2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
22 #include <gst/rtp/gstrtpbuffer.h>
23 #include <gst/rtp/gstrtcpbuffer.h>
25 #include "rtpjitterbuffer.h"
27 GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug);
28 #define GST_CAT_DEFAULT rtp_jitter_buffer_debug
30 #define MAX_WINDOW RTP_JITTER_BUFFER_MAX_WINDOW
31 #define MAX_TIME (2 * GST_SECOND)
33 /* signals and args */
44 /* GObject vmethods */
45 static void rtp_jitter_buffer_finalize (GObject * object);
48 rtp_jitter_buffer_mode_get_type (void)
50 static GType jitter_buffer_mode_type = 0;
51 static const GEnumValue jitter_buffer_modes[] = {
52 {RTP_JITTER_BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
53 {RTP_JITTER_BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
54 {RTP_JITTER_BUFFER_MODE_BUFFER, "Do low/high watermark buffering",
59 if (!jitter_buffer_mode_type) {
60 jitter_buffer_mode_type =
61 g_enum_register_static ("RTPJitterBufferMode", jitter_buffer_modes);
63 return jitter_buffer_mode_type;
66 /* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
68 G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
71 rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
73 GObjectClass *gobject_class;
75 gobject_class = (GObjectClass *) klass;
77 gobject_class->finalize = rtp_jitter_buffer_finalize;
79 GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
84 rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
86 jbuf->packets = g_queue_new ();
87 jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
89 rtp_jitter_buffer_reset_skew (jbuf);
93 rtp_jitter_buffer_finalize (GObject * object)
95 RTPJitterBuffer *jbuf;
97 jbuf = RTP_JITTER_BUFFER_CAST (object);
99 rtp_jitter_buffer_flush (jbuf);
100 g_queue_free (jbuf->packets);
102 G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
106 * rtp_jitter_buffer_new:
108 * Create an #RTPJitterBuffer.
110 * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
113 rtp_jitter_buffer_new (void)
115 RTPJitterBuffer *jbuf;
117 jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
123 * rtp_jitter_buffer_get_mode:
124 * @jbuf: an #RTPJitterBuffer
126 * Get the current jitterbuffer mode.
128 * Returns: the current jitterbuffer mode.
131 rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
137 * rtp_jitter_buffer_set_mode:
138 * @jbuf: an #RTPJitterBuffer
139 * @mode: a #RTPJitterBufferMode
141 * Set the buffering and clock slaving algorithm used in the @jbuf.
144 rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
150 rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
156 rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
159 jbuf->low_level = (delay * 15) / 100;
160 /* the high level is at 90% in order to release packets before we fill up the
161 * buffer up to the latency */
162 jbuf->high_level = (delay * 90) / 100;
164 GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
165 GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
166 GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
170 * rtp_jitter_buffer_set_clock_rate:
171 * @jbuf: an #RTPJitterBuffer
173 * Set the clock rate in the jitterbuffer.
176 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
178 if (jbuf->clock_rate != clock_rate) {
179 if (jbuf->clock_rate == -1) {
180 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
181 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
183 GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
184 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
186 jbuf->base_time = -1;
187 jbuf->base_rtptime = -1;
188 jbuf->clock_rate = clock_rate;
189 jbuf->prev_out_time = -1;
190 jbuf->prev_send_diff = -1;
195 * rtp_jitter_buffer_get_clock_rate:
196 * @jbuf: an #RTPJitterBuffer
198 * Get the currently configure clock rate in @jbuf.
200 * Returns: the current clock-rate
203 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
205 return jbuf->clock_rate;
209 * rtp_jitter_buffer_reset_skew:
210 * @jbuf: an #RTPJitterBuffer
212 * Reset the skew calculations in @jbuf.
215 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
217 jbuf->base_time = -1;
218 jbuf->base_rtptime = -1;
219 jbuf->base_extrtp = -1;
220 jbuf->clock_rate = -1;
221 jbuf->ext_rtptime = -1;
222 jbuf->last_rtptime = -1;
223 jbuf->window_pos = 0;
224 jbuf->window_filling = TRUE;
225 jbuf->window_min = 0;
227 jbuf->prev_send_diff = -1;
228 jbuf->prev_out_time = -1;
229 GST_DEBUG ("reset skew correction");
233 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
234 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
236 jbuf->base_time = time;
237 jbuf->base_rtptime = gstrtptime;
238 jbuf->base_extrtp = ext_rtptime;
239 jbuf->prev_out_time = -1;
240 jbuf->prev_send_diff = -1;
242 jbuf->window_filling = TRUE;
243 jbuf->window_pos = 0;
244 jbuf->window_min = 0;
245 jbuf->window_size = 0;
251 get_buffer_level (RTPJitterBuffer * jbuf)
253 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
256 /* first first buffer with timestamp */
257 high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
259 if (high_buf->dts != -1)
262 high_buf = (RTPJitterBufferItem *) g_list_next (high_buf);
265 low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
267 if (low_buf->dts != -1)
270 low_buf = (RTPJitterBufferItem *) g_list_previous (low_buf);
273 if (!high_buf || !low_buf || high_buf == low_buf) {
276 guint64 high_ts, low_ts;
278 high_ts = high_buf->dts;
279 low_ts = low_buf->dts;
281 if (high_ts > low_ts)
282 level = high_ts - low_ts;
286 GST_LOG_OBJECT (jbuf,
287 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
288 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
295 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
297 gboolean post = FALSE;
300 level = get_buffer_level (jbuf);
301 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
303 if (jbuf->buffering) {
305 if (level > jbuf->high_level) {
306 GST_DEBUG ("buffering finished");
307 jbuf->buffering = FALSE;
310 if (level < jbuf->low_level) {
311 GST_DEBUG ("buffering started");
312 jbuf->buffering = TRUE;
319 if (jbuf->buffering && (jbuf->high_level != 0)) {
320 perc = (level * 100 / jbuf->high_level);
321 perc = MIN (perc, 100);
329 GST_DEBUG ("buffering %d", perc);
333 /* For the clock skew we use a windowed low point averaging algorithm as can be
334 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
335 * over Network Delays":
336 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
337 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
339 * The idea is that the jitter is composed of:
343 * N : a constant network delay.
344 * n : random added noise. The noise is concentrated around 0
346 * In the receiver we can track the elapsed time at the sender with:
348 * send_diff(i) = (Tsi - Ts0);
350 * Tsi : The time at the sender at packet i
351 * Ts0 : The time at the sender at the first packet
353 * This is the difference between the RTP timestamp in the first received packet
354 * and the current packet.
356 * At the receiver we have to deal with the jitter introduced by the network.
358 * recv_diff(i) = (Tri - Tr0)
360 * Tri : The time at the receiver at packet i
361 * Tr0 : The time at the receiver at the first packet
363 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
366 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
368 * Cri : The time of the clock at the receiver for packet i
369 * D + ni : The jitter when receiving packet i
371 * We see that the network delay is irrelevant here as we can elliminate D:
373 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
375 * The drift is now expressed as:
377 * Drift(i) = recv_diff(i) - send_diff(i);
379 * We now keep the W latest values of Drift and find the minimum (this is the
380 * one with the lowest network jitter and thus the one which is least affected
381 * by it). We average this lowest value to smooth out the resulting network skew.
383 * Both the window and the weighting used for averaging influence the accuracy
384 * of the drift estimation. Finding the correct parameters turns out to be a
385 * compromise between accuracy and inertia.
387 * We use a 2 second window or up to 512 data points, which is statistically big
388 * enough to catch spikes (FIXME, detect spikes).
389 * We also use a rather large weighting factor (125) to smoothly adapt. During
390 * startup, when filling the window, we use a parabolic weighting factor, the
391 * more the window is filled, the faster we move to the detected possible skew.
393 * Returns: @time adjusted with the clock skew.
396 calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
399 guint64 send_diff, recv_diff;
403 GstClockTime gstrtptime, out_time;
406 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
408 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
409 return jbuf->prev_out_time;
412 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
414 /* keep track of the last extended rtptime */
415 jbuf->last_rtptime = ext_rtptime;
417 /* first time, lock on to time and gstrtptime */
418 if (G_UNLIKELY (jbuf->base_time == -1)) {
419 jbuf->base_time = time;
420 jbuf->prev_out_time = -1;
421 GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
423 if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
424 jbuf->base_rtptime = gstrtptime;
425 jbuf->base_extrtp = ext_rtptime;
426 jbuf->prev_send_diff = -1;
427 GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
428 GST_TIME_ARGS (gstrtptime));
431 if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
432 send_diff = gstrtptime - jbuf->base_rtptime;
433 else if (time != -1) {
434 /* elapsed time at sender, timestamps can go backwards and thus be smaller
435 * than our base time, take a new base time in that case. */
436 GST_WARNING ("backward timestamps at server, taking new base time");
437 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
440 GST_WARNING ("backward timestamps at server but no timestamps");
442 /* at least try to get a new timestamp.. */
443 jbuf->base_time = -1;
446 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
447 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
448 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
449 GST_TIME_ARGS (send_diff));
451 /* we don't have an arrival timestamp so we can't do skew detection. we
452 * should still apply a timestamp based on RTP timestamp and base_time */
453 if (time == -1 || jbuf->base_time == -1)
456 /* elapsed time at receiver, includes the jitter */
457 recv_diff = time - jbuf->base_time;
459 /* measure the diff */
460 delta = ((gint64) recv_diff) - ((gint64) send_diff);
462 /* measure the slope, this gives a rought estimate between the sender speed
463 * and the receiver speed. This should be approximately 8, higher values
464 * indicate a burst (especially when the connection starts) */
466 slope = (send_diff * 8) / recv_diff;
470 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
471 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
472 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
474 /* if the difference between the sender timeline and the receiver timeline
475 * changed too quickly we have to resync because the server likely restarted
477 if (ABS (delta - jbuf->skew) > GST_SECOND) {
478 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
479 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
480 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
485 pos = jbuf->window_pos;
487 if (G_UNLIKELY (jbuf->window_filling)) {
488 /* we are filling the window */
489 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
490 jbuf->window[pos++] = delta;
491 /* calc the min delta we observed */
492 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
493 jbuf->window_min = delta;
495 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
496 jbuf->window_size = pos;
499 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
501 /* the skew is now the min */
502 jbuf->skew = jbuf->window_min;
503 jbuf->window_filling = FALSE;
505 gint perc_time, perc_window, perc;
507 /* figure out how much we filled the window, this depends on the amount of
508 * time we have or the max number of points we keep. */
509 perc_time = send_diff * 100 / MAX_TIME;
510 perc_window = pos * 100 / MAX_WINDOW;
511 perc = MAX (perc_time, perc_window);
513 /* make a parabolic function, the closer we get to the MAX, the more value
514 * we give to the scaling factor of the new value */
517 /* quickly go to the min value when we are filling up, slowly when we are
518 * just starting because we're not sure it's a good value yet. */
520 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
521 jbuf->window_size = pos + 1;
524 /* pick old value and store new value. We keep the previous value in order
525 * to quickly check if the min of the window changed */
526 old = jbuf->window[pos];
527 jbuf->window[pos++] = delta;
529 if (G_UNLIKELY (delta <= jbuf->window_min)) {
530 /* if the new value we inserted is smaller or equal to the current min,
531 * it becomes the new min */
532 jbuf->window_min = delta;
533 } else if (G_UNLIKELY (old == jbuf->window_min)) {
534 gint64 min = G_MAXINT64;
536 /* if we removed the old min, we have to find a new min */
537 for (i = 0; i < jbuf->window_size; i++) {
538 /* we found another value equal to the old min, we can stop searching now */
539 if (jbuf->window[i] == old) {
543 if (jbuf->window[i] < min)
544 min = jbuf->window[i];
546 jbuf->window_min = min;
548 /* average the min values */
549 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
550 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
551 delta, jbuf->window_min);
553 /* wrap around in the window */
554 if (G_UNLIKELY (pos >= jbuf->window_size))
556 jbuf->window_pos = pos;
559 /* the output time is defined as the base timestamp plus the RTP time
560 * adjusted for the clock skew .*/
561 if (jbuf->base_time != -1) {
562 out_time = jbuf->base_time + send_diff;
563 /* skew can be negative and we don't want to make invalid timestamps */
564 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
567 out_time += jbuf->skew;
569 /* check if timestamps are not going backwards, we can only check this if we
570 * have a previous out time and a previous send_diff */
571 if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
572 /* now check for backwards timestamps */
574 /* if the server timestamps went up and the out_time backwards */
575 (send_diff > jbuf->prev_send_diff
576 && out_time < jbuf->prev_out_time) ||
577 /* if the server timestamps went backwards and the out_time forwards */
578 (send_diff < jbuf->prev_send_diff
579 && out_time > jbuf->prev_out_time) ||
580 /* if the server timestamps did not change */
581 send_diff == jbuf->prev_send_diff)) {
582 GST_DEBUG ("backwards timestamps, using previous time");
583 out_time = jbuf->prev_out_time;
586 if (time != -1 && out_time + jbuf->delay < time) {
587 /* if we are going to produce a timestamp that is later than the input
588 * timestamp, we need to reset the jitterbuffer. Likely the server paused
590 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
591 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
592 jbuf->delay, GST_TIME_ARGS (time));
593 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
600 jbuf->prev_out_time = out_time;
601 jbuf->prev_send_diff = send_diff;
603 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
604 jbuf->skew, GST_TIME_ARGS (out_time));
610 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
612 GQueue *queue = jbuf->packets;
615 /* It's more likely that the packet was inserted in the front of the buffer */
616 if (G_LIKELY (list)) {
617 item->prev = list->prev;
621 item->prev->next = item;
626 queue->tail = g_list_concat (queue->tail, item);
627 if (queue->tail->next)
628 queue->tail = queue->tail->next;
630 queue->head = queue->tail;
634 GST_DEBUG ("head %p, tail %p", queue->head, queue->tail);
635 for (walk = queue->head; walk; walk = walk->next) {
636 RTPJitterBufferItem *item = (RTPJitterBufferItem *) walk;
637 GST_DEBUG ("item %p, next %p, prev %p, #%d",
638 item, item->next, item->prev, item->seqnum);
643 * rtp_jitter_buffer_insert:
644 * @jbuf: an #RTPJitterBuffer
645 * @item: an #RTPJitterBufferItem to insert
646 * @tail: TRUE when the tail element changed.
647 * @percent: the buffering percent after insertion
649 * Inserts @item into the packet queue of @jbuf. The sequence number of the
650 * packet will be used to sort the packets. This function takes ownerhip of
651 * @buf when the function returns %TRUE.
653 * Returns: %FALSE if a packet with the same number already existed.
656 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
657 gboolean * tail, gint * percent)
664 g_return_val_if_fail (jbuf != NULL, FALSE);
665 g_return_val_if_fail (item != NULL, FALSE);
667 seqnum = item->seqnum;
668 /* no seqnum, simply append then */
672 /* loop the list to skip strictly smaller seqnum buffers */
673 for (list = jbuf->packets->head; list; list = g_list_next (list)) {
676 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
678 qseq = qitem->seqnum;
682 /* compare the new seqnum to the one in the buffer */
683 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
685 /* we hit a packet with the same seqnum, notify a duplicate */
686 if (G_UNLIKELY (gap == 0))
689 /* seqnum > qseq, we can stop looking */
690 if (G_LIKELY (gap < 0))
695 rtptime = item->rtptime;
697 /* rtp time jumps are checked for during skew calculation, but bypassed
698 * in other mode, so mind those here and reset jb if needed.
699 * Only reset if valid input time, which is likely for UDP input
700 * where we expect this might happen due to async thread effects
701 * (in seek and state change cycles), but not so much for TCP input */
702 if (GST_CLOCK_TIME_IS_VALID (dts) &&
703 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
704 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
705 GstClockTime ext_rtptime = jbuf->ext_rtptime;
707 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
708 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
709 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
710 /* reset even if we don't have valid incoming time;
711 * still better than producing possibly very bogus output timestamp */
712 GST_WARNING ("rtp delta too big, reset skew");
713 rtp_jitter_buffer_reset_skew (jbuf);
717 switch (jbuf->mode) {
718 case RTP_JITTER_BUFFER_MODE_NONE:
719 case RTP_JITTER_BUFFER_MODE_BUFFER:
720 /* send 0 as the first timestamp and -1 for the other ones. This will
721 * interpollate them from the RTP timestamps with a 0 origin. In buffering
722 * mode we will adjust the outgoing timestamps according to the amount of
723 * time we spent buffering. */
724 if (jbuf->base_time == -1)
729 case RTP_JITTER_BUFFER_MODE_SLAVE:
733 /* do skew calculation by measuring the difference between rtptime and the
734 * receive dts, this function will return the skew corrected rtptime. */
735 item->pts = calculate_skew (jbuf, rtptime, dts);
738 queue_do_insert (jbuf, list, (GList *) item);
740 /* buffering mode, update buffer stats */
741 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
742 update_buffer_level (jbuf, percent);
746 /* tail was changed when we did not find a previous packet, we set the return
747 * flag when requested. */
749 *tail = (list == NULL);
756 GST_WARNING ("duplicate packet %d found", (gint) seqnum);
762 * rtp_jitter_buffer_pop:
763 * @jbuf: an #RTPJitterBuffer
764 * @percent: the buffering percent
766 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
767 * have its timestamp adjusted with the incomming running_time and the detected
770 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
772 RTPJitterBufferItem *
773 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
778 g_return_val_if_fail (jbuf != NULL, NULL);
780 queue = jbuf->packets;
784 queue->tail = item->prev;
786 queue->tail->next = NULL;
792 /* buffering mode, update buffer stats */
793 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
794 update_buffer_level (jbuf, percent);
798 return (RTPJitterBufferItem *) item;
802 * rtp_jitter_buffer_peek:
803 * @jbuf: an #RTPJitterBuffer
805 * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
806 * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
807 * was inserted in the queue.
809 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
811 RTPJitterBufferItem *
812 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
814 g_return_val_if_fail (jbuf != NULL, NULL);
816 return (RTPJitterBufferItem *) jbuf->packets->tail;
820 * rtp_jitter_buffer_flush:
821 * @jbuf: an #RTPJitterBuffer
823 * Flush all packets from the jitterbuffer.
826 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
830 g_return_if_fail (jbuf != NULL);
832 while ((buffer = g_queue_pop_head (jbuf->packets)))
833 gst_buffer_unref (buffer);
837 * rtp_jitter_buffer_is_buffering:
838 * @jbuf: an #RTPJitterBuffer
840 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
841 * pop packets while in buffering mode.
843 * Returns: the buffering state of @jbuf
846 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
848 return jbuf->buffering;
852 * rtp_jitter_buffer_set_buffering:
853 * @jbuf: an #RTPJitterBuffer
854 * @buffering: the new buffering state
856 * Forces @jbuf to go into the buffering state.
859 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
861 jbuf->buffering = buffering;
865 * rtp_jitter_buffer_get_percent:
866 * @jbuf: an #RTPJitterBuffer
868 * Get the buffering percent of the jitterbuffer.
870 * Returns: the buffering percent
873 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
878 if (G_UNLIKELY (jbuf->high_level == 0))
881 level = get_buffer_level (jbuf);
882 percent = (level * 100 / jbuf->high_level);
883 percent = MIN (percent, 100);
889 * rtp_jitter_buffer_num_packets:
890 * @jbuf: an #RTPJitterBuffer
892 * Get the number of packets currently in "jbuf.
894 * Returns: The number of packets in @jbuf.
897 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
899 g_return_val_if_fail (jbuf != NULL, 0);
901 return jbuf->packets->length;
905 * rtp_jitter_buffer_get_ts_diff:
906 * @jbuf: an #RTPJitterBuffer
908 * Get the difference between the timestamps of first and last packet in the
911 * Returns: The difference expressed in the timestamp units of the packets.
914 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
916 guint64 high_ts, low_ts;
917 GstBuffer *high_buf, *low_buf;
919 GstRTPBuffer rtp = { NULL };
921 g_return_val_if_fail (jbuf != NULL, 0);
923 high_buf = g_queue_peek_head (jbuf->packets);
924 low_buf = g_queue_peek_tail (jbuf->packets);
926 if (!high_buf || !low_buf || high_buf == low_buf)
929 gst_rtp_buffer_map (high_buf, GST_MAP_READ, &rtp);
930 high_ts = gst_rtp_buffer_get_timestamp (&rtp);
931 gst_rtp_buffer_unmap (&rtp);
932 gst_rtp_buffer_map (low_buf, GST_MAP_READ, &rtp);
933 low_ts = gst_rtp_buffer_get_timestamp (&rtp);
934 gst_rtp_buffer_unmap (&rtp);
936 /* it needs to work if ts wraps */
937 if (high_ts >= low_ts) {
938 result = (guint32) (high_ts - low_ts);
940 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
946 * rtp_jitter_buffer_get_sync:
947 * @jbuf: an #RTPJitterBuffer
948 * @rtptime: result RTP time
949 * @timestamp: result GStreamer timestamp
950 * @clock_rate: clock-rate of @rtptime
951 * @last_rtptime: last seen rtptime.
953 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
954 * used for constructing timestamps.
956 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
957 * the GStreamer timestamp is currently @timestamp.
959 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
963 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
964 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
967 *rtptime = jbuf->base_extrtp;
969 *timestamp = jbuf->base_time + jbuf->skew;
971 *clock_rate = jbuf->clock_rate;
973 *last_rtptime = jbuf->last_rtptime;