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* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
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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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* 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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* 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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#ifndef __RTP_STATS_H__
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#define __RTP_STATS_H__
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#include <gst/netbuffer/gstnetbuffer.h>
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* A sender report structure.
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* A receiver report structure.
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guint32 ssrc; /* who the report is from */
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guint32 exthighestseq;
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* @time: arrival time of a packet
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* @address: address of the sender of the packet
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* @bytes: bytes of the packet including lowlevel overhead
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* @payload_len: bytes of the RTP payload
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* Structure holding information about the arrival stats of a packet.
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gboolean have_address;
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GstNetAddress address;
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* @packetsreceived: number of received packets in total
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* @prevpacketsreceived: number of packets received in previous reporting
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* @octetsreceived: number of payload bytes received
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* @bytesreceived: number of total bytes received including headers and lower
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* protocol level overhead
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* @max_seqnr: highest sequence number received
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* @transit: previous transit time used for calculating @jitter
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* @jitter: current jitter
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* @prev_rtptime: previous time when an RTP packet was received
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* @prev_rtcptime: previous time when an RTCP packet was received
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* @last_rtptime: time when last RTP packet received
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* @last_rtcptime: time when last RTCP packet received
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* @curr_rr: index of current @rr block
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* @rr: previous and current receiver report block
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* @curr_sr: index of current @sr block
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* @sr: previous and current sender report block
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* Stats about a source.
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guint64 packets_received;
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guint64 octets_received;
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guint64 bytes_received;
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guint32 prev_expected;
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guint32 prev_received;
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guint64 packets_sent;
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/* when we received stuff */
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GstClockTime prev_rtptime;
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GstClockTime prev_rtcptime;
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GstClockTime last_rtptime;
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GstClockTime last_rtcptime;
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/* sender and receiver reports */
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RTPReceiverReport rr[2];
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RTPSenderReport sr[2];
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#define RTP_STATS_BANDWIDTH 64000.0
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#define RTP_STATS_RTCP_BANDWIDTH 3000.0
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* Minimum average time between RTCP packets from this site (in
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* seconds). This time prevents the reports from `clumping' when
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* sessions are small and the law of large numbers isn't helping
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* to smooth out the traffic. It also keeps the report interval
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* from becoming ridiculously small during transient outages like
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* a network partition.
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#define RTP_STATS_MIN_INTERVAL 5.0
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* Fraction of the RTCP bandwidth to be shared among active
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* senders. (This fraction was chosen so that in a typical
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* session with one or two active senders, the computed report
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* time would be roughly equal to the minimum report time so that
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* we don't unnecessarily slow down receiver reports.) The
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* receiver fraction must be 1 - the sender fraction.
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#define RTP_STATS_SENDER_FRACTION (0.25)
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#define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
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* When receiving a BYE from a source, remove the source fomr the database
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* after this timeout.
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#define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
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* Stats kept for a session and used to produce RTCP packet timeouts.
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gdouble sender_fraction;
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gdouble receiver_fraction;
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gdouble rtcp_bandwidth;
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gdouble min_interval;
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GstClockTime bye_timeout;
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guint sender_sources;
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guint active_sources;
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guint avg_rtcp_packet_size;
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void rtp_stats_init_defaults (RTPSessionStats *stats);
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GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, gboolean first);
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GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);
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GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats);
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#endif /* __RTP_STATS_H__ */
added
removed
gst/rtpmanager/rtpstats.h
