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9d800cad43
According to RFC3550 lower-level headers should be considered for
bandwidth calculation.
See https://tools.ietf.org/html/rfc3550#section-6.2 paragraph 4:
Bandwidth calculations for control and data traffic include
lower-layer transport and network protocols (e.g., UDP and IP) since
that is what the resource reservation system would need to know.
Fix the source data to accommodate that.
Assume UDPv4 over IP for now, this is a simplification but it's good
enough for now.
While at it define a constant and use that instead of a magic number.
NOTE: this change basically reverts the logic of commit 529f443a6
(rtpsource: use payload size to estimate bitrate, 2010-03-02)
267 lines
8.6 KiB
C
267 lines
8.6 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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#ifndef __RTP_STATS_H__
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#define __RTP_STATS_H__
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#include <gst/gst.h>
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#include <gst/net/gstnetaddressmeta.h>
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#include <gst/rtp/rtp.h>
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#include <gio/gio.h>
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/* UDP/IP is assumed for bandwidth calculation */
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#define UDP_IP_HEADER_OVERHEAD 28
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/**
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* RTPSenderReport:
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*
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* A sender report structure.
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*/
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typedef struct {
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gboolean is_valid;
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guint64 ntptime;
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guint32 rtptime;
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guint32 packet_count;
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guint32 octet_count;
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GstClockTime time;
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} RTPSenderReport;
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/**
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* RTPReceiverReport:
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*
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* A receiver report structure.
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*/
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typedef struct {
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gboolean is_valid;
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guint32 ssrc; /* who the report is from */
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guint8 fractionlost;
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guint32 packetslost;
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guint32 exthighestseq;
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guint32 jitter;
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guint32 lsr;
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guint32 dlsr;
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guint32 round_trip;
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} RTPReceiverReport;
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/**
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* RTPPacketInfo:
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* @send: if this is a packet for sending
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* @rtp: if this info is about an RTP packet
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* @is_list: if this is a bufferlist
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* @data: a #GstBuffer or #GstBufferList
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* @address: address of the sender of the packet
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* @current_time: current time according to the system clock
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* @running_time: time of a packet as buffer running_time
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* @ntpnstime: time of a packet NTP time in nanoseconds
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* @header_len: number of overhead bytes per 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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* @seqnum: the seqnum of the packet
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* @pt: the payload type of the packet
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* @rtptime: the RTP time of the packet
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*
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* Structure holding information about the packet.
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*/
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typedef struct {
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gboolean send;
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gboolean rtp;
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gboolean is_list;
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gpointer data;
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GSocketAddress *address;
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GstClockTime current_time;
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GstClockTime running_time;
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guint64 ntpnstime;
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guint header_len;
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guint bytes;
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guint packets;
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guint payload_len;
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guint32 ssrc;
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guint16 seqnum;
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guint8 pt;
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guint32 rtptime;
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guint32 csrc_count;
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guint32 csrcs[16];
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} RTPPacketInfo;
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/**
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* RTPSourceStats:
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* @packets_received: number of received packets in total
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* @prev_received: number of packets received in previous reporting
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* interval
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* @octets_received: number of payload bytes received
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* @bytes_received: 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 (in clock rate units scaled by 16 for precision)
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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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*
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* Stats about a source.
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*/
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typedef struct {
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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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guint16 max_seq;
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guint64 cycles;
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guint32 base_seq;
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guint32 bad_seq;
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guint32 transit;
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guint32 jitter;
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guint64 packets_sent;
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guint64 octets_sent;
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guint sent_pli_count;
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guint recv_pli_count;
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guint sent_fir_count;
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guint recv_fir_count;
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guint sent_nack_count;
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guint recv_nack_count;
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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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gint curr_rr;
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RTPReceiverReport rr[2];
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gint curr_sr;
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RTPSenderReport sr[2];
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} RTPSourceStats;
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#define RTP_STATS_BANDWIDTH 64000
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#define RTP_STATS_RTCP_FRACTION 0.05
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/*
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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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*/
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#define RTP_STATS_MIN_INTERVAL 5.0
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/*
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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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*/
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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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/*
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* When receiving a BYE from a source, remove the source from the database
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* after this timeout.
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*/
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#define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
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/*
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* The default and minimum values of the maximum number of missing packets we tolerate.
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* These are packets with asequence number bigger than the last seen packet.
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*/
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#define RTP_DEF_DROPOUT 3000
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#define RTP_MIN_DROPOUT 30
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/*
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* The default and minimum values of the maximum number of misordered packets we tolerate.
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* These are packets with a sequence number smaller than the last seen packet.
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*/
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#define RTP_DEF_MISORDER 100
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#define RTP_MIN_MISORDER 10
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/**
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* RTPPacketRateCtx:
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*
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* Context to calculate the pseudo-average packet rate.
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*/
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typedef struct {
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gboolean probed;
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gint32 clock_rate;
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guint16 last_seqnum;
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guint64 last_ts;
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guint32 avg_packet_rate;
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} RTPPacketRateCtx;
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void gst_rtp_packet_rate_ctx_reset (RTPPacketRateCtx * ctx, gint32 clock_rate);
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guint32 gst_rtp_packet_rate_ctx_update (RTPPacketRateCtx *ctx, guint16 seqnum, guint32 ts);
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guint32 gst_rtp_packet_rate_ctx_get (RTPPacketRateCtx *ctx);
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guint32 gst_rtp_packet_rate_ctx_get_max_dropout (RTPPacketRateCtx *ctx, gint32 time_ms);
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guint32 gst_rtp_packet_rate_ctx_get_max_misorder (RTPPacketRateCtx *ctx, gint32 time_ms);
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/**
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* RTPSessionStats:
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*
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* Stats kept for a session and used to produce RTCP packet timeouts.
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*/
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typedef struct {
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guint bandwidth;
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guint rtcp_bandwidth;
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gdouble sender_fraction;
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gdouble receiver_fraction;
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gdouble min_interval;
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GstClockTime bye_timeout;
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guint internal_sources;
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guint sender_sources;
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guint internal_sender_sources;
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guint active_sources;
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guint avg_rtcp_packet_size;
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guint bye_members;
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guint nacks_dropped;
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guint nacks_sent;
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guint nacks_received;
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} RTPSessionStats;
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void rtp_stats_init_defaults (RTPSessionStats *stats);
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void rtp_stats_set_bandwidths (RTPSessionStats *stats,
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guint rtp_bw,
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gdouble rtcp_bw,
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guint rs, guint rr);
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GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, GstRTPProfile profile, gboolean ptp, 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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gint64 rtp_stats_get_packets_lost (const RTPSourceStats *stats);
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void rtp_stats_set_min_interval (RTPSessionStats *stats,
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gdouble min_interval);
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gboolean __g_socket_address_equal (GSocketAddress *a, GSocketAddress *b);
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gchar * __g_socket_address_to_string (GSocketAddress * addr);
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#endif /* __RTP_STATS_H__ */
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