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 "rtpsession.h"
33 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
34 #define GST_CAT_DEFAULT rtp_session_debug
36 /* signals and args */
39 SIGNAL_GET_SOURCE_BY_SSRC,
41 SIGNAL_ON_SSRC_COLLISION,
42 SIGNAL_ON_SSRC_VALIDATED,
43 SIGNAL_ON_SSRC_ACTIVE,
46 SIGNAL_ON_BYE_TIMEOUT,
48 SIGNAL_ON_SENDER_TIMEOUT,
49 SIGNAL_ON_SENDING_RTCP,
50 SIGNAL_ON_FEEDBACK_RTCP,
55 #define DEFAULT_INTERNAL_SOURCE NULL
56 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
57 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
58 #define DEFAULT_RTCP_RR_BANDWIDTH -1
59 #define DEFAULT_RTCP_RS_BANDWIDTH -1
60 #define DEFAULT_RTCP_MTU 1400
61 #define DEFAULT_SDES NULL
62 #define DEFAULT_NUM_SOURCES 0
63 #define DEFAULT_NUM_ACTIVE_SOURCES 0
64 #define DEFAULT_SOURCES NULL
65 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
66 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
67 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
68 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
77 PROP_RTCP_RR_BANDWIDTH,
78 PROP_RTCP_RS_BANDWIDTH,
82 PROP_NUM_ACTIVE_SOURCES,
85 PROP_RTCP_MIN_INTERVAL,
86 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
87 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 void rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay);
117 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
119 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
121 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
122 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
123 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
124 static RTPSource *obtain_internal_source (RTPSession * sess,
125 guint32 ssrc, gboolean * created);
126 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
127 GstClockTime current_time);
128 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
129 gboolean deterministic, gboolean first);
132 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
133 const GValue * handler_return, gpointer data)
135 if (g_value_get_boolean (handler_return))
136 g_value_set_boolean (return_accu, TRUE);
142 rtp_session_class_init (RTPSessionClass * klass)
144 GObjectClass *gobject_class;
146 gobject_class = (GObjectClass *) klass;
148 gobject_class->finalize = rtp_session_finalize;
149 gobject_class->set_property = rtp_session_set_property;
150 gobject_class->get_property = rtp_session_get_property;
153 * RTPSession::get-source-by-ssrc:
154 * @session: the object which received the signal
155 * @ssrc: the SSRC of the RTPSource
157 * Request the #RTPSource object with SSRC @ssrc in @session.
159 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
160 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
161 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
162 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
163 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
166 * RTPSession::on-new-ssrc:
167 * @session: the object which received the signal
168 * @src: the new RTPSource
170 * Notify of a new SSRC that entered @session.
172 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
173 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
174 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
175 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
178 * RTPSession::on-ssrc-collision:
179 * @session: the object which received the signal
180 * @src: the #RTPSource that caused a collision
182 * Notify when we have an SSRC collision
184 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
185 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
186 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
187 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
190 * RTPSession::on-ssrc-validated:
191 * @session: the object which received the signal
192 * @src: the new validated RTPSource
194 * Notify of a new SSRC that became validated.
196 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
197 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
198 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
199 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
202 * RTPSession::on-ssrc-active:
203 * @session: the object which received the signal
204 * @src: the active RTPSource
206 * Notify of a SSRC that is active, i.e., sending RTCP.
208 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
209 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
210 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
211 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
214 * RTPSession::on-ssrc-sdes:
215 * @session: the object which received the signal
216 * @src: the RTPSource
218 * Notify that a new SDES was received for SSRC.
220 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
221 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
222 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
223 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
226 * RTPSession::on-bye-ssrc:
227 * @session: the object which received the signal
228 * @src: the RTPSource that went away
230 * Notify of an SSRC that became inactive because of a BYE packet.
232 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
233 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
234 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
235 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
238 * RTPSession::on-bye-timeout:
239 * @session: the object which received the signal
240 * @src: the RTPSource that timed out
242 * Notify of an SSRC that has timed out because of BYE
244 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
245 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
246 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
247 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
250 * RTPSession::on-timeout:
251 * @session: the object which received the signal
252 * @src: the RTPSource that timed out
254 * Notify of an SSRC that has timed out
256 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
257 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
258 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
259 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
262 * RTPSession::on-sender-timeout:
263 * @session: the object which received the signal
264 * @src: the RTPSource that timed out
266 * Notify of an SSRC that was a sender but timed out and became a receiver.
268 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
269 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
270 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
271 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
275 * RTPSession::on-sending-rtcp
276 * @session: the object which received the signal
277 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
278 * @early: %TRUE if the packet is early, %FALSE if it is regular
280 * This signal is emitted before sending an RTCP packet, it can be used
281 * to add extra RTCP Packets.
283 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
284 * if suppressing it is acceptable
286 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
287 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
288 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
289 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
290 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
293 * RTPSession::on-feedback-rtcp:
294 * @session: the object which received the signal
295 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
296 * %GST_RTCP_TYPE_RTPFB
297 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
298 * @sender_ssrc: The SSRC of the sender
299 * @media_ssrc: The SSRC of the media this refers to
300 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
303 * Notify that a RTCP feedback packet has been received
305 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
306 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
307 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
308 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
309 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
312 * RTPSession::send-rtcp:
313 * @session: the object which received the signal
314 * @max_delay: The maximum delay after which the feedback will not be useful
317 * Requests that the #RTPSession initiate a new RTCP packet as soon as
318 * possible within the requested delay.
320 rtp_session_signals[SIGNAL_SEND_RTCP] =
321 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
322 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
323 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
324 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
326 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
327 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
328 "The internal SSRC used for the session (deprecated)",
329 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
332 g_param_spec_object ("internal-source", "Internal Source",
333 "The internal source element of the session (deprecated)",
334 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
336 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
337 g_param_spec_double ("bandwidth", "Bandwidth",
338 "The bandwidth of the session (0 for auto-discover)",
339 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
340 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
343 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
344 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
345 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
346 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
348 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
349 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
350 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
351 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
352 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
354 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
355 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
356 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
357 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
358 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
361 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
362 "The maximum size of the RTCP packets",
363 16, G_MAXINT16, DEFAULT_RTCP_MTU,
364 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
366 g_object_class_install_property (gobject_class, PROP_SDES,
367 g_param_spec_boxed ("sdes", "SDES",
368 "The SDES items of this session",
369 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
371 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
372 g_param_spec_uint ("num-sources", "Num Sources",
373 "The number of sources in the session", 0, G_MAXUINT,
374 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
377 g_param_spec_uint ("num-active-sources", "Num Active Sources",
378 "The number of active sources in the session", 0, G_MAXUINT,
379 DEFAULT_NUM_ACTIVE_SOURCES,
380 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
384 * Get a GValue Array of all sources in the session.
387 * <title>Getting the #RTPSources of a session
394 * g_object_get (sess, "sources", &arr, NULL);
396 * for (i = 0; i < arr->n_values; i++) {
399 * val = g_value_array_get_nth (arr, i);
400 * source = g_value_get_object (val);
402 * g_value_array_free (arr);
407 g_object_class_install_property (gobject_class, PROP_SOURCES,
408 g_param_spec_boxed ("sources", "Sources",
409 "An array of all known sources in the session",
410 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
412 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
413 g_param_spec_boolean ("favor-new", "Favor new sources",
414 "Resolve SSRC conflict in favor of new sources", FALSE,
415 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
417 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
418 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
419 "Minimum interval between Regular RTCP packet (in ns)",
420 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
421 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
423 g_object_class_install_property (gobject_class,
424 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
425 g_param_spec_uint64 ("rtcp-feedback-retention-window",
426 "RTCP Feedback retention window",
427 "Duration during which RTCP Feedback packets are retained (in ns)",
428 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
429 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
431 g_object_class_install_property (gobject_class,
432 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
433 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
434 "RTCP Immediate Feedback threshold",
435 "The maximum number of members of a RTP session for which immediate"
437 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
438 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
440 g_object_class_install_property (gobject_class, PROP_PROBATION,
441 g_param_spec_uint ("probation", "Number of probations",
442 "Consecutive packet sequence numbers to accept the source",
443 0, G_MAXUINT, DEFAULT_PROBATION,
444 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
449 * Various session statistics. This property returns a GstStructure
450 * with name application/x-rtp-session-stats with the following fields:
452 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
453 * dropped (due to bandwidth constraints)
454 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
455 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
459 g_object_class_install_property (gobject_class, PROP_STATS,
460 g_param_spec_boxed ("stats", "Statistics",
461 "Various statistics", GST_TYPE_STRUCTURE,
462 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
464 klass->get_source_by_ssrc =
465 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
466 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
468 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
472 rtp_session_init (RTPSession * sess)
477 g_mutex_init (&sess->lock);
478 sess->key = g_random_int ();
482 for (i = 0; i < 32; i++) {
484 g_hash_table_new_full (NULL, NULL, NULL,
485 (GDestroyNotify) g_object_unref);
488 rtp_stats_init_defaults (&sess->stats);
489 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
490 rtp_stats_set_min_interval (&sess->stats,
491 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
493 sess->recalc_bandwidth = TRUE;
494 sess->bandwidth = DEFAULT_BANDWIDTH;
495 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
496 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
497 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
499 /* default UDP header length */
500 sess->header_len = 28;
501 sess->mtu = DEFAULT_RTCP_MTU;
503 sess->probation = DEFAULT_PROBATION;
505 /* some default SDES entries */
506 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
508 /* we do not want to leak details like the username or hostname here */
509 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
510 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
514 /* we do not want to leak the user's real name here */
515 str = g_strdup_printf ("Anon%u", g_random_int ());
516 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
520 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
522 /* this is the SSRC we suggest */
523 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
525 sess->first_rtcp = TRUE;
526 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
528 sess->allow_early = TRUE;
529 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
530 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
531 sess->rtcp_immediate_feedback_threshold =
532 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
534 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
536 sess->is_doing_ptp = TRUE;
540 rtp_session_finalize (GObject * object)
545 sess = RTP_SESSION_CAST (object);
547 gst_structure_free (sess->sdes);
549 for (i = 0; i < 32; i++)
550 g_hash_table_destroy (sess->ssrcs[i]);
552 g_mutex_clear (&sess->lock);
554 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
558 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
560 GValue value = { 0 };
562 g_value_init (&value, RTP_TYPE_SOURCE);
563 g_value_take_object (&value, source);
564 /* copies the value */
565 g_value_array_append (arr, &value);
569 rtp_session_create_sources (RTPSession * sess)
574 RTP_SESSION_LOCK (sess);
575 /* get number of elements in the table */
576 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
577 /* create the result value array */
578 res = g_value_array_new (size);
580 /* and copy all values into the array */
581 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
582 RTP_SESSION_UNLOCK (sess);
587 static GstStructure *
588 rtp_session_create_stats (RTPSession * sess)
592 s = gst_structure_new ("application/x-rtp-session-stats",
593 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
594 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
595 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
601 rtp_session_set_property (GObject * object, guint prop_id,
602 const GValue * value, GParamSpec * pspec)
606 sess = RTP_SESSION (object);
609 case PROP_INTERNAL_SSRC:
610 RTP_SESSION_LOCK (sess);
611 sess->suggested_ssrc = g_value_get_uint (value);
612 RTP_SESSION_UNLOCK (sess);
613 if (sess->callbacks.reconfigure)
614 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
617 RTP_SESSION_LOCK (sess);
618 sess->bandwidth = g_value_get_double (value);
619 sess->recalc_bandwidth = TRUE;
620 RTP_SESSION_UNLOCK (sess);
622 case PROP_RTCP_FRACTION:
623 RTP_SESSION_LOCK (sess);
624 sess->rtcp_bandwidth = g_value_get_double (value);
625 sess->recalc_bandwidth = TRUE;
626 RTP_SESSION_UNLOCK (sess);
628 case PROP_RTCP_RR_BANDWIDTH:
629 RTP_SESSION_LOCK (sess);
630 sess->rtcp_rr_bandwidth = g_value_get_int (value);
631 sess->recalc_bandwidth = TRUE;
632 RTP_SESSION_UNLOCK (sess);
634 case PROP_RTCP_RS_BANDWIDTH:
635 RTP_SESSION_LOCK (sess);
636 sess->rtcp_rs_bandwidth = g_value_get_int (value);
637 sess->recalc_bandwidth = TRUE;
638 RTP_SESSION_UNLOCK (sess);
641 sess->mtu = g_value_get_uint (value);
644 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
647 sess->favor_new = g_value_get_boolean (value);
649 case PROP_RTCP_MIN_INTERVAL:
650 rtp_stats_set_min_interval (&sess->stats,
651 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
652 /* trigger reconsideration */
653 RTP_SESSION_LOCK (sess);
654 sess->next_rtcp_check_time = 0;
655 RTP_SESSION_UNLOCK (sess);
656 if (sess->callbacks.reconsider)
657 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
659 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
660 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
663 sess->probation = g_value_get_uint (value);
666 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
672 rtp_session_get_property (GObject * object, guint prop_id,
673 GValue * value, GParamSpec * pspec)
677 sess = RTP_SESSION (object);
680 case PROP_INTERNAL_SSRC:
681 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
683 case PROP_INTERNAL_SOURCE:
684 /* FIXME, return a random source */
685 g_value_set_object (value, NULL);
688 g_value_set_double (value, sess->bandwidth);
690 case PROP_RTCP_FRACTION:
691 g_value_set_double (value, sess->rtcp_bandwidth);
693 case PROP_RTCP_RR_BANDWIDTH:
694 g_value_set_int (value, sess->rtcp_rr_bandwidth);
696 case PROP_RTCP_RS_BANDWIDTH:
697 g_value_set_int (value, sess->rtcp_rs_bandwidth);
700 g_value_set_uint (value, sess->mtu);
703 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
705 case PROP_NUM_SOURCES:
706 g_value_set_uint (value, rtp_session_get_num_sources (sess));
708 case PROP_NUM_ACTIVE_SOURCES:
709 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
712 g_value_take_boxed (value, rtp_session_create_sources (sess));
715 g_value_set_boolean (value, sess->favor_new);
717 case PROP_RTCP_MIN_INTERVAL:
718 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
720 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
721 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
724 g_value_set_uint (value, sess->probation);
727 g_value_take_boxed (value, rtp_session_create_stats (sess));
730 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
736 on_new_ssrc (RTPSession * sess, RTPSource * source)
738 g_object_ref (source);
739 RTP_SESSION_UNLOCK (sess);
740 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
741 RTP_SESSION_LOCK (sess);
742 g_object_unref (source);
746 on_ssrc_collision (RTPSession * sess, RTPSource * source)
748 g_object_ref (source);
749 RTP_SESSION_UNLOCK (sess);
750 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
752 RTP_SESSION_LOCK (sess);
753 g_object_unref (source);
757 on_ssrc_validated (RTPSession * sess, RTPSource * source)
759 g_object_ref (source);
760 RTP_SESSION_UNLOCK (sess);
761 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
763 RTP_SESSION_LOCK (sess);
764 g_object_unref (source);
768 on_ssrc_active (RTPSession * sess, RTPSource * source)
770 g_object_ref (source);
771 RTP_SESSION_UNLOCK (sess);
772 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
773 RTP_SESSION_LOCK (sess);
774 g_object_unref (source);
778 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
780 g_object_ref (source);
781 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
782 RTP_SESSION_UNLOCK (sess);
783 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
784 RTP_SESSION_LOCK (sess);
785 g_object_unref (source);
789 on_bye_ssrc (RTPSession * sess, RTPSource * source)
791 g_object_ref (source);
792 RTP_SESSION_UNLOCK (sess);
793 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
794 RTP_SESSION_LOCK (sess);
795 g_object_unref (source);
799 on_bye_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_BYE_TIMEOUT], 0, source);
804 RTP_SESSION_LOCK (sess);
805 g_object_unref (source);
809 on_timeout (RTPSession * sess, RTPSource * source)
811 g_object_ref (source);
812 RTP_SESSION_UNLOCK (sess);
813 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
814 RTP_SESSION_LOCK (sess);
815 g_object_unref (source);
819 on_sender_timeout (RTPSession * sess, RTPSource * source)
821 g_object_ref (source);
822 RTP_SESSION_UNLOCK (sess);
823 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
825 RTP_SESSION_LOCK (sess);
826 g_object_unref (source);
832 * Create a new session object.
834 * Returns: a new #RTPSession. g_object_unref() after usage.
837 rtp_session_new (void)
841 sess = g_object_new (RTP_TYPE_SESSION, NULL);
847 * rtp_session_set_callbacks:
848 * @sess: an #RTPSession
849 * @callbacks: callbacks to configure
850 * @user_data: user data passed in the callbacks
852 * Configure a set of callbacks to be notified of actions.
855 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
858 g_return_if_fail (RTP_IS_SESSION (sess));
860 if (callbacks->process_rtp) {
861 sess->callbacks.process_rtp = callbacks->process_rtp;
862 sess->process_rtp_user_data = user_data;
864 if (callbacks->send_rtp) {
865 sess->callbacks.send_rtp = callbacks->send_rtp;
866 sess->send_rtp_user_data = user_data;
868 if (callbacks->send_rtcp) {
869 sess->callbacks.send_rtcp = callbacks->send_rtcp;
870 sess->send_rtcp_user_data = user_data;
872 if (callbacks->sync_rtcp) {
873 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
874 sess->sync_rtcp_user_data = user_data;
876 if (callbacks->clock_rate) {
877 sess->callbacks.clock_rate = callbacks->clock_rate;
878 sess->clock_rate_user_data = user_data;
880 if (callbacks->reconsider) {
881 sess->callbacks.reconsider = callbacks->reconsider;
882 sess->reconsider_user_data = user_data;
884 if (callbacks->request_key_unit) {
885 sess->callbacks.request_key_unit = callbacks->request_key_unit;
886 sess->request_key_unit_user_data = user_data;
888 if (callbacks->request_time) {
889 sess->callbacks.request_time = callbacks->request_time;
890 sess->request_time_user_data = user_data;
892 if (callbacks->notify_nack) {
893 sess->callbacks.notify_nack = callbacks->notify_nack;
894 sess->notify_nack_user_data = user_data;
896 if (callbacks->reconfigure) {
897 sess->callbacks.reconfigure = callbacks->reconfigure;
898 sess->reconfigure_user_data = user_data;
903 * rtp_session_set_process_rtp_callback:
904 * @sess: an #RTPSession
905 * @callback: callback to set
906 * @user_data: user data passed in the callback
908 * Configure only the process_rtp callback to be notified of the process_rtp action.
911 rtp_session_set_process_rtp_callback (RTPSession * sess,
912 RTPSessionProcessRTP callback, gpointer user_data)
914 g_return_if_fail (RTP_IS_SESSION (sess));
916 sess->callbacks.process_rtp = callback;
917 sess->process_rtp_user_data = user_data;
921 * rtp_session_set_send_rtp_callback:
922 * @sess: an #RTPSession
923 * @callback: callback to set
924 * @user_data: user data passed in the callback
926 * Configure only the send_rtp callback to be notified of the send_rtp action.
929 rtp_session_set_send_rtp_callback (RTPSession * sess,
930 RTPSessionSendRTP callback, gpointer user_data)
932 g_return_if_fail (RTP_IS_SESSION (sess));
934 sess->callbacks.send_rtp = callback;
935 sess->send_rtp_user_data = user_data;
939 * rtp_session_set_send_rtcp_callback:
940 * @sess: an #RTPSession
941 * @callback: callback to set
942 * @user_data: user data passed in the callback
944 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
947 rtp_session_set_send_rtcp_callback (RTPSession * sess,
948 RTPSessionSendRTCP callback, gpointer user_data)
950 g_return_if_fail (RTP_IS_SESSION (sess));
952 sess->callbacks.send_rtcp = callback;
953 sess->send_rtcp_user_data = user_data;
957 * rtp_session_set_sync_rtcp_callback:
958 * @sess: an #RTPSession
959 * @callback: callback to set
960 * @user_data: user data passed in the callback
962 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
965 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
966 RTPSessionSyncRTCP callback, gpointer user_data)
968 g_return_if_fail (RTP_IS_SESSION (sess));
970 sess->callbacks.sync_rtcp = callback;
971 sess->sync_rtcp_user_data = user_data;
975 * rtp_session_set_clock_rate_callback:
976 * @sess: an #RTPSession
977 * @callback: callback to set
978 * @user_data: user data passed in the callback
980 * Configure only the clock_rate callback to be notified of the clock_rate action.
983 rtp_session_set_clock_rate_callback (RTPSession * sess,
984 RTPSessionClockRate callback, gpointer user_data)
986 g_return_if_fail (RTP_IS_SESSION (sess));
988 sess->callbacks.clock_rate = callback;
989 sess->clock_rate_user_data = user_data;
993 * rtp_session_set_reconsider_callback:
994 * @sess: an #RTPSession
995 * @callback: callback to set
996 * @user_data: user data passed in the callback
998 * Configure only the reconsider callback to be notified of the reconsider action.
1001 rtp_session_set_reconsider_callback (RTPSession * sess,
1002 RTPSessionReconsider callback, gpointer user_data)
1004 g_return_if_fail (RTP_IS_SESSION (sess));
1006 sess->callbacks.reconsider = callback;
1007 sess->reconsider_user_data = user_data;
1011 * rtp_session_set_request_time_callback:
1012 * @sess: an #RTPSession
1013 * @callback: callback to set
1014 * @user_data: user data passed in the callback
1016 * Configure only the request_time callback
1019 rtp_session_set_request_time_callback (RTPSession * sess,
1020 RTPSessionRequestTime callback, gpointer user_data)
1022 g_return_if_fail (RTP_IS_SESSION (sess));
1024 sess->callbacks.request_time = callback;
1025 sess->request_time_user_data = user_data;
1029 * rtp_session_set_bandwidth:
1030 * @sess: an #RTPSession
1031 * @bandwidth: the bandwidth allocated
1033 * Set the session bandwidth in bytes per second.
1036 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1038 g_return_if_fail (RTP_IS_SESSION (sess));
1040 RTP_SESSION_LOCK (sess);
1041 sess->stats.bandwidth = bandwidth;
1042 RTP_SESSION_UNLOCK (sess);
1046 * rtp_session_get_bandwidth:
1047 * @sess: an #RTPSession
1049 * Get the session bandwidth.
1051 * Returns: the session bandwidth.
1054 rtp_session_get_bandwidth (RTPSession * sess)
1058 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1060 RTP_SESSION_LOCK (sess);
1061 result = sess->stats.bandwidth;
1062 RTP_SESSION_UNLOCK (sess);
1068 * rtp_session_set_rtcp_fraction:
1069 * @sess: an #RTPSession
1070 * @bandwidth: the RTCP bandwidth
1072 * Set the bandwidth in bytes per second that should be used for RTCP
1076 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1078 g_return_if_fail (RTP_IS_SESSION (sess));
1080 RTP_SESSION_LOCK (sess);
1081 sess->stats.rtcp_bandwidth = bandwidth;
1082 RTP_SESSION_UNLOCK (sess);
1086 * rtp_session_get_rtcp_fraction:
1087 * @sess: an #RTPSession
1089 * Get the session bandwidth used for RTCP.
1091 * Returns: The bandwidth used for RTCP messages.
1094 rtp_session_get_rtcp_fraction (RTPSession * sess)
1098 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1100 RTP_SESSION_LOCK (sess);
1101 result = sess->stats.rtcp_bandwidth;
1102 RTP_SESSION_UNLOCK (sess);
1108 * rtp_session_get_sdes_struct:
1109 * @sess: an #RTSPSession
1111 * Get the SDES data as a #GstStructure
1113 * Returns: a GstStructure with SDES items for @sess. This function returns a
1114 * copy of the SDES structure, use gst_structure_free() after usage.
1117 rtp_session_get_sdes_struct (RTPSession * sess)
1119 GstStructure *result = NULL;
1121 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1123 RTP_SESSION_LOCK (sess);
1125 result = gst_structure_copy (sess->sdes);
1126 RTP_SESSION_UNLOCK (sess);
1132 * rtp_session_set_sdes_struct:
1133 * @sess: an #RTSPSession
1134 * @sdes: a #GstStructure
1136 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1139 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1141 g_return_if_fail (sdes);
1142 g_return_if_fail (RTP_IS_SESSION (sess));
1144 RTP_SESSION_LOCK (sess);
1146 gst_structure_free (sess->sdes);
1147 sess->sdes = gst_structure_copy (sdes);
1148 RTP_SESSION_UNLOCK (sess);
1151 static GstFlowReturn
1152 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1154 GstFlowReturn result = GST_FLOW_OK;
1156 if (source->internal) {
1157 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1159 RTP_SESSION_UNLOCK (session);
1161 if (session->callbacks.send_rtp)
1163 session->callbacks.send_rtp (session, source, data,
1164 session->send_rtp_user_data);
1166 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1169 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1170 RTP_SESSION_UNLOCK (session);
1172 if (session->callbacks.process_rtp)
1174 session->callbacks.process_rtp (session, source,
1175 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1177 gst_buffer_unref (GST_BUFFER_CAST (data));
1179 RTP_SESSION_LOCK (session);
1185 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1189 RTP_SESSION_UNLOCK (session);
1191 if (session->callbacks.clock_rate)
1193 session->callbacks.clock_rate (session, pt,
1194 session->clock_rate_user_data);
1198 RTP_SESSION_LOCK (session);
1200 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1205 static RTPSourceCallbacks callbacks = {
1206 (RTPSourcePushRTP) source_push_rtp,
1207 (RTPSourceClockRate) source_clock_rate,
1211 check_collision (RTPSession * sess, RTPSource * source,
1212 RTPPacketInfo * pinfo, gboolean rtp)
1216 /* If we have no pinfo address, we can't do collision checking */
1217 if (!pinfo->address)
1220 ssrc = rtp_source_get_ssrc (source);
1222 if (!source->internal) {
1223 GSocketAddress *from;
1225 /* This is not our local source, but lets check if two remote
1228 from = source->rtp_from;
1230 from = source->rtcp_from;
1234 if (__g_socket_address_equal (from, pinfo->address)) {
1235 /* Address is the same */
1238 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1239 if (sess->favor_new) {
1240 if (rtp_source_find_conflicting_address (source,
1241 pinfo->address, pinfo->current_time)) {
1244 buf1 = __g_socket_address_to_string (pinfo->address);
1245 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1253 /* Current address is not a known conflict, lets assume this is
1254 * a new source. Save old address in possible conflict list
1256 rtp_source_add_conflicting_address (source, from,
1257 pinfo->current_time);
1259 buf1 = __g_socket_address_to_string (from);
1260 buf2 = __g_socket_address_to_string (pinfo->address);
1262 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1263 " saving old as known conflict", ssrc, buf1, buf2);
1266 rtp_source_set_rtp_from (source, pinfo->address);
1268 rtp_source_set_rtcp_from (source, pinfo->address);
1276 /* Don't need to save old addresses, we ignore new sources */
1281 /* We don't already have a from address for RTP, just set it */
1283 rtp_source_set_rtp_from (source, pinfo->address);
1285 rtp_source_set_rtcp_from (source, pinfo->address);
1289 /* FIXME: Log 3rd party collision somehow
1290 * Maybe should be done in upper layer, only the SDES can tell us
1291 * if its a collision or a loop
1294 /* This is sending with our ssrc, is it an address we already know */
1295 if (rtp_source_find_conflicting_address (source, pinfo->address,
1296 pinfo->current_time)) {
1297 /* Its a known conflict, its probably a loop, not a collision
1298 * lets just drop the incoming packet
1300 GST_DEBUG ("Our packets are being looped back to us, dropping");
1302 /* Its a new collision, lets change our SSRC */
1303 rtp_source_add_conflicting_address (source, pinfo->address,
1304 pinfo->current_time);
1306 GST_DEBUG ("Collision for SSRC %x", ssrc);
1307 /* mark the source BYE */
1308 rtp_source_mark_bye (source, "SSRC Collision");
1309 /* if we were suggesting this SSRC, change to something else */
1310 if (sess->suggested_ssrc == ssrc)
1311 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1313 on_ssrc_collision (sess, source);
1315 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1324 gboolean is_doing_ptp;
1325 GSocketAddress *new_addr;
1328 /* check if the two given ip addr are the same (do not care about the port) */
1330 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1333 g_inet_address_equal (g_inet_socket_address_get_address
1334 (G_INET_SOCKET_ADDRESS (a)),
1335 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1339 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1340 CompareAddrData * data)
1342 /* only compare ip addr of remote sources which are also not closing */
1343 if (!source->internal && !source->closing && source->rtp_from) {
1344 /* look for the first rtp source */
1345 if (!data->new_addr)
1346 data->new_addr = source->rtp_from;
1347 /* compare current ip addr with the first one */
1349 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1354 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1355 CompareAddrData * data)
1357 /* only compare ip addr of remote sources which are also not closing */
1358 if (!source->internal && !source->closing && source->rtcp_from) {
1359 /* look for the first rtcp source */
1360 if (!data->new_addr)
1361 data->new_addr = source->rtcp_from;
1363 /* compare current ip addr with the first one */
1364 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1368 /* loop over our non-internal source to know if the session
1369 * is doing point-to-point */
1371 session_update_ptp (RTPSession * sess)
1373 /* to know if the session is doing point to point, the ip addr
1374 * of each non-internal (=remotes) source have to be compared
1377 gboolean is_doing_rtp_ptp = FALSE;
1378 gboolean is_doing_rtcp_ptp = FALSE;
1379 CompareAddrData data;
1381 /* compare the first remote source's ip addr that receive rtp packets
1382 * with other remote rtp source.
1383 * it's enough because the session just needs to know if they are all
1386 data.is_doing_ptp = TRUE;
1387 data.new_addr = NULL;
1388 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1389 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1390 is_doing_rtp_ptp = data.is_doing_ptp;
1392 /* same but about rtcp */
1393 data.is_doing_ptp = TRUE;
1394 data.new_addr = NULL;
1395 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1396 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1397 is_doing_rtcp_ptp = data.is_doing_ptp;
1399 /* the session is doing point-to-point if all rtp remote have the same
1400 * ip addr and if all rtcp remote sources have the same ip addr */
1401 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1403 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1407 add_source (RTPSession * sess, RTPSource * src)
1409 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1410 GINT_TO_POINTER (src->ssrc), src);
1411 /* report the new source ASAP */
1412 src->generation = sess->generation;
1413 /* we have one more source now */
1414 sess->total_sources++;
1415 if (RTP_SOURCE_IS_ACTIVE (src))
1416 sess->stats.active_sources++;
1417 if (src->internal) {
1418 sess->stats.internal_sources++;
1419 if (sess->suggested_ssrc != src->ssrc)
1420 sess->suggested_ssrc = src->ssrc;
1423 /* update point-to-point status */
1425 session_update_ptp (sess);
1429 find_source (RTPSession * sess, guint32 ssrc)
1431 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1432 GINT_TO_POINTER (ssrc));
1435 /* must be called with the session lock, the returned source needs to be
1436 * unreffed after usage. */
1438 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1439 RTPPacketInfo * pinfo, gboolean rtp)
1443 source = find_source (sess, ssrc);
1444 if (source == NULL) {
1445 /* make new Source in probation and insert */
1446 source = rtp_source_new (ssrc);
1448 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1450 /* for RTP packets we need to set the source in probation. Receiving RTCP
1451 * packets of an SSRC, on the other hand, is a strong indication that we
1452 * are dealing with a valid source. */
1454 g_object_set (source, "probation", sess->probation, NULL);
1456 g_object_set (source, "probation", 0, NULL);
1458 /* store from address, if any */
1459 if (pinfo->address) {
1461 rtp_source_set_rtp_from (source, pinfo->address);
1463 rtp_source_set_rtcp_from (source, pinfo->address);
1466 /* configure a callback on the source */
1467 rtp_source_set_callbacks (source, &callbacks, sess);
1469 add_source (sess, source);
1473 /* check for collision, this updates the address when not previously set */
1474 if (check_collision (sess, source, pinfo, rtp)) {
1477 /* Receiving RTCP packets of an SSRC is a strong indication that we
1478 * are dealing with a valid source. */
1480 g_object_set (source, "probation", 0, NULL);
1482 /* update last activity */
1483 source->last_activity = pinfo->current_time;
1485 source->last_rtp_activity = pinfo->current_time;
1486 g_object_ref (source);
1491 /* must be called with the session lock, the returned source needs to be
1492 * unreffed after usage. */
1494 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1498 source = find_source (sess, ssrc);
1499 if (source == NULL) {
1500 /* make new internal Source and insert */
1501 source = rtp_source_new (ssrc);
1503 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1505 source->validated = TRUE;
1506 source->internal = TRUE;
1507 source->probation = FALSE;
1508 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1509 rtp_source_set_callbacks (source, &callbacks, sess);
1511 add_source (sess, source);
1516 g_object_ref (source);
1522 * rtp_session_suggest_ssrc:
1523 * @sess: a #RTPSession
1525 * Suggest an unused SSRC in @sess.
1527 * Returns: a free unused SSRC
1530 rtp_session_suggest_ssrc (RTPSession * sess)
1534 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1536 RTP_SESSION_LOCK (sess);
1537 result = sess->suggested_ssrc;
1538 RTP_SESSION_UNLOCK (sess);
1544 * rtp_session_add_source:
1545 * @sess: a #RTPSession
1546 * @src: #RTPSource to add
1548 * Add @src to @session.
1550 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1551 * existed in the session.
1554 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1556 gboolean result = FALSE;
1559 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1560 g_return_val_if_fail (src != NULL, FALSE);
1562 RTP_SESSION_LOCK (sess);
1563 find = find_source (sess, src->ssrc);
1565 add_source (sess, src);
1568 RTP_SESSION_UNLOCK (sess);
1574 * rtp_session_get_num_sources:
1575 * @sess: an #RTPSession
1577 * Get the number of sources in @sess.
1579 * Returns: The number of sources in @sess.
1582 rtp_session_get_num_sources (RTPSession * sess)
1586 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1588 RTP_SESSION_LOCK (sess);
1589 result = sess->total_sources;
1590 RTP_SESSION_UNLOCK (sess);
1596 * rtp_session_get_num_active_sources:
1597 * @sess: an #RTPSession
1599 * Get the number of active sources in @sess. A source is considered active when
1600 * it has been validated and has not yet received a BYE RTCP message.
1602 * Returns: The number of active sources in @sess.
1605 rtp_session_get_num_active_sources (RTPSession * sess)
1609 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1611 RTP_SESSION_LOCK (sess);
1612 result = sess->stats.active_sources;
1613 RTP_SESSION_UNLOCK (sess);
1619 * rtp_session_get_source_by_ssrc:
1620 * @sess: an #RTPSession
1623 * Find the source with @ssrc in @sess.
1625 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1626 * g_object_unref() after usage.
1629 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1633 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1635 RTP_SESSION_LOCK (sess);
1636 result = find_source (sess, ssrc);
1638 g_object_ref (result);
1639 RTP_SESSION_UNLOCK (sess);
1644 /* should be called with the SESSION lock */
1646 rtp_session_create_new_ssrc (RTPSession * sess)
1651 ssrc = g_random_int ();
1653 /* see if it exists in the session, we're done if it doesn't */
1654 if (find_source (sess, ssrc) == NULL)
1662 * rtp_session_create_source:
1663 * @sess: an #RTPSession
1665 * Create an #RTPSource for use in @sess. This function will create a source
1666 * with an ssrc that is currently not used by any participants in the session.
1668 * Returns: an #RTPSource.
1671 rtp_session_create_source (RTPSession * sess)
1676 RTP_SESSION_LOCK (sess);
1677 ssrc = rtp_session_create_new_ssrc (sess);
1678 source = rtp_source_new (ssrc);
1679 rtp_source_set_callbacks (source, &callbacks, sess);
1680 /* we need an additional ref for the source in the hashtable */
1681 g_object_ref (source);
1682 add_source (sess, source);
1683 RTP_SESSION_UNLOCK (sess);
1689 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1691 GstNetAddressMeta *meta;
1693 /* get packet size including header overhead */
1694 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1698 GstRTPBuffer rtp = { NULL };
1700 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1701 goto invalid_packet;
1703 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1707 /* only keep info for first buffer */
1708 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1709 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1710 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1711 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1712 /* copy available csrc */
1713 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1714 for (i = 0; i < pinfo->csrc_count; i++)
1715 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1717 gst_rtp_buffer_unmap (&rtp);
1721 /* for netbuffer we can store the IP address to check for collisions */
1722 meta = gst_buffer_get_net_address_meta (*buffer);
1724 g_object_unref (pinfo->address);
1726 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1728 pinfo->address = NULL;
1736 GST_DEBUG ("invalid RTP packet received");
1741 /* update the RTPPacketInfo structure with the current time and other bits
1742 * about the current buffer we are handling.
1743 * This function is typically called when a validated packet is received.
1744 * This function should be called with the SESSION_LOCK
1747 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1748 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1749 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1755 pinfo->is_list = is_list;
1757 pinfo->current_time = current_time;
1758 pinfo->running_time = running_time;
1759 pinfo->ntpnstime = ntpnstime;
1760 pinfo->header_len = sess->header_len;
1762 pinfo->payload_len = 0;
1766 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1768 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1771 GstBuffer *buffer = GST_BUFFER_CAST (data);
1772 res = update_packet (&buffer, 0, pinfo);
1778 clean_packet_info (RTPPacketInfo * pinfo)
1781 g_object_unref (pinfo->address);
1783 gst_mini_object_unref (pinfo->data);
1789 source_update_active (RTPSession * sess, RTPSource * source,
1790 gboolean prevactive)
1792 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1793 guint32 ssrc = source->ssrc;
1795 if (prevactive == active)
1799 sess->stats.active_sources++;
1800 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1801 sess->stats.active_sources);
1803 sess->stats.active_sources--;
1804 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1805 sess->stats.active_sources);
1811 source_update_sender (RTPSession * sess, RTPSource * source,
1812 gboolean prevsender)
1814 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1815 guint32 ssrc = source->ssrc;
1817 if (prevsender == sender)
1821 sess->stats.sender_sources++;
1822 if (source->internal)
1823 sess->stats.internal_sender_sources++;
1824 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1825 sess->stats.sender_sources);
1827 sess->stats.sender_sources--;
1828 if (source->internal)
1829 sess->stats.internal_sender_sources--;
1830 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1831 sess->stats.sender_sources);
1837 * rtp_session_process_rtp:
1838 * @sess: and #RTPSession
1839 * @buffer: an RTP buffer
1840 * @current_time: the current system time
1841 * @running_time: the running_time of @buffer
1843 * Process an RTP buffer in the session manager. This function takes ownership
1846 * Returns: a #GstFlowReturn.
1849 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1850 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1852 GstFlowReturn result;
1856 gboolean prevsender, prevactive;
1857 RTPPacketInfo pinfo = { 0, };
1860 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1861 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1863 RTP_SESSION_LOCK (sess);
1865 /* update pinfo stats */
1866 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1867 current_time, running_time, ntpnstime)) {
1868 GST_DEBUG ("invalid RTP packet received");
1869 RTP_SESSION_UNLOCK (sess);
1870 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1875 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1879 prevsender = RTP_SOURCE_IS_SENDER (source);
1880 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1881 oldrate = source->bitrate;
1883 /* let source process the packet */
1884 result = rtp_source_process_rtp (source, &pinfo);
1886 /* source became active */
1887 if (source_update_active (sess, source, prevactive))
1888 on_ssrc_validated (sess, source);
1890 source_update_sender (sess, source, prevsender);
1892 if (oldrate != source->bitrate)
1893 sess->recalc_bandwidth = TRUE;
1896 on_new_ssrc (sess, source);
1898 if (source->validated) {
1902 /* for validated sources, we add the CSRCs as well */
1903 for (i = 0; i < pinfo.csrc_count; i++) {
1905 RTPSource *csrc_src;
1907 csrc = pinfo.csrcs[i];
1910 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
1915 GST_DEBUG ("created new CSRC: %08x", csrc);
1916 rtp_source_set_as_csrc (csrc_src);
1917 source_update_active (sess, csrc_src, FALSE);
1918 on_new_ssrc (sess, csrc_src);
1920 g_object_unref (csrc_src);
1923 g_object_unref (source);
1925 RTP_SESSION_UNLOCK (sess);
1927 clean_packet_info (&pinfo);
1934 RTP_SESSION_UNLOCK (sess);
1935 clean_packet_info (&pinfo);
1936 GST_DEBUG ("ignoring packet because its collisioning");
1942 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1943 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
1947 count = gst_rtcp_packet_get_rb_count (packet);
1948 for (i = 0; i < count; i++) {
1949 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1950 guint8 fractionlost;
1954 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1955 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1957 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1959 /* find our own source */
1960 src = find_source (sess, ssrc);
1964 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
1965 /* only deal with report blocks for our session, we update the stats of
1966 * the sender of the RTCP message. We could also compare our stats against
1967 * the other sender to see if we are better or worse. */
1968 /* FIXME, need to keep track who the RB block is from */
1969 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
1970 packetslost, exthighestseq, jitter, lsr, dlsr);
1973 on_ssrc_active (sess, source);
1976 /* A Sender report contains statistics about how the sender is doing. This
1977 * includes timing informataion such as the relation between RTP and NTP
1978 * timestamps and the number of packets/bytes it sent to us.
1980 * In this report is also included a set of report blocks related to how this
1981 * sender is receiving data (in case we (or somebody else) is also sending stuff
1982 * to it). This info includes the packet loss, jitter and seqnum. It also
1983 * contains information to calculate the round trip time (LSR/DLSR).
1986 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1987 RTPPacketInfo * pinfo, gboolean * do_sync)
1989 guint32 senderssrc, rtptime, packet_count, octet_count;
1992 gboolean created, prevsender;
1994 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1995 &packet_count, &octet_count);
1997 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1998 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2000 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2004 /* skip non-bye packets for sources that are marked BYE */
2005 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2008 /* don't try to do lip-sync for sources that sent a BYE */
2009 if (RTP_SOURCE_IS_MARKED_BYE (source))
2014 prevsender = RTP_SOURCE_IS_SENDER (source);
2016 /* first update the source */
2017 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2018 packet_count, octet_count);
2020 source_update_sender (sess, source, prevsender);
2023 on_new_ssrc (sess, source);
2025 rtp_session_process_rb (sess, source, packet, pinfo);
2028 g_object_unref (source);
2031 /* A receiver report contains statistics about how a receiver is doing. It
2032 * includes stuff like packet loss, jitter and the seqnum it received last. It
2033 * also contains info to calculate the round trip time.
2035 * We are only interested in how the sender of this report is doing wrt to us.
2038 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2039 RTPPacketInfo * pinfo)
2045 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2047 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2049 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2053 /* skip non-bye packets for sources that are marked BYE */
2054 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2058 on_new_ssrc (sess, source);
2060 rtp_session_process_rb (sess, source, packet, pinfo);
2063 g_object_unref (source);
2066 /* Get SDES items and store them in the SSRC */
2068 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2069 RTPPacketInfo * pinfo)
2072 gboolean more_items, more_entries;
2074 items = gst_rtcp_packet_sdes_get_item_count (packet);
2075 GST_DEBUG ("got SDES packet with %d items", items);
2077 more_items = gst_rtcp_packet_sdes_first_item (packet);
2079 while (more_items) {
2081 gboolean changed, created, prevactive;
2085 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2087 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2091 /* find src, no probation when dealing with RTCP */
2092 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2096 /* skip non-bye packets for sources that are marked BYE */
2097 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2100 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2102 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2104 while (more_entries) {
2105 GstRTCPSDESType type;
2111 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2113 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2116 if (type == GST_RTCP_SDES_PRIV) {
2117 name = g_strndup ((const gchar *) &data[1], data[0]);
2119 data += data[0] + 1;
2121 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2124 value = g_strndup ((const gchar *) data, len);
2126 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2131 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2135 /* takes ownership of sdes */
2136 changed = rtp_source_set_sdes_struct (source, sdes);
2138 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2139 source->validated = TRUE;
2142 on_new_ssrc (sess, source);
2144 /* source became active */
2145 if (source_update_active (sess, source, prevactive))
2146 on_ssrc_validated (sess, source);
2149 on_ssrc_sdes (sess, source);
2152 g_object_unref (source);
2154 more_items = gst_rtcp_packet_sdes_next_item (packet);
2159 /* BYE is sent when a client leaves the session
2162 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2163 RTPPacketInfo * pinfo)
2167 gboolean reconsider = FALSE;
2169 reason = gst_rtcp_packet_bye_get_reason (packet);
2170 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2172 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2173 for (i = 0; i < count; i++) {
2176 gboolean created, prevactive, prevsender;
2177 guint pmembers, members;
2179 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2180 GST_DEBUG ("SSRC: %08x", ssrc);
2182 /* find src and mark bye, no probation when dealing with RTCP */
2183 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2187 if (source->internal) {
2188 /* our own source, something weird with this packet */
2189 g_object_unref (source);
2193 /* store time for when we need to time out this source */
2194 source->bye_time = pinfo->current_time;
2196 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2197 prevsender = RTP_SOURCE_IS_SENDER (source);
2199 /* mark the source BYE */
2200 rtp_source_mark_bye (source, reason);
2202 pmembers = sess->stats.active_sources;
2204 source_update_active (sess, source, prevactive);
2205 source_update_sender (sess, source, prevsender);
2207 members = sess->stats.active_sources;
2209 if (!sess->scheduled_bye && members < pmembers) {
2210 /* some members went away since the previous timeout estimate.
2211 * Perform reverse reconsideration but only when we are not scheduling a
2213 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2214 pinfo->current_time < sess->next_rtcp_check_time) {
2215 GstClockTime time_remaining;
2217 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2218 sess->next_rtcp_check_time =
2219 gst_util_uint64_scale (time_remaining, members, pmembers);
2221 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2222 GST_TIME_ARGS (sess->next_rtcp_check_time));
2224 sess->next_rtcp_check_time += pinfo->current_time;
2226 /* mark pending reconsider. We only want to signal the reconsideration
2227 * once after we handled all the source in the bye packet */
2233 on_new_ssrc (sess, source);
2235 on_bye_ssrc (sess, source);
2237 g_object_unref (source);
2240 RTP_SESSION_UNLOCK (sess);
2241 /* notify app of reconsideration */
2242 if (sess->callbacks.reconsider)
2243 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2244 RTP_SESSION_LOCK (sess);
2250 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2251 RTPPacketInfo * pinfo)
2253 GST_DEBUG ("received APP");
2257 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2258 gboolean fir, GstClockTime current_time)
2260 guint32 round_trip = 0;
2262 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2264 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2265 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2268 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2269 GST_DEBUG ("Ignoring %s request because one was send without one "
2270 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2271 fir ? "FIR" : "PLI",
2272 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2273 GST_TIME_ARGS (round_trip_in_ns));;
2278 sess->last_keyframe_request = current_time;
2280 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2281 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2282 sess->callbacks.request_key_unit);
2284 RTP_SESSION_UNLOCK (sess);
2285 sess->callbacks.request_key_unit (sess, fir,
2286 sess->request_key_unit_user_data);
2287 RTP_SESSION_LOCK (sess);
2293 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2294 guint32 media_ssrc, GstClockTime current_time)
2298 if (!sess->callbacks.request_key_unit)
2301 src = find_source (sess, sender_ssrc);
2305 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2309 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2310 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2315 gboolean our_request = FALSE;
2317 if (!sess->callbacks.request_key_unit)
2323 src = find_source (sess, sender_ssrc);
2325 /* Hack because Google fails to set the sender_ssrc correctly */
2326 if (!src && sender_ssrc == 1) {
2327 GHashTableIter iter;
2329 /* we can't find the source if there are multiple */
2330 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2333 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2334 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2335 if (!src->internal && rtp_source_is_sender (src))
2343 for (position = 0; position < fci_length; position += 8) {
2344 guint8 *data = fci_data + position;
2347 ssrc = GST_READ_UINT32_BE (data);
2349 own = find_source (sess, ssrc);
2353 if (own->internal) {
2361 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2365 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2366 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2367 GstClockTime current_time)
2369 sess->stats.nacks_received++;
2371 if (!sess->callbacks.notify_nack)
2374 while (fci_length > 0) {
2375 guint16 seqnum, blp;
2377 seqnum = GST_READ_UINT16_BE (fci_data);
2378 blp = GST_READ_UINT16_BE (fci_data + 2);
2380 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2382 RTP_SESSION_UNLOCK (sess);
2383 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2384 sess->notify_nack_user_data);
2385 RTP_SESSION_LOCK (sess);
2393 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2394 RTPPacketInfo * pinfo, GstClockTime current_time)
2396 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2397 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2398 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2399 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2400 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2401 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2404 src = find_source (sess, media_ssrc);
2406 /* skip non-bye packets for sources that are marked BYE */
2407 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2410 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2411 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2413 if (g_signal_has_handler_pending (sess,
2414 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2415 GstBuffer *fci_buffer = NULL;
2417 if (fci_length > 0) {
2418 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2419 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2421 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2424 RTP_SESSION_UNLOCK (sess);
2425 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2426 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2427 RTP_SESSION_LOCK (sess);
2430 gst_buffer_unref (fci_buffer);
2433 if (src && sess->rtcp_feedback_retention_window) {
2434 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2437 if ((src && src->internal) ||
2438 /* PSFB FIR puts the media ssrc inside the FCI */
2439 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2441 case GST_RTCP_TYPE_PSFB:
2443 case GST_RTCP_PSFB_TYPE_PLI:
2444 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2447 case GST_RTCP_PSFB_TYPE_FIR:
2448 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2455 case GST_RTCP_TYPE_RTPFB:
2457 case GST_RTCP_RTPFB_TYPE_NACK:
2458 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2459 fci_data, fci_length, current_time);
2471 * rtp_session_process_rtcp:
2472 * @sess: and #RTPSession
2473 * @buffer: an RTCP buffer
2474 * @current_time: the current system time
2475 * @ntpnstime: the current NTP time in nanoseconds
2477 * Process an RTCP buffer in the session manager. This function takes ownership
2480 * Returns: a #GstFlowReturn.
2483 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2484 GstClockTime current_time, guint64 ntpnstime)
2486 GstRTCPPacket packet;
2487 gboolean more, is_bye = FALSE, do_sync = FALSE;
2488 RTPPacketInfo pinfo = { 0, };
2489 GstFlowReturn result = GST_FLOW_OK;
2490 GstRTCPBuffer rtcp = { NULL, };
2492 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2493 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2495 if (!gst_rtcp_buffer_validate (buffer))
2496 goto invalid_packet;
2498 GST_DEBUG ("received RTCP packet");
2500 RTP_SESSION_LOCK (sess);
2501 /* update pinfo stats */
2502 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2505 /* start processing the compound packet */
2506 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2507 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2511 type = gst_rtcp_packet_get_type (&packet);
2514 case GST_RTCP_TYPE_SR:
2515 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2517 case GST_RTCP_TYPE_RR:
2518 rtp_session_process_rr (sess, &packet, &pinfo);
2520 case GST_RTCP_TYPE_SDES:
2521 rtp_session_process_sdes (sess, &packet, &pinfo);
2523 case GST_RTCP_TYPE_BYE:
2525 /* don't try to attempt lip-sync anymore for streams with a BYE */
2527 rtp_session_process_bye (sess, &packet, &pinfo);
2529 case GST_RTCP_TYPE_APP:
2530 rtp_session_process_app (sess, &packet, &pinfo);
2532 case GST_RTCP_TYPE_RTPFB:
2533 case GST_RTCP_TYPE_PSFB:
2534 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2537 GST_WARNING ("got unknown RTCP packet");
2540 more = gst_rtcp_packet_move_to_next (&packet);
2543 gst_rtcp_buffer_unmap (&rtcp);
2545 /* if we are scheduling a BYE, we only want to count bye packets, else we
2546 * count everything */
2547 if (sess->scheduled_bye && is_bye) {
2548 sess->bye_stats.bye_members++;
2549 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2552 /* keep track of average packet size */
2553 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2555 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2556 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2557 RTP_SESSION_UNLOCK (sess);
2560 clean_packet_info (&pinfo);
2562 /* notify caller of sr packets in the callback */
2563 if (do_sync && sess->callbacks.sync_rtcp) {
2564 result = sess->callbacks.sync_rtcp (sess, buffer,
2565 sess->sync_rtcp_user_data);
2567 gst_buffer_unref (buffer);
2574 GST_DEBUG ("invalid RTCP packet received");
2575 gst_buffer_unref (buffer);
2581 * rtp_session_update_send_caps:
2582 * @sess: an #RTPSession
2585 * Update the caps of the sender in the rtp session.
2588 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2593 g_return_if_fail (RTP_IS_SESSION (sess));
2594 g_return_if_fail (GST_IS_CAPS (caps));
2596 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2598 s = gst_caps_get_structure (caps, 0);
2600 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2604 RTP_SESSION_LOCK (sess);
2605 source = obtain_internal_source (sess, ssrc, &created);
2607 rtp_source_update_caps (source, caps);
2608 g_object_unref (source);
2610 RTP_SESSION_UNLOCK (sess);
2615 * rtp_session_send_rtp:
2616 * @sess: an #RTPSession
2617 * @data: pointer to either an RTP buffer or a list of RTP buffers
2618 * @is_list: TRUE when @data is a buffer list
2619 * @current_time: the current system time
2620 * @running_time: the running time of @data
2622 * Send the RTP buffer in the session manager. This function takes ownership of
2625 * Returns: a #GstFlowReturn.
2628 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2629 GstClockTime current_time, GstClockTime running_time)
2631 GstFlowReturn result;
2633 gboolean prevsender;
2635 RTPPacketInfo pinfo = { 0, };
2638 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2639 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2641 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2643 RTP_SESSION_LOCK (sess);
2644 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2645 current_time, running_time, -1))
2646 goto invalid_packet;
2648 source = obtain_internal_source (sess, pinfo.ssrc, &created);
2650 /* update last activity */
2651 source->last_rtp_activity = current_time;
2653 prevsender = RTP_SOURCE_IS_SENDER (source);
2654 oldrate = source->bitrate;
2656 /* we use our own source to send */
2657 result = rtp_source_send_rtp (source, &pinfo);
2659 source_update_sender (sess, source, prevsender);
2661 if (oldrate != source->bitrate)
2662 sess->recalc_bandwidth = TRUE;
2663 RTP_SESSION_UNLOCK (sess);
2665 g_object_unref (source);
2666 clean_packet_info (&pinfo);
2672 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2673 RTP_SESSION_UNLOCK (sess);
2674 GST_DEBUG ("invalid RTP packet received");
2680 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2682 *bandwidth += source->bitrate;
2685 /* must be called with session lock */
2687 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2690 GstClockTime result;
2691 RTPSessionStats *stats;
2693 /* recalculate bandwidth when it changed */
2694 if (sess->recalc_bandwidth) {
2697 if (sess->bandwidth > 0)
2698 bandwidth = sess->bandwidth;
2700 /* If it is <= 0, then try to estimate the actual bandwidth */
2703 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2704 (GHFunc) add_bitrates, &bandwidth);
2707 if (bandwidth < 8000)
2708 bandwidth = RTP_STATS_BANDWIDTH;
2710 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2711 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2713 sess->recalc_bandwidth = FALSE;
2716 if (sess->scheduled_bye) {
2717 stats = &sess->bye_stats;
2718 result = rtp_stats_calculate_bye_interval (stats);
2720 stats = &sess->stats;
2721 result = rtp_stats_calculate_rtcp_interval (stats,
2722 stats->internal_sender_sources > 0, first);
2725 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2726 GST_TIME_ARGS (result), first);
2728 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2729 result = rtp_stats_add_rtcp_jitter (stats, result);
2731 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2737 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2739 if (source->internal)
2740 rtp_source_mark_bye (source, reason);
2744 * rtp_session_mark_all_bye:
2745 * @sess: an #RTPSession
2748 * Mark all internal sources of the session as BYE with @reason.
2751 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2753 g_return_if_fail (RTP_IS_SESSION (sess));
2755 RTP_SESSION_LOCK (sess);
2756 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2757 (GHFunc) source_mark_bye, (gpointer) reason);
2758 RTP_SESSION_UNLOCK (sess);
2761 /* Stop the current @sess and schedule a BYE message for the other members.
2762 * One must have the session lock to call this function
2764 static GstFlowReturn
2765 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2767 GstFlowReturn result = GST_FLOW_OK;
2768 GstClockTime interval;
2770 /* nothing to do it we already scheduled bye */
2771 if (sess->scheduled_bye)
2774 /* we schedule BYE now */
2775 sess->scheduled_bye = TRUE;
2776 /* at least one member wants to send a BYE */
2777 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2778 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2779 sess->bye_stats.bye_members = 1;
2780 sess->first_rtcp = TRUE;
2781 sess->allow_early = TRUE;
2783 /* reschedule transmission */
2784 sess->last_rtcp_send_time = current_time;
2785 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2787 if (interval != GST_CLOCK_TIME_NONE)
2788 sess->next_rtcp_check_time = current_time + interval;
2790 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2792 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2793 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2795 RTP_SESSION_UNLOCK (sess);
2796 /* notify app of reconsideration */
2797 if (sess->callbacks.reconsider)
2798 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2799 RTP_SESSION_LOCK (sess);
2806 * rtp_session_schedule_bye:
2807 * @sess: an #RTPSession
2808 * @current_time: the current system time
2810 * Schedule a BYE message for all sources marked as BYE in @sess.
2812 * Returns: a #GstFlowReturn.
2815 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2817 GstFlowReturn result = GST_FLOW_OK;
2819 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2821 RTP_SESSION_LOCK (sess);
2822 result = rtp_session_schedule_bye_locked (sess, current_time);
2823 RTP_SESSION_UNLOCK (sess);
2829 * rtp_session_next_timeout:
2830 * @sess: an #RTPSession
2831 * @current_time: the current system time
2833 * Get the next time we should perform session maintenance tasks.
2835 * Returns: a time when rtp_session_on_timeout() should be called with the
2836 * current system time.
2839 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2841 GstClockTime result, interval = 0;
2843 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2845 RTP_SESSION_LOCK (sess);
2847 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2848 GST_DEBUG ("have early rtcp time");
2849 result = sess->next_early_rtcp_time;
2853 result = sess->next_rtcp_check_time;
2855 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2856 ", next time: %" GST_TIME_FORMAT,
2857 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2859 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2860 GST_DEBUG ("take current time as base");
2861 /* our previous check time expired, start counting from the current time
2863 result = current_time;
2866 if (sess->scheduled_bye) {
2867 if (sess->bye_stats.active_sources >= 50) {
2868 GST_DEBUG ("reconsider BYE, more than 50 sources");
2869 /* reconsider BYE if members >= 50 */
2870 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2873 if (sess->first_rtcp) {
2874 GST_DEBUG ("first RTCP packet");
2875 /* we are called for the first time */
2876 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2877 } else if (sess->next_rtcp_check_time < current_time) {
2878 GST_DEBUG ("old check time expired, getting new timeout");
2879 /* get a new timeout when we need to */
2880 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2884 if (interval != GST_CLOCK_TIME_NONE)
2887 result = GST_CLOCK_TIME_NONE;
2889 sess->next_rtcp_check_time = result;
2893 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2894 ", next time: %" GST_TIME_FORMAT,
2895 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2896 RTP_SESSION_UNLOCK (sess);
2910 GstRTCPBuffer rtcpbuf;
2913 guint num_to_report;
2918 GstClockTime current_time;
2920 GstClockTime running_time;
2921 GstClockTime interval;
2922 GstRTCPPacket packet;
2925 gboolean may_suppress;
2927 guint nacked_seqnums;
2931 session_start_rtcp (RTPSession * sess, ReportData * data)
2933 GstRTCPPacket *packet = &data->packet;
2934 RTPSource *own = data->source;
2935 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2937 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2938 data->has_sdes = FALSE;
2940 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2942 if (RTP_SOURCE_IS_SENDER (own)) {
2945 guint32 packet_count, octet_count;
2947 /* we are a sender, create SR */
2948 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2949 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2951 /* get latest stats */
2952 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2953 &ntptime, &rtptime, &packet_count, &octet_count);
2955 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2956 packet_count, octet_count);
2958 /* fill in sender report info */
2959 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2960 ntptime, rtptime, packet_count, octet_count);
2962 /* we are only receiver, create RR */
2963 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2964 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2965 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2969 /* construct a Sender or Receiver Report */
2971 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2973 RTPSession *sess = data->sess;
2974 GstRTCPPacket *packet = &data->packet;
2975 guint8 fractionlost;
2977 guint32 exthighestseq, jitter;
2980 /* don't report for sources in future generations */
2981 if (((gint16) (source->generation - sess->generation)) > 0) {
2982 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
2983 source->generation, sess->generation);
2987 if (g_hash_table_contains (source->reported_in_sr_of,
2988 GUINT_TO_POINTER (data->source->ssrc))) {
2989 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
2993 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
2994 GST_DEBUG ("max RB count reached");
2998 /* only report about other sender */
2999 if (source == data->source)
3002 if (!RTP_SOURCE_IS_SENDER (source)) {
3003 GST_DEBUG ("source %08x not sender", source->ssrc);
3007 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3010 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3011 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3013 /* store last generated RR packet */
3014 source->last_rr.is_valid = TRUE;
3015 source->last_rr.fractionlost = fractionlost;
3016 source->last_rr.packetslost = packetslost;
3017 source->last_rr.exthighestseq = exthighestseq;
3018 source->last_rr.jitter = jitter;
3019 source->last_rr.lsr = lsr;
3020 source->last_rr.dlsr = dlsr;
3022 /* packet is not yet filled, add report block for this source. */
3023 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3024 exthighestseq, jitter, lsr, dlsr);
3027 g_hash_table_add (source->reported_in_sr_of,
3028 GUINT_TO_POINTER (data->source->ssrc));
3033 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3035 GstRTCPPacket *packet = &data->packet;
3039 if (!source->send_fir)
3042 len = gst_rtcp_packet_fb_get_fci_length (packet);
3043 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3044 /* exit because the packet is full, will put next request in a
3048 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3050 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3052 fci_data[0] = source->current_send_fir_seqnum;
3053 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3055 source->send_fir = FALSE;
3059 session_fir (RTPSession * sess, ReportData * data)
3061 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3062 GstRTCPPacket *packet = &data->packet;
3064 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3067 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3068 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3069 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3071 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3072 (GHFunc) session_add_fir, data);
3074 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3075 gst_rtcp_packet_remove (packet);
3077 data->may_suppress = FALSE;
3081 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3083 GstRTCPPacket packet;
3084 GstRTCPBuffer rtcp = { NULL, };
3085 gboolean ret = FALSE;
3087 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3089 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3090 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3091 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3095 gst_rtcp_buffer_unmap (&rtcp);
3102 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3104 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3105 GstRTCPPacket *packet = &data->packet;
3107 if (!source->send_pli)
3110 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3113 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3114 /* exit because the packet is full, will put next request in a
3118 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3119 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3120 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3122 source->send_pli = FALSE;
3123 data->may_suppress = FALSE;
3126 /* construct NACK */
3128 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3130 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3131 GstRTCPPacket *packet = &data->packet;
3136 if (!source->send_nack)
3139 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3140 /* exit because the packet is full, will put next request in a
3144 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3145 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3146 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3148 nacks = rtp_source_get_nacks (source, &n_nacks);
3149 GST_DEBUG ("%u NACKs", n_nacks);
3150 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3153 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3154 for (i = 0; i < n_nacks; i++) {
3155 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3157 data->nacked_seqnums++;
3160 rtp_source_clear_nacks (source);
3161 data->may_suppress = FALSE;
3164 /* perform cleanup of sources that timed out */
3166 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3168 gboolean remove = FALSE;
3169 gboolean byetimeout = FALSE;
3170 gboolean sendertimeout = FALSE;
3171 gboolean is_sender, is_active;
3172 RTPSession *sess = data->sess;
3173 GstClockTime interval, binterval;
3176 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3178 /* check for outdated collisions */
3179 if (source->internal) {
3180 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3181 rtp_source_timeout (source, data->current_time,
3182 /* "a relatively long time" -- RFC 3550 section 8.2 */
3183 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
3184 data->running_time - sess->rtcp_feedback_retention_window);
3187 /* nothing else to do when without RTCP */
3188 if (data->interval == GST_CLOCK_TIME_NONE)
3191 is_sender = RTP_SOURCE_IS_SENDER (source);
3192 is_active = RTP_SOURCE_IS_ACTIVE (source);
3194 /* our own rtcp interval may have been forced low by secondary configuration,
3195 * while sender side may still operate with higher interval,
3196 * so do not just take our interval to decide on timing out sender,
3197 * but take (if data->interval <= 5 * GST_SECOND):
3198 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3199 * where sender_interval is difference between last 2 received RTCP reports
3201 if (data->interval >= 5 * GST_SECOND || source->internal) {
3202 binterval = data->interval;
3204 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3205 GST_TIME_ARGS (source->stats.prev_rtcptime),
3206 GST_TIME_ARGS (source->stats.last_rtcptime));
3207 /* if not received enough yet, fallback to larger default */
3208 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3209 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3211 binterval = 5 * GST_SECOND;
3212 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3214 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3215 GST_TIME_ARGS (binterval));
3217 if (!source->internal) {
3218 if (source->marked_bye) {
3219 /* if we received a BYE from the source, remove the source after some
3221 if (data->current_time > source->bye_time &&
3222 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3223 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3228 /* sources that were inactive for more than 5 times the deterministic reporting
3229 * interval get timed out. the min timeout is 5 seconds. */
3230 /* mind old time that might pre-date last time going to PLAYING */
3231 btime = MAX (source->last_activity, sess->start_time);
3232 if (data->current_time > btime) {
3233 interval = MAX (binterval * 5, 5 * GST_SECOND);
3234 if (data->current_time - btime > interval) {
3235 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3236 source->ssrc, GST_TIME_ARGS (btime));
3242 /* senders that did not send for a long time become a receiver, this also
3243 * holds for our own sources. */
3245 /* mind old time that might pre-date last time going to PLAYING */
3246 btime = MAX (source->last_rtp_activity, sess->start_time);
3247 if (data->current_time > btime) {
3248 interval = MAX (binterval * 2, 5 * GST_SECOND);
3249 if (data->current_time - btime > interval) {
3250 if (source->internal && source->sent_bye) {
3251 /* an internal source is BYE and stopped sending RTP, remove */
3252 GST_DEBUG ("internal BYE source %08x timed out, last %"
3253 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3256 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3257 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3258 sendertimeout = TRUE;
3265 sess->total_sources--;
3267 sess->stats.sender_sources--;
3268 if (source->internal)
3269 sess->stats.internal_sender_sources--;
3272 sess->stats.active_sources--;
3274 if (source->internal)
3275 sess->stats.internal_sources--;
3278 on_bye_timeout (sess, source);
3280 on_timeout (sess, source);
3282 if (sendertimeout) {
3283 source->is_sender = FALSE;
3284 sess->stats.sender_sources--;
3285 if (source->internal)
3286 sess->stats.internal_sender_sources--;
3288 on_sender_timeout (sess, source);
3290 /* count how many source to report in this generation */
3291 if (((gint16) (source->generation - sess->generation)) <= 0)
3292 data->num_to_report++;
3294 source->closing = remove;
3298 session_sdes (RTPSession * sess, ReportData * data)
3300 GstRTCPPacket *packet = &data->packet;
3301 const GstStructure *sdes;
3303 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3305 /* add SDES packet */
3306 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3308 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3310 sdes = rtp_source_get_sdes_struct (data->source);
3312 /* add all fields in the structure, the order is not important. */
3313 n_fields = gst_structure_n_fields (sdes);
3314 for (i = 0; i < n_fields; ++i) {
3317 GstRTCPSDESType type;
3319 field = gst_structure_nth_field_name (sdes, i);
3322 value = gst_structure_get_string (sdes, field);
3325 type = gst_rtcp_sdes_name_to_type (field);
3327 /* Early packets are minimal and only include the CNAME */
3328 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3331 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3332 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3333 (const guint8 *) value);
3334 } else if (type == GST_RTCP_SDES_PRIV) {
3340 /* don't accept entries that are too big */
3341 prefix_len = strlen (field);
3342 if (prefix_len > 255)
3344 value_len = strlen (value);
3345 if (value_len > 255)
3347 data_len = 1 + prefix_len + value_len;
3351 data[0] = prefix_len;
3352 memcpy (&data[1], field, prefix_len);
3353 memcpy (&data[1 + prefix_len], value, value_len);
3355 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3359 data->has_sdes = TRUE;
3362 /* schedule a BYE packet */
3364 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3366 GstRTCPPacket *packet = &data->packet;
3367 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3370 session_sdes (sess, data);
3371 /* add a BYE packet */
3372 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3373 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3374 if (source->bye_reason)
3375 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3377 /* we have a BYE packet now */
3378 source->sent_bye = TRUE;
3382 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3384 GstClockTime new_send_time, elapsed;
3385 GstClockTime interval;
3386 RTPSessionStats *stats;
3388 if (sess->scheduled_bye)
3389 stats = &sess->bye_stats;
3391 stats = &sess->stats;
3393 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3394 data->is_early = TRUE;
3396 data->is_early = FALSE;
3398 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3399 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3400 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3401 GST_TIME_ARGS (current_time));
3405 /* no need to check yet */
3406 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3407 sess->next_rtcp_check_time > current_time) {
3408 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3409 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3410 GST_TIME_ARGS (current_time));
3415 /* get elapsed time since we last reported */
3416 elapsed = current_time - sess->last_rtcp_send_time;
3418 /* take interval and add jitter */
3419 interval = data->interval;
3420 if (interval != GST_CLOCK_TIME_NONE)
3421 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3423 /* perform forward reconsideration */
3424 if (interval != GST_CLOCK_TIME_NONE) {
3425 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3426 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3427 new_send_time = interval + sess->last_rtcp_send_time;
3429 new_send_time = sess->last_rtcp_send_time;
3432 if (!data->is_early) {
3433 /* check if reconsideration */
3434 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3435 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3436 GST_TIME_ARGS (new_send_time));
3437 /* store new check time */
3438 sess->next_rtcp_check_time = new_send_time;
3441 sess->next_rtcp_check_time = current_time + interval;
3442 } else if (interval != GST_CLOCK_TIME_NONE) {
3443 /* Apply the rules from RFC 4585 section 3.5.3 */
3444 if (stats->min_interval != 0 && !sess->first_rtcp) {
3445 GstClockTime T_rr_current_interval =
3446 g_random_double_range (0.5, 1.5) * stats->min_interval;
3448 /* This will caused the RTCP to be suppressed if no FB packets are added */
3449 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3450 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3451 " last: %" GST_TIME_FORMAT
3452 " + T_rr_current_interval: %" GST_TIME_FORMAT
3453 " > new_send_time: %" GST_TIME_FORMAT,
3454 GST_TIME_ARGS (stats->min_interval),
3455 GST_TIME_ARGS (sess->last_rtcp_send_time),
3456 GST_TIME_ARGS (T_rr_current_interval),
3457 GST_TIME_ARGS (new_send_time));
3458 data->may_suppress = TRUE;
3463 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3464 GST_TIME_ARGS (new_send_time));
3470 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3472 g_hash_table_insert (hash_table, key, g_object_ref (source));
3476 remove_closing_sources (const gchar * key, RTPSource * source,
3479 if (source->closing)
3482 if (source->send_fir)
3483 data->have_fir = TRUE;
3484 if (source->send_pli)
3485 data->have_pli = TRUE;
3486 if (source->send_nack)
3487 data->have_nack = TRUE;
3493 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3495 RTPSession *sess = data->sess;
3496 gboolean is_bye = FALSE;
3497 ReportOutput *output;
3499 /* only generate RTCP for active internal sources */
3500 if (!source->internal || source->sent_bye)
3503 /* ignore other sources when we do the timeout after a scheduled BYE */
3504 if (sess->scheduled_bye && !source->marked_bye)
3507 data->source = source;
3510 session_start_rtcp (sess, data);
3512 if (source->marked_bye) {
3514 make_source_bye (sess, source, data);
3516 } else if (!data->is_early) {
3517 /* loop over all known sources and add report blocks. If we are early, we
3518 * just make a minimal RTCP packet and skip this step */
3519 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3520 (GHFunc) session_report_blocks, data);
3522 if (!data->has_sdes)
3523 session_sdes (sess, data);
3526 session_fir (sess, data);
3529 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3530 (GHFunc) session_pli, data);
3532 if (data->have_nack)
3533 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3534 (GHFunc) session_nack, data);
3536 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3538 output = g_slice_new (ReportOutput);
3539 output->source = g_object_ref (source);
3540 output->is_bye = is_bye;
3541 output->buffer = data->rtcp;
3542 /* queue the RTCP packet to push later */
3543 g_queue_push_tail (&data->output, output);
3547 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3549 RTPSession *sess = data->sess;
3551 if (g_hash_table_size (source->reported_in_sr_of) >=
3552 sess->stats.internal_sources) {
3553 /* source is reported, move to next generation */
3554 source->generation = sess->generation + 1;
3555 g_hash_table_remove_all (source->reported_in_sr_of);
3557 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3558 source->generation);
3560 /* if we reported all sources in this generation, move to next */
3561 if (--data->num_to_report == 0) {
3563 GST_DEBUG ("all reported, generation now %u", sess->generation);
3569 * rtp_session_on_timeout:
3570 * @sess: an #RTPSession
3571 * @current_time: the current system time
3572 * @ntpnstime: the current NTP time in nanoseconds
3573 * @running_time: the current running_time of the pipeline
3575 * Perform maintenance actions after the timeout obtained with
3576 * rtp_session_next_timeout() expired.
3578 * This function will perform timeouts of receivers and senders, send a BYE
3579 * packet or generate RTCP packets with current session stats.
3581 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3582 * times, for each packet that should be processed.
3584 * Returns: a #GstFlowReturn.
3587 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3588 guint64 ntpnstime, GstClockTime running_time)
3590 GstFlowReturn result = GST_FLOW_OK;
3591 ReportData data = { GST_RTCP_BUFFER_INIT };
3592 GHashTable *table_copy;
3593 ReportOutput *output;
3595 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3597 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3598 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3599 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3602 data.current_time = current_time;
3603 data.ntpnstime = ntpnstime;
3604 data.running_time = running_time;
3605 data.num_to_report = 0;
3606 data.may_suppress = FALSE;
3607 data.nacked_seqnums = 0;
3608 g_queue_init (&data.output);
3610 RTP_SESSION_LOCK (sess);
3611 /* get a new interval, we need this for various cleanups etc */
3612 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3614 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3616 /* we need an internal source now */
3617 if (sess->stats.internal_sources == 0) {
3621 source = obtain_internal_source (sess, sess->suggested_ssrc, &created);
3622 g_object_unref (source);
3625 /* Make a local copy of the hashtable. We need to do this because the
3626 * cleanup stage below releases the session lock. */
3627 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3628 (GDestroyNotify) g_object_unref);
3629 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3630 (GHFunc) clone_ssrcs_hashtable, table_copy);
3632 /* Clean up the session, mark the source for removing, this might release the
3634 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3635 g_hash_table_destroy (table_copy);
3637 /* Now remove the marked sources */
3638 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3639 (GHRFunc) remove_closing_sources, &data);
3641 /* update point-to-point status */
3642 session_update_ptp (sess);
3644 /* see if we need to generate SR or RR packets */
3645 if (!is_rtcp_time (sess, current_time, &data))
3648 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3649 sess->generation, data.num_to_report, data.is_early);
3651 /* generate RTCP for all internal sources */
3652 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3653 (GHFunc) generate_rtcp, &data);
3655 /* update the generation for all the sources that have been reported */
3656 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3657 (GHFunc) update_generation, &data);
3659 /* we keep track of the last report time in order to timeout inactive
3660 * receivers or senders */
3661 if (!data.is_early && !data.may_suppress)
3662 sess->last_rtcp_send_time = data.current_time;
3663 sess->first_rtcp = FALSE;
3664 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3665 sess->scheduled_bye = FALSE;
3668 RTP_SESSION_UNLOCK (sess);
3670 /* push out the RTCP packets */
3671 while ((output = g_queue_pop_head (&data.output))) {
3672 gboolean do_not_suppress;
3673 GstBuffer *buffer = output->buffer;
3674 RTPSource *source = output->source;
3676 /* Give the user a change to add its own packet */
3677 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3678 buffer, data.is_early, &do_not_suppress);
3680 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3683 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3685 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3686 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3687 sess->stats.avg_rtcp_packet_size, packet_size);
3689 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3690 sess->send_rtcp_user_data);
3691 sess->stats.nacks_sent += data.nacked_seqnums;
3693 GST_DEBUG ("freeing packet callback: %p"
3694 " do_not_suppress: %d may_suppress: %d",
3695 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3696 sess->stats.nacks_dropped += data.nacked_seqnums;
3697 gst_buffer_unref (buffer);
3699 g_object_unref (source);
3700 g_slice_free (ReportOutput, output);
3706 * rtp_session_request_early_rtcp:
3707 * @sess: an #RTPSession
3708 * @current_time: the current system time
3709 * @max_delay: maximum delay
3711 * Request transmission of early RTCP
3714 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3715 GstClockTime max_delay)
3717 GstClockTime T_dither_max;
3719 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3721 RTP_SESSION_LOCK (sess);
3723 /* Check if already requested */
3724 /* RFC 4585 section 3.5.2 step 2 */
3725 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3726 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3730 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3731 GST_LOG_OBJECT (sess, "no next RTCP check time");
3735 /* Ignore the request a scheduled packet will be in time anyway */
3736 if (current_time + max_delay > sess->next_rtcp_check_time) {
3737 GST_LOG_OBJECT (sess, "next scheduled time is soon %" GST_TIME_FORMAT " + %"
3738 GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
3739 GST_TIME_ARGS (current_time),
3740 GST_TIME_ARGS (max_delay), GST_TIME_ARGS (sess->next_rtcp_check_time));
3744 /* RFC 4585 section 3.5.2 step 2b */
3745 /* If the total sources is <=2, then there is only us and one peer */
3746 /* When there is one auxiliary stream the session can still do point
3749 if (sess->is_doing_ptp) {
3752 /* Divide by 2 because l = 0.5 */
3753 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3757 /* RFC 4585 section 3.5.2 step 3 */
3758 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3759 GST_LOG_OBJECT (sess, "don't send because of dither");
3763 /* RFC 4585 section 3.5.2 step 4
3764 * Don't send if allow_early is FALSE, but not if we are in
3765 * immediate mode, meaning we are part of a group of at most the
3766 * application-specific threshold.
3768 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3769 sess->allow_early == FALSE) {
3770 GST_LOG_OBJECT (sess, "can't allow early feedback");
3775 /* Schedule an early transmission later */
3776 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3779 /* If no dithering, schedule it for NOW */
3780 sess->next_early_rtcp_time = current_time;
3783 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3784 GST_TIME_ARGS (sess->next_early_rtcp_time));
3785 RTP_SESSION_UNLOCK (sess);
3787 /* notify app of need to send packet early
3788 * and therefore of timeout change */
3789 if (sess->callbacks.reconsider)
3790 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3796 RTP_SESSION_UNLOCK (sess);
3800 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3804 if (!sess->callbacks.send_rtcp)
3807 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3809 rtp_session_request_early_rtcp (sess, now, max_delay);
3813 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3814 gboolean fir, gint count)
3818 RTP_SESSION_LOCK (sess);
3819 src = find_source (sess, ssrc);
3824 src->send_pli = FALSE;
3825 src->send_fir = TRUE;
3827 if (count == -1 || count != src->last_fir_count)
3828 src->current_send_fir_seqnum++;
3829 src->last_fir_count = count;
3830 } else if (!src->send_fir) {
3831 src->send_pli = TRUE;
3833 RTP_SESSION_UNLOCK (sess);
3835 rtp_session_send_rtcp (sess, 200 * GST_MSECOND);
3842 RTP_SESSION_UNLOCK (sess);
3848 * rtp_session_request_nack:
3849 * @sess: a #RTPSession
3851 * @seqnum: the missing seqnum
3852 * @max_delay: max delay to request NACK
3854 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
3856 * Returns: %TRUE if the NACK feedback could be scheduled
3859 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
3860 GstClockTime max_delay)
3864 RTP_SESSION_LOCK (sess);
3865 source = find_source (sess, ssrc);
3869 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
3870 rtp_source_register_nack (source, seqnum);
3871 RTP_SESSION_UNLOCK (sess);
3873 rtp_session_send_rtcp (sess, max_delay);
3880 RTP_SESSION_UNLOCK (sess);