2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "gstrtpbin-marshal.h"
32 #include "rtpsession.h"
34 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
35 #define GST_CAT_DEFAULT rtp_session_debug
37 /* signals and args */
40 SIGNAL_GET_SOURCE_BY_SSRC,
42 SIGNAL_ON_SSRC_COLLISION,
43 SIGNAL_ON_SSRC_VALIDATED,
44 SIGNAL_ON_SSRC_ACTIVE,
47 SIGNAL_ON_BYE_TIMEOUT,
49 SIGNAL_ON_SENDER_TIMEOUT,
50 SIGNAL_ON_SENDING_RTCP,
51 SIGNAL_ON_FEEDBACK_RTCP,
56 #define DEFAULT_INTERNAL_SOURCE NULL
57 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
58 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
59 #define DEFAULT_RTCP_RR_BANDWIDTH -1
60 #define DEFAULT_RTCP_RS_BANDWIDTH -1
61 #define DEFAULT_RTCP_MTU 1400
62 #define DEFAULT_SDES NULL
63 #define DEFAULT_NUM_SOURCES 0
64 #define DEFAULT_NUM_ACTIVE_SOURCES 0
65 #define DEFAULT_SOURCES NULL
66 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
67 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
68 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
69 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
78 PROP_RTCP_RR_BANDWIDTH,
79 PROP_RTCP_RS_BANDWIDTH,
83 PROP_NUM_ACTIVE_SOURCES,
86 PROP_RTCP_MIN_INTERVAL,
87 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
88 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
93 /* update average packet size */
94 #define INIT_AVG(avg, val) \
96 #define UPDATE_AVG(avg, val) \
100 (avg) = ((val) + (15 * (avg))) >> 4;
103 /* The number RTCP intervals after which to timeout entries in the
106 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
108 /* GObject vmethods */
109 static void rtp_session_finalize (GObject * object);
110 static void rtp_session_set_property (GObject * object, guint prop_id,
111 const GValue * value, GParamSpec * pspec);
112 static void rtp_session_get_property (GObject * object, guint prop_id,
113 GValue * value, GParamSpec * pspec);
115 static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
116 GstBuffer * buffer, gboolean early);
117 static void rtp_session_send_rtcp (RTPSession * sess,
118 GstClockTimeDiff max_delay);
121 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
123 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
125 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
126 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
127 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
128 static RTPSource *obtain_internal_source (RTPSession * sess,
129 guint32 ssrc, gboolean * created);
130 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
131 GstClockTime current_time);
132 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
133 gboolean deterministic, gboolean first);
136 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
137 const GValue * handler_return, gpointer data)
139 if (g_value_get_boolean (handler_return))
140 g_value_set_boolean (return_accu, TRUE);
146 rtp_session_class_init (RTPSessionClass * klass)
148 GObjectClass *gobject_class;
150 gobject_class = (GObjectClass *) klass;
152 gobject_class->finalize = rtp_session_finalize;
153 gobject_class->set_property = rtp_session_set_property;
154 gobject_class->get_property = rtp_session_get_property;
157 * RTPSession::get-source-by-ssrc:
158 * @session: the object which received the signal
159 * @ssrc: the SSRC of the RTPSource
161 * Request the #RTPSource object with SSRC @ssrc in @session.
163 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
164 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
165 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
166 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
167 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
170 * RTPSession::on-new-ssrc:
171 * @session: the object which received the signal
172 * @src: the new RTPSource
174 * Notify of a new SSRC that entered @session.
176 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
177 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
178 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
179 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
182 * RTPSession::on-ssrc-collision:
183 * @session: the object which received the signal
184 * @src: the #RTPSource that caused a collision
186 * Notify when we have an SSRC collision
188 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
189 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
190 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
191 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
194 * RTPSession::on-ssrc-validated:
195 * @session: the object which received the signal
196 * @src: the new validated RTPSource
198 * Notify of a new SSRC that became validated.
200 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
201 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
202 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
203 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
206 * RTPSession::on-ssrc-active:
207 * @session: the object which received the signal
208 * @src: the active RTPSource
210 * Notify of a SSRC that is active, i.e., sending RTCP.
212 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
213 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
214 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
215 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
218 * RTPSession::on-ssrc-sdes:
219 * @session: the object which received the signal
220 * @src: the RTPSource
222 * Notify that a new SDES was received for SSRC.
224 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
225 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
226 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
227 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
230 * RTPSession::on-bye-ssrc:
231 * @session: the object which received the signal
232 * @src: the RTPSource that went away
234 * Notify of an SSRC that became inactive because of a BYE packet.
236 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
237 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
238 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
239 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
242 * RTPSession::on-bye-timeout:
243 * @session: the object which received the signal
244 * @src: the RTPSource that timed out
246 * Notify of an SSRC that has timed out because of BYE
248 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
249 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
250 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
251 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
254 * RTPSession::on-timeout:
255 * @session: the object which received the signal
256 * @src: the RTPSource that timed out
258 * Notify of an SSRC that has timed out
260 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
261 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
262 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
263 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
266 * RTPSession::on-sender-timeout:
267 * @session: the object which received the signal
268 * @src: the RTPSource that timed out
270 * Notify of an SSRC that was a sender but timed out and became a receiver.
272 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
273 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
274 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
275 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
279 * RTPSession::on-sending-rtcp
280 * @session: the object which received the signal
281 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
282 * @early: %TRUE if the packet is early, %FALSE if it is regular
284 * This signal is emitted before sending an RTCP packet, it can be used
285 * to add extra RTCP Packets.
287 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
288 * if suppressing it is acceptable
290 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
291 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
292 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
293 accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__BOXED_BOOLEAN,
294 G_TYPE_BOOLEAN, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
298 * RTPSession::on-feedback-rtcp:
299 * @session: the object which received the signal
300 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
301 * %GST_RTCP_TYPE_RTPFB
302 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
303 * @sender_ssrc: The SSRC of the sender
304 * @media_ssrc: The SSRC of the media this refers to
305 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
308 * Notify that a RTCP feedback packet has been received
310 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
311 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
312 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
313 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_BOXED,
314 G_TYPE_NONE, 5, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
318 * RTPSession::send-rtcp:
319 * @session: the object which received the signal
320 * @max_delay: The maximum delay after which the feedback will not be useful
323 * Requests that the #RTPSession initiate a new RTCP packet as soon as
324 * possible within the requested delay.
327 rtp_session_signals[SIGNAL_SEND_RTCP] =
328 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
329 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
330 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
331 gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64);
333 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
334 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
335 "The internal SSRC used for the session (deprecated)",
336 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
339 g_param_spec_object ("internal-source", "Internal Source",
340 "The internal source element of the session (deprecated)",
341 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
344 g_param_spec_double ("bandwidth", "Bandwidth",
345 "The bandwidth of the session (0 for auto-discover)",
346 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
347 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
350 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
351 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
352 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
353 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
356 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
357 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
358 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
359 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
361 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
362 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
363 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
364 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
365 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
367 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
368 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
369 "The maximum size of the RTCP packets",
370 16, G_MAXINT16, DEFAULT_RTCP_MTU,
371 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
373 g_object_class_install_property (gobject_class, PROP_SDES,
374 g_param_spec_boxed ("sdes", "SDES",
375 "The SDES items of this session",
376 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
379 g_param_spec_uint ("num-sources", "Num Sources",
380 "The number of sources in the session", 0, G_MAXUINT,
381 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
383 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
384 g_param_spec_uint ("num-active-sources", "Num Active Sources",
385 "The number of active sources in the session", 0, G_MAXUINT,
386 DEFAULT_NUM_ACTIVE_SOURCES,
387 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
391 * Get a GValue Array of all sources in the session.
394 * <title>Getting the #RTPSources of a session
401 * g_object_get (sess, "sources", &arr, NULL);
403 * for (i = 0; i < arr->n_values; i++) {
406 * val = g_value_array_get_nth (arr, i);
407 * source = g_value_get_object (val);
409 * g_value_array_free (arr);
414 g_object_class_install_property (gobject_class, PROP_SOURCES,
415 g_param_spec_boxed ("sources", "Sources",
416 "An array of all known sources in the session",
417 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
419 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
420 g_param_spec_boolean ("favor-new", "Favor new sources",
421 "Resolve SSRC conflict in favor of new sources", FALSE,
422 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
424 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
425 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
426 "Minimum interval between Regular RTCP packet (in ns)",
427 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
428 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 g_object_class_install_property (gobject_class,
431 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
432 g_param_spec_uint64 ("rtcp-feedback-retention-window",
433 "RTCP Feedback retention window",
434 "Duration during which RTCP Feedback packets are retained (in ns)",
435 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
438 g_object_class_install_property (gobject_class,
439 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
440 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
441 "RTCP Immediate Feedback threshold",
442 "The maximum number of members of a RTP session for which immediate"
444 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
445 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
447 g_object_class_install_property (gobject_class, PROP_PROBATION,
448 g_param_spec_uint ("probation", "Number of probations",
449 "Consecutive packet sequence numbers to accept the source",
450 0, G_MAXUINT, DEFAULT_PROBATION,
451 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
453 klass->get_source_by_ssrc =
454 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
455 klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
456 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
458 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
462 rtp_session_init (RTPSession * sess)
469 g_mutex_init (&sess->lock);
470 sess->key = g_random_int ();
474 for (i = 0; i < 32; i++) {
476 g_hash_table_new_full (NULL, NULL, NULL,
477 (GDestroyNotify) g_object_unref);
480 rtp_stats_init_defaults (&sess->stats);
481 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
482 rtp_stats_set_min_interval (&sess->stats,
483 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
485 sess->recalc_bandwidth = TRUE;
486 sess->bandwidth = DEFAULT_BANDWIDTH;
487 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
488 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
489 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
491 /* default UDP header length */
492 sess->header_len = 28;
493 sess->mtu = DEFAULT_RTCP_MTU;
495 sess->probation = DEFAULT_PROBATION;
497 /* some default SDES entries */
498 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
500 /* we do not want to leak details like the username or hostname here */
501 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
502 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
506 /* we do not want to leak the user's real name here */
507 str = g_strdup_printf ("Anon%u", g_random_int ());
508 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
512 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
514 /* create an active SSRC for this session manager */
515 ssrc = rtp_session_create_new_ssrc (sess);
516 sess->source = obtain_internal_source (sess, ssrc, &created);
518 sess->first_rtcp = TRUE;
519 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
521 sess->allow_early = TRUE;
522 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
523 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
524 sess->rtcp_immediate_feedback_threshold =
525 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
527 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
531 rtp_session_finalize (GObject * object)
536 sess = RTP_SESSION_CAST (object);
538 gst_structure_free (sess->sdes);
540 for (i = 0; i < 32; i++)
541 g_hash_table_destroy (sess->ssrcs[i]);
543 g_object_unref (sess->source);
544 g_mutex_clear (&sess->lock);
546 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
550 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
552 GValue value = { 0 };
554 g_value_init (&value, RTP_TYPE_SOURCE);
555 g_value_take_object (&value, source);
556 /* copies the value */
557 g_value_array_append (arr, &value);
561 rtp_session_create_sources (RTPSession * sess)
566 RTP_SESSION_LOCK (sess);
567 /* get number of elements in the table */
568 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
569 /* create the result value array */
570 res = g_value_array_new (size);
572 /* and copy all values into the array */
573 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
574 RTP_SESSION_UNLOCK (sess);
580 rtp_session_set_property (GObject * object, guint prop_id,
581 const GValue * value, GParamSpec * pspec)
585 sess = RTP_SESSION (object);
588 case PROP_INTERNAL_SSRC:
591 RTP_SESSION_LOCK (sess);
592 sess->bandwidth = g_value_get_double (value);
593 sess->recalc_bandwidth = TRUE;
594 RTP_SESSION_UNLOCK (sess);
596 case PROP_RTCP_FRACTION:
597 RTP_SESSION_LOCK (sess);
598 sess->rtcp_bandwidth = g_value_get_double (value);
599 sess->recalc_bandwidth = TRUE;
600 RTP_SESSION_UNLOCK (sess);
602 case PROP_RTCP_RR_BANDWIDTH:
603 RTP_SESSION_LOCK (sess);
604 sess->rtcp_rr_bandwidth = g_value_get_int (value);
605 sess->recalc_bandwidth = TRUE;
606 RTP_SESSION_UNLOCK (sess);
608 case PROP_RTCP_RS_BANDWIDTH:
609 RTP_SESSION_LOCK (sess);
610 sess->rtcp_rs_bandwidth = g_value_get_int (value);
611 sess->recalc_bandwidth = TRUE;
612 RTP_SESSION_UNLOCK (sess);
615 sess->mtu = g_value_get_uint (value);
618 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
621 sess->favor_new = g_value_get_boolean (value);
623 case PROP_RTCP_MIN_INTERVAL:
624 rtp_stats_set_min_interval (&sess->stats,
625 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
626 /* trigger reconsideration */
627 RTP_SESSION_LOCK (sess);
628 sess->next_rtcp_check_time = 0;
629 RTP_SESSION_UNLOCK (sess);
630 if (sess->callbacks.reconsider)
631 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
633 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
634 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
637 sess->probation = g_value_get_uint (value);
640 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
646 rtp_session_get_property (GObject * object, guint prop_id,
647 GValue * value, GParamSpec * pspec)
651 sess = RTP_SESSION (object);
654 case PROP_INTERNAL_SSRC:
655 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
657 case PROP_INTERNAL_SOURCE:
658 g_value_set_object (value, sess->source);
661 g_value_set_double (value, sess->bandwidth);
663 case PROP_RTCP_FRACTION:
664 g_value_set_double (value, sess->rtcp_bandwidth);
666 case PROP_RTCP_RR_BANDWIDTH:
667 g_value_set_int (value, sess->rtcp_rr_bandwidth);
669 case PROP_RTCP_RS_BANDWIDTH:
670 g_value_set_int (value, sess->rtcp_rs_bandwidth);
673 g_value_set_uint (value, sess->mtu);
676 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
678 case PROP_NUM_SOURCES:
679 g_value_set_uint (value, rtp_session_get_num_sources (sess));
681 case PROP_NUM_ACTIVE_SOURCES:
682 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
685 g_value_take_boxed (value, rtp_session_create_sources (sess));
688 g_value_set_boolean (value, sess->favor_new);
690 case PROP_RTCP_MIN_INTERVAL:
691 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
693 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
694 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
697 g_value_set_uint (value, sess->probation);
700 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
706 on_new_ssrc (RTPSession * sess, RTPSource * source)
708 g_object_ref (source);
709 RTP_SESSION_UNLOCK (sess);
710 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
711 RTP_SESSION_LOCK (sess);
712 g_object_unref (source);
716 on_ssrc_collision (RTPSession * sess, RTPSource * source)
718 g_object_ref (source);
719 RTP_SESSION_UNLOCK (sess);
720 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
722 RTP_SESSION_LOCK (sess);
723 g_object_unref (source);
727 on_ssrc_validated (RTPSession * sess, RTPSource * source)
729 g_object_ref (source);
730 RTP_SESSION_UNLOCK (sess);
731 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
733 RTP_SESSION_LOCK (sess);
734 g_object_unref (source);
738 on_ssrc_active (RTPSession * sess, RTPSource * source)
740 g_object_ref (source);
741 RTP_SESSION_UNLOCK (sess);
742 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
743 RTP_SESSION_LOCK (sess);
744 g_object_unref (source);
748 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
750 g_object_ref (source);
751 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
752 RTP_SESSION_UNLOCK (sess);
753 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
754 RTP_SESSION_LOCK (sess);
755 g_object_unref (source);
759 on_bye_ssrc (RTPSession * sess, RTPSource * source)
761 g_object_ref (source);
762 RTP_SESSION_UNLOCK (sess);
763 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
764 RTP_SESSION_LOCK (sess);
765 g_object_unref (source);
769 on_bye_timeout (RTPSession * sess, RTPSource * source)
771 g_object_ref (source);
772 RTP_SESSION_UNLOCK (sess);
773 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
774 RTP_SESSION_LOCK (sess);
775 g_object_unref (source);
779 on_timeout (RTPSession * sess, RTPSource * source)
781 g_object_ref (source);
782 RTP_SESSION_UNLOCK (sess);
783 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
784 RTP_SESSION_LOCK (sess);
785 g_object_unref (source);
789 on_sender_timeout (RTPSession * sess, RTPSource * source)
791 g_object_ref (source);
792 RTP_SESSION_UNLOCK (sess);
793 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
795 RTP_SESSION_LOCK (sess);
796 g_object_unref (source);
802 * Create a new session object.
804 * Returns: a new #RTPSession. g_object_unref() after usage.
807 rtp_session_new (void)
811 sess = g_object_new (RTP_TYPE_SESSION, NULL);
817 * rtp_session_set_callbacks:
818 * @sess: an #RTPSession
819 * @callbacks: callbacks to configure
820 * @user_data: user data passed in the callbacks
822 * Configure a set of callbacks to be notified of actions.
825 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
828 g_return_if_fail (RTP_IS_SESSION (sess));
830 if (callbacks->process_rtp) {
831 sess->callbacks.process_rtp = callbacks->process_rtp;
832 sess->process_rtp_user_data = user_data;
834 if (callbacks->send_rtp) {
835 sess->callbacks.send_rtp = callbacks->send_rtp;
836 sess->send_rtp_user_data = user_data;
838 if (callbacks->send_rtcp) {
839 sess->callbacks.send_rtcp = callbacks->send_rtcp;
840 sess->send_rtcp_user_data = user_data;
842 if (callbacks->sync_rtcp) {
843 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
844 sess->sync_rtcp_user_data = user_data;
846 if (callbacks->clock_rate) {
847 sess->callbacks.clock_rate = callbacks->clock_rate;
848 sess->clock_rate_user_data = user_data;
850 if (callbacks->reconsider) {
851 sess->callbacks.reconsider = callbacks->reconsider;
852 sess->reconsider_user_data = user_data;
854 if (callbacks->request_key_unit) {
855 sess->callbacks.request_key_unit = callbacks->request_key_unit;
856 sess->request_key_unit_user_data = user_data;
858 if (callbacks->request_time) {
859 sess->callbacks.request_time = callbacks->request_time;
860 sess->request_time_user_data = user_data;
865 * rtp_session_set_process_rtp_callback:
866 * @sess: an #RTPSession
867 * @callback: callback to set
868 * @user_data: user data passed in the callback
870 * Configure only the process_rtp callback to be notified of the process_rtp action.
873 rtp_session_set_process_rtp_callback (RTPSession * sess,
874 RTPSessionProcessRTP callback, gpointer user_data)
876 g_return_if_fail (RTP_IS_SESSION (sess));
878 sess->callbacks.process_rtp = callback;
879 sess->process_rtp_user_data = user_data;
883 * rtp_session_set_send_rtp_callback:
884 * @sess: an #RTPSession
885 * @callback: callback to set
886 * @user_data: user data passed in the callback
888 * Configure only the send_rtp callback to be notified of the send_rtp action.
891 rtp_session_set_send_rtp_callback (RTPSession * sess,
892 RTPSessionSendRTP callback, gpointer user_data)
894 g_return_if_fail (RTP_IS_SESSION (sess));
896 sess->callbacks.send_rtp = callback;
897 sess->send_rtp_user_data = user_data;
901 * rtp_session_set_send_rtcp_callback:
902 * @sess: an #RTPSession
903 * @callback: callback to set
904 * @user_data: user data passed in the callback
906 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
909 rtp_session_set_send_rtcp_callback (RTPSession * sess,
910 RTPSessionSendRTCP callback, gpointer user_data)
912 g_return_if_fail (RTP_IS_SESSION (sess));
914 sess->callbacks.send_rtcp = callback;
915 sess->send_rtcp_user_data = user_data;
919 * rtp_session_set_sync_rtcp_callback:
920 * @sess: an #RTPSession
921 * @callback: callback to set
922 * @user_data: user data passed in the callback
924 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
927 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
928 RTPSessionSyncRTCP callback, gpointer user_data)
930 g_return_if_fail (RTP_IS_SESSION (sess));
932 sess->callbacks.sync_rtcp = callback;
933 sess->sync_rtcp_user_data = user_data;
937 * rtp_session_set_clock_rate_callback:
938 * @sess: an #RTPSession
939 * @callback: callback to set
940 * @user_data: user data passed in the callback
942 * Configure only the clock_rate callback to be notified of the clock_rate action.
945 rtp_session_set_clock_rate_callback (RTPSession * sess,
946 RTPSessionClockRate callback, gpointer user_data)
948 g_return_if_fail (RTP_IS_SESSION (sess));
950 sess->callbacks.clock_rate = callback;
951 sess->clock_rate_user_data = user_data;
955 * rtp_session_set_reconsider_callback:
956 * @sess: an #RTPSession
957 * @callback: callback to set
958 * @user_data: user data passed in the callback
960 * Configure only the reconsider callback to be notified of the reconsider action.
963 rtp_session_set_reconsider_callback (RTPSession * sess,
964 RTPSessionReconsider callback, gpointer user_data)
966 g_return_if_fail (RTP_IS_SESSION (sess));
968 sess->callbacks.reconsider = callback;
969 sess->reconsider_user_data = user_data;
973 * rtp_session_set_request_time_callback:
974 * @sess: an #RTPSession
975 * @callback: callback to set
976 * @user_data: user data passed in the callback
978 * Configure only the request_time callback
981 rtp_session_set_request_time_callback (RTPSession * sess,
982 RTPSessionRequestTime callback, gpointer user_data)
984 g_return_if_fail (RTP_IS_SESSION (sess));
986 sess->callbacks.request_time = callback;
987 sess->request_time_user_data = user_data;
991 * rtp_session_set_bandwidth:
992 * @sess: an #RTPSession
993 * @bandwidth: the bandwidth allocated
995 * Set the session bandwidth in bytes per second.
998 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1000 g_return_if_fail (RTP_IS_SESSION (sess));
1002 RTP_SESSION_LOCK (sess);
1003 sess->stats.bandwidth = bandwidth;
1004 RTP_SESSION_UNLOCK (sess);
1008 * rtp_session_get_bandwidth:
1009 * @sess: an #RTPSession
1011 * Get the session bandwidth.
1013 * Returns: the session bandwidth.
1016 rtp_session_get_bandwidth (RTPSession * sess)
1020 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1022 RTP_SESSION_LOCK (sess);
1023 result = sess->stats.bandwidth;
1024 RTP_SESSION_UNLOCK (sess);
1030 * rtp_session_set_rtcp_fraction:
1031 * @sess: an #RTPSession
1032 * @bandwidth: the RTCP bandwidth
1034 * Set the bandwidth in bytes per second that should be used for RTCP
1038 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1040 g_return_if_fail (RTP_IS_SESSION (sess));
1042 RTP_SESSION_LOCK (sess);
1043 sess->stats.rtcp_bandwidth = bandwidth;
1044 RTP_SESSION_UNLOCK (sess);
1048 * rtp_session_get_rtcp_fraction:
1049 * @sess: an #RTPSession
1051 * Get the session bandwidth used for RTCP.
1053 * Returns: The bandwidth used for RTCP messages.
1056 rtp_session_get_rtcp_fraction (RTPSession * sess)
1060 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1062 RTP_SESSION_LOCK (sess);
1063 result = sess->stats.rtcp_bandwidth;
1064 RTP_SESSION_UNLOCK (sess);
1070 * rtp_session_get_sdes_struct:
1071 * @sess: an #RTSPSession
1073 * Get the SDES data as a #GstStructure
1075 * Returns: a GstStructure with SDES items for @sess. This function returns a
1076 * copy of the SDES structure, use gst_structure_free() after usage.
1079 rtp_session_get_sdes_struct (RTPSession * sess)
1081 GstStructure *result = NULL;
1083 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1085 RTP_SESSION_LOCK (sess);
1087 result = gst_structure_copy (sess->sdes);
1088 RTP_SESSION_UNLOCK (sess);
1094 * rtp_session_set_sdes_struct:
1095 * @sess: an #RTSPSession
1096 * @sdes: a #GstStructure
1098 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1101 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1103 g_return_if_fail (sdes);
1104 g_return_if_fail (RTP_IS_SESSION (sess));
1106 RTP_SESSION_LOCK (sess);
1108 gst_structure_free (sess->sdes);
1109 sess->sdes = gst_structure_copy (sdes);
1110 RTP_SESSION_UNLOCK (sess);
1113 static GstFlowReturn
1114 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1116 GstFlowReturn result = GST_FLOW_OK;
1118 if (source->internal) {
1119 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1121 RTP_SESSION_UNLOCK (session);
1123 if (session->callbacks.send_rtp)
1125 session->callbacks.send_rtp (session, source, data,
1126 session->send_rtp_user_data);
1128 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1131 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1132 RTP_SESSION_UNLOCK (session);
1134 if (session->callbacks.process_rtp)
1136 session->callbacks.process_rtp (session, source,
1137 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1139 gst_buffer_unref (GST_BUFFER_CAST (data));
1141 RTP_SESSION_LOCK (session);
1147 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1151 RTP_SESSION_UNLOCK (session);
1153 if (session->callbacks.clock_rate)
1155 session->callbacks.clock_rate (session, pt,
1156 session->clock_rate_user_data);
1160 RTP_SESSION_LOCK (session);
1162 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1167 static RTPSourceCallbacks callbacks = {
1168 (RTPSourcePushRTP) source_push_rtp,
1169 (RTPSourceClockRate) source_clock_rate,
1173 check_collision (RTPSession * sess, RTPSource * source,
1174 RTPArrivalStats * arrival, gboolean rtp)
1176 /* If we have no arrival address, we can't do collision checking */
1177 if (!arrival->address)
1180 if (!source->internal) {
1181 GSocketAddress *from;
1183 /* This is not our local source, but lets check if two remote
1186 from = source->rtp_from;
1188 from = source->rtcp_from;
1192 if (__g_socket_address_equal (from, arrival->address)) {
1193 /* Address is the same */
1196 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1197 rtp_source_get_ssrc (source));
1198 if (sess->favor_new) {
1199 if (rtp_source_find_conflicting_address (source,
1200 arrival->address, arrival->current_time)) {
1203 buf1 = __g_socket_address_to_string (arrival->address);
1204 GST_LOG ("Known conflict on %x for %s, dropping packet",
1205 rtp_source_get_ssrc (source), buf1);
1212 /* Current address is not a known conflict, lets assume this is
1213 * a new source. Save old address in possible conflict list
1215 rtp_source_add_conflicting_address (source, from,
1216 arrival->current_time);
1218 buf1 = __g_socket_address_to_string (from);
1219 buf2 = __g_socket_address_to_string (arrival->address);
1221 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1222 " saving old as known conflict",
1223 rtp_source_get_ssrc (source), buf1, buf2);
1226 rtp_source_set_rtp_from (source, arrival->address);
1228 rtp_source_set_rtcp_from (source, arrival->address);
1236 /* Don't need to save old addresses, we ignore new sources */
1241 /* We don't already have a from address for RTP, just set it */
1243 rtp_source_set_rtp_from (source, arrival->address);
1245 rtp_source_set_rtcp_from (source, arrival->address);
1249 /* FIXME: Log 3rd party collision somehow
1250 * Maybe should be done in upper layer, only the SDES can tell us
1251 * if its a collision or a loop
1254 /* This is sending with our ssrc, is it an address we already know */
1256 if (rtp_source_find_conflicting_address (source, arrival->address,
1257 arrival->current_time)) {
1258 /* Its a known conflict, its probably a loop, not a collision
1259 * lets just drop the incoming packet
1261 GST_DEBUG ("Our packets are being looped back to us, dropping");
1263 /* Its a new collision, lets change our SSRC */
1265 rtp_source_add_conflicting_address (source, arrival->address,
1266 arrival->current_time);
1268 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1269 on_ssrc_collision (sess, source);
1271 sess->change_ssrc = TRUE;
1273 rtp_source_mark_bye (source, "SSRC Collision");
1274 rtp_session_schedule_bye_locked (sess, arrival->current_time);
1282 find_source (RTPSession * sess, guint32 ssrc)
1284 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1285 GINT_TO_POINTER (ssrc));
1289 add_source (RTPSession * sess, RTPSource * src)
1291 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1292 GINT_TO_POINTER (src->ssrc), src);
1293 /* we have one more source now */
1294 sess->total_sources++;
1295 if (RTP_SOURCE_IS_ACTIVE (src))
1296 sess->stats.active_sources++;
1297 if (src->internal) {
1298 sess->stats.internal_sources++;
1299 if (sess->suggested_ssrc != src->ssrc)
1300 sess->suggested_ssrc = src->ssrc;
1304 /* must be called with the session lock, the returned source needs to be
1305 * unreffed after usage. */
1307 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1308 RTPArrivalStats * arrival, gboolean rtp)
1312 source = find_source (sess, ssrc);
1313 if (source == NULL) {
1314 /* make new Source in probation and insert */
1315 source = rtp_source_new (ssrc);
1317 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1319 /* for RTP packets we need to set the source in probation. Receiving RTCP
1320 * packets of an SSRC, on the other hand, is a strong indication that we
1321 * are dealing with a valid source. */
1323 g_object_set (source, "probation", sess->probation, NULL);
1325 g_object_set (source, "probation", 0, NULL);
1327 /* store from address, if any */
1328 if (arrival->address) {
1330 rtp_source_set_rtp_from (source, arrival->address);
1332 rtp_source_set_rtcp_from (source, arrival->address);
1335 /* configure a callback on the source */
1336 rtp_source_set_callbacks (source, &callbacks, sess);
1338 add_source (sess, source);
1342 /* check for collision, this updates the address when not previously set */
1343 if (check_collision (sess, source, arrival, rtp)) {
1346 /* Receiving RTCP packets of an SSRC is a strong indication that we
1347 * are dealing with a valid source. */
1349 g_object_set (source, "probation", 0, NULL);
1351 /* update last activity */
1352 source->last_activity = arrival->current_time;
1354 source->last_rtp_activity = arrival->current_time;
1355 g_object_ref (source);
1360 /* must be called with the session lock, the returned source needs to be
1361 * unreffed after usage. */
1363 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1367 source = find_source (sess, ssrc);
1368 if (source == NULL) {
1369 /* make new internal Source and insert */
1370 source = rtp_source_new (ssrc);
1372 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1374 source->validated = TRUE;
1375 source->internal = TRUE;
1376 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1377 rtp_source_set_callbacks (source, &callbacks, sess);
1379 add_source (sess, source);
1384 g_object_ref (source);
1390 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1392 if (ssrc != sess->source->ssrc) {
1393 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1394 GINT_TO_POINTER (sess->source->ssrc));
1396 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1397 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1398 * packets will timeout on the old SSRC, we could potentially schedule a
1399 * BYE RTCP for the old SSRC... */
1400 sess->source->ssrc = ssrc;
1401 rtp_source_reset (sess->source);
1403 /* rehash with the new SSRC */
1404 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1405 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1410 * rtp_session_suggest_ssrc:
1411 * @sess: a #RTPSession
1413 * Suggest an unused SSRC in @sess.
1415 * Returns: a free unused SSRC
1418 rtp_session_suggest_ssrc (RTPSession * sess)
1422 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1424 RTP_SESSION_LOCK (sess);
1425 result = sess->suggested_ssrc;
1426 RTP_SESSION_UNLOCK (sess);
1432 * rtp_session_add_source:
1433 * @sess: a #RTPSession
1434 * @src: #RTPSource to add
1436 * Add @src to @session.
1438 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1439 * existed in the session.
1442 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1444 gboolean result = FALSE;
1447 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1448 g_return_val_if_fail (src != NULL, FALSE);
1450 RTP_SESSION_LOCK (sess);
1451 find = find_source (sess, src->ssrc);
1453 add_source (sess, src);
1456 RTP_SESSION_UNLOCK (sess);
1462 * rtp_session_get_num_sources:
1463 * @sess: an #RTPSession
1465 * Get the number of sources in @sess.
1467 * Returns: The number of sources in @sess.
1470 rtp_session_get_num_sources (RTPSession * sess)
1474 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1476 RTP_SESSION_LOCK (sess);
1477 result = sess->total_sources;
1478 RTP_SESSION_UNLOCK (sess);
1484 * rtp_session_get_num_active_sources:
1485 * @sess: an #RTPSession
1487 * Get the number of active sources in @sess. A source is considered active when
1488 * it has been validated and has not yet received a BYE RTCP message.
1490 * Returns: The number of active sources in @sess.
1493 rtp_session_get_num_active_sources (RTPSession * sess)
1497 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1499 RTP_SESSION_LOCK (sess);
1500 result = sess->stats.active_sources;
1501 RTP_SESSION_UNLOCK (sess);
1507 * rtp_session_get_source_by_ssrc:
1508 * @sess: an #RTPSession
1511 * Find the source with @ssrc in @sess.
1513 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1514 * g_object_unref() after usage.
1517 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1521 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1523 RTP_SESSION_LOCK (sess);
1524 result = find_source (sess, ssrc);
1526 g_object_ref (result);
1527 RTP_SESSION_UNLOCK (sess);
1532 /* should be called with the SESSION lock */
1534 rtp_session_create_new_ssrc (RTPSession * sess)
1539 ssrc = g_random_int ();
1541 /* see if it exists in the session, we're done if it doesn't */
1542 if (find_source (sess, ssrc) == NULL)
1550 * rtp_session_create_source:
1551 * @sess: an #RTPSession
1553 * Create an #RTPSource for use in @sess. This function will create a source
1554 * with an ssrc that is currently not used by any participants in the session.
1556 * Returns: an #RTPSource.
1559 rtp_session_create_source (RTPSession * sess)
1564 RTP_SESSION_LOCK (sess);
1565 ssrc = rtp_session_create_new_ssrc (sess);
1566 source = rtp_source_new (ssrc);
1567 rtp_source_set_callbacks (source, &callbacks, sess);
1568 /* we need an additional ref for the source in the hashtable */
1569 g_object_ref (source);
1570 add_source (sess, source);
1571 RTP_SESSION_UNLOCK (sess);
1576 /* update the RTPArrivalStats structure with the current time and other bits
1577 * about the current buffer we are handling.
1578 * This function is typically called when a validated packet is received.
1579 * This function should be called with the SESSION_LOCK
1582 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1583 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1584 GstClockTime running_time, guint64 ntpnstime)
1586 GstNetAddressMeta *meta;
1587 GstRTPBuffer rtpb = { NULL };
1589 /* get time of arrival */
1590 arrival->current_time = current_time;
1591 arrival->running_time = running_time;
1592 arrival->ntpnstime = ntpnstime;
1594 /* get packet size including header overhead */
1595 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1598 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1599 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1600 gst_rtp_buffer_unmap (&rtpb);
1602 arrival->payload_len = 0;
1605 /* for netbuffer we can store the IP address to check for collisions */
1606 meta = gst_buffer_get_net_address_meta (buffer);
1607 if (arrival->address)
1608 g_object_unref (arrival->address);
1610 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1612 arrival->address = NULL;
1617 clean_arrival_stats (RTPArrivalStats * arrival)
1619 if (arrival->address)
1620 g_object_unref (arrival->address);
1624 * rtp_session_process_rtp:
1625 * @sess: and #RTPSession
1626 * @buffer: an RTP buffer
1627 * @current_time: the current system time
1628 * @running_time: the running_time of @buffer
1630 * Process an RTP buffer in the session manager. This function takes ownership
1633 * Returns: a #GstFlowReturn.
1636 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1637 GstClockTime current_time, GstClockTime running_time)
1639 GstFlowReturn result;
1643 gboolean prevsender, prevactive;
1644 RTPArrivalStats arrival = { NULL, };
1648 GstRTPBuffer rtp = { NULL };
1650 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1651 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1653 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1654 goto invalid_packet;
1656 /* get SSRC to look up in session database */
1657 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1658 /* copy available csrc for later */
1659 count = gst_rtp_buffer_get_csrc_count (&rtp);
1660 /* make sure to not overflow our array. An RTP buffer can maximally contain
1662 count = MIN (count, 16);
1664 for (i = 0; i < count; i++)
1665 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1667 gst_rtp_buffer_unmap (&rtp);
1669 RTP_SESSION_LOCK (sess);
1670 /* ignore more RTP packets when we left the session */
1671 if (sess->source->marked_bye)
1674 /* update arrival stats */
1675 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1678 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1682 prevsender = RTP_SOURCE_IS_SENDER (source);
1683 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1684 oldrate = source->bitrate;
1686 /* let source process the packet */
1687 result = rtp_source_process_rtp (source, buffer, &arrival);
1689 /* source became active */
1690 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1691 sess->stats.active_sources++;
1692 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1693 sess->stats.active_sources);
1694 on_ssrc_validated (sess, source);
1696 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1697 sess->stats.sender_sources++;
1698 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1699 sess->stats.sender_sources);
1701 if (oldrate != source->bitrate)
1702 sess->recalc_bandwidth = TRUE;
1705 on_new_ssrc (sess, source);
1707 if (source->validated) {
1710 /* for validated sources, we add the CSRCs as well */
1711 for (i = 0; i < count; i++) {
1713 RTPSource *csrc_src;
1718 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1723 GST_DEBUG ("created new CSRC: %08x", csrc);
1724 rtp_source_set_as_csrc (csrc_src);
1725 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1726 sess->stats.active_sources++;
1727 on_new_ssrc (sess, csrc_src);
1729 g_object_unref (csrc_src);
1732 g_object_unref (source);
1734 RTP_SESSION_UNLOCK (sess);
1736 clean_arrival_stats (&arrival);
1743 gst_buffer_unref (buffer);
1744 GST_DEBUG ("invalid RTP packet received");
1749 RTP_SESSION_UNLOCK (sess);
1750 gst_buffer_unref (buffer);
1751 GST_DEBUG ("ignoring RTP packet because we are leaving");
1756 RTP_SESSION_UNLOCK (sess);
1757 gst_buffer_unref (buffer);
1758 clean_arrival_stats (&arrival);
1759 GST_DEBUG ("ignoring packet because its collisioning");
1765 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1766 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1770 count = gst_rtcp_packet_get_rb_count (packet);
1771 for (i = 0; i < count; i++) {
1772 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1773 guint8 fractionlost;
1776 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1777 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1779 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1781 if (ssrc == sess->source->ssrc) {
1782 /* only deal with report blocks for our session, we update the stats of
1783 * the sender of the RTCP message. We could also compare our stats against
1784 * the other sender to see if we are better or worse. */
1785 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1786 packetslost, exthighestseq, jitter, lsr, dlsr);
1789 on_ssrc_active (sess, source);
1792 /* A Sender report contains statistics about how the sender is doing. This
1793 * includes timing informataion such as the relation between RTP and NTP
1794 * timestamps and the number of packets/bytes it sent to us.
1796 * In this report is also included a set of report blocks related to how this
1797 * sender is receiving data (in case we (or somebody else) is also sending stuff
1798 * to it). This info includes the packet loss, jitter and seqnum. It also
1799 * contains information to calculate the round trip time (LSR/DLSR).
1802 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1803 RTPArrivalStats * arrival, gboolean * do_sync)
1805 guint32 senderssrc, rtptime, packet_count, octet_count;
1808 gboolean created, prevsender;
1810 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1811 &packet_count, &octet_count);
1813 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1814 senderssrc, GST_TIME_ARGS (arrival->current_time));
1816 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1820 /* don't try to do lip-sync for sources that sent a BYE */
1821 if (RTP_SOURCE_IS_MARKED_BYE (source))
1826 prevsender = RTP_SOURCE_IS_SENDER (source);
1828 /* first update the source */
1829 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1830 packet_count, octet_count);
1832 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1833 sess->stats.sender_sources++;
1834 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1835 sess->stats.sender_sources);
1839 on_new_ssrc (sess, source);
1841 rtp_session_process_rb (sess, source, packet, arrival);
1842 g_object_unref (source);
1845 /* A receiver report contains statistics about how a receiver is doing. It
1846 * includes stuff like packet loss, jitter and the seqnum it received last. It
1847 * also contains info to calculate the round trip time.
1849 * We are only interested in how the sender of this report is doing wrt to us.
1852 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1853 RTPArrivalStats * arrival)
1859 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1861 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1863 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1868 on_new_ssrc (sess, source);
1870 rtp_session_process_rb (sess, source, packet, arrival);
1871 g_object_unref (source);
1874 /* Get SDES items and store them in the SSRC */
1876 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1877 RTPArrivalStats * arrival)
1880 gboolean more_items, more_entries;
1882 items = gst_rtcp_packet_sdes_get_item_count (packet);
1883 GST_DEBUG ("got SDES packet with %d items", items);
1885 more_items = gst_rtcp_packet_sdes_first_item (packet);
1887 while (more_items) {
1889 gboolean changed, created, validated;
1893 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1895 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1899 /* find src, no probation when dealing with RTCP */
1900 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1904 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1906 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1908 while (more_entries) {
1909 GstRTCPSDESType type;
1915 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1917 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1920 if (type == GST_RTCP_SDES_PRIV) {
1921 name = g_strndup ((const gchar *) &data[1], data[0]);
1923 data += data[0] + 1;
1925 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1928 value = g_strndup ((const gchar *) data, len);
1930 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1935 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1939 /* takes ownership of sdes */
1940 changed = rtp_source_set_sdes_struct (source, sdes);
1942 validated = !RTP_SOURCE_IS_ACTIVE (source);
1943 source->validated = TRUE;
1946 on_new_ssrc (sess, source);
1948 /* source became active */
1950 sess->stats.active_sources++;
1951 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1952 sess->stats.active_sources);
1953 on_ssrc_validated (sess, source);
1957 on_ssrc_sdes (sess, source);
1959 g_object_unref (source);
1961 more_items = gst_rtcp_packet_sdes_next_item (packet);
1966 /* BYE is sent when a client leaves the session
1969 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1970 RTPArrivalStats * arrival)
1974 gboolean reconsider = FALSE;
1976 reason = gst_rtcp_packet_bye_get_reason (packet);
1977 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1979 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1980 for (i = 0; i < count; i++) {
1983 gboolean created, prevactive, prevsender;
1984 guint pmembers, members;
1986 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1987 GST_DEBUG ("SSRC: %08x", ssrc);
1989 /* find src and mark bye, no probation when dealing with RTCP */
1990 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1994 if (source->internal) {
1995 /* our own source, something weird with this packet */
1996 g_object_unref (source);
2000 /* store time for when we need to time out this source */
2001 source->bye_time = arrival->current_time;
2003 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2004 prevsender = RTP_SOURCE_IS_SENDER (source);
2006 /* mark the source BYE */
2007 rtp_source_mark_bye (source, reason);
2009 pmembers = sess->stats.active_sources;
2011 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2012 sess->stats.active_sources--;
2013 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2014 sess->stats.active_sources);
2016 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2017 sess->stats.sender_sources--;
2018 if (source->internal)
2019 sess->stats.internal_sender_sources--;
2020 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2021 sess->stats.sender_sources);
2023 members = sess->stats.active_sources;
2025 if (!sess->scheduled_bye && members < pmembers) {
2026 /* some members went away since the previous timeout estimate.
2027 * Perform reverse reconsideration but only when we are not scheduling a
2029 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2030 arrival->current_time < sess->next_rtcp_check_time) {
2031 GstClockTime time_remaining;
2033 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2034 sess->next_rtcp_check_time =
2035 gst_util_uint64_scale (time_remaining, members, pmembers);
2037 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2038 GST_TIME_ARGS (sess->next_rtcp_check_time));
2040 sess->next_rtcp_check_time += arrival->current_time;
2042 /* mark pending reconsider. We only want to signal the reconsideration
2043 * once after we handled all the source in the bye packet */
2049 on_new_ssrc (sess, source);
2051 on_bye_ssrc (sess, source);
2053 g_object_unref (source);
2056 RTP_SESSION_UNLOCK (sess);
2057 /* notify app of reconsideration */
2058 if (sess->callbacks.reconsider)
2059 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2060 RTP_SESSION_LOCK (sess);
2066 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2067 RTPArrivalStats * arrival)
2069 GST_DEBUG ("received APP");
2073 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2074 gboolean fir, GstClockTime current_time)
2076 guint32 round_trip = 0;
2078 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2080 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2081 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2084 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2085 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2086 GST_DEBUG ("Ignoring %s request because one was send without one "
2087 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2088 fir ? "FIR" : "PLI",
2089 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2090 GST_TIME_ARGS (round_trip_in_ns));;
2095 sess->last_keyframe_request = current_time;
2097 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2098 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2099 sess->callbacks.request_key_unit);
2101 RTP_SESSION_UNLOCK (sess);
2102 sess->callbacks.request_key_unit (sess, fir,
2103 sess->request_key_unit_user_data);
2104 RTP_SESSION_LOCK (sess);
2110 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2111 guint32 media_ssrc, GstClockTime current_time)
2115 if (!sess->callbacks.request_key_unit)
2118 src = find_source (sess, sender_ssrc);
2122 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2126 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2127 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2132 gboolean our_request = FALSE;
2134 if (!sess->callbacks.request_key_unit)
2140 src = find_source (sess, sender_ssrc);
2142 /* Hack because Google fails to set the sender_ssrc correctly */
2143 if (!src && sender_ssrc == 1) {
2144 GHashTableIter iter;
2146 if (sess->stats.sender_sources >
2147 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2150 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2152 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2153 if (src != sess->source && rtp_source_is_sender (src))
2162 for (position = 0; position < fci_length; position += 8) {
2163 guint8 *data = fci_data + position;
2166 ssrc = GST_READ_UINT32_BE (data);
2168 own = find_source (sess, ssrc);
2169 if (own->internal) {
2177 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2181 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2182 RTPArrivalStats * arrival, GstClockTime current_time)
2184 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2185 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2186 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2187 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2188 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2189 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2192 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2193 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2195 if (g_signal_has_handler_pending (sess,
2196 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2197 GstBuffer *fci_buffer = NULL;
2199 if (fci_length > 0) {
2200 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2201 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2203 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2206 RTP_SESSION_UNLOCK (sess);
2207 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2208 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2209 RTP_SESSION_LOCK (sess);
2212 gst_buffer_unref (fci_buffer);
2215 src = find_source (sess, media_ssrc);
2219 if (sess->rtcp_feedback_retention_window) {
2220 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2223 if (src->internal ||
2224 /* PSFB FIR puts the media ssrc inside the FCI */
2225 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2227 case GST_RTCP_TYPE_PSFB:
2229 case GST_RTCP_PSFB_TYPE_PLI:
2230 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2233 case GST_RTCP_PSFB_TYPE_FIR:
2234 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2241 case GST_RTCP_TYPE_RTPFB:
2249 * rtp_session_process_rtcp:
2250 * @sess: and #RTPSession
2251 * @buffer: an RTCP buffer
2252 * @current_time: the current system time
2253 * @ntpnstime: the current NTP time in nanoseconds
2255 * Process an RTCP buffer in the session manager. This function takes ownership
2258 * Returns: a #GstFlowReturn.
2261 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2262 GstClockTime current_time, guint64 ntpnstime)
2264 GstRTCPPacket packet;
2265 gboolean more, is_bye = FALSE, do_sync = FALSE;
2266 RTPArrivalStats arrival = { NULL, };
2267 GstFlowReturn result = GST_FLOW_OK;
2268 GstRTCPBuffer rtcp = { NULL, };
2270 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2271 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2273 if (!gst_rtcp_buffer_validate (buffer))
2274 goto invalid_packet;
2276 GST_DEBUG ("received RTCP packet");
2278 RTP_SESSION_LOCK (sess);
2279 /* update arrival stats */
2280 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2283 if (sess->source->sent_bye)
2286 /* start processing the compound packet */
2287 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2288 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2292 type = gst_rtcp_packet_get_type (&packet);
2294 /* when we are leaving the session, we should ignore all non-BYE messages */
2295 if (sess->scheduled_bye && type != GST_RTCP_TYPE_BYE) {
2296 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2301 case GST_RTCP_TYPE_SR:
2302 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2304 case GST_RTCP_TYPE_RR:
2305 rtp_session_process_rr (sess, &packet, &arrival);
2307 case GST_RTCP_TYPE_SDES:
2308 rtp_session_process_sdes (sess, &packet, &arrival);
2310 case GST_RTCP_TYPE_BYE:
2312 /* don't try to attempt lip-sync anymore for streams with a BYE */
2314 rtp_session_process_bye (sess, &packet, &arrival);
2316 case GST_RTCP_TYPE_APP:
2317 rtp_session_process_app (sess, &packet, &arrival);
2319 case GST_RTCP_TYPE_RTPFB:
2320 case GST_RTCP_TYPE_PSFB:
2321 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2324 GST_WARNING ("got unknown RTCP packet");
2328 more = gst_rtcp_packet_move_to_next (&packet);
2331 gst_rtcp_buffer_unmap (&rtcp);
2333 /* if we are scheduling a BYE, we only want to count bye packets, else we
2334 * count everything */
2335 if (sess->scheduled_bye) {
2337 sess->stats.bye_members++;
2338 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2341 /* keep track of average packet size */
2342 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2344 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2345 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2346 RTP_SESSION_UNLOCK (sess);
2348 clean_arrival_stats (&arrival);
2350 /* notify caller of sr packets in the callback */
2351 if (do_sync && sess->callbacks.sync_rtcp) {
2352 /* make writable, we might want to change the buffer */
2353 buffer = gst_buffer_make_writable (buffer);
2355 result = sess->callbacks.sync_rtcp (sess, buffer,
2356 sess->sync_rtcp_user_data);
2358 gst_buffer_unref (buffer);
2365 GST_DEBUG ("invalid RTCP packet received");
2366 gst_buffer_unref (buffer);
2371 RTP_SESSION_UNLOCK (sess);
2372 gst_buffer_unref (buffer);
2373 clean_arrival_stats (&arrival);
2374 GST_DEBUG ("ignoring RTCP packet because we left");
2380 * rtp_session_update_send_caps:
2381 * @sess: an #RTPSession
2384 * Update the caps of the sender in the rtp session.
2387 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2392 g_return_if_fail (RTP_IS_SESSION (sess));
2393 g_return_if_fail (GST_IS_CAPS (caps));
2395 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2397 s = gst_caps_get_structure (caps, 0);
2399 if (gst_structure_get_uint (s, "ssrc", &ssrc))
2400 rtp_session_set_internal_ssrc (sess, ssrc);
2402 RTP_SESSION_LOCK (sess);
2403 rtp_source_update_caps (sess->source, caps);
2404 RTP_SESSION_UNLOCK (sess);
2408 * rtp_session_send_rtp:
2409 * @sess: an #RTPSession
2410 * @data: pointer to either an RTP buffer or a list of RTP buffers
2411 * @is_list: TRUE when @data is a buffer list
2412 * @current_time: the current system time
2413 * @running_time: the running time of @data
2415 * Send the RTP buffer in the session manager. This function takes ownership of
2418 * Returns: a #GstFlowReturn.
2421 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2422 GstClockTime current_time, GstClockTime running_time)
2424 GstFlowReturn result;
2426 gboolean prevsender;
2429 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2430 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2432 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2434 RTP_SESSION_LOCK (sess);
2435 source = sess->source;
2437 /* update last activity */
2438 source->last_rtp_activity = current_time;
2440 prevsender = RTP_SOURCE_IS_SENDER (source);
2441 oldrate = source->bitrate;
2443 /* we use our own source to send */
2444 result = rtp_source_send_rtp (source, data, is_list, running_time);
2446 if (RTP_SOURCE_IS_SENDER (source) && !prevsender) {
2447 sess->stats.sender_sources++;
2448 sess->stats.internal_sender_sources++;
2450 if (oldrate != source->bitrate)
2451 sess->recalc_bandwidth = TRUE;
2452 RTP_SESSION_UNLOCK (sess);
2458 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2460 *bandwidth += source->bitrate;
2463 /* must be called with session lock */
2465 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2468 GstClockTime result;
2470 /* recalculate bandwidth when it changed */
2471 if (sess->recalc_bandwidth) {
2474 if (sess->bandwidth > 0)
2475 bandwidth = sess->bandwidth;
2477 /* If it is <= 0, then try to estimate the actual bandwidth */
2480 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2481 (GHFunc) add_bitrates, &bandwidth);
2484 if (bandwidth < 8000)
2485 bandwidth = RTP_STATS_BANDWIDTH;
2487 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2488 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2490 sess->recalc_bandwidth = FALSE;
2493 if (sess->scheduled_bye) {
2494 result = rtp_stats_calculate_bye_interval (&sess->stats);
2496 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2497 sess->stats.internal_sender_sources > 0, first);
2500 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2501 GST_TIME_ARGS (result), first);
2503 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2504 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2506 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2512 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2514 if (source->internal)
2515 rtp_source_mark_bye (source, reason);
2519 * rtp_session_mark_all_bye:
2520 * @sess: an #RTPSession
2523 * Mark all internal sources of the session as BYE with @reason.
2526 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2528 g_return_if_fail (RTP_IS_SESSION (sess));
2530 RTP_SESSION_LOCK (sess);
2531 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2532 (GHFunc) source_mark_bye, (gpointer) reason);
2533 RTP_SESSION_UNLOCK (sess);
2536 /* Stop the current @sess and schedule a BYE message for the other members.
2537 * One must have the session lock to call this function
2539 static GstFlowReturn
2540 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2542 GstFlowReturn result = GST_FLOW_OK;
2543 GstClockTime interval;
2545 /* nothing to do it we already scheduled bye */
2546 if (sess->scheduled_bye)
2549 /* we schedule BYE now */
2550 sess->scheduled_bye = TRUE;
2551 /* at least one member wants to send a BYE */
2552 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2553 sess->stats.bye_members = 1;
2554 sess->first_rtcp = TRUE;
2555 sess->allow_early = TRUE;
2557 /* reschedule transmission */
2558 sess->last_rtcp_send_time = current_time;
2559 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2561 if (interval != GST_CLOCK_TIME_NONE)
2562 sess->next_rtcp_check_time = current_time + interval;
2564 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2566 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2567 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2569 RTP_SESSION_UNLOCK (sess);
2570 /* notify app of reconsideration */
2571 if (sess->callbacks.reconsider)
2572 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2573 RTP_SESSION_LOCK (sess);
2580 * rtp_session_schedule_bye:
2581 * @sess: an #RTPSession
2582 * @current_time: the current system time
2584 * Schedule a BYE message for all sources marked as BYE in @sess.
2586 * Returns: a #GstFlowReturn.
2589 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2591 GstFlowReturn result = GST_FLOW_OK;
2593 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2595 RTP_SESSION_LOCK (sess);
2596 result = rtp_session_schedule_bye_locked (sess, current_time);
2597 RTP_SESSION_UNLOCK (sess);
2603 * rtp_session_next_timeout:
2604 * @sess: an #RTPSession
2605 * @current_time: the current system time
2607 * Get the next time we should perform session maintenance tasks.
2609 * Returns: a time when rtp_session_on_timeout() should be called with the
2610 * current system time.
2613 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2615 GstClockTime result, interval = 0;
2617 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2619 RTP_SESSION_LOCK (sess);
2621 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2622 result = sess->next_early_rtcp_time;
2626 result = sess->next_rtcp_check_time;
2628 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2629 ", next time: %" GST_TIME_FORMAT,
2630 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2632 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2633 GST_DEBUG ("take current time as base");
2634 /* our previous check time expired, start counting from the current time
2636 result = current_time;
2639 if (sess->scheduled_bye) {
2640 if (sess->source->sent_bye) {
2641 GST_DEBUG ("we sent BYE already");
2642 interval = GST_CLOCK_TIME_NONE;
2643 } else if (sess->stats.active_sources >= 50) {
2644 GST_DEBUG ("reconsider BYE, more than 50 sources");
2645 /* reconsider BYE if members >= 50 */
2646 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2649 if (sess->first_rtcp) {
2650 GST_DEBUG ("first RTCP packet");
2651 /* we are called for the first time */
2652 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2653 } else if (sess->next_rtcp_check_time < current_time) {
2654 GST_DEBUG ("old check time expired, getting new timeout");
2655 /* get a new timeout when we need to */
2656 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2660 if (interval != GST_CLOCK_TIME_NONE)
2663 result = GST_CLOCK_TIME_NONE;
2665 sess->next_rtcp_check_time = result;
2669 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2670 ", next time: %" GST_TIME_FORMAT,
2671 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2672 RTP_SESSION_UNLOCK (sess);
2686 GstRTCPBuffer rtcpbuf;
2690 GstClockTime current_time;
2692 GstClockTime running_time;
2693 GstClockTime interval;
2694 GstRTCPPacket packet;
2697 gboolean may_suppress;
2703 session_start_rtcp (RTPSession * sess, ReportData * data)
2705 GstRTCPPacket *packet = &data->packet;
2706 RTPSource *own = data->source;
2707 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2709 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2710 data->has_sdes = FALSE;
2712 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2714 if (RTP_SOURCE_IS_SENDER (own)) {
2717 guint32 packet_count, octet_count;
2719 /* we are a sender, create SR */
2720 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2721 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2723 /* get latest stats */
2724 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2725 &ntptime, &rtptime, &packet_count, &octet_count);
2727 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2728 packet_count, octet_count);
2730 /* fill in sender report info */
2731 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2732 ntptime, rtptime, packet_count, octet_count);
2734 /* we are only receiver, create RR */
2735 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2736 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2737 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2741 /* construct a Sender or Receiver Report */
2743 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2745 GstRTCPPacket *packet = &data->packet;
2747 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2748 /* only report about other sender sources */
2749 if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
2750 guint8 fractionlost;
2752 guint32 exthighestseq, jitter;
2756 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2757 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2759 /* store last generated RR packet */
2760 source->last_rr.is_valid = TRUE;
2761 source->last_rr.fractionlost = fractionlost;
2762 source->last_rr.packetslost = packetslost;
2763 source->last_rr.exthighestseq = exthighestseq;
2764 source->last_rr.jitter = jitter;
2765 source->last_rr.lsr = lsr;
2766 source->last_rr.dlsr = dlsr;
2768 /* packet is not yet filled, add report block for this source. */
2769 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2770 exthighestseq, jitter, lsr, dlsr);
2775 /* perform cleanup of sources that timed out */
2777 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2779 gboolean remove = FALSE;
2780 gboolean byetimeout = FALSE;
2781 gboolean sendertimeout = FALSE;
2782 gboolean is_sender, is_active;
2783 RTPSession *sess = data->sess;
2784 GstClockTime interval, binterval;
2787 /* check for outdated collisions */
2788 if (source->internal) {
2789 GST_DEBUG ("Timing out collisions");
2790 rtp_source_timeout (source, data->current_time,
2791 /* "a relatively long time" -- RFC 3550 section 8.2 */
2792 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
2793 data->running_time - sess->rtcp_feedback_retention_window);
2796 /* nothing else to do when without RTCP */
2797 if (data->interval == GST_CLOCK_TIME_NONE)
2800 is_sender = RTP_SOURCE_IS_SENDER (source);
2801 is_active = RTP_SOURCE_IS_ACTIVE (source);
2803 /* our own rtcp interval may have been forced low by secondary configuration,
2804 * while sender side may still operate with higher interval,
2805 * so do not just take our interval to decide on timing out sender,
2806 * but take (if data->interval <= 5 * GST_SECOND):
2807 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2808 * where sender_interval is difference between last 2 received RTCP reports
2810 if (data->interval >= 5 * GST_SECOND || source->internal) {
2811 binterval = data->interval;
2813 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2814 GST_TIME_ARGS (source->stats.prev_rtcptime),
2815 GST_TIME_ARGS (source->stats.last_rtcptime));
2816 /* if not received enough yet, fallback to larger default */
2817 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2818 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2820 binterval = 5 * GST_SECOND;
2821 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2823 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2824 GST_TIME_ARGS (binterval));
2826 /* check for our own source, we don't want to delete our own source. */
2827 if (!source->internal) {
2828 if (source->marked_bye) {
2829 /* if we received a BYE from the source, remove the source after some
2831 if (data->current_time > source->bye_time &&
2832 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2833 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2838 /* sources that were inactive for more than 5 times the deterministic reporting
2839 * interval get timed out. the min timeout is 5 seconds. */
2840 /* mind old time that might pre-date last time going to PLAYING */
2841 btime = MAX (source->last_activity, sess->start_time);
2842 if (data->current_time > btime) {
2843 interval = MAX (binterval * 5, 5 * GST_SECOND);
2844 if (data->current_time - btime > interval) {
2845 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2846 source->ssrc, GST_TIME_ARGS (btime));
2852 /* senders that did not send for a long time become a receiver, this also
2853 * holds for our own sources. */
2855 /* mind old time that might pre-date last time going to PLAYING */
2856 btime = MAX (source->last_rtp_activity, sess->start_time);
2857 if (data->current_time > btime) {
2858 interval = MAX (binterval * 2, 5 * GST_SECOND);
2859 if (data->current_time - btime > interval) {
2860 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2861 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2862 source->is_sender = FALSE;
2863 sess->stats.sender_sources--;
2864 if (source->internal)
2865 sess->stats.internal_sender_sources--;
2866 sendertimeout = TRUE;
2872 sess->total_sources--;
2874 sess->stats.sender_sources--;
2875 if (source->internal)
2876 sess->stats.internal_sender_sources--;
2879 sess->stats.active_sources--;
2881 if (source->internal)
2882 sess->stats.internal_sources--;
2885 on_bye_timeout (sess, source);
2887 on_timeout (sess, source);
2890 on_sender_timeout (sess, source);
2893 source->closing = remove;
2897 session_sdes (RTPSession * sess, ReportData * data)
2899 GstRTCPPacket *packet = &data->packet;
2900 const GstStructure *sdes;
2902 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2904 /* add SDES packet */
2905 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2907 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
2909 sdes = rtp_source_get_sdes_struct (data->source);
2911 /* add all fields in the structure, the order is not important. */
2912 n_fields = gst_structure_n_fields (sdes);
2913 for (i = 0; i < n_fields; ++i) {
2916 GstRTCPSDESType type;
2918 field = gst_structure_nth_field_name (sdes, i);
2921 value = gst_structure_get_string (sdes, field);
2924 type = gst_rtcp_sdes_name_to_type (field);
2926 /* Early packets are minimal and only include the CNAME */
2927 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2930 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2931 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2932 (const guint8 *) value);
2933 } else if (type == GST_RTCP_SDES_PRIV) {
2939 /* don't accept entries that are too big */
2940 prefix_len = strlen (field);
2941 if (prefix_len > 255)
2943 value_len = strlen (value);
2944 if (value_len > 255)
2946 data_len = 1 + prefix_len + value_len;
2950 data[0] = prefix_len;
2951 memcpy (&data[1], field, prefix_len);
2952 memcpy (&data[1 + prefix_len], value, value_len);
2954 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2958 data->has_sdes = TRUE;
2961 /* schedule a BYE packet */
2963 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
2965 GstRTCPPacket *packet = &data->packet;
2966 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2969 session_sdes (sess, data);
2970 /* add a BYE packet */
2971 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
2972 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
2973 if (source->bye_reason)
2974 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
2976 /* we have a BYE packet now */
2977 source->sent_bye = TRUE;
2981 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2983 GstClockTime new_send_time, elapsed;
2985 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
2986 data->is_early = TRUE;
2988 data->is_early = FALSE;
2990 if (data->is_early && sess->next_early_rtcp_time < current_time)
2993 /* no need to check yet */
2994 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
2995 sess->next_rtcp_check_time > current_time) {
2996 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2997 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2998 GST_TIME_ARGS (current_time));
3002 /* get elapsed time since we last reported */
3003 elapsed = current_time - sess->last_rtcp_send_time;
3005 new_send_time = data->interval;
3006 /* perform forward reconsideration */
3007 if (new_send_time != GST_CLOCK_TIME_NONE) {
3008 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
3010 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3011 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time),
3012 GST_TIME_ARGS (elapsed));
3014 new_send_time += sess->last_rtcp_send_time;
3017 /* check if reconsideration */
3018 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3019 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3020 GST_TIME_ARGS (new_send_time));
3021 /* store new check time */
3022 sess->next_rtcp_check_time = new_send_time;
3028 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
3030 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3031 GST_TIME_ARGS (new_send_time));
3033 sess->next_rtcp_check_time = new_send_time;
3034 if (new_send_time != GST_CLOCK_TIME_NONE) {
3035 sess->next_rtcp_check_time += current_time;
3037 /* Apply the rules from RFC 4585 section 3.5.3 */
3038 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
3039 GstClockTimeDiff T_rr_current_interval =
3040 g_random_double_range (0.5, 1.5) * sess->stats.min_interval;
3042 /* This will caused the RTCP to be suppressed if no FB packets are added */
3043 if (sess->last_rtcp_send_time + T_rr_current_interval >
3044 sess->next_rtcp_check_time) {
3045 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3046 " last: %" GST_TIME_FORMAT
3047 " + T_rr_current_interval: %" GST_TIME_FORMAT
3048 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3049 GST_TIME_ARGS (sess->stats.min_interval),
3050 GST_TIME_ARGS (sess->last_rtcp_send_time),
3051 GST_TIME_ARGS (T_rr_current_interval),
3052 GST_TIME_ARGS (sess->next_rtcp_check_time));
3053 data->may_suppress = TRUE;
3062 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3064 g_hash_table_insert (hash_table, key, g_object_ref (source));
3068 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3070 return source->closing;
3074 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3076 RTPSession *sess = data->sess;
3077 gboolean is_bye = FALSE;
3078 ReportOutput *output;
3080 /* only generate RTCP for active internal sources */
3081 if (!source->internal || source->sent_bye)
3084 data->source = source;
3087 session_start_rtcp (sess, data);
3089 if (source->marked_bye) {
3091 make_source_bye (sess, source, data);
3093 } else if (!data->is_early) {
3094 /* loop over all known sources and add report blocks. If we are ealy, we
3095 * just make a minimal RTCP packet and skip this step */
3096 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3097 (GHFunc) session_report_blocks, data);
3099 if (!data->has_sdes)
3100 session_sdes (sess, data);
3102 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3104 if (sess->change_ssrc) {
3105 GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
3106 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3107 GINT_TO_POINTER (source->ssrc));
3109 source->ssrc = rtp_session_create_new_ssrc (sess);
3110 rtp_source_reset (source);
3112 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3113 GINT_TO_POINTER (source->ssrc), source);
3115 sess->change_ssrc = FALSE;
3116 data->notify = TRUE;
3117 GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
3120 output = g_slice_new (ReportOutput);
3121 output->source = g_object_ref (source);
3122 output->is_bye = is_bye;
3123 output->buffer = data->rtcp;
3124 /* queue the RTCP packet to push later */
3125 g_queue_push_tail (&data->output, output);
3129 * rtp_session_on_timeout:
3130 * @sess: an #RTPSession
3131 * @current_time: the current system time
3132 * @ntpnstime: the current NTP time in nanoseconds
3133 * @running_time: the current running_time of the pipeline
3135 * Perform maintenance actions after the timeout obtained with
3136 * rtp_session_next_timeout() expired.
3138 * This function will perform timeouts of receivers and senders, send a BYE
3139 * packet or generate RTCP packets with current session stats.
3141 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3142 * times, for each packet that should be processed.
3144 * Returns: a #GstFlowReturn.
3147 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3148 guint64 ntpnstime, GstClockTime running_time)
3150 GstFlowReturn result = GST_FLOW_OK;
3151 ReportData data = { GST_RTCP_BUFFER_INIT };
3152 GHashTable *table_copy;
3153 ReportOutput *output;
3155 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3157 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3158 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3159 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3162 data.current_time = current_time;
3163 data.ntpnstime = ntpnstime;
3164 data.running_time = running_time;
3165 data.may_suppress = FALSE;
3166 data.notify = FALSE;
3167 g_queue_init (&data.output);
3169 RTP_SESSION_LOCK (sess);
3170 /* get a new interval, we need this for various cleanups etc */
3171 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3173 /* Make a local copy of the hashtable. We need to do this because the
3174 * cleanup stage below releases the session lock. */
3175 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3176 (GDestroyNotify) g_object_unref);
3177 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3178 (GHFunc) clone_ssrcs_hashtable, table_copy);
3180 /* Clean up the session, mark the source for removing, this might release the
3182 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3183 g_hash_table_destroy (table_copy);
3185 /* Now remove the marked sources */
3186 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3187 (GHRFunc) remove_closing_sources, NULL);
3189 /* see if we need to generate SR or RR packets */
3190 if (!is_rtcp_time (sess, current_time, &data))
3193 /* generate RTCP for all internal sources */
3194 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3195 (GHFunc) generate_rtcp, &data);
3197 /* we keep track of the last report time in order to timeout inactive
3198 * receivers or senders */
3199 if (!data.is_early && !data.may_suppress)
3200 sess->last_rtcp_send_time = data.current_time;
3201 sess->first_rtcp = FALSE;
3202 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3205 RTP_SESSION_UNLOCK (sess);
3208 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3210 /* push out the RTCP packets */
3211 while ((output = g_queue_pop_head (&data.output))) {
3212 gboolean do_not_suppress;
3213 GstBuffer *buffer = output->buffer;
3214 RTPSource *source = output->source;
3216 /* Give the user a change to add its own packet */
3217 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3218 buffer, data.is_early, &do_not_suppress);
3220 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3223 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3225 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3226 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3227 sess->stats.avg_rtcp_packet_size, packet_size);
3229 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3230 sess->send_rtcp_user_data);
3232 GST_DEBUG ("freeing packet callback: %p"
3233 " do_not_suppress: %d may_suppress: %d",
3234 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3235 gst_buffer_unref (buffer);
3237 g_object_unref (source);
3238 g_slice_free (ReportOutput, output);
3245 * rtp_session_request_early_rtcp:
3246 * @sess: an #RTPSession
3247 * @current_time: the current system time
3248 * @max_delay: maximum delay
3250 * Request transmission of early RTCP
3253 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3254 GstClockTimeDiff max_delay)
3256 GstClockTime T_dither_max;
3258 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3260 RTP_SESSION_LOCK (sess);
3262 /* Check if already requested */
3263 /* RFC 4585 section 3.5.2 step 2 */
3264 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3267 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time))
3270 /* Ignore the request a scheduled packet will be in time anyway */
3271 if (current_time + max_delay > sess->next_rtcp_check_time)
3274 /* RFC 4585 section 3.5.2 step 2b */
3275 /* If the total sources is <=2, then there is only us and one peer */
3276 if (sess->total_sources <= 2) {
3279 /* Divide by 2 because l = 0.5 */
3280 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3284 /* RFC 4585 section 3.5.2 step 3 */
3285 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3288 /* RFC 4585 section 3.5.2 step 4
3289 * Don't send if allow_early is FALSE, but not if we are in
3290 * immediate mode, meaning we are part of a group of at most the
3291 * application-specific threshold.
3293 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3294 sess->allow_early == FALSE)
3298 /* Schedule an early transmission later */
3299 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3302 /* If no dithering, schedule it for NOW */
3303 sess->next_early_rtcp_time = current_time;
3306 RTP_SESSION_UNLOCK (sess);
3308 /* notify app of need to send packet early
3309 * and therefore of timeout change */
3310 if (sess->callbacks.reconsider)
3311 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3317 RTP_SESSION_UNLOCK (sess);
3321 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3322 gboolean fir, gint count)
3324 RTPSource *src = find_source (sess, ssrc);
3330 src->send_pli = FALSE;
3331 src->send_fir = TRUE;
3333 if (count == -1 || count != src->last_fir_count)
3334 src->current_send_fir_seqnum++;
3335 src->last_fir_count = count;
3336 } else if (!src->send_fir) {
3337 src->send_pli = TRUE;
3340 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3346 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3348 GstRTCPPacket packet;
3349 GstRTCPBuffer rtcp = { NULL, };
3350 gboolean ret = FALSE;
3352 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3354 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3355 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3356 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3360 gst_rtcp_buffer_unmap (&rtcp);
3366 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3369 gboolean ret = FALSE;
3370 GHashTableIter iter;
3371 gpointer key, value;
3372 gboolean started_fir = FALSE;
3373 GstRTCPPacket fir_rtcppacket;
3374 GstRTCPPacket packet;
3375 GstRTCPBuffer rtcp = { NULL, };
3378 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3380 gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
3381 switch (gst_rtcp_packet_get_type (&packet)) {
3382 case GST_RTCP_TYPE_SR:
3383 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc,
3384 NULL, NULL, NULL, NULL);
3386 case GST_RTCP_TYPE_RR:
3387 ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
3393 RTP_SESSION_LOCK (sess);
3394 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3395 while (g_hash_table_iter_next (&iter, &key, &value)) {
3396 guint media_ssrc = GPOINTER_TO_UINT (key);
3397 RTPSource *media_src = value;
3400 if (media_src->send_fir) {
3402 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3405 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3406 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket, ssrc);
3407 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3409 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3410 gst_rtcp_packet_remove (&fir_rtcppacket);
3416 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3417 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3421 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3422 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3424 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3426 fci_data[0] = media_src->current_send_fir_seqnum;
3427 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3428 media_src->send_fir = FALSE;
3432 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3433 while (g_hash_table_iter_next (&iter, &key, &value)) {
3434 guint media_ssrc = GPOINTER_TO_UINT (key);
3435 RTPSource *media_src = value;
3436 GstRTCPPacket pli_rtcppacket;
3438 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3439 has_pli_compare_func, NULL)) {
3440 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3442 /* Break because the packet is full, will put next request in a
3445 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3446 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket, ssrc);
3447 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3450 media_src->send_pli = FALSE;
3452 RTP_SESSION_UNLOCK (sess);
3455 gst_rtcp_buffer_unmap (&rtcp);
3461 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3465 if (!sess->callbacks.send_rtcp)
3468 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3470 rtp_session_request_early_rtcp (sess, now, max_delay);