mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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Remove code used to change the SSRC after a collision. We now send a RECONFIGURE event upstream to make the upstream element change the SSRC.
1782 lines
51 KiB
C
1782 lines
51 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "rtpsource.h"
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GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
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#define GST_CAT_DEFAULT rtp_source_debug
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#define RTP_MAX_PROBATION_LEN 32
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/* signals and args */
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enum
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{
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LAST_SIGNAL
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};
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#define DEFAULT_SSRC 0
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#define DEFAULT_IS_CSRC FALSE
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#define DEFAULT_IS_VALIDATED FALSE
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#define DEFAULT_IS_SENDER FALSE
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#define DEFAULT_SDES NULL
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#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
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enum
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{
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PROP_0,
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PROP_SSRC,
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PROP_IS_CSRC,
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PROP_IS_VALIDATED,
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PROP_IS_SENDER,
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PROP_SDES,
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PROP_STATS,
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PROP_PROBATION,
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PROP_LAST
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};
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/* GObject vmethods */
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static void rtp_source_finalize (GObject * object);
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static void rtp_source_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void rtp_source_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
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G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
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static void
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rtp_source_class_init (RTPSourceClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->finalize = rtp_source_finalize;
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gobject_class->set_property = rtp_source_set_property;
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gobject_class->get_property = rtp_source_get_property;
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g_object_class_install_property (gobject_class, PROP_SSRC,
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g_param_spec_uint ("ssrc", "SSRC",
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"The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_IS_CSRC,
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g_param_spec_boolean ("is-csrc", "Is CSRC",
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"If this SSRC is acting as a contributing source",
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DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
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g_param_spec_boolean ("is-validated", "Is Validated",
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"If this SSRC is validated", DEFAULT_IS_VALIDATED,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_IS_SENDER,
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g_param_spec_boolean ("is-sender", "Is Sender",
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"If this SSRC is a sender", DEFAULT_IS_SENDER,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* RTPSource::sdes
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*
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* The current SDES items of the source. Returns a structure with name
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* application/x-rtp-source-sdes and may contain the following fields:
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*
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* 'cname' G_TYPE_STRING : The canonical name
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* 'name' G_TYPE_STRING : The user name
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* 'email' G_TYPE_STRING : The user's electronic mail address
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* 'phone' G_TYPE_STRING : The user's phone number
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* 'location' G_TYPE_STRING : The geographic user location
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* 'tool' G_TYPE_STRING : The name of application or tool
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* 'note' G_TYPE_STRING : A notice about the source
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*
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* other fields may be present and these represent private items in
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* the SDES where the field name is the prefix.
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*/
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g_object_class_install_property (gobject_class, PROP_SDES,
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g_param_spec_boxed ("sdes", "SDES",
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"The SDES information for this source",
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GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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/**
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* RTPSource::stats
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*
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* The statistics of the source. This property returns a GstStructure with
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* name application/x-rtp-source-stats with the following fields:
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*
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* "ssrc" G_TYPE_UINT The SSRC of this source
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* "internal" G_TYPE_BOOLEAN If this source is a source of the session
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* "validated" G_TYPE_BOOLEAN If the source is validated
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* "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source
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* "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC
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* "is-sender" G_TYPE_BOOLEAN If this source is a sender
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* "seqnum-base" G_TYPE_INT first seqnum if known
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* "clock-rate" G_TYPE_INT the clock rate of the media
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*
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* The following two fields are only present when known.
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*
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* "rtp-from" G_TYPE_STRING where we received the last RTP packet from
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* "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from
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*
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* The following fields make sense for internal sources and will only increase
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* when "is-sender" is TRUE:
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*
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* "octets-sent" G_TYPE_UINT64 number of bytes we sent
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* "packets-sent" G_TYPE_UINT64 number of packets we sent
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*
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* The following fields make sense for non-internal sources and will only
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* increase when "is-sender" is TRUE.
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*
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* "octets-received" G_TYPE_UINT64 total number of bytes received
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* "packets-received" G_TYPE_UINT64 total number of packets received
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*
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* Following fields are updated when "is-sender" is TRUE.
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*
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* "bitrate" G_TYPE_UINT64 bitrate in bits per second
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* "jitter" G_TYPE_UINT estimated jitter
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* "packets-lost" G_TYPE_INT estimated amount of packets lost
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*
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* The last SR report this source sent. This only updates when "is-sender" is
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* TRUE.
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*
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* "have-sr" G_TYPE_BOOLEAN the source has sent SR
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* "sr-ntptime" G_TYPE_UINT64 ntptime of SR
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* "sr-rtptime" G_TYPE_UINT rtptime of SR
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* "sr-octet-count" G_TYPE_UINT the number of bytes in the SR
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* "sr-packet-count" G_TYPE_UINT the number of packets in the SR
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*
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* The following fields are only present for non-internal sources and
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* represent the content of the last RB packet that was sent to this source.
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* These values are only updated when the source is sending.
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*
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* "sent-rb" G_TYPE_BOOLEAN we have sent an RB
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* "sent-rb-fractionlost" G_TYPE_UINT calculated lost fraction
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* "sent-rb-packetslost" G_TYPE_INT lost packets
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* "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum
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* "sent-rb-jitter" G_TYPE_UINT jitter
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* "sent-rb-lsr" G_TYPE_UINT last SR time
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* "sent-rb-dlsr" G_TYPE_UINT delay since last SR
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*
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* The following fields are only present for non-internal sources and
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* represents the last RB that this source sent. This is only updated
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* when the source is receiving data and sending RB blocks.
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*
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* "have-rb" G_TYPE_BOOLEAN the source has sent RB
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* "rb-fractionlost" G_TYPE_UINT lost fraction
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* "rb-packetslost" G_TYPE_INT lost packets
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* "rb-exthighestseq" G_TYPE_UINT highest received seqnum
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* "rb-jitter" G_TYPE_UINT reception jitter
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* "rb-lsr" G_TYPE_UINT last SR time
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* "rb-dlsr" G_TYPE_UINT delay since last SR
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*
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* The round trip of this source. This is calculated from the last RB
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* values and the recption time of the last RB packet. Only present for
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* non-internal sources.
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*
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* "rb-round-trip" G_TYPE_UINT the round trip time in nanoseconds
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*/
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g_object_class_install_property (gobject_class, PROP_STATS,
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g_param_spec_boxed ("stats", "Stats",
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"The stats of this source", GST_TYPE_STRUCTURE,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_PROBATION,
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g_param_spec_uint ("probation", "Number of probations",
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"Consecutive packet sequence numbers to accept the source",
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0, G_MAXUINT, DEFAULT_PROBATION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
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}
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/**
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* rtp_source_reset:
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* @src: an #RTPSource
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*
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* Reset the stats of @src.
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*/
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void
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rtp_source_reset (RTPSource * src)
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{
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src->marked_bye = FALSE;
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if (src->bye_reason)
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g_free (src->bye_reason);
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src->bye_reason = NULL;
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src->sent_bye = FALSE;
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src->stats.cycles = -1;
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src->stats.jitter = 0;
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src->stats.transit = -1;
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src->stats.curr_sr = 0;
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src->stats.sr[0].is_valid = FALSE;
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src->stats.curr_rr = 0;
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src->stats.rr[0].is_valid = FALSE;
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src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
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src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
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src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
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src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
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}
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static void
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rtp_source_init (RTPSource * src)
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{
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/* sources are initialy on probation until we receive enough valid RTP
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* packets or a valid RTCP packet */
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src->validated = FALSE;
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src->internal = FALSE;
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src->probation = DEFAULT_PROBATION;
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src->curr_probation = src->probation;
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src->closing = FALSE;
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src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
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src->payload = -1;
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src->clock_rate = -1;
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src->packets = g_queue_new ();
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src->seqnum_base = -1;
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src->last_rtptime = -1;
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src->retained_feedback = g_queue_new ();
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rtp_source_reset (src);
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}
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static void
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rtp_conflicting_address_free (RTPConflictingAddress * addr)
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{
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g_object_unref (addr->address);
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g_free (addr);
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}
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static void
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rtp_source_finalize (GObject * object)
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{
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RTPSource *src;
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GstBuffer *buffer;
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src = RTP_SOURCE_CAST (object);
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while ((buffer = g_queue_pop_head (src->packets)))
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gst_buffer_unref (buffer);
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g_queue_free (src->packets);
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gst_structure_free (src->sdes);
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g_free (src->bye_reason);
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gst_caps_replace (&src->caps, NULL);
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g_list_foreach (src->conflicting_addresses,
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(GFunc) rtp_conflicting_address_free, NULL);
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g_list_free (src->conflicting_addresses);
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while ((buffer = g_queue_pop_head (src->retained_feedback)))
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gst_buffer_unref (buffer);
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g_queue_free (src->retained_feedback);
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if (src->rtp_from)
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g_object_unref (src->rtp_from);
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if (src->rtcp_from)
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g_object_unref (src->rtcp_from);
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G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
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}
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static GstStructure *
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rtp_source_create_stats (RTPSource * src)
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{
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GstStructure *s;
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gboolean is_sender = src->is_sender;
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gboolean internal = src->internal;
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gchar *address_str;
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gboolean have_rb;
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guint8 fractionlost = 0;
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gint32 packetslost = 0;
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guint32 exthighestseq = 0;
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guint32 jitter = 0;
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guint32 lsr = 0;
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guint32 dlsr = 0;
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guint32 round_trip = 0;
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gboolean have_sr;
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GstClockTime time = 0;
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guint64 ntptime = 0;
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guint32 rtptime = 0;
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guint32 packet_count = 0;
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guint32 octet_count = 0;
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/* common data for all types of sources */
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s = gst_structure_new ("application/x-rtp-source-stats",
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"ssrc", G_TYPE_UINT, (guint) src->ssrc,
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"internal", G_TYPE_BOOLEAN, internal,
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"validated", G_TYPE_BOOLEAN, src->validated,
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"received-bye", G_TYPE_BOOLEAN, src->marked_bye,
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"is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
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"is-sender", G_TYPE_BOOLEAN, is_sender,
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"seqnum-base", G_TYPE_INT, src->seqnum_base,
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"clock-rate", G_TYPE_INT, src->clock_rate, NULL);
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/* add address and port */
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if (src->rtp_from) {
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address_str = __g_socket_address_to_string (src->rtp_from);
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gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
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g_free (address_str);
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}
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if (src->rtcp_from) {
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address_str = __g_socket_address_to_string (src->rtcp_from);
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gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
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g_free (address_str);
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}
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gst_structure_set (s,
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"octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
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"packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
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"octets-received", G_TYPE_UINT64, src->stats.octets_received,
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"packets-received", G_TYPE_UINT64, src->stats.packets_received,
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"bitrate", G_TYPE_UINT64, src->bitrate,
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"packets-lost", G_TYPE_INT,
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(gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
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(guint) (src->stats.jitter >> 4), NULL);
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/* get the last SR. */
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have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
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&packet_count, &octet_count);
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gst_structure_set (s,
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"have-sr", G_TYPE_BOOLEAN, have_sr,
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"sr-ntptime", G_TYPE_UINT64, ntptime,
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"sr-rtptime", G_TYPE_UINT, (guint) rtptime,
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"sr-octet-count", G_TYPE_UINT, (guint) octet_count,
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"sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
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if (!internal) {
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/* get the last RB we sent */
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gst_structure_set (s,
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"sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
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"sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
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"sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
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"sent-rb-exthighestseq", G_TYPE_UINT,
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(guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
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(guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
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(guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
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(guint) src->last_rr.dlsr, NULL);
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/* get the last RB */
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have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
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&exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
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gst_structure_set (s,
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"have-rb", G_TYPE_BOOLEAN, have_rb,
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"rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
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"rb-packetslost", G_TYPE_INT, (gint) packetslost,
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"rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
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"rb-jitter", G_TYPE_UINT, (guint) jitter,
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"rb-lsr", G_TYPE_UINT, (guint) lsr,
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"rb-dlsr", G_TYPE_UINT, (guint) dlsr,
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"rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
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}
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return s;
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}
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|
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/**
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* rtp_source_get_sdes_struct:
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* @src: an #RTPSource
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*
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* Get the SDES from @src. See the SDES property for more details.
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*
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* Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
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* valid until the SDES items of @src are modified.
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*/
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const GstStructure *
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rtp_source_get_sdes_struct (RTPSource * src)
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{
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g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
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return src->sdes;
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}
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|
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static gboolean
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sdes_struct_compare_func (GQuark field_id, const GValue * value,
|
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gpointer user_data)
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{
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GstStructure *old;
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const gchar *field;
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old = GST_STRUCTURE (user_data);
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field = g_quark_to_string (field_id);
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if (!gst_structure_has_field (old, field))
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return FALSE;
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g_assert (G_VALUE_HOLDS_STRING (value));
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return strcmp (g_value_get_string (value), gst_structure_get_string (old,
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field)) == 0;
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}
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|
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/**
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* rtp_source_set_sdes_struct:
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* @src: an #RTPSource
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* @sdes: the SDES structure
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*
|
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* Store the @sdes in @src. @sdes must be a structure of type
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* "application/x-rtp-source-sdes", see the SDES property for more details.
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*
|
|
* This function takes ownership of @sdes.
|
|
*
|
|
* Returns: %FALSE if the SDES was unchanged.
|
|
*/
|
|
gboolean
|
|
rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
|
|
{
|
|
gboolean changed;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
|
|
"application/x-rtp-source-sdes") == 0, FALSE);
|
|
|
|
changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
|
|
|
|
if (changed) {
|
|
gst_structure_free (src->sdes);
|
|
src->sdes = sdes;
|
|
} else {
|
|
gst_structure_free (sdes);
|
|
}
|
|
return changed;
|
|
}
|
|
|
|
static void
|
|
rtp_source_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
RTPSource *src;
|
|
|
|
src = RTP_SOURCE (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SSRC:
|
|
src->ssrc = g_value_get_uint (value);
|
|
break;
|
|
case PROP_PROBATION:
|
|
src->probation = g_value_get_uint (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
rtp_source_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
RTPSource *src;
|
|
|
|
src = RTP_SOURCE (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_SSRC:
|
|
g_value_set_uint (value, rtp_source_get_ssrc (src));
|
|
break;
|
|
case PROP_IS_CSRC:
|
|
g_value_set_boolean (value, rtp_source_is_as_csrc (src));
|
|
break;
|
|
case PROP_IS_VALIDATED:
|
|
g_value_set_boolean (value, rtp_source_is_validated (src));
|
|
break;
|
|
case PROP_IS_SENDER:
|
|
g_value_set_boolean (value, rtp_source_is_sender (src));
|
|
break;
|
|
case PROP_SDES:
|
|
g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
|
|
break;
|
|
case PROP_STATS:
|
|
g_value_take_boxed (value, rtp_source_create_stats (src));
|
|
break;
|
|
case PROP_PROBATION:
|
|
g_value_set_uint (value, src->probation);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_source_new:
|
|
* @ssrc: an SSRC
|
|
*
|
|
* Create a #RTPSource with @ssrc.
|
|
*
|
|
* Returns: a new #RTPSource. Use g_object_unref() after usage.
|
|
*/
|
|
RTPSource *
|
|
rtp_source_new (guint32 ssrc)
|
|
{
|
|
RTPSource *src;
|
|
|
|
src = g_object_new (RTP_TYPE_SOURCE, NULL);
|
|
src->ssrc = ssrc;
|
|
|
|
return src;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_set_callbacks:
|
|
* @src: an #RTPSource
|
|
* @cb: callback functions
|
|
* @user_data: user data
|
|
*
|
|
* Set the callbacks for the source.
|
|
*/
|
|
void
|
|
rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
|
|
gpointer user_data)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
src->callbacks.push_rtp = cb->push_rtp;
|
|
src->callbacks.clock_rate = cb->clock_rate;
|
|
src->user_data = user_data;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_ssrc:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Get the SSRC of @source.
|
|
*
|
|
* Returns: the SSRC of src.
|
|
*/
|
|
guint32
|
|
rtp_source_get_ssrc (RTPSource * src)
|
|
{
|
|
guint32 result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
|
|
|
|
result = src->ssrc;
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_set_as_csrc:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Configure @src as a CSRC, this will also validate @src.
|
|
*/
|
|
void
|
|
rtp_source_set_as_csrc (RTPSource * src)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
src->validated = TRUE;
|
|
src->is_csrc = TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_is_as_csrc:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Check if @src is a contributing source.
|
|
*
|
|
* Returns: %TRUE if @src is acting as a contributing source.
|
|
*/
|
|
gboolean
|
|
rtp_source_is_as_csrc (RTPSource * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
result = src->is_csrc;
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_is_active:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Check if @src is an active source. A source is active if it has been
|
|
* validated and has not yet received a BYE packet
|
|
*
|
|
* Returns: %TRUE if @src is an qactive source.
|
|
*/
|
|
gboolean
|
|
rtp_source_is_active (RTPSource * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
result = RTP_SOURCE_IS_ACTIVE (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_is_validated:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Check if @src is a validated source.
|
|
*
|
|
* Returns: %TRUE if @src is a validated source.
|
|
*/
|
|
gboolean
|
|
rtp_source_is_validated (RTPSource * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
result = src->validated;
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_is_sender:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Check if @src is a sending source.
|
|
*
|
|
* Returns: %TRUE if @src is a sending source.
|
|
*/
|
|
gboolean
|
|
rtp_source_is_sender (RTPSource * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
result = RTP_SOURCE_IS_SENDER (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_is_marked_bye:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Check if @src is marked as leaving the session with a BYE packet.
|
|
*
|
|
* Returns: %TRUE if @src has been marked BYE.
|
|
*/
|
|
gboolean
|
|
rtp_source_is_marked_bye (RTPSource * src)
|
|
{
|
|
gboolean result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
result = RTP_SOURCE_IS_MARKED_BYE (src);
|
|
|
|
return result;
|
|
}
|
|
|
|
|
|
/**
|
|
* rtp_source_get_bye_reason:
|
|
* @src: an #RTPSource
|
|
*
|
|
* Get the BYE reason for @src. Check if the source is marked as leaving the
|
|
* session with a BYE message first with rtp_source_is_marked_bye().
|
|
*
|
|
* Returns: The BYE reason or NULL when no reason was given or the source was
|
|
* not marked BYE yet. g_free() after usage.
|
|
*/
|
|
gchar *
|
|
rtp_source_get_bye_reason (RTPSource * src)
|
|
{
|
|
gchar *result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
|
|
|
|
result = g_strdup (src->bye_reason);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_update_caps:
|
|
* @src: an #RTPSource
|
|
* @caps: a #GstCaps
|
|
*
|
|
* Parse @caps and store all relevant information in @source.
|
|
*/
|
|
void
|
|
rtp_source_update_caps (RTPSource * src, GstCaps * caps)
|
|
{
|
|
GstStructure *s;
|
|
guint val;
|
|
gint ival;
|
|
|
|
/* nothing changed, return */
|
|
if (caps == NULL || src->caps == caps)
|
|
return;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (gst_structure_get_int (s, "payload", &ival))
|
|
src->payload = ival;
|
|
else
|
|
src->payload = -1;
|
|
GST_DEBUG ("got payload %d", src->payload);
|
|
|
|
if (gst_structure_get_int (s, "clock-rate", &ival))
|
|
src->clock_rate = ival;
|
|
else
|
|
src->clock_rate = -1;
|
|
|
|
GST_DEBUG ("got clock-rate %d", src->clock_rate);
|
|
|
|
if (gst_structure_get_uint (s, "seqnum-base", &val))
|
|
src->seqnum_base = val;
|
|
else
|
|
src->seqnum_base = -1;
|
|
|
|
GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
|
|
|
|
gst_caps_replace (&src->caps, caps);
|
|
}
|
|
|
|
/**
|
|
* rtp_source_set_rtp_from:
|
|
* @src: an #RTPSource
|
|
* @address: the RTP address to set
|
|
*
|
|
* Set that @src is receiving RTP packets from @address. This is used for
|
|
* collistion checking.
|
|
*/
|
|
void
|
|
rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
if (src->rtp_from)
|
|
g_object_unref (src->rtp_from);
|
|
src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
|
|
}
|
|
|
|
/**
|
|
* rtp_source_set_rtcp_from:
|
|
* @src: an #RTPSource
|
|
* @address: the RTCP address to set
|
|
*
|
|
* Set that @src is receiving RTCP packets from @address. This is used for
|
|
* collistion checking.
|
|
*/
|
|
void
|
|
rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
if (src->rtcp_from)
|
|
g_object_unref (src->rtcp_from);
|
|
src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
|
|
}
|
|
|
|
static GstFlowReturn
|
|
push_packet (RTPSource * src, GstBuffer * buffer)
|
|
{
|
|
GstFlowReturn ret = GST_FLOW_OK;
|
|
|
|
/* push queued packets first if any */
|
|
while (!g_queue_is_empty (src->packets)) {
|
|
GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
|
|
|
|
GST_LOG ("pushing queued packet");
|
|
if (src->callbacks.push_rtp)
|
|
src->callbacks.push_rtp (src, buffer, src->user_data);
|
|
else
|
|
gst_buffer_unref (buffer);
|
|
}
|
|
GST_LOG ("pushing new packet");
|
|
/* push packet */
|
|
if (src->callbacks.push_rtp)
|
|
ret = src->callbacks.push_rtp (src, buffer, src->user_data);
|
|
else
|
|
gst_buffer_unref (buffer);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gint
|
|
get_clock_rate (RTPSource * src, guint8 payload)
|
|
{
|
|
if (src->payload == -1) {
|
|
/* first payload received, nothing was in the caps, lock on to this payload */
|
|
src->payload = payload;
|
|
GST_DEBUG ("first payload %d", payload);
|
|
} else if (payload != src->payload) {
|
|
/* we have a different payload than before, reset the clock-rate */
|
|
GST_DEBUG ("new payload %d", payload);
|
|
src->payload = payload;
|
|
src->clock_rate = -1;
|
|
src->stats.transit = -1;
|
|
}
|
|
|
|
if (src->clock_rate == -1) {
|
|
gint clock_rate = -1;
|
|
|
|
if (src->callbacks.clock_rate)
|
|
clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
|
|
|
|
GST_DEBUG ("got clock-rate %d", clock_rate);
|
|
|
|
src->clock_rate = clock_rate;
|
|
}
|
|
return src->clock_rate;
|
|
}
|
|
|
|
/* Jitter is the variation in the delay of received packets in a flow. It is
|
|
* measured by comparing the interval when RTP packets were sent to the interval
|
|
* at which they were received. For instance, if packet #1 and packet #2 leave
|
|
* 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
|
|
* milliseconds. */
|
|
static void
|
|
calculate_jitter (RTPSource * src, GstBuffer * buffer,
|
|
RTPArrivalStats * arrival)
|
|
{
|
|
GstClockTime running_time;
|
|
guint32 rtparrival, transit, rtptime;
|
|
gint32 diff;
|
|
gint clock_rate;
|
|
guint8 pt;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
/* get arrival time */
|
|
if ((running_time = arrival->running_time) == GST_CLOCK_TIME_NONE)
|
|
goto no_time;
|
|
|
|
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
|
|
goto invalid_packet;
|
|
|
|
pt = gst_rtp_buffer_get_payload_type (&rtp);
|
|
|
|
GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
|
|
|
|
/* get clockrate */
|
|
if ((clock_rate = get_clock_rate (src, pt)) == -1) {
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
goto no_clock_rate;
|
|
}
|
|
|
|
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
|
|
/* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
|
|
* care about the absolute value, just the difference. */
|
|
rtparrival = gst_util_uint64_scale_int (running_time, clock_rate, GST_SECOND);
|
|
|
|
/* transit time is difference with RTP timestamp */
|
|
transit = rtparrival - rtptime;
|
|
|
|
/* get ABS diff with previous transit time */
|
|
if (src->stats.transit != -1) {
|
|
if (transit > src->stats.transit)
|
|
diff = transit - src->stats.transit;
|
|
else
|
|
diff = src->stats.transit - transit;
|
|
} else
|
|
diff = 0;
|
|
|
|
src->stats.transit = transit;
|
|
|
|
/* update jitter, the value we store is scaled up so we can keep precision. */
|
|
src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
|
|
|
|
src->stats.prev_rtptime = src->stats.last_rtptime;
|
|
src->stats.last_rtptime = rtparrival;
|
|
|
|
GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
|
|
rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
return;
|
|
|
|
/* ERRORS */
|
|
no_time:
|
|
{
|
|
GST_WARNING ("cannot get current running_time");
|
|
return;
|
|
}
|
|
invalid_packet:
|
|
{
|
|
GST_WARNING ("invalid RTP packet");
|
|
return;
|
|
}
|
|
no_clock_rate:
|
|
{
|
|
GST_WARNING ("cannot get clock-rate for pt %d", pt);
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
init_seq (RTPSource * src, guint16 seq)
|
|
{
|
|
src->stats.base_seq = seq;
|
|
src->stats.max_seq = seq;
|
|
src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
|
|
src->stats.cycles = 0;
|
|
src->stats.packets_received = 0;
|
|
src->stats.octets_received = 0;
|
|
src->stats.bytes_received = 0;
|
|
src->stats.prev_received = 0;
|
|
src->stats.prev_expected = 0;
|
|
|
|
GST_DEBUG ("base_seq %d", seq);
|
|
}
|
|
|
|
#define BITRATE_INTERVAL (2 * GST_SECOND)
|
|
|
|
static void
|
|
do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
|
|
guint64 * bytes_handled)
|
|
{
|
|
guint64 elapsed;
|
|
|
|
if (src->prev_rtime) {
|
|
elapsed = running_time - src->prev_rtime;
|
|
|
|
if (elapsed > BITRATE_INTERVAL) {
|
|
guint64 rate;
|
|
|
|
rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
|
|
|
|
GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
|
|
", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
|
|
|
|
if (src->bitrate == 0)
|
|
src->bitrate = rate;
|
|
else
|
|
src->bitrate = ((src->bitrate * 3) + rate) / 4;
|
|
|
|
src->prev_rtime = running_time;
|
|
*bytes_handled = 0;
|
|
}
|
|
} else {
|
|
GST_LOG ("Reset bitrate measurement");
|
|
src->prev_rtime = running_time;
|
|
src->bitrate = 0;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_rtp:
|
|
* @src: an #RTPSource
|
|
* @buffer: an RTP buffer
|
|
*
|
|
* Let @src handle the incomming RTP @buffer.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
|
|
RTPArrivalStats * arrival)
|
|
{
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
guint16 seqnr, udelta;
|
|
RTPSourceStats *stats;
|
|
guint16 expected;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
|
|
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
|
|
|
|
stats = &src->stats;
|
|
|
|
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
|
|
goto invalid_packet;
|
|
|
|
seqnr = gst_rtp_buffer_get_seq (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
if (stats->cycles == -1) {
|
|
GST_DEBUG ("received first buffer");
|
|
/* first time we heard of this source */
|
|
init_seq (src, seqnr);
|
|
src->stats.max_seq = seqnr - 1;
|
|
src->curr_probation = src->probation;
|
|
}
|
|
|
|
udelta = seqnr - stats->max_seq;
|
|
|
|
/* if we are still on probation, check seqnum */
|
|
if (src->curr_probation) {
|
|
expected = src->stats.max_seq + 1;
|
|
|
|
/* when in probation, we require consecutive seqnums */
|
|
if (seqnr == expected) {
|
|
/* expected packet */
|
|
GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
|
|
src->curr_probation--;
|
|
src->stats.max_seq = seqnr;
|
|
if (src->curr_probation == 0) {
|
|
GST_DEBUG ("probation done!");
|
|
init_seq (src, seqnr);
|
|
} else {
|
|
GstBuffer *q;
|
|
|
|
GST_DEBUG ("probation %d: queue buffer", src->curr_probation);
|
|
/* when still in probation, keep packets in a list. */
|
|
g_queue_push_tail (src->packets, buffer);
|
|
/* remove packets from queue if there are too many */
|
|
while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
|
|
q = g_queue_pop_head (src->packets);
|
|
gst_buffer_unref (q);
|
|
}
|
|
goto done;
|
|
}
|
|
} else {
|
|
/* unexpected seqnum in probation */
|
|
goto probation_seqnum;
|
|
}
|
|
} else if (udelta < RTP_MAX_DROPOUT) {
|
|
/* in order, with permissible gap */
|
|
if (seqnr < stats->max_seq) {
|
|
/* sequence number wrapped - count another 64K cycle. */
|
|
stats->cycles += RTP_SEQ_MOD;
|
|
}
|
|
stats->max_seq = seqnr;
|
|
} else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
|
|
/* the sequence number made a very large jump */
|
|
if (seqnr == stats->bad_seq) {
|
|
/* two sequential packets -- assume that the other side
|
|
* restarted without telling us so just re-sync
|
|
* (i.e., pretend this was the first packet). */
|
|
init_seq (src, seqnr);
|
|
} else {
|
|
/* unacceptable jump */
|
|
stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
|
|
goto bad_sequence;
|
|
}
|
|
} else {
|
|
/* duplicate or reordered packet, will be filtered by jitterbuffer. */
|
|
GST_WARNING ("duplicate or reordered packet (seqnr %d)", seqnr);
|
|
}
|
|
|
|
src->stats.octets_received += arrival->payload_len;
|
|
src->stats.bytes_received += arrival->bytes;
|
|
src->stats.packets_received++;
|
|
/* for the bitrate estimation */
|
|
src->bytes_received += arrival->payload_len;
|
|
/* the source that sent the packet must be a sender */
|
|
src->is_sender = TRUE;
|
|
src->validated = TRUE;
|
|
|
|
do_bitrate_estimation (src, arrival->running_time, &src->bytes_received);
|
|
|
|
GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
|
|
seqnr, src->stats.packets_received, src->stats.octets_received);
|
|
|
|
/* calculate jitter for the stats */
|
|
calculate_jitter (src, buffer, arrival);
|
|
|
|
/* we're ready to push the RTP packet now */
|
|
result = push_packet (src, buffer);
|
|
|
|
done:
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
invalid_packet:
|
|
{
|
|
GST_WARNING ("invalid packet received");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
bad_sequence:
|
|
{
|
|
GST_WARNING ("unacceptable seqnum received");
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
probation_seqnum:
|
|
{
|
|
GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
|
|
src->curr_probation = src->probation;
|
|
src->stats.max_seq = seqnr;
|
|
gst_buffer_unref (buffer);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_source_mark_bye:
|
|
* @src: an #RTPSource
|
|
* @reason: the reason for leaving
|
|
*
|
|
* Mark @src in the BYE state. This can happen when the source wants to
|
|
* leave the sesssion or when a BYE packets has been received.
|
|
*
|
|
* This will make the source inactive.
|
|
*/
|
|
void
|
|
rtp_source_mark_bye (RTPSource * src, const gchar * reason)
|
|
{
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
|
|
GST_STR_NULL (reason));
|
|
|
|
/* copy the reason and mark as bye */
|
|
g_free (src->bye_reason);
|
|
src->bye_reason = g_strdup (reason);
|
|
src->marked_bye = TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_send_rtp:
|
|
* @src: an #RTPSource
|
|
* @data: an RTP buffer or a list of RTP buffers
|
|
* @is_list: if @data is a buffer or list
|
|
* @running_time: the running time of @data
|
|
*
|
|
* Send @data (an RTP buffer or list of buffers) originating from @src.
|
|
* This will make @src a sender. This function takes ownership of @data and
|
|
* modifies the SSRC in the RTP packet to that of @src when needed.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
|
|
GstClockTime running_time)
|
|
{
|
|
GstFlowReturn result;
|
|
guint len;
|
|
guint32 rtptime;
|
|
guint64 ext_rtptime;
|
|
guint64 rt_diff, rtp_diff;
|
|
GstBufferList *list = NULL;
|
|
GstBuffer *buffer = NULL;
|
|
guint packets;
|
|
GstRTPBuffer rtp = { NULL };
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
|
|
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
|
|
|
|
if (is_list) {
|
|
list = GST_BUFFER_LIST_CAST (data);
|
|
|
|
/* We can grab the caps from the first group, since all
|
|
* groups of a buffer list have same caps. */
|
|
buffer = gst_buffer_list_get (list, 0);
|
|
if (!buffer)
|
|
goto no_buffer;
|
|
} else {
|
|
buffer = GST_BUFFER_CAST (data);
|
|
}
|
|
|
|
/* we are a sender now */
|
|
src->is_sender = TRUE;
|
|
|
|
if (is_list) {
|
|
gint i;
|
|
|
|
/* Each group makes up a network packet. */
|
|
packets = gst_buffer_list_length (list);
|
|
for (i = 0, len = 0; i < packets; i++) {
|
|
if (!gst_rtp_buffer_map (gst_buffer_list_get (list, i), GST_MAP_READ,
|
|
&rtp))
|
|
goto invalid_packet;
|
|
|
|
len += gst_rtp_buffer_get_payload_len (&rtp);
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
}
|
|
/* subsequent info taken from first list member */
|
|
gst_rtp_buffer_map (gst_buffer_list_get (list, 0), GST_MAP_READ, &rtp);
|
|
} else {
|
|
packets = 1;
|
|
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
|
|
goto invalid_packet;
|
|
|
|
len = gst_rtp_buffer_get_payload_len (&rtp);
|
|
}
|
|
|
|
/* update stats for the SR */
|
|
src->stats.packets_sent += packets;
|
|
src->stats.octets_sent += len;
|
|
src->bytes_sent += len;
|
|
|
|
do_bitrate_estimation (src, running_time, &src->bytes_sent);
|
|
|
|
rtptime = gst_rtp_buffer_get_timestamp (&rtp);
|
|
ext_rtptime = src->last_rtptime;
|
|
ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
|
|
|
|
GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
|
|
GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
|
|
|
|
if (ext_rtptime > src->last_rtptime) {
|
|
rtp_diff = ext_rtptime - src->last_rtptime;
|
|
rt_diff = running_time - src->last_rtime;
|
|
|
|
/* calc the diff so we can detect drift at the sender. This can also be used
|
|
* to guestimate the clock rate if the NTP time is locked to the RTP
|
|
* timestamps (as is the case when the capture device is providing the clock). */
|
|
GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
|
|
GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
|
|
}
|
|
|
|
/* we keep track of the last received RTP timestamp and the corresponding
|
|
* buffer running_time so that we can use this info when constructing SR reports */
|
|
src->last_rtime = running_time;
|
|
src->last_rtptime = ext_rtptime;
|
|
|
|
/* push packet */
|
|
if (!src->callbacks.push_rtp) {
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
goto no_callback;
|
|
}
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
|
|
src->stats.packets_sent);
|
|
|
|
result = src->callbacks.push_rtp (src, data, src->user_data);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
invalid_packet:
|
|
{
|
|
GST_WARNING ("invalid packet received");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_buffer:
|
|
{
|
|
GST_WARNING ("no buffers in buffer list");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_callback:
|
|
{
|
|
GST_WARNING ("no callback installed, dropping packet");
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_sr:
|
|
* @src: an #RTPSource
|
|
* @time: time of packet arrival
|
|
* @ntptime: the NTP time in 32.32 fixed point
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Update the sender report in @src.
|
|
*/
|
|
void
|
|
rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
|
|
guint32 rtptime, guint32 packet_count, guint32 octet_count)
|
|
{
|
|
RTPSenderReport *curr;
|
|
gint curridx;
|
|
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
|
|
", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
|
|
(guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
|
|
packet_count, octet_count);
|
|
|
|
curridx = src->stats.curr_sr ^ 1;
|
|
curr = &src->stats.sr[curridx];
|
|
|
|
/* this is a sender now */
|
|
src->is_sender = TRUE;
|
|
|
|
/* update current */
|
|
curr->is_valid = TRUE;
|
|
curr->ntptime = ntptime;
|
|
curr->rtptime = rtptime;
|
|
curr->packet_count = packet_count;
|
|
curr->octet_count = octet_count;
|
|
curr->time = time;
|
|
|
|
/* make current */
|
|
src->stats.curr_sr = curridx;
|
|
|
|
src->stats.prev_rtcptime = src->stats.last_rtcptime;
|
|
src->stats.last_rtcptime = time;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_process_rb:
|
|
* @src: an #RTPSource
|
|
* @ntpnstime: the current time in nanoseconds since 1970
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Update the report block in @src.
|
|
*/
|
|
void
|
|
rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
|
|
guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
|
|
guint32 jitter, guint32 lsr, guint32 dlsr)
|
|
{
|
|
RTPReceiverReport *curr;
|
|
gint curridx;
|
|
guint32 ntp, A;
|
|
guint64 f_ntp;
|
|
|
|
g_return_if_fail (RTP_IS_SOURCE (src));
|
|
|
|
GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
|
|
", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
|
|
src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
|
|
lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
|
|
|
|
curridx = src->stats.curr_rr ^ 1;
|
|
curr = &src->stats.rr[curridx];
|
|
|
|
/* update current */
|
|
curr->is_valid = TRUE;
|
|
curr->fractionlost = fractionlost;
|
|
curr->packetslost = packetslost;
|
|
curr->exthighestseq = exthighestseq;
|
|
curr->jitter = jitter;
|
|
curr->lsr = lsr;
|
|
curr->dlsr = dlsr;
|
|
|
|
/* convert the NTP time in nanoseconds to 32.32 fixed point */
|
|
f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
|
|
/* calculate round trip, round the time up */
|
|
ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
|
|
|
|
A = dlsr + lsr;
|
|
if (A > 0 && ntp > A)
|
|
A = ntp - A;
|
|
else
|
|
A = 0;
|
|
curr->round_trip = A;
|
|
|
|
GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
|
|
A >> 16, A & 0xffff);
|
|
|
|
/* make current */
|
|
src->stats.curr_rr = curridx;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_new_sr:
|
|
* @src: an #RTPSource
|
|
* @ntpnstime: the current time in nanoseconds since 1970
|
|
* @running_time: the current running_time of the pipeline.
|
|
* @ntptime: the NTP time in 32.32 fixed point
|
|
* @rtptime: the RTP time corresponding to @ntptime
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Get new values to put into a new SR report from this source.
|
|
*
|
|
* @running_time and @ntpnstime are captured at the same time and represent the
|
|
* running time of the pipeline clock and the absolute current system time in
|
|
* nanoseconds respectively. Together with the last running_time and rtp timestamp
|
|
* we have observed in the source, we can generate @ntptime and @rtptime for an SR
|
|
* packet. @ntptime is basically the fixed point representation of @ntpnstime
|
|
* and @rtptime the associated RTP timestamp.
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
|
|
GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
|
|
guint32 * packet_count, guint32 * octet_count)
|
|
{
|
|
guint64 t_rtp;
|
|
guint64 t_current_ntp;
|
|
GstClockTimeDiff diff;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
/* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
|
|
* and an NTP time, we can scale the RTP timestamps so that they match the
|
|
* given NTP time. for scaling, we assume that the slope of the rtptime vs
|
|
* running_time vs ntptime curve is close to 1, which is certainly
|
|
* sufficient for the frequency at which we report SR and the rate we send
|
|
* out RTP packets. */
|
|
t_rtp = src->last_rtptime;
|
|
|
|
GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
|
|
G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
|
|
|
|
if (src->clock_rate != -1) {
|
|
/* get the diff between the clock running_time and the buffer running_time.
|
|
* This is the elapsed time, as measured against the pipeline clock, between
|
|
* when the rtp timestamp was observed and the current running_time.
|
|
*
|
|
* We need to apply this diff to the RTP timestamp to get the RTP timestamp
|
|
* for the given ntpnstime. */
|
|
diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
|
|
|
|
/* now translate the diff to RTP time, handle positive and negative cases.
|
|
* If there is no diff, we already set rtptime correctly above. */
|
|
if (diff > 0) {
|
|
GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
|
|
t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
|
|
} else {
|
|
diff = -diff;
|
|
GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
|
|
t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
|
|
}
|
|
} else {
|
|
GST_WARNING ("no clock-rate, cannot interpolate rtp time");
|
|
}
|
|
|
|
/* convert the NTP time in nanoseconds to 32.32 fixed point */
|
|
t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
|
|
|
|
GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
|
|
(guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
|
|
(guint32) t_rtp);
|
|
|
|
if (ntptime)
|
|
*ntptime = t_current_ntp;
|
|
if (rtptime)
|
|
*rtptime = t_rtp;
|
|
if (packet_count)
|
|
*packet_count = src->stats.packets_sent;
|
|
if (octet_count)
|
|
*octet_count = src->stats.octets_sent;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_new_rb:
|
|
* @src: an #RTPSource
|
|
* @time: the current time of the system clock
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
*
|
|
* Get new values to put into a new report block from this source.
|
|
*
|
|
* Returns: %TRUE on success.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
|
|
guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
|
|
guint32 * jitter, guint32 * lsr, guint32 * dlsr)
|
|
{
|
|
RTPSourceStats *stats;
|
|
guint64 extended_max, expected;
|
|
guint64 expected_interval, received_interval, ntptime;
|
|
gint64 lost, lost_interval;
|
|
guint32 fraction, LSR, DLSR;
|
|
GstClockTime sr_time;
|
|
|
|
stats = &src->stats;
|
|
|
|
extended_max = stats->cycles + stats->max_seq;
|
|
expected = extended_max - stats->base_seq + 1;
|
|
|
|
GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
|
|
", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
|
|
extended_max, expected, stats->packets_received, stats->base_seq);
|
|
|
|
lost = expected - stats->packets_received;
|
|
lost = CLAMP (lost, -0x800000, 0x7fffff);
|
|
|
|
expected_interval = expected - stats->prev_expected;
|
|
stats->prev_expected = expected;
|
|
received_interval = stats->packets_received - stats->prev_received;
|
|
stats->prev_received = stats->packets_received;
|
|
|
|
lost_interval = expected_interval - received_interval;
|
|
|
|
if (expected_interval == 0 || lost_interval <= 0)
|
|
fraction = 0;
|
|
else
|
|
fraction = (lost_interval << 8) / expected_interval;
|
|
|
|
GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
|
|
/* we scaled the jitter up for additional precision */
|
|
GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
|
|
", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
|
|
extended_max, stats->jitter >> 4);
|
|
|
|
if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
|
|
GstClockTime diff;
|
|
|
|
/* LSR is middle 32 bits of the last ntptime */
|
|
LSR = (ntptime >> 16) & 0xffffffff;
|
|
diff = time - sr_time;
|
|
GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
|
|
/* DLSR, delay since last SR is expressed in 1/65536 second units */
|
|
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
|
|
} else {
|
|
/* No valid SR received, LSR/DLSR are set to 0 then */
|
|
GST_DEBUG ("no valid SR received");
|
|
LSR = 0;
|
|
DLSR = 0;
|
|
}
|
|
GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
|
|
DLSR >> 16, DLSR & 0xffff);
|
|
|
|
if (fractionlost)
|
|
*fractionlost = fraction;
|
|
if (packetslost)
|
|
*packetslost = lost;
|
|
if (exthighestseq)
|
|
*exthighestseq = extended_max;
|
|
if (jitter)
|
|
*jitter = stats->jitter >> 4;
|
|
if (lsr)
|
|
*lsr = LSR;
|
|
if (dlsr)
|
|
*dlsr = DLSR;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_last_sr:
|
|
* @src: an #RTPSource
|
|
* @time: time of packet arrival
|
|
* @ntptime: the NTP time in 32.32 fixed point
|
|
* @rtptime: the RTP time
|
|
* @packet_count: the packet count
|
|
* @octet_count: the octect count
|
|
*
|
|
* Get the values of the last sender report as set with rtp_source_process_sr().
|
|
*
|
|
* Returns: %TRUE if there was a valid SR report.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
|
|
guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
|
|
{
|
|
RTPSenderReport *curr;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
curr = &src->stats.sr[src->stats.curr_sr];
|
|
if (!curr->is_valid)
|
|
return FALSE;
|
|
|
|
if (ntptime)
|
|
*ntptime = curr->ntptime;
|
|
if (rtptime)
|
|
*rtptime = curr->rtptime;
|
|
if (packet_count)
|
|
*packet_count = curr->packet_count;
|
|
if (octet_count)
|
|
*octet_count = curr->octet_count;
|
|
if (time)
|
|
*time = curr->time;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_get_last_rb:
|
|
* @src: an #RTPSource
|
|
* @fractionlost: fraction lost since last SR/RR
|
|
* @packetslost: the cumululative number of packets lost
|
|
* @exthighestseq: the extended last sequence number received
|
|
* @jitter: the interarrival jitter
|
|
* @lsr: the last SR packet from this source
|
|
* @dlsr: the delay since last SR packet
|
|
* @round_trip: the round trip time
|
|
*
|
|
* Get the values of the last RB report set with rtp_source_process_rb().
|
|
*
|
|
* Returns: %TRUE if there was a valid SB report.
|
|
*/
|
|
gboolean
|
|
rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
|
|
gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
|
|
guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
|
|
{
|
|
RTPReceiverReport *curr;
|
|
|
|
g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
|
|
|
|
curr = &src->stats.rr[src->stats.curr_rr];
|
|
if (!curr->is_valid)
|
|
return FALSE;
|
|
|
|
if (fractionlost)
|
|
*fractionlost = curr->fractionlost;
|
|
if (packetslost)
|
|
*packetslost = curr->packetslost;
|
|
if (exthighestseq)
|
|
*exthighestseq = curr->exthighestseq;
|
|
if (jitter)
|
|
*jitter = curr->jitter;
|
|
if (lsr)
|
|
*lsr = curr->lsr;
|
|
if (dlsr)
|
|
*dlsr = curr->dlsr;
|
|
if (round_trip)
|
|
*round_trip = curr->round_trip;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_find_conflicting_address:
|
|
* @src: The source the packet came in
|
|
* @address: address to check for
|
|
* @time: The time when the packet that is possibly in conflict arrived
|
|
*
|
|
* Checks if an address which has a conflict is already known. If it is
|
|
* a known conflict, remember the time
|
|
*
|
|
* Returns: TRUE if it was a known conflict, FALSE otherwise
|
|
*/
|
|
gboolean
|
|
rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
|
|
GstClockTime time)
|
|
{
|
|
GList *item;
|
|
|
|
for (item = g_list_first (src->conflicting_addresses);
|
|
item; item = g_list_next (item)) {
|
|
RTPConflictingAddress *known_conflict = item->data;
|
|
|
|
if (__g_socket_address_equal (address, known_conflict->address)) {
|
|
known_conflict->time = time;
|
|
return TRUE;
|
|
}
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
/**
|
|
* rtp_source_add_conflicting_address:
|
|
* @src: The source the packet came in
|
|
* @address: address to remember
|
|
* @time: The time when the packet that is in conflict arrived
|
|
*
|
|
* Adds a new conflict address
|
|
*/
|
|
void
|
|
rtp_source_add_conflicting_address (RTPSource * src,
|
|
GSocketAddress * address, GstClockTime time)
|
|
{
|
|
RTPConflictingAddress *new_conflict;
|
|
|
|
new_conflict = g_new0 (RTPConflictingAddress, 1);
|
|
|
|
new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
|
|
new_conflict->time = time;
|
|
|
|
src->conflicting_addresses = g_list_prepend (src->conflicting_addresses,
|
|
new_conflict);
|
|
}
|
|
|
|
/**
|
|
* rtp_source_timeout:
|
|
* @src: The #RTPSource
|
|
* @current_time: The current time
|
|
* @collision_timeout: The amount of time after which a collision is timed out
|
|
* @feedback_retention_window: The running time before which retained feedback
|
|
* packets have to be discarded
|
|
*
|
|
* This is processed on each RTCP interval. It times out old collisions.
|
|
* It also times out old retained feedback packets
|
|
*/
|
|
void
|
|
rtp_source_timeout (RTPSource * src, GstClockTime current_time,
|
|
GstClockTime collision_timeout, GstClockTime feedback_retention_window)
|
|
{
|
|
GList *item;
|
|
GstRTCPPacket *pkt;
|
|
|
|
item = g_list_first (src->conflicting_addresses);
|
|
while (item) {
|
|
RTPConflictingAddress *known_conflict = item->data;
|
|
GList *next_item = g_list_next (item);
|
|
|
|
if (known_conflict->time < current_time - collision_timeout) {
|
|
gchar *buf;
|
|
|
|
src->conflicting_addresses =
|
|
g_list_delete_link (src->conflicting_addresses, item);
|
|
buf = __g_socket_address_to_string (known_conflict->address);
|
|
GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
|
|
g_free (buf);
|
|
g_object_unref (known_conflict->address);
|
|
g_free (known_conflict);
|
|
}
|
|
item = next_item;
|
|
}
|
|
|
|
/* Time out AVPF packets that are older than the desired length */
|
|
while ((pkt = g_queue_peek_tail (src->retained_feedback)) &&
|
|
GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window)
|
|
gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
|
|
}
|
|
|
|
static gint
|
|
compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
|
|
{
|
|
const GstBuffer *bufa = a;
|
|
const GstBuffer *bufb = b;
|
|
|
|
return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb);
|
|
}
|
|
|
|
void
|
|
rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
|
|
GstClockTime running_time)
|
|
{
|
|
GstBuffer *buffer;
|
|
|
|
buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
|
|
packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
|
|
|
|
GST_BUFFER_TIMESTAMP (buffer) = running_time;
|
|
|
|
g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
|
|
}
|
|
|
|
gboolean
|
|
rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
|
|
{
|
|
if (g_queue_find_custom (src->retained_feedback, data, func))
|
|
return TRUE;
|
|
else
|
|
return FALSE;
|
|
}
|