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",
56 {RTP_JITTER_BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks",
61 if (!jitter_buffer_mode_type) {
62 jitter_buffer_mode_type =
63 g_enum_register_static ("RTPJitterBufferMode", jitter_buffer_modes);
65 return jitter_buffer_mode_type;
68 /* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
70 G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
73 rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
75 GObjectClass *gobject_class;
77 gobject_class = (GObjectClass *) klass;
79 gobject_class->finalize = rtp_jitter_buffer_finalize;
81 GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
86 rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
88 jbuf->packets = g_queue_new ();
89 jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
91 rtp_jitter_buffer_reset_skew (jbuf);
95 rtp_jitter_buffer_finalize (GObject * object)
97 RTPJitterBuffer *jbuf;
99 jbuf = RTP_JITTER_BUFFER_CAST (object);
101 g_queue_free (jbuf->packets);
103 G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
107 * rtp_jitter_buffer_new:
109 * Create an #RTPJitterBuffer.
111 * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
114 rtp_jitter_buffer_new (void)
116 RTPJitterBuffer *jbuf;
118 jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
124 * rtp_jitter_buffer_get_mode:
125 * @jbuf: an #RTPJitterBuffer
127 * Get the current jitterbuffer mode.
129 * Returns: the current jitterbuffer mode.
132 rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
138 * rtp_jitter_buffer_set_mode:
139 * @jbuf: an #RTPJitterBuffer
140 * @mode: a #RTPJitterBufferMode
142 * Set the buffering and clock slaving algorithm used in the @jbuf.
145 rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
151 rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
157 rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
160 jbuf->low_level = (delay * 15) / 100;
161 /* the high level is at 90% in order to release packets before we fill up the
162 * buffer up to the latency */
163 jbuf->high_level = (delay * 90) / 100;
165 GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
166 GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
167 GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
171 * rtp_jitter_buffer_set_clock_rate:
172 * @jbuf: an #RTPJitterBuffer
173 * @clock_rate: the new clock rate
175 * Set the clock rate in the jitterbuffer.
178 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
180 if (jbuf->clock_rate != clock_rate) {
181 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
182 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
183 jbuf->clock_rate = clock_rate;
184 rtp_jitter_buffer_reset_skew (jbuf);
189 * rtp_jitter_buffer_get_clock_rate:
190 * @jbuf: an #RTPJitterBuffer
192 * Get the currently configure clock rate in @jbuf.
194 * Returns: the current clock-rate
197 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
199 return jbuf->clock_rate;
203 * rtp_jitter_buffer_reset_skew:
204 * @jbuf: an #RTPJitterBuffer
206 * Reset the skew calculations in @jbuf.
209 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
211 jbuf->base_time = -1;
212 jbuf->base_rtptime = -1;
213 jbuf->base_extrtp = -1;
214 jbuf->ext_rtptime = -1;
215 jbuf->last_rtptime = -1;
216 jbuf->window_pos = 0;
217 jbuf->window_filling = TRUE;
218 jbuf->window_min = 0;
220 jbuf->prev_send_diff = -1;
221 jbuf->prev_out_time = -1;
222 jbuf->need_resync = TRUE;
223 GST_DEBUG ("reset skew correction");
227 * rtp_jitter_buffer_disable_buffering:
228 * @jbuf: an #RTPJitterBuffer
229 * @disabled: the new state
231 * Enable or disable buffering on @jbuf.
234 rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
236 jbuf->buffering_disabled = disabled;
240 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
241 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
243 jbuf->base_time = time;
244 jbuf->base_rtptime = gstrtptime;
245 jbuf->base_extrtp = ext_rtptime;
246 jbuf->prev_out_time = -1;
247 jbuf->prev_send_diff = -1;
249 jbuf->window_filling = TRUE;
250 jbuf->window_pos = 0;
251 jbuf->window_min = 0;
252 jbuf->window_size = 0;
255 jbuf->need_resync = FALSE;
259 get_buffer_level (RTPJitterBuffer * jbuf)
261 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
264 /* first buffer with timestamp */
265 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
267 if (high_buf->dts != -1 || high_buf->pts != -1)
270 high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
273 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
275 if (low_buf->dts != -1 || low_buf->pts != -1)
278 low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
281 if (!high_buf || !low_buf || high_buf == low_buf) {
284 guint64 high_ts, low_ts;
286 high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
287 low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
289 if (high_ts > low_ts)
290 level = high_ts - low_ts;
294 GST_LOG_OBJECT (jbuf,
295 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
296 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
303 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
305 gboolean post = FALSE;
308 level = get_buffer_level (jbuf);
309 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
311 if (jbuf->buffering_disabled) {
312 GST_DEBUG ("buffering is disabled");
313 level = jbuf->high_level;
316 if (jbuf->buffering) {
318 if (level >= jbuf->high_level) {
319 GST_DEBUG ("buffering finished");
320 jbuf->buffering = FALSE;
323 if (level < jbuf->low_level) {
324 GST_DEBUG ("buffering started");
325 jbuf->buffering = TRUE;
332 if (jbuf->buffering && (jbuf->high_level != 0)) {
333 perc = (level * 100 / jbuf->high_level);
334 perc = MIN (perc, 100);
342 GST_DEBUG ("buffering %d", perc);
346 /* For the clock skew we use a windowed low point averaging algorithm as can be
347 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
348 * over Network Delays":
349 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
350 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
352 * The idea is that the jitter is composed of:
356 * N : a constant network delay.
357 * n : random added noise. The noise is concentrated around 0
359 * In the receiver we can track the elapsed time at the sender with:
361 * send_diff(i) = (Tsi - Ts0);
363 * Tsi : The time at the sender at packet i
364 * Ts0 : The time at the sender at the first packet
366 * This is the difference between the RTP timestamp in the first received packet
367 * and the current packet.
369 * At the receiver we have to deal with the jitter introduced by the network.
371 * recv_diff(i) = (Tri - Tr0)
373 * Tri : The time at the receiver at packet i
374 * Tr0 : The time at the receiver at the first packet
376 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
379 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
381 * Cri : The time of the clock at the receiver for packet i
382 * D + ni : The jitter when receiving packet i
384 * We see that the network delay is irrelevant here as we can elliminate D:
386 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
388 * The drift is now expressed as:
390 * Drift(i) = recv_diff(i) - send_diff(i);
392 * We now keep the W latest values of Drift and find the minimum (this is the
393 * one with the lowest network jitter and thus the one which is least affected
394 * by it). We average this lowest value to smooth out the resulting network skew.
396 * Both the window and the weighting used for averaging influence the accuracy
397 * of the drift estimation. Finding the correct parameters turns out to be a
398 * compromise between accuracy and inertia.
400 * We use a 2 second window or up to 512 data points, which is statistically big
401 * enough to catch spikes (FIXME, detect spikes).
402 * We also use a rather large weighting factor (125) to smoothly adapt. During
403 * startup, when filling the window, we use a parabolic weighting factor, the
404 * more the window is filled, the faster we move to the detected possible skew.
406 * Returns: @time adjusted with the clock skew.
409 calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
412 guint64 send_diff, recv_diff;
416 GstClockTime gstrtptime, out_time;
419 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
421 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
422 return jbuf->prev_out_time;
425 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
427 /* keep track of the last extended rtptime */
428 jbuf->last_rtptime = ext_rtptime;
431 if (G_LIKELY (jbuf->base_rtptime != -1)) {
432 /* check elapsed time in RTP units */
433 if (G_LIKELY (gstrtptime >= jbuf->base_rtptime)) {
434 send_diff = gstrtptime - jbuf->base_rtptime;
436 /* elapsed time at sender, timestamps can go backwards and thus be
437 * smaller than our base time, schedule to take a new base time in
439 GST_WARNING ("backward timestamps at server, schedule resync");
440 jbuf->need_resync = TRUE;
445 /* need resync, lock on to time and gstrtptime if we can, otherwise we
446 * do with the previous values */
447 if (G_UNLIKELY (jbuf->need_resync && time != -1)) {
448 GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
449 GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (gstrtptime));
450 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
454 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
455 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
456 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
457 GST_TIME_ARGS (send_diff));
459 /* we don't have an arrival timestamp so we can't do skew detection. we
460 * should still apply a timestamp based on RTP timestamp and base_time */
461 if (time == -1 || jbuf->base_time == -1)
464 /* elapsed time at receiver, includes the jitter */
465 recv_diff = time - jbuf->base_time;
467 /* measure the diff */
468 delta = ((gint64) recv_diff) - ((gint64) send_diff);
470 /* measure the slope, this gives a rought estimate between the sender speed
471 * and the receiver speed. This should be approximately 8, higher values
472 * indicate a burst (especially when the connection starts) */
474 slope = (send_diff * 8) / recv_diff;
478 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
479 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
480 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
482 /* if the difference between the sender timeline and the receiver timeline
483 * changed too quickly we have to resync because the server likely restarted
485 if (ABS (delta - jbuf->skew) > GST_SECOND) {
486 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
487 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
488 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
493 pos = jbuf->window_pos;
495 if (G_UNLIKELY (jbuf->window_filling)) {
496 /* we are filling the window */
497 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
498 jbuf->window[pos++] = delta;
499 /* calc the min delta we observed */
500 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
501 jbuf->window_min = delta;
503 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
504 jbuf->window_size = pos;
507 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
509 /* the skew is now the min */
510 jbuf->skew = jbuf->window_min;
511 jbuf->window_filling = FALSE;
513 gint perc_time, perc_window, perc;
515 /* figure out how much we filled the window, this depends on the amount of
516 * time we have or the max number of points we keep. */
517 perc_time = send_diff * 100 / MAX_TIME;
518 perc_window = pos * 100 / MAX_WINDOW;
519 perc = MAX (perc_time, perc_window);
521 /* make a parabolic function, the closer we get to the MAX, the more value
522 * we give to the scaling factor of the new value */
525 /* quickly go to the min value when we are filling up, slowly when we are
526 * just starting because we're not sure it's a good value yet. */
528 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
529 jbuf->window_size = pos + 1;
532 /* pick old value and store new value. We keep the previous value in order
533 * to quickly check if the min of the window changed */
534 old = jbuf->window[pos];
535 jbuf->window[pos++] = delta;
537 if (G_UNLIKELY (delta <= jbuf->window_min)) {
538 /* if the new value we inserted is smaller or equal to the current min,
539 * it becomes the new min */
540 jbuf->window_min = delta;
541 } else if (G_UNLIKELY (old == jbuf->window_min)) {
542 gint64 min = G_MAXINT64;
544 /* if we removed the old min, we have to find a new min */
545 for (i = 0; i < jbuf->window_size; i++) {
546 /* we found another value equal to the old min, we can stop searching now */
547 if (jbuf->window[i] == old) {
551 if (jbuf->window[i] < min)
552 min = jbuf->window[i];
554 jbuf->window_min = min;
556 /* average the min values */
557 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
558 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
559 delta, jbuf->window_min);
561 /* wrap around in the window */
562 if (G_UNLIKELY (pos >= jbuf->window_size))
564 jbuf->window_pos = pos;
567 /* the output time is defined as the base timestamp plus the RTP time
568 * adjusted for the clock skew .*/
569 if (jbuf->base_time != -1) {
570 out_time = jbuf->base_time + send_diff;
571 /* skew can be negative and we don't want to make invalid timestamps */
572 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
575 out_time += jbuf->skew;
577 /* check if timestamps are not going backwards, we can only check this if we
578 * have a previous out time and a previous send_diff */
579 if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
580 /* now check for backwards timestamps */
582 /* if the server timestamps went up and the out_time backwards */
583 (send_diff > jbuf->prev_send_diff
584 && out_time < jbuf->prev_out_time) ||
585 /* if the server timestamps went backwards and the out_time forwards */
586 (send_diff < jbuf->prev_send_diff
587 && out_time > jbuf->prev_out_time) ||
588 /* if the server timestamps did not change */
589 send_diff == jbuf->prev_send_diff)) {
590 GST_DEBUG ("backwards timestamps, using previous time");
591 out_time = jbuf->prev_out_time;
594 if (time != -1 && out_time + jbuf->delay < time) {
595 /* if we are going to produce a timestamp that is later than the input
596 * timestamp, we need to reset the jitterbuffer. Likely the server paused
598 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
599 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
600 jbuf->delay, GST_TIME_ARGS (time));
601 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
608 jbuf->prev_out_time = out_time;
609 jbuf->prev_send_diff = send_diff;
611 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
612 jbuf->skew, GST_TIME_ARGS (out_time));
618 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
620 GQueue *queue = jbuf->packets;
622 /* It's more likely that the packet was inserted at the tail of the queue */
623 if (G_LIKELY (list)) {
625 item->next = list->next;
629 item->next = queue->head;
633 item->next->prev = item;
640 * rtp_jitter_buffer_insert:
641 * @jbuf: an #RTPJitterBuffer
642 * @item: an #RTPJitterBufferItem to insert
643 * @head: TRUE when the head element changed.
644 * @percent: the buffering percent after insertion
646 * Inserts @item into the packet queue of @jbuf. The sequence number of the
647 * packet will be used to sort the packets. This function takes ownerhip of
648 * @buf when the function returns %TRUE.
650 * When @head is %TRUE, the new packet was added at the head of the queue and
651 * will be available with the next call to rtp_jitter_buffer_pop() and
652 * rtp_jitter_buffer_peek().
654 * Returns: %FALSE if a packet with the same number already existed.
657 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
658 gboolean * head, gint * percent)
660 GList *list, *event = NULL;
665 g_return_val_if_fail (jbuf != NULL, FALSE);
666 g_return_val_if_fail (item != NULL, FALSE);
668 list = jbuf->packets->tail;
670 /* no seqnum, simply append then */
671 if (item->seqnum == -1)
674 seqnum = item->seqnum;
676 /* loop the list to skip strictly larger seqnum buffers */
677 for (; list; list = g_list_previous (list)) {
680 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
682 if (qitem->seqnum == -1) {
683 /* keep a pointer to the first consecutive event if not already
684 * set. we will insert the packet after the event if we can't find
685 * a packet with lower sequence number before the event. */
691 qseq = qitem->seqnum;
693 /* compare the new seqnum to the one in the buffer */
694 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
696 /* we hit a packet with the same seqnum, notify a duplicate */
697 if (G_UNLIKELY (gap == 0))
700 /* seqnum > qseq, we can stop looking */
701 if (G_LIKELY (gap < 0))
704 /* if we've found a packet with greater sequence number, cleanup the
705 * event pointer as the packet will be inserted before the event */
709 /* if event is set it means that packets before the event had smaller
710 * sequence number, so we will insert our packet after the event */
715 if (item->rtptime == -1)
718 rtptime = item->rtptime;
720 /* rtp time jumps are checked for during skew calculation, but bypassed
721 * in other mode, so mind those here and reset jb if needed.
722 * Only reset if valid input time, which is likely for UDP input
723 * where we expect this might happen due to async thread effects
724 * (in seek and state change cycles), but not so much for TCP input */
725 if (GST_CLOCK_TIME_IS_VALID (dts) &&
726 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
727 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
728 GstClockTime ext_rtptime = jbuf->ext_rtptime;
730 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
731 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
732 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
733 /* reset even if we don't have valid incoming time;
734 * still better than producing possibly very bogus output timestamp */
735 GST_WARNING ("rtp delta too big, reset skew");
736 rtp_jitter_buffer_reset_skew (jbuf);
740 switch (jbuf->mode) {
741 case RTP_JITTER_BUFFER_MODE_NONE:
742 case RTP_JITTER_BUFFER_MODE_BUFFER:
743 /* send 0 as the first timestamp and -1 for the other ones. This will
744 * interpolate them from the RTP timestamps with a 0 origin. In buffering
745 * mode we will adjust the outgoing timestamps according to the amount of
746 * time we spent buffering. */
747 if (jbuf->base_time == -1)
752 case RTP_JITTER_BUFFER_MODE_SYNCED:
753 /* synchronized clocks, take first timestamp as base, use RTP timestamps
755 if (jbuf->base_time != -1)
758 case RTP_JITTER_BUFFER_MODE_SLAVE:
762 /* do skew calculation by measuring the difference between rtptime and the
763 * receive dts, this function will return the skew corrected rtptime. */
764 item->pts = calculate_skew (jbuf, rtptime, dts);
767 queue_do_insert (jbuf, list, (GList *) item);
769 /* buffering mode, update buffer stats */
770 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
771 update_buffer_level (jbuf, percent);
775 /* head was changed when we did not find a previous packet, we set the return
776 * flag when requested. */
778 *head = (list == NULL);
785 GST_WARNING ("duplicate packet %d found", (gint) seqnum);
791 * rtp_jitter_buffer_pop:
792 * @jbuf: an #RTPJitterBuffer
793 * @percent: the buffering percent
795 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
796 * have its timestamp adjusted with the incomming running_time and the detected
799 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
801 RTPJitterBufferItem *
802 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
807 g_return_val_if_fail (jbuf != NULL, NULL);
809 queue = jbuf->packets;
813 queue->head = item->next;
815 queue->head->prev = NULL;
821 /* buffering mode, update buffer stats */
822 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
823 update_buffer_level (jbuf, percent);
827 return (RTPJitterBufferItem *) item;
831 * rtp_jitter_buffer_peek:
832 * @jbuf: an #RTPJitterBuffer
834 * Peek the oldest buffer from the packet queue of @jbuf.
836 * See rtp_jitter_buffer_insert() to check when an older packet was
839 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
841 RTPJitterBufferItem *
842 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
844 g_return_val_if_fail (jbuf != NULL, NULL);
846 return (RTPJitterBufferItem *) jbuf->packets->head;
850 * rtp_jitter_buffer_flush:
851 * @jbuf: an #RTPJitterBuffer
852 * @free_func: function to free each item
853 * @user_data: user data passed to @free_func
855 * Flush all packets from the jitterbuffer.
858 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
863 g_return_if_fail (jbuf != NULL);
864 g_return_if_fail (free_func != NULL);
866 while ((item = g_queue_pop_head_link (jbuf->packets)))
867 free_func ((RTPJitterBufferItem *) item, user_data);
871 * rtp_jitter_buffer_is_buffering:
872 * @jbuf: an #RTPJitterBuffer
874 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
875 * pop packets while in buffering mode.
877 * Returns: the buffering state of @jbuf
880 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
882 return jbuf->buffering && !jbuf->buffering_disabled;
886 * rtp_jitter_buffer_set_buffering:
887 * @jbuf: an #RTPJitterBuffer
888 * @buffering: the new buffering state
890 * Forces @jbuf to go into the buffering state.
893 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
895 jbuf->buffering = buffering;
899 * rtp_jitter_buffer_get_percent:
900 * @jbuf: an #RTPJitterBuffer
902 * Get the buffering percent of the jitterbuffer.
904 * Returns: the buffering percent
907 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
912 if (G_UNLIKELY (jbuf->high_level == 0))
915 if (G_UNLIKELY (jbuf->buffering_disabled))
918 level = get_buffer_level (jbuf);
919 percent = (level * 100 / jbuf->high_level);
920 percent = MIN (percent, 100);
926 * rtp_jitter_buffer_num_packets:
927 * @jbuf: an #RTPJitterBuffer
929 * Get the number of packets currently in "jbuf.
931 * Returns: The number of packets in @jbuf.
934 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
936 g_return_val_if_fail (jbuf != NULL, 0);
938 return jbuf->packets->length;
942 * rtp_jitter_buffer_get_ts_diff:
943 * @jbuf: an #RTPJitterBuffer
945 * Get the difference between the timestamps of first and last packet in the
948 * Returns: The difference expressed in the timestamp units of the packets.
951 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
953 guint64 high_ts, low_ts;
954 RTPJitterBufferItem *high_buf, *low_buf;
957 g_return_val_if_fail (jbuf != NULL, 0);
959 high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
960 low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
962 if (!high_buf || !low_buf || high_buf == low_buf)
965 high_ts = high_buf->rtptime;
966 low_ts = low_buf->rtptime;
968 /* it needs to work if ts wraps */
969 if (high_ts >= low_ts) {
970 result = (guint32) (high_ts - low_ts);
972 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
978 * rtp_jitter_buffer_get_sync:
979 * @jbuf: an #RTPJitterBuffer
980 * @rtptime: result RTP time
981 * @timestamp: result GStreamer timestamp
982 * @clock_rate: clock-rate of @rtptime
983 * @last_rtptime: last seen rtptime.
985 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
986 * used for constructing timestamps.
988 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
989 * the GStreamer timestamp is currently @timestamp.
991 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
995 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
996 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
999 *rtptime = jbuf->base_extrtp;
1001 *timestamp = jbuf->base_time + jbuf->skew;
1003 *clock_rate = jbuf->clock_rate;
1005 *last_rtptime = jbuf->last_rtptime;