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,
52 SIGNAL_SEND_RTCP_FULL,
53 SIGNAL_ON_RECEIVING_RTCP,
57 #define DEFAULT_INTERNAL_SOURCE NULL
58 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
59 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
60 #define DEFAULT_RTCP_RR_BANDWIDTH -1
61 #define DEFAULT_RTCP_RS_BANDWIDTH -1
62 #define DEFAULT_RTCP_MTU 1400
63 #define DEFAULT_SDES NULL
64 #define DEFAULT_NUM_SOURCES 0
65 #define DEFAULT_NUM_ACTIVE_SOURCES 0
66 #define DEFAULT_SOURCES NULL
67 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
68 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
69 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
70 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
79 PROP_RTCP_RR_BANDWIDTH,
80 PROP_RTCP_RS_BANDWIDTH,
84 PROP_NUM_ACTIVE_SOURCES,
87 PROP_RTCP_MIN_INTERVAL,
88 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
89 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
95 /* update average packet size */
96 #define INIT_AVG(avg, val) \
98 #define UPDATE_AVG(avg, val) \
102 (avg) = ((val) + (15 * (avg))) >> 4;
105 /* GObject vmethods */
106 static void rtp_session_finalize (GObject * object);
107 static void rtp_session_set_property (GObject * object, guint prop_id,
108 const GValue * value, GParamSpec * pspec);
109 static void rtp_session_get_property (GObject * object, guint prop_id,
110 GValue * value, GParamSpec * pspec);
112 static gboolean rtp_session_send_rtcp (RTPSession * sess,
113 GstClockTime max_delay);
115 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
117 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
119 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
120 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
121 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
122 static RTPSource *obtain_internal_source (RTPSession * sess,
123 guint32 ssrc, gboolean * created, GstClockTime current_time);
124 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
125 GstClockTime current_time);
126 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
127 gboolean deterministic, gboolean first);
130 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
131 const GValue * handler_return, gpointer data)
133 if (g_value_get_boolean (handler_return))
134 g_value_set_boolean (return_accu, TRUE);
140 rtp_session_class_init (RTPSessionClass * klass)
142 GObjectClass *gobject_class;
144 gobject_class = (GObjectClass *) klass;
146 gobject_class->finalize = rtp_session_finalize;
147 gobject_class->set_property = rtp_session_set_property;
148 gobject_class->get_property = rtp_session_get_property;
151 * RTPSession::get-source-by-ssrc:
152 * @session: the object which received the signal
153 * @ssrc: the SSRC of the RTPSource
155 * Request the #RTPSource object with SSRC @ssrc in @session.
157 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
158 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
159 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
160 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
161 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
164 * RTPSession::on-new-ssrc:
165 * @session: the object which received the signal
166 * @src: the new RTPSource
168 * Notify of a new SSRC that entered @session.
170 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
171 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
172 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
173 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
176 * RTPSession::on-ssrc-collision:
177 * @session: the object which received the signal
178 * @src: the #RTPSource that caused a collision
180 * Notify when we have an SSRC collision
182 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
183 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
184 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
185 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
188 * RTPSession::on-ssrc-validated:
189 * @session: the object which received the signal
190 * @src: the new validated RTPSource
192 * Notify of a new SSRC that became validated.
194 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
195 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
196 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
197 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
200 * RTPSession::on-ssrc-active:
201 * @session: the object which received the signal
202 * @src: the active RTPSource
204 * Notify of a SSRC that is active, i.e., sending RTCP.
206 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
207 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
208 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
209 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
212 * RTPSession::on-ssrc-sdes:
213 * @session: the object which received the signal
214 * @src: the RTPSource
216 * Notify that a new SDES was received for SSRC.
218 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
219 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
220 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
221 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
224 * RTPSession::on-bye-ssrc:
225 * @session: the object which received the signal
226 * @src: the RTPSource that went away
228 * Notify of an SSRC that became inactive because of a BYE packet.
230 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
231 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
232 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
233 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
236 * RTPSession::on-bye-timeout:
237 * @session: the object which received the signal
238 * @src: the RTPSource that timed out
240 * Notify of an SSRC that has timed out because of BYE
242 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
243 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
244 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
245 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
248 * RTPSession::on-timeout:
249 * @session: the object which received the signal
250 * @src: the RTPSource that timed out
252 * Notify of an SSRC that has timed out
254 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
255 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
256 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
257 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
260 * RTPSession::on-sender-timeout:
261 * @session: the object which received the signal
262 * @src: the RTPSource that timed out
264 * Notify of an SSRC that was a sender but timed out and became a receiver.
266 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
267 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
268 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
269 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
273 * RTPSession::on-sending-rtcp
274 * @session: the object which received the signal
275 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
276 * @early: %TRUE if the packet is early, %FALSE if it is regular
278 * This signal is emitted before sending an RTCP packet, it can be used
279 * to add extra RTCP Packets.
281 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
282 * if suppressing it is acceptable
284 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
285 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
286 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
287 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
288 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
291 * RTPSession::on-feedback-rtcp:
292 * @session: the object which received the signal
293 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
294 * %GST_RTCP_TYPE_RTPFB
295 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
296 * @sender_ssrc: The SSRC of the sender
297 * @media_ssrc: The SSRC of the media this refers to
298 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
301 * Notify that a RTCP feedback packet has been received
303 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
304 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
305 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
306 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
307 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
310 * RTPSession::send-rtcp:
311 * @session: the object which received the signal
312 * @max_delay: The maximum delay after which the feedback will not be useful
315 * Requests that the #RTPSession initiate a new RTCP packet as soon as
316 * possible within the requested delay.
318 rtp_session_signals[SIGNAL_SEND_RTCP] =
319 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
320 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
321 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
322 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
325 * RTPSession::send-rtcp-full:
326 * @session: the object which received the signal
327 * @max_delay: The maximum delay after which the feedback will not be useful
330 * Requests that the #RTPSession initiate a new RTCP packet as soon as
331 * possible within the requested delay.
333 * Returns: TRUE if the new RTCP packet could be scheduled within the
334 * requested delay, FALSE otherwise.
338 rtp_session_signals[SIGNAL_SEND_RTCP_FULL] =
339 g_signal_new ("send-rtcp-full", G_TYPE_FROM_CLASS (klass),
340 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
341 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
342 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
345 * RTPSession::on-receiving-rtcp
346 * @session: the object which received the signal
347 * @buffer: the #GstBuffer containing the RTCP packet that was received
349 * This signal is emitted when receiving an RTCP packet before it is handled
350 * by the session. It can be used to extract custom information from RTCP packets.
354 rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP] =
355 g_signal_new ("on-receiving-rtcp", G_TYPE_FROM_CLASS (klass),
356 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_receiving_rtcp),
357 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
358 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
360 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
361 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
362 "The internal SSRC used for the session (deprecated)",
363 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
365 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
366 g_param_spec_object ("internal-source", "Internal Source",
367 "The internal source element of the session (deprecated)",
368 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
371 g_param_spec_double ("bandwidth", "Bandwidth",
372 "The bandwidth of the session (0 for auto-discover)",
373 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
374 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
377 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
378 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
379 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
380 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
383 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
384 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
385 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
386 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
389 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
390 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
391 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
392 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
395 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
396 "The maximum size of the RTCP packets",
397 16, G_MAXINT16, DEFAULT_RTCP_MTU,
398 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
400 g_object_class_install_property (gobject_class, PROP_SDES,
401 g_param_spec_boxed ("sdes", "SDES",
402 "The SDES items of this session",
403 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
405 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
406 g_param_spec_uint ("num-sources", "Num Sources",
407 "The number of sources in the session", 0, G_MAXUINT,
408 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
410 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
411 g_param_spec_uint ("num-active-sources", "Num Active Sources",
412 "The number of active sources in the session", 0, G_MAXUINT,
413 DEFAULT_NUM_ACTIVE_SOURCES,
414 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
418 * Get a GValue Array of all sources in the session.
421 * <title>Getting the #RTPSources of a session
428 * g_object_get (sess, "sources", &arr, NULL);
430 * for (i = 0; i < arr->n_values; i++) {
433 * val = g_value_array_get_nth (arr, i);
434 * source = g_value_get_object (val);
436 * g_value_array_free (arr);
441 g_object_class_install_property (gobject_class, PROP_SOURCES,
442 g_param_spec_boxed ("sources", "Sources",
443 "An array of all known sources in the session",
444 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
446 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
447 g_param_spec_boolean ("favor-new", "Favor new sources",
448 "Resolve SSRC conflict in favor of new sources", FALSE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
451 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
452 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
453 "Minimum interval between Regular RTCP packet (in ns)",
454 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
455 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
457 g_object_class_install_property (gobject_class,
458 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
459 g_param_spec_uint64 ("rtcp-feedback-retention-window",
460 "RTCP Feedback retention window",
461 "Duration during which RTCP Feedback packets are retained (in ns)",
462 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
463 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
465 g_object_class_install_property (gobject_class,
466 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
467 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
468 "RTCP Immediate Feedback threshold",
469 "The maximum number of members of a RTP session for which immediate"
470 " feedback is used (DEPRECATED: has no effect and is not needed)",
471 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
472 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
474 g_object_class_install_property (gobject_class, PROP_PROBATION,
475 g_param_spec_uint ("probation", "Number of probations",
476 "Consecutive packet sequence numbers to accept the source",
477 0, G_MAXUINT, DEFAULT_PROBATION,
478 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
483 * Various session statistics. This property returns a GstStructure
484 * with name application/x-rtp-session-stats with the following fields:
486 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
487 * dropped (due to bandwidth constraints)
488 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
489 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
493 g_object_class_install_property (gobject_class, PROP_STATS,
494 g_param_spec_boxed ("stats", "Statistics",
495 "Various statistics", GST_TYPE_STRUCTURE,
496 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
498 klass->get_source_by_ssrc =
499 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
500 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
502 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
506 rtp_session_init (RTPSession * sess)
511 g_mutex_init (&sess->lock);
512 sess->key = g_random_int ();
516 /* TODO: We currently only use the first hash table but this is the
517 * beginning of an implementation for RFC2762
518 for (i = 0; i < 32; i++) {
520 for (i = 0; i < 1; i++) {
522 g_hash_table_new_full (NULL, NULL, NULL,
523 (GDestroyNotify) g_object_unref);
526 rtp_stats_init_defaults (&sess->stats);
527 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
528 rtp_stats_set_min_interval (&sess->stats,
529 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
531 sess->recalc_bandwidth = TRUE;
532 sess->bandwidth = DEFAULT_BANDWIDTH;
533 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
534 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
535 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
537 /* default UDP header length */
538 sess->header_len = 28;
539 sess->mtu = DEFAULT_RTCP_MTU;
541 sess->probation = DEFAULT_PROBATION;
543 /* some default SDES entries */
544 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
546 /* we do not want to leak details like the username or hostname here */
547 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
548 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
552 /* we do not want to leak the user's real name here */
553 str = g_strdup_printf ("Anon%u", g_random_int ());
554 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
558 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
560 /* this is the SSRC we suggest */
561 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
563 sess->first_rtcp = TRUE;
564 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
565 sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
567 sess->allow_early = TRUE;
568 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
569 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
570 sess->rtcp_immediate_feedback_threshold =
571 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
573 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
575 sess->is_doing_ptp = TRUE;
579 rtp_session_finalize (GObject * object)
584 sess = RTP_SESSION_CAST (object);
586 gst_structure_free (sess->sdes);
588 g_list_free_full (sess->conflicting_addresses,
589 (GDestroyNotify) rtp_conflicting_address_free);
591 /* TODO: Change this again when implementing RFC 2762
592 * for (i = 0; i < 32; i++)
594 for (i = 0; i < 1; i++)
595 g_hash_table_destroy (sess->ssrcs[i]);
597 g_mutex_clear (&sess->lock);
599 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
603 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
605 GValue value = { 0 };
607 g_value_init (&value, RTP_TYPE_SOURCE);
608 g_value_take_object (&value, source);
609 /* copies the value */
610 g_value_array_append (arr, &value);
614 rtp_session_create_sources (RTPSession * sess)
619 RTP_SESSION_LOCK (sess);
620 /* get number of elements in the table */
621 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
622 /* create the result value array */
623 res = g_value_array_new (size);
625 /* and copy all values into the array */
626 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
627 RTP_SESSION_UNLOCK (sess);
632 static GstStructure *
633 rtp_session_create_stats (RTPSession * sess)
637 s = gst_structure_new ("application/x-rtp-session-stats",
638 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
639 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
640 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
646 rtp_session_set_property (GObject * object, guint prop_id,
647 const GValue * value, GParamSpec * pspec)
651 sess = RTP_SESSION (object);
654 case PROP_INTERNAL_SSRC:
655 RTP_SESSION_LOCK (sess);
656 sess->suggested_ssrc = g_value_get_uint (value);
657 RTP_SESSION_UNLOCK (sess);
658 if (sess->callbacks.reconfigure)
659 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
662 RTP_SESSION_LOCK (sess);
663 sess->bandwidth = g_value_get_double (value);
664 sess->recalc_bandwidth = TRUE;
665 RTP_SESSION_UNLOCK (sess);
667 case PROP_RTCP_FRACTION:
668 RTP_SESSION_LOCK (sess);
669 sess->rtcp_bandwidth = g_value_get_double (value);
670 sess->recalc_bandwidth = TRUE;
671 RTP_SESSION_UNLOCK (sess);
673 case PROP_RTCP_RR_BANDWIDTH:
674 RTP_SESSION_LOCK (sess);
675 sess->rtcp_rr_bandwidth = g_value_get_int (value);
676 sess->recalc_bandwidth = TRUE;
677 RTP_SESSION_UNLOCK (sess);
679 case PROP_RTCP_RS_BANDWIDTH:
680 RTP_SESSION_LOCK (sess);
681 sess->rtcp_rs_bandwidth = g_value_get_int (value);
682 sess->recalc_bandwidth = TRUE;
683 RTP_SESSION_UNLOCK (sess);
686 sess->mtu = g_value_get_uint (value);
689 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
692 sess->favor_new = g_value_get_boolean (value);
694 case PROP_RTCP_MIN_INTERVAL:
695 rtp_stats_set_min_interval (&sess->stats,
696 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
697 /* trigger reconsideration */
698 RTP_SESSION_LOCK (sess);
699 sess->next_rtcp_check_time = 0;
700 RTP_SESSION_UNLOCK (sess);
701 if (sess->callbacks.reconsider)
702 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
704 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
705 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
708 sess->probation = g_value_get_uint (value);
711 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
717 rtp_session_get_property (GObject * object, guint prop_id,
718 GValue * value, GParamSpec * pspec)
722 sess = RTP_SESSION (object);
725 case PROP_INTERNAL_SSRC:
726 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
728 case PROP_INTERNAL_SOURCE:
729 /* FIXME, return a random source */
730 g_value_set_object (value, NULL);
733 g_value_set_double (value, sess->bandwidth);
735 case PROP_RTCP_FRACTION:
736 g_value_set_double (value, sess->rtcp_bandwidth);
738 case PROP_RTCP_RR_BANDWIDTH:
739 g_value_set_int (value, sess->rtcp_rr_bandwidth);
741 case PROP_RTCP_RS_BANDWIDTH:
742 g_value_set_int (value, sess->rtcp_rs_bandwidth);
745 g_value_set_uint (value, sess->mtu);
748 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
750 case PROP_NUM_SOURCES:
751 g_value_set_uint (value, rtp_session_get_num_sources (sess));
753 case PROP_NUM_ACTIVE_SOURCES:
754 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
757 g_value_take_boxed (value, rtp_session_create_sources (sess));
760 g_value_set_boolean (value, sess->favor_new);
762 case PROP_RTCP_MIN_INTERVAL:
763 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
765 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
766 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
769 g_value_set_uint (value, sess->probation);
772 g_value_take_boxed (value, rtp_session_create_stats (sess));
775 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
781 on_new_ssrc (RTPSession * sess, RTPSource * source)
783 g_object_ref (source);
784 RTP_SESSION_UNLOCK (sess);
785 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
786 RTP_SESSION_LOCK (sess);
787 g_object_unref (source);
791 on_ssrc_collision (RTPSession * sess, RTPSource * source)
793 g_object_ref (source);
794 RTP_SESSION_UNLOCK (sess);
795 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
797 RTP_SESSION_LOCK (sess);
798 g_object_unref (source);
802 on_ssrc_validated (RTPSession * sess, RTPSource * source)
804 g_object_ref (source);
805 RTP_SESSION_UNLOCK (sess);
806 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
808 RTP_SESSION_LOCK (sess);
809 g_object_unref (source);
813 on_ssrc_active (RTPSession * sess, RTPSource * source)
815 g_object_ref (source);
816 RTP_SESSION_UNLOCK (sess);
817 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
818 RTP_SESSION_LOCK (sess);
819 g_object_unref (source);
823 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
825 g_object_ref (source);
826 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
827 RTP_SESSION_UNLOCK (sess);
828 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
829 RTP_SESSION_LOCK (sess);
830 g_object_unref (source);
834 on_bye_ssrc (RTPSession * sess, RTPSource * source)
836 g_object_ref (source);
837 RTP_SESSION_UNLOCK (sess);
838 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
839 RTP_SESSION_LOCK (sess);
840 g_object_unref (source);
844 on_bye_timeout (RTPSession * sess, RTPSource * source)
846 g_object_ref (source);
847 RTP_SESSION_UNLOCK (sess);
848 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
849 RTP_SESSION_LOCK (sess);
850 g_object_unref (source);
854 on_timeout (RTPSession * sess, RTPSource * source)
856 g_object_ref (source);
857 RTP_SESSION_UNLOCK (sess);
858 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
859 RTP_SESSION_LOCK (sess);
860 g_object_unref (source);
864 on_sender_timeout (RTPSession * sess, RTPSource * source)
866 g_object_ref (source);
867 RTP_SESSION_UNLOCK (sess);
868 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
870 RTP_SESSION_LOCK (sess);
871 g_object_unref (source);
877 * Create a new session object.
879 * Returns: a new #RTPSession. g_object_unref() after usage.
882 rtp_session_new (void)
886 sess = g_object_new (RTP_TYPE_SESSION, NULL);
892 * rtp_session_set_callbacks:
893 * @sess: an #RTPSession
894 * @callbacks: callbacks to configure
895 * @user_data: user data passed in the callbacks
897 * Configure a set of callbacks to be notified of actions.
900 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
903 g_return_if_fail (RTP_IS_SESSION (sess));
905 if (callbacks->process_rtp) {
906 sess->callbacks.process_rtp = callbacks->process_rtp;
907 sess->process_rtp_user_data = user_data;
909 if (callbacks->send_rtp) {
910 sess->callbacks.send_rtp = callbacks->send_rtp;
911 sess->send_rtp_user_data = user_data;
913 if (callbacks->send_rtcp) {
914 sess->callbacks.send_rtcp = callbacks->send_rtcp;
915 sess->send_rtcp_user_data = user_data;
917 if (callbacks->sync_rtcp) {
918 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
919 sess->sync_rtcp_user_data = user_data;
921 if (callbacks->clock_rate) {
922 sess->callbacks.clock_rate = callbacks->clock_rate;
923 sess->clock_rate_user_data = user_data;
925 if (callbacks->reconsider) {
926 sess->callbacks.reconsider = callbacks->reconsider;
927 sess->reconsider_user_data = user_data;
929 if (callbacks->request_key_unit) {
930 sess->callbacks.request_key_unit = callbacks->request_key_unit;
931 sess->request_key_unit_user_data = user_data;
933 if (callbacks->request_time) {
934 sess->callbacks.request_time = callbacks->request_time;
935 sess->request_time_user_data = user_data;
937 if (callbacks->notify_nack) {
938 sess->callbacks.notify_nack = callbacks->notify_nack;
939 sess->notify_nack_user_data = user_data;
941 if (callbacks->reconfigure) {
942 sess->callbacks.reconfigure = callbacks->reconfigure;
943 sess->reconfigure_user_data = user_data;
948 * rtp_session_set_process_rtp_callback:
949 * @sess: an #RTPSession
950 * @callback: callback to set
951 * @user_data: user data passed in the callback
953 * Configure only the process_rtp callback to be notified of the process_rtp action.
956 rtp_session_set_process_rtp_callback (RTPSession * sess,
957 RTPSessionProcessRTP callback, gpointer user_data)
959 g_return_if_fail (RTP_IS_SESSION (sess));
961 sess->callbacks.process_rtp = callback;
962 sess->process_rtp_user_data = user_data;
966 * rtp_session_set_send_rtp_callback:
967 * @sess: an #RTPSession
968 * @callback: callback to set
969 * @user_data: user data passed in the callback
971 * Configure only the send_rtp callback to be notified of the send_rtp action.
974 rtp_session_set_send_rtp_callback (RTPSession * sess,
975 RTPSessionSendRTP callback, gpointer user_data)
977 g_return_if_fail (RTP_IS_SESSION (sess));
979 sess->callbacks.send_rtp = callback;
980 sess->send_rtp_user_data = user_data;
984 * rtp_session_set_send_rtcp_callback:
985 * @sess: an #RTPSession
986 * @callback: callback to set
987 * @user_data: user data passed in the callback
989 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
992 rtp_session_set_send_rtcp_callback (RTPSession * sess,
993 RTPSessionSendRTCP callback, gpointer user_data)
995 g_return_if_fail (RTP_IS_SESSION (sess));
997 sess->callbacks.send_rtcp = callback;
998 sess->send_rtcp_user_data = user_data;
1002 * rtp_session_set_sync_rtcp_callback:
1003 * @sess: an #RTPSession
1004 * @callback: callback to set
1005 * @user_data: user data passed in the callback
1007 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
1010 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
1011 RTPSessionSyncRTCP callback, gpointer user_data)
1013 g_return_if_fail (RTP_IS_SESSION (sess));
1015 sess->callbacks.sync_rtcp = callback;
1016 sess->sync_rtcp_user_data = user_data;
1020 * rtp_session_set_clock_rate_callback:
1021 * @sess: an #RTPSession
1022 * @callback: callback to set
1023 * @user_data: user data passed in the callback
1025 * Configure only the clock_rate callback to be notified of the clock_rate action.
1028 rtp_session_set_clock_rate_callback (RTPSession * sess,
1029 RTPSessionClockRate callback, gpointer user_data)
1031 g_return_if_fail (RTP_IS_SESSION (sess));
1033 sess->callbacks.clock_rate = callback;
1034 sess->clock_rate_user_data = user_data;
1038 * rtp_session_set_reconsider_callback:
1039 * @sess: an #RTPSession
1040 * @callback: callback to set
1041 * @user_data: user data passed in the callback
1043 * Configure only the reconsider callback to be notified of the reconsider action.
1046 rtp_session_set_reconsider_callback (RTPSession * sess,
1047 RTPSessionReconsider callback, gpointer user_data)
1049 g_return_if_fail (RTP_IS_SESSION (sess));
1051 sess->callbacks.reconsider = callback;
1052 sess->reconsider_user_data = user_data;
1056 * rtp_session_set_request_time_callback:
1057 * @sess: an #RTPSession
1058 * @callback: callback to set
1059 * @user_data: user data passed in the callback
1061 * Configure only the request_time callback
1064 rtp_session_set_request_time_callback (RTPSession * sess,
1065 RTPSessionRequestTime callback, gpointer user_data)
1067 g_return_if_fail (RTP_IS_SESSION (sess));
1069 sess->callbacks.request_time = callback;
1070 sess->request_time_user_data = user_data;
1074 * rtp_session_set_bandwidth:
1075 * @sess: an #RTPSession
1076 * @bandwidth: the bandwidth allocated
1078 * Set the session bandwidth in bytes per second.
1081 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1083 g_return_if_fail (RTP_IS_SESSION (sess));
1085 RTP_SESSION_LOCK (sess);
1086 sess->stats.bandwidth = bandwidth;
1087 RTP_SESSION_UNLOCK (sess);
1091 * rtp_session_get_bandwidth:
1092 * @sess: an #RTPSession
1094 * Get the session bandwidth.
1096 * Returns: the session bandwidth.
1099 rtp_session_get_bandwidth (RTPSession * sess)
1103 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1105 RTP_SESSION_LOCK (sess);
1106 result = sess->stats.bandwidth;
1107 RTP_SESSION_UNLOCK (sess);
1113 * rtp_session_set_rtcp_fraction:
1114 * @sess: an #RTPSession
1115 * @bandwidth: the RTCP bandwidth
1117 * Set the bandwidth in bytes per second that should be used for RTCP
1121 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1123 g_return_if_fail (RTP_IS_SESSION (sess));
1125 RTP_SESSION_LOCK (sess);
1126 sess->stats.rtcp_bandwidth = bandwidth;
1127 RTP_SESSION_UNLOCK (sess);
1131 * rtp_session_get_rtcp_fraction:
1132 * @sess: an #RTPSession
1134 * Get the session bandwidth used for RTCP.
1136 * Returns: The bandwidth used for RTCP messages.
1139 rtp_session_get_rtcp_fraction (RTPSession * sess)
1143 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1145 RTP_SESSION_LOCK (sess);
1146 result = sess->stats.rtcp_bandwidth;
1147 RTP_SESSION_UNLOCK (sess);
1153 * rtp_session_get_sdes_struct:
1154 * @sess: an #RTSPSession
1156 * Get the SDES data as a #GstStructure
1158 * Returns: a GstStructure with SDES items for @sess. This function returns a
1159 * copy of the SDES structure, use gst_structure_free() after usage.
1162 rtp_session_get_sdes_struct (RTPSession * sess)
1164 GstStructure *result = NULL;
1166 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1168 RTP_SESSION_LOCK (sess);
1170 result = gst_structure_copy (sess->sdes);
1171 RTP_SESSION_UNLOCK (sess);
1177 * rtp_session_set_sdes_struct:
1178 * @sess: an #RTSPSession
1179 * @sdes: a #GstStructure
1181 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1184 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1186 g_return_if_fail (sdes);
1187 g_return_if_fail (RTP_IS_SESSION (sess));
1189 RTP_SESSION_LOCK (sess);
1191 gst_structure_free (sess->sdes);
1192 sess->sdes = gst_structure_copy (sdes);
1193 RTP_SESSION_UNLOCK (sess);
1196 static GstFlowReturn
1197 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1199 GstFlowReturn result = GST_FLOW_OK;
1201 if (source->internal) {
1202 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1204 RTP_SESSION_UNLOCK (session);
1206 if (session->callbacks.send_rtp)
1208 session->callbacks.send_rtp (session, source, data,
1209 session->send_rtp_user_data);
1211 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1214 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1215 RTP_SESSION_UNLOCK (session);
1217 if (session->callbacks.process_rtp)
1219 session->callbacks.process_rtp (session, source,
1220 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1222 gst_buffer_unref (GST_BUFFER_CAST (data));
1224 RTP_SESSION_LOCK (session);
1230 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1234 RTP_SESSION_UNLOCK (session);
1236 if (session->callbacks.clock_rate)
1238 session->callbacks.clock_rate (session, pt,
1239 session->clock_rate_user_data);
1243 RTP_SESSION_LOCK (session);
1245 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1250 static RTPSourceCallbacks callbacks = {
1251 (RTPSourcePushRTP) source_push_rtp,
1252 (RTPSourceClockRate) source_clock_rate,
1257 * rtp_session_find_conflicting_address:
1258 * @session: The session the packet came in
1259 * @address: address to check for
1260 * @time: The time when the packet that is possibly in conflict arrived
1262 * Checks if an address which has a conflict is already known. If it is
1263 * a known conflict, remember the time
1265 * Returns: TRUE if it was a known conflict, FALSE otherwise
1268 rtp_session_find_conflicting_address (RTPSession * session,
1269 GSocketAddress * address, GstClockTime time)
1271 return find_conflicting_address (session->conflicting_addresses, address,
1276 * rtp_session_add_conflicting_address:
1277 * @session: The session the packet came in
1278 * @address: address to remember
1279 * @time: The time when the packet that is in conflict arrived
1281 * Adds a new conflict address
1284 rtp_session_add_conflicting_address (RTPSession * sess,
1285 GSocketAddress * address, GstClockTime time)
1287 sess->conflicting_addresses =
1288 add_conflicting_address (sess->conflicting_addresses, address, time);
1293 check_collision (RTPSession * sess, RTPSource * source,
1294 RTPPacketInfo * pinfo, gboolean rtp)
1298 /* If we have no pinfo address, we can't do collision checking */
1299 if (!pinfo->address)
1302 ssrc = rtp_source_get_ssrc (source);
1304 if (!source->internal) {
1305 GSocketAddress *from;
1307 /* This is not our local source, but lets check if two remote
1310 from = source->rtp_from;
1312 from = source->rtcp_from;
1316 if (__g_socket_address_equal (from, pinfo->address)) {
1317 /* Address is the same */
1320 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1321 if (sess->favor_new) {
1322 if (rtp_source_find_conflicting_address (source,
1323 pinfo->address, pinfo->current_time)) {
1326 buf1 = __g_socket_address_to_string (pinfo->address);
1327 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1335 /* Current address is not a known conflict, lets assume this is
1336 * a new source. Save old address in possible conflict list
1338 rtp_source_add_conflicting_address (source, from,
1339 pinfo->current_time);
1341 buf1 = __g_socket_address_to_string (from);
1342 buf2 = __g_socket_address_to_string (pinfo->address);
1344 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1345 " saving old as known conflict", ssrc, buf1, buf2);
1348 rtp_source_set_rtp_from (source, pinfo->address);
1350 rtp_source_set_rtcp_from (source, pinfo->address);
1358 /* Don't need to save old addresses, we ignore new sources */
1363 /* We don't already have a from address for RTP, just set it */
1365 rtp_source_set_rtp_from (source, pinfo->address);
1367 rtp_source_set_rtcp_from (source, pinfo->address);
1371 /* FIXME: Log 3rd party collision somehow
1372 * Maybe should be done in upper layer, only the SDES can tell us
1373 * if its a collision or a loop
1376 /* This is sending with our ssrc, is it an address we already know */
1377 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1378 pinfo->current_time)) {
1379 /* Its a known conflict, its probably a loop, not a collision
1380 * lets just drop the incoming packet
1382 GST_DEBUG ("Our packets are being looped back to us, dropping");
1384 /* Its a new collision, lets change our SSRC */
1385 rtp_session_add_conflicting_address (sess, pinfo->address,
1386 pinfo->current_time);
1388 GST_DEBUG ("Collision for SSRC %x", ssrc);
1389 /* mark the source BYE */
1390 rtp_source_mark_bye (source, "SSRC Collision");
1391 /* if we were suggesting this SSRC, change to something else */
1392 if (sess->suggested_ssrc == ssrc)
1393 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1395 on_ssrc_collision (sess, source);
1397 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1406 gboolean is_doing_ptp;
1407 GSocketAddress *new_addr;
1410 /* check if the two given ip addr are the same (do not care about the port) */
1412 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1415 g_inet_address_equal (g_inet_socket_address_get_address
1416 (G_INET_SOCKET_ADDRESS (a)),
1417 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1421 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1422 CompareAddrData * data)
1424 /* only compare ip addr of remote sources which are also not closing */
1425 if (!source->internal && !source->closing && source->rtp_from) {
1426 /* look for the first rtp source */
1427 if (!data->new_addr)
1428 data->new_addr = source->rtp_from;
1429 /* compare current ip addr with the first one */
1431 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1436 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1437 CompareAddrData * data)
1439 /* only compare ip addr of remote sources which are also not closing */
1440 if (!source->internal && !source->closing && source->rtcp_from) {
1441 /* look for the first rtcp source */
1442 if (!data->new_addr)
1443 data->new_addr = source->rtcp_from;
1445 /* compare current ip addr with the first one */
1446 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1450 /* loop over our non-internal source to know if the session
1451 * is doing point-to-point */
1453 session_update_ptp (RTPSession * sess)
1455 /* to know if the session is doing point to point, the ip addr
1456 * of each non-internal (=remotes) source have to be compared
1459 gboolean is_doing_rtp_ptp;
1460 gboolean is_doing_rtcp_ptp;
1461 CompareAddrData data;
1463 /* compare the first remote source's ip addr that receive rtp packets
1464 * with other remote rtp source.
1465 * it's enough because the session just needs to know if they are all
1468 data.is_doing_ptp = TRUE;
1469 data.new_addr = NULL;
1470 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1471 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1472 is_doing_rtp_ptp = data.is_doing_ptp;
1474 /* same but about rtcp */
1475 data.is_doing_ptp = TRUE;
1476 data.new_addr = NULL;
1477 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1478 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1479 is_doing_rtcp_ptp = data.is_doing_ptp;
1481 /* the session is doing point-to-point if all rtp remote have the same
1482 * ip addr and if all rtcp remote sources have the same ip addr */
1483 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1485 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1489 add_source (RTPSession * sess, RTPSource * src)
1491 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1492 GINT_TO_POINTER (src->ssrc), src);
1493 /* report the new source ASAP */
1494 src->generation = sess->generation;
1495 /* we have one more source now */
1496 sess->total_sources++;
1497 if (RTP_SOURCE_IS_ACTIVE (src))
1498 sess->stats.active_sources++;
1499 if (src->internal) {
1500 sess->stats.internal_sources++;
1501 if (sess->suggested_ssrc != src->ssrc)
1502 sess->suggested_ssrc = src->ssrc;
1505 /* update point-to-point status */
1507 session_update_ptp (sess);
1511 find_source (RTPSession * sess, guint32 ssrc)
1513 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1514 GINT_TO_POINTER (ssrc));
1517 /* must be called with the session lock, the returned source needs to be
1518 * unreffed after usage. */
1520 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1521 RTPPacketInfo * pinfo, gboolean rtp)
1525 source = find_source (sess, ssrc);
1526 if (source == NULL) {
1527 /* make new Source in probation and insert */
1528 source = rtp_source_new (ssrc);
1530 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1532 /* for RTP packets we need to set the source in probation. Receiving RTCP
1533 * packets of an SSRC, on the other hand, is a strong indication that we
1534 * are dealing with a valid source. */
1536 g_object_set (source, "probation", sess->probation, NULL);
1538 g_object_set (source, "probation", 0, NULL);
1540 /* store from address, if any */
1541 if (pinfo->address) {
1543 rtp_source_set_rtp_from (source, pinfo->address);
1545 rtp_source_set_rtcp_from (source, pinfo->address);
1548 /* configure a callback on the source */
1549 rtp_source_set_callbacks (source, &callbacks, sess);
1551 add_source (sess, source);
1555 /* check for collision, this updates the address when not previously set */
1556 if (check_collision (sess, source, pinfo, rtp)) {
1559 /* Receiving RTCP packets of an SSRC is a strong indication that we
1560 * are dealing with a valid source. */
1562 g_object_set (source, "probation", 0, NULL);
1564 /* update last activity */
1565 source->last_activity = pinfo->current_time;
1567 source->last_rtp_activity = pinfo->current_time;
1568 g_object_ref (source);
1573 /* must be called with the session lock, the returned source needs to be
1574 * unreffed after usage. */
1576 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1577 GstClockTime current_time)
1581 source = find_source (sess, ssrc);
1582 if (source == NULL) {
1583 /* make new internal Source and insert */
1584 source = rtp_source_new (ssrc);
1586 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1588 source->validated = TRUE;
1589 source->internal = TRUE;
1590 source->probation = FALSE;
1591 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1592 rtp_source_set_callbacks (source, &callbacks, sess);
1594 add_source (sess, source);
1599 /* update last activity */
1600 if (current_time != GST_CLOCK_TIME_NONE) {
1601 source->last_activity = current_time;
1602 source->last_rtp_activity = current_time;
1604 g_object_ref (source);
1610 * rtp_session_suggest_ssrc:
1611 * @sess: a #RTPSession
1613 * Suggest an unused SSRC in @sess.
1615 * Returns: a free unused SSRC
1618 rtp_session_suggest_ssrc (RTPSession * sess)
1622 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1624 RTP_SESSION_LOCK (sess);
1625 result = sess->suggested_ssrc;
1626 RTP_SESSION_UNLOCK (sess);
1632 * rtp_session_add_source:
1633 * @sess: a #RTPSession
1634 * @src: #RTPSource to add
1636 * Add @src to @session.
1638 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1639 * existed in the session.
1642 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1644 gboolean result = FALSE;
1647 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1648 g_return_val_if_fail (src != NULL, FALSE);
1650 RTP_SESSION_LOCK (sess);
1651 find = find_source (sess, src->ssrc);
1653 add_source (sess, src);
1656 RTP_SESSION_UNLOCK (sess);
1662 * rtp_session_get_num_sources:
1663 * @sess: an #RTPSession
1665 * Get the number of sources in @sess.
1667 * Returns: The number of sources in @sess.
1670 rtp_session_get_num_sources (RTPSession * sess)
1674 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1676 RTP_SESSION_LOCK (sess);
1677 result = sess->total_sources;
1678 RTP_SESSION_UNLOCK (sess);
1684 * rtp_session_get_num_active_sources:
1685 * @sess: an #RTPSession
1687 * Get the number of active sources in @sess. A source is considered active when
1688 * it has been validated and has not yet received a BYE RTCP message.
1690 * Returns: The number of active sources in @sess.
1693 rtp_session_get_num_active_sources (RTPSession * sess)
1697 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1699 RTP_SESSION_LOCK (sess);
1700 result = sess->stats.active_sources;
1701 RTP_SESSION_UNLOCK (sess);
1707 * rtp_session_get_source_by_ssrc:
1708 * @sess: an #RTPSession
1711 * Find the source with @ssrc in @sess.
1713 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1714 * g_object_unref() after usage.
1717 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1721 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1723 RTP_SESSION_LOCK (sess);
1724 result = find_source (sess, ssrc);
1726 g_object_ref (result);
1727 RTP_SESSION_UNLOCK (sess);
1732 /* should be called with the SESSION lock */
1734 rtp_session_create_new_ssrc (RTPSession * sess)
1739 ssrc = g_random_int ();
1741 /* see if it exists in the session, we're done if it doesn't */
1742 if (find_source (sess, ssrc) == NULL)
1750 * rtp_session_create_source:
1751 * @sess: an #RTPSession
1753 * Create an #RTPSource for use in @sess. This function will create a source
1754 * with an ssrc that is currently not used by any participants in the session.
1756 * Returns: an #RTPSource.
1759 rtp_session_create_source (RTPSession * sess)
1764 RTP_SESSION_LOCK (sess);
1765 ssrc = rtp_session_create_new_ssrc (sess);
1766 source = rtp_source_new (ssrc);
1767 rtp_source_set_callbacks (source, &callbacks, sess);
1768 /* we need an additional ref for the source in the hashtable */
1769 g_object_ref (source);
1770 add_source (sess, source);
1771 RTP_SESSION_UNLOCK (sess);
1777 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1779 GstNetAddressMeta *meta;
1781 /* get packet size including header overhead */
1782 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1786 GstRTPBuffer rtp = { NULL };
1788 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1789 goto invalid_packet;
1791 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1795 /* only keep info for first buffer */
1796 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1797 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1798 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1799 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1800 /* copy available csrc */
1801 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1802 for (i = 0; i < pinfo->csrc_count; i++)
1803 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1805 gst_rtp_buffer_unmap (&rtp);
1809 /* for netbuffer we can store the IP address to check for collisions */
1810 meta = gst_buffer_get_net_address_meta (*buffer);
1812 g_object_unref (pinfo->address);
1814 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1816 pinfo->address = NULL;
1824 GST_DEBUG ("invalid RTP packet received");
1829 /* update the RTPPacketInfo structure with the current time and other bits
1830 * about the current buffer we are handling.
1831 * This function is typically called when a validated packet is received.
1832 * This function should be called with the SESSION_LOCK
1835 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1836 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1837 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1843 pinfo->is_list = is_list;
1845 pinfo->current_time = current_time;
1846 pinfo->running_time = running_time;
1847 pinfo->ntpnstime = ntpnstime;
1848 pinfo->header_len = sess->header_len;
1850 pinfo->payload_len = 0;
1854 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1856 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1859 GstBuffer *buffer = GST_BUFFER_CAST (data);
1860 res = update_packet (&buffer, 0, pinfo);
1866 clean_packet_info (RTPPacketInfo * pinfo)
1869 g_object_unref (pinfo->address);
1871 gst_mini_object_unref (pinfo->data);
1877 source_update_active (RTPSession * sess, RTPSource * source,
1878 gboolean prevactive)
1880 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1881 guint32 ssrc = source->ssrc;
1883 if (prevactive == active)
1887 sess->stats.active_sources++;
1888 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1889 sess->stats.active_sources);
1891 sess->stats.active_sources--;
1892 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1893 sess->stats.active_sources);
1899 source_update_sender (RTPSession * sess, RTPSource * source,
1900 gboolean prevsender)
1902 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1903 guint32 ssrc = source->ssrc;
1905 if (prevsender == sender)
1909 sess->stats.sender_sources++;
1910 if (source->internal)
1911 sess->stats.internal_sender_sources++;
1912 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1913 sess->stats.sender_sources);
1915 sess->stats.sender_sources--;
1916 if (source->internal)
1917 sess->stats.internal_sender_sources--;
1918 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1919 sess->stats.sender_sources);
1925 * rtp_session_process_rtp:
1926 * @sess: and #RTPSession
1927 * @buffer: an RTP buffer
1928 * @current_time: the current system time
1929 * @running_time: the running_time of @buffer
1931 * Process an RTP buffer in the session manager. This function takes ownership
1934 * Returns: a #GstFlowReturn.
1937 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1938 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1940 GstFlowReturn result;
1944 gboolean prevsender, prevactive;
1945 RTPPacketInfo pinfo = { 0, };
1948 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1949 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1951 RTP_SESSION_LOCK (sess);
1953 /* update pinfo stats */
1954 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1955 current_time, running_time, ntpnstime)) {
1956 GST_DEBUG ("invalid RTP packet received");
1957 RTP_SESSION_UNLOCK (sess);
1958 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1963 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1967 prevsender = RTP_SOURCE_IS_SENDER (source);
1968 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1969 oldrate = source->bitrate;
1971 /* let source process the packet */
1972 result = rtp_source_process_rtp (source, &pinfo);
1974 /* source became active */
1975 if (source_update_active (sess, source, prevactive))
1976 on_ssrc_validated (sess, source);
1978 source_update_sender (sess, source, prevsender);
1980 if (oldrate != source->bitrate)
1981 sess->recalc_bandwidth = TRUE;
1984 on_new_ssrc (sess, source);
1986 if (source->validated) {
1990 /* for validated sources, we add the CSRCs as well */
1991 for (i = 0; i < pinfo.csrc_count; i++) {
1993 RTPSource *csrc_src;
1995 csrc = pinfo.csrcs[i];
1998 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
2003 GST_DEBUG ("created new CSRC: %08x", csrc);
2004 rtp_source_set_as_csrc (csrc_src);
2005 source_update_active (sess, csrc_src, FALSE);
2006 on_new_ssrc (sess, csrc_src);
2008 g_object_unref (csrc_src);
2011 g_object_unref (source);
2013 RTP_SESSION_UNLOCK (sess);
2015 clean_packet_info (&pinfo);
2022 RTP_SESSION_UNLOCK (sess);
2023 clean_packet_info (&pinfo);
2024 GST_DEBUG ("ignoring packet because its collisioning");
2030 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
2031 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
2035 count = gst_rtcp_packet_get_rb_count (packet);
2036 for (i = 0; i < count; i++) {
2037 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
2038 guint8 fractionlost;
2042 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
2043 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2045 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2047 /* find our own source */
2048 src = find_source (sess, ssrc);
2052 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2053 /* only deal with report blocks for our session, we update the stats of
2054 * the sender of the RTCP message. We could also compare our stats against
2055 * the other sender to see if we are better or worse. */
2056 /* FIXME, need to keep track who the RB block is from */
2057 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2058 packetslost, exthighestseq, jitter, lsr, dlsr);
2061 on_ssrc_active (sess, source);
2064 /* A Sender report contains statistics about how the sender is doing. This
2065 * includes timing informataion such as the relation between RTP and NTP
2066 * timestamps and the number of packets/bytes it sent to us.
2068 * In this report is also included a set of report blocks related to how this
2069 * sender is receiving data (in case we (or somebody else) is also sending stuff
2070 * to it). This info includes the packet loss, jitter and seqnum. It also
2071 * contains information to calculate the round trip time (LSR/DLSR).
2074 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2075 RTPPacketInfo * pinfo, gboolean * do_sync)
2077 guint32 senderssrc, rtptime, packet_count, octet_count;
2080 gboolean created, prevsender;
2082 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2083 &packet_count, &octet_count);
2085 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2086 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2088 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2092 /* skip non-bye packets for sources that are marked BYE */
2093 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2096 /* don't try to do lip-sync for sources that sent a BYE */
2097 if (RTP_SOURCE_IS_MARKED_BYE (source))
2102 prevsender = RTP_SOURCE_IS_SENDER (source);
2104 /* first update the source */
2105 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2106 packet_count, octet_count);
2108 source_update_sender (sess, source, prevsender);
2111 on_new_ssrc (sess, source);
2113 rtp_session_process_rb (sess, source, packet, pinfo);
2116 g_object_unref (source);
2119 /* A receiver report contains statistics about how a receiver is doing. It
2120 * includes stuff like packet loss, jitter and the seqnum it received last. It
2121 * also contains info to calculate the round trip time.
2123 * We are only interested in how the sender of this report is doing wrt to us.
2126 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2127 RTPPacketInfo * pinfo)
2133 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2135 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2137 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2141 /* skip non-bye packets for sources that are marked BYE */
2142 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2146 on_new_ssrc (sess, source);
2148 rtp_session_process_rb (sess, source, packet, pinfo);
2151 g_object_unref (source);
2154 /* Get SDES items and store them in the SSRC */
2156 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2157 RTPPacketInfo * pinfo)
2160 gboolean more_items, more_entries;
2162 items = gst_rtcp_packet_sdes_get_item_count (packet);
2163 GST_DEBUG ("got SDES packet with %d items", items);
2165 more_items = gst_rtcp_packet_sdes_first_item (packet);
2167 while (more_items) {
2169 gboolean changed, created, prevactive;
2173 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2175 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2179 /* find src, no probation when dealing with RTCP */
2180 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2184 /* skip non-bye packets for sources that are marked BYE */
2185 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2188 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2190 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2192 while (more_entries) {
2193 GstRTCPSDESType type;
2199 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2201 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2204 if (type == GST_RTCP_SDES_PRIV) {
2205 name = g_strndup ((const gchar *) &data[1], data[0]);
2207 data += data[0] + 1;
2209 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2212 value = g_strndup ((const gchar *) data, len);
2214 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2219 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2223 /* takes ownership of sdes */
2224 changed = rtp_source_set_sdes_struct (source, sdes);
2226 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2227 source->validated = TRUE;
2230 on_new_ssrc (sess, source);
2232 /* source became active */
2233 if (source_update_active (sess, source, prevactive))
2234 on_ssrc_validated (sess, source);
2237 on_ssrc_sdes (sess, source);
2240 g_object_unref (source);
2242 more_items = gst_rtcp_packet_sdes_next_item (packet);
2247 /* BYE is sent when a client leaves the session
2250 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2251 RTPPacketInfo * pinfo)
2255 gboolean reconsider = FALSE;
2257 reason = gst_rtcp_packet_bye_get_reason (packet);
2258 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2260 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2261 for (i = 0; i < count; i++) {
2264 gboolean created, prevactive, prevsender;
2265 guint pmembers, members;
2267 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2268 GST_DEBUG ("SSRC: %08x", ssrc);
2270 /* find src and mark bye, no probation when dealing with RTCP */
2271 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2275 if (source->internal) {
2276 /* our own source, something weird with this packet */
2277 g_object_unref (source);
2281 /* store time for when we need to time out this source */
2282 source->bye_time = pinfo->current_time;
2284 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2285 prevsender = RTP_SOURCE_IS_SENDER (source);
2287 /* mark the source BYE */
2288 rtp_source_mark_bye (source, reason);
2290 pmembers = sess->stats.active_sources;
2292 source_update_active (sess, source, prevactive);
2293 source_update_sender (sess, source, prevsender);
2295 members = sess->stats.active_sources;
2297 if (!sess->scheduled_bye && members < pmembers) {
2298 /* some members went away since the previous timeout estimate.
2299 * Perform reverse reconsideration but only when we are not scheduling a
2301 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2302 pinfo->current_time < sess->next_rtcp_check_time) {
2303 GstClockTime time_remaining;
2305 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2306 sess->next_rtcp_check_time =
2307 gst_util_uint64_scale (time_remaining, members, pmembers);
2309 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2310 GST_TIME_ARGS (sess->next_rtcp_check_time));
2312 sess->next_rtcp_check_time += pinfo->current_time;
2314 /* mark pending reconsider. We only want to signal the reconsideration
2315 * once after we handled all the source in the bye packet */
2321 on_new_ssrc (sess, source);
2323 on_bye_ssrc (sess, source);
2325 g_object_unref (source);
2328 RTP_SESSION_UNLOCK (sess);
2329 /* notify app of reconsideration */
2330 if (sess->callbacks.reconsider)
2331 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2332 RTP_SESSION_LOCK (sess);
2338 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2339 RTPPacketInfo * pinfo)
2341 GST_DEBUG ("received APP");
2345 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2346 gboolean fir, GstClockTime current_time)
2348 guint32 round_trip = 0;
2350 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2352 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2353 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2356 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2357 GST_DEBUG ("Ignoring %s request because one was send without one "
2358 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2359 fir ? "FIR" : "PLI",
2360 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2361 GST_TIME_ARGS (round_trip_in_ns));;
2366 sess->last_keyframe_request = current_time;
2368 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2369 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2370 sess->callbacks.request_key_unit);
2372 RTP_SESSION_UNLOCK (sess);
2373 sess->callbacks.request_key_unit (sess, fir,
2374 sess->request_key_unit_user_data);
2375 RTP_SESSION_LOCK (sess);
2381 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2382 guint32 media_ssrc, GstClockTime current_time)
2386 if (!sess->callbacks.request_key_unit)
2389 src = find_source (sess, sender_ssrc);
2393 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2397 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2398 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2403 gboolean our_request = FALSE;
2405 if (!sess->callbacks.request_key_unit)
2411 src = find_source (sess, sender_ssrc);
2413 /* Hack because Google fails to set the sender_ssrc correctly */
2414 if (!src && sender_ssrc == 1) {
2415 GHashTableIter iter;
2417 /* we can't find the source if there are multiple */
2418 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2421 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2422 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2423 if (!src->internal && rtp_source_is_sender (src))
2431 for (position = 0; position < fci_length; position += 8) {
2432 guint8 *data = fci_data + position;
2435 ssrc = GST_READ_UINT32_BE (data);
2437 own = find_source (sess, ssrc);
2441 if (own->internal) {
2449 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2453 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2454 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2455 GstClockTime current_time)
2457 sess->stats.nacks_received++;
2459 if (!sess->callbacks.notify_nack)
2462 while (fci_length > 0) {
2463 guint16 seqnum, blp;
2465 seqnum = GST_READ_UINT16_BE (fci_data);
2466 blp = GST_READ_UINT16_BE (fci_data + 2);
2468 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2470 RTP_SESSION_UNLOCK (sess);
2471 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2472 sess->notify_nack_user_data);
2473 RTP_SESSION_LOCK (sess);
2481 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2482 RTPPacketInfo * pinfo, GstClockTime current_time)
2484 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2485 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2486 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2487 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2488 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2489 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2492 src = find_source (sess, media_ssrc);
2494 /* skip non-bye packets for sources that are marked BYE */
2495 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2498 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2499 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2501 if (g_signal_has_handler_pending (sess,
2502 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2503 GstBuffer *fci_buffer = NULL;
2505 if (fci_length > 0) {
2506 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2507 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2509 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2512 RTP_SESSION_UNLOCK (sess);
2513 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2514 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2515 RTP_SESSION_LOCK (sess);
2518 gst_buffer_unref (fci_buffer);
2521 if (src && sess->rtcp_feedback_retention_window) {
2522 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2525 if ((src && src->internal) ||
2526 /* PSFB FIR puts the media ssrc inside the FCI */
2527 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2529 case GST_RTCP_TYPE_PSFB:
2531 case GST_RTCP_PSFB_TYPE_PLI:
2532 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2535 case GST_RTCP_PSFB_TYPE_FIR:
2536 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2543 case GST_RTCP_TYPE_RTPFB:
2545 case GST_RTCP_RTPFB_TYPE_NACK:
2546 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2547 fci_data, fci_length, current_time);
2559 * rtp_session_process_rtcp:
2560 * @sess: and #RTPSession
2561 * @buffer: an RTCP buffer
2562 * @current_time: the current system time
2563 * @ntpnstime: the current NTP time in nanoseconds
2565 * Process an RTCP buffer in the session manager. This function takes ownership
2568 * Returns: a #GstFlowReturn.
2571 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2572 GstClockTime current_time, guint64 ntpnstime)
2574 GstRTCPPacket packet;
2575 gboolean more, is_bye = FALSE, do_sync = FALSE;
2576 RTPPacketInfo pinfo = { 0, };
2577 GstFlowReturn result = GST_FLOW_OK;
2578 GstRTCPBuffer rtcp = { NULL, };
2580 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2581 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2583 if (!gst_rtcp_buffer_validate (buffer))
2584 goto invalid_packet;
2586 GST_DEBUG ("received RTCP packet");
2588 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
2591 RTP_SESSION_LOCK (sess);
2592 /* update pinfo stats */
2593 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2596 /* start processing the compound packet */
2597 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2598 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2602 type = gst_rtcp_packet_get_type (&packet);
2605 case GST_RTCP_TYPE_SR:
2606 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2608 case GST_RTCP_TYPE_RR:
2609 rtp_session_process_rr (sess, &packet, &pinfo);
2611 case GST_RTCP_TYPE_SDES:
2612 rtp_session_process_sdes (sess, &packet, &pinfo);
2614 case GST_RTCP_TYPE_BYE:
2616 /* don't try to attempt lip-sync anymore for streams with a BYE */
2618 rtp_session_process_bye (sess, &packet, &pinfo);
2620 case GST_RTCP_TYPE_APP:
2621 rtp_session_process_app (sess, &packet, &pinfo);
2623 case GST_RTCP_TYPE_RTPFB:
2624 case GST_RTCP_TYPE_PSFB:
2625 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2628 GST_WARNING ("got unknown RTCP packet");
2631 more = gst_rtcp_packet_move_to_next (&packet);
2634 gst_rtcp_buffer_unmap (&rtcp);
2636 /* if we are scheduling a BYE, we only want to count bye packets, else we
2637 * count everything */
2638 if (sess->scheduled_bye && is_bye) {
2639 sess->bye_stats.bye_members++;
2640 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2643 /* keep track of average packet size */
2644 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2646 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2647 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2648 RTP_SESSION_UNLOCK (sess);
2651 clean_packet_info (&pinfo);
2653 /* notify caller of sr packets in the callback */
2654 if (do_sync && sess->callbacks.sync_rtcp) {
2655 result = sess->callbacks.sync_rtcp (sess, buffer,
2656 sess->sync_rtcp_user_data);
2658 gst_buffer_unref (buffer);
2665 GST_DEBUG ("invalid RTCP packet received");
2666 gst_buffer_unref (buffer);
2672 * rtp_session_update_send_caps:
2673 * @sess: an #RTPSession
2676 * Update the caps of the sender in the rtp session.
2679 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2684 g_return_if_fail (RTP_IS_SESSION (sess));
2685 g_return_if_fail (GST_IS_CAPS (caps));
2687 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2689 s = gst_caps_get_structure (caps, 0);
2691 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2695 RTP_SESSION_LOCK (sess);
2696 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2698 rtp_source_update_caps (source, caps);
2699 g_object_unref (source);
2701 RTP_SESSION_UNLOCK (sess);
2706 * rtp_session_send_rtp:
2707 * @sess: an #RTPSession
2708 * @data: pointer to either an RTP buffer or a list of RTP buffers
2709 * @is_list: TRUE when @data is a buffer list
2710 * @current_time: the current system time
2711 * @running_time: the running time of @data
2713 * Send the RTP buffer in the session manager. This function takes ownership of
2716 * Returns: a #GstFlowReturn.
2719 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2720 GstClockTime current_time, GstClockTime running_time)
2722 GstFlowReturn result;
2724 gboolean prevsender;
2726 RTPPacketInfo pinfo = { 0, };
2729 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2730 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2732 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2734 RTP_SESSION_LOCK (sess);
2735 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2736 current_time, running_time, -1))
2737 goto invalid_packet;
2739 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
2741 prevsender = RTP_SOURCE_IS_SENDER (source);
2742 oldrate = source->bitrate;
2744 /* we use our own source to send */
2745 result = rtp_source_send_rtp (source, &pinfo);
2747 source_update_sender (sess, source, prevsender);
2749 if (oldrate != source->bitrate)
2750 sess->recalc_bandwidth = TRUE;
2751 RTP_SESSION_UNLOCK (sess);
2753 g_object_unref (source);
2754 clean_packet_info (&pinfo);
2760 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2761 RTP_SESSION_UNLOCK (sess);
2762 GST_DEBUG ("invalid RTP packet received");
2768 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2770 *bandwidth += source->bitrate;
2773 /* must be called with session lock */
2775 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2778 GstClockTime result;
2779 RTPSessionStats *stats;
2781 /* recalculate bandwidth when it changed */
2782 if (sess->recalc_bandwidth) {
2785 if (sess->bandwidth > 0)
2786 bandwidth = sess->bandwidth;
2788 /* If it is <= 0, then try to estimate the actual bandwidth */
2791 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2792 (GHFunc) add_bitrates, &bandwidth);
2795 if (bandwidth < 8000)
2796 bandwidth = RTP_STATS_BANDWIDTH;
2798 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2799 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2801 sess->recalc_bandwidth = FALSE;
2804 if (sess->scheduled_bye) {
2805 stats = &sess->bye_stats;
2806 result = rtp_stats_calculate_bye_interval (stats);
2808 stats = &sess->stats;
2809 result = rtp_stats_calculate_rtcp_interval (stats,
2810 stats->internal_sender_sources > 0, first);
2813 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2814 GST_TIME_ARGS (result), first);
2816 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2817 result = rtp_stats_add_rtcp_jitter (stats, result);
2819 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2825 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2827 if (source->internal)
2828 rtp_source_mark_bye (source, reason);
2832 * rtp_session_mark_all_bye:
2833 * @sess: an #RTPSession
2836 * Mark all internal sources of the session as BYE with @reason.
2839 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2841 g_return_if_fail (RTP_IS_SESSION (sess));
2843 RTP_SESSION_LOCK (sess);
2844 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2845 (GHFunc) source_mark_bye, (gpointer) reason);
2846 RTP_SESSION_UNLOCK (sess);
2849 /* Stop the current @sess and schedule a BYE message for the other members.
2850 * One must have the session lock to call this function
2852 static GstFlowReturn
2853 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2855 GstFlowReturn result = GST_FLOW_OK;
2856 GstClockTime interval;
2858 /* nothing to do it we already scheduled bye */
2859 if (sess->scheduled_bye)
2862 /* we schedule BYE now */
2863 sess->scheduled_bye = TRUE;
2864 /* at least one member wants to send a BYE */
2865 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2866 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2867 sess->bye_stats.bye_members = 1;
2868 sess->first_rtcp = TRUE;
2869 sess->allow_early = TRUE;
2871 /* reschedule transmission */
2872 sess->last_rtcp_send_time = current_time;
2873 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2875 if (interval != GST_CLOCK_TIME_NONE)
2876 sess->next_rtcp_check_time = current_time + interval;
2878 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2880 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2881 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2883 RTP_SESSION_UNLOCK (sess);
2884 /* notify app of reconsideration */
2885 if (sess->callbacks.reconsider)
2886 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2887 RTP_SESSION_LOCK (sess);
2894 * rtp_session_schedule_bye:
2895 * @sess: an #RTPSession
2896 * @current_time: the current system time
2898 * Schedule a BYE message for all sources marked as BYE in @sess.
2900 * Returns: a #GstFlowReturn.
2903 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2905 GstFlowReturn result;
2907 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2909 RTP_SESSION_LOCK (sess);
2910 result = rtp_session_schedule_bye_locked (sess, current_time);
2911 RTP_SESSION_UNLOCK (sess);
2917 * rtp_session_next_timeout:
2918 * @sess: an #RTPSession
2919 * @current_time: the current system time
2921 * Get the next time we should perform session maintenance tasks.
2923 * Returns: a time when rtp_session_on_timeout() should be called with the
2924 * current system time.
2927 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2929 GstClockTime result, interval = 0;
2931 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2933 RTP_SESSION_LOCK (sess);
2935 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2936 GST_DEBUG ("have early rtcp time");
2937 result = sess->next_early_rtcp_time;
2941 result = sess->next_rtcp_check_time;
2943 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2944 ", next time: %" GST_TIME_FORMAT,
2945 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2947 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2948 GST_DEBUG ("take current time as base");
2949 /* our previous check time expired, start counting from the current time
2951 result = current_time;
2954 if (sess->scheduled_bye) {
2955 if (sess->bye_stats.active_sources >= 50) {
2956 GST_DEBUG ("reconsider BYE, more than 50 sources");
2957 /* reconsider BYE if members >= 50 */
2958 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2961 if (sess->first_rtcp) {
2962 GST_DEBUG ("first RTCP packet");
2963 /* we are called for the first time */
2964 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2965 } else if (sess->next_rtcp_check_time < current_time) {
2966 GST_DEBUG ("old check time expired, getting new timeout");
2967 /* get a new timeout when we need to */
2968 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2972 if (interval != GST_CLOCK_TIME_NONE)
2975 result = GST_CLOCK_TIME_NONE;
2977 sess->next_rtcp_check_time = result;
2981 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2982 ", next time: %" GST_TIME_FORMAT,
2983 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2984 RTP_SESSION_UNLOCK (sess);
2998 GstRTCPBuffer rtcpbuf;
3001 guint num_to_report;
3006 GstClockTime current_time;
3008 GstClockTime running_time;
3009 GstClockTime interval;
3010 GstRTCPPacket packet;
3013 gboolean may_suppress;
3015 guint nacked_seqnums;
3019 session_start_rtcp (RTPSession * sess, ReportData * data)
3021 GstRTCPPacket *packet = &data->packet;
3022 RTPSource *own = data->source;
3023 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3025 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
3026 data->has_sdes = FALSE;
3028 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3030 if (RTP_SOURCE_IS_SENDER (own)) {
3033 guint32 packet_count, octet_count;
3035 /* we are a sender, create SR */
3036 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
3037 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
3039 /* get latest stats */
3040 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
3041 &ntptime, &rtptime, &packet_count, &octet_count);
3043 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
3044 packet_count, octet_count);
3046 /* fill in sender report info */
3047 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
3048 ntptime, rtptime, packet_count, octet_count);
3050 /* we are only receiver, create RR */
3051 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3052 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3053 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3057 /* construct a Sender or Receiver Report */
3059 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3061 RTPSession *sess = data->sess;
3062 GstRTCPPacket *packet = &data->packet;
3063 guint8 fractionlost;
3065 guint32 exthighestseq, jitter;
3068 /* don't report for sources in future generations */
3069 if (((gint16) (source->generation - sess->generation)) > 0) {
3070 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3071 source->generation, sess->generation);
3075 if (g_hash_table_contains (source->reported_in_sr_of,
3076 GUINT_TO_POINTER (data->source->ssrc))) {
3077 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3081 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3082 GST_DEBUG ("max RB count reached");
3086 /* only report about other sender */
3087 if (source == data->source)
3090 if (!RTP_SOURCE_IS_SENDER (source)) {
3091 GST_DEBUG ("source %08x not sender", source->ssrc);
3095 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3098 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3099 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3101 /* store last generated RR packet */
3102 source->last_rr.is_valid = TRUE;
3103 source->last_rr.fractionlost = fractionlost;
3104 source->last_rr.packetslost = packetslost;
3105 source->last_rr.exthighestseq = exthighestseq;
3106 source->last_rr.jitter = jitter;
3107 source->last_rr.lsr = lsr;
3108 source->last_rr.dlsr = dlsr;
3110 /* packet is not yet filled, add report block for this source. */
3111 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3112 exthighestseq, jitter, lsr, dlsr);
3115 g_hash_table_add (source->reported_in_sr_of,
3116 GUINT_TO_POINTER (data->source->ssrc));
3121 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3123 GstRTCPPacket *packet = &data->packet;
3127 if (!source->send_fir)
3130 len = gst_rtcp_packet_fb_get_fci_length (packet);
3131 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3132 /* exit because the packet is full, will put next request in a
3136 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3138 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3140 fci_data[0] = source->current_send_fir_seqnum;
3141 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3143 source->send_fir = FALSE;
3147 session_fir (RTPSession * sess, ReportData * data)
3149 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3150 GstRTCPPacket *packet = &data->packet;
3152 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3155 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3156 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3157 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3159 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3160 (GHFunc) session_add_fir, data);
3162 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3163 gst_rtcp_packet_remove (packet);
3165 data->may_suppress = FALSE;
3169 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3171 GstRTCPPacket packet;
3172 GstRTCPBuffer rtcp = { NULL, };
3173 gboolean ret = FALSE;
3175 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3177 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3178 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3179 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3183 gst_rtcp_buffer_unmap (&rtcp);
3190 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3192 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3193 GstRTCPPacket *packet = &data->packet;
3195 if (!source->send_pli)
3198 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3201 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3202 /* exit because the packet is full, will put next request in a
3206 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3207 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3208 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3210 source->send_pli = FALSE;
3211 data->may_suppress = FALSE;
3214 /* construct NACK */
3216 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3218 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3219 GstRTCPPacket *packet = &data->packet;
3224 if (!source->send_nack)
3227 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3228 /* exit because the packet is full, will put next request in a
3232 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3233 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3234 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3236 nacks = rtp_source_get_nacks (source, &n_nacks);
3237 GST_DEBUG ("%u NACKs", n_nacks);
3238 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3241 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3242 for (i = 0; i < n_nacks; i++) {
3243 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3245 data->nacked_seqnums++;
3248 rtp_source_clear_nacks (source);
3249 data->may_suppress = FALSE;
3252 /* perform cleanup of sources that timed out */
3254 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3256 gboolean remove = FALSE;
3257 gboolean byetimeout = FALSE;
3258 gboolean sendertimeout = FALSE;
3259 gboolean is_sender, is_active;
3260 RTPSession *sess = data->sess;
3261 GstClockTime interval, binterval;
3264 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3266 /* check for outdated collisions */
3267 if (source->internal) {
3268 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3269 rtp_source_timeout (source, data->current_time,
3270 data->running_time - sess->rtcp_feedback_retention_window);
3273 /* nothing else to do when without RTCP */
3274 if (data->interval == GST_CLOCK_TIME_NONE)
3277 is_sender = RTP_SOURCE_IS_SENDER (source);
3278 is_active = RTP_SOURCE_IS_ACTIVE (source);
3280 /* our own rtcp interval may have been forced low by secondary configuration,
3281 * while sender side may still operate with higher interval,
3282 * so do not just take our interval to decide on timing out sender,
3283 * but take (if data->interval <= 5 * GST_SECOND):
3284 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3285 * where sender_interval is difference between last 2 received RTCP reports
3287 if (data->interval >= 5 * GST_SECOND || source->internal) {
3288 binterval = data->interval;
3290 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3291 GST_TIME_ARGS (source->stats.prev_rtcptime),
3292 GST_TIME_ARGS (source->stats.last_rtcptime));
3293 /* if not received enough yet, fallback to larger default */
3294 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3295 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3297 binterval = 5 * GST_SECOND;
3298 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3300 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3301 GST_TIME_ARGS (binterval));
3303 if (!source->internal && source->marked_bye) {
3304 /* if we received a BYE from the source, remove the source after some
3306 if (data->current_time > source->bye_time &&
3307 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3308 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3314 if (source->internal && source->sent_bye) {
3315 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3319 /* sources that were inactive for more than 5 times the deterministic reporting
3320 * interval get timed out. the min timeout is 5 seconds. */
3321 /* mind old time that might pre-date last time going to PLAYING */
3322 btime = MAX (source->last_activity, sess->start_time);
3323 if (data->current_time > btime) {
3324 interval = MAX (binterval * 5, 5 * GST_SECOND);
3325 if (data->current_time - btime > interval) {
3326 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3327 source->ssrc, GST_TIME_ARGS (btime));
3328 if (source->internal) {
3329 /* this is an internal source that is not using our suggested ssrc.
3330 * since there must be another source using this ssrc, we can remove
3331 * this one instead of making it a receiver forever */
3332 if (source->ssrc != sess->suggested_ssrc) {
3333 rtp_source_mark_bye (source, "timed out");
3334 /* do not schedule bye here, since we are inside the RTCP timeout
3335 * processing and scheduling bye will interfere with SR/RR sending */
3343 /* senders that did not send for a long time become a receiver, this also
3344 * holds for our own sources. */
3346 /* mind old time that might pre-date last time going to PLAYING */
3347 btime = MAX (source->last_rtp_activity, sess->start_time);
3348 if (data->current_time > btime) {
3349 interval = MAX (binterval * 2, 5 * GST_SECOND);
3350 if (data->current_time - btime > interval) {
3351 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3352 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3353 sendertimeout = TRUE;
3359 sess->total_sources--;
3361 sess->stats.sender_sources--;
3362 if (source->internal)
3363 sess->stats.internal_sender_sources--;
3366 sess->stats.active_sources--;
3368 if (source->internal)
3369 sess->stats.internal_sources--;
3372 on_bye_timeout (sess, source);
3374 on_timeout (sess, source);
3376 if (sendertimeout) {
3377 source->is_sender = FALSE;
3378 sess->stats.sender_sources--;
3379 if (source->internal)
3380 sess->stats.internal_sender_sources--;
3382 on_sender_timeout (sess, source);
3384 /* count how many source to report in this generation */
3385 if (((gint16) (source->generation - sess->generation)) <= 0)
3386 data->num_to_report++;
3388 source->closing = remove;
3392 session_sdes (RTPSession * sess, ReportData * data)
3394 GstRTCPPacket *packet = &data->packet;
3395 const GstStructure *sdes;
3397 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3399 /* add SDES packet */
3400 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3402 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3404 sdes = rtp_source_get_sdes_struct (data->source);
3406 /* add all fields in the structure, the order is not important. */
3407 n_fields = gst_structure_n_fields (sdes);
3408 for (i = 0; i < n_fields; ++i) {
3411 GstRTCPSDESType type;
3413 field = gst_structure_nth_field_name (sdes, i);
3416 value = gst_structure_get_string (sdes, field);
3419 type = gst_rtcp_sdes_name_to_type (field);
3421 /* Early packets are minimal and only include the CNAME */
3422 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3425 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3426 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3427 (const guint8 *) value);
3428 } else if (type == GST_RTCP_SDES_PRIV) {
3434 /* don't accept entries that are too big */
3435 prefix_len = strlen (field);
3436 if (prefix_len > 255)
3438 value_len = strlen (value);
3439 if (value_len > 255)
3441 data_len = 1 + prefix_len + value_len;
3445 data[0] = prefix_len;
3446 memcpy (&data[1], field, prefix_len);
3447 memcpy (&data[1 + prefix_len], value, value_len);
3449 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3453 data->has_sdes = TRUE;
3456 /* schedule a BYE packet */
3458 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3460 GstRTCPPacket *packet = &data->packet;
3461 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3464 session_sdes (sess, data);
3465 /* add a BYE packet */
3466 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3467 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3468 if (source->bye_reason)
3469 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3471 /* we have a BYE packet now */
3472 source->sent_bye = TRUE;
3476 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3478 GstClockTime new_send_time;
3479 GstClockTime interval;
3480 RTPSessionStats *stats;
3482 if (sess->scheduled_bye)
3483 stats = &sess->bye_stats;
3485 stats = &sess->stats;
3487 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3488 data->is_early = TRUE;
3490 data->is_early = FALSE;
3492 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3493 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3494 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3495 GST_TIME_ARGS (current_time));
3499 /* no need to check yet */
3500 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3501 sess->next_rtcp_check_time > current_time) {
3502 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3503 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3504 GST_TIME_ARGS (current_time));
3510 /* take interval and add jitter */
3511 interval = data->interval;
3512 if (interval != GST_CLOCK_TIME_NONE)
3513 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3515 if (sess->last_rtcp_send_time != GST_CLOCK_TIME_NONE) {
3516 /* perform forward reconsideration */
3517 if (interval != GST_CLOCK_TIME_NONE) {
3518 GstClockTime elapsed;
3520 /* get elapsed time since we last reported */
3521 elapsed = current_time - sess->last_rtcp_send_time;
3523 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3524 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3525 new_send_time = interval + sess->last_rtcp_send_time;
3527 new_send_time = sess->last_rtcp_send_time;
3530 /* If this is the first RTCP packet, we can reconsider anything based
3531 * on the last RTCP send time because there was none.
3533 g_warn_if_fail (!data->is_early);
3534 data->is_early = FALSE;
3535 new_send_time = current_time;
3538 if (!data->is_early) {
3539 /* check if reconsideration */
3540 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3541 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3542 GST_TIME_ARGS (new_send_time));
3543 /* store new check time */
3544 sess->next_rtcp_check_time = new_send_time;
3547 sess->next_rtcp_check_time = current_time + interval;
3548 } else if (interval != GST_CLOCK_TIME_NONE) {
3549 /* Apply the rules from RFC 4585 section 3.5.3 */
3550 if (stats->min_interval != 0 && !sess->first_rtcp) {
3551 GstClockTime T_rr_current_interval =
3552 g_random_double_range (0.5, 1.5) * stats->min_interval;
3554 /* This will caused the RTCP to be suppressed if no FB packets are added */
3555 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3556 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3557 " last: %" GST_TIME_FORMAT
3558 " + T_rr_current_interval: %" GST_TIME_FORMAT
3559 " > new_send_time: %" GST_TIME_FORMAT,
3560 GST_TIME_ARGS (stats->min_interval),
3561 GST_TIME_ARGS (sess->last_rtcp_send_time),
3562 GST_TIME_ARGS (T_rr_current_interval),
3563 GST_TIME_ARGS (new_send_time));
3564 data->may_suppress = TRUE;
3569 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3570 GST_TIME_ARGS (new_send_time));
3576 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3578 g_hash_table_insert (hash_table, key, g_object_ref (source));
3582 remove_closing_sources (const gchar * key, RTPSource * source,
3585 if (source->closing)
3588 if (source->send_fir)
3589 data->have_fir = TRUE;
3590 if (source->send_pli)
3591 data->have_pli = TRUE;
3592 if (source->send_nack)
3593 data->have_nack = TRUE;
3599 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3601 RTPSession *sess = data->sess;
3602 gboolean is_bye = FALSE;
3603 ReportOutput *output;
3605 /* only generate RTCP for active internal sources */
3606 if (!source->internal || source->sent_bye)
3609 /* ignore other sources when we do the timeout after a scheduled BYE */
3610 if (sess->scheduled_bye && !source->marked_bye)
3613 data->source = source;
3616 session_start_rtcp (sess, data);
3618 if (source->marked_bye) {
3620 make_source_bye (sess, source, data);
3622 } else if (!data->is_early) {
3623 /* loop over all known sources and add report blocks. If we are early, we
3624 * just make a minimal RTCP packet and skip this step */
3625 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3626 (GHFunc) session_report_blocks, data);
3628 if (!data->has_sdes)
3629 session_sdes (sess, data);
3632 session_fir (sess, data);
3635 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3636 (GHFunc) session_pli, data);
3638 if (data->have_nack)
3639 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3640 (GHFunc) session_nack, data);
3642 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3644 output = g_slice_new (ReportOutput);
3645 output->source = g_object_ref (source);
3646 output->is_bye = is_bye;
3647 output->buffer = data->rtcp;
3648 /* queue the RTCP packet to push later */
3649 g_queue_push_tail (&data->output, output);
3653 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3655 RTPSession *sess = data->sess;
3657 if (g_hash_table_size (source->reported_in_sr_of) >=
3658 sess->stats.internal_sources) {
3659 /* source is reported, move to next generation */
3660 source->generation = sess->generation + 1;
3661 g_hash_table_remove_all (source->reported_in_sr_of);
3663 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3664 source->generation);
3666 /* if we reported all sources in this generation, move to next */
3667 if (--data->num_to_report == 0) {
3669 GST_DEBUG ("all reported, generation now %u", sess->generation);
3675 * rtp_session_on_timeout:
3676 * @sess: an #RTPSession
3677 * @current_time: the current system time
3678 * @ntpnstime: the current NTP time in nanoseconds
3679 * @running_time: the current running_time of the pipeline
3681 * Perform maintenance actions after the timeout obtained with
3682 * rtp_session_next_timeout() expired.
3684 * This function will perform timeouts of receivers and senders, send a BYE
3685 * packet or generate RTCP packets with current session stats.
3687 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3688 * times, for each packet that should be processed.
3690 * Returns: a #GstFlowReturn.
3693 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3694 guint64 ntpnstime, GstClockTime running_time)
3696 GstFlowReturn result = GST_FLOW_OK;
3697 ReportData data = { GST_RTCP_BUFFER_INIT };
3698 GHashTable *table_copy;
3699 ReportOutput *output;
3701 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3703 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3704 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3705 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3708 data.current_time = current_time;
3709 data.ntpnstime = ntpnstime;
3710 data.running_time = running_time;
3711 data.num_to_report = 0;
3712 data.may_suppress = FALSE;
3713 data.nacked_seqnums = 0;
3714 g_queue_init (&data.output);
3716 RTP_SESSION_LOCK (sess);
3717 /* get a new interval, we need this for various cleanups etc */
3718 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3720 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3722 /* we need an internal source now */
3723 if (sess->stats.internal_sources == 0) {
3727 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
3729 g_object_unref (source);
3732 sess->conflicting_addresses =
3733 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
3735 /* Make a local copy of the hashtable. We need to do this because the
3736 * cleanup stage below releases the session lock. */
3737 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3738 (GDestroyNotify) g_object_unref);
3739 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3740 (GHFunc) clone_ssrcs_hashtable, table_copy);
3742 /* Clean up the session, mark the source for removing, this might release the
3744 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3745 g_hash_table_destroy (table_copy);
3747 /* Now remove the marked sources */
3748 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3749 (GHRFunc) remove_closing_sources, &data);
3751 /* update point-to-point status */
3752 session_update_ptp (sess);
3754 /* see if we need to generate SR or RR packets */
3755 if (!is_rtcp_time (sess, current_time, &data))
3758 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3759 sess->generation, data.num_to_report, data.is_early);
3761 /* generate RTCP for all internal sources */
3762 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3763 (GHFunc) generate_rtcp, &data);
3765 /* update the generation for all the sources that have been reported */
3766 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3767 (GHFunc) update_generation, &data);
3769 /* we keep track of the last report time in order to timeout inactive
3770 * receivers or senders */
3771 if (!data.is_early && !data.may_suppress)
3772 sess->last_rtcp_send_time = data.current_time;
3773 sess->first_rtcp = FALSE;
3774 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3775 sess->scheduled_bye = FALSE;
3777 /* RFC 4585 section 3.5.2 step 6 */
3778 if (!data.is_early) {
3779 sess->allow_early = TRUE;
3783 RTP_SESSION_UNLOCK (sess);
3785 /* push out the RTCP packets */
3786 while ((output = g_queue_pop_head (&data.output))) {
3787 gboolean do_not_suppress;
3788 GstBuffer *buffer = output->buffer;
3789 RTPSource *source = output->source;
3791 /* Give the user a change to add its own packet */
3792 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3793 buffer, data.is_early, &do_not_suppress);
3795 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3798 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3800 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3801 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3802 sess->stats.avg_rtcp_packet_size, packet_size);
3804 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3805 sess->send_rtcp_user_data);
3806 sess->stats.nacks_sent += data.nacked_seqnums;
3808 GST_DEBUG ("freeing packet callback: %p"
3809 " do_not_suppress: %d may_suppress: %d",
3810 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3811 sess->stats.nacks_dropped += data.nacked_seqnums;
3812 gst_buffer_unref (buffer);
3814 g_object_unref (source);
3815 g_slice_free (ReportOutput, output);
3821 * rtp_session_request_early_rtcp:
3822 * @sess: an #RTPSession
3823 * @current_time: the current system time
3824 * @max_delay: maximum delay
3826 * Request transmission of early RTCP
3828 * Returns: %TRUE if the related RTCP can be scheduled.
3831 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3832 GstClockTime max_delay)
3834 GstClockTime T_dither_max, T_rr;
3837 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3839 RTP_SESSION_LOCK (sess);
3841 /* Check if already requested */
3842 /* RFC 4585 section 3.5.2 step 2 */
3843 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3844 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3849 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3850 GST_LOG_OBJECT (sess, "no next RTCP check time");
3855 /* RFC 4585 section 3.5.3 step 1
3856 * If no regular RTCP packet has been sent before, then a regular
3857 * RTCP packet has to be scheduled first and FB messages might be
3860 if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
3861 GST_LOG_OBJECT (sess, "no RTCP sent yet");
3863 if (current_time + max_delay > sess->next_rtcp_check_time) {
3864 GST_LOG_OBJECT (sess,
3865 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
3866 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3867 GST_TIME_ARGS (max_delay),
3868 GST_TIME_ARGS (sess->next_rtcp_check_time));
3876 T_rr = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3878 /* RFC 4585 section 3.5.2 step 2b */
3879 /* If the total sources is <=2, then there is only us and one peer */
3880 /* When there is one auxiliary stream the session can still do point
3883 if (sess->is_doing_ptp) {
3886 /* Divide by 2 because l = 0.5 */
3887 T_dither_max = T_rr;
3891 /* RFC 4585 section 3.5.2 step 3 */
3892 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3893 GST_LOG_OBJECT (sess, "don't send because of dither");
3898 /* RFC 4585 section 3.5.2 step 4a */
3899 if (sess->allow_early == FALSE) {
3900 /* Ignore the request a scheduled packet will be in time anyway */
3901 if (current_time + max_delay > sess->next_rtcp_check_time) {
3902 GST_LOG_OBJECT (sess,
3903 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
3904 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3905 GST_TIME_ARGS (max_delay),
3906 GST_TIME_ARGS (sess->next_rtcp_check_time));
3909 GST_LOG_OBJECT (sess, "can't allow early feedback");
3915 /* RFC 4585 section 3.5.2 step 4b */
3917 /* Schedule an early transmission later */
3918 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3921 /* If no dithering, schedule it for NOW */
3922 sess->next_early_rtcp_time = current_time;
3925 /* RFC 4585 section 3.5.2 step 6 */
3926 sess->allow_early = FALSE;
3927 /* Delay next regular RTCP packet to not exceed the short-term
3928 * RTCP bandwidth when using early feedback as compared to
3930 sess->next_rtcp_check_time = sess->last_rtcp_send_time + 2 * T_rr;
3931 sess->last_rtcp_send_time += T_rr;
3933 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3934 GST_TIME_ARGS (sess->next_early_rtcp_time));
3935 RTP_SESSION_UNLOCK (sess);
3937 /* notify app of need to send packet early
3938 * and therefore of timeout change */
3939 if (sess->callbacks.reconsider)
3940 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3946 RTP_SESSION_UNLOCK (sess);
3952 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3956 if (!sess->callbacks.send_rtcp)
3959 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3961 return rtp_session_request_early_rtcp (sess, now, max_delay);
3965 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3966 gboolean fir, gint count)
3970 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
3971 GST_DEBUG ("FIR/PLI not sent");
3975 RTP_SESSION_LOCK (sess);
3976 src = find_source (sess, ssrc);
3981 src->send_pli = FALSE;
3982 src->send_fir = TRUE;
3984 if (count == -1 || count != src->last_fir_count)
3985 src->current_send_fir_seqnum++;
3986 src->last_fir_count = count;
3987 } else if (!src->send_fir) {
3988 src->send_pli = TRUE;
3990 RTP_SESSION_UNLOCK (sess);
3997 RTP_SESSION_UNLOCK (sess);
4003 * rtp_session_request_nack:
4004 * @sess: a #RTPSession
4006 * @seqnum: the missing seqnum
4007 * @max_delay: max delay to request NACK
4009 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
4011 * Returns: %TRUE if the NACK feedback could be scheduled
4014 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
4015 GstClockTime max_delay)
4019 if (!rtp_session_send_rtcp (sess, max_delay)) {
4020 GST_DEBUG ("NACK not sent");
4024 RTP_SESSION_LOCK (sess);
4025 source = find_source (sess, ssrc);
4029 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
4030 rtp_source_register_nack (source, seqnum);
4031 RTP_SESSION_UNLOCK (sess);
4038 RTP_SESSION_UNLOCK (sess);