2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
20 #ifndef __RTP_STATS_H__
21 #define __RTP_STATS_H__
24 #include <gst/netbuffer/gstnetbuffer.h>
29 * A sender report structure.
43 * A receiver report structure.
47 guint32 ssrc; /* who the report is from */
50 guint32 exthighestseq;
59 * @current_time: arrival time of a packet according to the system clock
60 * @running_time: arrival time of a packet as buffer running_time
61 * @have_address: if the @address field contains a valid address
62 * @address: address of the sender of the packet
63 * @bytes: bytes of the packet including lowlevel overhead
64 * @payload_len: bytes of the RTP payload
66 * Structure holding information about the arrival stats of a packet.
69 GstClockTime current_time;
70 GstClockTime running_time;
71 gboolean have_address;
72 GstNetAddress address;
79 * @packetsreceived: number of received packets in total
80 * @prevpacketsreceived: number of packets received in previous reporting
82 * @octetsreceived: number of payload bytes received
83 * @bytesreceived: number of total bytes received including headers and lower
84 * protocol level overhead
85 * @max_seqnr: highest sequence number received
86 * @transit: previous transit time used for calculating @jitter
87 * @jitter: current jitter
88 * @prev_rtptime: previous time when an RTP packet was received
89 * @prev_rtcptime: previous time when an RTCP packet was received
90 * @last_rtptime: time when last RTP packet received
91 * @last_rtcptime: time when last RTCP packet received
92 * @curr_rr: index of current @rr block
93 * @rr: previous and current receiver report block
94 * @curr_sr: index of current @sr block
95 * @sr: previous and current sender report block
97 * Stats about a source.
100 guint64 packets_received;
101 guint64 octets_received;
102 guint64 bytes_received;
104 guint32 prev_expected;
105 guint32 prev_received;
114 guint64 packets_sent;
117 /* when we received stuff */
118 GstClockTime prev_rtptime;
119 GstClockTime prev_rtcptime;
120 GstClockTime last_rtptime;
121 GstClockTime last_rtcptime;
123 /* sender and receiver reports */
125 RTPReceiverReport rr[2];
127 RTPSenderReport sr[2];
130 #define RTP_STATS_BANDWIDTH 64000
131 #define RTP_STATS_RTCP_BANDWIDTH 3200
133 * Minimum average time between RTCP packets from this site (in
134 * seconds). This time prevents the reports from `clumping' when
135 * sessions are small and the law of large numbers isn't helping
136 * to smooth out the traffic. It also keeps the report interval
137 * from becoming ridiculously small during transient outages like
138 * a network partition.
140 #define RTP_STATS_MIN_INTERVAL 5.0
142 * Fraction of the RTCP bandwidth to be shared among active
143 * senders. (This fraction was chosen so that in a typical
144 * session with one or two active senders, the computed report
145 * time would be roughly equal to the minimum report time so that
146 * we don't unnecessarily slow down receiver reports.) The
147 * receiver fraction must be 1 - the sender fraction.
149 #define RTP_STATS_SENDER_FRACTION (0.25)
150 #define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
153 * When receiving a BYE from a source, remove the source from the database
154 * after this timeout.
156 #define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
159 * The maximum number of missing packets we tollerate. These are packets with a
160 * sequence number bigger than the last seen packet.
162 #define RTP_MAX_DROPOUT 3000
164 * The maximum number of misordered packets we tollerate. These are packets with
165 * a sequence number smaller than the last seen packet.
167 #define RTP_MAX_MISORDER 100
172 * Stats kept for a session and used to produce RTCP packet timeouts.
176 guint rtcp_bandwidth;
177 gdouble sender_fraction;
178 gdouble receiver_fraction;
179 gdouble min_interval;
180 GstClockTime bye_timeout;
181 guint sender_sources;
182 guint active_sources;
183 guint avg_rtcp_packet_size;
187 void rtp_stats_init_defaults (RTPSessionStats *stats);
189 void rtp_stats_set_bandwidths (RTPSessionStats *stats, guint rtp_bw, guint rtcp_bw,
192 GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, gboolean first);
193 GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);
194 GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats);
196 #endif /* __RTP_STATS_H__ */