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;
228 src->stats.cycles = -1;
229 src->stats.jitter = 0;
230 src->stats.transit = -1;
231 src->stats.curr_sr = 0;
232 src->stats.sr[0].is_valid = FALSE;
233 src->stats.curr_rr = 0;
234 src->stats.rr[0].is_valid = FALSE;
235 src->stats.prev_rtptime = GST_CLOCK_TIME_NONE;
236 src->stats.prev_rtcptime = GST_CLOCK_TIME_NONE;
237 src->stats.last_rtptime = GST_CLOCK_TIME_NONE;
238 src->stats.last_rtcptime = GST_CLOCK_TIME_NONE;
242 rtp_source_init (RTPSource * src)
244 /* sources are initialy on probation until we receive enough valid RTP
245 * packets or a valid RTCP packet */
246 src->validated = FALSE;
247 src->internal = FALSE;
248 src->probation = DEFAULT_PROBATION;
249 src->curr_probation = src->probation;
250 src->closing = FALSE;
252 src->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
255 src->clock_rate = -1;
256 src->packets = g_queue_new ();
257 src->seqnum_base = -1;
258 src->last_rtptime = -1;
260 src->retained_feedback = g_queue_new ();
262 rtp_source_reset (src);
266 rtp_conflicting_address_free (RTPConflictingAddress * addr)
268 g_object_unref (addr->address);
273 rtp_source_finalize (GObject * object)
278 src = RTP_SOURCE_CAST (object);
280 while ((buffer = g_queue_pop_head (src->packets)))
281 gst_buffer_unref (buffer);
282 g_queue_free (src->packets);
284 gst_structure_free (src->sdes);
286 g_free (src->bye_reason);
288 gst_caps_replace (&src->caps, NULL);
290 g_list_foreach (src->conflicting_addresses,
291 (GFunc) rtp_conflicting_address_free, NULL);
292 g_list_free (src->conflicting_addresses);
294 while ((buffer = g_queue_pop_head (src->retained_feedback)))
295 gst_buffer_unref (buffer);
296 g_queue_free (src->retained_feedback);
299 g_object_unref (src->rtp_from);
301 g_object_unref (src->rtcp_from);
303 G_OBJECT_CLASS (rtp_source_parent_class)->finalize (object);
306 static GstStructure *
307 rtp_source_create_stats (RTPSource * src)
310 gboolean is_sender = src->is_sender;
311 gboolean internal = src->internal;
314 guint8 fractionlost = 0;
315 gint32 packetslost = 0;
316 guint32 exthighestseq = 0;
320 guint32 round_trip = 0;
322 GstClockTime time = 0;
325 guint32 packet_count = 0;
326 guint32 octet_count = 0;
329 /* common data for all types of sources */
330 s = gst_structure_new ("application/x-rtp-source-stats",
331 "ssrc", G_TYPE_UINT, (guint) src->ssrc,
332 "internal", G_TYPE_BOOLEAN, internal,
333 "validated", G_TYPE_BOOLEAN, src->validated,
334 "received-bye", G_TYPE_BOOLEAN, src->marked_bye,
335 "is-csrc", G_TYPE_BOOLEAN, src->is_csrc,
336 "is-sender", G_TYPE_BOOLEAN, is_sender,
337 "seqnum-base", G_TYPE_INT, src->seqnum_base,
338 "clock-rate", G_TYPE_INT, src->clock_rate, NULL);
340 /* add address and port */
342 address_str = __g_socket_address_to_string (src->rtp_from);
343 gst_structure_set (s, "rtp-from", G_TYPE_STRING, address_str, NULL);
344 g_free (address_str);
346 if (src->rtcp_from) {
347 address_str = __g_socket_address_to_string (src->rtcp_from);
348 gst_structure_set (s, "rtcp-from", G_TYPE_STRING, address_str, NULL);
349 g_free (address_str);
352 gst_structure_set (s,
353 "octets-sent", G_TYPE_UINT64, src->stats.octets_sent,
354 "packets-sent", G_TYPE_UINT64, src->stats.packets_sent,
355 "octets-received", G_TYPE_UINT64, src->stats.octets_received,
356 "packets-received", G_TYPE_UINT64, src->stats.packets_received,
357 "bitrate", G_TYPE_UINT64, src->bitrate,
358 "packets-lost", G_TYPE_INT,
359 (gint) rtp_stats_get_packets_lost (&src->stats), "jitter", G_TYPE_UINT,
360 (guint) (src->stats.jitter >> 4), NULL);
362 /* get the last SR. */
363 have_sr = rtp_source_get_last_sr (src, &time, &ntptime, &rtptime,
364 &packet_count, &octet_count);
365 gst_structure_set (s,
366 "have-sr", G_TYPE_BOOLEAN, have_sr,
367 "sr-ntptime", G_TYPE_UINT64, ntptime,
368 "sr-rtptime", G_TYPE_UINT, (guint) rtptime,
369 "sr-octet-count", G_TYPE_UINT, (guint) octet_count,
370 "sr-packet-count", G_TYPE_UINT, (guint) packet_count, NULL);
373 /* get the last RB we sent */
374 gst_structure_set (s,
375 "sent-rb", G_TYPE_BOOLEAN, src->last_rr.is_valid,
376 "sent-rb-fractionlost", G_TYPE_UINT, (guint) src->last_rr.fractionlost,
377 "sent-rb-packetslost", G_TYPE_INT, (gint) src->last_rr.packetslost,
378 "sent-rb-exthighestseq", G_TYPE_UINT,
379 (guint) src->last_rr.exthighestseq, "sent-rb-jitter", G_TYPE_UINT,
380 (guint) src->last_rr.jitter, "sent-rb-lsr", G_TYPE_UINT,
381 (guint) src->last_rr.lsr, "sent-rb-dlsr", G_TYPE_UINT,
382 (guint) src->last_rr.dlsr, NULL);
384 /* get the last RB */
385 have_rb = rtp_source_get_last_rb (src, &fractionlost, &packetslost,
386 &exthighestseq, &jitter, &lsr, &dlsr, &round_trip);
388 gst_structure_set (s,
389 "have-rb", G_TYPE_BOOLEAN, have_rb,
390 "rb-fractionlost", G_TYPE_UINT, (guint) fractionlost,
391 "rb-packetslost", G_TYPE_INT, (gint) packetslost,
392 "rb-exthighestseq", G_TYPE_UINT, (guint) exthighestseq,
393 "rb-jitter", G_TYPE_UINT, (guint) jitter,
394 "rb-lsr", G_TYPE_UINT, (guint) lsr,
395 "rb-dlsr", G_TYPE_UINT, (guint) dlsr,
396 "rb-round-trip", G_TYPE_UINT, (guint) round_trip, NULL);
403 * rtp_source_get_sdes_struct:
404 * @src: an #RTPSource
406 * Get the SDES from @src. See the SDES property for more details.
408 * Returns: %GstStructure of type "application/x-rtp-source-sdes". The result is
409 * valid until the SDES items of @src are modified.
412 rtp_source_get_sdes_struct (RTPSource * src)
414 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
420 sdes_struct_compare_func (GQuark field_id, const GValue * value,
426 old = GST_STRUCTURE (user_data);
427 field = g_quark_to_string (field_id);
429 if (!gst_structure_has_field (old, field))
432 g_assert (G_VALUE_HOLDS_STRING (value));
434 return strcmp (g_value_get_string (value), gst_structure_get_string (old,
439 * rtp_source_set_sdes_struct:
440 * @src: an #RTPSource
441 * @sdes: the SDES structure
443 * Store the @sdes in @src. @sdes must be a structure of type
444 * "application/x-rtp-source-sdes", see the SDES property for more details.
446 * This function takes ownership of @sdes.
448 * Returns: %FALSE if the SDES was unchanged.
451 rtp_source_set_sdes_struct (RTPSource * src, GstStructure * sdes)
455 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
456 g_return_val_if_fail (strcmp (gst_structure_get_name (sdes),
457 "application/x-rtp-source-sdes") == 0, FALSE);
459 changed = !gst_structure_foreach (sdes, sdes_struct_compare_func, src->sdes);
462 gst_structure_free (src->sdes);
465 gst_structure_free (sdes);
472 rtp_source_set_property (GObject * object, guint prop_id,
473 const GValue * value, GParamSpec * pspec)
477 src = RTP_SOURCE (object);
481 src->ssrc = g_value_get_uint (value);
484 src->probation = g_value_get_uint (value);
487 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
493 rtp_source_get_property (GObject * object, guint prop_id,
494 GValue * value, GParamSpec * pspec)
498 src = RTP_SOURCE (object);
502 g_value_set_uint (value, rtp_source_get_ssrc (src));
505 g_value_set_boolean (value, rtp_source_is_as_csrc (src));
507 case PROP_IS_VALIDATED:
508 g_value_set_boolean (value, rtp_source_is_validated (src));
511 g_value_set_boolean (value, rtp_source_is_sender (src));
514 g_value_set_boxed (value, rtp_source_get_sdes_struct (src));
517 g_value_take_boxed (value, rtp_source_create_stats (src));
520 g_value_set_uint (value, src->probation);
523 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
532 * Create a #RTPSource with @ssrc.
534 * Returns: a new #RTPSource. Use g_object_unref() after usage.
537 rtp_source_new (guint32 ssrc)
541 src = g_object_new (RTP_TYPE_SOURCE, NULL);
548 * rtp_source_set_callbacks:
549 * @src: an #RTPSource
550 * @cb: callback functions
551 * @user_data: user data
553 * Set the callbacks for the source.
556 rtp_source_set_callbacks (RTPSource * src, RTPSourceCallbacks * cb,
559 g_return_if_fail (RTP_IS_SOURCE (src));
561 src->callbacks.push_rtp = cb->push_rtp;
562 src->callbacks.clock_rate = cb->clock_rate;
563 src->user_data = user_data;
567 * rtp_source_get_ssrc:
568 * @src: an #RTPSource
570 * Get the SSRC of @source.
572 * Returns: the SSRC of src.
575 rtp_source_get_ssrc (RTPSource * src)
579 g_return_val_if_fail (RTP_IS_SOURCE (src), 0);
587 * rtp_source_set_as_csrc:
588 * @src: an #RTPSource
590 * Configure @src as a CSRC, this will also validate @src.
593 rtp_source_set_as_csrc (RTPSource * src)
595 g_return_if_fail (RTP_IS_SOURCE (src));
597 src->validated = TRUE;
602 * rtp_source_is_as_csrc:
603 * @src: an #RTPSource
605 * Check if @src is a contributing source.
607 * Returns: %TRUE if @src is acting as a contributing source.
610 rtp_source_is_as_csrc (RTPSource * src)
614 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
616 result = src->is_csrc;
622 * rtp_source_is_active:
623 * @src: an #RTPSource
625 * Check if @src is an active source. A source is active if it has been
626 * validated and has not yet received a BYE packet
628 * Returns: %TRUE if @src is an qactive source.
631 rtp_source_is_active (RTPSource * src)
635 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
637 result = RTP_SOURCE_IS_ACTIVE (src);
643 * rtp_source_is_validated:
644 * @src: an #RTPSource
646 * Check if @src is a validated source.
648 * Returns: %TRUE if @src is a validated source.
651 rtp_source_is_validated (RTPSource * src)
655 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
657 result = src->validated;
663 * rtp_source_is_sender:
664 * @src: an #RTPSource
666 * Check if @src is a sending source.
668 * Returns: %TRUE if @src is a sending source.
671 rtp_source_is_sender (RTPSource * src)
675 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
677 result = RTP_SOURCE_IS_SENDER (src);
683 * rtp_source_is_marked_bye:
684 * @src: an #RTPSource
686 * Check if @src is marked as leaving the session with a BYE packet.
688 * Returns: %TRUE if @src has been marked BYE.
691 rtp_source_is_marked_bye (RTPSource * src)
695 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
697 result = RTP_SOURCE_IS_MARKED_BYE (src);
704 * rtp_source_get_bye_reason:
705 * @src: an #RTPSource
707 * Get the BYE reason for @src. Check if the source is marked as leaving the
708 * session with a BYE message first with rtp_source_is_marked_bye().
710 * Returns: The BYE reason or NULL when no reason was given or the source was
711 * not marked BYE yet. g_free() after usage.
714 rtp_source_get_bye_reason (RTPSource * src)
718 g_return_val_if_fail (RTP_IS_SOURCE (src), NULL);
720 result = g_strdup (src->bye_reason);
726 * rtp_source_update_caps:
727 * @src: an #RTPSource
730 * Parse @caps and store all relevant information in @source.
733 rtp_source_update_caps (RTPSource * src, GstCaps * caps)
739 /* nothing changed, return */
740 if (caps == NULL || src->caps == caps)
743 s = gst_caps_get_structure (caps, 0);
745 if (gst_structure_get_int (s, "payload", &ival))
749 GST_DEBUG ("got payload %d", src->payload);
751 if (gst_structure_get_int (s, "clock-rate", &ival))
752 src->clock_rate = ival;
754 src->clock_rate = -1;
756 GST_DEBUG ("got clock-rate %d", src->clock_rate);
758 if (gst_structure_get_uint (s, "seqnum-base", &val))
759 src->seqnum_base = val;
761 src->seqnum_base = -1;
763 GST_DEBUG ("got seqnum-base %" G_GINT32_FORMAT, src->seqnum_base);
765 gst_caps_replace (&src->caps, caps);
769 * rtp_source_set_rtp_from:
770 * @src: an #RTPSource
771 * @address: the RTP address to set
773 * Set that @src is receiving RTP packets from @address. This is used for
774 * collistion checking.
777 rtp_source_set_rtp_from (RTPSource * src, GSocketAddress * address)
779 g_return_if_fail (RTP_IS_SOURCE (src));
782 g_object_unref (src->rtp_from);
783 src->rtp_from = G_SOCKET_ADDRESS (g_object_ref (address));
787 * rtp_source_set_rtcp_from:
788 * @src: an #RTPSource
789 * @address: the RTCP address to set
791 * Set that @src is receiving RTCP packets from @address. This is used for
792 * collistion checking.
795 rtp_source_set_rtcp_from (RTPSource * src, GSocketAddress * address)
797 g_return_if_fail (RTP_IS_SOURCE (src));
800 g_object_unref (src->rtcp_from);
801 src->rtcp_from = G_SOCKET_ADDRESS (g_object_ref (address));
805 push_packet (RTPSource * src, GstBuffer * buffer)
807 GstFlowReturn ret = GST_FLOW_OK;
809 /* push queued packets first if any */
810 while (!g_queue_is_empty (src->packets)) {
811 GstBuffer *buffer = GST_BUFFER_CAST (g_queue_pop_head (src->packets));
813 GST_LOG ("pushing queued packet");
814 if (src->callbacks.push_rtp)
815 src->callbacks.push_rtp (src, buffer, src->user_data);
817 gst_buffer_unref (buffer);
819 GST_LOG ("pushing new packet");
821 if (src->callbacks.push_rtp)
822 ret = src->callbacks.push_rtp (src, buffer, src->user_data);
824 gst_buffer_unref (buffer);
830 get_clock_rate (RTPSource * src, guint8 payload)
832 if (src->payload == -1) {
833 /* first payload received, nothing was in the caps, lock on to this payload */
834 src->payload = payload;
835 GST_DEBUG ("first payload %d", payload);
836 } else if (payload != src->payload) {
837 /* we have a different payload than before, reset the clock-rate */
838 GST_DEBUG ("new payload %d", payload);
839 src->payload = payload;
840 src->clock_rate = -1;
841 src->stats.transit = -1;
844 if (src->clock_rate == -1) {
845 gint clock_rate = -1;
847 if (src->callbacks.clock_rate)
848 clock_rate = src->callbacks.clock_rate (src, payload, src->user_data);
850 GST_DEBUG ("got clock-rate %d", clock_rate);
852 src->clock_rate = clock_rate;
854 return src->clock_rate;
857 /* Jitter is the variation in the delay of received packets in a flow. It is
858 * measured by comparing the interval when RTP packets were sent to the interval
859 * at which they were received. For instance, if packet #1 and packet #2 leave
860 * 50 milliseconds apart and arrive 60 milliseconds apart, then the jitter is 10
863 calculate_jitter (RTPSource * src, GstBuffer * buffer,
864 RTPArrivalStats * arrival)
866 GstClockTime running_time;
867 guint32 rtparrival, transit, rtptime;
871 GstRTPBuffer rtp = { NULL };
873 /* get arrival time */
874 if ((running_time = arrival->running_time) == GST_CLOCK_TIME_NONE)
877 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
880 pt = gst_rtp_buffer_get_payload_type (&rtp);
882 GST_LOG ("SSRC %08x got payload %d", src->ssrc, pt);
885 if ((clock_rate = get_clock_rate (src, pt)) == -1) {
886 gst_rtp_buffer_unmap (&rtp);
890 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
892 /* convert arrival time to RTP timestamp units, truncate to 32 bits, we don't
893 * care about the absolute value, just the difference. */
894 rtparrival = gst_util_uint64_scale_int (running_time, clock_rate, GST_SECOND);
896 /* transit time is difference with RTP timestamp */
897 transit = rtparrival - rtptime;
899 /* get ABS diff with previous transit time */
900 if (src->stats.transit != -1) {
901 if (transit > src->stats.transit)
902 diff = transit - src->stats.transit;
904 diff = src->stats.transit - transit;
908 src->stats.transit = transit;
910 /* update jitter, the value we store is scaled up so we can keep precision. */
911 src->stats.jitter += diff - ((src->stats.jitter + 8) >> 4);
913 src->stats.prev_rtptime = src->stats.last_rtptime;
914 src->stats.last_rtptime = rtparrival;
916 GST_LOG ("rtparrival %u, rtptime %u, clock-rate %d, diff %d, jitter: %f",
917 rtparrival, rtptime, clock_rate, diff, (src->stats.jitter) / 16.0);
919 gst_rtp_buffer_unmap (&rtp);
925 GST_WARNING ("cannot get current running_time");
930 GST_WARNING ("invalid RTP packet");
935 GST_WARNING ("cannot get clock-rate for pt %d", pt);
941 init_seq (RTPSource * src, guint16 seq)
943 src->stats.base_seq = seq;
944 src->stats.max_seq = seq;
945 src->stats.bad_seq = RTP_SEQ_MOD + 1; /* so seq == bad_seq is false */
946 src->stats.cycles = 0;
947 src->stats.packets_received = 0;
948 src->stats.octets_received = 0;
949 src->stats.bytes_received = 0;
950 src->stats.prev_received = 0;
951 src->stats.prev_expected = 0;
953 GST_DEBUG ("base_seq %d", seq);
956 #define BITRATE_INTERVAL (2 * GST_SECOND)
959 do_bitrate_estimation (RTPSource * src, GstClockTime running_time,
960 guint64 * bytes_handled)
964 if (src->prev_rtime) {
965 elapsed = running_time - src->prev_rtime;
967 if (elapsed > BITRATE_INTERVAL) {
970 rate = gst_util_uint64_scale (*bytes_handled, 8 * GST_SECOND, elapsed);
972 GST_LOG ("Elapsed %" G_GUINT64_FORMAT ", bytes %" G_GUINT64_FORMAT
973 ", rate %" G_GUINT64_FORMAT, elapsed, *bytes_handled, rate);
975 if (src->bitrate == 0)
978 src->bitrate = ((src->bitrate * 3) + rate) / 4;
980 src->prev_rtime = running_time;
984 GST_LOG ("Reset bitrate measurement");
985 src->prev_rtime = running_time;
991 * rtp_source_process_rtp:
992 * @src: an #RTPSource
993 * @buffer: an RTP buffer
995 * Let @src handle the incomming RTP @buffer.
997 * Returns: a #GstFlowReturn.
1000 rtp_source_process_rtp (RTPSource * src, GstBuffer * buffer,
1001 RTPArrivalStats * arrival)
1003 GstFlowReturn result = GST_FLOW_OK;
1004 guint16 seqnr, udelta;
1005 RTPSourceStats *stats;
1007 GstRTPBuffer rtp = { NULL };
1009 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1010 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1012 stats = &src->stats;
1014 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1015 goto invalid_packet;
1017 seqnr = gst_rtp_buffer_get_seq (&rtp);
1018 gst_rtp_buffer_unmap (&rtp);
1020 if (stats->cycles == -1) {
1021 GST_DEBUG ("received first buffer");
1022 /* first time we heard of this source */
1023 init_seq (src, seqnr);
1024 src->stats.max_seq = seqnr - 1;
1025 src->curr_probation = src->probation;
1028 udelta = seqnr - stats->max_seq;
1030 /* if we are still on probation, check seqnum */
1031 if (src->curr_probation) {
1032 expected = src->stats.max_seq + 1;
1034 /* when in probation, we require consecutive seqnums */
1035 if (seqnr == expected) {
1036 /* expected packet */
1037 GST_DEBUG ("probation: seqnr %d == expected %d", seqnr, expected);
1038 src->curr_probation--;
1039 src->stats.max_seq = seqnr;
1040 if (src->curr_probation == 0) {
1041 GST_DEBUG ("probation done!");
1042 init_seq (src, seqnr);
1046 GST_DEBUG ("probation %d: queue buffer", src->curr_probation);
1047 /* when still in probation, keep packets in a list. */
1048 g_queue_push_tail (src->packets, buffer);
1049 /* remove packets from queue if there are too many */
1050 while (g_queue_get_length (src->packets) > RTP_MAX_PROBATION_LEN) {
1051 q = g_queue_pop_head (src->packets);
1052 gst_buffer_unref (q);
1057 /* unexpected seqnum in probation */
1058 goto probation_seqnum;
1060 } else if (udelta < RTP_MAX_DROPOUT) {
1061 /* in order, with permissible gap */
1062 if (seqnr < stats->max_seq) {
1063 /* sequence number wrapped - count another 64K cycle. */
1064 stats->cycles += RTP_SEQ_MOD;
1066 stats->max_seq = seqnr;
1067 } else if (udelta <= RTP_SEQ_MOD - RTP_MAX_MISORDER) {
1068 /* the sequence number made a very large jump */
1069 if (seqnr == stats->bad_seq) {
1070 /* two sequential packets -- assume that the other side
1071 * restarted without telling us so just re-sync
1072 * (i.e., pretend this was the first packet). */
1073 init_seq (src, seqnr);
1075 /* unacceptable jump */
1076 stats->bad_seq = (seqnr + 1) & (RTP_SEQ_MOD - 1);
1080 /* duplicate or reordered packet, will be filtered by jitterbuffer. */
1081 GST_WARNING ("duplicate or reordered packet (seqnr %d)", seqnr);
1084 src->stats.octets_received += arrival->payload_len;
1085 src->stats.bytes_received += arrival->bytes;
1086 src->stats.packets_received++;
1087 /* for the bitrate estimation */
1088 src->bytes_received += arrival->payload_len;
1089 /* the source that sent the packet must be a sender */
1090 src->is_sender = TRUE;
1091 src->validated = TRUE;
1093 do_bitrate_estimation (src, arrival->running_time, &src->bytes_received);
1095 GST_LOG ("seq %d, PC: %" G_GUINT64_FORMAT ", OC: %" G_GUINT64_FORMAT,
1096 seqnr, src->stats.packets_received, src->stats.octets_received);
1098 /* calculate jitter for the stats */
1099 calculate_jitter (src, buffer, arrival);
1101 /* we're ready to push the RTP packet now */
1102 result = push_packet (src, buffer);
1110 GST_WARNING ("invalid packet received");
1111 gst_buffer_unref (buffer);
1116 GST_WARNING ("unacceptable seqnum received");
1117 gst_buffer_unref (buffer);
1122 GST_WARNING ("probation: seqnr %d != expected %d", seqnr, expected);
1123 src->curr_probation = src->probation;
1124 src->stats.max_seq = seqnr;
1125 gst_buffer_unref (buffer);
1131 * rtp_source_mark_bye:
1132 * @src: an #RTPSource
1133 * @reason: the reason for leaving
1135 * Mark @src in the BYE state. This can happen when the source wants to
1136 * leave the sesssion or when a BYE packets has been received.
1138 * This will make the source inactive.
1141 rtp_source_mark_bye (RTPSource * src, const gchar * reason)
1143 g_return_if_fail (RTP_IS_SOURCE (src));
1145 GST_DEBUG ("marking SSRC %08x as BYE, reason: %s", src->ssrc,
1146 GST_STR_NULL (reason));
1148 /* copy the reason and mark as bye */
1149 g_free (src->bye_reason);
1150 src->bye_reason = g_strdup (reason);
1151 src->marked_bye = TRUE;
1155 * rtp_source_send_rtp:
1156 * @src: an #RTPSource
1157 * @data: an RTP buffer or a list of RTP buffers
1158 * @is_list: if @data is a buffer or list
1159 * @running_time: the running time of @data
1161 * Send @data (an RTP buffer or list of buffers) originating from @src.
1162 * This will make @src a sender. This function takes ownership of @data and
1163 * modifies the SSRC in the RTP packet to that of @src when needed.
1165 * Returns: a #GstFlowReturn.
1168 rtp_source_send_rtp (RTPSource * src, gpointer data, gboolean is_list,
1169 GstClockTime running_time)
1171 GstFlowReturn result;
1174 guint64 ext_rtptime;
1175 guint64 rt_diff, rtp_diff;
1176 GstBufferList *list = NULL;
1177 GstBuffer *buffer = NULL;
1179 GstRTPBuffer rtp = { NULL };
1181 g_return_val_if_fail (RTP_IS_SOURCE (src), GST_FLOW_ERROR);
1182 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
1185 list = GST_BUFFER_LIST_CAST (data);
1187 /* We can grab the caps from the first group, since all
1188 * groups of a buffer list have same caps. */
1189 buffer = gst_buffer_list_get (list, 0);
1193 buffer = GST_BUFFER_CAST (data);
1196 /* we are a sender now */
1197 src->is_sender = TRUE;
1202 /* Each group makes up a network packet. */
1203 packets = gst_buffer_list_length (list);
1204 for (i = 0, len = 0; i < packets; i++) {
1205 if (!gst_rtp_buffer_map (gst_buffer_list_get (list, i), GST_MAP_READ,
1207 goto invalid_packet;
1209 len += gst_rtp_buffer_get_payload_len (&rtp);
1210 gst_rtp_buffer_unmap (&rtp);
1212 /* subsequent info taken from first list member */
1213 gst_rtp_buffer_map (gst_buffer_list_get (list, 0), GST_MAP_READ, &rtp);
1216 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1217 goto invalid_packet;
1219 len = gst_rtp_buffer_get_payload_len (&rtp);
1222 /* update stats for the SR */
1223 src->stats.packets_sent += packets;
1224 src->stats.octets_sent += len;
1225 src->bytes_sent += len;
1227 do_bitrate_estimation (src, running_time, &src->bytes_sent);
1229 rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1230 ext_rtptime = src->last_rtptime;
1231 ext_rtptime = gst_rtp_buffer_ext_timestamp (&ext_rtptime, rtptime);
1233 GST_LOG ("SSRC %08x, RTP %" G_GUINT64_FORMAT ", running_time %"
1234 GST_TIME_FORMAT, src->ssrc, ext_rtptime, GST_TIME_ARGS (running_time));
1236 if (ext_rtptime > src->last_rtptime) {
1237 rtp_diff = ext_rtptime - src->last_rtptime;
1238 rt_diff = running_time - src->last_rtime;
1240 /* calc the diff so we can detect drift at the sender. This can also be used
1241 * to guestimate the clock rate if the NTP time is locked to the RTP
1242 * timestamps (as is the case when the capture device is providing the clock). */
1243 GST_LOG ("SSRC %08x, diff RTP %" G_GUINT64_FORMAT ", diff running_time %"
1244 GST_TIME_FORMAT, src->ssrc, rtp_diff, GST_TIME_ARGS (rt_diff));
1247 /* we keep track of the last received RTP timestamp and the corresponding
1248 * buffer running_time so that we can use this info when constructing SR reports */
1249 src->last_rtime = running_time;
1250 src->last_rtptime = ext_rtptime;
1253 if (!src->callbacks.push_rtp) {
1254 gst_rtp_buffer_unmap (&rtp);
1257 gst_rtp_buffer_unmap (&rtp);
1259 GST_LOG ("pushing RTP %s %" G_GUINT64_FORMAT, is_list ? "list" : "packet",
1260 src->stats.packets_sent);
1262 result = src->callbacks.push_rtp (src, data, src->user_data);
1269 GST_WARNING ("invalid packet received");
1270 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1275 GST_WARNING ("no buffers in buffer list");
1276 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1281 GST_WARNING ("no callback installed, dropping packet");
1282 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1288 * rtp_source_process_sr:
1289 * @src: an #RTPSource
1290 * @time: time of packet arrival
1291 * @ntptime: the NTP time in 32.32 fixed point
1292 * @rtptime: the RTP time
1293 * @packet_count: the packet count
1294 * @octet_count: the octect count
1296 * Update the sender report in @src.
1299 rtp_source_process_sr (RTPSource * src, GstClockTime time, guint64 ntptime,
1300 guint32 rtptime, guint32 packet_count, guint32 octet_count)
1302 RTPSenderReport *curr;
1305 g_return_if_fail (RTP_IS_SOURCE (src));
1307 GST_DEBUG ("got SR packet: SSRC %08x, NTP %08x:%08x, RTP %" G_GUINT32_FORMAT
1308 ", PC %" G_GUINT32_FORMAT ", OC %" G_GUINT32_FORMAT, src->ssrc,
1309 (guint32) (ntptime >> 32), (guint32) (ntptime & 0xffffffff), rtptime,
1310 packet_count, octet_count);
1312 curridx = src->stats.curr_sr ^ 1;
1313 curr = &src->stats.sr[curridx];
1315 /* this is a sender now */
1316 src->is_sender = TRUE;
1318 /* update current */
1319 curr->is_valid = TRUE;
1320 curr->ntptime = ntptime;
1321 curr->rtptime = rtptime;
1322 curr->packet_count = packet_count;
1323 curr->octet_count = octet_count;
1327 src->stats.curr_sr = curridx;
1329 src->stats.prev_rtcptime = src->stats.last_rtcptime;
1330 src->stats.last_rtcptime = time;
1334 * rtp_source_process_rb:
1335 * @src: an #RTPSource
1336 * @ntpnstime: the current time in nanoseconds since 1970
1337 * @fractionlost: fraction lost since last SR/RR
1338 * @packetslost: the cumululative number of packets lost
1339 * @exthighestseq: the extended last sequence number received
1340 * @jitter: the interarrival jitter
1341 * @lsr: the last SR packet from this source
1342 * @dlsr: the delay since last SR packet
1344 * Update the report block in @src.
1347 rtp_source_process_rb (RTPSource * src, guint64 ntpnstime,
1348 guint8 fractionlost, gint32 packetslost, guint32 exthighestseq,
1349 guint32 jitter, guint32 lsr, guint32 dlsr)
1351 RTPReceiverReport *curr;
1356 g_return_if_fail (RTP_IS_SOURCE (src));
1358 GST_DEBUG ("got RB packet: SSRC %08x, FL %2x, PL %d, HS %" G_GUINT32_FORMAT
1359 ", jitter %" G_GUINT32_FORMAT ", LSR %04x:%04x, DLSR %04x:%04x",
1360 src->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr >> 16,
1361 lsr & 0xffff, dlsr >> 16, dlsr & 0xffff);
1363 curridx = src->stats.curr_rr ^ 1;
1364 curr = &src->stats.rr[curridx];
1366 /* update current */
1367 curr->is_valid = TRUE;
1368 curr->fractionlost = fractionlost;
1369 curr->packetslost = packetslost;
1370 curr->exthighestseq = exthighestseq;
1371 curr->jitter = jitter;
1375 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1376 f_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1377 /* calculate round trip, round the time up */
1378 ntp = ((f_ntp + 0xffff) >> 16) & 0xffffffff;
1381 if (A > 0 && ntp > A)
1385 curr->round_trip = A;
1387 GST_DEBUG ("NTP %04x:%04x, round trip %04x:%04x", ntp >> 16, ntp & 0xffff,
1388 A >> 16, A & 0xffff);
1391 src->stats.curr_rr = curridx;
1395 * rtp_source_get_new_sr:
1396 * @src: an #RTPSource
1397 * @ntpnstime: the current time in nanoseconds since 1970
1398 * @running_time: the current running_time of the pipeline.
1399 * @ntptime: the NTP time in 32.32 fixed point
1400 * @rtptime: the RTP time corresponding to @ntptime
1401 * @packet_count: the packet count
1402 * @octet_count: the octect count
1404 * Get new values to put into a new SR report from this source.
1406 * @running_time and @ntpnstime are captured at the same time and represent the
1407 * running time of the pipeline clock and the absolute current system time in
1408 * nanoseconds respectively. Together with the last running_time and rtp timestamp
1409 * we have observed in the source, we can generate @ntptime and @rtptime for an SR
1410 * packet. @ntptime is basically the fixed point representation of @ntpnstime
1411 * and @rtptime the associated RTP timestamp.
1413 * Returns: %TRUE on success.
1416 rtp_source_get_new_sr (RTPSource * src, guint64 ntpnstime,
1417 GstClockTime running_time, guint64 * ntptime, guint32 * rtptime,
1418 guint32 * packet_count, guint32 * octet_count)
1421 guint64 t_current_ntp;
1422 GstClockTimeDiff diff;
1424 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1426 /* We last saw a buffer with last_rtptime at last_rtime. Given a running_time
1427 * and an NTP time, we can scale the RTP timestamps so that they match the
1428 * given NTP time. for scaling, we assume that the slope of the rtptime vs
1429 * running_time vs ntptime curve is close to 1, which is certainly
1430 * sufficient for the frequency at which we report SR and the rate we send
1431 * out RTP packets. */
1432 t_rtp = src->last_rtptime;
1434 GST_DEBUG ("last_rtime %" GST_TIME_FORMAT ", last_rtptime %"
1435 G_GUINT64_FORMAT, GST_TIME_ARGS (src->last_rtime), t_rtp);
1437 if (src->clock_rate != -1) {
1438 /* get the diff between the clock running_time and the buffer running_time.
1439 * This is the elapsed time, as measured against the pipeline clock, between
1440 * when the rtp timestamp was observed and the current running_time.
1442 * We need to apply this diff to the RTP timestamp to get the RTP timestamp
1443 * for the given ntpnstime. */
1444 diff = GST_CLOCK_DIFF (src->last_rtime, running_time);
1446 /* now translate the diff to RTP time, handle positive and negative cases.
1447 * If there is no diff, we already set rtptime correctly above. */
1449 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff %" GST_TIME_FORMAT,
1450 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1451 t_rtp += gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1454 GST_DEBUG ("running_time %" GST_TIME_FORMAT ", diff -%" GST_TIME_FORMAT,
1455 GST_TIME_ARGS (running_time), GST_TIME_ARGS (diff));
1456 t_rtp -= gst_util_uint64_scale_int (diff, src->clock_rate, GST_SECOND);
1459 GST_WARNING ("no clock-rate, cannot interpolate rtp time");
1462 /* convert the NTP time in nanoseconds to 32.32 fixed point */
1463 t_current_ntp = gst_util_uint64_scale (ntpnstime, (1LL << 32), GST_SECOND);
1465 GST_DEBUG ("NTP %08x:%08x, RTP %" G_GUINT32_FORMAT,
1466 (guint32) (t_current_ntp >> 32), (guint32) (t_current_ntp & 0xffffffff),
1470 *ntptime = t_current_ntp;
1474 *packet_count = src->stats.packets_sent;
1476 *octet_count = src->stats.octets_sent;
1482 * rtp_source_get_new_rb:
1483 * @src: an #RTPSource
1484 * @time: the current time of the system clock
1485 * @fractionlost: fraction lost since last SR/RR
1486 * @packetslost: the cumululative number of packets lost
1487 * @exthighestseq: the extended last sequence number received
1488 * @jitter: the interarrival jitter
1489 * @lsr: the last SR packet from this source
1490 * @dlsr: the delay since last SR packet
1492 * Get new values to put into a new report block from this source.
1494 * Returns: %TRUE on success.
1497 rtp_source_get_new_rb (RTPSource * src, GstClockTime time,
1498 guint8 * fractionlost, gint32 * packetslost, guint32 * exthighestseq,
1499 guint32 * jitter, guint32 * lsr, guint32 * dlsr)
1501 RTPSourceStats *stats;
1502 guint64 extended_max, expected;
1503 guint64 expected_interval, received_interval, ntptime;
1504 gint64 lost, lost_interval;
1505 guint32 fraction, LSR, DLSR;
1506 GstClockTime sr_time;
1508 stats = &src->stats;
1510 extended_max = stats->cycles + stats->max_seq;
1511 expected = extended_max - stats->base_seq + 1;
1513 GST_DEBUG ("ext_max %" G_GUINT64_FORMAT ", expected %" G_GUINT64_FORMAT
1514 ", received %" G_GUINT64_FORMAT ", base_seq %" G_GUINT32_FORMAT,
1515 extended_max, expected, stats->packets_received, stats->base_seq);
1517 lost = expected - stats->packets_received;
1518 lost = CLAMP (lost, -0x800000, 0x7fffff);
1520 expected_interval = expected - stats->prev_expected;
1521 stats->prev_expected = expected;
1522 received_interval = stats->packets_received - stats->prev_received;
1523 stats->prev_received = stats->packets_received;
1525 lost_interval = expected_interval - received_interval;
1527 if (expected_interval == 0 || lost_interval <= 0)
1530 fraction = (lost_interval << 8) / expected_interval;
1532 GST_DEBUG ("add RR for SSRC %08x", src->ssrc);
1533 /* we scaled the jitter up for additional precision */
1534 GST_DEBUG ("fraction %" G_GUINT32_FORMAT ", lost %" G_GINT64_FORMAT
1535 ", extseq %" G_GUINT64_FORMAT ", jitter %d", fraction, lost,
1536 extended_max, stats->jitter >> 4);
1538 if (rtp_source_get_last_sr (src, &sr_time, &ntptime, NULL, NULL, NULL)) {
1541 /* LSR is middle 32 bits of the last ntptime */
1542 LSR = (ntptime >> 16) & 0xffffffff;
1543 diff = time - sr_time;
1544 GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
1545 /* DLSR, delay since last SR is expressed in 1/65536 second units */
1546 DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
1548 /* No valid SR received, LSR/DLSR are set to 0 then */
1549 GST_DEBUG ("no valid SR received");
1553 GST_DEBUG ("LSR %04x:%04x, DLSR %04x:%04x", LSR >> 16, LSR & 0xffff,
1554 DLSR >> 16, DLSR & 0xffff);
1557 *fractionlost = fraction;
1559 *packetslost = lost;
1561 *exthighestseq = extended_max;
1563 *jitter = stats->jitter >> 4;
1573 * rtp_source_get_last_sr:
1574 * @src: an #RTPSource
1575 * @time: time of packet arrival
1576 * @ntptime: the NTP time in 32.32 fixed point
1577 * @rtptime: the RTP time
1578 * @packet_count: the packet count
1579 * @octet_count: the octect count
1581 * Get the values of the last sender report as set with rtp_source_process_sr().
1583 * Returns: %TRUE if there was a valid SR report.
1586 rtp_source_get_last_sr (RTPSource * src, GstClockTime * time, guint64 * ntptime,
1587 guint32 * rtptime, guint32 * packet_count, guint32 * octet_count)
1589 RTPSenderReport *curr;
1591 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1593 curr = &src->stats.sr[src->stats.curr_sr];
1594 if (!curr->is_valid)
1598 *ntptime = curr->ntptime;
1600 *rtptime = curr->rtptime;
1602 *packet_count = curr->packet_count;
1604 *octet_count = curr->octet_count;
1612 * rtp_source_get_last_rb:
1613 * @src: an #RTPSource
1614 * @fractionlost: fraction lost since last SR/RR
1615 * @packetslost: the cumululative number of packets lost
1616 * @exthighestseq: the extended last sequence number received
1617 * @jitter: the interarrival jitter
1618 * @lsr: the last SR packet from this source
1619 * @dlsr: the delay since last SR packet
1620 * @round_trip: the round trip time
1622 * Get the values of the last RB report set with rtp_source_process_rb().
1624 * Returns: %TRUE if there was a valid SB report.
1627 rtp_source_get_last_rb (RTPSource * src, guint8 * fractionlost,
1628 gint32 * packetslost, guint32 * exthighestseq, guint32 * jitter,
1629 guint32 * lsr, guint32 * dlsr, guint32 * round_trip)
1631 RTPReceiverReport *curr;
1633 g_return_val_if_fail (RTP_IS_SOURCE (src), FALSE);
1635 curr = &src->stats.rr[src->stats.curr_rr];
1636 if (!curr->is_valid)
1640 *fractionlost = curr->fractionlost;
1642 *packetslost = curr->packetslost;
1644 *exthighestseq = curr->exthighestseq;
1646 *jitter = curr->jitter;
1652 *round_trip = curr->round_trip;
1658 * rtp_source_find_conflicting_address:
1659 * @src: The source the packet came in
1660 * @address: address to check for
1661 * @time: The time when the packet that is possibly in conflict arrived
1663 * Checks if an address which has a conflict is already known. If it is
1664 * a known conflict, remember the time
1666 * Returns: TRUE if it was a known conflict, FALSE otherwise
1669 rtp_source_find_conflicting_address (RTPSource * src, GSocketAddress * address,
1674 for (item = g_list_first (src->conflicting_addresses);
1675 item; item = g_list_next (item)) {
1676 RTPConflictingAddress *known_conflict = item->data;
1678 if (__g_socket_address_equal (address, known_conflict->address)) {
1679 known_conflict->time = time;
1688 * rtp_source_add_conflicting_address:
1689 * @src: The source the packet came in
1690 * @address: address to remember
1691 * @time: The time when the packet that is in conflict arrived
1693 * Adds a new conflict address
1696 rtp_source_add_conflicting_address (RTPSource * src,
1697 GSocketAddress * address, GstClockTime time)
1699 RTPConflictingAddress *new_conflict;
1701 new_conflict = g_new0 (RTPConflictingAddress, 1);
1703 new_conflict->address = G_SOCKET_ADDRESS (g_object_ref (address));
1704 new_conflict->time = time;
1706 src->conflicting_addresses = g_list_prepend (src->conflicting_addresses,
1711 * rtp_source_timeout:
1712 * @src: The #RTPSource
1713 * @current_time: The current time
1714 * @collision_timeout: The amount of time after which a collision is timed out
1715 * @feedback_retention_window: The running time before which retained feedback
1716 * packets have to be discarded
1718 * This is processed on each RTCP interval. It times out old collisions.
1719 * It also times out old retained feedback packets
1722 rtp_source_timeout (RTPSource * src, GstClockTime current_time,
1723 GstClockTime collision_timeout, GstClockTime feedback_retention_window)
1728 item = g_list_first (src->conflicting_addresses);
1730 RTPConflictingAddress *known_conflict = item->data;
1731 GList *next_item = g_list_next (item);
1733 if (known_conflict->time < current_time - collision_timeout) {
1736 src->conflicting_addresses =
1737 g_list_delete_link (src->conflicting_addresses, item);
1738 buf = __g_socket_address_to_string (known_conflict->address);
1739 GST_DEBUG ("collision %p timed out: %s", known_conflict, buf);
1741 g_object_unref (known_conflict->address);
1742 g_free (known_conflict);
1747 /* Time out AVPF packets that are older than the desired length */
1748 while ((pkt = g_queue_peek_tail (src->retained_feedback)) &&
1749 GST_BUFFER_TIMESTAMP (pkt) < feedback_retention_window)
1750 gst_buffer_unref (g_queue_pop_tail (src->retained_feedback));
1754 compare_buffers (gconstpointer a, gconstpointer b, gpointer user_data)
1756 const GstBuffer *bufa = a;
1757 const GstBuffer *bufb = b;
1759 return GST_BUFFER_TIMESTAMP (bufa) - GST_BUFFER_TIMESTAMP (bufb);
1763 rtp_source_retain_rtcp_packet (RTPSource * src, GstRTCPPacket * packet,
1764 GstClockTime running_time)
1768 buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY,
1769 packet->offset, (gst_rtcp_packet_get_length (packet) + 1) * 4);
1771 GST_BUFFER_TIMESTAMP (buffer) = running_time;
1773 g_queue_insert_sorted (src->retained_feedback, buffer, compare_buffers, NULL);
1777 rtp_source_has_retained (RTPSource * src, GCompareFunc func, gconstpointer data)
1779 if (g_queue_find_custom (src->retained_feedback, data, func))