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34534179a2
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (create_rtcp): fix for pad name change * gst/rtpmanager/gstrtpsession.c: (rtcp_thread), (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate): Fix for renamed methods. * gst/rtpmanager/rtpsession.c: (rtp_session_init), (rtp_session_finalize), (rtp_session_set_cname), (rtp_session_get_cname), (rtp_session_set_name), (rtp_session_get_name), (rtp_session_set_email), (rtp_session_get_email), (rtp_session_set_phone), (rtp_session_get_phone), (rtp_session_set_location), (rtp_session_get_location), (rtp_session_set_tool), (rtp_session_get_tool), (rtp_session_set_note), (rtp_session_get_note), (source_push_rtp), (obtain_source), (rtp_session_add_source), (rtp_session_get_source_by_ssrc), (rtp_session_create_source), (rtp_session_process_rtp), (rtp_session_process_sr), (rtp_session_process_sdes), (rtp_session_process_rtcp), (rtp_session_send_rtp), (rtp_session_get_reporting_interval), (session_report_blocks), (session_sdes), (rtp_session_perform_reporting): * gst/rtpmanager/rtpsession.h: Prepare for implementing SSRC sampling. Create SSRC for the session. Add methods to set the SDES entries. fix accounting of senders/receivers. Implement SR/RR/SDES RTCP reporting. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq), (rtp_source_process_rtp), (rtp_source_process_sr): * gst/rtpmanager/rtpsource.h: Implement extended sequence number. * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval): * gst/rtpmanager/rtpstats.h: Rename some fields.
111 lines
3.2 KiB
C
111 lines
3.2 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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#include "rtpstats.h"
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/**
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* rtp_stats_init_defaults:
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* @stats: an #RTPSessionStats struct
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*
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* Initialize @stats with its default values.
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*/
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void
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rtp_stats_init_defaults (RTPSessionStats * stats)
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{
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stats->bandwidth = RTP_STATS_BANDWIDTH;
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stats->sender_fraction = RTP_STATS_SENDER_FRACTION;
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stats->receiver_fraction = RTP_STATS_RECEIVER_FRACTION;
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stats->rtcp_bandwidth = RTP_STATS_RTCP_BANDWIDTH;
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stats->min_interval = RTP_STATS_MIN_INTERVAL;
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}
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/**
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* rtp_stats_calculate_rtcp_interval:
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* @stats: an #RTPSessionStats struct
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*
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* Calculate the RTCP interval. The result of this function is the amount of
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* time to wait (in seconds) before sender a new RTCP message.
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*
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* Returns: the RTCP interval.
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*/
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gdouble
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rtp_stats_calculate_rtcp_interval (RTPSessionStats * stats, gboolean sender)
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{
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gdouble active, senders, receivers, sfraction;
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gboolean avg_rtcp;
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gdouble interval;
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active = stats->active_sources;
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/* Try to avoid division by zero */
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if (stats->active_sources == 0)
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active += 1.0;
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senders = (gdouble) stats->sender_sources;
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receivers = (gdouble) (active - senders);
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avg_rtcp = (gdouble) stats->avg_rtcp_packet_size;
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sfraction = senders / active;
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GST_DEBUG ("senders: %f, receivers %f, avg_rtcp %f, sfraction %f",
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senders, receivers, avg_rtcp, sfraction);
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if (senders > 0 && sfraction <= stats->sender_fraction) {
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if (sender) {
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interval =
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(avg_rtcp * senders) / (stats->sender_fraction *
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stats->rtcp_bandwidth);
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} else {
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interval =
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(avg_rtcp * receivers) / ((1.0 -
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stats->sender_fraction) * stats->rtcp_bandwidth);
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}
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} else {
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interval = (avg_rtcp * active) / stats->rtcp_bandwidth;
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}
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if (interval < stats->min_interval)
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interval = stats->min_interval;
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if (!stats->sent_rtcp)
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interval /= 2.0;
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return interval;
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}
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/**
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* rtp_stats_calculate_rtcp_interval:
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* @stats: an #RTPSessionStats struct
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* @interval: an RTCP interval
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*
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* Apply a random jitter to the @interval. @interval is typically obtained with
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* rtp_stats_calculate_rtcp_interval().
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*
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* Returns: the new RTCP interval.
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*/
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gdouble
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rtp_stats_add_rtcp_jitter (RTPSessionStats * stats, gdouble interval)
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{
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/* see RFC 3550 p 30
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* To compensate for "unconditional reconsideration" converging to a
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* value below the intended average.
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*/
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#define COMPENSATION (2.71828 - 1.5);
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return (interval * g_random_double_range (0.5, 1.5)) / COMPENSATION;
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}
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