gstreamer/gst/rtpmanager/rtpstats.h
Wim Taymans 1d75a69ccf configure.ac: Disable rtpmanager for now because it depends on CVS -base.
Original commit message from CVS:
* configure.ac:
Disable rtpmanager for now because it depends on CVS -base.
* gst/rtpmanager/Makefile.am:
Added new files for session manager.
* gst/rtpmanager/gstrtpjitterbuffer.h:
* gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
(create_stream), (pt_map_requested), (new_ssrc_pad_found):
Some cleanups.
the session manager can now also request a pt-map.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
(gst_rtp_session_class_init), (gst_rtp_session_init),
(gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
(stop_rtcp_thread), (gst_rtp_session_change_state),
(gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
(gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
(gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
(gst_rtp_session_chain_recv_rtp),
(gst_rtp_session_event_recv_rtcp_sink),
(gst_rtp_session_chain_recv_rtcp),
(gst_rtp_session_event_send_rtp_sink),
(gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
(gst_rtp_session_request_new_pad):
* gst/rtpmanager/gstrtpsession.h:
We can ask for pt-map now too when the session manager needs it.
Hook up to the new session manager, implement the needed callbacks for
pushing data, getting clock time and requesting clock-rates.
Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
be send to clients.
Add code to start and stop the thread that will schedule RTCP through
the session manager.
* gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
(rtp_session_init), (rtp_session_finalize),
(rtp_session_set_property), (rtp_session_get_property),
(on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
(on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
(rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
(rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
(source_push_rtp), (source_clock_rate), (check_collision),
(obtain_source), (rtp_session_add_source),
(rtp_session_get_num_sources),
(rtp_session_get_num_active_sources),
(rtp_session_get_source_by_ssrc),
(rtp_session_get_source_by_cname), (rtp_session_create_source),
(update_arrival_stats), (rtp_session_process_rtp),
(rtp_session_process_sr), (rtp_session_process_rr),
(rtp_session_process_sdes), (rtp_session_process_bye),
(rtp_session_process_app), (rtp_session_process_rtcp),
(rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
(rtp_session_produce_rtcp):
* gst/rtpmanager/rtpsession.h:
The advanced beginnings of the main session manager that handles the
participant database of RTPSources, SSRC probation, SSRC collisions,
parse RTCP to update source stats. etc..
* gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
(rtp_source_init), (rtp_source_finalize), (rtp_source_new),
(rtp_source_set_callbacks), (rtp_source_set_as_csrc),
(rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
(push_packet), (get_clock_rate), (calculate_jitter),
(rtp_source_process_rtp), (rtp_source_process_bye),
(rtp_source_send_rtp), (rtp_source_process_sr),
(rtp_source_process_rb):
* gst/rtpmanager/rtpsource.h:
Object that encapsulates an SSRC and its state in the database.
Calculates the jitter and transit times of data packets.
* gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
(rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
* gst/rtpmanager/rtpstats.h:
Various stats regarding the session and sources.
Used to calculate the RTCP interval.
2007-04-18 18:58:53 +00:00

161 lines
5 KiB
C

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.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;
} 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;
} RTPReceiverReport;
/**
* RTPArrivalStats:
* @time: arrival time of a packet
* @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;
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 packetsreceived;
guint64 prevpacketsreceived;
guint64 octetsreceived;
guint64 bytesreceived;
guint16 max_seqnr;
guint32 transit;
guint32 jitter;
/* 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)
/**
* 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;
guint sender_sources;
guint active_sources;
guint avg_rtcp_packet_size;
guint avg_bye_packet_size;
gboolean sent_rtcp;
} RTPSessionStats;
void rtp_stats_init_defaults (RTPSessionStats *stats);
gdouble rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender);
gdouble rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, gdouble interval);
#endif /* __RTP_STATS_H__ */