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: current time according to the system clock
60 * @running_time: arrival time of a packet as buffer running_time
61 * @ntpnstime: arrival time of a packet NTP time in nanoseconds
62 * @have_address: if the @address field contains a valid address
63 * @address: address of the sender of the packet
64 * @bytes: bytes of the packet including lowlevel overhead
65 * @payload_len: bytes of the RTP payload
67 * Structure holding information about the arrival stats of a packet.
70 GstClockTime current_time;
71 GstClockTime running_time;
73 gboolean have_address;
74 GstNetAddress address;
81 * @packetsreceived: number of received packets in total
82 * @prevpacketsreceived: number of packets received in previous reporting
84 * @octetsreceived: number of payload bytes received
85 * @bytesreceived: number of total bytes received including headers and lower
86 * protocol level overhead
87 * @max_seqnr: highest sequence number received
88 * @transit: previous transit time used for calculating @jitter
89 * @jitter: current jitter
90 * @prev_rtptime: previous time when an RTP packet was received
91 * @prev_rtcptime: previous time when an RTCP packet was received
92 * @last_rtptime: time when last RTP packet received
93 * @last_rtcptime: time when last RTCP packet received
94 * @curr_rr: index of current @rr block
95 * @rr: previous and current receiver report block
96 * @curr_sr: index of current @sr block
97 * @sr: previous and current sender report block
99 * Stats about a source.
102 guint64 packets_received;
103 guint64 octets_received;
104 guint64 bytes_received;
106 guint32 prev_expected;
107 guint32 prev_received;
116 guint64 packets_sent;
119 /* when we received stuff */
120 GstClockTime prev_rtptime;
121 GstClockTime prev_rtcptime;
122 GstClockTime last_rtptime;
123 GstClockTime last_rtcptime;
125 /* sender and receiver reports */
127 RTPReceiverReport rr[2];
129 RTPSenderReport sr[2];
132 #define RTP_STATS_BANDWIDTH 64000
133 #define RTP_STATS_RTCP_FRACTION 0.05
135 * Minimum average time between RTCP packets from this site (in
136 * seconds). This time prevents the reports from `clumping' when
137 * sessions are small and the law of large numbers isn't helping
138 * to smooth out the traffic. It also keeps the report interval
139 * from becoming ridiculously small during transient outages like
140 * a network partition.
142 #define RTP_STATS_MIN_INTERVAL 5.0
144 * Fraction of the RTCP bandwidth to be shared among active
145 * senders. (This fraction was chosen so that in a typical
146 * session with one or two active senders, the computed report
147 * time would be roughly equal to the minimum report time so that
148 * we don't unnecessarily slow down receiver reports.) The
149 * receiver fraction must be 1 - the sender fraction.
151 #define RTP_STATS_SENDER_FRACTION (0.25)
152 #define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
155 * When receiving a BYE from a source, remove the source from the database
156 * after this timeout.
158 #define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
161 * The maximum number of missing packets we tollerate. These are packets with a
162 * sequence number bigger than the last seen packet.
164 #define RTP_MAX_DROPOUT 3000
166 * The maximum number of misordered packets we tollerate. These are packets with
167 * a sequence number smaller than the last seen packet.
169 #define RTP_MAX_MISORDER 100
174 * Stats kept for a session and used to produce RTCP packet timeouts.
178 guint rtcp_bandwidth;
179 gdouble sender_fraction;
180 gdouble receiver_fraction;
181 gdouble min_interval;
182 GstClockTime bye_timeout;
183 guint sender_sources;
184 guint active_sources;
185 guint avg_rtcp_packet_size;
189 void rtp_stats_init_defaults (RTPSessionStats *stats);
191 void rtp_stats_set_bandwidths (RTPSessionStats *stats,
196 GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, gboolean first);
197 GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);
198 GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats);
199 gint64 rtp_stats_get_packets_lost (const RTPSourceStats *stats);
201 void rtp_stats_set_min_interval (RTPSessionStats *stats,
202 gdouble min_interval);
203 #endif /* __RTP_STATS_H__ */