 msg = check_buffering_percent (jitterbuffer, percent);
3002 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer), msg);
3005 case ITEM_TYPE_BUFFER:
3007 GST_DEBUG_OBJECT (jitterbuffer,
3008 "Pushing buffer %d, dts %" GST_TIME_FORMAT ", pts %" GST_TIME_FORMAT,
3009 seqnum, GST_TIME_ARGS (GST_BUFFER_DTS (outbuf)),
3010 GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
3011 result = gst_pad_push (priv->srcpad, outbuf);
3013 JBUF_LOCK_CHECK (priv, out_flushing);
3015 case ITEM_TYPE_LOST:
3016 case ITEM_TYPE_EVENT:
3017 /* We got not enough consecutive packets with a huge gap, we can
3018 * as well just drop them here now on EOS */
3019 if (GST_EVENT_TYPE (outevent) == GST_EVENT_EOS) {
3020 GST_DEBUG_OBJECT (jitterbuffer, "Clearing gap packets on EOS");
3021 g_queue_foreach (&priv->gap_packets, (GFunc) gst_buffer_unref, NULL);
3022 g_queue_clear (&priv->gap_packets);
3025 GST_DEBUG_OBJECT (jitterbuffer, "%sPushing event %" GST_PTR_FORMAT
3026 ", seqnum %d", do_push ? "" : "NOT ", outevent, seqnum);
3029 gst_pad_push_event (priv->srcpad, outevent);
3031 gst_event_unref (outevent);
3033 result = GST_FLOW_OK;
3035 JBUF_LOCK_CHECK (priv, out_flushing);
3037 case ITEM_TYPE_QUERY:
3041 res = gst_pad_peer_query (priv->srcpad, outquery);
3043 JBUF_LOCK_CHECK (priv, out_flushing);
3044 result = GST_FLOW_OK;
3045 GST_LOG_OBJECT (jitterbuffer, "did query %p, return %d", outquery, res);
3046 JBUF_SIGNAL_QUERY (priv, res);
3056 return priv->srcresult;
3060 #define GST_FLOW_WAIT GST_FLOW_CUSTOM_SUCCESS
3062 /* Peek a buffer and compare the seqnum to the expected seqnum.
3063 * If all is fine, the buffer is pushed.
3064 * If something is wrong, we wait for some event
3066 static GstFlowReturn
3067 handle_next_buffer (GstRtpJitterBuffer * jitterbuffer)
3069 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3070 GstFlowReturn result;
3071 RTPJitterBufferItem *item;
3073 guint32 next_seqnum;
3075 /* only push buffers when PLAYING and active and not buffering */
3076 if (priv->blocked || !priv->active ||
3077 rtp_jitter_buffer_is_buffering (priv->jbuf)) {
3078 return GST_FLOW_WAIT;
3081 /* peek a buffer, we're just looking at the sequence number.
3082 * If all is fine, we'll pop and push it. If the sequence number is wrong we
3083 * wait for a timeout or something to change.
3084 * The peeked buffer is valid for as long as we hold the jitterbuffer lock. */
3085 item = rtp_jitter_buffer_peek (priv->jbuf);
3090 /* get the seqnum and the next expected seqnum */
3091 seqnum = item->seqnum;
3093 return pop_and_push_next (jitterbuffer, seqnum);
3096 next_seqnum = priv->next_seqnum;
3098 /* get the gap between this and the previous packet. If we don't know the
3099 * previous packet seqnum assume no gap. */
3100 if (G_UNLIKELY (next_seqnum == -1)) {
3101 GST_DEBUG_OBJECT (jitterbuffer, "First buffer #%d", seqnum);
3102 /* we don't know what the next_seqnum should be, the chain function should
3103 * have scheduled a DEADLINE timer that will increment next_seqnum when it
3104 * fires, so wait for that */
3105 result = GST_FLOW_WAIT;
3107 gint gap = gst_rtp_buffer_compare_seqnum (next_seqnum, seqnum);
3109 if (G_LIKELY (gap == 0)) {
3110 /* no missing packet, pop and push */
3111 result = pop_and_push_next (jitterbuffer, seqnum);
3112 } else if (G_UNLIKELY (gap < 0)) {
3113 /* if we have a packet that we already pushed or considered dropped, pop it
3114 * off and get the next packet */
3115 GST_DEBUG_OBJECT (jitterbuffer, "Old packet #%d, next #%d dropping",
3116 seqnum, next_seqnum);
3117 item = rtp_jitter_buffer_pop (priv->jbuf, NULL);
3119 result = GST_FLOW_OK;
3121 /* the chain function has scheduled timers to request retransmission or
3122 * when to consider the packet lost, wait for that */
3123 GST_DEBUG_OBJECT (jitterbuffer,
3124 "Sequence number GAP detected: expected %d instead of %d (%d missing)",
3125 next_seqnum, seqnum, gap);
3126 result = GST_FLOW_WAIT;
3134 GST_DEBUG_OBJECT (jitterbuffer, "no buffer, going to wait");
3136 return GST_FLOW_EOS;
3138 return GST_FLOW_WAIT;
3144 get_rtx_retry_timeout (GstRtpJitterBufferPrivate * priv)
3146 GstClockTime rtx_retry_timeout;
3147 GstClockTime rtx_min_retry_timeout;
3149 if (priv->rtx_retry_timeout == -1) {
3150 if (priv->avg_rtx_rtt == 0)
3151 rtx_retry_timeout = DEFAULT_AUTO_RTX_TIMEOUT;
3153 /* we want to ask for a retransmission after we waited for a
3154 * complete RTT and the additional jitter */
3155 rtx_retry_timeout = priv->avg_rtx_rtt + priv->avg_jitter * 2;
3157 rtx_retry_timeout = priv->rtx_retry_timeout * GST_MSECOND;
3159 /* make sure we don't retry too often. On very low latency networks,
3160 * the RTT and jitter can be very low. */
3161 if (priv->rtx_min_retry_timeout == -1) {
3162 rtx_min_retry_timeout = priv->packet_spacing;
3164 rtx_min_retry_timeout = priv->rtx_min_retry_timeout * GST_MSECOND;
3166 rtx_retry_timeout = MAX (rtx_retry_timeout, rtx_min_retry_timeout);
3168 return rtx_retry_timeout;
3172 get_rtx_retry_period (GstRtpJitterBufferPrivate * priv,
3173 GstClockTime rtx_retry_timeout)
3175 GstClockTime rtx_retry_period;
3177 if (priv->rtx_retry_period == -1) {
3178 /* we retry up to the configured jitterbuffer size but leaving some
3179 * room for the retransmission to arrive in time */
3180 if (rtx_retry_timeout > priv->latency_ns) {
3181 rtx_retry_period = 0;
3183 rtx_retry_period = priv->latency_ns - rtx_retry_timeout;
3186 rtx_retry_period = priv->rtx_retry_period * GST_MSECOND;
3188 return rtx_retry_period;
3191 /* the timeout for when we expected a packet expired */
3193 do_expected_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3196 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3198 guint delay, delay_ms, avg_rtx_rtt_ms;
3199 guint rtx_retry_timeout_ms, rtx_retry_period_ms;
3200 GstClockTime rtx_retry_period;
3201 GstClockTime rtx_retry_timeout;
3204 GST_DEBUG_OBJECT (jitterbuffer, "expected %d didn't arrive, now %"
3205 GST_TIME_FORMAT, timer->seqnum, GST_TIME_ARGS (now));
3207 rtx_retry_timeout = get_rtx_retry_timeout (priv);
3208 rtx_retry_period = get_rtx_retry_period (priv, rtx_retry_timeout);
3210 GST_DEBUG_OBJECT (jitterbuffer, "timeout %" GST_TIME_FORMAT ", period %"
3211 GST_TIME_FORMAT, GST_TIME_ARGS (rtx_retry_timeout),
3212 GST_TIME_ARGS (rtx_retry_period));
3214 delay = timer->rtx_delay + timer->rtx_retry;
3216 delay_ms = GST_TIME_AS_MSECONDS (delay);
3217 rtx_retry_timeout_ms = GST_TIME_AS_MSECONDS (rtx_retry_timeout);
3218 rtx_retry_period_ms = GST_TIME_AS_MSECONDS (rtx_retry_period);
3219 avg_rtx_rtt_ms = GST_TIME_AS_MSECONDS (priv->avg_rtx_rtt);
3221 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
3222 gst_structure_new ("GstRTPRetransmissionRequest",
3223 "seqnum", G_TYPE_UINT, (guint) timer->seqnum,
3224 "running-time", G_TYPE_UINT64, timer->rtx_base,
3225 "delay", G_TYPE_UINT, delay_ms,
3226 "retry", G_TYPE_UINT, timer->num_rtx_retry,
3227 "frequency", G_TYPE_UINT, rtx_retry_timeout_ms,
3228 "period", G_TYPE_UINT, rtx_retry_period_ms,
3229 "deadline", G_TYPE_UINT, priv->latency_ms,
3230 "packet-spacing", G_TYPE_UINT64, priv->packet_spacing,
3231 "avg-rtt", G_TYPE_UINT, avg_rtx_rtt_ms, NULL));
3233 priv->num_rtx_requests++;
3234 timer->num_rtx_retry++;
3236 GST_OBJECT_LOCK (jitterbuffer);
3237 if ((clock = GST_ELEMENT_CLOCK (jitterbuffer))) {
3238 timer->rtx_last = gst_clock_get_time (clock);
3239 timer->rtx_last -= GST_ELEMENT_CAST (jitterbuffer)->base_time;
3241 timer->rtx_last = now;
3243 GST_OBJECT_UNLOCK (jitterbuffer);
3245 /* calculate the timeout for the next retransmission attempt */
3246 timer->rtx_retry += rtx_retry_timeout;
3247 GST_DEBUG_OBJECT (jitterbuffer, "base %" GST_TIME_FORMAT ", delay %"
3248 GST_TIME_FORMAT ", retry %" GST_TIME_FORMAT ", num_retry %u",
3249 GST_TIME_ARGS (timer->rtx_base), GST_TIME_ARGS (timer->rtx_delay),
3250 GST_TIME_ARGS (timer->rtx_retry), timer->num_rtx_retry);
3251 if ((priv->rtx_max_retries != -1
3252 && timer->num_rtx_retry >= priv->rtx_max_retries)
3253 || (timer->rtx_retry + timer->rtx_delay > rtx_retry_period)) {
3254 GST_DEBUG_OBJECT (jitterbuffer, "reschedule as LOST timer");
3255 /* too many retransmission request, we now convert the timer
3256 * to a lost timer, leave the num_rtx_retry as it is for stats */
3257 timer->type = TIMER_TYPE_LOST;
3258 timer->rtx_delay = 0;
3259 timer->rtx_retry = 0;
3261 reschedule_timer (jitterbuffer, timer, timer->seqnum,
3262 timer->rtx_base + timer->rtx_retry, timer->rtx_delay, FALSE);
3265 gst_pad_push_event (priv->sinkpad, event);
3271 /* a packet is lost */
3273 do_lost_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3276 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3277 GstClockTime duration, timestamp;
3278 guint seqnum, lost_packets, num_rtx_retry, next_in_seqnum;
3281 RTPJitterBufferItem *item;
3283 seqnum = timer->seqnum;
3284 timestamp = apply_offset (jitterbuffer, timer->timeout);
3285 duration = timer->duration;
3286 if (duration == GST_CLOCK_TIME_NONE && priv->packet_spacing > 0)
3287 duration = priv->packet_spacing;
3288 lost_packets = MAX (timer->num, 1);
3289 num_rtx_retry = timer->num_rtx_retry;
3291 /* we had a gap and thus we lost some packets. Create an event for this. */
3292 if (lost_packets > 1)
3293 GST_DEBUG_OBJECT (jitterbuffer, "Packets #%d -> #%d lost", seqnum,
3294 seqnum + lost_packets - 1);
3296 GST_DEBUG_OBJECT (jitterbuffer, "Packet #%d lost", seqnum);
3298 priv->num_late += lost_packets;
3299 priv->num_rtx_failed += num_rtx_retry;
3301 next_in_seqnum = (seqnum + lost_packets) & 0xffff;
3303 /* we now only accept seqnum bigger than this */
3304 if (gst_rtp_buffer_compare_seqnum (priv->next_in_seqnum, next_in_seqnum) > 0)
3305 priv->next_in_seqnum = next_in_seqnum;
3307 /* create paket lost event */
3308 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
3309 gst_structure_new ("GstRTPPacketLost",
3310 "seqnum", G_TYPE_UINT, (guint) seqnum,
3311 "timestamp", G_TYPE_UINT64, timestamp,
3312 "duration", G_TYPE_UINT64, duration,
3313 "retry", G_TYPE_UINT, num_rtx_retry, NULL));
3315 item = alloc_item (event, ITEM_TYPE_LOST, -1, -1, seqnum, lost_packets, -1);
3316 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3318 /* remove timer now */
3319 remove_timer (jitterbuffer, timer);
3321 JBUF_SIGNAL_EVENT (priv);
3327 do_eos_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3330 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3332 GST_INFO_OBJECT (jitterbuffer, "got the NPT timeout");
3333 remove_timer (jitterbuffer, timer);
3335 /* there was no EOS in the buffer, put one in there now */
3336 queue_event (jitterbuffer, gst_event_new_eos ());
3338 JBUF_SIGNAL_EVENT (priv);
3344 do_deadline_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3347 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3349 GST_INFO_OBJECT (jitterbuffer, "got deadline timeout");
3351 /* timer seqnum might have been obsoleted by caps seqnum-base,
3352 * only mess with current ongoing seqnum if still unknown */
3353 if (priv->next_seqnum == -1)
3354 priv->next_seqnum = timer->seqnum;
3355 remove_timer (jitterbuffer, timer);
3356 JBUF_SIGNAL_EVENT (priv);
3362 do_timeout (GstRtpJitterBuffer * jitterbuffer, TimerData * timer,
3365 gboolean removed = FALSE;
3367 switch (timer->type) {
3368 case TIMER_TYPE_EXPECTED:
3369 removed = do_expected_timeout (jitterbuffer, timer, now);
3371 case TIMER_TYPE_LOST:
3372 removed = do_lost_timeout (jitterbuffer, timer, now);
3374 case TIMER_TYPE_DEADLINE:
3375 removed = do_deadline_timeout (jitterbuffer, timer, now);
3377 case TIMER_TYPE_EOS:
3378 removed = do_eos_timeout (jitterbuffer, timer, now);
3384 /* called when we need to wait for the next timeout.
3386 * We loop over the array of recorded timeouts and wait for the earliest one.
3387 * When it timed out, do the logic associated with the timer.
3389 * If there are no timers, we wait on a gcond until something new happens.
3392 wait_next_timeout (GstRtpJitterBuffer * jitterbuffer)
3394 GstRtpJitterBufferPrivate *priv = jitterbuffer->priv;
3395 GstClockTime now = 0;
3398 while (priv->timer_running) {
3399 TimerData *timer = NULL;
3400 GstClockTime timer_timeout = -1;
3403 /* If we have a clock, update "now" now with the very
3404 * latest running time we have. If timers are unscheduled below we
3405 * otherwise wouldn't update now (it's only updated when timers
3406 * expire), and also for the very first loop iteration now would
3407 * otherwise always be 0
3409 GST_OBJECT_LOCK (jitterbuffer);
3410 if (GST_ELEMENT_CLOCK (jitterbuffer)) {
3412 gst_clock_get_time (GST_ELEMENT_CLOCK (jitterbuffer)) -
3413 GST_ELEMENT_CAST (jitterbuffer)->base_time;
3415 GST_OBJECT_UNLOCK (jitterbuffer);
3417 GST_DEBUG_OBJECT (jitterbuffer, "now %" GST_TIME_FORMAT,
3418 GST_TIME_ARGS (now));
3420 len = priv->timers->len;
3421 for (i = 0; i < len; i++) {
3422 TimerData *test = &g_array_index (priv->timers, TimerData, i);
3423 GstClockTime test_timeout = get_timeout (jitterbuffer, test);
3424 gboolean save_best = FALSE;
3426 GST_DEBUG_OBJECT (jitterbuffer, "%d, %d, %d, %" GST_TIME_FORMAT,
3427 i, test->type, test->seqnum, GST_TIME_ARGS (test_timeout));
3429 /* find the smallest timeout */
3430 if (timer == NULL) {
3432 } else if (timer_timeout == -1) {
3433 /* we already have an immediate timeout, the new timer must be an
3434 * immediate timer with smaller seqnum to become the best */
3435 if (test_timeout == -1
3436 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3437 timer->seqnum) > 0))
3439 } else if (test_timeout == -1) {
3440 /* first immediate timer */
3442 } else if (test_timeout < timer_timeout) {
3445 } else if (test_timeout == timer_timeout
3446 && (gst_rtp_buffer_compare_seqnum (test->seqnum,
3447 timer->seqnum) > 0)) {
3448 /* same timer, smaller seqnum */
3452 GST_DEBUG_OBJECT (jitterbuffer, "new best %d", i);
3454 timer_timeout = test_timeout;
3457 if (timer && !priv->blocked) {
3459 GstClockTime sync_time;
3462 GstClockTimeDiff clock_jitter;
3464 if (timer_timeout == -1 || timer_timeout <= now) {
3465 do_timeout (jitterbuffer, timer, now);
3466 /* check here, do_timeout could have released the lock */
3467 if (!priv->timer_running)
3472 GST_OBJECT_LOCK (jitterbuffer);
3473 clock = GST_ELEMENT_CLOCK (jitterbuffer);
3475 GST_OBJECT_UNLOCK (jitterbuffer);
3476 /* let's just push if there is no clock */
3477 GST_DEBUG_OBJECT (jitterbuffer, "No clock, timeout right away");
3478 now = timer_timeout;
3482 /* prepare for sync against clock */
3483 sync_time = timer_timeout + GST_ELEMENT_CAST (jitterbuffer)->base_time;
3484 /* add latency of peer to get input time */
3485 sync_time += priv->peer_latency;
3487 GST_DEBUG_OBJECT (jitterbuffer, "sync to timestamp %" GST_TIME_FORMAT
3488 " with sync time %" GST_TIME_FORMAT,
3489 GST_TIME_ARGS (timer_timeout), GST_TIME_ARGS (sync_time));
3491 /* create an entry for the clock */
3492 id = priv->clock_id = gst_clock_new_single_shot_id (clock, sync_time);
3493 priv->timer_timeout = timer_timeout;
3494 priv->timer_seqnum = timer->seqnum;
3495 GST_OBJECT_UNLOCK (jitterbuffer);
3497 /* release the lock so that the other end can push stuff or unlock */
3500 ret = gst_clock_id_wait (id, &clock_jitter);
3503 if (!priv->timer_running) {
3504 gst_clock_id_unref (id);
3505 priv->clock_id = NULL;
3509 if (ret != GST_CLOCK_UNSCHEDULED) {
3510 now = timer_timeout + MAX (clock_jitter, 0);
3511 GST_DEBUG_OBJECT (jitterbuffer,
3512 "sync done, %d, #%d, %" GST_STIME_FORMAT, ret, priv->timer_seqnum,
3513 GST_STIME_ARGS (clock_jitter));
3515 GST_DEBUG_OBJECT (jitterbuffer, "sync unscheduled");
3517 /* and free the entry */
3518 gst_clock_id_unref (id);
3519 priv->clock_id = NULL;
3521 /* no timers, wait for activity */
3522 JBUF_WAIT_TIMER (priv);
3527 GST_DEBUG_OBJECT (jitterbuffer, "we are stopping");
3532 * This funcion implements the main pushing loop on the source pad.
3534 * It first tries to push as many buffers as possible. If there is a seqnum
3535 * mismatch, we wait for the next timeouts.
3538 gst_rtp_jitter_buffer_loop (GstRtpJitterBuffer * jitterbuffer)
3540 GstRtpJitterBufferPrivate *priv;
3541 GstFlowReturn result = GST_FLOW_OK;
3543 priv = jitterbuffer->priv;
3545 JBUF_LOCK_CHECK (priv, flushing);
3547 result = handle_next_buffer (jitterbuffer);
3548 if (G_LIKELY (result == GST_FLOW_WAIT)) {
3549 /* now wait for the next event */
3550 JBUF_WAIT_EVENT (priv, flushing);
3551 result = GST_FLOW_OK;
3553 } while (result == GST_FLOW_OK);
3554 /* store result for upstream */
3555 priv->srcresult = result;
3556 /* if we get here we need to pause */
3562 result = priv->srcresult;
3569 JBUF_SIGNAL_QUERY (priv, FALSE);
3572 GST_DEBUG_OBJECT (jitterbuffer, "pausing task, reason %s",
3573 gst_flow_get_name (result));
3574 gst_pad_pause_task (priv->srcpad);
3575 if (result == GST_FLOW_EOS) {
3576 event = gst_event_new_eos ();
3577 gst_pad_push_event (priv->srcpad, event);
3583 /* collect the info from the lastest RTCP packet and the jitterbuffer sync, do
3584 * some sanity checks and then emit the handle-sync signal with the parameters.
3585 * This function must be called with the LOCK */
3587 do_handle_sync (GstRtpJitterBuffer * jitterbuffer)
3589 GstRtpJitterBufferPrivate *priv;
3590 guint64 base_rtptime, base_time;
3592 guint64 last_rtptime;
3594 guint64 ext_rtptime, diff;
3595 gboolean valid = TRUE, keep = FALSE;
3597 priv = jitterbuffer->priv;
3599 /* get the last values from the jitterbuffer */
3600 rtp_jitter_buffer_get_sync (priv->jbuf, &base_rtptime, &base_time,
3601 &clock_rate, &last_rtptime);
3603 clock_base = priv->clock_base;
3604 ext_rtptime = priv->ext_rtptime;
3606 GST_DEBUG_OBJECT (jitterbuffer, "ext SR %" G_GUINT64_FORMAT ", base %"
3607 G_GUINT64_FORMAT ", clock-rate %" G_GUINT32_FORMAT
3608 ", clock-base %" G_GUINT64_FORMAT ", last-rtptime %" G_GUINT64_FORMAT,
3609 ext_rtptime, base_rtptime, clock_rate, clock_base, last_rtptime);
3611 if (base_rtptime == -1 || clock_rate == -1 || base_time == -1) {
3612 /* we keep this SR packet for later. When we get a valid RTP packet the
3613 * above values will be set and we can try to use the SR packet */
3614 GST_DEBUG_OBJECT (jitterbuffer, "keeping for later, no RTP values");
3617 /* we can't accept anything that happened before we did the last resync */
3618 if (base_rtptime > ext_rtptime) {
3619 GST_DEBUG_OBJECT (jitterbuffer, "dropping, older than base time");
3622 /* the SR RTP timestamp must be something close to what we last observed
3623 * in the jitterbuffer */
3624 if (ext_rtptime > last_rtptime) {
3625 /* check how far ahead it is to our RTP timestamps */
3626 diff = ext_rtptime - last_rtptime;
3627 /* if bigger than 1 second, we drop it */
3628 if (jitterbuffer->priv->max_rtcp_rtp_time_diff != -1 &&
3630 gst_util_uint64_scale (jitterbuffer->priv->max_rtcp_rtp_time_diff,
3631 clock_rate, 1000)) {
3632 GST_DEBUG_OBJECT (jitterbuffer, "too far ahead");
3633 /* should drop this, but some RTSP servers end up with bogus
3634 * way too ahead RTCP packet when repeated PAUSE/PLAY,
3635 * so still trigger rptbin sync but invalidate RTCP data
3636 * (sync might use other methods) */
3639 GST_DEBUG_OBJECT (jitterbuffer, "ext last %" G_GUINT64_FORMAT ", diff %"
3640 G_GUINT64_FORMAT, last_rtptime, diff);
3646 GST_DEBUG_OBJECT (jitterbuffer, "keeping RTCP packet for later");
3650 s = gst_structure_new ("application/x-rtp-sync",
3651 "base-rtptime", G_TYPE_UINT64, base_rtptime,
3652 "base-time", G_TYPE_UINT64, base_time,
3653 "clock-rate", G_TYPE_UINT, clock_rate,
3654 "clock-base", G_TYPE_UINT64, clock_base,
3655 "sr-ext-rtptime", G_TYPE_UINT64, ext_rtptime,
3656 "sr-buffer", GST_TYPE_BUFFER, priv->last_sr, NULL);
3658 GST_DEBUG_OBJECT (jitterbuffer, "signaling sync");
3659 gst_buffer_replace (&priv->last_sr, NULL);
3661 g_signal_emit (jitterbuffer,
3662 gst_rtp_jitter_buffer_signals[SIGNAL_HANDLE_SYNC], 0, s);
3664 gst_structure_free (s);
3666 GST_DEBUG_OBJECT (jitterbuffer, "dropping RTCP packet");
3667 gst_buffer_replace (&priv->last_sr, NULL);
3671 static GstFlowReturn
3672 gst_rtp_jitter_buffer_chain_rtcp (GstPad * pad, GstObject * parent,
3675 GstRtpJitterBuffer *jitterbuffer;
3676 GstRtpJitterBufferPrivate *priv;
3677 GstFlowReturn ret = GST_FLOW_OK;
3679 GstRTCPPacket packet;
3680 guint64 ext_rtptime;
3682 GstRTCPBuffer rtcp = { NULL, };
3684 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3686 if (G_UNLIKELY (!gst_rtcp_buffer_validate_reduced (buffer)))
3687 goto invalid_buffer;
3689 priv = jitterbuffer->priv;
3691 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
3693 if (!gst_rtcp_buffer_get_first_packet (&rtcp, &packet))
3696 /* first packet must be SR or RR or else the validate would have failed */
3697 switch (gst_rtcp_packet_get_type (&packet)) {
3698 case GST_RTCP_TYPE_SR:
3699 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, &rtptime,
3705 gst_rtcp_buffer_unmap (&rtcp);
3707 GST_DEBUG_OBJECT (jitterbuffer, "received RTCP of SSRC %08x", ssrc);
3710 /* convert the RTP timestamp to our extended timestamp, using the same offset
3711 * we used in the jitterbuffer */
3712 ext_rtptime = priv->jbuf->ext_rtptime;
3713 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
3715 priv->ext_rtptime = ext_rtptime;
3716 gst_buffer_replace (&priv->last_sr, buffer);
3718 do_handle_sync (jitterbuffer);
3722 gst_buffer_unref (buffer);
3728 /* this is not fatal but should be filtered earlier */
3729 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3730 ("Received invalid RTCP payload, dropping"));
3736 /* this is not fatal but should be filtered earlier */
3737 GST_ELEMENT_WARNING (jitterbuffer, STREAM, DECODE, (NULL),
3738 ("Received empty RTCP payload, dropping"));
3739 gst_rtcp_buffer_unmap (&rtcp);
3745 GST_DEBUG_OBJECT (jitterbuffer, "ignoring RTCP packet");
3746 gst_rtcp_buffer_unmap (&rtcp);
3753 gst_rtp_jitter_buffer_sink_query (GstPad * pad, GstObject * parent,
3756 gboolean res = FALSE;
3757 GstRtpJitterBuffer *jitterbuffer;
3758 GstRtpJitterBufferPrivate *priv;
3760 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3761 priv = jitterbuffer->priv;
3763 switch (GST_QUERY_TYPE (query)) {
3764 case GST_QUERY_CAPS:
3766 GstCaps *filter, *caps;
3768 gst_query_parse_caps (query, &filter);
3769 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3770 gst_query_set_caps_result (query, caps);
3771 gst_caps_unref (caps);
3776 if (GST_QUERY_IS_SERIALIZED (query)) {
3777 RTPJitterBufferItem *item;
3780 JBUF_LOCK_CHECK (priv, out_flushing);
3781 if (rtp_jitter_buffer_get_mode (priv->jbuf) !=
3782 RTP_JITTER_BUFFER_MODE_BUFFER) {
3783 GST_DEBUG_OBJECT (jitterbuffer, "adding serialized query");
3784 item = alloc_item (query, ITEM_TYPE_QUERY, -1, -1, -1, 0, -1);
3785 rtp_jitter_buffer_insert (priv->jbuf, item, &head, NULL);
3787 JBUF_SIGNAL_EVENT (priv);
3788 JBUF_WAIT_QUERY (priv, out_flushing);
3789 res = priv->last_query;
3791 GST_DEBUG_OBJECT (jitterbuffer, "refusing query, we are buffering");
3796 res = gst_pad_query_default (pad, parent, query);
3804 GST_DEBUG_OBJECT (jitterbuffer, "we are flushing");
3812 gst_rtp_jitter_buffer_src_query (GstPad * pad, GstObject * parent,
3815 GstRtpJitterBuffer *jitterbuffer;
3816 GstRtpJitterBufferPrivate *priv;
3817 gboolean res = FALSE;
3819 jitterbuffer = GST_RTP_JITTER_BUFFER (parent);
3820 priv = jitterbuffer->priv;
3822 switch (GST_QUERY_TYPE (query)) {
3823 case GST_QUERY_LATENCY:
3825 /* We need to send the query upstream and add the returned latency to our
3827 GstClockTime min_latency, max_latency;
3829 GstClockTime our_latency;
3831 if ((res = gst_pad_peer_query (priv->sinkpad, query))) {
3832 gst_query_parse_latency (query, &us_live, &min_latency, &max_latency);
3834 GST_DEBUG_OBJECT (jitterbuffer, "Peer latency: min %"
3835 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3836 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3838 /* store this so that we can safely sync on the peer buffers. */
3840 priv->peer_latency = min_latency;
3841 our_latency = priv->latency_ns;
3844 GST_DEBUG_OBJECT (jitterbuffer, "Our latency: %" GST_TIME_FORMAT,
3845 GST_TIME_ARGS (our_latency));
3847 /* we add some latency but can buffer an infinite amount of time */
3848 min_latency += our_latency;
3851 GST_DEBUG_OBJECT (jitterbuffer, "Calculated total latency : min %"
3852 GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
3853 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
3855 gst_query_set_latency (query, TRUE, min_latency, max_latency);
3859 case GST_QUERY_POSITION:
3861 GstClockTime start, last_out;
3864 gst_query_parse_position (query, &fmt, NULL);
3865 if (fmt != GST_FORMAT_TIME) {
3866 res = gst_pad_query_default (pad, parent, query);
3871 start = priv->npt_start;
3872 last_out = priv->last_out_time;
3875 GST_DEBUG_OBJECT (jitterbuffer, "npt start %" GST_TIME_FORMAT
3876 ", last out %" GST_TIME_FORMAT, GST_TIME_ARGS (start),
3877 GST_TIME_ARGS (last_out));
3879 if (GST_CLOCK_TIME_IS_VALID (start) && GST_CLOCK_TIME_IS_VALID (last_out)) {
3880 /* bring 0-based outgoing time to stream time */
3881 gst_query_set_position (query, GST_FORMAT_TIME, start + last_out);
3884 res = gst_pad_query_default (pad, parent, query);
3888 case GST_QUERY_CAPS:
3890 GstCaps *filter, *caps;
3892 gst_query_parse_caps (query, &filter);
3893 caps = gst_rtp_jitter_buffer_getcaps (pad, filter);
3894 gst_query_set_caps_result (query, caps);
3895 gst_caps_unref (caps);
3900 res = gst_pad_query_default (pad, parent, query);
3908 gst_rtp_jitter_buffer_set_property (GObject * object,
3909 guint prop_id, const GValue * value, GParamSpec * pspec)
3911 GstRtpJitterBuffer *jitterbuffer;
3912 GstRtpJitterBufferPrivate *priv;
3914 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
3915 priv = jitterbuffer->priv;
3920 guint new_latency, old_latency;
3922 new_latency = g_value_get_uint (value);
3925 old_latency = priv->latency_ms;
3926 priv->latency_ms = new_latency;
3927 priv->latency_ns = priv->latency_ms * GST_MSECOND;
3928 rtp_jitter_buffer_set_delay (priv->jbuf, priv->latency_ns);
3931 /* post message if latency changed, this will inform the parent pipeline
3932 * that a latency reconfiguration is possible/needed. */
3933 if (new_latency != old_latency) {
3934 GST_DEBUG_OBJECT (jitterbuffer, "latency changed to: %" GST_TIME_FORMAT,
3935 GST_TIME_ARGS (new_latency * GST_MSECOND));
3937 gst_element_post_message (GST_ELEMENT_CAST (jitterbuffer),
3938 gst_message_new_latency (GST_OBJECT_CAST (jitterbuffer)));
3942 case PROP_DROP_ON_LATENCY:
3944 priv->drop_on_latency = g_value_get_boolean (value);
3947 case PROP_TS_OFFSET:
3949 priv->ts_offset = g_value_get_int64 (value);
3950 priv->ts_discont = TRUE;
3955 priv->do_lost = g_value_get_boolean (value);
3960 rtp_jitter_buffer_set_mode (priv->jbuf, g_value_get_enum (value));
3963 case PROP_DO_RETRANSMISSION:
3965 priv->do_retransmission = g_value_get_boolean (value);
3968 case PROP_RTX_NEXT_SEQNUM:
3970 priv->rtx_next_seqnum = g_value_get_boolean (value);
3973 case PROP_RTX_DELAY:
3975 priv->rtx_delay = g_value_get_int (value);
3978 case PROP_RTX_MIN_DELAY:
3980 priv->rtx_min_delay = g_value_get_uint (value);
3983 case PROP_RTX_DELAY_REORDER:
3985 priv->rtx_delay_reorder = g_value_get_int (value);
3988 case PROP_RTX_RETRY_TIMEOUT:
3990 priv->rtx_retry_timeout = g_value_get_int (value);
3993 case PROP_RTX_MIN_RETRY_TIMEOUT:
3995 priv->rtx_min_retry_timeout = g_value_get_int (value);
3998 case PROP_RTX_RETRY_PERIOD:
4000 priv->rtx_retry_period = g_value_get_int (value);
4003 case PROP_RTX_MAX_RETRIES:
4005 priv->rtx_max_retries = g_value_get_int (value);
4008 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4010 priv->max_rtcp_rtp_time_diff = g_value_get_int (value);
4013 case PROP_MAX_DROPOUT_TIME:
4015 priv->max_dropout_time = g_value_get_uint (value);
4018 case PROP_MAX_MISORDER_TIME:
4020 priv->max_misorder_time = g_value_get_uint (value);
4024 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4030 gst_rtp_jitter_buffer_get_property (GObject * object,
4031 guint prop_id, GValue * value, GParamSpec * pspec)
4033 GstRtpJitterBuffer *jitterbuffer;
4034 GstRtpJitterBufferPrivate *priv;
4036 jitterbuffer = GST_RTP_JITTER_BUFFER (object);
4037 priv = jitterbuffer->priv;
4042 g_value_set_uint (value, priv->latency_ms);
4045 case PROP_DROP_ON_LATENCY:
4047 g_value_set_boolean (value, priv->drop_on_latency);
4050 case PROP_TS_OFFSET:
4052 g_value_set_int64 (value, priv->ts_offset);
4057 g_value_set_boolean (value, priv->do_lost);
4062 g_value_set_enum (value, rtp_jitter_buffer_get_mode (priv->jbuf));
4070 if (priv->srcresult != GST_FLOW_OK)
4073 percent = rtp_jitter_buffer_get_percent (priv->jbuf);
4075 g_value_set_int (value, percent);
4079 case PROP_DO_RETRANSMISSION:
4081 g_value_set_boolean (value, priv->do_retransmission);
4084 case PROP_RTX_NEXT_SEQNUM:
4086 g_value_set_boolean (value, priv->rtx_next_seqnum);
4089 case PROP_RTX_DELAY:
4091 g_value_set_int (value, priv->rtx_delay);
4094 case PROP_RTX_MIN_DELAY:
4096 g_value_set_uint (value, priv->rtx_min_delay);
4099 case PROP_RTX_DELAY_REORDER:
4101 g_value_set_int (value, priv->rtx_delay_reorder);
4104 case PROP_RTX_RETRY_TIMEOUT:
4106 g_value_set_int (value, priv->rtx_retry_timeout);
4109 case PROP_RTX_MIN_RETRY_TIMEOUT:
4111 g_value_set_int (value, priv->rtx_min_retry_timeout);
4114 case PROP_RTX_RETRY_PERIOD:
4116 g_value_set_int (value, priv->rtx_retry_period);
4119 case PROP_RTX_MAX_RETRIES:
4121 g_value_set_int (value, priv->rtx_max_retries);
4125 g_value_take_boxed (value,
4126 gst_rtp_jitter_buffer_create_stats (jitterbuffer));
4128 case PROP_MAX_RTCP_RTP_TIME_DIFF:
4130 g_value_set_int (value, priv->max_rtcp_rtp_time_diff);
4133 case PROP_MAX_DROPOUT_TIME:
4135 g_value_set_uint (value, priv->max_dropout_time);
4138 case PROP_MAX_MISORDER_TIME:
4140 g_value_set_uint (value, priv->max_misorder_time);
4144 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
4149 static GstStructure *
4150 gst_rtp_jitter_buffer_create_stats (GstRtpJitterBuffer * jbuf)
4154 JBUF_LOCK (jbuf->priv);
4155 s = gst_structure_new ("application/x-rtp-jitterbuffer-stats",
4156 "rtx-count", G_TYPE_UINT64, jbuf->priv->num_rtx_requests,
4157 "rtx-success-count", G_TYPE_UINT64, jbuf->priv->num_rtx_success,
4158 "rtx-per-packet", G_TYPE_DOUBLE, jbuf->priv->avg_rtx_num,
4159 "rtx-rtt", G_TYPE_UINT64, jbuf->priv->avg_rtx_rtt, NULL);
4160 JBUF_UNLOCK (jbuf->priv);