gstreamer/gst/rtpmanager/rtpjitterbuffer.c
Wim Taymans 2e7f5c08cf jitterbuffer: rework resync handling
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
2014-10-21 11:57:34 +02:00

1006 lines
29 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include <string.h>
#include <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include "rtpjitterbuffer.h"
GST_DEBUG_CATEGORY_STATIC (rtp_jitter_buffer_debug);
#define GST_CAT_DEFAULT rtp_jitter_buffer_debug
#define MAX_WINDOW RTP_JITTER_BUFFER_MAX_WINDOW
#define MAX_TIME (2 * GST_SECOND)
/* signals and args */
enum
{
LAST_SIGNAL
};
enum
{
PROP_0
};
/* GObject vmethods */
static void rtp_jitter_buffer_finalize (GObject * object);
GType
rtp_jitter_buffer_mode_get_type (void)
{
static GType jitter_buffer_mode_type = 0;
static const GEnumValue jitter_buffer_modes[] = {
{RTP_JITTER_BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
{RTP_JITTER_BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
{RTP_JITTER_BUFFER_MODE_BUFFER, "Do low/high watermark buffering",
"buffer"},
{RTP_JITTER_BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks",
"synced"},
{0, NULL, NULL},
};
if (!jitter_buffer_mode_type) {
jitter_buffer_mode_type =
g_enum_register_static ("RTPJitterBufferMode", jitter_buffer_modes);
}
return jitter_buffer_mode_type;
}
/* static guint rtp_jitter_buffer_signals[LAST_SIGNAL] = { 0 }; */
G_DEFINE_TYPE (RTPJitterBuffer, rtp_jitter_buffer, G_TYPE_OBJECT);
static void
rtp_jitter_buffer_class_init (RTPJitterBufferClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_jitter_buffer_finalize;
GST_DEBUG_CATEGORY_INIT (rtp_jitter_buffer_debug, "rtpjitterbuffer", 0,
"RTP Jitter Buffer");
}
static void
rtp_jitter_buffer_init (RTPJitterBuffer * jbuf)
{
jbuf->packets = g_queue_new ();
jbuf->mode = RTP_JITTER_BUFFER_MODE_SLAVE;
rtp_jitter_buffer_reset_skew (jbuf);
}
static void
rtp_jitter_buffer_finalize (GObject * object)
{
RTPJitterBuffer *jbuf;
jbuf = RTP_JITTER_BUFFER_CAST (object);
g_queue_free (jbuf->packets);
G_OBJECT_CLASS (rtp_jitter_buffer_parent_class)->finalize (object);
}
/**
* rtp_jitter_buffer_new:
*
* Create an #RTPJitterBuffer.
*
* Returns: a new #RTPJitterBuffer. Use g_object_unref() after usage.
*/
RTPJitterBuffer *
rtp_jitter_buffer_new (void)
{
RTPJitterBuffer *jbuf;
jbuf = g_object_new (RTP_TYPE_JITTER_BUFFER, NULL);
return jbuf;
}
/**
* rtp_jitter_buffer_get_mode:
* @jbuf: an #RTPJitterBuffer
*
* Get the current jitterbuffer mode.
*
* Returns: the current jitterbuffer mode.
*/
RTPJitterBufferMode
rtp_jitter_buffer_get_mode (RTPJitterBuffer * jbuf)
{
return jbuf->mode;
}
/**
* rtp_jitter_buffer_set_mode:
* @jbuf: an #RTPJitterBuffer
* @mode: a #RTPJitterBufferMode
*
* Set the buffering and clock slaving algorithm used in the @jbuf.
*/
void
rtp_jitter_buffer_set_mode (RTPJitterBuffer * jbuf, RTPJitterBufferMode mode)
{
jbuf->mode = mode;
}
GstClockTime
rtp_jitter_buffer_get_delay (RTPJitterBuffer * jbuf)
{
return jbuf->delay;
}
void
rtp_jitter_buffer_set_delay (RTPJitterBuffer * jbuf, GstClockTime delay)
{
jbuf->delay = delay;
jbuf->low_level = (delay * 15) / 100;
/* the high level is at 90% in order to release packets before we fill up the
* buffer up to the latency */
jbuf->high_level = (delay * 90) / 100;
GST_DEBUG ("delay %" GST_TIME_FORMAT ", min %" GST_TIME_FORMAT ", max %"
GST_TIME_FORMAT, GST_TIME_ARGS (jbuf->delay),
GST_TIME_ARGS (jbuf->low_level), GST_TIME_ARGS (jbuf->high_level));
}
/**
* rtp_jitter_buffer_set_clock_rate:
* @jbuf: an #RTPJitterBuffer
* @clock_rate: the new clock rate
*
* Set the clock rate in the jitterbuffer.
*/
void
rtp_jitter_buffer_set_clock_rate (RTPJitterBuffer * jbuf, guint32 clock_rate)
{
if (jbuf->clock_rate != clock_rate) {
GST_DEBUG ("Clock rate changed from %" G_GUINT32_FORMAT " to %"
G_GUINT32_FORMAT, jbuf->clock_rate, clock_rate);
jbuf->clock_rate = clock_rate;
rtp_jitter_buffer_reset_skew (jbuf);
}
}
/**
* rtp_jitter_buffer_get_clock_rate:
* @jbuf: an #RTPJitterBuffer
*
* Get the currently configure clock rate in @jbuf.
*
* Returns: the current clock-rate
*/
guint32
rtp_jitter_buffer_get_clock_rate (RTPJitterBuffer * jbuf)
{
return jbuf->clock_rate;
}
/**
* rtp_jitter_buffer_reset_skew:
* @jbuf: an #RTPJitterBuffer
*
* Reset the skew calculations in @jbuf.
*/
void
rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
{
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->base_extrtp = -1;
jbuf->ext_rtptime = -1;
jbuf->last_rtptime = -1;
jbuf->window_pos = 0;
jbuf->window_filling = TRUE;
jbuf->window_min = 0;
jbuf->skew = 0;
jbuf->prev_send_diff = -1;
jbuf->prev_out_time = -1;
jbuf->need_resync = TRUE;
GST_DEBUG ("reset skew correction");
}
/**
* rtp_jitter_buffer_disable_buffering:
* @jbuf: an #RTPJitterBuffer
* @disabled: the new state
*
* Enable or disable buffering on @jbuf.
*/
void
rtp_jitter_buffer_disable_buffering (RTPJitterBuffer * jbuf, gboolean disabled)
{
jbuf->buffering_disabled = disabled;
}
static void
rtp_jitter_buffer_resync (RTPJitterBuffer * jbuf, GstClockTime time,
GstClockTime gstrtptime, guint64 ext_rtptime, gboolean reset_skew)
{
jbuf->base_time = time;
jbuf->base_rtptime = gstrtptime;
jbuf->base_extrtp = ext_rtptime;
jbuf->prev_out_time = -1;
jbuf->prev_send_diff = -1;
if (reset_skew) {
jbuf->window_filling = TRUE;
jbuf->window_pos = 0;
jbuf->window_min = 0;
jbuf->window_size = 0;
jbuf->skew = 0;
}
jbuf->need_resync = FALSE;
}
static guint64
get_buffer_level (RTPJitterBuffer * jbuf)
{
RTPJitterBufferItem *high_buf = NULL, *low_buf = NULL;
guint64 level;
/* first first buffer with timestamp */
high_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
while (high_buf) {
if (high_buf->dts != -1 || high_buf->pts != -1)
break;
high_buf = (RTPJitterBufferItem *) g_list_previous (high_buf);
}
low_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
while (low_buf) {
if (low_buf->dts != -1 || low_buf->pts != -1)
break;
low_buf = (RTPJitterBufferItem *) g_list_next (low_buf);
}
if (!high_buf || !low_buf || high_buf == low_buf) {
level = 0;
} else {
guint64 high_ts, low_ts;
high_ts = high_buf->dts != -1 ? high_buf->dts : high_buf->pts;
low_ts = low_buf->dts != -1 ? low_buf->dts : low_buf->pts;
if (high_ts > low_ts)
level = high_ts - low_ts;
else
level = 0;
GST_LOG_OBJECT (jbuf,
"low %" GST_TIME_FORMAT " high %" GST_TIME_FORMAT " level %"
G_GUINT64_FORMAT, GST_TIME_ARGS (low_ts), GST_TIME_ARGS (high_ts),
level);
}
return level;
}
static void
update_buffer_level (RTPJitterBuffer * jbuf, gint * percent)
{
gboolean post = FALSE;
guint64 level;
level = get_buffer_level (jbuf);
GST_DEBUG ("buffer level %" GST_TIME_FORMAT, GST_TIME_ARGS (level));
if (jbuf->buffering_disabled) {
GST_DEBUG ("buffering is disabled");
level = jbuf->high_level;
}
if (jbuf->buffering) {
post = TRUE;
if (level >= jbuf->high_level) {
GST_DEBUG ("buffering finished");
jbuf->buffering = FALSE;
}
} else {
if (level < jbuf->low_level) {
GST_DEBUG ("buffering started");
jbuf->buffering = TRUE;
post = TRUE;
}
}
if (post) {
gint perc;
if (jbuf->buffering && (jbuf->high_level != 0)) {
perc = (level * 100 / jbuf->high_level);
perc = MIN (perc, 100);
} else {
perc = 100;
}
if (percent)
*percent = perc;
GST_DEBUG ("buffering %d", perc);
}
}
/* For the clock skew we use a windowed low point averaging algorithm as can be
* found in Fober, Orlarey and Letz, 2005, "Real Time Clock Skew Estimation
* over Network Delays":
* http://www.grame.fr/Ressources/pub/TR-050601.pdf
* http://citeseerx.ist.psu.edu/viewdoc/summary?doi=10.1.1.102.1546
*
* The idea is that the jitter is composed of:
*
* J = N + n
*
* N : a constant network delay.
* n : random added noise. The noise is concentrated around 0
*
* In the receiver we can track the elapsed time at the sender with:
*
* send_diff(i) = (Tsi - Ts0);
*
* Tsi : The time at the sender at packet i
* Ts0 : The time at the sender at the first packet
*
* This is the difference between the RTP timestamp in the first received packet
* and the current packet.
*
* At the receiver we have to deal with the jitter introduced by the network.
*
* recv_diff(i) = (Tri - Tr0)
*
* Tri : The time at the receiver at packet i
* Tr0 : The time at the receiver at the first packet
*
* Both of these values contain a jitter Ji, a jitter for packet i, so we can
* write:
*
* recv_diff(i) = (Cri + D + ni) - (Cr0 + D + n0))
*
* Cri : The time of the clock at the receiver for packet i
* D + ni : The jitter when receiving packet i
*
* We see that the network delay is irrelevant here as we can elliminate D:
*
* recv_diff(i) = (Cri + ni) - (Cr0 + n0))
*
* The drift is now expressed as:
*
* Drift(i) = recv_diff(i) - send_diff(i);
*
* We now keep the W latest values of Drift and find the minimum (this is the
* one with the lowest network jitter and thus the one which is least affected
* by it). We average this lowest value to smooth out the resulting network skew.
*
* Both the window and the weighting used for averaging influence the accuracy
* of the drift estimation. Finding the correct parameters turns out to be a
* compromise between accuracy and inertia.
*
* We use a 2 second window or up to 512 data points, which is statistically big
* enough to catch spikes (FIXME, detect spikes).
* We also use a rather large weighting factor (125) to smoothly adapt. During
* startup, when filling the window, we use a parabolic weighting factor, the
* more the window is filled, the faster we move to the detected possible skew.
*
* Returns: @time adjusted with the clock skew.
*/
static GstClockTime
calculate_skew (RTPJitterBuffer * jbuf, guint32 rtptime, GstClockTime time)
{
guint64 ext_rtptime;
guint64 send_diff, recv_diff;
gint64 delta;
gint64 old;
gint pos, i;
GstClockTime gstrtptime, out_time;
guint64 slope;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
if (jbuf->last_rtptime != -1 && ext_rtptime == jbuf->last_rtptime)
return jbuf->prev_out_time;
gstrtptime =
gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, jbuf->clock_rate);
/* keep track of the last extended rtptime */
jbuf->last_rtptime = ext_rtptime;
send_diff = 0;
if (G_LIKELY (jbuf->base_rtptime != -1)) {
/* check elapsed time in RTP units */
if (G_LIKELY (gstrtptime >= jbuf->base_rtptime)) {
send_diff = gstrtptime - jbuf->base_rtptime;
} else {
/* elapsed time at sender, timestamps can go backwards and thus be
* smaller than our base time, schedule to take a new base time in
* that case. */
GST_WARNING ("backward timestamps at server, schedule resync");
jbuf->need_resync = TRUE;
send_diff = 0;
}
}
/* need resync, lock on to time and gstrtptime if we can, otherwise we
* do with the previous values */
if (G_UNLIKELY (jbuf->need_resync && time != -1)) {
GST_WARNING ("resync to time %" GST_TIME_FORMAT ", rtptime %"
GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (gstrtptime));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, FALSE);
send_diff = 0;
}
GST_DEBUG ("extrtp %" G_GUINT64_FORMAT ", gstrtp %" GST_TIME_FORMAT ", base %"
GST_TIME_FORMAT ", send_diff %" GST_TIME_FORMAT, ext_rtptime,
GST_TIME_ARGS (gstrtptime), GST_TIME_ARGS (jbuf->base_rtptime),
GST_TIME_ARGS (send_diff));
/* we don't have an arrival timestamp so we can't do skew detection. we
* should still apply a timestamp based on RTP timestamp and base_time */
if (time == -1 || jbuf->base_time == -1)
goto no_skew;
/* elapsed time at receiver, includes the jitter */
recv_diff = time - jbuf->base_time;
/* measure the diff */
delta = ((gint64) recv_diff) - ((gint64) send_diff);
/* measure the slope, this gives a rought estimate between the sender speed
* and the receiver speed. This should be approximately 8, higher values
* indicate a burst (especially when the connection starts) */
if (recv_diff > 0)
slope = (send_diff * 8) / recv_diff;
else
slope = 8;
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
GST_TIME_FORMAT ", slope %" G_GUINT64_FORMAT, GST_TIME_ARGS (time),
GST_TIME_ARGS (jbuf->base_time), GST_TIME_ARGS (recv_diff), slope);
/* if the difference between the sender timeline and the receiver timeline
* changed too quickly we have to resync because the server likely restarted
* its timestamps. */
if (ABS (delta - jbuf->skew) > GST_SECOND) {
GST_WARNING ("delta - skew: %" GST_TIME_FORMAT " too big, reset skew",
GST_TIME_ARGS (ABS (delta - jbuf->skew)));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
send_diff = 0;
delta = 0;
}
pos = jbuf->window_pos;
if (G_UNLIKELY (jbuf->window_filling)) {
/* we are filling the window */
GST_DEBUG ("filling %d, delta %" G_GINT64_FORMAT, pos, delta);
jbuf->window[pos++] = delta;
/* calc the min delta we observed */
if (G_UNLIKELY (pos == 1 || delta < jbuf->window_min))
jbuf->window_min = delta;
if (G_UNLIKELY (send_diff >= MAX_TIME || pos >= MAX_WINDOW)) {
jbuf->window_size = pos;
/* window filled */
GST_DEBUG ("min %" G_GINT64_FORMAT, jbuf->window_min);
/* the skew is now the min */
jbuf->skew = jbuf->window_min;
jbuf->window_filling = FALSE;
} else {
gint perc_time, perc_window, perc;
/* figure out how much we filled the window, this depends on the amount of
* time we have or the max number of points we keep. */
perc_time = send_diff * 100 / MAX_TIME;
perc_window = pos * 100 / MAX_WINDOW;
perc = MAX (perc_time, perc_window);
/* make a parabolic function, the closer we get to the MAX, the more value
* we give to the scaling factor of the new value */
perc = perc * perc;
/* quickly go to the min value when we are filling up, slowly when we are
* just starting because we're not sure it's a good value yet. */
jbuf->skew =
(perc * jbuf->window_min + ((10000 - perc) * jbuf->skew)) / 10000;
jbuf->window_size = pos + 1;
}
} else {
/* pick old value and store new value. We keep the previous value in order
* to quickly check if the min of the window changed */
old = jbuf->window[pos];
jbuf->window[pos++] = delta;
if (G_UNLIKELY (delta <= jbuf->window_min)) {
/* if the new value we inserted is smaller or equal to the current min,
* it becomes the new min */
jbuf->window_min = delta;
} else if (G_UNLIKELY (old == jbuf->window_min)) {
gint64 min = G_MAXINT64;
/* if we removed the old min, we have to find a new min */
for (i = 0; i < jbuf->window_size; i++) {
/* we found another value equal to the old min, we can stop searching now */
if (jbuf->window[i] == old) {
min = old;
break;
}
if (jbuf->window[i] < min)
min = jbuf->window[i];
}
jbuf->window_min = min;
}
/* average the min values */
jbuf->skew = (jbuf->window_min + (124 * jbuf->skew)) / 125;
GST_DEBUG ("delta %" G_GINT64_FORMAT ", new min: %" G_GINT64_FORMAT,
delta, jbuf->window_min);
}
/* wrap around in the window */
if (G_UNLIKELY (pos >= jbuf->window_size))
pos = 0;
jbuf->window_pos = pos;
no_skew:
/* the output time is defined as the base timestamp plus the RTP time
* adjusted for the clock skew .*/
if (jbuf->base_time != -1) {
out_time = jbuf->base_time + send_diff;
/* skew can be negative and we don't want to make invalid timestamps */
if (jbuf->skew < 0 && out_time < -jbuf->skew) {
out_time = 0;
} else {
out_time += jbuf->skew;
}
/* check if timestamps are not going backwards, we can only check this if we
* have a previous out time and a previous send_diff */
if (G_LIKELY (jbuf->prev_out_time != -1 && jbuf->prev_send_diff != -1)) {
/* now check for backwards timestamps */
if (G_UNLIKELY (
/* if the server timestamps went up and the out_time backwards */
(send_diff > jbuf->prev_send_diff
&& out_time < jbuf->prev_out_time) ||
/* if the server timestamps went backwards and the out_time forwards */
(send_diff < jbuf->prev_send_diff
&& out_time > jbuf->prev_out_time) ||
/* if the server timestamps did not change */
send_diff == jbuf->prev_send_diff)) {
GST_DEBUG ("backwards timestamps, using previous time");
out_time = jbuf->prev_out_time;
}
}
if (time != -1 && out_time + jbuf->delay < time) {
/* if we are going to produce a timestamp that is later than the input
* timestamp, we need to reset the jitterbuffer. Likely the server paused
* temporarily */
GST_DEBUG ("out %" GST_TIME_FORMAT " + %" G_GUINT64_FORMAT " < time %"
GST_TIME_FORMAT ", reset jitterbuffer", GST_TIME_ARGS (out_time),
jbuf->delay, GST_TIME_ARGS (time));
rtp_jitter_buffer_resync (jbuf, time, gstrtptime, ext_rtptime, TRUE);
out_time = time;
send_diff = 0;
}
} else
out_time = -1;
jbuf->prev_out_time = out_time;
jbuf->prev_send_diff = send_diff;
GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
jbuf->skew, GST_TIME_ARGS (out_time));
return out_time;
}
static void
queue_do_insert (RTPJitterBuffer * jbuf, GList * list, GList * item)
{
GQueue *queue = jbuf->packets;
/* It's more likely that the packet was inserted at the tail of the queue */
if (G_LIKELY (list)) {
item->prev = list;
item->next = list->next;
list->next = item;
} else {
item->prev = NULL;
item->next = queue->head;
queue->head = item;
}
if (item->next)
item->next->prev = item;
else
queue->tail = item;
queue->length++;
}
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
* @item: an #RTPJitterBufferItem to insert
* @head: TRUE when the head element changed.
* @percent: the buffering percent after insertion
*
* Inserts @item into the packet queue of @jbuf. The sequence number of the
* packet will be used to sort the packets. This function takes ownerhip of
* @buf when the function returns %TRUE.
*
* When @head is %TRUE, the new packet was added at the head of the queue and
* will be available with the next call to rtp_jitter_buffer_pop() and
* rtp_jitter_buffer_peek().
*
* Returns: %FALSE if a packet with the same number already existed.
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, RTPJitterBufferItem * item,
gboolean * head, gint * percent)
{
GList *list, *event = NULL;
guint32 rtptime;
guint16 seqnum;
GstClockTime dts;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (item != NULL, FALSE);
list = jbuf->packets->tail;
/* no seqnum, simply append then */
if (item->seqnum == -1)
goto append;
seqnum = item->seqnum;
/* loop the list to skip strictly larger seqnum buffers */
for (; list; list = g_list_previous (list)) {
guint16 qseq;
gint gap;
RTPJitterBufferItem *qitem = (RTPJitterBufferItem *) list;
if (qitem->seqnum == -1) {
/* keep a pointer to the first consecutive event if not already
* set. we will insert the packet after the event if we can't find
* a packet with lower sequence number before the event. */
if (event == NULL)
event = list;
continue;
}
qseq = qitem->seqnum;
/* compare the new seqnum to the one in the buffer */
gap = gst_rtp_buffer_compare_seqnum (seqnum, qseq);
/* we hit a packet with the same seqnum, notify a duplicate */
if (G_UNLIKELY (gap == 0))
goto duplicate;
/* seqnum > qseq, we can stop looking */
if (G_LIKELY (gap < 0))
break;
/* if we've found a packet with greater sequence number, cleanup the
* event pointer as the packet will be inserted before the event */
event = NULL;
}
/* if event is set it means that packets before the event had smaller
* sequence number, so we will insert our packet after the event */
if (event)
list = event;
dts = item->dts;
if (item->rtptime == -1)
goto append;
rtptime = item->rtptime;
/* rtp time jumps are checked for during skew calculation, but bypassed
* in other mode, so mind those here and reset jb if needed.
* Only reset if valid input time, which is likely for UDP input
* where we expect this might happen due to async thread effects
* (in seek and state change cycles), but not so much for TCP input */
if (GST_CLOCK_TIME_IS_VALID (dts) &&
jbuf->mode != RTP_JITTER_BUFFER_MODE_SLAVE &&
jbuf->base_time != -1 && jbuf->last_rtptime != -1) {
GstClockTime ext_rtptime = jbuf->ext_rtptime;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
if (ext_rtptime > jbuf->last_rtptime + 3 * jbuf->clock_rate ||
ext_rtptime + 3 * jbuf->clock_rate < jbuf->last_rtptime) {
/* reset even if we don't have valid incoming time;
* still better than producing possibly very bogus output timestamp */
GST_WARNING ("rtp delta too big, reset skew");
rtp_jitter_buffer_reset_skew (jbuf);
}
}
switch (jbuf->mode) {
case RTP_JITTER_BUFFER_MODE_NONE:
case RTP_JITTER_BUFFER_MODE_BUFFER:
/* send 0 as the first timestamp and -1 for the other ones. This will
* interpollate them from the RTP timestamps with a 0 origin. In buffering
* mode we will adjust the outgoing timestamps according to the amount of
* time we spent buffering. */
if (jbuf->base_time == -1)
dts = 0;
else
dts = -1;
break;
case RTP_JITTER_BUFFER_MODE_SYNCED:
/* synchronized clocks, take first timestamp as base, use RTP timestamps
* to interpolate */
if (jbuf->base_time != -1)
dts = -1;
break;
case RTP_JITTER_BUFFER_MODE_SLAVE:
default:
break;
}
/* do skew calculation by measuring the difference between rtptime and the
* receive dts, this function will return the skew corrected rtptime. */
item->pts = calculate_skew (jbuf, rtptime, dts);
append:
queue_do_insert (jbuf, list, (GList *) item);
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
else if (percent)
*percent = -1;
/* head was changed when we did not find a previous packet, we set the return
* flag when requested. */
if (G_LIKELY (head))
*head = (list == NULL);
return TRUE;
/* ERRORS */
duplicate:
{
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
}
/**
* rtp_jitter_buffer_pop:
* @jbuf: an #RTPJitterBuffer
* @percent: the buffering percent
*
* Pops the oldest buffer from the packet queue of @jbuf. The popped buffer will
* have its timestamp adjusted with the incomming running_time and the detected
* clock skew.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
*/
RTPJitterBufferItem *
rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf, gint * percent)
{
GList *item = NULL;
GQueue *queue;
g_return_val_if_fail (jbuf != NULL, NULL);
queue = jbuf->packets;
item = queue->head;
if (item) {
queue->head = item->next;
if (queue->head)
queue->head->prev = NULL;
else
queue->tail = NULL;
queue->length--;
}
/* buffering mode, update buffer stats */
if (jbuf->mode == RTP_JITTER_BUFFER_MODE_BUFFER)
update_buffer_level (jbuf, percent);
else if (percent)
*percent = -1;
return (RTPJitterBufferItem *) item;
}
/**
* rtp_jitter_buffer_peek:
* @jbuf: an #RTPJitterBuffer
*
* Peek the oldest buffer from the packet queue of @jbuf.
*
* See rtp_jitter_buffer_insert() to check when an older packet was
* added.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
*/
RTPJitterBufferItem *
rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
{
g_return_val_if_fail (jbuf != NULL, NULL);
return (RTPJitterBufferItem *) jbuf->packets->head;
}
/**
* rtp_jitter_buffer_flush:
* @jbuf: an #RTPJitterBuffer
* @free_func: function to free each item
* @user_data: user data passed to @free_func
*
* Flush all packets from the jitterbuffer.
*/
void
rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf, GFunc free_func,
gpointer user_data)
{
GList *item;
g_return_if_fail (jbuf != NULL);
g_return_if_fail (free_func != NULL);
while ((item = g_queue_pop_head_link (jbuf->packets)))
free_func ((RTPJitterBufferItem *) item, user_data);
}
/**
* rtp_jitter_buffer_is_buffering:
* @jbuf: an #RTPJitterBuffer
*
* Check if @jbuf is buffering currently. Users of the jitterbuffer should not
* pop packets while in buffering mode.
*
* Returns: the buffering state of @jbuf
*/
gboolean
rtp_jitter_buffer_is_buffering (RTPJitterBuffer * jbuf)
{
return jbuf->buffering && !jbuf->buffering_disabled;
}
/**
* rtp_jitter_buffer_set_buffering:
* @jbuf: an #RTPJitterBuffer
* @buffering: the new buffering state
*
* Forces @jbuf to go into the buffering state.
*/
void
rtp_jitter_buffer_set_buffering (RTPJitterBuffer * jbuf, gboolean buffering)
{
jbuf->buffering = buffering;
}
/**
* rtp_jitter_buffer_get_percent:
* @jbuf: an #RTPJitterBuffer
*
* Get the buffering percent of the jitterbuffer.
*
* Returns: the buffering percent
*/
gint
rtp_jitter_buffer_get_percent (RTPJitterBuffer * jbuf)
{
gint percent;
guint64 level;
if (G_UNLIKELY (jbuf->high_level == 0))
return 100;
if (G_UNLIKELY (jbuf->buffering_disabled))
return 100;
level = get_buffer_level (jbuf);
percent = (level * 100 / jbuf->high_level);
percent = MIN (percent, 100);
return percent;
}
/**
* rtp_jitter_buffer_num_packets:
* @jbuf: an #RTPJitterBuffer
*
* Get the number of packets currently in "jbuf.
*
* Returns: The number of packets in @jbuf.
*/
guint
rtp_jitter_buffer_num_packets (RTPJitterBuffer * jbuf)
{
g_return_val_if_fail (jbuf != NULL, 0);
return jbuf->packets->length;
}
/**
* rtp_jitter_buffer_get_ts_diff:
* @jbuf: an #RTPJitterBuffer
*
* Get the difference between the timestamps of first and last packet in the
* jitterbuffer.
*
* Returns: The difference expressed in the timestamp units of the packets.
*/
guint32
rtp_jitter_buffer_get_ts_diff (RTPJitterBuffer * jbuf)
{
guint64 high_ts, low_ts;
RTPJitterBufferItem *high_buf, *low_buf;
guint32 result;
g_return_val_if_fail (jbuf != NULL, 0);
high_buf = (RTPJitterBufferItem *) g_queue_peek_head_link (jbuf->packets);
low_buf = (RTPJitterBufferItem *) g_queue_peek_tail_link (jbuf->packets);
if (!high_buf || !low_buf || high_buf == low_buf)
return 0;
high_ts = high_buf->rtptime;
low_ts = low_buf->rtptime;
/* it needs to work if ts wraps */
if (high_ts >= low_ts) {
result = (guint32) (high_ts - low_ts);
} else {
result = (guint32) (high_ts + G_MAXUINT32 + 1 - low_ts);
}
return result;
}
/**
* rtp_jitter_buffer_get_sync:
* @jbuf: an #RTPJitterBuffer
* @rtptime: result RTP time
* @timestamp: result GStreamer timestamp
* @clock_rate: clock-rate of @rtptime
* @last_rtptime: last seen rtptime.
*
* Calculates the relation between the RTP timestamp and the GStreamer timestamp
* used for constructing timestamps.
*
* For extended RTP timestamp @rtptime with a clock-rate of @clock_rate,
* the GStreamer timestamp is currently @timestamp.
*
* The last seen extended RTP timestamp with clock-rate @clock-rate is returned in
* @last_rtptime.
*/
void
rtp_jitter_buffer_get_sync (RTPJitterBuffer * jbuf, guint64 * rtptime,
guint64 * timestamp, guint32 * clock_rate, guint64 * last_rtptime)
{
if (rtptime)
*rtptime = jbuf->base_extrtp;
if (timestamp)
*timestamp = jbuf->base_time + jbuf->skew;
if (clock_rate)
*clock_rate = jbuf->clock_rate;
if (last_rtptime)
*last_rtptime = jbuf->last_rtptime;
}