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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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99723bc1c1
This is useful in the Bundle case because there may be multiple local and remote SSRCs in the same session. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/776>
301 lines
9.5 KiB
C
301 lines
9.5 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; /* which source is the report about */
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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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* @marker: the marker bit
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*
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* @tw_seqnum_ext_id: the extension-header ID for transport-wide seqnums
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* @tw_seqnum: the transport-wide seqnum 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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gboolean marker;
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guint32 csrc_count;
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guint32 csrcs[16];
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GBytes *header_ext;
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guint16 header_ext_bit_pattern;
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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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/**
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* RTPTWCCStats:
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*
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* Stats kept for a session and used to produce TWCC stats.
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*/
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typedef struct {
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GArray *packets;
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GstClockTime window_size;
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GstClockTime last_local_ts;
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GstClockTime last_remote_ts;
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guint bitrate_sent;
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guint bitrate_recv;
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guint packets_sent;
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guint packets_recv;
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gfloat packet_loss_pct;
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GstClockTimeDiff avg_delta_of_delta;
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gfloat avg_delta_of_delta_change;
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} RTPTWCCStats;
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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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RTPTWCCStats * rtp_twcc_stats_new (void);
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void rtp_twcc_stats_free (RTPTWCCStats * stats);
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GstStructure * rtp_twcc_stats_process_packets (RTPTWCCStats * stats,
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GArray * twcc_packets);
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GstStructure * rtp_twcc_stats_get_packets_structure (GArray * twcc_packets);
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#endif /* __RTP_STATS_H__ */
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