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>
27 #include "rtpsession.h"
29 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
30 #define GST_CAT_DEFAULT rtp_session_debug
32 /* signals and args */
36 SIGNAL_ON_SSRC_COLLISION,
37 SIGNAL_ON_SSRC_VALIDATED,
38 SIGNAL_ON_SSRC_ACTIVE,
41 SIGNAL_ON_BYE_TIMEOUT,
46 #define DEFAULT_INTERNAL_SOURCE NULL
47 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
48 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
49 #define DEFAULT_SDES_CNAME NULL
50 #define DEFAULT_SDES_NAME NULL
51 #define DEFAULT_SDES_EMAIL NULL
52 #define DEFAULT_SDES_PHONE NULL
53 #define DEFAULT_SDES_LOCATION NULL
54 #define DEFAULT_SDES_TOOL NULL
55 #define DEFAULT_SDES_NOTE NULL
56 #define DEFAULT_NUM_SOURCES 0
57 #define DEFAULT_NUM_ACTIVE_SOURCES 0
73 PROP_NUM_ACTIVE_SOURCES,
77 /* update average packet size, we keep this scaled by 16 to keep enough
79 #define UPDATE_AVG(avg, val) \
83 (avg) = ((val) + (15 * (avg))) >> 4;
85 /* The number RTCP intervals after which to timeout entries in the
88 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
90 /* GObject vmethods */
91 static void rtp_session_finalize (GObject * object);
92 static void rtp_session_set_property (GObject * object, guint prop_id,
93 const GValue * value, GParamSpec * pspec);
94 static void rtp_session_get_property (GObject * object, guint prop_id,
95 GValue * value, GParamSpec * pspec);
97 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
99 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
101 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
102 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
103 static GstFlowReturn rtp_session_send_bye_locked (RTPSession * sess,
104 const gchar * reason);
105 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
106 gboolean deterministic, gboolean first);
109 rtp_session_class_init (RTPSessionClass * klass)
111 GObjectClass *gobject_class;
113 gobject_class = (GObjectClass *) klass;
115 gobject_class->finalize = rtp_session_finalize;
116 gobject_class->set_property = rtp_session_set_property;
117 gobject_class->get_property = rtp_session_get_property;
120 * RTPSession::on-new-ssrc:
121 * @session: the object which received the signal
122 * @src: the new RTPSource
124 * Notify of a new SSRC that entered @session.
126 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
127 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
128 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
129 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
132 * RTPSession::on-ssrc-collision:
133 * @session: the object which received the signal
134 * @src: the #RTPSource that caused a collision
136 * Notify when we have an SSRC collision
138 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
139 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
140 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
141 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
144 * RTPSession::on-ssrc-validated:
145 * @session: the object which received the signal
146 * @src: the new validated RTPSource
148 * Notify of a new SSRC that became validated.
150 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
151 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
152 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
153 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
156 * RTPSession::on-ssrc-active:
157 * @session: the object which received the signal
158 * @src: the active RTPSource
160 * Notify of a SSRC that is active, i.e., sending RTCP.
162 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
163 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
164 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
165 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
168 * RTPSession::on-ssrc-sdes:
169 * @session: the object which received the signal
170 * @src: the RTPSource
172 * Notify that a new SDES was received for SSRC.
174 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
175 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
176 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
177 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
180 * RTPSession::on-bye-ssrc:
181 * @session: the object which received the signal
182 * @src: the RTPSource that went away
184 * Notify of an SSRC that became inactive because of a BYE packet.
186 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
187 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
188 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
189 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
192 * RTPSession::on-bye-timeout:
193 * @session: the object which received the signal
194 * @src: the RTPSource that timed out
196 * Notify of an SSRC that has timed out because of BYE
198 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
199 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
200 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
201 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
204 * RTPSession::on-timeout:
205 * @session: the object which received the signal
206 * @src: the RTPSource that timed out
208 * Notify of an SSRC that has timed out
210 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
211 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
212 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
213 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
216 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
217 g_param_spec_object ("internal-source", "Internal Source",
218 "The internal source element of the session",
219 RTP_TYPE_SOURCE, G_PARAM_READABLE));
221 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
222 g_param_spec_double ("bandwidth", "Bandwidth",
223 "The bandwidth of the session",
224 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE));
226 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
227 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
228 "The fraction of the bandwidth used for RTCP",
229 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE));
231 g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
232 g_param_spec_string ("sdes-cname", "SDES CNAME",
233 "The CNAME to put in SDES messages of this session",
234 DEFAULT_SDES_CNAME, G_PARAM_READWRITE));
236 g_object_class_install_property (gobject_class, PROP_SDES_NAME,
237 g_param_spec_string ("sdes-name", "SDES NAME",
238 "The NAME to put in SDES messages of this session",
239 DEFAULT_SDES_NAME, G_PARAM_READWRITE));
241 g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
242 g_param_spec_string ("sdes-email", "SDES EMAIL",
243 "The EMAIL to put in SDES messages of this session",
244 DEFAULT_SDES_EMAIL, G_PARAM_READWRITE));
246 g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
247 g_param_spec_string ("sdes-phone", "SDES PHONE",
248 "The PHONE to put in SDES messages of this session",
249 DEFAULT_SDES_PHONE, G_PARAM_READWRITE));
251 g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
252 g_param_spec_string ("sdes-location", "SDES LOCATION",
253 "The LOCATION to put in SDES messages of this session",
254 DEFAULT_SDES_LOCATION, G_PARAM_READWRITE));
256 g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
257 g_param_spec_string ("sdes-tool", "SDES TOOL",
258 "The TOOL to put in SDES messages of this session",
259 DEFAULT_SDES_TOOL, G_PARAM_READWRITE));
261 g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
262 g_param_spec_string ("sdes-note", "SDES NOTE",
263 "The NOTE to put in SDES messages of this session",
264 DEFAULT_SDES_NOTE, G_PARAM_READWRITE));
266 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
267 g_param_spec_uint ("num-sources", "Num Sources",
268 "The number of sources in the session", 0, G_MAXUINT,
269 DEFAULT_NUM_SOURCES, G_PARAM_READABLE));
271 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
272 g_param_spec_uint ("num-active-sources", "Num Active Sources",
273 "The number of active sources in the session", 0, G_MAXUINT,
274 DEFAULT_NUM_ACTIVE_SOURCES, G_PARAM_READABLE));
276 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
280 rtp_session_init (RTPSession * sess)
285 sess->lock = g_mutex_new ();
286 sess->key = g_random_int ();
290 for (i = 0; i < 32; i++) {
292 g_hash_table_new_full (NULL, NULL, NULL,
293 (GDestroyNotify) g_object_unref);
295 sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
297 rtp_stats_init_defaults (&sess->stats);
299 /* create an active SSRC for this session manager */
300 sess->source = rtp_session_create_source (sess);
301 sess->source->validated = TRUE;
302 sess->stats.active_sources++;
304 /* default UDP header length */
305 sess->header_len = 28;
308 /* some default SDES entries */
309 str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
310 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
313 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
315 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
317 sess->first_rtcp = TRUE;
319 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
323 rtp_session_finalize (GObject * object)
328 sess = RTP_SESSION_CAST (object);
330 g_mutex_free (sess->lock);
331 for (i = 0; i < 32; i++)
332 g_hash_table_destroy (sess->ssrcs[i]);
334 g_free (sess->bye_reason);
336 g_hash_table_destroy (sess->cnames);
337 g_object_unref (sess->source);
339 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
343 rtp_session_set_property (GObject * object, guint prop_id,
344 const GValue * value, GParamSpec * pspec)
348 sess = RTP_SESSION (object);
352 rtp_session_set_bandwidth (sess, g_value_get_double (value));
354 case PROP_RTCP_FRACTION:
355 rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
357 case PROP_SDES_CNAME:
358 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_CNAME,
359 g_value_get_string (value));
362 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NAME,
363 g_value_get_string (value));
365 case PROP_SDES_EMAIL:
366 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_EMAIL,
367 g_value_get_string (value));
369 case PROP_SDES_PHONE:
370 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_PHONE,
371 g_value_get_string (value));
373 case PROP_SDES_LOCATION:
374 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_LOC,
375 g_value_get_string (value));
378 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_TOOL,
379 g_value_get_string (value));
382 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NOTE,
383 g_value_get_string (value));
386 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
392 rtp_session_get_property (GObject * object, guint prop_id,
393 GValue * value, GParamSpec * pspec)
397 sess = RTP_SESSION (object);
400 case PROP_INTERNAL_SOURCE:
401 g_value_take_object (value, rtp_session_get_internal_source (sess));
404 g_value_set_double (value, rtp_session_get_bandwidth (sess));
406 case PROP_RTCP_FRACTION:
407 g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
409 case PROP_SDES_CNAME:
410 g_value_take_string (value, rtp_session_get_sdes_string (sess,
411 GST_RTCP_SDES_CNAME));
414 g_value_take_string (value, rtp_session_get_sdes_string (sess,
415 GST_RTCP_SDES_NAME));
417 case PROP_SDES_EMAIL:
418 g_value_take_string (value, rtp_session_get_sdes_string (sess,
419 GST_RTCP_SDES_EMAIL));
421 case PROP_SDES_PHONE:
422 g_value_take_string (value, rtp_session_get_sdes_string (sess,
423 GST_RTCP_SDES_PHONE));
425 case PROP_SDES_LOCATION:
426 g_value_take_string (value, rtp_session_get_sdes_string (sess,
430 g_value_take_string (value, rtp_session_get_sdes_string (sess,
431 GST_RTCP_SDES_TOOL));
434 g_value_take_string (value, rtp_session_get_sdes_string (sess,
435 GST_RTCP_SDES_NOTE));
437 case PROP_NUM_SOURCES:
438 g_value_set_uint (value, rtp_session_get_num_sources (sess));
440 case PROP_NUM_ACTIVE_SOURCES:
441 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
444 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
450 on_new_ssrc (RTPSession * sess, RTPSource * source)
452 RTP_SESSION_UNLOCK (sess);
453 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
454 RTP_SESSION_LOCK (sess);
458 on_ssrc_collision (RTPSession * sess, RTPSource * source)
460 RTP_SESSION_UNLOCK (sess);
461 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
463 RTP_SESSION_LOCK (sess);
467 on_ssrc_validated (RTPSession * sess, RTPSource * source)
469 RTP_SESSION_UNLOCK (sess);
470 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
472 RTP_SESSION_LOCK (sess);
476 on_ssrc_active (RTPSession * sess, RTPSource * source)
478 RTP_SESSION_UNLOCK (sess);
479 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
480 RTP_SESSION_LOCK (sess);
484 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
486 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
487 RTP_SESSION_UNLOCK (sess);
488 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
489 RTP_SESSION_LOCK (sess);
493 on_bye_ssrc (RTPSession * sess, RTPSource * source)
495 RTP_SESSION_UNLOCK (sess);
496 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
497 RTP_SESSION_LOCK (sess);
501 on_bye_timeout (RTPSession * sess, RTPSource * source)
503 RTP_SESSION_UNLOCK (sess);
504 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
505 RTP_SESSION_LOCK (sess);
509 on_timeout (RTPSession * sess, RTPSource * source)
511 RTP_SESSION_UNLOCK (sess);
512 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
513 RTP_SESSION_LOCK (sess);
519 * Create a new session object.
521 * Returns: a new #RTPSession. g_object_unref() after usage.
524 rtp_session_new (void)
528 sess = g_object_new (RTP_TYPE_SESSION, NULL);
534 * rtp_session_set_callbacks:
535 * @sess: an #RTPSession
536 * @callbacks: callbacks to configure
537 * @user_data: user data passed in the callbacks
539 * Configure a set of callbacks to be notified of actions.
542 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
545 g_return_if_fail (RTP_IS_SESSION (sess));
547 if (callbacks->process_rtp) {
548 sess->callbacks.process_rtp = callbacks->process_rtp;
549 sess->process_rtp_user_data = user_data;
551 if (callbacks->send_rtp) {
552 sess->callbacks.send_rtp = callbacks->send_rtp;
553 sess->send_rtp_user_data = user_data;
555 if (callbacks->send_rtcp) {
556 sess->callbacks.send_rtcp = callbacks->send_rtcp;
557 sess->send_rtcp_user_data = user_data;
559 if (callbacks->sync_rtcp) {
560 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
561 sess->sync_rtcp_user_data = user_data;
563 if (callbacks->clock_rate) {
564 sess->callbacks.clock_rate = callbacks->clock_rate;
565 sess->clock_rate_user_data = user_data;
567 if (callbacks->reconsider) {
568 sess->callbacks.reconsider = callbacks->reconsider;
569 sess->reconsider_user_data = user_data;
574 * rtp_session_set_process_rtp_callback:
575 * @sess: an #RTPSession
576 * @callback: callback to set
577 * @user_data: user data passed in the callback
579 * Configure only the process_rtp callback to be notified of the process_rtp action.
582 rtp_session_set_process_rtp_callback (RTPSession * sess,
583 RTPSessionProcessRTP callback, gpointer user_data)
585 g_return_if_fail (RTP_IS_SESSION (sess));
587 sess->callbacks.process_rtp = callback;
588 sess->process_rtp_user_data = user_data;
592 * rtp_session_set_send_rtp_callback:
593 * @sess: an #RTPSession
594 * @callback: callback to set
595 * @user_data: user data passed in the callback
597 * Configure only the send_rtp callback to be notified of the send_rtp action.
600 rtp_session_set_send_rtp_callback (RTPSession * sess,
601 RTPSessionSendRTP callback, gpointer user_data)
603 g_return_if_fail (RTP_IS_SESSION (sess));
605 sess->callbacks.send_rtp = callback;
606 sess->send_rtp_user_data = user_data;
610 * rtp_session_set_send_rtcp_callback:
611 * @sess: an #RTPSession
612 * @callback: callback to set
613 * @user_data: user data passed in the callback
615 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
618 rtp_session_set_send_rtcp_callback (RTPSession * sess,
619 RTPSessionSendRTCP callback, gpointer user_data)
621 g_return_if_fail (RTP_IS_SESSION (sess));
623 sess->callbacks.send_rtcp = callback;
624 sess->send_rtcp_user_data = user_data;
628 * rtp_session_set_sync_rtcp_callback:
629 * @sess: an #RTPSession
630 * @callback: callback to set
631 * @user_data: user data passed in the callback
633 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
636 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
637 RTPSessionSyncRTCP callback, gpointer user_data)
639 g_return_if_fail (RTP_IS_SESSION (sess));
641 sess->callbacks.sync_rtcp = callback;
642 sess->sync_rtcp_user_data = user_data;
646 * rtp_session_set_clock_rate_callback:
647 * @sess: an #RTPSession
648 * @callback: callback to set
649 * @user_data: user data passed in the callback
651 * Configure only the clock_rate callback to be notified of the clock_rate action.
654 rtp_session_set_clock_rate_callback (RTPSession * sess,
655 RTPSessionClockRate callback, gpointer user_data)
657 g_return_if_fail (RTP_IS_SESSION (sess));
659 sess->callbacks.clock_rate = callback;
660 sess->clock_rate_user_data = user_data;
664 * rtp_session_set_reconsider_callback:
665 * @sess: an #RTPSession
666 * @callback: callback to set
667 * @user_data: user data passed in the callback
669 * Configure only the reconsider callback to be notified of the reconsider action.
672 rtp_session_set_reconsider_callback (RTPSession * sess,
673 RTPSessionReconsider callback, gpointer user_data)
675 g_return_if_fail (RTP_IS_SESSION (sess));
677 sess->callbacks.reconsider = callback;
678 sess->reconsider_user_data = user_data;
682 * rtp_session_set_bandwidth:
683 * @sess: an #RTPSession
684 * @bandwidth: the bandwidth allocated
686 * Set the session bandwidth in bytes per second.
689 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
691 g_return_if_fail (RTP_IS_SESSION (sess));
693 RTP_SESSION_LOCK (sess);
694 sess->stats.bandwidth = bandwidth;
695 RTP_SESSION_UNLOCK (sess);
699 * rtp_session_get_bandwidth:
700 * @sess: an #RTPSession
702 * Get the session bandwidth.
704 * Returns: the session bandwidth.
707 rtp_session_get_bandwidth (RTPSession * sess)
711 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
713 RTP_SESSION_LOCK (sess);
714 result = sess->stats.bandwidth;
715 RTP_SESSION_UNLOCK (sess);
721 * rtp_session_set_rtcp_fraction:
722 * @sess: an #RTPSession
723 * @bandwidth: the RTCP bandwidth
725 * Set the bandwidth that should be used for RTCP
729 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
731 g_return_if_fail (RTP_IS_SESSION (sess));
733 RTP_SESSION_LOCK (sess);
734 sess->stats.rtcp_bandwidth = bandwidth;
735 RTP_SESSION_UNLOCK (sess);
739 * rtp_session_get_rtcp_fraction:
740 * @sess: an #RTPSession
742 * Get the session bandwidth used for RTCP.
744 * Returns: The bandwidth used for RTCP messages.
747 rtp_session_get_rtcp_fraction (RTPSession * sess)
751 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
753 RTP_SESSION_LOCK (sess);
754 result = sess->stats.rtcp_bandwidth;
755 RTP_SESSION_UNLOCK (sess);
761 * rtp_session_set_sdes_string:
762 * @sess: an #RTPSession
763 * @type: the type of the SDES item
764 * @item: a null-terminated string to set.
766 * Store an SDES item of @type in @sess.
768 * Returns: %FALSE if the data was unchanged @type is invalid.
771 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
776 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
778 RTP_SESSION_LOCK (sess);
779 result = rtp_source_set_sdes_string (sess->source, type, item);
780 RTP_SESSION_UNLOCK (sess);
786 * rtp_session_get_sdes_string:
787 * @sess: an #RTPSession
788 * @type: the type of the SDES item
790 * Get the SDES item of @type from @sess.
792 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
793 * valid. g_free() after usage.
796 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
800 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
802 RTP_SESSION_LOCK (sess);
803 result = rtp_source_get_sdes_string (sess->source, type);
804 RTP_SESSION_UNLOCK (sess);
810 source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
812 GstFlowReturn result = GST_FLOW_OK;
814 if (source == session->source) {
815 GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
817 RTP_SESSION_UNLOCK (session);
819 if (session->callbacks.send_rtp)
821 session->callbacks.send_rtp (session, source, buffer,
822 session->send_rtp_user_data);
824 gst_buffer_unref (buffer);
827 GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
828 RTP_SESSION_UNLOCK (session);
830 if (session->callbacks.process_rtp)
832 session->callbacks.process_rtp (session, source, buffer,
833 session->process_rtp_user_data);
835 gst_buffer_unref (buffer);
837 RTP_SESSION_LOCK (session);
843 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
847 RTP_SESSION_UNLOCK (session);
849 if (session->callbacks.clock_rate)
851 session->callbacks.clock_rate (session, pt,
852 session->clock_rate_user_data);
856 RTP_SESSION_LOCK (session);
858 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
863 static RTPSourceCallbacks callbacks = {
864 (RTPSourcePushRTP) source_push_rtp,
865 (RTPSourceClockRate) source_clock_rate,
869 * find_add_conflicting_addresses:
870 * @sess: The session to check in
871 * @arrival: The arrival stats for the buffer
873 * Checks if an address which has a conflict is already known,
874 * otherwise remembers it to prevent loops.
876 * Returns: TRUE if it was a known conflict, FALSE otherwise
880 find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
883 RTPConflictingAddress *new_conflict;
885 for (item = g_list_first (sess->conflicting_addresses);
886 item; item = g_list_next (item)) {
887 RTPConflictingAddress *known_conflict = item->data;
889 if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
890 known_conflict->time = arrival->time;
895 new_conflict = g_new0 (RTPConflictingAddress, 1);
897 memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
898 new_conflict->time = arrival->time;
900 sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
907 check_collision (RTPSession * sess, RTPSource * source,
908 RTPArrivalStats * arrival, gboolean rtp)
910 /* If we have not arrival address, we can't do collision checking */
911 if (!arrival->have_address) {
915 if (sess->source != source) {
916 /* This is not our local source, but lets check if two remote
921 if (source->have_rtp_from) {
922 if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
923 /* Address is the same */
926 /* We don't already have a from address for RTP, just set it */
927 rtp_source_set_rtp_from (source, &arrival->address);
931 if (source->have_rtcp_from) {
932 if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
933 /* Address is the same */
936 /* We don't already have a from address for RTCP, just set it */
937 rtp_source_set_rtcp_from (source, &arrival->address);
942 /* In this case, we have third-party collision or loop */
944 /* FIXME: Log 3rd party collision somehow
945 * Maybe should be done in upper layer, only the SDES can tell us
946 * if its a collision or a loop
949 /* This is sending with our ssrc, is it an address we already know */
951 if (find_add_conflicting_addresses (sess, arrival)) {
952 /* Its a known conflict, its probably a loop, not a collision
953 * lets just drop the incoming packet
955 GST_DEBUG ("Our packets are being looped back to us, dropping");
957 /* Its a new collision, lets change our SSRC */
959 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
960 on_ssrc_collision (sess, source);
962 rtp_session_send_bye_locked (sess, "SSRC Collision");
964 sess->change_ssrc = TRUE;
972 /* must be called with the session lock */
974 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
975 RTPArrivalStats * arrival, gboolean rtp)
980 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
981 if (source == NULL) {
982 /* make new Source in probation and insert */
983 source = rtp_source_new (ssrc);
985 /* for RTP packets we need to set the source in probation. Receiving RTCP
986 * packets of an SSRC, on the other hand, is a strong indication that we
987 * are dealing with a valid source. */
989 source->probation = RTP_DEFAULT_PROBATION;
991 source->probation = 0;
993 /* store from address, if any */
994 if (arrival->have_address) {
996 rtp_source_set_rtp_from (source, &arrival->address);
998 rtp_source_set_rtcp_from (source, &arrival->address);
1001 /* configure a callback on the source */
1002 rtp_source_set_callbacks (source, &callbacks, sess);
1004 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1007 /* we have one more source now */
1008 sess->total_sources++;
1012 /* check for collision, this updates the address when not previously set */
1013 if (check_collision (sess, source, arrival, rtp)) {
1017 /* update last activity */
1018 source->last_activity = arrival->time;
1020 source->last_rtp_activity = arrival->time;
1026 * rtp_session_get_internal_source:
1027 * @sess: a #RTPSession
1029 * Get the internal #RTPSource of @session.
1031 * Returns: The internal #RTPSource. g_object_unref() after usage.
1034 rtp_session_get_internal_source (RTPSession * sess)
1038 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1040 result = g_object_ref (sess->source);
1046 * rtp_session_add_source:
1047 * @sess: a #RTPSession
1048 * @src: #RTPSource to add
1050 * Add @src to @session.
1052 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1053 * existed in the session.
1056 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1058 gboolean result = FALSE;
1061 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1062 g_return_val_if_fail (src != NULL, FALSE);
1064 RTP_SESSION_LOCK (sess);
1066 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1067 GINT_TO_POINTER (src->ssrc));
1069 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1070 GINT_TO_POINTER (src->ssrc), src);
1071 /* we have one more source now */
1072 sess->total_sources++;
1075 RTP_SESSION_UNLOCK (sess);
1081 * rtp_session_get_num_sources:
1082 * @sess: an #RTPSession
1084 * Get the number of sources in @sess.
1086 * Returns: The number of sources in @sess.
1089 rtp_session_get_num_sources (RTPSession * sess)
1093 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1095 RTP_SESSION_LOCK (sess);
1096 result = sess->total_sources;
1097 RTP_SESSION_UNLOCK (sess);
1103 * rtp_session_get_num_active_sources:
1104 * @sess: an #RTPSession
1106 * Get the number of active sources in @sess. A source is considered active when
1107 * it has been validated and has not yet received a BYE RTCP message.
1109 * Returns: The number of active sources in @sess.
1112 rtp_session_get_num_active_sources (RTPSession * sess)
1116 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1118 RTP_SESSION_LOCK (sess);
1119 result = sess->stats.active_sources;
1120 RTP_SESSION_UNLOCK (sess);
1126 * rtp_session_get_source_by_ssrc:
1127 * @sess: an #RTPSession
1130 * Find the source with @ssrc in @sess.
1132 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1133 * g_object_unref() after usage.
1136 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1140 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1142 RTP_SESSION_LOCK (sess);
1144 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1146 g_object_ref (result);
1147 RTP_SESSION_UNLOCK (sess);
1153 * rtp_session_get_source_by_cname:
1154 * @sess: a #RTPSession
1157 * Find the source with @cname in @sess.
1159 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1160 * g_object_unref() after usage.
1163 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1167 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1168 g_return_val_if_fail (cname != NULL, NULL);
1170 RTP_SESSION_LOCK (sess);
1171 result = g_hash_table_lookup (sess->cnames, cname);
1173 g_object_ref (result);
1174 RTP_SESSION_UNLOCK (sess);
1180 rtp_session_create_new_ssrc (RTPSession * sess)
1185 ssrc = g_random_int ();
1187 /* see if it exists in the session, we're done if it doesn't */
1188 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1189 GINT_TO_POINTER (ssrc)) == NULL)
1198 * rtp_session_create_source:
1199 * @sess: an #RTPSession
1201 * Create an #RTPSource for use in @sess. This function will create a source
1202 * with an ssrc that is currently not used by any participants in the session.
1204 * Returns: an #RTPSource.
1207 rtp_session_create_source (RTPSession * sess)
1212 RTP_SESSION_LOCK (sess);
1213 ssrc = rtp_session_create_new_ssrc (sess);
1214 source = rtp_source_new (ssrc);
1215 g_object_ref (source);
1216 rtp_source_set_callbacks (source, &callbacks, sess);
1217 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1219 /* we have one more source now */
1220 sess->total_sources++;
1221 RTP_SESSION_UNLOCK (sess);
1226 /* update the RTPArrivalStats structure with the current time and other bits
1227 * about the current buffer we are handling.
1228 * This function is typically called when a validated packet is received.
1229 * This function should be called with the SESSION_LOCK
1232 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1233 gboolean rtp, GstBuffer * buffer, guint64 ntpnstime)
1237 /* get time of arrival */
1238 g_get_current_time (¤t);
1239 arrival->time = GST_TIMEVAL_TO_TIME (current);
1240 arrival->timestamp = GST_BUFFER_TIMESTAMP (buffer);
1241 arrival->ntpnstime = ntpnstime;
1243 /* get packet size including header overhead */
1244 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1247 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1249 arrival->payload_len = 0;
1252 /* for netbuffer we can store the IP address to check for collisions */
1253 arrival->have_address = GST_IS_NETBUFFER (buffer);
1254 if (arrival->have_address) {
1255 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1257 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1262 * rtp_session_process_rtp:
1263 * @sess: and #RTPSession
1264 * @buffer: an RTP buffer
1265 * @ntpnstime: the NTP arrival time in nanoseconds
1267 * Process an RTP buffer in the session manager. This function takes ownership
1270 * Returns: a #GstFlowReturn.
1273 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1276 GstFlowReturn result;
1280 gboolean prevsender, prevactive;
1281 RTPArrivalStats arrival;
1283 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1284 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1286 if (!gst_rtp_buffer_validate (buffer))
1287 goto invalid_packet;
1289 RTP_SESSION_LOCK (sess);
1290 /* update arrival stats */
1291 update_arrival_stats (sess, &arrival, TRUE, buffer, ntpnstime);
1293 /* ignore more RTP packets when we left the session */
1294 if (sess->source->received_bye)
1297 /* get SSRC and look up in session database */
1298 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1299 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1304 prevsender = RTP_SOURCE_IS_SENDER (source);
1305 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1307 /* we need to ref so that we can process the CSRCs later */
1308 gst_buffer_ref (buffer);
1310 /* let source process the packet */
1311 result = rtp_source_process_rtp (source, buffer, &arrival);
1313 /* source became active */
1314 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1315 sess->stats.active_sources++;
1316 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1317 sess->stats.active_sources);
1318 on_ssrc_validated (sess, source);
1320 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1321 sess->stats.sender_sources++;
1322 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1323 sess->stats.sender_sources);
1327 on_new_ssrc (sess, source);
1329 if (source->validated) {
1333 /* for validated sources, we add the CSRCs as well */
1334 count = gst_rtp_buffer_get_csrc_count (buffer);
1336 for (i = 0; i < count; i++) {
1338 RTPSource *csrc_src;
1340 csrc = gst_rtp_buffer_get_csrc (buffer, i);
1343 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1346 GST_DEBUG ("created new CSRC: %08x", csrc);
1347 rtp_source_set_as_csrc (csrc_src);
1348 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1349 sess->stats.active_sources++;
1350 on_new_ssrc (sess, source);
1354 gst_buffer_unref (buffer);
1356 RTP_SESSION_UNLOCK (sess);
1363 gst_buffer_unref (buffer);
1364 GST_DEBUG ("invalid RTP packet received");
1369 gst_buffer_unref (buffer);
1370 RTP_SESSION_UNLOCK (sess);
1371 GST_DEBUG ("ignoring RTP packet because we are leaving");
1376 gst_buffer_unref (buffer);
1377 RTP_SESSION_UNLOCK (sess);
1378 GST_DEBUG ("ignoring packet because its collisioning");
1384 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1385 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1389 count = gst_rtcp_packet_get_rb_count (packet);
1390 for (i = 0; i < count; i++) {
1391 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1392 guint8 fractionlost;
1395 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1396 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1398 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1400 if (ssrc == sess->source->ssrc) {
1401 /* only deal with report blocks for our session, we update the stats of
1402 * the sender of the RTCP message. We could also compare our stats against
1403 * the other sender to see if we are better or worse. */
1404 rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1405 exthighestseq, jitter, lsr, dlsr);
1407 on_ssrc_active (sess, source);
1412 /* A Sender report contains statistics about how the sender is doing. This
1413 * includes timing informataion such as the relation between RTP and NTP
1414 * timestamps and the number of packets/bytes it sent to us.
1416 * In this report is also included a set of report blocks related to how this
1417 * sender is receiving data (in case we (or somebody else) is also sending stuff
1418 * to it). This info includes the packet loss, jitter and seqnum. It also
1419 * contains information to calculate the round trip time (LSR/DLSR).
1422 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1423 RTPArrivalStats * arrival)
1425 guint32 senderssrc, rtptime, packet_count, octet_count;
1428 gboolean created, prevsender;
1430 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1431 &packet_count, &octet_count);
1433 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1434 senderssrc, GST_TIME_ARGS (arrival->time));
1436 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1441 GST_BUFFER_OFFSET (packet->buffer) = source->clock_base;
1442 GST_BUFFER_OFFSET_END (packet->buffer) = source->clock_base_time;
1444 prevsender = RTP_SOURCE_IS_SENDER (source);
1446 /* first update the source */
1447 rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1450 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1451 sess->stats.sender_sources++;
1452 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1453 sess->stats.sender_sources);
1457 on_new_ssrc (sess, source);
1459 rtp_session_process_rb (sess, source, packet, arrival);
1462 /* A receiver report contains statistics about how a receiver is doing. It
1463 * includes stuff like packet loss, jitter and the seqnum it received last. It
1464 * also contains info to calculate the round trip time.
1466 * We are only interested in how the sender of this report is doing wrt to us.
1469 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1470 RTPArrivalStats * arrival)
1476 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1478 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1480 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1486 on_new_ssrc (sess, source);
1488 rtp_session_process_rb (sess, source, packet, arrival);
1491 /* Get SDES items and store them in the SSRC */
1493 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1494 RTPArrivalStats * arrival)
1497 gboolean more_items, more_entries;
1499 items = gst_rtcp_packet_sdes_get_item_count (packet);
1500 GST_DEBUG ("got SDES packet with %d items", items);
1502 more_items = gst_rtcp_packet_sdes_first_item (packet);
1504 while (more_items) {
1506 gboolean changed, created;
1509 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1511 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1513 /* find src, no probation when dealing with RTCP */
1514 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1520 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1522 while (more_entries) {
1523 GstRTCPSDESType type;
1527 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1529 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1532 changed |= rtp_source_set_sdes (source, type, data, len);
1534 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1539 on_new_ssrc (sess, source);
1541 on_ssrc_sdes (sess, source);
1543 more_items = gst_rtcp_packet_sdes_next_item (packet);
1548 /* BYE is sent when a client leaves the session
1551 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1552 RTPArrivalStats * arrival)
1557 reason = gst_rtcp_packet_bye_get_reason (packet);
1558 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1560 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1561 for (i = 0; i < count; i++) {
1564 gboolean created, prevactive, prevsender;
1565 guint pmembers, members;
1567 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1568 GST_DEBUG ("SSRC: %08x", ssrc);
1570 /* find src and mark bye, no probation when dealing with RTCP */
1571 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1576 /* store time for when we need to time out this source */
1577 source->bye_time = arrival->time;
1579 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1580 prevsender = RTP_SOURCE_IS_SENDER (source);
1582 /* let the source handle the rest */
1583 rtp_source_process_bye (source, reason);
1585 pmembers = sess->stats.active_sources;
1587 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1588 sess->stats.active_sources--;
1589 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1590 sess->stats.active_sources);
1592 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1593 sess->stats.sender_sources--;
1594 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1595 sess->stats.sender_sources);
1597 members = sess->stats.active_sources;
1599 if (!sess->source->received_bye && members < pmembers) {
1600 /* some members went away since the previous timeout estimate.
1601 * Perform reverse reconsideration but only when we are not scheduling a
1603 if (arrival->time < sess->next_rtcp_check_time) {
1604 GstClockTime time_remaining;
1606 time_remaining = sess->next_rtcp_check_time - arrival->time;
1607 sess->next_rtcp_check_time =
1608 gst_util_uint64_scale (time_remaining, members, pmembers);
1610 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1611 GST_TIME_ARGS (sess->next_rtcp_check_time));
1613 sess->next_rtcp_check_time += arrival->time;
1615 RTP_SESSION_UNLOCK (sess);
1616 /* notify app of reconsideration */
1617 if (sess->callbacks.reconsider)
1618 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1619 RTP_SESSION_LOCK (sess);
1624 on_new_ssrc (sess, source);
1626 on_bye_ssrc (sess, source);
1632 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1633 RTPArrivalStats * arrival)
1635 GST_DEBUG ("received APP");
1639 * rtp_session_process_rtcp:
1640 * @sess: and #RTPSession
1641 * @buffer: an RTCP buffer
1643 * Process an RTCP buffer in the session manager. This function takes ownership
1646 * Returns: a #GstFlowReturn.
1649 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
1651 GstRTCPPacket packet;
1652 gboolean more, is_bye = FALSE, is_sr = FALSE;
1653 RTPArrivalStats arrival;
1654 GstFlowReturn result = GST_FLOW_OK;
1656 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1657 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1659 if (!gst_rtcp_buffer_validate (buffer))
1660 goto invalid_packet;
1662 GST_DEBUG ("received RTCP packet");
1664 RTP_SESSION_LOCK (sess);
1665 /* update arrival stats */
1666 update_arrival_stats (sess, &arrival, FALSE, buffer, -1);
1671 /* start processing the compound packet */
1672 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1676 type = gst_rtcp_packet_get_type (&packet);
1678 /* when we are leaving the session, we should ignore all non-BYE messages */
1679 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1680 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1685 case GST_RTCP_TYPE_SR:
1686 rtp_session_process_sr (sess, &packet, &arrival);
1689 case GST_RTCP_TYPE_RR:
1690 rtp_session_process_rr (sess, &packet, &arrival);
1692 case GST_RTCP_TYPE_SDES:
1693 rtp_session_process_sdes (sess, &packet, &arrival);
1695 case GST_RTCP_TYPE_BYE:
1697 rtp_session_process_bye (sess, &packet, &arrival);
1699 case GST_RTCP_TYPE_APP:
1700 rtp_session_process_app (sess, &packet, &arrival);
1703 GST_WARNING ("got unknown RTCP packet");
1707 more = gst_rtcp_packet_move_to_next (&packet);
1710 /* if we are scheduling a BYE, we only want to count bye packets, else we
1711 * count everything */
1712 if (sess->source->received_bye) {
1714 sess->stats.bye_members++;
1715 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1718 /* keep track of average packet size */
1719 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1721 RTP_SESSION_UNLOCK (sess);
1723 /* notify caller of sr packets in the callback */
1724 if (is_sr && sess->callbacks.sync_rtcp)
1725 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1726 sess->sync_rtcp_user_data);
1728 gst_buffer_unref (buffer);
1735 GST_DEBUG ("invalid RTCP packet received");
1736 gst_buffer_unref (buffer);
1741 gst_buffer_unref (buffer);
1742 RTP_SESSION_UNLOCK (sess);
1743 GST_DEBUG ("ignoring RTP packet because we left");
1749 * rtp_session_send_rtp:
1750 * @sess: an #RTPSession
1751 * @buffer: an RTP buffer
1752 * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
1754 * Send the RTP buffer in the session manager. This function takes ownership of
1757 * Returns: a #GstFlowReturn.
1760 rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer, guint64 ntpnstime)
1762 GstFlowReturn result;
1764 gboolean prevsender;
1767 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1768 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1770 if (!gst_rtp_buffer_validate (buffer))
1771 goto invalid_packet;
1773 GST_DEBUG ("received RTP packet for sending");
1775 RTP_SESSION_LOCK (sess);
1776 source = sess->source;
1778 /* update last activity */
1779 g_get_current_time (¤t);
1780 source->last_rtp_activity = GST_TIMEVAL_TO_TIME (current);
1782 prevsender = RTP_SOURCE_IS_SENDER (source);
1784 /* we use our own source to send */
1785 result = rtp_source_send_rtp (source, buffer, ntpnstime);
1787 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
1788 sess->stats.sender_sources++;
1789 RTP_SESSION_UNLOCK (sess);
1796 gst_buffer_unref (buffer);
1797 GST_DEBUG ("invalid RTP packet received");
1803 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
1806 GstClockTime result;
1808 if (sess->source->received_bye) {
1809 result = rtp_stats_calculate_bye_interval (&sess->stats);
1811 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
1812 RTP_SOURCE_IS_SENDER (sess->source), first);
1815 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
1816 GST_TIME_ARGS (result), first);
1819 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
1821 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
1827 * rtp_session_send_bye_locked:
1828 * @sess: an #RTPSession
1829 * @reason: a reason or NULL
1831 * Stop the current @sess and schedule a BYE message for the other members.
1833 * One must have the session lock to call this function
1835 * Returns: a #GstFlowReturn.
1837 static GstFlowReturn
1838 rtp_session_send_bye_locked (RTPSession * sess, const gchar * reason)
1840 GstFlowReturn result = GST_FLOW_OK;
1842 GstClockTime current, interval;
1845 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1847 source = sess->source;
1849 /* ignore more BYEs */
1850 if (source->received_bye)
1853 /* we have BYE now */
1854 source->received_bye = TRUE;
1855 /* at least one member wants to send a BYE */
1856 g_free (sess->bye_reason);
1857 sess->bye_reason = g_strdup (reason);
1858 sess->stats.avg_rtcp_packet_size = 100;
1859 sess->stats.bye_members = 1;
1860 sess->first_rtcp = TRUE;
1861 sess->sent_bye = FALSE;
1863 /* get current time */
1864 g_get_current_time (&curtv);
1865 current = GST_TIMEVAL_TO_TIME (curtv);
1867 /* reschedule transmission */
1868 sess->last_rtcp_send_time = current;
1869 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
1870 sess->next_rtcp_check_time = current + interval;
1872 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
1873 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
1875 RTP_SESSION_UNLOCK (sess);
1876 /* notify app of reconsideration */
1877 if (sess->callbacks.reconsider)
1878 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1879 RTP_SESSION_LOCK (sess);
1886 * rtp_session_send_bye:
1887 * @sess: an #RTPSession
1888 * @reason: a reason or NULL
1890 * Stop the current @sess and schedule a BYE message for the other members.
1892 * One must have the session lock to call this function
1894 * Returns: a #GstFlowReturn.
1897 rtp_session_send_bye (RTPSession * sess, const gchar * reason)
1899 GstFlowReturn result = GST_FLOW_OK;
1901 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1903 RTP_SESSION_LOCK (sess);
1904 result = rtp_session_send_bye_locked (sess, reason);
1905 RTP_SESSION_UNLOCK (sess);
1911 * rtp_session_next_timeout:
1912 * @sess: an #RTPSession
1913 * @time: the current system time
1915 * Get the next time we should perform session maintenance tasks.
1917 * Returns: a time when rtp_session_on_timeout() should be called with the
1918 * current system time.
1921 rtp_session_next_timeout (RTPSession * sess, GstClockTime time)
1923 GstClockTime result;
1925 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1927 RTP_SESSION_LOCK (sess);
1929 result = sess->next_rtcp_check_time;
1931 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
1932 GST_TIME_ARGS (time), GST_TIME_ARGS (result));
1934 if (result < time) {
1935 GST_DEBUG ("take current time as base");
1936 /* our previous check time expired, start counting from the current time
1941 if (sess->source->received_bye) {
1942 if (sess->sent_bye) {
1943 GST_DEBUG ("we sent BYE already");
1944 result = GST_CLOCK_TIME_NONE;
1945 } else if (sess->stats.active_sources >= 50) {
1946 GST_DEBUG ("reconsider BYE, more than 50 sources");
1947 /* reconsider BYE if members >= 50 */
1948 result += calculate_rtcp_interval (sess, FALSE, TRUE);
1951 if (sess->first_rtcp) {
1952 GST_DEBUG ("first RTCP packet");
1953 /* we are called for the first time */
1954 result += calculate_rtcp_interval (sess, FALSE, TRUE);
1955 } else if (sess->next_rtcp_check_time < time) {
1956 GST_DEBUG ("old check time expired, getting new timeout");
1957 /* get a new timeout when we need to */
1958 result += calculate_rtcp_interval (sess, FALSE, FALSE);
1961 sess->next_rtcp_check_time = result;
1963 GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
1964 RTP_SESSION_UNLOCK (sess);
1975 GstClockTime interval;
1976 GstRTCPPacket packet;
1982 session_start_rtcp (RTPSession * sess, ReportData * data)
1984 GstRTCPPacket *packet = &data->packet;
1985 RTPSource *own = sess->source;
1987 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
1989 if (RTP_SOURCE_IS_SENDER (own)) {
1992 guint32 packet_count, octet_count;
1994 /* we are a sender, create SR */
1995 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
1996 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
1998 /* get latest stats */
1999 rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
2000 &packet_count, &octet_count);
2002 rtp_source_process_sr (own, data->ntpnstime, ntptime, rtptime, packet_count,
2005 /* fill in sender report info */
2006 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2007 ntptime, rtptime, packet_count, octet_count);
2009 /* we are only receiver, create RR */
2010 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2011 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2012 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2016 /* construct a Sender or Receiver Report */
2018 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2020 RTPSession *sess = data->sess;
2021 GstRTCPPacket *packet = &data->packet;
2023 /* create a new buffer if needed */
2024 if (data->rtcp == NULL) {
2025 session_start_rtcp (sess, data);
2027 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2028 /* only report about other sender sources */
2029 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2030 guint8 fractionlost;
2032 guint32 exthighestseq, jitter;
2036 rtp_source_get_new_rb (source, data->time, &fractionlost, &packetslost,
2037 &exthighestseq, &jitter, &lsr, &dlsr);
2039 /* packet is not yet filled, add report block for this source. */
2040 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2041 exthighestseq, jitter, lsr, dlsr);
2046 /* perform cleanup of sources that timed out */
2048 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2050 gboolean remove = FALSE;
2051 gboolean byetimeout = FALSE;
2052 gboolean is_sender, is_active;
2053 RTPSession *sess = data->sess;
2054 GstClockTime interval;
2056 is_sender = RTP_SOURCE_IS_SENDER (source);
2057 is_active = RTP_SOURCE_IS_ACTIVE (source);
2059 /* check for our own source, we don't want to delete our own source. */
2060 if (!(source == sess->source)) {
2061 if (source->received_bye) {
2062 /* if we received a BYE from the source, remove the source after some
2064 if (data->time > source->bye_time &&
2065 data->time - source->bye_time > sess->stats.bye_timeout) {
2066 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2071 /* sources that were inactive for more than 5 times the deterministic reporting
2072 * interval get timed out. the min timeout is 5 seconds. */
2073 if (data->time > source->last_activity) {
2074 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2075 if (data->time - source->last_activity > interval) {
2076 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2077 source->ssrc, GST_TIME_ARGS (source->last_activity));
2083 /* senders that did not send for a long time become a receiver, this also
2084 * holds for our own source. */
2086 if (data->time > source->last_rtp_activity) {
2087 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2088 if (data->time - source->last_rtp_activity > interval) {
2089 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2090 GST_TIME_FORMAT, source->ssrc,
2091 GST_TIME_ARGS (source->last_rtp_activity));
2092 source->is_sender = FALSE;
2093 sess->stats.sender_sources--;
2099 sess->total_sources--;
2101 sess->stats.sender_sources--;
2103 sess->stats.active_sources--;
2106 on_bye_timeout (sess, source);
2108 on_timeout (sess, source);
2114 session_sdes (RTPSession * sess, ReportData * data)
2116 GstRTCPPacket *packet = &data->packet;
2120 /* add SDES packet */
2121 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2123 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2125 rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
2127 gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
2130 /* other SDES items must only be added at regular intervals and only when the
2131 * user requests to since it might be a privacy problem */
2133 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
2134 strlen (sess->name), (guint8 *) sess->name);
2135 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
2136 strlen (sess->tool), (guint8 *) sess->tool);
2139 data->has_sdes = TRUE;
2142 /* schedule a BYE packet */
2144 session_bye (RTPSession * sess, ReportData * data)
2146 GstRTCPPacket *packet = &data->packet;
2149 session_start_rtcp (sess, data);
2152 session_sdes (sess, data);
2154 /* add a BYE packet */
2155 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2156 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2157 if (sess->bye_reason)
2158 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2160 /* we have a BYE packet now */
2161 data->is_bye = TRUE;
2165 is_rtcp_time (RTPSession * sess, GstClockTime time, ReportData * data)
2167 GstClockTime new_send_time, elapsed;
2170 /* no need to check yet */
2171 if (sess->next_rtcp_check_time > time) {
2172 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2173 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2174 GST_TIME_ARGS (time));
2178 /* get elapsed time since we last reported */
2179 elapsed = time - sess->last_rtcp_send_time;
2181 /* perform forward reconsideration */
2182 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2184 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2185 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2187 new_send_time += sess->last_rtcp_send_time;
2189 /* check if reconsideration */
2190 if (time < new_send_time) {
2191 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2192 GST_TIME_ARGS (new_send_time));
2194 /* store new check time */
2195 sess->next_rtcp_check_time = new_send_time;
2198 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2200 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2201 GST_TIME_ARGS (new_send_time));
2202 sess->next_rtcp_check_time = time + new_send_time;
2208 * rtp_session_on_timeout:
2209 * @sess: an #RTPSession
2210 * @time: the current system time
2211 * @ntpnstime: the current NTP time in nanoseconds
2213 * Perform maintenance actions after the timeout obtained with
2214 * rtp_session_next_timeout() expired.
2216 * This function will perform timeouts of receivers and senders, send a BYE
2217 * packet or generate RTCP packets with current session stats.
2219 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2220 * times, for each packet that should be processed.
2222 * Returns: a #GstFlowReturn.
2225 rtp_session_on_timeout (RTPSession * sess, GstClockTime time, guint64 ntpnstime)
2227 GstFlowReturn result = GST_FLOW_OK;
2231 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2233 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2234 GST_TIME_ARGS (time), GST_TIME_ARGS (ntpnstime));
2239 data.ntpnstime = ntpnstime;
2240 data.is_bye = FALSE;
2241 data.has_sdes = FALSE;
2243 RTP_SESSION_LOCK (sess);
2244 /* get a new interval, we need this for various cleanups etc */
2245 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2247 /* first perform cleanups */
2248 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2249 (GHRFunc) session_cleanup, &data);
2251 /* see if we need to generate SR or RR packets */
2252 if (is_rtcp_time (sess, time, &data)) {
2253 if (sess->source->received_bye) {
2254 /* generate BYE instead */
2255 session_bye (sess, &data);
2256 sess->sent_bye = TRUE;
2258 /* loop over all known sources and do something */
2259 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2260 (GHFunc) session_report_blocks, &data);
2267 /* we keep track of the last report time in order to timeout inactive
2268 * receivers or senders */
2269 sess->last_rtcp_send_time = data.time;
2270 sess->first_rtcp = FALSE;
2272 /* add SDES for this source when not already added */
2274 session_sdes (sess, &data);
2276 /* update average RTCP size before sending */
2277 size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2278 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2281 /* check for outdated collisions */
2282 item = g_list_first (sess->conflicting_addresses);
2284 RTPConflictingAddress *known_conflict = item->data;
2285 GList *next_item = g_list_next (item);
2287 if (known_conflict->time < time - (data.interval *
2288 RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2289 sess->conflicting_addresses =
2290 g_list_delete_link (sess->conflicting_addresses, item);
2291 g_free (known_conflict);
2296 if (sess->change_ssrc) {
2297 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2298 GINT_TO_POINTER (sess->source->ssrc));
2300 sess->source->ssrc = rtp_session_create_new_ssrc (sess);
2301 rtp_source_reset (sess->source);
2303 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2304 GINT_TO_POINTER (sess->source->ssrc), sess->source);
2306 g_free (sess->bye_reason);
2307 sess->bye_reason = NULL;
2308 sess->sent_bye = FALSE;
2309 sess->change_ssrc = FALSE;
2311 RTP_SESSION_UNLOCK (sess);
2313 /* push out the RTCP packet */
2315 /* close the RTCP packet */
2316 gst_rtcp_buffer_end (data.rtcp);
2318 if (sess->callbacks.send_rtcp)
2319 result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
2320 sess->send_rtcp_user_data);
2322 gst_buffer_unref (data.rtcp);