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 /* GObject vmethods */
104 static void rtp_session_finalize (GObject * object);
105 static void rtp_session_set_property (GObject * object, guint prop_id,
106 const GValue * value, GParamSpec * pspec);
107 static void rtp_session_get_property (GObject * object, guint prop_id,
108 GValue * value, GParamSpec * pspec);
110 static gboolean rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay);
112 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
114 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
116 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
117 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
118 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
119 static RTPSource *obtain_internal_source (RTPSession * sess,
120 guint32 ssrc, gboolean * created, GstClockTime current_time);
121 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
122 GstClockTime current_time);
123 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
124 gboolean deterministic, gboolean first);
127 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
128 const GValue * handler_return, gpointer data)
130 if (g_value_get_boolean (handler_return))
131 g_value_set_boolean (return_accu, TRUE);
137 rtp_session_class_init (RTPSessionClass * klass)
139 GObjectClass *gobject_class;
141 gobject_class = (GObjectClass *) klass;
143 gobject_class->finalize = rtp_session_finalize;
144 gobject_class->set_property = rtp_session_set_property;
145 gobject_class->get_property = rtp_session_get_property;
148 * RTPSession::get-source-by-ssrc:
149 * @session: the object which received the signal
150 * @ssrc: the SSRC of the RTPSource
152 * Request the #RTPSource object with SSRC @ssrc in @session.
154 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
155 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
156 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
157 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
158 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
161 * RTPSession::on-new-ssrc:
162 * @session: the object which received the signal
163 * @src: the new RTPSource
165 * Notify of a new SSRC that entered @session.
167 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
168 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
169 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
170 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
173 * RTPSession::on-ssrc-collision:
174 * @session: the object which received the signal
175 * @src: the #RTPSource that caused a collision
177 * Notify when we have an SSRC collision
179 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
180 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
181 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
182 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
185 * RTPSession::on-ssrc-validated:
186 * @session: the object which received the signal
187 * @src: the new validated RTPSource
189 * Notify of a new SSRC that became validated.
191 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
192 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
193 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
194 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
197 * RTPSession::on-ssrc-active:
198 * @session: the object which received the signal
199 * @src: the active RTPSource
201 * Notify of a SSRC that is active, i.e., sending RTCP.
203 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
204 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
206 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
209 * RTPSession::on-ssrc-sdes:
210 * @session: the object which received the signal
211 * @src: the RTPSource
213 * Notify that a new SDES was received for SSRC.
215 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
216 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
217 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
218 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
221 * RTPSession::on-bye-ssrc:
222 * @session: the object which received the signal
223 * @src: the RTPSource that went away
225 * Notify of an SSRC that became inactive because of a BYE packet.
227 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
228 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
229 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
230 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
233 * RTPSession::on-bye-timeout:
234 * @session: the object which received the signal
235 * @src: the RTPSource that timed out
237 * Notify of an SSRC that has timed out because of BYE
239 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
240 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
241 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
242 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
245 * RTPSession::on-timeout:
246 * @session: the object which received the signal
247 * @src: the RTPSource that timed out
249 * Notify of an SSRC that has timed out
251 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
252 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
253 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
254 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
257 * RTPSession::on-sender-timeout:
258 * @session: the object which received the signal
259 * @src: the RTPSource that timed out
261 * Notify of an SSRC that was a sender but timed out and became a receiver.
263 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
264 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
265 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
266 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
270 * RTPSession::on-sending-rtcp
271 * @session: the object which received the signal
272 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
273 * @early: %TRUE if the packet is early, %FALSE if it is regular
275 * This signal is emitted before sending an RTCP packet, it can be used
276 * to add extra RTCP Packets.
278 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
279 * if suppressing it is acceptable
281 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
282 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
283 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
284 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
285 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
288 * RTPSession::on-feedback-rtcp:
289 * @session: the object which received the signal
290 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
291 * %GST_RTCP_TYPE_RTPFB
292 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
293 * @sender_ssrc: The SSRC of the sender
294 * @media_ssrc: The SSRC of the media this refers to
295 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
298 * Notify that a RTCP feedback packet has been received
300 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
301 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
302 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
303 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
304 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
307 * RTPSession::send-rtcp:
308 * @session: the object which received the signal
309 * @max_delay: The maximum delay after which the feedback will not be useful
312 * Requests that the #RTPSession initiate a new RTCP packet as soon as
313 * possible within the requested delay.
315 rtp_session_signals[SIGNAL_SEND_RTCP] =
316 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
317 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
318 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
319 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
321 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
322 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
323 "The internal SSRC used for the session (deprecated)",
324 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
326 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
327 g_param_spec_object ("internal-source", "Internal Source",
328 "The internal source element of the session (deprecated)",
329 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
331 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
332 g_param_spec_double ("bandwidth", "Bandwidth",
333 "The bandwidth of the session (0 for auto-discover)",
334 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
335 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
337 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
338 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
339 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
340 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
341 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
344 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
345 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
346 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
347 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
350 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
351 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
352 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
353 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
356 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
357 "The maximum size of the RTCP packets",
358 16, G_MAXINT16, DEFAULT_RTCP_MTU,
359 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
361 g_object_class_install_property (gobject_class, PROP_SDES,
362 g_param_spec_boxed ("sdes", "SDES",
363 "The SDES items of this session",
364 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
366 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
367 g_param_spec_uint ("num-sources", "Num Sources",
368 "The number of sources in the session", 0, G_MAXUINT,
369 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
371 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
372 g_param_spec_uint ("num-active-sources", "Num Active Sources",
373 "The number of active sources in the session", 0, G_MAXUINT,
374 DEFAULT_NUM_ACTIVE_SOURCES,
375 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
379 * Get a GValue Array of all sources in the session.
382 * <title>Getting the #RTPSources of a session
389 * g_object_get (sess, "sources", &arr, NULL);
391 * for (i = 0; i < arr->n_values; i++) {
394 * val = g_value_array_get_nth (arr, i);
395 * source = g_value_get_object (val);
397 * g_value_array_free (arr);
402 g_object_class_install_property (gobject_class, PROP_SOURCES,
403 g_param_spec_boxed ("sources", "Sources",
404 "An array of all known sources in the session",
405 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
407 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
408 g_param_spec_boolean ("favor-new", "Favor new sources",
409 "Resolve SSRC conflict in favor of new sources", FALSE,
410 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
412 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
413 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
414 "Minimum interval between Regular RTCP packet (in ns)",
415 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
416 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
418 g_object_class_install_property (gobject_class,
419 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
420 g_param_spec_uint64 ("rtcp-feedback-retention-window",
421 "RTCP Feedback retention window",
422 "Duration during which RTCP Feedback packets are retained (in ns)",
423 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
424 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
426 g_object_class_install_property (gobject_class,
427 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
428 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
429 "RTCP Immediate Feedback threshold",
430 "The maximum number of members of a RTP session for which immediate"
432 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
433 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
435 g_object_class_install_property (gobject_class, PROP_PROBATION,
436 g_param_spec_uint ("probation", "Number of probations",
437 "Consecutive packet sequence numbers to accept the source",
438 0, G_MAXUINT, DEFAULT_PROBATION,
439 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
444 * Various session statistics. This property returns a GstStructure
445 * with name application/x-rtp-session-stats with the following fields:
447 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
448 * dropped (due to bandwidth constraints)
449 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
450 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
454 g_object_class_install_property (gobject_class, PROP_STATS,
455 g_param_spec_boxed ("stats", "Statistics",
456 "Various statistics", GST_TYPE_STRUCTURE,
457 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
459 klass->get_source_by_ssrc =
460 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
461 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
463 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
467 rtp_session_init (RTPSession * sess)
472 g_mutex_init (&sess->lock);
473 sess->key = g_random_int ();
477 for (i = 0; i < 32; i++) {
479 g_hash_table_new_full (NULL, NULL, NULL,
480 (GDestroyNotify) g_object_unref);
483 rtp_stats_init_defaults (&sess->stats);
484 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
485 rtp_stats_set_min_interval (&sess->stats,
486 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
488 sess->recalc_bandwidth = TRUE;
489 sess->bandwidth = DEFAULT_BANDWIDTH;
490 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
491 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
492 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
494 /* default UDP header length */
495 sess->header_len = 28;
496 sess->mtu = DEFAULT_RTCP_MTU;
498 sess->probation = DEFAULT_PROBATION;
500 /* some default SDES entries */
501 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
503 /* we do not want to leak details like the username or hostname here */
504 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
505 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
509 /* we do not want to leak the user's real name here */
510 str = g_strdup_printf ("Anon%u", g_random_int ());
511 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
515 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
517 /* this is the SSRC we suggest */
518 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
520 sess->first_rtcp = TRUE;
521 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
523 sess->allow_early = TRUE;
524 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
525 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
526 sess->rtcp_immediate_feedback_threshold =
527 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
529 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
531 sess->is_doing_ptp = TRUE;
535 rtp_session_finalize (GObject * object)
540 sess = RTP_SESSION_CAST (object);
542 gst_structure_free (sess->sdes);
544 g_list_free_full (sess->conflicting_addresses,
545 (GDestroyNotify) rtp_conflicting_address_free);
547 for (i = 0; i < 32; i++)
548 g_hash_table_destroy (sess->ssrcs[i]);
550 g_mutex_clear (&sess->lock);
552 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
556 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
558 GValue value = { 0 };
560 g_value_init (&value, RTP_TYPE_SOURCE);
561 g_value_take_object (&value, source);
562 /* copies the value */
563 g_value_array_append (arr, &value);
567 rtp_session_create_sources (RTPSession * sess)
572 RTP_SESSION_LOCK (sess);
573 /* get number of elements in the table */
574 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
575 /* create the result value array */
576 res = g_value_array_new (size);
578 /* and copy all values into the array */
579 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
580 RTP_SESSION_UNLOCK (sess);
585 static GstStructure *
586 rtp_session_create_stats (RTPSession * sess)
590 s = gst_structure_new ("application/x-rtp-session-stats",
591 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
592 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
593 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
599 rtp_session_set_property (GObject * object, guint prop_id,
600 const GValue * value, GParamSpec * pspec)
604 sess = RTP_SESSION (object);
607 case PROP_INTERNAL_SSRC:
608 RTP_SESSION_LOCK (sess);
609 sess->suggested_ssrc = g_value_get_uint (value);
610 RTP_SESSION_UNLOCK (sess);
611 if (sess->callbacks.reconfigure)
612 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
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;
894 if (callbacks->reconfigure) {
895 sess->callbacks.reconfigure = callbacks->reconfigure;
896 sess->reconfigure_user_data = user_data;
901 * rtp_session_set_process_rtp_callback:
902 * @sess: an #RTPSession
903 * @callback: callback to set
904 * @user_data: user data passed in the callback
906 * Configure only the process_rtp callback to be notified of the process_rtp action.
909 rtp_session_set_process_rtp_callback (RTPSession * sess,
910 RTPSessionProcessRTP callback, gpointer user_data)
912 g_return_if_fail (RTP_IS_SESSION (sess));
914 sess->callbacks.process_rtp = callback;
915 sess->process_rtp_user_data = user_data;
919 * rtp_session_set_send_rtp_callback:
920 * @sess: an #RTPSession
921 * @callback: callback to set
922 * @user_data: user data passed in the callback
924 * Configure only the send_rtp callback to be notified of the send_rtp action.
927 rtp_session_set_send_rtp_callback (RTPSession * sess,
928 RTPSessionSendRTP callback, gpointer user_data)
930 g_return_if_fail (RTP_IS_SESSION (sess));
932 sess->callbacks.send_rtp = callback;
933 sess->send_rtp_user_data = user_data;
937 * rtp_session_set_send_rtcp_callback:
938 * @sess: an #RTPSession
939 * @callback: callback to set
940 * @user_data: user data passed in the callback
942 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
945 rtp_session_set_send_rtcp_callback (RTPSession * sess,
946 RTPSessionSendRTCP callback, gpointer user_data)
948 g_return_if_fail (RTP_IS_SESSION (sess));
950 sess->callbacks.send_rtcp = callback;
951 sess->send_rtcp_user_data = user_data;
955 * rtp_session_set_sync_rtcp_callback:
956 * @sess: an #RTPSession
957 * @callback: callback to set
958 * @user_data: user data passed in the callback
960 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
963 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
964 RTPSessionSyncRTCP callback, gpointer user_data)
966 g_return_if_fail (RTP_IS_SESSION (sess));
968 sess->callbacks.sync_rtcp = callback;
969 sess->sync_rtcp_user_data = user_data;
973 * rtp_session_set_clock_rate_callback:
974 * @sess: an #RTPSession
975 * @callback: callback to set
976 * @user_data: user data passed in the callback
978 * Configure only the clock_rate callback to be notified of the clock_rate action.
981 rtp_session_set_clock_rate_callback (RTPSession * sess,
982 RTPSessionClockRate callback, gpointer user_data)
984 g_return_if_fail (RTP_IS_SESSION (sess));
986 sess->callbacks.clock_rate = callback;
987 sess->clock_rate_user_data = user_data;
991 * rtp_session_set_reconsider_callback:
992 * @sess: an #RTPSession
993 * @callback: callback to set
994 * @user_data: user data passed in the callback
996 * Configure only the reconsider callback to be notified of the reconsider action.
999 rtp_session_set_reconsider_callback (RTPSession * sess,
1000 RTPSessionReconsider callback, gpointer user_data)
1002 g_return_if_fail (RTP_IS_SESSION (sess));
1004 sess->callbacks.reconsider = callback;
1005 sess->reconsider_user_data = user_data;
1009 * rtp_session_set_request_time_callback:
1010 * @sess: an #RTPSession
1011 * @callback: callback to set
1012 * @user_data: user data passed in the callback
1014 * Configure only the request_time callback
1017 rtp_session_set_request_time_callback (RTPSession * sess,
1018 RTPSessionRequestTime callback, gpointer user_data)
1020 g_return_if_fail (RTP_IS_SESSION (sess));
1022 sess->callbacks.request_time = callback;
1023 sess->request_time_user_data = user_data;
1027 * rtp_session_set_bandwidth:
1028 * @sess: an #RTPSession
1029 * @bandwidth: the bandwidth allocated
1031 * Set the session bandwidth in bytes per second.
1034 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1036 g_return_if_fail (RTP_IS_SESSION (sess));
1038 RTP_SESSION_LOCK (sess);
1039 sess->stats.bandwidth = bandwidth;
1040 RTP_SESSION_UNLOCK (sess);
1044 * rtp_session_get_bandwidth:
1045 * @sess: an #RTPSession
1047 * Get the session bandwidth.
1049 * Returns: the session bandwidth.
1052 rtp_session_get_bandwidth (RTPSession * sess)
1056 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1058 RTP_SESSION_LOCK (sess);
1059 result = sess->stats.bandwidth;
1060 RTP_SESSION_UNLOCK (sess);
1066 * rtp_session_set_rtcp_fraction:
1067 * @sess: an #RTPSession
1068 * @bandwidth: the RTCP bandwidth
1070 * Set the bandwidth in bytes per second that should be used for RTCP
1074 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1076 g_return_if_fail (RTP_IS_SESSION (sess));
1078 RTP_SESSION_LOCK (sess);
1079 sess->stats.rtcp_bandwidth = bandwidth;
1080 RTP_SESSION_UNLOCK (sess);
1084 * rtp_session_get_rtcp_fraction:
1085 * @sess: an #RTPSession
1087 * Get the session bandwidth used for RTCP.
1089 * Returns: The bandwidth used for RTCP messages.
1092 rtp_session_get_rtcp_fraction (RTPSession * sess)
1096 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1098 RTP_SESSION_LOCK (sess);
1099 result = sess->stats.rtcp_bandwidth;
1100 RTP_SESSION_UNLOCK (sess);
1106 * rtp_session_get_sdes_struct:
1107 * @sess: an #RTSPSession
1109 * Get the SDES data as a #GstStructure
1111 * Returns: a GstStructure with SDES items for @sess. This function returns a
1112 * copy of the SDES structure, use gst_structure_free() after usage.
1115 rtp_session_get_sdes_struct (RTPSession * sess)
1117 GstStructure *result = NULL;
1119 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1121 RTP_SESSION_LOCK (sess);
1123 result = gst_structure_copy (sess->sdes);
1124 RTP_SESSION_UNLOCK (sess);
1130 * rtp_session_set_sdes_struct:
1131 * @sess: an #RTSPSession
1132 * @sdes: a #GstStructure
1134 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1137 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1139 g_return_if_fail (sdes);
1140 g_return_if_fail (RTP_IS_SESSION (sess));
1142 RTP_SESSION_LOCK (sess);
1144 gst_structure_free (sess->sdes);
1145 sess->sdes = gst_structure_copy (sdes);
1146 RTP_SESSION_UNLOCK (sess);
1149 static GstFlowReturn
1150 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1152 GstFlowReturn result = GST_FLOW_OK;
1154 if (source->internal) {
1155 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1157 RTP_SESSION_UNLOCK (session);
1159 if (session->callbacks.send_rtp)
1161 session->callbacks.send_rtp (session, source, data,
1162 session->send_rtp_user_data);
1164 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1167 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1168 RTP_SESSION_UNLOCK (session);
1170 if (session->callbacks.process_rtp)
1172 session->callbacks.process_rtp (session, source,
1173 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1175 gst_buffer_unref (GST_BUFFER_CAST (data));
1177 RTP_SESSION_LOCK (session);
1183 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1187 RTP_SESSION_UNLOCK (session);
1189 if (session->callbacks.clock_rate)
1191 session->callbacks.clock_rate (session, pt,
1192 session->clock_rate_user_data);
1196 RTP_SESSION_LOCK (session);
1198 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1203 static RTPSourceCallbacks callbacks = {
1204 (RTPSourcePushRTP) source_push_rtp,
1205 (RTPSourceClockRate) source_clock_rate,
1210 * rtp_session_find_conflicting_address:
1211 * @session: The session the packet came in
1212 * @address: address to check for
1213 * @time: The time when the packet that is possibly in conflict arrived
1215 * Checks if an address which has a conflict is already known. If it is
1216 * a known conflict, remember the time
1218 * Returns: TRUE if it was a known conflict, FALSE otherwise
1221 rtp_session_find_conflicting_address (RTPSession * session,
1222 GSocketAddress * address, GstClockTime time)
1224 return find_conflicting_address (session->conflicting_addresses, address,
1229 * rtp_session_add_conflicting_address:
1230 * @session: The session the packet came in
1231 * @address: address to remember
1232 * @time: The time when the packet that is in conflict arrived
1234 * Adds a new conflict address
1237 rtp_session_add_conflicting_address (RTPSession * sess,
1238 GSocketAddress * address, GstClockTime time)
1240 sess->conflicting_addresses =
1241 add_conflicting_address (sess->conflicting_addresses, address, time);
1246 check_collision (RTPSession * sess, RTPSource * source,
1247 RTPPacketInfo * pinfo, gboolean rtp)
1251 /* If we have no pinfo address, we can't do collision checking */
1252 if (!pinfo->address)
1255 ssrc = rtp_source_get_ssrc (source);
1257 if (!source->internal) {
1258 GSocketAddress *from;
1260 /* This is not our local source, but lets check if two remote
1263 from = source->rtp_from;
1265 from = source->rtcp_from;
1269 if (__g_socket_address_equal (from, pinfo->address)) {
1270 /* Address is the same */
1273 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1274 if (sess->favor_new) {
1275 if (rtp_source_find_conflicting_address (source,
1276 pinfo->address, pinfo->current_time)) {
1279 buf1 = __g_socket_address_to_string (pinfo->address);
1280 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1288 /* Current address is not a known conflict, lets assume this is
1289 * a new source. Save old address in possible conflict list
1291 rtp_source_add_conflicting_address (source, from,
1292 pinfo->current_time);
1294 buf1 = __g_socket_address_to_string (from);
1295 buf2 = __g_socket_address_to_string (pinfo->address);
1297 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1298 " saving old as known conflict", ssrc, buf1, buf2);
1301 rtp_source_set_rtp_from (source, pinfo->address);
1303 rtp_source_set_rtcp_from (source, pinfo->address);
1311 /* Don't need to save old addresses, we ignore new sources */
1316 /* We don't already have a from address for RTP, just set it */
1318 rtp_source_set_rtp_from (source, pinfo->address);
1320 rtp_source_set_rtcp_from (source, pinfo->address);
1324 /* FIXME: Log 3rd party collision somehow
1325 * Maybe should be done in upper layer, only the SDES can tell us
1326 * if its a collision or a loop
1329 /* This is sending with our ssrc, is it an address we already know */
1330 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1331 pinfo->current_time)) {
1332 /* Its a known conflict, its probably a loop, not a collision
1333 * lets just drop the incoming packet
1335 GST_DEBUG ("Our packets are being looped back to us, dropping");
1337 /* Its a new collision, lets change our SSRC */
1338 rtp_session_add_conflicting_address (sess, pinfo->address,
1339 pinfo->current_time);
1341 GST_DEBUG ("Collision for SSRC %x", ssrc);
1342 /* mark the source BYE */
1343 rtp_source_mark_bye (source, "SSRC Collision");
1344 /* if we were suggesting this SSRC, change to something else */
1345 if (sess->suggested_ssrc == ssrc)
1346 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1348 on_ssrc_collision (sess, source);
1350 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1359 gboolean is_doing_ptp;
1360 GSocketAddress *new_addr;
1363 /* check if the two given ip addr are the same (do not care about the port) */
1365 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1368 g_inet_address_equal (g_inet_socket_address_get_address
1369 (G_INET_SOCKET_ADDRESS (a)),
1370 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1374 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1375 CompareAddrData * data)
1377 /* only compare ip addr of remote sources which are also not closing */
1378 if (!source->internal && !source->closing && source->rtp_from) {
1379 /* look for the first rtp source */
1380 if (!data->new_addr)
1381 data->new_addr = source->rtp_from;
1382 /* compare current ip addr with the first one */
1384 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1389 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1390 CompareAddrData * data)
1392 /* only compare ip addr of remote sources which are also not closing */
1393 if (!source->internal && !source->closing && source->rtcp_from) {
1394 /* look for the first rtcp source */
1395 if (!data->new_addr)
1396 data->new_addr = source->rtcp_from;
1398 /* compare current ip addr with the first one */
1399 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1403 /* loop over our non-internal source to know if the session
1404 * is doing point-to-point */
1406 session_update_ptp (RTPSession * sess)
1408 /* to know if the session is doing point to point, the ip addr
1409 * of each non-internal (=remotes) source have to be compared
1412 gboolean is_doing_rtp_ptp;
1413 gboolean is_doing_rtcp_ptp;
1414 CompareAddrData data;
1416 /* compare the first remote source's ip addr that receive rtp packets
1417 * with other remote rtp source.
1418 * it's enough because the session just needs to know if they are all
1421 data.is_doing_ptp = TRUE;
1422 data.new_addr = NULL;
1423 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1424 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1425 is_doing_rtp_ptp = data.is_doing_ptp;
1427 /* same but about rtcp */
1428 data.is_doing_ptp = TRUE;
1429 data.new_addr = NULL;
1430 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1431 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1432 is_doing_rtcp_ptp = data.is_doing_ptp;
1434 /* the session is doing point-to-point if all rtp remote have the same
1435 * ip addr and if all rtcp remote sources have the same ip addr */
1436 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1438 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1442 add_source (RTPSession * sess, RTPSource * src)
1444 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1445 GINT_TO_POINTER (src->ssrc), src);
1446 /* report the new source ASAP */
1447 src->generation = sess->generation;
1448 /* we have one more source now */
1449 sess->total_sources++;
1450 if (RTP_SOURCE_IS_ACTIVE (src))
1451 sess->stats.active_sources++;
1452 if (src->internal) {
1453 sess->stats.internal_sources++;
1454 if (sess->suggested_ssrc != src->ssrc)
1455 sess->suggested_ssrc = src->ssrc;
1458 /* update point-to-point status */
1460 session_update_ptp (sess);
1464 find_source (RTPSession * sess, guint32 ssrc)
1466 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1467 GINT_TO_POINTER (ssrc));
1470 /* must be called with the session lock, the returned source needs to be
1471 * unreffed after usage. */
1473 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1474 RTPPacketInfo * pinfo, gboolean rtp)
1478 source = find_source (sess, ssrc);
1479 if (source == NULL) {
1480 /* make new Source in probation and insert */
1481 source = rtp_source_new (ssrc);
1483 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1485 /* for RTP packets we need to set the source in probation. Receiving RTCP
1486 * packets of an SSRC, on the other hand, is a strong indication that we
1487 * are dealing with a valid source. */
1489 g_object_set (source, "probation", sess->probation, NULL);
1491 g_object_set (source, "probation", 0, NULL);
1493 /* store from address, if any */
1494 if (pinfo->address) {
1496 rtp_source_set_rtp_from (source, pinfo->address);
1498 rtp_source_set_rtcp_from (source, pinfo->address);
1501 /* configure a callback on the source */
1502 rtp_source_set_callbacks (source, &callbacks, sess);
1504 add_source (sess, source);
1508 /* check for collision, this updates the address when not previously set */
1509 if (check_collision (sess, source, pinfo, rtp)) {
1512 /* Receiving RTCP packets of an SSRC is a strong indication that we
1513 * are dealing with a valid source. */
1515 g_object_set (source, "probation", 0, NULL);
1517 /* update last activity */
1518 source->last_activity = pinfo->current_time;
1520 source->last_rtp_activity = pinfo->current_time;
1521 g_object_ref (source);
1526 /* must be called with the session lock, the returned source needs to be
1527 * unreffed after usage. */
1529 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1530 GstClockTime current_time)
1534 source = find_source (sess, ssrc);
1535 if (source == NULL) {
1536 /* make new internal Source and insert */
1537 source = rtp_source_new (ssrc);
1539 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1541 source->validated = TRUE;
1542 source->internal = TRUE;
1543 source->probation = FALSE;
1544 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1545 rtp_source_set_callbacks (source, &callbacks, sess);
1547 add_source (sess, source);
1552 /* update last activity */
1553 if (current_time != GST_CLOCK_TIME_NONE) {
1554 source->last_activity = current_time;
1555 source->last_rtp_activity = current_time;
1557 g_object_ref (source);
1563 * rtp_session_suggest_ssrc:
1564 * @sess: a #RTPSession
1566 * Suggest an unused SSRC in @sess.
1568 * Returns: a free unused SSRC
1571 rtp_session_suggest_ssrc (RTPSession * sess)
1575 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1577 RTP_SESSION_LOCK (sess);
1578 result = sess->suggested_ssrc;
1579 RTP_SESSION_UNLOCK (sess);
1585 * rtp_session_add_source:
1586 * @sess: a #RTPSession
1587 * @src: #RTPSource to add
1589 * Add @src to @session.
1591 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1592 * existed in the session.
1595 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1597 gboolean result = FALSE;
1600 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1601 g_return_val_if_fail (src != NULL, FALSE);
1603 RTP_SESSION_LOCK (sess);
1604 find = find_source (sess, src->ssrc);
1606 add_source (sess, src);
1609 RTP_SESSION_UNLOCK (sess);
1615 * rtp_session_get_num_sources:
1616 * @sess: an #RTPSession
1618 * Get the number of sources in @sess.
1620 * Returns: The number of sources in @sess.
1623 rtp_session_get_num_sources (RTPSession * sess)
1627 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1629 RTP_SESSION_LOCK (sess);
1630 result = sess->total_sources;
1631 RTP_SESSION_UNLOCK (sess);
1637 * rtp_session_get_num_active_sources:
1638 * @sess: an #RTPSession
1640 * Get the number of active sources in @sess. A source is considered active when
1641 * it has been validated and has not yet received a BYE RTCP message.
1643 * Returns: The number of active sources in @sess.
1646 rtp_session_get_num_active_sources (RTPSession * sess)
1650 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1652 RTP_SESSION_LOCK (sess);
1653 result = sess->stats.active_sources;
1654 RTP_SESSION_UNLOCK (sess);
1660 * rtp_session_get_source_by_ssrc:
1661 * @sess: an #RTPSession
1664 * Find the source with @ssrc in @sess.
1666 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1667 * g_object_unref() after usage.
1670 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1674 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1676 RTP_SESSION_LOCK (sess);
1677 result = find_source (sess, ssrc);
1679 g_object_ref (result);
1680 RTP_SESSION_UNLOCK (sess);
1685 /* should be called with the SESSION lock */
1687 rtp_session_create_new_ssrc (RTPSession * sess)
1692 ssrc = g_random_int ();
1694 /* see if it exists in the session, we're done if it doesn't */
1695 if (find_source (sess, ssrc) == NULL)
1703 * rtp_session_create_source:
1704 * @sess: an #RTPSession
1706 * Create an #RTPSource for use in @sess. This function will create a source
1707 * with an ssrc that is currently not used by any participants in the session.
1709 * Returns: an #RTPSource.
1712 rtp_session_create_source (RTPSession * sess)
1717 RTP_SESSION_LOCK (sess);
1718 ssrc = rtp_session_create_new_ssrc (sess);
1719 source = rtp_source_new (ssrc);
1720 rtp_source_set_callbacks (source, &callbacks, sess);
1721 /* we need an additional ref for the source in the hashtable */
1722 g_object_ref (source);
1723 add_source (sess, source);
1724 RTP_SESSION_UNLOCK (sess);
1730 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1732 GstNetAddressMeta *meta;
1734 /* get packet size including header overhead */
1735 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1739 GstRTPBuffer rtp = { NULL };
1741 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1742 goto invalid_packet;
1744 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1748 /* only keep info for first buffer */
1749 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1750 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1751 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1752 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1753 /* copy available csrc */
1754 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1755 for (i = 0; i < pinfo->csrc_count; i++)
1756 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1758 gst_rtp_buffer_unmap (&rtp);
1762 /* for netbuffer we can store the IP address to check for collisions */
1763 meta = gst_buffer_get_net_address_meta (*buffer);
1765 g_object_unref (pinfo->address);
1767 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1769 pinfo->address = NULL;
1777 GST_DEBUG ("invalid RTP packet received");
1782 /* update the RTPPacketInfo structure with the current time and other bits
1783 * about the current buffer we are handling.
1784 * This function is typically called when a validated packet is received.
1785 * This function should be called with the SESSION_LOCK
1788 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1789 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1790 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1796 pinfo->is_list = is_list;
1798 pinfo->current_time = current_time;
1799 pinfo->running_time = running_time;
1800 pinfo->ntpnstime = ntpnstime;
1801 pinfo->header_len = sess->header_len;
1803 pinfo->payload_len = 0;
1807 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1809 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1812 GstBuffer *buffer = GST_BUFFER_CAST (data);
1813 res = update_packet (&buffer, 0, pinfo);
1819 clean_packet_info (RTPPacketInfo * pinfo)
1822 g_object_unref (pinfo->address);
1824 gst_mini_object_unref (pinfo->data);
1830 source_update_active (RTPSession * sess, RTPSource * source,
1831 gboolean prevactive)
1833 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1834 guint32 ssrc = source->ssrc;
1836 if (prevactive == active)
1840 sess->stats.active_sources++;
1841 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1842 sess->stats.active_sources);
1844 sess->stats.active_sources--;
1845 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1846 sess->stats.active_sources);
1852 source_update_sender (RTPSession * sess, RTPSource * source,
1853 gboolean prevsender)
1855 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1856 guint32 ssrc = source->ssrc;
1858 if (prevsender == sender)
1862 sess->stats.sender_sources++;
1863 if (source->internal)
1864 sess->stats.internal_sender_sources++;
1865 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1866 sess->stats.sender_sources);
1868 sess->stats.sender_sources--;
1869 if (source->internal)
1870 sess->stats.internal_sender_sources--;
1871 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1872 sess->stats.sender_sources);
1878 * rtp_session_process_rtp:
1879 * @sess: and #RTPSession
1880 * @buffer: an RTP buffer
1881 * @current_time: the current system time
1882 * @running_time: the running_time of @buffer
1884 * Process an RTP buffer in the session manager. This function takes ownership
1887 * Returns: a #GstFlowReturn.
1890 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1891 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1893 GstFlowReturn result;
1897 gboolean prevsender, prevactive;
1898 RTPPacketInfo pinfo = { 0, };
1901 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1902 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1904 RTP_SESSION_LOCK (sess);
1906 /* update pinfo stats */
1907 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1908 current_time, running_time, ntpnstime)) {
1909 GST_DEBUG ("invalid RTP packet received");
1910 RTP_SESSION_UNLOCK (sess);
1911 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1916 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1920 prevsender = RTP_SOURCE_IS_SENDER (source);
1921 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1922 oldrate = source->bitrate;
1924 /* let source process the packet */
1925 result = rtp_source_process_rtp (source, &pinfo);
1927 /* source became active */
1928 if (source_update_active (sess, source, prevactive))
1929 on_ssrc_validated (sess, source);
1931 source_update_sender (sess, source, prevsender);
1933 if (oldrate != source->bitrate)
1934 sess->recalc_bandwidth = TRUE;
1937 on_new_ssrc (sess, source);
1939 if (source->validated) {
1943 /* for validated sources, we add the CSRCs as well */
1944 for (i = 0; i < pinfo.csrc_count; i++) {
1946 RTPSource *csrc_src;
1948 csrc = pinfo.csrcs[i];
1951 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
1956 GST_DEBUG ("created new CSRC: %08x", csrc);
1957 rtp_source_set_as_csrc (csrc_src);
1958 source_update_active (sess, csrc_src, FALSE);
1959 on_new_ssrc (sess, csrc_src);
1961 g_object_unref (csrc_src);
1964 g_object_unref (source);
1966 RTP_SESSION_UNLOCK (sess);
1968 clean_packet_info (&pinfo);
1975 RTP_SESSION_UNLOCK (sess);
1976 clean_packet_info (&pinfo);
1977 GST_DEBUG ("ignoring packet because its collisioning");
1983 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1984 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
1988 count = gst_rtcp_packet_get_rb_count (packet);
1989 for (i = 0; i < count; i++) {
1990 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1991 guint8 fractionlost;
1995 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1996 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1998 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2000 /* find our own source */
2001 src = find_source (sess, ssrc);
2005 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2006 /* only deal with report blocks for our session, we update the stats of
2007 * the sender of the RTCP message. We could also compare our stats against
2008 * the other sender to see if we are better or worse. */
2009 /* FIXME, need to keep track who the RB block is from */
2010 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2011 packetslost, exthighestseq, jitter, lsr, dlsr);
2014 on_ssrc_active (sess, source);
2017 /* A Sender report contains statistics about how the sender is doing. This
2018 * includes timing informataion such as the relation between RTP and NTP
2019 * timestamps and the number of packets/bytes it sent to us.
2021 * In this report is also included a set of report blocks related to how this
2022 * sender is receiving data (in case we (or somebody else) is also sending stuff
2023 * to it). This info includes the packet loss, jitter and seqnum. It also
2024 * contains information to calculate the round trip time (LSR/DLSR).
2027 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2028 RTPPacketInfo * pinfo, gboolean * do_sync)
2030 guint32 senderssrc, rtptime, packet_count, octet_count;
2033 gboolean created, prevsender;
2035 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2036 &packet_count, &octet_count);
2038 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2039 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2041 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2045 /* skip non-bye packets for sources that are marked BYE */
2046 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2049 /* don't try to do lip-sync for sources that sent a BYE */
2050 if (RTP_SOURCE_IS_MARKED_BYE (source))
2055 prevsender = RTP_SOURCE_IS_SENDER (source);
2057 /* first update the source */
2058 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2059 packet_count, octet_count);
2061 source_update_sender (sess, source, prevsender);
2064 on_new_ssrc (sess, source);
2066 rtp_session_process_rb (sess, source, packet, pinfo);
2069 g_object_unref (source);
2072 /* A receiver report contains statistics about how a receiver is doing. It
2073 * includes stuff like packet loss, jitter and the seqnum it received last. It
2074 * also contains info to calculate the round trip time.
2076 * We are only interested in how the sender of this report is doing wrt to us.
2079 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2080 RTPPacketInfo * pinfo)
2086 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2088 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2090 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2094 /* skip non-bye packets for sources that are marked BYE */
2095 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2099 on_new_ssrc (sess, source);
2101 rtp_session_process_rb (sess, source, packet, pinfo);
2104 g_object_unref (source);
2107 /* Get SDES items and store them in the SSRC */
2109 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2110 RTPPacketInfo * pinfo)
2113 gboolean more_items, more_entries;
2115 items = gst_rtcp_packet_sdes_get_item_count (packet);
2116 GST_DEBUG ("got SDES packet with %d items", items);
2118 more_items = gst_rtcp_packet_sdes_first_item (packet);
2120 while (more_items) {
2122 gboolean changed, created, prevactive;
2126 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2128 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2132 /* find src, no probation when dealing with RTCP */
2133 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2137 /* skip non-bye packets for sources that are marked BYE */
2138 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2141 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2143 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2145 while (more_entries) {
2146 GstRTCPSDESType type;
2152 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2154 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2157 if (type == GST_RTCP_SDES_PRIV) {
2158 name = g_strndup ((const gchar *) &data[1], data[0]);
2160 data += data[0] + 1;
2162 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2165 value = g_strndup ((const gchar *) data, len);
2167 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2172 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2176 /* takes ownership of sdes */
2177 changed = rtp_source_set_sdes_struct (source, sdes);
2179 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2180 source->validated = TRUE;
2183 on_new_ssrc (sess, source);
2185 /* source became active */
2186 if (source_update_active (sess, source, prevactive))
2187 on_ssrc_validated (sess, source);
2190 on_ssrc_sdes (sess, source);
2193 g_object_unref (source);
2195 more_items = gst_rtcp_packet_sdes_next_item (packet);
2200 /* BYE is sent when a client leaves the session
2203 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2204 RTPPacketInfo * pinfo)
2208 gboolean reconsider = FALSE;
2210 reason = gst_rtcp_packet_bye_get_reason (packet);
2211 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2213 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2214 for (i = 0; i < count; i++) {
2217 gboolean created, prevactive, prevsender;
2218 guint pmembers, members;
2220 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2221 GST_DEBUG ("SSRC: %08x", ssrc);
2223 /* find src and mark bye, no probation when dealing with RTCP */
2224 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2228 if (source->internal) {
2229 /* our own source, something weird with this packet */
2230 g_object_unref (source);
2234 /* store time for when we need to time out this source */
2235 source->bye_time = pinfo->current_time;
2237 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2238 prevsender = RTP_SOURCE_IS_SENDER (source);
2240 /* mark the source BYE */
2241 rtp_source_mark_bye (source, reason);
2243 pmembers = sess->stats.active_sources;
2245 source_update_active (sess, source, prevactive);
2246 source_update_sender (sess, source, prevsender);
2248 members = sess->stats.active_sources;
2250 if (!sess->scheduled_bye && members < pmembers) {
2251 /* some members went away since the previous timeout estimate.
2252 * Perform reverse reconsideration but only when we are not scheduling a
2254 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2255 pinfo->current_time < sess->next_rtcp_check_time) {
2256 GstClockTime time_remaining;
2258 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2259 sess->next_rtcp_check_time =
2260 gst_util_uint64_scale (time_remaining, members, pmembers);
2262 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2263 GST_TIME_ARGS (sess->next_rtcp_check_time));
2265 sess->next_rtcp_check_time += pinfo->current_time;
2267 /* mark pending reconsider. We only want to signal the reconsideration
2268 * once after we handled all the source in the bye packet */
2274 on_new_ssrc (sess, source);
2276 on_bye_ssrc (sess, source);
2278 g_object_unref (source);
2281 RTP_SESSION_UNLOCK (sess);
2282 /* notify app of reconsideration */
2283 if (sess->callbacks.reconsider)
2284 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2285 RTP_SESSION_LOCK (sess);
2291 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2292 RTPPacketInfo * pinfo)
2294 GST_DEBUG ("received APP");
2298 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2299 gboolean fir, GstClockTime current_time)
2301 guint32 round_trip = 0;
2303 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2305 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2306 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2309 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2310 GST_DEBUG ("Ignoring %s request because one was send without one "
2311 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2312 fir ? "FIR" : "PLI",
2313 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2314 GST_TIME_ARGS (round_trip_in_ns));;
2319 sess->last_keyframe_request = current_time;
2321 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2322 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2323 sess->callbacks.request_key_unit);
2325 RTP_SESSION_UNLOCK (sess);
2326 sess->callbacks.request_key_unit (sess, fir,
2327 sess->request_key_unit_user_data);
2328 RTP_SESSION_LOCK (sess);
2334 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2335 guint32 media_ssrc, GstClockTime current_time)
2339 if (!sess->callbacks.request_key_unit)
2342 src = find_source (sess, sender_ssrc);
2346 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2350 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2351 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2356 gboolean our_request = FALSE;
2358 if (!sess->callbacks.request_key_unit)
2364 src = find_source (sess, sender_ssrc);
2366 /* Hack because Google fails to set the sender_ssrc correctly */
2367 if (!src && sender_ssrc == 1) {
2368 GHashTableIter iter;
2370 /* we can't find the source if there are multiple */
2371 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2374 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2375 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2376 if (!src->internal && rtp_source_is_sender (src))
2384 for (position = 0; position < fci_length; position += 8) {
2385 guint8 *data = fci_data + position;
2388 ssrc = GST_READ_UINT32_BE (data);
2390 own = find_source (sess, ssrc);
2394 if (own->internal) {
2402 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2406 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2407 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2408 GstClockTime current_time)
2410 sess->stats.nacks_received++;
2412 if (!sess->callbacks.notify_nack)
2415 while (fci_length > 0) {
2416 guint16 seqnum, blp;
2418 seqnum = GST_READ_UINT16_BE (fci_data);
2419 blp = GST_READ_UINT16_BE (fci_data + 2);
2421 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2423 RTP_SESSION_UNLOCK (sess);
2424 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2425 sess->notify_nack_user_data);
2426 RTP_SESSION_LOCK (sess);
2434 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2435 RTPPacketInfo * pinfo, GstClockTime current_time)
2437 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2438 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2439 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2440 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2441 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2442 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2445 src = find_source (sess, media_ssrc);
2447 /* skip non-bye packets for sources that are marked BYE */
2448 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2451 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2452 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2454 if (g_signal_has_handler_pending (sess,
2455 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2456 GstBuffer *fci_buffer = NULL;
2458 if (fci_length > 0) {
2459 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2460 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2462 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2465 RTP_SESSION_UNLOCK (sess);
2466 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2467 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2468 RTP_SESSION_LOCK (sess);
2471 gst_buffer_unref (fci_buffer);
2474 if (src && sess->rtcp_feedback_retention_window) {
2475 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2478 if ((src && src->internal) ||
2479 /* PSFB FIR puts the media ssrc inside the FCI */
2480 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2482 case GST_RTCP_TYPE_PSFB:
2484 case GST_RTCP_PSFB_TYPE_PLI:
2485 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2488 case GST_RTCP_PSFB_TYPE_FIR:
2489 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2496 case GST_RTCP_TYPE_RTPFB:
2498 case GST_RTCP_RTPFB_TYPE_NACK:
2499 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2500 fci_data, fci_length, current_time);
2512 * rtp_session_process_rtcp:
2513 * @sess: and #RTPSession
2514 * @buffer: an RTCP buffer
2515 * @current_time: the current system time
2516 * @ntpnstime: the current NTP time in nanoseconds
2518 * Process an RTCP buffer in the session manager. This function takes ownership
2521 * Returns: a #GstFlowReturn.
2524 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2525 GstClockTime current_time, guint64 ntpnstime)
2527 GstRTCPPacket packet;
2528 gboolean more, is_bye = FALSE, do_sync = FALSE;
2529 RTPPacketInfo pinfo = { 0, };
2530 GstFlowReturn result = GST_FLOW_OK;
2531 GstRTCPBuffer rtcp = { NULL, };
2533 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2534 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2536 if (!gst_rtcp_buffer_validate (buffer))
2537 goto invalid_packet;
2539 GST_DEBUG ("received RTCP packet");
2541 RTP_SESSION_LOCK (sess);
2542 /* update pinfo stats */
2543 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2546 /* start processing the compound packet */
2547 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2548 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2552 type = gst_rtcp_packet_get_type (&packet);
2555 case GST_RTCP_TYPE_SR:
2556 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2558 case GST_RTCP_TYPE_RR:
2559 rtp_session_process_rr (sess, &packet, &pinfo);
2561 case GST_RTCP_TYPE_SDES:
2562 rtp_session_process_sdes (sess, &packet, &pinfo);
2564 case GST_RTCP_TYPE_BYE:
2566 /* don't try to attempt lip-sync anymore for streams with a BYE */
2568 rtp_session_process_bye (sess, &packet, &pinfo);
2570 case GST_RTCP_TYPE_APP:
2571 rtp_session_process_app (sess, &packet, &pinfo);
2573 case GST_RTCP_TYPE_RTPFB:
2574 case GST_RTCP_TYPE_PSFB:
2575 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2578 GST_WARNING ("got unknown RTCP packet");
2581 more = gst_rtcp_packet_move_to_next (&packet);
2584 gst_rtcp_buffer_unmap (&rtcp);
2586 /* if we are scheduling a BYE, we only want to count bye packets, else we
2587 * count everything */
2588 if (sess->scheduled_bye && is_bye) {
2589 sess->bye_stats.bye_members++;
2590 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2593 /* keep track of average packet size */
2594 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2596 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2597 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2598 RTP_SESSION_UNLOCK (sess);
2601 clean_packet_info (&pinfo);
2603 /* notify caller of sr packets in the callback */
2604 if (do_sync && sess->callbacks.sync_rtcp) {
2605 result = sess->callbacks.sync_rtcp (sess, buffer,
2606 sess->sync_rtcp_user_data);
2608 gst_buffer_unref (buffer);
2615 GST_DEBUG ("invalid RTCP packet received");
2616 gst_buffer_unref (buffer);
2622 * rtp_session_update_send_caps:
2623 * @sess: an #RTPSession
2626 * Update the caps of the sender in the rtp session.
2629 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2634 g_return_if_fail (RTP_IS_SESSION (sess));
2635 g_return_if_fail (GST_IS_CAPS (caps));
2637 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2639 s = gst_caps_get_structure (caps, 0);
2641 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2645 RTP_SESSION_LOCK (sess);
2646 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2648 rtp_source_update_caps (source, caps);
2649 g_object_unref (source);
2651 RTP_SESSION_UNLOCK (sess);
2656 * rtp_session_send_rtp:
2657 * @sess: an #RTPSession
2658 * @data: pointer to either an RTP buffer or a list of RTP buffers
2659 * @is_list: TRUE when @data is a buffer list
2660 * @current_time: the current system time
2661 * @running_time: the running time of @data
2663 * Send the RTP buffer in the session manager. This function takes ownership of
2666 * Returns: a #GstFlowReturn.
2669 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2670 GstClockTime current_time, GstClockTime running_time)
2672 GstFlowReturn result;
2674 gboolean prevsender;
2676 RTPPacketInfo pinfo = { 0, };
2679 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2680 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2682 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2684 RTP_SESSION_LOCK (sess);
2685 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2686 current_time, running_time, -1))
2687 goto invalid_packet;
2689 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
2691 prevsender = RTP_SOURCE_IS_SENDER (source);
2692 oldrate = source->bitrate;
2694 /* we use our own source to send */
2695 result = rtp_source_send_rtp (source, &pinfo);
2697 source_update_sender (sess, source, prevsender);
2699 if (oldrate != source->bitrate)
2700 sess->recalc_bandwidth = TRUE;
2701 RTP_SESSION_UNLOCK (sess);
2703 g_object_unref (source);
2704 clean_packet_info (&pinfo);
2710 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2711 RTP_SESSION_UNLOCK (sess);
2712 GST_DEBUG ("invalid RTP packet received");
2718 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2720 *bandwidth += source->bitrate;
2723 /* must be called with session lock */
2725 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2728 GstClockTime result;
2729 RTPSessionStats *stats;
2731 /* recalculate bandwidth when it changed */
2732 if (sess->recalc_bandwidth) {
2735 if (sess->bandwidth > 0)
2736 bandwidth = sess->bandwidth;
2738 /* If it is <= 0, then try to estimate the actual bandwidth */
2741 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2742 (GHFunc) add_bitrates, &bandwidth);
2745 if (bandwidth < 8000)
2746 bandwidth = RTP_STATS_BANDWIDTH;
2748 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2749 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2751 sess->recalc_bandwidth = FALSE;
2754 if (sess->scheduled_bye) {
2755 stats = &sess->bye_stats;
2756 result = rtp_stats_calculate_bye_interval (stats);
2758 stats = &sess->stats;
2759 result = rtp_stats_calculate_rtcp_interval (stats,
2760 stats->internal_sender_sources > 0, first);
2763 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2764 GST_TIME_ARGS (result), first);
2766 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2767 result = rtp_stats_add_rtcp_jitter (stats, result);
2769 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2775 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2777 if (source->internal)
2778 rtp_source_mark_bye (source, reason);
2782 * rtp_session_mark_all_bye:
2783 * @sess: an #RTPSession
2786 * Mark all internal sources of the session as BYE with @reason.
2789 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2791 g_return_if_fail (RTP_IS_SESSION (sess));
2793 RTP_SESSION_LOCK (sess);
2794 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2795 (GHFunc) source_mark_bye, (gpointer) reason);
2796 RTP_SESSION_UNLOCK (sess);
2799 /* Stop the current @sess and schedule a BYE message for the other members.
2800 * One must have the session lock to call this function
2802 static GstFlowReturn
2803 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2805 GstFlowReturn result = GST_FLOW_OK;
2806 GstClockTime interval;
2808 /* nothing to do it we already scheduled bye */
2809 if (sess->scheduled_bye)
2812 /* we schedule BYE now */
2813 sess->scheduled_bye = TRUE;
2814 /* at least one member wants to send a BYE */
2815 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2816 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2817 sess->bye_stats.bye_members = 1;
2818 sess->first_rtcp = TRUE;
2819 sess->allow_early = TRUE;
2821 /* reschedule transmission */
2822 sess->last_rtcp_send_time = current_time;
2823 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2825 if (interval != GST_CLOCK_TIME_NONE)
2826 sess->next_rtcp_check_time = current_time + interval;
2828 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2830 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2831 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2833 RTP_SESSION_UNLOCK (sess);
2834 /* notify app of reconsideration */
2835 if (sess->callbacks.reconsider)
2836 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2837 RTP_SESSION_LOCK (sess);
2844 * rtp_session_schedule_bye:
2845 * @sess: an #RTPSession
2846 * @current_time: the current system time
2848 * Schedule a BYE message for all sources marked as BYE in @sess.
2850 * Returns: a #GstFlowReturn.
2853 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2855 GstFlowReturn result;
2857 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2859 RTP_SESSION_LOCK (sess);
2860 result = rtp_session_schedule_bye_locked (sess, current_time);
2861 RTP_SESSION_UNLOCK (sess);
2867 * rtp_session_next_timeout:
2868 * @sess: an #RTPSession
2869 * @current_time: the current system time
2871 * Get the next time we should perform session maintenance tasks.
2873 * Returns: a time when rtp_session_on_timeout() should be called with the
2874 * current system time.
2877 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2879 GstClockTime result, interval = 0;
2881 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2883 RTP_SESSION_LOCK (sess);
2885 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2886 GST_DEBUG ("have early rtcp time");
2887 result = sess->next_early_rtcp_time;
2891 result = sess->next_rtcp_check_time;
2893 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2894 ", next time: %" GST_TIME_FORMAT,
2895 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2897 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2898 GST_DEBUG ("take current time as base");
2899 /* our previous check time expired, start counting from the current time
2901 result = current_time;
2904 if (sess->scheduled_bye) {
2905 if (sess->bye_stats.active_sources >= 50) {
2906 GST_DEBUG ("reconsider BYE, more than 50 sources");
2907 /* reconsider BYE if members >= 50 */
2908 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2911 if (sess->first_rtcp) {
2912 GST_DEBUG ("first RTCP packet");
2913 /* we are called for the first time */
2914 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2915 } else if (sess->next_rtcp_check_time < current_time) {
2916 GST_DEBUG ("old check time expired, getting new timeout");
2917 /* get a new timeout when we need to */
2918 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2922 if (interval != GST_CLOCK_TIME_NONE)
2925 result = GST_CLOCK_TIME_NONE;
2927 sess->next_rtcp_check_time = result;
2931 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2932 ", next time: %" GST_TIME_FORMAT,
2933 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2934 RTP_SESSION_UNLOCK (sess);
2948 GstRTCPBuffer rtcpbuf;
2951 guint num_to_report;
2956 GstClockTime current_time;
2958 GstClockTime running_time;
2959 GstClockTime interval;
2960 GstRTCPPacket packet;
2963 gboolean may_suppress;
2965 guint nacked_seqnums;
2969 session_start_rtcp (RTPSession * sess, ReportData * data)
2971 GstRTCPPacket *packet = &data->packet;
2972 RTPSource *own = data->source;
2973 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2975 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2976 data->has_sdes = FALSE;
2978 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2980 if (RTP_SOURCE_IS_SENDER (own)) {
2983 guint32 packet_count, octet_count;
2985 /* we are a sender, create SR */
2986 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2987 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2989 /* get latest stats */
2990 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2991 &ntptime, &rtptime, &packet_count, &octet_count);
2993 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2994 packet_count, octet_count);
2996 /* fill in sender report info */
2997 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2998 ntptime, rtptime, packet_count, octet_count);
3000 /* we are only receiver, create RR */
3001 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3002 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3003 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3007 /* construct a Sender or Receiver Report */
3009 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3011 RTPSession *sess = data->sess;
3012 GstRTCPPacket *packet = &data->packet;
3013 guint8 fractionlost;
3015 guint32 exthighestseq, jitter;
3018 /* don't report for sources in future generations */
3019 if (((gint16) (source->generation - sess->generation)) > 0) {
3020 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3021 source->generation, sess->generation);
3025 if (g_hash_table_contains (source->reported_in_sr_of,
3026 GUINT_TO_POINTER (data->source->ssrc))) {
3027 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3031 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3032 GST_DEBUG ("max RB count reached");
3036 /* only report about other sender */
3037 if (source == data->source)
3040 if (!RTP_SOURCE_IS_SENDER (source)) {
3041 GST_DEBUG ("source %08x not sender", source->ssrc);
3045 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3048 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3049 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3051 /* store last generated RR packet */
3052 source->last_rr.is_valid = TRUE;
3053 source->last_rr.fractionlost = fractionlost;
3054 source->last_rr.packetslost = packetslost;
3055 source->last_rr.exthighestseq = exthighestseq;
3056 source->last_rr.jitter = jitter;
3057 source->last_rr.lsr = lsr;
3058 source->last_rr.dlsr = dlsr;
3060 /* packet is not yet filled, add report block for this source. */
3061 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3062 exthighestseq, jitter, lsr, dlsr);
3065 g_hash_table_add (source->reported_in_sr_of,
3066 GUINT_TO_POINTER (data->source->ssrc));
3071 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3073 GstRTCPPacket *packet = &data->packet;
3077 if (!source->send_fir)
3080 len = gst_rtcp_packet_fb_get_fci_length (packet);
3081 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3082 /* exit because the packet is full, will put next request in a
3086 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3088 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3090 fci_data[0] = source->current_send_fir_seqnum;
3091 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3093 source->send_fir = FALSE;
3097 session_fir (RTPSession * sess, ReportData * data)
3099 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3100 GstRTCPPacket *packet = &data->packet;
3102 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3105 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3106 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3107 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3109 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3110 (GHFunc) session_add_fir, data);
3112 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3113 gst_rtcp_packet_remove (packet);
3115 data->may_suppress = FALSE;
3119 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3121 GstRTCPPacket packet;
3122 GstRTCPBuffer rtcp = { NULL, };
3123 gboolean ret = FALSE;
3125 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3127 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3128 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3129 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3133 gst_rtcp_buffer_unmap (&rtcp);
3140 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3142 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3143 GstRTCPPacket *packet = &data->packet;
3145 if (!source->send_pli)
3148 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3151 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3152 /* exit because the packet is full, will put next request in a
3156 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3157 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3158 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3160 source->send_pli = FALSE;
3161 data->may_suppress = FALSE;
3164 /* construct NACK */
3166 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3168 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3169 GstRTCPPacket *packet = &data->packet;
3174 if (!source->send_nack)
3177 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3178 /* exit because the packet is full, will put next request in a
3182 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3183 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3184 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3186 nacks = rtp_source_get_nacks (source, &n_nacks);
3187 GST_DEBUG ("%u NACKs", n_nacks);
3188 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3191 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3192 for (i = 0; i < n_nacks; i++) {
3193 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3195 data->nacked_seqnums++;
3198 rtp_source_clear_nacks (source);
3199 data->may_suppress = FALSE;
3202 /* perform cleanup of sources that timed out */
3204 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3206 gboolean remove = FALSE;
3207 gboolean byetimeout = FALSE;
3208 gboolean sendertimeout = FALSE;
3209 gboolean is_sender, is_active;
3210 RTPSession *sess = data->sess;
3211 GstClockTime interval, binterval;
3214 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3216 /* check for outdated collisions */
3217 if (source->internal) {
3218 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3219 rtp_source_timeout (source, data->current_time,
3220 data->running_time - sess->rtcp_feedback_retention_window);
3223 /* nothing else to do when without RTCP */
3224 if (data->interval == GST_CLOCK_TIME_NONE)
3227 is_sender = RTP_SOURCE_IS_SENDER (source);
3228 is_active = RTP_SOURCE_IS_ACTIVE (source);
3230 /* our own rtcp interval may have been forced low by secondary configuration,
3231 * while sender side may still operate with higher interval,
3232 * so do not just take our interval to decide on timing out sender,
3233 * but take (if data->interval <= 5 * GST_SECOND):
3234 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3235 * where sender_interval is difference between last 2 received RTCP reports
3237 if (data->interval >= 5 * GST_SECOND || source->internal) {
3238 binterval = data->interval;
3240 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3241 GST_TIME_ARGS (source->stats.prev_rtcptime),
3242 GST_TIME_ARGS (source->stats.last_rtcptime));
3243 /* if not received enough yet, fallback to larger default */
3244 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3245 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3247 binterval = 5 * GST_SECOND;
3248 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3250 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3251 GST_TIME_ARGS (binterval));
3253 if (!source->internal && source->marked_bye) {
3254 /* if we received a BYE from the source, remove the source after some
3256 if (data->current_time > source->bye_time &&
3257 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3258 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3264 if (source->internal && source->sent_bye) {
3265 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3269 /* sources that were inactive for more than 5 times the deterministic reporting
3270 * interval get timed out. the min timeout is 5 seconds. */
3271 /* mind old time that might pre-date last time going to PLAYING */
3272 btime = MAX (source->last_activity, sess->start_time);
3273 if (data->current_time > btime) {
3274 interval = MAX (binterval * 5, 5 * GST_SECOND);
3275 if (data->current_time - btime > interval) {
3276 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3277 source->ssrc, GST_TIME_ARGS (btime));
3278 if (source->internal) {
3279 /* this is an internal source that is not using our suggested ssrc.
3280 * since there must be another source using this ssrc, we can remove
3281 * this one instead of making it a receiver forever */
3282 if (source->ssrc != sess->suggested_ssrc) {
3283 rtp_source_mark_bye (source, "timed out");
3284 /* do not schedule bye here, since we are inside the RTCP timeout
3285 * processing and scheduling bye will interfere with SR/RR sending */
3293 /* senders that did not send for a long time become a receiver, this also
3294 * holds for our own sources. */
3296 /* mind old time that might pre-date last time going to PLAYING */
3297 btime = MAX (source->last_rtp_activity, sess->start_time);
3298 if (data->current_time > btime) {
3299 interval = MAX (binterval * 2, 5 * GST_SECOND);
3300 if (data->current_time - btime > interval) {
3301 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3302 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3303 sendertimeout = TRUE;
3309 sess->total_sources--;
3311 sess->stats.sender_sources--;
3312 if (source->internal)
3313 sess->stats.internal_sender_sources--;
3316 sess->stats.active_sources--;
3318 if (source->internal)
3319 sess->stats.internal_sources--;
3322 on_bye_timeout (sess, source);
3324 on_timeout (sess, source);
3326 if (sendertimeout) {
3327 source->is_sender = FALSE;
3328 sess->stats.sender_sources--;
3329 if (source->internal)
3330 sess->stats.internal_sender_sources--;
3332 on_sender_timeout (sess, source);
3334 /* count how many source to report in this generation */
3335 if (((gint16) (source->generation - sess->generation)) <= 0)
3336 data->num_to_report++;
3338 source->closing = remove;
3342 session_sdes (RTPSession * sess, ReportData * data)
3344 GstRTCPPacket *packet = &data->packet;
3345 const GstStructure *sdes;
3347 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3349 /* add SDES packet */
3350 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3352 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3354 sdes = rtp_source_get_sdes_struct (data->source);
3356 /* add all fields in the structure, the order is not important. */
3357 n_fields = gst_structure_n_fields (sdes);
3358 for (i = 0; i < n_fields; ++i) {
3361 GstRTCPSDESType type;
3363 field = gst_structure_nth_field_name (sdes, i);
3366 value = gst_structure_get_string (sdes, field);
3369 type = gst_rtcp_sdes_name_to_type (field);
3371 /* Early packets are minimal and only include the CNAME */
3372 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3375 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3376 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3377 (const guint8 *) value);
3378 } else if (type == GST_RTCP_SDES_PRIV) {
3384 /* don't accept entries that are too big */
3385 prefix_len = strlen (field);
3386 if (prefix_len > 255)
3388 value_len = strlen (value);
3389 if (value_len > 255)
3391 data_len = 1 + prefix_len + value_len;
3395 data[0] = prefix_len;
3396 memcpy (&data[1], field, prefix_len);
3397 memcpy (&data[1 + prefix_len], value, value_len);
3399 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3403 data->has_sdes = TRUE;
3406 /* schedule a BYE packet */
3408 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3410 GstRTCPPacket *packet = &data->packet;
3411 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3414 session_sdes (sess, data);
3415 /* add a BYE packet */
3416 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3417 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3418 if (source->bye_reason)
3419 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3421 /* we have a BYE packet now */
3422 source->sent_bye = TRUE;
3426 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3428 GstClockTime new_send_time, elapsed;
3429 GstClockTime interval;
3430 RTPSessionStats *stats;
3432 if (sess->scheduled_bye)
3433 stats = &sess->bye_stats;
3435 stats = &sess->stats;
3437 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3438 data->is_early = TRUE;
3440 data->is_early = FALSE;
3442 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3443 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3444 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3445 GST_TIME_ARGS (current_time));
3449 /* no need to check yet */
3450 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3451 sess->next_rtcp_check_time > current_time) {
3452 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3453 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3454 GST_TIME_ARGS (current_time));
3459 /* get elapsed time since we last reported */
3460 elapsed = current_time - sess->last_rtcp_send_time;
3462 /* take interval and add jitter */
3463 interval = data->interval;
3464 if (interval != GST_CLOCK_TIME_NONE)
3465 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3467 /* perform forward reconsideration */
3468 if (interval != GST_CLOCK_TIME_NONE) {
3469 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3470 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3471 new_send_time = interval + sess->last_rtcp_send_time;
3473 new_send_time = sess->last_rtcp_send_time;
3476 if (!data->is_early) {
3477 /* check if reconsideration */
3478 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3479 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3480 GST_TIME_ARGS (new_send_time));
3481 /* store new check time */
3482 sess->next_rtcp_check_time = new_send_time;
3485 sess->next_rtcp_check_time = current_time + interval;
3486 } else if (interval != GST_CLOCK_TIME_NONE) {
3487 /* Apply the rules from RFC 4585 section 3.5.3 */
3488 if (stats->min_interval != 0 && !sess->first_rtcp) {
3489 GstClockTime T_rr_current_interval =
3490 g_random_double_range (0.5, 1.5) * stats->min_interval;
3492 /* This will caused the RTCP to be suppressed if no FB packets are added */
3493 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3494 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3495 " last: %" GST_TIME_FORMAT
3496 " + T_rr_current_interval: %" GST_TIME_FORMAT
3497 " > new_send_time: %" GST_TIME_FORMAT,
3498 GST_TIME_ARGS (stats->min_interval),
3499 GST_TIME_ARGS (sess->last_rtcp_send_time),
3500 GST_TIME_ARGS (T_rr_current_interval),
3501 GST_TIME_ARGS (new_send_time));
3502 data->may_suppress = TRUE;
3507 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3508 GST_TIME_ARGS (new_send_time));
3514 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3516 g_hash_table_insert (hash_table, key, g_object_ref (source));
3520 remove_closing_sources (const gchar * key, RTPSource * source,
3523 if (source->closing)
3526 if (source->send_fir)
3527 data->have_fir = TRUE;
3528 if (source->send_pli)
3529 data->have_pli = TRUE;
3530 if (source->send_nack)
3531 data->have_nack = TRUE;
3537 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3539 RTPSession *sess = data->sess;
3540 gboolean is_bye = FALSE;
3541 ReportOutput *output;
3543 /* only generate RTCP for active internal sources */
3544 if (!source->internal || source->sent_bye)
3547 /* ignore other sources when we do the timeout after a scheduled BYE */
3548 if (sess->scheduled_bye && !source->marked_bye)
3551 data->source = source;
3554 session_start_rtcp (sess, data);
3556 if (source->marked_bye) {
3558 make_source_bye (sess, source, data);
3560 } else if (!data->is_early) {
3561 /* loop over all known sources and add report blocks. If we are early, we
3562 * just make a minimal RTCP packet and skip this step */
3563 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3564 (GHFunc) session_report_blocks, data);
3566 if (!data->has_sdes)
3567 session_sdes (sess, data);
3570 session_fir (sess, data);
3573 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3574 (GHFunc) session_pli, data);
3576 if (data->have_nack)
3577 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3578 (GHFunc) session_nack, data);
3580 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3582 output = g_slice_new (ReportOutput);
3583 output->source = g_object_ref (source);
3584 output->is_bye = is_bye;
3585 output->buffer = data->rtcp;
3586 /* queue the RTCP packet to push later */
3587 g_queue_push_tail (&data->output, output);
3591 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3593 RTPSession *sess = data->sess;
3595 if (g_hash_table_size (source->reported_in_sr_of) >=
3596 sess->stats.internal_sources) {
3597 /* source is reported, move to next generation */
3598 source->generation = sess->generation + 1;
3599 g_hash_table_remove_all (source->reported_in_sr_of);
3601 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3602 source->generation);
3604 /* if we reported all sources in this generation, move to next */
3605 if (--data->num_to_report == 0) {
3607 GST_DEBUG ("all reported, generation now %u", sess->generation);
3613 * rtp_session_on_timeout:
3614 * @sess: an #RTPSession
3615 * @current_time: the current system time
3616 * @ntpnstime: the current NTP time in nanoseconds
3617 * @running_time: the current running_time of the pipeline
3619 * Perform maintenance actions after the timeout obtained with
3620 * rtp_session_next_timeout() expired.
3622 * This function will perform timeouts of receivers and senders, send a BYE
3623 * packet or generate RTCP packets with current session stats.
3625 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3626 * times, for each packet that should be processed.
3628 * Returns: a #GstFlowReturn.
3631 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3632 guint64 ntpnstime, GstClockTime running_time)
3634 GstFlowReturn result = GST_FLOW_OK;
3635 ReportData data = { GST_RTCP_BUFFER_INIT };
3636 GHashTable *table_copy;
3637 ReportOutput *output;
3639 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3641 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3642 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3643 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3646 data.current_time = current_time;
3647 data.ntpnstime = ntpnstime;
3648 data.running_time = running_time;
3649 data.num_to_report = 0;
3650 data.may_suppress = FALSE;
3651 data.nacked_seqnums = 0;
3652 g_queue_init (&data.output);
3654 RTP_SESSION_LOCK (sess);
3655 /* get a new interval, we need this for various cleanups etc */
3656 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3658 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3660 /* we need an internal source now */
3661 if (sess->stats.internal_sources == 0) {
3665 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
3667 g_object_unref (source);
3670 sess->conflicting_addresses =
3671 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
3673 /* Make a local copy of the hashtable. We need to do this because the
3674 * cleanup stage below releases the session lock. */
3675 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3676 (GDestroyNotify) g_object_unref);
3677 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3678 (GHFunc) clone_ssrcs_hashtable, table_copy);
3680 /* Clean up the session, mark the source for removing, this might release the
3682 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3683 g_hash_table_destroy (table_copy);
3685 /* Now remove the marked sources */
3686 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3687 (GHRFunc) remove_closing_sources, &data);
3689 /* update point-to-point status */
3690 session_update_ptp (sess);
3692 /* see if we need to generate SR or RR packets */
3693 if (!is_rtcp_time (sess, current_time, &data))
3696 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3697 sess->generation, data.num_to_report, data.is_early);
3699 /* generate RTCP for all internal sources */
3700 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3701 (GHFunc) generate_rtcp, &data);
3703 /* update the generation for all the sources that have been reported */
3704 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3705 (GHFunc) update_generation, &data);
3707 /* we keep track of the last report time in order to timeout inactive
3708 * receivers or senders */
3709 if (!data.is_early && !data.may_suppress)
3710 sess->last_rtcp_send_time = data.current_time;
3711 sess->first_rtcp = FALSE;
3712 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3713 sess->scheduled_bye = FALSE;
3715 /* RFC 4585 section 3.5.2 step 6 */
3716 if (!data.is_early) {
3717 sess->allow_early = TRUE;
3721 RTP_SESSION_UNLOCK (sess);
3723 /* push out the RTCP packets */
3724 while ((output = g_queue_pop_head (&data.output))) {
3725 gboolean do_not_suppress;
3726 GstBuffer *buffer = output->buffer;
3727 RTPSource *source = output->source;
3729 /* Give the user a change to add its own packet */
3730 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3731 buffer, data.is_early, &do_not_suppress);
3733 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3736 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3738 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3739 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3740 sess->stats.avg_rtcp_packet_size, packet_size);
3742 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3743 sess->send_rtcp_user_data);
3744 sess->stats.nacks_sent += data.nacked_seqnums;
3746 GST_DEBUG ("freeing packet callback: %p"
3747 " do_not_suppress: %d may_suppress: %d",
3748 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3749 sess->stats.nacks_dropped += data.nacked_seqnums;
3750 gst_buffer_unref (buffer);
3752 g_object_unref (source);
3753 g_slice_free (ReportOutput, output);
3759 * rtp_session_request_early_rtcp:
3760 * @sess: an #RTPSession
3761 * @current_time: the current system time
3762 * @max_delay: maximum delay
3764 * Request transmission of early RTCP
3766 * Returns: %TRUE if the related RTCP can be scheduled.
3769 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3770 GstClockTime max_delay)
3772 GstClockTime T_dither_max;
3775 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3777 RTP_SESSION_LOCK (sess);
3779 /* Check if already requested */
3780 /* RFC 4585 section 3.5.2 step 2 */
3781 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3782 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3787 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3788 GST_LOG_OBJECT (sess, "no next RTCP check time");
3793 /* RFC 4585 section 3.5.2 step 2b */
3794 /* If the total sources is <=2, then there is only us and one peer */
3795 /* When there is one auxiliary stream the session can still do point
3798 if (sess->is_doing_ptp) {
3801 /* Divide by 2 because l = 0.5 */
3802 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3806 /* RFC 4585 section 3.5.2 step 3 */
3807 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3808 GST_LOG_OBJECT (sess, "don't send because of dither");
3813 /* RFC 4585 section 3.5.2 step 4a */
3814 if (sess->allow_early == FALSE) {
3815 /* Ignore the request a scheduled packet will be in time anyway */
3816 if (current_time + max_delay > sess->next_rtcp_check_time) {
3817 GST_LOG_OBJECT (sess, "next scheduled time is soon %" GST_TIME_FORMAT " + %"
3818 GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
3819 GST_TIME_ARGS (current_time),
3820 GST_TIME_ARGS (max_delay), GST_TIME_ARGS (sess->next_rtcp_check_time));
3823 GST_LOG_OBJECT (sess, "can't allow early feedback");
3829 /* RFC 4585 section 3.5.2 step 4b */
3831 /* Schedule an early transmission later */
3832 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3835 /* If no dithering, schedule it for NOW */
3836 sess->next_early_rtcp_time = current_time;
3839 /* RFC 4585 section 3.5.2 step 6 */
3840 sess->allow_early = FALSE;
3841 /* TODO(mparis): "R MUST recalculate tn = tp + 2*T_rr,
3842 * and MUST set tp to the previous tn" */
3844 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3845 GST_TIME_ARGS (sess->next_early_rtcp_time));
3846 RTP_SESSION_UNLOCK (sess);
3848 /* notify app of need to send packet early
3849 * and therefore of timeout change */
3850 if (sess->callbacks.reconsider)
3851 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3857 RTP_SESSION_UNLOCK (sess);
3863 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3867 if (!sess->callbacks.send_rtcp)
3870 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3872 return rtp_session_request_early_rtcp (sess, now, max_delay);
3876 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3877 gboolean fir, gint count)
3881 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
3882 GST_DEBUG ("FIR/PLI not sent");
3886 RTP_SESSION_LOCK (sess);
3887 src = find_source (sess, ssrc);
3892 src->send_pli = FALSE;
3893 src->send_fir = TRUE;
3895 if (count == -1 || count != src->last_fir_count)
3896 src->current_send_fir_seqnum++;
3897 src->last_fir_count = count;
3898 } else if (!src->send_fir) {
3899 src->send_pli = TRUE;
3901 RTP_SESSION_UNLOCK (sess);
3908 RTP_SESSION_UNLOCK (sess);
3914 * rtp_session_request_nack:
3915 * @sess: a #RTPSession
3917 * @seqnum: the missing seqnum
3918 * @max_delay: max delay to request NACK
3920 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
3922 * Returns: %TRUE if the NACK feedback could be scheduled
3925 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
3926 GstClockTime max_delay)
3930 if (!rtp_session_send_rtcp (sess, max_delay)) {
3931 GST_DEBUG ("NACK not sent");
3935 RTP_SESSION_LOCK (sess);
3936 source = find_source (sess, ssrc);
3940 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
3941 rtp_source_register_nack (source, seqnum);
3942 RTP_SESSION_UNLOCK (sess);
3949 RTP_SESSION_UNLOCK (sess);