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 g_mutex_init (&jbuf->clock_lock);
90 jbuf->packets = g_queue_new ();
91 jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
93 rtp_jitter_buffer_reset_skew (jbuf);
97 rtp_jitter_buffer_finalize (GObject * object)
99 RTPJitterBuffer *jbuf;
101 jbuf = RTP_JITTER_BUFFER_CAST (object);
103 if (jbuf->media_clock_synced_id)
104 g_signal_handler_disconnect (jbuf->media_clock,
105 jbuf->media_clock_synced_id);
106 if (jbuf->media_clock) {
107 /* Make sure to clear any clock master before releasing the clock */
108 gst_clock_set_master (jbuf->media_clock, NULL);
109 gst_object_unref (jbuf->media_clock);
112 if (jbuf->pipeline_clock)
113 gst_object_unref (jbuf->pipeline_clock);
115 g_queue_free (jbuf->packets);
117 g_mutex_clear (&jbuf->clock_lock);
119 G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
123 * rtp_jitter_buffer_new:
125 * Create an #RTPJitterBuffer.
127 * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
130 rtp_jitter_buffer_new (void)
132 RTPJitterBuffer *jbuf;
134 jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
140 * rtp_jitter_buffer_get_mode:
141 * @jbuf: an #RTPJitterBuffer
143 * Get the current jitterbuffer mode.
145 * Returns: the current jitterbuffer mode.
148 rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
154 * rtp_jitter_buffer_set_mode:
155 * @jbuf: an #RTPJitterBuffer
156 * @mode: a #RTPJitterBufferMode
158 * Set the buffering and clock slaving algorithm used in the @jbuf.
161 rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
167 rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
173 rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
176 jbuf->low_level = (delay * 15) / 100;
177 /* the high level is at 90% in order to release packets before we fill up the
178 * buffer up to the latency */
179 jbuf->high_level = (delay * 90) / 100;
181 GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
182 GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
183 GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
187 * rtp_jitter_buffer_set_clock_rate:
188 * @jbuf: an #RTPJitterBuffer
189 * @clock_rate: the new clock rate
191 * Set the clock rate in the jitterbuffer.
194 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
196 if (jbuf->clock_rate != clock_rate) {
197 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
198 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
199 jbuf->clock_rate = clock_rate;
200 rtp_jitter_buffer_reset_skew (jbuf);
205 * rtp_jitter_buffer_get_clock_rate:
206 * @jbuf: an #RTPJitterBuffer
208 * Get the currently configure clock rate in @jbuf.
210 * Returns: the current clock-rate
213 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
215 return jbuf->clock_rate;
219 media_clock_synced_cb (GstClock * clock, gboolean synced,
220 RTPJitterBuffer * jbuf)
222 GstClockTime internal, external;
224 g_mutex_lock (&jbuf->clock_lock);
225 if (jbuf->pipeline_clock) {
226 internal = gst_clock_get_internal_time (jbuf->media_clock);
227 external = gst_clock_get_time (jbuf->pipeline_clock);
229 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
231 g_mutex_unlock (&jbuf->clock_lock);
235 * rtp_jitter_buffer_set_media_clock:
236 * @jbuf: an #RTPJitterBuffer
237 * @clock: (transfer full): media #GstClock
238 * @clock_offset: RTP time at clock epoch or -1
240 * Sets the media clock for the media and the clock offset
244 rtp_jitter_buffer_set_media_clock (RTPJitterBuffer * jbuf, GstClock * clock,
245 guint64 clock_offset)
247 g_mutex_lock (&jbuf->clock_lock);
248 if (jbuf->media_clock) {
249 if (jbuf->media_clock_synced_id)
250 g_signal_handler_disconnect (jbuf->media_clock,
251 jbuf->media_clock_synced_id);
252 jbuf->media_clock_synced_id = 0;
253 gst_object_unref (jbuf->media_clock);
255 jbuf->media_clock = clock;
256 jbuf->media_clock_offset = clock_offset;
258 if (jbuf->pipeline_clock && jbuf->media_clock &&
259 jbuf->pipeline_clock != jbuf->media_clock) {
260 jbuf->media_clock_synced_id =
261 g_signal_connect (jbuf->media_clock, "synced",
262 G_CALLBACK (media_clock_synced_cb), jbuf);
263 if (gst_clock_is_synced (jbuf->media_clock)) {
264 GstClockTime internal, external;
266 internal = gst_clock_get_internal_time (jbuf->media_clock);
267 external = gst_clock_get_time (jbuf->pipeline_clock);
269 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
272 gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
274 g_mutex_unlock (&jbuf->clock_lock);
278 * rtp_jitter_buffer_set_pipeline_clock:
279 * @jbuf: an #RTPJitterBuffer
280 * @clock: pipeline #GstClock
282 * Sets the pipeline clock
286 rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer * jbuf, GstClock * clock)
288 g_mutex_lock (&jbuf->clock_lock);
289 if (jbuf->pipeline_clock)
290 gst_object_unref (jbuf->pipeline_clock);
291 jbuf->pipeline_clock = clock ? gst_object_ref (clock) : NULL;
293 if (jbuf->pipeline_clock && jbuf->media_clock &&
294 jbuf->pipeline_clock != jbuf->media_clock) {
295 if (gst_clock_is_synced (jbuf->media_clock)) {
296 GstClockTime internal, external;
298 internal = gst_clock_get_internal_time (jbuf->media_clock);
299 external = gst_clock_get_time (jbuf->pipeline_clock);
301 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
304 gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
306 g_mutex_unlock (&jbuf->clock_lock);
310 rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer * jbuf)
312 return jbuf->rfc7273_sync;
316 rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer * jbuf,
317 gboolean rfc7273_sync)
319 jbuf->rfc7273_sync = rfc7273_sync;
323 * rtp_jitter_buffer_reset_skew:
324 * @jbuf: an #RTPJitterBuffer
326 * Reset the skew calculations in @jbuf.
329 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
331 jbuf->base_time = -1;
332 jbuf->base_rtptime = -1;
333 jbuf->base_extrtp = -1;
334 jbuf->media_clock_base_time = -1;
335 jbuf->ext_rtptime = -1;
336 jbuf->last_rtptime = -1;
337 jbuf->window_pos = 0;
338 jbuf->window_filling = TRUE;
339 jbuf->window_min = 0;
341 jbuf->prev_send_diff = -1;
342 jbuf->prev_out_time = -1;
343 jbuf->need_resync = TRUE;
345 GST_DEBUG ("reset skew correction");
349 * rtp_jitter_buffer_disable_buffering:
350 * @jbuf: an #RTPJitterBuffer
351 * @disabled: the new state
353 * Enable or disable buffering on @jbuf.
356 rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
358 jbuf->buffering_disabled = disabled;
362 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
363 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
365 jbuf->base_time = time;
366 jbuf->media_clock_base_time = -1;
367 jbuf->base_rtptime = gstrtptime;
368 jbuf->base_extrtp = ext_rtptime;
369 jbuf->prev_out_time = -1;
370 jbuf->prev_send_diff = -1;
372 jbuf->window_filling = TRUE;
373 jbuf->window_pos = 0;
374 jbuf->window_min = 0;
375 jbuf->window_size = 0;
378 jbuf->need_resync = FALSE;
382 get_buffer_level (RTPJitterBuffer * jbuf)
384 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
387 /* first buffer with timestamp */
388 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
390 if (high_buf->dts != -1 || high_buf->pts != -1)
393 high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
396 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
398 if (low_buf->dts != -1 || low_buf->pts != -1)
401 low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
404 if (!high_buf || !low_buf || high_buf == low_buf) {
407 guint64 high_ts, low_ts;
409 high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
410 low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
412 if (high_ts > low_ts)
413 level = high_ts - low_ts;
417 GST_LOG_OBJECT (jbuf,
418 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
419 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
426 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
428 gboolean post = FALSE;
431 level = get_buffer_level (jbuf);
432 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
434 if (jbuf->buffering_disabled) {
435 GST_DEBUG ("buffering is disabled");
436 level = jbuf->high_level;
439 if (jbuf->buffering) {
441 if (level >= jbuf->high_level) {
442 GST_DEBUG ("buffering finished");
443 jbuf->buffering = FALSE;
446 if (level < jbuf->low_level) {
447 GST_DEBUG ("buffering started");
448 jbuf->buffering = TRUE;
455 if (jbuf->buffering && (jbuf->high_level != 0)) {
456 perc = (level * 100 / jbuf->high_level);
457 perc = MIN (perc, 100);
465 GST_DEBUG ("buffering %d", perc);
469 /* For the clock skew we use a windowed low point averaging algorithm as can be
470 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
471 * over Network Delays":
472 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
473 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
475 * The idea is that the jitter is composed of:
479 * N : a constant network delay.
480 * n : random added noise. The noise is concentrated around 0
482 * In the receiver we can track the elapsed time at the sender with:
484 * send_diff(i) = (Tsi - Ts0);
486 * Tsi : The time at the sender at packet i
487 * Ts0 : The time at the sender at the first packet
489 * This is the difference between the RTP timestamp in the first received packet
490 * and the current packet.
492 * At the receiver we have to deal with the jitter introduced by the network.
494 * recv_diff(i) = (Tri - Tr0)
496 * Tri : The time at the receiver at packet i
497 * Tr0 : The time at the receiver at the first packet
499 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
502 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
504 * Cri : The time of the clock at the receiver for packet i
505 * D + ni : The jitter when receiving packet i
507 * We see that the network delay is irrelevant here as we can elliminate D:
509 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
511 * The drift is now expressed as:
513 * Drift(i) = recv_diff(i) - send_diff(i);
515 * We now keep the W latest values of Drift and find the minimum (this is the
516 * one with the lowest network jitter and thus the one which is least affected
517 * by it). We average this lowest value to smooth out the resulting network skew.
519 * Both the window and the weighting used for averaging influence the accuracy
520 * of the drift estimation. Finding the correct parameters turns out to be a
521 * compromise between accuracy and inertia.
523 * We use a 2 second window or up to 512 data points, which is statistically big
524 * enough to catch spikes (FIXME, detect spikes).
525 * We also use a rather large weighting factor (125) to smoothly adapt. During
526 * startup, when filling the window, we use a parabolic weighting factor, the
527 * more the window is filled, the faster we move to the detected possible skew.
529 * Returns: @time adjusted with the clock skew.
532 calculate_skew (RTPJitterBuffer * jbuf, guint64 ext_rtptime,
533 GstClockTime gstrtptime, GstClockTime time)
535 guint64 send_diff, recv_diff;
539 GstClockTime out_time;
542 /* elapsed time at sender */
543 send_diff = gstrtptime - jbuf->base_rtptime;
545 /* we don't have an arrival timestamp so we can't do skew detection. we
546 * should still apply a timestamp based on RTP timestamp and base_time */
547 if (time == -1 || jbuf->base_time == -1)
550 /* elapsed time at receiver, includes the jitter */
551 recv_diff = time - jbuf->base_time;
553 /* measure the diff */
554 delta = ((gint64) recv_diff) - ((gint64) send_diff);
556 /* measure the slope, this gives a rought estimate between the sender speed
557 * and the receiver speed. This should be approximately 8, higher values
558 * indicate a burst (especially when the connection starts) */
560 slope = (send_diff * 8) / recv_diff;
564 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
565 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
566 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
568 /* if the difference between the sender timeline and the receiver timeline
569 * changed too quickly we have to resync because the server likely restarted
571 if (ABS (delta - jbuf->skew) > GST_SECOND) {
572 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
573 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
574 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
579 pos = jbuf->window_pos;
581 if (G_UNLIKELY (jbuf->window_filling)) {
582 /* we are filling the window */
583 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
584 jbuf->window[pos++] = delta;
585 /* calc the min delta we observed */
586 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
587 jbuf->window_min = delta;
589 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
590 jbuf->window_size = pos;
593 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
595 /* the skew is now the min */
596 jbuf->skew = jbuf->window_min;
597 jbuf->window_filling = FALSE;
599 gint perc_time, perc_window, perc;
601 /* figure out how much we filled the window, this depends on the amount of
602 * time we have or the max number of points we keep. */
603 perc_time = send_diff * 100 / MAX_TIME;
604 perc_window = pos * 100 / MAX_WINDOW;
605 perc = MAX (perc_time, perc_window);
607 /* make a parabolic function, the closer we get to the MAX, the more value
608 * we give to the scaling factor of the new value */
611 /* quickly go to the min value when we are filling up, slowly when we are
612 * just starting because we're not sure it's a good value yet. */
614 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
615 jbuf->window_size = pos + 1;
618 /* pick old value and store new value. We keep the previous value in order
619 * to quickly check if the min of the window changed */
620 old = jbuf->window[pos];
621 jbuf->window[pos++] = delta;
623 if (G_UNLIKELY (delta <= jbuf->window_min)) {
624 /* if the new value we inserted is smaller or equal to the current min,
625 * it becomes the new min */
626 jbuf->window_min = delta;
627 } else if (G_UNLIKELY (old == jbuf->window_min)) {
628 gint64 min = G_MAXINT64;
630 /* if we removed the old min, we have to find a new min */
631 for (i = 0; i < jbuf->window_size; i++) {
632 /* we found another value equal to the old min, we can stop searching now */
633 if (jbuf->window[i] == old) {
637 if (jbuf->window[i] < min)
638 min = jbuf->window[i];
640 jbuf->window_min = min;
642 /* average the min values */
643 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
644 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
645 delta, jbuf->window_min);
647 /* wrap around in the window */
648 if (G_UNLIKELY (pos >= jbuf->window_size))
650 jbuf->window_pos = pos;
653 /* the output time is defined as the base timestamp plus the RTP time
654 * adjusted for the clock skew .*/
655 if (jbuf->base_time != -1) {
656 out_time = jbuf->base_time + send_diff;
657 /* skew can be negative and we don't want to make invalid timestamps */
658 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
661 out_time += jbuf->skew;
666 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
667 jbuf->skew, GST_TIME_ARGS (out_time));
673 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
675 GQueue *queue = jbuf->packets;
677 /* It's more likely that the packet was inserted at the tail of the queue */
678 if (G_LIKELY (list)) {
680 item->next = list->next;
684 item->next = queue->head;
688 item->next->prev = item;
695 rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
696 gboolean estimated_dts, guint32 rtptime, GstClockTime base_time)
699 GstClockTime gstrtptime, pts;
700 GstClock *media_clock, *pipeline_clock;
701 guint64 media_clock_offset;
702 gboolean rfc7273_mode;
704 /* rtp time jumps are checked for during skew calculation, but bypassed
705 * in other mode, so mind those here and reset jb if needed.
706 * Only reset if valid input time, which is likely for UDP input
707 * where we expect this might happen due to async thread effects
708 * (in seek and state change cycles), but not so much for TCP input */
709 if (GST_CLOCK_TIME_IS_VALID (dts) && !estimated_dts &&
710 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
711 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
712 GstClockTime ext_rtptime = jbuf->ext_rtptime;
714 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
715 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
716 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
717 /* reset even if we don't have valid incoming time;
718 * still better than producing possibly very bogus output timestamp */
719 GST_WARNING ("rtp delta too big, reset skew");
720 rtp_jitter_buffer_reset_skew (jbuf);
724 /* Return the last time if we got the same RTP timestamp again */
725 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
726 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime) {
727 return jbuf->prev_out_time;
730 /* keep track of the last extended rtptime */
731 jbuf->last_rtptime = ext_rtptime;
733 g_mutex_lock (&jbuf->clock_lock);
734 media_clock = jbuf->media_clock ? gst_object_ref (jbuf->media_clock) : NULL;
736 jbuf->pipeline_clock ? gst_object_ref (jbuf->pipeline_clock) : NULL;
737 media_clock_offset = jbuf->media_clock_offset;
738 g_mutex_unlock (&jbuf->clock_lock);
741 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
743 if (G_LIKELY (jbuf->base_rtptime != -1)) {
744 /* check elapsed time in RTP units */
745 if (gstrtptime < jbuf->base_rtptime) {
746 /* elapsed time at sender, timestamps can go backwards and thus be
747 * smaller than our base time, schedule to take a new base time in
749 GST_WARNING ("backward timestamps at server, schedule resync");
750 jbuf->need_resync = TRUE;
754 switch (jbuf->mode) {
755 case RTP_JITTER_BUFFER_MODE_NONE:
756 case RTP_JITTER_BUFFER_MODE_BUFFER:
757 /* send 0 as the first timestamp and -1 for the other ones. This will
758 * interpolate them from the RTP timestamps with a 0 origin. In buffering
759 * mode we will adjust the outgoing timestamps according to the amount of
760 * time we spent buffering. */
761 if (jbuf->base_time == -1)
766 case RTP_JITTER_BUFFER_MODE_SYNCED:
767 /* synchronized clocks, take first timestamp as base, use RTP timestamps
769 if (jbuf->base_time != -1 && !jbuf->need_resync)
772 case RTP_JITTER_BUFFER_MODE_SLAVE:
777 /* need resync, lock on to time and gstrtptime if we can, otherwise we
778 * do with the previous values */
779 if (G_UNLIKELY (jbuf->need_resync && dts != -1)) {
780 GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
781 GST_TIME_FORMAT, GST_TIME_ARGS (dts), GST_TIME_ARGS (gstrtptime));
782 rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, FALSE);
785 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
786 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
787 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
788 GST_TIME_ARGS (gstrtptime - jbuf->base_rtptime));
790 rfc7273_mode = media_clock && pipeline_clock
791 && gst_clock_is_synced (media_clock);
793 if (rfc7273_mode && jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
794 && (media_clock_offset == -1 || !jbuf->rfc7273_sync)) {
795 GstClockTime internal, external;
796 GstClockTime rate_num, rate_denom;
797 GstClockTime nsrtptimediff, rtpntptime, rtpsystime;
799 gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
802 /* Slave to the RFC7273 media clock instead of trying to estimate it
803 * based on receive times and RTP timestamps */
805 if (jbuf->media_clock_base_time == -1) {
806 if (jbuf->base_time != -1) {
807 jbuf->media_clock_base_time =
808 gst_clock_unadjust_with_calibration (media_clock,
809 jbuf->base_time + base_time, internal, external, rate_num,
813 jbuf->media_clock_base_time =
814 gst_clock_unadjust_with_calibration (media_clock, dts + base_time,
815 internal, external, rate_num, rate_denom);
817 jbuf->media_clock_base_time =
818 gst_clock_get_internal_time (media_clock);
819 jbuf->base_rtptime = gstrtptime;
823 if (gstrtptime > jbuf->base_rtptime)
824 nsrtptimediff = gstrtptime - jbuf->base_rtptime;
828 rtpntptime = nsrtptimediff + jbuf->media_clock_base_time;
831 gst_clock_adjust_with_calibration (media_clock, rtpntptime, internal,
832 external, rate_num, rate_denom);
834 if (rtpsystime > base_time)
835 pts = rtpsystime - base_time;
839 GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
840 GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
841 } else if (rfc7273_mode && (jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
842 || jbuf->mode == RTP_JITTER_BUFFER_MODE_SYNCED)
843 && media_clock_offset != -1 && jbuf->rfc7273_sync) {
844 GstClockTime ntptime, rtptime_tmp;
845 GstClockTime ntprtptime, rtpsystime;
846 GstClockTime internal, external;
847 GstClockTime rate_num, rate_denom;
849 /* Don't do any of the dts related adjustments further down */
852 /* Calculate the actual clock time on the sender side based on the
853 * RFC7273 clock and convert it to our pipeline clock
856 gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
859 ntptime = gst_clock_get_internal_time (media_clock);
861 ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
862 ntprtptime += media_clock_offset;
863 ntprtptime &= 0xffffffff;
865 rtptime_tmp = rtptime;
866 /* Check for wraparounds, we assume that the diff between current RTP
867 * timestamp and current media clock time can't be bigger than
868 * 2**31 clock units */
869 if (ntprtptime > rtptime_tmp && ntprtptime - rtptime_tmp >= 0x80000000)
870 rtptime_tmp += G_GUINT64_CONSTANT (0x100000000);
871 else if (rtptime_tmp > ntprtptime && rtptime_tmp - ntprtptime >= 0x80000000)
872 ntprtptime += G_GUINT64_CONSTANT (0x100000000);
874 if (ntprtptime > rtptime_tmp)
876 gst_util_uint64_scale (ntprtptime - rtptime_tmp, jbuf->clock_rate,
880 gst_util_uint64_scale (rtptime_tmp - ntprtptime, jbuf->clock_rate,
884 gst_clock_adjust_with_calibration (media_clock, ntptime, internal,
885 external, rate_num, rate_denom);
886 /* All this assumes that the pipeline has enough additional
887 * latency to cover for the network delay */
888 if (rtpsystime > base_time)
889 pts = rtpsystime - base_time;
893 GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
894 GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
896 /* If we used the RFC7273 clock before and not anymore,
897 * we need to resync it later again */
898 jbuf->media_clock_base_time = -1;
900 /* do skew calculation by measuring the difference between rtptime and the
901 * receive dts, this function will return the skew corrected rtptime. */
902 pts = calculate_skew (jbuf, ext_rtptime, gstrtptime, dts);
905 /* check if timestamps are not going backwards, we can only check this if we
906 * have a previous out time and a previous send_diff */
907 if (G_LIKELY (pts != -1 && jbuf->prev_out_time != -1
908 && jbuf->prev_send_diff != -1)) {
909 /* now check for backwards timestamps */
911 /* if the server timestamps went up and the out_time backwards */
912 (gstrtptime - jbuf->base_rtptime > jbuf->prev_send_diff
913 && pts < jbuf->prev_out_time) ||
914 /* if the server timestamps went backwards and the out_time forwards */
915 (gstrtptime - jbuf->base_rtptime < jbuf->prev_send_diff
916 && pts > jbuf->prev_out_time) ||
917 /* if the server timestamps did not change */
918 gstrtptime - jbuf->base_rtptime == jbuf->prev_send_diff)) {
919 GST_DEBUG ("backwards timestamps, using previous time");
920 pts = jbuf->prev_out_time;
924 if (dts != -1 && pts + jbuf->delay < dts) {
925 /* if we are going to produce a timestamp that is later than the input
926 * timestamp, we need to reset the jitterbuffer. Likely the server paused
928 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
929 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (pts),
930 jbuf->delay, GST_TIME_ARGS (dts));
931 rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, TRUE);
935 jbuf->prev_out_time = pts;
936 jbuf->prev_send_diff = gstrtptime - jbuf->base_rtptime;
939 gst_object_unref (media_clock);
941 gst_object_unref (pipeline_clock);
948 * rtp_jitter_buffer_insert:
949 * @jbuf: an #RTPJitterBuffer
950 * @item: an #RTPJitterBufferItem to insert
951 * @head: TRUE when the head element changed.
952 * @percent: the buffering percent after insertion
954 * Inserts @item into the packet queue of @jbuf. The sequence number of the
955 * packet will be used to sort the packets. This function takes ownerhip of
956 * @buf when the function returns %TRUE.
958 * When @head is %TRUE, the new packet was added at the head of the queue and
959 * will be available with the next call to rtp_jitter_buffer_pop() and
960 * rtp_jitter_buffer_peek().
962 * Returns: %FALSE if a packet with the same number already existed.
965 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
966 gboolean * head, gint * percent)
968 GList *list, *event = NULL;
971 g_return_val_if_fail (jbuf != NULL, FALSE);
972 g_return_val_if_fail (item != NULL, FALSE);
974 list = jbuf->packets->tail;
976 /* no seqnum, simply append then */
977 if (item->seqnum == -1)
980 seqnum = item->seqnum;
982 /* loop the list to skip strictly larger seqnum buffers */
983 for (; list; list = g_list_previous (list)) {
986 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
988 if (qitem->seqnum == -1) {
989 /* keep a pointer to the first consecutive event if not already
990 * set. we will insert the packet after the event if we can't find
991 * a packet with lower sequence number before the event. */
997 qseq = qitem->seqnum;
999 /* compare the new seqnum to the one in the buffer */
1000 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
1002 /* we hit a packet with the same seqnum, notify a duplicate */
1003 if (G_UNLIKELY (gap == 0))
1006 /* seqnum > qseq, we can stop looking */
1007 if (G_LIKELY (gap < 0))
1010 /* if we've found a packet with greater sequence number, cleanup the
1011 * event pointer as the packet will be inserted before the event */
1015 /* if event is set it means that packets before the event had smaller
1016 * sequence number, so we will insert our packet after the event */
1021 queue_do_insert (jbuf, list, (GList *) item);
1023 /* buffering mode, update buffer stats */
1024 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
1025 update_buffer_level (jbuf, percent);
1029 /* head was changed when we did not find a previous packet, we set the return
1030 * flag when requested. */
1031 if (G_LIKELY (head))
1032 *head = (list == NULL);
1039 GST_DEBUG ("duplicate packet %d found", (gint) seqnum);
1040 if (G_LIKELY (head))
1047 * rtp_jitter_buffer_pop:
1048 * @jbuf: an #RTPJitterBuffer
1049 * @percent: the buffering percent
1051 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
1052 * have its timestamp adjusted with the incoming running_time and the detected
1055 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
1057 RTPJitterBufferItem *
1058 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
1063 g_return_val_if_fail (jbuf != NULL, NULL);
1065 queue = jbuf->packets;
1069 queue->head = item->next;
1071 queue->head->prev = NULL;
1077 /* buffering mode, update buffer stats */
1078 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
1079 update_buffer_level (jbuf, percent);
1083 return (RTPJitterBufferItem *) item;
1087 * rtp_jitter_buffer_peek:
1088 * @jbuf: an #RTPJitterBuffer
1090 * Peek the oldest buffer from the packet queue of @jbuf.
1092 * See rtp_jitter_buffer_insert() to check when an older packet was
1095 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
1097 RTPJitterBufferItem *
1098 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
1100 g_return_val_if_fail (jbuf != NULL, NULL);
1102 return (RTPJitterBufferItem *) jbuf->packets->head;
1106 * rtp_jitter_buffer_flush:
1107 * @jbuf: an #RTPJitterBuffer
1108 * @free_func: function to free each item
1109 * @user_data: user data passed to @free_func
1111 * Flush all packets from the jitterbuffer.
1114 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
1119 g_return_if_fail (jbuf != NULL);
1120 g_return_if_fail (free_func != NULL);
1122 while ((item = g_queue_pop_head_link (jbuf->packets)))
1123 free_func ((RTPJitterBufferItem *) item, user_data);
1127 * rtp_jitter_buffer_is_buffering:
1128 * @jbuf: an #RTPJitterBuffer
1130 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
1131 * pop packets while in buffering mode.
1133 * Returns: the buffering state of @jbuf
1136 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
1138 return jbuf->buffering && !jbuf->buffering_disabled;
1142 * rtp_jitter_buffer_set_buffering:
1143 * @jbuf: an #RTPJitterBuffer
1144 * @buffering: the new buffering state
1146 * Forces @jbuf to go into the buffering state.
1149 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
1151 jbuf->buffering = buffering;
1155 * rtp_jitter_buffer_get_percent:
1156 * @jbuf: an #RTPJitterBuffer
1158 * Get the buffering percent of the jitterbuffer.
1160 * Returns: the buffering percent
1163 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
1168 if (G_UNLIKELY (jbuf->high_level == 0))
1171 if (G_UNLIKELY (jbuf->buffering_disabled))
1174 level = get_buffer_level (jbuf);
1175 percent = (level * 100 / jbuf->high_level);
1176 percent = MIN (percent, 100);
1182 * rtp_jitter_buffer_num_packets:
1183 * @jbuf: an #RTPJitterBuffer
1185 * Get the number of packets currently in "jbuf.
1187 * Returns: The number of packets in @jbuf.
1190 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
1192 g_return_val_if_fail (jbuf != NULL, 0);
1194 return jbuf->packets->length;
1198 * rtp_jitter_buffer_get_ts_diff:
1199 * @jbuf: an #RTPJitterBuffer
1201 * Get the difference between the timestamps of first and last packet in the
1204 * Returns: The difference expressed in the timestamp units of the packets.
1207 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
1209 guint64 high_ts, low_ts;
1210 RTPJitterBufferItem *high_buf, *low_buf;
1213 g_return_val_if_fail (jbuf != NULL, 0);
1215 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
1216 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
1218 if (!high_buf || !low_buf || high_buf == low_buf)
1221 high_ts = high_buf->rtptime;
1222 low_ts = low_buf->rtptime;
1224 /* it needs to work if ts wraps */
1225 if (high_ts >= low_ts) {
1226 result = (guint32) (high_ts - low_ts);
1228 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
1235 * rtp_jitter_buffer_get_seqnum_diff:
1236 * @jbuf: an #RTPJitterBuffer
1238 * Get the difference between the seqnum of first and last packet in the
1241 * Returns: The difference expressed in seqnum.
1244 rtp_jitter_buffer_get_seqnum_diff (RTPJitterBuffer * jbuf)
1246 guint32 high_seqnum, low_seqnum;
1247 RTPJitterBufferItem *high_buf, *low_buf;
1250 g_return_val_if_fail (jbuf != NULL, 0);
1252 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
1253 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
1255 while (high_buf && high_buf->seqnum == -1)
1256 high_buf = (RTPJitterBufferItem *) high_buf->prev;
1258 while (low_buf && low_buf->seqnum == -1)
1259 low_buf = (RTPJitterBufferItem *) low_buf->next;
1261 if (!high_buf || !low_buf || high_buf == low_buf)
1264 high_seqnum = high_buf->seqnum;
1265 low_seqnum = low_buf->seqnum;
1267 /* it needs to work if ts wraps */
1268 if (high_seqnum >= low_seqnum) {
1269 result = (guint32) (high_seqnum - low_seqnum);
1271 result = (guint32) (high_seqnum + G_MAXUINT16 + 1 - low_seqnum);
1277 * rtp_jitter_buffer_get_sync:
1278 * @jbuf: an #RTPJitterBuffer
1279 * @rtptime: result RTP time
1280 * @timestamp: result GStreamer timestamp
1281 * @clock_rate: clock-rate of @rtptime
1282 * @last_rtptime: last seen rtptime.
1284 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
1285 * used for constructing timestamps.
1287 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
1288 * the GStreamer timestamp is currently @timestamp.
1290 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
1294 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
1295 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
1298 *rtptime = jbuf->base_extrtp;
1300 *timestamp = jbuf->base_time + jbuf->skew;
1302 *clock_rate = jbuf->clock_rate;
1304 *last_rtptime = jbuf->last_rtptime;
1308 * rtp_jitter_buffer_can_fast_start:
1309 * @jbuf: an #RTPJitterBuffer
1310 * @num_packets: Number of consecutive packets needed
1312 * Check if in the queue if there is enough packets with consecutive seqnum in
1313 * order to start delivering them.
1315 * Returns: %TRUE if the required number of consecutive packets was found.
1318 rtp_jitter_buffer_can_fast_start (RTPJitterBuffer * jbuf, gint num_packet)
1320 gboolean ret = TRUE;
1321 RTPJitterBufferItem *last_item = NULL, *item;
1324 if (rtp_jitter_buffer_num_packets (jbuf) < num_packet)
1327 item = rtp_jitter_buffer_peek (jbuf);
1328 for (i = 0; i < num_packet; i++) {
1329 if (G_LIKELY (last_item)) {
1330 guint16 expected_seqnum = last_item->seqnum + 1;
1332 if (expected_seqnum != item->seqnum) {
1339 item = (RTPJitterBufferItem *) last_item->next;