gstreamer/gst/rtpmanager/rtpstats.h
Wim Taymans a35d1dde42 gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
Original commit message from CVS:
* gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
(create_session), (gst_rtp_bin_associate),
(gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
(gst_rtp_bin_request_new_pad):
* gst/rtpmanager/gstrtpbin.h:
Add signal to notify listeners when a sender becomes a receiver.
Tweak lip-sync code, don't store our own copy of the ts-offset of the
jitterbuffer, don't adjust sync if the change is less than 4msec.
Get the RTP timestamp <-> GStreamer timestamp relation directly from
the jitterbuffer instead of our inaccurate version from the source.
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
(gst_rtp_jitter_buffer_get_sync):
* gst/rtpmanager/gstrtpjitterbuffer.h:
Add G_LIKELY macros, use global defines for max packet reorder and
dropouts.
Reset the jitterbuffer clock skew detection when packets seqnums are
changed unexpectedly.
* gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
(gst_rtp_session_class_init), (gst_rtp_session_init):
* gst/rtpmanager/gstrtpsession.h:
Add sender timeout signal.
* gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
(calculate_skew), (rtp_jitter_buffer_insert),
(rtp_jitter_buffer_get_sync):
* gst/rtpmanager/rtpjitterbuffer.h:
Add some G_LIKELY macros.
Keep track of the extended RTP timestamp so that we can report the RTP
timestamp <-> GStreamer timestamp relation for lip-sync.
Remove server timestamp gap detection code, the server can sometimes
make a huge gap in timestamps (talk spurts,...) see #549774.
Detect timetamp weirdness instead by observing the sender/receiver
timestamp relation and resync if it changes more than 1 second.
Add method to report about the current rtp <-> gst timestamp relation
which is needed for lip-sync.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(on_sender_timeout), (check_collision), (rtp_session_process_sr),
(session_cleanup):
* gst/rtpmanager/rtpsession.h:
Add sender timeout signal.
Remove inaccurate rtp <-> gst timestamp relation code, the
jitterbuffer can now do an accurate reporting about this.
* gst/rtpmanager/rtpsource.c: (rtp_source_init),
(rtp_source_update_caps), (calculate_jitter),
(rtp_source_process_rtp):
* gst/rtpmanager/rtpsource.h:
Remove inaccurate rtp <-> gst timestamp relation code.
* gst/rtpmanager/rtpstats.h:
Define global max-reorder and max-dropout constants for use in various
subsystems.
2008-09-05 13:52:34 +00:00

193 lines
5.9 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* 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
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* 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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#ifndef __RTP_STATS_H__
#define __RTP_STATS_H__
#include <gst/gst.h>
#include <gst/netbuffer/gstnetbuffer.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;
/**
* RTPArrivalStats:
* @time: arrival time of a packet according to the system clock
* @timestamp: arrival time of a packet as buffer timestamp
* @address: address of the sender of the packet
* @bytes: bytes of the packet including lowlevel overhead
* @payload_len: bytes of the RTP payload
*
* Structure holding information about the arrival stats of a packet.
*/
typedef struct {
GstClockTime time;
GstClockTime timestamp;
guint64 ntpnstime;
gboolean have_address;
GstNetAddress address;
guint bytes;
guint payload_len;
} RTPArrivalStats;
/**
* 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
* @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;
/* 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;
#define RTP_STATS_BANDWIDTH 64000.0
#define RTP_STATS_RTCP_BANDWIDTH 3000.0
/*
* 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.
*/
#define RTP_STATS_MIN_INTERVAL 5.0
/*
* 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.
*/
#define RTP_STATS_SENDER_FRACTION (0.25)
#define RTP_STATS_RECEIVER_FRACTION (1.0 - RTP_STATS_SENDER_FRACTION)
/*
* When receiving a BYE from a source, remove the source from the database
* after this timeout.
*/
#define RTP_STATS_BYE_TIMEOUT (2 * GST_SECOND)
/*
* The maximum number of missing packets we tollerate. These are packets with a
* sequence number bigger than the last seen packet.
*/
#define RTP_MAX_DROPOUT 3000
/*
* The maximum number of misordered packets we tollerate. These are packets with
* a sequence number smaller than the last seen packet.
*/
#define RTP_MAX_MISORDER 100
/**
* RTPSessionStats:
*
* Stats kept for a session and used to produce RTCP packet timeouts.
*/
typedef struct {
gdouble bandwidth;
gdouble sender_fraction;
gdouble receiver_fraction;
gdouble rtcp_bandwidth;
gdouble min_interval;
GstClockTime bye_timeout;
guint sender_sources;
guint active_sources;
guint avg_rtcp_packet_size;
guint bye_members;
} RTPSessionStats;
void rtp_stats_init_defaults (RTPSessionStats *stats);
GstClockTime rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender, gboolean first);
GstClockTime rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, GstClockTime interval);
GstClockTime rtp_stats_calculate_bye_interval (RTPSessionStats *stats);
#endif /* __RTP_STATS_H__ */