2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "rtpsession.h"
33 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
34 #define GST_CAT_DEFAULT rtp_session_debug
36 /* signals and args */
39 SIGNAL_GET_SOURCE_BY_SSRC,
41 SIGNAL_ON_SSRC_COLLISION,
42 SIGNAL_ON_SSRC_VALIDATED,
43 SIGNAL_ON_SSRC_ACTIVE,
46 SIGNAL_ON_BYE_TIMEOUT,
48 SIGNAL_ON_SENDER_TIMEOUT,
49 SIGNAL_ON_SENDING_RTCP,
50 SIGNAL_ON_FEEDBACK_RTCP,
55 #define DEFAULT_INTERNAL_SOURCE NULL
56 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
57 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
58 #define DEFAULT_RTCP_RR_BANDWIDTH -1
59 #define DEFAULT_RTCP_RS_BANDWIDTH -1
60 #define DEFAULT_RTCP_MTU 1400
61 #define DEFAULT_SDES NULL
62 #define DEFAULT_NUM_SOURCES 0
63 #define DEFAULT_NUM_ACTIVE_SOURCES 0
64 #define DEFAULT_SOURCES NULL
65 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
66 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
67 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
68 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
77 PROP_RTCP_RR_BANDWIDTH,
78 PROP_RTCP_RS_BANDWIDTH,
82 PROP_NUM_ACTIVE_SOURCES,
85 PROP_RTCP_MIN_INTERVAL,
86 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
87 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
93 /* update average packet size */
94 #define INIT_AVG(avg, val) \
96 #define UPDATE_AVG(avg, val) \
100 (avg) = ((val) + (15 * (avg))) >> 4;
103 /* The number RTCP intervals after which to timeout entries in the
106 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
108 /* GObject vmethods */
109 static void rtp_session_finalize (GObject * object);
110 static void rtp_session_set_property (GObject * object, guint prop_id,
111 const GValue * value, GParamSpec * pspec);
112 static void rtp_session_get_property (GObject * object, guint prop_id,
113 GValue * value, GParamSpec * pspec);
115 static void rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay);
117 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
119 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
121 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
122 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
123 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
124 static RTPSource *obtain_internal_source (RTPSession * sess,
125 guint32 ssrc, gboolean * created);
126 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
127 GstClockTime current_time);
128 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
129 gboolean deterministic, gboolean first);
132 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
133 const GValue * handler_return, gpointer data)
135 if (g_value_get_boolean (handler_return))
136 g_value_set_boolean (return_accu, TRUE);
142 rtp_session_class_init (RTPSessionClass * klass)
144 GObjectClass *gobject_class;
146 gobject_class = (GObjectClass *) klass;
148 gobject_class->finalize = rtp_session_finalize;
149 gobject_class->set_property = rtp_session_set_property;
150 gobject_class->get_property = rtp_session_get_property;
153 * RTPSession::get-source-by-ssrc:
154 * @session: the object which received the signal
155 * @ssrc: the SSRC of the RTPSource
157 * Request the #RTPSource object with SSRC @ssrc in @session.
159 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
160 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
161 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
162 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
163 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
166 * RTPSession::on-new-ssrc:
167 * @session: the object which received the signal
168 * @src: the new RTPSource
170 * Notify of a new SSRC that entered @session.
172 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
173 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
174 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
175 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
178 * RTPSession::on-ssrc-collision:
179 * @session: the object which received the signal
180 * @src: the #RTPSource that caused a collision
182 * Notify when we have an SSRC collision
184 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
185 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
186 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
187 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
190 * RTPSession::on-ssrc-validated:
191 * @session: the object which received the signal
192 * @src: the new validated RTPSource
194 * Notify of a new SSRC that became validated.
196 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
197 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
198 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
199 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
202 * RTPSession::on-ssrc-active:
203 * @session: the object which received the signal
204 * @src: the active RTPSource
206 * Notify of a SSRC that is active, i.e., sending RTCP.
208 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
209 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
210 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
211 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
214 * RTPSession::on-ssrc-sdes:
215 * @session: the object which received the signal
216 * @src: the RTPSource
218 * Notify that a new SDES was received for SSRC.
220 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
221 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
222 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
223 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
226 * RTPSession::on-bye-ssrc:
227 * @session: the object which received the signal
228 * @src: the RTPSource that went away
230 * Notify of an SSRC that became inactive because of a BYE packet.
232 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
233 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
234 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
235 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
238 * RTPSession::on-bye-timeout:
239 * @session: the object which received the signal
240 * @src: the RTPSource that timed out
242 * Notify of an SSRC that has timed out because of BYE
244 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
245 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
246 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
247 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
250 * RTPSession::on-timeout:
251 * @session: the object which received the signal
252 * @src: the RTPSource that timed out
254 * Notify of an SSRC that has timed out
256 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
257 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
258 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
259 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
262 * RTPSession::on-sender-timeout:
263 * @session: the object which received the signal
264 * @src: the RTPSource that timed out
266 * Notify of an SSRC that was a sender but timed out and became a receiver.
268 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
269 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
270 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
271 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
275 * RTPSession::on-sending-rtcp
276 * @session: the object which received the signal
277 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
278 * @early: %TRUE if the packet is early, %FALSE if it is regular
280 * This signal is emitted before sending an RTCP packet, it can be used
281 * to add extra RTCP Packets.
283 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
284 * if suppressing it is acceptable
286 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
287 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
288 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
289 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
290 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
293 * RTPSession::on-feedback-rtcp:
294 * @session: the object which received the signal
295 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
296 * %GST_RTCP_TYPE_RTPFB
297 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
298 * @sender_ssrc: The SSRC of the sender
299 * @media_ssrc: The SSRC of the media this refers to
300 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
303 * Notify that a RTCP feedback packet has been received
305 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
306 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
307 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
308 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
309 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
312 * RTPSession::send-rtcp:
313 * @session: the object which received the signal
314 * @max_delay: The maximum delay after which the feedback will not be useful
317 * Requests that the #RTPSession initiate a new RTCP packet as soon as
318 * possible within the requested delay.
320 rtp_session_signals[SIGNAL_SEND_RTCP] =
321 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
322 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
323 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
324 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
326 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
327 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
328 "The internal SSRC used for the session (deprecated)",
329 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
332 g_param_spec_object ("internal-source", "Internal Source",
333 "The internal source element of the session (deprecated)",
334 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
336 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
337 g_param_spec_double ("bandwidth", "Bandwidth",
338 "The bandwidth of the session (0 for auto-discover)",
339 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
340 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
342 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
343 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
344 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
345 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
346 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
348 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
349 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
350 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
351 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
352 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
354 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
355 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
356 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
357 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
358 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
360 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
361 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
362 "The maximum size of the RTCP packets",
363 16, G_MAXINT16, DEFAULT_RTCP_MTU,
364 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
366 g_object_class_install_property (gobject_class, PROP_SDES,
367 g_param_spec_boxed ("sdes", "SDES",
368 "The SDES items of this session",
369 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
371 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
372 g_param_spec_uint ("num-sources", "Num Sources",
373 "The number of sources in the session", 0, G_MAXUINT,
374 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
377 g_param_spec_uint ("num-active-sources", "Num Active Sources",
378 "The number of active sources in the session", 0, G_MAXUINT,
379 DEFAULT_NUM_ACTIVE_SOURCES,
380 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
384 * Get a GValue Array of all sources in the session.
387 * <title>Getting the #RTPSources of a session
394 * g_object_get (sess, "sources", &arr, NULL);
396 * for (i = 0; i < arr->n_values; i++) {
399 * val = g_value_array_get_nth (arr, i);
400 * source = g_value_get_object (val);
402 * g_value_array_free (arr);
407 g_object_class_install_property (gobject_class, PROP_SOURCES,
408 g_param_spec_boxed ("sources", "Sources",
409 "An array of all known sources in the session",
410 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
412 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
413 g_param_spec_boolean ("favor-new", "Favor new sources",
414 "Resolve SSRC conflict in favor of new sources", FALSE,
415 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
417 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
418 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
419 "Minimum interval between Regular RTCP packet (in ns)",
420 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
421 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
423 g_object_class_install_property (gobject_class,
424 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
425 g_param_spec_uint64 ("rtcp-feedback-retention-window",
426 "RTCP Feedback retention window",
427 "Duration during which RTCP Feedback packets are retained (in ns)",
428 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
429 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
431 g_object_class_install_property (gobject_class,
432 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
433 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
434 "RTCP Immediate Feedback threshold",
435 "The maximum number of members of a RTP session for which immediate"
437 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
438 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
440 g_object_class_install_property (gobject_class, PROP_PROBATION,
441 g_param_spec_uint ("probation", "Number of probations",
442 "Consecutive packet sequence numbers to accept the source",
443 0, G_MAXUINT, DEFAULT_PROBATION,
444 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
449 * Various session statistics. This property returns a GstStructure
450 * with name application/x-rtp-session-stats with the following fields:
452 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
453 * dropped (due to bandwidth constraints)
454 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
455 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
459 g_object_class_install_property (gobject_class, PROP_STATS,
460 g_param_spec_boxed ("stats", "Statistics",
461 "Various statistics", GST_TYPE_STRUCTURE,
462 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
464 klass->get_source_by_ssrc =
465 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
466 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
468 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
472 rtp_session_init (RTPSession * sess)
477 g_mutex_init (&sess->lock);
478 sess->key = g_random_int ();
482 for (i = 0; i < 32; i++) {
484 g_hash_table_new_full (NULL, NULL, NULL,
485 (GDestroyNotify) g_object_unref);
488 rtp_stats_init_defaults (&sess->stats);
489 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
490 rtp_stats_set_min_interval (&sess->stats,
491 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
493 sess->recalc_bandwidth = TRUE;
494 sess->bandwidth = DEFAULT_BANDWIDTH;
495 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
496 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
497 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
499 /* default UDP header length */
500 sess->header_len = 28;
501 sess->mtu = DEFAULT_RTCP_MTU;
503 sess->probation = DEFAULT_PROBATION;
505 /* some default SDES entries */
506 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
508 /* we do not want to leak details like the username or hostname here */
509 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
510 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
514 /* we do not want to leak the user's real name here */
515 str = g_strdup_printf ("Anon%u", g_random_int ());
516 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
520 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
522 /* this is the SSRC we suggest */
523 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
525 sess->first_rtcp = TRUE;
526 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
528 sess->allow_early = TRUE;
529 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
530 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
531 sess->rtcp_immediate_feedback_threshold =
532 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
534 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
536 sess->is_doing_ptp = TRUE;
540 rtp_session_finalize (GObject * object)
545 sess = RTP_SESSION_CAST (object);
547 gst_structure_free (sess->sdes);
549 for (i = 0; i < 32; i++)
550 g_hash_table_destroy (sess->ssrcs[i]);
552 g_mutex_clear (&sess->lock);
554 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
558 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
560 GValue value = { 0 };
562 g_value_init (&value, RTP_TYPE_SOURCE);
563 g_value_take_object (&value, source);
564 /* copies the value */
565 g_value_array_append (arr, &value);
569 rtp_session_create_sources (RTPSession * sess)
574 RTP_SESSION_LOCK (sess);
575 /* get number of elements in the table */
576 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
577 /* create the result value array */
578 res = g_value_array_new (size);
580 /* and copy all values into the array */
581 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
582 RTP_SESSION_UNLOCK (sess);
587 static GstStructure *
588 rtp_session_create_stats (RTPSession * sess)
592 s = gst_structure_new ("application/x-rtp-session-stats",
593 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
594 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
595 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
601 rtp_session_set_property (GObject * object, guint prop_id,
602 const GValue * value, GParamSpec * pspec)
606 sess = RTP_SESSION (object);
609 case PROP_INTERNAL_SSRC:
610 GST_ERROR_OBJECT (object, "Setting the \"internal-ssrc\" property"
611 " is deprecated and ignored");
614 RTP_SESSION_LOCK (sess);
615 sess->bandwidth = g_value_get_double (value);
616 sess->recalc_bandwidth = TRUE;
617 RTP_SESSION_UNLOCK (sess);
619 case PROP_RTCP_FRACTION:
620 RTP_SESSION_LOCK (sess);
621 sess->rtcp_bandwidth = g_value_get_double (value);
622 sess->recalc_bandwidth = TRUE;
623 RTP_SESSION_UNLOCK (sess);
625 case PROP_RTCP_RR_BANDWIDTH:
626 RTP_SESSION_LOCK (sess);
627 sess->rtcp_rr_bandwidth = g_value_get_int (value);
628 sess->recalc_bandwidth = TRUE;
629 RTP_SESSION_UNLOCK (sess);
631 case PROP_RTCP_RS_BANDWIDTH:
632 RTP_SESSION_LOCK (sess);
633 sess->rtcp_rs_bandwidth = g_value_get_int (value);
634 sess->recalc_bandwidth = TRUE;
635 RTP_SESSION_UNLOCK (sess);
638 sess->mtu = g_value_get_uint (value);
641 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
644 sess->favor_new = g_value_get_boolean (value);
646 case PROP_RTCP_MIN_INTERVAL:
647 rtp_stats_set_min_interval (&sess->stats,
648 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
649 /* trigger reconsideration */
650 RTP_SESSION_LOCK (sess);
651 sess->next_rtcp_check_time = 0;
652 RTP_SESSION_UNLOCK (sess);
653 if (sess->callbacks.reconsider)
654 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
656 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
657 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
660 sess->probation = g_value_get_uint (value);
663 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
669 rtp_session_get_property (GObject * object, guint prop_id,
670 GValue * value, GParamSpec * pspec)
674 sess = RTP_SESSION (object);
677 case PROP_INTERNAL_SSRC:
678 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
680 case PROP_INTERNAL_SOURCE:
681 /* FIXME, return a random source */
682 g_value_set_object (value, NULL);
685 g_value_set_double (value, sess->bandwidth);
687 case PROP_RTCP_FRACTION:
688 g_value_set_double (value, sess->rtcp_bandwidth);
690 case PROP_RTCP_RR_BANDWIDTH:
691 g_value_set_int (value, sess->rtcp_rr_bandwidth);
693 case PROP_RTCP_RS_BANDWIDTH:
694 g_value_set_int (value, sess->rtcp_rs_bandwidth);
697 g_value_set_uint (value, sess->mtu);
700 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
702 case PROP_NUM_SOURCES:
703 g_value_set_uint (value, rtp_session_get_num_sources (sess));
705 case PROP_NUM_ACTIVE_SOURCES:
706 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
709 g_value_take_boxed (value, rtp_session_create_sources (sess));
712 g_value_set_boolean (value, sess->favor_new);
714 case PROP_RTCP_MIN_INTERVAL:
715 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
717 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
718 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
721 g_value_set_uint (value, sess->probation);
724 g_value_take_boxed (value, rtp_session_create_stats (sess));
727 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
733 on_new_ssrc (RTPSession * sess, RTPSource * source)
735 g_object_ref (source);
736 RTP_SESSION_UNLOCK (sess);
737 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
738 RTP_SESSION_LOCK (sess);
739 g_object_unref (source);
743 on_ssrc_collision (RTPSession * sess, RTPSource * source)
745 g_object_ref (source);
746 RTP_SESSION_UNLOCK (sess);
747 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
749 RTP_SESSION_LOCK (sess);
750 g_object_unref (source);
754 on_ssrc_validated (RTPSession * sess, RTPSource * source)
756 g_object_ref (source);
757 RTP_SESSION_UNLOCK (sess);
758 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
760 RTP_SESSION_LOCK (sess);
761 g_object_unref (source);
765 on_ssrc_active (RTPSession * sess, RTPSource * source)
767 g_object_ref (source);
768 RTP_SESSION_UNLOCK (sess);
769 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
770 RTP_SESSION_LOCK (sess);
771 g_object_unref (source);
775 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
777 g_object_ref (source);
778 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
779 RTP_SESSION_UNLOCK (sess);
780 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
781 RTP_SESSION_LOCK (sess);
782 g_object_unref (source);
786 on_bye_ssrc (RTPSession * sess, RTPSource * source)
788 g_object_ref (source);
789 RTP_SESSION_UNLOCK (sess);
790 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
791 RTP_SESSION_LOCK (sess);
792 g_object_unref (source);
796 on_bye_timeout (RTPSession * sess, RTPSource * source)
798 g_object_ref (source);
799 RTP_SESSION_UNLOCK (sess);
800 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
801 RTP_SESSION_LOCK (sess);
802 g_object_unref (source);
806 on_timeout (RTPSession * sess, RTPSource * source)
808 g_object_ref (source);
809 RTP_SESSION_UNLOCK (sess);
810 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
811 RTP_SESSION_LOCK (sess);
812 g_object_unref (source);
816 on_sender_timeout (RTPSession * sess, RTPSource * source)
818 g_object_ref (source);
819 RTP_SESSION_UNLOCK (sess);
820 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
822 RTP_SESSION_LOCK (sess);
823 g_object_unref (source);
829 * Create a new session object.
831 * Returns: a new #RTPSession. g_object_unref() after usage.
834 rtp_session_new (void)
838 sess = g_object_new (RTP_TYPE_SESSION, NULL);
844 * rtp_session_set_callbacks:
845 * @sess: an #RTPSession
846 * @callbacks: callbacks to configure
847 * @user_data: user data passed in the callbacks
849 * Configure a set of callbacks to be notified of actions.
852 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
855 g_return_if_fail (RTP_IS_SESSION (sess));
857 if (callbacks->process_rtp) {
858 sess->callbacks.process_rtp = callbacks->process_rtp;
859 sess->process_rtp_user_data = user_data;
861 if (callbacks->send_rtp) {
862 sess->callbacks.send_rtp = callbacks->send_rtp;
863 sess->send_rtp_user_data = user_data;
865 if (callbacks->send_rtcp) {
866 sess->callbacks.send_rtcp = callbacks->send_rtcp;
867 sess->send_rtcp_user_data = user_data;
869 if (callbacks->sync_rtcp) {
870 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
871 sess->sync_rtcp_user_data = user_data;
873 if (callbacks->clock_rate) {
874 sess->callbacks.clock_rate = callbacks->clock_rate;
875 sess->clock_rate_user_data = user_data;
877 if (callbacks->reconsider) {
878 sess->callbacks.reconsider = callbacks->reconsider;
879 sess->reconsider_user_data = user_data;
881 if (callbacks->request_key_unit) {
882 sess->callbacks.request_key_unit = callbacks->request_key_unit;
883 sess->request_key_unit_user_data = user_data;
885 if (callbacks->request_time) {
886 sess->callbacks.request_time = callbacks->request_time;
887 sess->request_time_user_data = user_data;
889 if (callbacks->notify_nack) {
890 sess->callbacks.notify_nack = callbacks->notify_nack;
891 sess->notify_nack_user_data = user_data;
896 * rtp_session_set_process_rtp_callback:
897 * @sess: an #RTPSession
898 * @callback: callback to set
899 * @user_data: user data passed in the callback
901 * Configure only the process_rtp callback to be notified of the process_rtp action.
904 rtp_session_set_process_rtp_callback (RTPSession * sess,
905 RTPSessionProcessRTP callback, gpointer user_data)
907 g_return_if_fail (RTP_IS_SESSION (sess));
909 sess->callbacks.process_rtp = callback;
910 sess->process_rtp_user_data = user_data;
914 * rtp_session_set_send_rtp_callback:
915 * @sess: an #RTPSession
916 * @callback: callback to set
917 * @user_data: user data passed in the callback
919 * Configure only the send_rtp callback to be notified of the send_rtp action.
922 rtp_session_set_send_rtp_callback (RTPSession * sess,
923 RTPSessionSendRTP callback, gpointer user_data)
925 g_return_if_fail (RTP_IS_SESSION (sess));
927 sess->callbacks.send_rtp = callback;
928 sess->send_rtp_user_data = user_data;
932 * rtp_session_set_send_rtcp_callback:
933 * @sess: an #RTPSession
934 * @callback: callback to set
935 * @user_data: user data passed in the callback
937 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
940 rtp_session_set_send_rtcp_callback (RTPSession * sess,
941 RTPSessionSendRTCP callback, gpointer user_data)
943 g_return_if_fail (RTP_IS_SESSION (sess));
945 sess->callbacks.send_rtcp = callback;
946 sess->send_rtcp_user_data = user_data;
950 * rtp_session_set_sync_rtcp_callback:
951 * @sess: an #RTPSession
952 * @callback: callback to set
953 * @user_data: user data passed in the callback
955 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
958 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
959 RTPSessionSyncRTCP callback, gpointer user_data)
961 g_return_if_fail (RTP_IS_SESSION (sess));
963 sess->callbacks.sync_rtcp = callback;
964 sess->sync_rtcp_user_data = user_data;
968 * rtp_session_set_clock_rate_callback:
969 * @sess: an #RTPSession
970 * @callback: callback to set
971 * @user_data: user data passed in the callback
973 * Configure only the clock_rate callback to be notified of the clock_rate action.
976 rtp_session_set_clock_rate_callback (RTPSession * sess,
977 RTPSessionClockRate callback, gpointer user_data)
979 g_return_if_fail (RTP_IS_SESSION (sess));
981 sess->callbacks.clock_rate = callback;
982 sess->clock_rate_user_data = user_data;
986 * rtp_session_set_reconsider_callback:
987 * @sess: an #RTPSession
988 * @callback: callback to set
989 * @user_data: user data passed in the callback
991 * Configure only the reconsider callback to be notified of the reconsider action.
994 rtp_session_set_reconsider_callback (RTPSession * sess,
995 RTPSessionReconsider callback, gpointer user_data)
997 g_return_if_fail (RTP_IS_SESSION (sess));
999 sess->callbacks.reconsider = callback;
1000 sess->reconsider_user_data = user_data;
1004 * rtp_session_set_request_time_callback:
1005 * @sess: an #RTPSession
1006 * @callback: callback to set
1007 * @user_data: user data passed in the callback
1009 * Configure only the request_time callback
1012 rtp_session_set_request_time_callback (RTPSession * sess,
1013 RTPSessionRequestTime callback, gpointer user_data)
1015 g_return_if_fail (RTP_IS_SESSION (sess));
1017 sess->callbacks.request_time = callback;
1018 sess->request_time_user_data = user_data;
1022 * rtp_session_set_bandwidth:
1023 * @sess: an #RTPSession
1024 * @bandwidth: the bandwidth allocated
1026 * Set the session bandwidth in bytes per second.
1029 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1031 g_return_if_fail (RTP_IS_SESSION (sess));
1033 RTP_SESSION_LOCK (sess);
1034 sess->stats.bandwidth = bandwidth;
1035 RTP_SESSION_UNLOCK (sess);
1039 * rtp_session_get_bandwidth:
1040 * @sess: an #RTPSession
1042 * Get the session bandwidth.
1044 * Returns: the session bandwidth.
1047 rtp_session_get_bandwidth (RTPSession * sess)
1051 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1053 RTP_SESSION_LOCK (sess);
1054 result = sess->stats.bandwidth;
1055 RTP_SESSION_UNLOCK (sess);
1061 * rtp_session_set_rtcp_fraction:
1062 * @sess: an #RTPSession
1063 * @bandwidth: the RTCP bandwidth
1065 * Set the bandwidth in bytes per second that should be used for RTCP
1069 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1071 g_return_if_fail (RTP_IS_SESSION (sess));
1073 RTP_SESSION_LOCK (sess);
1074 sess->stats.rtcp_bandwidth = bandwidth;
1075 RTP_SESSION_UNLOCK (sess);
1079 * rtp_session_get_rtcp_fraction:
1080 * @sess: an #RTPSession
1082 * Get the session bandwidth used for RTCP.
1084 * Returns: The bandwidth used for RTCP messages.
1087 rtp_session_get_rtcp_fraction (RTPSession * sess)
1091 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1093 RTP_SESSION_LOCK (sess);
1094 result = sess->stats.rtcp_bandwidth;
1095 RTP_SESSION_UNLOCK (sess);
1101 * rtp_session_get_sdes_struct:
1102 * @sess: an #RTSPSession
1104 * Get the SDES data as a #GstStructure
1106 * Returns: a GstStructure with SDES items for @sess. This function returns a
1107 * copy of the SDES structure, use gst_structure_free() after usage.
1110 rtp_session_get_sdes_struct (RTPSession * sess)
1112 GstStructure *result = NULL;
1114 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1116 RTP_SESSION_LOCK (sess);
1118 result = gst_structure_copy (sess->sdes);
1119 RTP_SESSION_UNLOCK (sess);
1125 * rtp_session_set_sdes_struct:
1126 * @sess: an #RTSPSession
1127 * @sdes: a #GstStructure
1129 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1132 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1134 g_return_if_fail (sdes);
1135 g_return_if_fail (RTP_IS_SESSION (sess));
1137 RTP_SESSION_LOCK (sess);
1139 gst_structure_free (sess->sdes);
1140 sess->sdes = gst_structure_copy (sdes);
1141 RTP_SESSION_UNLOCK (sess);
1144 static GstFlowReturn
1145 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1147 GstFlowReturn result = GST_FLOW_OK;
1149 if (source->internal) {
1150 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1152 RTP_SESSION_UNLOCK (session);
1154 if (session->callbacks.send_rtp)
1156 session->callbacks.send_rtp (session, source, data,
1157 session->send_rtp_user_data);
1159 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1162 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1163 RTP_SESSION_UNLOCK (session);
1165 if (session->callbacks.process_rtp)
1167 session->callbacks.process_rtp (session, source,
1168 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1170 gst_buffer_unref (GST_BUFFER_CAST (data));
1172 RTP_SESSION_LOCK (session);
1178 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1182 RTP_SESSION_UNLOCK (session);
1184 if (session->callbacks.clock_rate)
1186 session->callbacks.clock_rate (session, pt,
1187 session->clock_rate_user_data);
1191 RTP_SESSION_LOCK (session);
1193 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1198 static RTPSourceCallbacks callbacks = {
1199 (RTPSourcePushRTP) source_push_rtp,
1200 (RTPSourceClockRate) source_clock_rate,
1204 check_collision (RTPSession * sess, RTPSource * source,
1205 RTPPacketInfo * pinfo, gboolean rtp)
1209 /* If we have no pinfo address, we can't do collision checking */
1210 if (!pinfo->address)
1213 ssrc = rtp_source_get_ssrc (source);
1215 if (!source->internal) {
1216 GSocketAddress *from;
1218 /* This is not our local source, but lets check if two remote
1221 from = source->rtp_from;
1223 from = source->rtcp_from;
1227 if (__g_socket_address_equal (from, pinfo->address)) {
1228 /* Address is the same */
1231 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1232 if (sess->favor_new) {
1233 if (rtp_source_find_conflicting_address (source,
1234 pinfo->address, pinfo->current_time)) {
1237 buf1 = __g_socket_address_to_string (pinfo->address);
1238 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1246 /* Current address is not a known conflict, lets assume this is
1247 * a new source. Save old address in possible conflict list
1249 rtp_source_add_conflicting_address (source, from,
1250 pinfo->current_time);
1252 buf1 = __g_socket_address_to_string (from);
1253 buf2 = __g_socket_address_to_string (pinfo->address);
1255 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1256 " saving old as known conflict", ssrc, buf1, buf2);
1259 rtp_source_set_rtp_from (source, pinfo->address);
1261 rtp_source_set_rtcp_from (source, pinfo->address);
1269 /* Don't need to save old addresses, we ignore new sources */
1274 /* We don't already have a from address for RTP, just set it */
1276 rtp_source_set_rtp_from (source, pinfo->address);
1278 rtp_source_set_rtcp_from (source, pinfo->address);
1282 /* FIXME: Log 3rd party collision somehow
1283 * Maybe should be done in upper layer, only the SDES can tell us
1284 * if its a collision or a loop
1287 /* This is sending with our ssrc, is it an address we already know */
1288 if (rtp_source_find_conflicting_address (source, pinfo->address,
1289 pinfo->current_time)) {
1290 /* Its a known conflict, its probably a loop, not a collision
1291 * lets just drop the incoming packet
1293 GST_DEBUG ("Our packets are being looped back to us, dropping");
1295 /* Its a new collision, lets change our SSRC */
1296 rtp_source_add_conflicting_address (source, pinfo->address,
1297 pinfo->current_time);
1299 GST_DEBUG ("Collision for SSRC %x", ssrc);
1300 /* mark the source BYE */
1301 rtp_source_mark_bye (source, "SSRC Collision");
1302 /* if we were suggesting this SSRC, change to something else */
1303 if (sess->suggested_ssrc == ssrc)
1304 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1306 on_ssrc_collision (sess, source);
1308 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1317 gboolean is_doing_ptp;
1318 GSocketAddress *new_addr;
1321 /* check if the two given ip addr are the same (do not care about the port) */
1323 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1326 g_inet_address_equal (g_inet_socket_address_get_address
1327 (G_INET_SOCKET_ADDRESS (a)),
1328 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1332 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1333 CompareAddrData * data)
1335 /* only compare ip addr of remote sources which are also not closing */
1336 if (!source->internal && !source->closing && source->rtp_from) {
1337 /* look for the first rtp source */
1338 if (!data->new_addr)
1339 data->new_addr = source->rtp_from;
1340 /* compare current ip addr with the first one */
1342 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1347 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1348 CompareAddrData * data)
1350 /* only compare ip addr of remote sources which are also not closing */
1351 if (!source->internal && !source->closing && source->rtcp_from) {
1352 /* look for the first rtcp source */
1353 if (!data->new_addr)
1354 data->new_addr = source->rtcp_from;
1356 /* compare current ip addr with the first one */
1357 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1361 /* loop over our non-internal source to know if the session
1362 * is doing point-to-point */
1364 session_update_ptp (RTPSession * sess)
1366 /* to know if the session is doing point to point, the ip addr
1367 * of each non-internal (=remotes) source have to be compared
1370 gboolean is_doing_rtp_ptp = FALSE;
1371 gboolean is_doing_rtcp_ptp = FALSE;
1372 CompareAddrData data;
1374 /* compare the first remote source's ip addr that receive rtp packets
1375 * with other remote rtp source.
1376 * it's enough because the session just needs to know if they are all
1379 data.is_doing_ptp = TRUE;
1380 data.new_addr = NULL;
1381 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1382 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1383 is_doing_rtp_ptp = data.is_doing_ptp;
1385 /* same but about rtcp */
1386 data.is_doing_ptp = TRUE;
1387 data.new_addr = NULL;
1388 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1389 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1390 is_doing_rtcp_ptp = data.is_doing_ptp;
1392 /* the session is doing point-to-point if all rtp remote have the same
1393 * ip addr and if all rtcp remote sources have the same ip addr */
1394 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1396 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1400 add_source (RTPSession * sess, RTPSource * src)
1402 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1403 GINT_TO_POINTER (src->ssrc), src);
1404 /* report the new source ASAP */
1405 src->generation = sess->generation;
1406 /* we have one more source now */
1407 sess->total_sources++;
1408 if (RTP_SOURCE_IS_ACTIVE (src))
1409 sess->stats.active_sources++;
1410 if (src->internal) {
1411 sess->stats.internal_sources++;
1412 if (sess->suggested_ssrc != src->ssrc)
1413 sess->suggested_ssrc = src->ssrc;
1416 /* update point-to-point status */
1418 session_update_ptp (sess);
1422 find_source (RTPSession * sess, guint32 ssrc)
1424 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1425 GINT_TO_POINTER (ssrc));
1428 /* must be called with the session lock, the returned source needs to be
1429 * unreffed after usage. */
1431 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1432 RTPPacketInfo * pinfo, gboolean rtp)
1436 source = find_source (sess, ssrc);
1437 if (source == NULL) {
1438 /* make new Source in probation and insert */
1439 source = rtp_source_new (ssrc);
1441 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1443 /* for RTP packets we need to set the source in probation. Receiving RTCP
1444 * packets of an SSRC, on the other hand, is a strong indication that we
1445 * are dealing with a valid source. */
1447 g_object_set (source, "probation", sess->probation, NULL);
1449 g_object_set (source, "probation", 0, NULL);
1451 /* store from address, if any */
1452 if (pinfo->address) {
1454 rtp_source_set_rtp_from (source, pinfo->address);
1456 rtp_source_set_rtcp_from (source, pinfo->address);
1459 /* configure a callback on the source */
1460 rtp_source_set_callbacks (source, &callbacks, sess);
1462 add_source (sess, source);
1466 /* check for collision, this updates the address when not previously set */
1467 if (check_collision (sess, source, pinfo, rtp)) {
1470 /* Receiving RTCP packets of an SSRC is a strong indication that we
1471 * are dealing with a valid source. */
1473 g_object_set (source, "probation", 0, NULL);
1475 /* update last activity */
1476 source->last_activity = pinfo->current_time;
1478 source->last_rtp_activity = pinfo->current_time;
1479 g_object_ref (source);
1484 /* must be called with the session lock, the returned source needs to be
1485 * unreffed after usage. */
1487 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1491 source = find_source (sess, ssrc);
1492 if (source == NULL) {
1493 /* make new internal Source and insert */
1494 source = rtp_source_new (ssrc);
1496 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1498 source->validated = TRUE;
1499 source->internal = TRUE;
1500 source->probation = FALSE;
1501 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1502 rtp_source_set_callbacks (source, &callbacks, sess);
1504 add_source (sess, source);
1509 g_object_ref (source);
1515 * rtp_session_suggest_ssrc:
1516 * @sess: a #RTPSession
1518 * Suggest an unused SSRC in @sess.
1520 * Returns: a free unused SSRC
1523 rtp_session_suggest_ssrc (RTPSession * sess)
1527 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1529 RTP_SESSION_LOCK (sess);
1530 result = sess->suggested_ssrc;
1531 RTP_SESSION_UNLOCK (sess);
1537 * rtp_session_add_source:
1538 * @sess: a #RTPSession
1539 * @src: #RTPSource to add
1541 * Add @src to @session.
1543 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1544 * existed in the session.
1547 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1549 gboolean result = FALSE;
1552 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1553 g_return_val_if_fail (src != NULL, FALSE);
1555 RTP_SESSION_LOCK (sess);
1556 find = find_source (sess, src->ssrc);
1558 add_source (sess, src);
1561 RTP_SESSION_UNLOCK (sess);
1567 * rtp_session_get_num_sources:
1568 * @sess: an #RTPSession
1570 * Get the number of sources in @sess.
1572 * Returns: The number of sources in @sess.
1575 rtp_session_get_num_sources (RTPSession * sess)
1579 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1581 RTP_SESSION_LOCK (sess);
1582 result = sess->total_sources;
1583 RTP_SESSION_UNLOCK (sess);
1589 * rtp_session_get_num_active_sources:
1590 * @sess: an #RTPSession
1592 * Get the number of active sources in @sess. A source is considered active when
1593 * it has been validated and has not yet received a BYE RTCP message.
1595 * Returns: The number of active sources in @sess.
1598 rtp_session_get_num_active_sources (RTPSession * sess)
1602 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1604 RTP_SESSION_LOCK (sess);
1605 result = sess->stats.active_sources;
1606 RTP_SESSION_UNLOCK (sess);
1612 * rtp_session_get_source_by_ssrc:
1613 * @sess: an #RTPSession
1616 * Find the source with @ssrc in @sess.
1618 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1619 * g_object_unref() after usage.
1622 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1626 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1628 RTP_SESSION_LOCK (sess);
1629 result = find_source (sess, ssrc);
1631 g_object_ref (result);
1632 RTP_SESSION_UNLOCK (sess);
1637 /* should be called with the SESSION lock */
1639 rtp_session_create_new_ssrc (RTPSession * sess)
1644 ssrc = g_random_int ();
1646 /* see if it exists in the session, we're done if it doesn't */
1647 if (find_source (sess, ssrc) == NULL)
1655 * rtp_session_create_source:
1656 * @sess: an #RTPSession
1658 * Create an #RTPSource for use in @sess. This function will create a source
1659 * with an ssrc that is currently not used by any participants in the session.
1661 * Returns: an #RTPSource.
1664 rtp_session_create_source (RTPSession * sess)
1669 RTP_SESSION_LOCK (sess);
1670 ssrc = rtp_session_create_new_ssrc (sess);
1671 source = rtp_source_new (ssrc);
1672 rtp_source_set_callbacks (source, &callbacks, sess);
1673 /* we need an additional ref for the source in the hashtable */
1674 g_object_ref (source);
1675 add_source (sess, source);
1676 RTP_SESSION_UNLOCK (sess);
1682 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1684 GstNetAddressMeta *meta;
1686 /* get packet size including header overhead */
1687 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1691 GstRTPBuffer rtp = { NULL };
1693 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1694 goto invalid_packet;
1696 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1700 /* only keep info for first buffer */
1701 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1702 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1703 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1704 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1705 /* copy available csrc */
1706 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1707 for (i = 0; i < pinfo->csrc_count; i++)
1708 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1710 gst_rtp_buffer_unmap (&rtp);
1714 /* for netbuffer we can store the IP address to check for collisions */
1715 meta = gst_buffer_get_net_address_meta (*buffer);
1717 g_object_unref (pinfo->address);
1719 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1721 pinfo->address = NULL;
1729 GST_DEBUG ("invalid RTP packet received");
1734 /* update the RTPPacketInfo structure with the current time and other bits
1735 * about the current buffer we are handling.
1736 * This function is typically called when a validated packet is received.
1737 * This function should be called with the SESSION_LOCK
1740 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1741 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1742 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1748 pinfo->is_list = is_list;
1750 pinfo->current_time = current_time;
1751 pinfo->running_time = running_time;
1752 pinfo->ntpnstime = ntpnstime;
1753 pinfo->header_len = sess->header_len;
1755 pinfo->payload_len = 0;
1759 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1761 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1764 GstBuffer *buffer = GST_BUFFER_CAST (data);
1765 res = update_packet (&buffer, 0, pinfo);
1771 clean_packet_info (RTPPacketInfo * pinfo)
1774 g_object_unref (pinfo->address);
1776 gst_mini_object_unref (pinfo->data);
1782 source_update_active (RTPSession * sess, RTPSource * source,
1783 gboolean prevactive)
1785 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1786 guint32 ssrc = source->ssrc;
1788 if (prevactive == active)
1792 sess->stats.active_sources++;
1793 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1794 sess->stats.active_sources);
1796 sess->stats.active_sources--;
1797 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1798 sess->stats.active_sources);
1804 source_update_sender (RTPSession * sess, RTPSource * source,
1805 gboolean prevsender)
1807 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1808 guint32 ssrc = source->ssrc;
1810 if (prevsender == sender)
1814 sess->stats.sender_sources++;
1815 if (source->internal)
1816 sess->stats.internal_sender_sources++;
1817 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1818 sess->stats.sender_sources);
1820 sess->stats.sender_sources--;
1821 if (source->internal)
1822 sess->stats.internal_sender_sources--;
1823 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1824 sess->stats.sender_sources);
1830 * rtp_session_process_rtp:
1831 * @sess: and #RTPSession
1832 * @buffer: an RTP buffer
1833 * @current_time: the current system time
1834 * @running_time: the running_time of @buffer
1836 * Process an RTP buffer in the session manager. This function takes ownership
1839 * Returns: a #GstFlowReturn.
1842 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1843 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1845 GstFlowReturn result;
1849 gboolean prevsender, prevactive;
1850 RTPPacketInfo pinfo = { 0, };
1853 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1854 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1856 RTP_SESSION_LOCK (sess);
1858 /* update pinfo stats */
1859 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1860 current_time, running_time, ntpnstime)) {
1861 GST_DEBUG ("invalid RTP packet received");
1862 RTP_SESSION_UNLOCK (sess);
1863 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1868 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1872 prevsender = RTP_SOURCE_IS_SENDER (source);
1873 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1874 oldrate = source->bitrate;
1876 /* let source process the packet */
1877 result = rtp_source_process_rtp (source, &pinfo);
1879 /* source became active */
1880 if (source_update_active (sess, source, prevactive))
1881 on_ssrc_validated (sess, source);
1883 source_update_sender (sess, source, prevsender);
1885 if (oldrate != source->bitrate)
1886 sess->recalc_bandwidth = TRUE;
1889 on_new_ssrc (sess, source);
1891 if (source->validated) {
1895 /* for validated sources, we add the CSRCs as well */
1896 for (i = 0; i < pinfo.csrc_count; i++) {
1898 RTPSource *csrc_src;
1900 csrc = pinfo.csrcs[i];
1903 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
1908 GST_DEBUG ("created new CSRC: %08x", csrc);
1909 rtp_source_set_as_csrc (csrc_src);
1910 source_update_active (sess, csrc_src, FALSE);
1911 on_new_ssrc (sess, csrc_src);
1913 g_object_unref (csrc_src);
1916 g_object_unref (source);
1918 RTP_SESSION_UNLOCK (sess);
1920 clean_packet_info (&pinfo);
1927 RTP_SESSION_UNLOCK (sess);
1928 clean_packet_info (&pinfo);
1929 GST_DEBUG ("ignoring packet because its collisioning");
1935 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1936 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
1940 count = gst_rtcp_packet_get_rb_count (packet);
1941 for (i = 0; i < count; i++) {
1942 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1943 guint8 fractionlost;
1947 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1948 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1950 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1952 /* find our own source */
1953 src = find_source (sess, ssrc);
1957 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
1958 /* only deal with report blocks for our session, we update the stats of
1959 * the sender of the RTCP message. We could also compare our stats against
1960 * the other sender to see if we are better or worse. */
1961 /* FIXME, need to keep track who the RB block is from */
1962 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
1963 packetslost, exthighestseq, jitter, lsr, dlsr);
1966 on_ssrc_active (sess, source);
1969 /* A Sender report contains statistics about how the sender is doing. This
1970 * includes timing informataion such as the relation between RTP and NTP
1971 * timestamps and the number of packets/bytes it sent to us.
1973 * In this report is also included a set of report blocks related to how this
1974 * sender is receiving data (in case we (or somebody else) is also sending stuff
1975 * to it). This info includes the packet loss, jitter and seqnum. It also
1976 * contains information to calculate the round trip time (LSR/DLSR).
1979 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1980 RTPPacketInfo * pinfo, gboolean * do_sync)
1982 guint32 senderssrc, rtptime, packet_count, octet_count;
1985 gboolean created, prevsender;
1987 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1988 &packet_count, &octet_count);
1990 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1991 senderssrc, GST_TIME_ARGS (pinfo->current_time));
1993 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
1997 /* skip non-bye packets for sources that are marked BYE */
1998 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2001 /* don't try to do lip-sync for sources that sent a BYE */
2002 if (RTP_SOURCE_IS_MARKED_BYE (source))
2007 prevsender = RTP_SOURCE_IS_SENDER (source);
2009 /* first update the source */
2010 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2011 packet_count, octet_count);
2013 source_update_sender (sess, source, prevsender);
2016 on_new_ssrc (sess, source);
2018 rtp_session_process_rb (sess, source, packet, pinfo);
2021 g_object_unref (source);
2024 /* A receiver report contains statistics about how a receiver is doing. It
2025 * includes stuff like packet loss, jitter and the seqnum it received last. It
2026 * also contains info to calculate the round trip time.
2028 * We are only interested in how the sender of this report is doing wrt to us.
2031 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2032 RTPPacketInfo * pinfo)
2038 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2040 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2042 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2046 /* skip non-bye packets for sources that are marked BYE */
2047 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2051 on_new_ssrc (sess, source);
2053 rtp_session_process_rb (sess, source, packet, pinfo);
2056 g_object_unref (source);
2059 /* Get SDES items and store them in the SSRC */
2061 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2062 RTPPacketInfo * pinfo)
2065 gboolean more_items, more_entries;
2067 items = gst_rtcp_packet_sdes_get_item_count (packet);
2068 GST_DEBUG ("got SDES packet with %d items", items);
2070 more_items = gst_rtcp_packet_sdes_first_item (packet);
2072 while (more_items) {
2074 gboolean changed, created, prevactive;
2078 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2080 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2084 /* find src, no probation when dealing with RTCP */
2085 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2089 /* skip non-bye packets for sources that are marked BYE */
2090 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2093 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2095 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2097 while (more_entries) {
2098 GstRTCPSDESType type;
2104 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2106 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2109 if (type == GST_RTCP_SDES_PRIV) {
2110 name = g_strndup ((const gchar *) &data[1], data[0]);
2112 data += data[0] + 1;
2114 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2117 value = g_strndup ((const gchar *) data, len);
2119 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2124 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2128 /* takes ownership of sdes */
2129 changed = rtp_source_set_sdes_struct (source, sdes);
2131 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2132 source->validated = TRUE;
2135 on_new_ssrc (sess, source);
2137 /* source became active */
2138 if (source_update_active (sess, source, prevactive))
2139 on_ssrc_validated (sess, source);
2142 on_ssrc_sdes (sess, source);
2145 g_object_unref (source);
2147 more_items = gst_rtcp_packet_sdes_next_item (packet);
2152 /* BYE is sent when a client leaves the session
2155 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2156 RTPPacketInfo * pinfo)
2160 gboolean reconsider = FALSE;
2162 reason = gst_rtcp_packet_bye_get_reason (packet);
2163 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2165 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2166 for (i = 0; i < count; i++) {
2169 gboolean created, prevactive, prevsender;
2170 guint pmembers, members;
2172 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2173 GST_DEBUG ("SSRC: %08x", ssrc);
2175 /* find src and mark bye, no probation when dealing with RTCP */
2176 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2180 if (source->internal) {
2181 /* our own source, something weird with this packet */
2182 g_object_unref (source);
2186 /* store time for when we need to time out this source */
2187 source->bye_time = pinfo->current_time;
2189 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2190 prevsender = RTP_SOURCE_IS_SENDER (source);
2192 /* mark the source BYE */
2193 rtp_source_mark_bye (source, reason);
2195 pmembers = sess->stats.active_sources;
2197 source_update_active (sess, source, prevactive);
2198 source_update_sender (sess, source, prevsender);
2200 members = sess->stats.active_sources;
2202 if (!sess->scheduled_bye && members < pmembers) {
2203 /* some members went away since the previous timeout estimate.
2204 * Perform reverse reconsideration but only when we are not scheduling a
2206 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2207 pinfo->current_time < sess->next_rtcp_check_time) {
2208 GstClockTime time_remaining;
2210 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2211 sess->next_rtcp_check_time =
2212 gst_util_uint64_scale (time_remaining, members, pmembers);
2214 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2215 GST_TIME_ARGS (sess->next_rtcp_check_time));
2217 sess->next_rtcp_check_time += pinfo->current_time;
2219 /* mark pending reconsider. We only want to signal the reconsideration
2220 * once after we handled all the source in the bye packet */
2226 on_new_ssrc (sess, source);
2228 on_bye_ssrc (sess, source);
2230 g_object_unref (source);
2233 RTP_SESSION_UNLOCK (sess);
2234 /* notify app of reconsideration */
2235 if (sess->callbacks.reconsider)
2236 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2237 RTP_SESSION_LOCK (sess);
2243 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2244 RTPPacketInfo * pinfo)
2246 GST_DEBUG ("received APP");
2250 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2251 gboolean fir, GstClockTime current_time)
2253 guint32 round_trip = 0;
2255 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2257 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2258 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2261 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2262 GST_DEBUG ("Ignoring %s request because one was send without one "
2263 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2264 fir ? "FIR" : "PLI",
2265 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2266 GST_TIME_ARGS (round_trip_in_ns));;
2271 sess->last_keyframe_request = current_time;
2273 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2274 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2275 sess->callbacks.request_key_unit);
2277 RTP_SESSION_UNLOCK (sess);
2278 sess->callbacks.request_key_unit (sess, fir,
2279 sess->request_key_unit_user_data);
2280 RTP_SESSION_LOCK (sess);
2286 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2287 guint32 media_ssrc, GstClockTime current_time)
2291 if (!sess->callbacks.request_key_unit)
2294 src = find_source (sess, sender_ssrc);
2298 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2302 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2303 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2308 gboolean our_request = FALSE;
2310 if (!sess->callbacks.request_key_unit)
2316 src = find_source (sess, sender_ssrc);
2318 /* Hack because Google fails to set the sender_ssrc correctly */
2319 if (!src && sender_ssrc == 1) {
2320 GHashTableIter iter;
2322 /* we can't find the source if there are multiple */
2323 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2326 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2327 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2328 if (!src->internal && rtp_source_is_sender (src))
2336 for (position = 0; position < fci_length; position += 8) {
2337 guint8 *data = fci_data + position;
2340 ssrc = GST_READ_UINT32_BE (data);
2342 own = find_source (sess, ssrc);
2343 if (own->internal) {
2351 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2355 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2356 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2357 GstClockTime current_time)
2359 sess->stats.nacks_received++;
2361 if (!sess->callbacks.notify_nack)
2364 while (fci_length > 0) {
2365 guint16 seqnum, blp;
2367 seqnum = GST_READ_UINT16_BE (fci_data);
2368 blp = GST_READ_UINT16_BE (fci_data + 2);
2370 GST_DEBUG ("NACK #%u, blp %04x", seqnum, blp);
2372 RTP_SESSION_UNLOCK (sess);
2373 sess->callbacks.notify_nack (sess, seqnum, blp,
2374 sess->notify_nack_user_data);
2375 RTP_SESSION_LOCK (sess);
2383 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2384 RTPPacketInfo * pinfo, GstClockTime current_time)
2386 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2387 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2388 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2389 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2390 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2391 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2394 src = find_source (sess, media_ssrc);
2396 /* skip non-bye packets for sources that are marked BYE */
2397 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2400 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2401 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2403 if (g_signal_has_handler_pending (sess,
2404 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2405 GstBuffer *fci_buffer = NULL;
2407 if (fci_length > 0) {
2408 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2409 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2411 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2414 RTP_SESSION_UNLOCK (sess);
2415 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2416 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2417 RTP_SESSION_LOCK (sess);
2420 gst_buffer_unref (fci_buffer);
2423 if (src && sess->rtcp_feedback_retention_window) {
2424 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2427 if ((src && src->internal) ||
2428 /* PSFB FIR puts the media ssrc inside the FCI */
2429 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2431 case GST_RTCP_TYPE_PSFB:
2433 case GST_RTCP_PSFB_TYPE_PLI:
2434 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2437 case GST_RTCP_PSFB_TYPE_FIR:
2438 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2445 case GST_RTCP_TYPE_RTPFB:
2447 case GST_RTCP_RTPFB_TYPE_NACK:
2448 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2449 fci_data, fci_length, current_time);
2461 * rtp_session_process_rtcp:
2462 * @sess: and #RTPSession
2463 * @buffer: an RTCP buffer
2464 * @current_time: the current system time
2465 * @ntpnstime: the current NTP time in nanoseconds
2467 * Process an RTCP buffer in the session manager. This function takes ownership
2470 * Returns: a #GstFlowReturn.
2473 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2474 GstClockTime current_time, guint64 ntpnstime)
2476 GstRTCPPacket packet;
2477 gboolean more, is_bye = FALSE, do_sync = FALSE;
2478 RTPPacketInfo pinfo = { 0, };
2479 GstFlowReturn result = GST_FLOW_OK;
2480 GstRTCPBuffer rtcp = { NULL, };
2482 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2483 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2485 if (!gst_rtcp_buffer_validate (buffer))
2486 goto invalid_packet;
2488 GST_DEBUG ("received RTCP packet");
2490 RTP_SESSION_LOCK (sess);
2491 /* update pinfo stats */
2492 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2495 /* start processing the compound packet */
2496 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2497 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2501 type = gst_rtcp_packet_get_type (&packet);
2504 case GST_RTCP_TYPE_SR:
2505 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2507 case GST_RTCP_TYPE_RR:
2508 rtp_session_process_rr (sess, &packet, &pinfo);
2510 case GST_RTCP_TYPE_SDES:
2511 rtp_session_process_sdes (sess, &packet, &pinfo);
2513 case GST_RTCP_TYPE_BYE:
2515 /* don't try to attempt lip-sync anymore for streams with a BYE */
2517 rtp_session_process_bye (sess, &packet, &pinfo);
2519 case GST_RTCP_TYPE_APP:
2520 rtp_session_process_app (sess, &packet, &pinfo);
2522 case GST_RTCP_TYPE_RTPFB:
2523 case GST_RTCP_TYPE_PSFB:
2524 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2527 GST_WARNING ("got unknown RTCP packet");
2530 more = gst_rtcp_packet_move_to_next (&packet);
2533 gst_rtcp_buffer_unmap (&rtcp);
2535 /* if we are scheduling a BYE, we only want to count bye packets, else we
2536 * count everything */
2537 if (sess->scheduled_bye && is_bye) {
2538 sess->bye_stats.bye_members++;
2539 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2542 /* keep track of average packet size */
2543 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2545 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2546 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2547 RTP_SESSION_UNLOCK (sess);
2550 clean_packet_info (&pinfo);
2552 /* notify caller of sr packets in the callback */
2553 if (do_sync && sess->callbacks.sync_rtcp) {
2554 result = sess->callbacks.sync_rtcp (sess, buffer,
2555 sess->sync_rtcp_user_data);
2557 gst_buffer_unref (buffer);
2564 GST_DEBUG ("invalid RTCP packet received");
2565 gst_buffer_unref (buffer);
2571 * rtp_session_update_send_caps:
2572 * @sess: an #RTPSession
2575 * Update the caps of the sender in the rtp session.
2578 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2583 g_return_if_fail (RTP_IS_SESSION (sess));
2584 g_return_if_fail (GST_IS_CAPS (caps));
2586 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2588 s = gst_caps_get_structure (caps, 0);
2590 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2594 RTP_SESSION_LOCK (sess);
2595 source = obtain_internal_source (sess, ssrc, &created);
2597 rtp_source_update_caps (source, caps);
2598 g_object_unref (source);
2600 RTP_SESSION_UNLOCK (sess);
2605 * rtp_session_send_rtp:
2606 * @sess: an #RTPSession
2607 * @data: pointer to either an RTP buffer or a list of RTP buffers
2608 * @is_list: TRUE when @data is a buffer list
2609 * @current_time: the current system time
2610 * @running_time: the running time of @data
2612 * Send the RTP buffer in the session manager. This function takes ownership of
2615 * Returns: a #GstFlowReturn.
2618 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2619 GstClockTime current_time, GstClockTime running_time)
2621 GstFlowReturn result;
2623 gboolean prevsender;
2625 RTPPacketInfo pinfo = { 0, };
2628 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2629 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2631 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2633 RTP_SESSION_LOCK (sess);
2634 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2635 current_time, running_time, -1))
2636 goto invalid_packet;
2638 source = obtain_internal_source (sess, pinfo.ssrc, &created);
2640 /* update last activity */
2641 source->last_rtp_activity = current_time;
2643 prevsender = RTP_SOURCE_IS_SENDER (source);
2644 oldrate = source->bitrate;
2646 /* we use our own source to send */
2647 result = rtp_source_send_rtp (source, &pinfo);
2649 source_update_sender (sess, source, prevsender);
2651 if (oldrate != source->bitrate)
2652 sess->recalc_bandwidth = TRUE;
2653 RTP_SESSION_UNLOCK (sess);
2655 g_object_unref (source);
2656 clean_packet_info (&pinfo);
2662 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2663 RTP_SESSION_UNLOCK (sess);
2664 GST_DEBUG ("invalid RTP packet received");
2670 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2672 *bandwidth += source->bitrate;
2675 /* must be called with session lock */
2677 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2680 GstClockTime result;
2681 RTPSessionStats *stats;
2683 /* recalculate bandwidth when it changed */
2684 if (sess->recalc_bandwidth) {
2687 if (sess->bandwidth > 0)
2688 bandwidth = sess->bandwidth;
2690 /* If it is <= 0, then try to estimate the actual bandwidth */
2693 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2694 (GHFunc) add_bitrates, &bandwidth);
2697 if (bandwidth < 8000)
2698 bandwidth = RTP_STATS_BANDWIDTH;
2700 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2701 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2703 sess->recalc_bandwidth = FALSE;
2706 if (sess->scheduled_bye) {
2707 stats = &sess->bye_stats;
2708 result = rtp_stats_calculate_bye_interval (stats);
2710 stats = &sess->stats;
2711 result = rtp_stats_calculate_rtcp_interval (stats,
2712 stats->internal_sender_sources > 0, first);
2715 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2716 GST_TIME_ARGS (result), first);
2718 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2719 result = rtp_stats_add_rtcp_jitter (stats, result);
2721 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2727 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2729 if (source->internal)
2730 rtp_source_mark_bye (source, reason);
2734 * rtp_session_mark_all_bye:
2735 * @sess: an #RTPSession
2738 * Mark all internal sources of the session as BYE with @reason.
2741 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2743 g_return_if_fail (RTP_IS_SESSION (sess));
2745 RTP_SESSION_LOCK (sess);
2746 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2747 (GHFunc) source_mark_bye, (gpointer) reason);
2748 RTP_SESSION_UNLOCK (sess);
2751 /* Stop the current @sess and schedule a BYE message for the other members.
2752 * One must have the session lock to call this function
2754 static GstFlowReturn
2755 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2757 GstFlowReturn result = GST_FLOW_OK;
2758 GstClockTime interval;
2760 /* nothing to do it we already scheduled bye */
2761 if (sess->scheduled_bye)
2764 /* we schedule BYE now */
2765 sess->scheduled_bye = TRUE;
2766 /* at least one member wants to send a BYE */
2767 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2768 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2769 sess->bye_stats.bye_members = 1;
2770 sess->first_rtcp = TRUE;
2771 sess->allow_early = TRUE;
2773 /* reschedule transmission */
2774 sess->last_rtcp_send_time = current_time;
2775 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2777 if (interval != GST_CLOCK_TIME_NONE)
2778 sess->next_rtcp_check_time = current_time + interval;
2780 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2782 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2783 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2785 RTP_SESSION_UNLOCK (sess);
2786 /* notify app of reconsideration */
2787 if (sess->callbacks.reconsider)
2788 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2789 RTP_SESSION_LOCK (sess);
2796 * rtp_session_schedule_bye:
2797 * @sess: an #RTPSession
2798 * @current_time: the current system time
2800 * Schedule a BYE message for all sources marked as BYE in @sess.
2802 * Returns: a #GstFlowReturn.
2805 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2807 GstFlowReturn result = GST_FLOW_OK;
2809 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2811 RTP_SESSION_LOCK (sess);
2812 result = rtp_session_schedule_bye_locked (sess, current_time);
2813 RTP_SESSION_UNLOCK (sess);
2819 * rtp_session_next_timeout:
2820 * @sess: an #RTPSession
2821 * @current_time: the current system time
2823 * Get the next time we should perform session maintenance tasks.
2825 * Returns: a time when rtp_session_on_timeout() should be called with the
2826 * current system time.
2829 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2831 GstClockTime result, interval = 0;
2833 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2835 RTP_SESSION_LOCK (sess);
2837 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2838 GST_DEBUG ("have early rtcp time");
2839 result = sess->next_early_rtcp_time;
2843 result = sess->next_rtcp_check_time;
2845 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2846 ", next time: %" GST_TIME_FORMAT,
2847 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2849 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2850 GST_DEBUG ("take current time as base");
2851 /* our previous check time expired, start counting from the current time
2853 result = current_time;
2856 if (sess->scheduled_bye) {
2857 if (sess->bye_stats.active_sources >= 50) {
2858 GST_DEBUG ("reconsider BYE, more than 50 sources");
2859 /* reconsider BYE if members >= 50 */
2860 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2863 if (sess->first_rtcp) {
2864 GST_DEBUG ("first RTCP packet");
2865 /* we are called for the first time */
2866 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2867 } else if (sess->next_rtcp_check_time < current_time) {
2868 GST_DEBUG ("old check time expired, getting new timeout");
2869 /* get a new timeout when we need to */
2870 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2874 if (interval != GST_CLOCK_TIME_NONE)
2877 result = GST_CLOCK_TIME_NONE;
2879 sess->next_rtcp_check_time = result;
2883 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2884 ", next time: %" GST_TIME_FORMAT,
2885 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2886 RTP_SESSION_UNLOCK (sess);
2900 GstRTCPBuffer rtcpbuf;
2903 guint num_to_report;
2908 GstClockTime current_time;
2910 GstClockTime running_time;
2911 GstClockTime interval;
2912 GstRTCPPacket packet;
2915 gboolean may_suppress;
2917 guint nacked_seqnums;
2921 session_start_rtcp (RTPSession * sess, ReportData * data)
2923 GstRTCPPacket *packet = &data->packet;
2924 RTPSource *own = data->source;
2925 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2927 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2928 data->has_sdes = FALSE;
2930 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2932 if (RTP_SOURCE_IS_SENDER (own)) {
2935 guint32 packet_count, octet_count;
2937 /* we are a sender, create SR */
2938 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2939 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2941 /* get latest stats */
2942 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2943 &ntptime, &rtptime, &packet_count, &octet_count);
2945 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2946 packet_count, octet_count);
2948 /* fill in sender report info */
2949 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2950 ntptime, rtptime, packet_count, octet_count);
2952 /* we are only receiver, create RR */
2953 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2954 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2955 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2959 /* construct a Sender or Receiver Report */
2961 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2963 RTPSession *sess = data->sess;
2964 GstRTCPPacket *packet = &data->packet;
2965 guint8 fractionlost;
2967 guint32 exthighestseq, jitter;
2970 /* don't report for sources in future generations */
2971 if (((gint16) (source->generation - sess->generation)) > 0) {
2972 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
2973 source->generation, sess->generation);
2977 if (g_hash_table_contains (source->reported_in_sr_of,
2978 GUINT_TO_POINTER (data->source->ssrc))) {
2979 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
2983 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
2984 GST_DEBUG ("max RB count reached");
2988 /* only report about other sender */
2989 if (source == data->source)
2992 if (!RTP_SOURCE_IS_SENDER (source)) {
2993 GST_DEBUG ("source %08x not sender", source->ssrc);
2997 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3000 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3001 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3003 /* store last generated RR packet */
3004 source->last_rr.is_valid = TRUE;
3005 source->last_rr.fractionlost = fractionlost;
3006 source->last_rr.packetslost = packetslost;
3007 source->last_rr.exthighestseq = exthighestseq;
3008 source->last_rr.jitter = jitter;
3009 source->last_rr.lsr = lsr;
3010 source->last_rr.dlsr = dlsr;
3012 /* packet is not yet filled, add report block for this source. */
3013 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3014 exthighestseq, jitter, lsr, dlsr);
3017 g_hash_table_add (source->reported_in_sr_of,
3018 GUINT_TO_POINTER (data->source->ssrc));
3023 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3025 GstRTCPPacket *packet = &data->packet;
3029 if (!source->send_fir)
3032 len = gst_rtcp_packet_fb_get_fci_length (packet);
3033 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3034 /* exit because the packet is full, will put next request in a
3038 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3040 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3042 fci_data[0] = source->current_send_fir_seqnum;
3043 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3045 source->send_fir = FALSE;
3049 session_fir (RTPSession * sess, ReportData * data)
3051 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3052 GstRTCPPacket *packet = &data->packet;
3054 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3057 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3058 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3059 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3061 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3062 (GHFunc) session_add_fir, data);
3064 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3065 gst_rtcp_packet_remove (packet);
3067 data->may_suppress = FALSE;
3071 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3073 GstRTCPPacket packet;
3074 GstRTCPBuffer rtcp = { NULL, };
3075 gboolean ret = FALSE;
3077 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3079 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3080 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3081 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3085 gst_rtcp_buffer_unmap (&rtcp);
3092 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3094 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3095 GstRTCPPacket *packet = &data->packet;
3097 if (!source->send_pli)
3100 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3103 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3104 /* exit because the packet is full, will put next request in a
3108 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3109 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3110 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3112 source->send_pli = FALSE;
3113 data->may_suppress = FALSE;
3116 /* construct NACK */
3118 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3120 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3121 GstRTCPPacket *packet = &data->packet;
3126 if (!source->send_nack)
3129 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3130 /* exit because the packet is full, will put next request in a
3134 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3135 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3136 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3138 nacks = rtp_source_get_nacks (source, &n_nacks);
3139 GST_DEBUG ("%u NACKs", n_nacks);
3140 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3143 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3144 for (i = 0; i < n_nacks; i++) {
3145 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3147 data->nacked_seqnums++;
3150 rtp_source_clear_nacks (source);
3151 data->may_suppress = FALSE;
3154 /* perform cleanup of sources that timed out */
3156 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3158 gboolean remove = FALSE;
3159 gboolean byetimeout = FALSE;
3160 gboolean sendertimeout = FALSE;
3161 gboolean is_sender, is_active;
3162 RTPSession *sess = data->sess;
3163 GstClockTime interval, binterval;
3166 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3168 /* check for outdated collisions */
3169 if (source->internal) {
3170 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3171 rtp_source_timeout (source, data->current_time,
3172 /* "a relatively long time" -- RFC 3550 section 8.2 */
3173 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
3174 data->running_time - sess->rtcp_feedback_retention_window);
3177 /* nothing else to do when without RTCP */
3178 if (data->interval == GST_CLOCK_TIME_NONE)
3181 is_sender = RTP_SOURCE_IS_SENDER (source);
3182 is_active = RTP_SOURCE_IS_ACTIVE (source);
3184 /* our own rtcp interval may have been forced low by secondary configuration,
3185 * while sender side may still operate with higher interval,
3186 * so do not just take our interval to decide on timing out sender,
3187 * but take (if data->interval <= 5 * GST_SECOND):
3188 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3189 * where sender_interval is difference between last 2 received RTCP reports
3191 if (data->interval >= 5 * GST_SECOND || source->internal) {
3192 binterval = data->interval;
3194 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3195 GST_TIME_ARGS (source->stats.prev_rtcptime),
3196 GST_TIME_ARGS (source->stats.last_rtcptime));
3197 /* if not received enough yet, fallback to larger default */
3198 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3199 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3201 binterval = 5 * GST_SECOND;
3202 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3204 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3205 GST_TIME_ARGS (binterval));
3207 if (!source->internal) {
3208 if (source->marked_bye) {
3209 /* if we received a BYE from the source, remove the source after some
3211 if (data->current_time > source->bye_time &&
3212 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3213 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3218 /* sources that were inactive for more than 5 times the deterministic reporting
3219 * interval get timed out. the min timeout is 5 seconds. */
3220 /* mind old time that might pre-date last time going to PLAYING */
3221 btime = MAX (source->last_activity, sess->start_time);
3222 if (data->current_time > btime) {
3223 interval = MAX (binterval * 5, 5 * GST_SECOND);
3224 if (data->current_time - btime > interval) {
3225 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3226 source->ssrc, GST_TIME_ARGS (btime));
3232 /* senders that did not send for a long time become a receiver, this also
3233 * holds for our own sources. */
3235 /* mind old time that might pre-date last time going to PLAYING */
3236 btime = MAX (source->last_rtp_activity, sess->start_time);
3237 if (data->current_time > btime) {
3238 interval = MAX (binterval * 2, 5 * GST_SECOND);
3239 if (data->current_time - btime > interval) {
3240 if (source->internal && source->sent_bye) {
3241 /* an internal source is BYE and stopped sending RTP, remove */
3242 GST_DEBUG ("internal BYE source %08x timed out, last %"
3243 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3246 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3247 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3248 sendertimeout = TRUE;
3255 sess->total_sources--;
3257 sess->stats.sender_sources--;
3258 if (source->internal)
3259 sess->stats.internal_sender_sources--;
3262 sess->stats.active_sources--;
3264 if (source->internal)
3265 sess->stats.internal_sources--;
3268 on_bye_timeout (sess, source);
3270 on_timeout (sess, source);
3272 if (sendertimeout) {
3273 source->is_sender = FALSE;
3274 sess->stats.sender_sources--;
3275 if (source->internal)
3276 sess->stats.internal_sender_sources--;
3278 on_sender_timeout (sess, source);
3280 /* count how many source to report in this generation */
3281 if (((gint16) (source->generation - sess->generation)) <= 0)
3282 data->num_to_report++;
3284 source->closing = remove;
3288 session_sdes (RTPSession * sess, ReportData * data)
3290 GstRTCPPacket *packet = &data->packet;
3291 const GstStructure *sdes;
3293 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3295 /* add SDES packet */
3296 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3298 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3300 sdes = rtp_source_get_sdes_struct (data->source);
3302 /* add all fields in the structure, the order is not important. */
3303 n_fields = gst_structure_n_fields (sdes);
3304 for (i = 0; i < n_fields; ++i) {
3307 GstRTCPSDESType type;
3309 field = gst_structure_nth_field_name (sdes, i);
3312 value = gst_structure_get_string (sdes, field);
3315 type = gst_rtcp_sdes_name_to_type (field);
3317 /* Early packets are minimal and only include the CNAME */
3318 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3321 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3322 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3323 (const guint8 *) value);
3324 } else if (type == GST_RTCP_SDES_PRIV) {
3330 /* don't accept entries that are too big */
3331 prefix_len = strlen (field);
3332 if (prefix_len > 255)
3334 value_len = strlen (value);
3335 if (value_len > 255)
3337 data_len = 1 + prefix_len + value_len;
3341 data[0] = prefix_len;
3342 memcpy (&data[1], field, prefix_len);
3343 memcpy (&data[1 + prefix_len], value, value_len);
3345 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3349 data->has_sdes = TRUE;
3352 /* schedule a BYE packet */
3354 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3356 GstRTCPPacket *packet = &data->packet;
3357 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3360 session_sdes (sess, data);
3361 /* add a BYE packet */
3362 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3363 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3364 if (source->bye_reason)
3365 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3367 /* we have a BYE packet now */
3368 source->sent_bye = TRUE;
3372 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3374 GstClockTime new_send_time, elapsed;
3375 GstClockTime interval;
3376 RTPSessionStats *stats;
3378 if (sess->scheduled_bye)
3379 stats = &sess->bye_stats;
3381 stats = &sess->stats;
3383 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3384 data->is_early = TRUE;
3386 data->is_early = FALSE;
3388 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3389 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3390 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3391 GST_TIME_ARGS (current_time));
3395 /* no need to check yet */
3396 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3397 sess->next_rtcp_check_time > current_time) {
3398 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3399 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3400 GST_TIME_ARGS (current_time));
3405 /* get elapsed time since we last reported */
3406 elapsed = current_time - sess->last_rtcp_send_time;
3408 /* take interval and add jitter */
3409 interval = data->interval;
3410 if (interval != GST_CLOCK_TIME_NONE)
3411 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3413 /* perform forward reconsideration */
3414 if (interval != GST_CLOCK_TIME_NONE) {
3415 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3416 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3417 new_send_time = interval + sess->last_rtcp_send_time;
3419 new_send_time = sess->last_rtcp_send_time;
3422 if (!data->is_early) {
3423 /* check if reconsideration */
3424 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3425 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3426 GST_TIME_ARGS (new_send_time));
3427 /* store new check time */
3428 sess->next_rtcp_check_time = new_send_time;
3431 sess->next_rtcp_check_time = current_time + interval;
3432 } else if (interval != GST_CLOCK_TIME_NONE) {
3433 /* Apply the rules from RFC 4585 section 3.5.3 */
3434 if (stats->min_interval != 0 && !sess->first_rtcp) {
3435 GstClockTime T_rr_current_interval =
3436 g_random_double_range (0.5, 1.5) * stats->min_interval;
3438 /* This will caused the RTCP to be suppressed if no FB packets are added */
3439 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3440 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3441 " last: %" GST_TIME_FORMAT
3442 " + T_rr_current_interval: %" GST_TIME_FORMAT
3443 " > new_send_time: %" GST_TIME_FORMAT,
3444 GST_TIME_ARGS (stats->min_interval),
3445 GST_TIME_ARGS (sess->last_rtcp_send_time),
3446 GST_TIME_ARGS (T_rr_current_interval),
3447 GST_TIME_ARGS (new_send_time));
3448 data->may_suppress = TRUE;
3453 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3454 GST_TIME_ARGS (new_send_time));
3460 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3462 g_hash_table_insert (hash_table, key, g_object_ref (source));
3466 remove_closing_sources (const gchar * key, RTPSource * source,
3469 if (source->closing)
3472 if (source->send_fir)
3473 data->have_fir = TRUE;
3474 if (source->send_pli)
3475 data->have_pli = TRUE;
3476 if (source->send_nack)
3477 data->have_nack = TRUE;
3483 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3485 RTPSession *sess = data->sess;
3486 gboolean is_bye = FALSE;
3487 ReportOutput *output;
3489 /* only generate RTCP for active internal sources */
3490 if (!source->internal || source->sent_bye)
3493 /* ignore other sources when we do the timeout after a scheduled BYE */
3494 if (sess->scheduled_bye && !source->marked_bye)
3497 data->source = source;
3500 session_start_rtcp (sess, data);
3502 if (source->marked_bye) {
3504 make_source_bye (sess, source, data);
3506 } else if (!data->is_early) {
3507 /* loop over all known sources and add report blocks. If we are early, we
3508 * just make a minimal RTCP packet and skip this step */
3509 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3510 (GHFunc) session_report_blocks, data);
3512 if (!data->has_sdes)
3513 session_sdes (sess, data);
3516 session_fir (sess, data);
3519 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3520 (GHFunc) session_pli, data);
3522 if (data->have_nack)
3523 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3524 (GHFunc) session_nack, data);
3526 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3528 output = g_slice_new (ReportOutput);
3529 output->source = g_object_ref (source);
3530 output->is_bye = is_bye;
3531 output->buffer = data->rtcp;
3532 /* queue the RTCP packet to push later */
3533 g_queue_push_tail (&data->output, output);
3537 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3539 RTPSession *sess = data->sess;
3541 if (g_hash_table_size (source->reported_in_sr_of) >=
3542 sess->stats.internal_sources) {
3543 /* source is reported, move to next generation */
3544 source->generation = sess->generation + 1;
3545 g_hash_table_remove_all (source->reported_in_sr_of);
3547 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3548 source->generation);
3550 /* if we reported all sources in this generation, move to next */
3551 if (--data->num_to_report == 0) {
3553 GST_DEBUG ("all reported, generation now %u", sess->generation);
3559 * rtp_session_on_timeout:
3560 * @sess: an #RTPSession
3561 * @current_time: the current system time
3562 * @ntpnstime: the current NTP time in nanoseconds
3563 * @running_time: the current running_time of the pipeline
3565 * Perform maintenance actions after the timeout obtained with
3566 * rtp_session_next_timeout() expired.
3568 * This function will perform timeouts of receivers and senders, send a BYE
3569 * packet or generate RTCP packets with current session stats.
3571 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3572 * times, for each packet that should be processed.
3574 * Returns: a #GstFlowReturn.
3577 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3578 guint64 ntpnstime, GstClockTime running_time)
3580 GstFlowReturn result = GST_FLOW_OK;
3581 ReportData data = { GST_RTCP_BUFFER_INIT };
3582 GHashTable *table_copy;
3583 ReportOutput *output;
3585 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3587 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3588 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3589 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3592 data.current_time = current_time;
3593 data.ntpnstime = ntpnstime;
3594 data.running_time = running_time;
3595 data.num_to_report = 0;
3596 data.may_suppress = FALSE;
3597 data.nacked_seqnums = 0;
3598 g_queue_init (&data.output);
3600 RTP_SESSION_LOCK (sess);
3601 /* get a new interval, we need this for various cleanups etc */
3602 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3604 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3606 /* we need an internal source now */
3607 if (sess->stats.internal_sources == 0) {
3611 source = obtain_internal_source (sess, sess->suggested_ssrc, &created);
3612 g_object_unref (source);
3615 /* Make a local copy of the hashtable. We need to do this because the
3616 * cleanup stage below releases the session lock. */
3617 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3618 (GDestroyNotify) g_object_unref);
3619 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3620 (GHFunc) clone_ssrcs_hashtable, table_copy);
3622 /* Clean up the session, mark the source for removing, this might release the
3624 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3625 g_hash_table_destroy (table_copy);
3627 /* Now remove the marked sources */
3628 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3629 (GHRFunc) remove_closing_sources, &data);
3631 /* update point-to-point status */
3632 session_update_ptp (sess);
3634 /* see if we need to generate SR or RR packets */
3635 if (!is_rtcp_time (sess, current_time, &data))
3638 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3639 sess->generation, data.num_to_report, data.is_early);
3641 /* generate RTCP for all internal sources */
3642 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3643 (GHFunc) generate_rtcp, &data);
3645 /* update the generation for all the sources that have been reported */
3646 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3647 (GHFunc) update_generation, &data);
3649 /* we keep track of the last report time in order to timeout inactive
3650 * receivers or senders */
3651 if (!data.is_early && !data.may_suppress)
3652 sess->last_rtcp_send_time = data.current_time;
3653 sess->first_rtcp = FALSE;
3654 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3655 sess->scheduled_bye = FALSE;
3658 RTP_SESSION_UNLOCK (sess);
3660 /* push out the RTCP packets */
3661 while ((output = g_queue_pop_head (&data.output))) {
3662 gboolean do_not_suppress;
3663 GstBuffer *buffer = output->buffer;
3664 RTPSource *source = output->source;
3666 /* Give the user a change to add its own packet */
3667 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3668 buffer, data.is_early, &do_not_suppress);
3670 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3673 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3675 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3676 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3677 sess->stats.avg_rtcp_packet_size, packet_size);
3679 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3680 sess->send_rtcp_user_data);
3681 sess->stats.nacks_sent += data.nacked_seqnums;
3683 GST_DEBUG ("freeing packet callback: %p"
3684 " do_not_suppress: %d may_suppress: %d",
3685 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3686 sess->stats.nacks_dropped += data.nacked_seqnums;
3687 gst_buffer_unref (buffer);
3689 g_object_unref (source);
3690 g_slice_free (ReportOutput, output);
3696 * rtp_session_request_early_rtcp:
3697 * @sess: an #RTPSession
3698 * @current_time: the current system time
3699 * @max_delay: maximum delay
3701 * Request transmission of early RTCP
3704 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3705 GstClockTime max_delay)
3707 GstClockTime T_dither_max;
3709 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3711 RTP_SESSION_LOCK (sess);
3713 /* Check if already requested */
3714 /* RFC 4585 section 3.5.2 step 2 */
3715 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3716 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3720 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3721 GST_LOG_OBJECT (sess, "no next RTCP check time");
3725 /* Ignore the request a scheduled packet will be in time anyway */
3726 if (current_time + max_delay > sess->next_rtcp_check_time) {
3727 GST_LOG_OBJECT (sess, "next scheduled time is soon %" GST_TIME_FORMAT " + %"
3728 GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
3729 GST_TIME_ARGS (current_time),
3730 GST_TIME_ARGS (max_delay), GST_TIME_ARGS (sess->next_rtcp_check_time));
3734 /* RFC 4585 section 3.5.2 step 2b */
3735 /* If the total sources is <=2, then there is only us and one peer */
3736 /* When there is one auxiliary stream the session can still do point
3739 if (sess->is_doing_ptp) {
3742 /* Divide by 2 because l = 0.5 */
3743 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3747 /* RFC 4585 section 3.5.2 step 3 */
3748 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3749 GST_LOG_OBJECT (sess, "don't send because of dither");
3753 /* RFC 4585 section 3.5.2 step 4
3754 * Don't send if allow_early is FALSE, but not if we are in
3755 * immediate mode, meaning we are part of a group of at most the
3756 * application-specific threshold.
3758 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3759 sess->allow_early == FALSE) {
3760 GST_LOG_OBJECT (sess, "can't allow early feedback");
3765 /* Schedule an early transmission later */
3766 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3769 /* If no dithering, schedule it for NOW */
3770 sess->next_early_rtcp_time = current_time;
3773 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3774 GST_TIME_ARGS (sess->next_early_rtcp_time));
3775 RTP_SESSION_UNLOCK (sess);
3777 /* notify app of need to send packet early
3778 * and therefore of timeout change */
3779 if (sess->callbacks.reconsider)
3780 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3786 RTP_SESSION_UNLOCK (sess);
3790 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3794 if (!sess->callbacks.send_rtcp)
3797 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3799 rtp_session_request_early_rtcp (sess, now, max_delay);
3803 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3804 gboolean fir, gint count)
3808 RTP_SESSION_LOCK (sess);
3809 src = find_source (sess, ssrc);
3814 src->send_pli = FALSE;
3815 src->send_fir = TRUE;
3817 if (count == -1 || count != src->last_fir_count)
3818 src->current_send_fir_seqnum++;
3819 src->last_fir_count = count;
3820 } else if (!src->send_fir) {
3821 src->send_pli = TRUE;
3823 RTP_SESSION_UNLOCK (sess);
3825 rtp_session_send_rtcp (sess, 200 * GST_MSECOND);
3832 RTP_SESSION_UNLOCK (sess);
3838 * rtp_session_request_nack:
3839 * @sess: a #RTPSession
3841 * @seqnum: the missing seqnum
3842 * @max_delay: max delay to request NACK
3844 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
3846 * Returns: %TRUE if the NACK feedback could be scheduled
3849 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
3850 GstClockTime max_delay)
3854 RTP_SESSION_LOCK (sess);
3855 source = find_source (sess, ssrc);
3859 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
3860 rtp_source_register_nack (source, seqnum);
3861 RTP_SESSION_UNLOCK (sess);
3863 rtp_session_send_rtcp (sess, max_delay);
3870 RTP_SESSION_UNLOCK (sess);