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294 lines
8.7 KiB
C
294 lines
8.7 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#include "rtpstats.h"
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/**
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* rtp_stats_init_defaults:
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* @stats: an #RTPSessionStats struct
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*
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* Initialize @stats with its default values.
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*/
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void
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rtp_stats_init_defaults (RTPSessionStats * stats)
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{
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rtp_stats_set_bandwidths (stats, -1, -1, -1, -1);
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stats->min_interval = RTP_STATS_MIN_INTERVAL;
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stats->bye_timeout = RTP_STATS_BYE_TIMEOUT;
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}
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/**
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* rtp_stats_set_bandwidths:
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* @stats: an #RTPSessionStats struct
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* @rtp_bw: RTP bandwidth
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* @rtcp_bw: RTCP bandwidth
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* @rs: sender RTCP bandwidth
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* @rr: receiver RTCP bandwidth
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*
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* Configure the bandwidth parameters in the stats. When an input variable is
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* set to -1, it will be calculated from the other input variables and from the
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* defaults.
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*/
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void
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rtp_stats_set_bandwidths (RTPSessionStats * stats, guint rtp_bw,
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gdouble rtcp_bw, guint rs, guint rr)
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{
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GST_DEBUG ("recalc bandwidths: RTP %u, RTCP %f, RS %u, RR %u", rtp_bw,
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rtcp_bw, rs, rr);
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/* when given, sender and receive bandwidth add up to the total
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* rtcp bandwidth */
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if (rs != -1 && rr != -1)
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rtcp_bw = rs + rr;
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/* If rtcp_bw is between 0 and 1, it is a fraction of rtp_bw */
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if (rtcp_bw > 0.0 && rtcp_bw < 1.0) {
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if (rtp_bw > 0.0)
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rtcp_bw = rtp_bw * rtcp_bw;
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else
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rtcp_bw = -1.0;
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}
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/* RTCP is 5% of the RTP bandwidth */
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if (rtp_bw == -1 && rtcp_bw > 1.0)
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rtp_bw = rtcp_bw * 20;
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else if (rtp_bw != -1 && rtcp_bw < 0.0)
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rtcp_bw = rtp_bw / 20;
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else if (rtp_bw == -1 && rtcp_bw < 0.0) {
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/* nothing given, take defaults */
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rtp_bw = RTP_STATS_BANDWIDTH;
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rtcp_bw = rtp_bw * RTP_STATS_RTCP_FRACTION;
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}
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stats->bandwidth = rtp_bw;
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stats->rtcp_bandwidth = rtcp_bw;
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/* now figure out the fractions */
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if (rs == -1) {
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/* rs unknown */
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if (rr == -1) {
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/* both not given, use defaults */
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rs = stats->rtcp_bandwidth * RTP_STATS_SENDER_FRACTION;
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rr = stats->rtcp_bandwidth * RTP_STATS_RECEIVER_FRACTION;
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} else {
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/* rr known, calculate rs */
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if (stats->rtcp_bandwidth > rr)
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rs = stats->rtcp_bandwidth - rr;
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else
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rs = 0;
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}
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} else if (rr == -1) {
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/* rs known, calculate rr */
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if (stats->rtcp_bandwidth > rs)
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rr = stats->rtcp_bandwidth - rs;
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else
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rr = 0;
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}
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if (stats->rtcp_bandwidth > 0) {
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stats->sender_fraction = ((gdouble) rs) / ((gdouble) stats->rtcp_bandwidth);
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stats->receiver_fraction = 1.0 - stats->sender_fraction;
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} else {
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/* no RTCP bandwidth, set dummy values */
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stats->sender_fraction = 0.0;
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stats->receiver_fraction = 0.0;
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}
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GST_DEBUG ("bandwidths: RTP %u, RTCP %u, RS %f, RR %f", stats->bandwidth,
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stats->rtcp_bandwidth, stats->sender_fraction, stats->receiver_fraction);
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}
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/**
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* rtp_stats_calculate_rtcp_interval:
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* @stats: an #RTPSessionStats struct
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* @sender: if we are a sender
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* @first: if this is the first time
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*
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* Calculate the RTCP interval. The result of this function is the amount of
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* time to wait (in nanoseconds) before sending a new RTCP message.
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*
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* Returns: the RTCP interval.
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*/
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GstClockTime
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rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean we_send,
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gboolean first)
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{
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gdouble members, senders, n;
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gdouble avg_rtcp_size, rtcp_bw;
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gdouble interval;
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gdouble rtcp_min_time;
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/* Very first call at application start-up uses half the min
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* delay for quicker notification while still allowing some time
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* before reporting for randomization and to learn about other
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* sources so the report interval will converge to the correct
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* interval more quickly.
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*/
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rtcp_min_time = stats->min_interval;
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if (first)
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rtcp_min_time /= 2.0;
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/* Dedicate a fraction of the RTCP bandwidth to senders unless
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* the number of senders is large enough that their share is
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* more than that fraction.
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*/
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n = members = stats->active_sources;
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senders = (gdouble) stats->sender_sources;
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rtcp_bw = stats->rtcp_bandwidth;
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if (senders <= members * stats->sender_fraction) {
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if (we_send) {
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rtcp_bw *= stats->sender_fraction;
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n = senders;
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} else {
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rtcp_bw *= stats->receiver_fraction;
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n -= senders;
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}
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}
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/* no bandwidth for RTCP, return NONE to signal that we don't want to send
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* RTCP packets */
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if (rtcp_bw <= 0.00001)
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return GST_CLOCK_TIME_NONE;
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avg_rtcp_size = stats->avg_rtcp_packet_size;
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/*
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* The effective number of sites times the average packet size is
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* the total number of octets sent when each site sends a report.
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* Dividing this by the effective bandwidth gives the time
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* interval over which those packets must be sent in order to
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* meet the bandwidth target, with a minimum enforced. In that
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* time interval we send one report so this time is also our
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* average time between reports.
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*/
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GST_DEBUG ("avg size %f, n %f, rtcp_bw %f", avg_rtcp_size, n, rtcp_bw);
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interval = avg_rtcp_size * n / rtcp_bw;
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if (interval < rtcp_min_time)
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interval = rtcp_min_time;
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return interval * GST_SECOND;
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}
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/**
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* rtp_stats_add_rtcp_jitter:
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* @stats: an #RTPSessionStats struct
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* @interval: an RTCP interval
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*
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* Apply a random jitter to the @interval. @interval is typically obtained with
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* rtp_stats_calculate_rtcp_interval().
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*
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* Returns: the new RTCP interval.
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*/
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GstClockTime
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rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, GstClockTime interval)
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{
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gdouble temp;
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/* see RFC 3550 p 30
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* To compensate for "unconditional reconsideration" converging to a
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* value below the intended average.
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*/
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#define COMPENSATION (2.71828 - 1.5);
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temp = (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
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return (GstClockTime) temp;
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}
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/**
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* rtp_stats_calculate_bye_interval:
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* @stats: an #RTPSessionStats struct
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*
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* Calculate the BYE interval. The result of this function is the amount of
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* time to wait (in nanoseconds) before sending a BYE message.
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*
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* Returns: the BYE interval.
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*/
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GstClockTime
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rtp_stats_calculate_bye_interval (RTPSessionStats * stats)
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{
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gdouble members;
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gdouble avg_rtcp_size, rtcp_bw;
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gdouble interval;
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gdouble rtcp_min_time;
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/* no interval when we have less than 50 members */
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if (stats->active_sources < 50)
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return 0;
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rtcp_min_time = (stats->min_interval) / 2.0;
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/* Dedicate a fraction of the RTCP bandwidth to senders unless
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* the number of senders is large enough that their share is
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* more than that fraction.
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*/
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members = stats->bye_members;
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rtcp_bw = stats->rtcp_bandwidth * stats->receiver_fraction;
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/* no bandwidth for RTCP, return NONE to signal that we don't want to send
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* RTCP packets */
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if (rtcp_bw <= 0.0001)
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return GST_CLOCK_TIME_NONE;
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avg_rtcp_size = stats->avg_rtcp_packet_size;
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/*
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* The effective number of sites times the average packet size is
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* the total number of octets sent when each site sends a report.
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* Dividing this by the effective bandwidth gives the time
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* interval over which those packets must be sent in order to
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* meet the bandwidth target, with a minimum enforced. In that
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* time interval we send one report so this time is also our
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* average time between reports.
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*/
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interval = avg_rtcp_size * members / rtcp_bw;
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if (interval < rtcp_min_time)
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interval = rtcp_min_time;
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return interval * GST_SECOND;
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}
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/**
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* rtp_stats_get_packets_lost:
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* @stats: an #RTPSourceStats struct
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*
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* Calculate the total number of RTP packets lost since beginning of
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* reception. Packets that arrive late are not considered lost, and
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* duplicates are not taken into account. Hence, the loss may be negative
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* if there are duplicates.
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*
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* Returns: total RTP packets lost.
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*/
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gint64
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rtp_stats_get_packets_lost (const RTPSourceStats * stats)
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{
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gint64 lost;
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guint64 extended_max, expected;
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extended_max = stats->cycles + stats->max_seq;
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expected = extended_max - stats->base_seq + 1;
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lost = expected - stats->packets_received;
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return lost;
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}
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void
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rtp_stats_set_min_interval (RTPSessionStats * stats, gdouble min_interval)
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{
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stats->min_interval = min_interval;
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}
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