gstreamer/gst/rtpmanager/rtpstats.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 "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;
}