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_list_foreach (sess->conflicting_addresses, (GFunc) g_free, NULL);
373 g_list_free (sess->conflicting_addresses);
375 g_free (sess->bye_reason);
377 g_hash_table_destroy (sess->cnames);
378 g_object_unref (sess->source);
380 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
384 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
386 GValue value = { 0 };
388 g_value_init (&value, RTP_TYPE_SOURCE);
389 g_value_take_object (&value, source);
390 /* copies the value */
391 g_value_array_append (arr, &value);
395 rtp_session_create_sources (RTPSession * sess)
400 RTP_SESSION_LOCK (sess);
401 /* get number of elements in the table */
402 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
403 /* create the result value array */
404 res = g_value_array_new (size);
406 /* and copy all values into the array */
407 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
408 RTP_SESSION_UNLOCK (sess);
414 rtp_session_set_property (GObject * object, guint prop_id,
415 const GValue * value, GParamSpec * pspec)
419 sess = RTP_SESSION (object);
422 case PROP_INTERNAL_SSRC:
423 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
426 rtp_session_set_bandwidth (sess, g_value_get_double (value));
428 case PROP_RTCP_FRACTION:
429 rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
432 sess->mtu = g_value_get_uint (value);
435 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
438 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
444 rtp_session_get_property (GObject * object, guint prop_id,
445 GValue * value, GParamSpec * pspec)
449 sess = RTP_SESSION (object);
452 case PROP_INTERNAL_SSRC:
453 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
455 case PROP_INTERNAL_SOURCE:
456 g_value_take_object (value, rtp_session_get_internal_source (sess));
459 g_value_set_double (value, rtp_session_get_bandwidth (sess));
461 case PROP_RTCP_FRACTION:
462 g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
465 g_value_set_uint (value, sess->mtu);
468 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
470 case PROP_NUM_SOURCES:
471 g_value_set_uint (value, rtp_session_get_num_sources (sess));
473 case PROP_NUM_ACTIVE_SOURCES:
474 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
477 g_value_take_boxed (value, rtp_session_create_sources (sess));
480 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
486 on_new_ssrc (RTPSession * sess, RTPSource * source)
488 g_object_ref (source);
489 RTP_SESSION_UNLOCK (sess);
490 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
491 RTP_SESSION_LOCK (sess);
492 g_object_unref (source);
496 on_ssrc_collision (RTPSession * sess, RTPSource * source)
498 g_object_ref (source);
499 RTP_SESSION_UNLOCK (sess);
500 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
502 RTP_SESSION_LOCK (sess);
503 g_object_unref (source);
507 on_ssrc_validated (RTPSession * sess, RTPSource * source)
509 g_object_ref (source);
510 RTP_SESSION_UNLOCK (sess);
511 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
513 RTP_SESSION_LOCK (sess);
514 g_object_unref (source);
518 on_ssrc_active (RTPSession * sess, RTPSource * source)
520 g_object_ref (source);
521 RTP_SESSION_UNLOCK (sess);
522 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
523 RTP_SESSION_LOCK (sess);
524 g_object_unref (source);
528 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
530 g_object_ref (source);
531 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
532 RTP_SESSION_UNLOCK (sess);
533 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
534 RTP_SESSION_LOCK (sess);
535 g_object_unref (source);
539 on_bye_ssrc (RTPSession * sess, RTPSource * source)
541 g_object_ref (source);
542 RTP_SESSION_UNLOCK (sess);
543 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
544 RTP_SESSION_LOCK (sess);
545 g_object_unref (source);
549 on_bye_timeout (RTPSession * sess, RTPSource * source)
551 g_object_ref (source);
552 RTP_SESSION_UNLOCK (sess);
553 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
554 RTP_SESSION_LOCK (sess);
555 g_object_unref (source);
559 on_timeout (RTPSession * sess, RTPSource * source)
561 g_object_ref (source);
562 RTP_SESSION_UNLOCK (sess);
563 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
564 RTP_SESSION_LOCK (sess);
565 g_object_unref (source);
569 on_sender_timeout (RTPSession * sess, RTPSource * source)
571 g_object_ref (source);
572 RTP_SESSION_UNLOCK (sess);
573 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
575 RTP_SESSION_LOCK (sess);
576 g_object_unref (source);
582 * Create a new session object.
584 * Returns: a new #RTPSession. g_object_unref() after usage.
587 rtp_session_new (void)
591 sess = g_object_new (RTP_TYPE_SESSION, NULL);
597 * rtp_session_set_callbacks:
598 * @sess: an #RTPSession
599 * @callbacks: callbacks to configure
600 * @user_data: user data passed in the callbacks
602 * Configure a set of callbacks to be notified of actions.
605 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
608 g_return_if_fail (RTP_IS_SESSION (sess));
610 if (callbacks->process_rtp) {
611 sess->callbacks.process_rtp = callbacks->process_rtp;
612 sess->process_rtp_user_data = user_data;
614 if (callbacks->send_rtp) {
615 sess->callbacks.send_rtp = callbacks->send_rtp;
616 sess->send_rtp_user_data = user_data;
618 if (callbacks->send_rtcp) {
619 sess->callbacks.send_rtcp = callbacks->send_rtcp;
620 sess->send_rtcp_user_data = user_data;
622 if (callbacks->sync_rtcp) {
623 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
624 sess->sync_rtcp_user_data = user_data;
626 if (callbacks->clock_rate) {
627 sess->callbacks.clock_rate = callbacks->clock_rate;
628 sess->clock_rate_user_data = user_data;
630 if (callbacks->reconsider) {
631 sess->callbacks.reconsider = callbacks->reconsider;
632 sess->reconsider_user_data = user_data;
637 * rtp_session_set_process_rtp_callback:
638 * @sess: an #RTPSession
639 * @callback: callback to set
640 * @user_data: user data passed in the callback
642 * Configure only the process_rtp callback to be notified of the process_rtp action.
645 rtp_session_set_process_rtp_callback (RTPSession * sess,
646 RTPSessionProcessRTP callback, gpointer user_data)
648 g_return_if_fail (RTP_IS_SESSION (sess));
650 sess->callbacks.process_rtp = callback;
651 sess->process_rtp_user_data = user_data;
655 * rtp_session_set_send_rtp_callback:
656 * @sess: an #RTPSession
657 * @callback: callback to set
658 * @user_data: user data passed in the callback
660 * Configure only the send_rtp callback to be notified of the send_rtp action.
663 rtp_session_set_send_rtp_callback (RTPSession * sess,
664 RTPSessionSendRTP callback, gpointer user_data)
666 g_return_if_fail (RTP_IS_SESSION (sess));
668 sess->callbacks.send_rtp = callback;
669 sess->send_rtp_user_data = user_data;
673 * rtp_session_set_send_rtcp_callback:
674 * @sess: an #RTPSession
675 * @callback: callback to set
676 * @user_data: user data passed in the callback
678 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
681 rtp_session_set_send_rtcp_callback (RTPSession * sess,
682 RTPSessionSendRTCP callback, gpointer user_data)
684 g_return_if_fail (RTP_IS_SESSION (sess));
686 sess->callbacks.send_rtcp = callback;
687 sess->send_rtcp_user_data = user_data;
691 * rtp_session_set_sync_rtcp_callback:
692 * @sess: an #RTPSession
693 * @callback: callback to set
694 * @user_data: user data passed in the callback
696 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
699 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
700 RTPSessionSyncRTCP callback, gpointer user_data)
702 g_return_if_fail (RTP_IS_SESSION (sess));
704 sess->callbacks.sync_rtcp = callback;
705 sess->sync_rtcp_user_data = user_data;
709 * rtp_session_set_clock_rate_callback:
710 * @sess: an #RTPSession
711 * @callback: callback to set
712 * @user_data: user data passed in the callback
714 * Configure only the clock_rate callback to be notified of the clock_rate action.
717 rtp_session_set_clock_rate_callback (RTPSession * sess,
718 RTPSessionClockRate callback, gpointer user_data)
720 g_return_if_fail (RTP_IS_SESSION (sess));
722 sess->callbacks.clock_rate = callback;
723 sess->clock_rate_user_data = user_data;
727 * rtp_session_set_reconsider_callback:
728 * @sess: an #RTPSession
729 * @callback: callback to set
730 * @user_data: user data passed in the callback
732 * Configure only the reconsider callback to be notified of the reconsider action.
735 rtp_session_set_reconsider_callback (RTPSession * sess,
736 RTPSessionReconsider callback, gpointer user_data)
738 g_return_if_fail (RTP_IS_SESSION (sess));
740 sess->callbacks.reconsider = callback;
741 sess->reconsider_user_data = user_data;
745 * rtp_session_set_bandwidth:
746 * @sess: an #RTPSession
747 * @bandwidth: the bandwidth allocated
749 * Set the session bandwidth in bytes per second.
752 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
754 g_return_if_fail (RTP_IS_SESSION (sess));
756 RTP_SESSION_LOCK (sess);
757 sess->stats.bandwidth = bandwidth;
758 RTP_SESSION_UNLOCK (sess);
762 * rtp_session_get_bandwidth:
763 * @sess: an #RTPSession
765 * Get the session bandwidth.
767 * Returns: the session bandwidth.
770 rtp_session_get_bandwidth (RTPSession * sess)
774 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
776 RTP_SESSION_LOCK (sess);
777 result = sess->stats.bandwidth;
778 RTP_SESSION_UNLOCK (sess);
784 * rtp_session_set_rtcp_fraction:
785 * @sess: an #RTPSession
786 * @bandwidth: the RTCP bandwidth
788 * Set the bandwidth that should be used for RTCP
792 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
794 g_return_if_fail (RTP_IS_SESSION (sess));
796 RTP_SESSION_LOCK (sess);
797 sess->stats.rtcp_bandwidth = bandwidth;
798 RTP_SESSION_UNLOCK (sess);
802 * rtp_session_get_rtcp_fraction:
803 * @sess: an #RTPSession
805 * Get the session bandwidth used for RTCP.
807 * Returns: The bandwidth used for RTCP messages.
810 rtp_session_get_rtcp_fraction (RTPSession * sess)
814 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
816 RTP_SESSION_LOCK (sess);
817 result = sess->stats.rtcp_bandwidth;
818 RTP_SESSION_UNLOCK (sess);
824 * rtp_session_set_sdes_string:
825 * @sess: an #RTPSession
826 * @type: the type of the SDES item
827 * @item: a null-terminated string to set.
829 * Store an SDES item of @type in @sess.
831 * Returns: %FALSE if the data was unchanged @type is invalid.
834 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
839 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
841 RTP_SESSION_LOCK (sess);
842 result = rtp_source_set_sdes_string (sess->source, type, item);
843 RTP_SESSION_UNLOCK (sess);
849 * rtp_session_get_sdes_string:
850 * @sess: an #RTPSession
851 * @type: the type of the SDES item
853 * Get the SDES item of @type from @sess.
855 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
856 * valid. g_free() after usage.
859 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
863 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
865 RTP_SESSION_LOCK (sess);
866 result = rtp_source_get_sdes_string (sess->source, type);
867 RTP_SESSION_UNLOCK (sess);
873 * rtp_session_get_sdes_struct:
874 * @sess: an #RTSPSession
876 * Get the SDES data as a #GstStructure
878 * Returns: a GstStructure with SDES items for @sess. This function returns a
879 * copy of the SDES structure, use gst_structure_free() after usage.
882 rtp_session_get_sdes_struct (RTPSession * sess)
884 const GstStructure *sdes;
885 GstStructure *result = NULL;
887 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
889 RTP_SESSION_LOCK (sess);
890 sdes = rtp_source_get_sdes_struct (sess->source);
892 result = gst_structure_copy (sdes);
893 RTP_SESSION_UNLOCK (sess);
899 * rtp_session_set_sdes_struct:
900 * @sess: an #RTSPSession
901 * @sdes: a #GstStructure
903 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
906 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
908 g_return_if_fail (sdes);
909 g_return_if_fail (RTP_IS_SESSION (sess));
911 RTP_SESSION_LOCK (sess);
912 rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
913 RTP_SESSION_UNLOCK (sess);
917 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
919 GstFlowReturn result = GST_FLOW_OK;
921 if (source == session->source) {
922 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
924 RTP_SESSION_UNLOCK (session);
926 if (session->callbacks.send_rtp)
928 session->callbacks.send_rtp (session, source, data,
929 session->send_rtp_user_data);
931 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
934 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
935 RTP_SESSION_UNLOCK (session);
937 if (session->callbacks.process_rtp)
939 session->callbacks.process_rtp (session, source,
940 GST_BUFFER_CAST (data), session->process_rtp_user_data);
942 gst_buffer_unref (GST_BUFFER_CAST (data));
944 RTP_SESSION_LOCK (session);
950 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
954 RTP_SESSION_UNLOCK (session);
956 if (session->callbacks.clock_rate)
958 session->callbacks.clock_rate (session, pt,
959 session->clock_rate_user_data);
963 RTP_SESSION_LOCK (session);
965 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
970 static RTPSourceCallbacks callbacks = {
971 (RTPSourcePushRTP) source_push_rtp,
972 (RTPSourceClockRate) source_clock_rate,
976 * find_add_conflicting_addresses:
977 * @sess: The session to check in
978 * @arrival: The arrival stats for the buffer
980 * Checks if an address which has a conflict is already known,
981 * otherwise remembers it to prevent loops.
983 * Returns: TRUE if it was a known conflict, FALSE otherwise
987 find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
990 RTPConflictingAddress *new_conflict;
992 for (item = g_list_first (sess->conflicting_addresses);
993 item; item = g_list_next (item)) {
994 RTPConflictingAddress *known_conflict = item->data;
996 if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
997 known_conflict->time = arrival->time;
1002 new_conflict = g_new0 (RTPConflictingAddress, 1);
1004 memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
1005 new_conflict->time = arrival->time;
1007 sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
1014 check_collision (RTPSession * sess, RTPSource * source,
1015 RTPArrivalStats * arrival, gboolean rtp)
1017 /* If we have no arrival address, we can't do collision checking */
1018 if (!arrival->have_address)
1021 if (sess->source != source) {
1022 /* This is not our local source, but lets check if two remote
1026 if (source->have_rtp_from) {
1027 if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
1028 /* Address is the same */
1031 /* We don't already have a from address for RTP, just set it */
1032 rtp_source_set_rtp_from (source, &arrival->address);
1036 if (source->have_rtcp_from) {
1037 if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
1038 /* Address is the same */
1041 /* We don't already have a from address for RTCP, just set it */
1042 rtp_source_set_rtcp_from (source, &arrival->address);
1046 /* We received RTP or RTCP from this source before but the network address
1047 * changed. In this case, we have third-party collision or loop */
1048 GST_DEBUG ("we have a third-party collision or loop");
1050 /* FIXME: Log 3rd party collision somehow
1051 * Maybe should be done in upper layer, only the SDES can tell us
1052 * if its a collision or a loop
1055 /* This is sending with our ssrc, is it an address we already know */
1057 if (find_add_conflicting_addresses (sess, arrival)) {
1058 /* Its a known conflict, its probably a loop, not a collision
1059 * lets just drop the incoming packet
1061 GST_DEBUG ("Our packets are being looped back to us, dropping");
1063 /* Its a new collision, lets change our SSRC */
1065 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1066 on_ssrc_collision (sess, source);
1068 rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->time);
1070 sess->change_ssrc = TRUE;
1078 /* must be called with the session lock, the returned source needs to be
1079 * unreffed after usage. */
1081 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1082 RTPArrivalStats * arrival, gboolean rtp)
1087 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1088 if (source == NULL) {
1089 /* make new Source in probation and insert */
1090 source = rtp_source_new (ssrc);
1092 /* for RTP packets we need to set the source in probation. Receiving RTCP
1093 * packets of an SSRC, on the other hand, is a strong indication that we
1094 * are dealing with a valid source. */
1096 source->probation = RTP_DEFAULT_PROBATION;
1098 source->probation = 0;
1100 /* store from address, if any */
1101 if (arrival->have_address) {
1103 rtp_source_set_rtp_from (source, &arrival->address);
1105 rtp_source_set_rtcp_from (source, &arrival->address);
1108 /* configure a callback on the source */
1109 rtp_source_set_callbacks (source, &callbacks, sess);
1111 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1114 /* we have one more source now */
1115 sess->total_sources++;
1119 /* check for collision, this updates the address when not previously set */
1120 if (check_collision (sess, source, arrival, rtp)) {
1124 /* update last activity */
1125 source->last_activity = arrival->time;
1127 source->last_rtp_activity = arrival->time;
1128 g_object_ref (source);
1134 * rtp_session_get_internal_source:
1135 * @sess: a #RTPSession
1137 * Get the internal #RTPSource of @sess.
1139 * Returns: The internal #RTPSource. g_object_unref() after usage.
1142 rtp_session_get_internal_source (RTPSession * sess)
1146 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1148 result = g_object_ref (sess->source);
1154 * rtp_session_set_internal_ssrc:
1155 * @sess: a #RTPSession
1158 * Set the SSRC of @sess to @ssrc.
1161 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1163 RTP_SESSION_LOCK (sess);
1164 if (ssrc != sess->source->ssrc) {
1165 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1166 GINT_TO_POINTER (sess->source->ssrc));
1168 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1169 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1170 * packets will timeout on the old SSRC, we could potentially schedule a
1171 * BYE RTCP for the old SSRC... */
1172 sess->source->ssrc = ssrc;
1173 rtp_source_reset (sess->source);
1175 /* rehash with the new SSRC */
1176 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1177 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1179 RTP_SESSION_UNLOCK (sess);
1181 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1185 * rtp_session_get_internal_ssrc:
1186 * @sess: a #RTPSession
1188 * Get the internal SSRC of @sess.
1190 * Returns: The SSRC of the session.
1193 rtp_session_get_internal_ssrc (RTPSession * sess)
1197 RTP_SESSION_LOCK (sess);
1198 ssrc = sess->source->ssrc;
1199 RTP_SESSION_UNLOCK (sess);
1205 * rtp_session_add_source:
1206 * @sess: a #RTPSession
1207 * @src: #RTPSource to add
1209 * Add @src to @session.
1211 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1212 * existed in the session.
1215 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1217 gboolean result = FALSE;
1220 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1221 g_return_val_if_fail (src != NULL, FALSE);
1223 RTP_SESSION_LOCK (sess);
1225 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1226 GINT_TO_POINTER (src->ssrc));
1228 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1229 GINT_TO_POINTER (src->ssrc), src);
1230 /* we have one more source now */
1231 sess->total_sources++;
1234 RTP_SESSION_UNLOCK (sess);
1240 * rtp_session_get_num_sources:
1241 * @sess: an #RTPSession
1243 * Get the number of sources in @sess.
1245 * Returns: The number of sources in @sess.
1248 rtp_session_get_num_sources (RTPSession * sess)
1252 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1254 RTP_SESSION_LOCK (sess);
1255 result = sess->total_sources;
1256 RTP_SESSION_UNLOCK (sess);
1262 * rtp_session_get_num_active_sources:
1263 * @sess: an #RTPSession
1265 * Get the number of active sources in @sess. A source is considered active when
1266 * it has been validated and has not yet received a BYE RTCP message.
1268 * Returns: The number of active sources in @sess.
1271 rtp_session_get_num_active_sources (RTPSession * sess)
1275 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1277 RTP_SESSION_LOCK (sess);
1278 result = sess->stats.active_sources;
1279 RTP_SESSION_UNLOCK (sess);
1285 * rtp_session_get_source_by_ssrc:
1286 * @sess: an #RTPSession
1289 * Find the source with @ssrc in @sess.
1291 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1292 * g_object_unref() after usage.
1295 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1299 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1301 RTP_SESSION_LOCK (sess);
1303 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1305 g_object_ref (result);
1306 RTP_SESSION_UNLOCK (sess);
1312 * rtp_session_get_source_by_cname:
1313 * @sess: a #RTPSession
1316 * Find the source with @cname in @sess.
1318 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1319 * g_object_unref() after usage.
1322 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1326 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1327 g_return_val_if_fail (cname != NULL, NULL);
1329 RTP_SESSION_LOCK (sess);
1330 result = g_hash_table_lookup (sess->cnames, cname);
1332 g_object_ref (result);
1333 RTP_SESSION_UNLOCK (sess);
1339 rtp_session_create_new_ssrc (RTPSession * sess)
1344 ssrc = g_random_int ();
1346 /* see if it exists in the session, we're done if it doesn't */
1347 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1348 GINT_TO_POINTER (ssrc)) == NULL)
1356 * rtp_session_create_source:
1357 * @sess: an #RTPSession
1359 * Create an #RTPSource for use in @sess. This function will create a source
1360 * with an ssrc that is currently not used by any participants in the session.
1362 * Returns: an #RTPSource.
1365 rtp_session_create_source (RTPSession * sess)
1370 RTP_SESSION_LOCK (sess);
1371 ssrc = rtp_session_create_new_ssrc (sess);
1372 source = rtp_source_new (ssrc);
1373 rtp_source_set_callbacks (source, &callbacks, sess);
1374 /* we need an additional ref for the source in the hashtable */
1375 g_object_ref (source);
1376 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1378 /* we have one more source now */
1379 sess->total_sources++;
1380 RTP_SESSION_UNLOCK (sess);
1385 /* update the RTPArrivalStats structure with the current time and other bits
1386 * about the current buffer we are handling.
1387 * This function is typically called when a validated packet is received.
1388 * This function should be called with the SESSION_LOCK
1391 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1392 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1393 GstClockTime running_time, guint64 ntpnstime)
1395 /* get time of arrival */
1396 arrival->time = current_time;
1397 arrival->running_time = running_time;
1398 arrival->ntpnstime = ntpnstime;
1400 /* get packet size including header overhead */
1401 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1404 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1406 arrival->payload_len = 0;
1409 /* for netbuffer we can store the IP address to check for collisions */
1410 arrival->have_address = GST_IS_NETBUFFER (buffer);
1411 if (arrival->have_address) {
1412 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1414 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1419 * rtp_session_process_rtp:
1420 * @sess: and #RTPSession
1421 * @buffer: an RTP buffer
1422 * @current_time: the current system time
1423 * @ntpnstime: the NTP arrival time in nanoseconds
1425 * Process an RTP buffer in the session manager. This function takes ownership
1428 * Returns: a #GstFlowReturn.
1431 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1432 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1434 GstFlowReturn result;
1438 gboolean prevsender, prevactive;
1439 RTPArrivalStats arrival;
1443 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1444 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1446 if (!gst_rtp_buffer_validate (buffer))
1447 goto invalid_packet;
1449 RTP_SESSION_LOCK (sess);
1450 /* update arrival stats */
1451 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1452 running_time, ntpnstime);
1454 /* ignore more RTP packets when we left the session */
1455 if (sess->source->received_bye)
1458 /* get SSRC and look up in session database */
1459 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1460 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1464 prevsender = RTP_SOURCE_IS_SENDER (source);
1465 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1467 /* copy available csrc for later */
1468 count = gst_rtp_buffer_get_csrc_count (buffer);
1469 /* make sure to not overflow our array. An RTP buffer can maximally contain
1471 count = MIN (count, 16);
1473 for (i = 0; i < count; i++)
1474 csrcs[i] = gst_rtp_buffer_get_csrc (buffer, i);
1476 /* let source process the packet */
1477 result = rtp_source_process_rtp (source, buffer, &arrival);
1479 /* source became active */
1480 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1481 sess->stats.active_sources++;
1482 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1483 sess->stats.active_sources);
1484 on_ssrc_validated (sess, source);
1486 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1487 sess->stats.sender_sources++;
1488 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1489 sess->stats.sender_sources);
1493 on_new_ssrc (sess, source);
1495 if (source->validated) {
1498 /* for validated sources, we add the CSRCs as well */
1499 for (i = 0; i < count; i++) {
1501 RTPSource *csrc_src;
1506 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1511 GST_DEBUG ("created new CSRC: %08x", csrc);
1512 rtp_source_set_as_csrc (csrc_src);
1513 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1514 sess->stats.active_sources++;
1515 on_new_ssrc (sess, csrc_src);
1517 g_object_unref (csrc_src);
1520 g_object_unref (source);
1522 RTP_SESSION_UNLOCK (sess);
1529 gst_buffer_unref (buffer);
1530 GST_DEBUG ("invalid RTP packet received");
1535 gst_buffer_unref (buffer);
1536 RTP_SESSION_UNLOCK (sess);
1537 GST_DEBUG ("ignoring RTP packet because we are leaving");
1542 gst_buffer_unref (buffer);
1543 RTP_SESSION_UNLOCK (sess);
1544 GST_DEBUG ("ignoring packet because its collisioning");
1550 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1551 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1555 count = gst_rtcp_packet_get_rb_count (packet);
1556 for (i = 0; i < count; i++) {
1557 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1558 guint8 fractionlost;
1561 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1562 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1564 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1566 if (ssrc == sess->source->ssrc) {
1567 /* only deal with report blocks for our session, we update the stats of
1568 * the sender of the RTCP message. We could also compare our stats against
1569 * the other sender to see if we are better or worse. */
1570 rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1571 exthighestseq, jitter, lsr, dlsr);
1573 on_ssrc_active (sess, source);
1578 /* A Sender report contains statistics about how the sender is doing. This
1579 * includes timing informataion such as the relation between RTP and NTP
1580 * timestamps and the number of packets/bytes it sent to us.
1582 * In this report is also included a set of report blocks related to how this
1583 * sender is receiving data (in case we (or somebody else) is also sending stuff
1584 * to it). This info includes the packet loss, jitter and seqnum. It also
1585 * contains information to calculate the round trip time (LSR/DLSR).
1588 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1589 RTPArrivalStats * arrival, gboolean * do_sync)
1591 guint32 senderssrc, rtptime, packet_count, octet_count;
1594 gboolean created, prevsender;
1596 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1597 &packet_count, &octet_count);
1599 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1600 senderssrc, GST_TIME_ARGS (arrival->time));
1602 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1606 /* don't try to do lip-sync for sources that sent a BYE */
1607 if (rtp_source_received_bye (source))
1612 prevsender = RTP_SOURCE_IS_SENDER (source);
1614 /* first update the source */
1615 rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1618 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1619 sess->stats.sender_sources++;
1620 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1621 sess->stats.sender_sources);
1625 on_new_ssrc (sess, source);
1627 rtp_session_process_rb (sess, source, packet, arrival);
1628 g_object_unref (source);
1631 /* A receiver report contains statistics about how a receiver is doing. It
1632 * includes stuff like packet loss, jitter and the seqnum it received last. It
1633 * also contains info to calculate the round trip time.
1635 * We are only interested in how the sender of this report is doing wrt to us.
1638 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1639 RTPArrivalStats * arrival)
1645 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1647 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1649 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1654 on_new_ssrc (sess, source);
1656 rtp_session_process_rb (sess, source, packet, arrival);
1657 g_object_unref (source);
1660 /* Get SDES items and store them in the SSRC */
1662 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1663 RTPArrivalStats * arrival)
1666 gboolean more_items, more_entries;
1668 items = gst_rtcp_packet_sdes_get_item_count (packet);
1669 GST_DEBUG ("got SDES packet with %d items", items);
1671 more_items = gst_rtcp_packet_sdes_first_item (packet);
1673 while (more_items) {
1675 gboolean changed, created;
1679 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1681 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1685 /* find src, no probation when dealing with RTCP */
1686 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1690 sdes = gst_structure_new ("application/x-rtp-source-sdes", NULL);
1692 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1694 while (more_entries) {
1695 GstRTCPSDESType type;
1701 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1703 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1706 if (type == GST_RTCP_SDES_PRIV) {
1707 name = g_strndup ((const gchar *) &data[1], data[0]);
1709 data += data[0] + 1;
1711 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1714 value = g_strndup ((const gchar *) data, len);
1716 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1721 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1725 /* takes ownership of sdes */
1726 changed = rtp_source_set_sdes_struct (source, sdes);
1728 source->validated = TRUE;
1731 on_new_ssrc (sess, source);
1733 on_ssrc_sdes (sess, source);
1735 g_object_unref (source);
1737 more_items = gst_rtcp_packet_sdes_next_item (packet);
1742 /* BYE is sent when a client leaves the session
1745 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1746 RTPArrivalStats * arrival)
1750 gboolean reconsider = FALSE;
1752 reason = gst_rtcp_packet_bye_get_reason (packet);
1753 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1755 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1756 for (i = 0; i < count; i++) {
1759 gboolean created, prevactive, prevsender;
1760 guint pmembers, members;
1762 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1763 GST_DEBUG ("SSRC: %08x", ssrc);
1765 /* find src and mark bye, no probation when dealing with RTCP */
1766 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1770 /* store time for when we need to time out this source */
1771 source->bye_time = arrival->time;
1773 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1774 prevsender = RTP_SOURCE_IS_SENDER (source);
1776 /* let the source handle the rest */
1777 rtp_source_process_bye (source, reason);
1779 pmembers = sess->stats.active_sources;
1781 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1782 sess->stats.active_sources--;
1783 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1784 sess->stats.active_sources);
1786 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1787 sess->stats.sender_sources--;
1788 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1789 sess->stats.sender_sources);
1791 members = sess->stats.active_sources;
1793 if (!sess->source->received_bye && members < pmembers) {
1794 /* some members went away since the previous timeout estimate.
1795 * Perform reverse reconsideration but only when we are not scheduling a
1797 if (arrival->time < sess->next_rtcp_check_time) {
1798 GstClockTime time_remaining;
1800 time_remaining = sess->next_rtcp_check_time - arrival->time;
1801 sess->next_rtcp_check_time =
1802 gst_util_uint64_scale (time_remaining, members, pmembers);
1804 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1805 GST_TIME_ARGS (sess->next_rtcp_check_time));
1807 sess->next_rtcp_check_time += arrival->time;
1809 /* mark pending reconsider. We only want to signal the reconsideration
1810 * once after we handled all the source in the bye packet */
1816 on_new_ssrc (sess, source);
1818 on_bye_ssrc (sess, source);
1820 g_object_unref (source);
1823 RTP_SESSION_UNLOCK (sess);
1824 /* notify app of reconsideration */
1825 if (sess->callbacks.reconsider)
1826 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1827 RTP_SESSION_LOCK (sess);
1833 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1834 RTPArrivalStats * arrival)
1836 GST_DEBUG ("received APP");
1840 * rtp_session_process_rtcp:
1841 * @sess: and #RTPSession
1842 * @buffer: an RTCP buffer
1843 * @current_time: the current system time
1845 * Process an RTCP buffer in the session manager. This function takes ownership
1848 * Returns: a #GstFlowReturn.
1851 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
1852 GstClockTime current_time)
1854 GstRTCPPacket packet;
1855 gboolean more, is_bye = FALSE, do_sync = FALSE;
1856 RTPArrivalStats arrival;
1857 GstFlowReturn result = GST_FLOW_OK;
1859 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1860 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1862 if (!gst_rtcp_buffer_validate (buffer))
1863 goto invalid_packet;
1865 GST_DEBUG ("received RTCP packet");
1867 RTP_SESSION_LOCK (sess);
1868 /* update arrival stats */
1869 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
1874 /* make writable, we might want to change the buffer */
1875 buffer = gst_buffer_make_metadata_writable (buffer);
1877 /* start processing the compound packet */
1878 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1882 type = gst_rtcp_packet_get_type (&packet);
1884 /* when we are leaving the session, we should ignore all non-BYE messages */
1885 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1886 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1891 case GST_RTCP_TYPE_SR:
1892 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
1894 case GST_RTCP_TYPE_RR:
1895 rtp_session_process_rr (sess, &packet, &arrival);
1897 case GST_RTCP_TYPE_SDES:
1898 rtp_session_process_sdes (sess, &packet, &arrival);
1900 case GST_RTCP_TYPE_BYE:
1902 /* don't try to attempt lip-sync anymore for streams with a BYE */
1904 rtp_session_process_bye (sess, &packet, &arrival);
1906 case GST_RTCP_TYPE_APP:
1907 rtp_session_process_app (sess, &packet, &arrival);
1910 GST_WARNING ("got unknown RTCP packet");
1914 more = gst_rtcp_packet_move_to_next (&packet);
1917 /* if we are scheduling a BYE, we only want to count bye packets, else we
1918 * count everything */
1919 if (sess->source->received_bye) {
1921 sess->stats.bye_members++;
1922 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1925 /* keep track of average packet size */
1926 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1928 RTP_SESSION_UNLOCK (sess);
1930 /* notify caller of sr packets in the callback */
1931 if (do_sync && sess->callbacks.sync_rtcp)
1932 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1933 sess->sync_rtcp_user_data);
1935 gst_buffer_unref (buffer);
1942 GST_DEBUG ("invalid RTCP packet received");
1943 gst_buffer_unref (buffer);
1948 gst_buffer_unref (buffer);
1949 RTP_SESSION_UNLOCK (sess);
1950 GST_DEBUG ("ignoring RTP packet because we left");
1956 * rtp_session_send_rtp:
1957 * @sess: an #RTPSession
1958 * @data: pointer to either an RTP buffer or a list of RTP buffers
1959 * @is_list: TRUE when @data is a buffer list
1960 * @current_time: the current system time
1961 * @running_time: the running time of @data
1963 * Send the RTP buffer in the session manager. This function takes ownership of
1966 * Returns: a #GstFlowReturn.
1969 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
1970 GstClockTime current_time, GstClockTime running_time)
1972 GstFlowReturn result;
1974 gboolean prevsender;
1975 gboolean valid_packet;
1977 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1978 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
1981 valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
1983 valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
1987 goto invalid_packet;
1989 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
1991 RTP_SESSION_LOCK (sess);
1992 source = sess->source;
1994 /* update last activity */
1995 source->last_rtp_activity = current_time;
1997 prevsender = RTP_SOURCE_IS_SENDER (source);
1999 /* we use our own source to send */
2000 result = rtp_source_send_rtp (source, data, is_list, running_time);
2002 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2003 sess->stats.sender_sources++;
2004 RTP_SESSION_UNLOCK (sess);
2011 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2012 GST_DEBUG ("invalid RTP packet received");
2018 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2021 GstClockTime result;
2023 if (sess->source->received_bye) {
2024 result = rtp_stats_calculate_bye_interval (&sess->stats);
2026 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2027 RTP_SOURCE_IS_SENDER (sess->source), first);
2030 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2031 GST_TIME_ARGS (result), first);
2034 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2036 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2041 /* Stop the current @sess and schedule a BYE message for the other members.
2042 * One must have the session lock to call this function
2044 static GstFlowReturn
2045 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2046 GstClockTime current_time)
2048 GstFlowReturn result = GST_FLOW_OK;
2050 GstClockTime interval;
2052 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2054 source = sess->source;
2056 /* ignore more BYEs */
2057 if (source->received_bye)
2060 /* we have BYE now */
2061 source->received_bye = TRUE;
2062 /* at least one member wants to send a BYE */
2063 g_free (sess->bye_reason);
2064 sess->bye_reason = g_strdup (reason);
2065 sess->stats.avg_rtcp_packet_size = 100;
2066 sess->stats.bye_members = 1;
2067 sess->first_rtcp = TRUE;
2068 sess->sent_bye = FALSE;
2070 /* reschedule transmission */
2071 sess->last_rtcp_send_time = current_time;
2072 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2073 sess->next_rtcp_check_time = current_time + interval;
2075 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2076 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2078 RTP_SESSION_UNLOCK (sess);
2079 /* notify app of reconsideration */
2080 if (sess->callbacks.reconsider)
2081 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2082 RTP_SESSION_LOCK (sess);
2089 * rtp_session_schedule_bye:
2090 * @sess: an #RTPSession
2091 * @reason: a reason or NULL
2092 * @current_time: the current system time
2094 * Stop the current @sess and schedule a BYE message for the other members.
2096 * Returns: a #GstFlowReturn.
2099 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2100 GstClockTime current_time)
2102 GstFlowReturn result = GST_FLOW_OK;
2104 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2106 RTP_SESSION_LOCK (sess);
2107 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2108 RTP_SESSION_UNLOCK (sess);
2114 * rtp_session_next_timeout:
2115 * @sess: an #RTPSession
2116 * @current_time: the current system time
2118 * Get the next time we should perform session maintenance tasks.
2120 * Returns: a time when rtp_session_on_timeout() should be called with the
2121 * current system time.
2124 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2126 GstClockTime result;
2128 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2130 RTP_SESSION_LOCK (sess);
2132 result = sess->next_rtcp_check_time;
2134 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
2135 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2137 if (result < current_time) {
2138 GST_DEBUG ("take current time as base");
2139 /* our previous check time expired, start counting from the current time
2141 result = current_time;
2144 if (sess->source->received_bye) {
2145 if (sess->sent_bye) {
2146 GST_DEBUG ("we sent BYE already");
2147 result = GST_CLOCK_TIME_NONE;
2148 } else if (sess->stats.active_sources >= 50) {
2149 GST_DEBUG ("reconsider BYE, more than 50 sources");
2150 /* reconsider BYE if members >= 50 */
2151 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2154 if (sess->first_rtcp) {
2155 GST_DEBUG ("first RTCP packet");
2156 /* we are called for the first time */
2157 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2158 } else if (sess->next_rtcp_check_time < current_time) {
2159 GST_DEBUG ("old check time expired, getting new timeout");
2160 /* get a new timeout when we need to */
2161 result += calculate_rtcp_interval (sess, FALSE, FALSE);
2164 sess->next_rtcp_check_time = result;
2166 GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2167 RTP_SESSION_UNLOCK (sess);
2176 GstClockTime current_time;
2178 GstClockTime running_time;
2179 GstClockTime interval;
2180 GstRTCPPacket packet;
2186 session_start_rtcp (RTPSession * sess, ReportData * data)
2188 GstRTCPPacket *packet = &data->packet;
2189 RTPSource *own = sess->source;
2191 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2193 if (RTP_SOURCE_IS_SENDER (own)) {
2196 guint32 packet_count, octet_count;
2198 /* we are a sender, create SR */
2199 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2200 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
2202 /* get latest stats */
2203 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2204 &ntptime, &rtptime, &packet_count, &octet_count);
2206 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2207 packet_count, octet_count);
2209 /* fill in sender report info */
2210 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2211 ntptime, rtptime, packet_count, octet_count);
2213 /* we are only receiver, create RR */
2214 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2215 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2216 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2220 /* construct a Sender or Receiver Report */
2222 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2224 RTPSession *sess = data->sess;
2225 GstRTCPPacket *packet = &data->packet;
2227 /* create a new buffer if needed */
2228 if (data->rtcp == NULL) {
2229 session_start_rtcp (sess, data);
2231 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2232 /* only report about other sender sources */
2233 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2234 guint8 fractionlost;
2236 guint32 exthighestseq, jitter;
2240 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2241 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2243 /* packet is not yet filled, add report block for this source. */
2244 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2245 exthighestseq, jitter, lsr, dlsr);
2250 /* perform cleanup of sources that timed out */
2252 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2254 gboolean remove = FALSE;
2255 gboolean byetimeout = FALSE;
2256 gboolean sendertimeout = FALSE;
2257 gboolean is_sender, is_active;
2258 RTPSession *sess = data->sess;
2259 GstClockTime interval;
2261 is_sender = RTP_SOURCE_IS_SENDER (source);
2262 is_active = RTP_SOURCE_IS_ACTIVE (source);
2264 /* check for our own source, we don't want to delete our own source. */
2265 if (!(source == sess->source)) {
2266 if (source->received_bye) {
2267 /* if we received a BYE from the source, remove the source after some
2269 if (data->current_time > source->bye_time &&
2270 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2271 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2276 /* sources that were inactive for more than 5 times the deterministic reporting
2277 * interval get timed out. the min timeout is 5 seconds. */
2278 if (data->current_time > source->last_activity) {
2279 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2280 if (data->current_time - source->last_activity > interval) {
2281 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2282 source->ssrc, GST_TIME_ARGS (source->last_activity));
2288 /* senders that did not send for a long time become a receiver, this also
2289 * holds for our own source. */
2291 if (data->current_time > source->last_rtp_activity) {
2292 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2293 if (data->current_time - source->last_rtp_activity > interval) {
2294 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2295 GST_TIME_FORMAT, source->ssrc,
2296 GST_TIME_ARGS (source->last_rtp_activity));
2297 source->is_sender = FALSE;
2298 sess->stats.sender_sources--;
2299 sendertimeout = TRUE;
2305 sess->total_sources--;
2307 sess->stats.sender_sources--;
2309 sess->stats.active_sources--;
2312 on_bye_timeout (sess, source);
2314 on_timeout (sess, source);
2317 on_sender_timeout (sess, source);
2323 session_sdes (RTPSession * sess, ReportData * data)
2325 GstRTCPPacket *packet = &data->packet;
2326 const GstStructure *sdes;
2329 /* add SDES packet */
2330 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2332 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2334 sdes = rtp_source_get_sdes_struct (sess->source);
2336 /* add all fields in the structure, the order is not important. */
2337 n_fields = gst_structure_n_fields (sdes);
2338 for (i = 0; i < n_fields; ++i) {
2341 GstRTCPSDESType type;
2343 field = gst_structure_nth_field_name (sdes, i);
2346 value = gst_structure_get_string (sdes, field);
2349 type = gst_rtcp_sdes_name_to_type (field);
2351 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2352 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2353 (const guint8 *) value);
2354 } else if (type == GST_RTCP_SDES_PRIV) {
2360 /* don't accept entries that are too big */
2361 prefix_len = strlen (field);
2362 if (prefix_len > 255)
2364 value_len = strlen (value);
2365 if (value_len > 255)
2367 data_len = 1 + prefix_len + value_len;
2371 data[0] = prefix_len;
2372 memcpy (&data[1], field, prefix_len);
2373 memcpy (&data[1 + prefix_len], value, value_len);
2375 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2379 data->has_sdes = TRUE;
2382 /* schedule a BYE packet */
2384 session_bye (RTPSession * sess, ReportData * data)
2386 GstRTCPPacket *packet = &data->packet;
2389 session_start_rtcp (sess, data);
2392 session_sdes (sess, data);
2394 /* add a BYE packet */
2395 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2396 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2397 if (sess->bye_reason)
2398 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2400 /* we have a BYE packet now */
2401 data->is_bye = TRUE;
2405 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2407 GstClockTime new_send_time, elapsed;
2410 /* no need to check yet */
2411 if (sess->next_rtcp_check_time > current_time) {
2412 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2413 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2414 GST_TIME_ARGS (current_time));
2418 /* get elapsed time since we last reported */
2419 elapsed = current_time - sess->last_rtcp_send_time;
2421 /* perform forward reconsideration */
2422 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2424 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2425 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2427 new_send_time += sess->last_rtcp_send_time;
2429 /* check if reconsideration */
2430 if (current_time < new_send_time) {
2431 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2432 GST_TIME_ARGS (new_send_time));
2434 /* store new check time */
2435 sess->next_rtcp_check_time = new_send_time;
2438 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2440 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2441 GST_TIME_ARGS (new_send_time));
2442 sess->next_rtcp_check_time = current_time + new_send_time;
2448 * rtp_session_on_timeout:
2449 * @sess: an #RTPSession
2450 * @current_time: the current system time
2451 * @ntpnstime: the current NTP time in nanoseconds
2452 * @running_time: the current running_time of the pipeline
2454 * Perform maintenance actions after the timeout obtained with
2455 * rtp_session_next_timeout() expired.
2457 * This function will perform timeouts of receivers and senders, send a BYE
2458 * packet or generate RTCP packets with current session stats.
2460 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2461 * times, for each packet that should be processed.
2463 * Returns: a #GstFlowReturn.
2466 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
2467 guint64 ntpnstime, GstClockTime running_time)
2469 GstFlowReturn result = GST_FLOW_OK;
2473 gboolean notify = FALSE;
2475 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2477 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2478 GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
2482 data.current_time = current_time;
2483 data.ntpnstime = ntpnstime;
2484 data.is_bye = FALSE;
2485 data.has_sdes = FALSE;
2486 data.running_time = running_time;
2490 RTP_SESSION_LOCK (sess);
2491 /* get a new interval, we need this for various cleanups etc */
2492 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2494 /* first perform cleanups */
2495 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2496 (GHRFunc) session_cleanup, &data);
2498 /* see if we need to generate SR or RR packets */
2499 if (is_rtcp_time (sess, current_time, &data)) {
2500 if (own->received_bye) {
2501 /* generate BYE instead */
2502 GST_DEBUG ("generating BYE message");
2503 session_bye (sess, &data);
2504 sess->sent_bye = TRUE;
2506 /* loop over all known sources and do something */
2507 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2508 (GHFunc) session_report_blocks, &data);
2515 /* we keep track of the last report time in order to timeout inactive
2516 * receivers or senders */
2517 sess->last_rtcp_send_time = data.current_time;
2518 sess->first_rtcp = FALSE;
2520 /* add SDES for this source when not already added */
2522 session_sdes (sess, &data);
2524 /* update average RTCP size before sending */
2525 size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2526 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2529 /* check for outdated collisions */
2530 GST_DEBUG ("checking collision list");
2531 item = g_list_first (sess->conflicting_addresses);
2533 RTPConflictingAddress *known_conflict = item->data;
2534 GList *next_item = g_list_next (item);
2536 if (known_conflict->time < current_time - (data.interval *
2537 RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2538 sess->conflicting_addresses =
2539 g_list_delete_link (sess->conflicting_addresses, item);
2540 GST_DEBUG ("collision %p timed out", known_conflict);
2541 g_free (known_conflict);
2546 if (sess->change_ssrc) {
2547 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
2548 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2549 GINT_TO_POINTER (own->ssrc));
2551 own->ssrc = rtp_session_create_new_ssrc (sess);
2552 rtp_source_reset (own);
2554 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2555 GINT_TO_POINTER (own->ssrc), own);
2557 g_free (sess->bye_reason);
2558 sess->bye_reason = NULL;
2559 sess->sent_bye = FALSE;
2560 sess->change_ssrc = FALSE;
2562 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
2564 RTP_SESSION_UNLOCK (sess);
2567 g_object_notify (G_OBJECT (sess), "internal-ssrc");
2569 /* push out the RTCP packet */
2571 /* close the RTCP packet */
2572 gst_rtcp_buffer_end (data.rtcp);
2574 GST_DEBUG ("sending packet");
2575 if (sess->callbacks.send_rtcp)
2576 result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
2577 sess->sent_bye, sess->send_rtcp_user_data);
2579 GST_DEBUG ("freeing packet");
2580 gst_buffer_unref (data.rtcp);