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 if (session->callbacks.clock_rate)
849 session->callbacks.clock_rate (session, pt,
850 session->clock_rate_user_data);
854 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
859 static RTPSourceCallbacks callbacks = {
860 (RTPSourcePushRTP) source_push_rtp,
861 (RTPSourceClockRate) source_clock_rate,
865 * find_add_conflicting_addresses:
866 * @sess: The session to check in
867 * @arrival: The arrival stats for the buffer
869 * Checks if an address which has a conflict is already known,
870 * otherwise remembers it to prevent loops.
872 * Returns: TRUE if it was a known conflict, FALSE otherwise
876 find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
879 RTPConflictingAddress *new_conflict;
881 for (item = g_list_first (sess->conflicting_addresses);
882 item; item = g_list_next (item)) {
883 RTPConflictingAddress *known_conflict = item->data;
885 if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
886 known_conflict->time = arrival->time;
891 new_conflict = g_new0 (RTPConflictingAddress, 1);
893 memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
894 new_conflict->time = arrival->time;
896 sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
903 check_collision (RTPSession * sess, RTPSource * source,
904 RTPArrivalStats * arrival, gboolean rtp)
906 /* If we have not arrival address, we can't do collision checking */
907 if (!arrival->have_address) {
911 if (sess->source != source) {
912 /* This is not our local source, but lets check if two remote
917 if (source->have_rtp_from) {
918 if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
919 /* Address is the same */
922 /* We don't already have a from address for RTP, just set it */
923 rtp_source_set_rtp_from (source, &arrival->address);
927 if (source->have_rtcp_from) {
928 if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
929 /* Address is the same */
932 /* We don't already have a from address for RTCP, just set it */
933 rtp_source_set_rtcp_from (source, &arrival->address);
938 /* In this case, we have third-party collision or loop */
940 /* FIXME: Log 3rd party collision somehow
941 * Maybe should be done in upper layer, only the SDES can tell us
942 * if its a collision or a loop
945 /* This is sending with our ssrc, is it an address we already know */
947 if (find_add_conflicting_addresses (sess, arrival)) {
948 /* Its a known conflict, its probably a loop, not a collision
949 * lets just drop the incoming packet
951 GST_DEBUG ("Our packets are being looped back to us, dropping");
953 /* Its a new collision, lets change our SSRC */
955 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
956 on_ssrc_collision (sess, source);
958 rtp_session_send_bye_locked (sess, "SSRC Collision");
960 sess->change_ssrc = TRUE;
968 /* must be called with the session lock */
970 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
971 RTPArrivalStats * arrival, gboolean rtp)
976 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
977 if (source == NULL) {
978 /* make new Source in probation and insert */
979 source = rtp_source_new (ssrc);
981 /* for RTP packets we need to set the source in probation. Receiving RTCP
982 * packets of an SSRC, on the other hand, is a strong indication that we
983 * are dealing with a valid source. */
985 source->probation = RTP_DEFAULT_PROBATION;
987 source->probation = 0;
989 /* store from address, if any */
990 if (arrival->have_address) {
992 rtp_source_set_rtp_from (source, &arrival->address);
994 rtp_source_set_rtcp_from (source, &arrival->address);
997 /* configure a callback on the source */
998 rtp_source_set_callbacks (source, &callbacks, sess);
1000 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1003 /* we have one more source now */
1004 sess->total_sources++;
1008 /* check for collision, this updates the address when not previously set */
1009 if (check_collision (sess, source, arrival, rtp)) {
1013 /* update last activity */
1014 source->last_activity = arrival->time;
1016 source->last_rtp_activity = arrival->time;
1022 * rtp_session_get_internal_source:
1023 * @sess: a #RTPSession
1025 * Get the internal #RTPSource of @session.
1027 * Returns: The internal #RTPSource. g_object_unref() after usage.
1030 rtp_session_get_internal_source (RTPSession * sess)
1034 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1036 result = g_object_ref (sess->source);
1042 * rtp_session_add_source:
1043 * @sess: a #RTPSession
1044 * @src: #RTPSource to add
1046 * Add @src to @session.
1048 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1049 * existed in the session.
1052 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1054 gboolean result = FALSE;
1057 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1058 g_return_val_if_fail (src != NULL, FALSE);
1060 RTP_SESSION_LOCK (sess);
1062 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1063 GINT_TO_POINTER (src->ssrc));
1065 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1066 GINT_TO_POINTER (src->ssrc), src);
1067 /* we have one more source now */
1068 sess->total_sources++;
1071 RTP_SESSION_UNLOCK (sess);
1077 * rtp_session_get_num_sources:
1078 * @sess: an #RTPSession
1080 * Get the number of sources in @sess.
1082 * Returns: The number of sources in @sess.
1085 rtp_session_get_num_sources (RTPSession * sess)
1089 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1091 RTP_SESSION_LOCK (sess);
1092 result = sess->total_sources;
1093 RTP_SESSION_UNLOCK (sess);
1099 * rtp_session_get_num_active_sources:
1100 * @sess: an #RTPSession
1102 * Get the number of active sources in @sess. A source is considered active when
1103 * it has been validated and has not yet received a BYE RTCP message.
1105 * Returns: The number of active sources in @sess.
1108 rtp_session_get_num_active_sources (RTPSession * sess)
1112 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1114 RTP_SESSION_LOCK (sess);
1115 result = sess->stats.active_sources;
1116 RTP_SESSION_UNLOCK (sess);
1122 * rtp_session_get_source_by_ssrc:
1123 * @sess: an #RTPSession
1126 * Find the source with @ssrc in @sess.
1128 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1129 * g_object_unref() after usage.
1132 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1136 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1138 RTP_SESSION_LOCK (sess);
1140 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1142 g_object_ref (result);
1143 RTP_SESSION_UNLOCK (sess);
1149 * rtp_session_get_source_by_cname:
1150 * @sess: a #RTPSession
1153 * Find the source with @cname in @sess.
1155 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1156 * g_object_unref() after usage.
1159 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1163 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1164 g_return_val_if_fail (cname != NULL, NULL);
1166 RTP_SESSION_LOCK (sess);
1167 result = g_hash_table_lookup (sess->cnames, cname);
1169 g_object_ref (result);
1170 RTP_SESSION_UNLOCK (sess);
1176 rtp_session_create_new_ssrc (RTPSession * sess)
1181 ssrc = g_random_int ();
1183 /* see if it exists in the session, we're done if it doesn't */
1184 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1185 GINT_TO_POINTER (ssrc)) == NULL)
1194 * rtp_session_create_source:
1195 * @sess: an #RTPSession
1197 * Create an #RTPSource for use in @sess. This function will create a source
1198 * with an ssrc that is currently not used by any participants in the session.
1200 * Returns: an #RTPSource.
1203 rtp_session_create_source (RTPSession * sess)
1208 RTP_SESSION_LOCK (sess);
1209 ssrc = rtp_session_create_new_ssrc (sess);
1210 source = rtp_source_new (ssrc);
1211 g_object_ref (source);
1212 rtp_source_set_callbacks (source, &callbacks, sess);
1213 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1215 /* we have one more source now */
1216 sess->total_sources++;
1217 RTP_SESSION_UNLOCK (sess);
1222 /* update the RTPArrivalStats structure with the current time and other bits
1223 * about the current buffer we are handling.
1224 * This function is typically called when a validated packet is received.
1225 * This function should be called with the SESSION_LOCK
1228 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1229 gboolean rtp, GstBuffer * buffer, guint64 ntpnstime)
1233 /* get time of arrival */
1234 g_get_current_time (¤t);
1235 arrival->time = GST_TIMEVAL_TO_TIME (current);
1236 arrival->ntpnstime = ntpnstime;
1238 /* get packet size including header overhead */
1239 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1242 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1244 arrival->payload_len = 0;
1247 /* for netbuffer we can store the IP address to check for collisions */
1248 arrival->have_address = GST_IS_NETBUFFER (buffer);
1249 if (arrival->have_address) {
1250 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1252 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1257 * rtp_session_process_rtp:
1258 * @sess: and #RTPSession
1259 * @buffer: an RTP buffer
1260 * @ntpnstime: the NTP arrival time in nanoseconds
1262 * Process an RTP buffer in the session manager. This function takes ownership
1265 * Returns: a #GstFlowReturn.
1268 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1271 GstFlowReturn result;
1275 gboolean prevsender, prevactive;
1276 RTPArrivalStats arrival;
1278 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1279 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1281 if (!gst_rtp_buffer_validate (buffer))
1282 goto invalid_packet;
1284 RTP_SESSION_LOCK (sess);
1285 /* update arrival stats */
1286 update_arrival_stats (sess, &arrival, TRUE, buffer, ntpnstime);
1288 /* ignore more RTP packets when we left the session */
1289 if (sess->source->received_bye)
1292 /* get SSRC and look up in session database */
1293 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1294 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1299 prevsender = RTP_SOURCE_IS_SENDER (source);
1300 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1302 /* we need to ref so that we can process the CSRCs later */
1303 gst_buffer_ref (buffer);
1305 /* let source process the packet */
1306 result = rtp_source_process_rtp (source, buffer, &arrival);
1308 /* source became active */
1309 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1310 sess->stats.active_sources++;
1311 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1312 sess->stats.active_sources);
1313 on_ssrc_validated (sess, source);
1315 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1316 sess->stats.sender_sources++;
1317 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1318 sess->stats.sender_sources);
1322 on_new_ssrc (sess, source);
1324 if (source->validated) {
1328 /* for validated sources, we add the CSRCs as well */
1329 count = gst_rtp_buffer_get_csrc_count (buffer);
1331 for (i = 0; i < count; i++) {
1333 RTPSource *csrc_src;
1335 csrc = gst_rtp_buffer_get_csrc (buffer, i);
1338 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1341 GST_DEBUG ("created new CSRC: %08x", csrc);
1342 rtp_source_set_as_csrc (csrc_src);
1343 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1344 sess->stats.active_sources++;
1345 on_new_ssrc (sess, source);
1349 gst_buffer_unref (buffer);
1351 RTP_SESSION_UNLOCK (sess);
1358 gst_buffer_unref (buffer);
1359 GST_DEBUG ("invalid RTP packet received");
1364 gst_buffer_unref (buffer);
1365 RTP_SESSION_UNLOCK (sess);
1366 GST_DEBUG ("ignoring RTP packet because we are leaving");
1371 gst_buffer_unref (buffer);
1372 RTP_SESSION_UNLOCK (sess);
1373 GST_DEBUG ("ignoring packet because its collisioning");
1379 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1380 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1384 count = gst_rtcp_packet_get_rb_count (packet);
1385 for (i = 0; i < count; i++) {
1386 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1387 guint8 fractionlost;
1390 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1391 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1393 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1395 if (ssrc == sess->source->ssrc) {
1396 /* only deal with report blocks for our session, we update the stats of
1397 * the sender of the RTCP message. We could also compare our stats against
1398 * the other sender to see if we are better or worse. */
1399 rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1400 exthighestseq, jitter, lsr, dlsr);
1402 on_ssrc_active (sess, source);
1407 /* A Sender report contains statistics about how the sender is doing. This
1408 * includes timing informataion such as the relation between RTP and NTP
1409 * timestamps and the number of packets/bytes it sent to us.
1411 * In this report is also included a set of report blocks related to how this
1412 * sender is receiving data (in case we (or somebody else) is also sending stuff
1413 * to it). This info includes the packet loss, jitter and seqnum. It also
1414 * contains information to calculate the round trip time (LSR/DLSR).
1417 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1418 RTPArrivalStats * arrival)
1420 guint32 senderssrc, rtptime, packet_count, octet_count;
1423 gboolean created, prevsender;
1425 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1426 &packet_count, &octet_count);
1428 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1429 senderssrc, GST_TIME_ARGS (arrival->time));
1431 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1436 GST_BUFFER_OFFSET (packet->buffer) = source->clock_base;
1438 prevsender = RTP_SOURCE_IS_SENDER (source);
1440 /* first update the source */
1441 rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1444 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1445 sess->stats.sender_sources++;
1446 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1447 sess->stats.sender_sources);
1451 on_new_ssrc (sess, source);
1453 rtp_session_process_rb (sess, source, packet, arrival);
1456 /* A receiver report contains statistics about how a receiver is doing. It
1457 * includes stuff like packet loss, jitter and the seqnum it received last. It
1458 * also contains info to calculate the round trip time.
1460 * We are only interested in how the sender of this report is doing wrt to us.
1463 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1464 RTPArrivalStats * arrival)
1470 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1472 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1474 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1480 on_new_ssrc (sess, source);
1482 rtp_session_process_rb (sess, source, packet, arrival);
1485 /* Get SDES items and store them in the SSRC */
1487 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1488 RTPArrivalStats * arrival)
1491 gboolean more_items, more_entries;
1493 items = gst_rtcp_packet_sdes_get_item_count (packet);
1494 GST_DEBUG ("got SDES packet with %d items", items);
1496 more_items = gst_rtcp_packet_sdes_first_item (packet);
1498 while (more_items) {
1500 gboolean changed, created;
1503 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1505 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1507 /* find src, no probation when dealing with RTCP */
1508 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1514 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1516 while (more_entries) {
1517 GstRTCPSDESType type;
1521 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1523 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1526 changed |= rtp_source_set_sdes (source, type, data, len);
1528 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1533 on_new_ssrc (sess, source);
1535 on_ssrc_sdes (sess, source);
1537 more_items = gst_rtcp_packet_sdes_next_item (packet);
1542 /* BYE is sent when a client leaves the session
1545 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1546 RTPArrivalStats * arrival)
1551 reason = gst_rtcp_packet_bye_get_reason (packet);
1552 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1554 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1555 for (i = 0; i < count; i++) {
1558 gboolean created, prevactive, prevsender;
1559 guint pmembers, members;
1561 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1562 GST_DEBUG ("SSRC: %08x", ssrc);
1564 /* find src and mark bye, no probation when dealing with RTCP */
1565 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1570 /* store time for when we need to time out this source */
1571 source->bye_time = arrival->time;
1573 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1574 prevsender = RTP_SOURCE_IS_SENDER (source);
1576 /* let the source handle the rest */
1577 rtp_source_process_bye (source, reason);
1579 pmembers = sess->stats.active_sources;
1581 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1582 sess->stats.active_sources--;
1583 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1584 sess->stats.active_sources);
1586 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1587 sess->stats.sender_sources--;
1588 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1589 sess->stats.sender_sources);
1591 members = sess->stats.active_sources;
1593 if (!sess->source->received_bye && members < pmembers) {
1594 /* some members went away since the previous timeout estimate.
1595 * Perform reverse reconsideration but only when we are not scheduling a
1597 if (arrival->time < sess->next_rtcp_check_time) {
1598 GstClockTime time_remaining;
1600 time_remaining = sess->next_rtcp_check_time - arrival->time;
1601 sess->next_rtcp_check_time =
1602 gst_util_uint64_scale (time_remaining, members, pmembers);
1604 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1605 GST_TIME_ARGS (sess->next_rtcp_check_time));
1607 sess->next_rtcp_check_time += arrival->time;
1609 /* notify app of reconsideration */
1610 if (sess->callbacks.reconsider)
1611 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1616 on_new_ssrc (sess, source);
1618 on_bye_ssrc (sess, source);
1624 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1625 RTPArrivalStats * arrival)
1627 GST_DEBUG ("received APP");
1631 * rtp_session_process_rtcp:
1632 * @sess: and #RTPSession
1633 * @buffer: an RTCP buffer
1635 * Process an RTCP buffer in the session manager. This function takes ownership
1638 * Returns: a #GstFlowReturn.
1641 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
1643 GstRTCPPacket packet;
1644 gboolean more, is_bye = FALSE, is_sr = FALSE;
1645 RTPArrivalStats arrival;
1646 GstFlowReturn result = GST_FLOW_OK;
1648 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1649 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1651 if (!gst_rtcp_buffer_validate (buffer))
1652 goto invalid_packet;
1654 GST_DEBUG ("received RTCP packet");
1656 RTP_SESSION_LOCK (sess);
1657 /* update arrival stats */
1658 update_arrival_stats (sess, &arrival, FALSE, buffer, -1);
1663 /* start processing the compound packet */
1664 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1668 type = gst_rtcp_packet_get_type (&packet);
1670 /* when we are leaving the session, we should ignore all non-BYE messages */
1671 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1672 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1677 case GST_RTCP_TYPE_SR:
1678 rtp_session_process_sr (sess, &packet, &arrival);
1681 case GST_RTCP_TYPE_RR:
1682 rtp_session_process_rr (sess, &packet, &arrival);
1684 case GST_RTCP_TYPE_SDES:
1685 rtp_session_process_sdes (sess, &packet, &arrival);
1687 case GST_RTCP_TYPE_BYE:
1689 rtp_session_process_bye (sess, &packet, &arrival);
1691 case GST_RTCP_TYPE_APP:
1692 rtp_session_process_app (sess, &packet, &arrival);
1695 GST_WARNING ("got unknown RTCP packet");
1699 more = gst_rtcp_packet_move_to_next (&packet);
1702 /* if we are scheduling a BYE, we only want to count bye packets, else we
1703 * count everything */
1704 if (sess->source->received_bye) {
1706 sess->stats.bye_members++;
1707 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1710 /* keep track of average packet size */
1711 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1713 RTP_SESSION_UNLOCK (sess);
1715 /* notify caller of sr packets in the callback */
1716 if (is_sr && sess->callbacks.sync_rtcp)
1717 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1718 sess->sync_rtcp_user_data);
1720 gst_buffer_unref (buffer);
1727 GST_DEBUG ("invalid RTCP packet received");
1728 gst_buffer_unref (buffer);
1733 gst_buffer_unref (buffer);
1734 RTP_SESSION_UNLOCK (sess);
1735 GST_DEBUG ("ignoring RTP packet because we left");
1741 * rtp_session_send_rtp:
1742 * @sess: an #RTPSession
1743 * @buffer: an RTP buffer
1744 * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
1746 * Send the RTP buffer in the session manager. This function takes ownership of
1749 * Returns: a #GstFlowReturn.
1752 rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer, guint64 ntpnstime)
1754 GstFlowReturn result;
1756 gboolean prevsender;
1759 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1760 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1762 if (!gst_rtp_buffer_validate (buffer))
1763 goto invalid_packet;
1765 GST_DEBUG ("received RTP packet for sending");
1767 RTP_SESSION_LOCK (sess);
1768 source = sess->source;
1770 /* update last activity */
1771 g_get_current_time (¤t);
1772 source->last_rtp_activity = GST_TIMEVAL_TO_TIME (current);
1774 prevsender = RTP_SOURCE_IS_SENDER (source);
1776 /* we use our own source to send */
1777 result = rtp_source_send_rtp (source, buffer, ntpnstime);
1779 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
1780 sess->stats.sender_sources++;
1781 RTP_SESSION_UNLOCK (sess);
1788 gst_buffer_unref (buffer);
1789 GST_DEBUG ("invalid RTP packet received");
1795 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
1798 GstClockTime result;
1800 if (sess->source->received_bye) {
1801 result = rtp_stats_calculate_bye_interval (&sess->stats);
1803 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
1804 RTP_SOURCE_IS_SENDER (sess->source), first);
1807 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
1808 GST_TIME_ARGS (result), first);
1811 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
1813 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
1819 * rtp_session_send_bye_locked:
1820 * @sess: an #RTPSession
1821 * @reason: a reason or NULL
1823 * Stop the current @sess and schedule a BYE message for the other members.
1825 * One must have the session lock to call this function
1827 * Returns: a #GstFlowReturn.
1829 static GstFlowReturn
1830 rtp_session_send_bye_locked (RTPSession * sess, const gchar * reason)
1832 GstFlowReturn result = GST_FLOW_OK;
1834 GstClockTime current, interval;
1837 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1839 source = sess->source;
1841 /* ignore more BYEs */
1842 if (source->received_bye)
1845 /* we have BYE now */
1846 source->received_bye = TRUE;
1847 /* at least one member wants to send a BYE */
1848 g_free (sess->bye_reason);
1849 sess->bye_reason = g_strdup (reason);
1850 sess->stats.avg_rtcp_packet_size = 100;
1851 sess->stats.bye_members = 1;
1852 sess->first_rtcp = TRUE;
1853 sess->sent_bye = FALSE;
1855 /* get current time */
1856 g_get_current_time (&curtv);
1857 current = GST_TIMEVAL_TO_TIME (curtv);
1859 /* reschedule transmission */
1860 sess->last_rtcp_send_time = current;
1861 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
1862 sess->next_rtcp_check_time = current + interval;
1864 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
1865 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
1867 /* notify app of reconsideration */
1868 if (sess->callbacks.reconsider)
1869 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1876 * rtp_session_send_bye:
1877 * @sess: an #RTPSession
1878 * @reason: a reason or NULL
1880 * Stop the current @sess and schedule a BYE message for the other members.
1882 * One must have the session lock to call this function
1884 * Returns: a #GstFlowReturn.
1887 rtp_session_send_bye (RTPSession * sess, const gchar * reason)
1889 GstFlowReturn result = GST_FLOW_OK;
1891 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1893 RTP_SESSION_LOCK (sess);
1894 result = rtp_session_send_bye_locked (sess, reason);
1895 RTP_SESSION_UNLOCK (sess);
1901 * rtp_session_next_timeout:
1902 * @sess: an #RTPSession
1903 * @time: the current system time
1905 * Get the next time we should perform session maintenance tasks.
1907 * Returns: a time when rtp_session_on_timeout() should be called with the
1908 * current system time.
1911 rtp_session_next_timeout (RTPSession * sess, GstClockTime time)
1913 GstClockTime result;
1915 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1917 RTP_SESSION_LOCK (sess);
1919 result = sess->next_rtcp_check_time;
1921 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
1922 GST_TIME_ARGS (time), GST_TIME_ARGS (result));
1924 if (result < time) {
1925 GST_DEBUG ("take current time as base");
1926 /* our previous check time expired, start counting from the current time
1931 if (sess->source->received_bye) {
1932 if (sess->sent_bye) {
1933 GST_DEBUG ("we sent BYE already");
1934 result = GST_CLOCK_TIME_NONE;
1935 } else if (sess->stats.active_sources >= 50) {
1936 GST_DEBUG ("reconsider BYE, more than 50 sources");
1937 /* reconsider BYE if members >= 50 */
1938 result += calculate_rtcp_interval (sess, FALSE, TRUE);
1941 if (sess->first_rtcp) {
1942 GST_DEBUG ("first RTCP packet");
1943 /* we are called for the first time */
1944 result += calculate_rtcp_interval (sess, FALSE, TRUE);
1945 } else if (sess->next_rtcp_check_time < time) {
1946 GST_DEBUG ("old check time expired, getting new timeout");
1947 /* get a new timeout when we need to */
1948 result += calculate_rtcp_interval (sess, FALSE, FALSE);
1951 sess->next_rtcp_check_time = result;
1953 GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
1954 RTP_SESSION_UNLOCK (sess);
1965 GstClockTime interval;
1966 GstRTCPPacket packet;
1972 session_start_rtcp (RTPSession * sess, ReportData * data)
1974 GstRTCPPacket *packet = &data->packet;
1975 RTPSource *own = sess->source;
1977 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
1979 if (RTP_SOURCE_IS_SENDER (own)) {
1982 guint32 packet_count, octet_count;
1984 /* we are a sender, create SR */
1985 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
1986 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
1988 /* get latest stats */
1989 rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
1990 &packet_count, &octet_count);
1992 rtp_source_process_sr (own, data->ntpnstime, ntptime, rtptime, packet_count,
1995 /* fill in sender report info */
1996 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
1997 ntptime, rtptime, packet_count, octet_count);
1999 /* we are only receiver, create RR */
2000 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2001 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2002 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2006 /* construct a Sender or Receiver Report */
2008 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2010 RTPSession *sess = data->sess;
2011 GstRTCPPacket *packet = &data->packet;
2013 /* create a new buffer if needed */
2014 if (data->rtcp == NULL) {
2015 session_start_rtcp (sess, data);
2017 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2018 /* only report about other sender sources */
2019 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2020 guint8 fractionlost;
2022 guint32 exthighestseq, jitter;
2026 rtp_source_get_new_rb (source, data->time, &fractionlost, &packetslost,
2027 &exthighestseq, &jitter, &lsr, &dlsr);
2029 /* packet is not yet filled, add report block for this source. */
2030 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2031 exthighestseq, jitter, lsr, dlsr);
2036 /* perform cleanup of sources that timed out */
2038 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2040 gboolean remove = FALSE;
2041 gboolean byetimeout = FALSE;
2042 gboolean is_sender, is_active;
2043 RTPSession *sess = data->sess;
2044 GstClockTime interval;
2046 is_sender = RTP_SOURCE_IS_SENDER (source);
2047 is_active = RTP_SOURCE_IS_ACTIVE (source);
2049 /* check for our own source, we don't want to delete our own source. */
2050 if (!(source == sess->source)) {
2051 if (source->received_bye) {
2052 /* if we received a BYE from the source, remove the source after some
2054 if (data->time > source->bye_time &&
2055 data->time - source->bye_time > sess->stats.bye_timeout) {
2056 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2061 /* sources that were inactive for more than 5 times the deterministic reporting
2062 * interval get timed out. the min timeout is 5 seconds. */
2063 if (data->time > source->last_activity) {
2064 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2065 if (data->time - source->last_activity > interval) {
2066 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2067 source->ssrc, GST_TIME_ARGS (source->last_activity));
2073 /* senders that did not send for a long time become a receiver, this also
2074 * holds for our own source. */
2076 if (data->time > source->last_rtp_activity) {
2077 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2078 if (data->time - source->last_rtp_activity > interval) {
2079 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2080 GST_TIME_FORMAT, source->ssrc,
2081 GST_TIME_ARGS (source->last_rtp_activity));
2082 source->is_sender = FALSE;
2083 sess->stats.sender_sources--;
2089 sess->total_sources--;
2091 sess->stats.sender_sources--;
2093 sess->stats.active_sources--;
2096 on_bye_timeout (sess, source);
2098 on_timeout (sess, source);
2104 session_sdes (RTPSession * sess, ReportData * data)
2106 GstRTCPPacket *packet = &data->packet;
2110 /* add SDES packet */
2111 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2113 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2115 rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
2117 gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
2120 /* other SDES items must only be added at regular intervals and only when the
2121 * user requests to since it might be a privacy problem */
2123 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
2124 strlen (sess->name), (guint8 *) sess->name);
2125 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
2126 strlen (sess->tool), (guint8 *) sess->tool);
2129 data->has_sdes = TRUE;
2132 /* schedule a BYE packet */
2134 session_bye (RTPSession * sess, ReportData * data)
2136 GstRTCPPacket *packet = &data->packet;
2139 session_start_rtcp (sess, data);
2142 session_sdes (sess, data);
2144 /* add a BYE packet */
2145 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2146 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2147 if (sess->bye_reason)
2148 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2150 /* we have a BYE packet now */
2151 data->is_bye = TRUE;
2155 is_rtcp_time (RTPSession * sess, GstClockTime time, ReportData * data)
2157 GstClockTime new_send_time, elapsed;
2160 /* no need to check yet */
2161 if (sess->next_rtcp_check_time > time) {
2162 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2163 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2164 GST_TIME_ARGS (time));
2168 /* get elapsed time since we last reported */
2169 elapsed = time - sess->last_rtcp_send_time;
2171 /* perform forward reconsideration */
2172 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2174 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2175 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2177 new_send_time += sess->last_rtcp_send_time;
2179 /* check if reconsideration */
2180 if (time < new_send_time) {
2181 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2182 GST_TIME_ARGS (new_send_time));
2184 /* store new check time */
2185 sess->next_rtcp_check_time = new_send_time;
2188 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2190 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2191 GST_TIME_ARGS (new_send_time));
2192 sess->next_rtcp_check_time = time + new_send_time;
2198 * rtp_session_on_timeout:
2199 * @sess: an #RTPSession
2200 * @time: the current system time
2201 * @ntpnstime: the current NTP time in nanoseconds
2203 * Perform maintenance actions after the timeout obtained with
2204 * rtp_session_next_timeout() expired.
2206 * This function will perform timeouts of receivers and senders, send a BYE
2207 * packet or generate RTCP packets with current session stats.
2209 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2210 * times, for each packet that should be processed.
2212 * Returns: a #GstFlowReturn.
2215 rtp_session_on_timeout (RTPSession * sess, GstClockTime time, guint64 ntpnstime)
2217 GstFlowReturn result = GST_FLOW_OK;
2221 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2223 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2224 GST_TIME_ARGS (time), GST_TIME_ARGS (ntpnstime));
2229 data.ntpnstime = ntpnstime;
2230 data.is_bye = FALSE;
2231 data.has_sdes = FALSE;
2233 RTP_SESSION_LOCK (sess);
2234 /* get a new interval, we need this for various cleanups etc */
2235 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2237 /* first perform cleanups */
2238 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2239 (GHRFunc) session_cleanup, &data);
2241 /* see if we need to generate SR or RR packets */
2242 if (is_rtcp_time (sess, time, &data)) {
2243 if (sess->source->received_bye) {
2244 /* generate BYE instead */
2245 session_bye (sess, &data);
2246 sess->sent_bye = TRUE;
2248 /* loop over all known sources and do something */
2249 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2250 (GHFunc) session_report_blocks, &data);
2257 /* we keep track of the last report time in order to timeout inactive
2258 * receivers or senders */
2259 sess->last_rtcp_send_time = data.time;
2260 sess->first_rtcp = FALSE;
2262 /* add SDES for this source when not already added */
2264 session_sdes (sess, &data);
2266 /* update average RTCP size before sending */
2267 size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2268 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2271 /* check for outdated collisions */
2272 item = g_list_first (sess->conflicting_addresses);
2274 RTPConflictingAddress *known_conflict = item->data;
2275 GList *next_item = g_list_next (item);
2277 if (known_conflict->time < time - (data.interval *
2278 RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2279 sess->conflicting_addresses =
2280 g_list_delete_link (sess->conflicting_addresses, item);
2281 g_free (known_conflict);
2286 if (sess->change_ssrc) {
2287 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2288 GINT_TO_POINTER (sess->source->ssrc));
2290 sess->source->ssrc = rtp_session_create_new_ssrc (sess);
2291 rtp_source_reset (sess->source);
2293 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2294 GINT_TO_POINTER (sess->source->ssrc), sess->source);
2296 g_free (sess->bye_reason);
2297 sess->bye_reason = NULL;
2298 sess->sent_bye = FALSE;
2299 sess->change_ssrc = FALSE;
2301 RTP_SESSION_UNLOCK (sess);
2303 /* push out the RTCP packet */
2305 /* close the RTCP packet */
2306 gst_rtcp_buffer_end (data.rtcp);
2308 if (sess->callbacks.send_rtcp)
2309 result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
2310 sess->send_rtcp_user_data);
2312 gst_buffer_unref (data.rtcp);