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 g_queue_free (jbuf->packets);
101 G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
105 * rtp_jitter_buffer_new:
107 * Create an #RTPJitterBuffer.
109 * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
112 rtp_jitter_buffer_new (void)
114 RTPJitterBuffer *jbuf;
116 jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
122 * rtp_jitter_buffer_get_mode:
123 * @jbuf: an #RTPJitterBuffer
125 * Get the current jitterbuffer mode.
127 * Returns: the current jitterbuffer mode.
130 rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
136 * rtp_jitter_buffer_set_mode:
137 * @jbuf: an #RTPJitterBuffer
138 * @mode: a #RTPJitterBufferMode
140 * Set the buffering and clock slaving algorithm used in the @jbuf.
143 rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
149 rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
155 rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
158 jbuf->low_level = (delay * 15) / 100;
159 /* the high level is at 90% in order to release packets before we fill up the
160 * buffer up to the latency */
161 jbuf->high_level = (delay * 90) / 100;
163 GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
164 GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
165 GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
169 * rtp_jitter_buffer_set_clock_rate:
170 * @jbuf: an #RTPJitterBuffer
172 * Set the clock rate in the jitterbuffer.
175 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
177 if (jbuf->clock_rate != clock_rate) {
178 if (jbuf->clock_rate == -1) {
179 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
180 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
182 GST_WARNING ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
183 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
185 jbuf->clock_rate = clock_rate;
186 rtp_jitter_buffer_reset_skew (jbuf);
191 * rtp_jitter_buffer_get_clock_rate:
192 * @jbuf: an #RTPJitterBuffer
194 * Get the currently configure clock rate in @jbuf.
196 * Returns: the current clock-rate
199 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
201 return jbuf->clock_rate;
205 * rtp_jitter_buffer_reset_skew:
206 * @jbuf: an #RTPJitterBuffer
208 * Reset the skew calculations in @jbuf.
211 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
213 jbuf->base_time = -1;
214 jbuf->base_rtptime = -1;
215 jbuf->base_extrtp = -1;
216 jbuf->ext_rtptime = -1;
217 jbuf->last_rtptime = -1;
218 jbuf->window_pos = 0;
219 jbuf->window_filling = TRUE;
220 jbuf->window_min = 0;
222 jbuf->prev_send_diff = -1;
223 jbuf->prev_out_time = -1;
224 GST_DEBUG ("reset skew correction");
228 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
229 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
231 jbuf->base_time = time;
232 jbuf->base_rtptime = gstrtptime;
233 jbuf->base_extrtp = ext_rtptime;
234 jbuf->prev_out_time = -1;
235 jbuf->prev_send_diff = -1;
237 jbuf->window_filling = TRUE;
238 jbuf->window_pos = 0;
239 jbuf->window_min = 0;
240 jbuf->window_size = 0;
246 get_buffer_level (RTPJitterBuffer * jbuf)
248 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
251 /* first first buffer with timestamp */
252 high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
254 if (high_buf->dts != -1)
257 high_buf = (RTPJitterBufferItem *) g_list_next (high_buf);
260 low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
262 if (low_buf->dts != -1)
265 low_buf = (RTPJitterBufferItem *) g_list_previous (low_buf);
268 if (!high_buf || !low_buf || high_buf == low_buf) {
271 guint64 high_ts, low_ts;
273 high_ts = high_buf->dts;
274 low_ts = low_buf->dts;
276 if (high_ts > low_ts)
277 level = high_ts - low_ts;
281 GST_LOG_OBJECT (jbuf,
282 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
283 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
290 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
292 gboolean post = FALSE;
295 level = get_buffer_level (jbuf);
296 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
298 if (jbuf->buffering) {
300 if (level > jbuf->high_level) {
301 GST_DEBUG ("buffering finished");
302 jbuf->buffering = FALSE;
305 if (level < jbuf->low_level) {
306 GST_DEBUG ("buffering started");
307 jbuf->buffering = TRUE;
314 if (jbuf->buffering && (jbuf->high_level != 0)) {
315 perc = (level * 100 / jbuf->high_level);
316 perc = MIN (perc, 100);
324 GST_DEBUG ("buffering %d", perc);
328 /* For the clock skew we use a windowed low point averaging algorithm as can be
329 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
330 * over Network Delays":
331 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
332 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
334 * The idea is that the jitter is composed of:
338 * N : a constant network delay.
339 * n : random added noise. The noise is concentrated around 0
341 * In the receiver we can track the elapsed time at the sender with:
343 * send_diff(i) = (Tsi - Ts0);
345 * Tsi : The time at the sender at packet i
346 * Ts0 : The time at the sender at the first packet
348 * This is the difference between the RTP timestamp in the first received packet
349 * and the current packet.
351 * At the receiver we have to deal with the jitter introduced by the network.
353 * recv_diff(i) = (Tri - Tr0)
355 * Tri : The time at the receiver at packet i
356 * Tr0 : The time at the receiver at the first packet
358 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
361 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
363 * Cri : The time of the clock at the receiver for packet i
364 * D + ni : The jitter when receiving packet i
366 * We see that the network delay is irrelevant here as we can elliminate D:
368 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
370 * The drift is now expressed as:
372 * Drift(i) = recv_diff(i) - send_diff(i);
374 * We now keep the W latest values of Drift and find the minimum (this is the
375 * one with the lowest network jitter and thus the one which is least affected
376 * by it). We average this lowest value to smooth out the resulting network skew.
378 * Both the window and the weighting used for averaging influence the accuracy
379 * of the drift estimation. Finding the correct parameters turns out to be a
380 * compromise between accuracy and inertia.
382 * We use a 2 second window or up to 512 data points, which is statistically big
383 * enough to catch spikes (FIXME, detect spikes).
384 * We also use a rather large weighting factor (125) to smoothly adapt. During
385 * startup, when filling the window, we use a parabolic weighting factor, the
386 * more the window is filled, the faster we move to the detected possible skew.
388 * Returns: @time adjusted with the clock skew.
391 calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
394 guint64 send_diff, recv_diff;
398 GstClockTime gstrtptime, out_time;
401 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
403 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
404 return jbuf->prev_out_time;
407 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
409 /* keep track of the last extended rtptime */
410 jbuf->last_rtptime = ext_rtptime;
412 /* first time, lock on to time and gstrtptime */
413 if (G_UNLIKELY (jbuf->base_time == -1)) {
414 jbuf->base_time = time;
415 jbuf->prev_out_time = -1;
416 GST_DEBUG ("Taking new base time %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
418 if (G_UNLIKELY (jbuf->base_rtptime == -1)) {
419 jbuf->base_rtptime = gstrtptime;
420 jbuf->base_extrtp = ext_rtptime;
421 jbuf->prev_send_diff = -1;
422 GST_DEBUG ("Taking new base rtptime %" GST_TIME_FORMAT,
423 GST_TIME_ARGS (gstrtptime));
426 if (G_LIKELY (gstrtptime >= jbuf->base_rtptime))
427 send_diff = gstrtptime - jbuf->base_rtptime;
428 else if (time != -1) {
429 /* elapsed time at sender, timestamps can go backwards and thus be smaller
430 * than our base time, take a new base time in that case. */
431 GST_WARNING ("backward timestamps at server, taking new base time");
432 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
435 GST_WARNING ("backward timestamps at server but no timestamps");
437 /* at least try to get a new timestamp.. */
438 jbuf->base_time = -1;
441 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
442 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
443 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
444 GST_TIME_ARGS (send_diff));
446 /* we don't have an arrival timestamp so we can't do skew detection. we
447 * should still apply a timestamp based on RTP timestamp and base_time */
448 if (time == -1 || jbuf->base_time == -1)
451 /* elapsed time at receiver, includes the jitter */
452 recv_diff = time - jbuf->base_time;
454 /* measure the diff */
455 delta = ((gint64) recv_diff) - ((gint64) send_diff);
457 /* measure the slope, this gives a rought estimate between the sender speed
458 * and the receiver speed. This should be approximately 8, higher values
459 * indicate a burst (especially when the connection starts) */
461 slope = (send_diff * 8) / recv_diff;
465 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
466 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
467 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
469 /* if the difference between the sender timeline and the receiver timeline
470 * changed too quickly we have to resync because the server likely restarted
472 if (ABS (delta - jbuf->skew) > GST_SECOND) {
473 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
474 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
475 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
480 pos = jbuf->window_pos;
482 if (G_UNLIKELY (jbuf->window_filling)) {
483 /* we are filling the window */
484 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
485 jbuf->window[pos++] = delta;
486 /* calc the min delta we observed */
487 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
488 jbuf->window_min = delta;
490 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
491 jbuf->window_size = pos;
494 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
496 /* the skew is now the min */
497 jbuf->skew = jbuf->window_min;
498 jbuf->window_filling = FALSE;
500 gint perc_time, perc_window, perc;
502 /* figure out how much we filled the window, this depends on the amount of
503 * time we have or the max number of points we keep. */
504 perc_time = send_diff * 100 / MAX_TIME;
505 perc_window = pos * 100 / MAX_WINDOW;
506 perc = MAX (perc_time, perc_window);
508 /* make a parabolic function, the closer we get to the MAX, the more value
509 * we give to the scaling factor of the new value */
512 /* quickly go to the min value when we are filling up, slowly when we are
513 * just starting because we're not sure it's a good value yet. */
515 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
516 jbuf->window_size = pos + 1;
519 /* pick old value and store new value. We keep the previous value in order
520 * to quickly check if the min of the window changed */
521 old = jbuf->window[pos];
522 jbuf->window[pos++] = delta;
524 if (G_UNLIKELY (delta <= jbuf->window_min)) {
525 /* if the new value we inserted is smaller or equal to the current min,
526 * it becomes the new min */
527 jbuf->window_min = delta;
528 } else if (G_UNLIKELY (old == jbuf->window_min)) {
529 gint64 min = G_MAXINT64;
531 /* if we removed the old min, we have to find a new min */
532 for (i = 0; i < jbuf->window_size; i++) {
533 /* we found another value equal to the old min, we can stop searching now */
534 if (jbuf->window[i] == old) {
538 if (jbuf->window[i] < min)
539 min = jbuf->window[i];
541 jbuf->window_min = min;
543 /* average the min values */
544 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
545 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
546 delta, jbuf->window_min);
548 /* wrap around in the window */
549 if (G_UNLIKELY (pos >= jbuf->window_size))
551 jbuf->window_pos = pos;
554 /* the output time is defined as the base timestamp plus the RTP time
555 * adjusted for the clock skew .*/
556 if (jbuf->base_time != -1) {
557 out_time = jbuf->base_time + send_diff;
558 /* skew can be negative and we don't want to make invalid timestamps */
559 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
562 out_time += jbuf->skew;
564 /* check if timestamps are not going backwards, we can only check this if we
565 * have a previous out time and a previous send_diff */
566 if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
567 /* now check for backwards timestamps */
569 /* if the server timestamps went up and the out_time backwards */
570 (send_diff > jbuf->prev_send_diff
571 && out_time < jbuf->prev_out_time) ||
572 /* if the server timestamps went backwards and the out_time forwards */
573 (send_diff < jbuf->prev_send_diff
574 && out_time > jbuf->prev_out_time) ||
575 /* if the server timestamps did not change */
576 send_diff == jbuf->prev_send_diff)) {
577 GST_DEBUG ("backwards timestamps, using previous time");
578 out_time = jbuf->prev_out_time;
581 if (time != -1 && out_time + jbuf->delay < time) {
582 /* if we are going to produce a timestamp that is later than the input
583 * timestamp, we need to reset the jitterbuffer. Likely the server paused
585 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
586 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
587 jbuf->delay, GST_TIME_ARGS (time));
588 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
595 jbuf->prev_out_time = out_time;
596 jbuf->prev_send_diff = send_diff;
598 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
599 jbuf->skew, GST_TIME_ARGS (out_time));
605 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
607 GQueue *queue = jbuf->packets;
609 /* It's more likely that the packet was inserted in the front of the buffer */
610 if (G_LIKELY (list)) {
611 item->prev = list->prev;
615 item->prev->next = item;
620 queue->tail = g_list_concat (queue->tail, item);
621 if (queue->tail->next)
622 queue->tail = queue->tail->next;
624 queue->head = queue->tail;
630 * rtp_jitter_buffer_insert:
631 * @jbuf: an #RTPJitterBuffer
632 * @item: an #RTPJitterBufferItem to insert
633 * @tail: TRUE when the tail element changed.
634 * @percent: the buffering percent after insertion
636 * Inserts @item into the packet queue of @jbuf. The sequence number of the
637 * packet will be used to sort the packets. This function takes ownerhip of
638 * @buf when the function returns %TRUE.
640 * Returns: %FALSE if a packet with the same number already existed.
643 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
644 gboolean * tail, gint * percent)
651 g_return_val_if_fail (jbuf != NULL, FALSE);
652 g_return_val_if_fail (item != NULL, FALSE);
654 seqnum = item->seqnum;
655 /* no seqnum, simply append then */
659 /* loop the list to skip strictly smaller seqnum buffers */
660 for (list = jbuf->packets->head; list; list = g_list_next (list)) {
663 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
665 qseq = qitem->seqnum;
669 /* compare the new seqnum to the one in the buffer */
670 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
672 /* we hit a packet with the same seqnum, notify a duplicate */
673 if (G_UNLIKELY (gap == 0))
676 /* seqnum > qseq, we can stop looking */
677 if (G_LIKELY (gap < 0))
682 rtptime = item->rtptime;
687 /* rtp time jumps are checked for during skew calculation, but bypassed
688 * in other mode, so mind those here and reset jb if needed.
689 * Only reset if valid input time, which is likely for UDP input
690 * where we expect this might happen due to async thread effects
691 * (in seek and state change cycles), but not so much for TCP input */
692 if (GST_CLOCK_TIME_IS_VALID (dts) &&
693 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
694 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
695 GstClockTime ext_rtptime = jbuf->ext_rtptime;
697 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
698 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
699 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
700 /* reset even if we don't have valid incoming time;
701 * still better than producing possibly very bogus output timestamp */
702 GST_WARNING ("rtp delta too big, reset skew");
703 rtp_jitter_buffer_reset_skew (jbuf);
707 switch (jbuf->mode) {
708 case RTP_JITTER_BUFFER_MODE_NONE:
709 case RTP_JITTER_BUFFER_MODE_BUFFER:
710 /* send 0 as the first timestamp and -1 for the other ones. This will
711 * interpollate them from the RTP timestamps with a 0 origin. In buffering
712 * mode we will adjust the outgoing timestamps according to the amount of
713 * time we spent buffering. */
714 if (jbuf->base_time == -1)
719 case RTP_JITTER_BUFFER_MODE_SLAVE:
723 /* do skew calculation by measuring the difference between rtptime and the
724 * receive dts, this function will return the skew corrected rtptime. */
725 item->pts = calculate_skew (jbuf, rtptime, dts);
728 queue_do_insert (jbuf, list, (GList *) item);
730 /* buffering mode, update buffer stats */
731 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
732 update_buffer_level (jbuf, percent);
736 /* tail was changed when we did not find a previous packet, we set the return
737 * flag when requested. */
739 *tail = (list == NULL);
746 GST_WARNING ("duplicate packet %d found", (gint) seqnum);
752 * rtp_jitter_buffer_pop:
753 * @jbuf: an #RTPJitterBuffer
754 * @percent: the buffering percent
756 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
757 * have its timestamp adjusted with the incomming running_time and the detected
760 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
762 RTPJitterBufferItem *
763 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
768 g_return_val_if_fail (jbuf != NULL, NULL);
770 queue = jbuf->packets;
774 queue->tail = item->prev;
776 queue->tail->next = NULL;
782 /* buffering mode, update buffer stats */
783 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
784 update_buffer_level (jbuf, percent);
788 return (RTPJitterBufferItem *) item;
792 * rtp_jitter_buffer_peek:
793 * @jbuf: an #RTPJitterBuffer
795 * Peek the oldest buffer from the packet queue of @jbuf. Register a callback
796 * with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
797 * was inserted in the queue.
799 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
801 RTPJitterBufferItem *
802 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
804 g_return_val_if_fail (jbuf != NULL, NULL);
806 return (RTPJitterBufferItem *) jbuf->packets->tail;
810 * rtp_jitter_buffer_flush:
811 * @jbuf: an #RTPJitterBuffer
812 * @free_func: function to free each item
813 * @user_data: user data passed to @free_func
815 * Flush all packets from the jitterbuffer.
818 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
823 g_return_if_fail (jbuf != NULL);
824 g_return_if_fail (free_func != NULL);
826 while ((item = g_queue_pop_head_link (jbuf->packets)))
827 free_func ((RTPJitterBufferItem *) item, user_data);
831 * rtp_jitter_buffer_is_buffering:
832 * @jbuf: an #RTPJitterBuffer
834 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
835 * pop packets while in buffering mode.
837 * Returns: the buffering state of @jbuf
840 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
842 return jbuf->buffering;
846 * rtp_jitter_buffer_set_buffering:
847 * @jbuf: an #RTPJitterBuffer
848 * @buffering: the new buffering state
850 * Forces @jbuf to go into the buffering state.
853 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
855 jbuf->buffering = buffering;
859 * rtp_jitter_buffer_get_percent:
860 * @jbuf: an #RTPJitterBuffer
862 * Get the buffering percent of the jitterbuffer.
864 * Returns: the buffering percent
867 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
872 if (G_UNLIKELY (jbuf->high_level == 0))
875 level = get_buffer_level (jbuf);
876 percent = (level * 100 / jbuf->high_level);
877 percent = MIN (percent, 100);
883 * rtp_jitter_buffer_num_packets:
884 * @jbuf: an #RTPJitterBuffer
886 * Get the number of packets currently in "jbuf.
888 * Returns: The number of packets in @jbuf.
891 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
893 g_return_val_if_fail (jbuf != NULL, 0);
895 return jbuf->packets->length;
899 * rtp_jitter_buffer_get_ts_diff:
900 * @jbuf: an #RTPJitterBuffer
902 * Get the difference between the timestamps of first and last packet in the
905 * Returns: The difference expressed in the timestamp units of the packets.
908 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
910 guint64 high_ts, low_ts;
911 GstBuffer *high_buf, *low_buf;
913 GstRTPBuffer rtp = { NULL };
915 g_return_val_if_fail (jbuf != NULL, 0);
917 high_buf = g_queue_peek_head (jbuf->packets);
918 low_buf = g_queue_peek_tail (jbuf->packets);
920 if (!high_buf || !low_buf || high_buf == low_buf)
923 gst_rtp_buffer_map (high_buf, GST_MAP_READ, &rtp);
924 high_ts = gst_rtp_buffer_get_timestamp (&rtp);
925 gst_rtp_buffer_unmap (&rtp);
926 gst_rtp_buffer_map (low_buf, GST_MAP_READ, &rtp);
927 low_ts = gst_rtp_buffer_get_timestamp (&rtp);
928 gst_rtp_buffer_unmap (&rtp);
930 /* it needs to work if ts wraps */
931 if (high_ts >= low_ts) {
932 result = (guint32) (high_ts - low_ts);
934 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
940 * rtp_jitter_buffer_get_sync:
941 * @jbuf: an #RTPJitterBuffer
942 * @rtptime: result RTP time
943 * @timestamp: result GStreamer timestamp
944 * @clock_rate: clock-rate of @rtptime
945 * @last_rtptime: last seen rtptime.
947 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
948 * used for constructing timestamps.
950 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
951 * the GStreamer timestamp is currently @timestamp.
953 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
957 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
958 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
961 *rtptime = jbuf->base_extrtp;
963 *timestamp = jbuf->base_time + jbuf->skew;
965 *clock_rate = jbuf->clock_rate;
967 *last_rtptime = jbuf->last_rtptime;