/* GStreamer * Copyright (C) <2007> Wim Taymans * Copyright (C) 2015 Kurento (http://kurento.org/) * @author: Miguel París * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ #include #include #include #include "rtpsource.h" GST_DEBUG_CATEGORY_STATIC (rtp_source_debug); #define GST_CAT_DEFAULT rtp_source_debug #define RTP_MAX_PROBATION_LEN 32 /* signals and args */ enum { LAST_SIGNAL }; #define DEFAULT_SSRC 0 #define DEFAULT_IS_CSRC FALSE #define DEFAULT_IS_VALIDATED FALSE #define DEFAULT_IS_SENDER FALSE #define DEFAULT_SDES NULL #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION #define DEFAULT_MAX_DROPOUT_TIME 60000 #define DEFAULT_MAX_MISORDER_TIME 2000 #define DEFAULT_DISABLE_RTCP FALSE enum { PROP_0, PROP_SSRC, PROP_IS_CSRC, PROP_IS_VALIDATED, PROP_IS_SENDER, PROP_SDES, PROP_STATS, PROP_PROBATION, PROP_MAX_DROPOUT_TIME, PROP_MAX_MISORDER_TIME, PROP_DISABLE_RTCP }; /* GObject vmethods */ static void rtp_source_finalize (GObject * object); static void rtp_source_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void rtp_source_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */ G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT); static void rtp_source_class_init (RTPSourceClass * klass) { GObjectClass *gobject_class; gobject_class = (GObjectClass *) klass; gobject_class->finalize = rtp_source_finalize; gobject_class->set_property = rtp_source_set_property; gobject_class->get_property = rtp_source_get_property; g_object_class_install_property (gobject_class, PROP_SSRC, g_param_spec_uint ("ssrc", "SSRC", "The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC, G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_IS_CSRC, g_param_spec_boolean ("is-csrc", "Is CSRC", "If this SSRC is acting as a contributing source", DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_IS_VALIDATED, g_param_spec_boolean ("is-validated", "Is Validated", "If this SSRC is validated", DEFAULT_IS_VALIDATED, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_IS_SENDER, g_param_spec_boolean ("is-sender", "Is Sender", "If this SSRC is a sender", DEFAULT_IS_SENDER, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * RTPSource::sdes * * The current SDES items of the source. Returns a structure with name * application/x-rtp-source-sdes and may contain the following fields: * * 'cname' G_TYPE_STRING : The canonical name in the form user@host * 'name' G_TYPE_STRING : The user name * 'email' G_TYPE_STRING : The user's electronic mail address * 'phone' G_TYPE_STRING : The user's phone number * 'location' G_TYPE_STRING : The geographic user location * 'tool' G_TYPE_STRING : The name of application or tool * 'note' G_TYPE_STRING : A notice about the source * * Other fields may be present and these represent private items in * the SDES where the field name is the prefix. */ g_object_class_install_property (gobject_class, PROP_SDES, g_param_spec_boxed ("sdes", "SDES", "The SDES information for this source", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * RTPSource::stats * * This property returns a GstStructure named application/x-rtp-source-stats with * fields useful for statistics and diagnostics. * * Take note of each respective field's units: * * - NTP times are in the appropriate 32-bit or 64-bit fixed-point format * starting from January 1, 1970 (except for timespans). * - RTP times are in clock rate units (i.e. clock rate = 1 second) * starting at a random offset. * - For fields indicating packet loss, note that late packets are not considered lost, * and duplicates are not taken into account. Hence, the loss may be negative * if there are duplicates. * * The following fields are always present. * * "ssrc" G_TYPE_UINT the SSRC of this source * "internal" G_TYPE_BOOLEAN this source is a source of the session * "validated" G_TYPE_BOOLEAN the source is validated * "received-bye" G_TYPE_BOOLEAN we received a BYE from this source * "is-csrc" G_TYPE_BOOLEAN this source was found as CSRC * "is-sender" G_TYPE_BOOLEAN this source is a sender * "seqnum-base" G_TYPE_INT first seqnum if known * "clock-rate" G_TYPE_INT the clock rate of the media * * The following fields are only present when known. * * "rtp-from" G_TYPE_STRING where we received the last RTP packet from * "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from * * The following fields make sense for internal sources and will only increase * when "is-sender" is TRUE. * * "octets-sent" G_TYPE_UINT64 number of payload bytes we sent * "packets-sent" G_TYPE_UINT64 number of packets we sent * * The following fields make sense for non-internal sources and will only * increase when "is-sender" is TRUE. * * "octets-received" G_TYPE_UINT64 total number of payload bytes received * "packets-received" G_TYPE_UINT64 total number of packets received * "bytes-received" G_TYPE_UINT64 total number of bytes received including lower level headers overhead * * Following fields are updated when "is-sender" is TRUE. * * "bitrate" G_TYPE_UINT64 bitrate in bits per second * "jitter" G_TYPE_UINT estimated jitter (in clock rate units) * "packets-lost" G_TYPE_INT estimated amount of packets lost * * The last SR report this source sent. This only updates when "is-sender" is * TRUE. * * "have-sr" G_TYPE_BOOLEAN the source has sent SR * "sr-ntptime" G_TYPE_UINT64 NTP time of SR (in NTP Timestamp Format, 32.32 fixed point) * "sr-rtptime" G_TYPE_UINT RTP time of SR (in clock rate units) * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR * "sr-packet-count" G_TYPE_UINT the number of packets in the SR * * The following fields are only present for non-internal sources and * represent the content of the last RB packet that was sent to this source. * These values are only updated when the source is sending. * * "sent-rb" G_TYPE_BOOLEAN we have sent an RB * "sent-rb-fractionlost" G_TYPE_UINT calculated lost 8-bit fraction * "sent-rb-packetslost" G_TYPE_INT lost packets * "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum * "sent-rb-jitter" G_TYPE_UINT jitter (in clock rate units) * "sent-rb-lsr" G_TYPE_UINT last SR time (seconds in NTP Short Format, 16.16 fixed point) * "sent-rb-dlsr" G_TYPE_UINT delay since last SR (seconds in NTP Short Format, 16.16 fixed point) * * The following fields are only present for non-internal sources and * represents the last RB that this source sent. This is only updated * when the source is receiving data and sending RB blocks. * * "have-rb" G_TYPE_BOOLEAN the source has sent RB * "rb-fractionlost" G_TYPE_UINT lost 8-bit fraction * "rb-packetslost" G_TYPE_INT lost packets * "rb-exthighestseq" G_TYPE_UINT highest received seqnum * "rb-jitter" G_TYPE_UINT reception jitter (in clock rate units) * "rb-lsr" G_TYPE_UINT last SR time (seconds in NTP Short Format, 16.16 fixed point) * "rb-dlsr" G_TYPE_UINT delay since last SR (seconds in NTP Short Format, 16.16 fixed point) * * The round trip of this source is calculated from the last RB * values and the reception time of the last RB packet. It is only present for * non-internal sources. * * "rb-round-trip" G_TYPE_UINT the round-trip time (seconds in NTP Short Format, 16.16 fixed point) * */ g_object_class_install_property (gobject_class, PROP_STATS, g_param_spec_boxed ("stats", "Stats", "The stats of this source", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROBATION, g_param_spec_uint ("probation", "Number of probations", "Consecutive packet sequence numbers to accept the source", 0, G_MAXUINT, DEFAULT_PROBATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME, g_param_spec_uint ("max-dropout-time", "Max dropout time", "The maximum time (milliseconds) of missing packets tolerated.", 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME, g_param_spec_uint ("max-misorder-time", "Max misorder time", "The maximum time (milliseconds) of misordered packets tolerated.", 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * RTPSession::disable-rtcp: * * Allow disabling the sending of RTCP packets for this source. */ g_object_class_install_property (gobject_class, PROP_DISABLE_RTCP, g_param_spec_boolean ("disable-rtcp", "Disable RTCP", "Disable sending RTCP packets for this source", DEFAULT_DISABLE_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source"); } /** * rtp_source_reset: * @src: an #RTPSource * * Reset the stats of @src. */ void rtp_source_reset (RTPSource * src) { src->marked_bye = FALSE; if (src->bye_reason) g_free (src->bye_reason); src->bye_reason = NULL; src->sent_bye = FALSE; g_hash_table_remove_all (src->reported_in_sr_of); g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL); g_queue_clear (src->retained_feedback); src->last_rtptime = -1; src->stats.cycles = -1; src->stats.jitter = 0; src->stats.transit = -1; src->stats.curr_sr = 0; src->stats.sr[0].is_valid = FALSE; src->stats.curr_rr = 0; src->stats.rr[0].is_valid = FALSE; src->stats.prev_rtptime = GST_CLOCK_TIME_NONE; src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE; src->stats.last_rtptime = GST_CLOCK_TIME_NONE; src->stats.last_rtcptime = GST_CLOCK_TIME_NONE; g_array_set_size (src->nacks, 0); src->stats.sent_pli_count = 0; src->stats.sent_fir_count = 0; src->stats.sent_nack_count = 0; src->stats.recv_nack_count = 0; } static void rtp_source_init (RTPSource * src) { /* sources are initially on probation until we receive enough valid RTP * packets or a valid RTCP packet */ src->validated = FALSE; src->internal = FALSE; src->probation = DEFAULT_PROBATION; src->curr_probation = src->probation; src->closing = FALSE; src->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME; src->max_misorder_time = DEFAULT_MAX_MISORDER_TIME; src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes"); src->payload = -1; src->clock_rate = -1; src->packets = g_queue_new (); src->seqnum_offset = -1; src->retained_feedback = g_queue_new (); src->nacks = g_array_new (FALSE, FALSE, sizeof (guint16)); src->nack_deadlines = g_array_new (FALSE, FALSE, sizeof (GstClockTime)); src->reported_in_sr_of = g_hash_table_new (g_direct_hash, g_direct_equal); src->last_keyframe_request = GST_CLOCK_TIME_NONE; rtp_source_reset (src); src->pt_set = FALSE; } void rtp_conflicting_address_free (RTPConflictingAddress * addr) { g_object_unref (addr->address); g_slice_free (RTPConflictingAddress, addr); } static void rtp_source_finalize (GObject * object) { RTPSource *src; src = RTP_SOURCE_CAST (object); g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL); g_queue_free (src->packets); gst_structure_free (src->sdes); g_free (src->bye_reason); gst_caps_replace (&src->caps, NULL); g_list_free_full (src->conflicting_addresses, (GDestroyNotify) rtp_conflicting_address_free); g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL); g_queue_free (src->retained_feedback); g_array_free (src->nacks, TRUE); g_array_free (src->nack_deadlines, TRUE); if (src->rtp_from) g_object_unref (src->rtp_from); if (src->rtcp_from) g_object_unref (src->rtcp_from); g_hash_table_unref (src->reported_in_sr_of); G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object); } static GstStructure * rtp_source_create_stats (RTPSource * src) { GstStructure *s; gboolean is_sender = src->is_sender; gboolean internal = src->internal; gchar *address_str; gboolean have_rb; guint8 fractionlost = 0; gint32 packetslost = 0; guint32 exthighestseq = 0; guint32 jitter = 0; guint32 lsr = 0; guint32 dlsr = 0; guint32 round_trip = 0; gboolean have_sr; GstClockTime time = 0; guint64 ntptime = 0; guint32 rtptime = 0; guint32 packet_count = 0; guint32 octet_count = 0; /* common data for all types of sources */ s = gst_structure_new ("application/x-rtp-source-stats", "ssrc", G_TYPE_UINT, (guint) src->ssrc, "internal", G_TYPE_BOOLEAN, internal, "validated", G_TYPE_BOOLEAN, src->validated, "received-bye", G_TYPE_BOOLEAN, src->marked_bye, "is-csrc", G_TYPE_BOOLEAN, src->is_csrc, "is-sender", G_TYPE_BOOLEAN, is_sender, "seqnum-base", G_TYPE_INT, src->seqnum_offset, "clock-rate", G_TYPE_INT, src->clock_rate, NULL); /* add address and port */ if (src->rtp_from) { address_str = __g_socket_address_to_string (src->rtp_from); gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL); g_free (address_str); } if (src->rtcp_from) { address_str = __g_socket_address_to_string (src->rtcp_from); gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL); g_free (address_str); } gst_structure_set (s, "octets-sent", G_TYPE_UINT64, src->stats.octets_sent, "packets-sent", G_TYPE_UINT64, src->stats.packets_sent, "octets-received", G_TYPE_UINT64, src->stats.octets_received, "packets-received", G_TYPE_UINT64, src->stats.packets_received, "bytes-received", G_TYPE_UINT64, src->stats.bytes_received, "bitrate", G_TYPE_UINT64, src->bitrate, "packets-lost", G_TYPE_INT, (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT, (guint) (src->stats.jitter >> 4), "sent-pli-count", G_TYPE_UINT, src->stats.sent_pli_count, "recv-pli-count", G_TYPE_UINT, src->stats.recv_pli_count, "sent-fir-count", G_TYPE_UINT, src->stats.sent_fir_count, "recv-fir-count", G_TYPE_UINT, src->stats.recv_fir_count, "sent-nack-count", G_TYPE_UINT, src->stats.sent_nack_count, "recv-nack-count", G_TYPE_UINT, src->stats.recv_nack_count, NULL); /* get the last SR. */ have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime, &packet_count, &octet_count); gst_structure_set (s, "have-sr", G_TYPE_BOOLEAN, have_sr, "sr-ntptime", G_TYPE_UINT64, ntptime, "sr-rtptime", G_TYPE_UINT, (guint) rtptime, "sr-octet-count", G_TYPE_UINT, (guint) octet_count, "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL); if (!internal) { /* get the last RB we sent */ gst_structure_set (s, "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid, "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost, "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost, "sent-rb-exthighestseq", G_TYPE_UINT, (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT, (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT, (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT, (guint) src->last_rr.dlsr, NULL); /* get the last RB */ have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr, &round_trip); gst_structure_set (s, "have-rb", G_TYPE_BOOLEAN, have_rb, "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost, "rb-packetslost", G_TYPE_INT, (gint) packetslost, "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq, "rb-jitter", G_TYPE_UINT, (guint) jitter, "rb-lsr", G_TYPE_UINT, (guint) lsr, "rb-dlsr", G_TYPE_UINT, (guint) dlsr, "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL); } return s; } /** * rtp_source_get_sdes_struct: * @src: an #RTPSource * * Get the SDES from @src. See the SDES property for more details. * * Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is * valid until the SDES items of @src are modified. */ const GstStructure * rtp_source_get_sdes_struct (RTPSource * src) { g_return_val_if_fail (RTP_IS_SOURCE (src), NULL); return src->sdes; } static gboolean sdes_struct_compare_func (GQuark field_id, const GValue * value, gpointer user_data) { GstStructure *old; const gchar *field; old = GST_STRUCTURE (user_data); field = g_quark_to_string (field_id); if (!gst_structure_has_field (old, field)) return FALSE; g_assert (G_VALUE_HOLDS_STRING (value)); return strcmp (g_value_get_string (value), gst_structure_get_string (old, field)) == 0; } /** * rtp_source_set_sdes_struct: * @src: an #RTPSource * @sdes: the SDES structure * * Store the @sdes in @src. @sdes must be a structure of type * "application/x-rtp-source-sdes", see the SDES property for more details. * * 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; case PROP_MAX_DROPOUT_TIME: src->max_dropout_time = g_value_get_uint (value); break; case PROP_MAX_MISORDER_TIME: src->max_misorder_time = g_value_get_uint (value); break; case PROP_DISABLE_RTCP: src->disable_rtcp = g_value_get_boolean (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; case PROP_MAX_DROPOUT_TIME: g_value_set_uint (value, src->max_dropout_time); break; case PROP_MAX_MISORDER_TIME: g_value_set_uint (value, src->max_misorder_time); break; case PROP_DISABLE_RTCP: g_value_set_boolean (value, src->disable_rtcp); 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; gboolean rtx; /* nothing changed, return */ if (caps == NULL || src->caps == caps) return; s = gst_caps_get_structure (caps, 0); rtx = (gst_structure_get_uint (s, "rtx-ssrc", &val) && val == src->ssrc); if (gst_structure_get_int (s, rtx ? "rtx-payload" : "payload", &ival)) src->payload = ival; else src->payload = -1; GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", 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, rtx ? "rtx-seqnum-offset" : "seqnum-offset", &val)) src->seqnum_offset = val; else src->seqnum_offset = -1; GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "", src->seqnum_offset); 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; gst_rtp_packet_rate_ctx_reset (&src->packet_rate_ctx, 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, RTPPacketInfo * pinfo) { GstClockTime running_time; guint32 rtparrival, transit, rtptime; gint32 diff; gint clock_rate; guint8 pt; /* get arrival time */ if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE) goto no_time; pt = pinfo->pt; GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt); /* get clockrate */ if ((clock_rate = get_clock_rate (src, pt)) == -1) goto no_clock_rate; rtptime = pinfo->rtptime; /* 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); return; /* ERRORS */ no_time: { GST_WARNING ("cannot get current running_time"); return; } no_clock_rate: { GST_WARNING ("cannot get clock-rate for pt %d", pt); return; } } static void update_queued_stats (GstBuffer * buffer, RTPSource * src) { GstRTPBuffer rtp = { NULL }; guint payload_len; guint64 bytes; /* no need to check the return value, a queued packet is a valid RTP one */ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp); payload_len = gst_rtp_buffer_get_payload_len (&rtp); bytes = gst_buffer_get_size (buffer) + UDP_IP_HEADER_OVERHEAD; src->stats.octets_received += payload_len; src->stats.bytes_received += bytes; src->stats.packets_received++; /* for the bitrate estimation consider all lower level headers */ src->bytes_received += bytes; gst_rtp_buffer_unmap (&rtp); } 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; src->stats.recv_pli_count = 0; src->stats.recv_fir_count = 0; /* if there are queued packets, consider them too in the stats */ g_queue_foreach (src->packets, (GFunc) update_queued_stats, src); 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; } } static gboolean update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo, gboolean is_receive) { guint16 seqnr, expected; RTPSourceStats *stats; gint16 delta; gint32 packet_rate, max_dropout, max_misorder; stats = &src->stats; seqnr = pinfo->seqnum; packet_rate = gst_rtp_packet_rate_ctx_update (&src->packet_rate_ctx, pinfo->seqnum, pinfo->rtptime); max_dropout = gst_rtp_packet_rate_ctx_get_max_dropout (&src->packet_rate_ctx, src->max_dropout_time); max_misorder = gst_rtp_packet_rate_ctx_get_max_misorder (&src->packet_rate_ctx, src->max_misorder_time); GST_TRACE ("SSRC %08x, packet_rate: %d, max_dropout: %d, max_misorder: %d", src->ssrc, packet_rate, max_dropout, max_misorder); if (stats->cycles == -1) { GST_DEBUG ("received first packet"); /* first time we heard of this source */ init_seq (src, seqnr); src->stats.max_seq = seqnr - 1; src->curr_probation = src->probation; } if (is_receive) { expected = src->stats.max_seq + 1; delta = gst_rtp_buffer_compare_seqnum (expected, seqnr); /* if we are still on probation, check seqnum */ if (src->curr_probation) { /* when in probation, we require consecutive seqnums */ if (delta == 0) { /* expected packet */ GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected); src->curr_probation--; if (seqnr < stats->max_seq) { /* sequence number wrapped - count another 64K cycle. */ stats->cycles += RTP_SEQ_MOD; } 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 packet", src->curr_probation); /* when still in probation, keep packets in a list. */ g_queue_push_tail (src->packets, pinfo->data); pinfo->data = NULL; /* 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 * * There is no need to clean the queue at this point because the * invalid packets in the queue are not going to be pushed as we are * still in probation, and some cleanup will be performed at future * probation attempts anyway if there are too many old packets in the * queue. */ goto probation_seqnum; } } else if (delta >= 0 && delta < max_dropout) { /* Clear bad packets */ stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */ g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (src->packets); /* 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 (delta < -max_misorder || delta >= max_dropout) { /* the sequence number made a very large jump */ if (seqnr == stats->bad_seq && src->packets->head) { /* 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); g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (src->packets); g_queue_push_tail (src->packets, pinfo->data); pinfo->data = NULL; goto bad_sequence; } } else { /* delta < 0 && delta >= -max_misorder */ /* Clear bad packets */ stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */ g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL); g_queue_clear (src->packets); /* duplicate or reordered packet, will be filtered by jitterbuffer. */ GST_INFO ("duplicate or reordered packet (seqnr %u, expected %u)", seqnr, expected); } } src->stats.octets_received += pinfo->payload_len; src->stats.bytes_received += pinfo->bytes; src->stats.packets_received += pinfo->packets; /* for the bitrate estimation consider all lower level headers */ src->bytes_received += pinfo->bytes; GST_LOG ("seq %u, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT, seqnr, src->stats.packets_received, src->stats.octets_received); return TRUE; /* ERRORS */ done: { return FALSE; } bad_sequence: { GST_WARNING ("unacceptable seqnum received (seqnr %u, delta %d, packet_rate: %d, max_dropout: %d, max_misorder: %d)", seqnr, delta, packet_rate, max_dropout, max_misorder); return FALSE; } probation_seqnum: { GST_WARNING ("probation: seqnr %d != expected %d " "(SSRC %u curr_probation %i probation %i)", seqnr, expected, src->ssrc, src->curr_probation, src->probation); src->curr_probation = src->probation; src->stats.max_seq = seqnr; return FALSE; } } /** * rtp_source_process_rtp: * @src: an #RTPSource * @pinfo: an #RTPPacketInfo * * Let @src handle the incoming RTP packet described in @pinfo. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo) { GstFlowReturn result; g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR); g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR); if (!update_receiver_stats (src, pinfo, TRUE)) return GST_FLOW_OK; /* the source that sent the packet must be a sender */ src->is_sender = TRUE; src->validated = TRUE; do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received); /* calculate jitter for the stats */ calculate_jitter (src, pinfo); /* we're ready to push the RTP packet now */ result = push_packet (src, pinfo->data); pinfo->data = NULL; return result; } /** * 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 * @pinfo: an #RTPPacketInfo * * Send data (an RTP buffer or buffer list from @pinfo) originating from @src. * This will make @src a sender. This function takes ownership of the 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, RTPPacketInfo * pinfo) { GstFlowReturn result; GstClockTime running_time; guint32 rtptime; guint64 ext_rtptime; guint64 rt_diff, rtp_diff; g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR); /* we are a sender now */ src->is_sender = TRUE; /* we are also a receiver of our packets */ if (!update_receiver_stats (src, pinfo, FALSE)) return GST_FLOW_OK; if (src->pt_set && src->pt != pinfo->pt) { GST_WARNING ("Changing pt from %u to %u for SSRC %u", src->pt, pinfo->pt, src->ssrc); } src->pt = pinfo->pt; src->pt_set = TRUE; /* update stats for the SR */ src->stats.packets_sent += pinfo->packets; src->stats.octets_sent += pinfo->payload_len; src->bytes_sent += pinfo->bytes; running_time = pinfo->running_time; do_bitrate_estimation (src, running_time, &src->bytes_sent); rtptime = pinfo->rtptime; 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) goto no_callback; GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, pinfo->is_list ? "list" : "packet", src->stats.packets_sent); result = src->callbacks.push_rtp (src, pinfo->data, src->user_data); pinfo->data = NULL; return result; /* ERRORS */ no_callback: { GST_WARNING ("no callback installed, dropping packet"); return GST_FLOW_OK; } } /** * rtp_source_process_sr: * @src: an #RTPSource * @time: time of packet arrival * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point) * @rtptime: the RTP time (in clock rate units) * @packet_count: the packet count * @octet_count: the octet 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 cumulative number of packets lost * @exthighestseq: the extended last sequence number received * @jitter: the interarrival jitter (in clock rate units) * @lsr: the time of the last SR packet on this source * (in NTP Short Format, 16.16 fixed point) * @dlsr: the delay since the last SR packet * (in NTP Short Format, 16.16 fixed point) * * 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 NTP Timestamp Format, 32.32 fixed point) * @rtptime: the RTP time corresponding to @ntptime (in clock rate units) * @packet_count: the packet count * @octet_count: the octet 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 && src->pt_set) { GST_INFO ("no clock-rate, getting for pt %u and SSRC %u", src->pt, src->ssrc); get_clock_rate (src, src->pt); } 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); GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_STIME_FORMAT, GST_TIME_ARGS (running_time), GST_STIME_ARGS (diff)); /* 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) { t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND); } else { diff = -diff; t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND); } } else { GST_WARNING ("no clock-rate, cannot interpolate rtp time for SSRC %u", src->ssrc); } /* 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 cumulative number of packets lost * @exthighestseq: the extended last sequence number received * @jitter: the interarrival jitter (in clock rate units) * @lsr: the time of the last SR packet on this source * (in NTP Short Format, 16.16 fixed point) * @dlsr: the delay since the last SR packet * (in NTP Short Format, 16.16 fixed point) * * 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 NTP Timestamp Format, 32.32 fixed point) * @rtptime: the RTP time (in clock rate units) * @packet_count: the packet count * @octet_count: the octet 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 cumulative number of packets lost * @exthighestseq: the extended last sequence number received * @jitter: the interarrival jitter (in clock rate units) * @lsr: the time of the last SR packet on this source * (in NTP Short Format, 16.16 fixed point) * @dlsr: the delay since the last SR packet * (in NTP Short Format, 16.16 fixed point) * @round_trip: the round-trip time * (in NTP Short Format, 16.16 fixed point) * * 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; } gboolean find_conflicting_address (GList * conflicting_addresses, GSocketAddress * address, GstClockTime time) { GList *item; for (item = 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; } GList * add_conflicting_address (GList * conflicting_addresses, GSocketAddress * address, GstClockTime time) { RTPConflictingAddress *new_conflict; new_conflict = g_slice_new (RTPConflictingAddress); new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address)); new_conflict->time = time; return g_list_prepend (conflicting_addresses, new_conflict); } GList * timeout_conflicting_addresses (GList * conflicting_addresses, GstClockTime current_time) { GList *item; /* "a relatively long time" -- RFC 3550 section 8.2 */ const GstClockTime collision_timeout = RTP_STATS_MIN_INTERVAL * GST_SECOND * 10; item = g_list_first (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; conflicting_addresses = g_list_delete_link (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); rtp_conflicting_address_free (known_conflict); } item = next_item; } return conflicting_addresses; } /** * 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) { return find_conflicting_address (src->conflicting_addresses, address, time); } /** * 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) { src->conflicting_addresses = add_conflicting_address (src->conflicting_addresses, address, time); } /** * rtp_source_timeout: * @src: The #RTPSource * @current_time: The current time * @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 running_time, GstClockTime feedback_retention_window) { GstRTCPPacket *pkt; GstClockTime max_pts_window; guint pruned = 0; src->conflicting_addresses = timeout_conflicting_addresses (src->conflicting_addresses, current_time); if (feedback_retention_window == GST_CLOCK_TIME_NONE || running_time < feedback_retention_window) { return; } max_pts_window = running_time - feedback_retention_window; /* Time out AVPF packets that are older than the desired length */ while ((pkt = g_queue_peek_head (src->retained_feedback)) && GST_BUFFER_PTS (pkt) < max_pts_window) { gst_buffer_unref (g_queue_pop_head (src->retained_feedback)); pruned++; } GST_LOG_OBJECT (src, "%u RTCP packets pruned with PTS less than %" GST_TIME_FORMAT ", queue len: %u", pruned, GST_TIME_ARGS (max_pts_window), g_queue_get_length (src->retained_feedback)); } static gint compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data) { const GstBuffer *bufa = a; const GstBuffer *bufb = b; g_return_val_if_fail (GST_BUFFER_PTS (bufa) != GST_CLOCK_TIME_NONE, -1); g_return_val_if_fail (GST_BUFFER_PTS (bufb) != GST_CLOCK_TIME_NONE, 1); if (GST_BUFFER_PTS (bufa) < GST_BUFFER_PTS (bufb)) { return -1; } else if (GST_BUFFER_PTS (bufa) > GST_BUFFER_PTS (bufb)) { return 1; } return 0; } void rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet, GstClockTime running_time) { GstBuffer *buffer; g_return_if_fail (running_time != GST_CLOCK_TIME_NONE); buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY, packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4); GST_BUFFER_PTS (buffer) = running_time; g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL); GST_LOG_OBJECT (src, "RTCP packet retained with PTS: %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time)); } 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; } /** * rtp_source_register_nack: * @src: The #RTPSource * @seqnum: a seqnum * @deadline: the deadline before which RTX is still possible * * Register that @seqnum has not been received from @src. */ void rtp_source_register_nack (RTPSource * src, guint16 seqnum, GstClockTime deadline) { gint i; guint len; gint diff = -1; guint16 tseq; len = src->nacks->len; for (i = len - 1; i >= 0; i--) { tseq = g_array_index (src->nacks, guint16, i); diff = gst_rtp_buffer_compare_seqnum (tseq, seqnum); GST_TRACE ("[%u] %u %u diff %i len %u", i, tseq, seqnum, diff, len); if (diff >= 0) break; } if (diff == 0) { GST_DEBUG ("update NACK #%u deadline to %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (deadline)); g_array_index (src->nack_deadlines, GstClockTime, i) = deadline; } else if (i == len - 1) { GST_DEBUG ("append NACK #%u with deadline %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (deadline)); g_array_append_val (src->nacks, seqnum); g_array_append_val (src->nack_deadlines, deadline); } else { GST_DEBUG ("insert NACK #%u with deadline %" GST_TIME_FORMAT, seqnum, GST_TIME_ARGS (deadline)); g_array_insert_val (src->nacks, i + 1, seqnum); g_array_insert_val (src->nack_deadlines, i + 1, deadline); } src->send_nack = TRUE; } /** * rtp_source_get_nacks: * @src: The #RTPSource * @n_nacks: result number of nacks * * Get the registered NACKS since the last rtp_source_clear_nacks(). * * Returns: an array of @n_nacks seqnum values. */ guint16 * rtp_source_get_nacks (RTPSource * src, guint * n_nacks) { if (n_nacks) *n_nacks = src->nacks->len; return (guint16 *) src->nacks->data; } /** * rtp_source_get_nacks: * @src: The #RTPSource * @n_nacks: result number of nacks * * Get the registered NACKS deadlines. * * Returns: an array of @n_nacks deadline values. */ GstClockTime * rtp_source_get_nack_deadlines (RTPSource * src, guint * n_nacks) { if (n_nacks) *n_nacks = src->nack_deadlines->len; return (GstClockTime *) src->nack_deadlines->data; } /** * rtp_source_clear_nacks: * @src: The #RTPSource * @n_nacks: number of nacks * * Remove @n_nacks oldest NACKS form array. */ void rtp_source_clear_nacks (RTPSource * src, guint n_nacks) { g_return_if_fail (n_nacks <= src->nacks->len); if (src->nacks->len == n_nacks) { g_array_set_size (src->nacks, 0); g_array_set_size (src->nack_deadlines, 0); src->send_nack = FALSE; } else { g_array_remove_range (src->nacks, 0, n_nacks); g_array_remove_range (src->nack_deadlines, 0, n_nacks); } }