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_resync (RTPJitterBuffer * jbuf, GstClockTime time,
231 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
233 jbuf->base_time = time;
234 jbuf->base_rtptime = gstrtptime;
235 jbuf->base_extrtp = ext_rtptime;
236 jbuf->prev_out_time = -1;
237 jbuf->prev_send_diff = -1;
239 jbuf->window_filling = TRUE;
240 jbuf->window_pos = 0;
241 jbuf->window_min = 0;
242 jbuf->window_size = 0;
248 get_buffer_level (RTPJitterBuffer * jbuf)
250 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
253 /* first first buffer with timestamp */
254 high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
256 if (high_buf->dts != -1)
259 high_buf = (RTPJitterBufferItem *) g_list_next (high_buf);
262 low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
264 if (low_buf->dts != -1)
267 low_buf = (RTPJitterBufferItem *) g_list_previous (low_buf);
270 if (!high_buf || !low_buf || high_buf == low_buf) {
273 guint64 high_ts, low_ts;
275 high_ts = high_buf->dts;
276 low_ts = low_buf->dts;
278 if (high_ts > low_ts)
279 level = high_ts - low_ts;
283 GST_LOG_OBJECT (jbuf,
284 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
285 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
292 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
294 gboolean post = FALSE;
297 level = get_buffer_level (jbuf);
298 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
300 if (jbuf->buffering) {
302 if (level > jbuf->high_level) {
303 GST_DEBUG ("buffering finished");
304 jbuf->buffering = FALSE;
307 if (level < jbuf->low_level) {
308 GST_DEBUG ("buffering started");
309 jbuf->buffering = TRUE;
316 if (jbuf->buffering && (jbuf->high_level != 0)) {
317 perc = (level * 100 / jbuf->high_level);
318 perc = MIN (perc, 100);
326 GST_DEBUG ("buffering %d", perc);
330 /* For the clock skew we use a windowed low point averaging algorithm as can be
331 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
332 * over Network Delays":
333 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
334 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
336 * The idea is that the jitter is composed of:
340 * N : a constant network delay.
341 * n : random added noise. The noise is concentrated around 0
343 * In the receiver we can track the elapsed time at the sender with:
345 * send_diff(i) = (Tsi - Ts0);
347 * Tsi : The time at the sender at packet i
348 * Ts0 : The time at the sender at the first packet
350 * This is the difference between the RTP timestamp in the first received packet
351 * and the current packet.
353 * At the receiver we have to deal with the jitter introduced by the network.
355 * recv_diff(i) = (Tri - Tr0)
357 * Tri : The time at the receiver at packet i
358 * Tr0 : The time at the receiver at the first packet
360 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
363 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
365 * Cri : The time of the clock at the receiver for packet i
366 * D + ni : The jitter when receiving packet i
368 * We see that the network delay is irrelevant here as we can elliminate D:
370 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
372 * The drift is now expressed as:
374 * Drift(i) = recv_diff(i) - send_diff(i);
376 * We now keep the W latest values of Drift and find the minimum (this is the
377 * one with the lowest network jitter and thus the one which is least affected
378 * by it). We average this lowest value to smooth out the resulting network skew.
380 * Both the window and the weighting used for averaging influence the accuracy
381 * of the drift estimation. Finding the correct parameters turns out to be a
382 * compromise between accuracy and inertia.
384 * We use a 2 second window or up to 512 data points, which is statistically big
385 * enough to catch spikes (FIXME, detect spikes).
386 * We also use a rather large weighting factor (125) to smoothly adapt. During
387 * startup, when filling the window, we use a parabolic weighting factor, the
388 * more the window is filled, the faster we move to the detected possible skew.
390 * Returns: @time adjusted with the clock skew.
393 calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
396 guint64 send_diff, recv_diff;
400 GstClockTime gstrtptime, out_time;
403 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
405 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
406 return jbuf->prev_out_time;
409 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
411 /* keep track of the last extended rtptime */
412 jbuf->last_rtptime = ext_rtptime;
414 /* first time, lock on to time and gstrtptime */
415 if (G_UNLIKELY (jbuf->base_time == -1)) {
416 jbuf->base_time = time;
417 jbuf->prev_out_time = -1;
418 GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
420 if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
421 jbuf->base_rtptime = gstrtptime;
422 jbuf->base_extrtp = ext_rtptime;
423 jbuf->prev_send_diff = -1;
424 GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
425 GST_TIME_ARGS (gstrtptime));
428 if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
429 send_diff = gstrtptime - jbuf->base_rtptime;
430 else if (time != -1) {
431 /* elapsed time at sender, timestamps can go backwards and thus be smaller
432 * than our base time, take a new base time in that case. */
433 GST_WARNING ("backward timestamps at server, taking new base time");
434 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
437 GST_WARNING ("backward timestamps at server but no timestamps");
439 /* at least try to get a new timestamp.. */
440 jbuf->base_time = -1;
443 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
444 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
445 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
446 GST_TIME_ARGS (send_diff));
448 /* we don't have an arrival timestamp so we can't do skew detection. we
449 * should still apply a timestamp based on RTP timestamp and base_time */
450 if (time == -1 || jbuf->base_time == -1)
453 /* elapsed time at receiver, includes the jitter */
454 recv_diff = time - jbuf->base_time;
456 /* measure the diff */
457 delta = ((gint64) recv_diff) - ((gint64) send_diff);
459 /* measure the slope, this gives a rought estimate between the sender speed
460 * and the receiver speed. This should be approximately 8, higher values
461 * indicate a burst (especially when the connection starts) */
463 slope = (send_diff * 8) / recv_diff;
467 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
468 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
469 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
471 /* if the difference between the sender timeline and the receiver timeline
472 * changed too quickly we have to resync because the server likely restarted
474 if (ABS (delta - jbuf->skew) > GST_SECOND) {
475 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
476 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
477 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
482 pos = jbuf->window_pos;
484 if (G_UNLIKELY (jbuf->window_filling)) {
485 /* we are filling the window */
486 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
487 jbuf->window[pos++] = delta;
488 /* calc the min delta we observed */
489 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
490 jbuf->window_min = delta;
492 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
493 jbuf->window_size = pos;
496 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
498 /* the skew is now the min */
499 jbuf->skew = jbuf->window_min;
500 jbuf->window_filling = FALSE;
502 gint perc_time, perc_window, perc;
504 /* figure out how much we filled the window, this depends on the amount of
505 * time we have or the max number of points we keep. */
506 perc_time = send_diff * 100 / MAX_TIME;
507 perc_window = pos * 100 / MAX_WINDOW;
508 perc = MAX (perc_time, perc_window);
510 /* make a parabolic function, the closer we get to the MAX, the more value
511 * we give to the scaling factor of the new value */
514 /* quickly go to the min value when we are filling up, slowly when we are
515 * just starting because we're not sure it's a good value yet. */
517 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
518 jbuf->window_size = pos + 1;
521 /* pick old value and store new value. We keep the previous value in order
522 * to quickly check if the min of the window changed */
523 old = jbuf->window[pos];
524 jbuf->window[pos++] = delta;
526 if (G_UNLIKELY (delta <= jbuf->window_min)) {
527 /* if the new value we inserted is smaller or equal to the current min,
528 * it becomes the new min */
529 jbuf->window_min = delta;
530 } else if (G_UNLIKELY (old == jbuf->window_min)) {
531 gint64 min = G_MAXINT64;
533 /* if we removed the old min, we have to find a new min */
534 for (i = 0; i < jbuf->window_size; i++) {
535 /* we found another value equal to the old min, we can stop searching now */
536 if (jbuf->window[i] == old) {
540 if (jbuf->window[i] < min)
541 min = jbuf->window[i];
543 jbuf->window_min = min;
545 /* average the min values */
546 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
547 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
548 delta, jbuf->window_min);
550 /* wrap around in the window */
551 if (G_UNLIKELY (pos >= jbuf->window_size))
553 jbuf->window_pos = pos;
556 /* the output time is defined as the base timestamp plus the RTP time
557 * adjusted for the clock skew .*/
558 if (jbuf->base_time != -1) {
559 out_time = jbuf->base_time + send_diff;
560 /* skew can be negative and we don't want to make invalid timestamps */
561 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
564 out_time += jbuf->skew;
566 /* check if timestamps are not going backwards, we can only check this if we
567 * have a previous out time and a previous send_diff */
568 if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
569 /* now check for backwards timestamps */
571 /* if the server timestamps went up and the out_time backwards */
572 (send_diff > jbuf->prev_send_diff
573 && out_time < jbuf->prev_out_time) ||
574 /* if the server timestamps went backwards and the out_time forwards */
575 (send_diff < jbuf->prev_send_diff
576 && out_time > jbuf->prev_out_time) ||
577 /* if the server timestamps did not change */
578 send_diff == jbuf->prev_send_diff)) {
579 GST_DEBUG ("backwards timestamps, using previous time");
580 out_time = jbuf->prev_out_time;
583 if (time != -1 && out_time + jbuf->delay < time) {
584 /* if we are going to produce a timestamp that is later than the input
585 * timestamp, we need to reset the jitterbuffer. Likely the server paused
587 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
588 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
589 jbuf->delay, GST_TIME_ARGS (time));
590 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
597 jbuf->prev_out_time = out_time;
598 jbuf->prev_send_diff = send_diff;
600 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
601 jbuf->skew, GST_TIME_ARGS (out_time));
607 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
609 GQueue *queue = jbuf->packets;
611 /* It's more likely that the packet was inserted in the front of the buffer */
612 if (G_LIKELY (list)) {
613 item->prev = list->prev;
617 item->prev->next = item;
622 queue->tail = g_list_concat (queue->tail, item);
623 if (queue->tail->next)
624 queue->tail = queue->tail->next;
626 queue->head = queue->tail;
632 * rtp_jitter_buffer_insert:
633 * @jbuf: an #RTPJitterBuffer
634 * @item: an #RTPJitterBufferItem to insert
635 * @tail: TRUE when the tail element changed.
636 * @percent: the buffering percent after insertion
638 * Inserts @item into the packet queue of @jbuf. The sequence number of the
639 * packet will be used to sort the packets. This function takes ownerhip of
640 * @buf when the function returns %TRUE.
642 * Returns: %FALSE if a packet with the same number already existed.
645 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
646 gboolean * tail, gint * percent)
653 g_return_val_if_fail (jbuf != NULL, FALSE);
654 g_return_val_if_fail (item != NULL, FALSE);
656 seqnum = item->seqnum;
657 /* no seqnum, simply append then */
661 /* loop the list to skip strictly smaller seqnum buffers */
662 for (list = jbuf->packets->head; list; list = g_list_next (list)) {
665 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
667 qseq = qitem->seqnum;
671 /* compare the new seqnum to the one in the buffer */
672 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
674 /* we hit a packet with the same seqnum, notify a duplicate */
675 if (G_UNLIKELY (gap == 0))
678 /* seqnum > qseq, we can stop looking */
679 if (G_LIKELY (gap < 0))
684 rtptime = item->rtptime;
689 /* rtp time jumps are checked for during skew calculation, but bypassed
690 * in other mode, so mind those here and reset jb if needed.
691 * Only reset if valid input time, which is likely for UDP input
692 * where we expect this might happen due to async thread effects
693 * (in seek and state change cycles), but not so much for TCP input */
694 if (GST_CLOCK_TIME_IS_VALID (dts) &&
695 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
696 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
697 GstClockTime ext_rtptime = jbuf->ext_rtptime;
699 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
700 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
701 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
702 /* reset even if we don't have valid incoming time;
703 * still better than producing possibly very bogus output timestamp */
704 GST_WARNING ("rtp delta too big, reset skew");
705 rtp_jitter_buffer_reset_skew (jbuf);
709 switch (jbuf->mode) {
710 case RTP_JITTER_BUFFER_MODE_NONE:
711 case RTP_JITTER_BUFFER_MODE_BUFFER:
712 /* send 0 as the first timestamp and -1 for the other ones. This will
713 * interpollate them from the RTP timestamps with a 0 origin. In buffering
714 * mode we will adjust the outgoing timestamps according to the amount of
715 * time we spent buffering. */
716 if (jbuf->base_time == -1)
721 case RTP_JITTER_BUFFER_MODE_SYNCED:
722 /* synchronized clocks, take first timestamp as base, use RTP timestamps
724 if (jbuf->base_time != -1)
727 case RTP_JITTER_BUFFER_MODE_SLAVE:
731 /* do skew calculation by measuring the difference between rtptime and the
732 * receive dts, this function will return the skew corrected rtptime. */
733 item->pts = calculate_skew (jbuf, rtptime, dts);
736 queue_do_insert (jbuf, list, (GList *) item);
738 /* buffering mode, update buffer stats */
739 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
740 update_buffer_level (jbuf, percent);
744 /* tail was changed when we did not find a previous packet, we set the return
745 * flag when requested. */
747 *tail = (list == NULL);
754 GST_WARNING ("duplicate packet %d found", (gint) seqnum);
760 * rtp_jitter_buffer_pop:
761 * @jbuf: an #RTPJitterBuffer
762 * @percent: the buffering percent
764 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
765 * have its timestamp adjusted with the incomming running_time and the detected
768 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
770 RTPJitterBufferItem *
771 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
776 g_return_val_if_fail (jbuf != NULL, NULL);
778 queue = jbuf->packets;
782 queue->tail = item->prev;
784 queue->tail->next = NULL;
790 /* buffering mode, update buffer stats */
791 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
792 update_buffer_level (jbuf, percent);
796 return (RTPJitterBufferItem *) item;
800 * rtp_jitter_buffer_peek:
801 * @jbuf: an #RTPJitterBuffer
803 * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
804 * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
805 * was inserted in the queue.
807 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
809 RTPJitterBufferItem *
810 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
812 g_return_val_if_fail (jbuf != NULL, NULL);
814 return (RTPJitterBufferItem *) jbuf->packets->tail;
818 * rtp_jitter_buffer_flush:
819 * @jbuf: an #RTPJitterBuffer
820 * @free_func: function to free each item
821 * @user_data: user data passed to @free_func
823 * Flush all packets from the jitterbuffer.
826 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
831 g_return_if_fail (jbuf != NULL);
832 g_return_if_fail (free_func != NULL);
834 while ((item = g_queue_pop_head_link (jbuf->packets)))
835 free_func ((RTPJitterBufferItem *) item, user_data);
839 * rtp_jitter_buffer_is_buffering:
840 * @jbuf: an #RTPJitterBuffer
842 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
843 * pop packets while in buffering mode.
845 * Returns: the buffering state of @jbuf
848 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
850 return jbuf->buffering;
854 * rtp_jitter_buffer_set_buffering:
855 * @jbuf: an #RTPJitterBuffer
856 * @buffering: the new buffering state
858 * Forces @jbuf to go into the buffering state.
861 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
863 jbuf->buffering = buffering;
867 * rtp_jitter_buffer_get_percent:
868 * @jbuf: an #RTPJitterBuffer
870 * Get the buffering percent of the jitterbuffer.
872 * Returns: the buffering percent
875 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
880 if (G_UNLIKELY (jbuf->high_level == 0))
883 level = get_buffer_level (jbuf);
884 percent = (level * 100 / jbuf->high_level);
885 percent = MIN (percent, 100);
891 * rtp_jitter_buffer_num_packets:
892 * @jbuf: an #RTPJitterBuffer
894 * Get the number of packets currently in "jbuf.
896 * Returns: The number of packets in @jbuf.
899 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
901 g_return_val_if_fail (jbuf != NULL, 0);
903 return jbuf->packets->length;
907 * rtp_jitter_buffer_get_ts_diff:
908 * @jbuf: an #RTPJitterBuffer
910 * Get the difference between the timestamps of first and last packet in the
913 * Returns: The difference expressed in the timestamp units of the packets.
916 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
918 guint64 high_ts, low_ts;
919 GstBuffer *high_buf, *low_buf;
921 GstRTPBuffer rtp = { NULL };
923 g_return_val_if_fail (jbuf != NULL, 0);
925 high_buf = g_queue_peek_head (jbuf->packets);
926 low_buf = g_queue_peek_tail (jbuf->packets);
928 if (!high_buf || !low_buf || high_buf == low_buf)
931 gst_rtp_buffer_map (high_buf, GST_MAP_READ, &rtp);
932 high_ts = gst_rtp_buffer_get_timestamp (&rtp);
933 gst_rtp_buffer_unmap (&rtp);
934 gst_rtp_buffer_map (low_buf, GST_MAP_READ, &rtp);
935 low_ts = gst_rtp_buffer_get_timestamp (&rtp);
936 gst_rtp_buffer_unmap (&rtp);
938 /* it needs to work if ts wraps */
939 if (high_ts >= low_ts) {
940 result = (guint32) (high_ts - low_ts);
942 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
948 * rtp_jitter_buffer_get_sync:
949 * @jbuf: an #RTPJitterBuffer
950 * @rtptime: result RTP time
951 * @timestamp: result GStreamer timestamp
952 * @clock_rate: clock-rate of @rtptime
953 * @last_rtptime: last seen rtptime.
955 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
956 * used for constructing timestamps.
958 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
959 * the GStreamer timestamp is currently @timestamp.
961 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
965 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
966 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
969 *rtptime = jbuf->base_extrtp;
971 *timestamp = jbuf->base_time + jbuf->skew;
973 *clock_rate = jbuf->clock_rate;
975 *last_rtptime = jbuf->last_rtptime;