2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 #include <gst/rtp/gstrtpbuffer.h>
22 #include <gst/rtp/gstrtcpbuffer.h>
24 #include "rtpsource.h"
26 GST_DEBUG_CATEGORY_STATIC (rtp_source_debug);
27 #define GST_CAT_DEFAULT rtp_source_debug
29 #define RTP_MAX_PROBATION_LEN 32
31 /* signals and args */
37 #define DEFAULT_SSRC 0
38 #define DEFAULT_IS_CSRC FALSE
39 #define DEFAULT_IS_VALIDATED FALSE
40 #define DEFAULT_IS_SENDER FALSE
41 #define DEFAULT_SDES NULL
42 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
56 /* GObject vmethods */
57 static void rtp_source_finalize (GObject * object);
58 static void rtp_source_set_property (GObject * object, guint prop_id,
59 const GValue * value, GParamSpec * pspec);
60 static void rtp_source_get_property (GObject * object, guint prop_id,
61 GValue * value, GParamSpec * pspec);
63 /* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
65 G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
68 rtp_source_class_init (RTPSourceClass * klass)
70 GObjectClass *gobject_class;
72 gobject_class = (GObjectClass *) klass;
74 gobject_class->finalize = rtp_source_finalize;
76 gobject_class->set_property = rtp_source_set_property;
77 gobject_class->get_property = rtp_source_get_property;
79 g_object_class_install_property (gobject_class, PROP_SSRC,
80 g_param_spec_uint ("ssrc", "SSRC",
81 "The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC,
82 G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
84 g_object_class_install_property (gobject_class, PROP_IS_CSRC,
85 g_param_spec_boolean ("is-csrc", "Is CSRC",
86 "If this SSRC is acting as a contributing source",
87 DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
89 g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
90 g_param_spec_boolean ("is-validated", "Is Validated",
91 "If this SSRC is validated", DEFAULT_IS_VALIDATED,
92 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
94 g_object_class_install_property (gobject_class, PROP_IS_SENDER,
95 g_param_spec_boolean ("is-sender", "Is Sender",
96 "If this SSRC is a sender", DEFAULT_IS_SENDER,
97 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
102 * The current SDES items of the source. Returns a structure with name
103 * application/x-rtp-source-sdes and may contain the following fields:
105 * 'cname' G_TYPE_STRING : The canonical name
106 * 'name' G_TYPE_STRING : The user name
107 * 'email' G_TYPE_STRING : The user's electronic mail address
108 * 'phone' G_TYPE_STRING : The user's phone number
109 * 'location' G_TYPE_STRING : The geographic user location
110 * 'tool' G_TYPE_STRING : The name of application or tool
111 * 'note' G_TYPE_STRING : A notice about the source
113 * other fields may be present and these represent private items in
114 * the SDES where the field name is the prefix.
116 g_object_class_install_property (gobject_class, PROP_SDES,
117 g_param_spec_boxed ("sdes", "SDES",
118 "The SDES information for this source",
119 GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
124 * The statistics of the source. This property returns a GstStructure with
125 * name application/x-rtp-source-stats with the following fields:
127 * "ssrc" G_TYPE_UINT the SSRC of this source
128 * "internal" G_TYPE_BOOLEAN this source is a source of the session
129 * "validated" G_TYPE_BOOLEAN the source is validated
130 * "received-bye" G_TYPE_BOOLEAN we received a BYE from this source
131 * "is-csrc" G_TYPE_BOOLEAN this source was found as CSRC
132 * "is-sender" G_TYPE_BOOLEAN this source is a sender
133 * "seqnum-base" G_TYPE_INT first seqnum if known
134 * "clock-rate" G_TYPE_INT the clock rate of the media
136 * The following two fields are only present when known.
138 * "rtp-from" G_TYPE_STRING where we received the last RTP packet from
139 * "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from
141 * The following fields make sense for internal sources and will only increase
142 * when "is-sender" is TRUE:
144 * "octets-sent" G_TYPE_UINT64 number of bytes we sent
145 * "packets-sent" G_TYPE_UINT64 number of packets we sent
147 * The following fields make sense for non-internal sources and will only
148 * increase when "is-sender" is TRUE.
150 * "octets-received" G_TYPE_UINT64 total number of bytes received
151 * "packets-received" G_TYPE_UINT64 total number of packets received
153 * Following fields are updated when "is-sender" is TRUE.
155 * "bitrate" G_TYPE_UINT64 bitrate in bits per second
156 * "jitter" G_TYPE_UINT estimated jitter (in clock rate units)
157 * "packets-lost" G_TYPE_INT estimated amount of packets lost
159 * The last SR report this source sent. This only updates when "is-sender" is
162 * "have-sr" G_TYPE_BOOLEAN the source has sent SR
163 * "sr-ntptime" G_TYPE_UINT64 NTP time of SR (in NTP Timestamp Format, 32.32 fixed point)
164 * "sr-rtptime" G_TYPE_UINT RTP time of SR (in clock rate units)
165 * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR
166 * "sr-packet-count" G_TYPE_UINT the number of packets in the SR
168 * The following fields are only present for non-internal sources and
169 * represent the content of the last RB packet that was sent to this source.
170 * These values are only updated when the source is sending.
172 * "sent-rb" G_TYPE_BOOLEAN we have sent an RB
173 * "sent-rb-fractionlost" G_TYPE_UINT calculated lost fraction
174 * "sent-rb-packetslost" G_TYPE_INT lost packets
175 * "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum
176 * "sent-rb-jitter" G_TYPE_UINT jitter (in clock rate units)
177 * "sent-rb-lsr" G_TYPE_UINT last SR time (in NTP Short Format, 16.16 fixed point)
178 * "sent-rb-dlsr" G_TYPE_UINT delay since last SR (in NTP Short Format, 16.16 fixed point)
180 * The following fields are only present for non-internal sources and
181 * represents the last RB that this source sent. This is only updated
182 * when the source is receiving data and sending RB blocks.
184 * "have-rb" G_TYPE_BOOLEAN the source has sent RB
185 * "rb-fractionlost" G_TYPE_UINT lost fraction
186 * "rb-packetslost" G_TYPE_INT lost packets
187 * "rb-exthighestseq" G_TYPE_UINT highest received seqnum
188 * "rb-jitter" G_TYPE_UINT reception jitter (in clock rate units)
189 * "rb-lsr" G_TYPE_UINT last SR time (in NTP Short Format, 16.16 fixed point)
190 * "rb-dlsr" G_TYPE_UINT delay since last SR (in NTP Short Format, 16.16 fixed point)
192 * The round trip of this source is calculated from the last RB
193 * values and the reception time of the last RB packet. It is only present for
194 * non-internal sources.
196 * "rb-round-trip" G_TYPE_UINT the round-trip time (in NTP Short Format, 16.16 fixed point)
198 * In all fields above, NTP times are in the appropriate 32-bit or 64-bit fixed-point format
199 * starting from January 1, 1970 (except for timespans), and RTP times are in clock rate units
200 * (i.e. clock rate = 1 second) starting from a random offset.
202 g_object_class_install_property (gobject_class, PROP_STATS,
203 g_param_spec_boxed ("stats", "Stats",
204 "The stats of this source", GST_TYPE_STRUCTURE,
205 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
207 g_object_class_install_property (gobject_class, PROP_PROBATION,
208 g_param_spec_uint ("probation", "Number of probations",
209 "Consecutive packet sequence numbers to accept the source",
210 0, G_MAXUINT, DEFAULT_PROBATION,
211 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
213 GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
218 * @src: an #RTPSource
220 * Reset the stats of @src.
223 rtp_source_reset (RTPSource * src)
225 src->marked_bye = FALSE;
227 g_free (src->bye_reason);
228 src->bye_reason = NULL;
229 src->sent_bye = FALSE;
230 g_hash_table_remove_all (src->reported_in_sr_of);
232 src->stats.cycles = -1;
233 src->stats.jitter = 0;
234 src->stats.transit = -1;
235 src->stats.curr_sr = 0;
236 src->stats.sr[0].is_valid = FALSE;
237 src->stats.curr_rr = 0;
238 src->stats.rr[0].is_valid = FALSE;
239 src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
240 src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
241 src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
242 src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
243 g_array_set_size (src->nacks, 0);
245 src->stats.sent_pli_count = 0;
246 src->stats.sent_fir_count = 0;
250 rtp_source_init (RTPSource * src)
252 /* sources are initialy on probation until we receive enough valid RTP
253 * packets or a valid RTCP packet */
254 src->validated = FALSE;
255 src->internal = FALSE;
256 src->probation = DEFAULT_PROBATION;
257 src->curr_probation = src->probation;
258 src->closing = FALSE;
260 src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
263 src->clock_rate = -1;
264 src->packets = g_queue_new ();
265 src->seqnum_offset = -1;
266 src->last_rtptime = -1;
268 src->retained_feedback = g_queue_new ();
269 src->nacks = g_array_new (FALSE, FALSE, sizeof (guint32));
271 src->reported_in_sr_of = g_hash_table_new (g_direct_hash, g_direct_equal);
273 rtp_source_reset (src);
277 rtp_conflicting_address_free (RTPConflictingAddress * addr)
279 g_object_unref (addr->address);
280 g_slice_free (RTPConflictingAddress, addr);
284 rtp_source_finalize (GObject * object)
288 src = RTP_SOURCE_CAST (object);
290 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
291 g_queue_free (src->packets);
293 gst_structure_free (src->sdes);
295 g_free (src->bye_reason);
297 gst_caps_replace (&src->caps, NULL);
299 g_list_free_full (src->conflicting_addresses,
300 (GDestroyNotify) rtp_conflicting_address_free);
301 g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL);
302 g_queue_free (src->retained_feedback);
304 g_array_free (src->nacks, TRUE);
307 g_object_unref (src->rtp_from);
309 g_object_unref (src->rtcp_from);
311 g_hash_table_unref (src->reported_in_sr_of);
313 G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
316 static GstStructure *
317 rtp_source_create_stats (RTPSource * src)
320 gboolean is_sender = src->is_sender;
321 gboolean internal = src->internal;
324 guint8 fractionlost = 0;
325 gint32 packetslost = 0;
326 guint32 exthighestseq = 0;
330 guint32 round_trip = 0;
332 GstClockTime time = 0;
335 guint32 packet_count = 0;
336 guint32 octet_count = 0;
339 /* common data for all types of sources */
340 s = gst_structure_new ("application/x-rtp-source-stats",
341 "ssrc", G_TYPE_UINT, (guint) src->ssrc,
342 "internal", G_TYPE_BOOLEAN, internal,
343 "validated", G_TYPE_BOOLEAN, src->validated,
344 "received-bye", G_TYPE_BOOLEAN, src->marked_bye,
345 "is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
346 "is-sender", G_TYPE_BOOLEAN, is_sender,
347 "seqnum-base", G_TYPE_INT, src->seqnum_offset,
348 "clock-rate", G_TYPE_INT, src->clock_rate, NULL);
350 /* add address and port */
352 address_str = __g_socket_address_to_string (src->rtp_from);
353 gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
354 g_free (address_str);
356 if (src->rtcp_from) {
357 address_str = __g_socket_address_to_string (src->rtcp_from);
358 gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
359 g_free (address_str);
362 gst_structure_set (s,
363 "octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
364 "packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
365 "octets-received", G_TYPE_UINT64, src->stats.octets_received,
366 "packets-received", G_TYPE_UINT64, src->stats.packets_received,
367 "bitrate", G_TYPE_UINT64, src->bitrate,
368 "packets-lost", G_TYPE_INT,
369 (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
370 (guint) (src->stats.jitter >> 4),
371 "sent-pli-count", G_TYPE_UINT, src->stats.sent_pli_count,
372 "recv-pli-count", G_TYPE_UINT, src->stats.recv_pli_count,
373 "sent-fir-count", G_TYPE_UINT, src->stats.sent_fir_count,
374 "recv-fir-count", G_TYPE_UINT, src->stats.recv_fir_count, NULL);
376 /* get the last SR. */
377 have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
378 &packet_count, &octet_count);
379 gst_structure_set (s,
380 "have-sr", G_TYPE_BOOLEAN, have_sr,
381 "sr-ntptime", G_TYPE_UINT64, ntptime,
382 "sr-rtptime", G_TYPE_UINT, (guint) rtptime,
383 "sr-octet-count", G_TYPE_UINT, (guint) octet_count,
384 "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
387 /* get the last RB we sent */
388 gst_structure_set (s,
389 "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
390 "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
391 "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
392 "sent-rb-exthighestseq", G_TYPE_UINT,
393 (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
394 (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
395 (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
396 (guint) src->last_rr.dlsr, NULL);
398 /* get the last RB */
399 have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
400 &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
402 gst_structure_set (s,
403 "have-rb", G_TYPE_BOOLEAN, have_rb,
404 "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
405 "rb-packetslost", G_TYPE_INT, (gint) packetslost,
406 "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
407 "rb-jitter", G_TYPE_UINT, (guint) jitter,
408 "rb-lsr", G_TYPE_UINT, (guint) lsr,
409 "rb-dlsr", G_TYPE_UINT, (guint) dlsr,
410 "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
417 * rtp_source_get_sdes_struct:
418 * @src: an #RTPSource
420 * Get the SDES from @src. See the SDES property for more details.
422 * Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
423 * valid until the SDES items of @src are modified.
426 rtp_source_get_sdes_struct (RTPSource * src)
428 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
434 sdes_struct_compare_func (GQuark field_id, const GValue * value,
440 old = GST_STRUCTURE (user_data);
441 field = g_quark_to_string (field_id);
443 if (!gst_structure_has_field (old, field))
446 g_assert (G_VALUE_HOLDS_STRING (value));
448 return strcmp (g_value_get_string (value), gst_structure_get_string (old,
453 * rtp_source_set_sdes_struct:
454 * @src: an #RTPSource
455 * @sdes: the SDES structure
457 * Store the @sdes in @src. @sdes must be a structure of type
458 * "application/x-rtp-source-sdes", see the SDES property for more details.
460 * This function takes ownership of @sdes.
462 * Returns: %FALSE if the SDES was unchanged.
465 rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
469 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
470 g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
471 "application/x-rtp-source-sdes") == 0, FALSE);
473 changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
476 gst_structure_free (src->sdes);
479 gst_structure_free (sdes);
485 rtp_source_set_property (GObject * object, guint prop_id,
486 const GValue * value, GParamSpec * pspec)
490 src = RTP_SOURCE (object);
494 src->ssrc = g_value_get_uint (value);
497 src->probation = g_value_get_uint (value);
500 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
506 rtp_source_get_property (GObject * object, guint prop_id,
507 GValue * value, GParamSpec * pspec)
511 src = RTP_SOURCE (object);
515 g_value_set_uint (value, rtp_source_get_ssrc (src));
518 g_value_set_boolean (value, rtp_source_is_as_csrc (src));
520 case PROP_IS_VALIDATED:
521 g_value_set_boolean (value, rtp_source_is_validated (src));
524 g_value_set_boolean (value, rtp_source_is_sender (src));
527 g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
530 g_value_take_boxed (value, rtp_source_create_stats (src));
533 g_value_set_uint (value, src->probation);
536 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
545 * Create a #RTPSource with @ssrc.
547 * Returns: a new #RTPSource. Use g_object_unref() after usage.
550 rtp_source_new (guint32 ssrc)
554 src = g_object_new (RTP_TYPE_SOURCE, NULL);
561 * rtp_source_set_callbacks:
562 * @src: an #RTPSource
563 * @cb: callback functions
564 * @user_data: user data
566 * Set the callbacks for the source.
569 rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
572 g_return_if_fail (RTP_IS_SOURCE (src));
574 src->callbacks.push_rtp = cb->push_rtp;
575 src->callbacks.clock_rate = cb->clock_rate;
576 src->user_data = user_data;
580 * rtp_source_get_ssrc:
581 * @src: an #RTPSource
583 * Get the SSRC of @source.
585 * Returns: the SSRC of src.
588 rtp_source_get_ssrc (RTPSource * src)
592 g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
600 * rtp_source_set_as_csrc:
601 * @src: an #RTPSource
603 * Configure @src as a CSRC, this will also validate @src.
606 rtp_source_set_as_csrc (RTPSource * src)
608 g_return_if_fail (RTP_IS_SOURCE (src));
610 src->validated = TRUE;
615 * rtp_source_is_as_csrc:
616 * @src: an #RTPSource
618 * Check if @src is a contributing source.
620 * Returns: %TRUE if @src is acting as a contributing source.
623 rtp_source_is_as_csrc (RTPSource * src)
627 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
629 result = src->is_csrc;
635 * rtp_source_is_active:
636 * @src: an #RTPSource
638 * Check if @src is an active source. A source is active if it has been
639 * validated and has not yet received a BYE packet
641 * Returns: %TRUE if @src is an qactive source.
644 rtp_source_is_active (RTPSource * src)
648 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
650 result = RTP_SOURCE_IS_ACTIVE (src);
656 * rtp_source_is_validated:
657 * @src: an #RTPSource
659 * Check if @src is a validated source.
661 * Returns: %TRUE if @src is a validated source.
664 rtp_source_is_validated (RTPSource * src)
668 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
670 result = src->validated;
676 * rtp_source_is_sender:
677 * @src: an #RTPSource
679 * Check if @src is a sending source.
681 * Returns: %TRUE if @src is a sending source.
684 rtp_source_is_sender (RTPSource * src)
688 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
690 result = RTP_SOURCE_IS_SENDER (src);
696 * rtp_source_is_marked_bye:
697 * @src: an #RTPSource
699 * Check if @src is marked as leaving the session with a BYE packet.
701 * Returns: %TRUE if @src has been marked BYE.
704 rtp_source_is_marked_bye (RTPSource * src)
708 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
710 result = RTP_SOURCE_IS_MARKED_BYE (src);
717 * rtp_source_get_bye_reason:
718 * @src: an #RTPSource
720 * Get the BYE reason for @src. Check if the source is marked as leaving the
721 * session with a BYE message first with rtp_source_is_marked_bye().
723 * Returns: The BYE reason or NULL when no reason was given or the source was
724 * not marked BYE yet. g_free() after usage.
727 rtp_source_get_bye_reason (RTPSource * src)
731 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
733 result = g_strdup (src->bye_reason);
739 * rtp_source_update_caps:
740 * @src: an #RTPSource
743 * Parse @caps and store all relevant information in @source.
746 rtp_source_update_caps (RTPSource * src, GstCaps * caps)
753 /* nothing changed, return */
754 if (caps == NULL || src->caps == caps)
757 s = gst_caps_get_structure (caps, 0);
759 rtx = (gst_structure_get_uint (s, "rtx-ssrc", &val) && val == src->ssrc);
761 if (gst_structure_get_int (s, rtx ? "rtx-payload" : "payload", &ival))
766 GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", src->payload);
768 if (gst_structure_get_int (s, "clock-rate", &ival))
769 src->clock_rate = ival;
771 src->clock_rate = -1;
773 GST_DEBUG ("got clock-rate %d", src->clock_rate);
775 if (gst_structure_get_uint (s, rtx ? "rtx-seqnum-offset" : "seqnum-offset",
777 src->seqnum_offset = val;
779 src->seqnum_offset = -1;
781 GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "",
784 gst_caps_replace (&src->caps, caps);
788 * rtp_source_set_rtp_from:
789 * @src: an #RTPSource
790 * @address: the RTP address to set
792 * Set that @src is receiving RTP packets from @address. This is used for
793 * collistion checking.
796 rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
798 g_return_if_fail (RTP_IS_SOURCE (src));
801 g_object_unref (src->rtp_from);
802 src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
806 * rtp_source_set_rtcp_from:
807 * @src: an #RTPSource
808 * @address: the RTCP address to set
810 * Set that @src is receiving RTCP packets from @address. This is used for
811 * collistion checking.
814 rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
816 g_return_if_fail (RTP_IS_SOURCE (src));
819 g_object_unref (src->rtcp_from);
820 src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
824 push_packet (RTPSource * src, GstBuffer * buffer)
826 GstFlowReturn ret = GST_FLOW_OK;
828 /* push queued packets first if any */
829 while (!g_queue_is_empty (src->packets)) {
830 GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
832 GST_LOG ("pushing queued packet");
833 if (src->callbacks.push_rtp)
834 src->callbacks.push_rtp (src, buffer, src->user_data);
836 gst_buffer_unref (buffer);
838 GST_LOG ("pushing new packet");
840 if (src->callbacks.push_rtp)
841 ret = src->callbacks.push_rtp (src, buffer, src->user_data);
843 gst_buffer_unref (buffer);
849 get_clock_rate (RTPSource * src, guint8 payload)
851 if (src->payload == -1) {
852 /* first payload received, nothing was in the caps, lock on to this payload */
853 src->payload = payload;
854 GST_DEBUG ("first payload %d", payload);
855 } else if (payload != src->payload) {
856 /* we have a different payload than before, reset the clock-rate */
857 GST_DEBUG ("new payload %d", payload);
858 src->payload = payload;
859 src->clock_rate = -1;
860 src->stats.transit = -1;
863 if (src->clock_rate == -1) {
864 gint clock_rate = -1;
866 if (src->callbacks.clock_rate)
867 clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
869 GST_DEBUG ("got clock-rate %d", clock_rate);
871 src->clock_rate = clock_rate;
873 return src->clock_rate;
876 /* Jitter is the variation in the delay of received packets in a flow. It is
877 * measured by comparing the interval when RTP packets were sent to the interval
878 * at which they were received. For instance, if packet #1 and packet #2 leave
879 * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
882 calculate_jitter (RTPSource * src, RTPPacketInfo * pinfo)
884 GstClockTime running_time;
885 guint32 rtparrival, transit, rtptime;
890 /* get arrival time */
891 if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
896 GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
899 if ((clock_rate = get_clock_rate (src, pt)) == -1)
902 rtptime = pinfo->rtptime;
904 /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
905 * care about the absolute value, just the difference. */
906 rtparrival = gst_util_uint64_scale_int (running_time, clock_rate, GST_SECOND);
908 /* transit time is difference with RTP timestamp */
909 transit = rtparrival - rtptime;
911 /* get ABS diff with previous transit time */
912 if (src->stats.transit != -1) {
913 if (transit > src->stats.transit)
914 diff = transit - src->stats.transit;
916 diff = src->stats.transit - transit;
920 src->stats.transit = transit;
922 /* update jitter, the value we store is scaled up so we can keep precision. */
923 src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
925 src->stats.prev_rtptime = src->stats.last_rtptime;
926 src->stats.last_rtptime = rtparrival;
928 GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
929 rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
936 GST_WARNING ("cannot get current running_time");
941 GST_WARNING ("cannot get clock-rate for pt %d", pt);
947 init_seq (RTPSource * src, guint16 seq)
949 src->stats.base_seq = seq;
950 src->stats.max_seq = seq;
951 src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
952 src->stats.cycles = 0;
953 src->stats.packets_received = 0;
954 src->stats.octets_received = 0;
955 src->stats.bytes_received = 0;
956 src->stats.prev_received = 0;
957 src->stats.prev_expected = 0;
958 src->stats.recv_pli_count = 0;
959 src->stats.recv_fir_count = 0;
961 GST_DEBUG ("base_seq %d", seq);
964 #define BITRATE_INTERVAL (2 * GST_SECOND)
967 do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
968 guint64 * bytes_handled)
972 if (src->prev_rtime) {
973 elapsed = running_time - src->prev_rtime;
975 if (elapsed > BITRATE_INTERVAL) {
978 rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
980 GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
981 ", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
983 if (src->bitrate == 0)
986 src->bitrate = ((src->bitrate * 3) + rate) / 4;
988 src->prev_rtime = running_time;
992 GST_LOG ("Reset bitrate measurement");
993 src->prev_rtime = running_time;
999 update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo)
1001 guint16 seqnr, expected;
1002 RTPSourceStats *stats;
1005 stats = &src->stats;
1007 seqnr = pinfo->seqnum;
1009 if (stats->cycles == -1) {
1010 GST_DEBUG ("received first packet");
1011 /* first time we heard of this source */
1012 init_seq (src, seqnr);
1013 src->stats.max_seq = seqnr - 1;
1014 src->curr_probation = src->probation;
1017 expected = src->stats.max_seq + 1;
1018 delta = gst_rtp_buffer_compare_seqnum (expected, seqnr);
1020 /* if we are still on probation, check seqnum */
1021 if (src->curr_probation) {
1022 /* when in probation, we require consecutive seqnums */
1024 /* expected packet */
1025 GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
1026 src->curr_probation--;
1027 if (seqnr < stats->max_seq) {
1028 /* sequence number wrapped - count another 64K cycle. */
1029 stats->cycles += RTP_SEQ_MOD;
1031 src->stats.max_seq = seqnr;
1033 if (src->curr_probation == 0) {
1034 GST_DEBUG ("probation done!");
1035 init_seq (src, seqnr);
1039 GST_DEBUG ("probation %d: queue packet", src->curr_probation);
1040 /* when still in probation, keep packets in a list. */
1041 g_queue_push_tail (src->packets, pinfo->data);
1043 /* remove packets from queue if there are too many */
1044 while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
1045 q = g_queue_pop_head (src->packets);
1046 gst_buffer_unref (q);
1051 /* unexpected seqnum in probation */
1052 goto probation_seqnum;
1054 } else if (delta >= 0 && delta < RTP_MAX_DROPOUT) {
1055 /* Clear bad packets */
1056 stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
1057 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1058 g_queue_clear (src->packets);
1060 /* in order, with permissible gap */
1061 if (seqnr < stats->max_seq) {
1062 /* sequence number wrapped - count another 64K cycle. */
1063 stats->cycles += RTP_SEQ_MOD;
1065 stats->max_seq = seqnr;
1066 } else if (delta < -RTP_MAX_MISORDER || delta >= RTP_MAX_DROPOUT) {
1067 /* the sequence number made a very large jump */
1068 if (seqnr == stats->bad_seq && src->packets->head) {
1069 /* two sequential packets -- assume that the other side
1070 * restarted without telling us so just re-sync
1071 * (i.e., pretend this was the first packet). */
1072 init_seq (src, seqnr);
1074 /* unacceptable jump */
1075 stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
1076 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1077 g_queue_clear (src->packets);
1078 g_queue_push_tail (src->packets, pinfo->data);
1082 } else { /* delta < 0 && delta >= -RTP_MAX_MISORDER */
1083 /* Clear bad packets */
1084 stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
1085 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1086 g_queue_clear (src->packets);
1088 /* duplicate or reordered packet, will be filtered by jitterbuffer. */
1089 GST_WARNING ("duplicate or reordered packet (seqnr %u, expected %u)", seqnr,
1093 src->stats.octets_received += pinfo->payload_len;
1094 src->stats.bytes_received += pinfo->bytes;
1095 src->stats.packets_received++;
1096 /* for the bitrate estimation */
1097 src->bytes_received += pinfo->payload_len;
1099 GST_LOG ("seq %u, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
1100 seqnr, src->stats.packets_received, src->stats.octets_received);
1111 GST_WARNING ("unacceptable seqnum received");
1116 GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
1117 src->curr_probation = src->probation;
1118 src->stats.max_seq = seqnr;
1124 * rtp_source_process_rtp:
1125 * @src: an #RTPSource
1126 * @pinfo: an #RTPPacketInfo
1128 * Let @src handle the incomming RTP packet described in @pinfo.
1130 * Returns: a #GstFlowReturn.
1133 rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1135 GstFlowReturn result;
1137 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1138 g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
1140 if (!update_receiver_stats (src, pinfo))
1143 /* the source that sent the packet must be a sender */
1144 src->is_sender = TRUE;
1145 src->validated = TRUE;
1147 do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
1149 /* calculate jitter for the stats */
1150 calculate_jitter (src, pinfo);
1152 /* we're ready to push the RTP packet now */
1153 result = push_packet (src, pinfo->data);
1160 * rtp_source_mark_bye:
1161 * @src: an #RTPSource
1162 * @reason: the reason for leaving
1164 * Mark @src in the BYE state. This can happen when the source wants to
1165 * leave the sesssion or when a BYE packets has been received.
1167 * This will make the source inactive.
1170 rtp_source_mark_bye (RTPSource * src, const gchar * reason)
1172 g_return_if_fail (RTP_IS_SOURCE (src));
1174 GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
1175 GST_STR_NULL (reason));
1177 /* copy the reason and mark as bye */
1178 g_free (src->bye_reason);
1179 src->bye_reason = g_strdup (reason);
1180 src->marked_bye = TRUE;
1184 * rtp_source_send_rtp:
1185 * @src: an #RTPSource
1186 * @data: an RTP buffer or a list of RTP buffers
1187 * @is_list: if @data is a buffer or list
1188 * @running_time: the running time of @data
1190 * Send @data (an RTP buffer or list of buffers) originating from @src.
1191 * This will make @src a sender. This function takes ownership of @data and
1192 * modifies the SSRC in the RTP packet to that of @src when needed.
1194 * Returns: a #GstFlowReturn.
1197 rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1199 GstFlowReturn result;
1200 GstClockTime running_time;
1202 guint64 ext_rtptime;
1203 guint64 rt_diff, rtp_diff;
1205 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1207 /* we are a sender now */
1208 src->is_sender = TRUE;
1210 /* update stats for the SR */
1211 src->stats.packets_sent += pinfo->packets;
1212 src->stats.octets_sent += pinfo->payload_len;
1213 src->bytes_sent += pinfo->payload_len;
1214 /* we are also a receiver of our packets */
1215 update_receiver_stats (src, pinfo);
1217 running_time = pinfo->running_time;
1219 do_bitrate_estimation (src, running_time, &src->bytes_sent);
1221 rtptime = pinfo->rtptime;
1223 ext_rtptime = src->last_rtptime;
1224 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
1226 GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
1227 GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
1229 if (ext_rtptime > src->last_rtptime) {
1230 rtp_diff = ext_rtptime - src->last_rtptime;
1231 rt_diff = running_time - src->last_rtime;
1233 /* calc the diff so we can detect drift at the sender. This can also be used
1234 * to guestimate the clock rate if the NTP time is locked to the RTP
1235 * timestamps (as is the case when the capture device is providing the clock). */
1236 GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
1237 GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
1240 /* we keep track of the last received RTP timestamp and the corresponding
1241 * buffer running_time so that we can use this info when constructing SR reports */
1242 src->last_rtime = running_time;
1243 src->last_rtptime = ext_rtptime;
1246 if (!src->callbacks.push_rtp)
1249 GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT,
1250 pinfo->is_list ? "list" : "packet", src->stats.packets_sent);
1252 result = src->callbacks.push_rtp (src, pinfo->data, src->user_data);
1260 GST_WARNING ("no callback installed, dropping packet");
1266 * rtp_source_process_sr:
1267 * @src: an #RTPSource
1268 * @time: time of packet arrival
1269 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1270 * @rtptime: the RTP time (in clock rate units)
1271 * @packet_count: the packet count
1272 * @octet_count: the octet count
1274 * Update the sender report in @src.
1277 rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
1278 guint32 rtptime, guint32 packet_count, guint32 octet_count)
1280 RTPSenderReport *curr;
1283 g_return_if_fail (RTP_IS_SOURCE (src));
1285 GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
1286 ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
1287 (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
1288 packet_count, octet_count);
1290 curridx = src->stats.curr_sr ^ 1;
1291 curr = &src->stats.sr[curridx];
1293 /* this is a sender now */
1294 src->is_sender = TRUE;
1296 /* update current */
1297 curr->is_valid = TRUE;
1298 curr->ntptime = ntptime;
1299 curr->rtptime = rtptime;
1300 curr->packet_count = packet_count;
1301 curr->octet_count = octet_count;
1305 src->stats.curr_sr = curridx;
1307 src->stats.prev_rtcptime = src->stats.last_rtcptime;
1308 src->stats.last_rtcptime = time;
1312 * rtp_source_process_rb:
1313 * @src: an #RTPSource
1314 * @ntpnstime: the current time in nanoseconds since 1970
1315 * @fractionlost: fraction lost since last SR/RR
1316 * @packetslost: the cumulative number of packets lost
1317 * @exthighestseq: the extended last sequence number received
1318 * @jitter: the interarrival jitter (in clock rate units)
1319 * @lsr: the time of the last SR packet on this source
1320 * (in NTP Short Format, 16.16 fixed point)
1321 * @dlsr: the delay since the last SR packet
1322 * (in NTP Short Format, 16.16 fixed point)
1324 * Update the report block in @src.
1327 rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
1328 guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
1329 guint32 jitter, guint32 lsr, guint32 dlsr)
1331 RTPReceiverReport *curr;
1336 g_return_if_fail (RTP_IS_SOURCE (src));
1338 GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
1339 ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
1340 src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
1341 lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
1343 curridx = src->stats.curr_rr ^ 1;
1344 curr = &src->stats.rr[curridx];
1346 /* update current */
1347 curr->is_valid = TRUE;
1348 curr->fractionlost = fractionlost;
1349 curr->packetslost = packetslost;
1350 curr->exthighestseq = exthighestseq;
1351 curr->jitter = jitter;
1355 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1356 f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1357 /* calculate round trip, round the time up */
1358 ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
1361 if (A > 0 && ntp > A)
1365 curr->round_trip = A;
1367 GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
1368 A >> 16, A & 0xffff);
1371 src->stats.curr_rr = curridx;
1375 * rtp_source_get_new_sr:
1376 * @src: an #RTPSource
1377 * @ntpnstime: the current time in nanoseconds since 1970
1378 * @running_time: the current running_time of the pipeline
1379 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1380 * @rtptime: the RTP time corresponding to @ntptime (in clock rate units)
1381 * @packet_count: the packet count
1382 * @octet_count: the octet count
1384 * Get new values to put into a new SR report from this source.
1386 * @running_time and @ntpnstime are captured at the same time and represent the
1387 * running time of the pipeline clock and the absolute current system time in
1388 * nanoseconds respectively. Together with the last running_time and RTP timestamp
1389 * we have observed in the source, we can generate @ntptime and @rtptime for an SR
1390 * packet. @ntptime is basically the fixed point representation of @ntpnstime
1391 * and @rtptime the associated RTP timestamp.
1393 * Returns: %TRUE on success.
1396 rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
1397 GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
1398 guint32 * packet_count, guint32 * octet_count)
1401 guint64 t_current_ntp;
1402 GstClockTimeDiff diff;
1404 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1406 /* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
1407 * and an NTP time, we can scale the RTP timestamps so that they match the
1408 * given NTP time. for scaling, we assume that the slope of the rtptime vs
1409 * running_time vs ntptime curve is close to 1, which is certainly
1410 * sufficient for the frequency at which we report SR and the rate we send
1411 * out RTP packets. */
1412 t_rtp = src->last_rtptime;
1414 GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
1415 G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
1417 if (src->clock_rate != -1) {
1418 /* get the diff between the clock running_time and the buffer running_time.
1419 * This is the elapsed time, as measured against the pipeline clock, between
1420 * when the rtp timestamp was observed and the current running_time.
1422 * We need to apply this diff to the RTP timestamp to get the RTP timestamp
1423 * for the given ntpnstime. */
1424 diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
1426 /* now translate the diff to RTP time, handle positive and negative cases.
1427 * If there is no diff, we already set rtptime correctly above. */
1429 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
1430 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1431 t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1434 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
1435 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1436 t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1439 GST_WARNING ("no clock-rate, cannot interpolate rtp time");
1442 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1443 t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1445 GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
1446 (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
1450 *ntptime = t_current_ntp;
1454 *packet_count = src->stats.packets_sent;
1456 *octet_count = src->stats.octets_sent;
1462 * rtp_source_get_new_rb:
1463 * @src: an #RTPSource
1464 * @time: the current time of the system clock
1465 * @fractionlost: fraction lost since last SR/RR
1466 * @packetslost: the cumulative number of packets lost
1467 * @exthighestseq: the extended last sequence number received
1468 * @jitter: the interarrival jitter (in clock rate units)
1469 * @lsr: the time of the last SR packet on this source
1470 * in NTP Short Format (16.16 fixed point)
1471 * @dlsr: the delay since the last SR packet
1472 * in NTP Short Format (16.16 fixed point)
1474 * Get new values to put into a new report block from this source.
1476 * Returns: %TRUE on success.
1479 rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
1480 guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
1481 guint32 * jitter, guint32 * lsr, guint32 * dlsr)
1483 RTPSourceStats *stats;
1484 guint64 extended_max, expected;
1485 guint64 expected_interval, received_interval, ntptime;
1486 gint64 lost, lost_interval;
1487 guint32 fraction, LSR, DLSR;
1488 GstClockTime sr_time;
1490 stats = &src->stats;
1492 extended_max = stats->cycles + stats->max_seq;
1493 expected = extended_max - stats->base_seq + 1;
1495 GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
1496 ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
1497 extended_max, expected, stats->packets_received, stats->base_seq);
1499 lost = expected - stats->packets_received;
1500 lost = CLAMP (lost, -0x800000, 0x7fffff);
1502 expected_interval = expected - stats->prev_expected;
1503 stats->prev_expected = expected;
1504 received_interval = stats->packets_received - stats->prev_received;
1505 stats->prev_received = stats->packets_received;
1507 lost_interval = expected_interval - received_interval;
1509 if (expected_interval == 0 || lost_interval <= 0)
1512 fraction = (lost_interval << 8) / expected_interval;
1514 GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
1515 /* we scaled the jitter up for additional precision */
1516 GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
1517 ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
1518 extended_max, stats->jitter >> 4);
1520 if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
1523 /* LSR is middle 32 bits of the last ntptime */
1524 LSR = (ntptime >> 16) & 0xffffffff;
1525 diff = time - sr_time;
1526 GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
1527 /* DLSR, delay since last SR is expressed in 1/65536 second units */
1528 DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
1530 /* No valid SR received, LSR/DLSR are set to 0 then */
1531 GST_DEBUG ("no valid SR received");
1535 GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
1536 DLSR >> 16, DLSR & 0xffff);
1539 *fractionlost = fraction;
1541 *packetslost = lost;
1543 *exthighestseq = extended_max;
1545 *jitter = stats->jitter >> 4;
1555 * rtp_source_get_last_sr:
1556 * @src: an #RTPSource
1557 * @time: time of packet arrival
1558 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1559 * @rtptime: the RTP time (in clock rate units)
1560 * @packet_count: the packet count
1561 * @octet_count: the octet count
1563 * Get the values of the last sender report as set with rtp_source_process_sr().
1565 * Returns: %TRUE if there was a valid SR report.
1568 rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
1569 guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
1571 RTPSenderReport *curr;
1573 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1575 curr = &src->stats.sr[src->stats.curr_sr];
1576 if (!curr->is_valid)
1580 *ntptime = curr->ntptime;
1582 *rtptime = curr->rtptime;
1584 *packet_count = curr->packet_count;
1586 *octet_count = curr->octet_count;
1594 * rtp_source_get_last_rb:
1595 * @src: an #RTPSource
1596 * @fractionlost: fraction lost since last SR/RR
1597 * @packetslost: the cumulative number of packets lost
1598 * @exthighestseq: the extended last sequence number received
1599 * @jitter: the interarrival jitter (in clock rate units)
1600 * @lsr: the time of the last SR packet on this source
1601 * (in NTP Short Format, 16.16 fixed point)
1602 * @dlsr: the delay since the last SR packet
1603 * (in NTP Short Format, 16.16 fixed point)
1604 * @round_trip: the round-trip time
1605 * (in NTP Short Format, 16.16 fixed point)
1607 * Get the values of the last RB report set with rtp_source_process_rb().
1609 * Returns: %TRUE if there was a valid SB report.
1612 rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
1613 gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
1614 guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
1616 RTPReceiverReport *curr;
1618 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1620 curr = &src->stats.rr[src->stats.curr_rr];
1621 if (!curr->is_valid)
1625 *fractionlost = curr->fractionlost;
1627 *packetslost = curr->packetslost;
1629 *exthighestseq = curr->exthighestseq;
1631 *jitter = curr->jitter;
1637 *round_trip = curr->round_trip;
1643 find_conflicting_address (GList * conflicting_addresses,
1644 GSocketAddress * address, GstClockTime time)
1648 for (item = conflicting_addresses; item; item = g_list_next (item)) {
1649 RTPConflictingAddress *known_conflict = item->data;
1651 if (__g_socket_address_equal (address, known_conflict->address)) {
1652 known_conflict->time = time;
1661 add_conflicting_address (GList * conflicting_addresses,
1662 GSocketAddress * address, GstClockTime time)
1664 RTPConflictingAddress *new_conflict;
1666 new_conflict = g_slice_new (RTPConflictingAddress);
1668 new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
1669 new_conflict->time = time;
1671 return g_list_prepend (conflicting_addresses, new_conflict);
1675 timeout_conflicting_addresses (GList * conflicting_addresses,
1676 GstClockTime current_time)
1679 /* "a relatively long time" -- RFC 3550 section 8.2 */
1680 const GstClockTime collision_timeout =
1681 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10;
1683 item = g_list_first (conflicting_addresses);
1685 RTPConflictingAddress *known_conflict = item->data;
1686 GList *next_item = g_list_next (item);
1688 if (known_conflict->time < current_time - collision_timeout) {
1691 conflicting_addresses = g_list_delete_link (conflicting_addresses, item);
1692 buf = __g_socket_address_to_string (known_conflict->address);
1693 GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
1695 rtp_conflicting_address_free (known_conflict);
1700 return conflicting_addresses;
1704 * rtp_source_find_conflicting_address:
1705 * @src: The source the packet came in
1706 * @address: address to check for
1707 * @time: The time when the packet that is possibly in conflict arrived
1709 * Checks if an address which has a conflict is already known. If it is
1710 * a known conflict, remember the time
1712 * Returns: TRUE if it was a known conflict, FALSE otherwise
1715 rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
1718 return find_conflicting_address (src->conflicting_addresses, address, time);
1722 * rtp_source_add_conflicting_address:
1723 * @src: The source the packet came in
1724 * @address: address to remember
1725 * @time: The time when the packet that is in conflict arrived
1727 * Adds a new conflict address
1730 rtp_source_add_conflicting_address (RTPSource * src,
1731 GSocketAddress * address, GstClockTime time)
1733 src->conflicting_addresses =
1734 add_conflicting_address (src->conflicting_addresses, address, time);
1738 * rtp_source_timeout:
1739 * @src: The #RTPSource
1740 * @current_time: The current time
1741 * @feedback_retention_window: The running time before which retained feedback
1742 * packets have to be discarded
1744 * This is processed on each RTCP interval. It times out old collisions.
1745 * It also times out old retained feedback packets
1748 rtp_source_timeout (RTPSource * src, GstClockTime current_time,
1749 GstClockTime feedback_retention_window)
1753 src->conflicting_addresses =
1754 timeout_conflicting_addresses (src->conflicting_addresses, current_time);
1756 /* Time out AVPF packets that are older than the desired length */
1757 while ((pkt = g_queue_peek_tail (src->retained_feedback)) &&
1758 GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window)
1759 gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
1763 compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
1765 const GstBuffer *bufa = a;
1766 const GstBuffer *bufb = b;
1768 return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb);
1772 rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
1773 GstClockTime running_time)
1777 buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
1778 packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
1780 GST_BUFFER_TIMESTAMP (buffer) = running_time;
1782 g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
1786 rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
1788 if (g_queue_find_custom (src->retained_feedback, data, func))
1795 * @src: The #RTPSource
1798 * Register that @seqnum has not been received from @src.
1801 rtp_source_register_nack (RTPSource * src, guint16 seqnum)
1804 guint32 dword = seqnum << 16;
1807 len = src->nacks->len;
1808 for (i = 0; i < len; i++) {
1812 tdword = g_array_index (src->nacks, guint32, i);
1813 tseq = tdword >> 16;
1815 diff = gst_rtp_buffer_compare_seqnum (tseq, seqnum);
1819 /* we already have this seqnum */
1822 /* it comes before the recorded seqnum, FIXME, we could merge it
1823 * if not to far away */
1825 GST_DEBUG ("insert NACK #%u at %u", seqnum, i);
1826 g_array_insert_val (src->nacks, i, dword);
1827 } else if (diff < 16) {
1828 /* we can merge it */
1829 dword = g_array_index (src->nacks, guint32, i);
1830 dword |= 1 << (diff - 1);
1831 GST_DEBUG ("merge NACK #%u at %u with NACK #%u -> 0x%08x", seqnum, i,
1832 dword >> 16, dword);
1833 g_array_index (src->nacks, guint32, i) = dword;
1835 GST_DEBUG ("append NACK #%u", seqnum);
1836 g_array_append_val (src->nacks, dword);
1838 src->send_nack = TRUE;
1842 * @src: The #RTPSource
1843 * @n_nacks: result number of nacks
1845 * Get the registered NACKS since the last rtp_source_clear_nacks().
1847 * Returns: an array of @n_nacks seqnum values.
1850 rtp_source_get_nacks (RTPSource * src, guint * n_nacks)
1853 *n_nacks = src->nacks->len;
1855 return (guint32 *) src->nacks->data;
1859 rtp_source_clear_nacks (RTPSource * src)
1861 g_array_set_size (src->nacks, 0);
1862 src->send_nack = FALSE;