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 gst_object_unref (jbuf->media_clock);
109 if (jbuf->pipeline_clock)
110 gst_object_unref (jbuf->pipeline_clock);
112 g_queue_free (jbuf->packets);
114 g_mutex_clear (&jbuf->clock_lock);
116 G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
120 * rtp_jitter_buffer_new:
122 * Create an #RTPJitterBuffer.
124 * Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
127 rtp_jitter_buffer_new (void)
129 RTPJitterBuffer *jbuf;
131 jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
137 * rtp_jitter_buffer_get_mode:
138 * @jbuf: an #RTPJitterBuffer
140 * Get the current jitterbuffer mode.
142 * Returns: the current jitterbuffer mode.
145 rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
151 * rtp_jitter_buffer_set_mode:
152 * @jbuf: an #RTPJitterBuffer
153 * @mode: a #RTPJitterBufferMode
155 * Set the buffering and clock slaving algorithm used in the @jbuf.
158 rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
164 rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
170 rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
173 jbuf->low_level = (delay * 15) / 100;
174 /* the high level is at 90% in order to release packets before we fill up the
175 * buffer up to the latency */
176 jbuf->high_level = (delay * 90) / 100;
178 GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
179 GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
180 GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
184 * rtp_jitter_buffer_set_clock_rate:
185 * @jbuf: an #RTPJitterBuffer
186 * @clock_rate: the new clock rate
188 * Set the clock rate in the jitterbuffer.
191 rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
193 if (jbuf->clock_rate != clock_rate) {
194 GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
195 G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
196 jbuf->clock_rate = clock_rate;
197 rtp_jitter_buffer_reset_skew (jbuf);
202 * rtp_jitter_buffer_get_clock_rate:
203 * @jbuf: an #RTPJitterBuffer
205 * Get the currently configure clock rate in @jbuf.
207 * Returns: the current clock-rate
210 rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
212 return jbuf->clock_rate;
216 media_clock_synced_cb (GstClock * clock, gboolean synced,
217 RTPJitterBuffer * jbuf)
219 GstClockTime internal, external;
221 g_mutex_lock (&jbuf->clock_lock);
222 if (jbuf->pipeline_clock) {
223 internal = gst_clock_get_internal_time (jbuf->media_clock);
224 external = gst_clock_get_time (jbuf->pipeline_clock);
226 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
228 g_mutex_unlock (&jbuf->clock_lock);
232 * rtp_jitter_buffer_set_media_clock:
233 * @jbuf: an #RTPJitterBuffer
234 * @clock: (transfer full): media #GstClock
235 * @clock_offset: RTP time at clock epoch or -1
237 * Sets the media clock for the media and the clock offset
241 rtp_jitter_buffer_set_media_clock (RTPJitterBuffer * jbuf, GstClock * clock,
242 guint64 clock_offset)
244 g_mutex_lock (&jbuf->clock_lock);
245 if (jbuf->media_clock) {
246 if (jbuf->media_clock_synced_id)
247 g_signal_handler_disconnect (jbuf->media_clock,
248 jbuf->media_clock_synced_id);
249 jbuf->media_clock_synced_id = 0;
250 gst_object_unref (jbuf->media_clock);
252 jbuf->media_clock = clock;
253 jbuf->media_clock_offset = clock_offset;
255 if (jbuf->pipeline_clock && jbuf->media_clock &&
256 jbuf->pipeline_clock != jbuf->media_clock) {
257 jbuf->media_clock_synced_id =
258 g_signal_connect (jbuf->media_clock, "synced",
259 G_CALLBACK (media_clock_synced_cb), jbuf);
260 if (gst_clock_is_synced (jbuf->media_clock)) {
261 GstClockTime internal, external;
263 internal = gst_clock_get_internal_time (jbuf->media_clock);
264 external = gst_clock_get_time (jbuf->pipeline_clock);
266 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
269 gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
271 g_mutex_unlock (&jbuf->clock_lock);
275 * rtp_jitter_buffer_set_pipeline_clock:
276 * @jbuf: an #RTPJitterBuffer
277 * @clock: pipeline #GstClock
279 * Sets the pipeline clock
283 rtp_jitter_buffer_set_pipeline_clock (RTPJitterBuffer * jbuf, GstClock * clock)
285 g_mutex_lock (&jbuf->clock_lock);
286 if (jbuf->pipeline_clock)
287 gst_object_unref (jbuf->pipeline_clock);
288 jbuf->pipeline_clock = clock ? gst_object_ref (clock) : NULL;
290 if (jbuf->pipeline_clock && jbuf->media_clock &&
291 jbuf->pipeline_clock != jbuf->media_clock) {
292 if (gst_clock_is_synced (jbuf->media_clock)) {
293 GstClockTime internal, external;
295 internal = gst_clock_get_internal_time (jbuf->media_clock);
296 external = gst_clock_get_time (jbuf->pipeline_clock);
298 gst_clock_set_calibration (jbuf->media_clock, internal, external, 1, 1);
301 gst_clock_set_master (jbuf->media_clock, jbuf->pipeline_clock);
303 g_mutex_unlock (&jbuf->clock_lock);
307 rtp_jitter_buffer_get_rfc7273_sync (RTPJitterBuffer * jbuf)
309 return jbuf->rfc7273_sync;
313 rtp_jitter_buffer_set_rfc7273_sync (RTPJitterBuffer * jbuf,
314 gboolean rfc7273_sync)
316 jbuf->rfc7273_sync = rfc7273_sync;
320 * rtp_jitter_buffer_reset_skew:
321 * @jbuf: an #RTPJitterBuffer
323 * Reset the skew calculations in @jbuf.
326 rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
328 jbuf->base_time = -1;
329 jbuf->base_rtptime = -1;
330 jbuf->base_extrtp = -1;
331 jbuf->media_clock_base_time = -1;
332 jbuf->ext_rtptime = -1;
333 jbuf->last_rtptime = -1;
334 jbuf->window_pos = 0;
335 jbuf->window_filling = TRUE;
336 jbuf->window_min = 0;
338 jbuf->prev_send_diff = -1;
339 jbuf->prev_out_time = -1;
340 jbuf->need_resync = TRUE;
342 GST_DEBUG ("reset skew correction");
346 * rtp_jitter_buffer_disable_buffering:
347 * @jbuf: an #RTPJitterBuffer
348 * @disabled: the new state
350 * Enable or disable buffering on @jbuf.
353 rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
355 jbuf->buffering_disabled = disabled;
359 rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
360 GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
362 jbuf->base_time = time;
363 jbuf->media_clock_base_time = -1;
364 jbuf->base_rtptime = gstrtptime;
365 jbuf->base_extrtp = ext_rtptime;
366 jbuf->prev_out_time = -1;
367 jbuf->prev_send_diff = -1;
369 jbuf->window_filling = TRUE;
370 jbuf->window_pos = 0;
371 jbuf->window_min = 0;
372 jbuf->window_size = 0;
375 jbuf->need_resync = FALSE;
379 get_buffer_level (RTPJitterBuffer * jbuf)
381 RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
384 /* first buffer with timestamp */
385 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
387 if (high_buf->dts != -1 || high_buf->pts != -1)
390 high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
393 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
395 if (low_buf->dts != -1 || low_buf->pts != -1)
398 low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
401 if (!high_buf || !low_buf || high_buf == low_buf) {
404 guint64 high_ts, low_ts;
406 high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
407 low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
409 if (high_ts > low_ts)
410 level = high_ts - low_ts;
414 GST_LOG_OBJECT (jbuf,
415 "low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
416 G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
423 update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
425 gboolean post = FALSE;
428 level = get_buffer_level (jbuf);
429 GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
431 if (jbuf->buffering_disabled) {
432 GST_DEBUG ("buffering is disabled");
433 level = jbuf->high_level;
436 if (jbuf->buffering) {
438 if (level >= jbuf->high_level) {
439 GST_DEBUG ("buffering finished");
440 jbuf->buffering = FALSE;
443 if (level < jbuf->low_level) {
444 GST_DEBUG ("buffering started");
445 jbuf->buffering = TRUE;
452 if (jbuf->buffering && (jbuf->high_level != 0)) {
453 perc = (level * 100 / jbuf->high_level);
454 perc = MIN (perc, 100);
462 GST_DEBUG ("buffering %d", perc);
466 /* For the clock skew we use a windowed low point averaging algorithm as can be
467 * found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
468 * over Network Delays":
469 * http://www.grame.fr/Ressources/pub/TR-050601.pdf
470 * http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
472 * The idea is that the jitter is composed of:
476 * N : a constant network delay.
477 * n : random added noise. The noise is concentrated around 0
479 * In the receiver we can track the elapsed time at the sender with:
481 * send_diff(i) = (Tsi - Ts0);
483 * Tsi : The time at the sender at packet i
484 * Ts0 : The time at the sender at the first packet
486 * This is the difference between the RTP timestamp in the first received packet
487 * and the current packet.
489 * At the receiver we have to deal with the jitter introduced by the network.
491 * recv_diff(i) = (Tri - Tr0)
493 * Tri : The time at the receiver at packet i
494 * Tr0 : The time at the receiver at the first packet
496 * Both of these values contain a jitter Ji, a jitter for packet i, so we can
499 * recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
501 * Cri : The time of the clock at the receiver for packet i
502 * D + ni : The jitter when receiving packet i
504 * We see that the network delay is irrelevant here as we can elliminate D:
506 * recv_diff(i) = (Cri + ni) - (Cr0 + n0))
508 * The drift is now expressed as:
510 * Drift(i) = recv_diff(i) - send_diff(i);
512 * We now keep the W latest values of Drift and find the minimum (this is the
513 * one with the lowest network jitter and thus the one which is least affected
514 * by it). We average this lowest value to smooth out the resulting network skew.
516 * Both the window and the weighting used for averaging influence the accuracy
517 * of the drift estimation. Finding the correct parameters turns out to be a
518 * compromise between accuracy and inertia.
520 * We use a 2 second window or up to 512 data points, which is statistically big
521 * enough to catch spikes (FIXME, detect spikes).
522 * We also use a rather large weighting factor (125) to smoothly adapt. During
523 * startup, when filling the window, we use a parabolic weighting factor, the
524 * more the window is filled, the faster we move to the detected possible skew.
526 * Returns: @time adjusted with the clock skew.
529 calculate_skew (RTPJitterBuffer * jbuf, guint64 ext_rtptime,
530 GstClockTime gstrtptime, GstClockTime time)
532 guint64 send_diff, recv_diff;
536 GstClockTime out_time;
539 /* elapsed time at sender */
540 send_diff = gstrtptime - jbuf->base_rtptime;
542 /* we don't have an arrival timestamp so we can't do skew detection. we
543 * should still apply a timestamp based on RTP timestamp and base_time */
544 if (time == -1 || jbuf->base_time == -1)
547 /* elapsed time at receiver, includes the jitter */
548 recv_diff = time - jbuf->base_time;
550 /* measure the diff */
551 delta = ((gint64) recv_diff) - ((gint64) send_diff);
553 /* measure the slope, this gives a rought estimate between the sender speed
554 * and the receiver speed. This should be approximately 8, higher values
555 * indicate a burst (especially when the connection starts) */
557 slope = (send_diff * 8) / recv_diff;
561 GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
562 GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
563 GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
565 /* if the difference between the sender timeline and the receiver timeline
566 * changed too quickly we have to resync because the server likely restarted
568 if (ABS (delta - jbuf->skew) > GST_SECOND) {
569 GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
570 GST_TIME_ARGS (ABS (delta - jbuf->skew)));
571 rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
576 pos = jbuf->window_pos;
578 if (G_UNLIKELY (jbuf->window_filling)) {
579 /* we are filling the window */
580 GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
581 jbuf->window[pos++] = delta;
582 /* calc the min delta we observed */
583 if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
584 jbuf->window_min = delta;
586 if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
587 jbuf->window_size = pos;
590 GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
592 /* the skew is now the min */
593 jbuf->skew = jbuf->window_min;
594 jbuf->window_filling = FALSE;
596 gint perc_time, perc_window, perc;
598 /* figure out how much we filled the window, this depends on the amount of
599 * time we have or the max number of points we keep. */
600 perc_time = send_diff * 100 / MAX_TIME;
601 perc_window = pos * 100 / MAX_WINDOW;
602 perc = MAX (perc_time, perc_window);
604 /* make a parabolic function, the closer we get to the MAX, the more value
605 * we give to the scaling factor of the new value */
608 /* quickly go to the min value when we are filling up, slowly when we are
609 * just starting because we're not sure it's a good value yet. */
611 (perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
612 jbuf->window_size = pos + 1;
615 /* pick old value and store new value. We keep the previous value in order
616 * to quickly check if the min of the window changed */
617 old = jbuf->window[pos];
618 jbuf->window[pos++] = delta;
620 if (G_UNLIKELY (delta <= jbuf->window_min)) {
621 /* if the new value we inserted is smaller or equal to the current min,
622 * it becomes the new min */
623 jbuf->window_min = delta;
624 } else if (G_UNLIKELY (old == jbuf->window_min)) {
625 gint64 min = G_MAXINT64;
627 /* if we removed the old min, we have to find a new min */
628 for (i = 0; i < jbuf->window_size; i++) {
629 /* we found another value equal to the old min, we can stop searching now */
630 if (jbuf->window[i] == old) {
634 if (jbuf->window[i] < min)
635 min = jbuf->window[i];
637 jbuf->window_min = min;
639 /* average the min values */
640 jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
641 GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
642 delta, jbuf->window_min);
644 /* wrap around in the window */
645 if (G_UNLIKELY (pos >= jbuf->window_size))
647 jbuf->window_pos = pos;
650 /* the output time is defined as the base timestamp plus the RTP time
651 * adjusted for the clock skew .*/
652 if (jbuf->base_time != -1) {
653 out_time = jbuf->base_time + send_diff;
654 /* skew can be negative and we don't want to make invalid timestamps */
655 if (jbuf->skew < 0 && out_time < -jbuf->skew) {
658 out_time += jbuf->skew;
663 GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
664 jbuf->skew, GST_TIME_ARGS (out_time));
670 queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
672 GQueue *queue = jbuf->packets;
674 /* It's more likely that the packet was inserted at the tail of the queue */
675 if (G_LIKELY (list)) {
677 item->next = list->next;
681 item->next = queue->head;
685 item->next->prev = item;
692 rtp_jitter_buffer_calculate_pts (RTPJitterBuffer * jbuf, GstClockTime dts,
693 gboolean estimated_dts, guint32 rtptime, GstClockTime base_time)
696 GstClockTime gstrtptime, pts;
697 GstClock *media_clock, *pipeline_clock;
698 guint64 media_clock_offset;
699 gboolean rfc7273_mode;
701 /* rtp time jumps are checked for during skew calculation, but bypassed
702 * in other mode, so mind those here and reset jb if needed.
703 * Only reset if valid input time, which is likely for UDP input
704 * where we expect this might happen due to async thread effects
705 * (in seek and state change cycles), but not so much for TCP input */
706 if (GST_CLOCK_TIME_IS_VALID (dts) && !estimated_dts &&
707 jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
708 jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
709 GstClockTime ext_rtptime = jbuf->ext_rtptime;
711 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
712 if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
713 ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
714 /* reset even if we don't have valid incoming time;
715 * still better than producing possibly very bogus output timestamp */
716 GST_WARNING ("rtp delta too big, reset skew");
717 rtp_jitter_buffer_reset_skew (jbuf);
721 /* Return the last time if we got the same RTP timestamp again */
722 ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
723 if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime) {
724 return jbuf->prev_out_time;
727 /* keep track of the last extended rtptime */
728 jbuf->last_rtptime = ext_rtptime;
730 g_mutex_lock (&jbuf->clock_lock);
731 media_clock = jbuf->media_clock ? gst_object_ref (jbuf->media_clock) : NULL;
733 jbuf->pipeline_clock ? gst_object_ref (jbuf->pipeline_clock) : NULL;
734 media_clock_offset = jbuf->media_clock_offset;
735 g_mutex_unlock (&jbuf->clock_lock);
738 gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
740 if (G_LIKELY (jbuf->base_rtptime != -1)) {
741 /* check elapsed time in RTP units */
742 if (gstrtptime < jbuf->base_rtptime) {
743 /* elapsed time at sender, timestamps can go backwards and thus be
744 * smaller than our base time, schedule to take a new base time in
746 GST_WARNING ("backward timestamps at server, schedule resync");
747 jbuf->need_resync = TRUE;
751 switch (jbuf->mode) {
752 case RTP_JITTER_BUFFER_MODE_NONE:
753 case RTP_JITTER_BUFFER_MODE_BUFFER:
754 /* send 0 as the first timestamp and -1 for the other ones. This will
755 * interpolate them from the RTP timestamps with a 0 origin. In buffering
756 * mode we will adjust the outgoing timestamps according to the amount of
757 * time we spent buffering. */
758 if (jbuf->base_time == -1)
763 case RTP_JITTER_BUFFER_MODE_SYNCED:
764 /* synchronized clocks, take first timestamp as base, use RTP timestamps
766 if (jbuf->base_time != -1 && !jbuf->need_resync)
769 case RTP_JITTER_BUFFER_MODE_SLAVE:
774 /* need resync, lock on to time and gstrtptime if we can, otherwise we
775 * do with the previous values */
776 if (G_UNLIKELY (jbuf->need_resync && dts != -1)) {
777 GST_INFO ("resync to time %" GST_TIME_FORMAT ", rtptime %"
778 GST_TIME_FORMAT, GST_TIME_ARGS (dts), GST_TIME_ARGS (gstrtptime));
779 rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, FALSE);
782 GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
783 GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
784 GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
785 GST_TIME_ARGS (gstrtptime - jbuf->base_rtptime));
787 rfc7273_mode = media_clock && pipeline_clock
788 && gst_clock_is_synced (media_clock);
790 if (rfc7273_mode && jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
791 && (media_clock_offset == -1 || !jbuf->rfc7273_sync)) {
792 GstClockTime internal, external;
793 GstClockTime rate_num, rate_denom;
794 GstClockTime nsrtptimediff, rtpntptime, rtpsystime;
796 gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
799 /* Slave to the RFC7273 media clock instead of trying to estimate it
800 * based on receive times and RTP timestamps */
802 if (jbuf->media_clock_base_time == -1) {
803 if (jbuf->base_time != -1) {
804 jbuf->media_clock_base_time =
805 gst_clock_unadjust_with_calibration (media_clock,
806 jbuf->base_time + base_time, internal, external, rate_num,
810 jbuf->media_clock_base_time =
811 gst_clock_unadjust_with_calibration (media_clock, dts + base_time,
812 internal, external, rate_num, rate_denom);
814 jbuf->media_clock_base_time =
815 gst_clock_get_internal_time (media_clock);
816 jbuf->base_rtptime = gstrtptime;
820 if (gstrtptime > jbuf->base_rtptime)
821 nsrtptimediff = gstrtptime - jbuf->base_rtptime;
825 rtpntptime = nsrtptimediff + jbuf->media_clock_base_time;
828 gst_clock_adjust_with_calibration (media_clock, rtpntptime, internal,
829 external, rate_num, rate_denom);
831 if (rtpsystime > base_time)
832 pts = rtpsystime - base_time;
836 GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
837 GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
838 } else if (rfc7273_mode && (jbuf->mode == RTP_JITTER_BUFFER_MODE_SLAVE
839 || jbuf->mode == RTP_JITTER_BUFFER_MODE_SYNCED)
840 && media_clock_offset != -1 && jbuf->rfc7273_sync) {
841 GstClockTime ntptime, rtptime_tmp;
842 GstClockTime ntprtptime, rtpsystime;
843 GstClockTime internal, external;
844 GstClockTime rate_num, rate_denom;
846 /* Don't do any of the dts related adjustments further down */
849 /* Calculate the actual clock time on the sender side based on the
850 * RFC7273 clock and convert it to our pipeline clock
853 gst_clock_get_calibration (media_clock, &internal, &external, &rate_num,
856 ntptime = gst_clock_get_internal_time (media_clock);
858 ntprtptime = gst_util_uint64_scale (ntptime, jbuf->clock_rate, GST_SECOND);
859 ntprtptime += media_clock_offset;
860 ntprtptime &= 0xffffffff;
862 rtptime_tmp = rtptime;
863 /* Check for wraparounds, we assume that the diff between current RTP
864 * timestamp and current media clock time can't be bigger than
865 * 2**31 clock units */
866 if (ntprtptime > rtptime_tmp && ntprtptime - rtptime_tmp >= 0x80000000)
867 rtptime_tmp += G_GUINT64_CONSTANT (0x100000000);
868 else if (rtptime_tmp > ntprtptime && rtptime_tmp - ntprtptime >= 0x80000000)
869 ntprtptime += G_GUINT64_CONSTANT (0x100000000);
871 if (ntprtptime > rtptime_tmp)
873 gst_util_uint64_scale (ntprtptime - rtptime_tmp, jbuf->clock_rate,
877 gst_util_uint64_scale (rtptime_tmp - ntprtptime, jbuf->clock_rate,
881 gst_clock_adjust_with_calibration (media_clock, ntptime, internal,
882 external, rate_num, rate_denom);
883 /* All this assumes that the pipeline has enough additional
884 * latency to cover for the network delay */
885 if (rtpsystime > base_time)
886 pts = rtpsystime - base_time;
890 GST_DEBUG ("RFC7273 clock time %" GST_TIME_FORMAT ", out %" GST_TIME_FORMAT,
891 GST_TIME_ARGS (rtpsystime), GST_TIME_ARGS (pts));
893 /* If we used the RFC7273 clock before and not anymore,
894 * we need to resync it later again */
895 jbuf->media_clock_base_time = -1;
897 /* do skew calculation by measuring the difference between rtptime and the
898 * receive dts, this function will return the skew corrected rtptime. */
899 pts = calculate_skew (jbuf, ext_rtptime, gstrtptime, dts);
902 /* check if timestamps are not going backwards, we can only check this if we
903 * have a previous out time and a previous send_diff */
904 if (G_LIKELY (pts != -1 && jbuf->prev_out_time != -1
905 && jbuf->prev_send_diff != -1)) {
906 /* now check for backwards timestamps */
908 /* if the server timestamps went up and the out_time backwards */
909 (gstrtptime - jbuf->base_rtptime > jbuf->prev_send_diff
910 && pts < jbuf->prev_out_time) ||
911 /* if the server timestamps went backwards and the out_time forwards */
912 (gstrtptime - jbuf->base_rtptime < jbuf->prev_send_diff
913 && pts > jbuf->prev_out_time) ||
914 /* if the server timestamps did not change */
915 gstrtptime - jbuf->base_rtptime == jbuf->prev_send_diff)) {
916 GST_DEBUG ("backwards timestamps, using previous time");
917 pts = jbuf->prev_out_time;
921 if (dts != -1 && pts + jbuf->delay < dts) {
922 /* if we are going to produce a timestamp that is later than the input
923 * timestamp, we need to reset the jitterbuffer. Likely the server paused
925 GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
926 GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (pts),
927 jbuf->delay, GST_TIME_ARGS (dts));
928 rtp_jitter_buffer_resync (jbuf, dts, gstrtptime, ext_rtptime, TRUE);
932 jbuf->prev_out_time = pts;
933 jbuf->prev_send_diff = gstrtptime - jbuf->base_rtptime;
936 gst_object_unref (media_clock);
938 gst_object_unref (pipeline_clock);
945 * rtp_jitter_buffer_insert:
946 * @jbuf: an #RTPJitterBuffer
947 * @item: an #RTPJitterBufferItem to insert
948 * @head: TRUE when the head element changed.
949 * @percent: the buffering percent after insertion
951 * Inserts @item into the packet queue of @jbuf. The sequence number of the
952 * packet will be used to sort the packets. This function takes ownerhip of
953 * @buf when the function returns %TRUE.
955 * When @head is %TRUE, the new packet was added at the head of the queue and
956 * will be available with the next call to rtp_jitter_buffer_pop() and
957 * rtp_jitter_buffer_peek().
959 * Returns: %FALSE if a packet with the same number already existed.
962 rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
963 gboolean * head, gint * percent)
965 GList *list, *event = NULL;
968 g_return_val_if_fail (jbuf != NULL, FALSE);
969 g_return_val_if_fail (item != NULL, FALSE);
971 list = jbuf->packets->tail;
973 /* no seqnum, simply append then */
974 if (item->seqnum == -1)
977 seqnum = item->seqnum;
979 /* loop the list to skip strictly larger seqnum buffers */
980 for (; list; list = g_list_previous (list)) {
983 RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
985 if (qitem->seqnum == -1) {
986 /* keep a pointer to the first consecutive event if not already
987 * set. we will insert the packet after the event if we can't find
988 * a packet with lower sequence number before the event. */
994 qseq = qitem->seqnum;
996 /* compare the new seqnum to the one in the buffer */
997 gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
999 /* we hit a packet with the same seqnum, notify a duplicate */
1000 if (G_UNLIKELY (gap == 0))
1003 /* seqnum > qseq, we can stop looking */
1004 if (G_LIKELY (gap < 0))
1007 /* if we've found a packet with greater sequence number, cleanup the
1008 * event pointer as the packet will be inserted before the event */
1012 /* if event is set it means that packets before the event had smaller
1013 * sequence number, so we will insert our packet after the event */
1018 queue_do_insert (jbuf, list, (GList *) item);
1020 /* buffering mode, update buffer stats */
1021 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
1022 update_buffer_level (jbuf, percent);
1026 /* head was changed when we did not find a previous packet, we set the return
1027 * flag when requested. */
1028 if (G_LIKELY (head))
1029 *head = (list == NULL);
1036 GST_DEBUG ("duplicate packet %d found", (gint) seqnum);
1037 if (G_LIKELY (head))
1044 * rtp_jitter_buffer_pop:
1045 * @jbuf: an #RTPJitterBuffer
1046 * @percent: the buffering percent
1048 * Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
1049 * have its timestamp adjusted with the incomming running_time and the detected
1052 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
1054 RTPJitterBufferItem *
1055 rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
1060 g_return_val_if_fail (jbuf != NULL, NULL);
1062 queue = jbuf->packets;
1066 queue->head = item->next;
1068 queue->head->prev = NULL;
1074 /* buffering mode, update buffer stats */
1075 if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
1076 update_buffer_level (jbuf, percent);
1080 return (RTPJitterBufferItem *) item;
1084 * rtp_jitter_buffer_peek:
1085 * @jbuf: an #RTPJitterBuffer
1087 * Peek the oldest buffer from the packet queue of @jbuf.
1089 * See rtp_jitter_buffer_insert() to check when an older packet was
1092 * Returns: a #GstBuffer or %NULL when there was no packet in the queue.
1094 RTPJitterBufferItem *
1095 rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
1097 g_return_val_if_fail (jbuf != NULL, NULL);
1099 return (RTPJitterBufferItem *) jbuf->packets->head;
1103 * rtp_jitter_buffer_flush:
1104 * @jbuf: an #RTPJitterBuffer
1105 * @free_func: function to free each item
1106 * @user_data: user data passed to @free_func
1108 * Flush all packets from the jitterbuffer.
1111 rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
1116 g_return_if_fail (jbuf != NULL);
1117 g_return_if_fail (free_func != NULL);
1119 while ((item = g_queue_pop_head_link (jbuf->packets)))
1120 free_func ((RTPJitterBufferItem *) item, user_data);
1124 * rtp_jitter_buffer_is_buffering:
1125 * @jbuf: an #RTPJitterBuffer
1127 * Check if @jbuf is buffering currently. Users of the jitterbuffer should not
1128 * pop packets while in buffering mode.
1130 * Returns: the buffering state of @jbuf
1133 rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
1135 return jbuf->buffering && !jbuf->buffering_disabled;
1139 * rtp_jitter_buffer_set_buffering:
1140 * @jbuf: an #RTPJitterBuffer
1141 * @buffering: the new buffering state
1143 * Forces @jbuf to go into the buffering state.
1146 rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
1148 jbuf->buffering = buffering;
1152 * rtp_jitter_buffer_get_percent:
1153 * @jbuf: an #RTPJitterBuffer
1155 * Get the buffering percent of the jitterbuffer.
1157 * Returns: the buffering percent
1160 rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
1165 if (G_UNLIKELY (jbuf->high_level == 0))
1168 if (G_UNLIKELY (jbuf->buffering_disabled))
1171 level = get_buffer_level (jbuf);
1172 percent = (level * 100 / jbuf->high_level);
1173 percent = MIN (percent, 100);
1179 * rtp_jitter_buffer_num_packets:
1180 * @jbuf: an #RTPJitterBuffer
1182 * Get the number of packets currently in "jbuf.
1184 * Returns: The number of packets in @jbuf.
1187 rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
1189 g_return_val_if_fail (jbuf != NULL, 0);
1191 return jbuf->packets->length;
1195 * rtp_jitter_buffer_get_ts_diff:
1196 * @jbuf: an #RTPJitterBuffer
1198 * Get the difference between the timestamps of first and last packet in the
1201 * Returns: The difference expressed in the timestamp units of the packets.
1204 rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
1206 guint64 high_ts, low_ts;
1207 RTPJitterBufferItem *high_buf, *low_buf;
1210 g_return_val_if_fail (jbuf != NULL, 0);
1212 high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
1213 low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
1215 if (!high_buf || !low_buf || high_buf == low_buf)
1218 high_ts = high_buf->rtptime;
1219 low_ts = low_buf->rtptime;
1221 /* it needs to work if ts wraps */
1222 if (high_ts >= low_ts) {
1223 result = (guint32) (high_ts - low_ts);
1225 result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
1231 * rtp_jitter_buffer_get_sync:
1232 * @jbuf: an #RTPJitterBuffer
1233 * @rtptime: result RTP time
1234 * @timestamp: result GStreamer timestamp
1235 * @clock_rate: clock-rate of @rtptime
1236 * @last_rtptime: last seen rtptime.
1238 * Calculates the relation between the RTP timestamp and the GStreamer timestamp
1239 * used for constructing timestamps.
1241 * For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
1242 * the GStreamer timestamp is currently @timestamp.
1244 * The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
1248 rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
1249 guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
1252 *rtptime = jbuf->base_extrtp;
1254 *timestamp = jbuf->base_time + jbuf->skew;
1256 *clock_rate = jbuf->clock_rate;
1258 *last_rtptime = jbuf->last_rtptime;
1262 * rtp_jitter_buffer_can_fast_start:
1263 * @jbuf: an #RTPJitterBuffer
1264 * @num_packets: Number of consecutive packets needed
1266 * Check if in the queue if there is enough packets with consecutive seqnum in
1267 * order to start delivering them.
1269 * Returns: %TRUE if the required number of consecutive packets was found.
1272 rtp_jitter_buffer_can_fast_start (RTPJitterBuffer * jbuf, gint num_packet)
1274 gboolean ret = TRUE;
1275 RTPJitterBufferItem *last_item = NULL, *item;
1278 if (rtp_jitter_buffer_num_packets (jbuf) < num_packet)
1281 item = rtp_jitter_buffer_peek (jbuf);
1282 for (i = 0; i < num_packet; i++) {
1283 if (G_LIKELY (last_item)) {
1284 guint16 expected_seqnum = last_item->seqnum + 1;
1286 if (expected_seqnum != item->seqnum) {
1293 item = (RTPJitterBufferItem *) last_item->next;