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,
48 #define DEFAULT_INTERNAL_SOURCE NULL
49 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
50 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
51 #define DEFAULT_RTCP_MTU 1400
52 #define DEFAULT_SDES NULL
53 #define DEFAULT_NUM_SOURCES 0
54 #define DEFAULT_NUM_ACTIVE_SOURCES 0
55 #define DEFAULT_SOURCES NULL
67 PROP_NUM_ACTIVE_SOURCES,
72 /* update average packet size, we keep this scaled by 16 to keep enough
74 #define UPDATE_AVG(avg, val) \
78 (avg) = ((val) + (15 * (avg))) >> 4;
80 /* The number RTCP intervals after which to timeout entries in the
83 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
85 /* GObject vmethods */
86 static void rtp_session_finalize (GObject * object);
87 static void rtp_session_set_property (GObject * object, guint prop_id,
88 const GValue * value, GParamSpec * pspec);
89 static void rtp_session_get_property (GObject * object, guint prop_id,
90 GValue * value, GParamSpec * pspec);
92 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
94 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
96 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
97 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
98 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
99 const gchar * reason, GstClockTime current_time);
100 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
101 gboolean deterministic, gboolean first);
104 rtp_session_class_init (RTPSessionClass * klass)
106 GObjectClass *gobject_class;
108 gobject_class = (GObjectClass *) klass;
110 gobject_class->finalize = rtp_session_finalize;
111 gobject_class->set_property = rtp_session_set_property;
112 gobject_class->get_property = rtp_session_get_property;
115 * RTPSession::get-source-by-ssrc:
116 * @session: the object which received the signal
117 * @ssrc: the SSRC of the RTPSource
119 * Request the #RTPSource object with SSRC @ssrc in @session.
121 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
122 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
123 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
124 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
125 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
128 * RTPSession::on-new-ssrc:
129 * @session: the object which received the signal
130 * @src: the new RTPSource
132 * Notify of a new SSRC that entered @session.
134 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
135 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
136 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
137 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
140 * RTPSession::on-ssrc-collision:
141 * @session: the object which received the signal
142 * @src: the #RTPSource that caused a collision
144 * Notify when we have an SSRC collision
146 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
147 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
148 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
149 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
152 * RTPSession::on-ssrc-validated:
153 * @session: the object which received the signal
154 * @src: the new validated RTPSource
156 * Notify of a new SSRC that became validated.
158 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
159 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
160 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
161 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
164 * RTPSession::on-ssrc-active:
165 * @session: the object which received the signal
166 * @src: the active RTPSource
168 * Notify of a SSRC that is active, i.e., sending RTCP.
170 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
171 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
172 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
173 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
176 * RTPSession::on-ssrc-sdes:
177 * @session: the object which received the signal
178 * @src: the RTPSource
180 * Notify that a new SDES was received for SSRC.
182 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
183 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
184 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
185 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
188 * RTPSession::on-bye-ssrc:
189 * @session: the object which received the signal
190 * @src: the RTPSource that went away
192 * Notify of an SSRC that became inactive because of a BYE packet.
194 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
195 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
196 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
197 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
200 * RTPSession::on-bye-timeout:
201 * @session: the object which received the signal
202 * @src: the RTPSource that timed out
204 * Notify of an SSRC that has timed out because of BYE
206 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
207 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
208 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
209 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
212 * RTPSession::on-timeout:
213 * @session: the object which received the signal
214 * @src: the RTPSource that timed out
216 * Notify of an SSRC that has timed out
218 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
219 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
220 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
221 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
224 * RTPSession::on-sender-timeout:
225 * @session: the object which received the signal
226 * @src: the RTPSource that timed out
228 * Notify of an SSRC that was a sender but timed out and became a receiver.
230 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
231 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
232 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
233 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
236 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
237 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
238 "The internal SSRC used for the session",
239 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
241 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
242 g_param_spec_object ("internal-source", "Internal Source",
243 "The internal source element of the session",
244 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
246 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
247 g_param_spec_double ("bandwidth", "Bandwidth",
248 "The bandwidth of the session",
249 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
250 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
252 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
253 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
254 "The fraction of the bandwidth used for RTCP",
255 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
256 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
258 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
259 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
260 "The maximum size of the RTCP packets",
261 16, G_MAXINT16, DEFAULT_RTCP_MTU,
262 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
264 g_object_class_install_property (gobject_class, PROP_SDES,
265 g_param_spec_boxed ("sdes", "SDES",
266 "The SDES items of this session",
267 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
269 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
270 g_param_spec_uint ("num-sources", "Num Sources",
271 "The number of sources in the session", 0, G_MAXUINT,
272 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
274 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
275 g_param_spec_uint ("num-active-sources", "Num Active Sources",
276 "The number of active sources in the session", 0, G_MAXUINT,
277 DEFAULT_NUM_ACTIVE_SOURCES,
278 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
282 * Get a GValue Array of all sources in the session.
285 * <title>Getting the #RTPSources of a session
292 * g_object_get (sess, "sources", &arr, NULL);
294 * for (i = 0; i < arr->n_values; i++) {
297 * val = g_value_array_get_nth (arr, i);
298 * source = g_value_get_object (val);
300 * g_value_array_free (arr);
305 g_object_class_install_property (gobject_class, PROP_SOURCES,
306 g_param_spec_boxed ("sources", "Sources",
307 "An array of all known sources in the session",
308 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
310 klass->get_source_by_ssrc =
311 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
313 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
317 rtp_session_init (RTPSession * sess)
322 sess->lock = g_mutex_new ();
323 sess->key = g_random_int ();
327 for (i = 0; i < 32; i++) {
329 g_hash_table_new_full (NULL, NULL, NULL,
330 (GDestroyNotify) g_object_unref);
332 sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
334 rtp_stats_init_defaults (&sess->stats);
336 /* create an active SSRC for this session manager */
337 sess->source = rtp_session_create_source (sess);
338 sess->source->validated = TRUE;
339 sess->source->internal = TRUE;
340 sess->stats.active_sources++;
342 /* default UDP header length */
343 sess->header_len = 28;
344 sess->mtu = DEFAULT_RTCP_MTU;
346 /* some default SDES entries */
347 str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
348 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
351 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
353 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
355 sess->first_rtcp = TRUE;
357 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
361 rtp_session_finalize (GObject * object)
366 sess = RTP_SESSION_CAST (object);
368 g_mutex_free (sess->lock);
369 for (i = 0; i < 32; i++)
370 g_hash_table_destroy (sess->ssrcs[i]);
372 g_free (sess->bye_reason);
374 g_hash_table_destroy (sess->cnames);
375 g_object_unref (sess->source);
377 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
381 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
383 GValue value = { 0 };
385 g_value_init (&value, RTP_TYPE_SOURCE);
386 g_value_take_object (&value, source);
387 g_value_array_append (arr, &value);
391 rtp_session_create_sources (RTPSession * sess)
396 RTP_SESSION_LOCK (sess);
397 /* get number of elements in the table */
398 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
399 /* create the result value array */
400 res = g_value_array_new (size);
402 /* and copy all values into the array */
403 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
404 RTP_SESSION_UNLOCK (sess);
410 rtp_session_set_property (GObject * object, guint prop_id,
411 const GValue * value, GParamSpec * pspec)
415 sess = RTP_SESSION (object);
418 case PROP_INTERNAL_SSRC:
419 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
422 rtp_session_set_bandwidth (sess, g_value_get_double (value));
424 case PROP_RTCP_FRACTION:
425 rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
428 sess->mtu = g_value_get_uint (value);
431 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
434 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
440 rtp_session_get_property (GObject * object, guint prop_id,
441 GValue * value, GParamSpec * pspec)
445 sess = RTP_SESSION (object);
448 case PROP_INTERNAL_SSRC:
449 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
451 case PROP_INTERNAL_SOURCE:
452 g_value_take_object (value, rtp_session_get_internal_source (sess));
455 g_value_set_double (value, rtp_session_get_bandwidth (sess));
457 case PROP_RTCP_FRACTION:
458 g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
461 g_value_set_uint (value, sess->mtu);
464 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
466 case PROP_NUM_SOURCES:
467 g_value_set_uint (value, rtp_session_get_num_sources (sess));
469 case PROP_NUM_ACTIVE_SOURCES:
470 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
473 g_value_take_boxed (value, rtp_session_create_sources (sess));
476 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
482 on_new_ssrc (RTPSession * sess, RTPSource * source)
484 g_object_ref (source);
485 RTP_SESSION_UNLOCK (sess);
486 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
487 RTP_SESSION_LOCK (sess);
488 g_object_unref (source);
492 on_ssrc_collision (RTPSession * sess, RTPSource * source)
494 g_object_ref (source);
495 RTP_SESSION_UNLOCK (sess);
496 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
498 RTP_SESSION_LOCK (sess);
499 g_object_unref (source);
503 on_ssrc_validated (RTPSession * sess, RTPSource * source)
505 g_object_ref (source);
506 RTP_SESSION_UNLOCK (sess);
507 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
509 RTP_SESSION_LOCK (sess);
510 g_object_unref (source);
514 on_ssrc_active (RTPSession * sess, RTPSource * source)
516 g_object_ref (source);
517 RTP_SESSION_UNLOCK (sess);
518 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
519 RTP_SESSION_LOCK (sess);
520 g_object_unref (source);
524 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
526 g_object_ref (source);
527 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
528 RTP_SESSION_UNLOCK (sess);
529 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
530 RTP_SESSION_LOCK (sess);
531 g_object_unref (source);
535 on_bye_ssrc (RTPSession * sess, RTPSource * source)
537 g_object_ref (source);
538 RTP_SESSION_UNLOCK (sess);
539 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
540 RTP_SESSION_LOCK (sess);
541 g_object_unref (source);
545 on_bye_timeout (RTPSession * sess, RTPSource * source)
547 g_object_ref (source);
548 RTP_SESSION_UNLOCK (sess);
549 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
550 RTP_SESSION_LOCK (sess);
551 g_object_unref (source);
555 on_timeout (RTPSession * sess, RTPSource * source)
557 g_object_ref (source);
558 RTP_SESSION_UNLOCK (sess);
559 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
560 RTP_SESSION_LOCK (sess);
561 g_object_unref (source);
565 on_sender_timeout (RTPSession * sess, RTPSource * source)
567 g_object_ref (source);
568 RTP_SESSION_UNLOCK (sess);
569 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
571 RTP_SESSION_LOCK (sess);
572 g_object_unref (source);
578 * Create a new session object.
580 * Returns: a new #RTPSession. g_object_unref() after usage.
583 rtp_session_new (void)
587 sess = g_object_new (RTP_TYPE_SESSION, NULL);
593 * rtp_session_set_callbacks:
594 * @sess: an #RTPSession
595 * @callbacks: callbacks to configure
596 * @user_data: user data passed in the callbacks
598 * Configure a set of callbacks to be notified of actions.
601 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
604 g_return_if_fail (RTP_IS_SESSION (sess));
606 if (callbacks->process_rtp) {
607 sess->callbacks.process_rtp = callbacks->process_rtp;
608 sess->process_rtp_user_data = user_data;
610 if (callbacks->send_rtp) {
611 sess->callbacks.send_rtp = callbacks->send_rtp;
612 sess->send_rtp_user_data = user_data;
614 if (callbacks->send_rtcp) {
615 sess->callbacks.send_rtcp = callbacks->send_rtcp;
616 sess->send_rtcp_user_data = user_data;
618 if (callbacks->sync_rtcp) {
619 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
620 sess->sync_rtcp_user_data = user_data;
622 if (callbacks->clock_rate) {
623 sess->callbacks.clock_rate = callbacks->clock_rate;
624 sess->clock_rate_user_data = user_data;
626 if (callbacks->reconsider) {
627 sess->callbacks.reconsider = callbacks->reconsider;
628 sess->reconsider_user_data = user_data;
633 * rtp_session_set_process_rtp_callback:
634 * @sess: an #RTPSession
635 * @callback: callback to set
636 * @user_data: user data passed in the callback
638 * Configure only the process_rtp callback to be notified of the process_rtp action.
641 rtp_session_set_process_rtp_callback (RTPSession * sess,
642 RTPSessionProcessRTP callback, gpointer user_data)
644 g_return_if_fail (RTP_IS_SESSION (sess));
646 sess->callbacks.process_rtp = callback;
647 sess->process_rtp_user_data = user_data;
651 * rtp_session_set_send_rtp_callback:
652 * @sess: an #RTPSession
653 * @callback: callback to set
654 * @user_data: user data passed in the callback
656 * Configure only the send_rtp callback to be notified of the send_rtp action.
659 rtp_session_set_send_rtp_callback (RTPSession * sess,
660 RTPSessionSendRTP callback, gpointer user_data)
662 g_return_if_fail (RTP_IS_SESSION (sess));
664 sess->callbacks.send_rtp = callback;
665 sess->send_rtp_user_data = user_data;
669 * rtp_session_set_send_rtcp_callback:
670 * @sess: an #RTPSession
671 * @callback: callback to set
672 * @user_data: user data passed in the callback
674 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
677 rtp_session_set_send_rtcp_callback (RTPSession * sess,
678 RTPSessionSendRTCP callback, gpointer user_data)
680 g_return_if_fail (RTP_IS_SESSION (sess));
682 sess->callbacks.send_rtcp = callback;
683 sess->send_rtcp_user_data = user_data;
687 * rtp_session_set_sync_rtcp_callback:
688 * @sess: an #RTPSession
689 * @callback: callback to set
690 * @user_data: user data passed in the callback
692 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
695 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
696 RTPSessionSyncRTCP callback, gpointer user_data)
698 g_return_if_fail (RTP_IS_SESSION (sess));
700 sess->callbacks.sync_rtcp = callback;
701 sess->sync_rtcp_user_data = user_data;
705 * rtp_session_set_clock_rate_callback:
706 * @sess: an #RTPSession
707 * @callback: callback to set
708 * @user_data: user data passed in the callback
710 * Configure only the clock_rate callback to be notified of the clock_rate action.
713 rtp_session_set_clock_rate_callback (RTPSession * sess,
714 RTPSessionClockRate callback, gpointer user_data)
716 g_return_if_fail (RTP_IS_SESSION (sess));
718 sess->callbacks.clock_rate = callback;
719 sess->clock_rate_user_data = user_data;
723 * rtp_session_set_reconsider_callback:
724 * @sess: an #RTPSession
725 * @callback: callback to set
726 * @user_data: user data passed in the callback
728 * Configure only the reconsider callback to be notified of the reconsider action.
731 rtp_session_set_reconsider_callback (RTPSession * sess,
732 RTPSessionReconsider callback, gpointer user_data)
734 g_return_if_fail (RTP_IS_SESSION (sess));
736 sess->callbacks.reconsider = callback;
737 sess->reconsider_user_data = user_data;
741 * rtp_session_set_bandwidth:
742 * @sess: an #RTPSession
743 * @bandwidth: the bandwidth allocated
745 * Set the session bandwidth in bytes per second.
748 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
750 g_return_if_fail (RTP_IS_SESSION (sess));
752 RTP_SESSION_LOCK (sess);
753 sess->stats.bandwidth = bandwidth;
754 RTP_SESSION_UNLOCK (sess);
758 * rtp_session_get_bandwidth:
759 * @sess: an #RTPSession
761 * Get the session bandwidth.
763 * Returns: the session bandwidth.
766 rtp_session_get_bandwidth (RTPSession * sess)
770 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
772 RTP_SESSION_LOCK (sess);
773 result = sess->stats.bandwidth;
774 RTP_SESSION_UNLOCK (sess);
780 * rtp_session_set_rtcp_fraction:
781 * @sess: an #RTPSession
782 * @bandwidth: the RTCP bandwidth
784 * Set the bandwidth that should be used for RTCP
788 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
790 g_return_if_fail (RTP_IS_SESSION (sess));
792 RTP_SESSION_LOCK (sess);
793 sess->stats.rtcp_bandwidth = bandwidth;
794 RTP_SESSION_UNLOCK (sess);
798 * rtp_session_get_rtcp_fraction:
799 * @sess: an #RTPSession
801 * Get the session bandwidth used for RTCP.
803 * Returns: The bandwidth used for RTCP messages.
806 rtp_session_get_rtcp_fraction (RTPSession * sess)
810 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
812 RTP_SESSION_LOCK (sess);
813 result = sess->stats.rtcp_bandwidth;
814 RTP_SESSION_UNLOCK (sess);
820 * rtp_session_set_sdes_string:
821 * @sess: an #RTPSession
822 * @type: the type of the SDES item
823 * @item: a null-terminated string to set.
825 * Store an SDES item of @type in @sess.
827 * Returns: %FALSE if the data was unchanged @type is invalid.
830 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
835 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
837 RTP_SESSION_LOCK (sess);
838 result = rtp_source_set_sdes_string (sess->source, type, item);
839 RTP_SESSION_UNLOCK (sess);
845 * rtp_session_get_sdes_string:
846 * @sess: an #RTPSession
847 * @type: the type of the SDES item
849 * Get the SDES item of @type from @sess.
851 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
852 * valid. g_free() after usage.
855 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
859 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
861 RTP_SESSION_LOCK (sess);
862 result = rtp_source_get_sdes_string (sess->source, type);
863 RTP_SESSION_UNLOCK (sess);
869 * rtp_session_get_sdes_struct:
870 * @sess: an #RTSPSession
872 * Get the SDES data as a #GstStructure
874 * Returns: a GstStructure with SDES items for @sess.
877 rtp_session_get_sdes_struct (RTPSession * sess)
879 GstStructure *result;
881 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
883 RTP_SESSION_LOCK (sess);
884 result = rtp_source_get_sdes_struct (sess->source);
885 RTP_SESSION_UNLOCK (sess);
891 * rtp_session_set_sdes_struct:
892 * @sess: an #RTSPSession
893 * @sdes: a #GstStructure
895 * Set the SDES data as a #GstStructure.
898 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
900 g_return_if_fail (RTP_IS_SESSION (sess));
902 RTP_SESSION_LOCK (sess);
903 rtp_source_set_sdes_struct (sess->source, sdes);
904 RTP_SESSION_UNLOCK (sess);
908 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
910 GstFlowReturn result = GST_FLOW_OK;
912 if (source == session->source) {
913 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
915 RTP_SESSION_UNLOCK (session);
917 if (session->callbacks.send_rtp)
919 session->callbacks.send_rtp (session, source, data,
920 session->send_rtp_user_data);
922 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
925 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
926 RTP_SESSION_UNLOCK (session);
928 if (session->callbacks.process_rtp)
930 session->callbacks.process_rtp (session, source,
931 GST_BUFFER_CAST (data), session->process_rtp_user_data);
933 gst_buffer_unref (GST_BUFFER_CAST (data));
935 RTP_SESSION_LOCK (session);
941 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
945 RTP_SESSION_UNLOCK (session);
947 if (session->callbacks.clock_rate)
949 session->callbacks.clock_rate (session, pt,
950 session->clock_rate_user_data);
954 RTP_SESSION_LOCK (session);
956 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
961 static RTPSourceCallbacks callbacks = {
962 (RTPSourcePushRTP) source_push_rtp,
963 (RTPSourceClockRate) source_clock_rate,
967 * find_add_conflicting_addresses:
968 * @sess: The session to check in
969 * @arrival: The arrival stats for the buffer
971 * Checks if an address which has a conflict is already known,
972 * otherwise remembers it to prevent loops.
974 * Returns: TRUE if it was a known conflict, FALSE otherwise
978 find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
981 RTPConflictingAddress *new_conflict;
983 for (item = g_list_first (sess->conflicting_addresses);
984 item; item = g_list_next (item)) {
985 RTPConflictingAddress *known_conflict = item->data;
987 if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
988 known_conflict->time = arrival->time;
993 new_conflict = g_new0 (RTPConflictingAddress, 1);
995 memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
996 new_conflict->time = arrival->time;
998 sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
1005 check_collision (RTPSession * sess, RTPSource * source,
1006 RTPArrivalStats * arrival, gboolean rtp)
1008 /* If we have no arrival address, we can't do collision checking */
1009 if (!arrival->have_address)
1012 if (sess->source != source) {
1013 /* This is not our local source, but lets check if two remote
1017 if (source->have_rtp_from) {
1018 if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
1019 /* Address is the same */
1022 /* We don't already have a from address for RTP, just set it */
1023 rtp_source_set_rtp_from (source, &arrival->address);
1027 if (source->have_rtcp_from) {
1028 if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
1029 /* Address is the same */
1032 /* We don't already have a from address for RTCP, just set it */
1033 rtp_source_set_rtcp_from (source, &arrival->address);
1037 /* We received RTP or RTCP from this source before but the network address
1038 * changed. In this case, we have third-party collision or loop */
1039 GST_DEBUG ("we have a third-party collision or loop");
1041 /* FIXME: Log 3rd party collision somehow
1042 * Maybe should be done in upper layer, only the SDES can tell us
1043 * if its a collision or a loop
1046 /* This is sending with our ssrc, is it an address we already know */
1048 if (find_add_conflicting_addresses (sess, arrival)) {
1049 /* Its a known conflict, its probably a loop, not a collision
1050 * lets just drop the incoming packet
1052 GST_DEBUG ("Our packets are being looped back to us, dropping");
1054 /* Its a new collision, lets change our SSRC */
1056 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1057 on_ssrc_collision (sess, source);
1059 rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->time);
1061 sess->change_ssrc = TRUE;
1069 /* must be called with the session lock, the returned source needs to be
1070 * unreffed after usage. */
1072 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1073 RTPArrivalStats * arrival, gboolean rtp)
1078 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1079 if (source == NULL) {
1080 /* make new Source in probation and insert */
1081 source = rtp_source_new (ssrc);
1083 /* for RTP packets we need to set the source in probation. Receiving RTCP
1084 * packets of an SSRC, on the other hand, is a strong indication that we
1085 * are dealing with a valid source. */
1087 source->probation = RTP_DEFAULT_PROBATION;
1089 source->probation = 0;
1091 /* store from address, if any */
1092 if (arrival->have_address) {
1094 rtp_source_set_rtp_from (source, &arrival->address);
1096 rtp_source_set_rtcp_from (source, &arrival->address);
1099 /* configure a callback on the source */
1100 rtp_source_set_callbacks (source, &callbacks, sess);
1102 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1105 /* we have one more source now */
1106 sess->total_sources++;
1110 /* check for collision, this updates the address when not previously set */
1111 if (check_collision (sess, source, arrival, rtp)) {
1115 /* update last activity */
1116 source->last_activity = arrival->time;
1118 source->last_rtp_activity = arrival->time;
1119 g_object_ref (source);
1125 * rtp_session_get_internal_source:
1126 * @sess: a #RTPSession
1128 * Get the internal #RTPSource of @sess.
1130 * Returns: The internal #RTPSource. g_object_unref() after usage.
1133 rtp_session_get_internal_source (RTPSession * sess)
1137 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1139 result = g_object_ref (sess->source);
1145 * rtp_session_set_internal_ssrc:
1146 * @sess: a #RTPSession
1149 * Set the SSRC of @sess to @ssrc.
1152 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1154 RTP_SESSION_LOCK (sess);
1155 if (ssrc != sess->source->ssrc) {
1156 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1157 GINT_TO_POINTER (sess->source->ssrc));
1159 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1160 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1161 * packets will timeout on the old SSRC, we could potentially schedule a
1162 * BYE RTCP for the old SSRC... */
1163 sess->source->ssrc = ssrc;
1164 rtp_source_reset (sess->source);
1166 /* rehash with the new SSRC */
1167 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1168 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1170 RTP_SESSION_UNLOCK (sess);
1172 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1176 * rtp_session_get_internal_ssrc:
1177 * @sess: a #RTPSession
1179 * Get the internal SSRC of @sess.
1181 * Returns: The SSRC of the session.
1184 rtp_session_get_internal_ssrc (RTPSession * sess)
1188 RTP_SESSION_LOCK (sess);
1189 ssrc = sess->source->ssrc;
1190 RTP_SESSION_UNLOCK (sess);
1196 * rtp_session_add_source:
1197 * @sess: a #RTPSession
1198 * @src: #RTPSource to add
1200 * Add @src to @session.
1202 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1203 * existed in the session.
1206 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1208 gboolean result = FALSE;
1211 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1212 g_return_val_if_fail (src != NULL, FALSE);
1214 RTP_SESSION_LOCK (sess);
1216 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1217 GINT_TO_POINTER (src->ssrc));
1219 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1220 GINT_TO_POINTER (src->ssrc), src);
1221 /* we have one more source now */
1222 sess->total_sources++;
1225 RTP_SESSION_UNLOCK (sess);
1231 * rtp_session_get_num_sources:
1232 * @sess: an #RTPSession
1234 * Get the number of sources in @sess.
1236 * Returns: The number of sources in @sess.
1239 rtp_session_get_num_sources (RTPSession * sess)
1243 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1245 RTP_SESSION_LOCK (sess);
1246 result = sess->total_sources;
1247 RTP_SESSION_UNLOCK (sess);
1253 * rtp_session_get_num_active_sources:
1254 * @sess: an #RTPSession
1256 * Get the number of active sources in @sess. A source is considered active when
1257 * it has been validated and has not yet received a BYE RTCP message.
1259 * Returns: The number of active sources in @sess.
1262 rtp_session_get_num_active_sources (RTPSession * sess)
1266 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1268 RTP_SESSION_LOCK (sess);
1269 result = sess->stats.active_sources;
1270 RTP_SESSION_UNLOCK (sess);
1276 * rtp_session_get_source_by_ssrc:
1277 * @sess: an #RTPSession
1280 * Find the source with @ssrc in @sess.
1282 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1283 * g_object_unref() after usage.
1286 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1290 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1292 RTP_SESSION_LOCK (sess);
1294 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1296 g_object_ref (result);
1297 RTP_SESSION_UNLOCK (sess);
1303 * rtp_session_get_source_by_cname:
1304 * @sess: a #RTPSession
1307 * Find the source with @cname in @sess.
1309 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1310 * g_object_unref() after usage.
1313 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1317 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1318 g_return_val_if_fail (cname != NULL, NULL);
1320 RTP_SESSION_LOCK (sess);
1321 result = g_hash_table_lookup (sess->cnames, cname);
1323 g_object_ref (result);
1324 RTP_SESSION_UNLOCK (sess);
1330 rtp_session_create_new_ssrc (RTPSession * sess)
1335 ssrc = g_random_int ();
1337 /* see if it exists in the session, we're done if it doesn't */
1338 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1339 GINT_TO_POINTER (ssrc)) == NULL)
1348 * rtp_session_create_source:
1349 * @sess: an #RTPSession
1351 * Create an #RTPSource for use in @sess. This function will create a source
1352 * with an ssrc that is currently not used by any participants in the session.
1354 * Returns: an #RTPSource.
1357 rtp_session_create_source (RTPSession * sess)
1362 RTP_SESSION_LOCK (sess);
1363 ssrc = rtp_session_create_new_ssrc (sess);
1364 source = rtp_source_new (ssrc);
1365 rtp_source_set_callbacks (source, &callbacks, sess);
1366 /* we need an additional ref for the source in the hashtable */
1367 g_object_ref (source);
1368 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1370 /* we have one more source now */
1371 sess->total_sources++;
1372 RTP_SESSION_UNLOCK (sess);
1377 /* update the RTPArrivalStats structure with the current time and other bits
1378 * about the current buffer we are handling.
1379 * This function is typically called when a validated packet is received.
1380 * This function should be called with the SESSION_LOCK
1383 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1384 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1385 GstClockTime running_time, guint64 ntpnstime)
1387 /* get time of arrival */
1388 arrival->time = current_time;
1389 arrival->running_time = running_time;
1390 arrival->ntpnstime = ntpnstime;
1392 /* get packet size including header overhead */
1393 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1396 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1398 arrival->payload_len = 0;
1401 /* for netbuffer we can store the IP address to check for collisions */
1402 arrival->have_address = GST_IS_NETBUFFER (buffer);
1403 if (arrival->have_address) {
1404 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1406 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1411 * rtp_session_process_rtp:
1412 * @sess: and #RTPSession
1413 * @buffer: an RTP buffer
1414 * @current_time: the current system time
1415 * @ntpnstime: the NTP arrival time in nanoseconds
1417 * Process an RTP buffer in the session manager. This function takes ownership
1420 * Returns: a #GstFlowReturn.
1423 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1424 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1426 GstFlowReturn result;
1430 gboolean prevsender, prevactive;
1431 RTPArrivalStats arrival;
1433 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1434 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1436 if (!gst_rtp_buffer_validate (buffer))
1437 goto invalid_packet;
1439 RTP_SESSION_LOCK (sess);
1440 /* update arrival stats */
1441 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1442 running_time, ntpnstime);
1444 /* ignore more RTP packets when we left the session */
1445 if (sess->source->received_bye)
1448 /* get SSRC and look up in session database */
1449 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1450 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1454 prevsender = RTP_SOURCE_IS_SENDER (source);
1455 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1457 /* we need to ref so that we can process the CSRCs later */
1458 gst_buffer_ref (buffer);
1460 /* let source process the packet */
1461 result = rtp_source_process_rtp (source, buffer, &arrival);
1463 /* source became active */
1464 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1465 sess->stats.active_sources++;
1466 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1467 sess->stats.active_sources);
1468 on_ssrc_validated (sess, source);
1470 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1471 sess->stats.sender_sources++;
1472 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1473 sess->stats.sender_sources);
1477 on_new_ssrc (sess, source);
1479 if (source->validated) {
1483 /* for validated sources, we add the CSRCs as well */
1484 count = gst_rtp_buffer_get_csrc_count (buffer);
1486 for (i = 0; i < count; i++) {
1488 RTPSource *csrc_src;
1490 csrc = gst_rtp_buffer_get_csrc (buffer, i);
1493 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1498 GST_DEBUG ("created new CSRC: %08x", csrc);
1499 rtp_source_set_as_csrc (csrc_src);
1500 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1501 sess->stats.active_sources++;
1502 on_new_ssrc (sess, csrc_src);
1504 g_object_unref (csrc_src);
1507 g_object_unref (source);
1508 gst_buffer_unref (buffer);
1510 RTP_SESSION_UNLOCK (sess);
1517 gst_buffer_unref (buffer);
1518 GST_DEBUG ("invalid RTP packet received");
1523 gst_buffer_unref (buffer);
1524 RTP_SESSION_UNLOCK (sess);
1525 GST_DEBUG ("ignoring RTP packet because we are leaving");
1530 gst_buffer_unref (buffer);
1531 RTP_SESSION_UNLOCK (sess);
1532 GST_DEBUG ("ignoring packet because its collisioning");
1538 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1539 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1543 count = gst_rtcp_packet_get_rb_count (packet);
1544 for (i = 0; i < count; i++) {
1545 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1546 guint8 fractionlost;
1549 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1550 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1552 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1554 if (ssrc == sess->source->ssrc) {
1555 /* only deal with report blocks for our session, we update the stats of
1556 * the sender of the RTCP message. We could also compare our stats against
1557 * the other sender to see if we are better or worse. */
1558 rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1559 exthighestseq, jitter, lsr, dlsr);
1561 on_ssrc_active (sess, source);
1566 /* A Sender report contains statistics about how the sender is doing. This
1567 * includes timing informataion such as the relation between RTP and NTP
1568 * timestamps and the number of packets/bytes it sent to us.
1570 * In this report is also included a set of report blocks related to how this
1571 * sender is receiving data (in case we (or somebody else) is also sending stuff
1572 * to it). This info includes the packet loss, jitter and seqnum. It also
1573 * contains information to calculate the round trip time (LSR/DLSR).
1576 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1577 RTPArrivalStats * arrival)
1579 guint32 senderssrc, rtptime, packet_count, octet_count;
1582 gboolean created, prevsender;
1584 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1585 &packet_count, &octet_count);
1587 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1588 senderssrc, GST_TIME_ARGS (arrival->time));
1590 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1594 prevsender = RTP_SOURCE_IS_SENDER (source);
1596 /* first update the source */
1597 rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1600 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1601 sess->stats.sender_sources++;
1602 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1603 sess->stats.sender_sources);
1607 on_new_ssrc (sess, source);
1609 rtp_session_process_rb (sess, source, packet, arrival);
1610 g_object_unref (source);
1613 /* A receiver report contains statistics about how a receiver is doing. It
1614 * includes stuff like packet loss, jitter and the seqnum it received last. It
1615 * also contains info to calculate the round trip time.
1617 * We are only interested in how the sender of this report is doing wrt to us.
1620 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1621 RTPArrivalStats * arrival)
1627 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1629 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1631 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1636 on_new_ssrc (sess, source);
1638 rtp_session_process_rb (sess, source, packet, arrival);
1639 g_object_unref (source);
1642 /* Get SDES items and store them in the SSRC */
1644 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1645 RTPArrivalStats * arrival)
1648 gboolean more_items, more_entries;
1650 items = gst_rtcp_packet_sdes_get_item_count (packet);
1651 GST_DEBUG ("got SDES packet with %d items", items);
1653 more_items = gst_rtcp_packet_sdes_first_item (packet);
1655 while (more_items) {
1657 gboolean changed, created;
1660 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1662 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1666 /* find src, no probation when dealing with RTCP */
1667 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1671 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1673 while (more_entries) {
1674 GstRTCPSDESType type;
1678 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1680 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1683 changed |= rtp_source_set_sdes (source, type, data, len);
1685 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1689 source->validated = TRUE;
1692 on_new_ssrc (sess, source);
1694 on_ssrc_sdes (sess, source);
1696 g_object_unref (source);
1698 more_items = gst_rtcp_packet_sdes_next_item (packet);
1703 /* BYE is sent when a client leaves the session
1706 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1707 RTPArrivalStats * arrival)
1712 reason = gst_rtcp_packet_bye_get_reason (packet);
1713 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1715 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1716 for (i = 0; i < count; i++) {
1719 gboolean created, prevactive, prevsender;
1720 guint pmembers, members;
1722 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1723 GST_DEBUG ("SSRC: %08x", ssrc);
1725 /* find src and mark bye, no probation when dealing with RTCP */
1726 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1730 /* store time for when we need to time out this source */
1731 source->bye_time = arrival->time;
1733 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1734 prevsender = RTP_SOURCE_IS_SENDER (source);
1736 /* let the source handle the rest */
1737 rtp_source_process_bye (source, reason);
1739 pmembers = sess->stats.active_sources;
1741 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1742 sess->stats.active_sources--;
1743 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1744 sess->stats.active_sources);
1746 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1747 sess->stats.sender_sources--;
1748 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1749 sess->stats.sender_sources);
1751 members = sess->stats.active_sources;
1753 if (!sess->source->received_bye && members < pmembers) {
1754 /* some members went away since the previous timeout estimate.
1755 * Perform reverse reconsideration but only when we are not scheduling a
1757 if (arrival->time < sess->next_rtcp_check_time) {
1758 GstClockTime time_remaining;
1760 time_remaining = sess->next_rtcp_check_time - arrival->time;
1761 sess->next_rtcp_check_time =
1762 gst_util_uint64_scale (time_remaining, members, pmembers);
1764 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1765 GST_TIME_ARGS (sess->next_rtcp_check_time));
1767 sess->next_rtcp_check_time += arrival->time;
1769 RTP_SESSION_UNLOCK (sess);
1770 /* notify app of reconsideration */
1771 if (sess->callbacks.reconsider)
1772 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1773 RTP_SESSION_LOCK (sess);
1778 on_new_ssrc (sess, source);
1780 on_bye_ssrc (sess, source);
1782 g_object_unref (source);
1788 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1789 RTPArrivalStats * arrival)
1791 GST_DEBUG ("received APP");
1795 * rtp_session_process_rtcp:
1796 * @sess: and #RTPSession
1797 * @buffer: an RTCP buffer
1798 * @current_time: the current system time
1800 * Process an RTCP buffer in the session manager. This function takes ownership
1803 * Returns: a #GstFlowReturn.
1806 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
1807 GstClockTime current_time)
1809 GstRTCPPacket packet;
1810 gboolean more, is_bye = FALSE, is_sr = FALSE;
1811 RTPArrivalStats arrival;
1812 GstFlowReturn result = GST_FLOW_OK;
1814 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1815 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1817 if (!gst_rtcp_buffer_validate (buffer))
1818 goto invalid_packet;
1820 GST_DEBUG ("received RTCP packet");
1822 RTP_SESSION_LOCK (sess);
1823 /* update arrival stats */
1824 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
1829 /* make writable, we might want to change the buffer */
1830 buffer = gst_buffer_make_metadata_writable (buffer);
1832 /* start processing the compound packet */
1833 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1837 type = gst_rtcp_packet_get_type (&packet);
1839 /* when we are leaving the session, we should ignore all non-BYE messages */
1840 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1841 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1846 case GST_RTCP_TYPE_SR:
1847 rtp_session_process_sr (sess, &packet, &arrival);
1850 case GST_RTCP_TYPE_RR:
1851 rtp_session_process_rr (sess, &packet, &arrival);
1853 case GST_RTCP_TYPE_SDES:
1854 rtp_session_process_sdes (sess, &packet, &arrival);
1856 case GST_RTCP_TYPE_BYE:
1858 rtp_session_process_bye (sess, &packet, &arrival);
1860 case GST_RTCP_TYPE_APP:
1861 rtp_session_process_app (sess, &packet, &arrival);
1864 GST_WARNING ("got unknown RTCP packet");
1868 more = gst_rtcp_packet_move_to_next (&packet);
1871 /* if we are scheduling a BYE, we only want to count bye packets, else we
1872 * count everything */
1873 if (sess->source->received_bye) {
1875 sess->stats.bye_members++;
1876 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1879 /* keep track of average packet size */
1880 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1882 RTP_SESSION_UNLOCK (sess);
1884 /* notify caller of sr packets in the callback */
1885 if (is_sr && sess->callbacks.sync_rtcp)
1886 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1887 sess->sync_rtcp_user_data);
1889 gst_buffer_unref (buffer);
1896 GST_DEBUG ("invalid RTCP packet received");
1897 gst_buffer_unref (buffer);
1902 gst_buffer_unref (buffer);
1903 RTP_SESSION_UNLOCK (sess);
1904 GST_DEBUG ("ignoring RTP packet because we left");
1910 * rtp_session_send_rtp:
1911 * @sess: an #RTPSession
1912 * @data: pointer to either an RTP buffer or a list of RTP buffers
1913 * @current_time: the current system time
1914 * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
1915 * This is the buffer timestamp converted to NTP time.
1917 * Send the RTP buffer in the session manager. This function takes ownership of
1920 * Returns: a #GstFlowReturn.
1923 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
1924 GstClockTime current_time, guint64 ntpnstime)
1926 GstFlowReturn result;
1928 gboolean prevsender;
1929 gboolean valid_packet;
1931 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1932 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
1935 valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
1937 valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
1941 goto invalid_packet;
1943 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
1945 RTP_SESSION_LOCK (sess);
1946 source = sess->source;
1948 /* update last activity */
1949 source->last_rtp_activity = current_time;
1951 prevsender = RTP_SOURCE_IS_SENDER (source);
1953 /* we use our own source to send */
1954 result = rtp_source_send_rtp (source, data, is_list, ntpnstime);
1956 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
1957 sess->stats.sender_sources++;
1958 RTP_SESSION_UNLOCK (sess);
1965 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1966 GST_DEBUG ("invalid RTP packet received");
1972 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
1975 GstClockTime result;
1977 if (sess->source->received_bye) {
1978 result = rtp_stats_calculate_bye_interval (&sess->stats);
1980 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
1981 RTP_SOURCE_IS_SENDER (sess->source), first);
1984 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
1985 GST_TIME_ARGS (result), first);
1988 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
1990 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
1995 /* Stop the current @sess and schedule a BYE message for the other members.
1996 * One must have the session lock to call this function
1998 static GstFlowReturn
1999 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2000 GstClockTime current_time)
2002 GstFlowReturn result = GST_FLOW_OK;
2004 GstClockTime interval;
2006 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2008 source = sess->source;
2010 /* ignore more BYEs */
2011 if (source->received_bye)
2014 /* we have BYE now */
2015 source->received_bye = TRUE;
2016 /* at least one member wants to send a BYE */
2017 g_free (sess->bye_reason);
2018 sess->bye_reason = g_strdup (reason);
2019 sess->stats.avg_rtcp_packet_size = 100;
2020 sess->stats.bye_members = 1;
2021 sess->first_rtcp = TRUE;
2022 sess->sent_bye = FALSE;
2024 /* reschedule transmission */
2025 sess->last_rtcp_send_time = current_time;
2026 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2027 sess->next_rtcp_check_time = current_time + interval;
2029 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2030 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2032 RTP_SESSION_UNLOCK (sess);
2033 /* notify app of reconsideration */
2034 if (sess->callbacks.reconsider)
2035 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2036 RTP_SESSION_LOCK (sess);
2043 * rtp_session_schedule_bye:
2044 * @sess: an #RTPSession
2045 * @reason: a reason or NULL
2046 * @current_time: the current system time
2048 * Stop the current @sess and schedule a BYE message for the other members.
2050 * Returns: a #GstFlowReturn.
2053 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2054 GstClockTime current_time)
2056 GstFlowReturn result = GST_FLOW_OK;
2058 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2060 RTP_SESSION_LOCK (sess);
2061 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2062 RTP_SESSION_UNLOCK (sess);
2068 * rtp_session_next_timeout:
2069 * @sess: an #RTPSession
2070 * @current_time: the current system time
2072 * Get the next time we should perform session maintenance tasks.
2074 * Returns: a time when rtp_session_on_timeout() should be called with the
2075 * current system time.
2078 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2080 GstClockTime result;
2082 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2084 RTP_SESSION_LOCK (sess);
2086 result = sess->next_rtcp_check_time;
2088 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
2089 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2091 if (result < current_time) {
2092 GST_DEBUG ("take current time as base");
2093 /* our previous check time expired, start counting from the current time
2095 result = current_time;
2098 if (sess->source->received_bye) {
2099 if (sess->sent_bye) {
2100 GST_DEBUG ("we sent BYE already");
2101 result = GST_CLOCK_TIME_NONE;
2102 } else if (sess->stats.active_sources >= 50) {
2103 GST_DEBUG ("reconsider BYE, more than 50 sources");
2104 /* reconsider BYE if members >= 50 */
2105 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2108 if (sess->first_rtcp) {
2109 GST_DEBUG ("first RTCP packet");
2110 /* we are called for the first time */
2111 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2112 } else if (sess->next_rtcp_check_time < current_time) {
2113 GST_DEBUG ("old check time expired, getting new timeout");
2114 /* get a new timeout when we need to */
2115 result += calculate_rtcp_interval (sess, FALSE, FALSE);
2118 sess->next_rtcp_check_time = result;
2120 GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2121 RTP_SESSION_UNLOCK (sess);
2130 GstClockTime current_time;
2132 GstClockTime interval;
2133 GstRTCPPacket packet;
2139 session_start_rtcp (RTPSession * sess, ReportData * data)
2141 GstRTCPPacket *packet = &data->packet;
2142 RTPSource *own = sess->source;
2144 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2146 if (RTP_SOURCE_IS_SENDER (own)) {
2149 guint32 packet_count, octet_count;
2151 /* we are a sender, create SR */
2152 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2153 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
2155 /* get latest stats */
2156 rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
2157 &packet_count, &octet_count);
2159 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2160 packet_count, octet_count);
2162 /* fill in sender report info */
2163 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2164 ntptime, rtptime, packet_count, octet_count);
2166 /* we are only receiver, create RR */
2167 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2168 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2169 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2173 /* construct a Sender or Receiver Report */
2175 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2177 RTPSession *sess = data->sess;
2178 GstRTCPPacket *packet = &data->packet;
2180 /* create a new buffer if needed */
2181 if (data->rtcp == NULL) {
2182 session_start_rtcp (sess, data);
2184 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2185 /* only report about other sender sources */
2186 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2187 guint8 fractionlost;
2189 guint32 exthighestseq, jitter;
2193 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2194 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2196 /* packet is not yet filled, add report block for this source. */
2197 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2198 exthighestseq, jitter, lsr, dlsr);
2203 /* perform cleanup of sources that timed out */
2205 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2207 gboolean remove = FALSE;
2208 gboolean byetimeout = FALSE;
2209 gboolean sendertimeout = FALSE;
2210 gboolean is_sender, is_active;
2211 RTPSession *sess = data->sess;
2212 GstClockTime interval;
2214 is_sender = RTP_SOURCE_IS_SENDER (source);
2215 is_active = RTP_SOURCE_IS_ACTIVE (source);
2217 /* check for our own source, we don't want to delete our own source. */
2218 if (!(source == sess->source)) {
2219 if (source->received_bye) {
2220 /* if we received a BYE from the source, remove the source after some
2222 if (data->current_time > source->bye_time &&
2223 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2224 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2229 /* sources that were inactive for more than 5 times the deterministic reporting
2230 * interval get timed out. the min timeout is 5 seconds. */
2231 if (data->current_time > source->last_activity) {
2232 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2233 if (data->current_time - source->last_activity > interval) {
2234 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2235 source->ssrc, GST_TIME_ARGS (source->last_activity));
2241 /* senders that did not send for a long time become a receiver, this also
2242 * holds for our own source. */
2244 if (data->current_time > source->last_rtp_activity) {
2245 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2246 if (data->current_time - source->last_rtp_activity > interval) {
2247 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2248 GST_TIME_FORMAT, source->ssrc,
2249 GST_TIME_ARGS (source->last_rtp_activity));
2250 source->is_sender = FALSE;
2251 sess->stats.sender_sources--;
2252 sendertimeout = TRUE;
2258 sess->total_sources--;
2260 sess->stats.sender_sources--;
2262 sess->stats.active_sources--;
2265 on_bye_timeout (sess, source);
2267 on_timeout (sess, source);
2270 on_sender_timeout (sess, source);
2276 session_sdes (RTPSession * sess, ReportData * data)
2278 GstRTCPPacket *packet = &data->packet;
2282 /* add SDES packet */
2283 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2285 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2287 rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
2289 gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
2292 /* other SDES items must only be added at regular intervals and only when the
2293 * user requests to since it might be a privacy problem */
2295 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
2296 strlen (sess->name), (guint8 *) sess->name);
2297 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
2298 strlen (sess->tool), (guint8 *) sess->tool);
2301 data->has_sdes = TRUE;
2304 /* schedule a BYE packet */
2306 session_bye (RTPSession * sess, ReportData * data)
2308 GstRTCPPacket *packet = &data->packet;
2311 session_start_rtcp (sess, data);
2314 session_sdes (sess, data);
2316 /* add a BYE packet */
2317 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2318 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2319 if (sess->bye_reason)
2320 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2322 /* we have a BYE packet now */
2323 data->is_bye = TRUE;
2327 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2329 GstClockTime new_send_time, elapsed;
2332 /* no need to check yet */
2333 if (sess->next_rtcp_check_time > current_time) {
2334 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2335 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2336 GST_TIME_ARGS (current_time));
2340 /* get elapsed time since we last reported */
2341 elapsed = current_time - sess->last_rtcp_send_time;
2343 /* perform forward reconsideration */
2344 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2346 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2347 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2349 new_send_time += sess->last_rtcp_send_time;
2351 /* check if reconsideration */
2352 if (current_time < new_send_time) {
2353 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2354 GST_TIME_ARGS (new_send_time));
2356 /* store new check time */
2357 sess->next_rtcp_check_time = new_send_time;
2360 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2362 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2363 GST_TIME_ARGS (new_send_time));
2364 sess->next_rtcp_check_time = current_time + new_send_time;
2370 * rtp_session_on_timeout:
2371 * @sess: an #RTPSession
2372 * @current_time: the current system time
2373 * @ntpnstime: the current NTP time in nanoseconds
2375 * Perform maintenance actions after the timeout obtained with
2376 * rtp_session_next_timeout() expired.
2378 * This function will perform timeouts of receivers and senders, send a BYE
2379 * packet or generate RTCP packets with current session stats.
2381 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2382 * times, for each packet that should be processed.
2384 * Returns: a #GstFlowReturn.
2387 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
2390 GstFlowReturn result = GST_FLOW_OK;
2394 gboolean notify = FALSE;
2396 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2398 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2399 GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
2403 data.current_time = current_time;
2404 data.ntpnstime = ntpnstime;
2405 data.is_bye = FALSE;
2406 data.has_sdes = FALSE;
2410 RTP_SESSION_LOCK (sess);
2411 /* get a new interval, we need this for various cleanups etc */
2412 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2414 /* first perform cleanups */
2415 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2416 (GHRFunc) session_cleanup, &data);
2418 /* see if we need to generate SR or RR packets */
2419 if (is_rtcp_time (sess, current_time, &data)) {
2420 if (own->received_bye) {
2421 /* generate BYE instead */
2422 GST_DEBUG ("generating BYE message");
2423 session_bye (sess, &data);
2424 sess->sent_bye = TRUE;
2426 /* loop over all known sources and do something */
2427 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2428 (GHFunc) session_report_blocks, &data);
2435 /* we keep track of the last report time in order to timeout inactive
2436 * receivers or senders */
2437 sess->last_rtcp_send_time = data.current_time;
2438 sess->first_rtcp = FALSE;
2440 /* add SDES for this source when not already added */
2442 session_sdes (sess, &data);
2444 /* update average RTCP size before sending */
2445 size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2446 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2449 /* check for outdated collisions */
2450 GST_DEBUG ("checking collision list");
2451 item = g_list_first (sess->conflicting_addresses);
2453 RTPConflictingAddress *known_conflict = item->data;
2454 GList *next_item = g_list_next (item);
2456 if (known_conflict->time < current_time - (data.interval *
2457 RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2458 sess->conflicting_addresses =
2459 g_list_delete_link (sess->conflicting_addresses, item);
2460 GST_DEBUG ("collision %p timed out", known_conflict);
2461 g_free (known_conflict);
2466 if (sess->change_ssrc) {
2467 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
2468 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2469 GINT_TO_POINTER (own->ssrc));
2471 own->ssrc = rtp_session_create_new_ssrc (sess);
2472 rtp_source_reset (own);
2474 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2475 GINT_TO_POINTER (own->ssrc), own);
2477 g_free (sess->bye_reason);
2478 sess->bye_reason = NULL;
2479 sess->sent_bye = FALSE;
2480 sess->change_ssrc = FALSE;
2482 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
2484 RTP_SESSION_UNLOCK (sess);
2487 g_object_notify (G_OBJECT (sess), "internal-ssrc");
2489 /* push out the RTCP packet */
2491 /* close the RTCP packet */
2492 gst_rtcp_buffer_end (data.rtcp);
2494 GST_DEBUG ("sending packet");
2495 if (sess->callbacks.send_rtcp)
2496 result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
2497 sess->sent_bye, sess->send_rtcp_user_data);
2499 GST_DEBUG ("freeing packet");
2500 gst_buffer_unref (data.rtcp);