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 * This property returns a GstStructure named application/x-rtp-source-stats with
125 * fields useful for statistics and diagnostics.
127 * Take note of each respective field's units:
129 * - NTP times are in the appropriate 32-bit or 64-bit fixed-point format
130 * starting from January 1, 1970 (except for timespans).
131 * - RTP times are in clock rate units (i.e. clock rate = 1 second)
132 * starting at a random offset.
133 * - For fields indicating packet loss, note that late packets are not considered lost,
134 * and duplicates are not taken into account. Hence, the loss may be negative
135 * if there are duplicates.
137 * The following fields are always present.
139 * "ssrc" G_TYPE_UINT the SSRC of this source
140 * "internal" G_TYPE_BOOLEAN this source is a source of the session
141 * "validated" G_TYPE_BOOLEAN the source is validated
142 * "received-bye" G_TYPE_BOOLEAN we received a BYE from this source
143 * "is-csrc" G_TYPE_BOOLEAN this source was found as CSRC
144 * "is-sender" G_TYPE_BOOLEAN this source is a sender
145 * "seqnum-base" G_TYPE_INT first seqnum if known
146 * "clock-rate" G_TYPE_INT the clock rate of the media
148 * The following fields are only present when known.
150 * "rtp-from" G_TYPE_STRING where we received the last RTP packet from
151 * "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from
153 * The following fields make sense for internal sources and will only increase
154 * when "is-sender" is TRUE.
156 * "octets-sent" G_TYPE_UINT64 number of bytes we sent
157 * "packets-sent" G_TYPE_UINT64 number of packets we sent
159 * The following fields make sense for non-internal sources and will only
160 * increase when "is-sender" is TRUE.
162 * "octets-received" G_TYPE_UINT64 total number of bytes received
163 * "packets-received" G_TYPE_UINT64 total number of packets received
165 * Following fields are updated when "is-sender" is TRUE.
167 * "bitrate" G_TYPE_UINT64 bitrate in bits per second
168 * "jitter" G_TYPE_UINT estimated jitter (in clock rate units)
169 * "packets-lost" G_TYPE_INT estimated amount of packets lost
171 * The last SR report this source sent. This only updates when "is-sender" is
174 * "have-sr" G_TYPE_BOOLEAN the source has sent SR
175 * "sr-ntptime" G_TYPE_UINT64 NTP time of SR (in NTP Timestamp Format, 32.32 fixed point)
176 * "sr-rtptime" G_TYPE_UINT RTP time of SR (in clock rate units)
177 * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR
178 * "sr-packet-count" G_TYPE_UINT the number of packets in the SR
180 * The following fields are only present for non-internal sources and
181 * represent the content of the last RB packet that was sent to this source.
182 * These values are only updated when the source is sending.
184 * "sent-rb" G_TYPE_BOOLEAN we have sent an RB
185 * "sent-rb-fractionlost" G_TYPE_UINT calculated lost fraction
186 * "sent-rb-packetslost" G_TYPE_INT lost packets
187 * "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum
188 * "sent-rb-jitter" G_TYPE_UINT jitter (in clock rate units)
189 * "sent-rb-lsr" G_TYPE_UINT last SR time (in NTP Short Format, 16.16 fixed point)
190 * "sent-rb-dlsr" G_TYPE_UINT delay since last SR (in NTP Short Format, 16.16 fixed point)
192 * The following fields are only present for non-internal sources and
193 * represents the last RB that this source sent. This is only updated
194 * when the source is receiving data and sending RB blocks.
196 * "have-rb" G_TYPE_BOOLEAN the source has sent RB
197 * "rb-fractionlost" G_TYPE_UINT lost fraction
198 * "rb-packetslost" G_TYPE_INT lost packets
199 * "rb-exthighestseq" G_TYPE_UINT highest received seqnum
200 * "rb-jitter" G_TYPE_UINT reception jitter (in clock rate units)
201 * "rb-lsr" G_TYPE_UINT last SR time (in NTP Short Format, 16.16 fixed point)
202 * "rb-dlsr" G_TYPE_UINT delay since last SR (in NTP Short Format, 16.16 fixed point)
204 * The round trip of this source is calculated from the last RB
205 * values and the reception time of the last RB packet. It is only present for
206 * non-internal sources.
208 * "rb-round-trip" G_TYPE_UINT the round-trip time (in NTP Short Format, 16.16 fixed point)
211 g_object_class_install_property (gobject_class, PROP_STATS,
212 g_param_spec_boxed ("stats", "Stats",
213 "The stats of this source", GST_TYPE_STRUCTURE,
214 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
216 g_object_class_install_property (gobject_class, PROP_PROBATION,
217 g_param_spec_uint ("probation", "Number of probations",
218 "Consecutive packet sequence numbers to accept the source",
219 0, G_MAXUINT, DEFAULT_PROBATION,
220 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
222 GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
227 * @src: an #RTPSource
229 * Reset the stats of @src.
232 rtp_source_reset (RTPSource * src)
234 src->marked_bye = FALSE;
236 g_free (src->bye_reason);
237 src->bye_reason = NULL;
238 src->sent_bye = FALSE;
239 g_hash_table_remove_all (src->reported_in_sr_of);
241 src->stats.cycles = -1;
242 src->stats.jitter = 0;
243 src->stats.transit = -1;
244 src->stats.curr_sr = 0;
245 src->stats.sr[0].is_valid = FALSE;
246 src->stats.curr_rr = 0;
247 src->stats.rr[0].is_valid = FALSE;
248 src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
249 src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
250 src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
251 src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
252 g_array_set_size (src->nacks, 0);
254 src->stats.sent_pli_count = 0;
255 src->stats.sent_fir_count = 0;
259 rtp_source_init (RTPSource * src)
261 /* sources are initialy on probation until we receive enough valid RTP
262 * packets or a valid RTCP packet */
263 src->validated = FALSE;
264 src->internal = FALSE;
265 src->probation = DEFAULT_PROBATION;
266 src->curr_probation = src->probation;
267 src->closing = FALSE;
269 src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
272 src->clock_rate = -1;
273 src->packets = g_queue_new ();
274 src->seqnum_offset = -1;
275 src->last_rtptime = -1;
277 src->retained_feedback = g_queue_new ();
278 src->nacks = g_array_new (FALSE, FALSE, sizeof (guint32));
280 src->reported_in_sr_of = g_hash_table_new (g_direct_hash, g_direct_equal);
282 rtp_source_reset (src);
286 rtp_conflicting_address_free (RTPConflictingAddress * addr)
288 g_object_unref (addr->address);
289 g_slice_free (RTPConflictingAddress, addr);
293 rtp_source_finalize (GObject * object)
297 src = RTP_SOURCE_CAST (object);
299 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
300 g_queue_free (src->packets);
302 gst_structure_free (src->sdes);
304 g_free (src->bye_reason);
306 gst_caps_replace (&src->caps, NULL);
308 g_list_free_full (src->conflicting_addresses,
309 (GDestroyNotify) rtp_conflicting_address_free);
310 g_queue_foreach (src->retained_feedback, (GFunc) gst_buffer_unref, NULL);
311 g_queue_free (src->retained_feedback);
313 g_array_free (src->nacks, TRUE);
316 g_object_unref (src->rtp_from);
318 g_object_unref (src->rtcp_from);
320 g_hash_table_unref (src->reported_in_sr_of);
322 G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
325 static GstStructure *
326 rtp_source_create_stats (RTPSource * src)
329 gboolean is_sender = src->is_sender;
330 gboolean internal = src->internal;
333 guint8 fractionlost = 0;
334 gint32 packetslost = 0;
335 guint32 exthighestseq = 0;
339 guint32 round_trip = 0;
341 GstClockTime time = 0;
344 guint32 packet_count = 0;
345 guint32 octet_count = 0;
348 /* common data for all types of sources */
349 s = gst_structure_new ("application/x-rtp-source-stats",
350 "ssrc", G_TYPE_UINT, (guint) src->ssrc,
351 "internal", G_TYPE_BOOLEAN, internal,
352 "validated", G_TYPE_BOOLEAN, src->validated,
353 "received-bye", G_TYPE_BOOLEAN, src->marked_bye,
354 "is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
355 "is-sender", G_TYPE_BOOLEAN, is_sender,
356 "seqnum-base", G_TYPE_INT, src->seqnum_offset,
357 "clock-rate", G_TYPE_INT, src->clock_rate, NULL);
359 /* add address and port */
361 address_str = __g_socket_address_to_string (src->rtp_from);
362 gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
363 g_free (address_str);
365 if (src->rtcp_from) {
366 address_str = __g_socket_address_to_string (src->rtcp_from);
367 gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
368 g_free (address_str);
371 gst_structure_set (s,
372 "octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
373 "packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
374 "octets-received", G_TYPE_UINT64, src->stats.octets_received,
375 "packets-received", G_TYPE_UINT64, src->stats.packets_received,
376 "bitrate", G_TYPE_UINT64, src->bitrate,
377 "packets-lost", G_TYPE_INT,
378 (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
379 (guint) (src->stats.jitter >> 4),
380 "sent-pli-count", G_TYPE_UINT, src->stats.sent_pli_count,
381 "recv-pli-count", G_TYPE_UINT, src->stats.recv_pli_count,
382 "sent-fir-count", G_TYPE_UINT, src->stats.sent_fir_count,
383 "recv-fir-count", G_TYPE_UINT, src->stats.recv_fir_count, NULL);
385 /* get the last SR. */
386 have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
387 &packet_count, &octet_count);
388 gst_structure_set (s,
389 "have-sr", G_TYPE_BOOLEAN, have_sr,
390 "sr-ntptime", G_TYPE_UINT64, ntptime,
391 "sr-rtptime", G_TYPE_UINT, (guint) rtptime,
392 "sr-octet-count", G_TYPE_UINT, (guint) octet_count,
393 "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
396 /* get the last RB we sent */
397 gst_structure_set (s,
398 "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
399 "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
400 "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
401 "sent-rb-exthighestseq", G_TYPE_UINT,
402 (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
403 (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
404 (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
405 (guint) src->last_rr.dlsr, NULL);
407 /* get the last RB */
408 have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
409 &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
411 gst_structure_set (s,
412 "have-rb", G_TYPE_BOOLEAN, have_rb,
413 "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
414 "rb-packetslost", G_TYPE_INT, (gint) packetslost,
415 "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
416 "rb-jitter", G_TYPE_UINT, (guint) jitter,
417 "rb-lsr", G_TYPE_UINT, (guint) lsr,
418 "rb-dlsr", G_TYPE_UINT, (guint) dlsr,
419 "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
426 * rtp_source_get_sdes_struct:
427 * @src: an #RTPSource
429 * Get the SDES from @src. See the SDES property for more details.
431 * Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
432 * valid until the SDES items of @src are modified.
435 rtp_source_get_sdes_struct (RTPSource * src)
437 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
443 sdes_struct_compare_func (GQuark field_id, const GValue * value,
449 old = GST_STRUCTURE (user_data);
450 field = g_quark_to_string (field_id);
452 if (!gst_structure_has_field (old, field))
455 g_assert (G_VALUE_HOLDS_STRING (value));
457 return strcmp (g_value_get_string (value), gst_structure_get_string (old,
462 * rtp_source_set_sdes_struct:
463 * @src: an #RTPSource
464 * @sdes: the SDES structure
466 * Store the @sdes in @src. @sdes must be a structure of type
467 * "application/x-rtp-source-sdes", see the SDES property for more details.
469 * This function takes ownership of @sdes.
471 * Returns: %FALSE if the SDES was unchanged.
474 rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
478 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
479 g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
480 "application/x-rtp-source-sdes") == 0, FALSE);
482 changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
485 gst_structure_free (src->sdes);
488 gst_structure_free (sdes);
494 rtp_source_set_property (GObject * object, guint prop_id,
495 const GValue * value, GParamSpec * pspec)
499 src = RTP_SOURCE (object);
503 src->ssrc = g_value_get_uint (value);
506 src->probation = g_value_get_uint (value);
509 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
515 rtp_source_get_property (GObject * object, guint prop_id,
516 GValue * value, GParamSpec * pspec)
520 src = RTP_SOURCE (object);
524 g_value_set_uint (value, rtp_source_get_ssrc (src));
527 g_value_set_boolean (value, rtp_source_is_as_csrc (src));
529 case PROP_IS_VALIDATED:
530 g_value_set_boolean (value, rtp_source_is_validated (src));
533 g_value_set_boolean (value, rtp_source_is_sender (src));
536 g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
539 g_value_take_boxed (value, rtp_source_create_stats (src));
542 g_value_set_uint (value, src->probation);
545 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
554 * Create a #RTPSource with @ssrc.
556 * Returns: a new #RTPSource. Use g_object_unref() after usage.
559 rtp_source_new (guint32 ssrc)
563 src = g_object_new (RTP_TYPE_SOURCE, NULL);
570 * rtp_source_set_callbacks:
571 * @src: an #RTPSource
572 * @cb: callback functions
573 * @user_data: user data
575 * Set the callbacks for the source.
578 rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
581 g_return_if_fail (RTP_IS_SOURCE (src));
583 src->callbacks.push_rtp = cb->push_rtp;
584 src->callbacks.clock_rate = cb->clock_rate;
585 src->user_data = user_data;
589 * rtp_source_get_ssrc:
590 * @src: an #RTPSource
592 * Get the SSRC of @source.
594 * Returns: the SSRC of src.
597 rtp_source_get_ssrc (RTPSource * src)
601 g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
609 * rtp_source_set_as_csrc:
610 * @src: an #RTPSource
612 * Configure @src as a CSRC, this will also validate @src.
615 rtp_source_set_as_csrc (RTPSource * src)
617 g_return_if_fail (RTP_IS_SOURCE (src));
619 src->validated = TRUE;
624 * rtp_source_is_as_csrc:
625 * @src: an #RTPSource
627 * Check if @src is a contributing source.
629 * Returns: %TRUE if @src is acting as a contributing source.
632 rtp_source_is_as_csrc (RTPSource * src)
636 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
638 result = src->is_csrc;
644 * rtp_source_is_active:
645 * @src: an #RTPSource
647 * Check if @src is an active source. A source is active if it has been
648 * validated and has not yet received a BYE packet
650 * Returns: %TRUE if @src is an qactive source.
653 rtp_source_is_active (RTPSource * src)
657 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
659 result = RTP_SOURCE_IS_ACTIVE (src);
665 * rtp_source_is_validated:
666 * @src: an #RTPSource
668 * Check if @src is a validated source.
670 * Returns: %TRUE if @src is a validated source.
673 rtp_source_is_validated (RTPSource * src)
677 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
679 result = src->validated;
685 * rtp_source_is_sender:
686 * @src: an #RTPSource
688 * Check if @src is a sending source.
690 * Returns: %TRUE if @src is a sending source.
693 rtp_source_is_sender (RTPSource * src)
697 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
699 result = RTP_SOURCE_IS_SENDER (src);
705 * rtp_source_is_marked_bye:
706 * @src: an #RTPSource
708 * Check if @src is marked as leaving the session with a BYE packet.
710 * Returns: %TRUE if @src has been marked BYE.
713 rtp_source_is_marked_bye (RTPSource * src)
717 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
719 result = RTP_SOURCE_IS_MARKED_BYE (src);
726 * rtp_source_get_bye_reason:
727 * @src: an #RTPSource
729 * Get the BYE reason for @src. Check if the source is marked as leaving the
730 * session with a BYE message first with rtp_source_is_marked_bye().
732 * Returns: The BYE reason or NULL when no reason was given or the source was
733 * not marked BYE yet. g_free() after usage.
736 rtp_source_get_bye_reason (RTPSource * src)
740 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
742 result = g_strdup (src->bye_reason);
748 * rtp_source_update_caps:
749 * @src: an #RTPSource
752 * Parse @caps and store all relevant information in @source.
755 rtp_source_update_caps (RTPSource * src, GstCaps * caps)
762 /* nothing changed, return */
763 if (caps == NULL || src->caps == caps)
766 s = gst_caps_get_structure (caps, 0);
768 rtx = (gst_structure_get_uint (s, "rtx-ssrc", &val) && val == src->ssrc);
770 if (gst_structure_get_int (s, rtx ? "rtx-payload" : "payload", &ival))
775 GST_DEBUG ("got %spayload %d", rtx ? "rtx " : "", src->payload);
777 if (gst_structure_get_int (s, "clock-rate", &ival))
778 src->clock_rate = ival;
780 src->clock_rate = -1;
782 GST_DEBUG ("got clock-rate %d", src->clock_rate);
784 if (gst_structure_get_uint (s, rtx ? "rtx-seqnum-offset" : "seqnum-offset",
786 src->seqnum_offset = val;
788 src->seqnum_offset = -1;
790 GST_DEBUG ("got %sseqnum-offset %" G_GINT32_FORMAT, rtx ? "rtx " : "",
793 gst_caps_replace (&src->caps, caps);
797 * rtp_source_set_rtp_from:
798 * @src: an #RTPSource
799 * @address: the RTP address to set
801 * Set that @src is receiving RTP packets from @address. This is used for
802 * collistion checking.
805 rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
807 g_return_if_fail (RTP_IS_SOURCE (src));
810 g_object_unref (src->rtp_from);
811 src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
815 * rtp_source_set_rtcp_from:
816 * @src: an #RTPSource
817 * @address: the RTCP address to set
819 * Set that @src is receiving RTCP packets from @address. This is used for
820 * collistion checking.
823 rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
825 g_return_if_fail (RTP_IS_SOURCE (src));
828 g_object_unref (src->rtcp_from);
829 src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
833 push_packet (RTPSource * src, GstBuffer * buffer)
835 GstFlowReturn ret = GST_FLOW_OK;
837 /* push queued packets first if any */
838 while (!g_queue_is_empty (src->packets)) {
839 GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
841 GST_LOG ("pushing queued packet");
842 if (src->callbacks.push_rtp)
843 src->callbacks.push_rtp (src, buffer, src->user_data);
845 gst_buffer_unref (buffer);
847 GST_LOG ("pushing new packet");
849 if (src->callbacks.push_rtp)
850 ret = src->callbacks.push_rtp (src, buffer, src->user_data);
852 gst_buffer_unref (buffer);
858 get_clock_rate (RTPSource * src, guint8 payload)
860 if (src->payload == -1) {
861 /* first payload received, nothing was in the caps, lock on to this payload */
862 src->payload = payload;
863 GST_DEBUG ("first payload %d", payload);
864 } else if (payload != src->payload) {
865 /* we have a different payload than before, reset the clock-rate */
866 GST_DEBUG ("new payload %d", payload);
867 src->payload = payload;
868 src->clock_rate = -1;
869 src->stats.transit = -1;
872 if (src->clock_rate == -1) {
873 gint clock_rate = -1;
875 if (src->callbacks.clock_rate)
876 clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
878 GST_DEBUG ("got clock-rate %d", clock_rate);
880 src->clock_rate = clock_rate;
882 return src->clock_rate;
885 /* Jitter is the variation in the delay of received packets in a flow. It is
886 * measured by comparing the interval when RTP packets were sent to the interval
887 * at which they were received. For instance, if packet #1 and packet #2 leave
888 * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
891 calculate_jitter (RTPSource * src, RTPPacketInfo * pinfo)
893 GstClockTime running_time;
894 guint32 rtparrival, transit, rtptime;
899 /* get arrival time */
900 if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
905 GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
908 if ((clock_rate = get_clock_rate (src, pt)) == -1)
911 rtptime = pinfo->rtptime;
913 /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
914 * care about the absolute value, just the difference. */
915 rtparrival = gst_util_uint64_scale_int (running_time, clock_rate, GST_SECOND);
917 /* transit time is difference with RTP timestamp */
918 transit = rtparrival - rtptime;
920 /* get ABS diff with previous transit time */
921 if (src->stats.transit != -1) {
922 if (transit > src->stats.transit)
923 diff = transit - src->stats.transit;
925 diff = src->stats.transit - transit;
929 src->stats.transit = transit;
931 /* update jitter, the value we store is scaled up so we can keep precision. */
932 src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
934 src->stats.prev_rtptime = src->stats.last_rtptime;
935 src->stats.last_rtptime = rtparrival;
937 GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
938 rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
945 GST_WARNING ("cannot get current running_time");
950 GST_WARNING ("cannot get clock-rate for pt %d", pt);
956 init_seq (RTPSource * src, guint16 seq)
958 src->stats.base_seq = seq;
959 src->stats.max_seq = seq;
960 src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
961 src->stats.cycles = 0;
962 src->stats.packets_received = 0;
963 src->stats.octets_received = 0;
964 src->stats.bytes_received = 0;
965 src->stats.prev_received = 0;
966 src->stats.prev_expected = 0;
967 src->stats.recv_pli_count = 0;
968 src->stats.recv_fir_count = 0;
970 GST_DEBUG ("base_seq %d", seq);
973 #define BITRATE_INTERVAL (2 * GST_SECOND)
976 do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
977 guint64 * bytes_handled)
981 if (src->prev_rtime) {
982 elapsed = running_time - src->prev_rtime;
984 if (elapsed > BITRATE_INTERVAL) {
987 rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
989 GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
990 ", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
992 if (src->bitrate == 0)
995 src->bitrate = ((src->bitrate * 3) + rate) / 4;
997 src->prev_rtime = running_time;
1001 GST_LOG ("Reset bitrate measurement");
1002 src->prev_rtime = running_time;
1008 update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo,
1009 gboolean is_receive)
1011 guint16 seqnr, expected;
1012 RTPSourceStats *stats;
1015 stats = &src->stats;
1017 seqnr = pinfo->seqnum;
1019 if (stats->cycles == -1) {
1020 GST_DEBUG ("received first packet");
1021 /* first time we heard of this source */
1022 init_seq (src, seqnr);
1023 src->stats.max_seq = seqnr - 1;
1024 src->curr_probation = src->probation;
1028 expected = src->stats.max_seq + 1;
1029 delta = gst_rtp_buffer_compare_seqnum (expected, seqnr);
1031 /* if we are still on probation, check seqnum */
1032 if (src->curr_probation) {
1033 /* when in probation, we require consecutive seqnums */
1035 /* expected packet */
1036 GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
1037 src->curr_probation--;
1038 if (seqnr < stats->max_seq) {
1039 /* sequence number wrapped - count another 64K cycle. */
1040 stats->cycles += RTP_SEQ_MOD;
1042 src->stats.max_seq = seqnr;
1044 if (src->curr_probation == 0) {
1045 GST_DEBUG ("probation done!");
1046 init_seq (src, seqnr);
1050 GST_DEBUG ("probation %d: queue packet", src->curr_probation);
1051 /* when still in probation, keep packets in a list. */
1052 g_queue_push_tail (src->packets, pinfo->data);
1054 /* remove packets from queue if there are too many */
1055 while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
1056 q = g_queue_pop_head (src->packets);
1057 gst_buffer_unref (q);
1062 /* unexpected seqnum in probation */
1063 goto probation_seqnum;
1065 } else if (delta >= 0 && delta < RTP_MAX_DROPOUT) {
1066 /* Clear bad packets */
1067 stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
1068 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1069 g_queue_clear (src->packets);
1071 /* in order, with permissible gap */
1072 if (seqnr < stats->max_seq) {
1073 /* sequence number wrapped - count another 64K cycle. */
1074 stats->cycles += RTP_SEQ_MOD;
1076 stats->max_seq = seqnr;
1077 } else if (delta < -RTP_MAX_MISORDER || delta >= RTP_MAX_DROPOUT) {
1078 /* the sequence number made a very large jump */
1079 if (seqnr == stats->bad_seq && src->packets->head) {
1080 /* two sequential packets -- assume that the other side
1081 * restarted without telling us so just re-sync
1082 * (i.e., pretend this was the first packet). */
1083 init_seq (src, seqnr);
1085 /* unacceptable jump */
1086 stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
1087 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1088 g_queue_clear (src->packets);
1089 g_queue_push_tail (src->packets, pinfo->data);
1093 } else { /* delta < 0 && delta >= -RTP_MAX_MISORDER */
1094 /* Clear bad packets */
1095 stats->bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
1096 g_queue_foreach (src->packets, (GFunc) gst_buffer_unref, NULL);
1097 g_queue_clear (src->packets);
1099 /* duplicate or reordered packet, will be filtered by jitterbuffer. */
1100 GST_WARNING ("duplicate or reordered packet (seqnr %u, expected %u)",
1105 src->stats.octets_received += pinfo->payload_len;
1106 src->stats.bytes_received += pinfo->bytes;
1107 src->stats.packets_received++;
1108 /* for the bitrate estimation */
1109 src->bytes_received += pinfo->payload_len;
1111 GST_LOG ("seq %u, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
1112 seqnr, src->stats.packets_received, src->stats.octets_received);
1123 GST_WARNING ("unacceptable seqnum received");
1128 GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
1129 src->curr_probation = src->probation;
1130 src->stats.max_seq = seqnr;
1136 * rtp_source_process_rtp:
1137 * @src: an #RTPSource
1138 * @pinfo: an #RTPPacketInfo
1140 * Let @src handle the incomming RTP packet described in @pinfo.
1142 * Returns: a #GstFlowReturn.
1145 rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1147 GstFlowReturn result;
1149 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1150 g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
1152 if (!update_receiver_stats (src, pinfo, TRUE))
1155 /* the source that sent the packet must be a sender */
1156 src->is_sender = TRUE;
1157 src->validated = TRUE;
1159 do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
1161 /* calculate jitter for the stats */
1162 calculate_jitter (src, pinfo);
1164 /* we're ready to push the RTP packet now */
1165 result = push_packet (src, pinfo->data);
1172 * rtp_source_mark_bye:
1173 * @src: an #RTPSource
1174 * @reason: the reason for leaving
1176 * Mark @src in the BYE state. This can happen when the source wants to
1177 * leave the sesssion or when a BYE packets has been received.
1179 * This will make the source inactive.
1182 rtp_source_mark_bye (RTPSource * src, const gchar * reason)
1184 g_return_if_fail (RTP_IS_SOURCE (src));
1186 GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
1187 GST_STR_NULL (reason));
1189 /* copy the reason and mark as bye */
1190 g_free (src->bye_reason);
1191 src->bye_reason = g_strdup (reason);
1192 src->marked_bye = TRUE;
1196 * rtp_source_send_rtp:
1197 * @src: an #RTPSource
1198 * @data: an RTP buffer or a list of RTP buffers
1199 * @is_list: if @data is a buffer or list
1200 * @running_time: the running time of @data
1202 * Send @data (an RTP buffer or list of buffers) originating from @src.
1203 * This will make @src a sender. This function takes ownership of @data and
1204 * modifies the SSRC in the RTP packet to that of @src when needed.
1206 * Returns: a #GstFlowReturn.
1209 rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1211 GstFlowReturn result;
1212 GstClockTime running_time;
1214 guint64 ext_rtptime;
1215 guint64 rt_diff, rtp_diff;
1217 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1219 /* we are a sender now */
1220 src->is_sender = TRUE;
1222 /* we are also a receiver of our packets */
1223 if (!update_receiver_stats (src, pinfo, FALSE))
1226 /* update stats for the SR */
1227 src->stats.packets_sent += pinfo->packets;
1228 src->stats.octets_sent += pinfo->payload_len;
1229 src->bytes_sent += pinfo->payload_len;
1231 running_time = pinfo->running_time;
1233 do_bitrate_estimation (src, running_time, &src->bytes_sent);
1235 rtptime = pinfo->rtptime;
1237 ext_rtptime = src->last_rtptime;
1238 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
1240 GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
1241 GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
1243 if (ext_rtptime > src->last_rtptime) {
1244 rtp_diff = ext_rtptime - src->last_rtptime;
1245 rt_diff = running_time - src->last_rtime;
1247 /* calc the diff so we can detect drift at the sender. This can also be used
1248 * to guestimate the clock rate if the NTP time is locked to the RTP
1249 * timestamps (as is the case when the capture device is providing the clock). */
1250 GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
1251 GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
1254 /* we keep track of the last received RTP timestamp and the corresponding
1255 * buffer running_time so that we can use this info when constructing SR reports */
1256 src->last_rtime = running_time;
1257 src->last_rtptime = ext_rtptime;
1260 if (!src->callbacks.push_rtp)
1263 GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT,
1264 pinfo->is_list ? "list" : "packet", src->stats.packets_sent);
1266 result = src->callbacks.push_rtp (src, pinfo->data, src->user_data);
1274 GST_WARNING ("no callback installed, dropping packet");
1280 * rtp_source_process_sr:
1281 * @src: an #RTPSource
1282 * @time: time of packet arrival
1283 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1284 * @rtptime: the RTP time (in clock rate units)
1285 * @packet_count: the packet count
1286 * @octet_count: the octet count
1288 * Update the sender report in @src.
1291 rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
1292 guint32 rtptime, guint32 packet_count, guint32 octet_count)
1294 RTPSenderReport *curr;
1297 g_return_if_fail (RTP_IS_SOURCE (src));
1299 GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
1300 ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
1301 (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
1302 packet_count, octet_count);
1304 curridx = src->stats.curr_sr ^ 1;
1305 curr = &src->stats.sr[curridx];
1307 /* this is a sender now */
1308 src->is_sender = TRUE;
1310 /* update current */
1311 curr->is_valid = TRUE;
1312 curr->ntptime = ntptime;
1313 curr->rtptime = rtptime;
1314 curr->packet_count = packet_count;
1315 curr->octet_count = octet_count;
1319 src->stats.curr_sr = curridx;
1321 src->stats.prev_rtcptime = src->stats.last_rtcptime;
1322 src->stats.last_rtcptime = time;
1326 * rtp_source_process_rb:
1327 * @src: an #RTPSource
1328 * @ntpnstime: the current time in nanoseconds since 1970
1329 * @fractionlost: fraction lost since last SR/RR
1330 * @packetslost: the cumulative number of packets lost
1331 * @exthighestseq: the extended last sequence number received
1332 * @jitter: the interarrival jitter (in clock rate units)
1333 * @lsr: the time of the last SR packet on this source
1334 * (in NTP Short Format, 16.16 fixed point)
1335 * @dlsr: the delay since the last SR packet
1336 * (in NTP Short Format, 16.16 fixed point)
1338 * Update the report block in @src.
1341 rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
1342 guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
1343 guint32 jitter, guint32 lsr, guint32 dlsr)
1345 RTPReceiverReport *curr;
1350 g_return_if_fail (RTP_IS_SOURCE (src));
1352 GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
1353 ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
1354 src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
1355 lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
1357 curridx = src->stats.curr_rr ^ 1;
1358 curr = &src->stats.rr[curridx];
1360 /* update current */
1361 curr->is_valid = TRUE;
1362 curr->fractionlost = fractionlost;
1363 curr->packetslost = packetslost;
1364 curr->exthighestseq = exthighestseq;
1365 curr->jitter = jitter;
1369 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1370 f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1371 /* calculate round trip, round the time up */
1372 ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
1375 if (A > 0 && ntp > A)
1379 curr->round_trip = A;
1381 GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
1382 A >> 16, A & 0xffff);
1385 src->stats.curr_rr = curridx;
1389 * rtp_source_get_new_sr:
1390 * @src: an #RTPSource
1391 * @ntpnstime: the current time in nanoseconds since 1970
1392 * @running_time: the current running_time of the pipeline
1393 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1394 * @rtptime: the RTP time corresponding to @ntptime (in clock rate units)
1395 * @packet_count: the packet count
1396 * @octet_count: the octet count
1398 * Get new values to put into a new SR report from this source.
1400 * @running_time and @ntpnstime are captured at the same time and represent the
1401 * running time of the pipeline clock and the absolute current system time in
1402 * nanoseconds respectively. Together with the last running_time and RTP timestamp
1403 * we have observed in the source, we can generate @ntptime and @rtptime for an SR
1404 * packet. @ntptime is basically the fixed point representation of @ntpnstime
1405 * and @rtptime the associated RTP timestamp.
1407 * Returns: %TRUE on success.
1410 rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
1411 GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
1412 guint32 * packet_count, guint32 * octet_count)
1415 guint64 t_current_ntp;
1416 GstClockTimeDiff diff;
1418 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1420 /* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
1421 * and an NTP time, we can scale the RTP timestamps so that they match the
1422 * given NTP time. for scaling, we assume that the slope of the rtptime vs
1423 * running_time vs ntptime curve is close to 1, which is certainly
1424 * sufficient for the frequency at which we report SR and the rate we send
1425 * out RTP packets. */
1426 t_rtp = src->last_rtptime;
1428 GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
1429 G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
1431 if (src->clock_rate != -1) {
1432 /* get the diff between the clock running_time and the buffer running_time.
1433 * This is the elapsed time, as measured against the pipeline clock, between
1434 * when the rtp timestamp was observed and the current running_time.
1436 * We need to apply this diff to the RTP timestamp to get the RTP timestamp
1437 * for the given ntpnstime. */
1438 diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
1440 /* now translate the diff to RTP time, handle positive and negative cases.
1441 * If there is no diff, we already set rtptime correctly above. */
1443 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
1444 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1445 t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1448 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
1449 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1450 t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1453 GST_WARNING ("no clock-rate, cannot interpolate rtp time");
1456 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1457 t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1459 GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
1460 (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
1464 *ntptime = t_current_ntp;
1468 *packet_count = src->stats.packets_sent;
1470 *octet_count = src->stats.octets_sent;
1476 * rtp_source_get_new_rb:
1477 * @src: an #RTPSource
1478 * @time: the current time of the system clock
1479 * @fractionlost: fraction lost since last SR/RR
1480 * @packetslost: the cumulative number of packets lost
1481 * @exthighestseq: the extended last sequence number received
1482 * @jitter: the interarrival jitter (in clock rate units)
1483 * @lsr: the time of the last SR packet on this source
1484 * (in NTP Short Format, 16.16 fixed point)
1485 * @dlsr: the delay since the last SR packet
1486 * (in NTP Short Format, 16.16 fixed point)
1488 * Get new values to put into a new report block from this source.
1490 * Returns: %TRUE on success.
1493 rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
1494 guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
1495 guint32 * jitter, guint32 * lsr, guint32 * dlsr)
1497 RTPSourceStats *stats;
1498 guint64 extended_max, expected;
1499 guint64 expected_interval, received_interval, ntptime;
1500 gint64 lost, lost_interval;
1501 guint32 fraction, LSR, DLSR;
1502 GstClockTime sr_time;
1504 stats = &src->stats;
1506 extended_max = stats->cycles + stats->max_seq;
1507 expected = extended_max - stats->base_seq + 1;
1509 GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
1510 ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
1511 extended_max, expected, stats->packets_received, stats->base_seq);
1513 lost = expected - stats->packets_received;
1514 lost = CLAMP (lost, -0x800000, 0x7fffff);
1516 expected_interval = expected - stats->prev_expected;
1517 stats->prev_expected = expected;
1518 received_interval = stats->packets_received - stats->prev_received;
1519 stats->prev_received = stats->packets_received;
1521 lost_interval = expected_interval - received_interval;
1523 if (expected_interval == 0 || lost_interval <= 0)
1526 fraction = (lost_interval << 8) / expected_interval;
1528 GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
1529 /* we scaled the jitter up for additional precision */
1530 GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
1531 ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
1532 extended_max, stats->jitter >> 4);
1534 if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
1537 /* LSR is middle 32 bits of the last ntptime */
1538 LSR = (ntptime >> 16) & 0xffffffff;
1539 diff = time - sr_time;
1540 GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
1541 /* DLSR, delay since last SR is expressed in 1/65536 second units */
1542 DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
1544 /* No valid SR received, LSR/DLSR are set to 0 then */
1545 GST_DEBUG ("no valid SR received");
1549 GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
1550 DLSR >> 16, DLSR & 0xffff);
1553 *fractionlost = fraction;
1555 *packetslost = lost;
1557 *exthighestseq = extended_max;
1559 *jitter = stats->jitter >> 4;
1569 * rtp_source_get_last_sr:
1570 * @src: an #RTPSource
1571 * @time: time of packet arrival
1572 * @ntptime: the NTP time (in NTP Timestamp Format, 32.32 fixed point)
1573 * @rtptime: the RTP time (in clock rate units)
1574 * @packet_count: the packet count
1575 * @octet_count: the octet count
1577 * Get the values of the last sender report as set with rtp_source_process_sr().
1579 * Returns: %TRUE if there was a valid SR report.
1582 rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
1583 guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
1585 RTPSenderReport *curr;
1587 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1589 curr = &src->stats.sr[src->stats.curr_sr];
1590 if (!curr->is_valid)
1594 *ntptime = curr->ntptime;
1596 *rtptime = curr->rtptime;
1598 *packet_count = curr->packet_count;
1600 *octet_count = curr->octet_count;
1608 * rtp_source_get_last_rb:
1609 * @src: an #RTPSource
1610 * @fractionlost: fraction lost since last SR/RR
1611 * @packetslost: the cumulative number of packets lost
1612 * @exthighestseq: the extended last sequence number received
1613 * @jitter: the interarrival jitter (in clock rate units)
1614 * @lsr: the time of the last SR packet on this source
1615 * (in NTP Short Format, 16.16 fixed point)
1616 * @dlsr: the delay since the last SR packet
1617 * (in NTP Short Format, 16.16 fixed point)
1618 * @round_trip: the round-trip time
1619 * (in NTP Short Format, 16.16 fixed point)
1621 * Get the values of the last RB report set with rtp_source_process_rb().
1623 * Returns: %TRUE if there was a valid SB report.
1626 rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
1627 gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
1628 guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
1630 RTPReceiverReport *curr;
1632 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1634 curr = &src->stats.rr[src->stats.curr_rr];
1635 if (!curr->is_valid)
1639 *fractionlost = curr->fractionlost;
1641 *packetslost = curr->packetslost;
1643 *exthighestseq = curr->exthighestseq;
1645 *jitter = curr->jitter;
1651 *round_trip = curr->round_trip;
1657 find_conflicting_address (GList * conflicting_addresses,
1658 GSocketAddress * address, GstClockTime time)
1662 for (item = conflicting_addresses; item; item = g_list_next (item)) {
1663 RTPConflictingAddress *known_conflict = item->data;
1665 if (__g_socket_address_equal (address, known_conflict->address)) {
1666 known_conflict->time = time;
1675 add_conflicting_address (GList * conflicting_addresses,
1676 GSocketAddress * address, GstClockTime time)
1678 RTPConflictingAddress *new_conflict;
1680 new_conflict = g_slice_new (RTPConflictingAddress);
1682 new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
1683 new_conflict->time = time;
1685 return g_list_prepend (conflicting_addresses, new_conflict);
1689 timeout_conflicting_addresses (GList * conflicting_addresses,
1690 GstClockTime current_time)
1693 /* "a relatively long time" -- RFC 3550 section 8.2 */
1694 const GstClockTime collision_timeout =
1695 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10;
1697 item = g_list_first (conflicting_addresses);
1699 RTPConflictingAddress *known_conflict = item->data;
1700 GList *next_item = g_list_next (item);
1702 if (known_conflict->time < current_time - collision_timeout) {
1705 conflicting_addresses = g_list_delete_link (conflicting_addresses, item);
1706 buf = __g_socket_address_to_string (known_conflict->address);
1707 GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
1709 rtp_conflicting_address_free (known_conflict);
1714 return conflicting_addresses;
1718 * rtp_source_find_conflicting_address:
1719 * @src: The source the packet came in
1720 * @address: address to check for
1721 * @time: The time when the packet that is possibly in conflict arrived
1723 * Checks if an address which has a conflict is already known. If it is
1724 * a known conflict, remember the time
1726 * Returns: TRUE if it was a known conflict, FALSE otherwise
1729 rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
1732 return find_conflicting_address (src->conflicting_addresses, address, time);
1736 * rtp_source_add_conflicting_address:
1737 * @src: The source the packet came in
1738 * @address: address to remember
1739 * @time: The time when the packet that is in conflict arrived
1741 * Adds a new conflict address
1744 rtp_source_add_conflicting_address (RTPSource * src,
1745 GSocketAddress * address, GstClockTime time)
1747 src->conflicting_addresses =
1748 add_conflicting_address (src->conflicting_addresses, address, time);
1752 * rtp_source_timeout:
1753 * @src: The #RTPSource
1754 * @current_time: The current time
1755 * @feedback_retention_window: The running time before which retained feedback
1756 * packets have to be discarded
1758 * This is processed on each RTCP interval. It times out old collisions.
1759 * It also times out old retained feedback packets
1762 rtp_source_timeout (RTPSource * src, GstClockTime current_time,
1763 GstClockTime feedback_retention_window)
1767 src->conflicting_addresses =
1768 timeout_conflicting_addresses (src->conflicting_addresses, current_time);
1770 /* Time out AVPF packets that are older than the desired length */
1771 while ((pkt = g_queue_peek_tail (src->retained_feedback)) &&
1772 GST_BUFFER_PTS (pkt) < feedback_retention_window)
1773 gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
1777 compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
1779 const GstBuffer *bufa = a;
1780 const GstBuffer *bufb = b;
1782 return GST_BUFFER_PTS (bufa) - GST_BUFFER_PTS (bufb);
1786 rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
1787 GstClockTime running_time)
1791 buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
1792 packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
1794 GST_BUFFER_PTS (buffer) = running_time;
1796 g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
1800 rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
1802 if (g_queue_find_custom (src->retained_feedback, data, func))
1809 * rtp_source_register_nack:
1810 * @src: The #RTPSource
1813 * Register that @seqnum has not been received from @src.
1816 rtp_source_register_nack (RTPSource * src, guint16 seqnum)
1819 guint32 dword = seqnum << 16;
1822 len = src->nacks->len;
1823 for (i = 0; i < len; i++) {
1827 tdword = g_array_index (src->nacks, guint32, i);
1828 tseq = tdword >> 16;
1830 diff = gst_rtp_buffer_compare_seqnum (tseq, seqnum);
1834 /* we already have this seqnum */
1837 /* it comes before the recorded seqnum, FIXME, we could merge it
1838 * if not to far away */
1840 GST_DEBUG ("insert NACK #%u at %u", seqnum, i);
1841 g_array_insert_val (src->nacks, i, dword);
1842 } else if (diff < 16) {
1843 /* we can merge it */
1844 dword = g_array_index (src->nacks, guint32, i);
1845 dword |= 1 << (diff - 1);
1846 GST_DEBUG ("merge NACK #%u at %u with NACK #%u -> 0x%08x", seqnum, i,
1847 dword >> 16, dword);
1848 g_array_index (src->nacks, guint32, i) = dword;
1850 GST_DEBUG ("append NACK #%u", seqnum);
1851 g_array_append_val (src->nacks, dword);
1853 src->send_nack = TRUE;
1857 * rtp_source_get_nacks:
1858 * @src: The #RTPSource
1859 * @n_nacks: result number of nacks
1861 * Get the registered NACKS since the last rtp_source_clear_nacks().
1863 * Returns: an array of @n_nacks seqnum values.
1866 rtp_source_get_nacks (RTPSource * src, guint * n_nacks)
1869 *n_nacks = src->nacks->len;
1871 return (guint32 *) src->nacks->data;
1875 rtp_source_clear_nacks (RTPSource * src)
1877 g_array_set_size (src->nacks, 0);
1878 src->send_nack = FALSE;