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
57 /* GObject vmethods */
58 static void rtp_source_finalize (GObject * object);
59 static void rtp_source_set_property (GObject * object, guint prop_id,
60 const GValue * value, GParamSpec * pspec);
61 static void rtp_source_get_property (GObject * object, guint prop_id,
62 GValue * value, GParamSpec * pspec);
64 /* static guint rtp_source_signals[LAST_SIGNAL] = { 0 }; */
66 G_DEFINE_TYPE (RTPSource, rtp_source, G_TYPE_OBJECT);
69 rtp_source_class_init (RTPSourceClass * klass)
71 GObjectClass *gobject_class;
73 gobject_class = (GObjectClass *) klass;
75 gobject_class->finalize = rtp_source_finalize;
77 gobject_class->set_property = rtp_source_set_property;
78 gobject_class->get_property = rtp_source_get_property;
80 g_object_class_install_property (gobject_class, PROP_SSRC,
81 g_param_spec_uint ("ssrc", "SSRC",
82 "The SSRC of this source", 0, G_MAXUINT, DEFAULT_SSRC,
83 G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
85 g_object_class_install_property (gobject_class, PROP_IS_CSRC,
86 g_param_spec_boolean ("is-csrc", "Is CSRC",
87 "If this SSRC is acting as a contributing source",
88 DEFAULT_IS_CSRC, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
90 g_object_class_install_property (gobject_class, PROP_IS_VALIDATED,
91 g_param_spec_boolean ("is-validated", "Is Validated",
92 "If this SSRC is validated", DEFAULT_IS_VALIDATED,
93 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
95 g_object_class_install_property (gobject_class, PROP_IS_SENDER,
96 g_param_spec_boolean ("is-sender", "Is Sender",
97 "If this SSRC is a sender", DEFAULT_IS_SENDER,
98 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
103 * The current SDES items of the source. Returns a structure with name
104 * application/x-rtp-source-sdes and may contain the following fields:
106 * 'cname' G_TYPE_STRING : The canonical name
107 * 'name' G_TYPE_STRING : The user name
108 * 'email' G_TYPE_STRING : The user's electronic mail address
109 * 'phone' G_TYPE_STRING : The user's phone number
110 * 'location' G_TYPE_STRING : The geographic user location
111 * 'tool' G_TYPE_STRING : The name of application or tool
112 * 'note' G_TYPE_STRING : A notice about the source
114 * other fields may be present and these represent private items in
115 * the SDES where the field name is the prefix.
117 g_object_class_install_property (gobject_class, PROP_SDES,
118 g_param_spec_boxed ("sdes", "SDES",
119 "The SDES information for this source",
120 GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
125 * The statistics of the source. This property returns a GstStructure with
126 * name application/x-rtp-source-stats with the following fields:
128 * "ssrc" G_TYPE_UINT The SSRC of this source
129 * "internal" G_TYPE_BOOLEAN If this source is a source of the session
130 * "validated" G_TYPE_BOOLEAN If the source is validated
131 * "received-bye" G_TYPE_BOOLEAN If we received a BYE from this source
132 * "is-csrc" G_TYPE_BOOLEAN If this source was found as CSRC
133 * "is-sender" G_TYPE_BOOLEAN If this source is a sender
134 * "seqnum-base" G_TYPE_INT first seqnum if known
135 * "clock-rate" G_TYPE_INT the clock rate of the media
137 * The following two fields are only present when known.
139 * "rtp-from" G_TYPE_STRING where we received the last RTP packet from
140 * "rtcp-from" G_TYPE_STRING where we received the last RTCP packet from
142 * The following fields make sense for internal sources and will only increase
143 * when "is-sender" is TRUE:
145 * "octets-sent" G_TYPE_UINT64 number of bytes we sent
146 * "packets-sent" G_TYPE_UINT64 number of packets we sent
148 * The following fields make sense for non-internal sources and will only
149 * increase when "is-sender" is TRUE.
151 * "octets-received" G_TYPE_UINT64 total number of bytes received
152 * "packets-received" G_TYPE_UINT64 total number of packets received
154 * Following fields are updated when "is-sender" is TRUE.
156 * "bitrate" G_TYPE_UINT64 bitrate in bits per second
157 * "jitter" G_TYPE_UINT estimated jitter
158 * "packets-lost" G_TYPE_INT estimated amount of packets lost
160 * The last SR report this source sent. This only updates when "is-sender" is
163 * "have-sr" G_TYPE_BOOLEAN the source has sent SR
164 * "sr-ntptime" G_TYPE_UINT64 ntptime of SR
165 * "sr-rtptime" G_TYPE_UINT rtptime of SR
166 * "sr-octet-count" G_TYPE_UINT the number of bytes in the SR
167 * "sr-packet-count" G_TYPE_UINT the number of packets in the SR
169 * The following fields are only present for non-internal sources and
170 * represent the content of the last RB packet that was sent to this source.
171 * These values are only updated when the source is sending.
173 * "sent-rb" G_TYPE_BOOLEAN we have sent an RB
174 * "sent-rb-fractionlost" G_TYPE_UINT calculated lost fraction
175 * "sent-rb-packetslost" G_TYPE_INT lost packets
176 * "sent-rb-exthighestseq" G_TYPE_UINT last seen seqnum
177 * "sent-rb-jitter" G_TYPE_UINT jitter
178 * "sent-rb-lsr" G_TYPE_UINT last SR time
179 * "sent-rb-dlsr" G_TYPE_UINT delay since last SR
181 * The following fields are only present for non-internal sources and
182 * represents the last RB that this source sent. This is only updated
183 * when the source is receiving data and sending RB blocks.
185 * "have-rb" G_TYPE_BOOLEAN the source has sent RB
186 * "rb-fractionlost" G_TYPE_UINT lost fraction
187 * "rb-packetslost" G_TYPE_INT lost packets
188 * "rb-exthighestseq" G_TYPE_UINT highest received seqnum
189 * "rb-jitter" G_TYPE_UINT reception jitter
190 * "rb-lsr" G_TYPE_UINT last SR time
191 * "rb-dlsr" G_TYPE_UINT delay since last SR
193 * The round trip of this source. This is calculated from the last RB
194 * values and the recption time of the last RB packet. Only present for
195 * non-internal sources.
197 * "rb-round-trip" G_TYPE_UINT the round trip time in nanoseconds
199 g_object_class_install_property (gobject_class, PROP_STATS,
200 g_param_spec_boxed ("stats", "Stats",
201 "The stats of this source", GST_TYPE_STRUCTURE,
202 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
204 g_object_class_install_property (gobject_class, PROP_PROBATION,
205 g_param_spec_uint ("probation", "Number of probations",
206 "Consecutive packet sequence numbers to accept the source",
207 0, G_MAXUINT, DEFAULT_PROBATION,
208 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
210 GST_DEBUG_CATEGORY_INIT (rtp_source_debug, "rtpsource", 0, "RTP Source");
215 * @src: an #RTPSource
217 * Reset the stats of @src.
220 rtp_source_reset (RTPSource * src)
222 src->marked_bye = FALSE;
224 g_free (src->bye_reason);
225 src->bye_reason = NULL;
226 src->sent_bye = FALSE;
227 g_hash_table_remove_all (src->reported_in_sr_of);
229 src->stats.cycles = -1;
230 src->stats.jitter = 0;
231 src->stats.transit = -1;
232 src->stats.curr_sr = 0;
233 src->stats.sr[0].is_valid = FALSE;
234 src->stats.curr_rr = 0;
235 src->stats.rr[0].is_valid = FALSE;
236 src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
237 src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
238 src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
239 src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
240 g_array_set_size (src->nacks, 0);
244 rtp_source_init (RTPSource * src)
246 /* sources are initialy on probation until we receive enough valid RTP
247 * packets or a valid RTCP packet */
248 src->validated = FALSE;
249 src->internal = FALSE;
250 src->probation = DEFAULT_PROBATION;
251 src->curr_probation = src->probation;
252 src->closing = FALSE;
254 src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
257 src->clock_rate = -1;
258 src->packets = g_queue_new ();
259 src->seqnum_base = -1;
260 src->last_rtptime = -1;
262 src->retained_feedback = g_queue_new ();
263 src->nacks = g_array_new (FALSE, FALSE, sizeof (guint32));
265 src->reported_in_sr_of = g_hash_table_new (g_direct_hash, g_direct_equal);
267 rtp_source_reset (src);
271 rtp_conflicting_address_free (RTPConflictingAddress * addr)
273 g_object_unref (addr->address);
278 rtp_source_finalize (GObject * object)
283 src = RTP_SOURCE_CAST (object);
285 while ((buffer = g_queue_pop_head (src->packets)))
286 gst_buffer_unref (buffer);
287 g_queue_free (src->packets);
289 gst_structure_free (src->sdes);
291 g_free (src->bye_reason);
293 gst_caps_replace (&src->caps, NULL);
295 g_list_foreach (src->conflicting_addresses,
296 (GFunc) rtp_conflicting_address_free, NULL);
297 g_list_free (src->conflicting_addresses);
299 while ((buffer = g_queue_pop_head (src->retained_feedback)))
300 gst_buffer_unref (buffer);
301 g_queue_free (src->retained_feedback);
303 g_array_free (src->nacks, TRUE);
306 g_object_unref (src->rtp_from);
308 g_object_unref (src->rtcp_from);
310 g_hash_table_unref (src->reported_in_sr_of);
312 G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
315 static GstStructure *
316 rtp_source_create_stats (RTPSource * src)
319 gboolean is_sender = src->is_sender;
320 gboolean internal = src->internal;
323 guint8 fractionlost = 0;
324 gint32 packetslost = 0;
325 guint32 exthighestseq = 0;
329 guint32 round_trip = 0;
331 GstClockTime time = 0;
334 guint32 packet_count = 0;
335 guint32 octet_count = 0;
338 /* common data for all types of sources */
339 s = gst_structure_new ("application/x-rtp-source-stats",
340 "ssrc", G_TYPE_UINT, (guint) src->ssrc,
341 "internal", G_TYPE_BOOLEAN, internal,
342 "validated", G_TYPE_BOOLEAN, src->validated,
343 "received-bye", G_TYPE_BOOLEAN, src->marked_bye,
344 "is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
345 "is-sender", G_TYPE_BOOLEAN, is_sender,
346 "seqnum-base", G_TYPE_INT, src->seqnum_base,
347 "clock-rate", G_TYPE_INT, src->clock_rate, NULL);
349 /* add address and port */
351 address_str = __g_socket_address_to_string (src->rtp_from);
352 gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
353 g_free (address_str);
355 if (src->rtcp_from) {
356 address_str = __g_socket_address_to_string (src->rtcp_from);
357 gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
358 g_free (address_str);
361 gst_structure_set (s,
362 "octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
363 "packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
364 "octets-received", G_TYPE_UINT64, src->stats.octets_received,
365 "packets-received", G_TYPE_UINT64, src->stats.packets_received,
366 "bitrate", G_TYPE_UINT64, src->bitrate,
367 "packets-lost", G_TYPE_INT,
368 (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
369 (guint) (src->stats.jitter >> 4), NULL);
371 /* get the last SR. */
372 have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
373 &packet_count, &octet_count);
374 gst_structure_set (s,
375 "have-sr", G_TYPE_BOOLEAN, have_sr,
376 "sr-ntptime", G_TYPE_UINT64, ntptime,
377 "sr-rtptime", G_TYPE_UINT, (guint) rtptime,
378 "sr-octet-count", G_TYPE_UINT, (guint) octet_count,
379 "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
382 /* get the last RB we sent */
383 gst_structure_set (s,
384 "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
385 "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
386 "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
387 "sent-rb-exthighestseq", G_TYPE_UINT,
388 (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
389 (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
390 (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
391 (guint) src->last_rr.dlsr, NULL);
393 /* get the last RB */
394 have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
395 &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
397 gst_structure_set (s,
398 "have-rb", G_TYPE_BOOLEAN, have_rb,
399 "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
400 "rb-packetslost", G_TYPE_INT, (gint) packetslost,
401 "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
402 "rb-jitter", G_TYPE_UINT, (guint) jitter,
403 "rb-lsr", G_TYPE_UINT, (guint) lsr,
404 "rb-dlsr", G_TYPE_UINT, (guint) dlsr,
405 "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
412 * rtp_source_get_sdes_struct:
413 * @src: an #RTPSource
415 * Get the SDES from @src. See the SDES property for more details.
417 * Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
418 * valid until the SDES items of @src are modified.
421 rtp_source_get_sdes_struct (RTPSource * src)
423 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
429 sdes_struct_compare_func (GQuark field_id, const GValue * value,
435 old = GST_STRUCTURE (user_data);
436 field = g_quark_to_string (field_id);
438 if (!gst_structure_has_field (old, field))
441 g_assert (G_VALUE_HOLDS_STRING (value));
443 return strcmp (g_value_get_string (value), gst_structure_get_string (old,
448 * rtp_source_set_sdes_struct:
449 * @src: an #RTPSource
450 * @sdes: the SDES structure
452 * Store the @sdes in @src. @sdes must be a structure of type
453 * "application/x-rtp-source-sdes", see the SDES property for more details.
455 * This function takes ownership of @sdes.
457 * Returns: %FALSE if the SDES was unchanged.
460 rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
464 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
465 g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
466 "application/x-rtp-source-sdes") == 0, FALSE);
468 changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
471 gst_structure_free (src->sdes);
474 gst_structure_free (sdes);
480 rtp_source_set_property (GObject * object, guint prop_id,
481 const GValue * value, GParamSpec * pspec)
485 src = RTP_SOURCE (object);
489 src->ssrc = g_value_get_uint (value);
492 src->probation = g_value_get_uint (value);
495 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
501 rtp_source_get_property (GObject * object, guint prop_id,
502 GValue * value, GParamSpec * pspec)
506 src = RTP_SOURCE (object);
510 g_value_set_uint (value, rtp_source_get_ssrc (src));
513 g_value_set_boolean (value, rtp_source_is_as_csrc (src));
515 case PROP_IS_VALIDATED:
516 g_value_set_boolean (value, rtp_source_is_validated (src));
519 g_value_set_boolean (value, rtp_source_is_sender (src));
522 g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
525 g_value_take_boxed (value, rtp_source_create_stats (src));
528 g_value_set_uint (value, src->probation);
531 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
540 * Create a #RTPSource with @ssrc.
542 * Returns: a new #RTPSource. Use g_object_unref() after usage.
545 rtp_source_new (guint32 ssrc)
549 src = g_object_new (RTP_TYPE_SOURCE, NULL);
556 * rtp_source_set_callbacks:
557 * @src: an #RTPSource
558 * @cb: callback functions
559 * @user_data: user data
561 * Set the callbacks for the source.
564 rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
567 g_return_if_fail (RTP_IS_SOURCE (src));
569 src->callbacks.push_rtp = cb->push_rtp;
570 src->callbacks.clock_rate = cb->clock_rate;
571 src->user_data = user_data;
575 * rtp_source_get_ssrc:
576 * @src: an #RTPSource
578 * Get the SSRC of @source.
580 * Returns: the SSRC of src.
583 rtp_source_get_ssrc (RTPSource * src)
587 g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
595 * rtp_source_set_as_csrc:
596 * @src: an #RTPSource
598 * Configure @src as a CSRC, this will also validate @src.
601 rtp_source_set_as_csrc (RTPSource * src)
603 g_return_if_fail (RTP_IS_SOURCE (src));
605 src->validated = TRUE;
610 * rtp_source_is_as_csrc:
611 * @src: an #RTPSource
613 * Check if @src is a contributing source.
615 * Returns: %TRUE if @src is acting as a contributing source.
618 rtp_source_is_as_csrc (RTPSource * src)
622 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
624 result = src->is_csrc;
630 * rtp_source_is_active:
631 * @src: an #RTPSource
633 * Check if @src is an active source. A source is active if it has been
634 * validated and has not yet received a BYE packet
636 * Returns: %TRUE if @src is an qactive source.
639 rtp_source_is_active (RTPSource * src)
643 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
645 result = RTP_SOURCE_IS_ACTIVE (src);
651 * rtp_source_is_validated:
652 * @src: an #RTPSource
654 * Check if @src is a validated source.
656 * Returns: %TRUE if @src is a validated source.
659 rtp_source_is_validated (RTPSource * src)
663 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
665 result = src->validated;
671 * rtp_source_is_sender:
672 * @src: an #RTPSource
674 * Check if @src is a sending source.
676 * Returns: %TRUE if @src is a sending source.
679 rtp_source_is_sender (RTPSource * src)
683 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
685 result = RTP_SOURCE_IS_SENDER (src);
691 * rtp_source_is_marked_bye:
692 * @src: an #RTPSource
694 * Check if @src is marked as leaving the session with a BYE packet.
696 * Returns: %TRUE if @src has been marked BYE.
699 rtp_source_is_marked_bye (RTPSource * src)
703 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
705 result = RTP_SOURCE_IS_MARKED_BYE (src);
712 * rtp_source_get_bye_reason:
713 * @src: an #RTPSource
715 * Get the BYE reason for @src. Check if the source is marked as leaving the
716 * session with a BYE message first with rtp_source_is_marked_bye().
718 * Returns: The BYE reason or NULL when no reason was given or the source was
719 * not marked BYE yet. g_free() after usage.
722 rtp_source_get_bye_reason (RTPSource * src)
726 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
728 result = g_strdup (src->bye_reason);
734 * rtp_source_update_caps:
735 * @src: an #RTPSource
738 * Parse @caps and store all relevant information in @source.
741 rtp_source_update_caps (RTPSource * src, GstCaps * caps)
747 /* nothing changed, return */
748 if (caps == NULL || src->caps == caps)
751 s = gst_caps_get_structure (caps, 0);
753 if (gst_structure_get_int (s, "payload", &ival))
757 GST_DEBUG ("got payload %d", src->payload);
759 if (gst_structure_get_int (s, "clock-rate", &ival))
760 src->clock_rate = ival;
762 src->clock_rate = -1;
764 GST_DEBUG ("got clock-rate %d", src->clock_rate);
766 if (gst_structure_get_uint (s, "seqnum-base", &val))
767 src->seqnum_base = val;
769 src->seqnum_base = -1;
771 GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
773 gst_caps_replace (&src->caps, caps);
777 * rtp_source_set_rtp_from:
778 * @src: an #RTPSource
779 * @address: the RTP address to set
781 * Set that @src is receiving RTP packets from @address. This is used for
782 * collistion checking.
785 rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
787 g_return_if_fail (RTP_IS_SOURCE (src));
790 g_object_unref (src->rtp_from);
791 src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
795 * rtp_source_set_rtcp_from:
796 * @src: an #RTPSource
797 * @address: the RTCP address to set
799 * Set that @src is receiving RTCP packets from @address. This is used for
800 * collistion checking.
803 rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
805 g_return_if_fail (RTP_IS_SOURCE (src));
808 g_object_unref (src->rtcp_from);
809 src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
813 push_packet (RTPSource * src, GstBuffer * buffer)
815 GstFlowReturn ret = GST_FLOW_OK;
817 /* push queued packets first if any */
818 while (!g_queue_is_empty (src->packets)) {
819 GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
821 GST_LOG ("pushing queued packet");
822 if (src->callbacks.push_rtp)
823 src->callbacks.push_rtp (src, buffer, src->user_data);
825 gst_buffer_unref (buffer);
827 GST_LOG ("pushing new packet");
829 if (src->callbacks.push_rtp)
830 ret = src->callbacks.push_rtp (src, buffer, src->user_data);
832 gst_buffer_unref (buffer);
838 get_clock_rate (RTPSource * src, guint8 payload)
840 if (src->payload == -1) {
841 /* first payload received, nothing was in the caps, lock on to this payload */
842 src->payload = payload;
843 GST_DEBUG ("first payload %d", payload);
844 } else if (payload != src->payload) {
845 /* we have a different payload than before, reset the clock-rate */
846 GST_DEBUG ("new payload %d", payload);
847 src->payload = payload;
848 src->clock_rate = -1;
849 src->stats.transit = -1;
852 if (src->clock_rate == -1) {
853 gint clock_rate = -1;
855 if (src->callbacks.clock_rate)
856 clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
858 GST_DEBUG ("got clock-rate %d", clock_rate);
860 src->clock_rate = clock_rate;
862 return src->clock_rate;
865 /* Jitter is the variation in the delay of received packets in a flow. It is
866 * measured by comparing the interval when RTP packets were sent to the interval
867 * at which they were received. For instance, if packet #1 and packet #2 leave
868 * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
871 calculate_jitter (RTPSource * src, RTPPacketInfo * pinfo)
873 GstClockTime running_time;
874 guint32 rtparrival, transit, rtptime;
879 /* get arrival time */
880 if ((running_time = pinfo->running_time) == GST_CLOCK_TIME_NONE)
885 GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
888 if ((clock_rate = get_clock_rate (src, pt)) == -1)
891 rtptime = pinfo->rtptime;
893 /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
894 * care about the absolute value, just the difference. */
895 rtparrival = gst_util_uint64_scale_int (running_time, clock_rate, GST_SECOND);
897 /* transit time is difference with RTP timestamp */
898 transit = rtparrival - rtptime;
900 /* get ABS diff with previous transit time */
901 if (src->stats.transit != -1) {
902 if (transit > src->stats.transit)
903 diff = transit - src->stats.transit;
905 diff = src->stats.transit - transit;
909 src->stats.transit = transit;
911 /* update jitter, the value we store is scaled up so we can keep precision. */
912 src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
914 src->stats.prev_rtptime = src->stats.last_rtptime;
915 src->stats.last_rtptime = rtparrival;
917 GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
918 rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
925 GST_WARNING ("cannot get current running_time");
930 GST_WARNING ("cannot get clock-rate for pt %d", pt);
936 init_seq (RTPSource * src, guint16 seq)
938 src->stats.base_seq = seq;
939 src->stats.max_seq = seq;
940 src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
941 src->stats.cycles = 0;
942 src->stats.packets_received = 0;
943 src->stats.octets_received = 0;
944 src->stats.bytes_received = 0;
945 src->stats.prev_received = 0;
946 src->stats.prev_expected = 0;
948 GST_DEBUG ("base_seq %d", seq);
951 #define BITRATE_INTERVAL (2 * GST_SECOND)
954 do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
955 guint64 * bytes_handled)
959 if (src->prev_rtime) {
960 elapsed = running_time - src->prev_rtime;
962 if (elapsed > BITRATE_INTERVAL) {
965 rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
967 GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
968 ", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
970 if (src->bitrate == 0)
973 src->bitrate = ((src->bitrate * 3) + rate) / 4;
975 src->prev_rtime = running_time;
979 GST_LOG ("Reset bitrate measurement");
980 src->prev_rtime = running_time;
986 update_receiver_stats (RTPSource * src, RTPPacketInfo * pinfo)
988 guint16 seqnr, udelta;
989 RTPSourceStats *stats;
994 seqnr = pinfo->seqnum;
996 if (stats->cycles == -1) {
997 GST_DEBUG ("received first packet");
998 /* first time we heard of this source */
999 init_seq (src, seqnr);
1000 src->stats.max_seq = seqnr - 1;
1001 src->curr_probation = src->probation;
1004 udelta = seqnr - stats->max_seq;
1006 /* if we are still on probation, check seqnum */
1007 if (src->curr_probation) {
1008 expected = src->stats.max_seq + 1;
1010 /* when in probation, we require consecutive seqnums */
1011 if (seqnr == expected) {
1012 /* expected packet */
1013 GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
1014 src->curr_probation--;
1015 src->stats.max_seq = seqnr;
1016 if (src->curr_probation == 0) {
1017 GST_DEBUG ("probation done!");
1018 init_seq (src, seqnr);
1022 GST_DEBUG ("probation %d: queue packet", src->curr_probation);
1023 /* when still in probation, keep packets in a list. */
1024 g_queue_push_tail (src->packets, pinfo->data);
1026 /* remove packets from queue if there are too many */
1027 while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
1028 q = g_queue_pop_head (src->packets);
1029 gst_buffer_unref (q);
1034 /* unexpected seqnum in probation */
1035 goto probation_seqnum;
1037 } else if (udelta < RTP_MAX_DROPOUT) {
1038 /* in order, with permissible gap */
1039 if (seqnr < stats->max_seq) {
1040 /* sequence number wrapped - count another 64K cycle. */
1041 stats->cycles += RTP_SEQ_MOD;
1043 stats->max_seq = seqnr;
1044 } else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
1045 /* the sequence number made a very large jump */
1046 if (seqnr == stats->bad_seq) {
1047 /* two sequential packets -- assume that the other side
1048 * restarted without telling us so just re-sync
1049 * (i.e., pretend this was the first packet). */
1050 init_seq (src, seqnr);
1052 /* unacceptable jump */
1053 stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
1057 /* duplicate or reordered packet, will be filtered by jitterbuffer. */
1058 GST_WARNING ("duplicate or reordered packet (seqnr %d)", seqnr);
1061 src->stats.octets_received += pinfo->payload_len;
1062 src->stats.bytes_received += pinfo->bytes;
1063 src->stats.packets_received++;
1064 /* for the bitrate estimation */
1065 src->bytes_received += pinfo->payload_len;
1067 GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
1068 seqnr, src->stats.packets_received, src->stats.octets_received);
1079 GST_WARNING ("unacceptable seqnum received");
1084 GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
1085 src->curr_probation = src->probation;
1086 src->stats.max_seq = seqnr;
1092 * rtp_source_process_rtp:
1093 * @src: an #RTPSource
1094 * @pinfo: an #RTPPacketInfo
1096 * Let @src handle the incomming RTP packet described in @pinfo.
1098 * Returns: a #GstFlowReturn.
1101 rtp_source_process_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1103 GstFlowReturn result;
1105 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1106 g_return_val_if_fail (pinfo != NULL, GST_FLOW_ERROR);
1108 if (!update_receiver_stats (src, pinfo))
1111 /* the source that sent the packet must be a sender */
1112 src->is_sender = TRUE;
1113 src->validated = TRUE;
1115 do_bitrate_estimation (src, pinfo->running_time, &src->bytes_received);
1117 /* calculate jitter for the stats */
1118 calculate_jitter (src, pinfo);
1120 /* we're ready to push the RTP packet now */
1121 result = push_packet (src, pinfo->data);
1128 * rtp_source_mark_bye:
1129 * @src: an #RTPSource
1130 * @reason: the reason for leaving
1132 * Mark @src in the BYE state. This can happen when the source wants to
1133 * leave the sesssion or when a BYE packets has been received.
1135 * This will make the source inactive.
1138 rtp_source_mark_bye (RTPSource * src, const gchar * reason)
1140 g_return_if_fail (RTP_IS_SOURCE (src));
1142 GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
1143 GST_STR_NULL (reason));
1145 /* copy the reason and mark as bye */
1146 g_free (src->bye_reason);
1147 src->bye_reason = g_strdup (reason);
1148 src->marked_bye = TRUE;
1152 * rtp_source_send_rtp:
1153 * @src: an #RTPSource
1154 * @data: an RTP buffer or a list of RTP buffers
1155 * @is_list: if @data is a buffer or list
1156 * @running_time: the running time of @data
1158 * Send @data (an RTP buffer or list of buffers) originating from @src.
1159 * This will make @src a sender. This function takes ownership of @data and
1160 * modifies the SSRC in the RTP packet to that of @src when needed.
1162 * Returns: a #GstFlowReturn.
1165 rtp_source_send_rtp (RTPSource * src, RTPPacketInfo * pinfo)
1167 GstFlowReturn result;
1168 GstClockTime running_time;
1170 guint64 ext_rtptime;
1171 guint64 rt_diff, rtp_diff;
1173 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1175 /* we are a sender now */
1176 src->is_sender = TRUE;
1178 /* update stats for the SR */
1179 src->stats.packets_sent += pinfo->packets;
1180 src->stats.octets_sent += pinfo->payload_len;
1181 src->bytes_sent += pinfo->payload_len;
1182 /* we are also a receiver of our packets */
1183 update_receiver_stats (src, pinfo);
1185 running_time = pinfo->running_time;
1187 do_bitrate_estimation (src, running_time, &src->bytes_sent);
1189 rtptime = pinfo->rtptime;
1191 ext_rtptime = src->last_rtptime;
1192 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
1194 GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
1195 GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
1197 if (ext_rtptime > src->last_rtptime) {
1198 rtp_diff = ext_rtptime - src->last_rtptime;
1199 rt_diff = running_time - src->last_rtime;
1201 /* calc the diff so we can detect drift at the sender. This can also be used
1202 * to guestimate the clock rate if the NTP time is locked to the RTP
1203 * timestamps (as is the case when the capture device is providing the clock). */
1204 GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
1205 GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
1208 /* we keep track of the last received RTP timestamp and the corresponding
1209 * buffer running_time so that we can use this info when constructing SR reports */
1210 src->last_rtime = running_time;
1211 src->last_rtptime = ext_rtptime;
1214 if (!src->callbacks.push_rtp)
1217 GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT,
1218 pinfo->is_list ? "list" : "packet", src->stats.packets_sent);
1220 result = src->callbacks.push_rtp (src, pinfo->data, src->user_data);
1228 GST_WARNING ("no callback installed, dropping packet");
1234 * rtp_source_process_sr:
1235 * @src: an #RTPSource
1236 * @time: time of packet arrival
1237 * @ntptime: the NTP time in 32.32 fixed point
1238 * @rtptime: the RTP time
1239 * @packet_count: the packet count
1240 * @octet_count: the octect count
1242 * Update the sender report in @src.
1245 rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
1246 guint32 rtptime, guint32 packet_count, guint32 octet_count)
1248 RTPSenderReport *curr;
1251 g_return_if_fail (RTP_IS_SOURCE (src));
1253 GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
1254 ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
1255 (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
1256 packet_count, octet_count);
1258 curridx = src->stats.curr_sr ^ 1;
1259 curr = &src->stats.sr[curridx];
1261 /* this is a sender now */
1262 src->is_sender = TRUE;
1264 /* update current */
1265 curr->is_valid = TRUE;
1266 curr->ntptime = ntptime;
1267 curr->rtptime = rtptime;
1268 curr->packet_count = packet_count;
1269 curr->octet_count = octet_count;
1273 src->stats.curr_sr = curridx;
1275 src->stats.prev_rtcptime = src->stats.last_rtcptime;
1276 src->stats.last_rtcptime = time;
1280 * rtp_source_process_rb:
1281 * @src: an #RTPSource
1282 * @ntpnstime: the current time in nanoseconds since 1970
1283 * @fractionlost: fraction lost since last SR/RR
1284 * @packetslost: the cumululative number of packets lost
1285 * @exthighestseq: the extended last sequence number received
1286 * @jitter: the interarrival jitter
1287 * @lsr: the last SR packet from this source
1288 * @dlsr: the delay since last SR packet
1290 * Update the report block in @src.
1293 rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
1294 guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
1295 guint32 jitter, guint32 lsr, guint32 dlsr)
1297 RTPReceiverReport *curr;
1302 g_return_if_fail (RTP_IS_SOURCE (src));
1304 GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
1305 ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
1306 src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
1307 lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
1309 curridx = src->stats.curr_rr ^ 1;
1310 curr = &src->stats.rr[curridx];
1312 /* update current */
1313 curr->is_valid = TRUE;
1314 curr->fractionlost = fractionlost;
1315 curr->packetslost = packetslost;
1316 curr->exthighestseq = exthighestseq;
1317 curr->jitter = jitter;
1321 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1322 f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1323 /* calculate round trip, round the time up */
1324 ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
1327 if (A > 0 && ntp > A)
1331 curr->round_trip = A;
1333 GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
1334 A >> 16, A & 0xffff);
1337 src->stats.curr_rr = curridx;
1341 * rtp_source_get_new_sr:
1342 * @src: an #RTPSource
1343 * @ntpnstime: the current time in nanoseconds since 1970
1344 * @running_time: the current running_time of the pipeline.
1345 * @ntptime: the NTP time in 32.32 fixed point
1346 * @rtptime: the RTP time corresponding to @ntptime
1347 * @packet_count: the packet count
1348 * @octet_count: the octect count
1350 * Get new values to put into a new SR report from this source.
1352 * @running_time and @ntpnstime are captured at the same time and represent the
1353 * running time of the pipeline clock and the absolute current system time in
1354 * nanoseconds respectively. Together with the last running_time and rtp timestamp
1355 * we have observed in the source, we can generate @ntptime and @rtptime for an SR
1356 * packet. @ntptime is basically the fixed point representation of @ntpnstime
1357 * and @rtptime the associated RTP timestamp.
1359 * Returns: %TRUE on success.
1362 rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
1363 GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
1364 guint32 * packet_count, guint32 * octet_count)
1367 guint64 t_current_ntp;
1368 GstClockTimeDiff diff;
1370 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1372 /* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
1373 * and an NTP time, we can scale the RTP timestamps so that they match the
1374 * given NTP time. for scaling, we assume that the slope of the rtptime vs
1375 * running_time vs ntptime curve is close to 1, which is certainly
1376 * sufficient for the frequency at which we report SR and the rate we send
1377 * out RTP packets. */
1378 t_rtp = src->last_rtptime;
1380 GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
1381 G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
1383 if (src->clock_rate != -1) {
1384 /* get the diff between the clock running_time and the buffer running_time.
1385 * This is the elapsed time, as measured against the pipeline clock, between
1386 * when the rtp timestamp was observed and the current running_time.
1388 * We need to apply this diff to the RTP timestamp to get the RTP timestamp
1389 * for the given ntpnstime. */
1390 diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
1392 /* now translate the diff to RTP time, handle positive and negative cases.
1393 * If there is no diff, we already set rtptime correctly above. */
1395 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
1396 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1397 t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1400 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
1401 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1402 t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1405 GST_WARNING ("no clock-rate, cannot interpolate rtp time");
1408 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1409 t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1411 GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
1412 (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
1416 *ntptime = t_current_ntp;
1420 *packet_count = src->stats.packets_sent;
1422 *octet_count = src->stats.octets_sent;
1428 * rtp_source_get_new_rb:
1429 * @src: an #RTPSource
1430 * @time: the current time of the system clock
1431 * @fractionlost: fraction lost since last SR/RR
1432 * @packetslost: the cumululative number of packets lost
1433 * @exthighestseq: the extended last sequence number received
1434 * @jitter: the interarrival jitter
1435 * @lsr: the last SR packet from this source
1436 * @dlsr: the delay since last SR packet
1438 * Get new values to put into a new report block from this source.
1440 * Returns: %TRUE on success.
1443 rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
1444 guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
1445 guint32 * jitter, guint32 * lsr, guint32 * dlsr)
1447 RTPSourceStats *stats;
1448 guint64 extended_max, expected;
1449 guint64 expected_interval, received_interval, ntptime;
1450 gint64 lost, lost_interval;
1451 guint32 fraction, LSR, DLSR;
1452 GstClockTime sr_time;
1454 stats = &src->stats;
1456 extended_max = stats->cycles + stats->max_seq;
1457 expected = extended_max - stats->base_seq + 1;
1459 GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
1460 ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
1461 extended_max, expected, stats->packets_received, stats->base_seq);
1463 lost = expected - stats->packets_received;
1464 lost = CLAMP (lost, -0x800000, 0x7fffff);
1466 expected_interval = expected - stats->prev_expected;
1467 stats->prev_expected = expected;
1468 received_interval = stats->packets_received - stats->prev_received;
1469 stats->prev_received = stats->packets_received;
1471 lost_interval = expected_interval - received_interval;
1473 if (expected_interval == 0 || lost_interval <= 0)
1476 fraction = (lost_interval << 8) / expected_interval;
1478 GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
1479 /* we scaled the jitter up for additional precision */
1480 GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
1481 ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
1482 extended_max, stats->jitter >> 4);
1484 if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
1487 /* LSR is middle 32 bits of the last ntptime */
1488 LSR = (ntptime >> 16) & 0xffffffff;
1489 diff = time - sr_time;
1490 GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
1491 /* DLSR, delay since last SR is expressed in 1/65536 second units */
1492 DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
1494 /* No valid SR received, LSR/DLSR are set to 0 then */
1495 GST_DEBUG ("no valid SR received");
1499 GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
1500 DLSR >> 16, DLSR & 0xffff);
1503 *fractionlost = fraction;
1505 *packetslost = lost;
1507 *exthighestseq = extended_max;
1509 *jitter = stats->jitter >> 4;
1519 * rtp_source_get_last_sr:
1520 * @src: an #RTPSource
1521 * @time: time of packet arrival
1522 * @ntptime: the NTP time in 32.32 fixed point
1523 * @rtptime: the RTP time
1524 * @packet_count: the packet count
1525 * @octet_count: the octect count
1527 * Get the values of the last sender report as set with rtp_source_process_sr().
1529 * Returns: %TRUE if there was a valid SR report.
1532 rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
1533 guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
1535 RTPSenderReport *curr;
1537 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1539 curr = &src->stats.sr[src->stats.curr_sr];
1540 if (!curr->is_valid)
1544 *ntptime = curr->ntptime;
1546 *rtptime = curr->rtptime;
1548 *packet_count = curr->packet_count;
1550 *octet_count = curr->octet_count;
1558 * rtp_source_get_last_rb:
1559 * @src: an #RTPSource
1560 * @fractionlost: fraction lost since last SR/RR
1561 * @packetslost: the cumululative number of packets lost
1562 * @exthighestseq: the extended last sequence number received
1563 * @jitter: the interarrival jitter
1564 * @lsr: the last SR packet from this source
1565 * @dlsr: the delay since last SR packet
1566 * @round_trip: the round trip time
1568 * Get the values of the last RB report set with rtp_source_process_rb().
1570 * Returns: %TRUE if there was a valid SB report.
1573 rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
1574 gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
1575 guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
1577 RTPReceiverReport *curr;
1579 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1581 curr = &src->stats.rr[src->stats.curr_rr];
1582 if (!curr->is_valid)
1586 *fractionlost = curr->fractionlost;
1588 *packetslost = curr->packetslost;
1590 *exthighestseq = curr->exthighestseq;
1592 *jitter = curr->jitter;
1598 *round_trip = curr->round_trip;
1604 * rtp_source_find_conflicting_address:
1605 * @src: The source the packet came in
1606 * @address: address to check for
1607 * @time: The time when the packet that is possibly in conflict arrived
1609 * Checks if an address which has a conflict is already known. If it is
1610 * a known conflict, remember the time
1612 * Returns: TRUE if it was a known conflict, FALSE otherwise
1615 rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
1620 for (item = g_list_first (src->conflicting_addresses);
1621 item; item = g_list_next (item)) {
1622 RTPConflictingAddress *known_conflict = item->data;
1624 if (__g_socket_address_equal (address, known_conflict->address)) {
1625 known_conflict->time = time;
1634 * rtp_source_add_conflicting_address:
1635 * @src: The source the packet came in
1636 * @address: address to remember
1637 * @time: The time when the packet that is in conflict arrived
1639 * Adds a new conflict address
1642 rtp_source_add_conflicting_address (RTPSource * src,
1643 GSocketAddress * address, GstClockTime time)
1645 RTPConflictingAddress *new_conflict;
1647 new_conflict = g_new0 (RTPConflictingAddress, 1);
1649 new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
1650 new_conflict->time = time;
1652 src->conflicting_addresses = g_list_prepend (src->conflicting_addresses,
1657 * rtp_source_timeout:
1658 * @src: The #RTPSource
1659 * @current_time: The current time
1660 * @collision_timeout: The amount of time after which a collision is timed out
1661 * @feedback_retention_window: The running time before which retained feedback
1662 * packets have to be discarded
1664 * This is processed on each RTCP interval. It times out old collisions.
1665 * It also times out old retained feedback packets
1668 rtp_source_timeout (RTPSource * src, GstClockTime current_time,
1669 GstClockTime collision_timeout, GstClockTime feedback_retention_window)
1674 item = g_list_first (src->conflicting_addresses);
1676 RTPConflictingAddress *known_conflict = item->data;
1677 GList *next_item = g_list_next (item);
1679 if (known_conflict->time < current_time - collision_timeout) {
1682 src->conflicting_addresses =
1683 g_list_delete_link (src->conflicting_addresses, item);
1684 buf = __g_socket_address_to_string (known_conflict->address);
1685 GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
1687 g_object_unref (known_conflict->address);
1688 g_free (known_conflict);
1693 /* Time out AVPF packets that are older than the desired length */
1694 while ((pkt = g_queue_peek_tail (src->retained_feedback)) &&
1695 GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window)
1696 gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
1700 compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
1702 const GstBuffer *bufa = a;
1703 const GstBuffer *bufb = b;
1705 return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb);
1709 rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
1710 GstClockTime running_time)
1714 buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
1715 packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
1717 GST_BUFFER_TIMESTAMP (buffer) = running_time;
1719 g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
1723 rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
1725 if (g_queue_find_custom (src->retained_feedback, data, func))
1732 * @src: The #RTPSource
1735 * Register that @seqnum has not been received from @src.
1738 rtp_source_register_nack (RTPSource * src, guint16 seqnum)
1741 guint32 dword = seqnum << 16;
1744 len = src->nacks->len;
1745 for (i = 0; i < len; i++) {
1749 tdword = g_array_index (src->nacks, guint32, i);
1750 tseq = tdword >> 16;
1752 diff = gst_rtp_buffer_compare_seqnum (tseq, seqnum);
1756 /* we already have this seqnum */
1759 /* it comes before the recorded seqnum, FIXME, we could merge it
1760 * if not to far away */
1762 GST_DEBUG ("insert NACK #%u at %u", seqnum, i);
1763 g_array_insert_val (src->nacks, i, dword);
1764 } else if (diff < 16) {
1765 /* we can merge it */
1766 dword = g_array_index (src->nacks, guint32, i);
1767 dword |= 1 << (diff - 1);
1768 GST_DEBUG ("merge NACK #%u at %u with NACK #%u -> 0x%08x", seqnum, i,
1769 dword >> 16, dword);
1770 g_array_index (src->nacks, guint32, i) = dword;
1772 GST_DEBUG ("append NACK #%u", seqnum);
1773 g_array_append_val (src->nacks, dword);
1775 src->send_nack = TRUE;
1779 * @src: The #RTPSource
1780 * @n_nacks: result number of nacks
1782 * Get the registered NACKS since the last rtp_source_clear_nacks().
1784 * Returns: an array of @n_nacks seqnum values.
1787 rtp_source_get_nacks (RTPSource * src, guint * n_nacks)
1790 *n_nacks = src->nacks->len;
1792 return (guint32 *) src->nacks->data;
1796 rtp_source_clear_nacks (RTPSource * src)
1798 g_array_set_size (src->nacks, 0);
1799 src->send_nack = FALSE;