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",
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 RTP_SESSION_LOCK (sess);
611 sess->suggested_ssrc = g_value_get_uint (value);
612 RTP_SESSION_UNLOCK (sess);
615 RTP_SESSION_LOCK (sess);
616 sess->bandwidth = g_value_get_double (value);
617 sess->recalc_bandwidth = TRUE;
618 RTP_SESSION_UNLOCK (sess);
620 case PROP_RTCP_FRACTION:
621 RTP_SESSION_LOCK (sess);
622 sess->rtcp_bandwidth = g_value_get_double (value);
623 sess->recalc_bandwidth = TRUE;
624 RTP_SESSION_UNLOCK (sess);
626 case PROP_RTCP_RR_BANDWIDTH:
627 RTP_SESSION_LOCK (sess);
628 sess->rtcp_rr_bandwidth = g_value_get_int (value);
629 sess->recalc_bandwidth = TRUE;
630 RTP_SESSION_UNLOCK (sess);
632 case PROP_RTCP_RS_BANDWIDTH:
633 RTP_SESSION_LOCK (sess);
634 sess->rtcp_rs_bandwidth = g_value_get_int (value);
635 sess->recalc_bandwidth = TRUE;
636 RTP_SESSION_UNLOCK (sess);
639 sess->mtu = g_value_get_uint (value);
642 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
645 sess->favor_new = g_value_get_boolean (value);
647 case PROP_RTCP_MIN_INTERVAL:
648 rtp_stats_set_min_interval (&sess->stats,
649 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
650 /* trigger reconsideration */
651 RTP_SESSION_LOCK (sess);
652 sess->next_rtcp_check_time = 0;
653 RTP_SESSION_UNLOCK (sess);
654 if (sess->callbacks.reconsider)
655 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
657 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
658 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
661 sess->probation = g_value_get_uint (value);
664 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
670 rtp_session_get_property (GObject * object, guint prop_id,
671 GValue * value, GParamSpec * pspec)
675 sess = RTP_SESSION (object);
678 case PROP_INTERNAL_SSRC:
679 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
681 case PROP_INTERNAL_SOURCE:
682 /* FIXME, return a random source */
683 g_value_set_object (value, NULL);
686 g_value_set_double (value, sess->bandwidth);
688 case PROP_RTCP_FRACTION:
689 g_value_set_double (value, sess->rtcp_bandwidth);
691 case PROP_RTCP_RR_BANDWIDTH:
692 g_value_set_int (value, sess->rtcp_rr_bandwidth);
694 case PROP_RTCP_RS_BANDWIDTH:
695 g_value_set_int (value, sess->rtcp_rs_bandwidth);
698 g_value_set_uint (value, sess->mtu);
701 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
703 case PROP_NUM_SOURCES:
704 g_value_set_uint (value, rtp_session_get_num_sources (sess));
706 case PROP_NUM_ACTIVE_SOURCES:
707 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
710 g_value_take_boxed (value, rtp_session_create_sources (sess));
713 g_value_set_boolean (value, sess->favor_new);
715 case PROP_RTCP_MIN_INTERVAL:
716 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
718 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
719 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
722 g_value_set_uint (value, sess->probation);
725 g_value_take_boxed (value, rtp_session_create_stats (sess));
728 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
734 on_new_ssrc (RTPSession * sess, RTPSource * source)
736 g_object_ref (source);
737 RTP_SESSION_UNLOCK (sess);
738 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
739 RTP_SESSION_LOCK (sess);
740 g_object_unref (source);
744 on_ssrc_collision (RTPSession * sess, RTPSource * source)
746 g_object_ref (source);
747 RTP_SESSION_UNLOCK (sess);
748 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
750 RTP_SESSION_LOCK (sess);
751 g_object_unref (source);
755 on_ssrc_validated (RTPSession * sess, RTPSource * source)
757 g_object_ref (source);
758 RTP_SESSION_UNLOCK (sess);
759 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
761 RTP_SESSION_LOCK (sess);
762 g_object_unref (source);
766 on_ssrc_active (RTPSession * sess, RTPSource * source)
768 g_object_ref (source);
769 RTP_SESSION_UNLOCK (sess);
770 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
771 RTP_SESSION_LOCK (sess);
772 g_object_unref (source);
776 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
778 g_object_ref (source);
779 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
780 RTP_SESSION_UNLOCK (sess);
781 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
782 RTP_SESSION_LOCK (sess);
783 g_object_unref (source);
787 on_bye_ssrc (RTPSession * sess, RTPSource * source)
789 g_object_ref (source);
790 RTP_SESSION_UNLOCK (sess);
791 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
792 RTP_SESSION_LOCK (sess);
793 g_object_unref (source);
797 on_bye_timeout (RTPSession * sess, RTPSource * source)
799 g_object_ref (source);
800 RTP_SESSION_UNLOCK (sess);
801 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
802 RTP_SESSION_LOCK (sess);
803 g_object_unref (source);
807 on_timeout (RTPSession * sess, RTPSource * source)
809 g_object_ref (source);
810 RTP_SESSION_UNLOCK (sess);
811 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
812 RTP_SESSION_LOCK (sess);
813 g_object_unref (source);
817 on_sender_timeout (RTPSession * sess, RTPSource * source)
819 g_object_ref (source);
820 RTP_SESSION_UNLOCK (sess);
821 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
823 RTP_SESSION_LOCK (sess);
824 g_object_unref (source);
830 * Create a new session object.
832 * Returns: a new #RTPSession. g_object_unref() after usage.
835 rtp_session_new (void)
839 sess = g_object_new (RTP_TYPE_SESSION, NULL);
845 * rtp_session_set_callbacks:
846 * @sess: an #RTPSession
847 * @callbacks: callbacks to configure
848 * @user_data: user data passed in the callbacks
850 * Configure a set of callbacks to be notified of actions.
853 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
856 g_return_if_fail (RTP_IS_SESSION (sess));
858 if (callbacks->process_rtp) {
859 sess->callbacks.process_rtp = callbacks->process_rtp;
860 sess->process_rtp_user_data = user_data;
862 if (callbacks->send_rtp) {
863 sess->callbacks.send_rtp = callbacks->send_rtp;
864 sess->send_rtp_user_data = user_data;
866 if (callbacks->send_rtcp) {
867 sess->callbacks.send_rtcp = callbacks->send_rtcp;
868 sess->send_rtcp_user_data = user_data;
870 if (callbacks->sync_rtcp) {
871 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
872 sess->sync_rtcp_user_data = user_data;
874 if (callbacks->clock_rate) {
875 sess->callbacks.clock_rate = callbacks->clock_rate;
876 sess->clock_rate_user_data = user_data;
878 if (callbacks->reconsider) {
879 sess->callbacks.reconsider = callbacks->reconsider;
880 sess->reconsider_user_data = user_data;
882 if (callbacks->request_key_unit) {
883 sess->callbacks.request_key_unit = callbacks->request_key_unit;
884 sess->request_key_unit_user_data = user_data;
886 if (callbacks->request_time) {
887 sess->callbacks.request_time = callbacks->request_time;
888 sess->request_time_user_data = user_data;
890 if (callbacks->notify_nack) {
891 sess->callbacks.notify_nack = callbacks->notify_nack;
892 sess->notify_nack_user_data = user_data;
897 * rtp_session_set_process_rtp_callback:
898 * @sess: an #RTPSession
899 * @callback: callback to set
900 * @user_data: user data passed in the callback
902 * Configure only the process_rtp callback to be notified of the process_rtp action.
905 rtp_session_set_process_rtp_callback (RTPSession * sess,
906 RTPSessionProcessRTP callback, gpointer user_data)
908 g_return_if_fail (RTP_IS_SESSION (sess));
910 sess->callbacks.process_rtp = callback;
911 sess->process_rtp_user_data = user_data;
915 * rtp_session_set_send_rtp_callback:
916 * @sess: an #RTPSession
917 * @callback: callback to set
918 * @user_data: user data passed in the callback
920 * Configure only the send_rtp callback to be notified of the send_rtp action.
923 rtp_session_set_send_rtp_callback (RTPSession * sess,
924 RTPSessionSendRTP callback, gpointer user_data)
926 g_return_if_fail (RTP_IS_SESSION (sess));
928 sess->callbacks.send_rtp = callback;
929 sess->send_rtp_user_data = user_data;
933 * rtp_session_set_send_rtcp_callback:
934 * @sess: an #RTPSession
935 * @callback: callback to set
936 * @user_data: user data passed in the callback
938 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
941 rtp_session_set_send_rtcp_callback (RTPSession * sess,
942 RTPSessionSendRTCP callback, gpointer user_data)
944 g_return_if_fail (RTP_IS_SESSION (sess));
946 sess->callbacks.send_rtcp = callback;
947 sess->send_rtcp_user_data = user_data;
951 * rtp_session_set_sync_rtcp_callback:
952 * @sess: an #RTPSession
953 * @callback: callback to set
954 * @user_data: user data passed in the callback
956 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
959 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
960 RTPSessionSyncRTCP callback, gpointer user_data)
962 g_return_if_fail (RTP_IS_SESSION (sess));
964 sess->callbacks.sync_rtcp = callback;
965 sess->sync_rtcp_user_data = user_data;
969 * rtp_session_set_clock_rate_callback:
970 * @sess: an #RTPSession
971 * @callback: callback to set
972 * @user_data: user data passed in the callback
974 * Configure only the clock_rate callback to be notified of the clock_rate action.
977 rtp_session_set_clock_rate_callback (RTPSession * sess,
978 RTPSessionClockRate callback, gpointer user_data)
980 g_return_if_fail (RTP_IS_SESSION (sess));
982 sess->callbacks.clock_rate = callback;
983 sess->clock_rate_user_data = user_data;
987 * rtp_session_set_reconsider_callback:
988 * @sess: an #RTPSession
989 * @callback: callback to set
990 * @user_data: user data passed in the callback
992 * Configure only the reconsider callback to be notified of the reconsider action.
995 rtp_session_set_reconsider_callback (RTPSession * sess,
996 RTPSessionReconsider callback, gpointer user_data)
998 g_return_if_fail (RTP_IS_SESSION (sess));
1000 sess->callbacks.reconsider = callback;
1001 sess->reconsider_user_data = user_data;
1005 * rtp_session_set_request_time_callback:
1006 * @sess: an #RTPSession
1007 * @callback: callback to set
1008 * @user_data: user data passed in the callback
1010 * Configure only the request_time callback
1013 rtp_session_set_request_time_callback (RTPSession * sess,
1014 RTPSessionRequestTime callback, gpointer user_data)
1016 g_return_if_fail (RTP_IS_SESSION (sess));
1018 sess->callbacks.request_time = callback;
1019 sess->request_time_user_data = user_data;
1023 * rtp_session_set_bandwidth:
1024 * @sess: an #RTPSession
1025 * @bandwidth: the bandwidth allocated
1027 * Set the session bandwidth in bytes per second.
1030 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1032 g_return_if_fail (RTP_IS_SESSION (sess));
1034 RTP_SESSION_LOCK (sess);
1035 sess->stats.bandwidth = bandwidth;
1036 RTP_SESSION_UNLOCK (sess);
1040 * rtp_session_get_bandwidth:
1041 * @sess: an #RTPSession
1043 * Get the session bandwidth.
1045 * Returns: the session bandwidth.
1048 rtp_session_get_bandwidth (RTPSession * sess)
1052 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1054 RTP_SESSION_LOCK (sess);
1055 result = sess->stats.bandwidth;
1056 RTP_SESSION_UNLOCK (sess);
1062 * rtp_session_set_rtcp_fraction:
1063 * @sess: an #RTPSession
1064 * @bandwidth: the RTCP bandwidth
1066 * Set the bandwidth in bytes per second that should be used for RTCP
1070 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1072 g_return_if_fail (RTP_IS_SESSION (sess));
1074 RTP_SESSION_LOCK (sess);
1075 sess->stats.rtcp_bandwidth = bandwidth;
1076 RTP_SESSION_UNLOCK (sess);
1080 * rtp_session_get_rtcp_fraction:
1081 * @sess: an #RTPSession
1083 * Get the session bandwidth used for RTCP.
1085 * Returns: The bandwidth used for RTCP messages.
1088 rtp_session_get_rtcp_fraction (RTPSession * sess)
1092 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1094 RTP_SESSION_LOCK (sess);
1095 result = sess->stats.rtcp_bandwidth;
1096 RTP_SESSION_UNLOCK (sess);
1102 * rtp_session_get_sdes_struct:
1103 * @sess: an #RTSPSession
1105 * Get the SDES data as a #GstStructure
1107 * Returns: a GstStructure with SDES items for @sess. This function returns a
1108 * copy of the SDES structure, use gst_structure_free() after usage.
1111 rtp_session_get_sdes_struct (RTPSession * sess)
1113 GstStructure *result = NULL;
1115 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1117 RTP_SESSION_LOCK (sess);
1119 result = gst_structure_copy (sess->sdes);
1120 RTP_SESSION_UNLOCK (sess);
1126 * rtp_session_set_sdes_struct:
1127 * @sess: an #RTSPSession
1128 * @sdes: a #GstStructure
1130 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1133 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1135 g_return_if_fail (sdes);
1136 g_return_if_fail (RTP_IS_SESSION (sess));
1138 RTP_SESSION_LOCK (sess);
1140 gst_structure_free (sess->sdes);
1141 sess->sdes = gst_structure_copy (sdes);
1142 RTP_SESSION_UNLOCK (sess);
1145 static GstFlowReturn
1146 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1148 GstFlowReturn result = GST_FLOW_OK;
1150 if (source->internal) {
1151 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1153 RTP_SESSION_UNLOCK (session);
1155 if (session->callbacks.send_rtp)
1157 session->callbacks.send_rtp (session, source, data,
1158 session->send_rtp_user_data);
1160 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1163 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1164 RTP_SESSION_UNLOCK (session);
1166 if (session->callbacks.process_rtp)
1168 session->callbacks.process_rtp (session, source,
1169 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1171 gst_buffer_unref (GST_BUFFER_CAST (data));
1173 RTP_SESSION_LOCK (session);
1179 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1183 RTP_SESSION_UNLOCK (session);
1185 if (session->callbacks.clock_rate)
1187 session->callbacks.clock_rate (session, pt,
1188 session->clock_rate_user_data);
1192 RTP_SESSION_LOCK (session);
1194 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1199 static RTPSourceCallbacks callbacks = {
1200 (RTPSourcePushRTP) source_push_rtp,
1201 (RTPSourceClockRate) source_clock_rate,
1205 check_collision (RTPSession * sess, RTPSource * source,
1206 RTPPacketInfo * pinfo, gboolean rtp)
1210 /* If we have no pinfo address, we can't do collision checking */
1211 if (!pinfo->address)
1214 ssrc = rtp_source_get_ssrc (source);
1216 if (!source->internal) {
1217 GSocketAddress *from;
1219 /* This is not our local source, but lets check if two remote
1222 from = source->rtp_from;
1224 from = source->rtcp_from;
1228 if (__g_socket_address_equal (from, pinfo->address)) {
1229 /* Address is the same */
1232 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1233 if (sess->favor_new) {
1234 if (rtp_source_find_conflicting_address (source,
1235 pinfo->address, pinfo->current_time)) {
1238 buf1 = __g_socket_address_to_string (pinfo->address);
1239 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1247 /* Current address is not a known conflict, lets assume this is
1248 * a new source. Save old address in possible conflict list
1250 rtp_source_add_conflicting_address (source, from,
1251 pinfo->current_time);
1253 buf1 = __g_socket_address_to_string (from);
1254 buf2 = __g_socket_address_to_string (pinfo->address);
1256 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1257 " saving old as known conflict", ssrc, buf1, buf2);
1260 rtp_source_set_rtp_from (source, pinfo->address);
1262 rtp_source_set_rtcp_from (source, pinfo->address);
1270 /* Don't need to save old addresses, we ignore new sources */
1275 /* We don't already have a from address for RTP, just set it */
1277 rtp_source_set_rtp_from (source, pinfo->address);
1279 rtp_source_set_rtcp_from (source, pinfo->address);
1283 /* FIXME: Log 3rd party collision somehow
1284 * Maybe should be done in upper layer, only the SDES can tell us
1285 * if its a collision or a loop
1288 /* This is sending with our ssrc, is it an address we already know */
1289 if (rtp_source_find_conflicting_address (source, pinfo->address,
1290 pinfo->current_time)) {
1291 /* Its a known conflict, its probably a loop, not a collision
1292 * lets just drop the incoming packet
1294 GST_DEBUG ("Our packets are being looped back to us, dropping");
1296 /* Its a new collision, lets change our SSRC */
1297 rtp_source_add_conflicting_address (source, pinfo->address,
1298 pinfo->current_time);
1300 GST_DEBUG ("Collision for SSRC %x", ssrc);
1301 /* mark the source BYE */
1302 rtp_source_mark_bye (source, "SSRC Collision");
1303 /* if we were suggesting this SSRC, change to something else */
1304 if (sess->suggested_ssrc == ssrc)
1305 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1307 on_ssrc_collision (sess, source);
1309 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1318 gboolean is_doing_ptp;
1319 GSocketAddress *new_addr;
1322 /* check if the two given ip addr are the same (do not care about the port) */
1324 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1327 g_inet_address_equal (g_inet_socket_address_get_address
1328 (G_INET_SOCKET_ADDRESS (a)),
1329 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1333 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1334 CompareAddrData * data)
1336 /* only compare ip addr of remote sources which are also not closing */
1337 if (!source->internal && !source->closing && source->rtp_from) {
1338 /* look for the first rtp source */
1339 if (!data->new_addr)
1340 data->new_addr = source->rtp_from;
1341 /* compare current ip addr with the first one */
1343 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1348 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1349 CompareAddrData * data)
1351 /* only compare ip addr of remote sources which are also not closing */
1352 if (!source->internal && !source->closing && source->rtcp_from) {
1353 /* look for the first rtcp source */
1354 if (!data->new_addr)
1355 data->new_addr = source->rtcp_from;
1357 /* compare current ip addr with the first one */
1358 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1362 /* loop over our non-internal source to know if the session
1363 * is doing point-to-point */
1365 session_update_ptp (RTPSession * sess)
1367 /* to know if the session is doing point to point, the ip addr
1368 * of each non-internal (=remotes) source have to be compared
1371 gboolean is_doing_rtp_ptp = FALSE;
1372 gboolean is_doing_rtcp_ptp = FALSE;
1373 CompareAddrData data;
1375 /* compare the first remote source's ip addr that receive rtp packets
1376 * with other remote rtp source.
1377 * it's enough because the session just needs to know if they are all
1380 data.is_doing_ptp = TRUE;
1381 data.new_addr = NULL;
1382 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1383 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1384 is_doing_rtp_ptp = data.is_doing_ptp;
1386 /* same but about rtcp */
1387 data.is_doing_ptp = TRUE;
1388 data.new_addr = NULL;
1389 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1390 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1391 is_doing_rtcp_ptp = data.is_doing_ptp;
1393 /* the session is doing point-to-point if all rtp remote have the same
1394 * ip addr and if all rtcp remote sources have the same ip addr */
1395 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1397 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1401 add_source (RTPSession * sess, RTPSource * src)
1403 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1404 GINT_TO_POINTER (src->ssrc), src);
1405 /* report the new source ASAP */
1406 src->generation = sess->generation;
1407 /* we have one more source now */
1408 sess->total_sources++;
1409 if (RTP_SOURCE_IS_ACTIVE (src))
1410 sess->stats.active_sources++;
1411 if (src->internal) {
1412 sess->stats.internal_sources++;
1413 if (sess->suggested_ssrc != src->ssrc)
1414 sess->suggested_ssrc = src->ssrc;
1417 /* update point-to-point status */
1419 session_update_ptp (sess);
1423 find_source (RTPSession * sess, guint32 ssrc)
1425 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1426 GINT_TO_POINTER (ssrc));
1429 /* must be called with the session lock, the returned source needs to be
1430 * unreffed after usage. */
1432 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1433 RTPPacketInfo * pinfo, gboolean rtp)
1437 source = find_source (sess, ssrc);
1438 if (source == NULL) {
1439 /* make new Source in probation and insert */
1440 source = rtp_source_new (ssrc);
1442 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1444 /* for RTP packets we need to set the source in probation. Receiving RTCP
1445 * packets of an SSRC, on the other hand, is a strong indication that we
1446 * are dealing with a valid source. */
1448 g_object_set (source, "probation", sess->probation, NULL);
1450 g_object_set (source, "probation", 0, NULL);
1452 /* store from address, if any */
1453 if (pinfo->address) {
1455 rtp_source_set_rtp_from (source, pinfo->address);
1457 rtp_source_set_rtcp_from (source, pinfo->address);
1460 /* configure a callback on the source */
1461 rtp_source_set_callbacks (source, &callbacks, sess);
1463 add_source (sess, source);
1467 /* check for collision, this updates the address when not previously set */
1468 if (check_collision (sess, source, pinfo, rtp)) {
1471 /* Receiving RTCP packets of an SSRC is a strong indication that we
1472 * are dealing with a valid source. */
1474 g_object_set (source, "probation", 0, NULL);
1476 /* update last activity */
1477 source->last_activity = pinfo->current_time;
1479 source->last_rtp_activity = pinfo->current_time;
1480 g_object_ref (source);
1485 /* must be called with the session lock, the returned source needs to be
1486 * unreffed after usage. */
1488 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1492 source = find_source (sess, ssrc);
1493 if (source == NULL) {
1494 /* make new internal Source and insert */
1495 source = rtp_source_new (ssrc);
1497 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1499 source->validated = TRUE;
1500 source->internal = TRUE;
1501 source->probation = FALSE;
1502 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1503 rtp_source_set_callbacks (source, &callbacks, sess);
1505 add_source (sess, source);
1510 g_object_ref (source);
1516 * rtp_session_suggest_ssrc:
1517 * @sess: a #RTPSession
1519 * Suggest an unused SSRC in @sess.
1521 * Returns: a free unused SSRC
1524 rtp_session_suggest_ssrc (RTPSession * sess)
1528 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1530 RTP_SESSION_LOCK (sess);
1531 result = sess->suggested_ssrc;
1532 RTP_SESSION_UNLOCK (sess);
1538 * rtp_session_add_source:
1539 * @sess: a #RTPSession
1540 * @src: #RTPSource to add
1542 * Add @src to @session.
1544 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1545 * existed in the session.
1548 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1550 gboolean result = FALSE;
1553 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1554 g_return_val_if_fail (src != NULL, FALSE);
1556 RTP_SESSION_LOCK (sess);
1557 find = find_source (sess, src->ssrc);
1559 add_source (sess, src);
1562 RTP_SESSION_UNLOCK (sess);
1568 * rtp_session_get_num_sources:
1569 * @sess: an #RTPSession
1571 * Get the number of sources in @sess.
1573 * Returns: The number of sources in @sess.
1576 rtp_session_get_num_sources (RTPSession * sess)
1580 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1582 RTP_SESSION_LOCK (sess);
1583 result = sess->total_sources;
1584 RTP_SESSION_UNLOCK (sess);
1590 * rtp_session_get_num_active_sources:
1591 * @sess: an #RTPSession
1593 * Get the number of active sources in @sess. A source is considered active when
1594 * it has been validated and has not yet received a BYE RTCP message.
1596 * Returns: The number of active sources in @sess.
1599 rtp_session_get_num_active_sources (RTPSession * sess)
1603 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1605 RTP_SESSION_LOCK (sess);
1606 result = sess->stats.active_sources;
1607 RTP_SESSION_UNLOCK (sess);
1613 * rtp_session_get_source_by_ssrc:
1614 * @sess: an #RTPSession
1617 * Find the source with @ssrc in @sess.
1619 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1620 * g_object_unref() after usage.
1623 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1627 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1629 RTP_SESSION_LOCK (sess);
1630 result = find_source (sess, ssrc);
1632 g_object_ref (result);
1633 RTP_SESSION_UNLOCK (sess);
1638 /* should be called with the SESSION lock */
1640 rtp_session_create_new_ssrc (RTPSession * sess)
1645 ssrc = g_random_int ();
1647 /* see if it exists in the session, we're done if it doesn't */
1648 if (find_source (sess, ssrc) == NULL)
1656 * rtp_session_create_source:
1657 * @sess: an #RTPSession
1659 * Create an #RTPSource for use in @sess. This function will create a source
1660 * with an ssrc that is currently not used by any participants in the session.
1662 * Returns: an #RTPSource.
1665 rtp_session_create_source (RTPSession * sess)
1670 RTP_SESSION_LOCK (sess);
1671 ssrc = rtp_session_create_new_ssrc (sess);
1672 source = rtp_source_new (ssrc);
1673 rtp_source_set_callbacks (source, &callbacks, sess);
1674 /* we need an additional ref for the source in the hashtable */
1675 g_object_ref (source);
1676 add_source (sess, source);
1677 RTP_SESSION_UNLOCK (sess);
1683 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1685 GstNetAddressMeta *meta;
1687 /* get packet size including header overhead */
1688 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1692 GstRTPBuffer rtp = { NULL };
1694 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1695 goto invalid_packet;
1697 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1701 /* only keep info for first buffer */
1702 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1703 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1704 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1705 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1706 /* copy available csrc */
1707 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1708 for (i = 0; i < pinfo->csrc_count; i++)
1709 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1711 gst_rtp_buffer_unmap (&rtp);
1715 /* for netbuffer we can store the IP address to check for collisions */
1716 meta = gst_buffer_get_net_address_meta (*buffer);
1718 g_object_unref (pinfo->address);
1720 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1722 pinfo->address = NULL;
1730 GST_DEBUG ("invalid RTP packet received");
1735 /* update the RTPPacketInfo structure with the current time and other bits
1736 * about the current buffer we are handling.
1737 * This function is typically called when a validated packet is received.
1738 * This function should be called with the SESSION_LOCK
1741 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1742 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1743 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1749 pinfo->is_list = is_list;
1751 pinfo->current_time = current_time;
1752 pinfo->running_time = running_time;
1753 pinfo->ntpnstime = ntpnstime;
1754 pinfo->header_len = sess->header_len;
1756 pinfo->payload_len = 0;
1760 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1762 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1765 GstBuffer *buffer = GST_BUFFER_CAST (data);
1766 res = update_packet (&buffer, 0, pinfo);
1772 clean_packet_info (RTPPacketInfo * pinfo)
1775 g_object_unref (pinfo->address);
1777 gst_mini_object_unref (pinfo->data);
1783 source_update_active (RTPSession * sess, RTPSource * source,
1784 gboolean prevactive)
1786 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1787 guint32 ssrc = source->ssrc;
1789 if (prevactive == active)
1793 sess->stats.active_sources++;
1794 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1795 sess->stats.active_sources);
1797 sess->stats.active_sources--;
1798 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1799 sess->stats.active_sources);
1805 source_update_sender (RTPSession * sess, RTPSource * source,
1806 gboolean prevsender)
1808 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1809 guint32 ssrc = source->ssrc;
1811 if (prevsender == sender)
1815 sess->stats.sender_sources++;
1816 if (source->internal)
1817 sess->stats.internal_sender_sources++;
1818 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1819 sess->stats.sender_sources);
1821 sess->stats.sender_sources--;
1822 if (source->internal)
1823 sess->stats.internal_sender_sources--;
1824 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1825 sess->stats.sender_sources);
1831 * rtp_session_process_rtp:
1832 * @sess: and #RTPSession
1833 * @buffer: an RTP buffer
1834 * @current_time: the current system time
1835 * @running_time: the running_time of @buffer
1837 * Process an RTP buffer in the session manager. This function takes ownership
1840 * Returns: a #GstFlowReturn.
1843 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1844 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1846 GstFlowReturn result;
1850 gboolean prevsender, prevactive;
1851 RTPPacketInfo pinfo = { 0, };
1854 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1855 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1857 RTP_SESSION_LOCK (sess);
1859 /* update pinfo stats */
1860 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1861 current_time, running_time, ntpnstime)) {
1862 GST_DEBUG ("invalid RTP packet received");
1863 RTP_SESSION_UNLOCK (sess);
1864 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1869 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1873 prevsender = RTP_SOURCE_IS_SENDER (source);
1874 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1875 oldrate = source->bitrate;
1877 /* let source process the packet */
1878 result = rtp_source_process_rtp (source, &pinfo);
1880 /* source became active */
1881 if (source_update_active (sess, source, prevactive))
1882 on_ssrc_validated (sess, source);
1884 source_update_sender (sess, source, prevsender);
1886 if (oldrate != source->bitrate)
1887 sess->recalc_bandwidth = TRUE;
1890 on_new_ssrc (sess, source);
1892 if (source->validated) {
1896 /* for validated sources, we add the CSRCs as well */
1897 for (i = 0; i < pinfo.csrc_count; i++) {
1899 RTPSource *csrc_src;
1901 csrc = pinfo.csrcs[i];
1904 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
1909 GST_DEBUG ("created new CSRC: %08x", csrc);
1910 rtp_source_set_as_csrc (csrc_src);
1911 source_update_active (sess, csrc_src, FALSE);
1912 on_new_ssrc (sess, csrc_src);
1914 g_object_unref (csrc_src);
1917 g_object_unref (source);
1919 RTP_SESSION_UNLOCK (sess);
1921 clean_packet_info (&pinfo);
1928 RTP_SESSION_UNLOCK (sess);
1929 clean_packet_info (&pinfo);
1930 GST_DEBUG ("ignoring packet because its collisioning");
1936 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1937 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
1941 count = gst_rtcp_packet_get_rb_count (packet);
1942 for (i = 0; i < count; i++) {
1943 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1944 guint8 fractionlost;
1948 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1949 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1951 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1953 /* find our own source */
1954 src = find_source (sess, ssrc);
1958 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
1959 /* only deal with report blocks for our session, we update the stats of
1960 * the sender of the RTCP message. We could also compare our stats against
1961 * the other sender to see if we are better or worse. */
1962 /* FIXME, need to keep track who the RB block is from */
1963 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
1964 packetslost, exthighestseq, jitter, lsr, dlsr);
1967 on_ssrc_active (sess, source);
1970 /* A Sender report contains statistics about how the sender is doing. This
1971 * includes timing informataion such as the relation between RTP and NTP
1972 * timestamps and the number of packets/bytes it sent to us.
1974 * In this report is also included a set of report blocks related to how this
1975 * sender is receiving data (in case we (or somebody else) is also sending stuff
1976 * to it). This info includes the packet loss, jitter and seqnum. It also
1977 * contains information to calculate the round trip time (LSR/DLSR).
1980 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1981 RTPPacketInfo * pinfo, gboolean * do_sync)
1983 guint32 senderssrc, rtptime, packet_count, octet_count;
1986 gboolean created, prevsender;
1988 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1989 &packet_count, &octet_count);
1991 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1992 senderssrc, GST_TIME_ARGS (pinfo->current_time));
1994 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
1998 /* skip non-bye packets for sources that are marked BYE */
1999 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2002 /* don't try to do lip-sync for sources that sent a BYE */
2003 if (RTP_SOURCE_IS_MARKED_BYE (source))
2008 prevsender = RTP_SOURCE_IS_SENDER (source);
2010 /* first update the source */
2011 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2012 packet_count, octet_count);
2014 source_update_sender (sess, source, prevsender);
2017 on_new_ssrc (sess, source);
2019 rtp_session_process_rb (sess, source, packet, pinfo);
2022 g_object_unref (source);
2025 /* A receiver report contains statistics about how a receiver is doing. It
2026 * includes stuff like packet loss, jitter and the seqnum it received last. It
2027 * also contains info to calculate the round trip time.
2029 * We are only interested in how the sender of this report is doing wrt to us.
2032 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2033 RTPPacketInfo * pinfo)
2039 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2041 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2043 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2047 /* skip non-bye packets for sources that are marked BYE */
2048 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2052 on_new_ssrc (sess, source);
2054 rtp_session_process_rb (sess, source, packet, pinfo);
2057 g_object_unref (source);
2060 /* Get SDES items and store them in the SSRC */
2062 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2063 RTPPacketInfo * pinfo)
2066 gboolean more_items, more_entries;
2068 items = gst_rtcp_packet_sdes_get_item_count (packet);
2069 GST_DEBUG ("got SDES packet with %d items", items);
2071 more_items = gst_rtcp_packet_sdes_first_item (packet);
2073 while (more_items) {
2075 gboolean changed, created, prevactive;
2079 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2081 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2085 /* find src, no probation when dealing with RTCP */
2086 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2090 /* skip non-bye packets for sources that are marked BYE */
2091 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2094 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2096 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2098 while (more_entries) {
2099 GstRTCPSDESType type;
2105 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2107 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2110 if (type == GST_RTCP_SDES_PRIV) {
2111 name = g_strndup ((const gchar *) &data[1], data[0]);
2113 data += data[0] + 1;
2115 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2118 value = g_strndup ((const gchar *) data, len);
2120 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2125 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2129 /* takes ownership of sdes */
2130 changed = rtp_source_set_sdes_struct (source, sdes);
2132 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2133 source->validated = TRUE;
2136 on_new_ssrc (sess, source);
2138 /* source became active */
2139 if (source_update_active (sess, source, prevactive))
2140 on_ssrc_validated (sess, source);
2143 on_ssrc_sdes (sess, source);
2146 g_object_unref (source);
2148 more_items = gst_rtcp_packet_sdes_next_item (packet);
2153 /* BYE is sent when a client leaves the session
2156 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2157 RTPPacketInfo * pinfo)
2161 gboolean reconsider = FALSE;
2163 reason = gst_rtcp_packet_bye_get_reason (packet);
2164 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2166 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2167 for (i = 0; i < count; i++) {
2170 gboolean created, prevactive, prevsender;
2171 guint pmembers, members;
2173 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2174 GST_DEBUG ("SSRC: %08x", ssrc);
2176 /* find src and mark bye, no probation when dealing with RTCP */
2177 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2181 if (source->internal) {
2182 /* our own source, something weird with this packet */
2183 g_object_unref (source);
2187 /* store time for when we need to time out this source */
2188 source->bye_time = pinfo->current_time;
2190 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2191 prevsender = RTP_SOURCE_IS_SENDER (source);
2193 /* mark the source BYE */
2194 rtp_source_mark_bye (source, reason);
2196 pmembers = sess->stats.active_sources;
2198 source_update_active (sess, source, prevactive);
2199 source_update_sender (sess, source, prevsender);
2201 members = sess->stats.active_sources;
2203 if (!sess->scheduled_bye && members < pmembers) {
2204 /* some members went away since the previous timeout estimate.
2205 * Perform reverse reconsideration but only when we are not scheduling a
2207 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2208 pinfo->current_time < sess->next_rtcp_check_time) {
2209 GstClockTime time_remaining;
2211 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2212 sess->next_rtcp_check_time =
2213 gst_util_uint64_scale (time_remaining, members, pmembers);
2215 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2216 GST_TIME_ARGS (sess->next_rtcp_check_time));
2218 sess->next_rtcp_check_time += pinfo->current_time;
2220 /* mark pending reconsider. We only want to signal the reconsideration
2221 * once after we handled all the source in the bye packet */
2227 on_new_ssrc (sess, source);
2229 on_bye_ssrc (sess, source);
2231 g_object_unref (source);
2234 RTP_SESSION_UNLOCK (sess);
2235 /* notify app of reconsideration */
2236 if (sess->callbacks.reconsider)
2237 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2238 RTP_SESSION_LOCK (sess);
2244 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2245 RTPPacketInfo * pinfo)
2247 GST_DEBUG ("received APP");
2251 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2252 gboolean fir, GstClockTime current_time)
2254 guint32 round_trip = 0;
2256 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2258 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2259 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2262 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2263 GST_DEBUG ("Ignoring %s request because one was send without one "
2264 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2265 fir ? "FIR" : "PLI",
2266 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2267 GST_TIME_ARGS (round_trip_in_ns));;
2272 sess->last_keyframe_request = current_time;
2274 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2275 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2276 sess->callbacks.request_key_unit);
2278 RTP_SESSION_UNLOCK (sess);
2279 sess->callbacks.request_key_unit (sess, fir,
2280 sess->request_key_unit_user_data);
2281 RTP_SESSION_LOCK (sess);
2287 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2288 guint32 media_ssrc, GstClockTime current_time)
2292 if (!sess->callbacks.request_key_unit)
2295 src = find_source (sess, sender_ssrc);
2299 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2303 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2304 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2309 gboolean our_request = FALSE;
2311 if (!sess->callbacks.request_key_unit)
2317 src = find_source (sess, sender_ssrc);
2319 /* Hack because Google fails to set the sender_ssrc correctly */
2320 if (!src && sender_ssrc == 1) {
2321 GHashTableIter iter;
2323 /* we can't find the source if there are multiple */
2324 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2327 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2328 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2329 if (!src->internal && rtp_source_is_sender (src))
2337 for (position = 0; position < fci_length; position += 8) {
2338 guint8 *data = fci_data + position;
2341 ssrc = GST_READ_UINT32_BE (data);
2343 own = find_source (sess, ssrc);
2344 if (own->internal) {
2352 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2356 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2357 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2358 GstClockTime current_time)
2360 sess->stats.nacks_received++;
2362 if (!sess->callbacks.notify_nack)
2365 while (fci_length > 0) {
2366 guint16 seqnum, blp;
2368 seqnum = GST_READ_UINT16_BE (fci_data);
2369 blp = GST_READ_UINT16_BE (fci_data + 2);
2371 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2373 RTP_SESSION_UNLOCK (sess);
2374 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2375 sess->notify_nack_user_data);
2376 RTP_SESSION_LOCK (sess);
2384 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2385 RTPPacketInfo * pinfo, GstClockTime current_time)
2387 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2388 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2389 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2390 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2391 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2392 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2395 src = find_source (sess, media_ssrc);
2397 /* skip non-bye packets for sources that are marked BYE */
2398 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2401 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2402 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2404 if (g_signal_has_handler_pending (sess,
2405 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2406 GstBuffer *fci_buffer = NULL;
2408 if (fci_length > 0) {
2409 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2410 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2412 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2415 RTP_SESSION_UNLOCK (sess);
2416 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2417 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2418 RTP_SESSION_LOCK (sess);
2421 gst_buffer_unref (fci_buffer);
2424 if (src && sess->rtcp_feedback_retention_window) {
2425 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2428 if ((src && src->internal) ||
2429 /* PSFB FIR puts the media ssrc inside the FCI */
2430 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2432 case GST_RTCP_TYPE_PSFB:
2434 case GST_RTCP_PSFB_TYPE_PLI:
2435 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2438 case GST_RTCP_PSFB_TYPE_FIR:
2439 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2446 case GST_RTCP_TYPE_RTPFB:
2448 case GST_RTCP_RTPFB_TYPE_NACK:
2449 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2450 fci_data, fci_length, current_time);
2462 * rtp_session_process_rtcp:
2463 * @sess: and #RTPSession
2464 * @buffer: an RTCP buffer
2465 * @current_time: the current system time
2466 * @ntpnstime: the current NTP time in nanoseconds
2468 * Process an RTCP buffer in the session manager. This function takes ownership
2471 * Returns: a #GstFlowReturn.
2474 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2475 GstClockTime current_time, guint64 ntpnstime)
2477 GstRTCPPacket packet;
2478 gboolean more, is_bye = FALSE, do_sync = FALSE;
2479 RTPPacketInfo pinfo = { 0, };
2480 GstFlowReturn result = GST_FLOW_OK;
2481 GstRTCPBuffer rtcp = { NULL, };
2483 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2484 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2486 if (!gst_rtcp_buffer_validate (buffer))
2487 goto invalid_packet;
2489 GST_DEBUG ("received RTCP packet");
2491 RTP_SESSION_LOCK (sess);
2492 /* update pinfo stats */
2493 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2496 /* start processing the compound packet */
2497 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2498 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2502 type = gst_rtcp_packet_get_type (&packet);
2505 case GST_RTCP_TYPE_SR:
2506 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2508 case GST_RTCP_TYPE_RR:
2509 rtp_session_process_rr (sess, &packet, &pinfo);
2511 case GST_RTCP_TYPE_SDES:
2512 rtp_session_process_sdes (sess, &packet, &pinfo);
2514 case GST_RTCP_TYPE_BYE:
2516 /* don't try to attempt lip-sync anymore for streams with a BYE */
2518 rtp_session_process_bye (sess, &packet, &pinfo);
2520 case GST_RTCP_TYPE_APP:
2521 rtp_session_process_app (sess, &packet, &pinfo);
2523 case GST_RTCP_TYPE_RTPFB:
2524 case GST_RTCP_TYPE_PSFB:
2525 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2528 GST_WARNING ("got unknown RTCP packet");
2531 more = gst_rtcp_packet_move_to_next (&packet);
2534 gst_rtcp_buffer_unmap (&rtcp);
2536 /* if we are scheduling a BYE, we only want to count bye packets, else we
2537 * count everything */
2538 if (sess->scheduled_bye && is_bye) {
2539 sess->bye_stats.bye_members++;
2540 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2543 /* keep track of average packet size */
2544 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2546 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2547 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2548 RTP_SESSION_UNLOCK (sess);
2551 clean_packet_info (&pinfo);
2553 /* notify caller of sr packets in the callback */
2554 if (do_sync && sess->callbacks.sync_rtcp) {
2555 result = sess->callbacks.sync_rtcp (sess, buffer,
2556 sess->sync_rtcp_user_data);
2558 gst_buffer_unref (buffer);
2565 GST_DEBUG ("invalid RTCP packet received");
2566 gst_buffer_unref (buffer);
2572 * rtp_session_update_send_caps:
2573 * @sess: an #RTPSession
2576 * Update the caps of the sender in the rtp session.
2579 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2584 g_return_if_fail (RTP_IS_SESSION (sess));
2585 g_return_if_fail (GST_IS_CAPS (caps));
2587 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2589 s = gst_caps_get_structure (caps, 0);
2591 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2595 RTP_SESSION_LOCK (sess);
2596 source = obtain_internal_source (sess, ssrc, &created);
2598 rtp_source_update_caps (source, caps);
2599 g_object_unref (source);
2601 RTP_SESSION_UNLOCK (sess);
2606 * rtp_session_send_rtp:
2607 * @sess: an #RTPSession
2608 * @data: pointer to either an RTP buffer or a list of RTP buffers
2609 * @is_list: TRUE when @data is a buffer list
2610 * @current_time: the current system time
2611 * @running_time: the running time of @data
2613 * Send the RTP buffer in the session manager. This function takes ownership of
2616 * Returns: a #GstFlowReturn.
2619 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2620 GstClockTime current_time, GstClockTime running_time)
2622 GstFlowReturn result;
2624 gboolean prevsender;
2626 RTPPacketInfo pinfo = { 0, };
2629 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2630 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2632 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2634 RTP_SESSION_LOCK (sess);
2635 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2636 current_time, running_time, -1))
2637 goto invalid_packet;
2639 source = obtain_internal_source (sess, pinfo.ssrc, &created);
2641 /* update last activity */
2642 source->last_rtp_activity = current_time;
2644 prevsender = RTP_SOURCE_IS_SENDER (source);
2645 oldrate = source->bitrate;
2647 /* we use our own source to send */
2648 result = rtp_source_send_rtp (source, &pinfo);
2650 source_update_sender (sess, source, prevsender);
2652 if (oldrate != source->bitrate)
2653 sess->recalc_bandwidth = TRUE;
2654 RTP_SESSION_UNLOCK (sess);
2656 g_object_unref (source);
2657 clean_packet_info (&pinfo);
2663 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2664 RTP_SESSION_UNLOCK (sess);
2665 GST_DEBUG ("invalid RTP packet received");
2671 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2673 *bandwidth += source->bitrate;
2676 /* must be called with session lock */
2678 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2681 GstClockTime result;
2682 RTPSessionStats *stats;
2684 /* recalculate bandwidth when it changed */
2685 if (sess->recalc_bandwidth) {
2688 if (sess->bandwidth > 0)
2689 bandwidth = sess->bandwidth;
2691 /* If it is <= 0, then try to estimate the actual bandwidth */
2694 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2695 (GHFunc) add_bitrates, &bandwidth);
2698 if (bandwidth < 8000)
2699 bandwidth = RTP_STATS_BANDWIDTH;
2701 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2702 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2704 sess->recalc_bandwidth = FALSE;
2707 if (sess->scheduled_bye) {
2708 stats = &sess->bye_stats;
2709 result = rtp_stats_calculate_bye_interval (stats);
2711 stats = &sess->stats;
2712 result = rtp_stats_calculate_rtcp_interval (stats,
2713 stats->internal_sender_sources > 0, first);
2716 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2717 GST_TIME_ARGS (result), first);
2719 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2720 result = rtp_stats_add_rtcp_jitter (stats, result);
2722 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2728 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2730 if (source->internal)
2731 rtp_source_mark_bye (source, reason);
2735 * rtp_session_mark_all_bye:
2736 * @sess: an #RTPSession
2739 * Mark all internal sources of the session as BYE with @reason.
2742 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2744 g_return_if_fail (RTP_IS_SESSION (sess));
2746 RTP_SESSION_LOCK (sess);
2747 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2748 (GHFunc) source_mark_bye, (gpointer) reason);
2749 RTP_SESSION_UNLOCK (sess);
2752 /* Stop the current @sess and schedule a BYE message for the other members.
2753 * One must have the session lock to call this function
2755 static GstFlowReturn
2756 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2758 GstFlowReturn result = GST_FLOW_OK;
2759 GstClockTime interval;
2761 /* nothing to do it we already scheduled bye */
2762 if (sess->scheduled_bye)
2765 /* we schedule BYE now */
2766 sess->scheduled_bye = TRUE;
2767 /* at least one member wants to send a BYE */
2768 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2769 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2770 sess->bye_stats.bye_members = 1;
2771 sess->first_rtcp = TRUE;
2772 sess->allow_early = TRUE;
2774 /* reschedule transmission */
2775 sess->last_rtcp_send_time = current_time;
2776 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2778 if (interval != GST_CLOCK_TIME_NONE)
2779 sess->next_rtcp_check_time = current_time + interval;
2781 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2783 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2784 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2786 RTP_SESSION_UNLOCK (sess);
2787 /* notify app of reconsideration */
2788 if (sess->callbacks.reconsider)
2789 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2790 RTP_SESSION_LOCK (sess);
2797 * rtp_session_schedule_bye:
2798 * @sess: an #RTPSession
2799 * @current_time: the current system time
2801 * Schedule a BYE message for all sources marked as BYE in @sess.
2803 * Returns: a #GstFlowReturn.
2806 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2808 GstFlowReturn result = GST_FLOW_OK;
2810 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2812 RTP_SESSION_LOCK (sess);
2813 result = rtp_session_schedule_bye_locked (sess, current_time);
2814 RTP_SESSION_UNLOCK (sess);
2820 * rtp_session_next_timeout:
2821 * @sess: an #RTPSession
2822 * @current_time: the current system time
2824 * Get the next time we should perform session maintenance tasks.
2826 * Returns: a time when rtp_session_on_timeout() should be called with the
2827 * current system time.
2830 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2832 GstClockTime result, interval = 0;
2834 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2836 RTP_SESSION_LOCK (sess);
2838 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2839 GST_DEBUG ("have early rtcp time");
2840 result = sess->next_early_rtcp_time;
2844 result = sess->next_rtcp_check_time;
2846 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2847 ", next time: %" GST_TIME_FORMAT,
2848 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2850 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2851 GST_DEBUG ("take current time as base");
2852 /* our previous check time expired, start counting from the current time
2854 result = current_time;
2857 if (sess->scheduled_bye) {
2858 if (sess->bye_stats.active_sources >= 50) {
2859 GST_DEBUG ("reconsider BYE, more than 50 sources");
2860 /* reconsider BYE if members >= 50 */
2861 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2864 if (sess->first_rtcp) {
2865 GST_DEBUG ("first RTCP packet");
2866 /* we are called for the first time */
2867 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2868 } else if (sess->next_rtcp_check_time < current_time) {
2869 GST_DEBUG ("old check time expired, getting new timeout");
2870 /* get a new timeout when we need to */
2871 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2875 if (interval != GST_CLOCK_TIME_NONE)
2878 result = GST_CLOCK_TIME_NONE;
2880 sess->next_rtcp_check_time = result;
2884 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2885 ", next time: %" GST_TIME_FORMAT,
2886 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2887 RTP_SESSION_UNLOCK (sess);
2901 GstRTCPBuffer rtcpbuf;
2904 guint num_to_report;
2909 GstClockTime current_time;
2911 GstClockTime running_time;
2912 GstClockTime interval;
2913 GstRTCPPacket packet;
2916 gboolean may_suppress;
2918 guint nacked_seqnums;
2922 session_start_rtcp (RTPSession * sess, ReportData * data)
2924 GstRTCPPacket *packet = &data->packet;
2925 RTPSource *own = data->source;
2926 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2928 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2929 data->has_sdes = FALSE;
2931 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2933 if (RTP_SOURCE_IS_SENDER (own)) {
2936 guint32 packet_count, octet_count;
2938 /* we are a sender, create SR */
2939 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2940 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2942 /* get latest stats */
2943 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2944 &ntptime, &rtptime, &packet_count, &octet_count);
2946 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2947 packet_count, octet_count);
2949 /* fill in sender report info */
2950 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2951 ntptime, rtptime, packet_count, octet_count);
2953 /* we are only receiver, create RR */
2954 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2955 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2956 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2960 /* construct a Sender or Receiver Report */
2962 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2964 RTPSession *sess = data->sess;
2965 GstRTCPPacket *packet = &data->packet;
2966 guint8 fractionlost;
2968 guint32 exthighestseq, jitter;
2971 /* don't report for sources in future generations */
2972 if (((gint16) (source->generation - sess->generation)) > 0) {
2973 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
2974 source->generation, sess->generation);
2978 if (g_hash_table_contains (source->reported_in_sr_of,
2979 GUINT_TO_POINTER (data->source->ssrc))) {
2980 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
2984 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
2985 GST_DEBUG ("max RB count reached");
2989 /* only report about other sender */
2990 if (source == data->source)
2993 if (!RTP_SOURCE_IS_SENDER (source)) {
2994 GST_DEBUG ("source %08x not sender", source->ssrc);
2998 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3001 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3002 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3004 /* store last generated RR packet */
3005 source->last_rr.is_valid = TRUE;
3006 source->last_rr.fractionlost = fractionlost;
3007 source->last_rr.packetslost = packetslost;
3008 source->last_rr.exthighestseq = exthighestseq;
3009 source->last_rr.jitter = jitter;
3010 source->last_rr.lsr = lsr;
3011 source->last_rr.dlsr = dlsr;
3013 /* packet is not yet filled, add report block for this source. */
3014 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3015 exthighestseq, jitter, lsr, dlsr);
3018 g_hash_table_add (source->reported_in_sr_of,
3019 GUINT_TO_POINTER (data->source->ssrc));
3024 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3026 GstRTCPPacket *packet = &data->packet;
3030 if (!source->send_fir)
3033 len = gst_rtcp_packet_fb_get_fci_length (packet);
3034 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3035 /* exit because the packet is full, will put next request in a
3039 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3041 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3043 fci_data[0] = source->current_send_fir_seqnum;
3044 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3046 source->send_fir = FALSE;
3050 session_fir (RTPSession * sess, ReportData * data)
3052 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3053 GstRTCPPacket *packet = &data->packet;
3055 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3058 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3059 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3060 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3062 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3063 (GHFunc) session_add_fir, data);
3065 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3066 gst_rtcp_packet_remove (packet);
3068 data->may_suppress = FALSE;
3072 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3074 GstRTCPPacket packet;
3075 GstRTCPBuffer rtcp = { NULL, };
3076 gboolean ret = FALSE;
3078 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3080 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3081 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3082 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3086 gst_rtcp_buffer_unmap (&rtcp);
3093 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3095 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3096 GstRTCPPacket *packet = &data->packet;
3098 if (!source->send_pli)
3101 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3104 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3105 /* exit because the packet is full, will put next request in a
3109 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3110 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3111 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3113 source->send_pli = FALSE;
3114 data->may_suppress = FALSE;
3117 /* construct NACK */
3119 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3121 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3122 GstRTCPPacket *packet = &data->packet;
3127 if (!source->send_nack)
3130 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3131 /* exit because the packet is full, will put next request in a
3135 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3136 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3137 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3139 nacks = rtp_source_get_nacks (source, &n_nacks);
3140 GST_DEBUG ("%u NACKs", n_nacks);
3141 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3144 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3145 for (i = 0; i < n_nacks; i++) {
3146 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3148 data->nacked_seqnums++;
3151 rtp_source_clear_nacks (source);
3152 data->may_suppress = FALSE;
3155 /* perform cleanup of sources that timed out */
3157 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3159 gboolean remove = FALSE;
3160 gboolean byetimeout = FALSE;
3161 gboolean sendertimeout = FALSE;
3162 gboolean is_sender, is_active;
3163 RTPSession *sess = data->sess;
3164 GstClockTime interval, binterval;
3167 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3169 /* check for outdated collisions */
3170 if (source->internal) {
3171 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3172 rtp_source_timeout (source, data->current_time,
3173 /* "a relatively long time" -- RFC 3550 section 8.2 */
3174 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
3175 data->running_time - sess->rtcp_feedback_retention_window);
3178 /* nothing else to do when without RTCP */
3179 if (data->interval == GST_CLOCK_TIME_NONE)
3182 is_sender = RTP_SOURCE_IS_SENDER (source);
3183 is_active = RTP_SOURCE_IS_ACTIVE (source);
3185 /* our own rtcp interval may have been forced low by secondary configuration,
3186 * while sender side may still operate with higher interval,
3187 * so do not just take our interval to decide on timing out sender,
3188 * but take (if data->interval <= 5 * GST_SECOND):
3189 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3190 * where sender_interval is difference between last 2 received RTCP reports
3192 if (data->interval >= 5 * GST_SECOND || source->internal) {
3193 binterval = data->interval;
3195 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3196 GST_TIME_ARGS (source->stats.prev_rtcptime),
3197 GST_TIME_ARGS (source->stats.last_rtcptime));
3198 /* if not received enough yet, fallback to larger default */
3199 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3200 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3202 binterval = 5 * GST_SECOND;
3203 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3205 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3206 GST_TIME_ARGS (binterval));
3208 if (!source->internal) {
3209 if (source->marked_bye) {
3210 /* if we received a BYE from the source, remove the source after some
3212 if (data->current_time > source->bye_time &&
3213 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3214 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3219 /* sources that were inactive for more than 5 times the deterministic reporting
3220 * interval get timed out. the min timeout is 5 seconds. */
3221 /* mind old time that might pre-date last time going to PLAYING */
3222 btime = MAX (source->last_activity, sess->start_time);
3223 if (data->current_time > btime) {
3224 interval = MAX (binterval * 5, 5 * GST_SECOND);
3225 if (data->current_time - btime > interval) {
3226 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3227 source->ssrc, GST_TIME_ARGS (btime));
3233 /* senders that did not send for a long time become a receiver, this also
3234 * holds for our own sources. */
3236 /* mind old time that might pre-date last time going to PLAYING */
3237 btime = MAX (source->last_rtp_activity, sess->start_time);
3238 if (data->current_time > btime) {
3239 interval = MAX (binterval * 2, 5 * GST_SECOND);
3240 if (data->current_time - btime > interval) {
3241 if (source->internal && source->sent_bye) {
3242 /* an internal source is BYE and stopped sending RTP, remove */
3243 GST_DEBUG ("internal BYE source %08x timed out, last %"
3244 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3247 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3248 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3249 sendertimeout = TRUE;
3256 sess->total_sources--;
3258 sess->stats.sender_sources--;
3259 if (source->internal)
3260 sess->stats.internal_sender_sources--;
3263 sess->stats.active_sources--;
3265 if (source->internal)
3266 sess->stats.internal_sources--;
3269 on_bye_timeout (sess, source);
3271 on_timeout (sess, source);
3273 if (sendertimeout) {
3274 source->is_sender = FALSE;
3275 sess->stats.sender_sources--;
3276 if (source->internal)
3277 sess->stats.internal_sender_sources--;
3279 on_sender_timeout (sess, source);
3281 /* count how many source to report in this generation */
3282 if (((gint16) (source->generation - sess->generation)) <= 0)
3283 data->num_to_report++;
3285 source->closing = remove;
3289 session_sdes (RTPSession * sess, ReportData * data)
3291 GstRTCPPacket *packet = &data->packet;
3292 const GstStructure *sdes;
3294 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3296 /* add SDES packet */
3297 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3299 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3301 sdes = rtp_source_get_sdes_struct (data->source);
3303 /* add all fields in the structure, the order is not important. */
3304 n_fields = gst_structure_n_fields (sdes);
3305 for (i = 0; i < n_fields; ++i) {
3308 GstRTCPSDESType type;
3310 field = gst_structure_nth_field_name (sdes, i);
3313 value = gst_structure_get_string (sdes, field);
3316 type = gst_rtcp_sdes_name_to_type (field);
3318 /* Early packets are minimal and only include the CNAME */
3319 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3322 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3323 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3324 (const guint8 *) value);
3325 } else if (type == GST_RTCP_SDES_PRIV) {
3331 /* don't accept entries that are too big */
3332 prefix_len = strlen (field);
3333 if (prefix_len > 255)
3335 value_len = strlen (value);
3336 if (value_len > 255)
3338 data_len = 1 + prefix_len + value_len;
3342 data[0] = prefix_len;
3343 memcpy (&data[1], field, prefix_len);
3344 memcpy (&data[1 + prefix_len], value, value_len);
3346 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3350 data->has_sdes = TRUE;
3353 /* schedule a BYE packet */
3355 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3357 GstRTCPPacket *packet = &data->packet;
3358 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3361 session_sdes (sess, data);
3362 /* add a BYE packet */
3363 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3364 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3365 if (source->bye_reason)
3366 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3368 /* we have a BYE packet now */
3369 source->sent_bye = TRUE;
3373 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3375 GstClockTime new_send_time, elapsed;
3376 GstClockTime interval;
3377 RTPSessionStats *stats;
3379 if (sess->scheduled_bye)
3380 stats = &sess->bye_stats;
3382 stats = &sess->stats;
3384 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3385 data->is_early = TRUE;
3387 data->is_early = FALSE;
3389 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3390 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3391 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3392 GST_TIME_ARGS (current_time));
3396 /* no need to check yet */
3397 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3398 sess->next_rtcp_check_time > current_time) {
3399 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3400 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3401 GST_TIME_ARGS (current_time));
3406 /* get elapsed time since we last reported */
3407 elapsed = current_time - sess->last_rtcp_send_time;
3409 /* take interval and add jitter */
3410 interval = data->interval;
3411 if (interval != GST_CLOCK_TIME_NONE)
3412 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3414 /* perform forward reconsideration */
3415 if (interval != GST_CLOCK_TIME_NONE) {
3416 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3417 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3418 new_send_time = interval + sess->last_rtcp_send_time;
3420 new_send_time = sess->last_rtcp_send_time;
3423 if (!data->is_early) {
3424 /* check if reconsideration */
3425 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3426 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3427 GST_TIME_ARGS (new_send_time));
3428 /* store new check time */
3429 sess->next_rtcp_check_time = new_send_time;
3432 sess->next_rtcp_check_time = current_time + interval;
3433 } else if (interval != GST_CLOCK_TIME_NONE) {
3434 /* Apply the rules from RFC 4585 section 3.5.3 */
3435 if (stats->min_interval != 0 && !sess->first_rtcp) {
3436 GstClockTime T_rr_current_interval =
3437 g_random_double_range (0.5, 1.5) * stats->min_interval;
3439 /* This will caused the RTCP to be suppressed if no FB packets are added */
3440 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3441 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3442 " last: %" GST_TIME_FORMAT
3443 " + T_rr_current_interval: %" GST_TIME_FORMAT
3444 " > new_send_time: %" GST_TIME_FORMAT,
3445 GST_TIME_ARGS (stats->min_interval),
3446 GST_TIME_ARGS (sess->last_rtcp_send_time),
3447 GST_TIME_ARGS (T_rr_current_interval),
3448 GST_TIME_ARGS (new_send_time));
3449 data->may_suppress = TRUE;
3454 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3455 GST_TIME_ARGS (new_send_time));
3461 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3463 g_hash_table_insert (hash_table, key, g_object_ref (source));
3467 remove_closing_sources (const gchar * key, RTPSource * source,
3470 if (source->closing)
3473 if (source->send_fir)
3474 data->have_fir = TRUE;
3475 if (source->send_pli)
3476 data->have_pli = TRUE;
3477 if (source->send_nack)
3478 data->have_nack = TRUE;
3484 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3486 RTPSession *sess = data->sess;
3487 gboolean is_bye = FALSE;
3488 ReportOutput *output;
3490 /* only generate RTCP for active internal sources */
3491 if (!source->internal || source->sent_bye)
3494 /* ignore other sources when we do the timeout after a scheduled BYE */
3495 if (sess->scheduled_bye && !source->marked_bye)
3498 data->source = source;
3501 session_start_rtcp (sess, data);
3503 if (source->marked_bye) {
3505 make_source_bye (sess, source, data);
3507 } else if (!data->is_early) {
3508 /* loop over all known sources and add report blocks. If we are early, we
3509 * just make a minimal RTCP packet and skip this step */
3510 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3511 (GHFunc) session_report_blocks, data);
3513 if (!data->has_sdes)
3514 session_sdes (sess, data);
3517 session_fir (sess, data);
3520 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3521 (GHFunc) session_pli, data);
3523 if (data->have_nack)
3524 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3525 (GHFunc) session_nack, data);
3527 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3529 output = g_slice_new (ReportOutput);
3530 output->source = g_object_ref (source);
3531 output->is_bye = is_bye;
3532 output->buffer = data->rtcp;
3533 /* queue the RTCP packet to push later */
3534 g_queue_push_tail (&data->output, output);
3538 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3540 RTPSession *sess = data->sess;
3542 if (g_hash_table_size (source->reported_in_sr_of) >=
3543 sess->stats.internal_sources) {
3544 /* source is reported, move to next generation */
3545 source->generation = sess->generation + 1;
3546 g_hash_table_remove_all (source->reported_in_sr_of);
3548 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3549 source->generation);
3551 /* if we reported all sources in this generation, move to next */
3552 if (--data->num_to_report == 0) {
3554 GST_DEBUG ("all reported, generation now %u", sess->generation);
3560 * rtp_session_on_timeout:
3561 * @sess: an #RTPSession
3562 * @current_time: the current system time
3563 * @ntpnstime: the current NTP time in nanoseconds
3564 * @running_time: the current running_time of the pipeline
3566 * Perform maintenance actions after the timeout obtained with
3567 * rtp_session_next_timeout() expired.
3569 * This function will perform timeouts of receivers and senders, send a BYE
3570 * packet or generate RTCP packets with current session stats.
3572 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3573 * times, for each packet that should be processed.
3575 * Returns: a #GstFlowReturn.
3578 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3579 guint64 ntpnstime, GstClockTime running_time)
3581 GstFlowReturn result = GST_FLOW_OK;
3582 ReportData data = { GST_RTCP_BUFFER_INIT };
3583 GHashTable *table_copy;
3584 ReportOutput *output;
3586 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3588 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3589 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3590 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3593 data.current_time = current_time;
3594 data.ntpnstime = ntpnstime;
3595 data.running_time = running_time;
3596 data.num_to_report = 0;
3597 data.may_suppress = FALSE;
3598 data.nacked_seqnums = 0;
3599 g_queue_init (&data.output);
3601 RTP_SESSION_LOCK (sess);
3602 /* get a new interval, we need this for various cleanups etc */
3603 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3605 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3607 /* we need an internal source now */
3608 if (sess->stats.internal_sources == 0) {
3612 source = obtain_internal_source (sess, sess->suggested_ssrc, &created);
3613 g_object_unref (source);
3616 /* Make a local copy of the hashtable. We need to do this because the
3617 * cleanup stage below releases the session lock. */
3618 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3619 (GDestroyNotify) g_object_unref);
3620 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3621 (GHFunc) clone_ssrcs_hashtable, table_copy);
3623 /* Clean up the session, mark the source for removing, this might release the
3625 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3626 g_hash_table_destroy (table_copy);
3628 /* Now remove the marked sources */
3629 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3630 (GHRFunc) remove_closing_sources, &data);
3632 /* update point-to-point status */
3633 session_update_ptp (sess);
3635 /* see if we need to generate SR or RR packets */
3636 if (!is_rtcp_time (sess, current_time, &data))
3639 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3640 sess->generation, data.num_to_report, data.is_early);
3642 /* generate RTCP for all internal sources */
3643 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3644 (GHFunc) generate_rtcp, &data);
3646 /* update the generation for all the sources that have been reported */
3647 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3648 (GHFunc) update_generation, &data);
3650 /* we keep track of the last report time in order to timeout inactive
3651 * receivers or senders */
3652 if (!data.is_early && !data.may_suppress)
3653 sess->last_rtcp_send_time = data.current_time;
3654 sess->first_rtcp = FALSE;
3655 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3656 sess->scheduled_bye = FALSE;
3659 RTP_SESSION_UNLOCK (sess);
3661 /* push out the RTCP packets */
3662 while ((output = g_queue_pop_head (&data.output))) {
3663 gboolean do_not_suppress;
3664 GstBuffer *buffer = output->buffer;
3665 RTPSource *source = output->source;
3667 /* Give the user a change to add its own packet */
3668 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3669 buffer, data.is_early, &do_not_suppress);
3671 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3674 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3676 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3677 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3678 sess->stats.avg_rtcp_packet_size, packet_size);
3680 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3681 sess->send_rtcp_user_data);
3682 sess->stats.nacks_sent += data.nacked_seqnums;
3684 GST_DEBUG ("freeing packet callback: %p"
3685 " do_not_suppress: %d may_suppress: %d",
3686 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3687 sess->stats.nacks_dropped += data.nacked_seqnums;
3688 gst_buffer_unref (buffer);
3690 g_object_unref (source);
3691 g_slice_free (ReportOutput, output);
3697 * rtp_session_request_early_rtcp:
3698 * @sess: an #RTPSession
3699 * @current_time: the current system time
3700 * @max_delay: maximum delay
3702 * Request transmission of early RTCP
3705 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3706 GstClockTime max_delay)
3708 GstClockTime T_dither_max;
3710 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3712 RTP_SESSION_LOCK (sess);
3714 /* Check if already requested */
3715 /* RFC 4585 section 3.5.2 step 2 */
3716 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3717 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3721 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3722 GST_LOG_OBJECT (sess, "no next RTCP check time");
3726 /* Ignore the request a scheduled packet will be in time anyway */
3727 if (current_time + max_delay > sess->next_rtcp_check_time) {
3728 GST_LOG_OBJECT (sess, "next scheduled time is soon %" GST_TIME_FORMAT " + %"
3729 GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
3730 GST_TIME_ARGS (current_time),
3731 GST_TIME_ARGS (max_delay), GST_TIME_ARGS (sess->next_rtcp_check_time));
3735 /* RFC 4585 section 3.5.2 step 2b */
3736 /* If the total sources is <=2, then there is only us and one peer */
3737 /* When there is one auxiliary stream the session can still do point
3740 if (sess->is_doing_ptp) {
3743 /* Divide by 2 because l = 0.5 */
3744 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3748 /* RFC 4585 section 3.5.2 step 3 */
3749 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3750 GST_LOG_OBJECT (sess, "don't send because of dither");
3754 /* RFC 4585 section 3.5.2 step 4
3755 * Don't send if allow_early is FALSE, but not if we are in
3756 * immediate mode, meaning we are part of a group of at most the
3757 * application-specific threshold.
3759 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3760 sess->allow_early == FALSE) {
3761 GST_LOG_OBJECT (sess, "can't allow early feedback");
3766 /* Schedule an early transmission later */
3767 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3770 /* If no dithering, schedule it for NOW */
3771 sess->next_early_rtcp_time = current_time;
3774 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3775 GST_TIME_ARGS (sess->next_early_rtcp_time));
3776 RTP_SESSION_UNLOCK (sess);
3778 /* notify app of need to send packet early
3779 * and therefore of timeout change */
3780 if (sess->callbacks.reconsider)
3781 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3787 RTP_SESSION_UNLOCK (sess);
3791 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3795 if (!sess->callbacks.send_rtcp)
3798 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3800 rtp_session_request_early_rtcp (sess, now, max_delay);
3804 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3805 gboolean fir, gint count)
3809 RTP_SESSION_LOCK (sess);
3810 src = find_source (sess, ssrc);
3815 src->send_pli = FALSE;
3816 src->send_fir = TRUE;
3818 if (count == -1 || count != src->last_fir_count)
3819 src->current_send_fir_seqnum++;
3820 src->last_fir_count = count;
3821 } else if (!src->send_fir) {
3822 src->send_pli = TRUE;
3824 RTP_SESSION_UNLOCK (sess);
3826 rtp_session_send_rtcp (sess, 200 * GST_MSECOND);
3833 RTP_SESSION_UNLOCK (sess);
3839 * rtp_session_request_nack:
3840 * @sess: a #RTPSession
3842 * @seqnum: the missing seqnum
3843 * @max_delay: max delay to request NACK
3845 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
3847 * Returns: %TRUE if the NACK feedback could be scheduled
3850 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
3851 GstClockTime max_delay)
3855 RTP_SESSION_LOCK (sess);
3856 source = find_source (sess, ssrc);
3860 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
3861 rtp_source_register_nack (source, seqnum);
3862 RTP_SESSION_UNLOCK (sess);
3864 rtp_session_send_rtcp (sess, max_delay);
3871 RTP_SESSION_UNLOCK (sess);