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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.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;
}
/**
* rtp_stats_calculate_rtcp_interval:
* @stats: an #RTPSessionStats struct
*
* Calculate the RTCP interval. The result of this function is the amount of
* time to wait (in seconds) before sender a new RTCP message.
*
* Returns: the RTCP interval.
*/
gdouble
rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
{
gdouble active, senders, receivers, sfraction;
gboolean avg_rtcp;
gdouble interval;
active = stats->active_sources;
/* Try to avoid division by zero */
if (stats->active_sources == 0)
active += 1.0;
senders = (gdouble) stats->sender_sources;
receivers = (gdouble) (active - senders);
avg_rtcp = (gdouble) stats->avg_rtcp_packet_size;
sfraction = senders / active;
GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
senders, receivers, avg_rtcp, sfraction);
if (sfraction <= stats->sender_fraction) {
if (sender) {
interval =
(avg_rtcp * senders) / (stats->sender_fraction *
stats->rtcp_bandwidth);
} else {
interval =
(avg_rtcp * receivers) / ((1.0 -
stats->sender_fraction) * stats->rtcp_bandwidth);
}
} else {
interval = (avg_rtcp * active) / stats->rtcp_bandwidth;
}
if (interval < stats->min_interval)
interval = stats->min_interval;
if (!stats->sent_rtcp)
interval /= 2.0;
return interval;
}
/**
* rtp_stats_calculate_rtcp_interval:
* @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.
*/
gdouble
rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, gdouble interval)
{
/* see RFC 3550 p 30
* To compensate for "unconditional reconsideration" converging to a
* value below the intended average.
*/
#define COMPENSATION (2.71828 - 1.5);
return (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
}
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