gstreamer/gst/rtpmanager/rtpjitterbuffer.c
Wim Taymans bd1e0ebfc0 gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ...
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
When checking the seqnum, reset the jitterbuffer if the gap is too big,
we need to do this so that we can better handle a restarted source.
Fix some comments.
* gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
(rtp_jitter_buffer_insert):
Tweak the skew resync diff.
Use our working seqnum compare function in -base.
Rework the jitterbuffer insert code to make it clearer and more
performant by only retrieving the seqnum of the input buffer once and by
adding some G_LIKELY compiler hints.
Improve debugging for duplicate packets.
* gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
Fix a comment, we don't do skew correction here..
2009-08-11 02:30:36 +01:00

515 lines
15 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, 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);
/* 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 ();
rtp_jitter_buffer_reset_skew (jbuf);
}
static void
rtp_jitter_buffer_finalize (GObject * object)
{
RTPJitterBuffer *jbuf;
jbuf = RTP_JITTER_BUFFER_CAST (object);
rtp_jitter_buffer_flush (jbuf);
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;
}
void
rtp_jitter_buffer_reset_skew (RTPJitterBuffer * jbuf)
{
jbuf->base_time = -1;
jbuf->base_rtptime = -1;
jbuf->ext_rtptime = -1;
jbuf->window_pos = 0;
jbuf->window_filling = TRUE;
jbuf->window_min = 0;
jbuf->skew = 0;
jbuf->prev_send_diff = -1;
}
/* For the clock skew we use a windowed low point averaging algorithm as can be
* found in http://www.grame.fr/pub/TR-050601.pdf. 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,
guint32 clock_rate)
{
guint64 ext_rtptime;
guint64 send_diff, recv_diff;
gint64 delta;
gint64 old;
gint pos, i;
GstClockTime gstrtptime, out_time;
ext_rtptime = gst_rtp_buffer_ext_timestamp (&jbuf->ext_rtptime, rtptime);
gstrtptime = gst_util_uint64_scale_int (ext_rtptime, GST_SECOND, clock_rate);
again:
/* first time, lock on to time and gstrtptime */
if (jbuf->base_time == -1)
jbuf->base_time = time;
if (jbuf->base_rtptime == -1)
jbuf->base_rtptime = gstrtptime;
if (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, take a new base time in that case. */
GST_DEBUG ("backward timestamps at server, taking new base time");
jbuf->base_rtptime = gstrtptime;
jbuf->base_time = time;
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));
if (jbuf->prev_send_diff != -1 && time != -1) {
gint64 delta_diff;
if (send_diff > jbuf->prev_send_diff)
delta_diff = send_diff - jbuf->prev_send_diff;
else
delta_diff = jbuf->prev_send_diff - send_diff;
/* server changed rtp timestamps too quickly, reset skew detection and start
* again. This value is sortof arbitrary and can be a bad measurement up if
* there are many packets missing because then we get a big gap that is
* unrelated to a timestamp switch. */
if (delta_diff > GST_SECOND) {
GST_DEBUG ("delta changed too quickly %" GST_TIME_FORMAT " reset skew",
GST_TIME_ARGS (delta_diff));
rtp_jitter_buffer_reset_skew (jbuf);
goto again;
}
}
jbuf->prev_send_diff = 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)
goto no_skew;
/* elapsed time at receiver, includes the jitter */
recv_diff = time - jbuf->base_time;
GST_DEBUG ("time %" GST_TIME_FORMAT ", base %" GST_TIME_FORMAT ", recv_diff %"
GST_TIME_FORMAT, GST_TIME_ARGS (time), GST_TIME_ARGS (jbuf->base_time),
GST_TIME_ARGS (recv_diff));
/* measure the diff */
delta = ((gint64) recv_diff) - ((gint64) send_diff);
pos = jbuf->window_pos;
if (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 (pos == 1 || delta < jbuf->window_min)
jbuf->window_min = delta;
if (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 (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 (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 (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 .*/
out_time = jbuf->base_time + send_diff + jbuf->skew;
GST_DEBUG ("skew %" G_GINT64_FORMAT ", out %" GST_TIME_FORMAT,
jbuf->skew, GST_TIME_ARGS (out_time));
return out_time;
}
/**
* rtp_jitter_buffer_insert:
* @jbuf: an #RTPJitterBuffer
* @buf: a buffer
* @time: a running_time when this buffer was received in nanoseconds
* @clock_rate: the clock-rate of the payload of @buf
* @tail: TRUE when the tail element changed.
*
* Inserts @buf 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.
*
* Returns: %FALSE if a packet with the same number already existed.
*/
gboolean
rtp_jitter_buffer_insert (RTPJitterBuffer * jbuf, GstBuffer * buf,
GstClockTime time, guint32 clock_rate, gboolean * tail)
{
GList *list;
guint32 rtptime;
guint16 seqnum;
g_return_val_if_fail (jbuf != NULL, FALSE);
g_return_val_if_fail (buf != NULL, FALSE);
seqnum = gst_rtp_buffer_get_seq (buf);
/* loop the list to skip strictly smaller seqnum buffers */
for (list = jbuf->packets->head; list; list = g_list_next (list)) {
guint16 qseq;
gint gap;
qseq = gst_rtp_buffer_get_seq (GST_BUFFER_CAST (list->data));
/* 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;
}
/* do skew calculation by measuring the difference between rtptime and the
* receive time, this function will retimestamp @buf with the skew corrected
* running time. */
rtptime = gst_rtp_buffer_get_timestamp (buf);
time = calculate_skew (jbuf, rtptime, time, clock_rate);
GST_BUFFER_TIMESTAMP (buf) = time;
if (list)
g_queue_insert_before (jbuf->packets, list, buf);
else
g_queue_push_tail (jbuf->packets, buf);
/* tail was changed when we did not find a previous packet, we set the return
* flag when requested. */
if (tail)
*tail = (list == NULL);
return TRUE;
/* ERRORS */
duplicate:
{
GST_WARNING ("duplicate packet %d found", (gint) seqnum);
return FALSE;
}
}
/**
* rtp_jitter_buffer_pop:
* @jbuf: an #RTPJitterBuffer
*
* 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.
*/
GstBuffer *
rtp_jitter_buffer_pop (RTPJitterBuffer * jbuf)
{
GstBuffer *buf;
g_return_val_if_fail (jbuf != NULL, FALSE);
buf = g_queue_pop_tail (jbuf->packets);
return buf;
}
/**
* rtp_jitter_buffer_peek:
* @jbuf: an #RTPJitterBuffer
*
* Peek the oldest buffer from the packet queue of @jbuf. Register a callback
* with rtp_jitter_buffer_set_tail_changed() to be notified when an older packet
* was inserted in the queue.
*
* Returns: a #GstBuffer or %NULL when there was no packet in the queue.
*/
GstBuffer *
rtp_jitter_buffer_peek (RTPJitterBuffer * jbuf)
{
GstBuffer *buf;
g_return_val_if_fail (jbuf != NULL, FALSE);
buf = g_queue_peek_tail (jbuf->packets);
return buf;
}
/**
* rtp_jitter_buffer_flush:
* @jbuf: an #RTPJitterBuffer
*
* Flush all packets from the jitterbuffer.
*/
void
rtp_jitter_buffer_flush (RTPJitterBuffer * jbuf)
{
GstBuffer *buffer;
g_return_if_fail (jbuf != NULL);
while ((buffer = g_queue_pop_head (jbuf->packets)))
gst_buffer_unref (buffer);
}
/**
* 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;
GstBuffer *high_buf, *low_buf;
guint32 result;
g_return_val_if_fail (jbuf != NULL, 0);
high_buf = g_queue_peek_head (jbuf->packets);
low_buf = g_queue_peek_tail (jbuf->packets);
if (!high_buf || !low_buf || high_buf == low_buf)
return 0;
high_ts = gst_rtp_buffer_get_timestamp (high_buf);
low_ts = gst_rtp_buffer_get_timestamp (low_buf);
/* 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;
}