2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
22 #include <gst/rtp/gstrtpbuffer.h>
23 #include <gst/rtp/gstrtcpbuffer.h>
24 #include <gst/netbuffer/gstnetbuffer.h>
26 #include "gstrtpbin-marshal.h"
27 #include "rtpsession.h"
29 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
30 #define GST_CAT_DEFAULT rtp_session_debug
32 /* signals and args */
35 SIGNAL_GET_SOURCE_BY_SSRC,
37 SIGNAL_ON_SSRC_COLLISION,
38 SIGNAL_ON_SSRC_VALIDATED,
39 SIGNAL_ON_SSRC_ACTIVE,
42 SIGNAL_ON_BYE_TIMEOUT,
44 SIGNAL_ON_SENDER_TIMEOUT,
48 #define DEFAULT_INTERNAL_SOURCE NULL
49 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
50 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
51 #define DEFAULT_RTCP_MTU 1400
52 #define DEFAULT_SDES_CNAME NULL
53 #define DEFAULT_SDES_NAME NULL
54 #define DEFAULT_SDES_EMAIL NULL
55 #define DEFAULT_SDES_PHONE NULL
56 #define DEFAULT_SDES_LOCATION NULL
57 #define DEFAULT_SDES_TOOL NULL
58 #define DEFAULT_SDES_NOTE NULL
59 #define DEFAULT_NUM_SOURCES 0
60 #define DEFAULT_NUM_ACTIVE_SOURCES 0
61 #define DEFAULT_SOURCES NULL
79 PROP_NUM_ACTIVE_SOURCES,
84 /* update average packet size, we keep this scaled by 16 to keep enough
86 #define UPDATE_AVG(avg, val) \
90 (avg) = ((val) + (15 * (avg))) >> 4;
92 /* The number RTCP intervals after which to timeout entries in the
95 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
97 /* GObject vmethods */
98 static void rtp_session_finalize (GObject * object);
99 static void rtp_session_set_property (GObject * object, guint prop_id,
100 const GValue * value, GParamSpec * pspec);
101 static void rtp_session_get_property (GObject * object, guint prop_id,
102 GValue * value, GParamSpec * pspec);
104 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
106 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
108 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
109 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
110 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
111 const gchar * reason, GstClockTime current_time);
112 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
113 gboolean deterministic, gboolean first);
116 rtp_session_class_init (RTPSessionClass * klass)
118 GObjectClass *gobject_class;
120 gobject_class = (GObjectClass *) klass;
122 gobject_class->finalize = rtp_session_finalize;
123 gobject_class->set_property = rtp_session_set_property;
124 gobject_class->get_property = rtp_session_get_property;
127 * RTPSession::get-source-by-ssrc:
128 * @session: the object which received the signal
129 * @ssrc: the SSRC of the RTPSource
131 * Request the #RTPSource object with SSRC @ssrc in @session.
133 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
134 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
135 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
136 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
137 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
140 * RTPSession::on-new-ssrc:
141 * @session: the object which received the signal
142 * @src: the new RTPSource
144 * Notify of a new SSRC that entered @session.
146 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
147 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
148 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
149 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
152 * RTPSession::on-ssrc-collision:
153 * @session: the object which received the signal
154 * @src: the #RTPSource that caused a collision
156 * Notify when we have an SSRC collision
158 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
159 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
160 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
161 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
164 * RTPSession::on-ssrc-validated:
165 * @session: the object which received the signal
166 * @src: the new validated RTPSource
168 * Notify of a new SSRC that became validated.
170 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
171 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
172 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
173 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
176 * RTPSession::on-ssrc-active:
177 * @session: the object which received the signal
178 * @src: the active RTPSource
180 * Notify of a SSRC that is active, i.e., sending RTCP.
182 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
183 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
184 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
185 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
188 * RTPSession::on-ssrc-sdes:
189 * @session: the object which received the signal
190 * @src: the RTPSource
192 * Notify that a new SDES was received for SSRC.
194 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
195 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
196 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
197 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
200 * RTPSession::on-bye-ssrc:
201 * @session: the object which received the signal
202 * @src: the RTPSource that went away
204 * Notify of an SSRC that became inactive because of a BYE packet.
206 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
207 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
208 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
209 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
212 * RTPSession::on-bye-timeout:
213 * @session: the object which received the signal
214 * @src: the RTPSource that timed out
216 * Notify of an SSRC that has timed out because of BYE
218 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
219 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
220 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
221 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
224 * RTPSession::on-timeout:
225 * @session: the object which received the signal
226 * @src: the RTPSource that timed out
228 * Notify of an SSRC that has timed out
230 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
231 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
232 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
233 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
236 * RTPSession::on-sender-timeout:
237 * @session: the object which received the signal
238 * @src: the RTPSource that timed out
240 * Notify of an SSRC that was a sender but timed out and became a receiver.
242 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
243 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
244 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
245 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
248 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
249 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
250 "The internal SSRC used for the session",
251 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
253 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
254 g_param_spec_object ("internal-source", "Internal Source",
255 "The internal source element of the session",
256 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
258 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
259 g_param_spec_double ("bandwidth", "Bandwidth",
260 "The bandwidth of the session",
261 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
262 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
264 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
265 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
266 "The fraction of the bandwidth used for RTCP",
267 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
268 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
270 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
271 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
272 "The maximum size of the RTCP packets",
273 16, G_MAXINT16, DEFAULT_RTCP_MTU,
274 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
276 g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
277 g_param_spec_string ("sdes-cname", "SDES CNAME",
278 "The CNAME to put in SDES messages of this session",
279 DEFAULT_SDES_CNAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
281 g_object_class_install_property (gobject_class, PROP_SDES_NAME,
282 g_param_spec_string ("sdes-name", "SDES NAME",
283 "The NAME to put in SDES messages of this session",
284 DEFAULT_SDES_NAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
286 g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
287 g_param_spec_string ("sdes-email", "SDES EMAIL",
288 "The EMAIL to put in SDES messages of this session",
289 DEFAULT_SDES_EMAIL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
291 g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
292 g_param_spec_string ("sdes-phone", "SDES PHONE",
293 "The PHONE to put in SDES messages of this session",
294 DEFAULT_SDES_PHONE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
296 g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
297 g_param_spec_string ("sdes-location", "SDES LOCATION",
298 "The LOCATION to put in SDES messages of this session",
299 DEFAULT_SDES_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
301 g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
302 g_param_spec_string ("sdes-tool", "SDES TOOL",
303 "The TOOL to put in SDES messages of this session",
304 DEFAULT_SDES_TOOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
306 g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
307 g_param_spec_string ("sdes-note", "SDES NOTE",
308 "The NOTE to put in SDES messages of this session",
309 DEFAULT_SDES_NOTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
311 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
312 g_param_spec_uint ("num-sources", "Num Sources",
313 "The number of sources in the session", 0, G_MAXUINT,
314 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
316 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
317 g_param_spec_uint ("num-active-sources", "Num Active Sources",
318 "The number of active sources in the session", 0, G_MAXUINT,
319 DEFAULT_NUM_ACTIVE_SOURCES,
320 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
324 * Get a GValue Array of all sources in the session.
327 * <title>Getting the #RTPSources of a session
334 * g_object_get (sess, "sources", &arr, NULL);
336 * for (i = 0; i < arr->n_values; i++) {
339 * val = g_value_array_get_nth (arr, i);
340 * source = g_value_get_object (val);
342 * g_value_array_free (arr);
347 g_object_class_install_property (gobject_class, PROP_SOURCES,
348 g_param_spec_boxed ("sources", "Sources",
349 "An array of all known sources in the session",
350 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
352 klass->get_source_by_ssrc =
353 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
355 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
359 rtp_session_init (RTPSession * sess)
364 sess->lock = g_mutex_new ();
365 sess->key = g_random_int ();
369 for (i = 0; i < 32; i++) {
371 g_hash_table_new_full (NULL, NULL, NULL,
372 (GDestroyNotify) g_object_unref);
374 sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
376 rtp_stats_init_defaults (&sess->stats);
378 /* create an active SSRC for this session manager */
379 sess->source = rtp_session_create_source (sess);
380 sess->source->validated = TRUE;
381 sess->source->internal = TRUE;
382 sess->stats.active_sources++;
384 /* default UDP header length */
385 sess->header_len = 28;
386 sess->mtu = DEFAULT_RTCP_MTU;
388 /* some default SDES entries */
389 str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
390 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
393 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
395 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
397 sess->first_rtcp = TRUE;
399 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
403 rtp_session_finalize (GObject * object)
408 sess = RTP_SESSION_CAST (object);
410 g_mutex_free (sess->lock);
411 for (i = 0; i < 32; i++)
412 g_hash_table_destroy (sess->ssrcs[i]);
414 g_free (sess->bye_reason);
416 g_hash_table_destroy (sess->cnames);
417 g_object_unref (sess->source);
419 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
423 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
425 GValue value = { 0 };
427 g_value_init (&value, RTP_TYPE_SOURCE);
428 g_value_take_object (&value, source);
429 g_value_array_append (arr, &value);
433 rtp_session_create_sources (RTPSession * sess)
438 RTP_SESSION_LOCK (sess);
439 /* get number of elements in the table */
440 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
441 /* create the result value array */
442 res = g_value_array_new (size);
444 /* and copy all values into the array */
445 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
446 RTP_SESSION_UNLOCK (sess);
452 rtp_session_set_property (GObject * object, guint prop_id,
453 const GValue * value, GParamSpec * pspec)
457 sess = RTP_SESSION (object);
460 case PROP_INTERNAL_SSRC:
461 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
464 rtp_session_set_bandwidth (sess, g_value_get_double (value));
466 case PROP_RTCP_FRACTION:
467 rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
470 sess->mtu = g_value_get_uint (value);
472 case PROP_SDES_CNAME:
473 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_CNAME,
474 g_value_get_string (value));
477 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NAME,
478 g_value_get_string (value));
480 case PROP_SDES_EMAIL:
481 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_EMAIL,
482 g_value_get_string (value));
484 case PROP_SDES_PHONE:
485 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_PHONE,
486 g_value_get_string (value));
488 case PROP_SDES_LOCATION:
489 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_LOC,
490 g_value_get_string (value));
493 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_TOOL,
494 g_value_get_string (value));
497 rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NOTE,
498 g_value_get_string (value));
501 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
507 rtp_session_get_property (GObject * object, guint prop_id,
508 GValue * value, GParamSpec * pspec)
512 sess = RTP_SESSION (object);
515 case PROP_INTERNAL_SSRC:
516 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
518 case PROP_INTERNAL_SOURCE:
519 g_value_take_object (value, rtp_session_get_internal_source (sess));
522 g_value_set_double (value, rtp_session_get_bandwidth (sess));
524 case PROP_RTCP_FRACTION:
525 g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
528 g_value_set_uint (value, sess->mtu);
530 case PROP_SDES_CNAME:
531 g_value_take_string (value, rtp_session_get_sdes_string (sess,
532 GST_RTCP_SDES_CNAME));
535 g_value_take_string (value, rtp_session_get_sdes_string (sess,
536 GST_RTCP_SDES_NAME));
538 case PROP_SDES_EMAIL:
539 g_value_take_string (value, rtp_session_get_sdes_string (sess,
540 GST_RTCP_SDES_EMAIL));
542 case PROP_SDES_PHONE:
543 g_value_take_string (value, rtp_session_get_sdes_string (sess,
544 GST_RTCP_SDES_PHONE));
546 case PROP_SDES_LOCATION:
547 g_value_take_string (value, rtp_session_get_sdes_string (sess,
551 g_value_take_string (value, rtp_session_get_sdes_string (sess,
552 GST_RTCP_SDES_TOOL));
555 g_value_take_string (value, rtp_session_get_sdes_string (sess,
556 GST_RTCP_SDES_NOTE));
558 case PROP_NUM_SOURCES:
559 g_value_set_uint (value, rtp_session_get_num_sources (sess));
561 case PROP_NUM_ACTIVE_SOURCES:
562 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
565 g_value_take_boxed (value, rtp_session_create_sources (sess));
568 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
574 on_new_ssrc (RTPSession * sess, RTPSource * source)
576 g_object_ref (source);
577 RTP_SESSION_UNLOCK (sess);
578 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
579 RTP_SESSION_LOCK (sess);
580 g_object_unref (source);
584 on_ssrc_collision (RTPSession * sess, RTPSource * source)
586 g_object_ref (source);
587 RTP_SESSION_UNLOCK (sess);
588 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
590 RTP_SESSION_LOCK (sess);
591 g_object_unref (source);
595 on_ssrc_validated (RTPSession * sess, RTPSource * source)
597 g_object_ref (source);
598 RTP_SESSION_UNLOCK (sess);
599 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
601 RTP_SESSION_LOCK (sess);
602 g_object_unref (source);
606 on_ssrc_active (RTPSession * sess, RTPSource * source)
608 g_object_ref (source);
609 RTP_SESSION_UNLOCK (sess);
610 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
611 RTP_SESSION_LOCK (sess);
612 g_object_unref (source);
616 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
618 g_object_ref (source);
619 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
620 RTP_SESSION_UNLOCK (sess);
621 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
622 RTP_SESSION_LOCK (sess);
623 g_object_unref (source);
627 on_bye_ssrc (RTPSession * sess, RTPSource * source)
629 g_object_ref (source);
630 RTP_SESSION_UNLOCK (sess);
631 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
632 RTP_SESSION_LOCK (sess);
633 g_object_unref (source);
637 on_bye_timeout (RTPSession * sess, RTPSource * source)
639 g_object_ref (source);
640 RTP_SESSION_UNLOCK (sess);
641 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
642 RTP_SESSION_LOCK (sess);
643 g_object_unref (source);
647 on_timeout (RTPSession * sess, RTPSource * source)
649 g_object_ref (source);
650 RTP_SESSION_UNLOCK (sess);
651 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
652 RTP_SESSION_LOCK (sess);
653 g_object_unref (source);
657 on_sender_timeout (RTPSession * sess, RTPSource * source)
659 g_object_ref (source);
660 RTP_SESSION_UNLOCK (sess);
661 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
663 RTP_SESSION_LOCK (sess);
664 g_object_unref (source);
670 * Create a new session object.
672 * Returns: a new #RTPSession. g_object_unref() after usage.
675 rtp_session_new (void)
679 sess = g_object_new (RTP_TYPE_SESSION, NULL);
685 * rtp_session_set_callbacks:
686 * @sess: an #RTPSession
687 * @callbacks: callbacks to configure
688 * @user_data: user data passed in the callbacks
690 * Configure a set of callbacks to be notified of actions.
693 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
696 g_return_if_fail (RTP_IS_SESSION (sess));
698 if (callbacks->process_rtp) {
699 sess->callbacks.process_rtp = callbacks->process_rtp;
700 sess->process_rtp_user_data = user_data;
702 if (callbacks->send_rtp) {
703 sess->callbacks.send_rtp = callbacks->send_rtp;
704 sess->send_rtp_user_data = user_data;
706 if (callbacks->send_rtcp) {
707 sess->callbacks.send_rtcp = callbacks->send_rtcp;
708 sess->send_rtcp_user_data = user_data;
710 if (callbacks->sync_rtcp) {
711 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
712 sess->sync_rtcp_user_data = user_data;
714 if (callbacks->clock_rate) {
715 sess->callbacks.clock_rate = callbacks->clock_rate;
716 sess->clock_rate_user_data = user_data;
718 if (callbacks->reconsider) {
719 sess->callbacks.reconsider = callbacks->reconsider;
720 sess->reconsider_user_data = user_data;
725 * rtp_session_set_process_rtp_callback:
726 * @sess: an #RTPSession
727 * @callback: callback to set
728 * @user_data: user data passed in the callback
730 * Configure only the process_rtp callback to be notified of the process_rtp action.
733 rtp_session_set_process_rtp_callback (RTPSession * sess,
734 RTPSessionProcessRTP callback, gpointer user_data)
736 g_return_if_fail (RTP_IS_SESSION (sess));
738 sess->callbacks.process_rtp = callback;
739 sess->process_rtp_user_data = user_data;
743 * rtp_session_set_send_rtp_callback:
744 * @sess: an #RTPSession
745 * @callback: callback to set
746 * @user_data: user data passed in the callback
748 * Configure only the send_rtp callback to be notified of the send_rtp action.
751 rtp_session_set_send_rtp_callback (RTPSession * sess,
752 RTPSessionSendRTP callback, gpointer user_data)
754 g_return_if_fail (RTP_IS_SESSION (sess));
756 sess->callbacks.send_rtp = callback;
757 sess->send_rtp_user_data = user_data;
761 * rtp_session_set_send_rtcp_callback:
762 * @sess: an #RTPSession
763 * @callback: callback to set
764 * @user_data: user data passed in the callback
766 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
769 rtp_session_set_send_rtcp_callback (RTPSession * sess,
770 RTPSessionSendRTCP callback, gpointer user_data)
772 g_return_if_fail (RTP_IS_SESSION (sess));
774 sess->callbacks.send_rtcp = callback;
775 sess->send_rtcp_user_data = user_data;
779 * rtp_session_set_sync_rtcp_callback:
780 * @sess: an #RTPSession
781 * @callback: callback to set
782 * @user_data: user data passed in the callback
784 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
787 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
788 RTPSessionSyncRTCP callback, gpointer user_data)
790 g_return_if_fail (RTP_IS_SESSION (sess));
792 sess->callbacks.sync_rtcp = callback;
793 sess->sync_rtcp_user_data = user_data;
797 * rtp_session_set_clock_rate_callback:
798 * @sess: an #RTPSession
799 * @callback: callback to set
800 * @user_data: user data passed in the callback
802 * Configure only the clock_rate callback to be notified of the clock_rate action.
805 rtp_session_set_clock_rate_callback (RTPSession * sess,
806 RTPSessionClockRate callback, gpointer user_data)
808 g_return_if_fail (RTP_IS_SESSION (sess));
810 sess->callbacks.clock_rate = callback;
811 sess->clock_rate_user_data = user_data;
815 * rtp_session_set_reconsider_callback:
816 * @sess: an #RTPSession
817 * @callback: callback to set
818 * @user_data: user data passed in the callback
820 * Configure only the reconsider callback to be notified of the reconsider action.
823 rtp_session_set_reconsider_callback (RTPSession * sess,
824 RTPSessionReconsider callback, gpointer user_data)
826 g_return_if_fail (RTP_IS_SESSION (sess));
828 sess->callbacks.reconsider = callback;
829 sess->reconsider_user_data = user_data;
833 * rtp_session_set_bandwidth:
834 * @sess: an #RTPSession
835 * @bandwidth: the bandwidth allocated
837 * Set the session bandwidth in bytes per second.
840 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
842 g_return_if_fail (RTP_IS_SESSION (sess));
844 RTP_SESSION_LOCK (sess);
845 sess->stats.bandwidth = bandwidth;
846 RTP_SESSION_UNLOCK (sess);
850 * rtp_session_get_bandwidth:
851 * @sess: an #RTPSession
853 * Get the session bandwidth.
855 * Returns: the session bandwidth.
858 rtp_session_get_bandwidth (RTPSession * sess)
862 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
864 RTP_SESSION_LOCK (sess);
865 result = sess->stats.bandwidth;
866 RTP_SESSION_UNLOCK (sess);
872 * rtp_session_set_rtcp_fraction:
873 * @sess: an #RTPSession
874 * @bandwidth: the RTCP bandwidth
876 * Set the bandwidth that should be used for RTCP
880 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
882 g_return_if_fail (RTP_IS_SESSION (sess));
884 RTP_SESSION_LOCK (sess);
885 sess->stats.rtcp_bandwidth = bandwidth;
886 RTP_SESSION_UNLOCK (sess);
890 * rtp_session_get_rtcp_fraction:
891 * @sess: an #RTPSession
893 * Get the session bandwidth used for RTCP.
895 * Returns: The bandwidth used for RTCP messages.
898 rtp_session_get_rtcp_fraction (RTPSession * sess)
902 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
904 RTP_SESSION_LOCK (sess);
905 result = sess->stats.rtcp_bandwidth;
906 RTP_SESSION_UNLOCK (sess);
912 * rtp_session_set_sdes_string:
913 * @sess: an #RTPSession
914 * @type: the type of the SDES item
915 * @item: a null-terminated string to set.
917 * Store an SDES item of @type in @sess.
919 * Returns: %FALSE if the data was unchanged @type is invalid.
922 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
927 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
929 RTP_SESSION_LOCK (sess);
930 result = rtp_source_set_sdes_string (sess->source, type, item);
931 RTP_SESSION_UNLOCK (sess);
937 * rtp_session_get_sdes_string:
938 * @sess: an #RTPSession
939 * @type: the type of the SDES item
941 * Get the SDES item of @type from @sess.
943 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
944 * valid. g_free() after usage.
947 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
951 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
953 RTP_SESSION_LOCK (sess);
954 result = rtp_source_get_sdes_string (sess->source, type);
955 RTP_SESSION_UNLOCK (sess);
961 source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
963 GstFlowReturn result = GST_FLOW_OK;
965 if (source == session->source) {
966 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
968 RTP_SESSION_UNLOCK (session);
970 if (session->callbacks.send_rtp)
972 session->callbacks.send_rtp (session, source, buffer,
973 session->send_rtp_user_data);
975 gst_buffer_unref (buffer);
978 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
979 RTP_SESSION_UNLOCK (session);
981 if (session->callbacks.process_rtp)
983 session->callbacks.process_rtp (session, source, buffer,
984 session->process_rtp_user_data);
986 gst_buffer_unref (buffer);
988 RTP_SESSION_LOCK (session);
994 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
998 RTP_SESSION_UNLOCK (session);
1000 if (session->callbacks.clock_rate)
1002 session->callbacks.clock_rate (session, pt,
1003 session->clock_rate_user_data);
1007 RTP_SESSION_LOCK (session);
1009 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1014 static RTPSourceCallbacks callbacks = {
1015 (RTPSourcePushRTP) source_push_rtp,
1016 (RTPSourceClockRate) source_clock_rate,
1020 * find_add_conflicting_addresses:
1021 * @sess: The session to check in
1022 * @arrival: The arrival stats for the buffer
1024 * Checks if an address which has a conflict is already known,
1025 * otherwise remembers it to prevent loops.
1027 * Returns: TRUE if it was a known conflict, FALSE otherwise
1031 find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
1034 RTPConflictingAddress *new_conflict;
1036 for (item = g_list_first (sess->conflicting_addresses);
1037 item; item = g_list_next (item)) {
1038 RTPConflictingAddress *known_conflict = item->data;
1040 if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
1041 known_conflict->time = arrival->time;
1046 new_conflict = g_new0 (RTPConflictingAddress, 1);
1048 memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
1049 new_conflict->time = arrival->time;
1051 sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
1058 check_collision (RTPSession * sess, RTPSource * source,
1059 RTPArrivalStats * arrival, gboolean rtp)
1061 /* If we have no arrival address, we can't do collision checking */
1062 if (!arrival->have_address)
1065 if (sess->source != source) {
1066 /* This is not our local source, but lets check if two remote
1070 if (source->have_rtp_from) {
1071 if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
1072 /* Address is the same */
1075 /* We don't already have a from address for RTP, just set it */
1076 rtp_source_set_rtp_from (source, &arrival->address);
1080 if (source->have_rtcp_from) {
1081 if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
1082 /* Address is the same */
1085 /* We don't already have a from address for RTCP, just set it */
1086 rtp_source_set_rtcp_from (source, &arrival->address);
1090 /* We received RTP or RTCP from this source before but the network address
1091 * changed. In this case, we have third-party collision or loop */
1092 GST_DEBUG ("we have a third-party collision or loop");
1094 /* FIXME: Log 3rd party collision somehow
1095 * Maybe should be done in upper layer, only the SDES can tell us
1096 * if its a collision or a loop
1099 /* This is sending with our ssrc, is it an address we already know */
1101 if (find_add_conflicting_addresses (sess, arrival)) {
1102 /* Its a known conflict, its probably a loop, not a collision
1103 * lets just drop the incoming packet
1105 GST_DEBUG ("Our packets are being looped back to us, dropping");
1107 /* Its a new collision, lets change our SSRC */
1109 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1110 on_ssrc_collision (sess, source);
1112 rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->time);
1114 sess->change_ssrc = TRUE;
1122 /* must be called with the session lock, the returned source needs to be
1123 * unreffed after usage. */
1125 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1126 RTPArrivalStats * arrival, gboolean rtp)
1131 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1132 if (source == NULL) {
1133 /* make new Source in probation and insert */
1134 source = rtp_source_new (ssrc);
1136 /* for RTP packets we need to set the source in probation. Receiving RTCP
1137 * packets of an SSRC, on the other hand, is a strong indication that we
1138 * are dealing with a valid source. */
1140 source->probation = RTP_DEFAULT_PROBATION;
1142 source->probation = 0;
1144 /* store from address, if any */
1145 if (arrival->have_address) {
1147 rtp_source_set_rtp_from (source, &arrival->address);
1149 rtp_source_set_rtcp_from (source, &arrival->address);
1152 /* configure a callback on the source */
1153 rtp_source_set_callbacks (source, &callbacks, sess);
1155 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1158 /* we have one more source now */
1159 sess->total_sources++;
1163 /* check for collision, this updates the address when not previously set */
1164 if (check_collision (sess, source, arrival, rtp)) {
1168 /* update last activity */
1169 source->last_activity = arrival->time;
1171 source->last_rtp_activity = arrival->time;
1172 g_object_ref (source);
1178 * rtp_session_get_internal_source:
1179 * @sess: a #RTPSession
1181 * Get the internal #RTPSource of @sess.
1183 * Returns: The internal #RTPSource. g_object_unref() after usage.
1186 rtp_session_get_internal_source (RTPSession * sess)
1190 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1192 result = g_object_ref (sess->source);
1198 * rtp_session_set_internal_ssrc:
1199 * @sess: a #RTPSession
1202 * Set the SSRC of @sess to @ssrc.
1205 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1207 RTP_SESSION_LOCK (sess);
1208 if (ssrc != sess->source->ssrc) {
1209 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1210 GINT_TO_POINTER (sess->source->ssrc));
1212 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1213 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1214 * packets will timeout on the old SSRC, we could potentially schedule a
1215 * BYE RTCP for the old SSRC... */
1216 sess->source->ssrc = ssrc;
1217 rtp_source_reset (sess->source);
1219 /* rehash with the new SSRC */
1220 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1221 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1223 RTP_SESSION_UNLOCK (sess);
1225 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1229 * rtp_session_get_internal_ssrc:
1230 * @sess: a #RTPSession
1232 * Get the internal SSRC of @sess.
1234 * Returns: The SSRC of the session.
1237 rtp_session_get_internal_ssrc (RTPSession * sess)
1241 RTP_SESSION_LOCK (sess);
1242 ssrc = sess->source->ssrc;
1243 RTP_SESSION_UNLOCK (sess);
1249 * rtp_session_add_source:
1250 * @sess: a #RTPSession
1251 * @src: #RTPSource to add
1253 * Add @src to @session.
1255 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1256 * existed in the session.
1259 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1261 gboolean result = FALSE;
1264 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1265 g_return_val_if_fail (src != NULL, FALSE);
1267 RTP_SESSION_LOCK (sess);
1269 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1270 GINT_TO_POINTER (src->ssrc));
1272 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1273 GINT_TO_POINTER (src->ssrc), src);
1274 /* we have one more source now */
1275 sess->total_sources++;
1278 RTP_SESSION_UNLOCK (sess);
1284 * rtp_session_get_num_sources:
1285 * @sess: an #RTPSession
1287 * Get the number of sources in @sess.
1289 * Returns: The number of sources in @sess.
1292 rtp_session_get_num_sources (RTPSession * sess)
1296 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1298 RTP_SESSION_LOCK (sess);
1299 result = sess->total_sources;
1300 RTP_SESSION_UNLOCK (sess);
1306 * rtp_session_get_num_active_sources:
1307 * @sess: an #RTPSession
1309 * Get the number of active sources in @sess. A source is considered active when
1310 * it has been validated and has not yet received a BYE RTCP message.
1312 * Returns: The number of active sources in @sess.
1315 rtp_session_get_num_active_sources (RTPSession * sess)
1319 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1321 RTP_SESSION_LOCK (sess);
1322 result = sess->stats.active_sources;
1323 RTP_SESSION_UNLOCK (sess);
1329 * rtp_session_get_source_by_ssrc:
1330 * @sess: an #RTPSession
1333 * Find the source with @ssrc in @sess.
1335 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1336 * g_object_unref() after usage.
1339 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1343 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1345 RTP_SESSION_LOCK (sess);
1347 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1349 g_object_ref (result);
1350 RTP_SESSION_UNLOCK (sess);
1356 * rtp_session_get_source_by_cname:
1357 * @sess: a #RTPSession
1360 * Find the source with @cname in @sess.
1362 * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1363 * g_object_unref() after usage.
1366 rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1370 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1371 g_return_val_if_fail (cname != NULL, NULL);
1373 RTP_SESSION_LOCK (sess);
1374 result = g_hash_table_lookup (sess->cnames, cname);
1376 g_object_ref (result);
1377 RTP_SESSION_UNLOCK (sess);
1383 rtp_session_create_new_ssrc (RTPSession * sess)
1388 ssrc = g_random_int ();
1390 /* see if it exists in the session, we're done if it doesn't */
1391 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1392 GINT_TO_POINTER (ssrc)) == NULL)
1401 * rtp_session_create_source:
1402 * @sess: an #RTPSession
1404 * Create an #RTPSource for use in @sess. This function will create a source
1405 * with an ssrc that is currently not used by any participants in the session.
1407 * Returns: an #RTPSource.
1410 rtp_session_create_source (RTPSession * sess)
1415 RTP_SESSION_LOCK (sess);
1416 ssrc = rtp_session_create_new_ssrc (sess);
1417 source = rtp_source_new (ssrc);
1418 rtp_source_set_callbacks (source, &callbacks, sess);
1419 /* we need an additional ref for the source in the hashtable */
1420 g_object_ref (source);
1421 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1423 /* we have one more source now */
1424 sess->total_sources++;
1425 RTP_SESSION_UNLOCK (sess);
1430 /* update the RTPArrivalStats structure with the current time and other bits
1431 * about the current buffer we are handling.
1432 * This function is typically called when a validated packet is received.
1433 * This function should be called with the SESSION_LOCK
1436 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1437 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1438 GstClockTime running_time, guint64 ntpnstime)
1440 /* get time of arrival */
1441 arrival->time = current_time;
1442 arrival->running_time = running_time;
1443 arrival->ntpnstime = ntpnstime;
1445 /* get packet size including header overhead */
1446 arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1449 arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1451 arrival->payload_len = 0;
1454 /* for netbuffer we can store the IP address to check for collisions */
1455 arrival->have_address = GST_IS_NETBUFFER (buffer);
1456 if (arrival->have_address) {
1457 GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1459 memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1464 * rtp_session_process_rtp:
1465 * @sess: and #RTPSession
1466 * @buffer: an RTP buffer
1467 * @current_time: the current system time
1468 * @ntpnstime: the NTP arrival time in nanoseconds
1470 * Process an RTP buffer in the session manager. This function takes ownership
1473 * Returns: a #GstFlowReturn.
1476 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1477 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1479 GstFlowReturn result;
1483 gboolean prevsender, prevactive;
1484 RTPArrivalStats arrival;
1486 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1487 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1489 if (!gst_rtp_buffer_validate (buffer))
1490 goto invalid_packet;
1492 RTP_SESSION_LOCK (sess);
1493 /* update arrival stats */
1494 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1495 running_time, ntpnstime);
1497 /* ignore more RTP packets when we left the session */
1498 if (sess->source->received_bye)
1501 /* get SSRC and look up in session database */
1502 ssrc = gst_rtp_buffer_get_ssrc (buffer);
1503 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1507 prevsender = RTP_SOURCE_IS_SENDER (source);
1508 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1510 /* we need to ref so that we can process the CSRCs later */
1511 gst_buffer_ref (buffer);
1513 /* let source process the packet */
1514 result = rtp_source_process_rtp (source, buffer, &arrival);
1516 /* source became active */
1517 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1518 sess->stats.active_sources++;
1519 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1520 sess->stats.active_sources);
1521 on_ssrc_validated (sess, source);
1523 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1524 sess->stats.sender_sources++;
1525 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1526 sess->stats.sender_sources);
1530 on_new_ssrc (sess, source);
1532 if (source->validated) {
1536 /* for validated sources, we add the CSRCs as well */
1537 count = gst_rtp_buffer_get_csrc_count (buffer);
1539 for (i = 0; i < count; i++) {
1541 RTPSource *csrc_src;
1543 csrc = gst_rtp_buffer_get_csrc (buffer, i);
1546 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1551 GST_DEBUG ("created new CSRC: %08x", csrc);
1552 rtp_source_set_as_csrc (csrc_src);
1553 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1554 sess->stats.active_sources++;
1555 on_new_ssrc (sess, csrc_src);
1557 g_object_unref (csrc_src);
1560 g_object_unref (source);
1561 gst_buffer_unref (buffer);
1563 RTP_SESSION_UNLOCK (sess);
1570 gst_buffer_unref (buffer);
1571 GST_DEBUG ("invalid RTP packet received");
1576 gst_buffer_unref (buffer);
1577 RTP_SESSION_UNLOCK (sess);
1578 GST_DEBUG ("ignoring RTP packet because we are leaving");
1583 gst_buffer_unref (buffer);
1584 RTP_SESSION_UNLOCK (sess);
1585 GST_DEBUG ("ignoring packet because its collisioning");
1591 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1592 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1596 count = gst_rtcp_packet_get_rb_count (packet);
1597 for (i = 0; i < count; i++) {
1598 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1599 guint8 fractionlost;
1602 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1603 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1605 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1607 if (ssrc == sess->source->ssrc) {
1608 /* only deal with report blocks for our session, we update the stats of
1609 * the sender of the RTCP message. We could also compare our stats against
1610 * the other sender to see if we are better or worse. */
1611 rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1612 exthighestseq, jitter, lsr, dlsr);
1614 on_ssrc_active (sess, source);
1619 /* A Sender report contains statistics about how the sender is doing. This
1620 * includes timing informataion such as the relation between RTP and NTP
1621 * timestamps and the number of packets/bytes it sent to us.
1623 * In this report is also included a set of report blocks related to how this
1624 * sender is receiving data (in case we (or somebody else) is also sending stuff
1625 * to it). This info includes the packet loss, jitter and seqnum. It also
1626 * contains information to calculate the round trip time (LSR/DLSR).
1629 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1630 RTPArrivalStats * arrival)
1632 guint32 senderssrc, rtptime, packet_count, octet_count;
1635 gboolean created, prevsender;
1637 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1638 &packet_count, &octet_count);
1640 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1641 senderssrc, GST_TIME_ARGS (arrival->time));
1643 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1647 prevsender = RTP_SOURCE_IS_SENDER (source);
1649 /* first update the source */
1650 rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1653 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1654 sess->stats.sender_sources++;
1655 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1656 sess->stats.sender_sources);
1660 on_new_ssrc (sess, source);
1662 rtp_session_process_rb (sess, source, packet, arrival);
1663 g_object_unref (source);
1666 /* A receiver report contains statistics about how a receiver is doing. It
1667 * includes stuff like packet loss, jitter and the seqnum it received last. It
1668 * also contains info to calculate the round trip time.
1670 * We are only interested in how the sender of this report is doing wrt to us.
1673 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1674 RTPArrivalStats * arrival)
1680 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1682 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1684 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1689 on_new_ssrc (sess, source);
1691 rtp_session_process_rb (sess, source, packet, arrival);
1692 g_object_unref (source);
1695 /* Get SDES items and store them in the SSRC */
1697 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1698 RTPArrivalStats * arrival)
1701 gboolean more_items, more_entries;
1703 items = gst_rtcp_packet_sdes_get_item_count (packet);
1704 GST_DEBUG ("got SDES packet with %d items", items);
1706 more_items = gst_rtcp_packet_sdes_first_item (packet);
1708 while (more_items) {
1710 gboolean changed, created;
1713 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1715 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1719 /* find src, no probation when dealing with RTCP */
1720 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1724 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1726 while (more_entries) {
1727 GstRTCPSDESType type;
1731 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1733 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1736 changed |= rtp_source_set_sdes (source, type, data, len);
1738 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1742 source->validated = TRUE;
1745 on_new_ssrc (sess, source);
1747 on_ssrc_sdes (sess, source);
1749 g_object_unref (source);
1751 more_items = gst_rtcp_packet_sdes_next_item (packet);
1756 /* BYE is sent when a client leaves the session
1759 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1760 RTPArrivalStats * arrival)
1765 reason = gst_rtcp_packet_bye_get_reason (packet);
1766 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1768 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1769 for (i = 0; i < count; i++) {
1772 gboolean created, prevactive, prevsender;
1773 guint pmembers, members;
1775 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1776 GST_DEBUG ("SSRC: %08x", ssrc);
1778 /* find src and mark bye, no probation when dealing with RTCP */
1779 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1783 /* store time for when we need to time out this source */
1784 source->bye_time = arrival->time;
1786 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1787 prevsender = RTP_SOURCE_IS_SENDER (source);
1789 /* let the source handle the rest */
1790 rtp_source_process_bye (source, reason);
1792 pmembers = sess->stats.active_sources;
1794 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1795 sess->stats.active_sources--;
1796 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1797 sess->stats.active_sources);
1799 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1800 sess->stats.sender_sources--;
1801 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1802 sess->stats.sender_sources);
1804 members = sess->stats.active_sources;
1806 if (!sess->source->received_bye && members < pmembers) {
1807 /* some members went away since the previous timeout estimate.
1808 * Perform reverse reconsideration but only when we are not scheduling a
1810 if (arrival->time < sess->next_rtcp_check_time) {
1811 GstClockTime time_remaining;
1813 time_remaining = sess->next_rtcp_check_time - arrival->time;
1814 sess->next_rtcp_check_time =
1815 gst_util_uint64_scale (time_remaining, members, pmembers);
1817 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1818 GST_TIME_ARGS (sess->next_rtcp_check_time));
1820 sess->next_rtcp_check_time += arrival->time;
1822 RTP_SESSION_UNLOCK (sess);
1823 /* notify app of reconsideration */
1824 if (sess->callbacks.reconsider)
1825 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1826 RTP_SESSION_LOCK (sess);
1831 on_new_ssrc (sess, source);
1833 on_bye_ssrc (sess, source);
1835 g_object_unref (source);
1841 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1842 RTPArrivalStats * arrival)
1844 GST_DEBUG ("received APP");
1848 * rtp_session_process_rtcp:
1849 * @sess: and #RTPSession
1850 * @buffer: an RTCP buffer
1851 * @current_time: the current system time
1853 * Process an RTCP buffer in the session manager. This function takes ownership
1856 * Returns: a #GstFlowReturn.
1859 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
1860 GstClockTime current_time)
1862 GstRTCPPacket packet;
1863 gboolean more, is_bye = FALSE, is_sr = FALSE;
1864 RTPArrivalStats arrival;
1865 GstFlowReturn result = GST_FLOW_OK;
1867 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1868 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1870 if (!gst_rtcp_buffer_validate (buffer))
1871 goto invalid_packet;
1873 GST_DEBUG ("received RTCP packet");
1875 RTP_SESSION_LOCK (sess);
1876 /* update arrival stats */
1877 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
1882 /* make writable, we might want to change the buffer */
1883 buffer = gst_buffer_make_metadata_writable (buffer);
1885 /* start processing the compound packet */
1886 more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1890 type = gst_rtcp_packet_get_type (&packet);
1892 /* when we are leaving the session, we should ignore all non-BYE messages */
1893 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1894 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1899 case GST_RTCP_TYPE_SR:
1900 rtp_session_process_sr (sess, &packet, &arrival);
1903 case GST_RTCP_TYPE_RR:
1904 rtp_session_process_rr (sess, &packet, &arrival);
1906 case GST_RTCP_TYPE_SDES:
1907 rtp_session_process_sdes (sess, &packet, &arrival);
1909 case GST_RTCP_TYPE_BYE:
1911 rtp_session_process_bye (sess, &packet, &arrival);
1913 case GST_RTCP_TYPE_APP:
1914 rtp_session_process_app (sess, &packet, &arrival);
1917 GST_WARNING ("got unknown RTCP packet");
1921 more = gst_rtcp_packet_move_to_next (&packet);
1924 /* if we are scheduling a BYE, we only want to count bye packets, else we
1925 * count everything */
1926 if (sess->source->received_bye) {
1928 sess->stats.bye_members++;
1929 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1932 /* keep track of average packet size */
1933 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1935 RTP_SESSION_UNLOCK (sess);
1937 /* notify caller of sr packets in the callback */
1938 if (is_sr && sess->callbacks.sync_rtcp)
1939 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1940 sess->sync_rtcp_user_data);
1942 gst_buffer_unref (buffer);
1949 GST_DEBUG ("invalid RTCP packet received");
1950 gst_buffer_unref (buffer);
1955 gst_buffer_unref (buffer);
1956 RTP_SESSION_UNLOCK (sess);
1957 GST_DEBUG ("ignoring RTP packet because we left");
1963 * rtp_session_send_rtp:
1964 * @sess: an #RTPSession
1965 * @buffer: an RTP buffer
1966 * @current_time: the current system time
1967 * @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
1968 * This is the buffer timestamp converted to NTP time.
1970 * Send the RTP buffer in the session manager. This function takes ownership of
1973 * Returns: a #GstFlowReturn.
1976 rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
1977 GstClockTime current_time, guint64 ntpnstime)
1979 GstFlowReturn result;
1981 gboolean prevsender;
1983 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1984 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1986 if (!gst_rtp_buffer_validate (buffer))
1987 goto invalid_packet;
1989 GST_LOG ("received RTP packet for sending");
1991 RTP_SESSION_LOCK (sess);
1992 source = sess->source;
1994 /* update last activity */
1995 source->last_rtp_activity = current_time;
1997 prevsender = RTP_SOURCE_IS_SENDER (source);
1999 /* we use our own source to send */
2000 result = rtp_source_send_rtp (source, buffer, ntpnstime);
2002 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2003 sess->stats.sender_sources++;
2004 RTP_SESSION_UNLOCK (sess);
2011 gst_buffer_unref (buffer);
2012 GST_DEBUG ("invalid RTP packet received");
2018 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2021 GstClockTime result;
2023 if (sess->source->received_bye) {
2024 result = rtp_stats_calculate_bye_interval (&sess->stats);
2026 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2027 RTP_SOURCE_IS_SENDER (sess->source), first);
2030 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2031 GST_TIME_ARGS (result), first);
2034 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2036 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2041 /* Stop the current @sess and schedule a BYE message for the other members.
2042 * One must have the session lock to call this function
2044 static GstFlowReturn
2045 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2046 GstClockTime current_time)
2048 GstFlowReturn result = GST_FLOW_OK;
2050 GstClockTime interval;
2052 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2054 source = sess->source;
2056 /* ignore more BYEs */
2057 if (source->received_bye)
2060 /* we have BYE now */
2061 source->received_bye = TRUE;
2062 /* at least one member wants to send a BYE */
2063 g_free (sess->bye_reason);
2064 sess->bye_reason = g_strdup (reason);
2065 sess->stats.avg_rtcp_packet_size = 100;
2066 sess->stats.bye_members = 1;
2067 sess->first_rtcp = TRUE;
2068 sess->sent_bye = FALSE;
2070 /* reschedule transmission */
2071 sess->last_rtcp_send_time = current_time;
2072 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2073 sess->next_rtcp_check_time = current_time + interval;
2075 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2076 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2078 RTP_SESSION_UNLOCK (sess);
2079 /* notify app of reconsideration */
2080 if (sess->callbacks.reconsider)
2081 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2082 RTP_SESSION_LOCK (sess);
2089 * rtp_session_schedule_bye:
2090 * @sess: an #RTPSession
2091 * @reason: a reason or NULL
2092 * @current_time: the current system time
2094 * Stop the current @sess and schedule a BYE message for the other members.
2096 * Returns: a #GstFlowReturn.
2099 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2100 GstClockTime current_time)
2102 GstFlowReturn result = GST_FLOW_OK;
2104 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2106 RTP_SESSION_LOCK (sess);
2107 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2108 RTP_SESSION_UNLOCK (sess);
2114 * rtp_session_next_timeout:
2115 * @sess: an #RTPSession
2116 * @current_time: the current system time
2118 * Get the next time we should perform session maintenance tasks.
2120 * Returns: a time when rtp_session_on_timeout() should be called with the
2121 * current system time.
2124 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2126 GstClockTime result;
2128 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2130 RTP_SESSION_LOCK (sess);
2132 result = sess->next_rtcp_check_time;
2134 GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
2135 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2137 if (result < current_time) {
2138 GST_DEBUG ("take current time as base");
2139 /* our previous check time expired, start counting from the current time
2141 result = current_time;
2144 if (sess->source->received_bye) {
2145 if (sess->sent_bye) {
2146 GST_DEBUG ("we sent BYE already");
2147 result = GST_CLOCK_TIME_NONE;
2148 } else if (sess->stats.active_sources >= 50) {
2149 GST_DEBUG ("reconsider BYE, more than 50 sources");
2150 /* reconsider BYE if members >= 50 */
2151 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2154 if (sess->first_rtcp) {
2155 GST_DEBUG ("first RTCP packet");
2156 /* we are called for the first time */
2157 result += calculate_rtcp_interval (sess, FALSE, TRUE);
2158 } else if (sess->next_rtcp_check_time < current_time) {
2159 GST_DEBUG ("old check time expired, getting new timeout");
2160 /* get a new timeout when we need to */
2161 result += calculate_rtcp_interval (sess, FALSE, FALSE);
2164 sess->next_rtcp_check_time = result;
2166 GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2167 RTP_SESSION_UNLOCK (sess);
2176 GstClockTime current_time;
2178 GstClockTime interval;
2179 GstRTCPPacket packet;
2185 session_start_rtcp (RTPSession * sess, ReportData * data)
2187 GstRTCPPacket *packet = &data->packet;
2188 RTPSource *own = sess->source;
2190 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2192 if (RTP_SOURCE_IS_SENDER (own)) {
2195 guint32 packet_count, octet_count;
2197 /* we are a sender, create SR */
2198 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2199 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
2201 /* get latest stats */
2202 rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
2203 &packet_count, &octet_count);
2205 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2206 packet_count, octet_count);
2208 /* fill in sender report info */
2209 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2210 ntptime, rtptime, packet_count, octet_count);
2212 /* we are only receiver, create RR */
2213 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2214 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2215 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2219 /* construct a Sender or Receiver Report */
2221 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2223 RTPSession *sess = data->sess;
2224 GstRTCPPacket *packet = &data->packet;
2226 /* create a new buffer if needed */
2227 if (data->rtcp == NULL) {
2228 session_start_rtcp (sess, data);
2230 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2231 /* only report about other sender sources */
2232 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2233 guint8 fractionlost;
2235 guint32 exthighestseq, jitter;
2239 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2240 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2242 /* packet is not yet filled, add report block for this source. */
2243 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2244 exthighestseq, jitter, lsr, dlsr);
2249 /* perform cleanup of sources that timed out */
2251 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2253 gboolean remove = FALSE;
2254 gboolean byetimeout = FALSE;
2255 gboolean sendertimeout = FALSE;
2256 gboolean is_sender, is_active;
2257 RTPSession *sess = data->sess;
2258 GstClockTime interval;
2260 is_sender = RTP_SOURCE_IS_SENDER (source);
2261 is_active = RTP_SOURCE_IS_ACTIVE (source);
2263 /* check for our own source, we don't want to delete our own source. */
2264 if (!(source == sess->source)) {
2265 if (source->received_bye) {
2266 /* if we received a BYE from the source, remove the source after some
2268 if (data->current_time > source->bye_time &&
2269 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2270 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2275 /* sources that were inactive for more than 5 times the deterministic reporting
2276 * interval get timed out. the min timeout is 5 seconds. */
2277 if (data->current_time > source->last_activity) {
2278 interval = MAX (data->interval * 5, 5 * GST_SECOND);
2279 if (data->current_time - source->last_activity > interval) {
2280 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2281 source->ssrc, GST_TIME_ARGS (source->last_activity));
2287 /* senders that did not send for a long time become a receiver, this also
2288 * holds for our own source. */
2290 if (data->current_time > source->last_rtp_activity) {
2291 interval = MAX (data->interval * 2, 5 * GST_SECOND);
2292 if (data->current_time - source->last_rtp_activity > interval) {
2293 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2294 GST_TIME_FORMAT, source->ssrc,
2295 GST_TIME_ARGS (source->last_rtp_activity));
2296 source->is_sender = FALSE;
2297 sess->stats.sender_sources--;
2298 sendertimeout = TRUE;
2304 sess->total_sources--;
2306 sess->stats.sender_sources--;
2308 sess->stats.active_sources--;
2311 on_bye_timeout (sess, source);
2313 on_timeout (sess, source);
2316 on_sender_timeout (sess, source);
2322 session_sdes (RTPSession * sess, ReportData * data)
2324 GstRTCPPacket *packet = &data->packet;
2328 /* add SDES packet */
2329 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2331 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2333 rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
2335 gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
2338 /* other SDES items must only be added at regular intervals and only when the
2339 * user requests to since it might be a privacy problem */
2341 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
2342 strlen (sess->name), (guint8 *) sess->name);
2343 gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
2344 strlen (sess->tool), (guint8 *) sess->tool);
2347 data->has_sdes = TRUE;
2350 /* schedule a BYE packet */
2352 session_bye (RTPSession * sess, ReportData * data)
2354 GstRTCPPacket *packet = &data->packet;
2357 session_start_rtcp (sess, data);
2360 session_sdes (sess, data);
2362 /* add a BYE packet */
2363 gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2364 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2365 if (sess->bye_reason)
2366 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2368 /* we have a BYE packet now */
2369 data->is_bye = TRUE;
2373 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2375 GstClockTime new_send_time, elapsed;
2378 /* no need to check yet */
2379 if (sess->next_rtcp_check_time > current_time) {
2380 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2381 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2382 GST_TIME_ARGS (current_time));
2386 /* get elapsed time since we last reported */
2387 elapsed = current_time - sess->last_rtcp_send_time;
2389 /* perform forward reconsideration */
2390 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2392 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2393 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2395 new_send_time += sess->last_rtcp_send_time;
2397 /* check if reconsideration */
2398 if (current_time < new_send_time) {
2399 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2400 GST_TIME_ARGS (new_send_time));
2402 /* store new check time */
2403 sess->next_rtcp_check_time = new_send_time;
2406 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2408 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2409 GST_TIME_ARGS (new_send_time));
2410 sess->next_rtcp_check_time = current_time + new_send_time;
2416 * rtp_session_on_timeout:
2417 * @sess: an #RTPSession
2418 * @current_time: the current system time
2419 * @ntpnstime: the current NTP time in nanoseconds
2421 * Perform maintenance actions after the timeout obtained with
2422 * rtp_session_next_timeout() expired.
2424 * This function will perform timeouts of receivers and senders, send a BYE
2425 * packet or generate RTCP packets with current session stats.
2427 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
2428 * times, for each packet that should be processed.
2430 * Returns: a #GstFlowReturn.
2433 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
2436 GstFlowReturn result = GST_FLOW_OK;
2440 gboolean notify = FALSE;
2442 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2444 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2445 GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
2449 data.current_time = current_time;
2450 data.ntpnstime = ntpnstime;
2451 data.is_bye = FALSE;
2452 data.has_sdes = FALSE;
2456 RTP_SESSION_LOCK (sess);
2457 /* get a new interval, we need this for various cleanups etc */
2458 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2460 /* first perform cleanups */
2461 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2462 (GHRFunc) session_cleanup, &data);
2464 /* see if we need to generate SR or RR packets */
2465 if (is_rtcp_time (sess, current_time, &data)) {
2466 if (own->received_bye) {
2467 /* generate BYE instead */
2468 GST_DEBUG ("generating BYE message");
2469 session_bye (sess, &data);
2470 sess->sent_bye = TRUE;
2472 /* loop over all known sources and do something */
2473 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2474 (GHFunc) session_report_blocks, &data);
2481 /* we keep track of the last report time in order to timeout inactive
2482 * receivers or senders */
2483 sess->last_rtcp_send_time = data.current_time;
2484 sess->first_rtcp = FALSE;
2486 /* add SDES for this source when not already added */
2488 session_sdes (sess, &data);
2490 /* update average RTCP size before sending */
2491 size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2492 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2495 /* check for outdated collisions */
2496 GST_DEBUG ("checking collision list");
2497 item = g_list_first (sess->conflicting_addresses);
2499 RTPConflictingAddress *known_conflict = item->data;
2500 GList *next_item = g_list_next (item);
2502 if (known_conflict->time < current_time - (data.interval *
2503 RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2504 sess->conflicting_addresses =
2505 g_list_delete_link (sess->conflicting_addresses, item);
2506 GST_DEBUG ("collision %p timed out", known_conflict);
2507 g_free (known_conflict);
2512 if (sess->change_ssrc) {
2513 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
2514 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2515 GINT_TO_POINTER (own->ssrc));
2517 own->ssrc = rtp_session_create_new_ssrc (sess);
2518 rtp_source_reset (own);
2520 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2521 GINT_TO_POINTER (own->ssrc), own);
2523 g_free (sess->bye_reason);
2524 sess->bye_reason = NULL;
2525 sess->sent_bye = FALSE;
2526 sess->change_ssrc = FALSE;
2528 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
2530 RTP_SESSION_UNLOCK (sess);
2533 g_object_notify (G_OBJECT (sess), "internal-ssrc");
2535 /* push out the RTCP packet */
2537 /* close the RTCP packet */
2538 gst_rtcp_buffer_end (data.rtcp);
2540 GST_DEBUG ("sending packet");
2541 if (sess->callbacks.send_rtcp)
2542 result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
2543 sess->sent_bye, sess->send_rtcp_user_data);
2545 GST_DEBUG ("freeing packet");
2546 gst_buffer_unref (data.rtcp);