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54b3dec1f5
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.
161 lines
5 KiB
C
161 lines
5 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim@fluendo.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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, 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/netbuffer/gstnetbuffer.h>
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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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} 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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} RTPReceiverReport;
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/**
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* RTPArrivalStats:
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* @time: arrival time of a packet
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* @address: address of the sender of the 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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*
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* Structure holding information about the arrival stats of a packet.
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*/
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typedef struct {
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GstClockTime time;
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gboolean have_address;
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GstNetAddress address;
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guint bytes;
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guint payload_len;
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} RTPArrivalStats;
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/**
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* RTPSourceStats:
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* @packetsreceived: number of received packets in total
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* @prevpacketsreceived: number of packets received in previous reporting
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* interval
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* @octetsreceived: number of payload bytes received
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* @bytesreceived: 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
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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 packetsreceived;
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guint64 prevpacketsreceived;
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guint64 octetsreceived;
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guint64 bytesreceived;
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guint16 max_seqnr;
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guint32 transit;
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guint32 jitter;
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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 64000.0
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#define RTP_STATS_RTCP_BANDWIDTH 3000.0
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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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* 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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gdouble bandwidth;
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gdouble sender_fraction;
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gdouble receiver_fraction;
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gdouble rtcp_bandwidth;
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gdouble min_interval;
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guint sender_sources;
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guint active_sources;
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guint avg_rtcp_packet_size;
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guint avg_bye_packet_size;
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gboolean sent_rtcp;
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} RTPSessionStats;
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void rtp_stats_init_defaults (RTPSessionStats *stats);
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gdouble rtp_stats_calculate_rtcp_interval (RTPSessionStats *stats, gboolean sender);
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gdouble rtp_stats_add_rtcp_jitter (RTPSessionStats *stats, gdouble interval);
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#endif /* __RTP_STATS_H__ */
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