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",
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",
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:
589 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
592 RTP_SESSION_LOCK (sess);
593 sess->bandwidth = g_value_get_double (value);
594 sess->recalc_bandwidth = TRUE;
595 RTP_SESSION_UNLOCK (sess);
597 case PROP_RTCP_FRACTION:
598 RTP_SESSION_LOCK (sess);
599 sess->rtcp_bandwidth = g_value_get_double (value);
600 sess->recalc_bandwidth = TRUE;
601 RTP_SESSION_UNLOCK (sess);
603 case PROP_RTCP_RR_BANDWIDTH:
604 RTP_SESSION_LOCK (sess);
605 sess->rtcp_rr_bandwidth = g_value_get_int (value);
606 sess->recalc_bandwidth = TRUE;
607 RTP_SESSION_UNLOCK (sess);
609 case PROP_RTCP_RS_BANDWIDTH:
610 RTP_SESSION_LOCK (sess);
611 sess->rtcp_rs_bandwidth = g_value_get_int (value);
612 sess->recalc_bandwidth = TRUE;
613 RTP_SESSION_UNLOCK (sess);
616 sess->mtu = g_value_get_uint (value);
619 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
622 sess->favor_new = g_value_get_boolean (value);
624 case PROP_RTCP_MIN_INTERVAL:
625 rtp_stats_set_min_interval (&sess->stats,
626 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
627 /* trigger reconsideration */
628 RTP_SESSION_LOCK (sess);
629 sess->next_rtcp_check_time = 0;
630 RTP_SESSION_UNLOCK (sess);
631 if (sess->callbacks.reconsider)
632 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
634 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
635 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
638 sess->probation = g_value_get_uint (value);
639 g_object_set_property (G_OBJECT (sess->source), "probation", value);
642 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
648 rtp_session_get_property (GObject * object, guint prop_id,
649 GValue * value, GParamSpec * pspec)
653 sess = RTP_SESSION (object);
656 case PROP_INTERNAL_SSRC:
657 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
659 case PROP_INTERNAL_SOURCE:
660 g_value_take_object (value, rtp_session_get_internal_source (sess));
663 g_value_set_double (value, sess->bandwidth);
665 case PROP_RTCP_FRACTION:
666 g_value_set_double (value, sess->rtcp_bandwidth);
668 case PROP_RTCP_RR_BANDWIDTH:
669 g_value_set_int (value, sess->rtcp_rr_bandwidth);
671 case PROP_RTCP_RS_BANDWIDTH:
672 g_value_set_int (value, sess->rtcp_rs_bandwidth);
675 g_value_set_uint (value, sess->mtu);
678 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
680 case PROP_NUM_SOURCES:
681 g_value_set_uint (value, rtp_session_get_num_sources (sess));
683 case PROP_NUM_ACTIVE_SOURCES:
684 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
687 g_value_take_boxed (value, rtp_session_create_sources (sess));
690 g_value_set_boolean (value, sess->favor_new);
692 case PROP_RTCP_MIN_INTERVAL:
693 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
695 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
696 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
699 g_value_set_uint (value, sess->probation);
700 g_object_get_property (G_OBJECT (sess->source), "probation", value);
703 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
709 on_new_ssrc (RTPSession * sess, RTPSource * source)
711 g_object_ref (source);
712 RTP_SESSION_UNLOCK (sess);
713 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
714 RTP_SESSION_LOCK (sess);
715 g_object_unref (source);
719 on_ssrc_collision (RTPSession * sess, RTPSource * source)
721 g_object_ref (source);
722 RTP_SESSION_UNLOCK (sess);
723 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
725 RTP_SESSION_LOCK (sess);
726 g_object_unref (source);
730 on_ssrc_validated (RTPSession * sess, RTPSource * source)
732 g_object_ref (source);
733 RTP_SESSION_UNLOCK (sess);
734 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
736 RTP_SESSION_LOCK (sess);
737 g_object_unref (source);
741 on_ssrc_active (RTPSession * sess, RTPSource * source)
743 g_object_ref (source);
744 RTP_SESSION_UNLOCK (sess);
745 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
746 RTP_SESSION_LOCK (sess);
747 g_object_unref (source);
751 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
753 g_object_ref (source);
754 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
755 RTP_SESSION_UNLOCK (sess);
756 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
757 RTP_SESSION_LOCK (sess);
758 g_object_unref (source);
762 on_bye_ssrc (RTPSession * sess, RTPSource * source)
764 g_object_ref (source);
765 RTP_SESSION_UNLOCK (sess);
766 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
767 RTP_SESSION_LOCK (sess);
768 g_object_unref (source);
772 on_bye_timeout (RTPSession * sess, RTPSource * source)
774 g_object_ref (source);
775 RTP_SESSION_UNLOCK (sess);
776 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
777 RTP_SESSION_LOCK (sess);
778 g_object_unref (source);
782 on_timeout (RTPSession * sess, RTPSource * source)
784 g_object_ref (source);
785 RTP_SESSION_UNLOCK (sess);
786 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
787 RTP_SESSION_LOCK (sess);
788 g_object_unref (source);
792 on_sender_timeout (RTPSession * sess, RTPSource * source)
794 g_object_ref (source);
795 RTP_SESSION_UNLOCK (sess);
796 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
798 RTP_SESSION_LOCK (sess);
799 g_object_unref (source);
805 * Create a new session object.
807 * Returns: a new #RTPSession. g_object_unref() after usage.
810 rtp_session_new (void)
814 sess = g_object_new (RTP_TYPE_SESSION, NULL);
820 * rtp_session_set_callbacks:
821 * @sess: an #RTPSession
822 * @callbacks: callbacks to configure
823 * @user_data: user data passed in the callbacks
825 * Configure a set of callbacks to be notified of actions.
828 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
831 g_return_if_fail (RTP_IS_SESSION (sess));
833 if (callbacks->process_rtp) {
834 sess->callbacks.process_rtp = callbacks->process_rtp;
835 sess->process_rtp_user_data = user_data;
837 if (callbacks->send_rtp) {
838 sess->callbacks.send_rtp = callbacks->send_rtp;
839 sess->send_rtp_user_data = user_data;
841 if (callbacks->send_rtcp) {
842 sess->callbacks.send_rtcp = callbacks->send_rtcp;
843 sess->send_rtcp_user_data = user_data;
845 if (callbacks->sync_rtcp) {
846 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
847 sess->sync_rtcp_user_data = user_data;
849 if (callbacks->clock_rate) {
850 sess->callbacks.clock_rate = callbacks->clock_rate;
851 sess->clock_rate_user_data = user_data;
853 if (callbacks->reconsider) {
854 sess->callbacks.reconsider = callbacks->reconsider;
855 sess->reconsider_user_data = user_data;
857 if (callbacks->request_key_unit) {
858 sess->callbacks.request_key_unit = callbacks->request_key_unit;
859 sess->request_key_unit_user_data = user_data;
861 if (callbacks->request_time) {
862 sess->callbacks.request_time = callbacks->request_time;
863 sess->request_time_user_data = user_data;
868 * rtp_session_set_process_rtp_callback:
869 * @sess: an #RTPSession
870 * @callback: callback to set
871 * @user_data: user data passed in the callback
873 * Configure only the process_rtp callback to be notified of the process_rtp action.
876 rtp_session_set_process_rtp_callback (RTPSession * sess,
877 RTPSessionProcessRTP callback, gpointer user_data)
879 g_return_if_fail (RTP_IS_SESSION (sess));
881 sess->callbacks.process_rtp = callback;
882 sess->process_rtp_user_data = user_data;
886 * rtp_session_set_send_rtp_callback:
887 * @sess: an #RTPSession
888 * @callback: callback to set
889 * @user_data: user data passed in the callback
891 * Configure only the send_rtp callback to be notified of the send_rtp action.
894 rtp_session_set_send_rtp_callback (RTPSession * sess,
895 RTPSessionSendRTP callback, gpointer user_data)
897 g_return_if_fail (RTP_IS_SESSION (sess));
899 sess->callbacks.send_rtp = callback;
900 sess->send_rtp_user_data = user_data;
904 * rtp_session_set_send_rtcp_callback:
905 * @sess: an #RTPSession
906 * @callback: callback to set
907 * @user_data: user data passed in the callback
909 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
912 rtp_session_set_send_rtcp_callback (RTPSession * sess,
913 RTPSessionSendRTCP callback, gpointer user_data)
915 g_return_if_fail (RTP_IS_SESSION (sess));
917 sess->callbacks.send_rtcp = callback;
918 sess->send_rtcp_user_data = user_data;
922 * rtp_session_set_sync_rtcp_callback:
923 * @sess: an #RTPSession
924 * @callback: callback to set
925 * @user_data: user data passed in the callback
927 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
930 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
931 RTPSessionSyncRTCP callback, gpointer user_data)
933 g_return_if_fail (RTP_IS_SESSION (sess));
935 sess->callbacks.sync_rtcp = callback;
936 sess->sync_rtcp_user_data = user_data;
940 * rtp_session_set_clock_rate_callback:
941 * @sess: an #RTPSession
942 * @callback: callback to set
943 * @user_data: user data passed in the callback
945 * Configure only the clock_rate callback to be notified of the clock_rate action.
948 rtp_session_set_clock_rate_callback (RTPSession * sess,
949 RTPSessionClockRate callback, gpointer user_data)
951 g_return_if_fail (RTP_IS_SESSION (sess));
953 sess->callbacks.clock_rate = callback;
954 sess->clock_rate_user_data = user_data;
958 * rtp_session_set_reconsider_callback:
959 * @sess: an #RTPSession
960 * @callback: callback to set
961 * @user_data: user data passed in the callback
963 * Configure only the reconsider callback to be notified of the reconsider action.
966 rtp_session_set_reconsider_callback (RTPSession * sess,
967 RTPSessionReconsider callback, gpointer user_data)
969 g_return_if_fail (RTP_IS_SESSION (sess));
971 sess->callbacks.reconsider = callback;
972 sess->reconsider_user_data = user_data;
976 * rtp_session_set_request_time_callback:
977 * @sess: an #RTPSession
978 * @callback: callback to set
979 * @user_data: user data passed in the callback
981 * Configure only the request_time callback
984 rtp_session_set_request_time_callback (RTPSession * sess,
985 RTPSessionRequestTime callback, gpointer user_data)
987 g_return_if_fail (RTP_IS_SESSION (sess));
989 sess->callbacks.request_time = callback;
990 sess->request_time_user_data = user_data;
994 * rtp_session_set_bandwidth:
995 * @sess: an #RTPSession
996 * @bandwidth: the bandwidth allocated
998 * Set the session bandwidth in bytes per second.
1001 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1003 g_return_if_fail (RTP_IS_SESSION (sess));
1005 RTP_SESSION_LOCK (sess);
1006 sess->stats.bandwidth = bandwidth;
1007 RTP_SESSION_UNLOCK (sess);
1011 * rtp_session_get_bandwidth:
1012 * @sess: an #RTPSession
1014 * Get the session bandwidth.
1016 * Returns: the session bandwidth.
1019 rtp_session_get_bandwidth (RTPSession * sess)
1023 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1025 RTP_SESSION_LOCK (sess);
1026 result = sess->stats.bandwidth;
1027 RTP_SESSION_UNLOCK (sess);
1033 * rtp_session_set_rtcp_fraction:
1034 * @sess: an #RTPSession
1035 * @bandwidth: the RTCP bandwidth
1037 * Set the bandwidth in bytes per second that should be used for RTCP
1041 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1043 g_return_if_fail (RTP_IS_SESSION (sess));
1045 RTP_SESSION_LOCK (sess);
1046 sess->stats.rtcp_bandwidth = bandwidth;
1047 RTP_SESSION_UNLOCK (sess);
1051 * rtp_session_get_rtcp_fraction:
1052 * @sess: an #RTPSession
1054 * Get the session bandwidth used for RTCP.
1056 * Returns: The bandwidth used for RTCP messages.
1059 rtp_session_get_rtcp_fraction (RTPSession * sess)
1063 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1065 RTP_SESSION_LOCK (sess);
1066 result = sess->stats.rtcp_bandwidth;
1067 RTP_SESSION_UNLOCK (sess);
1073 * rtp_session_get_sdes_struct:
1074 * @sess: an #RTSPSession
1076 * Get the SDES data as a #GstStructure
1078 * Returns: a GstStructure with SDES items for @sess. This function returns a
1079 * copy of the SDES structure, use gst_structure_free() after usage.
1082 rtp_session_get_sdes_struct (RTPSession * sess)
1084 GstStructure *result = NULL;
1086 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1088 RTP_SESSION_LOCK (sess);
1090 result = gst_structure_copy (sess->sdes);
1091 RTP_SESSION_UNLOCK (sess);
1097 * rtp_session_set_sdes_struct:
1098 * @sess: an #RTSPSession
1099 * @sdes: a #GstStructure
1101 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1104 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1106 g_return_if_fail (sdes);
1107 g_return_if_fail (RTP_IS_SESSION (sess));
1109 RTP_SESSION_LOCK (sess);
1111 gst_structure_free (sess->sdes);
1112 sess->sdes = gst_structure_copy (sdes);
1113 RTP_SESSION_UNLOCK (sess);
1116 static GstFlowReturn
1117 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1119 GstFlowReturn result = GST_FLOW_OK;
1121 if (source->internal) {
1122 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1124 RTP_SESSION_UNLOCK (session);
1126 if (session->callbacks.send_rtp)
1128 session->callbacks.send_rtp (session, source, data,
1129 session->send_rtp_user_data);
1131 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1134 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1135 RTP_SESSION_UNLOCK (session);
1137 if (session->callbacks.process_rtp)
1139 session->callbacks.process_rtp (session, source,
1140 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1142 gst_buffer_unref (GST_BUFFER_CAST (data));
1144 RTP_SESSION_LOCK (session);
1150 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1154 RTP_SESSION_UNLOCK (session);
1156 if (session->callbacks.clock_rate)
1158 session->callbacks.clock_rate (session, pt,
1159 session->clock_rate_user_data);
1163 RTP_SESSION_LOCK (session);
1165 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1170 static RTPSourceCallbacks callbacks = {
1171 (RTPSourcePushRTP) source_push_rtp,
1172 (RTPSourceClockRate) source_clock_rate,
1176 check_collision (RTPSession * sess, RTPSource * source,
1177 RTPArrivalStats * arrival, gboolean rtp)
1179 /* If we have no arrival address, we can't do collision checking */
1180 if (!arrival->address)
1183 if (!source->internal) {
1184 GSocketAddress *from;
1186 /* This is not our local source, but lets check if two remote
1189 from = source->rtp_from;
1191 from = source->rtcp_from;
1195 if (__g_socket_address_equal (from, arrival->address)) {
1196 /* Address is the same */
1199 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1200 rtp_source_get_ssrc (source));
1201 if (sess->favor_new) {
1202 if (rtp_source_find_conflicting_address (source,
1203 arrival->address, arrival->current_time)) {
1206 buf1 = __g_socket_address_to_string (arrival->address);
1207 GST_LOG ("Known conflict on %x for %s, dropping packet",
1208 rtp_source_get_ssrc (source), buf1);
1215 /* Current address is not a known conflict, lets assume this is
1216 * a new source. Save old address in possible conflict list
1218 rtp_source_add_conflicting_address (source, from,
1219 arrival->current_time);
1221 buf1 = __g_socket_address_to_string (from);
1222 buf2 = __g_socket_address_to_string (arrival->address);
1224 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1225 " saving old as known conflict",
1226 rtp_source_get_ssrc (source), buf1, buf2);
1229 rtp_source_set_rtp_from (source, arrival->address);
1231 rtp_source_set_rtcp_from (source, arrival->address);
1239 /* Don't need to save old addresses, we ignore new sources */
1244 /* We don't already have a from address for RTP, just set it */
1246 rtp_source_set_rtp_from (source, arrival->address);
1248 rtp_source_set_rtcp_from (source, arrival->address);
1252 /* FIXME: Log 3rd party collision somehow
1253 * Maybe should be done in upper layer, only the SDES can tell us
1254 * if its a collision or a loop
1257 /* This is sending with our ssrc, is it an address we already know */
1259 if (rtp_source_find_conflicting_address (source, arrival->address,
1260 arrival->current_time)) {
1261 /* Its a known conflict, its probably a loop, not a collision
1262 * lets just drop the incoming packet
1264 GST_DEBUG ("Our packets are being looped back to us, dropping");
1266 /* Its a new collision, lets change our SSRC */
1268 rtp_source_add_conflicting_address (source, arrival->address,
1269 arrival->current_time);
1271 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1272 on_ssrc_collision (sess, source);
1274 sess->change_ssrc = TRUE;
1276 rtp_source_mark_bye (source, "SSRC Collision");
1277 rtp_session_schedule_bye_locked (sess, arrival->current_time);
1285 find_source (RTPSession * sess, guint32 ssrc)
1287 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1288 GINT_TO_POINTER (ssrc));
1292 add_source (RTPSession * sess, RTPSource * src)
1294 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1295 GINT_TO_POINTER (src->ssrc), src);
1296 /* we have one more source now */
1297 sess->total_sources++;
1298 if (RTP_SOURCE_IS_ACTIVE (src))
1299 sess->stats.active_sources++;
1300 if (src->internal) {
1301 sess->stats.internal_sources++;
1302 if (sess->suggested_ssrc != src->ssrc)
1303 sess->suggested_ssrc = src->ssrc;
1307 /* must be called with the session lock, the returned source needs to be
1308 * unreffed after usage. */
1310 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1311 RTPArrivalStats * arrival, gboolean rtp)
1315 source = find_source (sess, ssrc);
1316 if (source == NULL) {
1317 /* make new Source in probation and insert */
1318 source = rtp_source_new (ssrc);
1320 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1322 /* for RTP packets we need to set the source in probation. Receiving RTCP
1323 * packets of an SSRC, on the other hand, is a strong indication that we
1324 * are dealing with a valid source. */
1326 g_object_set (source, "probation", sess->probation, NULL);
1328 g_object_set (source, "probation", 0, NULL);
1330 /* store from address, if any */
1331 if (arrival->address) {
1333 rtp_source_set_rtp_from (source, arrival->address);
1335 rtp_source_set_rtcp_from (source, arrival->address);
1338 /* configure a callback on the source */
1339 rtp_source_set_callbacks (source, &callbacks, sess);
1341 add_source (sess, source);
1345 /* check for collision, this updates the address when not previously set */
1346 if (check_collision (sess, source, arrival, rtp)) {
1349 /* Receiving RTCP packets of an SSRC is a strong indication that we
1350 * are dealing with a valid source. */
1352 g_object_set (source, "probation", 0, NULL);
1354 /* update last activity */
1355 source->last_activity = arrival->current_time;
1357 source->last_rtp_activity = arrival->current_time;
1358 g_object_ref (source);
1363 /* must be called with the session lock, the returned source needs to be
1364 * unreffed after usage. */
1366 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1370 source = find_source (sess, ssrc);
1371 if (source == NULL) {
1372 /* make new internal Source and insert */
1373 source = rtp_source_new (ssrc);
1375 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1377 source->validated = TRUE;
1378 source->internal = TRUE;
1379 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1380 rtp_source_set_callbacks (source, &callbacks, sess);
1382 add_source (sess, source);
1387 g_object_ref (source);
1393 * rtp_session_get_internal_source:
1394 * @sess: a #RTPSession
1396 * Get the internal #RTPSource of @sess.
1398 * Returns: The internal #RTPSource. g_object_unref() after usage.
1401 rtp_session_get_internal_source (RTPSession * sess)
1405 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1407 result = g_object_ref (sess->source);
1413 * rtp_session_set_internal_ssrc:
1414 * @sess: a #RTPSession
1417 * Set the SSRC of @sess to @ssrc.
1420 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1422 RTP_SESSION_LOCK (sess);
1423 if (ssrc != sess->source->ssrc) {
1424 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1425 GINT_TO_POINTER (sess->source->ssrc));
1427 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1428 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1429 * packets will timeout on the old SSRC, we could potentially schedule a
1430 * BYE RTCP for the old SSRC... */
1431 sess->source->ssrc = ssrc;
1432 rtp_source_reset (sess->source);
1434 /* rehash with the new SSRC */
1435 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1436 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1438 RTP_SESSION_UNLOCK (sess);
1440 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1444 * rtp_session_get_internal_ssrc:
1445 * @sess: a #RTPSession
1447 * Get the internal SSRC of @sess.
1449 * Returns: The SSRC of the session.
1452 rtp_session_get_internal_ssrc (RTPSession * sess)
1456 RTP_SESSION_LOCK (sess);
1457 ssrc = sess->source->ssrc;
1458 RTP_SESSION_UNLOCK (sess);
1464 * rtp_session_suggest_ssrc:
1465 * @sess: a #RTPSession
1467 * Suggest an unused SSRC in @sess.
1469 * Returns: a free unused SSRC
1472 rtp_session_suggest_ssrc (RTPSession * sess)
1476 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1478 RTP_SESSION_LOCK (sess);
1479 result = sess->suggested_ssrc;
1480 RTP_SESSION_UNLOCK (sess);
1486 * rtp_session_add_source:
1487 * @sess: a #RTPSession
1488 * @src: #RTPSource to add
1490 * Add @src to @session.
1492 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1493 * existed in the session.
1496 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1498 gboolean result = FALSE;
1501 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1502 g_return_val_if_fail (src != NULL, FALSE);
1504 RTP_SESSION_LOCK (sess);
1505 find = find_source (sess, src->ssrc);
1507 add_source (sess, src);
1510 RTP_SESSION_UNLOCK (sess);
1516 * rtp_session_get_num_sources:
1517 * @sess: an #RTPSession
1519 * Get the number of sources in @sess.
1521 * Returns: The number of sources in @sess.
1524 rtp_session_get_num_sources (RTPSession * sess)
1528 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1530 RTP_SESSION_LOCK (sess);
1531 result = sess->total_sources;
1532 RTP_SESSION_UNLOCK (sess);
1538 * rtp_session_get_num_active_sources:
1539 * @sess: an #RTPSession
1541 * Get the number of active sources in @sess. A source is considered active when
1542 * it has been validated and has not yet received a BYE RTCP message.
1544 * Returns: The number of active sources in @sess.
1547 rtp_session_get_num_active_sources (RTPSession * sess)
1551 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1553 RTP_SESSION_LOCK (sess);
1554 result = sess->stats.active_sources;
1555 RTP_SESSION_UNLOCK (sess);
1561 * rtp_session_get_source_by_ssrc:
1562 * @sess: an #RTPSession
1565 * Find the source with @ssrc in @sess.
1567 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1568 * g_object_unref() after usage.
1571 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1575 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1577 RTP_SESSION_LOCK (sess);
1578 result = find_source (sess, ssrc);
1580 g_object_ref (result);
1581 RTP_SESSION_UNLOCK (sess);
1586 /* should be called with the SESSION lock */
1588 rtp_session_create_new_ssrc (RTPSession * sess)
1593 ssrc = g_random_int ();
1595 /* see if it exists in the session, we're done if it doesn't */
1596 if (find_source (sess, ssrc) == NULL)
1604 * rtp_session_create_source:
1605 * @sess: an #RTPSession
1607 * Create an #RTPSource for use in @sess. This function will create a source
1608 * with an ssrc that is currently not used by any participants in the session.
1610 * Returns: an #RTPSource.
1613 rtp_session_create_source (RTPSession * sess)
1618 RTP_SESSION_LOCK (sess);
1619 ssrc = rtp_session_create_new_ssrc (sess);
1620 source = rtp_source_new (ssrc);
1621 rtp_source_set_callbacks (source, &callbacks, sess);
1622 /* we need an additional ref for the source in the hashtable */
1623 g_object_ref (source);
1624 add_source (sess, source);
1625 RTP_SESSION_UNLOCK (sess);
1630 /* update the RTPArrivalStats structure with the current time and other bits
1631 * about the current buffer we are handling.
1632 * This function is typically called when a validated packet is received.
1633 * This function should be called with the SESSION_LOCK
1636 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1637 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1638 GstClockTime running_time, guint64 ntpnstime)
1640 GstNetAddressMeta *meta;
1641 GstRTPBuffer rtpb = { NULL };
1643 /* get time of arrival */
1644 arrival->current_time = current_time;
1645 arrival->running_time = running_time;
1646 arrival->ntpnstime = ntpnstime;
1648 /* get packet size including header overhead */
1649 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1652 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1653 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1654 gst_rtp_buffer_unmap (&rtpb);
1656 arrival->payload_len = 0;
1659 /* for netbuffer we can store the IP address to check for collisions */
1660 meta = gst_buffer_get_net_address_meta (buffer);
1661 if (arrival->address)
1662 g_object_unref (arrival->address);
1664 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1666 arrival->address = NULL;
1671 clean_arrival_stats (RTPArrivalStats * arrival)
1673 if (arrival->address)
1674 g_object_unref (arrival->address);
1678 * rtp_session_process_rtp:
1679 * @sess: and #RTPSession
1680 * @buffer: an RTP buffer
1681 * @current_time: the current system time
1682 * @running_time: the running_time of @buffer
1684 * Process an RTP buffer in the session manager. This function takes ownership
1687 * Returns: a #GstFlowReturn.
1690 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1691 GstClockTime current_time, GstClockTime running_time)
1693 GstFlowReturn result;
1697 gboolean prevsender, prevactive;
1698 RTPArrivalStats arrival = { NULL, };
1702 GstRTPBuffer rtp = { NULL };
1704 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1705 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1707 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1708 goto invalid_packet;
1710 /* get SSRC to look up in session database */
1711 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1712 /* copy available csrc for later */
1713 count = gst_rtp_buffer_get_csrc_count (&rtp);
1714 /* make sure to not overflow our array. An RTP buffer can maximally contain
1716 count = MIN (count, 16);
1718 for (i = 0; i < count; i++)
1719 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1721 gst_rtp_buffer_unmap (&rtp);
1723 RTP_SESSION_LOCK (sess);
1724 /* ignore more RTP packets when we left the session */
1725 if (sess->source->marked_bye)
1728 /* update arrival stats */
1729 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1732 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1736 prevsender = RTP_SOURCE_IS_SENDER (source);
1737 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1738 oldrate = source->bitrate;
1740 /* let source process the packet */
1741 result = rtp_source_process_rtp (source, buffer, &arrival);
1743 /* source became active */
1744 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1745 sess->stats.active_sources++;
1746 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1747 sess->stats.active_sources);
1748 on_ssrc_validated (sess, source);
1750 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1751 sess->stats.sender_sources++;
1752 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1753 sess->stats.sender_sources);
1755 if (oldrate != source->bitrate)
1756 sess->recalc_bandwidth = TRUE;
1759 on_new_ssrc (sess, source);
1761 if (source->validated) {
1764 /* for validated sources, we add the CSRCs as well */
1765 for (i = 0; i < count; i++) {
1767 RTPSource *csrc_src;
1772 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1777 GST_DEBUG ("created new CSRC: %08x", csrc);
1778 rtp_source_set_as_csrc (csrc_src);
1779 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1780 sess->stats.active_sources++;
1781 on_new_ssrc (sess, csrc_src);
1783 g_object_unref (csrc_src);
1786 g_object_unref (source);
1788 RTP_SESSION_UNLOCK (sess);
1790 clean_arrival_stats (&arrival);
1797 gst_buffer_unref (buffer);
1798 GST_DEBUG ("invalid RTP packet received");
1803 RTP_SESSION_UNLOCK (sess);
1804 gst_buffer_unref (buffer);
1805 GST_DEBUG ("ignoring RTP packet because we are leaving");
1810 RTP_SESSION_UNLOCK (sess);
1811 gst_buffer_unref (buffer);
1812 clean_arrival_stats (&arrival);
1813 GST_DEBUG ("ignoring packet because its collisioning");
1819 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1820 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1824 count = gst_rtcp_packet_get_rb_count (packet);
1825 for (i = 0; i < count; i++) {
1826 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1827 guint8 fractionlost;
1830 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1831 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1833 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1835 if (ssrc == sess->source->ssrc) {
1836 /* only deal with report blocks for our session, we update the stats of
1837 * the sender of the RTCP message. We could also compare our stats against
1838 * the other sender to see if we are better or worse. */
1839 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1840 packetslost, exthighestseq, jitter, lsr, dlsr);
1843 on_ssrc_active (sess, source);
1846 /* A Sender report contains statistics about how the sender is doing. This
1847 * includes timing informataion such as the relation between RTP and NTP
1848 * timestamps and the number of packets/bytes it sent to us.
1850 * In this report is also included a set of report blocks related to how this
1851 * sender is receiving data (in case we (or somebody else) is also sending stuff
1852 * to it). This info includes the packet loss, jitter and seqnum. It also
1853 * contains information to calculate the round trip time (LSR/DLSR).
1856 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1857 RTPArrivalStats * arrival, gboolean * do_sync)
1859 guint32 senderssrc, rtptime, packet_count, octet_count;
1862 gboolean created, prevsender;
1864 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1865 &packet_count, &octet_count);
1867 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1868 senderssrc, GST_TIME_ARGS (arrival->current_time));
1870 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1874 /* don't try to do lip-sync for sources that sent a BYE */
1875 if (RTP_SOURCE_IS_MARKED_BYE (source))
1880 prevsender = RTP_SOURCE_IS_SENDER (source);
1882 /* first update the source */
1883 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1884 packet_count, octet_count);
1886 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1887 sess->stats.sender_sources++;
1888 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1889 sess->stats.sender_sources);
1893 on_new_ssrc (sess, source);
1895 rtp_session_process_rb (sess, source, packet, arrival);
1896 g_object_unref (source);
1899 /* A receiver report contains statistics about how a receiver is doing. It
1900 * includes stuff like packet loss, jitter and the seqnum it received last. It
1901 * also contains info to calculate the round trip time.
1903 * We are only interested in how the sender of this report is doing wrt to us.
1906 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1907 RTPArrivalStats * arrival)
1913 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1915 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1917 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1922 on_new_ssrc (sess, source);
1924 rtp_session_process_rb (sess, source, packet, arrival);
1925 g_object_unref (source);
1928 /* Get SDES items and store them in the SSRC */
1930 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1931 RTPArrivalStats * arrival)
1934 gboolean more_items, more_entries;
1936 items = gst_rtcp_packet_sdes_get_item_count (packet);
1937 GST_DEBUG ("got SDES packet with %d items", items);
1939 more_items = gst_rtcp_packet_sdes_first_item (packet);
1941 while (more_items) {
1943 gboolean changed, created, validated;
1947 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1949 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1953 /* find src, no probation when dealing with RTCP */
1954 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1958 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1960 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1962 while (more_entries) {
1963 GstRTCPSDESType type;
1969 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1971 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1974 if (type == GST_RTCP_SDES_PRIV) {
1975 name = g_strndup ((const gchar *) &data[1], data[0]);
1977 data += data[0] + 1;
1979 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1982 value = g_strndup ((const gchar *) data, len);
1984 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1989 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1993 /* takes ownership of sdes */
1994 changed = rtp_source_set_sdes_struct (source, sdes);
1996 validated = !RTP_SOURCE_IS_ACTIVE (source);
1997 source->validated = TRUE;
2000 on_new_ssrc (sess, source);
2002 /* source became active */
2004 sess->stats.active_sources++;
2005 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
2006 sess->stats.active_sources);
2007 on_ssrc_validated (sess, source);
2011 on_ssrc_sdes (sess, source);
2013 g_object_unref (source);
2015 more_items = gst_rtcp_packet_sdes_next_item (packet);
2020 /* BYE is sent when a client leaves the session
2023 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2024 RTPArrivalStats * arrival)
2028 gboolean reconsider = FALSE;
2030 reason = gst_rtcp_packet_bye_get_reason (packet);
2031 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2033 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2034 for (i = 0; i < count; i++) {
2037 gboolean created, prevactive, prevsender;
2038 guint pmembers, members;
2040 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2041 GST_DEBUG ("SSRC: %08x", ssrc);
2043 /* find src and mark bye, no probation when dealing with RTCP */
2044 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
2048 if (source->internal) {
2049 /* our own source, something weird with this packet */
2050 g_object_unref (source);
2054 /* store time for when we need to time out this source */
2055 source->bye_time = arrival->current_time;
2057 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2058 prevsender = RTP_SOURCE_IS_SENDER (source);
2060 /* mark the source BYE */
2061 rtp_source_mark_bye (source, reason);
2063 pmembers = sess->stats.active_sources;
2065 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2066 sess->stats.active_sources--;
2067 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2068 sess->stats.active_sources);
2070 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2071 sess->stats.sender_sources--;
2072 if (source->internal)
2073 sess->stats.internal_sender_sources--;
2074 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2075 sess->stats.sender_sources);
2077 members = sess->stats.active_sources;
2079 if (!sess->scheduled_bye && members < pmembers) {
2080 /* some members went away since the previous timeout estimate.
2081 * Perform reverse reconsideration but only when we are not scheduling a
2083 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2084 arrival->current_time < sess->next_rtcp_check_time) {
2085 GstClockTime time_remaining;
2087 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2088 sess->next_rtcp_check_time =
2089 gst_util_uint64_scale (time_remaining, members, pmembers);
2091 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2092 GST_TIME_ARGS (sess->next_rtcp_check_time));
2094 sess->next_rtcp_check_time += arrival->current_time;
2096 /* mark pending reconsider. We only want to signal the reconsideration
2097 * once after we handled all the source in the bye packet */
2103 on_new_ssrc (sess, source);
2105 on_bye_ssrc (sess, source);
2107 g_object_unref (source);
2110 RTP_SESSION_UNLOCK (sess);
2111 /* notify app of reconsideration */
2112 if (sess->callbacks.reconsider)
2113 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2114 RTP_SESSION_LOCK (sess);
2120 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2121 RTPArrivalStats * arrival)
2123 GST_DEBUG ("received APP");
2127 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2128 gboolean fir, GstClockTime current_time)
2130 guint32 round_trip = 0;
2132 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2134 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2135 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2138 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2139 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2140 GST_DEBUG ("Ignoring %s request because one was send without one "
2141 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2142 fir ? "FIR" : "PLI",
2143 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2144 GST_TIME_ARGS (round_trip_in_ns));;
2149 sess->last_keyframe_request = current_time;
2151 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2152 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2153 sess->callbacks.request_key_unit);
2155 RTP_SESSION_UNLOCK (sess);
2156 sess->callbacks.request_key_unit (sess, fir,
2157 sess->request_key_unit_user_data);
2158 RTP_SESSION_LOCK (sess);
2164 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2165 guint32 media_ssrc, GstClockTime current_time)
2169 if (!sess->callbacks.request_key_unit)
2172 src = find_source (sess, sender_ssrc);
2176 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2180 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2181 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2186 gboolean our_request = FALSE;
2188 if (!sess->callbacks.request_key_unit)
2194 src = find_source (sess, sender_ssrc);
2196 /* Hack because Google fails to set the sender_ssrc correctly */
2197 if (!src && sender_ssrc == 1) {
2198 GHashTableIter iter;
2200 if (sess->stats.sender_sources >
2201 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2204 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2206 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2207 if (src != sess->source && rtp_source_is_sender (src))
2216 for (position = 0; position < fci_length; position += 8) {
2217 guint8 *data = fci_data + position;
2220 ssrc = GST_READ_UINT32_BE (data);
2222 own = find_source (sess, ssrc);
2223 if (own->internal) {
2231 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2235 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2236 RTPArrivalStats * arrival, GstClockTime current_time)
2238 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2239 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2240 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2241 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2242 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2243 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2246 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2247 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2249 if (g_signal_has_handler_pending (sess,
2250 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2251 GstBuffer *fci_buffer = NULL;
2253 if (fci_length > 0) {
2254 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2255 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2257 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2260 RTP_SESSION_UNLOCK (sess);
2261 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2262 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2263 RTP_SESSION_LOCK (sess);
2266 gst_buffer_unref (fci_buffer);
2269 src = find_source (sess, media_ssrc);
2273 if (sess->rtcp_feedback_retention_window) {
2274 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2277 if (src->internal ||
2278 /* PSFB FIR puts the media ssrc inside the FCI */
2279 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2281 case GST_RTCP_TYPE_PSFB:
2283 case GST_RTCP_PSFB_TYPE_PLI:
2284 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2287 case GST_RTCP_PSFB_TYPE_FIR:
2288 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2295 case GST_RTCP_TYPE_RTPFB:
2303 * rtp_session_process_rtcp:
2304 * @sess: and #RTPSession
2305 * @buffer: an RTCP buffer
2306 * @current_time: the current system time
2307 * @ntpnstime: the current NTP time in nanoseconds
2309 * Process an RTCP buffer in the session manager. This function takes ownership
2312 * Returns: a #GstFlowReturn.
2315 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2316 GstClockTime current_time, guint64 ntpnstime)
2318 GstRTCPPacket packet;
2319 gboolean more, is_bye = FALSE, do_sync = FALSE;
2320 RTPArrivalStats arrival = { NULL, };
2321 GstFlowReturn result = GST_FLOW_OK;
2322 GstRTCPBuffer rtcp = { NULL, };
2324 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2325 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2327 if (!gst_rtcp_buffer_validate (buffer))
2328 goto invalid_packet;
2330 GST_DEBUG ("received RTCP packet");
2332 RTP_SESSION_LOCK (sess);
2333 /* update arrival stats */
2334 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2337 if (sess->source->sent_bye)
2340 /* start processing the compound packet */
2341 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2342 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2346 type = gst_rtcp_packet_get_type (&packet);
2348 /* when we are leaving the session, we should ignore all non-BYE messages */
2349 if (sess->scheduled_bye && type != GST_RTCP_TYPE_BYE) {
2350 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2355 case GST_RTCP_TYPE_SR:
2356 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2358 case GST_RTCP_TYPE_RR:
2359 rtp_session_process_rr (sess, &packet, &arrival);
2361 case GST_RTCP_TYPE_SDES:
2362 rtp_session_process_sdes (sess, &packet, &arrival);
2364 case GST_RTCP_TYPE_BYE:
2366 /* don't try to attempt lip-sync anymore for streams with a BYE */
2368 rtp_session_process_bye (sess, &packet, &arrival);
2370 case GST_RTCP_TYPE_APP:
2371 rtp_session_process_app (sess, &packet, &arrival);
2373 case GST_RTCP_TYPE_RTPFB:
2374 case GST_RTCP_TYPE_PSFB:
2375 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2378 GST_WARNING ("got unknown RTCP packet");
2382 more = gst_rtcp_packet_move_to_next (&packet);
2385 gst_rtcp_buffer_unmap (&rtcp);
2387 /* if we are scheduling a BYE, we only want to count bye packets, else we
2388 * count everything */
2389 if (sess->scheduled_bye) {
2391 sess->stats.bye_members++;
2392 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2395 /* keep track of average packet size */
2396 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2398 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2399 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2400 RTP_SESSION_UNLOCK (sess);
2402 clean_arrival_stats (&arrival);
2404 /* notify caller of sr packets in the callback */
2405 if (do_sync && sess->callbacks.sync_rtcp) {
2406 /* make writable, we might want to change the buffer */
2407 buffer = gst_buffer_make_writable (buffer);
2409 result = sess->callbacks.sync_rtcp (sess, buffer,
2410 sess->sync_rtcp_user_data);
2412 gst_buffer_unref (buffer);
2419 GST_DEBUG ("invalid RTCP packet received");
2420 gst_buffer_unref (buffer);
2425 RTP_SESSION_UNLOCK (sess);
2426 gst_buffer_unref (buffer);
2427 clean_arrival_stats (&arrival);
2428 GST_DEBUG ("ignoring RTCP packet because we left");
2434 * rtp_session_update_send_caps:
2435 * @sess: an #RTPSession
2438 * Update the caps of the sender in the rtp session.
2441 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2443 g_return_if_fail (RTP_IS_SESSION (sess));
2444 g_return_if_fail (GST_IS_CAPS (caps));
2446 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2448 RTP_SESSION_LOCK (sess);
2449 rtp_source_update_caps (sess->source, caps);
2450 RTP_SESSION_UNLOCK (sess);
2454 * rtp_session_send_rtp:
2455 * @sess: an #RTPSession
2456 * @data: pointer to either an RTP buffer or a list of RTP buffers
2457 * @is_list: TRUE when @data is a buffer list
2458 * @current_time: the current system time
2459 * @running_time: the running time of @data
2461 * Send the RTP buffer in the session manager. This function takes ownership of
2464 * Returns: a #GstFlowReturn.
2467 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2468 GstClockTime current_time, GstClockTime running_time)
2470 GstFlowReturn result;
2472 gboolean prevsender;
2475 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2476 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2478 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2480 RTP_SESSION_LOCK (sess);
2481 source = sess->source;
2483 /* update last activity */
2484 source->last_rtp_activity = current_time;
2486 prevsender = RTP_SOURCE_IS_SENDER (source);
2487 oldrate = source->bitrate;
2489 /* we use our own source to send */
2490 result = rtp_source_send_rtp (source, data, is_list, running_time);
2492 if (RTP_SOURCE_IS_SENDER (source) && !prevsender) {
2493 sess->stats.sender_sources++;
2494 sess->stats.internal_sender_sources++;
2496 if (oldrate != source->bitrate)
2497 sess->recalc_bandwidth = TRUE;
2498 RTP_SESSION_UNLOCK (sess);
2504 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2506 *bandwidth += source->bitrate;
2509 /* must be called with session lock */
2511 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2514 GstClockTime result;
2516 /* recalculate bandwidth when it changed */
2517 if (sess->recalc_bandwidth) {
2520 if (sess->bandwidth > 0)
2521 bandwidth = sess->bandwidth;
2523 /* If it is <= 0, then try to estimate the actual bandwidth */
2526 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2527 (GHFunc) add_bitrates, &bandwidth);
2530 if (bandwidth < 8000)
2531 bandwidth = RTP_STATS_BANDWIDTH;
2533 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2534 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2536 sess->recalc_bandwidth = FALSE;
2539 if (sess->scheduled_bye) {
2540 result = rtp_stats_calculate_bye_interval (&sess->stats);
2542 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2543 sess->stats.internal_sender_sources > 0, first);
2546 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2547 GST_TIME_ARGS (result), first);
2549 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2550 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2552 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2558 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2560 if (source->internal)
2561 rtp_source_mark_bye (source, reason);
2565 * rtp_session_mark_all_bye:
2566 * @sess: an #RTPSession
2569 * Mark all internal sources of the session as BYE with @reason.
2572 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2574 g_return_if_fail (RTP_IS_SESSION (sess));
2576 RTP_SESSION_LOCK (sess);
2577 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2578 (GHFunc) source_mark_bye, (gpointer) reason);
2579 RTP_SESSION_UNLOCK (sess);
2582 /* Stop the current @sess and schedule a BYE message for the other members.
2583 * One must have the session lock to call this function
2585 static GstFlowReturn
2586 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2588 GstFlowReturn result = GST_FLOW_OK;
2589 GstClockTime interval;
2591 /* nothing to do it we already scheduled bye */
2592 if (sess->scheduled_bye)
2595 /* we schedule BYE now */
2596 sess->scheduled_bye = TRUE;
2597 /* at least one member wants to send a BYE */
2598 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2599 sess->stats.bye_members = 1;
2600 sess->first_rtcp = TRUE;
2601 sess->allow_early = TRUE;
2603 /* reschedule transmission */
2604 sess->last_rtcp_send_time = current_time;
2605 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2607 if (interval != GST_CLOCK_TIME_NONE)
2608 sess->next_rtcp_check_time = current_time + interval;
2610 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2612 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2613 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2615 RTP_SESSION_UNLOCK (sess);
2616 /* notify app of reconsideration */
2617 if (sess->callbacks.reconsider)
2618 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2619 RTP_SESSION_LOCK (sess);
2626 * rtp_session_schedule_bye:
2627 * @sess: an #RTPSession
2628 * @current_time: the current system time
2630 * Schedule a BYE message for all sources marked as BYE in @sess.
2632 * Returns: a #GstFlowReturn.
2635 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2637 GstFlowReturn result = GST_FLOW_OK;
2639 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2641 RTP_SESSION_LOCK (sess);
2642 result = rtp_session_schedule_bye_locked (sess, current_time);
2643 RTP_SESSION_UNLOCK (sess);
2649 * rtp_session_next_timeout:
2650 * @sess: an #RTPSession
2651 * @current_time: the current system time
2653 * Get the next time we should perform session maintenance tasks.
2655 * Returns: a time when rtp_session_on_timeout() should be called with the
2656 * current system time.
2659 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2661 GstClockTime result, interval = 0;
2663 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2665 RTP_SESSION_LOCK (sess);
2667 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2668 result = sess->next_early_rtcp_time;
2672 result = sess->next_rtcp_check_time;
2674 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2675 ", next time: %" GST_TIME_FORMAT,
2676 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2678 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2679 GST_DEBUG ("take current time as base");
2680 /* our previous check time expired, start counting from the current time
2682 result = current_time;
2685 if (sess->scheduled_bye) {
2686 if (sess->source->sent_bye) {
2687 GST_DEBUG ("we sent BYE already");
2688 interval = GST_CLOCK_TIME_NONE;
2689 } else if (sess->stats.active_sources >= 50) {
2690 GST_DEBUG ("reconsider BYE, more than 50 sources");
2691 /* reconsider BYE if members >= 50 */
2692 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2695 if (sess->first_rtcp) {
2696 GST_DEBUG ("first RTCP packet");
2697 /* we are called for the first time */
2698 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2699 } else if (sess->next_rtcp_check_time < current_time) {
2700 GST_DEBUG ("old check time expired, getting new timeout");
2701 /* get a new timeout when we need to */
2702 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2706 if (interval != GST_CLOCK_TIME_NONE)
2709 result = GST_CLOCK_TIME_NONE;
2711 sess->next_rtcp_check_time = result;
2715 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2716 ", next time: %" GST_TIME_FORMAT,
2717 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2718 RTP_SESSION_UNLOCK (sess);
2725 GstRTCPBuffer rtcpbuf;
2729 GstClockTime current_time;
2731 GstClockTime running_time;
2732 GstClockTime interval;
2733 GstRTCPPacket packet;
2737 gboolean may_suppress;
2742 session_start_rtcp (RTPSession * sess, ReportData * data)
2744 GstRTCPPacket *packet = &data->packet;
2745 RTPSource *own = data->source;
2746 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2748 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2749 data->is_bye = FALSE;
2750 data->has_sdes = FALSE;
2752 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2754 if (RTP_SOURCE_IS_SENDER (own)) {
2757 guint32 packet_count, octet_count;
2759 /* we are a sender, create SR */
2760 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2761 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2763 /* get latest stats */
2764 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2765 &ntptime, &rtptime, &packet_count, &octet_count);
2767 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2768 packet_count, octet_count);
2770 /* fill in sender report info */
2771 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2772 ntptime, rtptime, packet_count, octet_count);
2774 /* we are only receiver, create RR */
2775 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2776 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2777 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2781 /* construct a Sender or Receiver Report */
2783 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2785 GstRTCPPacket *packet = &data->packet;
2787 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2788 /* only report about other sender sources */
2789 if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
2790 guint8 fractionlost;
2792 guint32 exthighestseq, jitter;
2796 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2797 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2799 /* store last generated RR packet */
2800 source->last_rr.is_valid = TRUE;
2801 source->last_rr.fractionlost = fractionlost;
2802 source->last_rr.packetslost = packetslost;
2803 source->last_rr.exthighestseq = exthighestseq;
2804 source->last_rr.jitter = jitter;
2805 source->last_rr.lsr = lsr;
2806 source->last_rr.dlsr = dlsr;
2808 /* packet is not yet filled, add report block for this source. */
2809 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2810 exthighestseq, jitter, lsr, dlsr);
2815 /* perform cleanup of sources that timed out */
2817 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2819 gboolean remove = FALSE;
2820 gboolean byetimeout = FALSE;
2821 gboolean sendertimeout = FALSE;
2822 gboolean is_sender, is_active;
2823 RTPSession *sess = data->sess;
2824 GstClockTime interval, binterval;
2827 /* check for outdated collisions */
2828 if (source->internal) {
2829 GST_DEBUG ("Timing out collisions");
2830 rtp_source_timeout (source, data->current_time,
2831 /* "a relatively long time" -- RFC 3550 section 8.2 */
2832 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
2833 data->running_time - sess->rtcp_feedback_retention_window);
2836 /* nothing else to do when without RTCP */
2837 if (data->interval == GST_CLOCK_TIME_NONE)
2840 is_sender = RTP_SOURCE_IS_SENDER (source);
2841 is_active = RTP_SOURCE_IS_ACTIVE (source);
2843 /* our own rtcp interval may have been forced low by secondary configuration,
2844 * while sender side may still operate with higher interval,
2845 * so do not just take our interval to decide on timing out sender,
2846 * but take (if data->interval <= 5 * GST_SECOND):
2847 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2848 * where sender_interval is difference between last 2 received RTCP reports
2850 if (data->interval >= 5 * GST_SECOND || source->internal) {
2851 binterval = data->interval;
2853 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2854 GST_TIME_ARGS (source->stats.prev_rtcptime),
2855 GST_TIME_ARGS (source->stats.last_rtcptime));
2856 /* if not received enough yet, fallback to larger default */
2857 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2858 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2860 binterval = 5 * GST_SECOND;
2861 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2863 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2864 GST_TIME_ARGS (binterval));
2866 /* check for our own source, we don't want to delete our own source. */
2867 if (!source->internal) {
2868 if (source->marked_bye) {
2869 /* if we received a BYE from the source, remove the source after some
2871 if (data->current_time > source->bye_time &&
2872 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2873 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2878 /* sources that were inactive for more than 5 times the deterministic reporting
2879 * interval get timed out. the min timeout is 5 seconds. */
2880 /* mind old time that might pre-date last time going to PLAYING */
2881 btime = MAX (source->last_activity, sess->start_time);
2882 if (data->current_time > btime) {
2883 interval = MAX (binterval * 5, 5 * GST_SECOND);
2884 if (data->current_time - btime > interval) {
2885 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2886 source->ssrc, GST_TIME_ARGS (btime));
2892 /* senders that did not send for a long time become a receiver, this also
2893 * holds for our own sources. */
2895 /* mind old time that might pre-date last time going to PLAYING */
2896 btime = MAX (source->last_rtp_activity, sess->start_time);
2897 if (data->current_time > btime) {
2898 interval = MAX (binterval * 2, 5 * GST_SECOND);
2899 if (data->current_time - btime > interval) {
2900 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2901 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2902 source->is_sender = FALSE;
2903 sess->stats.sender_sources--;
2904 if (source->internal)
2905 sess->stats.internal_sender_sources--;
2906 sendertimeout = TRUE;
2912 sess->total_sources--;
2914 sess->stats.sender_sources--;
2915 if (source->internal)
2916 sess->stats.internal_sender_sources--;
2919 sess->stats.active_sources--;
2921 if (source->internal)
2922 sess->stats.internal_sources--;
2925 on_bye_timeout (sess, source);
2927 on_timeout (sess, source);
2930 on_sender_timeout (sess, source);
2933 source->closing = remove;
2937 session_sdes (RTPSession * sess, ReportData * data)
2939 GstRTCPPacket *packet = &data->packet;
2940 const GstStructure *sdes;
2942 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2944 /* add SDES packet */
2945 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2947 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
2949 sdes = rtp_source_get_sdes_struct (data->source);
2951 /* add all fields in the structure, the order is not important. */
2952 n_fields = gst_structure_n_fields (sdes);
2953 for (i = 0; i < n_fields; ++i) {
2956 GstRTCPSDESType type;
2958 field = gst_structure_nth_field_name (sdes, i);
2961 value = gst_structure_get_string (sdes, field);
2964 type = gst_rtcp_sdes_name_to_type (field);
2966 /* Early packets are minimal and only include the CNAME */
2967 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2970 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2971 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2972 (const guint8 *) value);
2973 } else if (type == GST_RTCP_SDES_PRIV) {
2979 /* don't accept entries that are too big */
2980 prefix_len = strlen (field);
2981 if (prefix_len > 255)
2983 value_len = strlen (value);
2984 if (value_len > 255)
2986 data_len = 1 + prefix_len + value_len;
2990 data[0] = prefix_len;
2991 memcpy (&data[1], field, prefix_len);
2992 memcpy (&data[1 + prefix_len], value, value_len);
2994 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2998 data->has_sdes = TRUE;
3001 /* schedule a BYE packet */
3003 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3005 GstRTCPPacket *packet = &data->packet;
3006 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3009 session_sdes (sess, data);
3010 /* add a BYE packet */
3011 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3012 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3013 if (source->bye_reason)
3014 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3016 /* we have a BYE packet now */
3017 data->is_bye = TRUE;
3018 source->sent_bye = TRUE;
3022 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3024 GstClockTime new_send_time, elapsed;
3026 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3027 data->is_early = TRUE;
3029 data->is_early = FALSE;
3031 if (data->is_early && sess->next_early_rtcp_time < current_time)
3034 /* no need to check yet */
3035 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3036 sess->next_rtcp_check_time > current_time) {
3037 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3038 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3039 GST_TIME_ARGS (current_time));
3043 /* get elapsed time since we last reported */
3044 elapsed = current_time - sess->last_rtcp_send_time;
3046 new_send_time = data->interval;
3047 /* perform forward reconsideration */
3048 if (new_send_time != GST_CLOCK_TIME_NONE) {
3049 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
3051 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3052 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time),
3053 GST_TIME_ARGS (elapsed));
3055 new_send_time += sess->last_rtcp_send_time;
3058 /* check if reconsideration */
3059 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3060 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3061 GST_TIME_ARGS (new_send_time));
3062 /* store new check time */
3063 sess->next_rtcp_check_time = new_send_time;
3069 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
3071 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3072 GST_TIME_ARGS (new_send_time));
3074 sess->next_rtcp_check_time = new_send_time;
3075 if (new_send_time != GST_CLOCK_TIME_NONE) {
3076 sess->next_rtcp_check_time += current_time;
3078 /* Apply the rules from RFC 4585 section 3.5.3 */
3079 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
3080 GstClockTimeDiff T_rr_current_interval =
3081 g_random_double_range (0.5, 1.5) * sess->stats.min_interval;
3083 /* This will caused the RTCP to be suppressed if no FB packets are added */
3084 if (sess->last_rtcp_send_time + T_rr_current_interval >
3085 sess->next_rtcp_check_time) {
3086 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3087 " last: %" GST_TIME_FORMAT
3088 " + T_rr_current_interval: %" GST_TIME_FORMAT
3089 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3090 GST_TIME_ARGS (sess->stats.min_interval),
3091 GST_TIME_ARGS (sess->last_rtcp_send_time),
3092 GST_TIME_ARGS (T_rr_current_interval),
3093 GST_TIME_ARGS (sess->next_rtcp_check_time));
3094 data->may_suppress = TRUE;
3103 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3105 g_hash_table_insert (hash_table, key, g_object_ref (source));
3109 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3111 return source->closing;
3115 generate_rtcp (RTPSource * source, ReportData * data)
3117 RTPSession *sess = data->sess;
3119 /* only generate RTCP for active internal sources */
3120 if (!source->internal || source->sent_bye)
3123 data->source = source;
3126 session_start_rtcp (sess, data);
3128 if (source->marked_bye) {
3130 make_source_bye (sess, source, data);
3131 } else if (!data->is_early) {
3132 /* loop over all known sources and add report blocks. If we are ealy, we
3133 * just make a minimal RTCP packet and skip this step */
3134 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3135 (GHFunc) session_report_blocks, data);
3137 if (!data->has_sdes)
3138 session_sdes (sess, data);
3140 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3142 if (sess->change_ssrc) {
3143 GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
3144 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3145 GINT_TO_POINTER (source->ssrc));
3147 source->ssrc = rtp_session_create_new_ssrc (sess);
3148 rtp_source_reset (source);
3150 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3151 GINT_TO_POINTER (source->ssrc), source);
3153 sess->change_ssrc = FALSE;
3154 data->notify = TRUE;
3155 GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
3160 * rtp_session_on_timeout:
3161 * @sess: an #RTPSession
3162 * @current_time: the current system time
3163 * @ntpnstime: the current NTP time in nanoseconds
3164 * @running_time: the current running_time of the pipeline
3166 * Perform maintenance actions after the timeout obtained with
3167 * rtp_session_next_timeout() expired.
3169 * This function will perform timeouts of receivers and senders, send a BYE
3170 * packet or generate RTCP packets with current session stats.
3172 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3173 * times, for each packet that should be processed.
3175 * Returns: a #GstFlowReturn.
3178 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3179 guint64 ntpnstime, GstClockTime running_time)
3181 GstFlowReturn result = GST_FLOW_OK;
3182 ReportData data = { GST_RTCP_BUFFER_INIT };
3184 GHashTable *table_copy;
3186 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3188 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3189 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3190 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3193 data.current_time = current_time;
3194 data.ntpnstime = ntpnstime;
3195 data.running_time = running_time;
3196 data.may_suppress = FALSE;
3197 data.notify = FALSE;
3201 RTP_SESSION_LOCK (sess);
3202 /* get a new interval, we need this for various cleanups etc */
3203 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3205 /* Make a local copy of the hashtable. We need to do this because the
3206 * cleanup stage below releases the session lock. */
3207 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3208 (GDestroyNotify) g_object_unref);
3209 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3210 (GHFunc) clone_ssrcs_hashtable, table_copy);
3212 /* Clean up the session, mark the source for removing, this might release the
3214 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3215 g_hash_table_destroy (table_copy);
3217 /* Now remove the marked sources */
3218 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3219 (GHRFunc) remove_closing_sources, NULL);
3221 /* see if we need to generate SR or RR packets */
3222 if (!is_rtcp_time (sess, current_time, &data))
3225 generate_rtcp (own, &data);
3227 /* we keep track of the last report time in order to timeout inactive
3228 * receivers or senders */
3229 if (!data.is_early && !data.may_suppress)
3230 sess->last_rtcp_send_time = data.current_time;
3231 sess->first_rtcp = FALSE;
3232 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3235 RTP_SESSION_UNLOCK (sess);
3238 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3240 /* push out the RTCP packet */
3242 gboolean do_not_suppress;
3243 GstBuffer *buffer = data.rtcp;
3245 /* Give the user a change to add its own packet */
3246 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3247 buffer, data.is_early, &do_not_suppress);
3249 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3252 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3254 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3255 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3256 sess->stats.avg_rtcp_packet_size, packet_size);
3258 sess->callbacks.send_rtcp (sess, own, buffer, data.is_bye,
3259 sess->send_rtcp_user_data);
3261 GST_DEBUG ("freeing packet callback: %p"
3262 " do_not_suppress: %d may_suppress: %d",
3263 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3264 gst_buffer_unref (buffer);
3272 * rtp_session_request_early_rtcp:
3273 * @sess: an #RTPSession
3274 * @current_time: the current system time
3275 * @max_delay: maximum delay
3277 * Request transmission of early RTCP
3280 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3281 GstClockTimeDiff max_delay)
3283 GstClockTime T_dither_max;
3285 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3287 RTP_SESSION_LOCK (sess);
3289 /* Check if already requested */
3290 /* RFC 4585 section 3.5.2 step 2 */
3291 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3294 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time))
3297 /* Ignore the request a scheduled packet will be in time anyway */
3298 if (current_time + max_delay > sess->next_rtcp_check_time)
3301 /* RFC 4585 section 3.5.2 step 2b */
3302 /* If the total sources is <=2, then there is only us and one peer */
3303 if (sess->total_sources <= 2) {
3306 /* Divide by 2 because l = 0.5 */
3307 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3311 /* RFC 4585 section 3.5.2 step 3 */
3312 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3315 /* RFC 4585 section 3.5.2 step 4
3316 * Don't send if allow_early is FALSE, but not if we are in
3317 * immediate mode, meaning we are part of a group of at most the
3318 * application-specific threshold.
3320 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3321 sess->allow_early == FALSE)
3325 /* Schedule an early transmission later */
3326 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3329 /* If no dithering, schedule it for NOW */
3330 sess->next_early_rtcp_time = current_time;
3333 RTP_SESSION_UNLOCK (sess);
3335 /* notify app of need to send packet early
3336 * and therefore of timeout change */
3337 if (sess->callbacks.reconsider)
3338 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3344 RTP_SESSION_UNLOCK (sess);
3348 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3349 gboolean fir, gint count)
3351 RTPSource *src = find_source (sess, ssrc);
3357 src->send_pli = FALSE;
3358 src->send_fir = TRUE;
3360 if (count == -1 || count != src->last_fir_count)
3361 src->current_send_fir_seqnum++;
3362 src->last_fir_count = count;
3363 } else if (!src->send_fir) {
3364 src->send_pli = TRUE;
3367 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3373 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3375 GstRTCPPacket packet;
3376 GstRTCPBuffer rtcp = { NULL, };
3377 gboolean ret = FALSE;
3379 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3381 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3382 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3383 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3387 gst_rtcp_buffer_unmap (&rtcp);
3393 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3396 gboolean ret = FALSE;
3397 GHashTableIter iter;
3398 gpointer key, value;
3399 gboolean started_fir = FALSE;
3400 GstRTCPPacket fir_rtcppacket;
3401 GstRTCPPacket packet;
3402 GstRTCPBuffer rtcp = { NULL, };
3405 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3407 gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
3408 switch (gst_rtcp_packet_get_type (&packet)) {
3409 case GST_RTCP_TYPE_SR:
3410 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc,
3411 NULL, NULL, NULL, NULL);
3413 case GST_RTCP_TYPE_RR:
3414 ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
3420 RTP_SESSION_LOCK (sess);
3421 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3422 while (g_hash_table_iter_next (&iter, &key, &value)) {
3423 guint media_ssrc = GPOINTER_TO_UINT (key);
3424 RTPSource *media_src = value;
3427 if (media_src->send_fir) {
3429 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3432 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3433 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket, ssrc);
3434 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3436 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3437 gst_rtcp_packet_remove (&fir_rtcppacket);
3443 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3444 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3448 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3449 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3451 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3453 fci_data[0] = media_src->current_send_fir_seqnum;
3454 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3455 media_src->send_fir = FALSE;
3459 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3460 while (g_hash_table_iter_next (&iter, &key, &value)) {
3461 guint media_ssrc = GPOINTER_TO_UINT (key);
3462 RTPSource *media_src = value;
3463 GstRTCPPacket pli_rtcppacket;
3465 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3466 has_pli_compare_func, NULL)) {
3467 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3469 /* Break because the packet is full, will put next request in a
3472 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3473 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket, ssrc);
3474 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3477 media_src->send_pli = FALSE;
3479 RTP_SESSION_UNLOCK (sess);
3482 gst_rtcp_buffer_unmap (&rtcp);
3488 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3492 if (!sess->callbacks.send_rtcp)
3495 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3497 rtp_session_request_early_rtcp (sess, now, max_delay);