1 files changed, 0 insertions, 176 deletions
diff --git a/gst/rtpmanager/rtpstats.c b/gst/rtpmanager/rtpstats.c
deleted file mode 100644
index 640c3194..00000000
--- a/gst/rtpmanager/rtpstats.c
+++ /dev/null
@@ -1,176 +0,0 @@
-/* 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;
-}