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
174 * Set the clock rate in the jitterbuffer.
177 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
179 if (jbuf->clock_rate != clock_rate) {
180 if (jbuf->clock_rate == -1) {
181 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
182 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
184 GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
185 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
187 jbuf->clock_rate = clock_rate;
188 rtp_jitter_buffer_reset_skew (jbuf);
193 * rtp_jitter_buffer_get_clock_rate:
194 * @jbuf: an #RTPJitterBuffer
196 * Get the currently configure clock rate in @jbuf.
198 * Returns: the current clock-rate
201 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
203 return jbuf->clock_rate;
207 * rtp_jitter_buffer_reset_skew:
208 * @jbuf: an #RTPJitterBuffer
210 * Reset the skew calculations in @jbuf.
213 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
215 jbuf->base_time = -1;
216 jbuf->base_rtptime = -1;
217 jbuf->base_extrtp = -1;
218 jbuf->ext_rtptime = -1;
219 jbuf->last_rtptime = -1;
220 jbuf->window_pos = 0;
221 jbuf->window_filling = TRUE;
222 jbuf->window_min = 0;
224 jbuf->prev_send_diff = -1;
225 jbuf->prev_out_time = -1;
226 GST_DEBUG ("reset skew correction");
230 * rtp_jitter_buffer_disable_buffering:
231 * @jbuf: an #RTPJitterBuffer
232 * @disabled: the new state
234 * Enable or disable buffering on @jbuf.
237 rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
239 jbuf->buffering_disabled = disabled;
243 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
244 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
246 jbuf->base_time = time;
247 jbuf->base_rtptime = gstrtptime;
248 jbuf->base_extrtp = ext_rtptime;
249 jbuf->prev_out_time = -1;
250 jbuf->prev_send_diff = -1;
252 jbuf->window_filling = TRUE;
253 jbuf->window_pos = 0;
254 jbuf->window_min = 0;
255 jbuf->window_size = 0;
261 get_buffer_level (RTPJitterBuffer * jbuf)
263 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
266 /* first first buffer with timestamp */
267 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
269 if (high_buf->dts != -1 || high_buf->pts != -1)
272 high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
275 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
277 if (low_buf->dts != -1 || low_buf->pts != -1)
280 low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
283 if (!high_buf || !low_buf || high_buf == low_buf) {
286 guint64 high_ts, low_ts;
288 high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
289 low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
291 if (high_ts > low_ts)
292 level = high_ts - low_ts;
296 GST_LOG_OBJECT (jbuf,
297 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
298 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
305 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
307 gboolean post = FALSE;
310 level = get_buffer_level (jbuf);
311 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
313 if (jbuf->buffering_disabled) {
314 GST_DEBUG ("buffering is disabled");
315 level = jbuf->high_level;
318 if (jbuf->buffering) {
320 if (level >= jbuf->high_level) {
321 GST_DEBUG ("buffering finished");
322 jbuf->buffering = FALSE;
325 if (level < jbuf->low_level) {
326 GST_DEBUG ("buffering started");
327 jbuf->buffering = TRUE;
334 if (jbuf->buffering && (jbuf->high_level != 0)) {
335 perc = (level * 100 / jbuf->high_level);
336 perc = MIN (perc, 100);
344 GST_DEBUG ("buffering %d", perc);
348 /* For the clock skew we use a windowed low point averaging algorithm as can be
349 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
350 * over Network Delays":
351 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
352 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
354 * The idea is that the jitter is composed of:
358 * N : a constant network delay.
359 * n : random added noise. The noise is concentrated around 0
361 * In the receiver we can track the elapsed time at the sender with:
363 * send_diff(i) = (Tsi - Ts0);
365 * Tsi : The time at the sender at packet i
366 * Ts0 : The time at the sender at the first packet
368 * This is the difference between the RTP timestamp in the first received packet
369 * and the current packet.
371 * At the receiver we have to deal with the jitter introduced by the network.
373 * recv_diff(i) = (Tri - Tr0)
375 * Tri : The time at the receiver at packet i
376 * Tr0 : The time at the receiver at the first packet
378 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
381 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
383 * Cri : The time of the clock at the receiver for packet i
384 * D + ni : The jitter when receiving packet i
386 * We see that the network delay is irrelevant here as we can elliminate D:
388 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
390 * The drift is now expressed as:
392 * Drift(i) = recv_diff(i) - send_diff(i);
394 * We now keep the W latest values of Drift and find the minimum (this is the
395 * one with the lowest network jitter and thus the one which is least affected
396 * by it). We average this lowest value to smooth out the resulting network skew.
398 * Both the window and the weighting used for averaging influence the accuracy
399 * of the drift estimation. Finding the correct parameters turns out to be a
400 * compromise between accuracy and inertia.
402 * We use a 2 second window or up to 512 data points, which is statistically big
403 * enough to catch spikes (FIXME, detect spikes).
404 * We also use a rather large weighting factor (125) to smoothly adapt. During
405 * startup, when filling the window, we use a parabolic weighting factor, the
406 * more the window is filled, the faster we move to the detected possible skew.
408 * Returns: @time adjusted with the clock skew.
411 calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
414 guint64 send_diff, recv_diff;
418 GstClockTime gstrtptime, out_time;
421 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
423 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
424 return jbuf->prev_out_time;
427 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
429 /* keep track of the last extended rtptime */
430 jbuf->last_rtptime = ext_rtptime;
432 /* first time, lock on to time and gstrtptime */
433 if (G_UNLIKELY (jbuf->base_time == -1)) {
434 jbuf->base_time = time;
435 jbuf->prev_out_time = -1;
436 GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
438 if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
439 jbuf->base_rtptime = gstrtptime;
440 jbuf->base_extrtp = ext_rtptime;
441 jbuf->prev_send_diff = -1;
442 GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
443 GST_TIME_ARGS (gstrtptime));
446 if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
447 send_diff = gstrtptime - jbuf->base_rtptime;
448 else if (time != -1) {
449 /* elapsed time at sender, timestamps can go backwards and thus be smaller
450 * than our base time, take a new base time in that case. */
451 GST_WARNING ("backward timestamps at server, taking new base time");
452 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
455 GST_WARNING ("backward timestamps at server but no timestamps");
457 /* at least try to get a new timestamp.. */
458 jbuf->base_time = -1;
461 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
462 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
463 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
464 GST_TIME_ARGS (send_diff));
466 /* we don't have an arrival timestamp so we can't do skew detection. we
467 * should still apply a timestamp based on RTP timestamp and base_time */
468 if (time == -1 || jbuf->base_time == -1)
471 /* elapsed time at receiver, includes the jitter */
472 recv_diff = time - jbuf->base_time;
474 /* measure the diff */
475 delta = ((gint64) recv_diff) - ((gint64) send_diff);
477 /* measure the slope, this gives a rought estimate between the sender speed
478 * and the receiver speed. This should be approximately 8, higher values
479 * indicate a burst (especially when the connection starts) */
481 slope = (send_diff * 8) / recv_diff;
485 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
486 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
487 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
489 /* if the difference between the sender timeline and the receiver timeline
490 * changed too quickly we have to resync because the server likely restarted
492 if (ABS (delta - jbuf->skew) > GST_SECOND) {
493 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
494 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
495 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
500 pos = jbuf->window_pos;
502 if (G_UNLIKELY (jbuf->window_filling)) {
503 /* we are filling the window */
504 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
505 jbuf->window[pos++] = delta;
506 /* calc the min delta we observed */
507 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
508 jbuf->window_min = delta;
510 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
511 jbuf->window_size = pos;
514 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
516 /* the skew is now the min */
517 jbuf->skew = jbuf->window_min;
518 jbuf->window_filling = FALSE;
520 gint perc_time, perc_window, perc;
522 /* figure out how much we filled the window, this depends on the amount of
523 * time we have or the max number of points we keep. */
524 perc_time = send_diff * 100 / MAX_TIME;
525 perc_window = pos * 100 / MAX_WINDOW;
526 perc = MAX (perc_time, perc_window);
528 /* make a parabolic function, the closer we get to the MAX, the more value
529 * we give to the scaling factor of the new value */
532 /* quickly go to the min value when we are filling up, slowly when we are
533 * just starting because we're not sure it's a good value yet. */
535 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
536 jbuf->window_size = pos + 1;
539 /* pick old value and store new value. We keep the previous value in order
540 * to quickly check if the min of the window changed */
541 old = jbuf->window[pos];
542 jbuf->window[pos++] = delta;
544 if (G_UNLIKELY (delta <= jbuf->window_min)) {
545 /* if the new value we inserted is smaller or equal to the current min,
546 * it becomes the new min */
547 jbuf->window_min = delta;
548 } else if (G_UNLIKELY (old == jbuf->window_min)) {
549 gint64 min = G_MAXINT64;
551 /* if we removed the old min, we have to find a new min */
552 for (i = 0; i < jbuf->window_size; i++) {
553 /* we found another value equal to the old min, we can stop searching now */
554 if (jbuf->window[i] == old) {
558 if (jbuf->window[i] < min)
559 min = jbuf->window[i];
561 jbuf->window_min = min;
563 /* average the min values */
564 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
565 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
566 delta, jbuf->window_min);
568 /* wrap around in the window */
569 if (G_UNLIKELY (pos >= jbuf->window_size))
571 jbuf->window_pos = pos;
574 /* the output time is defined as the base timestamp plus the RTP time
575 * adjusted for the clock skew .*/
576 if (jbuf->base_time != -1) {
577 out_time = jbuf->base_time + send_diff;
578 /* skew can be negative and we don't want to make invalid timestamps */
579 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
582 out_time += jbuf->skew;
584 /* check if timestamps are not going backwards, we can only check this if we
585 * have a previous out time and a previous send_diff */
586 if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
587 /* now check for backwards timestamps */
589 /* if the server timestamps went up and the out_time backwards */
590 (send_diff > jbuf->prev_send_diff
591 && out_time < jbuf->prev_out_time) ||
592 /* if the server timestamps went backwards and the out_time forwards */
593 (send_diff < jbuf->prev_send_diff
594 && out_time > jbuf->prev_out_time) ||
595 /* if the server timestamps did not change */
596 send_diff == jbuf->prev_send_diff)) {
597 GST_DEBUG ("backwards timestamps, using previous time");
598 out_time = jbuf->prev_out_time;
601 if (time != -1 && out_time + jbuf->delay < time) {
602 /* if we are going to produce a timestamp that is later than the input
603 * timestamp, we need to reset the jitterbuffer. Likely the server paused
605 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
606 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
607 jbuf->delay, GST_TIME_ARGS (time));
608 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
615 jbuf->prev_out_time = out_time;
616 jbuf->prev_send_diff = send_diff;
618 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
619 jbuf->skew, GST_TIME_ARGS (out_time));
625 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
627 GQueue *queue = jbuf->packets;
629 /* It's more likely that the packet was inserted in the front of the buffer */
630 if (G_LIKELY (list)) {
631 item->prev = list->prev;
635 item->prev->next = item;
640 queue->tail = g_list_concat (queue->tail, item);
641 if (queue->tail->next)
642 queue->tail = queue->tail->next;
644 queue->head = queue->tail;
650 * rtp_jitter_buffer_insert:
651 * @jbuf: an #RTPJitterBuffer
652 * @item: an #RTPJitterBufferItem to insert
653 * @tail: TRUE when the tail element changed.
654 * @percent: the buffering percent after insertion
656 * Inserts @item into the packet queue of @jbuf. The sequence number of the
657 * packet will be used to sort the packets. This function takes ownerhip of
658 * @buf when the function returns %TRUE.
660 * Returns: %FALSE if a packet with the same number already existed.
663 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
664 gboolean * tail, gint * percent)
671 g_return_val_if_fail (jbuf != NULL, FALSE);
672 g_return_val_if_fail (item != NULL, FALSE);
674 /* no seqnum, simply append then */
675 if (item->seqnum == -1) {
679 seqnum = item->seqnum;
681 /* loop the list to skip strictly smaller seqnum buffers */
682 for (list = jbuf->packets->head; list; list = g_list_next (list)) {
685 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
687 if (qitem->seqnum == -1)
690 qseq = qitem->seqnum;
692 /* compare the new seqnum to the one in the buffer */
693 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
695 /* we hit a packet with the same seqnum, notify a duplicate */
696 if (G_UNLIKELY (gap == 0))
699 /* seqnum < qseq, we can stop looking */
700 if (G_LIKELY (gap > 0))
705 if (item->rtptime == -1)
708 rtptime = item->rtptime;
710 /* rtp time jumps are checked for during skew calculation, but bypassed
711 * in other mode, so mind those here and reset jb if needed.
712 * Only reset if valid input time, which is likely for UDP input
713 * where we expect this might happen due to async thread effects
714 * (in seek and state change cycles), but not so much for TCP input */
715 if (GST_CLOCK_TIME_IS_VALID (dts) &&
716 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
717 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
718 GstClockTime ext_rtptime = jbuf->ext_rtptime;
720 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
721 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
722 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
723 /* reset even if we don't have valid incoming time;
724 * still better than producing possibly very bogus output timestamp */
725 GST_WARNING ("rtp delta too big, reset skew");
726 rtp_jitter_buffer_reset_skew (jbuf);
730 switch (jbuf->mode) {
731 case RTP_JITTER_BUFFER_MODE_NONE:
732 case RTP_JITTER_BUFFER_MODE_BUFFER:
733 /* send 0 as the first timestamp and -1 for the other ones. This will
734 * interpollate them from the RTP timestamps with a 0 origin. In buffering
735 * mode we will adjust the outgoing timestamps according to the amount of
736 * time we spent buffering. */
737 if (jbuf->base_time == -1)
742 case RTP_JITTER_BUFFER_MODE_SYNCED:
743 /* synchronized clocks, take first timestamp as base, use RTP timestamps
745 if (jbuf->base_time != -1)
748 case RTP_JITTER_BUFFER_MODE_SLAVE:
752 /* do skew calculation by measuring the difference between rtptime and the
753 * receive dts, this function will return the skew corrected rtptime. */
754 item->pts = calculate_skew (jbuf, rtptime, dts);
757 queue_do_insert (jbuf, list, (GList *) item);
759 /* buffering mode, update buffer stats */
760 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
761 update_buffer_level (jbuf, percent);
765 /* tail was changed when we did not find a previous packet, we set the return
766 * flag when requested. */
768 *tail = (list == NULL);
775 GST_WARNING ("duplicate packet %d found", (gint) seqnum);
781 * rtp_jitter_buffer_pop:
782 * @jbuf: an #RTPJitterBuffer
783 * @percent: the buffering percent
785 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
786 * have its timestamp adjusted with the incomming running_time and the detected
789 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
791 RTPJitterBufferItem *
792 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
797 g_return_val_if_fail (jbuf != NULL, NULL);
799 queue = jbuf->packets;
803 queue->head = item->next;
805 queue->head->prev = NULL;
811 /* buffering mode, update buffer stats */
812 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
813 update_buffer_level (jbuf, percent);
817 return (RTPJitterBufferItem *) item;
821 * rtp_jitter_buffer_peek:
822 * @jbuf: an #RTPJitterBuffer
824 * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
825 * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
826 * was inserted in the queue.
828 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
830 RTPJitterBufferItem *
831 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
833 g_return_val_if_fail (jbuf != NULL, NULL);
835 return (RTPJitterBufferItem *) jbuf->packets->head;
839 * rtp_jitter_buffer_flush:
840 * @jbuf: an #RTPJitterBuffer
841 * @free_func: function to free each item
842 * @user_data: user data passed to @free_func
844 * Flush all packets from the jitterbuffer.
847 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
852 g_return_if_fail (jbuf != NULL);
853 g_return_if_fail (free_func != NULL);
855 while ((item = g_queue_pop_head_link (jbuf->packets)))
856 free_func ((RTPJitterBufferItem *) item, user_data);
860 * rtp_jitter_buffer_is_buffering:
861 * @jbuf: an #RTPJitterBuffer
863 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
864 * pop packets while in buffering mode.
866 * Returns: the buffering state of @jbuf
869 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
871 return jbuf->buffering && !jbuf->buffering_disabled;
875 * rtp_jitter_buffer_set_buffering:
876 * @jbuf: an #RTPJitterBuffer
877 * @buffering: the new buffering state
879 * Forces @jbuf to go into the buffering state.
882 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
884 jbuf->buffering = buffering;
888 * rtp_jitter_buffer_get_percent:
889 * @jbuf: an #RTPJitterBuffer
891 * Get the buffering percent of the jitterbuffer.
893 * Returns: the buffering percent
896 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
901 if (G_UNLIKELY (jbuf->high_level == 0))
904 if (G_UNLIKELY (jbuf->buffering_disabled))
907 level = get_buffer_level (jbuf);
908 percent = (level * 100 / jbuf->high_level);
909 percent = MIN (percent, 100);
915 * rtp_jitter_buffer_num_packets:
916 * @jbuf: an #RTPJitterBuffer
918 * Get the number of packets currently in "jbuf.
920 * Returns: The number of packets in @jbuf.
923 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
925 g_return_val_if_fail (jbuf != NULL, 0);
927 return jbuf->packets->length;
931 * rtp_jitter_buffer_get_ts_diff:
932 * @jbuf: an #RTPJitterBuffer
934 * Get the difference between the timestamps of first and last packet in the
937 * Returns: The difference expressed in the timestamp units of the packets.
940 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
942 guint64 high_ts, low_ts;
943 RTPJitterBufferItem *high_buf, *low_buf;
946 g_return_val_if_fail (jbuf != NULL, 0);
948 high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
949 low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
951 if (!high_buf || !low_buf || high_buf == low_buf)
954 high_ts = high_buf->rtptime;
955 low_ts = low_buf->rtptime;
957 /* it needs to work if ts wraps */
958 if (high_ts >= low_ts) {
959 result = (guint32) (high_ts - low_ts);
961 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
967 * rtp_jitter_buffer_get_sync:
968 * @jbuf: an #RTPJitterBuffer
969 * @rtptime: result RTP time
970 * @timestamp: result GStreamer timestamp
971 * @clock_rate: clock-rate of @rtptime
972 * @last_rtptime: last seen rtptime.
974 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
975 * used for constructing timestamps.
977 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
978 * the GStreamer timestamp is currently @timestamp.
980 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
984 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
985 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
988 *rtptime = jbuf->base_extrtp;
990 *timestamp = jbuf->base_time + jbuf->skew;
992 *clock_rate = jbuf->clock_rate;
994 *last_rtptime = jbuf->last_rtptime;