mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-09-09 13:48:55 +00:00
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/646>
679 lines
20 KiB
C
679 lines
20 KiB
C
/* GStreamer
|
|
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
|
|
* Copyright (C) 2015 Kurento (http://kurento.org/)
|
|
* @author: Miguel París <mparisdiaz@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.
|
|
*/
|
|
|
|
#define GLIB_DISABLE_DEPRECATION_WARNINGS
|
|
|
|
#include "rtpstats.h"
|
|
#include "rtptwcc.h"
|
|
|
|
void
|
|
gst_rtp_packet_rate_ctx_reset (RTPPacketRateCtx * ctx, gint32 clock_rate)
|
|
{
|
|
ctx->clock_rate = clock_rate;
|
|
ctx->probed = FALSE;
|
|
ctx->avg_packet_rate = -1;
|
|
ctx->last_ts = -1;
|
|
}
|
|
|
|
guint32
|
|
gst_rtp_packet_rate_ctx_update (RTPPacketRateCtx * ctx, guint16 seqnum,
|
|
guint32 ts)
|
|
{
|
|
guint64 new_ts, diff_ts;
|
|
gint diff_seqnum;
|
|
gint32 new_packet_rate;
|
|
gint32 base;
|
|
|
|
if (ctx->clock_rate <= 0) {
|
|
return ctx->avg_packet_rate;
|
|
}
|
|
|
|
new_ts = ctx->last_ts;
|
|
gst_rtp_buffer_ext_timestamp (&new_ts, ts);
|
|
|
|
if (!ctx->probed) {
|
|
ctx->probed = TRUE;
|
|
goto done_but_save;
|
|
}
|
|
|
|
diff_seqnum = gst_rtp_buffer_compare_seqnum (ctx->last_seqnum, seqnum);
|
|
/* Ignore seqnums that are over 15,000 away from the latest one, it's close
|
|
* to 2^14 but far enough to avoid any risk of computing error.
|
|
*/
|
|
if (diff_seqnum > 15000)
|
|
goto done_but_save;
|
|
|
|
/* Ignore any packet that is in the past, we're only interested in newer
|
|
* packets to compute the packet rate.
|
|
*/
|
|
if (diff_seqnum <= 0 || new_ts <= ctx->last_ts)
|
|
goto done;
|
|
|
|
diff_ts = new_ts - ctx->last_ts;
|
|
diff_ts = gst_util_uint64_scale_int (diff_ts, GST_SECOND, ctx->clock_rate);
|
|
new_packet_rate = gst_util_uint64_scale (diff_seqnum, GST_SECOND, diff_ts);
|
|
|
|
/* The goal is that higher packet rates "win".
|
|
* If there's a sudden burst, the average will go up fast,
|
|
* but it will go down again slowly.
|
|
* This is useful for bursty cases, where a lot of packets are close
|
|
* to each other and should allow a higher reorder/dropout there.
|
|
* Round up the new average.
|
|
* We do it on different rates depending on the packet rate, so it's not too
|
|
* jumpy.
|
|
*/
|
|
if (ctx->avg_packet_rate > new_packet_rate)
|
|
base = MAX (ctx->avg_packet_rate / 3, 8); /* about 333 ms */
|
|
else
|
|
base = MAX (ctx->avg_packet_rate / 15, 2); /* about 66 ms */
|
|
|
|
diff_seqnum = MIN (diff_seqnum, base - 1);
|
|
|
|
ctx->avg_packet_rate = (((base - diff_seqnum) * ctx->avg_packet_rate) +
|
|
(new_packet_rate * diff_seqnum)) / base;
|
|
|
|
|
|
done_but_save:
|
|
|
|
ctx->last_seqnum = seqnum;
|
|
ctx->last_ts = new_ts;
|
|
done:
|
|
|
|
return ctx->avg_packet_rate;
|
|
}
|
|
|
|
guint32
|
|
gst_rtp_packet_rate_ctx_get (RTPPacketRateCtx * ctx)
|
|
{
|
|
return ctx->avg_packet_rate;
|
|
}
|
|
|
|
guint32
|
|
gst_rtp_packet_rate_ctx_get_max_dropout (RTPPacketRateCtx * ctx, gint32 time_ms)
|
|
{
|
|
if (time_ms <= 0 || !ctx->probed || ctx->avg_packet_rate == -1) {
|
|
return RTP_DEF_DROPOUT;
|
|
}
|
|
|
|
return MAX (RTP_MIN_DROPOUT, ctx->avg_packet_rate * time_ms / 1000);
|
|
}
|
|
|
|
guint32
|
|
gst_rtp_packet_rate_ctx_get_max_misorder (RTPPacketRateCtx * ctx,
|
|
gint32 time_ms)
|
|
{
|
|
if (time_ms <= 0 || !ctx->probed || ctx->avg_packet_rate == -1) {
|
|
return RTP_DEF_MISORDER;
|
|
}
|
|
|
|
return MAX (RTP_MIN_MISORDER, ctx->avg_packet_rate * time_ms / 1000);
|
|
}
|
|
|
|
/**
|
|
* rtp_stats_init_defaults:
|
|
* @stats: an #RTPSessionStats struct
|
|
*
|
|
* Initialize @stats with its default values.
|
|
*/
|
|
void
|
|
rtp_stats_init_defaults (RTPSessionStats * stats)
|
|
{
|
|
rtp_stats_set_bandwidths (stats, -1, -1, -1, -1);
|
|
stats->min_interval = RTP_STATS_MIN_INTERVAL;
|
|
stats->bye_timeout = RTP_STATS_BYE_TIMEOUT;
|
|
stats->nacks_dropped = 0;
|
|
stats->nacks_sent = 0;
|
|
stats->nacks_received = 0;
|
|
}
|
|
|
|
/**
|
|
* rtp_stats_set_bandwidths:
|
|
* @stats: an #RTPSessionStats struct
|
|
* @rtp_bw: RTP bandwidth
|
|
* @rtcp_bw: RTCP bandwidth
|
|
* @rs: sender RTCP bandwidth
|
|
* @rr: receiver RTCP bandwidth
|
|
*
|
|
* Configure the bandwidth parameters in the stats. When an input variable is
|
|
* set to -1, it will be calculated from the other input variables and from the
|
|
* defaults.
|
|
*/
|
|
void
|
|
rtp_stats_set_bandwidths (RTPSessionStats * stats, guint rtp_bw,
|
|
gdouble rtcp_bw, guint rs, guint rr)
|
|
{
|
|
GST_DEBUG ("recalc bandwidths: RTP %u, RTCP %f, RS %u, RR %u", rtp_bw,
|
|
rtcp_bw, rs, rr);
|
|
|
|
/* when given, sender and receive bandwidth add up to the total
|
|
* rtcp bandwidth */
|
|
if (rs != -1 && rr != -1)
|
|
rtcp_bw = rs + rr;
|
|
|
|
/* If rtcp_bw is between 0 and 1, it is a fraction of rtp_bw */
|
|
if (rtcp_bw > 0.0 && rtcp_bw < 1.0) {
|
|
if (rtp_bw > 0.0)
|
|
rtcp_bw = rtp_bw * rtcp_bw;
|
|
else
|
|
rtcp_bw = -1.0;
|
|
}
|
|
|
|
/* RTCP is 5% of the RTP bandwidth */
|
|
if (rtp_bw == -1 && rtcp_bw > 1.0)
|
|
rtp_bw = rtcp_bw * 20;
|
|
else if (rtp_bw != -1 && rtcp_bw < 0.0)
|
|
rtcp_bw = rtp_bw / 20;
|
|
else if (rtp_bw == -1 && rtcp_bw < 0.0) {
|
|
/* nothing given, take defaults */
|
|
rtp_bw = RTP_STATS_BANDWIDTH;
|
|
rtcp_bw = rtp_bw * RTP_STATS_RTCP_FRACTION;
|
|
}
|
|
|
|
stats->bandwidth = rtp_bw;
|
|
stats->rtcp_bandwidth = rtcp_bw;
|
|
|
|
/* now figure out the fractions */
|
|
if (rs == -1) {
|
|
/* rs unknown */
|
|
if (rr == -1) {
|
|
/* both not given, use defaults */
|
|
rs = stats->rtcp_bandwidth * RTP_STATS_SENDER_FRACTION;
|
|
rr = stats->rtcp_bandwidth * RTP_STATS_RECEIVER_FRACTION;
|
|
} else {
|
|
/* rr known, calculate rs */
|
|
if (stats->rtcp_bandwidth > rr)
|
|
rs = stats->rtcp_bandwidth - rr;
|
|
else
|
|
rs = 0;
|
|
}
|
|
} else if (rr == -1) {
|
|
/* rs known, calculate rr */
|
|
if (stats->rtcp_bandwidth > rs)
|
|
rr = stats->rtcp_bandwidth - rs;
|
|
else
|
|
rr = 0;
|
|
}
|
|
|
|
if (stats->rtcp_bandwidth > 0) {
|
|
stats->sender_fraction = ((gdouble) rs) / ((gdouble) stats->rtcp_bandwidth);
|
|
stats->receiver_fraction = 1.0 - stats->sender_fraction;
|
|
} else {
|
|
/* no RTCP bandwidth, set dummy values */
|
|
stats->sender_fraction = 0.0;
|
|
stats->receiver_fraction = 0.0;
|
|
}
|
|
GST_DEBUG ("bandwidths: RTP %u, RTCP %u, RS %f, RR %f", stats->bandwidth,
|
|
stats->rtcp_bandwidth, stats->sender_fraction, stats->receiver_fraction);
|
|
}
|
|
|
|
/**
|
|
* rtp_stats_calculate_rtcp_interval:
|
|
* @stats: an #RTPSessionStats struct
|
|
* @sender: if we are a sender
|
|
* @profile: RTP profile of this session
|
|
* @ptp: if this session is a point-to-point session
|
|
* @first: if this is the first time
|
|
*
|
|
* Calculate the RTCP interval. The result of this function is the amount of
|
|
* time to wait (in nanoseconds) before sending a new RTCP message.
|
|
*
|
|
* Returns: the RTCP interval.
|
|
*/
|
|
GstClockTime
|
|
rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
|
|
GstRTPProfile profile, gboolean ptp, gboolean first)
|
|
{
|
|
gdouble members, senders, n;
|
|
gdouble avg_rtcp_size, rtcp_bw;
|
|
gdouble interval;
|
|
gdouble rtcp_min_time;
|
|
|
|
if (profile == GST_RTP_PROFILE_AVPF || profile == GST_RTP_PROFILE_SAVPF) {
|
|
/* RFC 4585 3.4d), 3.5.1 */
|
|
|
|
if (first && !ptp)
|
|
rtcp_min_time = 1.0;
|
|
else
|
|
rtcp_min_time = 0.0;
|
|
} else {
|
|
/* Very first call at application start-up uses half the min
|
|
* delay for quicker notification while still allowing some time
|
|
* before reporting for randomization and to learn about other
|
|
* sources so the report interval will converge to the correct
|
|
* interval more quickly.
|
|
*/
|
|
rtcp_min_time = stats->min_interval;
|
|
if (first)
|
|
rtcp_min_time /= 2.0;
|
|
}
|
|
|
|
/* Dedicate a fraction of the RTCP bandwidth to senders unless
|
|
* the number of senders is large enough that their share is
|
|
* more than that fraction.
|
|
*/
|
|
n = members = stats->active_sources;
|
|
senders = (gdouble) stats->sender_sources;
|
|
rtcp_bw = stats->rtcp_bandwidth;
|
|
|
|
if (senders <= members * stats->sender_fraction) {
|
|
if (we_send) {
|
|
rtcp_bw *= stats->sender_fraction;
|
|
n = senders;
|
|
} else {
|
|
rtcp_bw *= stats->receiver_fraction;
|
|
n -= senders;
|
|
}
|
|
}
|
|
|
|
/* no bandwidth for RTCP, return NONE to signal that we don't want to send
|
|
* RTCP packets */
|
|
if (rtcp_bw <= 0.0001)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
avg_rtcp_size = 8.0 * stats->avg_rtcp_packet_size;
|
|
/*
|
|
* The effective number of sites times the average packet size is
|
|
* the total number of octets sent when each site sends a report.
|
|
* Dividing this by the effective bandwidth gives the time
|
|
* interval over which those packets must be sent in order to
|
|
* meet the bandwidth target, with a minimum enforced. In that
|
|
* time interval we send one report so this time is also our
|
|
* average time between reports.
|
|
*/
|
|
GST_DEBUG ("avg size %f, n %f, rtcp_bw %f", avg_rtcp_size, n, rtcp_bw);
|
|
interval = avg_rtcp_size * n / rtcp_bw;
|
|
if (interval < rtcp_min_time)
|
|
interval = rtcp_min_time;
|
|
|
|
return interval * GST_SECOND;
|
|
}
|
|
|
|
/**
|
|
* rtp_stats_add_rtcp_jitter:
|
|
* @stats: an #RTPSessionStats struct
|
|
* @interval: an RTCP interval
|
|
*
|
|
* Apply a random jitter to the @interval. @interval is typically obtained with
|
|
* rtp_stats_calculate_rtcp_interval().
|
|
*
|
|
* Returns: the new RTCP interval.
|
|
*/
|
|
GstClockTime
|
|
rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, GstClockTime interval)
|
|
{
|
|
gdouble temp;
|
|
|
|
/* see RFC 3550 p 30
|
|
* To compensate for "unconditional reconsideration" converging to a
|
|
* value below the intended average.
|
|
*/
|
|
#define COMPENSATION (2.71828 - 1.5);
|
|
|
|
temp = (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
|
|
|
|
return (GstClockTime) temp;
|
|
}
|
|
|
|
|
|
/**
|
|
* rtp_stats_calculate_bye_interval:
|
|
* @stats: an #RTPSessionStats struct
|
|
*
|
|
* Calculate the BYE interval. The result of this function is the amount of
|
|
* time to wait (in nanoseconds) before sending a BYE message.
|
|
*
|
|
* Returns: the BYE interval.
|
|
*/
|
|
GstClockTime
|
|
rtp_stats_calculate_bye_interval (RTPSessionStats * stats)
|
|
{
|
|
gdouble members;
|
|
gdouble avg_rtcp_size, rtcp_bw;
|
|
gdouble interval;
|
|
gdouble rtcp_min_time;
|
|
|
|
/* no interval when we have less than 50 members */
|
|
if (stats->active_sources < 50)
|
|
return 0;
|
|
|
|
rtcp_min_time = (stats->min_interval) / 2.0;
|
|
|
|
/* Dedicate a fraction of the RTCP bandwidth to senders unless
|
|
* the number of senders is large enough that their share is
|
|
* more than that fraction.
|
|
*/
|
|
members = stats->bye_members;
|
|
rtcp_bw = stats->rtcp_bandwidth * stats->receiver_fraction;
|
|
|
|
/* no bandwidth for RTCP, return NONE to signal that we don't want to send
|
|
* RTCP packets */
|
|
if (rtcp_bw <= 0.0001)
|
|
return GST_CLOCK_TIME_NONE;
|
|
|
|
avg_rtcp_size = 8.0 * stats->avg_rtcp_packet_size;
|
|
/*
|
|
* The effective number of sites times the average packet size is
|
|
* the total number of octets sent when each site sends a report.
|
|
* Dividing this by the effective bandwidth gives the time
|
|
* interval over which those packets must be sent in order to
|
|
* meet the bandwidth target, with a minimum enforced. In that
|
|
* time interval we send one report so this time is also our
|
|
* average time between reports.
|
|
*/
|
|
interval = avg_rtcp_size * members / rtcp_bw;
|
|
if (interval < rtcp_min_time)
|
|
interval = rtcp_min_time;
|
|
|
|
return interval * GST_SECOND;
|
|
}
|
|
|
|
/**
|
|
* rtp_stats_get_packets_lost:
|
|
* @stats: an #RTPSourceStats struct
|
|
*
|
|
* Calculate the total number of RTP packets lost since beginning of
|
|
* reception. Packets that arrive late are not considered lost, and
|
|
* duplicates are not taken into account. Hence, the loss may be negative
|
|
* if there are duplicates.
|
|
*
|
|
* Returns: total RTP packets lost.
|
|
*/
|
|
gint64
|
|
rtp_stats_get_packets_lost (const RTPSourceStats * stats)
|
|
{
|
|
gint64 lost;
|
|
guint64 extended_max, expected;
|
|
|
|
extended_max = stats->cycles + stats->max_seq;
|
|
expected = extended_max - stats->base_seq + 1;
|
|
lost = expected - stats->packets_received;
|
|
|
|
return lost;
|
|
}
|
|
|
|
void
|
|
rtp_stats_set_min_interval (RTPSessionStats * stats, gdouble min_interval)
|
|
{
|
|
stats->min_interval = min_interval;
|
|
}
|
|
|
|
gboolean
|
|
__g_socket_address_equal (GSocketAddress * a, GSocketAddress * b)
|
|
{
|
|
GInetSocketAddress *ia, *ib;
|
|
GInetAddress *iaa, *iab;
|
|
|
|
ia = G_INET_SOCKET_ADDRESS (a);
|
|
ib = G_INET_SOCKET_ADDRESS (b);
|
|
|
|
if (g_inet_socket_address_get_port (ia) !=
|
|
g_inet_socket_address_get_port (ib))
|
|
return FALSE;
|
|
|
|
iaa = g_inet_socket_address_get_address (ia);
|
|
iab = g_inet_socket_address_get_address (ib);
|
|
|
|
return g_inet_address_equal (iaa, iab);
|
|
}
|
|
|
|
gchar *
|
|
__g_socket_address_to_string (GSocketAddress * addr)
|
|
{
|
|
GInetSocketAddress *ia;
|
|
gchar *ret, *tmp;
|
|
|
|
ia = G_INET_SOCKET_ADDRESS (addr);
|
|
|
|
tmp = g_inet_address_to_string (g_inet_socket_address_get_address (ia));
|
|
ret = g_strdup_printf ("%s:%u", tmp, g_inet_socket_address_get_port (ia));
|
|
g_free (tmp);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
_append_structure_to_value_array (GValueArray * array, GstStructure * s)
|
|
{
|
|
GValue *val;
|
|
g_value_array_append (array, NULL);
|
|
val = g_value_array_get_nth (array, array->n_values - 1);
|
|
g_value_init (val, GST_TYPE_STRUCTURE);
|
|
g_value_take_boxed (val, s);
|
|
}
|
|
|
|
static void
|
|
_structure_take_value_array (GstStructure * s,
|
|
const gchar * field_name, GValueArray * array)
|
|
{
|
|
GValue value = G_VALUE_INIT;
|
|
g_value_init (&value, G_TYPE_VALUE_ARRAY);
|
|
g_value_take_boxed (&value, array);
|
|
gst_structure_take_value (s, field_name, &value);
|
|
g_value_unset (&value);
|
|
}
|
|
|
|
GstStructure *
|
|
rtp_twcc_stats_get_packets_structure (GArray * twcc_packets)
|
|
{
|
|
GstStructure *ret = gst_structure_new_empty ("RTPTWCCPackets");
|
|
GValueArray *array = g_value_array_new (0);
|
|
guint i;
|
|
|
|
for (i = 0; i < twcc_packets->len; i++) {
|
|
RTPTWCCPacket *pkt = &g_array_index (twcc_packets, RTPTWCCPacket, i);
|
|
|
|
GstStructure *pkt_s = gst_structure_new ("RTPTWCCPacket",
|
|
"seqnum", G_TYPE_UINT, pkt->seqnum,
|
|
"local-ts", G_TYPE_UINT64, pkt->local_ts,
|
|
"remote-ts", G_TYPE_UINT64, pkt->remote_ts,
|
|
"size", G_TYPE_UINT, pkt->size,
|
|
"lost", G_TYPE_BOOLEAN, pkt->status == RTP_TWCC_PACKET_STATUS_NOT_RECV,
|
|
NULL);
|
|
_append_structure_to_value_array (array, pkt_s);
|
|
}
|
|
|
|
_structure_take_value_array (ret, "packets", array);
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
rtp_twcc_stats_calculate_stats (RTPTWCCStats * stats, GArray * twcc_packets)
|
|
{
|
|
guint packets_recv = 0;
|
|
guint i;
|
|
|
|
for (i = 0; i < twcc_packets->len; i++) {
|
|
RTPTWCCPacket *pkt = &g_array_index (twcc_packets, RTPTWCCPacket, i);
|
|
|
|
if (pkt->status != RTP_TWCC_PACKET_STATUS_NOT_RECV)
|
|
packets_recv++;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts) &&
|
|
GST_CLOCK_TIME_IS_VALID (stats->last_local_ts)) {
|
|
pkt->local_delta = GST_CLOCK_DIFF (stats->last_local_ts, pkt->local_ts);
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->remote_ts) &&
|
|
GST_CLOCK_TIME_IS_VALID (stats->last_remote_ts)) {
|
|
pkt->remote_delta =
|
|
GST_CLOCK_DIFF (stats->last_remote_ts, pkt->remote_ts);
|
|
}
|
|
|
|
if (GST_CLOCK_STIME_IS_VALID (pkt->local_delta) &&
|
|
GST_CLOCK_STIME_IS_VALID (pkt->remote_delta)) {
|
|
pkt->delta_delta = pkt->remote_delta - pkt->local_delta;
|
|
}
|
|
|
|
stats->last_local_ts = pkt->local_ts;
|
|
stats->last_remote_ts = pkt->remote_ts;
|
|
}
|
|
|
|
stats->packets_sent = twcc_packets->len;
|
|
stats->packets_recv = packets_recv;
|
|
}
|
|
|
|
static gint
|
|
_get_window_start_index (RTPTWCCStats * stats, GstClockTime duration,
|
|
GstClockTime * local_duration, GstClockTime * remote_duration)
|
|
{
|
|
RTPTWCCPacket *last = NULL;
|
|
guint i;
|
|
|
|
if (stats->packets->len < 2)
|
|
return -1;
|
|
|
|
for (i = 0; i < stats->packets->len; i++) {
|
|
guint start_index = stats->packets->len - 1 - i;
|
|
RTPTWCCPacket *pkt =
|
|
&g_array_index (stats->packets, RTPTWCCPacket, start_index);
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts)
|
|
&& GST_CLOCK_TIME_IS_VALID (pkt->remote_ts)) {
|
|
/* first find the last valid packet */
|
|
if (last == NULL) {
|
|
last = pkt;
|
|
} else {
|
|
/* and then get the duration in local ts */
|
|
GstClockTimeDiff ld = GST_CLOCK_DIFF (pkt->local_ts, last->local_ts);
|
|
if (ld >= duration) {
|
|
*local_duration = ld;
|
|
*remote_duration = GST_CLOCK_DIFF (pkt->remote_ts, last->remote_ts);
|
|
return start_index;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static void
|
|
rtp_twcc_stats_calculate_windowed_stats (RTPTWCCStats * stats)
|
|
{
|
|
guint i;
|
|
gint start_idx;
|
|
guint bits_sent = 0;
|
|
guint bits_recv = 0;
|
|
guint packets_sent = 0;
|
|
guint packets_recv = 0;
|
|
guint packets_lost;
|
|
GstClockTimeDiff delta_delta_sum = 0;
|
|
guint delta_delta_count = 0;
|
|
GstClockTime local_duration;
|
|
GstClockTime remote_duration;
|
|
|
|
start_idx = _get_window_start_index (stats, stats->window_size,
|
|
&local_duration, &remote_duration);
|
|
if (start_idx == -1) {
|
|
return;
|
|
}
|
|
|
|
/* remove the old packets */
|
|
if (start_idx > 0)
|
|
g_array_remove_range (stats->packets, 0, start_idx);
|
|
|
|
packets_sent = stats->packets->len - 1;
|
|
|
|
for (i = 0; i < packets_sent; i++) {
|
|
RTPTWCCPacket *pkt = &g_array_index (stats->packets, RTPTWCCPacket, i);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts)) {
|
|
bits_sent += pkt->size * 8;
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->remote_ts)) {
|
|
bits_recv += pkt->size * 8;
|
|
packets_recv++;
|
|
}
|
|
|
|
if (GST_CLOCK_STIME_IS_VALID (pkt->delta_delta)) {
|
|
delta_delta_sum += pkt->delta_delta;
|
|
delta_delta_count++;
|
|
}
|
|
}
|
|
|
|
packets_lost = packets_sent - packets_recv;
|
|
stats->packet_loss_pct = (packets_lost * 100) / (gfloat) packets_sent;
|
|
|
|
if (delta_delta_count) {
|
|
GstClockTimeDiff avg_delta_of_delta = delta_delta_sum / delta_delta_count;
|
|
if (GST_CLOCK_STIME_IS_VALID (stats->avg_delta_of_delta)) {
|
|
stats->avg_delta_of_delta_change =
|
|
(avg_delta_of_delta -
|
|
stats->avg_delta_of_delta) / (250 * GST_USECOND);
|
|
}
|
|
stats->avg_delta_of_delta = avg_delta_of_delta;
|
|
}
|
|
|
|
if (local_duration > 0)
|
|
stats->bitrate_sent =
|
|
gst_util_uint64_scale (bits_sent, GST_SECOND, local_duration);
|
|
if (remote_duration > 0)
|
|
stats->bitrate_recv =
|
|
gst_util_uint64_scale (bits_recv, GST_SECOND, remote_duration);
|
|
|
|
GST_DEBUG ("Got stats: bits_sent: %u, bits_recv: %u, packets_sent = %u, "
|
|
"packets_recv: %u, packetlost_pct = %f, sent_bitrate = %u, "
|
|
"recv_bitrate = %u, delta-delta-avg = %" GST_STIME_FORMAT ", "
|
|
"delta-delta-change: %f", bits_sent, bits_recv, stats->packets_sent,
|
|
packets_recv, stats->packet_loss_pct, stats->bitrate_sent,
|
|
stats->bitrate_recv, GST_STIME_ARGS (stats->avg_delta_of_delta),
|
|
stats->avg_delta_of_delta_change);
|
|
}
|
|
|
|
RTPTWCCStats *
|
|
rtp_twcc_stats_new (void)
|
|
{
|
|
RTPTWCCStats *stats = g_new0 (RTPTWCCStats, 1);
|
|
stats->packets = g_array_new (FALSE, FALSE, sizeof (RTPTWCCPacket));
|
|
stats->last_local_ts = GST_CLOCK_TIME_NONE;
|
|
stats->last_remote_ts = GST_CLOCK_TIME_NONE;
|
|
stats->avg_delta_of_delta = GST_CLOCK_STIME_NONE;
|
|
stats->window_size = 300 * GST_MSECOND; /* FIXME: could be configurable? */
|
|
return stats;
|
|
}
|
|
|
|
void
|
|
rtp_twcc_stats_free (RTPTWCCStats * stats)
|
|
{
|
|
g_array_unref (stats->packets);
|
|
g_free (stats);
|
|
}
|
|
|
|
static GstStructure *
|
|
rtp_twcc_stats_get_stats_structure (RTPTWCCStats * stats)
|
|
{
|
|
return gst_structure_new ("RTPTWCCStats",
|
|
"bitrate-sent", G_TYPE_UINT, stats->bitrate_sent,
|
|
"bitrate-recv", G_TYPE_UINT, stats->bitrate_recv,
|
|
"packets-sent", G_TYPE_UINT, stats->packets_sent,
|
|
"packets-recv", G_TYPE_UINT, stats->packets_recv,
|
|
"packet-loss-pct", G_TYPE_DOUBLE, stats->packet_loss_pct,
|
|
"avg-delta-of-delta", G_TYPE_INT64, stats->avg_delta_of_delta, NULL);
|
|
}
|
|
|
|
GstStructure *
|
|
rtp_twcc_stats_process_packets (RTPTWCCStats * stats, GArray * twcc_packets)
|
|
{
|
|
rtp_twcc_stats_calculate_stats (stats, twcc_packets);
|
|
g_array_append_vals (stats->packets, twcc_packets->data, twcc_packets->len);
|
|
rtp_twcc_stats_calculate_windowed_stats (stats);
|
|
return rtp_twcc_stats_get_stats_structure (stats);
|
|
}
|