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/* 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 "rtpstats.h"
/**
* rtp_stats_init_defaults:
* @stats: an #RTPSessionStats struct
*
* Initialize @stats with its default values.
*/
void
rtp_stats_init_defaults (RTPSessionStats * stats)
{
stats->bandwidth = RTP_STATS_BANDWIDTH;
stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
stats->min_interval = RTP_STATS_MIN_INTERVAL;
stats->bye_timeout = RTP_STATS_BYE_TIMEOUT;
}
/**
* rtp_stats_calculate_rtcp_interval:
* @stats: an #RTPSessionStats struct
* @sender: if we are a sender
* @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,
gboolean first)
{
gdouble members, senders, n;
gdouble avg_rtcp_size, rtcp_bw;
gdouble interval;
gdouble rtcp_min_time;
/* 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 * RTP_STATS_SENDER_FRACTION) {
if (we_send) {
rtcp_bw *= RTP_STATS_SENDER_FRACTION;
n = senders;
} else {
rtcp_bw *= RTP_STATS_RECEIVER_FRACTION;
n -= senders;
}
}
avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
/*
* 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 * 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 * RTP_STATS_RECEIVER_FRACTION;
avg_rtcp_size = stats->avg_rtcp_packet_size / 16.0;
/*
* 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;
}
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