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);
641 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
647 rtp_session_get_property (GObject * object, guint prop_id,
648 GValue * value, GParamSpec * pspec)
652 sess = RTP_SESSION (object);
655 case PROP_INTERNAL_SSRC:
656 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
658 case PROP_INTERNAL_SOURCE:
659 g_value_take_object (value, rtp_session_get_internal_source (sess));
662 g_value_set_double (value, sess->bandwidth);
664 case PROP_RTCP_FRACTION:
665 g_value_set_double (value, sess->rtcp_bandwidth);
667 case PROP_RTCP_RR_BANDWIDTH:
668 g_value_set_int (value, sess->rtcp_rr_bandwidth);
670 case PROP_RTCP_RS_BANDWIDTH:
671 g_value_set_int (value, sess->rtcp_rs_bandwidth);
674 g_value_set_uint (value, sess->mtu);
677 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
679 case PROP_NUM_SOURCES:
680 g_value_set_uint (value, rtp_session_get_num_sources (sess));
682 case PROP_NUM_ACTIVE_SOURCES:
683 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
686 g_value_take_boxed (value, rtp_session_create_sources (sess));
689 g_value_set_boolean (value, sess->favor_new);
691 case PROP_RTCP_MIN_INTERVAL:
692 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
694 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
695 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
698 g_value_set_uint (value, sess->probation);
701 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
707 on_new_ssrc (RTPSession * sess, RTPSource * source)
709 g_object_ref (source);
710 RTP_SESSION_UNLOCK (sess);
711 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
712 RTP_SESSION_LOCK (sess);
713 g_object_unref (source);
717 on_ssrc_collision (RTPSession * sess, RTPSource * source)
719 g_object_ref (source);
720 RTP_SESSION_UNLOCK (sess);
721 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
723 RTP_SESSION_LOCK (sess);
724 g_object_unref (source);
728 on_ssrc_validated (RTPSession * sess, RTPSource * source)
730 g_object_ref (source);
731 RTP_SESSION_UNLOCK (sess);
732 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
734 RTP_SESSION_LOCK (sess);
735 g_object_unref (source);
739 on_ssrc_active (RTPSession * sess, RTPSource * source)
741 g_object_ref (source);
742 RTP_SESSION_UNLOCK (sess);
743 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
744 RTP_SESSION_LOCK (sess);
745 g_object_unref (source);
749 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
751 g_object_ref (source);
752 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
753 RTP_SESSION_UNLOCK (sess);
754 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
755 RTP_SESSION_LOCK (sess);
756 g_object_unref (source);
760 on_bye_ssrc (RTPSession * sess, RTPSource * source)
762 g_object_ref (source);
763 RTP_SESSION_UNLOCK (sess);
764 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
765 RTP_SESSION_LOCK (sess);
766 g_object_unref (source);
770 on_bye_timeout (RTPSession * sess, RTPSource * source)
772 g_object_ref (source);
773 RTP_SESSION_UNLOCK (sess);
774 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
775 RTP_SESSION_LOCK (sess);
776 g_object_unref (source);
780 on_timeout (RTPSession * sess, RTPSource * source)
782 g_object_ref (source);
783 RTP_SESSION_UNLOCK (sess);
784 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
785 RTP_SESSION_LOCK (sess);
786 g_object_unref (source);
790 on_sender_timeout (RTPSession * sess, RTPSource * source)
792 g_object_ref (source);
793 RTP_SESSION_UNLOCK (sess);
794 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
796 RTP_SESSION_LOCK (sess);
797 g_object_unref (source);
803 * Create a new session object.
805 * Returns: a new #RTPSession. g_object_unref() after usage.
808 rtp_session_new (void)
812 sess = g_object_new (RTP_TYPE_SESSION, NULL);
818 * rtp_session_set_callbacks:
819 * @sess: an #RTPSession
820 * @callbacks: callbacks to configure
821 * @user_data: user data passed in the callbacks
823 * Configure a set of callbacks to be notified of actions.
826 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
829 g_return_if_fail (RTP_IS_SESSION (sess));
831 if (callbacks->process_rtp) {
832 sess->callbacks.process_rtp = callbacks->process_rtp;
833 sess->process_rtp_user_data = user_data;
835 if (callbacks->send_rtp) {
836 sess->callbacks.send_rtp = callbacks->send_rtp;
837 sess->send_rtp_user_data = user_data;
839 if (callbacks->send_rtcp) {
840 sess->callbacks.send_rtcp = callbacks->send_rtcp;
841 sess->send_rtcp_user_data = user_data;
843 if (callbacks->sync_rtcp) {
844 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
845 sess->sync_rtcp_user_data = user_data;
847 if (callbacks->clock_rate) {
848 sess->callbacks.clock_rate = callbacks->clock_rate;
849 sess->clock_rate_user_data = user_data;
851 if (callbacks->reconsider) {
852 sess->callbacks.reconsider = callbacks->reconsider;
853 sess->reconsider_user_data = user_data;
855 if (callbacks->request_key_unit) {
856 sess->callbacks.request_key_unit = callbacks->request_key_unit;
857 sess->request_key_unit_user_data = user_data;
859 if (callbacks->request_time) {
860 sess->callbacks.request_time = callbacks->request_time;
861 sess->request_time_user_data = user_data;
866 * rtp_session_set_process_rtp_callback:
867 * @sess: an #RTPSession
868 * @callback: callback to set
869 * @user_data: user data passed in the callback
871 * Configure only the process_rtp callback to be notified of the process_rtp action.
874 rtp_session_set_process_rtp_callback (RTPSession * sess,
875 RTPSessionProcessRTP callback, gpointer user_data)
877 g_return_if_fail (RTP_IS_SESSION (sess));
879 sess->callbacks.process_rtp = callback;
880 sess->process_rtp_user_data = user_data;
884 * rtp_session_set_send_rtp_callback:
885 * @sess: an #RTPSession
886 * @callback: callback to set
887 * @user_data: user data passed in the callback
889 * Configure only the send_rtp callback to be notified of the send_rtp action.
892 rtp_session_set_send_rtp_callback (RTPSession * sess,
893 RTPSessionSendRTP callback, gpointer user_data)
895 g_return_if_fail (RTP_IS_SESSION (sess));
897 sess->callbacks.send_rtp = callback;
898 sess->send_rtp_user_data = user_data;
902 * rtp_session_set_send_rtcp_callback:
903 * @sess: an #RTPSession
904 * @callback: callback to set
905 * @user_data: user data passed in the callback
907 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
910 rtp_session_set_send_rtcp_callback (RTPSession * sess,
911 RTPSessionSendRTCP callback, gpointer user_data)
913 g_return_if_fail (RTP_IS_SESSION (sess));
915 sess->callbacks.send_rtcp = callback;
916 sess->send_rtcp_user_data = user_data;
920 * rtp_session_set_sync_rtcp_callback:
921 * @sess: an #RTPSession
922 * @callback: callback to set
923 * @user_data: user data passed in the callback
925 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
928 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
929 RTPSessionSyncRTCP callback, gpointer user_data)
931 g_return_if_fail (RTP_IS_SESSION (sess));
933 sess->callbacks.sync_rtcp = callback;
934 sess->sync_rtcp_user_data = user_data;
938 * rtp_session_set_clock_rate_callback:
939 * @sess: an #RTPSession
940 * @callback: callback to set
941 * @user_data: user data passed in the callback
943 * Configure only the clock_rate callback to be notified of the clock_rate action.
946 rtp_session_set_clock_rate_callback (RTPSession * sess,
947 RTPSessionClockRate callback, gpointer user_data)
949 g_return_if_fail (RTP_IS_SESSION (sess));
951 sess->callbacks.clock_rate = callback;
952 sess->clock_rate_user_data = user_data;
956 * rtp_session_set_reconsider_callback:
957 * @sess: an #RTPSession
958 * @callback: callback to set
959 * @user_data: user data passed in the callback
961 * Configure only the reconsider callback to be notified of the reconsider action.
964 rtp_session_set_reconsider_callback (RTPSession * sess,
965 RTPSessionReconsider callback, gpointer user_data)
967 g_return_if_fail (RTP_IS_SESSION (sess));
969 sess->callbacks.reconsider = callback;
970 sess->reconsider_user_data = user_data;
974 * rtp_session_set_request_time_callback:
975 * @sess: an #RTPSession
976 * @callback: callback to set
977 * @user_data: user data passed in the callback
979 * Configure only the request_time callback
982 rtp_session_set_request_time_callback (RTPSession * sess,
983 RTPSessionRequestTime callback, gpointer user_data)
985 g_return_if_fail (RTP_IS_SESSION (sess));
987 sess->callbacks.request_time = callback;
988 sess->request_time_user_data = user_data;
992 * rtp_session_set_bandwidth:
993 * @sess: an #RTPSession
994 * @bandwidth: the bandwidth allocated
996 * Set the session bandwidth in bytes per second.
999 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1001 g_return_if_fail (RTP_IS_SESSION (sess));
1003 RTP_SESSION_LOCK (sess);
1004 sess->stats.bandwidth = bandwidth;
1005 RTP_SESSION_UNLOCK (sess);
1009 * rtp_session_get_bandwidth:
1010 * @sess: an #RTPSession
1012 * Get the session bandwidth.
1014 * Returns: the session bandwidth.
1017 rtp_session_get_bandwidth (RTPSession * sess)
1021 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1023 RTP_SESSION_LOCK (sess);
1024 result = sess->stats.bandwidth;
1025 RTP_SESSION_UNLOCK (sess);
1031 * rtp_session_set_rtcp_fraction:
1032 * @sess: an #RTPSession
1033 * @bandwidth: the RTCP bandwidth
1035 * Set the bandwidth in bytes per second that should be used for RTCP
1039 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1041 g_return_if_fail (RTP_IS_SESSION (sess));
1043 RTP_SESSION_LOCK (sess);
1044 sess->stats.rtcp_bandwidth = bandwidth;
1045 RTP_SESSION_UNLOCK (sess);
1049 * rtp_session_get_rtcp_fraction:
1050 * @sess: an #RTPSession
1052 * Get the session bandwidth used for RTCP.
1054 * Returns: The bandwidth used for RTCP messages.
1057 rtp_session_get_rtcp_fraction (RTPSession * sess)
1061 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1063 RTP_SESSION_LOCK (sess);
1064 result = sess->stats.rtcp_bandwidth;
1065 RTP_SESSION_UNLOCK (sess);
1071 * rtp_session_get_sdes_struct:
1072 * @sess: an #RTSPSession
1074 * Get the SDES data as a #GstStructure
1076 * Returns: a GstStructure with SDES items for @sess. This function returns a
1077 * copy of the SDES structure, use gst_structure_free() after usage.
1080 rtp_session_get_sdes_struct (RTPSession * sess)
1082 GstStructure *result = NULL;
1084 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1086 RTP_SESSION_LOCK (sess);
1088 result = gst_structure_copy (sess->sdes);
1089 RTP_SESSION_UNLOCK (sess);
1095 * rtp_session_set_sdes_struct:
1096 * @sess: an #RTSPSession
1097 * @sdes: a #GstStructure
1099 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1102 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1104 g_return_if_fail (sdes);
1105 g_return_if_fail (RTP_IS_SESSION (sess));
1107 RTP_SESSION_LOCK (sess);
1109 gst_structure_free (sess->sdes);
1110 sess->sdes = gst_structure_copy (sdes);
1111 RTP_SESSION_UNLOCK (sess);
1114 static GstFlowReturn
1115 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1117 GstFlowReturn result = GST_FLOW_OK;
1119 if (source->internal) {
1120 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1122 RTP_SESSION_UNLOCK (session);
1124 if (session->callbacks.send_rtp)
1126 session->callbacks.send_rtp (session, source, data,
1127 session->send_rtp_user_data);
1129 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1132 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1133 RTP_SESSION_UNLOCK (session);
1135 if (session->callbacks.process_rtp)
1137 session->callbacks.process_rtp (session, source,
1138 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1140 gst_buffer_unref (GST_BUFFER_CAST (data));
1142 RTP_SESSION_LOCK (session);
1148 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1152 RTP_SESSION_UNLOCK (session);
1154 if (session->callbacks.clock_rate)
1156 session->callbacks.clock_rate (session, pt,
1157 session->clock_rate_user_data);
1161 RTP_SESSION_LOCK (session);
1163 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1168 static RTPSourceCallbacks callbacks = {
1169 (RTPSourcePushRTP) source_push_rtp,
1170 (RTPSourceClockRate) source_clock_rate,
1174 check_collision (RTPSession * sess, RTPSource * source,
1175 RTPArrivalStats * arrival, gboolean rtp)
1177 /* If we have no arrival address, we can't do collision checking */
1178 if (!arrival->address)
1181 if (!source->internal) {
1182 GSocketAddress *from;
1184 /* This is not our local source, but lets check if two remote
1187 from = source->rtp_from;
1189 from = source->rtcp_from;
1193 if (__g_socket_address_equal (from, arrival->address)) {
1194 /* Address is the same */
1197 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1198 rtp_source_get_ssrc (source));
1199 if (sess->favor_new) {
1200 if (rtp_source_find_conflicting_address (source,
1201 arrival->address, arrival->current_time)) {
1204 buf1 = __g_socket_address_to_string (arrival->address);
1205 GST_LOG ("Known conflict on %x for %s, dropping packet",
1206 rtp_source_get_ssrc (source), buf1);
1213 /* Current address is not a known conflict, lets assume this is
1214 * a new source. Save old address in possible conflict list
1216 rtp_source_add_conflicting_address (source, from,
1217 arrival->current_time);
1219 buf1 = __g_socket_address_to_string (from);
1220 buf2 = __g_socket_address_to_string (arrival->address);
1222 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1223 " saving old as known conflict",
1224 rtp_source_get_ssrc (source), buf1, buf2);
1227 rtp_source_set_rtp_from (source, arrival->address);
1229 rtp_source_set_rtcp_from (source, arrival->address);
1237 /* Don't need to save old addresses, we ignore new sources */
1242 /* We don't already have a from address for RTP, just set it */
1244 rtp_source_set_rtp_from (source, arrival->address);
1246 rtp_source_set_rtcp_from (source, arrival->address);
1250 /* FIXME: Log 3rd party collision somehow
1251 * Maybe should be done in upper layer, only the SDES can tell us
1252 * if its a collision or a loop
1255 /* This is sending with our ssrc, is it an address we already know */
1257 if (rtp_source_find_conflicting_address (source, arrival->address,
1258 arrival->current_time)) {
1259 /* Its a known conflict, its probably a loop, not a collision
1260 * lets just drop the incoming packet
1262 GST_DEBUG ("Our packets are being looped back to us, dropping");
1264 /* Its a new collision, lets change our SSRC */
1266 rtp_source_add_conflicting_address (source, arrival->address,
1267 arrival->current_time);
1269 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1270 on_ssrc_collision (sess, source);
1272 sess->change_ssrc = TRUE;
1274 rtp_source_mark_bye (source, "SSRC Collision");
1275 rtp_session_schedule_bye_locked (sess, arrival->current_time);
1283 find_source (RTPSession * sess, guint32 ssrc)
1285 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1286 GINT_TO_POINTER (ssrc));
1290 add_source (RTPSession * sess, RTPSource * src)
1292 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1293 GINT_TO_POINTER (src->ssrc), src);
1294 /* we have one more source now */
1295 sess->total_sources++;
1296 if (RTP_SOURCE_IS_ACTIVE (src))
1297 sess->stats.active_sources++;
1298 if (src->internal) {
1299 sess->stats.internal_sources++;
1300 if (sess->suggested_ssrc != src->ssrc)
1301 sess->suggested_ssrc = src->ssrc;
1305 /* must be called with the session lock, the returned source needs to be
1306 * unreffed after usage. */
1308 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1309 RTPArrivalStats * arrival, gboolean rtp)
1313 source = find_source (sess, ssrc);
1314 if (source == NULL) {
1315 /* make new Source in probation and insert */
1316 source = rtp_source_new (ssrc);
1318 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1320 /* for RTP packets we need to set the source in probation. Receiving RTCP
1321 * packets of an SSRC, on the other hand, is a strong indication that we
1322 * are dealing with a valid source. */
1324 g_object_set (source, "probation", sess->probation, NULL);
1326 g_object_set (source, "probation", 0, NULL);
1328 /* store from address, if any */
1329 if (arrival->address) {
1331 rtp_source_set_rtp_from (source, arrival->address);
1333 rtp_source_set_rtcp_from (source, arrival->address);
1336 /* configure a callback on the source */
1337 rtp_source_set_callbacks (source, &callbacks, sess);
1339 add_source (sess, source);
1343 /* check for collision, this updates the address when not previously set */
1344 if (check_collision (sess, source, arrival, rtp)) {
1347 /* Receiving RTCP packets of an SSRC is a strong indication that we
1348 * are dealing with a valid source. */
1350 g_object_set (source, "probation", 0, NULL);
1352 /* update last activity */
1353 source->last_activity = arrival->current_time;
1355 source->last_rtp_activity = arrival->current_time;
1356 g_object_ref (source);
1361 /* must be called with the session lock, the returned source needs to be
1362 * unreffed after usage. */
1364 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1368 source = find_source (sess, ssrc);
1369 if (source == NULL) {
1370 /* make new internal Source and insert */
1371 source = rtp_source_new (ssrc);
1373 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1375 source->validated = TRUE;
1376 source->internal = TRUE;
1377 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1378 rtp_source_set_callbacks (source, &callbacks, sess);
1380 add_source (sess, source);
1385 g_object_ref (source);
1391 * rtp_session_get_internal_source:
1392 * @sess: a #RTPSession
1394 * Get the internal #RTPSource of @sess.
1396 * Returns: The internal #RTPSource. g_object_unref() after usage.
1399 rtp_session_get_internal_source (RTPSession * sess)
1403 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1405 result = g_object_ref (sess->source);
1411 * rtp_session_set_internal_ssrc:
1412 * @sess: a #RTPSession
1415 * Set the SSRC of @sess to @ssrc.
1418 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1420 RTP_SESSION_LOCK (sess);
1421 if (ssrc != sess->source->ssrc) {
1422 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1423 GINT_TO_POINTER (sess->source->ssrc));
1425 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1426 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1427 * packets will timeout on the old SSRC, we could potentially schedule a
1428 * BYE RTCP for the old SSRC... */
1429 sess->source->ssrc = ssrc;
1430 rtp_source_reset (sess->source);
1432 /* rehash with the new SSRC */
1433 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1434 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1436 RTP_SESSION_UNLOCK (sess);
1438 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1442 * rtp_session_get_internal_ssrc:
1443 * @sess: a #RTPSession
1445 * Get the internal SSRC of @sess.
1447 * Returns: The SSRC of the session.
1450 rtp_session_get_internal_ssrc (RTPSession * sess)
1454 RTP_SESSION_LOCK (sess);
1455 ssrc = sess->source->ssrc;
1456 RTP_SESSION_UNLOCK (sess);
1462 * rtp_session_suggest_ssrc:
1463 * @sess: a #RTPSession
1465 * Suggest an unused SSRC in @sess.
1467 * Returns: a free unused SSRC
1470 rtp_session_suggest_ssrc (RTPSession * sess)
1474 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1476 RTP_SESSION_LOCK (sess);
1477 result = sess->suggested_ssrc;
1478 RTP_SESSION_UNLOCK (sess);
1484 * rtp_session_add_source:
1485 * @sess: a #RTPSession
1486 * @src: #RTPSource to add
1488 * Add @src to @session.
1490 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1491 * existed in the session.
1494 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1496 gboolean result = FALSE;
1499 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1500 g_return_val_if_fail (src != NULL, FALSE);
1502 RTP_SESSION_LOCK (sess);
1503 find = find_source (sess, src->ssrc);
1505 add_source (sess, src);
1508 RTP_SESSION_UNLOCK (sess);
1514 * rtp_session_get_num_sources:
1515 * @sess: an #RTPSession
1517 * Get the number of sources in @sess.
1519 * Returns: The number of sources in @sess.
1522 rtp_session_get_num_sources (RTPSession * sess)
1526 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1528 RTP_SESSION_LOCK (sess);
1529 result = sess->total_sources;
1530 RTP_SESSION_UNLOCK (sess);
1536 * rtp_session_get_num_active_sources:
1537 * @sess: an #RTPSession
1539 * Get the number of active sources in @sess. A source is considered active when
1540 * it has been validated and has not yet received a BYE RTCP message.
1542 * Returns: The number of active sources in @sess.
1545 rtp_session_get_num_active_sources (RTPSession * sess)
1549 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1551 RTP_SESSION_LOCK (sess);
1552 result = sess->stats.active_sources;
1553 RTP_SESSION_UNLOCK (sess);
1559 * rtp_session_get_source_by_ssrc:
1560 * @sess: an #RTPSession
1563 * Find the source with @ssrc in @sess.
1565 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1566 * g_object_unref() after usage.
1569 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1573 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1575 RTP_SESSION_LOCK (sess);
1576 result = find_source (sess, ssrc);
1578 g_object_ref (result);
1579 RTP_SESSION_UNLOCK (sess);
1584 /* should be called with the SESSION lock */
1586 rtp_session_create_new_ssrc (RTPSession * sess)
1591 ssrc = g_random_int ();
1593 /* see if it exists in the session, we're done if it doesn't */
1594 if (find_source (sess, ssrc) == NULL)
1602 * rtp_session_create_source:
1603 * @sess: an #RTPSession
1605 * Create an #RTPSource for use in @sess. This function will create a source
1606 * with an ssrc that is currently not used by any participants in the session.
1608 * Returns: an #RTPSource.
1611 rtp_session_create_source (RTPSession * sess)
1616 RTP_SESSION_LOCK (sess);
1617 ssrc = rtp_session_create_new_ssrc (sess);
1618 source = rtp_source_new (ssrc);
1619 rtp_source_set_callbacks (source, &callbacks, sess);
1620 /* we need an additional ref for the source in the hashtable */
1621 g_object_ref (source);
1622 add_source (sess, source);
1623 RTP_SESSION_UNLOCK (sess);
1628 /* update the RTPArrivalStats structure with the current time and other bits
1629 * about the current buffer we are handling.
1630 * This function is typically called when a validated packet is received.
1631 * This function should be called with the SESSION_LOCK
1634 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1635 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1636 GstClockTime running_time, guint64 ntpnstime)
1638 GstNetAddressMeta *meta;
1639 GstRTPBuffer rtpb = { NULL };
1641 /* get time of arrival */
1642 arrival->current_time = current_time;
1643 arrival->running_time = running_time;
1644 arrival->ntpnstime = ntpnstime;
1646 /* get packet size including header overhead */
1647 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1650 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1651 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1652 gst_rtp_buffer_unmap (&rtpb);
1654 arrival->payload_len = 0;
1657 /* for netbuffer we can store the IP address to check for collisions */
1658 meta = gst_buffer_get_net_address_meta (buffer);
1659 if (arrival->address)
1660 g_object_unref (arrival->address);
1662 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1664 arrival->address = NULL;
1669 clean_arrival_stats (RTPArrivalStats * arrival)
1671 if (arrival->address)
1672 g_object_unref (arrival->address);
1676 * rtp_session_process_rtp:
1677 * @sess: and #RTPSession
1678 * @buffer: an RTP buffer
1679 * @current_time: the current system time
1680 * @running_time: the running_time of @buffer
1682 * Process an RTP buffer in the session manager. This function takes ownership
1685 * Returns: a #GstFlowReturn.
1688 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1689 GstClockTime current_time, GstClockTime running_time)
1691 GstFlowReturn result;
1695 gboolean prevsender, prevactive;
1696 RTPArrivalStats arrival = { NULL, };
1700 GstRTPBuffer rtp = { NULL };
1702 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1703 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1705 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1706 goto invalid_packet;
1708 /* get SSRC to look up in session database */
1709 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1710 /* copy available csrc for later */
1711 count = gst_rtp_buffer_get_csrc_count (&rtp);
1712 /* make sure to not overflow our array. An RTP buffer can maximally contain
1714 count = MIN (count, 16);
1716 for (i = 0; i < count; i++)
1717 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1719 gst_rtp_buffer_unmap (&rtp);
1721 RTP_SESSION_LOCK (sess);
1722 /* ignore more RTP packets when we left the session */
1723 if (sess->source->marked_bye)
1726 /* update arrival stats */
1727 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1730 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1734 prevsender = RTP_SOURCE_IS_SENDER (source);
1735 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1736 oldrate = source->bitrate;
1738 /* let source process the packet */
1739 result = rtp_source_process_rtp (source, buffer, &arrival);
1741 /* source became active */
1742 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1743 sess->stats.active_sources++;
1744 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1745 sess->stats.active_sources);
1746 on_ssrc_validated (sess, source);
1748 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1749 sess->stats.sender_sources++;
1750 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1751 sess->stats.sender_sources);
1753 if (oldrate != source->bitrate)
1754 sess->recalc_bandwidth = TRUE;
1757 on_new_ssrc (sess, source);
1759 if (source->validated) {
1762 /* for validated sources, we add the CSRCs as well */
1763 for (i = 0; i < count; i++) {
1765 RTPSource *csrc_src;
1770 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1775 GST_DEBUG ("created new CSRC: %08x", csrc);
1776 rtp_source_set_as_csrc (csrc_src);
1777 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1778 sess->stats.active_sources++;
1779 on_new_ssrc (sess, csrc_src);
1781 g_object_unref (csrc_src);
1784 g_object_unref (source);
1786 RTP_SESSION_UNLOCK (sess);
1788 clean_arrival_stats (&arrival);
1795 gst_buffer_unref (buffer);
1796 GST_DEBUG ("invalid RTP packet received");
1801 RTP_SESSION_UNLOCK (sess);
1802 gst_buffer_unref (buffer);
1803 GST_DEBUG ("ignoring RTP packet because we are leaving");
1808 RTP_SESSION_UNLOCK (sess);
1809 gst_buffer_unref (buffer);
1810 clean_arrival_stats (&arrival);
1811 GST_DEBUG ("ignoring packet because its collisioning");
1817 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1818 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1822 count = gst_rtcp_packet_get_rb_count (packet);
1823 for (i = 0; i < count; i++) {
1824 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1825 guint8 fractionlost;
1828 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1829 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1831 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1833 if (ssrc == sess->source->ssrc) {
1834 /* only deal with report blocks for our session, we update the stats of
1835 * the sender of the RTCP message. We could also compare our stats against
1836 * the other sender to see if we are better or worse. */
1837 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1838 packetslost, exthighestseq, jitter, lsr, dlsr);
1841 on_ssrc_active (sess, source);
1844 /* A Sender report contains statistics about how the sender is doing. This
1845 * includes timing informataion such as the relation between RTP and NTP
1846 * timestamps and the number of packets/bytes it sent to us.
1848 * In this report is also included a set of report blocks related to how this
1849 * sender is receiving data (in case we (or somebody else) is also sending stuff
1850 * to it). This info includes the packet loss, jitter and seqnum. It also
1851 * contains information to calculate the round trip time (LSR/DLSR).
1854 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1855 RTPArrivalStats * arrival, gboolean * do_sync)
1857 guint32 senderssrc, rtptime, packet_count, octet_count;
1860 gboolean created, prevsender;
1862 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1863 &packet_count, &octet_count);
1865 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1866 senderssrc, GST_TIME_ARGS (arrival->current_time));
1868 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1872 /* don't try to do lip-sync for sources that sent a BYE */
1873 if (RTP_SOURCE_IS_MARKED_BYE (source))
1878 prevsender = RTP_SOURCE_IS_SENDER (source);
1880 /* first update the source */
1881 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1882 packet_count, octet_count);
1884 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1885 sess->stats.sender_sources++;
1886 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1887 sess->stats.sender_sources);
1891 on_new_ssrc (sess, source);
1893 rtp_session_process_rb (sess, source, packet, arrival);
1894 g_object_unref (source);
1897 /* A receiver report contains statistics about how a receiver is doing. It
1898 * includes stuff like packet loss, jitter and the seqnum it received last. It
1899 * also contains info to calculate the round trip time.
1901 * We are only interested in how the sender of this report is doing wrt to us.
1904 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1905 RTPArrivalStats * arrival)
1911 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1913 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1915 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1920 on_new_ssrc (sess, source);
1922 rtp_session_process_rb (sess, source, packet, arrival);
1923 g_object_unref (source);
1926 /* Get SDES items and store them in the SSRC */
1928 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1929 RTPArrivalStats * arrival)
1932 gboolean more_items, more_entries;
1934 items = gst_rtcp_packet_sdes_get_item_count (packet);
1935 GST_DEBUG ("got SDES packet with %d items", items);
1937 more_items = gst_rtcp_packet_sdes_first_item (packet);
1939 while (more_items) {
1941 gboolean changed, created, validated;
1945 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1947 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1951 /* find src, no probation when dealing with RTCP */
1952 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1956 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1958 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1960 while (more_entries) {
1961 GstRTCPSDESType type;
1967 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1969 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1972 if (type == GST_RTCP_SDES_PRIV) {
1973 name = g_strndup ((const gchar *) &data[1], data[0]);
1975 data += data[0] + 1;
1977 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1980 value = g_strndup ((const gchar *) data, len);
1982 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1987 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1991 /* takes ownership of sdes */
1992 changed = rtp_source_set_sdes_struct (source, sdes);
1994 validated = !RTP_SOURCE_IS_ACTIVE (source);
1995 source->validated = TRUE;
1998 on_new_ssrc (sess, source);
2000 /* source became active */
2002 sess->stats.active_sources++;
2003 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
2004 sess->stats.active_sources);
2005 on_ssrc_validated (sess, source);
2009 on_ssrc_sdes (sess, source);
2011 g_object_unref (source);
2013 more_items = gst_rtcp_packet_sdes_next_item (packet);
2018 /* BYE is sent when a client leaves the session
2021 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2022 RTPArrivalStats * arrival)
2026 gboolean reconsider = FALSE;
2028 reason = gst_rtcp_packet_bye_get_reason (packet);
2029 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2031 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2032 for (i = 0; i < count; i++) {
2035 gboolean created, prevactive, prevsender;
2036 guint pmembers, members;
2038 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2039 GST_DEBUG ("SSRC: %08x", ssrc);
2041 /* find src and mark bye, no probation when dealing with RTCP */
2042 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
2046 if (source->internal) {
2047 /* our own source, something weird with this packet */
2048 g_object_unref (source);
2052 /* store time for when we need to time out this source */
2053 source->bye_time = arrival->current_time;
2055 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2056 prevsender = RTP_SOURCE_IS_SENDER (source);
2058 /* mark the source BYE */
2059 rtp_source_mark_bye (source, reason);
2061 pmembers = sess->stats.active_sources;
2063 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2064 sess->stats.active_sources--;
2065 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2066 sess->stats.active_sources);
2068 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2069 sess->stats.sender_sources--;
2070 if (source->internal)
2071 sess->stats.internal_sender_sources--;
2072 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2073 sess->stats.sender_sources);
2075 members = sess->stats.active_sources;
2077 if (!sess->scheduled_bye && members < pmembers) {
2078 /* some members went away since the previous timeout estimate.
2079 * Perform reverse reconsideration but only when we are not scheduling a
2081 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2082 arrival->current_time < sess->next_rtcp_check_time) {
2083 GstClockTime time_remaining;
2085 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2086 sess->next_rtcp_check_time =
2087 gst_util_uint64_scale (time_remaining, members, pmembers);
2089 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2090 GST_TIME_ARGS (sess->next_rtcp_check_time));
2092 sess->next_rtcp_check_time += arrival->current_time;
2094 /* mark pending reconsider. We only want to signal the reconsideration
2095 * once after we handled all the source in the bye packet */
2101 on_new_ssrc (sess, source);
2103 on_bye_ssrc (sess, source);
2105 g_object_unref (source);
2108 RTP_SESSION_UNLOCK (sess);
2109 /* notify app of reconsideration */
2110 if (sess->callbacks.reconsider)
2111 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2112 RTP_SESSION_LOCK (sess);
2118 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2119 RTPArrivalStats * arrival)
2121 GST_DEBUG ("received APP");
2125 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2126 gboolean fir, GstClockTime current_time)
2128 guint32 round_trip = 0;
2130 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2132 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2133 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2136 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2137 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2138 GST_DEBUG ("Ignoring %s request because one was send without one "
2139 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2140 fir ? "FIR" : "PLI",
2141 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2142 GST_TIME_ARGS (round_trip_in_ns));;
2147 sess->last_keyframe_request = current_time;
2149 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2150 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2151 sess->callbacks.request_key_unit);
2153 RTP_SESSION_UNLOCK (sess);
2154 sess->callbacks.request_key_unit (sess, fir,
2155 sess->request_key_unit_user_data);
2156 RTP_SESSION_LOCK (sess);
2162 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2163 guint32 media_ssrc, GstClockTime current_time)
2167 if (!sess->callbacks.request_key_unit)
2170 src = find_source (sess, sender_ssrc);
2174 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2178 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2179 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2184 gboolean our_request = FALSE;
2186 if (!sess->callbacks.request_key_unit)
2192 src = find_source (sess, sender_ssrc);
2194 /* Hack because Google fails to set the sender_ssrc correctly */
2195 if (!src && sender_ssrc == 1) {
2196 GHashTableIter iter;
2198 if (sess->stats.sender_sources >
2199 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2202 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2204 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2205 if (src != sess->source && rtp_source_is_sender (src))
2214 for (position = 0; position < fci_length; position += 8) {
2215 guint8 *data = fci_data + position;
2218 ssrc = GST_READ_UINT32_BE (data);
2220 own = find_source (sess, ssrc);
2221 if (own->internal) {
2229 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2233 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2234 RTPArrivalStats * arrival, GstClockTime current_time)
2236 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2237 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2238 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2239 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2240 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2241 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2244 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2245 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2247 if (g_signal_has_handler_pending (sess,
2248 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2249 GstBuffer *fci_buffer = NULL;
2251 if (fci_length > 0) {
2252 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2253 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2255 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2258 RTP_SESSION_UNLOCK (sess);
2259 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2260 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2261 RTP_SESSION_LOCK (sess);
2264 gst_buffer_unref (fci_buffer);
2267 src = find_source (sess, media_ssrc);
2271 if (sess->rtcp_feedback_retention_window) {
2272 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2275 if (src->internal ||
2276 /* PSFB FIR puts the media ssrc inside the FCI */
2277 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2279 case GST_RTCP_TYPE_PSFB:
2281 case GST_RTCP_PSFB_TYPE_PLI:
2282 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2285 case GST_RTCP_PSFB_TYPE_FIR:
2286 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2293 case GST_RTCP_TYPE_RTPFB:
2301 * rtp_session_process_rtcp:
2302 * @sess: and #RTPSession
2303 * @buffer: an RTCP buffer
2304 * @current_time: the current system time
2305 * @ntpnstime: the current NTP time in nanoseconds
2307 * Process an RTCP buffer in the session manager. This function takes ownership
2310 * Returns: a #GstFlowReturn.
2313 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2314 GstClockTime current_time, guint64 ntpnstime)
2316 GstRTCPPacket packet;
2317 gboolean more, is_bye = FALSE, do_sync = FALSE;
2318 RTPArrivalStats arrival = { NULL, };
2319 GstFlowReturn result = GST_FLOW_OK;
2320 GstRTCPBuffer rtcp = { NULL, };
2322 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2323 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2325 if (!gst_rtcp_buffer_validate (buffer))
2326 goto invalid_packet;
2328 GST_DEBUG ("received RTCP packet");
2330 RTP_SESSION_LOCK (sess);
2331 /* update arrival stats */
2332 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2335 if (sess->source->sent_bye)
2338 /* start processing the compound packet */
2339 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2340 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2344 type = gst_rtcp_packet_get_type (&packet);
2346 /* when we are leaving the session, we should ignore all non-BYE messages */
2347 if (sess->scheduled_bye && type != GST_RTCP_TYPE_BYE) {
2348 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2353 case GST_RTCP_TYPE_SR:
2354 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2356 case GST_RTCP_TYPE_RR:
2357 rtp_session_process_rr (sess, &packet, &arrival);
2359 case GST_RTCP_TYPE_SDES:
2360 rtp_session_process_sdes (sess, &packet, &arrival);
2362 case GST_RTCP_TYPE_BYE:
2364 /* don't try to attempt lip-sync anymore for streams with a BYE */
2366 rtp_session_process_bye (sess, &packet, &arrival);
2368 case GST_RTCP_TYPE_APP:
2369 rtp_session_process_app (sess, &packet, &arrival);
2371 case GST_RTCP_TYPE_RTPFB:
2372 case GST_RTCP_TYPE_PSFB:
2373 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2376 GST_WARNING ("got unknown RTCP packet");
2380 more = gst_rtcp_packet_move_to_next (&packet);
2383 gst_rtcp_buffer_unmap (&rtcp);
2385 /* if we are scheduling a BYE, we only want to count bye packets, else we
2386 * count everything */
2387 if (sess->scheduled_bye) {
2389 sess->stats.bye_members++;
2390 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2393 /* keep track of average packet size */
2394 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2396 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2397 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2398 RTP_SESSION_UNLOCK (sess);
2400 clean_arrival_stats (&arrival);
2402 /* notify caller of sr packets in the callback */
2403 if (do_sync && sess->callbacks.sync_rtcp) {
2404 /* make writable, we might want to change the buffer */
2405 buffer = gst_buffer_make_writable (buffer);
2407 result = sess->callbacks.sync_rtcp (sess, buffer,
2408 sess->sync_rtcp_user_data);
2410 gst_buffer_unref (buffer);
2417 GST_DEBUG ("invalid RTCP packet received");
2418 gst_buffer_unref (buffer);
2423 RTP_SESSION_UNLOCK (sess);
2424 gst_buffer_unref (buffer);
2425 clean_arrival_stats (&arrival);
2426 GST_DEBUG ("ignoring RTCP packet because we left");
2432 * rtp_session_update_send_caps:
2433 * @sess: an #RTPSession
2436 * Update the caps of the sender in the rtp session.
2439 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2444 g_return_if_fail (RTP_IS_SESSION (sess));
2445 g_return_if_fail (GST_IS_CAPS (caps));
2447 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2449 s = gst_caps_get_structure (caps, 0);
2451 if (gst_structure_get_uint (s, "ssrc", &ssrc))
2452 rtp_session_set_internal_ssrc (sess, ssrc);
2454 RTP_SESSION_LOCK (sess);
2455 rtp_source_update_caps (sess->source, caps);
2456 RTP_SESSION_UNLOCK (sess);
2460 * rtp_session_send_rtp:
2461 * @sess: an #RTPSession
2462 * @data: pointer to either an RTP buffer or a list of RTP buffers
2463 * @is_list: TRUE when @data is a buffer list
2464 * @current_time: the current system time
2465 * @running_time: the running time of @data
2467 * Send the RTP buffer in the session manager. This function takes ownership of
2470 * Returns: a #GstFlowReturn.
2473 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2474 GstClockTime current_time, GstClockTime running_time)
2476 GstFlowReturn result;
2478 gboolean prevsender;
2481 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2482 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2484 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2486 RTP_SESSION_LOCK (sess);
2487 source = sess->source;
2489 /* update last activity */
2490 source->last_rtp_activity = current_time;
2492 prevsender = RTP_SOURCE_IS_SENDER (source);
2493 oldrate = source->bitrate;
2495 /* we use our own source to send */
2496 result = rtp_source_send_rtp (source, data, is_list, running_time);
2498 if (RTP_SOURCE_IS_SENDER (source) && !prevsender) {
2499 sess->stats.sender_sources++;
2500 sess->stats.internal_sender_sources++;
2502 if (oldrate != source->bitrate)
2503 sess->recalc_bandwidth = TRUE;
2504 RTP_SESSION_UNLOCK (sess);
2510 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2512 *bandwidth += source->bitrate;
2515 /* must be called with session lock */
2517 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2520 GstClockTime result;
2522 /* recalculate bandwidth when it changed */
2523 if (sess->recalc_bandwidth) {
2526 if (sess->bandwidth > 0)
2527 bandwidth = sess->bandwidth;
2529 /* If it is <= 0, then try to estimate the actual bandwidth */
2532 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2533 (GHFunc) add_bitrates, &bandwidth);
2536 if (bandwidth < 8000)
2537 bandwidth = RTP_STATS_BANDWIDTH;
2539 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2540 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2542 sess->recalc_bandwidth = FALSE;
2545 if (sess->scheduled_bye) {
2546 result = rtp_stats_calculate_bye_interval (&sess->stats);
2548 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2549 sess->stats.internal_sender_sources > 0, first);
2552 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2553 GST_TIME_ARGS (result), first);
2555 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2556 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2558 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2564 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2566 if (source->internal)
2567 rtp_source_mark_bye (source, reason);
2571 * rtp_session_mark_all_bye:
2572 * @sess: an #RTPSession
2575 * Mark all internal sources of the session as BYE with @reason.
2578 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2580 g_return_if_fail (RTP_IS_SESSION (sess));
2582 RTP_SESSION_LOCK (sess);
2583 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2584 (GHFunc) source_mark_bye, (gpointer) reason);
2585 RTP_SESSION_UNLOCK (sess);
2588 /* Stop the current @sess and schedule a BYE message for the other members.
2589 * One must have the session lock to call this function
2591 static GstFlowReturn
2592 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2594 GstFlowReturn result = GST_FLOW_OK;
2595 GstClockTime interval;
2597 /* nothing to do it we already scheduled bye */
2598 if (sess->scheduled_bye)
2601 /* we schedule BYE now */
2602 sess->scheduled_bye = TRUE;
2603 /* at least one member wants to send a BYE */
2604 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2605 sess->stats.bye_members = 1;
2606 sess->first_rtcp = TRUE;
2607 sess->allow_early = TRUE;
2609 /* reschedule transmission */
2610 sess->last_rtcp_send_time = current_time;
2611 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2613 if (interval != GST_CLOCK_TIME_NONE)
2614 sess->next_rtcp_check_time = current_time + interval;
2616 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2618 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2619 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2621 RTP_SESSION_UNLOCK (sess);
2622 /* notify app of reconsideration */
2623 if (sess->callbacks.reconsider)
2624 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2625 RTP_SESSION_LOCK (sess);
2632 * rtp_session_schedule_bye:
2633 * @sess: an #RTPSession
2634 * @current_time: the current system time
2636 * Schedule a BYE message for all sources marked as BYE in @sess.
2638 * Returns: a #GstFlowReturn.
2641 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2643 GstFlowReturn result = GST_FLOW_OK;
2645 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2647 RTP_SESSION_LOCK (sess);
2648 result = rtp_session_schedule_bye_locked (sess, current_time);
2649 RTP_SESSION_UNLOCK (sess);
2655 * rtp_session_next_timeout:
2656 * @sess: an #RTPSession
2657 * @current_time: the current system time
2659 * Get the next time we should perform session maintenance tasks.
2661 * Returns: a time when rtp_session_on_timeout() should be called with the
2662 * current system time.
2665 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2667 GstClockTime result, interval = 0;
2669 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2671 RTP_SESSION_LOCK (sess);
2673 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2674 result = sess->next_early_rtcp_time;
2678 result = sess->next_rtcp_check_time;
2680 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2681 ", next time: %" GST_TIME_FORMAT,
2682 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2684 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2685 GST_DEBUG ("take current time as base");
2686 /* our previous check time expired, start counting from the current time
2688 result = current_time;
2691 if (sess->scheduled_bye) {
2692 if (sess->source->sent_bye) {
2693 GST_DEBUG ("we sent BYE already");
2694 interval = GST_CLOCK_TIME_NONE;
2695 } else if (sess->stats.active_sources >= 50) {
2696 GST_DEBUG ("reconsider BYE, more than 50 sources");
2697 /* reconsider BYE if members >= 50 */
2698 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2701 if (sess->first_rtcp) {
2702 GST_DEBUG ("first RTCP packet");
2703 /* we are called for the first time */
2704 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2705 } else if (sess->next_rtcp_check_time < current_time) {
2706 GST_DEBUG ("old check time expired, getting new timeout");
2707 /* get a new timeout when we need to */
2708 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2712 if (interval != GST_CLOCK_TIME_NONE)
2715 result = GST_CLOCK_TIME_NONE;
2717 sess->next_rtcp_check_time = result;
2721 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2722 ", next time: %" GST_TIME_FORMAT,
2723 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2724 RTP_SESSION_UNLOCK (sess);
2731 GstRTCPBuffer rtcpbuf;
2735 GstClockTime current_time;
2737 GstClockTime running_time;
2738 GstClockTime interval;
2739 GstRTCPPacket packet;
2743 gboolean may_suppress;
2748 session_start_rtcp (RTPSession * sess, ReportData * data)
2750 GstRTCPPacket *packet = &data->packet;
2751 RTPSource *own = data->source;
2752 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2754 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2755 data->is_bye = FALSE;
2756 data->has_sdes = FALSE;
2758 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2760 if (RTP_SOURCE_IS_SENDER (own)) {
2763 guint32 packet_count, octet_count;
2765 /* we are a sender, create SR */
2766 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2767 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2769 /* get latest stats */
2770 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2771 &ntptime, &rtptime, &packet_count, &octet_count);
2773 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2774 packet_count, octet_count);
2776 /* fill in sender report info */
2777 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2778 ntptime, rtptime, packet_count, octet_count);
2780 /* we are only receiver, create RR */
2781 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2782 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2783 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2787 /* construct a Sender or Receiver Report */
2789 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2791 GstRTCPPacket *packet = &data->packet;
2793 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2794 /* only report about other sender sources */
2795 if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
2796 guint8 fractionlost;
2798 guint32 exthighestseq, jitter;
2802 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2803 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2805 /* store last generated RR packet */
2806 source->last_rr.is_valid = TRUE;
2807 source->last_rr.fractionlost = fractionlost;
2808 source->last_rr.packetslost = packetslost;
2809 source->last_rr.exthighestseq = exthighestseq;
2810 source->last_rr.jitter = jitter;
2811 source->last_rr.lsr = lsr;
2812 source->last_rr.dlsr = dlsr;
2814 /* packet is not yet filled, add report block for this source. */
2815 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2816 exthighestseq, jitter, lsr, dlsr);
2821 /* perform cleanup of sources that timed out */
2823 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2825 gboolean remove = FALSE;
2826 gboolean byetimeout = FALSE;
2827 gboolean sendertimeout = FALSE;
2828 gboolean is_sender, is_active;
2829 RTPSession *sess = data->sess;
2830 GstClockTime interval, binterval;
2833 /* check for outdated collisions */
2834 if (source->internal) {
2835 GST_DEBUG ("Timing out collisions");
2836 rtp_source_timeout (source, data->current_time,
2837 /* "a relatively long time" -- RFC 3550 section 8.2 */
2838 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
2839 data->running_time - sess->rtcp_feedback_retention_window);
2842 /* nothing else to do when without RTCP */
2843 if (data->interval == GST_CLOCK_TIME_NONE)
2846 is_sender = RTP_SOURCE_IS_SENDER (source);
2847 is_active = RTP_SOURCE_IS_ACTIVE (source);
2849 /* our own rtcp interval may have been forced low by secondary configuration,
2850 * while sender side may still operate with higher interval,
2851 * so do not just take our interval to decide on timing out sender,
2852 * but take (if data->interval <= 5 * GST_SECOND):
2853 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2854 * where sender_interval is difference between last 2 received RTCP reports
2856 if (data->interval >= 5 * GST_SECOND || source->internal) {
2857 binterval = data->interval;
2859 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2860 GST_TIME_ARGS (source->stats.prev_rtcptime),
2861 GST_TIME_ARGS (source->stats.last_rtcptime));
2862 /* if not received enough yet, fallback to larger default */
2863 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2864 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2866 binterval = 5 * GST_SECOND;
2867 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2869 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2870 GST_TIME_ARGS (binterval));
2872 /* check for our own source, we don't want to delete our own source. */
2873 if (!source->internal) {
2874 if (source->marked_bye) {
2875 /* if we received a BYE from the source, remove the source after some
2877 if (data->current_time > source->bye_time &&
2878 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2879 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2884 /* sources that were inactive for more than 5 times the deterministic reporting
2885 * interval get timed out. the min timeout is 5 seconds. */
2886 /* mind old time that might pre-date last time going to PLAYING */
2887 btime = MAX (source->last_activity, sess->start_time);
2888 if (data->current_time > btime) {
2889 interval = MAX (binterval * 5, 5 * GST_SECOND);
2890 if (data->current_time - btime > interval) {
2891 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2892 source->ssrc, GST_TIME_ARGS (btime));
2898 /* senders that did not send for a long time become a receiver, this also
2899 * holds for our own sources. */
2901 /* mind old time that might pre-date last time going to PLAYING */
2902 btime = MAX (source->last_rtp_activity, sess->start_time);
2903 if (data->current_time > btime) {
2904 interval = MAX (binterval * 2, 5 * GST_SECOND);
2905 if (data->current_time - btime > interval) {
2906 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2907 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2908 source->is_sender = FALSE;
2909 sess->stats.sender_sources--;
2910 if (source->internal)
2911 sess->stats.internal_sender_sources--;
2912 sendertimeout = TRUE;
2918 sess->total_sources--;
2920 sess->stats.sender_sources--;
2921 if (source->internal)
2922 sess->stats.internal_sender_sources--;
2925 sess->stats.active_sources--;
2927 if (source->internal)
2928 sess->stats.internal_sources--;
2931 on_bye_timeout (sess, source);
2933 on_timeout (sess, source);
2936 on_sender_timeout (sess, source);
2939 source->closing = remove;
2943 session_sdes (RTPSession * sess, ReportData * data)
2945 GstRTCPPacket *packet = &data->packet;
2946 const GstStructure *sdes;
2948 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2950 /* add SDES packet */
2951 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2953 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
2955 sdes = rtp_source_get_sdes_struct (data->source);
2957 /* add all fields in the structure, the order is not important. */
2958 n_fields = gst_structure_n_fields (sdes);
2959 for (i = 0; i < n_fields; ++i) {
2962 GstRTCPSDESType type;
2964 field = gst_structure_nth_field_name (sdes, i);
2967 value = gst_structure_get_string (sdes, field);
2970 type = gst_rtcp_sdes_name_to_type (field);
2972 /* Early packets are minimal and only include the CNAME */
2973 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2976 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2977 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2978 (const guint8 *) value);
2979 } else if (type == GST_RTCP_SDES_PRIV) {
2985 /* don't accept entries that are too big */
2986 prefix_len = strlen (field);
2987 if (prefix_len > 255)
2989 value_len = strlen (value);
2990 if (value_len > 255)
2992 data_len = 1 + prefix_len + value_len;
2996 data[0] = prefix_len;
2997 memcpy (&data[1], field, prefix_len);
2998 memcpy (&data[1 + prefix_len], value, value_len);
3000 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3004 data->has_sdes = TRUE;
3007 /* schedule a BYE packet */
3009 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3011 GstRTCPPacket *packet = &data->packet;
3012 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3015 session_sdes (sess, data);
3016 /* add a BYE packet */
3017 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3018 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3019 if (source->bye_reason)
3020 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3022 /* we have a BYE packet now */
3023 data->is_bye = TRUE;
3024 source->sent_bye = TRUE;
3028 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3030 GstClockTime new_send_time, elapsed;
3032 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3033 data->is_early = TRUE;
3035 data->is_early = FALSE;
3037 if (data->is_early && sess->next_early_rtcp_time < current_time)
3040 /* no need to check yet */
3041 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3042 sess->next_rtcp_check_time > current_time) {
3043 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3044 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3045 GST_TIME_ARGS (current_time));
3049 /* get elapsed time since we last reported */
3050 elapsed = current_time - sess->last_rtcp_send_time;
3052 new_send_time = data->interval;
3053 /* perform forward reconsideration */
3054 if (new_send_time != GST_CLOCK_TIME_NONE) {
3055 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
3057 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3058 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time),
3059 GST_TIME_ARGS (elapsed));
3061 new_send_time += sess->last_rtcp_send_time;
3064 /* check if reconsideration */
3065 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3066 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3067 GST_TIME_ARGS (new_send_time));
3068 /* store new check time */
3069 sess->next_rtcp_check_time = new_send_time;
3075 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
3077 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3078 GST_TIME_ARGS (new_send_time));
3080 sess->next_rtcp_check_time = new_send_time;
3081 if (new_send_time != GST_CLOCK_TIME_NONE) {
3082 sess->next_rtcp_check_time += current_time;
3084 /* Apply the rules from RFC 4585 section 3.5.3 */
3085 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
3086 GstClockTimeDiff T_rr_current_interval =
3087 g_random_double_range (0.5, 1.5) * sess->stats.min_interval;
3089 /* This will caused the RTCP to be suppressed if no FB packets are added */
3090 if (sess->last_rtcp_send_time + T_rr_current_interval >
3091 sess->next_rtcp_check_time) {
3092 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3093 " last: %" GST_TIME_FORMAT
3094 " + T_rr_current_interval: %" GST_TIME_FORMAT
3095 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3096 GST_TIME_ARGS (sess->stats.min_interval),
3097 GST_TIME_ARGS (sess->last_rtcp_send_time),
3098 GST_TIME_ARGS (T_rr_current_interval),
3099 GST_TIME_ARGS (sess->next_rtcp_check_time));
3100 data->may_suppress = TRUE;
3109 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3111 g_hash_table_insert (hash_table, key, g_object_ref (source));
3115 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3117 return source->closing;
3121 generate_rtcp (RTPSource * source, ReportData * data)
3123 RTPSession *sess = data->sess;
3125 /* only generate RTCP for active internal sources */
3126 if (!source->internal || source->sent_bye)
3129 data->source = source;
3132 session_start_rtcp (sess, data);
3134 if (source->marked_bye) {
3136 make_source_bye (sess, source, data);
3137 } else if (!data->is_early) {
3138 /* loop over all known sources and add report blocks. If we are ealy, we
3139 * just make a minimal RTCP packet and skip this step */
3140 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3141 (GHFunc) session_report_blocks, data);
3143 if (!data->has_sdes)
3144 session_sdes (sess, data);
3146 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3148 if (sess->change_ssrc) {
3149 GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
3150 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3151 GINT_TO_POINTER (source->ssrc));
3153 source->ssrc = rtp_session_create_new_ssrc (sess);
3154 rtp_source_reset (source);
3156 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3157 GINT_TO_POINTER (source->ssrc), source);
3159 sess->change_ssrc = FALSE;
3160 data->notify = TRUE;
3161 GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
3166 * rtp_session_on_timeout:
3167 * @sess: an #RTPSession
3168 * @current_time: the current system time
3169 * @ntpnstime: the current NTP time in nanoseconds
3170 * @running_time: the current running_time of the pipeline
3172 * Perform maintenance actions after the timeout obtained with
3173 * rtp_session_next_timeout() expired.
3175 * This function will perform timeouts of receivers and senders, send a BYE
3176 * packet or generate RTCP packets with current session stats.
3178 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3179 * times, for each packet that should be processed.
3181 * Returns: a #GstFlowReturn.
3184 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3185 guint64 ntpnstime, GstClockTime running_time)
3187 GstFlowReturn result = GST_FLOW_OK;
3188 ReportData data = { GST_RTCP_BUFFER_INIT };
3190 GHashTable *table_copy;
3192 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3194 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3195 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3196 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3199 data.current_time = current_time;
3200 data.ntpnstime = ntpnstime;
3201 data.running_time = running_time;
3202 data.may_suppress = FALSE;
3203 data.notify = FALSE;
3207 RTP_SESSION_LOCK (sess);
3208 /* get a new interval, we need this for various cleanups etc */
3209 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3211 /* Make a local copy of the hashtable. We need to do this because the
3212 * cleanup stage below releases the session lock. */
3213 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3214 (GDestroyNotify) g_object_unref);
3215 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3216 (GHFunc) clone_ssrcs_hashtable, table_copy);
3218 /* Clean up the session, mark the source for removing, this might release the
3220 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3221 g_hash_table_destroy (table_copy);
3223 /* Now remove the marked sources */
3224 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3225 (GHRFunc) remove_closing_sources, NULL);
3227 /* see if we need to generate SR or RR packets */
3228 if (!is_rtcp_time (sess, current_time, &data))
3231 generate_rtcp (own, &data);
3233 /* we keep track of the last report time in order to timeout inactive
3234 * receivers or senders */
3235 if (!data.is_early && !data.may_suppress)
3236 sess->last_rtcp_send_time = data.current_time;
3237 sess->first_rtcp = FALSE;
3238 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3241 RTP_SESSION_UNLOCK (sess);
3244 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3246 /* push out the RTCP packet */
3248 gboolean do_not_suppress;
3249 GstBuffer *buffer = data.rtcp;
3251 /* Give the user a change to add its own packet */
3252 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3253 buffer, data.is_early, &do_not_suppress);
3255 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3258 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3260 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3261 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3262 sess->stats.avg_rtcp_packet_size, packet_size);
3264 sess->callbacks.send_rtcp (sess, own, buffer, data.is_bye,
3265 sess->send_rtcp_user_data);
3267 GST_DEBUG ("freeing packet callback: %p"
3268 " do_not_suppress: %d may_suppress: %d",
3269 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3270 gst_buffer_unref (buffer);
3278 * rtp_session_request_early_rtcp:
3279 * @sess: an #RTPSession
3280 * @current_time: the current system time
3281 * @max_delay: maximum delay
3283 * Request transmission of early RTCP
3286 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3287 GstClockTimeDiff max_delay)
3289 GstClockTime T_dither_max;
3291 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3293 RTP_SESSION_LOCK (sess);
3295 /* Check if already requested */
3296 /* RFC 4585 section 3.5.2 step 2 */
3297 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3300 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time))
3303 /* Ignore the request a scheduled packet will be in time anyway */
3304 if (current_time + max_delay > sess->next_rtcp_check_time)
3307 /* RFC 4585 section 3.5.2 step 2b */
3308 /* If the total sources is <=2, then there is only us and one peer */
3309 if (sess->total_sources <= 2) {
3312 /* Divide by 2 because l = 0.5 */
3313 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3317 /* RFC 4585 section 3.5.2 step 3 */
3318 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3321 /* RFC 4585 section 3.5.2 step 4
3322 * Don't send if allow_early is FALSE, but not if we are in
3323 * immediate mode, meaning we are part of a group of at most the
3324 * application-specific threshold.
3326 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3327 sess->allow_early == FALSE)
3331 /* Schedule an early transmission later */
3332 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3335 /* If no dithering, schedule it for NOW */
3336 sess->next_early_rtcp_time = current_time;
3339 RTP_SESSION_UNLOCK (sess);
3341 /* notify app of need to send packet early
3342 * and therefore of timeout change */
3343 if (sess->callbacks.reconsider)
3344 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3350 RTP_SESSION_UNLOCK (sess);
3354 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3355 gboolean fir, gint count)
3357 RTPSource *src = find_source (sess, ssrc);
3363 src->send_pli = FALSE;
3364 src->send_fir = TRUE;
3366 if (count == -1 || count != src->last_fir_count)
3367 src->current_send_fir_seqnum++;
3368 src->last_fir_count = count;
3369 } else if (!src->send_fir) {
3370 src->send_pli = TRUE;
3373 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3379 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3381 GstRTCPPacket packet;
3382 GstRTCPBuffer rtcp = { NULL, };
3383 gboolean ret = FALSE;
3385 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3387 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3388 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3389 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3393 gst_rtcp_buffer_unmap (&rtcp);
3399 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3402 gboolean ret = FALSE;
3403 GHashTableIter iter;
3404 gpointer key, value;
3405 gboolean started_fir = FALSE;
3406 GstRTCPPacket fir_rtcppacket;
3407 GstRTCPPacket packet;
3408 GstRTCPBuffer rtcp = { NULL, };
3411 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3413 gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
3414 switch (gst_rtcp_packet_get_type (&packet)) {
3415 case GST_RTCP_TYPE_SR:
3416 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc,
3417 NULL, NULL, NULL, NULL);
3419 case GST_RTCP_TYPE_RR:
3420 ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
3426 RTP_SESSION_LOCK (sess);
3427 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3428 while (g_hash_table_iter_next (&iter, &key, &value)) {
3429 guint media_ssrc = GPOINTER_TO_UINT (key);
3430 RTPSource *media_src = value;
3433 if (media_src->send_fir) {
3435 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3438 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3439 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket, ssrc);
3440 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3442 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3443 gst_rtcp_packet_remove (&fir_rtcppacket);
3449 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3450 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3454 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3455 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3457 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3459 fci_data[0] = media_src->current_send_fir_seqnum;
3460 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3461 media_src->send_fir = FALSE;
3465 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3466 while (g_hash_table_iter_next (&iter, &key, &value)) {
3467 guint media_ssrc = GPOINTER_TO_UINT (key);
3468 RTPSource *media_src = value;
3469 GstRTCPPacket pli_rtcppacket;
3471 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3472 has_pli_compare_func, NULL)) {
3473 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3475 /* Break because the packet is full, will put next request in a
3478 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3479 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket, ssrc);
3480 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3483 media_src->send_pli = FALSE;
3485 RTP_SESSION_UNLOCK (sess);
3488 gst_rtcp_buffer_unmap (&rtcp);
3494 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3498 if (!sess->callbacks.send_rtcp)
3501 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3503 rtp_session_request_early_rtcp (sess, now, max_delay);