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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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680 lines
20 KiB
C
680 lines
20 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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* Copyright (C) 2015 Kurento (http://kurento.org/)
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* @author: Miguel París <mparisdiaz@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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include "rtpstats.h"
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#include "rtptwcc.h"
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void
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gst_rtp_packet_rate_ctx_reset (RTPPacketRateCtx * ctx, gint32 clock_rate)
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{
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ctx->clock_rate = clock_rate;
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ctx->probed = FALSE;
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ctx->avg_packet_rate = -1;
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ctx->last_ts = -1;
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}
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guint32
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gst_rtp_packet_rate_ctx_update (RTPPacketRateCtx * ctx, guint16 seqnum,
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guint32 ts)
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{
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guint64 new_ts, diff_ts;
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gint diff_seqnum;
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gint32 new_packet_rate;
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gint32 base;
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if (ctx->clock_rate <= 0) {
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return ctx->avg_packet_rate;
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}
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new_ts = ctx->last_ts;
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gst_rtp_buffer_ext_timestamp (&new_ts, ts);
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if (!ctx->probed) {
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ctx->probed = TRUE;
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goto done_but_save;
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}
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diff_seqnum = gst_rtp_buffer_compare_seqnum (ctx->last_seqnum, seqnum);
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/* Ignore seqnums that are over 15,000 away from the latest one, it's close
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* to 2^14 but far enough to avoid any risk of computing error.
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*/
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if (diff_seqnum > 15000)
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goto done_but_save;
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/* Ignore any packet that is in the past, we're only interested in newer
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* packets to compute the packet rate.
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*/
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if (diff_seqnum <= 0 || new_ts <= ctx->last_ts)
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goto done;
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diff_ts = new_ts - ctx->last_ts;
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diff_ts = gst_util_uint64_scale_int (diff_ts, GST_SECOND, ctx->clock_rate);
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new_packet_rate = gst_util_uint64_scale (diff_seqnum, GST_SECOND, diff_ts);
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/* The goal is that higher packet rates "win".
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* If there's a sudden burst, the average will go up fast,
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* but it will go down again slowly.
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* This is useful for bursty cases, where a lot of packets are close
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* to each other and should allow a higher reorder/dropout there.
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* Round up the new average.
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* We do it on different rates depending on the packet rate, so it's not too
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* jumpy.
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*/
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if (ctx->avg_packet_rate > new_packet_rate)
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base = MAX (ctx->avg_packet_rate / 3, 8); /* about 333 ms */
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else
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base = MAX (ctx->avg_packet_rate / 15, 2); /* about 66 ms */
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diff_seqnum = MIN (diff_seqnum, base - 1);
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ctx->avg_packet_rate = (((base - diff_seqnum) * ctx->avg_packet_rate) +
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(new_packet_rate * diff_seqnum)) / base;
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done_but_save:
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ctx->last_seqnum = seqnum;
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ctx->last_ts = new_ts;
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done:
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return ctx->avg_packet_rate;
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}
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guint32
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gst_rtp_packet_rate_ctx_get (RTPPacketRateCtx * ctx)
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{
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return ctx->avg_packet_rate;
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}
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guint32
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gst_rtp_packet_rate_ctx_get_max_dropout (RTPPacketRateCtx * ctx, gint32 time_ms)
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{
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if (time_ms <= 0 || !ctx->probed || ctx->avg_packet_rate == -1) {
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return RTP_DEF_DROPOUT;
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}
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return MAX (RTP_MIN_DROPOUT, ctx->avg_packet_rate * time_ms / 1000);
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}
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guint32
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gst_rtp_packet_rate_ctx_get_max_misorder (RTPPacketRateCtx * ctx,
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gint32 time_ms)
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{
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if (time_ms <= 0 || !ctx->probed || ctx->avg_packet_rate == -1) {
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return RTP_DEF_MISORDER;
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}
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return MAX (RTP_MIN_MISORDER, ctx->avg_packet_rate * time_ms / 1000);
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}
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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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stats->nacks_dropped = 0;
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stats->nacks_sent = 0;
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stats->nacks_received = 0;
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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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* @profile: RTP profile of this session
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* @ptp: if this session is a point-to-point session
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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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GstRTPProfile profile, gboolean ptp, 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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if (profile == GST_RTP_PROFILE_AVPF || profile == GST_RTP_PROFILE_SAVPF) {
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/* RFC 4585 3.4d), 3.5.1 */
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if (first && !ptp)
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rtcp_min_time = 1.0;
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else
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rtcp_min_time = 0.0;
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} else {
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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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}
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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 <= 0.0001)
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return GST_CLOCK_TIME_NONE;
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avg_rtcp_size = 8.0 * 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 = 8.0 * 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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gboolean
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__g_socket_address_equal (GSocketAddress * a, GSocketAddress * b)
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{
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GInetSocketAddress *ia, *ib;
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GInetAddress *iaa, *iab;
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ia = G_INET_SOCKET_ADDRESS (a);
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ib = G_INET_SOCKET_ADDRESS (b);
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if (g_inet_socket_address_get_port (ia) !=
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g_inet_socket_address_get_port (ib))
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return FALSE;
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iaa = g_inet_socket_address_get_address (ia);
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iab = g_inet_socket_address_get_address (ib);
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return g_inet_address_equal (iaa, iab);
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}
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gchar *
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__g_socket_address_to_string (GSocketAddress * addr)
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{
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GInetSocketAddress *ia;
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gchar *ret, *tmp;
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ia = G_INET_SOCKET_ADDRESS (addr);
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tmp = g_inet_address_to_string (g_inet_socket_address_get_address (ia));
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ret = g_strdup_printf ("%s:%u", tmp, g_inet_socket_address_get_port (ia));
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g_free (tmp);
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return ret;
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}
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static void
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_append_structure_to_value_array (GValueArray * array, GstStructure * s)
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{
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GValue *val;
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g_value_array_append (array, NULL);
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val = g_value_array_get_nth (array, array->n_values - 1);
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g_value_init (val, GST_TYPE_STRUCTURE);
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g_value_take_boxed (val, s);
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}
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static void
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_structure_take_value_array (GstStructure * s,
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const gchar * field_name, GValueArray * array)
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{
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GValue value = G_VALUE_INIT;
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g_value_init (&value, G_TYPE_VALUE_ARRAY);
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g_value_take_boxed (&value, array);
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gst_structure_take_value (s, field_name, &value);
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g_value_unset (&value);
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}
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GstStructure *
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rtp_twcc_stats_get_packets_structure (GArray * twcc_packets)
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{
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GstStructure *ret = gst_structure_new_empty ("RTPTWCCPackets");
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GValueArray *array = g_value_array_new (0);
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guint i;
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for (i = 0; i < twcc_packets->len; i++) {
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RTPTWCCPacket *pkt = &g_array_index (twcc_packets, RTPTWCCPacket, i);
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GstStructure *pkt_s = gst_structure_new ("RTPTWCCPacket",
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"seqnum", G_TYPE_UINT, pkt->seqnum,
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"local-ts", G_TYPE_UINT64, pkt->local_ts,
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"remote-ts", G_TYPE_UINT64, pkt->remote_ts,
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"payload-type", G_TYPE_UCHAR, pkt->pt,
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"size", G_TYPE_UINT, pkt->size,
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"lost", G_TYPE_BOOLEAN, pkt->status == RTP_TWCC_PACKET_STATUS_NOT_RECV,
|
|
NULL);
|
|
_append_structure_to_value_array (array, pkt_s);
|
|
}
|
|
|
|
_structure_take_value_array (ret, "packets", array);
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
rtp_twcc_stats_calculate_stats (RTPTWCCStats * stats, GArray * twcc_packets)
|
|
{
|
|
guint packets_recv = 0;
|
|
guint i;
|
|
|
|
for (i = 0; i < twcc_packets->len; i++) {
|
|
RTPTWCCPacket *pkt = &g_array_index (twcc_packets, RTPTWCCPacket, i);
|
|
|
|
if (pkt->status != RTP_TWCC_PACKET_STATUS_NOT_RECV)
|
|
packets_recv++;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts) &&
|
|
GST_CLOCK_TIME_IS_VALID (stats->last_local_ts)) {
|
|
pkt->local_delta = GST_CLOCK_DIFF (stats->last_local_ts, pkt->local_ts);
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->remote_ts) &&
|
|
GST_CLOCK_TIME_IS_VALID (stats->last_remote_ts)) {
|
|
pkt->remote_delta =
|
|
GST_CLOCK_DIFF (stats->last_remote_ts, pkt->remote_ts);
|
|
}
|
|
|
|
if (GST_CLOCK_STIME_IS_VALID (pkt->local_delta) &&
|
|
GST_CLOCK_STIME_IS_VALID (pkt->remote_delta)) {
|
|
pkt->delta_delta = pkt->remote_delta - pkt->local_delta;
|
|
}
|
|
|
|
stats->last_local_ts = pkt->local_ts;
|
|
stats->last_remote_ts = pkt->remote_ts;
|
|
}
|
|
|
|
stats->packets_sent = twcc_packets->len;
|
|
stats->packets_recv = packets_recv;
|
|
}
|
|
|
|
static gint
|
|
_get_window_start_index (RTPTWCCStats * stats, GstClockTime duration,
|
|
GstClockTime * local_duration, GstClockTime * remote_duration)
|
|
{
|
|
RTPTWCCPacket *last = NULL;
|
|
guint i;
|
|
|
|
if (stats->packets->len < 2)
|
|
return -1;
|
|
|
|
for (i = 0; i < stats->packets->len; i++) {
|
|
guint start_index = stats->packets->len - 1 - i;
|
|
RTPTWCCPacket *pkt =
|
|
&g_array_index (stats->packets, RTPTWCCPacket, start_index);
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts)
|
|
&& GST_CLOCK_TIME_IS_VALID (pkt->remote_ts)) {
|
|
/* first find the last valid packet */
|
|
if (last == NULL) {
|
|
last = pkt;
|
|
} else {
|
|
/* and then get the duration in local ts */
|
|
GstClockTimeDiff ld = GST_CLOCK_DIFF (pkt->local_ts, last->local_ts);
|
|
if (ld >= duration) {
|
|
*local_duration = ld;
|
|
*remote_duration = GST_CLOCK_DIFF (pkt->remote_ts, last->remote_ts);
|
|
return start_index;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
return -1;
|
|
}
|
|
|
|
static void
|
|
rtp_twcc_stats_calculate_windowed_stats (RTPTWCCStats * stats)
|
|
{
|
|
guint i;
|
|
gint start_idx;
|
|
guint bits_sent = 0;
|
|
guint bits_recv = 0;
|
|
guint packets_sent = 0;
|
|
guint packets_recv = 0;
|
|
guint packets_lost;
|
|
GstClockTimeDiff delta_delta_sum = 0;
|
|
guint delta_delta_count = 0;
|
|
GstClockTime local_duration;
|
|
GstClockTime remote_duration;
|
|
|
|
start_idx = _get_window_start_index (stats, stats->window_size,
|
|
&local_duration, &remote_duration);
|
|
if (start_idx == -1) {
|
|
return;
|
|
}
|
|
|
|
/* remove the old packets */
|
|
if (start_idx > 0)
|
|
g_array_remove_range (stats->packets, 0, start_idx);
|
|
|
|
packets_sent = stats->packets->len - 1;
|
|
|
|
for (i = 0; i < packets_sent; i++) {
|
|
RTPTWCCPacket *pkt = &g_array_index (stats->packets, RTPTWCCPacket, i);
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->local_ts)) {
|
|
bits_sent += pkt->size * 8;
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (pkt->remote_ts)) {
|
|
bits_recv += pkt->size * 8;
|
|
packets_recv++;
|
|
}
|
|
|
|
if (GST_CLOCK_STIME_IS_VALID (pkt->delta_delta)) {
|
|
delta_delta_sum += pkt->delta_delta;
|
|
delta_delta_count++;
|
|
}
|
|
}
|
|
|
|
packets_lost = packets_sent - packets_recv;
|
|
stats->packet_loss_pct = (packets_lost * 100) / (gfloat) packets_sent;
|
|
|
|
if (delta_delta_count) {
|
|
GstClockTimeDiff avg_delta_of_delta = delta_delta_sum / delta_delta_count;
|
|
if (GST_CLOCK_STIME_IS_VALID (stats->avg_delta_of_delta)) {
|
|
stats->avg_delta_of_delta_change =
|
|
(avg_delta_of_delta -
|
|
stats->avg_delta_of_delta) / (250 * GST_USECOND);
|
|
}
|
|
stats->avg_delta_of_delta = avg_delta_of_delta;
|
|
}
|
|
|
|
if (local_duration > 0)
|
|
stats->bitrate_sent =
|
|
gst_util_uint64_scale (bits_sent, GST_SECOND, local_duration);
|
|
if (remote_duration > 0)
|
|
stats->bitrate_recv =
|
|
gst_util_uint64_scale (bits_recv, GST_SECOND, remote_duration);
|
|
|
|
GST_DEBUG ("Got stats: bits_sent: %u, bits_recv: %u, packets_sent = %u, "
|
|
"packets_recv: %u, packetlost_pct = %f, sent_bitrate = %u, "
|
|
"recv_bitrate = %u, delta-delta-avg = %" GST_STIME_FORMAT ", "
|
|
"delta-delta-change: %f", bits_sent, bits_recv, stats->packets_sent,
|
|
packets_recv, stats->packet_loss_pct, stats->bitrate_sent,
|
|
stats->bitrate_recv, GST_STIME_ARGS (stats->avg_delta_of_delta),
|
|
stats->avg_delta_of_delta_change);
|
|
}
|
|
|
|
RTPTWCCStats *
|
|
rtp_twcc_stats_new (void)
|
|
{
|
|
RTPTWCCStats *stats = g_new0 (RTPTWCCStats, 1);
|
|
stats->packets = g_array_new (FALSE, FALSE, sizeof (RTPTWCCPacket));
|
|
stats->last_local_ts = GST_CLOCK_TIME_NONE;
|
|
stats->last_remote_ts = GST_CLOCK_TIME_NONE;
|
|
stats->avg_delta_of_delta = GST_CLOCK_STIME_NONE;
|
|
stats->window_size = 300 * GST_MSECOND; /* FIXME: could be configurable? */
|
|
return stats;
|
|
}
|
|
|
|
void
|
|
rtp_twcc_stats_free (RTPTWCCStats * stats)
|
|
{
|
|
g_array_unref (stats->packets);
|
|
g_free (stats);
|
|
}
|
|
|
|
static GstStructure *
|
|
rtp_twcc_stats_get_stats_structure (RTPTWCCStats * stats)
|
|
{
|
|
return gst_structure_new ("RTPTWCCStats",
|
|
"bitrate-sent", G_TYPE_UINT, stats->bitrate_sent,
|
|
"bitrate-recv", G_TYPE_UINT, stats->bitrate_recv,
|
|
"packets-sent", G_TYPE_UINT, stats->packets_sent,
|
|
"packets-recv", G_TYPE_UINT, stats->packets_recv,
|
|
"packet-loss-pct", G_TYPE_DOUBLE, stats->packet_loss_pct,
|
|
"avg-delta-of-delta", G_TYPE_INT64, stats->avg_delta_of_delta, NULL);
|
|
}
|
|
|
|
GstStructure *
|
|
rtp_twcc_stats_process_packets (RTPTWCCStats * stats, GArray * twcc_packets)
|
|
{
|
|
rtp_twcc_stats_calculate_stats (stats, twcc_packets);
|
|
g_array_append_vals (stats->packets, twcc_packets->data, twcc_packets->len);
|
|
rtp_twcc_stats_calculate_windowed_stats (stats);
|
|
return rtp_twcc_stats_get_stats_structure (stats);
|
|
}
|