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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
22 #include <gst/rtp/gstrtpbuffer.h>
23 #include <gst/rtp/gstrtcpbuffer.h>
24 #include <gst/netbuffer/gstnetbuffer.h>
26 #include "gstrtpbin-marshal.h"
27 #include "rtpsession.h"
29 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
30 #define GST_CAT_DEFAULT rtp_session_debug
32 /* signals and args */
35 SIGNAL_GET_SOURCE_BY_SSRC,
37 SIGNAL_ON_SSRC_COLLISION,
38 SIGNAL_ON_SSRC_VALIDATED,
39 SIGNAL_ON_SSRC_ACTIVE,
42 SIGNAL_ON_BYE_TIMEOUT,
44 SIGNAL_ON_SENDER_TIMEOUT,
45 SIGNAL_ON_SENDING_RTCP,
46 SIGNAL_ON_FEEDBACK_RTCP,
50 #define DEFAULT_INTERNAL_SOURCE NULL
51 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
52 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
53 #define DEFAULT_RTCP_RR_BANDWIDTH -1
54 #define DEFAULT_RTCP_RS_BANDWIDTH -1
55 #define DEFAULT_RTCP_MTU 1400
56 #define DEFAULT_SDES NULL
57 #define DEFAULT_NUM_SOURCES 0
58 #define DEFAULT_NUM_ACTIVE_SOURCES 0
59 #define DEFAULT_SOURCES NULL
60 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
61 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
70 PROP_RTCP_RR_BANDWIDTH,
71 PROP_RTCP_RS_BANDWIDTH,
75 PROP_NUM_ACTIVE_SOURCES,
78 PROP_RTCP_MIN_INTERVAL,
79 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
83 /* update average packet size */
84 #define INIT_AVG(avg, val) \
86 #define UPDATE_AVG(avg, val) \
90 (avg) = ((val) + (15 * (avg))) >> 4;
93 /* The number RTCP intervals after which to timeout entries in the
96 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
98 /* GObject vmethods */
99 static void rtp_session_finalize (GObject * object);
100 static void rtp_session_set_property (GObject * object, guint prop_id,
101 const GValue * value, GParamSpec * pspec);
102 static void rtp_session_get_property (GObject * object, guint prop_id,
103 GValue * value, GParamSpec * pspec);
105 static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
106 GstBuffer * buffer, gboolean early);
109 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
111 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
113 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
114 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
115 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
116 const gchar * reason, GstClockTime current_time);
117 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
118 gboolean deterministic, gboolean first);
121 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
122 const GValue * handler_return, gpointer data)
124 if (g_value_get_boolean (handler_return))
125 g_value_set_boolean (return_accu, TRUE);
131 rtp_session_class_init (RTPSessionClass * klass)
133 GObjectClass *gobject_class;
135 gobject_class = (GObjectClass *) klass;
137 gobject_class->finalize = rtp_session_finalize;
138 gobject_class->set_property = rtp_session_set_property;
139 gobject_class->get_property = rtp_session_get_property;
142 * RTPSession::get-source-by-ssrc:
143 * @session: the object which received the signal
144 * @ssrc: the SSRC of the RTPSource
146 * Request the #RTPSource object with SSRC @ssrc in @session.
148 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
149 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
150 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
151 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
152 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
155 * RTPSession::on-new-ssrc:
156 * @session: the object which received the signal
157 * @src: the new RTPSource
159 * Notify of a new SSRC that entered @session.
161 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
162 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
163 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
164 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
167 * RTPSession::on-ssrc-collision:
168 * @session: the object which received the signal
169 * @src: the #RTPSource that caused a collision
171 * Notify when we have an SSRC collision
173 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
174 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
175 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
176 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
179 * RTPSession::on-ssrc-validated:
180 * @session: the object which received the signal
181 * @src: the new validated RTPSource
183 * Notify of a new SSRC that became validated.
185 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
186 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
188 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
191 * RTPSession::on-ssrc-active:
192 * @session: the object which received the signal
193 * @src: the active RTPSource
195 * Notify of a SSRC that is active, i.e., sending RTCP.
197 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
198 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
200 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
203 * RTPSession::on-ssrc-sdes:
204 * @session: the object which received the signal
205 * @src: the RTPSource
207 * Notify that a new SDES was received for SSRC.
209 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
210 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
212 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
215 * RTPSession::on-bye-ssrc:
216 * @session: the object which received the signal
217 * @src: the RTPSource that went away
219 * Notify of an SSRC that became inactive because of a BYE packet.
221 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
222 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
224 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
227 * RTPSession::on-bye-timeout:
228 * @session: the object which received the signal
229 * @src: the RTPSource that timed out
231 * Notify of an SSRC that has timed out because of BYE
233 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
234 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
236 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
239 * RTPSession::on-timeout:
240 * @session: the object which received the signal
241 * @src: the RTPSource that timed out
243 * Notify of an SSRC that has timed out
245 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
246 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
248 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
251 * RTPSession::on-sender-timeout:
252 * @session: the object which received the signal
253 * @src: the RTPSource that timed out
255 * Notify of an SSRC that was a sender but timed out and became a receiver.
257 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
258 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
259 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
260 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
264 * RTPSession::on-sending-rtcp
265 * @session: the object which received the signal
266 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
267 * @early: %TRUE if the packet is early, %FALSE if it is regular
269 * This signal is emitted before sending an RTCP packet, it can be used
270 * to add extra RTCP Packets.
272 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
273 * if suppressing it is acceptable
275 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
276 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
277 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
278 accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__POINTER_BOOLEAN,
279 G_TYPE_BOOLEAN, 2, G_TYPE_POINTER, G_TYPE_BOOLEAN);
282 * RTPSession::on-feedback-rtcp:
283 * @session: the object which received the signal
284 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
285 * %GST_RTCP_TYPE_RTPFB
286 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
287 * @sender_ssrc: The SSRC of the sender
288 * @media_ssrc: The SSRC of the media this refers to
289 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
292 * Notify that a RTCP feedback packet has been received
295 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
296 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
297 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
298 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_POINTER,
299 G_TYPE_NONE, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
302 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
303 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
304 "The internal SSRC used for the session",
305 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
307 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
308 g_param_spec_object ("internal-source", "Internal Source",
309 "The internal source element of the session",
310 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
312 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
313 g_param_spec_double ("bandwidth", "Bandwidth",
314 "The bandwidth of the session (0 for auto-discover)",
315 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
316 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
318 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
319 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
320 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
321 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
322 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
324 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
325 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
326 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
327 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
328 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
330 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
331 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
332 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
333 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
334 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
336 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
337 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
338 "The maximum size of the RTCP packets",
339 16, G_MAXINT16, DEFAULT_RTCP_MTU,
340 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 g_object_class_install_property (gobject_class, PROP_SDES,
343 g_param_spec_boxed ("sdes", "SDES",
344 "The SDES items of this session",
345 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
347 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
348 g_param_spec_uint ("num-sources", "Num Sources",
349 "The number of sources in the session", 0, G_MAXUINT,
350 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
352 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
353 g_param_spec_uint ("num-active-sources", "Num Active Sources",
354 "The number of active sources in the session", 0, G_MAXUINT,
355 DEFAULT_NUM_ACTIVE_SOURCES,
356 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
360 * Get a GValue Array of all sources in the session.
363 * <title>Getting the #RTPSources of a session
370 * g_object_get (sess, "sources", &arr, NULL);
372 * for (i = 0; i < arr->n_values; i++) {
375 * val = g_value_array_get_nth (arr, i);
376 * source = g_value_get_object (val);
378 * g_value_array_free (arr);
383 g_object_class_install_property (gobject_class, PROP_SOURCES,
384 g_param_spec_boxed ("sources", "Sources",
385 "An array of all known sources in the session",
386 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
389 g_param_spec_boolean ("favor-new", "Favor new sources",
390 "Resolve SSRC conflict in favor of new sources", FALSE,
391 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
393 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
394 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
395 "Minimum interval between Regular RTCP packet (in ns)",
396 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
397 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
399 g_object_class_install_property (gobject_class,
400 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
401 g_param_spec_uint64 ("rtcp-feedback-retention-window",
402 "RTCP Feedback retention window",
403 "Duration during which RTCP Feedback packets are retained (in ns)",
404 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
405 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
408 klass->get_source_by_ssrc =
409 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
410 klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
412 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
416 rtp_session_init (RTPSession * sess)
421 sess->lock = g_mutex_new ();
422 sess->key = g_random_int ();
426 for (i = 0; i < 32; i++) {
428 g_hash_table_new_full (NULL, NULL, NULL,
429 (GDestroyNotify) g_object_unref);
431 sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
433 rtp_stats_init_defaults (&sess->stats);
435 sess->recalc_bandwidth = TRUE;
436 sess->bandwidth = DEFAULT_BANDWIDTH;
437 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
438 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
439 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
441 /* create an active SSRC for this session manager */
442 sess->source = rtp_session_create_source (sess);
443 sess->source->validated = TRUE;
444 sess->source->internal = TRUE;
445 sess->stats.active_sources++;
446 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
448 /* default UDP header length */
449 sess->header_len = 28;
450 sess->mtu = DEFAULT_RTCP_MTU;
452 /* some default SDES entries */
453 str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
454 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
457 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
459 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
461 sess->first_rtcp = TRUE;
462 sess->allow_early = TRUE;
463 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
465 sess->rtcp_pli_requests = g_array_new (FALSE, FALSE, sizeof (guint32));
467 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
471 rtp_session_finalize (GObject * object)
476 sess = RTP_SESSION_CAST (object);
478 g_mutex_free (sess->lock);
479 for (i = 0; i < 32; i++)
480 g_hash_table_destroy (sess->ssrcs[i]);
482 g_free (sess->bye_reason);
484 g_hash_table_destroy (sess->cnames);
485 g_object_unref (sess->source);
487 g_array_free (sess->rtcp_pli_requests, TRUE);
489 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
493 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
495 GValue value = { 0 };
497 g_value_init (&value, RTP_TYPE_SOURCE);
498 g_value_take_object (&value, source);
499 /* copies the value */
500 g_value_array_append (arr, &value);
504 rtp_session_create_sources (RTPSession * sess)
509 RTP_SESSION_LOCK (sess);
510 /* get number of elements in the table */
511 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
512 /* create the result value array */
513 res = g_value_array_new (size);
515 /* and copy all values into the array */
516 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
517 RTP_SESSION_UNLOCK (sess);
523 rtp_session_set_property (GObject * object, guint prop_id,
524 const GValue * value, GParamSpec * pspec)
528 sess = RTP_SESSION (object);
531 case PROP_INTERNAL_SSRC:
532 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
535 sess->bandwidth = g_value_get_double (value);
536 sess->recalc_bandwidth = TRUE;
538 case PROP_RTCP_FRACTION:
539 sess->rtcp_bandwidth = g_value_get_double (value);
540 sess->recalc_bandwidth = TRUE;
542 case PROP_RTCP_RR_BANDWIDTH:
543 sess->rtcp_rr_bandwidth = g_value_get_int (value);
544 sess->recalc_bandwidth = TRUE;
546 case PROP_RTCP_RS_BANDWIDTH:
547 sess->rtcp_rs_bandwidth = g_value_get_int (value);
548 sess->recalc_bandwidth = TRUE;
551 sess->mtu = g_value_get_uint (value);
554 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
557 sess->favor_new = g_value_get_boolean (value);
559 case PROP_RTCP_MIN_INTERVAL:
560 rtp_stats_set_min_interval (&sess->stats,
561 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
564 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
570 rtp_session_get_property (GObject * object, guint prop_id,
571 GValue * value, GParamSpec * pspec)
575 sess = RTP_SESSION (object);
578 case PROP_INTERNAL_SSRC:
579 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
581 case PROP_INTERNAL_SOURCE:
582 g_value_take_object (value, rtp_session_get_internal_source (sess));
585 g_value_set_double (value, sess->bandwidth);
587 case PROP_RTCP_FRACTION:
588 g_value_set_double (value, sess->rtcp_bandwidth);
590 case PROP_RTCP_RR_BANDWIDTH:
591 g_value_set_int (value, sess->rtcp_rr_bandwidth);
593 case PROP_RTCP_RS_BANDWIDTH:
594 g_value_set_int (value, sess->rtcp_rs_bandwidth);
597 g_value_set_uint (value, sess->mtu);
600 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
602 case PROP_NUM_SOURCES:
603 g_value_set_uint (value, rtp_session_get_num_sources (sess));
605 case PROP_NUM_ACTIVE_SOURCES:
606 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
609 g_value_take_boxed (value, rtp_session_create_sources (sess));
612 g_value_set_boolean (value, sess->favor_new);
614 case PROP_RTCP_MIN_INTERVAL:
615 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
618 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
624 on_new_ssrc (RTPSession * sess, RTPSource * source)
626 g_object_ref (source);
627 RTP_SESSION_UNLOCK (sess);
628 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
629 RTP_SESSION_LOCK (sess);
630 g_object_unref (source);
634 on_ssrc_collision (RTPSession * sess, RTPSource * source)
636 g_object_ref (source);
637 RTP_SESSION_UNLOCK (sess);
638 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
640 RTP_SESSION_LOCK (sess);
641 g_object_unref (source);
645 on_ssrc_validated (RTPSession * sess, RTPSource * source)
647 g_object_ref (source);
648 RTP_SESSION_UNLOCK (sess);
649 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
651 RTP_SESSION_LOCK (sess);
652 g_object_unref (source);
656 on_ssrc_active (RTPSession * sess, RTPSource * source)
658 g_object_ref (source);
659 RTP_SESSION_UNLOCK (sess);
660 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
661 RTP_SESSION_LOCK (sess);
662 g_object_unref (source);
666 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
668 g_object_ref (source);
669 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
670 RTP_SESSION_UNLOCK (sess);
671 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
672 RTP_SESSION_LOCK (sess);
673 g_object_unref (source);
677 on_bye_ssrc (RTPSession * sess, RTPSource * source)
679 g_object_ref (source);
680 RTP_SESSION_UNLOCK (sess);
681 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
682 RTP_SESSION_LOCK (sess);
683 g_object_unref (source);
687 on_bye_timeout (RTPSession * sess, RTPSource * source)
689 g_object_ref (source);
690 RTP_SESSION_UNLOCK (sess);
691 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
692 RTP_SESSION_LOCK (sess);
693 g_object_unref (source);
697 on_timeout (RTPSession * sess, RTPSource * source)
699 g_object_ref (source);
700 RTP_SESSION_UNLOCK (sess);
701 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
702 RTP_SESSION_LOCK (sess);
703 g_object_unref (source);
707 on_sender_timeout (RTPSession * sess, RTPSource * source)
709 g_object_ref (source);
710 RTP_SESSION_UNLOCK (sess);
711 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
713 RTP_SESSION_LOCK (sess);
714 g_object_unref (source);
720 * Create a new session object.
722 * Returns: a new #RTPSession. g_object_unref() after usage.
725 rtp_session_new (void)
729 sess = g_object_new (RTP_TYPE_SESSION, NULL);
735 * rtp_session_set_callbacks:
736 * @sess: an #RTPSession
737 * @callbacks: callbacks to configure
738 * @user_data: user data passed in the callbacks
740 * Configure a set of callbacks to be notified of actions.
743 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
746 g_return_if_fail (RTP_IS_SESSION (sess));
748 if (callbacks->process_rtp) {
749 sess->callbacks.process_rtp = callbacks->process_rtp;
750 sess->process_rtp_user_data = user_data;
752 if (callbacks->send_rtp) {
753 sess->callbacks.send_rtp = callbacks->send_rtp;
754 sess->send_rtp_user_data = user_data;
756 if (callbacks->send_rtcp) {
757 sess->callbacks.send_rtcp = callbacks->send_rtcp;
758 sess->send_rtcp_user_data = user_data;
760 if (callbacks->sync_rtcp) {
761 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
762 sess->sync_rtcp_user_data = user_data;
764 if (callbacks->clock_rate) {
765 sess->callbacks.clock_rate = callbacks->clock_rate;
766 sess->clock_rate_user_data = user_data;
768 if (callbacks->reconsider) {
769 sess->callbacks.reconsider = callbacks->reconsider;
770 sess->reconsider_user_data = user_data;
772 if (callbacks->request_key_unit) {
773 sess->callbacks.request_key_unit = callbacks->request_key_unit;
774 sess->request_key_unit_user_data = user_data;
779 * rtp_session_set_process_rtp_callback:
780 * @sess: an #RTPSession
781 * @callback: callback to set
782 * @user_data: user data passed in the callback
784 * Configure only the process_rtp callback to be notified of the process_rtp action.
787 rtp_session_set_process_rtp_callback (RTPSession * sess,
788 RTPSessionProcessRTP callback, gpointer user_data)
790 g_return_if_fail (RTP_IS_SESSION (sess));
792 sess->callbacks.process_rtp = callback;
793 sess->process_rtp_user_data = user_data;
797 * rtp_session_set_send_rtp_callback:
798 * @sess: an #RTPSession
799 * @callback: callback to set
800 * @user_data: user data passed in the callback
802 * Configure only the send_rtp callback to be notified of the send_rtp action.
805 rtp_session_set_send_rtp_callback (RTPSession * sess,
806 RTPSessionSendRTP callback, gpointer user_data)
808 g_return_if_fail (RTP_IS_SESSION (sess));
810 sess->callbacks.send_rtp = callback;
811 sess->send_rtp_user_data = user_data;
815 * rtp_session_set_send_rtcp_callback:
816 * @sess: an #RTPSession
817 * @callback: callback to set
818 * @user_data: user data passed in the callback
820 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
823 rtp_session_set_send_rtcp_callback (RTPSession * sess,
824 RTPSessionSendRTCP callback, gpointer user_data)
826 g_return_if_fail (RTP_IS_SESSION (sess));
828 sess->callbacks.send_rtcp = callback;
829 sess->send_rtcp_user_data = user_data;
833 * rtp_session_set_sync_rtcp_callback:
834 * @sess: an #RTPSession
835 * @callback: callback to set
836 * @user_data: user data passed in the callback
838 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
841 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
842 RTPSessionSyncRTCP callback, gpointer user_data)
844 g_return_if_fail (RTP_IS_SESSION (sess));
846 sess->callbacks.sync_rtcp = callback;
847 sess->sync_rtcp_user_data = user_data;
851 * rtp_session_set_clock_rate_callback:
852 * @sess: an #RTPSession
853 * @callback: callback to set
854 * @user_data: user data passed in the callback
856 * Configure only the clock_rate callback to be notified of the clock_rate action.
859 rtp_session_set_clock_rate_callback (RTPSession * sess,
860 RTPSessionClockRate callback, gpointer user_data)
862 g_return_if_fail (RTP_IS_SESSION (sess));
864 sess->callbacks.clock_rate = callback;
865 sess->clock_rate_user_data = user_data;
869 * rtp_session_set_reconsider_callback:
870 * @sess: an #RTPSession
871 * @callback: callback to set
872 * @user_data: user data passed in the callback
874 * Configure only the reconsider callback to be notified of the reconsider action.
877 rtp_session_set_reconsider_callback (RTPSession * sess,
878 RTPSessionReconsider callback, gpointer user_data)
880 g_return_if_fail (RTP_IS_SESSION (sess));
882 sess->callbacks.reconsider = callback;
883 sess->reconsider_user_data = user_data;
887 * rtp_session_set_bandwidth:
888 * @sess: an #RTPSession
889 * @bandwidth: the bandwidth allocated
891 * Set the session bandwidth in bytes per second.
894 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
896 g_return_if_fail (RTP_IS_SESSION (sess));
898 RTP_SESSION_LOCK (sess);
899 sess->stats.bandwidth = bandwidth;
900 RTP_SESSION_UNLOCK (sess);
904 * rtp_session_get_bandwidth:
905 * @sess: an #RTPSession
907 * Get the session bandwidth.
909 * Returns: the session bandwidth.
912 rtp_session_get_bandwidth (RTPSession * sess)
916 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
918 RTP_SESSION_LOCK (sess);
919 result = sess->stats.bandwidth;
920 RTP_SESSION_UNLOCK (sess);
926 * rtp_session_set_rtcp_fraction:
927 * @sess: an #RTPSession
928 * @bandwidth: the RTCP bandwidth
930 * Set the bandwidth in bytes per second that should be used for RTCP
934 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
936 g_return_if_fail (RTP_IS_SESSION (sess));
938 RTP_SESSION_LOCK (sess);
939 sess->stats.rtcp_bandwidth = bandwidth;
940 RTP_SESSION_UNLOCK (sess);
944 * rtp_session_get_rtcp_fraction:
945 * @sess: an #RTPSession
947 * Get the session bandwidth used for RTCP.
949 * Returns: The bandwidth used for RTCP messages.
952 rtp_session_get_rtcp_fraction (RTPSession * sess)
956 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
958 RTP_SESSION_LOCK (sess);
959 result = sess->stats.rtcp_bandwidth;
960 RTP_SESSION_UNLOCK (sess);
966 * rtp_session_set_sdes_string:
967 * @sess: an #RTPSession
968 * @type: the type of the SDES item
969 * @item: a null-terminated string to set.
971 * Store an SDES item of @type in @sess.
973 * Returns: %FALSE if the data was unchanged @type is invalid.
976 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
981 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
983 RTP_SESSION_LOCK (sess);
984 result = rtp_source_set_sdes_string (sess->source, type, item);
985 RTP_SESSION_UNLOCK (sess);
991 * rtp_session_get_sdes_string:
992 * @sess: an #RTPSession
993 * @type: the type of the SDES item
995 * Get the SDES item of @type from @sess.
997 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
998 * valid. g_free() after usage.
1001 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
1005 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1007 RTP_SESSION_LOCK (sess);
1008 result = rtp_source_get_sdes_string (sess->source, type);
1009 RTP_SESSION_UNLOCK (sess);
1015 * rtp_session_get_sdes_struct:
1016 * @sess: an #RTSPSession
1018 * Get the SDES data as a #GstStructure
1020 * Returns: a GstStructure with SDES items for @sess. This function returns a
1021 * copy of the SDES structure, use gst_structure_free() after usage.
1024 rtp_session_get_sdes_struct (RTPSession * sess)
1026 const GstStructure *sdes;
1027 GstStructure *result = NULL;
1029 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1031 RTP_SESSION_LOCK (sess);
1032 sdes = rtp_source_get_sdes_struct (sess->source);
1034 result = gst_structure_copy (sdes);
1035 RTP_SESSION_UNLOCK (sess);
1041 * rtp_session_set_sdes_struct:
1042 * @sess: an #RTSPSession
1043 * @sdes: a #GstStructure
1045 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1048 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1050 g_return_if_fail (sdes);
1051 g_return_if_fail (RTP_IS_SESSION (sess));
1053 RTP_SESSION_LOCK (sess);
1054 rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
1055 RTP_SESSION_UNLOCK (sess);
1058 static GstFlowReturn
1059 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1061 GstFlowReturn result = GST_FLOW_OK;
1063 if (source == session->source) {
1064 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1066 RTP_SESSION_UNLOCK (session);
1068 if (session->callbacks.send_rtp)
1070 session->callbacks.send_rtp (session, source, data,
1071 session->send_rtp_user_data);
1073 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1076 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1077 RTP_SESSION_UNLOCK (session);
1079 if (session->callbacks.process_rtp)
1081 session->callbacks.process_rtp (session, source,
1082 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1084 gst_buffer_unref (GST_BUFFER_CAST (data));
1086 RTP_SESSION_LOCK (session);
1092 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1096 RTP_SESSION_UNLOCK (session);
1098 if (session->callbacks.clock_rate)
1100 session->callbacks.clock_rate (session, pt,
1101 session->clock_rate_user_data);
1105 RTP_SESSION_LOCK (session);
1107 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1112 static RTPSourceCallbacks callbacks = {
1113 (RTPSourcePushRTP) source_push_rtp,
1114 (RTPSourceClockRate) source_clock_rate,
1118 check_collision (RTPSession * sess, RTPSource * source,
1119 RTPArrivalStats * arrival, gboolean rtp)
1121 /* If we have no arrival address, we can't do collision checking */
1122 if (!arrival->have_address)
1125 if (sess->source != source) {
1126 GstNetAddress *from;
1129 /* This is not our local source, but lets check if two remote
1134 from = &source->rtp_from;
1135 have_from = source->have_rtp_from;
1137 from = &source->rtcp_from;
1138 have_from = source->have_rtcp_from;
1142 if (gst_netaddress_equal (from, &arrival->address)) {
1143 /* Address is the same */
1146 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1147 rtp_source_get_ssrc (source));
1148 if (sess->favor_new) {
1149 if (rtp_source_find_conflicting_address (source,
1150 &arrival->address, arrival->current_time)) {
1152 gst_netaddress_to_string (&arrival->address, buf1, 40);
1153 GST_LOG ("Known conflict on %x for %s, dropping packet",
1154 rtp_source_get_ssrc (source), buf1);
1157 gchar buf1[40], buf2[40];
1159 /* Current address is not a known conflict, lets assume this is
1160 * a new source. Save old address in possible conflict list
1162 rtp_source_add_conflicting_address (source, from,
1163 arrival->current_time);
1165 gst_netaddress_to_string (from, buf1, 40);
1166 gst_netaddress_to_string (&arrival->address, buf2, 40);
1167 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1168 " saving old as known conflict",
1169 rtp_source_get_ssrc (source), buf1, buf2);
1172 rtp_source_set_rtp_from (source, &arrival->address);
1174 rtp_source_set_rtcp_from (source, &arrival->address);
1178 /* Don't need to save old addresses, we ignore new sources */
1183 /* We don't already have a from address for RTP, just set it */
1185 rtp_source_set_rtp_from (source, &arrival->address);
1187 rtp_source_set_rtcp_from (source, &arrival->address);
1191 /* FIXME: Log 3rd party collision somehow
1192 * Maybe should be done in upper layer, only the SDES can tell us
1193 * if its a collision or a loop
1196 /* If the source has been inactive for some time, we assume that it has
1197 * simply changed its transport source address. Hence, there is no true
1198 * third-party collision - only a simulated one. */
1199 if (arrival->current_time > source->last_activity) {
1200 GstClockTime inactivity_period =
1201 arrival->current_time - source->last_activity;
1202 if (inactivity_period > 1 * GST_SECOND) {
1203 /* Use new network address */
1205 g_assert (source->have_rtp_from);
1206 rtp_source_set_rtp_from (source, &arrival->address);
1208 g_assert (source->have_rtcp_from);
1209 rtp_source_set_rtcp_from (source, &arrival->address);
1215 /* This is sending with our ssrc, is it an address we already know */
1217 if (rtp_source_find_conflicting_address (source, &arrival->address,
1218 arrival->current_time)) {
1219 /* Its a known conflict, its probably a loop, not a collision
1220 * lets just drop the incoming packet
1222 GST_DEBUG ("Our packets are being looped back to us, dropping");
1224 /* Its a new collision, lets change our SSRC */
1226 rtp_source_add_conflicting_address (source, &arrival->address,
1227 arrival->current_time);
1229 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1230 on_ssrc_collision (sess, source);
1232 rtp_session_schedule_bye_locked (sess, "SSRC Collision",
1233 arrival->current_time);
1235 sess->change_ssrc = TRUE;
1243 /* must be called with the session lock, the returned source needs to be
1244 * unreffed after usage. */
1246 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1247 RTPArrivalStats * arrival, gboolean rtp)
1252 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1253 if (source == NULL) {
1254 /* make new Source in probation and insert */
1255 source = rtp_source_new (ssrc);
1257 /* for RTP packets we need to set the source in probation. Receiving RTCP
1258 * packets of an SSRC, on the other hand, is a strong indication that we
1259 * are dealing with a valid source. */
1261 source->probation = RTP_DEFAULT_PROBATION;
1263 source->probation = 0;
1265 /* store from address, if any */
1266 if (arrival->have_address) {
1268 rtp_source_set_rtp_from (source, &arrival->address);
1270 rtp_source_set_rtcp_from (source, &arrival->address);
1273 /* configure a callback on the source */
1274 rtp_source_set_callbacks (source, &callbacks, sess);
1276 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1279 /* we have one more source now */
1280 sess->total_sources++;
1284 /* check for collision, this updates the address when not previously set */
1285 if (check_collision (sess, source, arrival, rtp)) {
1289 /* update last activity */
1290 source->last_activity = arrival->current_time;
1292 source->last_rtp_activity = arrival->current_time;
1293 g_object_ref (source);
1299 * rtp_session_get_internal_source:
1300 * @sess: a #RTPSession
1302 * Get the internal #RTPSource of @sess.
1304 * Returns: The internal #RTPSource. g_object_unref() after usage.
1307 rtp_session_get_internal_source (RTPSession * sess)
1311 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1313 result = g_object_ref (sess->source);
1319 * rtp_session_set_internal_ssrc:
1320 * @sess: a #RTPSession
1323 * Set the SSRC of @sess to @ssrc.
1326 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1328 RTP_SESSION_LOCK (sess);
1329 if (ssrc != sess->source->ssrc) {
1330 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1331 GINT_TO_POINTER (sess->source->ssrc));
1333 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1334 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1335 * packets will timeout on the old SSRC, we could potentially schedule a
1336 * BYE RTCP for the old SSRC... */
1337 sess->source->ssrc = ssrc;
1338 rtp_source_reset (sess->source);
1340 /* rehash with the new SSRC */
1341 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1342 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1344 RTP_SESSION_UNLOCK (sess);
1346 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1350 * rtp_session_get_internal_ssrc:
1351 * @sess: a #RTPSession
1353 * Get the internal SSRC of @sess.
1355 * Returns: The SSRC of the session.
1358 rtp_session_get_internal_ssrc (RTPSession * sess)
1362 RTP_SESSION_LOCK (sess);
1363 ssrc = sess->source->ssrc;
1364 RTP_SESSION_UNLOCK (sess);
1370 * rtp_session_add_source:
1371 * @sess: a #RTPSession
1372 * @src: #RTPSource to add
1374 * Add @src to @session.
1376 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1377 * existed in the session.
1380 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1382 gboolean result = FALSE;
1385 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1386 g_return_val_if_fail (src != NULL, FALSE);
1388 RTP_SESSION_LOCK (sess);
1390 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1391 GINT_TO_POINTER (src->ssrc));
1393 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1394 GINT_TO_POINTER (src->ssrc), src);
1395 /* we have one more source now */
1396 sess->total_sources++;
1399 RTP_SESSION_UNLOCK (sess);
1405 * rtp_session_get_num_sources:
1406 * @sess: an #RTPSession
1408 * Get the number of sources in @sess.
1410 * Returns: The number of sources in @sess.
1413 rtp_session_get_num_sources (RTPSession * sess)
1417 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1419 RTP_SESSION_LOCK (sess);
1420 result = sess->total_sources;
1421 RTP_SESSION_UNLOCK (sess);
1427 * rtp_session_get_num_active_sources:
1428 * @sess: an #RTPSession
1430 * Get the number of active sources in @sess. A source is considered active when
1431 * it has been validated and has not yet received a BYE RTCP message.
1433 * Returns: The number of active sources in @sess.
1436 rtp_session_get_num_active_sources (RTPSession * sess)
1440 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1442 RTP_SESSION_LOCK (sess);
1443 result = sess->stats.active_sources;
1444 RTP_SESSION_UNLOCK (sess);
1450 * rtp_session_get_source_by_ssrc:
1451 * @sess: an #RTPSession
1454 * Find the source with @ssrc in @sess.
1456 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1457 * g_object_unref() after usage.
1460 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1464 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1466 RTP_SESSION_LOCK (sess);
1468 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1470 g_object_ref (result);
1471 RTP_SESSION_UNLOCK (sess);
1477 * rtp_session_get_source_by_cname:
1478 * @sess: a #RTPSession
1481 * Find the source with @cname in @sess.
1483 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1484 * g_object_unref() after usage.
1487 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1491 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1492 g_return_val_if_fail (cname != NULL, NULL);
1494 RTP_SESSION_LOCK (sess);
1495 result = g_hash_table_lookup (sess->cnames, cname);
1497 g_object_ref (result);
1498 RTP_SESSION_UNLOCK (sess);
1503 /* should be called with the SESSION lock */
1505 rtp_session_create_new_ssrc (RTPSession * sess)
1510 ssrc = g_random_int ();
1512 /* see if it exists in the session, we're done if it doesn't */
1513 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1514 GINT_TO_POINTER (ssrc)) == NULL)
1522 * rtp_session_create_source:
1523 * @sess: an #RTPSession
1525 * Create an #RTPSource for use in @sess. This function will create a source
1526 * with an ssrc that is currently not used by any participants in the session.
1528 * Returns: an #RTPSource.
1531 rtp_session_create_source (RTPSession * sess)
1536 RTP_SESSION_LOCK (sess);
1537 ssrc = rtp_session_create_new_ssrc (sess);
1538 source = rtp_source_new (ssrc);
1539 rtp_source_set_callbacks (source, &callbacks, sess);
1540 /* we need an additional ref for the source in the hashtable */
1541 g_object_ref (source);
1542 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1544 /* we have one more source now */
1545 sess->total_sources++;
1546 RTP_SESSION_UNLOCK (sess);
1551 /* update the RTPArrivalStats structure with the current time and other bits
1552 * about the current buffer we are handling.
1553 * This function is typically called when a validated packet is received.
1554 * This function should be called with the SESSION_LOCK
1557 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1558 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1559 GstClockTime running_time)
1561 /* get time of arrival */
1562 arrival->current_time = current_time;
1563 arrival->running_time = running_time;
1565 /* get packet size including header overhead */
1566 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1569 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1571 arrival->payload_len = 0;
1574 /* for netbuffer we can store the IP address to check for collisions */
1575 arrival->have_address = GST_IS_NETBUFFER (buffer);
1576 if (arrival->have_address) {
1577 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1579 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1584 * rtp_session_process_rtp:
1585 * @sess: and #RTPSession
1586 * @buffer: an RTP buffer
1587 * @current_time: the current system time
1588 * @running_time: the running_time of @buffer
1590 * Process an RTP buffer in the session manager. This function takes ownership
1593 * Returns: a #GstFlowReturn.
1596 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1597 GstClockTime current_time, GstClockTime running_time)
1599 GstFlowReturn result;
1603 gboolean prevsender, prevactive;
1604 RTPArrivalStats arrival;
1609 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1610 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1612 if (!gst_rtp_buffer_validate (buffer))
1613 goto invalid_packet;
1615 RTP_SESSION_LOCK (sess);
1616 /* update arrival stats */
1617 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1620 /* ignore more RTP packets when we left the session */
1621 if (sess->source->received_bye)
1624 /* get SSRC and look up in session database */
1625 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1626 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1630 prevsender = RTP_SOURCE_IS_SENDER (source);
1631 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1632 oldrate = source->bitrate;
1634 /* copy available csrc for later */
1635 count = gst_rtp_buffer_get_csrc_count (buffer);
1636 /* make sure to not overflow our array. An RTP buffer can maximally contain
1638 count = MIN (count, 16);
1640 for (i = 0; i < count; i++)
1641 csrcs[i] = gst_rtp_buffer_get_csrc (buffer, i);
1643 /* let source process the packet */
1644 result = rtp_source_process_rtp (source, buffer, &arrival);
1646 /* source became active */
1647 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1648 sess->stats.active_sources++;
1649 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1650 sess->stats.active_sources);
1651 on_ssrc_validated (sess, source);
1653 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1654 sess->stats.sender_sources++;
1655 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1656 sess->stats.sender_sources);
1658 if (oldrate != source->bitrate)
1659 sess->recalc_bandwidth = TRUE;
1662 on_new_ssrc (sess, source);
1664 if (source->validated) {
1667 /* for validated sources, we add the CSRCs as well */
1668 for (i = 0; i < count; i++) {
1670 RTPSource *csrc_src;
1675 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1680 GST_DEBUG ("created new CSRC: %08x", csrc);
1681 rtp_source_set_as_csrc (csrc_src);
1682 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1683 sess->stats.active_sources++;
1684 on_new_ssrc (sess, csrc_src);
1686 g_object_unref (csrc_src);
1689 g_object_unref (source);
1691 RTP_SESSION_UNLOCK (sess);
1698 gst_buffer_unref (buffer);
1699 GST_DEBUG ("invalid RTP packet received");
1704 gst_buffer_unref (buffer);
1705 RTP_SESSION_UNLOCK (sess);
1706 GST_DEBUG ("ignoring RTP packet because we are leaving");
1711 gst_buffer_unref (buffer);
1712 RTP_SESSION_UNLOCK (sess);
1713 GST_DEBUG ("ignoring packet because its collisioning");
1719 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1720 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1724 count = gst_rtcp_packet_get_rb_count (packet);
1725 for (i = 0; i < count; i++) {
1726 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1727 guint8 fractionlost;
1730 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1731 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1733 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1735 if (ssrc == sess->source->ssrc) {
1736 /* only deal with report blocks for our session, we update the stats of
1737 * the sender of the RTCP message. We could also compare our stats against
1738 * the other sender to see if we are better or worse. */
1739 rtp_source_process_rb (source, arrival->current_time, fractionlost,
1740 packetslost, exthighestseq, jitter, lsr, dlsr);
1743 on_ssrc_active (sess, source);
1746 /* A Sender report contains statistics about how the sender is doing. This
1747 * includes timing informataion such as the relation between RTP and NTP
1748 * timestamps and the number of packets/bytes it sent to us.
1750 * In this report is also included a set of report blocks related to how this
1751 * sender is receiving data (in case we (or somebody else) is also sending stuff
1752 * to it). This info includes the packet loss, jitter and seqnum. It also
1753 * contains information to calculate the round trip time (LSR/DLSR).
1756 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1757 RTPArrivalStats * arrival, gboolean * do_sync)
1759 guint32 senderssrc, rtptime, packet_count, octet_count;
1762 gboolean created, prevsender;
1764 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1765 &packet_count, &octet_count);
1767 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1768 senderssrc, GST_TIME_ARGS (arrival->current_time));
1770 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1774 /* don't try to do lip-sync for sources that sent a BYE */
1775 if (rtp_source_received_bye (source))
1780 prevsender = RTP_SOURCE_IS_SENDER (source);
1782 /* first update the source */
1783 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1784 packet_count, octet_count);
1786 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1787 sess->stats.sender_sources++;
1788 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1789 sess->stats.sender_sources);
1793 on_new_ssrc (sess, source);
1795 rtp_session_process_rb (sess, source, packet, arrival);
1796 g_object_unref (source);
1799 /* A receiver report contains statistics about how a receiver is doing. It
1800 * includes stuff like packet loss, jitter and the seqnum it received last. It
1801 * also contains info to calculate the round trip time.
1803 * We are only interested in how the sender of this report is doing wrt to us.
1806 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1807 RTPArrivalStats * arrival)
1813 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1815 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1817 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1822 on_new_ssrc (sess, source);
1824 rtp_session_process_rb (sess, source, packet, arrival);
1825 g_object_unref (source);
1828 /* Get SDES items and store them in the SSRC */
1830 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1831 RTPArrivalStats * arrival)
1834 gboolean more_items, more_entries;
1836 items = gst_rtcp_packet_sdes_get_item_count (packet);
1837 GST_DEBUG ("got SDES packet with %d items", items);
1839 more_items = gst_rtcp_packet_sdes_first_item (packet);
1841 while (more_items) {
1843 gboolean changed, created, validated;
1847 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1849 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1853 /* find src, no probation when dealing with RTCP */
1854 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1858 sdes = gst_structure_new ("application/x-rtp-source-sdes", NULL);
1860 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1862 while (more_entries) {
1863 GstRTCPSDESType type;
1869 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1871 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1874 if (type == GST_RTCP_SDES_PRIV) {
1875 name = g_strndup ((const gchar *) &data[1], data[0]);
1877 data += data[0] + 1;
1879 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1882 value = g_strndup ((const gchar *) data, len);
1884 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1889 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1893 /* takes ownership of sdes */
1894 changed = rtp_source_set_sdes_struct (source, sdes);
1896 validated = !RTP_SOURCE_IS_ACTIVE (source);
1897 source->validated = TRUE;
1900 on_new_ssrc (sess, source);
1902 on_ssrc_validated (sess, source);
1904 on_ssrc_sdes (sess, source);
1906 g_object_unref (source);
1908 more_items = gst_rtcp_packet_sdes_next_item (packet);
1913 /* BYE is sent when a client leaves the session
1916 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1917 RTPArrivalStats * arrival)
1921 gboolean reconsider = FALSE;
1923 reason = gst_rtcp_packet_bye_get_reason (packet);
1924 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1926 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1927 for (i = 0; i < count; i++) {
1930 gboolean created, prevactive, prevsender;
1931 guint pmembers, members;
1933 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1934 GST_DEBUG ("SSRC: %08x", ssrc);
1936 /* find src and mark bye, no probation when dealing with RTCP */
1937 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1941 /* store time for when we need to time out this source */
1942 source->bye_time = arrival->current_time;
1944 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1945 prevsender = RTP_SOURCE_IS_SENDER (source);
1947 /* let the source handle the rest */
1948 rtp_source_process_bye (source, reason);
1950 pmembers = sess->stats.active_sources;
1952 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1953 sess->stats.active_sources--;
1954 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1955 sess->stats.active_sources);
1957 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1958 sess->stats.sender_sources--;
1959 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1960 sess->stats.sender_sources);
1962 members = sess->stats.active_sources;
1964 if (!sess->source->received_bye && members < pmembers) {
1965 /* some members went away since the previous timeout estimate.
1966 * Perform reverse reconsideration but only when we are not scheduling a
1968 if (arrival->current_time < sess->next_rtcp_check_time) {
1969 GstClockTime time_remaining;
1971 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
1972 sess->next_rtcp_check_time =
1973 gst_util_uint64_scale (time_remaining, members, pmembers);
1975 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1976 GST_TIME_ARGS (sess->next_rtcp_check_time));
1978 sess->next_rtcp_check_time += arrival->current_time;
1980 /* mark pending reconsider. We only want to signal the reconsideration
1981 * once after we handled all the source in the bye packet */
1987 on_new_ssrc (sess, source);
1989 on_bye_ssrc (sess, source);
1991 g_object_unref (source);
1994 RTP_SESSION_UNLOCK (sess);
1995 /* notify app of reconsideration */
1996 if (sess->callbacks.reconsider)
1997 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1998 RTP_SESSION_LOCK (sess);
2004 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2005 RTPArrivalStats * arrival)
2007 GST_DEBUG ("received APP");
2012 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2013 RTPArrivalStats * arrival)
2015 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2016 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2017 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2018 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2019 guint length = 4 * (gst_rtcp_packet_get_length (packet) - 2);
2021 GST_DEBUG ("received feedback %d:%d from %08X about %08X"
2022 " with FCI of length %d", type, fbtype, sender_ssrc, media_ssrc, length);
2024 if (g_signal_has_handler_pending (sess,
2025 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2026 GstBuffer *fci = NULL;
2029 fci = gst_buffer_create_sub (packet->buffer, packet->offset + 72, length);
2030 GST_BUFFER_TIMESTAMP (fci) = arrival->running_time;
2033 RTP_SESSION_UNLOCK (sess);
2034 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2035 type, fbtype, sender_ssrc, media_ssrc, fci);
2036 RTP_SESSION_LOCK (sess);
2039 gst_buffer_unref (fci);
2042 if (sess->rtcp_feedback_retention_window) {
2043 RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
2044 GINT_TO_POINTER (media_ssrc));
2047 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2050 if (rtp_source_get_ssrc (sess->source) == media_ssrc) {
2052 case GST_RTCP_TYPE_PSFB:
2054 case GST_RTCP_PSFB_TYPE_PLI:
2055 if (sess->callbacks.request_key_unit)
2056 sess->callbacks.request_key_unit (sess, FALSE,
2057 sess->request_key_unit_user_data);
2063 case GST_RTCP_TYPE_RTPFB:
2071 * rtp_session_process_rtcp:
2072 * @sess: and #RTPSession
2073 * @buffer: an RTCP buffer
2074 * @current_time: the current system time
2076 * Process an RTCP buffer in the session manager. This function takes ownership
2079 * Returns: a #GstFlowReturn.
2082 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2083 GstClockTime current_time)
2085 GstRTCPPacket packet;
2086 gboolean more, is_bye = FALSE, do_sync = FALSE;
2087 RTPArrivalStats arrival;
2088 GstFlowReturn result = GST_FLOW_OK;
2090 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2091 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2093 if (!gst_rtcp_buffer_validate (buffer))
2094 goto invalid_packet;
2096 GST_DEBUG ("received RTCP packet");
2098 RTP_SESSION_LOCK (sess);
2099 /* update arrival stats */
2100 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1);
2105 /* start processing the compound packet */
2106 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
2110 type = gst_rtcp_packet_get_type (&packet);
2112 /* when we are leaving the session, we should ignore all non-BYE messages */
2113 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
2114 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2119 case GST_RTCP_TYPE_SR:
2120 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2122 case GST_RTCP_TYPE_RR:
2123 rtp_session_process_rr (sess, &packet, &arrival);
2125 case GST_RTCP_TYPE_SDES:
2126 rtp_session_process_sdes (sess, &packet, &arrival);
2128 case GST_RTCP_TYPE_BYE:
2130 /* don't try to attempt lip-sync anymore for streams with a BYE */
2132 rtp_session_process_bye (sess, &packet, &arrival);
2134 case GST_RTCP_TYPE_APP:
2135 rtp_session_process_app (sess, &packet, &arrival);
2137 case GST_RTCP_TYPE_RTPFB:
2138 case GST_RTCP_TYPE_PSFB:
2139 rtp_session_process_feedback (sess, &packet, &arrival);
2142 GST_WARNING ("got unknown RTCP packet");
2146 more = gst_rtcp_packet_move_to_next (&packet);
2149 /* if we are scheduling a BYE, we only want to count bye packets, else we
2150 * count everything */
2151 if (sess->source->received_bye) {
2153 sess->stats.bye_members++;
2154 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2157 /* keep track of average packet size */
2158 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2160 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2161 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2162 RTP_SESSION_UNLOCK (sess);
2164 /* notify caller of sr packets in the callback */
2165 if (do_sync && sess->callbacks.sync_rtcp) {
2166 /* make writable, we might want to change the buffer */
2167 buffer = gst_buffer_make_metadata_writable (buffer);
2169 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
2170 sess->sync_rtcp_user_data);
2172 gst_buffer_unref (buffer);
2179 GST_DEBUG ("invalid RTCP packet received");
2180 gst_buffer_unref (buffer);
2185 gst_buffer_unref (buffer);
2186 RTP_SESSION_UNLOCK (sess);
2187 GST_DEBUG ("ignoring RTP packet because we left");
2193 * rtp_session_send_rtp:
2194 * @sess: an #RTPSession
2195 * @data: pointer to either an RTP buffer or a list of RTP buffers
2196 * @is_list: TRUE when @data is a buffer list
2197 * @current_time: the current system time
2198 * @running_time: the running time of @data
2200 * Send the RTP buffer in the session manager. This function takes ownership of
2203 * Returns: a #GstFlowReturn.
2206 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2207 GstClockTime current_time, GstClockTime running_time)
2209 GstFlowReturn result;
2211 gboolean prevsender;
2212 gboolean valid_packet;
2215 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2216 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2219 valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
2221 valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
2225 goto invalid_packet;
2227 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2229 RTP_SESSION_LOCK (sess);
2230 source = sess->source;
2232 /* update last activity */
2233 source->last_rtp_activity = current_time;
2235 prevsender = RTP_SOURCE_IS_SENDER (source);
2236 oldrate = source->bitrate;
2238 /* we use our own source to send */
2239 result = rtp_source_send_rtp (source, data, is_list, running_time);
2241 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2242 sess->stats.sender_sources++;
2243 if (oldrate != source->bitrate)
2244 sess->recalc_bandwidth = TRUE;
2245 RTP_SESSION_UNLOCK (sess);
2252 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2253 GST_DEBUG ("invalid RTP packet received");
2259 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2261 *bandwidth += source->bitrate;
2265 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2268 GstClockTime result;
2270 /* recalculate bandwidth when it changed */
2271 if (sess->recalc_bandwidth) {
2274 if (sess->bandwidth > 0)
2275 bandwidth = sess->bandwidth;
2277 /* If it is <= 0, then try to estimate the actual bandwidth */
2278 bandwidth = sess->source->bitrate;
2280 g_hash_table_foreach (sess->cnames, (GHFunc) add_bitrates, &bandwidth);
2284 bandwidth = RTP_STATS_BANDWIDTH;
2286 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2287 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2289 sess->recalc_bandwidth = FALSE;
2292 if (sess->source->received_bye) {
2293 result = rtp_stats_calculate_bye_interval (&sess->stats);
2295 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2296 RTP_SOURCE_IS_SENDER (sess->source), first);
2299 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2300 GST_TIME_ARGS (result), first);
2302 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2303 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2305 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2310 /* Stop the current @sess and schedule a BYE message for the other members.
2311 * One must have the session lock to call this function
2313 static GstFlowReturn
2314 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2315 GstClockTime current_time)
2317 GstFlowReturn result = GST_FLOW_OK;
2319 GstClockTime interval;
2321 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2323 source = sess->source;
2325 /* ignore more BYEs */
2326 if (source->received_bye)
2329 /* we have BYE now */
2330 source->received_bye = TRUE;
2331 /* at least one member wants to send a BYE */
2332 g_free (sess->bye_reason);
2333 sess->bye_reason = g_strdup (reason);
2334 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2335 sess->stats.bye_members = 1;
2336 sess->first_rtcp = TRUE;
2337 sess->sent_bye = FALSE;
2338 sess->allow_early = TRUE;
2340 /* reschedule transmission */
2341 sess->last_rtcp_send_time = current_time;
2342 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2343 sess->next_rtcp_check_time = current_time + interval;
2345 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2346 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2348 RTP_SESSION_UNLOCK (sess);
2349 /* notify app of reconsideration */
2350 if (sess->callbacks.reconsider)
2351 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2352 RTP_SESSION_LOCK (sess);
2359 * rtp_session_schedule_bye:
2360 * @sess: an #RTPSession
2361 * @reason: a reason or NULL
2362 * @current_time: the current system time
2364 * Stop the current @sess and schedule a BYE message for the other members.
2366 * Returns: a #GstFlowReturn.
2369 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2370 GstClockTime current_time)
2372 GstFlowReturn result = GST_FLOW_OK;
2374 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2376 RTP_SESSION_LOCK (sess);
2377 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2378 RTP_SESSION_UNLOCK (sess);
2384 * rtp_session_next_timeout:
2385 * @sess: an #RTPSession
2386 * @current_time: the current system time
2388 * Get the next time we should perform session maintenance tasks.
2390 * Returns: a time when rtp_session_on_timeout() should be called with the
2391 * current system time.
2394 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2396 GstClockTime result, interval = 0;
2398 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2400 RTP_SESSION_LOCK (sess);
2402 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2403 result = sess->next_early_rtcp_time;
2407 result = sess->next_rtcp_check_time;
2409 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
2410 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2412 if (result < current_time) {
2413 GST_DEBUG ("take current time as base");
2414 /* our previous check time expired, start counting from the current time
2416 result = current_time;
2419 if (sess->source->received_bye) {
2420 if (sess->sent_bye) {
2421 GST_DEBUG ("we sent BYE already");
2422 interval = GST_CLOCK_TIME_NONE;
2423 } else if (sess->stats.active_sources >= 50) {
2424 GST_DEBUG ("reconsider BYE, more than 50 sources");
2425 /* reconsider BYE if members >= 50 */
2426 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2429 if (sess->first_rtcp) {
2430 GST_DEBUG ("first RTCP packet");
2431 /* we are called for the first time */
2432 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2433 } else if (sess->next_rtcp_check_time < current_time) {
2434 GST_DEBUG ("old check time expired, getting new timeout");
2435 /* get a new timeout when we need to */
2436 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2440 if (interval != GST_CLOCK_TIME_NONE)
2443 result = GST_CLOCK_TIME_NONE;
2445 sess->next_rtcp_check_time = result;
2449 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2450 ", next time: %" GST_TIME_FORMAT,
2451 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2452 RTP_SESSION_UNLOCK (sess);
2461 GstClockTime current_time;
2463 GstClockTime running_time;
2464 GstClockTime interval;
2465 GstRTCPPacket packet;
2469 gboolean may_suppress;
2473 session_start_rtcp (RTPSession * sess, ReportData * data)
2475 GstRTCPPacket *packet = &data->packet;
2476 RTPSource *own = sess->source;
2478 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2480 if (RTP_SOURCE_IS_SENDER (own)) {
2483 guint32 packet_count, octet_count;
2485 /* we are a sender, create SR */
2486 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2487 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
2489 /* get latest stats */
2490 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2491 &ntptime, &rtptime, &packet_count, &octet_count);
2493 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2494 packet_count, octet_count);
2496 /* fill in sender report info */
2497 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2498 ntptime, rtptime, packet_count, octet_count);
2500 /* we are only receiver, create RR */
2501 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2502 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2503 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2507 /* construct a Sender or Receiver Report */
2509 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2511 RTPSession *sess = data->sess;
2512 GstRTCPPacket *packet = &data->packet;
2514 /* create a new buffer if needed */
2515 if (data->rtcp == NULL) {
2516 session_start_rtcp (sess, data);
2517 } else if (data->is_early) {
2518 /* Put a single RR or SR in minimal compound packets */
2521 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2522 /* only report about other sender sources */
2523 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2524 guint8 fractionlost;
2526 guint32 exthighestseq, jitter;
2530 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2531 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2533 /* store last generated RR packet */
2534 source->last_rr.is_valid = TRUE;
2535 source->last_rr.fractionlost = fractionlost;
2536 source->last_rr.packetslost = packetslost;
2537 source->last_rr.exthighestseq = exthighestseq;
2538 source->last_rr.jitter = jitter;
2539 source->last_rr.lsr = lsr;
2540 source->last_rr.dlsr = dlsr;
2542 /* packet is not yet filled, add report block for this source. */
2543 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2544 exthighestseq, jitter, lsr, dlsr);
2549 /* perform cleanup of sources that timed out */
2551 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2553 gboolean remove = FALSE;
2554 gboolean byetimeout = FALSE;
2555 gboolean sendertimeout = FALSE;
2556 gboolean is_sender, is_active;
2557 RTPSession *sess = data->sess;
2558 GstClockTime interval;
2560 is_sender = RTP_SOURCE_IS_SENDER (source);
2561 is_active = RTP_SOURCE_IS_ACTIVE (source);
2563 /* check for our own source, we don't want to delete our own source. */
2564 if (!(source == sess->source)) {
2565 if (source->received_bye) {
2566 /* if we received a BYE from the source, remove the source after some
2568 if (data->current_time > source->bye_time &&
2569 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2570 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2575 /* sources that were inactive for more than 5 times the deterministic reporting
2576 * interval get timed out. the min timeout is 5 seconds. */
2577 if (data->current_time > source->last_activity) {
2578 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2579 if (data->current_time - source->last_activity > interval) {
2580 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2581 source->ssrc, GST_TIME_ARGS (source->last_activity));
2587 /* senders that did not send for a long time become a receiver, this also
2588 * holds for our own source. */
2590 if (data->current_time > source->last_rtp_activity) {
2591 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2592 if (data->current_time - source->last_rtp_activity > interval) {
2593 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2594 GST_TIME_FORMAT, source->ssrc,
2595 GST_TIME_ARGS (source->last_rtp_activity));
2596 source->is_sender = FALSE;
2597 sess->stats.sender_sources--;
2598 sendertimeout = TRUE;
2604 sess->total_sources--;
2606 sess->stats.sender_sources--;
2608 sess->stats.active_sources--;
2611 on_bye_timeout (sess, source);
2613 on_timeout (sess, source);
2616 on_sender_timeout (sess, source);
2619 source->closing = remove;
2623 session_sdes (RTPSession * sess, ReportData * data)
2625 GstRTCPPacket *packet = &data->packet;
2626 const GstStructure *sdes;
2629 /* add SDES packet */
2630 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2632 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2634 sdes = rtp_source_get_sdes_struct (sess->source);
2636 /* add all fields in the structure, the order is not important. */
2637 n_fields = gst_structure_n_fields (sdes);
2638 for (i = 0; i < n_fields; ++i) {
2641 GstRTCPSDESType type;
2643 field = gst_structure_nth_field_name (sdes, i);
2646 value = gst_structure_get_string (sdes, field);
2649 type = gst_rtcp_sdes_name_to_type (field);
2651 /* Early packets are minimal and only include the CNAME */
2652 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2655 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2656 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2657 (const guint8 *) value);
2658 } else if (type == GST_RTCP_SDES_PRIV) {
2664 /* don't accept entries that are too big */
2665 prefix_len = strlen (field);
2666 if (prefix_len > 255)
2668 value_len = strlen (value);
2669 if (value_len > 255)
2671 data_len = 1 + prefix_len + value_len;
2675 data[0] = prefix_len;
2676 memcpy (&data[1], field, prefix_len);
2677 memcpy (&data[1 + prefix_len], value, value_len);
2679 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2683 data->has_sdes = TRUE;
2686 /* schedule a BYE packet */
2688 session_bye (RTPSession * sess, ReportData * data)
2690 GstRTCPPacket *packet = &data->packet;
2693 session_start_rtcp (sess, data);
2696 session_sdes (sess, data);
2698 /* add a BYE packet */
2699 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2700 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2701 if (sess->bye_reason)
2702 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2704 /* we have a BYE packet now */
2705 data->is_bye = TRUE;
2709 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2711 GstClockTime new_send_time, elapsed;
2713 if (data->is_early && sess->next_early_rtcp_time < current_time)
2716 /* no need to check yet */
2717 if (sess->next_rtcp_check_time > current_time) {
2718 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2719 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2720 GST_TIME_ARGS (current_time));
2724 /* get elapsed time since we last reported */
2725 elapsed = current_time - sess->last_rtcp_send_time;
2727 /* perform forward reconsideration */
2728 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2730 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2731 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2733 new_send_time += sess->last_rtcp_send_time;
2735 /* check if reconsideration */
2736 if (current_time < new_send_time) {
2737 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2738 GST_TIME_ARGS (new_send_time));
2739 /* store new check time */
2740 sess->next_rtcp_check_time = new_send_time;
2746 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2748 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2749 GST_TIME_ARGS (new_send_time));
2750 sess->next_rtcp_check_time = current_time + new_send_time;
2752 /* Apply the rules from RFC 4585 section 3.5.3 */
2753 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
2754 GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) *
2755 sess->stats.min_interval;
2757 /* This will caused the RTCP to be suppressed if no FB packets are added */
2758 if (sess->last_rtcp_send_time + T_rr_current_interval >
2759 sess->next_rtcp_check_time) {
2760 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
2761 " last: %" GST_TIME_FORMAT
2762 " + T_rr_current_interval: %" GST_TIME_FORMAT
2763 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
2764 GST_TIME_ARGS (sess->stats.min_interval),
2765 GST_TIME_ARGS (sess->last_rtcp_send_time),
2766 GST_TIME_ARGS (T_rr_current_interval),
2767 GST_TIME_ARGS (sess->next_rtcp_check_time));
2768 data->may_suppress = TRUE;
2776 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
2778 g_hash_table_insert (hash_table, key, g_object_ref (source));
2782 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
2784 return source->closing;
2788 * rtp_session_on_timeout:
2789 * @sess: an #RTPSession
2790 * @current_time: the current system time
2791 * @ntpnstime: the current NTP time in nanoseconds
2792 * @running_time: the current running_time of the pipeline
2794 * Perform maintenance actions after the timeout obtained with
2795 * rtp_session_next_timeout() expired.
2797 * This function will perform timeouts of receivers and senders, send a BYE
2798 * packet or generate RTCP packets with current session stats.
2800 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2801 * times, for each packet that should be processed.
2803 * Returns: a #GstFlowReturn.
2806 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
2807 guint64 ntpnstime, GstClockTime running_time)
2809 GstFlowReturn result = GST_FLOW_OK;
2812 GHashTable *table_copy;
2813 gboolean notify = FALSE;
2815 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2817 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2818 GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
2822 data.current_time = current_time;
2823 data.ntpnstime = ntpnstime;
2824 data.is_bye = FALSE;
2825 data.has_sdes = FALSE;
2826 data.may_suppress = FALSE;
2827 data.running_time = running_time;
2831 RTP_SESSION_LOCK (sess);
2832 /* get a new interval, we need this for various cleanups etc */
2833 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2835 /* Make a local copy of the hashtable. We need to do this because the
2836 * cleanup stage below releases the session lock. */
2837 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
2838 (GDestroyNotify) g_object_unref);
2839 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2840 (GHFunc) clone_ssrcs_hashtable, table_copy);
2842 /* Clean up the session, mark the source for removing, this might release the
2844 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
2845 g_hash_table_destroy (table_copy);
2847 /* Now remove the marked sources */
2848 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2849 (GHRFunc) remove_closing_sources, NULL);
2851 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
2852 data.is_early = TRUE;
2854 data.is_early = FALSE;
2856 /* see if we need to generate SR or RR packets */
2857 if (is_rtcp_time (sess, current_time, &data)) {
2858 if (own->received_bye) {
2859 /* generate BYE instead */
2860 GST_DEBUG ("generating BYE message");
2861 session_bye (sess, &data);
2862 sess->sent_bye = TRUE;
2864 /* loop over all known sources and do something */
2865 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2866 (GHFunc) session_report_blocks, &data);
2871 /* we keep track of the last report time in order to timeout inactive
2872 * receivers or senders */
2873 if (!data.is_early && !data.may_suppress)
2874 sess->last_rtcp_send_time = data.current_time;
2875 sess->first_rtcp = FALSE;
2876 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
2878 /* add SDES for this source when not already added */
2880 session_sdes (sess, &data);
2883 /* check for outdated collisions */
2884 GST_DEBUG ("Timing out collisions");
2885 rtp_source_timeout (sess->source, current_time,
2886 data.interval * RTCP_INTERVAL_COLLISION_TIMEOUT,
2887 running_time - sess->rtcp_feedback_retention_window);
2889 if (sess->change_ssrc) {
2890 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
2891 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2892 GINT_TO_POINTER (own->ssrc));
2894 own->ssrc = rtp_session_create_new_ssrc (sess);
2895 rtp_source_reset (own);
2897 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2898 GINT_TO_POINTER (own->ssrc), own);
2900 g_free (sess->bye_reason);
2901 sess->bye_reason = NULL;
2902 sess->sent_bye = FALSE;
2903 sess->change_ssrc = FALSE;
2905 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
2908 sess->allow_early = TRUE;
2910 RTP_SESSION_UNLOCK (sess);
2913 g_object_notify (G_OBJECT (sess), "internal-ssrc");
2915 /* push out the RTCP packet */
2917 gboolean do_not_suppress;
2919 /* Give the user a change to add its own packet */
2920 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
2921 data.rtcp, data.is_early, &do_not_suppress);
2923 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
2926 /* close the RTCP packet */
2927 gst_rtcp_buffer_end (data.rtcp);
2929 packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2931 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
2932 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
2933 sess->stats.avg_rtcp_packet_size, packet_size);
2935 sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye,
2936 sess->send_rtcp_user_data);
2938 GST_DEBUG ("freeing packet callback: %p"
2939 " do_not_suppress: %d may_suppress: %d",
2940 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
2941 gst_buffer_unref (data.rtcp);
2949 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
2950 GstClockTimeDiff max_delay)
2952 GstClockTime T_dither_max;
2954 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
2956 RTP_SESSION_LOCK (sess);
2958 /* Check if already requested */
2959 /* RFC 4585 section 3.5.2 step 2 */
2960 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
2963 /* Ignore the request a scheduled packet will be in time anyway */
2964 if (current_time + max_delay > sess->next_rtcp_check_time)
2967 /* RFC 4585 section 3.5.2 step 2b */
2968 /* If the total sources is <=2, then there is only us and one peer */
2969 if (sess->total_sources <= 2) {
2972 /* Divide by 2 because l = 0.5 */
2973 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
2977 /* RFC 4585 section 3.5.2 step 3 */
2978 if (current_time + T_dither_max > sess->next_rtcp_check_time)
2981 /* RFC 4585 section 3.5.2 step 4 */
2982 if (sess->allow_early == FALSE)
2986 /* Schedule an early transmission later */
2987 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
2990 /* If no dithering, schedule it for NOW */
2991 sess->next_early_rtcp_time = current_time;
2994 RTP_SESSION_UNLOCK (sess);
2996 /* notify app of need to send packet early
2997 * and therefore of timeout change */
2998 if (sess->callbacks.reconsider)
2999 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3005 RTP_SESSION_UNLOCK (sess);
3010 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean fir)
3017 for (i = 0; i < sess->rtcp_pli_requests->len; i++)
3018 if (ssrc == g_array_index (sess->rtcp_pli_requests, guint32, i))
3021 g_array_append_val (sess->rtcp_pli_requests, ssrc);
3025 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3027 GstRTCPPacket packet;
3029 packet.buffer = (GstBuffer *) a;
3032 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3033 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3040 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3043 gboolean ret = FALSE;
3045 RTP_SESSION_LOCK (sess);
3047 while (sess->rtcp_pli_requests->len) {
3048 GstRTCPPacket rtcppacket;
3049 guint media_ssrc = g_array_index (sess->rtcp_pli_requests, guint32, 0);
3050 RTPSource *media_src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
3051 GUINT_TO_POINTER (media_ssrc));
3053 if (media_src && !rtp_source_has_retained (media_src,
3054 has_pli_compare_func, NULL)) {
3055 if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) {
3056 gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3057 gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket,
3058 rtp_source_get_ssrc (sess->source));
3059 gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc);
3062 /* Break because the packet is full, will put next request in a
3069 g_array_remove_index (sess->rtcp_pli_requests, 0);
3072 RTP_SESSION_UNLOCK (sess);