blob: b0fbddb48599ec89f8f1569b2e1e488cf12c31f6 [file] [log] [blame]
/* GStreamer
* Copyright (C) <2007> Wim Taymans <>
* Copyright (C) 2015 Kurento (
* @author: Miguel ParĂ­s <>
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* Library General Public License for more details.
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
#ifndef __RTP_STATS_H__
#define __RTP_STATS_H__
#include <gst/gst.h>
#include <gst/net/gstnetaddressmeta.h>
#include <gst/rtp/rtp.h>
#include <gio/gio.h>
* RTPSenderReport:
* A sender report structure.
typedef struct {
gboolean is_valid;
guint64 ntptime;
guint32 rtptime;
guint32 packet_count;
guint32 octet_count;
GstClockTime time;
} RTPSenderReport;
* RTPReceiverReport:
* A receiver report structure.
typedef struct {
gboolean is_valid;
guint32 ssrc; /* who the report is from */
guint8 fractionlost;
guint32 packetslost;
guint32 exthighestseq;
guint32 jitter;
guint32 lsr;
guint32 dlsr;
guint32 round_trip;
} RTPReceiverReport;
* RTPPacketInfo:
* @send: if this is a packet for sending
* @rtp: if this info is about an RTP packet
* @is_list: if this is a bufferlist
* @data: a #GstBuffer or #GstBufferList
* @address: address of the sender of the packet
* @current_time: current time according to the system clock
* @running_time: time of a packet as buffer running_time
* @ntpnstime: time of a packet NTP time in nanoseconds
* @header_len: number of overhead bytes per packet
* @bytes: bytes of the packet including lowlevel overhead
* @payload_len: bytes of the RTP payload
* @seqnum: the seqnum of the packet
* @pt: the payload type of the packet
* @rtptime: the RTP time of the packet
* Structure holding information about the packet.
typedef struct {
gboolean send;
gboolean rtp;
gboolean is_list;
gpointer data;
GSocketAddress *address;
GstClockTime current_time;
GstClockTime running_time;
guint64 ntpnstime;
guint header_len;
guint bytes;
guint packets;
guint payload_len;
guint32 ssrc;
guint16 seqnum;
guint8 pt;
guint32 rtptime;
guint32 csrc_count;
guint32 csrcs[16];
} RTPPacketInfo;
* RTPSourceStats:
* @packetsreceived: number of received packets in total
* @prevpacketsreceived: number of packets received in previous reporting
* interval
* @octetsreceived: number of payload bytes received
* @bytesreceived: number of total bytes received including headers and lower
* protocol level overhead
* @max_seqnr: highest sequence number received
* @transit: previous transit time used for calculating @jitter
* @jitter: current jitter (in clock rate units scaled by 16 for precision)
* @prev_rtptime: previous time when an RTP packet was received
* @prev_rtcptime: previous time when an RTCP packet was received
* @last_rtptime: time when last RTP packet received
* @last_rtcptime: time when last RTCP packet received
* @curr_rr: index of current @rr block
* @rr: previous and current receiver report block
* @curr_sr: index of current @sr block
* @sr: previous and current sender report block
* Stats about a source.
typedef struct {
guint64 packets_received;
guint64 octets_received;
guint64 bytes_received;
guint32 prev_expected;
guint32 prev_received;
guint16 max_seq;
guint64 cycles;
guint32 base_seq;
guint32 bad_seq;
guint32 transit;
guint32 jitter;
guint64 packets_sent;
guint64 octets_sent;
guint sent_pli_count;
guint recv_pli_count;
guint sent_fir_count;
guint recv_fir_count;
guint sent_nack_count;
guint recv_nack_count;
/* when we received stuff */
GstClockTime prev_rtptime;
GstClockTime prev_rtcptime;
GstClockTime last_rtptime;
GstClockTime last_rtcptime;
/* sender and receiver reports */
gint curr_rr;
RTPReceiverReport rr[2];
gint curr_sr;
RTPSenderReport sr[2];
} RTPSourceStats;
* Minimum average time between RTCP packets from this site (in
* seconds). This time prevents the reports from `clumping' when
* sessions are small and the law of large numbers isn't helping
* to smooth out the traffic. It also keeps the report interval
* from becoming ridiculously small during transient outages like
* a network partition.
* Fraction of the RTCP bandwidth to be shared among active
* senders. (This fraction was chosen so that in a typical
* session with one or two active senders, the computed report
* time would be roughly equal to the minimum report time so that
* we don't unnecessarily slow down receiver reports.) The
* receiver fraction must be 1 - the sender fraction.
* When receiving a BYE from a source, remove the source from the database
* after this timeout.
* The default and minimum values of the maximum number of missing packets we tolerate.
* These are packets with asequence number bigger than the last seen packet.
#define RTP_DEF_DROPOUT 3000
#define RTP_MIN_DROPOUT 30
* The default and minimum values of the maximum number of misordered packets we tolerate.
* These are packets with a sequence number smaller than the last seen packet.
#define RTP_DEF_MISORDER 100
* RTPPacketRateCtx:
* Context to calculate the pseudo-average packet rate.
typedef struct {
gboolean probed;
gint32 clock_rate;
guint16 last_seqnum;
guint64 last_ts;
guint32 avg_packet_rate;
} RTPPacketRateCtx;
void gst_rtp_packet_rate_ctx_reset (RTPPacketRateCtx * ctx, gint32 clock_rate);
guint32 gst_rtp_packet_rate_ctx_update (RTPPacketRateCtx *ctx, guint16 seqnum, guint32 ts);
guint32 gst_rtp_packet_rate_ctx_get (RTPPacketRateCtx *ctx);
guint32 gst_rtp_packet_rate_ctx_get_max_dropout (RTPPacketRateCtx *ctx, gint32 time_ms);
guint32 gst_rtp_packet_rate_ctx_get_max_misorder (RTPPacketRateCtx *ctx, gint32 time_ms);
* RTPSessionStats:
* Stats kept for a session and used to produce RTCP packet timeouts.
typedef struct {
guint bandwidth;
guint rtcp_bandwidth;
gdouble sender_fraction;
gdouble receiver_fraction;
gdouble min_interval;
GstClockTime bye_timeout;
guint internal_sources;
guint sender_sources;
guint internal_sender_sources;
guint active_sources;
guint avg_rtcp_packet_size;
guint bye_members;
guint nacks_dropped;
guint nacks_sent;
guint nacks_received;
} RTPSessionStats;
void rtp_stats_init_defaults (RTPSessionStats *stats);
void rtp_stats_set_bandwidths (RTPSessionStats *stats,
guint rtp_bw,
gdouble rtcp_bw,
guint rs, guint rr);
GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, GstRTPProfile profile, gboolean ptp, gboolean first);
GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);
GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats);
gint64 rtp_stats_get_packets_lost (const RTPSourceStats *stats);
void rtp_stats_set_min_interval (RTPSessionStats *stats,
gdouble min_interval);
gboolean __g_socket_address_equal (GSocketAddress *a, GSocketAddress *b);
gchar * __g_socket_address_to_string (GSocketAddress * addr);
#endif /* __RTP_STATS_H__ */