gstreamer/gst/rtpmanager/rtpstats.h

/* 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.
 */

#ifndef __RTP_STATS_H__
#define __RTP_STATS_H__

#include <gst/gst.h>
#include <gst/netbuffer/gstnetbuffer.h>

/**
 * RTPSenderReport:
 *
 * A sender report structure.
 */
typedef struct {
  gboolean is_valid;
  guint64 ntptime;
  guint32 rtptime;
  guint32 packet_count;
  guint32 octet_count;
  GstClockTime time;
} RTPSenderReport;

/**
 * RTPReceiverReport:
 *
 * A receiver report structure.
 */
typedef struct {
  gboolean is_valid;
  guint32 ssrc; /* who the report is from */
  guint8  fractionlost;
  guint32 packetslost;
  guint32 exthighestseq;
  guint32 jitter;
  guint32 lsr;
  guint32 dlsr;
} RTPReceiverReport;

/**
 * RTPArrivalStats:
 * @time: arrival time of a packet
 * @address: address of the sender of the packet
 * @bytes: bytes of the packet including lowlevel overhead
 * @payload_len: bytes of the RTP payload
 *
 * Structure holding information about the arrival stats of a packet.
 */
typedef struct {
  GstClockTime  time;
  gboolean      have_address;
  GstNetAddress address;
  guint         bytes;
  guint         payload_len;
} RTPArrivalStats;

/**
 * RTPSourceStats:
 * @packetsreceived: number of received packets in total
 * @prevpacketsreceived: number of packets received in previous reporting
 *                       interval
 * @octetsreceived: number of payload bytes received
 * @bytesreceived: number of total bytes received including headers and lower
 *                 protocol level overhead
 * @max_seqnr: highest sequence number received
 * @transit: previous transit time used for calculating @jitter
 * @jitter: current jitter
 * @prev_rtptime: previous time when an RTP packet was received
 * @prev_rtcptime: previous time when an RTCP packet was received
 * @last_rtptime: time when last RTP packet received
 * @last_rtcptime: time when last RTCP packet received
 * @curr_rr: index of current @rr block
 * @rr: previous and current receiver report block
 * @curr_sr: index of current @sr block
 * @sr: previous and current sender report block
 *
 * Stats about a source.
 */
typedef struct {
  guint64      packets_received;
  guint64      octets_received;
  guint64      bytes_received;

  guint32      prev_expected;
  guint32      prev_received;

  guint16      max_seq;
  guint64      cycles;
  guint32      base_seq;
  guint32      bad_seq;
  guint32      transit;
  guint32      jitter;

  guint64      packets_sent;
  guint64      octets_sent;

  /* when we received stuff */
  GstClockTime prev_rtptime;
  GstClockTime prev_rtcptime;
  GstClockTime last_rtptime;
  GstClockTime last_rtcptime;

  /* sender and receiver reports */
  gint              curr_rr;
  RTPReceiverReport rr[2];
  gint              curr_sr;
  RTPSenderReport   sr[2];
} RTPSourceStats;

#define RTP_STATS_BANDWIDTH           64000.0
#define RTP_STATS_RTCP_BANDWIDTH      3000.0
/*
 * Minimum average time between RTCP packets from this site (in
 * seconds).  This time prevents the reports from `clumping' when
 * sessions are small and the law of large numbers isn't helping
 * to smooth out the traffic.  It also keeps the report interval
 * from becoming ridiculously small during transient outages like
 * a network partition.
 */
#define RTP_STATS_MIN_INTERVAL      5.0
/*
 * Fraction of the RTCP bandwidth to be shared among active
 * senders.  (This fraction was chosen so that in a typical
 * session with one or two active senders, the computed report
 * time would be roughly equal to the minimum report time so that
 * we don't unnecessarily slow down receiver reports.) The
 * receiver fraction must be 1 - the sender fraction.
 */
#define RTP_STATS_SENDER_FRACTION       (0.25)
#define RTP_STATS_RECEIVER_FRACTION     (1.0 - RTP_STATS_SENDER_FRACTION)

/*
 * When receiving a BYE from a source, remove the source fomr the database
 * after this timeout.
 */
#define RTP_STATS_BYE_TIMEOUT           (2 * GST_SECOND)

/**
 * RTPSessionStats:
 *
 * Stats kept for a session and used to produce RTCP packet timeouts.
 */
typedef struct {
  gdouble       bandwidth;
  gdouble       sender_fraction;
  gdouble       receiver_fraction;
  gdouble       rtcp_bandwidth;
  gdouble       min_interval;
  GstClockTime  bye_timeout;
  guint         sender_sources;
  guint         active_sources;
  guint         avg_rtcp_packet_size;
  guint         bye_members;
} RTPSessionStats;

void           rtp_stats_init_defaults               (RTPSessionStats *stats);

GstClockTime   rtp_stats_calculate_rtcp_interval    (RTPSessionStats *stats, gboolean sender, gboolean first);
GstClockTime   rtp_stats_add_rtcp_jitter            (RTPSessionStats *stats, GstClockTime interval);
GstClockTime   rtp_stats_calculate_bye_interval     (RTPSessionStats *stats);

#endif /* __RTP_STATS_H__ */