/* GStreamer * Copyright (C) <2007> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ #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 enum { PROP_0, PROP_SSRC, PROP_IS_CSRC, PROP_IS_VALIDATED, PROP_IS_SENDER, PROP_SDES, PROP_STATS, PROP_LAST }; /* 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 the * following fields: * * 'cname' G_TYPE_STRING : The canonical name * '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 */ 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 * * The statistics of the source. This property returns a GstStructure with * name application/x-rtp-source-stats with the following fields: * */ 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)); 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->received_bye = FALSE; src->stats.cycles = -1; src->stats.jitter = 0; src->stats.transit = -1; src->stats.curr_sr = 0; src->stats.curr_rr = 0; } static void rtp_source_init (RTPSource * src) { /* sources are initialy on probation until we receive enough valid RTP * packets or a valid RTCP packet */ src->validated = FALSE; src->internal = FALSE; src->probation = RTP_DEFAULT_PROBATION; src->payload = -1; src->clock_rate = -1; src->packets = g_queue_new (); src->seqnum_base = -1; src->last_rtptime = -1; rtp_source_reset (src); } static void rtp_source_finalize (GObject * object) { RTPSource *src; GstBuffer *buffer; gint i; src = RTP_SOURCE_CAST (object); while ((buffer = g_queue_pop_head (src->packets))) gst_buffer_unref (buffer); g_queue_free (src->packets); for (i = 0; i < 9; i++) g_free (src->sdes[i]); g_free (src->bye_reason); gst_caps_replace (&src->caps, NULL); 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[GST_NETADDRESS_MAX_LEN]; /* 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->received_bye, "is-csrc", G_TYPE_BOOLEAN, src->is_csrc, "is-sender", G_TYPE_BOOLEAN, is_sender, NULL); /* add address and port */ if (src->have_rtp_from) { gst_netaddress_to_string (&src->rtp_from, address_str, sizeof (address_str)); gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL); } if (src->have_rtcp_from) { gst_netaddress_to_string (&src->rtcp_from, address_str, sizeof (address_str)); gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL); } if (internal) { /* our internal source */ if (is_sender) { /* if we are sending, report about how much we sent, other sources will * have a RB with info on reception. */ gst_structure_set (s, "octets-sent", G_TYPE_UINT64, src->stats.octets_sent, "packets-sent", G_TYPE_UINT64, src->stats.packets_sent, "bitrate", G_TYPE_UINT64, src->bitrate, NULL); } else { /* if we are not sending we have nothing more to report */ } } else { 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; /* other sources */ if (is_sender) { gboolean have_sr; GstClockTime time = 0; guint64 ntptime = 0; guint32 rtptime = 0; guint32 packet_count = 0; guint32 octet_count = 0; /* this source is sending to us, get the last SR. */ have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime, &packet_count, &octet_count); gst_structure_set (s, "octets-received", G_TYPE_UINT64, src->stats.octets_received, "packets-received", G_TYPE_UINT64, src->stats.packets_received, "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); } /* we might be sending to this SSRC so we report about how it is * receiving our data */ 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 #RTSPSource * * Get the SDES data as a GstStructure * * Returns: a GstStructure with SDES items for @src. */ GstStructure * rtp_source_get_sdes_struct (RTPSource * src) { GstStructure *s; gchar *str; s = gst_structure_new ("application/x-rtp-source-sdes", "ssrc", G_TYPE_UINT, (guint) src->ssrc, NULL); if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_CNAME))) { gst_structure_set (s, "cname", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NAME))) { gst_structure_set (s, "name", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_EMAIL))) { gst_structure_set (s, "email", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PHONE))) { gst_structure_set (s, "phone", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_LOC))) { gst_structure_set (s, "location", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_TOOL))) { gst_structure_set (s, "tool", G_TYPE_STRING, str, NULL); g_free (str); } if ((str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NOTE))) { gst_structure_set (s, "note", G_TYPE_STRING, str, NULL); g_free (str); } return s; } /** * rtp_source_set_sdes_struct: * @src: an #RTSPSource * @sdes: a #GstStructure with SDES info * * Set the SDES items from @sdes. */ void rtp_source_set_sdes_struct (RTPSource * src, const GstStructure * sdes) { const gchar *str; if (!gst_structure_has_name (sdes, "application/x-rtp-source-sdes")) return; if ((str = gst_structure_get_string (sdes, "cname"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_CNAME, str); } if ((str = gst_structure_get_string (sdes, "name"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_NAME, str); } if ((str = gst_structure_get_string (sdes, "email"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_EMAIL, str); } if ((str = gst_structure_get_string (sdes, "phone"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_PHONE, str); } if ((str = gst_structure_get_string (sdes, "location"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_LOC, str); } if ((str = gst_structure_get_string (sdes, "tool"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_TOOL, str); } if ((str = gst_structure_get_string (sdes, "note"))) { rtp_source_set_sdes_string (src, GST_RTCP_SDES_NOTE, str); } } 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; 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_take_boxed (value, rtp_source_get_sdes_struct (src)); break; case PROP_STATS: g_value_take_boxed (value, rtp_source_create_stats (src)); 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_received_bye: * @src: an #RTPSource * * Check if @src has receoved a BYE packet. * * Returns: %TRUE if @src has received a BYE packet. */ gboolean rtp_source_received_bye (RTPSource * src) { gboolean result; g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); result = src->received_bye; return result; } /** * rtp_source_get_bye_reason: * @src: an #RTPSource * * Get the BYE reason for @src. Check if the source receoved a BYE message first * with rtp_source_received_bye(). * * Returns: The BYE reason or NULL when no reason was given or the source did * not receive a BYE message yet. g_fee() 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 (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_sdes: * @src: an #RTPSource * @type: the type of the SDES item * @data: the SDES data * @len: the SDES length * * Store an SDES item of @type in @src. * * Returns: %FALSE if the SDES item was unchanged or @type is unknown. */ gboolean rtp_source_set_sdes (RTPSource * src, GstRTCPSDESType type, const guint8 * data, guint len) { guint8 *old; g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); if (type < 0 || type > GST_RTCP_SDES_PRIV) return FALSE; old = src->sdes[type]; /* lengths are the same, check if the data is the same */ if ((src->sdes_len[type] == len)) if (data != NULL && old != NULL && (memcmp (old, data, len) == 0)) return FALSE; /* NULL data, make sure we store 0 length or if no length is given, * take strlen */ if (data == NULL) len = 0; g_free (src->sdes[type]); src->sdes[type] = g_memdup (data, len); src->sdes_len[type] = len; return TRUE; } /** * rtp_source_set_sdes_string: * @src: an #RTPSource * @type: the type of the SDES item * @data: the SDES data * * Store an SDES item of @type in @src. This function is similar to * rtp_source_set_sdes() but takes a null-terminated string for convenience. * * Returns: %FALSE if the SDES item was unchanged or @type is unknown. */ gboolean rtp_source_set_sdes_string (RTPSource * src, GstRTCPSDESType type, const gchar * data) { guint len; gboolean result; if (data) len = strlen (data); else len = 0; result = rtp_source_set_sdes (src, type, (guint8 *) data, len); return result; } /** * rtp_source_get_sdes: * @src: an #RTPSource * @type: the type of the SDES item * @data: location to store the SDES data or NULL * @len: location to store the SDES length or NULL * * Get the SDES item of @type from @src. Note that @data does not always point * to a null-terminated string, use rtp_source_get_sdes_string() to retrieve a * null-terminated string instead. * * @data remains valid until the next call to rtp_source_set_sdes(). * * Returns: %TRUE if @type was valid and @data and @len contain valid * data. @data can be NULL when the item was unset. */ gboolean rtp_source_get_sdes (RTPSource * src, GstRTCPSDESType type, guint8 ** data, guint * len) { g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE); if (type < 0 || type > GST_RTCP_SDES_PRIV) return FALSE; if (data) *data = src->sdes[type]; if (len) *len = src->sdes_len[type]; return TRUE; } /** * rtp_source_get_sdes_string: * @src: an #RTPSource * @type: the type of the SDES item * * Get the SDES item of @type from @src. * * Returns: a null-terminated copy of the SDES item or NULL when @type was not * valid or the SDES item was unset. g_free() after usage. */ gchar * rtp_source_get_sdes_string (RTPSource * src, GstRTCPSDESType type) { gchar *result; g_return_val_if_fail (RTP_IS_SOURCE (src), NULL); if (type < 0 || type > GST_RTCP_SDES_PRIV) return NULL; result = g_strndup ((const gchar *) src->sdes[type], src->sdes_len[type]); return result; } /** * 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, GstNetAddress * address) { g_return_if_fail (RTP_IS_SOURCE (src)); src->have_rtp_from = TRUE; memcpy (&src->rtp_from, address, sizeof (GstNetAddress)); } /** * 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, GstNetAddress * address) { g_return_if_fail (RTP_IS_SOURCE (src)); src->have_rtcp_from = TRUE; memcpy (&src->rtcp_from, address, sizeof (GstNetAddress)); } 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) { guint64 ntpnstime; guint32 rtparrival, transit, rtptime; gint32 diff; gint clock_rate; guint8 pt; /* get arrival time */ if ((ntpnstime = arrival->ntpnstime) == GST_CLOCK_TIME_NONE) goto no_time; pt = gst_rtp_buffer_get_payload_type (buffer); 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 = gst_rtp_buffer_get_timestamp (buffer); /* 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 (ntpnstime, 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 time"); 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); } /** * 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; 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; seqnr = gst_rtp_buffer_get_seq (buffer); rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); 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->probation = RTP_DEFAULT_PROBATION; } udelta = seqnr - stats->max_seq; /* if we are still on probation, check seqnum */ if (src->probation) { guint16 expected; 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->probation--; src->stats.max_seq = seqnr; if (src->probation == 0) { GST_DEBUG ("probation done!"); init_seq (src, seqnr); } else { GstBuffer *q; GST_DEBUG ("probation %d: queue buffer", src->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 { GST_DEBUG ("probation: seqnr %d != expected %d", seqnr, expected); src->probation = RTP_DEFAULT_PROBATION; src->stats.max_seq = seqnr; goto done; } } 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"); } src->stats.octets_received += arrival->payload_len; src->stats.bytes_received += arrival->bytes; src->stats.packets_received++; /* the source that sent the packet must be a sender */ src->is_sender = TRUE; src->validated = TRUE; 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 */ bad_sequence: { GST_WARNING ("unacceptable seqnum received"); return GST_FLOW_OK; } } /** * rtp_source_process_bye: * @src: an #RTPSource * @reason: the reason for leaving * * Notify @src that a BYE packet has been received. This will make the source * inactive. */ void rtp_source_process_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 received_bye */ g_free (src->bye_reason); src->bye_reason = g_strdup (reason); src->received_bye = TRUE; } static GstBufferListItem set_ssrc (GstBuffer ** buffer, guint group, guint idx, RTPSource * src) { *buffer = gst_buffer_make_writable (*buffer); gst_rtp_buffer_set_ssrc (*buffer, src->ssrc); return GST_BUFFER_LIST_SKIP_GROUP; } /** * 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 * @ntpnstime: the NTP time when this buffer was captured in nanoseconds. This * is the buffer timestamp converted to NTP time. * * 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, guint64 ntpnstime) { GstFlowReturn result; guint len; guint32 rtptime; guint64 ext_rtptime; guint64 ntp_diff, rtp_diff; guint64 elapsed; GstBufferList *list = NULL; GstBuffer *buffer = NULL; guint packets; guint32 ssrc; 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, 0); if (!buffer) goto no_buffer; } else { buffer = GST_BUFFER_CAST (data); } rtp_source_update_caps (src, GST_BUFFER_CAPS (buffer)); /* we are a sender now */ src->is_sender = TRUE; if (is_list) { /* Each group makes up a network packet. */ packets = gst_buffer_list_n_groups (list); len = gst_rtp_buffer_list_get_payload_len (list); } else { packets = 1; len = gst_rtp_buffer_get_payload_len (buffer); } /* update stats for the SR */ src->stats.packets_sent += packets; src->stats.octets_sent += len; src->bytes_sent += len; if (src->prev_ntpnstime) { elapsed = ntpnstime - src->prev_ntpnstime; if (elapsed > (G_GINT64_CONSTANT (1) << 31)) { guint64 rate; rate = gst_util_uint64_scale (src->bytes_sent, elapsed, (G_GINT64_CONSTANT (1) << 29)); GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT ", rate %" G_GUINT64_FORMAT, elapsed, src->bytes_sent, rate); if (src->bitrate == 0) src->bitrate = rate; else src->bitrate = ((src->bitrate * 3) + rate) / 4; src->prev_ntpnstime = ntpnstime; src->bytes_sent = 0; } } else { GST_LOG ("Reset bitrate measurement"); src->prev_ntpnstime = ntpnstime; src->bitrate = 0; } if (is_list) { rtptime = gst_rtp_buffer_list_get_timestamp (list); } else { rtptime = gst_rtp_buffer_get_timestamp (buffer); } ext_rtptime = src->last_rtptime; ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime); GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", NTP %" GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (ntpnstime)); if (ext_rtptime > src->last_rtptime) { rtp_diff = ext_rtptime - src->last_rtptime; ntp_diff = ntpnstime - src->last_ntpnstime; /* 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 NTP %" GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (ntp_diff)); } /* we keep track of the last received RTP timestamp and the corresponding * NTP timestamp so that we can use this info when constructing SR reports */ src->last_rtptime = ext_rtptime; src->last_ntpnstime = ntpnstime; /* push packet */ if (!src->callbacks.push_rtp) goto no_callback; if (is_list) { ssrc = gst_rtp_buffer_list_get_ssrc (list); } else { ssrc = gst_rtp_buffer_get_ssrc (buffer); } if (ssrc != src->ssrc) { /* the SSRC of the packet is not correct, make a writable buffer and * update the SSRC. This could involve a complete copy of the packet when * it is not writable. Usually the payloader will use caps negotiation to * get the correct SSRC from the session manager before pushing anything. */ /* FIXME, we don't want to warn yet because we can't inform any payloader * of the changes SSRC yet because we don't implement pad-alloc. */ GST_LOG ("updating SSRC from %08x to %08x, fix the payloader", ssrc, src->ssrc); if (is_list) { list = gst_buffer_list_make_writable (list); gst_buffer_list_foreach (list, (GstBufferListFunc) set_ssrc, src); } else { set_ssrc (&buffer, 0, 0, src); } } 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 */ 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; } /** * rtp_source_process_rb: * @src: an #RTPSource * @time: 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, GstClockTime time, guint8 fractionlost, gint32 packetslost, guint32 exthighestseq, guint32 jitter, guint32 lsr, guint32 dlsr) { RTPReceiverReport *curr; gint curridx; guint32 ntp, A; 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; /* calculate round trip, round the time up */ ntp = ((gst_rtcp_unix_to_ntp (time) + 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 * @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. * * Returns: %TRUE on success. */ gboolean rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime, 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); /* use the sync params to interpolate the date->time member to rtptime. We * use the last sent timestamp and rtptime as reference points. We assume * that the slope of the rtptime vs timestamp curve is 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_ntpnstime %" GST_TIME_FORMAT ", last_rtptime %" G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_ntpnstime), t_rtp); if (src->clock_rate != -1) { /* get the diff with the SR time */ diff = GST_CLOCK_DIFF (src->last_ntpnstime, ntpnstime); /* 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 ("ntpnstime %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT, GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (diff)); t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND); } else { diff = -diff; GST_DEBUG ("ntpnstime %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT, GST_TIME_ARGS (ntpnstime), 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; }