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., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "gstrtpbin-marshal.h"
32 #include "rtpsession.h"
34 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
35 #define GST_CAT_DEFAULT rtp_session_debug
37 /* signals and args */
40 SIGNAL_GET_SOURCE_BY_SSRC,
42 SIGNAL_ON_SSRC_COLLISION,
43 SIGNAL_ON_SSRC_VALIDATED,
44 SIGNAL_ON_SSRC_ACTIVE,
47 SIGNAL_ON_BYE_TIMEOUT,
49 SIGNAL_ON_SENDER_TIMEOUT,
50 SIGNAL_ON_SENDING_RTCP,
51 SIGNAL_ON_FEEDBACK_RTCP,
56 #define DEFAULT_INTERNAL_SOURCE NULL
57 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
58 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
59 #define DEFAULT_RTCP_RR_BANDWIDTH -1
60 #define DEFAULT_RTCP_RS_BANDWIDTH -1
61 #define DEFAULT_RTCP_MTU 1400
62 #define DEFAULT_SDES NULL
63 #define DEFAULT_NUM_SOURCES 0
64 #define DEFAULT_NUM_ACTIVE_SOURCES 0
65 #define DEFAULT_SOURCES NULL
66 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
67 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
68 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
69 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
78 PROP_RTCP_RR_BANDWIDTH,
79 PROP_RTCP_RS_BANDWIDTH,
83 PROP_NUM_ACTIVE_SOURCES,
86 PROP_RTCP_MIN_INTERVAL,
87 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
88 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
93 /* update average packet size */
94 #define INIT_AVG(avg, val) \
96 #define UPDATE_AVG(avg, val) \
100 (avg) = ((val) + (15 * (avg))) >> 4;
103 /* The number RTCP intervals after which to timeout entries in the
106 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
108 /* GObject vmethods */
109 static void rtp_session_finalize (GObject * object);
110 static void rtp_session_set_property (GObject * object, guint prop_id,
111 const GValue * value, GParamSpec * pspec);
112 static void rtp_session_get_property (GObject * object, guint prop_id,
113 GValue * value, GParamSpec * pspec);
115 static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
116 GstBuffer * buffer, gboolean early);
117 static void rtp_session_send_rtcp (RTPSession * sess,
118 GstClockTimeDiff max_delay);
121 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
123 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
125 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
126 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
127 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
128 const gchar * reason, GstClockTime current_time);
129 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
130 gboolean deterministic, gboolean first);
133 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
134 const GValue * handler_return, gpointer data)
136 if (g_value_get_boolean (handler_return))
137 g_value_set_boolean (return_accu, TRUE);
143 rtp_session_class_init (RTPSessionClass * klass)
145 GObjectClass *gobject_class;
147 gobject_class = (GObjectClass *) klass;
149 gobject_class->finalize = rtp_session_finalize;
150 gobject_class->set_property = rtp_session_set_property;
151 gobject_class->get_property = rtp_session_get_property;
154 * RTPSession::get-source-by-ssrc:
155 * @session: the object which received the signal
156 * @ssrc: the SSRC of the RTPSource
158 * Request the #RTPSource object with SSRC @ssrc in @session.
160 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
161 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
162 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
163 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
164 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
167 * RTPSession::on-new-ssrc:
168 * @session: the object which received the signal
169 * @src: the new RTPSource
171 * Notify of a new SSRC that entered @session.
173 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
174 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
175 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
176 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
179 * RTPSession::on-ssrc-collision:
180 * @session: the object which received the signal
181 * @src: the #RTPSource that caused a collision
183 * Notify when we have an SSRC collision
185 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
186 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
188 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
191 * RTPSession::on-ssrc-validated:
192 * @session: the object which received the signal
193 * @src: the new validated RTPSource
195 * Notify of a new SSRC that became validated.
197 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
198 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
200 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
203 * RTPSession::on-ssrc-active:
204 * @session: the object which received the signal
205 * @src: the active RTPSource
207 * Notify of a SSRC that is active, i.e., sending RTCP.
209 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
210 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
212 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
215 * RTPSession::on-ssrc-sdes:
216 * @session: the object which received the signal
217 * @src: the RTPSource
219 * Notify that a new SDES was received for SSRC.
221 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
222 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
224 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
227 * RTPSession::on-bye-ssrc:
228 * @session: the object which received the signal
229 * @src: the RTPSource that went away
231 * Notify of an SSRC that became inactive because of a BYE packet.
233 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
234 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
236 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
239 * RTPSession::on-bye-timeout:
240 * @session: the object which received the signal
241 * @src: the RTPSource that timed out
243 * Notify of an SSRC that has timed out because of BYE
245 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
246 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
248 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
251 * RTPSession::on-timeout:
252 * @session: the object which received the signal
253 * @src: the RTPSource that timed out
255 * Notify of an SSRC that has timed out
257 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
258 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
259 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
260 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
263 * RTPSession::on-sender-timeout:
264 * @session: the object which received the signal
265 * @src: the RTPSource that timed out
267 * Notify of an SSRC that was a sender but timed out and became a receiver.
269 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
270 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
271 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
272 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
276 * RTPSession::on-sending-rtcp
277 * @session: the object which received the signal
278 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
279 * @early: %TRUE if the packet is early, %FALSE if it is regular
281 * This signal is emitted before sending an RTCP packet, it can be used
282 * to add extra RTCP Packets.
284 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
285 * if suppressing it is acceptable
287 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
288 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
289 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
290 accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__BOXED_BOOLEAN,
291 G_TYPE_BOOLEAN, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
295 * RTPSession::on-feedback-rtcp:
296 * @session: the object which received the signal
297 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
298 * %GST_RTCP_TYPE_RTPFB
299 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
300 * @sender_ssrc: The SSRC of the sender
301 * @media_ssrc: The SSRC of the media this refers to
302 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
305 * Notify that a RTCP feedback packet has been received
307 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
308 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
309 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
310 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_BOXED,
311 G_TYPE_NONE, 5, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
315 * RTPSession::send-rtcp:
316 * @session: the object which received the signal
317 * @max_delay: The maximum delay after which the feedback will not be useful
320 * Requests that the #RTPSession initiate a new RTCP packet as soon as
321 * possible within the requested delay.
324 rtp_session_signals[SIGNAL_SEND_RTCP] =
325 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
326 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
327 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
328 gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64);
330 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
331 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
332 "The internal SSRC used for the session",
333 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
336 g_param_spec_object ("internal-source", "Internal Source",
337 "The internal source element of the session",
338 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
340 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
341 g_param_spec_double ("bandwidth", "Bandwidth",
342 "The bandwidth of the session (0 for auto-discover)",
343 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
344 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
346 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
347 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
348 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
349 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
350 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
352 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
353 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
354 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
355 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
356 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
358 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
359 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
360 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
361 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
362 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
364 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
365 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
366 "The maximum size of the RTCP packets",
367 16, G_MAXINT16, DEFAULT_RTCP_MTU,
368 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_SDES,
371 g_param_spec_boxed ("sdes", "SDES",
372 "The SDES items of this session",
373 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
375 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
376 g_param_spec_uint ("num-sources", "Num Sources",
377 "The number of sources in the session", 0, G_MAXUINT,
378 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
380 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
381 g_param_spec_uint ("num-active-sources", "Num Active Sources",
382 "The number of active sources in the session", 0, G_MAXUINT,
383 DEFAULT_NUM_ACTIVE_SOURCES,
384 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
388 * Get a GValue Array of all sources in the session.
391 * <title>Getting the #RTPSources of a session
398 * g_object_get (sess, "sources", &arr, NULL);
400 * for (i = 0; i < arr->n_values; i++) {
403 * val = g_value_array_get_nth (arr, i);
404 * source = g_value_get_object (val);
406 * g_value_array_free (arr);
411 g_object_class_install_property (gobject_class, PROP_SOURCES,
412 g_param_spec_boxed ("sources", "Sources",
413 "An array of all known sources in the session",
414 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
416 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
417 g_param_spec_boolean ("favor-new", "Favor new sources",
418 "Resolve SSRC conflict in favor of new sources", FALSE,
419 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
421 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
422 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
423 "Minimum interval between Regular RTCP packet (in ns)",
424 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
425 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
427 g_object_class_install_property (gobject_class,
428 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
429 g_param_spec_uint64 ("rtcp-feedback-retention-window",
430 "RTCP Feedback retention window",
431 "Duration during which RTCP Feedback packets are retained (in ns)",
432 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
433 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
435 g_object_class_install_property (gobject_class,
436 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
437 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
438 "RTCP Immediate Feedback threshold",
439 "The maximum number of members of a RTP session for which immediate"
441 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
442 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
444 g_object_class_install_property (gobject_class, PROP_PROBATION,
445 g_param_spec_uint ("probation", "Number of probations",
446 "Consecutive packet sequence numbers to accept the source",
447 0, G_MAXUINT, DEFAULT_PROBATION,
448 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
450 klass->get_source_by_ssrc =
451 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
452 klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
453 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
455 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
459 rtp_session_init (RTPSession * sess)
465 g_mutex_init (&sess->lock);
466 sess->key = g_random_int ();
470 for (i = 0; i < 32; i++) {
472 g_hash_table_new_full (NULL, NULL, NULL,
473 (GDestroyNotify) g_object_unref);
476 rtp_stats_init_defaults (&sess->stats);
478 sess->recalc_bandwidth = TRUE;
479 sess->bandwidth = DEFAULT_BANDWIDTH;
480 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
481 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
482 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
484 /* create an active SSRC for this session manager */
485 sess->source = rtp_session_create_source (sess);
486 sess->source->validated = TRUE;
487 sess->source->internal = TRUE;
488 sess->stats.active_sources++;
489 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
490 sess->source->stats.prev_rtcptime = 0;
491 sess->source->stats.last_rtcptime = 1;
493 rtp_stats_set_min_interval (&sess->stats,
494 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
496 /* default UDP header length */
497 sess->header_len = 28;
498 sess->mtu = DEFAULT_RTCP_MTU;
500 sess->probation = DEFAULT_PROBATION;
502 /* some default SDES entries */
503 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
505 /* we do not want to leak details like the username or hostname here */
506 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
507 gst_structure_set (sdes, "cname", G_TYPE_STRING, str, NULL);
511 /* we do not want to leak the user's real name here */
512 str = g_strdup_printf ("Anon%u", g_random_int ());
513 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
517 gst_structure_set (sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
519 /* and configure in the source */
520 rtp_source_set_sdes_struct (sess->source, sdes);
522 sess->first_rtcp = TRUE;
523 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
525 sess->allow_early = TRUE;
526 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
527 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
528 sess->rtcp_immediate_feedback_threshold =
529 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
531 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
533 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
537 rtp_session_finalize (GObject * object)
542 sess = RTP_SESSION_CAST (object);
544 g_mutex_clear (&sess->lock);
546 for (i = 0; i < 32; i++)
547 g_hash_table_destroy (sess->ssrcs[i]);
549 g_free (sess->bye_reason);
551 g_object_unref (sess->source);
553 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
557 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
559 GValue value = { 0 };
561 g_value_init (&value, RTP_TYPE_SOURCE);
562 g_value_take_object (&value, source);
563 /* copies the value */
564 g_value_array_append (arr, &value);
568 rtp_session_create_sources (RTPSession * sess)
573 RTP_SESSION_LOCK (sess);
574 /* get number of elements in the table */
575 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
576 /* create the result value array */
577 res = g_value_array_new (size);
579 /* and copy all values into the array */
580 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
581 RTP_SESSION_UNLOCK (sess);
587 rtp_session_set_property (GObject * object, guint prop_id,
588 const GValue * value, GParamSpec * pspec)
592 sess = RTP_SESSION (object);
595 case PROP_INTERNAL_SSRC:
596 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
599 RTP_SESSION_LOCK (sess);
600 sess->bandwidth = g_value_get_double (value);
601 sess->recalc_bandwidth = TRUE;
602 RTP_SESSION_UNLOCK (sess);
604 case PROP_RTCP_FRACTION:
605 RTP_SESSION_LOCK (sess);
606 sess->rtcp_bandwidth = g_value_get_double (value);
607 sess->recalc_bandwidth = TRUE;
608 RTP_SESSION_UNLOCK (sess);
610 case PROP_RTCP_RR_BANDWIDTH:
611 RTP_SESSION_LOCK (sess);
612 sess->rtcp_rr_bandwidth = g_value_get_int (value);
613 sess->recalc_bandwidth = TRUE;
614 RTP_SESSION_UNLOCK (sess);
616 case PROP_RTCP_RS_BANDWIDTH:
617 RTP_SESSION_LOCK (sess);
618 sess->rtcp_rs_bandwidth = g_value_get_int (value);
619 sess->recalc_bandwidth = TRUE;
620 RTP_SESSION_UNLOCK (sess);
623 sess->mtu = g_value_get_uint (value);
626 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
629 sess->favor_new = g_value_get_boolean (value);
631 case PROP_RTCP_MIN_INTERVAL:
632 rtp_stats_set_min_interval (&sess->stats,
633 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
634 /* trigger reconsideration */
635 RTP_SESSION_LOCK (sess);
636 sess->next_rtcp_check_time = 0;
637 RTP_SESSION_UNLOCK (sess);
638 if (sess->callbacks.reconsider)
639 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
641 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
642 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
645 sess->probation = g_value_get_uint (value);
646 g_object_set_property (G_OBJECT (sess->source), "probation", value);
649 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
655 rtp_session_get_property (GObject * object, guint prop_id,
656 GValue * value, GParamSpec * pspec)
660 sess = RTP_SESSION (object);
663 case PROP_INTERNAL_SSRC:
664 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
666 case PROP_INTERNAL_SOURCE:
667 g_value_take_object (value, rtp_session_get_internal_source (sess));
670 g_value_set_double (value, sess->bandwidth);
672 case PROP_RTCP_FRACTION:
673 g_value_set_double (value, sess->rtcp_bandwidth);
675 case PROP_RTCP_RR_BANDWIDTH:
676 g_value_set_int (value, sess->rtcp_rr_bandwidth);
678 case PROP_RTCP_RS_BANDWIDTH:
679 g_value_set_int (value, sess->rtcp_rs_bandwidth);
682 g_value_set_uint (value, sess->mtu);
685 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
687 case PROP_NUM_SOURCES:
688 g_value_set_uint (value, rtp_session_get_num_sources (sess));
690 case PROP_NUM_ACTIVE_SOURCES:
691 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
694 g_value_take_boxed (value, rtp_session_create_sources (sess));
697 g_value_set_boolean (value, sess->favor_new);
699 case PROP_RTCP_MIN_INTERVAL:
700 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
702 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
703 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
706 g_value_set_uint (value, sess->probation);
707 g_object_get_property (G_OBJECT (sess->source), "probation", value);
710 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
716 on_new_ssrc (RTPSession * sess, RTPSource * source)
718 g_object_ref (source);
719 RTP_SESSION_UNLOCK (sess);
720 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
721 RTP_SESSION_LOCK (sess);
722 g_object_unref (source);
726 on_ssrc_collision (RTPSession * sess, RTPSource * source)
728 g_object_ref (source);
729 RTP_SESSION_UNLOCK (sess);
730 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
732 RTP_SESSION_LOCK (sess);
733 g_object_unref (source);
737 on_ssrc_validated (RTPSession * sess, RTPSource * source)
739 g_object_ref (source);
740 RTP_SESSION_UNLOCK (sess);
741 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
743 RTP_SESSION_LOCK (sess);
744 g_object_unref (source);
748 on_ssrc_active (RTPSession * sess, RTPSource * source)
750 g_object_ref (source);
751 RTP_SESSION_UNLOCK (sess);
752 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
753 RTP_SESSION_LOCK (sess);
754 g_object_unref (source);
758 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
760 g_object_ref (source);
761 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
762 RTP_SESSION_UNLOCK (sess);
763 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
764 RTP_SESSION_LOCK (sess);
765 g_object_unref (source);
769 on_bye_ssrc (RTPSession * sess, RTPSource * source)
771 g_object_ref (source);
772 RTP_SESSION_UNLOCK (sess);
773 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
774 RTP_SESSION_LOCK (sess);
775 g_object_unref (source);
779 on_bye_timeout (RTPSession * sess, RTPSource * source)
781 g_object_ref (source);
782 RTP_SESSION_UNLOCK (sess);
783 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
784 RTP_SESSION_LOCK (sess);
785 g_object_unref (source);
789 on_timeout (RTPSession * sess, RTPSource * source)
791 g_object_ref (source);
792 RTP_SESSION_UNLOCK (sess);
793 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
794 RTP_SESSION_LOCK (sess);
795 g_object_unref (source);
799 on_sender_timeout (RTPSession * sess, RTPSource * source)
801 g_object_ref (source);
802 RTP_SESSION_UNLOCK (sess);
803 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
805 RTP_SESSION_LOCK (sess);
806 g_object_unref (source);
812 * Create a new session object.
814 * Returns: a new #RTPSession. g_object_unref() after usage.
817 rtp_session_new (void)
821 sess = g_object_new (RTP_TYPE_SESSION, NULL);
827 * rtp_session_set_callbacks:
828 * @sess: an #RTPSession
829 * @callbacks: callbacks to configure
830 * @user_data: user data passed in the callbacks
832 * Configure a set of callbacks to be notified of actions.
835 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
838 g_return_if_fail (RTP_IS_SESSION (sess));
840 if (callbacks->process_rtp) {
841 sess->callbacks.process_rtp = callbacks->process_rtp;
842 sess->process_rtp_user_data = user_data;
844 if (callbacks->send_rtp) {
845 sess->callbacks.send_rtp = callbacks->send_rtp;
846 sess->send_rtp_user_data = user_data;
848 if (callbacks->send_rtcp) {
849 sess->callbacks.send_rtcp = callbacks->send_rtcp;
850 sess->send_rtcp_user_data = user_data;
852 if (callbacks->sync_rtcp) {
853 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
854 sess->sync_rtcp_user_data = user_data;
856 if (callbacks->clock_rate) {
857 sess->callbacks.clock_rate = callbacks->clock_rate;
858 sess->clock_rate_user_data = user_data;
860 if (callbacks->reconsider) {
861 sess->callbacks.reconsider = callbacks->reconsider;
862 sess->reconsider_user_data = user_data;
864 if (callbacks->request_key_unit) {
865 sess->callbacks.request_key_unit = callbacks->request_key_unit;
866 sess->request_key_unit_user_data = user_data;
868 if (callbacks->request_time) {
869 sess->callbacks.request_time = callbacks->request_time;
870 sess->request_time_user_data = user_data;
875 * rtp_session_set_process_rtp_callback:
876 * @sess: an #RTPSession
877 * @callback: callback to set
878 * @user_data: user data passed in the callback
880 * Configure only the process_rtp callback to be notified of the process_rtp action.
883 rtp_session_set_process_rtp_callback (RTPSession * sess,
884 RTPSessionProcessRTP callback, gpointer user_data)
886 g_return_if_fail (RTP_IS_SESSION (sess));
888 sess->callbacks.process_rtp = callback;
889 sess->process_rtp_user_data = user_data;
893 * rtp_session_set_send_rtp_callback:
894 * @sess: an #RTPSession
895 * @callback: callback to set
896 * @user_data: user data passed in the callback
898 * Configure only the send_rtp callback to be notified of the send_rtp action.
901 rtp_session_set_send_rtp_callback (RTPSession * sess,
902 RTPSessionSendRTP callback, gpointer user_data)
904 g_return_if_fail (RTP_IS_SESSION (sess));
906 sess->callbacks.send_rtp = callback;
907 sess->send_rtp_user_data = user_data;
911 * rtp_session_set_send_rtcp_callback:
912 * @sess: an #RTPSession
913 * @callback: callback to set
914 * @user_data: user data passed in the callback
916 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
919 rtp_session_set_send_rtcp_callback (RTPSession * sess,
920 RTPSessionSendRTCP callback, gpointer user_data)
922 g_return_if_fail (RTP_IS_SESSION (sess));
924 sess->callbacks.send_rtcp = callback;
925 sess->send_rtcp_user_data = user_data;
929 * rtp_session_set_sync_rtcp_callback:
930 * @sess: an #RTPSession
931 * @callback: callback to set
932 * @user_data: user data passed in the callback
934 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
937 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
938 RTPSessionSyncRTCP callback, gpointer user_data)
940 g_return_if_fail (RTP_IS_SESSION (sess));
942 sess->callbacks.sync_rtcp = callback;
943 sess->sync_rtcp_user_data = user_data;
947 * rtp_session_set_clock_rate_callback:
948 * @sess: an #RTPSession
949 * @callback: callback to set
950 * @user_data: user data passed in the callback
952 * Configure only the clock_rate callback to be notified of the clock_rate action.
955 rtp_session_set_clock_rate_callback (RTPSession * sess,
956 RTPSessionClockRate callback, gpointer user_data)
958 g_return_if_fail (RTP_IS_SESSION (sess));
960 sess->callbacks.clock_rate = callback;
961 sess->clock_rate_user_data = user_data;
965 * rtp_session_set_reconsider_callback:
966 * @sess: an #RTPSession
967 * @callback: callback to set
968 * @user_data: user data passed in the callback
970 * Configure only the reconsider callback to be notified of the reconsider action.
973 rtp_session_set_reconsider_callback (RTPSession * sess,
974 RTPSessionReconsider callback, gpointer user_data)
976 g_return_if_fail (RTP_IS_SESSION (sess));
978 sess->callbacks.reconsider = callback;
979 sess->reconsider_user_data = user_data;
983 * rtp_session_set_request_time_callback:
984 * @sess: an #RTPSession
985 * @callback: callback to set
986 * @user_data: user data passed in the callback
988 * Configure only the request_time callback
991 rtp_session_set_request_time_callback (RTPSession * sess,
992 RTPSessionRequestTime callback, gpointer user_data)
994 g_return_if_fail (RTP_IS_SESSION (sess));
996 sess->callbacks.request_time = callback;
997 sess->request_time_user_data = user_data;
1001 * rtp_session_set_bandwidth:
1002 * @sess: an #RTPSession
1003 * @bandwidth: the bandwidth allocated
1005 * Set the session bandwidth in bytes per second.
1008 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1010 g_return_if_fail (RTP_IS_SESSION (sess));
1012 RTP_SESSION_LOCK (sess);
1013 sess->stats.bandwidth = bandwidth;
1014 RTP_SESSION_UNLOCK (sess);
1018 * rtp_session_get_bandwidth:
1019 * @sess: an #RTPSession
1021 * Get the session bandwidth.
1023 * Returns: the session bandwidth.
1026 rtp_session_get_bandwidth (RTPSession * sess)
1030 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1032 RTP_SESSION_LOCK (sess);
1033 result = sess->stats.bandwidth;
1034 RTP_SESSION_UNLOCK (sess);
1040 * rtp_session_set_rtcp_fraction:
1041 * @sess: an #RTPSession
1042 * @bandwidth: the RTCP bandwidth
1044 * Set the bandwidth in bytes per second that should be used for RTCP
1048 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1050 g_return_if_fail (RTP_IS_SESSION (sess));
1052 RTP_SESSION_LOCK (sess);
1053 sess->stats.rtcp_bandwidth = bandwidth;
1054 RTP_SESSION_UNLOCK (sess);
1058 * rtp_session_get_rtcp_fraction:
1059 * @sess: an #RTPSession
1061 * Get the session bandwidth used for RTCP.
1063 * Returns: The bandwidth used for RTCP messages.
1066 rtp_session_get_rtcp_fraction (RTPSession * sess)
1070 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1072 RTP_SESSION_LOCK (sess);
1073 result = sess->stats.rtcp_bandwidth;
1074 RTP_SESSION_UNLOCK (sess);
1080 * rtp_session_get_sdes_struct:
1081 * @sess: an #RTSPSession
1083 * Get the SDES data as a #GstStructure
1085 * Returns: a GstStructure with SDES items for @sess. This function returns a
1086 * copy of the SDES structure, use gst_structure_free() after usage.
1089 rtp_session_get_sdes_struct (RTPSession * sess)
1091 const GstStructure *sdes;
1092 GstStructure *result = NULL;
1094 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1096 RTP_SESSION_LOCK (sess);
1097 sdes = rtp_source_get_sdes_struct (sess->source);
1099 result = gst_structure_copy (sdes);
1100 RTP_SESSION_UNLOCK (sess);
1106 * rtp_session_set_sdes_struct:
1107 * @sess: an #RTSPSession
1108 * @sdes: a #GstStructure
1110 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1113 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1115 g_return_if_fail (sdes);
1116 g_return_if_fail (RTP_IS_SESSION (sess));
1118 RTP_SESSION_LOCK (sess);
1119 rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
1120 RTP_SESSION_UNLOCK (sess);
1123 static GstFlowReturn
1124 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1126 GstFlowReturn result = GST_FLOW_OK;
1128 if (source == session->source) {
1129 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1131 RTP_SESSION_UNLOCK (session);
1133 if (session->callbacks.send_rtp)
1135 session->callbacks.send_rtp (session, source, data,
1136 session->send_rtp_user_data);
1138 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1141 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1142 RTP_SESSION_UNLOCK (session);
1144 if (session->callbacks.process_rtp)
1146 session->callbacks.process_rtp (session, source,
1147 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1149 gst_buffer_unref (GST_BUFFER_CAST (data));
1151 RTP_SESSION_LOCK (session);
1157 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1161 RTP_SESSION_UNLOCK (session);
1163 if (session->callbacks.clock_rate)
1165 session->callbacks.clock_rate (session, pt,
1166 session->clock_rate_user_data);
1170 RTP_SESSION_LOCK (session);
1172 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1177 static RTPSourceCallbacks callbacks = {
1178 (RTPSourcePushRTP) source_push_rtp,
1179 (RTPSourceClockRate) source_clock_rate,
1183 check_collision (RTPSession * sess, RTPSource * source,
1184 RTPArrivalStats * arrival, gboolean rtp)
1186 /* If we have no arrival address, we can't do collision checking */
1187 if (!arrival->address)
1190 if (sess->source != source) {
1191 GSocketAddress *from;
1193 /* This is not our local source, but lets check if two remote
1196 from = source->rtp_from;
1198 from = source->rtcp_from;
1202 if (__g_socket_address_equal (from, arrival->address)) {
1203 /* Address is the same */
1206 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1207 rtp_source_get_ssrc (source));
1208 if (sess->favor_new) {
1209 if (rtp_source_find_conflicting_address (source,
1210 arrival->address, arrival->current_time)) {
1213 buf1 = __g_socket_address_to_string (arrival->address);
1214 GST_LOG ("Known conflict on %x for %s, dropping packet",
1215 rtp_source_get_ssrc (source), buf1);
1222 /* Current address is not a known conflict, lets assume this is
1223 * a new source. Save old address in possible conflict list
1225 rtp_source_add_conflicting_address (source, from,
1226 arrival->current_time);
1228 buf1 = __g_socket_address_to_string (from);
1229 buf2 = __g_socket_address_to_string (arrival->address);
1231 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1232 " saving old as known conflict",
1233 rtp_source_get_ssrc (source), buf1, buf2);
1236 rtp_source_set_rtp_from (source, arrival->address);
1238 rtp_source_set_rtcp_from (source, arrival->address);
1246 /* Don't need to save old addresses, we ignore new sources */
1251 /* We don't already have a from address for RTP, just set it */
1253 rtp_source_set_rtp_from (source, arrival->address);
1255 rtp_source_set_rtcp_from (source, arrival->address);
1259 /* FIXME: Log 3rd party collision somehow
1260 * Maybe should be done in upper layer, only the SDES can tell us
1261 * if its a collision or a loop
1264 /* This is sending with our ssrc, is it an address we already know */
1266 if (rtp_source_find_conflicting_address (source, arrival->address,
1267 arrival->current_time)) {
1268 /* Its a known conflict, its probably a loop, not a collision
1269 * lets just drop the incoming packet
1271 GST_DEBUG ("Our packets are being looped back to us, dropping");
1273 /* Its a new collision, lets change our SSRC */
1275 rtp_source_add_conflicting_address (source, arrival->address,
1276 arrival->current_time);
1278 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1279 on_ssrc_collision (sess, source);
1281 sess->change_ssrc = TRUE;
1283 rtp_session_schedule_bye_locked (sess, "SSRC Collision",
1284 arrival->current_time);
1292 find_source (RTPSession * sess, guint32 ssrc)
1294 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1295 GINT_TO_POINTER (ssrc));
1299 add_source (RTPSession * sess, RTPSource * src)
1301 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1302 GINT_TO_POINTER (src->ssrc), src);
1303 /* we have one more source now */
1304 sess->total_sources++;
1307 /* must be called with the session lock, the returned source needs to be
1308 * unreffed after usage. */
1310 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1311 RTPArrivalStats * arrival, gboolean rtp)
1315 source = find_source (sess, ssrc);
1316 if (source == NULL) {
1317 /* make new Source in probation and insert */
1318 source = rtp_source_new (ssrc);
1320 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1322 /* for RTP packets we need to set the source in probation. Receiving RTCP
1323 * packets of an SSRC, on the other hand, is a strong indication that we
1324 * are dealing with a valid source. */
1326 g_object_set (source, "probation", sess->probation, NULL);
1328 g_object_set (source, "probation", 0, NULL);
1330 /* store from address, if any */
1331 if (arrival->address) {
1333 rtp_source_set_rtp_from (source, arrival->address);
1335 rtp_source_set_rtcp_from (source, arrival->address);
1338 /* configure a callback on the source */
1339 rtp_source_set_callbacks (source, &callbacks, sess);
1341 add_source (sess, source);
1345 /* check for collision, this updates the address when not previously set */
1346 if (check_collision (sess, source, arrival, rtp)) {
1349 /* Receiving RTCP packets of an SSRC is a strong indication that we
1350 * are dealing with a valid source. */
1352 g_object_set (source, "probation", 0, NULL);
1354 /* update last activity */
1355 source->last_activity = arrival->current_time;
1357 source->last_rtp_activity = arrival->current_time;
1358 g_object_ref (source);
1364 * rtp_session_get_internal_source:
1365 * @sess: a #RTPSession
1367 * Get the internal #RTPSource of @sess.
1369 * Returns: The internal #RTPSource. g_object_unref() after usage.
1372 rtp_session_get_internal_source (RTPSession * sess)
1376 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1378 result = g_object_ref (sess->source);
1384 * rtp_session_set_internal_ssrc:
1385 * @sess: a #RTPSession
1388 * Set the SSRC of @sess to @ssrc.
1391 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1393 RTP_SESSION_LOCK (sess);
1394 if (ssrc != sess->source->ssrc) {
1395 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1396 GINT_TO_POINTER (sess->source->ssrc));
1398 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1399 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1400 * packets will timeout on the old SSRC, we could potentially schedule a
1401 * BYE RTCP for the old SSRC... */
1402 sess->source->ssrc = ssrc;
1403 rtp_source_reset (sess->source);
1405 /* rehash with the new SSRC */
1406 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1407 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1409 RTP_SESSION_UNLOCK (sess);
1411 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1415 * rtp_session_get_internal_ssrc:
1416 * @sess: a #RTPSession
1418 * Get the internal SSRC of @sess.
1420 * Returns: The SSRC of the session.
1423 rtp_session_get_internal_ssrc (RTPSession * sess)
1427 RTP_SESSION_LOCK (sess);
1428 ssrc = sess->source->ssrc;
1429 RTP_SESSION_UNLOCK (sess);
1435 * rtp_session_add_source:
1436 * @sess: a #RTPSession
1437 * @src: #RTPSource to add
1439 * Add @src to @session.
1441 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1442 * existed in the session.
1445 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1447 gboolean result = FALSE;
1450 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1451 g_return_val_if_fail (src != NULL, FALSE);
1453 RTP_SESSION_LOCK (sess);
1454 find = find_source (sess, src->ssrc);
1456 add_source (sess, src);
1459 RTP_SESSION_UNLOCK (sess);
1465 * rtp_session_get_num_sources:
1466 * @sess: an #RTPSession
1468 * Get the number of sources in @sess.
1470 * Returns: The number of sources in @sess.
1473 rtp_session_get_num_sources (RTPSession * sess)
1477 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1479 RTP_SESSION_LOCK (sess);
1480 result = sess->total_sources;
1481 RTP_SESSION_UNLOCK (sess);
1487 * rtp_session_get_num_active_sources:
1488 * @sess: an #RTPSession
1490 * Get the number of active sources in @sess. A source is considered active when
1491 * it has been validated and has not yet received a BYE RTCP message.
1493 * Returns: The number of active sources in @sess.
1496 rtp_session_get_num_active_sources (RTPSession * sess)
1500 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1502 RTP_SESSION_LOCK (sess);
1503 result = sess->stats.active_sources;
1504 RTP_SESSION_UNLOCK (sess);
1510 * rtp_session_get_source_by_ssrc:
1511 * @sess: an #RTPSession
1514 * Find the source with @ssrc in @sess.
1516 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1517 * g_object_unref() after usage.
1520 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1524 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1526 RTP_SESSION_LOCK (sess);
1527 result = find_source (sess, ssrc);
1529 g_object_ref (result);
1530 RTP_SESSION_UNLOCK (sess);
1535 /* should be called with the SESSION lock */
1537 rtp_session_create_new_ssrc (RTPSession * sess)
1542 ssrc = g_random_int ();
1544 /* see if it exists in the session, we're done if it doesn't */
1545 if (find_source (sess, ssrc) == NULL)
1553 * rtp_session_create_source:
1554 * @sess: an #RTPSession
1556 * Create an #RTPSource for use in @sess. This function will create a source
1557 * with an ssrc that is currently not used by any participants in the session.
1559 * Returns: an #RTPSource.
1562 rtp_session_create_source (RTPSession * sess)
1567 RTP_SESSION_LOCK (sess);
1568 ssrc = rtp_session_create_new_ssrc (sess);
1569 source = rtp_source_new (ssrc);
1570 rtp_source_set_callbacks (source, &callbacks, sess);
1571 /* we need an additional ref for the source in the hashtable */
1572 g_object_ref (source);
1573 add_source (sess, source);
1574 RTP_SESSION_UNLOCK (sess);
1579 /* update the RTPArrivalStats structure with the current time and other bits
1580 * about the current buffer we are handling.
1581 * This function is typically called when a validated packet is received.
1582 * This function should be called with the SESSION_LOCK
1585 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1586 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1587 GstClockTime running_time, guint64 ntpnstime)
1589 GstNetAddressMeta *meta;
1590 GstRTPBuffer rtpb = { NULL };
1592 /* get time of arrival */
1593 arrival->current_time = current_time;
1594 arrival->running_time = running_time;
1595 arrival->ntpnstime = ntpnstime;
1597 /* get packet size including header overhead */
1598 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1601 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1602 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1603 gst_rtp_buffer_unmap (&rtpb);
1605 arrival->payload_len = 0;
1608 /* for netbuffer we can store the IP address to check for collisions */
1609 meta = gst_buffer_get_net_address_meta (buffer);
1610 if (arrival->address)
1611 g_object_unref (arrival->address);
1613 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1615 arrival->address = NULL;
1620 clean_arrival_stats (RTPArrivalStats * arrival)
1622 if (arrival->address)
1623 g_object_unref (arrival->address);
1627 * rtp_session_process_rtp:
1628 * @sess: and #RTPSession
1629 * @buffer: an RTP buffer
1630 * @current_time: the current system time
1631 * @running_time: the running_time of @buffer
1633 * Process an RTP buffer in the session manager. This function takes ownership
1636 * Returns: a #GstFlowReturn.
1639 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1640 GstClockTime current_time, GstClockTime running_time)
1642 GstFlowReturn result;
1646 gboolean prevsender, prevactive;
1647 RTPArrivalStats arrival = { NULL, };
1651 GstRTPBuffer rtp = { NULL };
1653 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1654 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1656 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1657 goto invalid_packet;
1659 RTP_SESSION_LOCK (sess);
1660 /* ignore more RTP packets when we left the session */
1661 if (sess->source->received_bye)
1664 /* update arrival stats */
1665 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1668 /* get SSRC and look up in session database */
1669 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1670 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1674 /* copy available csrc for later */
1675 count = gst_rtp_buffer_get_csrc_count (&rtp);
1676 /* make sure to not overflow our array. An RTP buffer can maximally contain
1678 count = MIN (count, 16);
1680 for (i = 0; i < count; i++)
1681 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1683 gst_rtp_buffer_unmap (&rtp);
1685 prevsender = RTP_SOURCE_IS_SENDER (source);
1686 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1687 oldrate = source->bitrate;
1689 /* let source process the packet */
1690 result = rtp_source_process_rtp (source, buffer, &arrival);
1692 /* source became active */
1693 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1694 sess->stats.active_sources++;
1695 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1696 sess->stats.active_sources);
1697 on_ssrc_validated (sess, source);
1699 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1700 sess->stats.sender_sources++;
1701 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1702 sess->stats.sender_sources);
1704 if (oldrate != source->bitrate)
1705 sess->recalc_bandwidth = TRUE;
1708 on_new_ssrc (sess, source);
1710 if (source->validated) {
1713 /* for validated sources, we add the CSRCs as well */
1714 for (i = 0; i < count; i++) {
1716 RTPSource *csrc_src;
1721 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1726 GST_DEBUG ("created new CSRC: %08x", csrc);
1727 rtp_source_set_as_csrc (csrc_src);
1728 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1729 sess->stats.active_sources++;
1730 on_new_ssrc (sess, csrc_src);
1732 g_object_unref (csrc_src);
1735 g_object_unref (source);
1737 RTP_SESSION_UNLOCK (sess);
1739 clean_arrival_stats (&arrival);
1746 gst_buffer_unref (buffer);
1747 GST_DEBUG ("invalid RTP packet received");
1752 RTP_SESSION_UNLOCK (sess);
1753 gst_rtp_buffer_unmap (&rtp);
1754 gst_buffer_unref (buffer);
1755 GST_DEBUG ("ignoring RTP packet because we are leaving");
1760 RTP_SESSION_UNLOCK (sess);
1761 gst_rtp_buffer_unmap (&rtp);
1762 gst_buffer_unref (buffer);
1763 clean_arrival_stats (&arrival);
1764 GST_DEBUG ("ignoring packet because its collisioning");
1770 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1771 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1775 count = gst_rtcp_packet_get_rb_count (packet);
1776 for (i = 0; i < count; i++) {
1777 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1778 guint8 fractionlost;
1781 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1782 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1784 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1786 if (ssrc == sess->source->ssrc) {
1787 /* only deal with report blocks for our session, we update the stats of
1788 * the sender of the RTCP message. We could also compare our stats against
1789 * the other sender to see if we are better or worse. */
1790 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1791 packetslost, exthighestseq, jitter, lsr, dlsr);
1794 on_ssrc_active (sess, source);
1797 /* A Sender report contains statistics about how the sender is doing. This
1798 * includes timing informataion such as the relation between RTP and NTP
1799 * timestamps and the number of packets/bytes it sent to us.
1801 * In this report is also included a set of report blocks related to how this
1802 * sender is receiving data (in case we (or somebody else) is also sending stuff
1803 * to it). This info includes the packet loss, jitter and seqnum. It also
1804 * contains information to calculate the round trip time (LSR/DLSR).
1807 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1808 RTPArrivalStats * arrival, gboolean * do_sync)
1810 guint32 senderssrc, rtptime, packet_count, octet_count;
1813 gboolean created, prevsender;
1815 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1816 &packet_count, &octet_count);
1818 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1819 senderssrc, GST_TIME_ARGS (arrival->current_time));
1821 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1825 /* don't try to do lip-sync for sources that sent a BYE */
1826 if (rtp_source_received_bye (source))
1831 prevsender = RTP_SOURCE_IS_SENDER (source);
1833 /* first update the source */
1834 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1835 packet_count, octet_count);
1837 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1838 sess->stats.sender_sources++;
1839 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1840 sess->stats.sender_sources);
1844 on_new_ssrc (sess, source);
1846 rtp_session_process_rb (sess, source, packet, arrival);
1847 g_object_unref (source);
1850 /* A receiver report contains statistics about how a receiver is doing. It
1851 * includes stuff like packet loss, jitter and the seqnum it received last. It
1852 * also contains info to calculate the round trip time.
1854 * We are only interested in how the sender of this report is doing wrt to us.
1857 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1858 RTPArrivalStats * arrival)
1864 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1866 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1868 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1873 on_new_ssrc (sess, source);
1875 rtp_session_process_rb (sess, source, packet, arrival);
1876 g_object_unref (source);
1879 /* Get SDES items and store them in the SSRC */
1881 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1882 RTPArrivalStats * arrival)
1885 gboolean more_items, more_entries;
1887 items = gst_rtcp_packet_sdes_get_item_count (packet);
1888 GST_DEBUG ("got SDES packet with %d items", items);
1890 more_items = gst_rtcp_packet_sdes_first_item (packet);
1892 while (more_items) {
1894 gboolean changed, created, validated;
1898 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1900 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1904 /* find src, no probation when dealing with RTCP */
1905 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1909 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1911 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1913 while (more_entries) {
1914 GstRTCPSDESType type;
1920 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1922 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1925 if (type == GST_RTCP_SDES_PRIV) {
1926 name = g_strndup ((const gchar *) &data[1], data[0]);
1928 data += data[0] + 1;
1930 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1933 value = g_strndup ((const gchar *) data, len);
1935 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1940 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1944 /* takes ownership of sdes */
1945 changed = rtp_source_set_sdes_struct (source, sdes);
1947 validated = !RTP_SOURCE_IS_ACTIVE (source);
1948 source->validated = TRUE;
1951 on_new_ssrc (sess, source);
1953 /* source became active */
1955 sess->stats.active_sources++;
1956 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1957 sess->stats.active_sources);
1958 on_ssrc_validated (sess, source);
1962 on_ssrc_sdes (sess, source);
1964 g_object_unref (source);
1966 more_items = gst_rtcp_packet_sdes_next_item (packet);
1971 /* BYE is sent when a client leaves the session
1974 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1975 RTPArrivalStats * arrival)
1979 gboolean reconsider = FALSE;
1981 reason = gst_rtcp_packet_bye_get_reason (packet);
1982 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1984 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1985 for (i = 0; i < count; i++) {
1988 gboolean created, prevactive, prevsender;
1989 guint pmembers, members;
1991 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1992 GST_DEBUG ("SSRC: %08x", ssrc);
1994 if (ssrc == sess->source->ssrc)
1997 /* find src and mark bye, no probation when dealing with RTCP */
1998 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
2002 /* store time for when we need to time out this source */
2003 source->bye_time = arrival->current_time;
2005 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2006 prevsender = RTP_SOURCE_IS_SENDER (source);
2008 /* let the source handle the rest */
2009 rtp_source_process_bye (source, reason);
2011 pmembers = sess->stats.active_sources;
2013 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2014 sess->stats.active_sources--;
2015 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2016 sess->stats.active_sources);
2018 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2019 sess->stats.sender_sources--;
2020 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2021 sess->stats.sender_sources);
2023 members = sess->stats.active_sources;
2025 if (!sess->source->received_bye && members < pmembers) {
2026 /* some members went away since the previous timeout estimate.
2027 * Perform reverse reconsideration but only when we are not scheduling a
2029 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2030 arrival->current_time < sess->next_rtcp_check_time) {
2031 GstClockTime time_remaining;
2033 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2034 sess->next_rtcp_check_time =
2035 gst_util_uint64_scale (time_remaining, members, pmembers);
2037 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2038 GST_TIME_ARGS (sess->next_rtcp_check_time));
2040 sess->next_rtcp_check_time += arrival->current_time;
2042 /* mark pending reconsider. We only want to signal the reconsideration
2043 * once after we handled all the source in the bye packet */
2049 on_new_ssrc (sess, source);
2051 on_bye_ssrc (sess, source);
2053 g_object_unref (source);
2056 RTP_SESSION_UNLOCK (sess);
2057 /* notify app of reconsideration */
2058 if (sess->callbacks.reconsider)
2059 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2060 RTP_SESSION_LOCK (sess);
2066 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2067 RTPArrivalStats * arrival)
2069 GST_DEBUG ("received APP");
2073 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2074 gboolean fir, GstClockTime current_time)
2076 guint32 round_trip = 0;
2078 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2080 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2081 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2084 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2085 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2086 GST_DEBUG ("Ignoring %s request because one was send without one "
2087 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2088 fir ? "FIR" : "PLI",
2089 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2090 GST_TIME_ARGS (round_trip_in_ns));;
2095 sess->last_keyframe_request = current_time;
2097 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2098 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2099 sess->callbacks.request_key_unit);
2101 RTP_SESSION_UNLOCK (sess);
2102 sess->callbacks.request_key_unit (sess, fir,
2103 sess->request_key_unit_user_data);
2104 RTP_SESSION_LOCK (sess);
2110 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2111 guint32 media_ssrc, GstClockTime current_time)
2115 if (!sess->callbacks.request_key_unit)
2118 src = find_source (sess, sender_ssrc);
2122 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2126 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2127 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2132 gboolean our_request = FALSE;
2134 if (!sess->callbacks.request_key_unit)
2140 src = find_source (sess, sender_ssrc);
2142 /* Hack because Google fails to set the sender_ssrc correctly */
2143 if (!src && sender_ssrc == 1) {
2144 GHashTableIter iter;
2146 if (sess->stats.sender_sources >
2147 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2150 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2152 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2153 if (src != sess->source && rtp_source_is_sender (src))
2162 for (position = 0; position < fci_length; position += 8) {
2163 guint8 *data = fci_data + position;
2165 ssrc = GST_READ_UINT32_BE (data);
2167 if (ssrc == rtp_source_get_ssrc (sess->source)) {
2175 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2179 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2180 RTPArrivalStats * arrival, GstClockTime current_time)
2182 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2183 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2184 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2185 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2186 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2187 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2189 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2190 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2192 if (g_signal_has_handler_pending (sess,
2193 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2194 GstBuffer *fci_buffer = NULL;
2196 if (fci_length > 0) {
2197 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2198 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2200 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2203 RTP_SESSION_UNLOCK (sess);
2204 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2205 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2206 RTP_SESSION_LOCK (sess);
2209 gst_buffer_unref (fci_buffer);
2212 if (sess->rtcp_feedback_retention_window) {
2213 RTPSource *src = find_source (sess, media_ssrc);
2216 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2219 if (rtp_source_get_ssrc (sess->source) == media_ssrc ||
2220 /* PSFB FIR puts the media ssrc inside the FCI */
2221 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2223 case GST_RTCP_TYPE_PSFB:
2225 case GST_RTCP_PSFB_TYPE_PLI:
2226 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2229 case GST_RTCP_PSFB_TYPE_FIR:
2230 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2237 case GST_RTCP_TYPE_RTPFB:
2245 * rtp_session_process_rtcp:
2246 * @sess: and #RTPSession
2247 * @buffer: an RTCP buffer
2248 * @current_time: the current system time
2249 * @ntpnstime: the current NTP time in nanoseconds
2251 * Process an RTCP buffer in the session manager. This function takes ownership
2254 * Returns: a #GstFlowReturn.
2257 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2258 GstClockTime current_time, guint64 ntpnstime)
2260 GstRTCPPacket packet;
2261 gboolean more, is_bye = FALSE, do_sync = FALSE;
2262 RTPArrivalStats arrival = { NULL, };
2263 GstFlowReturn result = GST_FLOW_OK;
2264 GstRTCPBuffer rtcp = { NULL, };
2266 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2267 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2269 if (!gst_rtcp_buffer_validate (buffer))
2270 goto invalid_packet;
2272 GST_DEBUG ("received RTCP packet");
2274 RTP_SESSION_LOCK (sess);
2275 /* update arrival stats */
2276 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2282 /* start processing the compound packet */
2283 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2284 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2288 type = gst_rtcp_packet_get_type (&packet);
2290 /* when we are leaving the session, we should ignore all non-BYE messages */
2291 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
2292 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2297 case GST_RTCP_TYPE_SR:
2298 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2300 case GST_RTCP_TYPE_RR:
2301 rtp_session_process_rr (sess, &packet, &arrival);
2303 case GST_RTCP_TYPE_SDES:
2304 rtp_session_process_sdes (sess, &packet, &arrival);
2306 case GST_RTCP_TYPE_BYE:
2308 /* don't try to attempt lip-sync anymore for streams with a BYE */
2310 rtp_session_process_bye (sess, &packet, &arrival);
2312 case GST_RTCP_TYPE_APP:
2313 rtp_session_process_app (sess, &packet, &arrival);
2315 case GST_RTCP_TYPE_RTPFB:
2316 case GST_RTCP_TYPE_PSFB:
2317 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2320 GST_WARNING ("got unknown RTCP packet");
2324 more = gst_rtcp_packet_move_to_next (&packet);
2327 gst_rtcp_buffer_unmap (&rtcp);
2329 /* if we are scheduling a BYE, we only want to count bye packets, else we
2330 * count everything */
2331 if (sess->source->received_bye) {
2333 sess->stats.bye_members++;
2334 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2337 /* keep track of average packet size */
2338 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2340 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2341 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2342 RTP_SESSION_UNLOCK (sess);
2344 clean_arrival_stats (&arrival);
2346 /* notify caller of sr packets in the callback */
2347 if (do_sync && sess->callbacks.sync_rtcp) {
2348 /* make writable, we might want to change the buffer */
2349 buffer = gst_buffer_make_writable (buffer);
2351 result = sess->callbacks.sync_rtcp (sess, buffer,
2352 sess->sync_rtcp_user_data);
2354 gst_buffer_unref (buffer);
2361 GST_DEBUG ("invalid RTCP packet received");
2362 gst_buffer_unref (buffer);
2367 RTP_SESSION_UNLOCK (sess);
2368 gst_buffer_unref (buffer);
2369 clean_arrival_stats (&arrival);
2370 GST_DEBUG ("ignoring RTCP packet because we left");
2376 * rtp_session_update_send_caps:
2377 * @sess: an #RTPSession
2380 * Update the caps of the sender in the rtp session.
2383 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2385 g_return_if_fail (RTP_IS_SESSION (sess));
2386 g_return_if_fail (GST_IS_CAPS (caps));
2388 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2390 RTP_SESSION_LOCK (sess);
2391 rtp_source_update_caps (sess->source, caps);
2392 RTP_SESSION_UNLOCK (sess);
2396 * rtp_session_send_rtp:
2397 * @sess: an #RTPSession
2398 * @data: pointer to either an RTP buffer or a list of RTP buffers
2399 * @is_list: TRUE when @data is a buffer list
2400 * @current_time: the current system time
2401 * @running_time: the running time of @data
2403 * Send the RTP buffer in the session manager. This function takes ownership of
2406 * Returns: a #GstFlowReturn.
2409 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2410 GstClockTime current_time, GstClockTime running_time)
2412 GstFlowReturn result;
2414 gboolean prevsender;
2417 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2418 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2420 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2422 RTP_SESSION_LOCK (sess);
2423 source = sess->source;
2425 /* update last activity */
2426 source->last_rtp_activity = current_time;
2428 prevsender = RTP_SOURCE_IS_SENDER (source);
2429 oldrate = source->bitrate;
2431 /* we use our own source to send */
2432 result = rtp_source_send_rtp (source, data, is_list, running_time);
2434 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2435 sess->stats.sender_sources++;
2436 if (oldrate != source->bitrate)
2437 sess->recalc_bandwidth = TRUE;
2438 RTP_SESSION_UNLOCK (sess);
2444 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2446 *bandwidth += source->bitrate;
2449 /* must be called with session lock */
2451 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2454 GstClockTime result;
2456 /* recalculate bandwidth when it changed */
2457 if (sess->recalc_bandwidth) {
2460 if (sess->bandwidth > 0)
2461 bandwidth = sess->bandwidth;
2463 /* If it is <= 0, then try to estimate the actual bandwidth */
2464 bandwidth = sess->source->bitrate;
2466 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2467 (GHFunc) add_bitrates, &bandwidth);
2470 if (bandwidth < 8000)
2471 bandwidth = RTP_STATS_BANDWIDTH;
2473 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2474 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2476 sess->recalc_bandwidth = FALSE;
2479 if (sess->source->received_bye) {
2480 result = rtp_stats_calculate_bye_interval (&sess->stats);
2482 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2483 RTP_SOURCE_IS_SENDER (sess->source), first);
2486 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2487 GST_TIME_ARGS (result), first);
2489 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2490 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2492 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2497 /* Stop the current @sess and schedule a BYE message for the other members.
2498 * One must have the session lock to call this function
2500 static GstFlowReturn
2501 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2502 GstClockTime current_time)
2504 GstFlowReturn result = GST_FLOW_OK;
2506 GstClockTime interval;
2508 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2510 source = sess->source;
2512 /* ignore more BYEs */
2513 if (source->received_bye)
2516 /* we have BYE now */
2517 source->received_bye = TRUE;
2518 /* at least one member wants to send a BYE */
2519 g_free (sess->bye_reason);
2520 sess->bye_reason = g_strdup (reason);
2521 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2522 sess->stats.bye_members = 1;
2523 sess->first_rtcp = TRUE;
2524 sess->sent_bye = FALSE;
2525 sess->allow_early = TRUE;
2527 /* reschedule transmission */
2528 sess->last_rtcp_send_time = current_time;
2529 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2531 if (interval != GST_CLOCK_TIME_NONE)
2532 sess->next_rtcp_check_time = current_time + interval;
2534 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2536 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2537 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2539 RTP_SESSION_UNLOCK (sess);
2540 /* notify app of reconsideration */
2541 if (sess->callbacks.reconsider)
2542 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2543 RTP_SESSION_LOCK (sess);
2550 * rtp_session_schedule_bye:
2551 * @sess: an #RTPSession
2552 * @reason: a reason or NULL
2553 * @current_time: the current system time
2555 * Stop the current @sess and schedule a BYE message for the other members.
2557 * Returns: a #GstFlowReturn.
2560 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2561 GstClockTime current_time)
2563 GstFlowReturn result = GST_FLOW_OK;
2565 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2567 RTP_SESSION_LOCK (sess);
2568 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2569 RTP_SESSION_UNLOCK (sess);
2575 * rtp_session_next_timeout:
2576 * @sess: an #RTPSession
2577 * @current_time: the current system time
2579 * Get the next time we should perform session maintenance tasks.
2581 * Returns: a time when rtp_session_on_timeout() should be called with the
2582 * current system time.
2585 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2587 GstClockTime result, interval = 0;
2589 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2591 RTP_SESSION_LOCK (sess);
2593 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2594 result = sess->next_early_rtcp_time;
2598 result = sess->next_rtcp_check_time;
2600 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2601 ", next time: %" GST_TIME_FORMAT,
2602 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2604 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2605 GST_DEBUG ("take current time as base");
2606 /* our previous check time expired, start counting from the current time
2608 result = current_time;
2611 if (sess->source->received_bye) {
2612 if (sess->sent_bye) {
2613 GST_DEBUG ("we sent BYE already");
2614 interval = GST_CLOCK_TIME_NONE;
2615 } else if (sess->stats.active_sources >= 50) {
2616 GST_DEBUG ("reconsider BYE, more than 50 sources");
2617 /* reconsider BYE if members >= 50 */
2618 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2621 if (sess->first_rtcp) {
2622 GST_DEBUG ("first RTCP packet");
2623 /* we are called for the first time */
2624 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2625 } else if (sess->next_rtcp_check_time < current_time) {
2626 GST_DEBUG ("old check time expired, getting new timeout");
2627 /* get a new timeout when we need to */
2628 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2632 if (interval != GST_CLOCK_TIME_NONE)
2635 result = GST_CLOCK_TIME_NONE;
2637 sess->next_rtcp_check_time = result;
2641 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2642 ", next time: %" GST_TIME_FORMAT,
2643 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2644 RTP_SESSION_UNLOCK (sess);
2651 GstRTCPBuffer rtcpbuf;
2654 GstClockTime current_time;
2656 GstClockTime running_time;
2657 GstClockTime interval;
2658 GstRTCPPacket packet;
2662 gboolean may_suppress;
2666 session_start_rtcp (RTPSession * sess, ReportData * data)
2668 GstRTCPPacket *packet = &data->packet;
2669 RTPSource *own = sess->source;
2670 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2672 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2674 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2676 if (RTP_SOURCE_IS_SENDER (own)) {
2679 guint32 packet_count, octet_count;
2681 /* we are a sender, create SR */
2682 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2683 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2685 /* get latest stats */
2686 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2687 &ntptime, &rtptime, &packet_count, &octet_count);
2689 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2690 packet_count, octet_count);
2692 /* fill in sender report info */
2693 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2694 ntptime, rtptime, packet_count, octet_count);
2696 /* we are only receiver, create RR */
2697 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2698 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2699 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2703 /* construct a Sender or Receiver Report */
2705 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2707 RTPSession *sess = data->sess;
2708 GstRTCPPacket *packet = &data->packet;
2710 /* create a new buffer if needed */
2711 if (data->rtcp == NULL) {
2712 session_start_rtcp (sess, data);
2713 } else if (data->is_early) {
2714 /* Put a single RR or SR in minimal compound packets */
2717 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2718 /* only report about other sender sources */
2719 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2720 guint8 fractionlost;
2722 guint32 exthighestseq, jitter;
2726 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2727 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2729 /* store last generated RR packet */
2730 source->last_rr.is_valid = TRUE;
2731 source->last_rr.fractionlost = fractionlost;
2732 source->last_rr.packetslost = packetslost;
2733 source->last_rr.exthighestseq = exthighestseq;
2734 source->last_rr.jitter = jitter;
2735 source->last_rr.lsr = lsr;
2736 source->last_rr.dlsr = dlsr;
2738 /* packet is not yet filled, add report block for this source. */
2739 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2740 exthighestseq, jitter, lsr, dlsr);
2745 /* perform cleanup of sources that timed out */
2747 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2749 gboolean remove = FALSE;
2750 gboolean byetimeout = FALSE;
2751 gboolean sendertimeout = FALSE;
2752 gboolean is_sender, is_active;
2753 RTPSession *sess = data->sess;
2754 GstClockTime interval, binterval;
2757 is_sender = RTP_SOURCE_IS_SENDER (source);
2758 is_active = RTP_SOURCE_IS_ACTIVE (source);
2760 /* nothing to do when without RTCP */
2761 if (data->interval == GST_CLOCK_TIME_NONE)
2764 /* our own rtcp interval may have been forced low by secondary configuration,
2765 * while sender side may still operate with higher interval,
2766 * so do not just take our interval to decide on timing out sender,
2767 * but take (if data->interval <= 5 * GST_SECOND):
2768 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2769 * where sender_interval is difference between last 2 received RTCP reports
2771 if (data->interval >= 5 * GST_SECOND || (source == sess->source)) {
2772 binterval = data->interval;
2774 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2775 GST_TIME_ARGS (source->stats.prev_rtcptime),
2776 GST_TIME_ARGS (source->stats.last_rtcptime));
2777 /* if not received enough yet, fallback to larger default */
2778 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2779 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2781 binterval = 5 * GST_SECOND;
2782 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2784 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2785 GST_TIME_ARGS (binterval));
2787 /* check for our own source, we don't want to delete our own source. */
2788 if (!(source == sess->source)) {
2789 if (source->received_bye) {
2790 /* if we received a BYE from the source, remove the source after some
2792 if (data->current_time > source->bye_time &&
2793 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2794 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2799 /* sources that were inactive for more than 5 times the deterministic reporting
2800 * interval get timed out. the min timeout is 5 seconds. */
2801 /* mind old time that might pre-date last time going to PLAYING */
2802 btime = MAX (source->last_activity, sess->start_time);
2803 if (data->current_time > btime) {
2804 interval = MAX (binterval * 5, 5 * GST_SECOND);
2805 if (data->current_time - btime > interval) {
2806 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2807 source->ssrc, GST_TIME_ARGS (btime));
2813 /* senders that did not send for a long time become a receiver, this also
2814 * holds for our own source. */
2816 /* mind old time that might pre-date last time going to PLAYING */
2817 btime = MAX (source->last_rtp_activity, sess->start_time);
2818 if (data->current_time > btime) {
2819 interval = MAX (binterval * 2, 5 * GST_SECOND);
2820 if (data->current_time - btime > interval) {
2821 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2822 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2823 source->is_sender = FALSE;
2824 sess->stats.sender_sources--;
2825 sendertimeout = TRUE;
2831 sess->total_sources--;
2833 sess->stats.sender_sources--;
2835 sess->stats.active_sources--;
2838 on_bye_timeout (sess, source);
2840 on_timeout (sess, source);
2843 on_sender_timeout (sess, source);
2846 source->closing = remove;
2850 session_sdes (RTPSession * sess, ReportData * data)
2852 GstRTCPPacket *packet = &data->packet;
2853 const GstStructure *sdes;
2855 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2857 /* add SDES packet */
2858 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2860 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2862 sdes = rtp_source_get_sdes_struct (sess->source);
2864 /* add all fields in the structure, the order is not important. */
2865 n_fields = gst_structure_n_fields (sdes);
2866 for (i = 0; i < n_fields; ++i) {
2869 GstRTCPSDESType type;
2871 field = gst_structure_nth_field_name (sdes, i);
2874 value = gst_structure_get_string (sdes, field);
2877 type = gst_rtcp_sdes_name_to_type (field);
2879 /* Early packets are minimal and only include the CNAME */
2880 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2883 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2884 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2885 (const guint8 *) value);
2886 } else if (type == GST_RTCP_SDES_PRIV) {
2892 /* don't accept entries that are too big */
2893 prefix_len = strlen (field);
2894 if (prefix_len > 255)
2896 value_len = strlen (value);
2897 if (value_len > 255)
2899 data_len = 1 + prefix_len + value_len;
2903 data[0] = prefix_len;
2904 memcpy (&data[1], field, prefix_len);
2905 memcpy (&data[1 + prefix_len], value, value_len);
2907 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2911 data->has_sdes = TRUE;
2914 /* schedule a BYE packet */
2916 session_bye (RTPSession * sess, ReportData * data)
2918 GstRTCPPacket *packet = &data->packet;
2919 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2922 session_start_rtcp (sess, data);
2925 session_sdes (sess, data);
2927 /* add a BYE packet */
2928 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
2929 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2930 if (sess->bye_reason)
2931 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2933 /* we have a BYE packet now */
2934 data->is_bye = TRUE;
2938 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2940 GstClockTime new_send_time, elapsed;
2942 if (data->is_early && sess->next_early_rtcp_time < current_time)
2945 /* no need to check yet */
2946 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
2947 sess->next_rtcp_check_time > current_time) {
2948 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2949 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2950 GST_TIME_ARGS (current_time));
2954 /* get elapsed time since we last reported */
2955 elapsed = current_time - sess->last_rtcp_send_time;
2957 new_send_time = data->interval;
2958 /* perform forward reconsideration */
2959 if (new_send_time != GST_CLOCK_TIME_NONE) {
2960 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
2962 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2963 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time),
2964 GST_TIME_ARGS (elapsed));
2966 new_send_time += sess->last_rtcp_send_time;
2969 /* check if reconsideration */
2970 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
2971 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2972 GST_TIME_ARGS (new_send_time));
2973 /* store new check time */
2974 sess->next_rtcp_check_time = new_send_time;
2980 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2982 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2983 GST_TIME_ARGS (new_send_time));
2985 sess->next_rtcp_check_time = new_send_time;
2986 if (new_send_time != GST_CLOCK_TIME_NONE) {
2987 sess->next_rtcp_check_time += current_time;
2989 /* Apply the rules from RFC 4585 section 3.5.3 */
2990 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
2991 GstClockTimeDiff T_rr_current_interval =
2992 g_random_double_range (0.5, 1.5) * sess->stats.min_interval;
2994 /* This will caused the RTCP to be suppressed if no FB packets are added */
2995 if (sess->last_rtcp_send_time + T_rr_current_interval >
2996 sess->next_rtcp_check_time) {
2997 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
2998 " last: %" GST_TIME_FORMAT
2999 " + T_rr_current_interval: %" GST_TIME_FORMAT
3000 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3001 GST_TIME_ARGS (sess->stats.min_interval),
3002 GST_TIME_ARGS (sess->last_rtcp_send_time),
3003 GST_TIME_ARGS (T_rr_current_interval),
3004 GST_TIME_ARGS (sess->next_rtcp_check_time));
3005 data->may_suppress = TRUE;
3014 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3016 g_hash_table_insert (hash_table, key, g_object_ref (source));
3020 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3022 return source->closing;
3026 * rtp_session_on_timeout:
3027 * @sess: an #RTPSession
3028 * @current_time: the current system time
3029 * @ntpnstime: the current NTP time in nanoseconds
3030 * @running_time: the current running_time of the pipeline
3032 * Perform maintenance actions after the timeout obtained with
3033 * rtp_session_next_timeout() expired.
3035 * This function will perform timeouts of receivers and senders, send a BYE
3036 * packet or generate RTCP packets with current session stats.
3038 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3039 * times, for each packet that should be processed.
3041 * Returns: a #GstFlowReturn.
3044 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3045 guint64 ntpnstime, GstClockTime running_time)
3047 GstFlowReturn result = GST_FLOW_OK;
3048 ReportData data = { GST_RTCP_BUFFER_INIT };
3050 GHashTable *table_copy;
3051 gboolean notify = FALSE;
3053 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3055 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3056 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3057 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3061 data.current_time = current_time;
3062 data.ntpnstime = ntpnstime;
3063 data.is_bye = FALSE;
3064 data.has_sdes = FALSE;
3065 data.may_suppress = FALSE;
3066 data.running_time = running_time;
3070 RTP_SESSION_LOCK (sess);
3071 /* get a new interval, we need this for various cleanups etc */
3072 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3074 /* Make a local copy of the hashtable. We need to do this because the
3075 * cleanup stage below releases the session lock. */
3076 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3077 (GDestroyNotify) g_object_unref);
3078 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3079 (GHFunc) clone_ssrcs_hashtable, table_copy);
3081 /* Clean up the session, mark the source for removing, this might release the
3083 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3084 g_hash_table_destroy (table_copy);
3086 /* Now remove the marked sources */
3087 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3088 (GHRFunc) remove_closing_sources, NULL);
3090 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3091 data.is_early = TRUE;
3093 data.is_early = FALSE;
3095 /* see if we need to generate SR or RR packets */
3096 if (is_rtcp_time (sess, current_time, &data)) {
3097 if (own->received_bye) {
3098 /* generate BYE instead */
3099 GST_DEBUG ("generating BYE message");
3100 session_bye (sess, &data);
3101 sess->sent_bye = TRUE;
3103 /* loop over all known sources and do something */
3104 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3105 (GHFunc) session_report_blocks, &data);
3110 /* we keep track of the last report time in order to timeout inactive
3111 * receivers or senders */
3112 if (!data.is_early && !data.may_suppress)
3113 sess->last_rtcp_send_time = data.current_time;
3114 sess->first_rtcp = FALSE;
3115 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3117 /* add SDES for this source when not already added */
3119 session_sdes (sess, &data);
3122 /* check for outdated collisions */
3123 GST_DEBUG ("Timing out collisions");
3124 rtp_source_timeout (sess->source, current_time,
3125 /* "a relatively long time" -- RFC 3550 section 8.2 */
3126 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
3127 running_time - sess->rtcp_feedback_retention_window);
3129 if (sess->change_ssrc) {
3130 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
3131 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3132 GINT_TO_POINTER (own->ssrc));
3134 own->ssrc = rtp_session_create_new_ssrc (sess);
3135 rtp_source_reset (own);
3137 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3138 GINT_TO_POINTER (own->ssrc), own);
3140 g_free (sess->bye_reason);
3141 sess->bye_reason = NULL;
3142 sess->sent_bye = FALSE;
3143 sess->change_ssrc = FALSE;
3145 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
3148 sess->allow_early = TRUE;
3150 RTP_SESSION_UNLOCK (sess);
3153 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3155 /* push out the RTCP packet */
3157 gboolean do_not_suppress;
3159 gst_rtcp_buffer_unmap (&data.rtcpbuf);
3161 /* Give the user a change to add its own packet */
3162 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3163 data.rtcp, data.is_early, &do_not_suppress);
3165 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3168 packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len;
3170 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3171 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3172 sess->stats.avg_rtcp_packet_size, packet_size);
3174 sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye,
3175 sess->send_rtcp_user_data);
3177 GST_DEBUG ("freeing packet callback: %p"
3178 " do_not_suppress: %d may_suppress: %d",
3179 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3180 gst_buffer_unref (data.rtcp);
3188 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3189 GstClockTimeDiff max_delay)
3191 GstClockTime T_dither_max;
3193 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3195 RTP_SESSION_LOCK (sess);
3197 /* Check if already requested */
3198 /* RFC 4585 section 3.5.2 step 2 */
3199 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3202 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time))
3205 /* Ignore the request a scheduled packet will be in time anyway */
3206 if (current_time + max_delay > sess->next_rtcp_check_time)
3209 /* RFC 4585 section 3.5.2 step 2b */
3210 /* If the total sources is <=2, then there is only us and one peer */
3211 if (sess->total_sources <= 2) {
3214 /* Divide by 2 because l = 0.5 */
3215 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3219 /* RFC 4585 section 3.5.2 step 3 */
3220 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3223 /* RFC 4585 section 3.5.2 step 4
3224 * Don't send if allow_early is FALSE, but not if we are in
3225 * immediate mode, meaning we are part of a group of at most the
3226 * application-specific threshold.
3228 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3229 sess->allow_early == FALSE)
3233 /* Schedule an early transmission later */
3234 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3237 /* If no dithering, schedule it for NOW */
3238 sess->next_early_rtcp_time = current_time;
3241 RTP_SESSION_UNLOCK (sess);
3243 /* notify app of need to send packet early
3244 * and therefore of timeout change */
3245 if (sess->callbacks.reconsider)
3246 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3252 RTP_SESSION_UNLOCK (sess);
3256 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3257 gboolean fir, gint count)
3259 RTPSource *src = find_source (sess, ssrc);
3265 src->send_pli = FALSE;
3266 src->send_fir = TRUE;
3268 if (count == -1 || count != src->last_fir_count)
3269 src->current_send_fir_seqnum++;
3270 src->last_fir_count = count;
3271 } else if (!src->send_fir) {
3272 src->send_pli = TRUE;
3275 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3281 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3283 GstRTCPPacket packet;
3284 GstRTCPBuffer rtcp = { NULL, };
3285 gboolean ret = FALSE;
3287 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3289 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3290 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3291 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3295 gst_rtcp_buffer_unmap (&rtcp);
3301 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3304 gboolean ret = FALSE;
3305 GHashTableIter iter;
3306 gpointer key, value;
3307 gboolean started_fir = FALSE;
3308 GstRTCPPacket fir_rtcppacket;
3309 GstRTCPBuffer rtcp = { NULL, };
3311 RTP_SESSION_LOCK (sess);
3313 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3315 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3316 while (g_hash_table_iter_next (&iter, &key, &value)) {
3317 guint media_ssrc = GPOINTER_TO_UINT (key);
3318 RTPSource *media_src = value;
3321 if (media_src->send_fir) {
3323 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3326 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3327 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket,
3328 rtp_source_get_ssrc (sess->source));
3329 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3331 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3332 gst_rtcp_packet_remove (&fir_rtcppacket);
3338 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3339 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3343 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3344 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3346 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3348 fci_data[0] = media_src->current_send_fir_seqnum;
3349 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3350 media_src->send_fir = FALSE;
3354 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3355 while (g_hash_table_iter_next (&iter, &key, &value)) {
3356 guint media_ssrc = GPOINTER_TO_UINT (key);
3357 RTPSource *media_src = value;
3358 GstRTCPPacket pli_rtcppacket;
3360 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3361 has_pli_compare_func, NULL)) {
3362 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3364 /* Break because the packet is full, will put next request in a
3367 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3368 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket,
3369 rtp_source_get_ssrc (sess->source));
3370 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3373 media_src->send_pli = FALSE;
3375 gst_rtcp_buffer_unmap (&rtcp);
3377 RTP_SESSION_UNLOCK (sess);
3383 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3387 if (!sess->callbacks.send_rtcp)
3390 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3392 rtp_session_request_early_rtcp (sess, now, max_delay);