2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "gstrtpbin-marshal.h"
32 #include "rtpsession.h"
34 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
35 #define GST_CAT_DEFAULT rtp_session_debug
37 /* signals and args */
40 SIGNAL_GET_SOURCE_BY_SSRC,
42 SIGNAL_ON_SSRC_COLLISION,
43 SIGNAL_ON_SSRC_VALIDATED,
44 SIGNAL_ON_SSRC_ACTIVE,
47 SIGNAL_ON_BYE_TIMEOUT,
49 SIGNAL_ON_SENDER_TIMEOUT,
50 SIGNAL_ON_SENDING_RTCP,
51 SIGNAL_ON_FEEDBACK_RTCP,
56 #define DEFAULT_INTERNAL_SOURCE NULL
57 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
58 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
59 #define DEFAULT_RTCP_RR_BANDWIDTH -1
60 #define DEFAULT_RTCP_RS_BANDWIDTH -1
61 #define DEFAULT_RTCP_MTU 1400
62 #define DEFAULT_SDES NULL
63 #define DEFAULT_NUM_SOURCES 0
64 #define DEFAULT_NUM_ACTIVE_SOURCES 0
65 #define DEFAULT_SOURCES NULL
66 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
67 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
68 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
69 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
78 PROP_RTCP_RR_BANDWIDTH,
79 PROP_RTCP_RS_BANDWIDTH,
83 PROP_NUM_ACTIVE_SOURCES,
86 PROP_RTCP_MIN_INTERVAL,
87 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
88 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
93 /* update average packet size */
94 #define INIT_AVG(avg, val) \
96 #define UPDATE_AVG(avg, val) \
100 (avg) = ((val) + (15 * (avg))) >> 4;
103 /* The number RTCP intervals after which to timeout entries in the
106 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
108 /* GObject vmethods */
109 static void rtp_session_finalize (GObject * object);
110 static void rtp_session_set_property (GObject * object, guint prop_id,
111 const GValue * value, GParamSpec * pspec);
112 static void rtp_session_get_property (GObject * object, guint prop_id,
113 GValue * value, GParamSpec * pspec);
115 static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
116 GstBuffer * buffer, gboolean early);
117 static void rtp_session_send_rtcp (RTPSession * sess,
118 GstClockTimeDiff max_delay);
121 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
123 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
125 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
126 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
127 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
128 const gchar * reason, GstClockTime current_time);
129 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
130 gboolean deterministic, gboolean first);
133 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
134 const GValue * handler_return, gpointer data)
136 if (g_value_get_boolean (handler_return))
137 g_value_set_boolean (return_accu, TRUE);
143 rtp_session_class_init (RTPSessionClass * klass)
145 GObjectClass *gobject_class;
147 gobject_class = (GObjectClass *) klass;
149 gobject_class->finalize = rtp_session_finalize;
150 gobject_class->set_property = rtp_session_set_property;
151 gobject_class->get_property = rtp_session_get_property;
154 * RTPSession::get-source-by-ssrc:
155 * @session: the object which received the signal
156 * @ssrc: the SSRC of the RTPSource
158 * Request the #RTPSource object with SSRC @ssrc in @session.
160 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
161 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
162 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
163 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
164 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
167 * RTPSession::on-new-ssrc:
168 * @session: the object which received the signal
169 * @src: the new RTPSource
171 * Notify of a new SSRC that entered @session.
173 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
174 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
175 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
176 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
179 * RTPSession::on-ssrc-collision:
180 * @session: the object which received the signal
181 * @src: the #RTPSource that caused a collision
183 * Notify when we have an SSRC collision
185 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
186 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
188 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
191 * RTPSession::on-ssrc-validated:
192 * @session: the object which received the signal
193 * @src: the new validated RTPSource
195 * Notify of a new SSRC that became validated.
197 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
198 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
200 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
203 * RTPSession::on-ssrc-active:
204 * @session: the object which received the signal
205 * @src: the active RTPSource
207 * Notify of a SSRC that is active, i.e., sending RTCP.
209 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
210 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
212 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
215 * RTPSession::on-ssrc-sdes:
216 * @session: the object which received the signal
217 * @src: the RTPSource
219 * Notify that a new SDES was received for SSRC.
221 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
222 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
224 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
227 * RTPSession::on-bye-ssrc:
228 * @session: the object which received the signal
229 * @src: the RTPSource that went away
231 * Notify of an SSRC that became inactive because of a BYE packet.
233 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
234 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
236 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
239 * RTPSession::on-bye-timeout:
240 * @session: the object which received the signal
241 * @src: the RTPSource that timed out
243 * Notify of an SSRC that has timed out because of BYE
245 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
246 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
248 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
251 * RTPSession::on-timeout:
252 * @session: the object which received the signal
253 * @src: the RTPSource that timed out
255 * Notify of an SSRC that has timed out
257 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
258 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
259 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
260 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
263 * RTPSession::on-sender-timeout:
264 * @session: the object which received the signal
265 * @src: the RTPSource that timed out
267 * Notify of an SSRC that was a sender but timed out and became a receiver.
269 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
270 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
271 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
272 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
276 * RTPSession::on-sending-rtcp
277 * @session: the object which received the signal
278 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
279 * @early: %TRUE if the packet is early, %FALSE if it is regular
281 * This signal is emitted before sending an RTCP packet, it can be used
282 * to add extra RTCP Packets.
284 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
285 * if suppressing it is acceptable
287 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
288 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
289 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
290 accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__BOXED_BOOLEAN,
291 G_TYPE_BOOLEAN, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
295 * RTPSession::on-feedback-rtcp:
296 * @session: the object which received the signal
297 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
298 * %GST_RTCP_TYPE_RTPFB
299 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
300 * @sender_ssrc: The SSRC of the sender
301 * @media_ssrc: The SSRC of the media this refers to
302 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
305 * Notify that a RTCP feedback packet has been received
307 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
308 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
309 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
310 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_BOXED,
311 G_TYPE_NONE, 5, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
315 * RTPSession::send-rtcp:
316 * @session: the object which received the signal
317 * @max_delay: The maximum delay after which the feedback will not be useful
320 * Requests that the #RTPSession initiate a new RTCP packet as soon as
321 * possible within the requested delay.
324 rtp_session_signals[SIGNAL_SEND_RTCP] =
325 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
326 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
327 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
328 gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64);
330 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
331 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
332 "The internal SSRC used for the session",
333 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
335 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
336 g_param_spec_object ("internal-source", "Internal Source",
337 "The internal source element of the session",
338 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
340 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
341 g_param_spec_double ("bandwidth", "Bandwidth",
342 "The bandwidth of the session (0 for auto-discover)",
343 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
344 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
346 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
347 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
348 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
349 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
350 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
352 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
353 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
354 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
355 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
356 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
358 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
359 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
360 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
361 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
362 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
364 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
365 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
366 "The maximum size of the RTCP packets",
367 16, G_MAXINT16, DEFAULT_RTCP_MTU,
368 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_SDES,
371 g_param_spec_boxed ("sdes", "SDES",
372 "The SDES items of this session",
373 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
375 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
376 g_param_spec_uint ("num-sources", "Num Sources",
377 "The number of sources in the session", 0, G_MAXUINT,
378 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
380 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
381 g_param_spec_uint ("num-active-sources", "Num Active Sources",
382 "The number of active sources in the session", 0, G_MAXUINT,
383 DEFAULT_NUM_ACTIVE_SOURCES,
384 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
388 * Get a GValue Array of all sources in the session.
391 * <title>Getting the #RTPSources of a session
398 * g_object_get (sess, "sources", &arr, NULL);
400 * for (i = 0; i < arr->n_values; i++) {
403 * val = g_value_array_get_nth (arr, i);
404 * source = g_value_get_object (val);
406 * g_value_array_free (arr);
411 g_object_class_install_property (gobject_class, PROP_SOURCES,
412 g_param_spec_boxed ("sources", "Sources",
413 "An array of all known sources in the session",
414 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
416 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
417 g_param_spec_boolean ("favor-new", "Favor new sources",
418 "Resolve SSRC conflict in favor of new sources", FALSE,
419 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
421 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
422 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
423 "Minimum interval between Regular RTCP packet (in ns)",
424 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
425 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
427 g_object_class_install_property (gobject_class,
428 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
429 g_param_spec_uint64 ("rtcp-feedback-retention-window",
430 "RTCP Feedback retention window",
431 "Duration during which RTCP Feedback packets are retained (in ns)",
432 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
433 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
435 g_object_class_install_property (gobject_class,
436 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
437 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
438 "RTCP Immediate Feedback threshold",
439 "The maximum number of members of a RTP session for which immediate"
441 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
442 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
444 g_object_class_install_property (gobject_class, PROP_PROBATION,
445 g_param_spec_uint ("probation", "Number of probations",
446 "Consecutive packet sequence numbers to accept the source",
447 0, G_MAXUINT, DEFAULT_PROBATION,
448 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
450 klass->get_source_by_ssrc =
451 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
452 klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
453 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
455 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
459 rtp_session_init (RTPSession * sess)
464 g_mutex_init (&sess->lock);
465 sess->key = g_random_int ();
469 for (i = 0; i < 32; i++) {
471 g_hash_table_new_full (NULL, NULL, NULL,
472 (GDestroyNotify) g_object_unref);
475 rtp_stats_init_defaults (&sess->stats);
477 sess->recalc_bandwidth = TRUE;
478 sess->bandwidth = DEFAULT_BANDWIDTH;
479 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
480 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
481 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
483 /* create an active SSRC for this session manager */
484 sess->source = rtp_session_create_source (sess);
485 sess->source->validated = TRUE;
486 sess->source->internal = TRUE;
487 sess->stats.active_sources++;
488 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
489 sess->source->stats.prev_rtcptime = 0;
490 sess->source->stats.last_rtcptime = 1;
492 rtp_stats_set_min_interval (&sess->stats,
493 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
495 /* default UDP header length */
496 sess->header_len = 28;
497 sess->mtu = DEFAULT_RTCP_MTU;
499 sess->probation = DEFAULT_PROBATION;
501 /* some default SDES entries */
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 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
509 /* we do not want to leak the user's real name here */
510 str = g_strdup_printf ("Anon%u", g_random_int ());
511 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME, str);
515 rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
517 sess->first_rtcp = TRUE;
518 sess->allow_early = TRUE;
519 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
520 sess->rtcp_immediate_feedback_threshold =
521 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
523 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
525 GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
529 rtp_session_finalize (GObject * object)
534 sess = RTP_SESSION_CAST (object);
536 g_mutex_clear (&sess->lock);
537 for (i = 0; i < 32; i++)
538 g_hash_table_destroy (sess->ssrcs[i]);
540 g_free (sess->bye_reason);
542 g_object_unref (sess->source);
544 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
548 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
550 GValue value = { 0 };
552 g_value_init (&value, RTP_TYPE_SOURCE);
553 g_value_take_object (&value, source);
554 /* copies the value */
555 g_value_array_append (arr, &value);
559 rtp_session_create_sources (RTPSession * sess)
564 RTP_SESSION_LOCK (sess);
565 /* get number of elements in the table */
566 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
567 /* create the result value array */
568 res = g_value_array_new (size);
570 /* and copy all values into the array */
571 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
572 RTP_SESSION_UNLOCK (sess);
578 rtp_session_set_property (GObject * object, guint prop_id,
579 const GValue * value, GParamSpec * pspec)
583 sess = RTP_SESSION (object);
586 case PROP_INTERNAL_SSRC:
587 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
590 sess->bandwidth = g_value_get_double (value);
591 sess->recalc_bandwidth = TRUE;
593 case PROP_RTCP_FRACTION:
594 sess->rtcp_bandwidth = g_value_get_double (value);
595 sess->recalc_bandwidth = TRUE;
597 case PROP_RTCP_RR_BANDWIDTH:
598 sess->rtcp_rr_bandwidth = g_value_get_int (value);
599 sess->recalc_bandwidth = TRUE;
601 case PROP_RTCP_RS_BANDWIDTH:
602 sess->rtcp_rs_bandwidth = g_value_get_int (value);
603 sess->recalc_bandwidth = TRUE;
606 sess->mtu = g_value_get_uint (value);
609 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
612 sess->favor_new = g_value_get_boolean (value);
614 case PROP_RTCP_MIN_INTERVAL:
615 rtp_stats_set_min_interval (&sess->stats,
616 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
617 /* trigger reconsideration */
618 RTP_SESSION_LOCK (sess);
619 sess->next_rtcp_check_time = 0;
620 RTP_SESSION_UNLOCK (sess);
621 if (sess->callbacks.reconsider)
622 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
624 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
625 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
628 sess->probation = g_value_get_uint (value);
629 g_object_set_property (G_OBJECT (sess->source), "probation", value);
632 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
638 rtp_session_get_property (GObject * object, guint prop_id,
639 GValue * value, GParamSpec * pspec)
643 sess = RTP_SESSION (object);
646 case PROP_INTERNAL_SSRC:
647 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
649 case PROP_INTERNAL_SOURCE:
650 g_value_take_object (value, rtp_session_get_internal_source (sess));
653 g_value_set_double (value, sess->bandwidth);
655 case PROP_RTCP_FRACTION:
656 g_value_set_double (value, sess->rtcp_bandwidth);
658 case PROP_RTCP_RR_BANDWIDTH:
659 g_value_set_int (value, sess->rtcp_rr_bandwidth);
661 case PROP_RTCP_RS_BANDWIDTH:
662 g_value_set_int (value, sess->rtcp_rs_bandwidth);
665 g_value_set_uint (value, sess->mtu);
668 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
670 case PROP_NUM_SOURCES:
671 g_value_set_uint (value, rtp_session_get_num_sources (sess));
673 case PROP_NUM_ACTIVE_SOURCES:
674 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
677 g_value_take_boxed (value, rtp_session_create_sources (sess));
680 g_value_set_boolean (value, sess->favor_new);
682 case PROP_RTCP_MIN_INTERVAL:
683 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
685 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
686 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
689 g_value_set_uint (value, sess->probation);
690 g_object_get_property (G_OBJECT (sess->source), "probation", value);
693 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
699 on_new_ssrc (RTPSession * sess, RTPSource * source)
701 g_object_ref (source);
702 RTP_SESSION_UNLOCK (sess);
703 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
704 RTP_SESSION_LOCK (sess);
705 g_object_unref (source);
709 on_ssrc_collision (RTPSession * sess, RTPSource * source)
711 g_object_ref (source);
712 RTP_SESSION_UNLOCK (sess);
713 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
715 RTP_SESSION_LOCK (sess);
716 g_object_unref (source);
720 on_ssrc_validated (RTPSession * sess, RTPSource * source)
722 g_object_ref (source);
723 RTP_SESSION_UNLOCK (sess);
724 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
726 RTP_SESSION_LOCK (sess);
727 g_object_unref (source);
731 on_ssrc_active (RTPSession * sess, RTPSource * source)
733 g_object_ref (source);
734 RTP_SESSION_UNLOCK (sess);
735 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
736 RTP_SESSION_LOCK (sess);
737 g_object_unref (source);
741 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
743 g_object_ref (source);
744 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
745 RTP_SESSION_UNLOCK (sess);
746 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
747 RTP_SESSION_LOCK (sess);
748 g_object_unref (source);
752 on_bye_ssrc (RTPSession * sess, RTPSource * source)
754 g_object_ref (source);
755 RTP_SESSION_UNLOCK (sess);
756 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
757 RTP_SESSION_LOCK (sess);
758 g_object_unref (source);
762 on_bye_timeout (RTPSession * sess, RTPSource * source)
764 g_object_ref (source);
765 RTP_SESSION_UNLOCK (sess);
766 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
767 RTP_SESSION_LOCK (sess);
768 g_object_unref (source);
772 on_timeout (RTPSession * sess, RTPSource * source)
774 g_object_ref (source);
775 RTP_SESSION_UNLOCK (sess);
776 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
777 RTP_SESSION_LOCK (sess);
778 g_object_unref (source);
782 on_sender_timeout (RTPSession * sess, RTPSource * source)
784 g_object_ref (source);
785 RTP_SESSION_UNLOCK (sess);
786 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
788 RTP_SESSION_LOCK (sess);
789 g_object_unref (source);
795 * Create a new session object.
797 * Returns: a new #RTPSession. g_object_unref() after usage.
800 rtp_session_new (void)
804 sess = g_object_new (RTP_TYPE_SESSION, NULL);
810 * rtp_session_set_callbacks:
811 * @sess: an #RTPSession
812 * @callbacks: callbacks to configure
813 * @user_data: user data passed in the callbacks
815 * Configure a set of callbacks to be notified of actions.
818 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
821 g_return_if_fail (RTP_IS_SESSION (sess));
823 if (callbacks->process_rtp) {
824 sess->callbacks.process_rtp = callbacks->process_rtp;
825 sess->process_rtp_user_data = user_data;
827 if (callbacks->send_rtp) {
828 sess->callbacks.send_rtp = callbacks->send_rtp;
829 sess->send_rtp_user_data = user_data;
831 if (callbacks->send_rtcp) {
832 sess->callbacks.send_rtcp = callbacks->send_rtcp;
833 sess->send_rtcp_user_data = user_data;
835 if (callbacks->sync_rtcp) {
836 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
837 sess->sync_rtcp_user_data = user_data;
839 if (callbacks->clock_rate) {
840 sess->callbacks.clock_rate = callbacks->clock_rate;
841 sess->clock_rate_user_data = user_data;
843 if (callbacks->reconsider) {
844 sess->callbacks.reconsider = callbacks->reconsider;
845 sess->reconsider_user_data = user_data;
847 if (callbacks->request_key_unit) {
848 sess->callbacks.request_key_unit = callbacks->request_key_unit;
849 sess->request_key_unit_user_data = user_data;
851 if (callbacks->request_time) {
852 sess->callbacks.request_time = callbacks->request_time;
853 sess->request_time_user_data = user_data;
858 * rtp_session_set_process_rtp_callback:
859 * @sess: an #RTPSession
860 * @callback: callback to set
861 * @user_data: user data passed in the callback
863 * Configure only the process_rtp callback to be notified of the process_rtp action.
866 rtp_session_set_process_rtp_callback (RTPSession * sess,
867 RTPSessionProcessRTP callback, gpointer user_data)
869 g_return_if_fail (RTP_IS_SESSION (sess));
871 sess->callbacks.process_rtp = callback;
872 sess->process_rtp_user_data = user_data;
876 * rtp_session_set_send_rtp_callback:
877 * @sess: an #RTPSession
878 * @callback: callback to set
879 * @user_data: user data passed in the callback
881 * Configure only the send_rtp callback to be notified of the send_rtp action.
884 rtp_session_set_send_rtp_callback (RTPSession * sess,
885 RTPSessionSendRTP callback, gpointer user_data)
887 g_return_if_fail (RTP_IS_SESSION (sess));
889 sess->callbacks.send_rtp = callback;
890 sess->send_rtp_user_data = user_data;
894 * rtp_session_set_send_rtcp_callback:
895 * @sess: an #RTPSession
896 * @callback: callback to set
897 * @user_data: user data passed in the callback
899 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
902 rtp_session_set_send_rtcp_callback (RTPSession * sess,
903 RTPSessionSendRTCP callback, gpointer user_data)
905 g_return_if_fail (RTP_IS_SESSION (sess));
907 sess->callbacks.send_rtcp = callback;
908 sess->send_rtcp_user_data = user_data;
912 * rtp_session_set_sync_rtcp_callback:
913 * @sess: an #RTPSession
914 * @callback: callback to set
915 * @user_data: user data passed in the callback
917 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
920 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
921 RTPSessionSyncRTCP callback, gpointer user_data)
923 g_return_if_fail (RTP_IS_SESSION (sess));
925 sess->callbacks.sync_rtcp = callback;
926 sess->sync_rtcp_user_data = user_data;
930 * rtp_session_set_clock_rate_callback:
931 * @sess: an #RTPSession
932 * @callback: callback to set
933 * @user_data: user data passed in the callback
935 * Configure only the clock_rate callback to be notified of the clock_rate action.
938 rtp_session_set_clock_rate_callback (RTPSession * sess,
939 RTPSessionClockRate callback, gpointer user_data)
941 g_return_if_fail (RTP_IS_SESSION (sess));
943 sess->callbacks.clock_rate = callback;
944 sess->clock_rate_user_data = user_data;
948 * rtp_session_set_reconsider_callback:
949 * @sess: an #RTPSession
950 * @callback: callback to set
951 * @user_data: user data passed in the callback
953 * Configure only the reconsider callback to be notified of the reconsider action.
956 rtp_session_set_reconsider_callback (RTPSession * sess,
957 RTPSessionReconsider callback, gpointer user_data)
959 g_return_if_fail (RTP_IS_SESSION (sess));
961 sess->callbacks.reconsider = callback;
962 sess->reconsider_user_data = user_data;
966 * rtp_session_set_request_time_callback:
967 * @sess: an #RTPSession
968 * @callback: callback to set
969 * @user_data: user data passed in the callback
971 * Configure only the request_time callback
974 rtp_session_set_request_time_callback (RTPSession * sess,
975 RTPSessionRequestTime callback, gpointer user_data)
977 g_return_if_fail (RTP_IS_SESSION (sess));
979 sess->callbacks.request_time = callback;
980 sess->request_time_user_data = user_data;
984 * rtp_session_set_bandwidth:
985 * @sess: an #RTPSession
986 * @bandwidth: the bandwidth allocated
988 * Set the session bandwidth in bytes per second.
991 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
993 g_return_if_fail (RTP_IS_SESSION (sess));
995 RTP_SESSION_LOCK (sess);
996 sess->stats.bandwidth = bandwidth;
997 RTP_SESSION_UNLOCK (sess);
1001 * rtp_session_get_bandwidth:
1002 * @sess: an #RTPSession
1004 * Get the session bandwidth.
1006 * Returns: the session bandwidth.
1009 rtp_session_get_bandwidth (RTPSession * sess)
1013 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1015 RTP_SESSION_LOCK (sess);
1016 result = sess->stats.bandwidth;
1017 RTP_SESSION_UNLOCK (sess);
1023 * rtp_session_set_rtcp_fraction:
1024 * @sess: an #RTPSession
1025 * @bandwidth: the RTCP bandwidth
1027 * Set the bandwidth in bytes per second that should be used for RTCP
1031 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1033 g_return_if_fail (RTP_IS_SESSION (sess));
1035 RTP_SESSION_LOCK (sess);
1036 sess->stats.rtcp_bandwidth = bandwidth;
1037 RTP_SESSION_UNLOCK (sess);
1041 * rtp_session_get_rtcp_fraction:
1042 * @sess: an #RTPSession
1044 * Get the session bandwidth used for RTCP.
1046 * Returns: The bandwidth used for RTCP messages.
1049 rtp_session_get_rtcp_fraction (RTPSession * sess)
1053 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1055 RTP_SESSION_LOCK (sess);
1056 result = sess->stats.rtcp_bandwidth;
1057 RTP_SESSION_UNLOCK (sess);
1063 * rtp_session_set_sdes_string:
1064 * @sess: an #RTPSession
1065 * @type: the type of the SDES item
1066 * @item: a null-terminated string to set.
1068 * Store an SDES item of @type in @sess.
1070 * Returns: %FALSE if the data was unchanged @type is invalid.
1073 rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
1078 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1080 RTP_SESSION_LOCK (sess);
1081 result = rtp_source_set_sdes_string (sess->source, type, item);
1082 RTP_SESSION_UNLOCK (sess);
1088 * rtp_session_get_sdes_string:
1089 * @sess: an #RTPSession
1090 * @type: the type of the SDES item
1092 * Get the SDES item of @type from @sess.
1094 * Returns: a null-terminated copy of the SDES item or NULL when @type was not
1095 * valid. g_free() after usage.
1098 rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
1102 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1104 RTP_SESSION_LOCK (sess);
1105 result = rtp_source_get_sdes_string (sess->source, type);
1106 RTP_SESSION_UNLOCK (sess);
1112 * rtp_session_get_sdes_struct:
1113 * @sess: an #RTSPSession
1115 * Get the SDES data as a #GstStructure
1117 * Returns: a GstStructure with SDES items for @sess. This function returns a
1118 * copy of the SDES structure, use gst_structure_free() after usage.
1121 rtp_session_get_sdes_struct (RTPSession * sess)
1123 const GstStructure *sdes;
1124 GstStructure *result = NULL;
1126 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1128 RTP_SESSION_LOCK (sess);
1129 sdes = rtp_source_get_sdes_struct (sess->source);
1131 result = gst_structure_copy (sdes);
1132 RTP_SESSION_UNLOCK (sess);
1138 * rtp_session_set_sdes_struct:
1139 * @sess: an #RTSPSession
1140 * @sdes: a #GstStructure
1142 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1145 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1147 g_return_if_fail (sdes);
1148 g_return_if_fail (RTP_IS_SESSION (sess));
1150 RTP_SESSION_LOCK (sess);
1151 rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
1152 RTP_SESSION_UNLOCK (sess);
1155 static GstFlowReturn
1156 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1158 GstFlowReturn result = GST_FLOW_OK;
1160 if (source == session->source) {
1161 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1163 RTP_SESSION_UNLOCK (session);
1165 if (session->callbacks.send_rtp)
1167 session->callbacks.send_rtp (session, source, data,
1168 session->send_rtp_user_data);
1170 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1173 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1174 RTP_SESSION_UNLOCK (session);
1176 if (session->callbacks.process_rtp)
1178 session->callbacks.process_rtp (session, source,
1179 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1181 gst_buffer_unref (GST_BUFFER_CAST (data));
1183 RTP_SESSION_LOCK (session);
1189 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1193 RTP_SESSION_UNLOCK (session);
1195 if (session->callbacks.clock_rate)
1197 session->callbacks.clock_rate (session, pt,
1198 session->clock_rate_user_data);
1202 RTP_SESSION_LOCK (session);
1204 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1209 static RTPSourceCallbacks callbacks = {
1210 (RTPSourcePushRTP) source_push_rtp,
1211 (RTPSourceClockRate) source_clock_rate,
1215 check_collision (RTPSession * sess, RTPSource * source,
1216 RTPArrivalStats * arrival, gboolean rtp)
1218 /* If we have no arrival address, we can't do collision checking */
1219 if (!arrival->address)
1222 if (sess->source != source) {
1223 GSocketAddress *from;
1225 /* This is not our local source, but lets check if two remote
1228 from = source->rtp_from;
1230 from = source->rtcp_from;
1234 if (__g_socket_address_equal (from, arrival->address)) {
1235 /* Address is the same */
1238 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1239 rtp_source_get_ssrc (source));
1240 if (sess->favor_new) {
1241 if (rtp_source_find_conflicting_address (source,
1242 arrival->address, arrival->current_time)) {
1245 buf1 = __g_socket_address_to_string (arrival->address);
1246 GST_LOG ("Known conflict on %x for %s, dropping packet",
1247 rtp_source_get_ssrc (source), buf1);
1254 /* Current address is not a known conflict, lets assume this is
1255 * a new source. Save old address in possible conflict list
1257 rtp_source_add_conflicting_address (source, from,
1258 arrival->current_time);
1260 buf1 = __g_socket_address_to_string (from);
1261 buf2 = __g_socket_address_to_string (arrival->address);
1263 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1264 " saving old as known conflict",
1265 rtp_source_get_ssrc (source), buf1, buf2);
1268 rtp_source_set_rtp_from (source, arrival->address);
1270 rtp_source_set_rtcp_from (source, arrival->address);
1278 /* Don't need to save old addresses, we ignore new sources */
1283 /* We don't already have a from address for RTP, just set it */
1285 rtp_source_set_rtp_from (source, arrival->address);
1287 rtp_source_set_rtcp_from (source, arrival->address);
1291 /* FIXME: Log 3rd party collision somehow
1292 * Maybe should be done in upper layer, only the SDES can tell us
1293 * if its a collision or a loop
1296 /* This is sending with our ssrc, is it an address we already know */
1298 if (rtp_source_find_conflicting_address (source, arrival->address,
1299 arrival->current_time)) {
1300 /* Its a known conflict, its probably a loop, not a collision
1301 * lets just drop the incoming packet
1303 GST_DEBUG ("Our packets are being looped back to us, dropping");
1305 /* Its a new collision, lets change our SSRC */
1307 rtp_source_add_conflicting_address (source, arrival->address,
1308 arrival->current_time);
1310 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1311 on_ssrc_collision (sess, source);
1313 sess->change_ssrc = TRUE;
1315 rtp_session_schedule_bye_locked (sess, "SSRC Collision",
1316 arrival->current_time);
1324 /* must be called with the session lock, the returned source needs to be
1325 * unreffed after usage. */
1327 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1328 RTPArrivalStats * arrival, gboolean rtp)
1333 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1334 if (source == NULL) {
1335 /* make new Source in probation and insert */
1336 source = rtp_source_new (ssrc);
1338 /* for RTP packets we need to set the source in probation. Receiving RTCP
1339 * packets of an SSRC, on the other hand, is a strong indication that we
1340 * are dealing with a valid source. */
1342 g_object_set (source, "probation", sess->probation, NULL);
1344 g_object_set (source, "probation", 0, NULL);
1346 /* store from address, if any */
1347 if (arrival->address) {
1349 rtp_source_set_rtp_from (source, arrival->address);
1351 rtp_source_set_rtcp_from (source, arrival->address);
1354 /* configure a callback on the source */
1355 rtp_source_set_callbacks (source, &callbacks, sess);
1357 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1360 /* we have one more source now */
1361 sess->total_sources++;
1365 /* check for collision, this updates the address when not previously set */
1366 if (check_collision (sess, source, arrival, rtp)) {
1369 /* Receiving RTCP packets of an SSRC is a strong indication that we
1370 * are dealing with a valid source. */
1372 g_object_set (source, "probation", 0, NULL);
1374 /* update last activity */
1375 source->last_activity = arrival->current_time;
1377 source->last_rtp_activity = arrival->current_time;
1378 g_object_ref (source);
1384 * rtp_session_get_internal_source:
1385 * @sess: a #RTPSession
1387 * Get the internal #RTPSource of @sess.
1389 * Returns: The internal #RTPSource. g_object_unref() after usage.
1392 rtp_session_get_internal_source (RTPSession * sess)
1396 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1398 result = g_object_ref (sess->source);
1404 * rtp_session_set_internal_ssrc:
1405 * @sess: a #RTPSession
1408 * Set the SSRC of @sess to @ssrc.
1411 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1413 RTP_SESSION_LOCK (sess);
1414 if (ssrc != sess->source->ssrc) {
1415 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1416 GINT_TO_POINTER (sess->source->ssrc));
1418 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1419 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1420 * packets will timeout on the old SSRC, we could potentially schedule a
1421 * BYE RTCP for the old SSRC... */
1422 sess->source->ssrc = ssrc;
1423 rtp_source_reset (sess->source);
1425 /* rehash with the new SSRC */
1426 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1427 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1429 RTP_SESSION_UNLOCK (sess);
1431 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1435 * rtp_session_get_internal_ssrc:
1436 * @sess: a #RTPSession
1438 * Get the internal SSRC of @sess.
1440 * Returns: The SSRC of the session.
1443 rtp_session_get_internal_ssrc (RTPSession * sess)
1447 RTP_SESSION_LOCK (sess);
1448 ssrc = sess->source->ssrc;
1449 RTP_SESSION_UNLOCK (sess);
1455 * rtp_session_add_source:
1456 * @sess: a #RTPSession
1457 * @src: #RTPSource to add
1459 * Add @src to @session.
1461 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1462 * existed in the session.
1465 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1467 gboolean result = FALSE;
1470 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1471 g_return_val_if_fail (src != NULL, FALSE);
1473 RTP_SESSION_LOCK (sess);
1475 g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1476 GINT_TO_POINTER (src->ssrc));
1478 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1479 GINT_TO_POINTER (src->ssrc), src);
1480 /* we have one more source now */
1481 sess->total_sources++;
1484 RTP_SESSION_UNLOCK (sess);
1490 * rtp_session_get_num_sources:
1491 * @sess: an #RTPSession
1493 * Get the number of sources in @sess.
1495 * Returns: The number of sources in @sess.
1498 rtp_session_get_num_sources (RTPSession * sess)
1502 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1504 RTP_SESSION_LOCK (sess);
1505 result = sess->total_sources;
1506 RTP_SESSION_UNLOCK (sess);
1512 * rtp_session_get_num_active_sources:
1513 * @sess: an #RTPSession
1515 * Get the number of active sources in @sess. A source is considered active when
1516 * it has been validated and has not yet received a BYE RTCP message.
1518 * Returns: The number of active sources in @sess.
1521 rtp_session_get_num_active_sources (RTPSession * sess)
1525 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1527 RTP_SESSION_LOCK (sess);
1528 result = sess->stats.active_sources;
1529 RTP_SESSION_UNLOCK (sess);
1535 * rtp_session_get_source_by_ssrc:
1536 * @sess: an #RTPSession
1539 * Find the source with @ssrc in @sess.
1541 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1542 * g_object_unref() after usage.
1545 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1549 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1551 RTP_SESSION_LOCK (sess);
1553 g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1555 g_object_ref (result);
1556 RTP_SESSION_UNLOCK (sess);
1561 /* should be called with the SESSION lock */
1563 rtp_session_create_new_ssrc (RTPSession * sess)
1568 ssrc = g_random_int ();
1570 /* see if it exists in the session, we're done if it doesn't */
1571 if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1572 GINT_TO_POINTER (ssrc)) == NULL)
1580 * rtp_session_create_source:
1581 * @sess: an #RTPSession
1583 * Create an #RTPSource for use in @sess. This function will create a source
1584 * with an ssrc that is currently not used by any participants in the session.
1586 * Returns: an #RTPSource.
1589 rtp_session_create_source (RTPSession * sess)
1594 RTP_SESSION_LOCK (sess);
1595 ssrc = rtp_session_create_new_ssrc (sess);
1596 source = rtp_source_new (ssrc);
1597 rtp_source_set_callbacks (source, &callbacks, sess);
1598 /* we need an additional ref for the source in the hashtable */
1599 g_object_ref (source);
1600 g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1602 /* we have one more source now */
1603 sess->total_sources++;
1604 RTP_SESSION_UNLOCK (sess);
1609 /* update the RTPArrivalStats structure with the current time and other bits
1610 * about the current buffer we are handling.
1611 * This function is typically called when a validated packet is received.
1612 * This function should be called with the SESSION_LOCK
1615 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1616 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1617 GstClockTime running_time, guint64 ntpnstime)
1619 GstNetAddressMeta *meta;
1620 GstRTPBuffer rtpb = { NULL };
1622 /* get time of arrival */
1623 arrival->current_time = current_time;
1624 arrival->running_time = running_time;
1625 arrival->ntpnstime = ntpnstime;
1627 /* get packet size including header overhead */
1628 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1631 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1632 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1633 gst_rtp_buffer_unmap (&rtpb);
1635 arrival->payload_len = 0;
1638 /* for netbuffer we can store the IP address to check for collisions */
1639 meta = gst_buffer_get_net_address_meta (buffer);
1640 if (arrival->address)
1641 g_object_unref (arrival->address);
1643 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1645 arrival->address = NULL;
1650 clean_arrival_stats (RTPArrivalStats * arrival)
1652 if (arrival->address)
1653 g_object_unref (arrival->address);
1657 * rtp_session_process_rtp:
1658 * @sess: and #RTPSession
1659 * @buffer: an RTP buffer
1660 * @current_time: the current system time
1661 * @running_time: the running_time of @buffer
1663 * Process an RTP buffer in the session manager. This function takes ownership
1666 * Returns: a #GstFlowReturn.
1669 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1670 GstClockTime current_time, GstClockTime running_time)
1672 GstFlowReturn result;
1676 gboolean prevsender, prevactive;
1677 RTPArrivalStats arrival = { NULL, };
1681 GstRTPBuffer rtp = { NULL };
1683 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1684 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1686 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1687 goto invalid_packet;
1689 RTP_SESSION_LOCK (sess);
1690 /* ignore more RTP packets when we left the session */
1691 if (sess->source->received_bye)
1694 /* update arrival stats */
1695 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1698 /* get SSRC and look up in session database */
1699 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1700 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1704 /* copy available csrc for later */
1705 count = gst_rtp_buffer_get_csrc_count (&rtp);
1706 /* make sure to not overflow our array. An RTP buffer can maximally contain
1708 count = MIN (count, 16);
1710 for (i = 0; i < count; i++)
1711 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1713 gst_rtp_buffer_unmap (&rtp);
1715 prevsender = RTP_SOURCE_IS_SENDER (source);
1716 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1717 oldrate = source->bitrate;
1719 /* let source process the packet */
1720 result = rtp_source_process_rtp (source, buffer, &arrival);
1722 /* source became active */
1723 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1724 sess->stats.active_sources++;
1725 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1726 sess->stats.active_sources);
1727 on_ssrc_validated (sess, source);
1729 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1730 sess->stats.sender_sources++;
1731 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1732 sess->stats.sender_sources);
1734 if (oldrate != source->bitrate)
1735 sess->recalc_bandwidth = TRUE;
1738 on_new_ssrc (sess, source);
1740 if (source->validated) {
1743 /* for validated sources, we add the CSRCs as well */
1744 for (i = 0; i < count; i++) {
1746 RTPSource *csrc_src;
1751 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1756 GST_DEBUG ("created new CSRC: %08x", csrc);
1757 rtp_source_set_as_csrc (csrc_src);
1758 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1759 sess->stats.active_sources++;
1760 on_new_ssrc (sess, csrc_src);
1762 g_object_unref (csrc_src);
1765 g_object_unref (source);
1767 RTP_SESSION_UNLOCK (sess);
1769 clean_arrival_stats (&arrival);
1776 gst_buffer_unref (buffer);
1777 GST_DEBUG ("invalid RTP packet received");
1782 RTP_SESSION_UNLOCK (sess);
1783 gst_rtp_buffer_unmap (&rtp);
1784 gst_buffer_unref (buffer);
1785 GST_DEBUG ("ignoring RTP packet because we are leaving");
1790 RTP_SESSION_UNLOCK (sess);
1791 gst_rtp_buffer_unmap (&rtp);
1792 gst_buffer_unref (buffer);
1793 clean_arrival_stats (&arrival);
1794 GST_DEBUG ("ignoring packet because its collisioning");
1800 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1801 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1805 count = gst_rtcp_packet_get_rb_count (packet);
1806 for (i = 0; i < count; i++) {
1807 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1808 guint8 fractionlost;
1811 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1812 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1814 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1816 if (ssrc == sess->source->ssrc) {
1817 /* only deal with report blocks for our session, we update the stats of
1818 * the sender of the RTCP message. We could also compare our stats against
1819 * the other sender to see if we are better or worse. */
1820 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1821 packetslost, exthighestseq, jitter, lsr, dlsr);
1824 on_ssrc_active (sess, source);
1827 /* A Sender report contains statistics about how the sender is doing. This
1828 * includes timing informataion such as the relation between RTP and NTP
1829 * timestamps and the number of packets/bytes it sent to us.
1831 * In this report is also included a set of report blocks related to how this
1832 * sender is receiving data (in case we (or somebody else) is also sending stuff
1833 * to it). This info includes the packet loss, jitter and seqnum. It also
1834 * contains information to calculate the round trip time (LSR/DLSR).
1837 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1838 RTPArrivalStats * arrival, gboolean * do_sync)
1840 guint32 senderssrc, rtptime, packet_count, octet_count;
1843 gboolean created, prevsender;
1845 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1846 &packet_count, &octet_count);
1848 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1849 senderssrc, GST_TIME_ARGS (arrival->current_time));
1851 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1855 /* don't try to do lip-sync for sources that sent a BYE */
1856 if (rtp_source_received_bye (source))
1861 prevsender = RTP_SOURCE_IS_SENDER (source);
1863 /* first update the source */
1864 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1865 packet_count, octet_count);
1867 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1868 sess->stats.sender_sources++;
1869 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1870 sess->stats.sender_sources);
1874 on_new_ssrc (sess, source);
1876 rtp_session_process_rb (sess, source, packet, arrival);
1877 g_object_unref (source);
1880 /* A receiver report contains statistics about how a receiver is doing. It
1881 * includes stuff like packet loss, jitter and the seqnum it received last. It
1882 * also contains info to calculate the round trip time.
1884 * We are only interested in how the sender of this report is doing wrt to us.
1887 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1888 RTPArrivalStats * arrival)
1894 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1896 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1898 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1903 on_new_ssrc (sess, source);
1905 rtp_session_process_rb (sess, source, packet, arrival);
1906 g_object_unref (source);
1909 /* Get SDES items and store them in the SSRC */
1911 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1912 RTPArrivalStats * arrival)
1915 gboolean more_items, more_entries;
1917 items = gst_rtcp_packet_sdes_get_item_count (packet);
1918 GST_DEBUG ("got SDES packet with %d items", items);
1920 more_items = gst_rtcp_packet_sdes_first_item (packet);
1922 while (more_items) {
1924 gboolean changed, created, validated;
1928 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1930 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1934 /* find src, no probation when dealing with RTCP */
1935 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1939 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1941 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1943 while (more_entries) {
1944 GstRTCPSDESType type;
1950 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1952 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1955 if (type == GST_RTCP_SDES_PRIV) {
1956 name = g_strndup ((const gchar *) &data[1], data[0]);
1958 data += data[0] + 1;
1960 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1963 value = g_strndup ((const gchar *) data, len);
1965 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1970 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1974 /* takes ownership of sdes */
1975 changed = rtp_source_set_sdes_struct (source, sdes);
1977 validated = !RTP_SOURCE_IS_ACTIVE (source);
1978 source->validated = TRUE;
1981 on_new_ssrc (sess, source);
1983 /* source became active */
1985 sess->stats.active_sources++;
1986 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1987 sess->stats.active_sources);
1988 on_ssrc_validated (sess, source);
1992 on_ssrc_sdes (sess, source);
1994 g_object_unref (source);
1996 more_items = gst_rtcp_packet_sdes_next_item (packet);
2001 /* BYE is sent when a client leaves the session
2004 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2005 RTPArrivalStats * arrival)
2009 gboolean reconsider = FALSE;
2011 reason = gst_rtcp_packet_bye_get_reason (packet);
2012 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2014 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2015 for (i = 0; i < count; i++) {
2018 gboolean created, prevactive, prevsender;
2019 guint pmembers, members;
2021 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2022 GST_DEBUG ("SSRC: %08x", ssrc);
2024 if (ssrc == sess->source->ssrc)
2027 /* find src and mark bye, no probation when dealing with RTCP */
2028 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
2032 /* store time for when we need to time out this source */
2033 source->bye_time = arrival->current_time;
2035 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2036 prevsender = RTP_SOURCE_IS_SENDER (source);
2038 /* let the source handle the rest */
2039 rtp_source_process_bye (source, reason);
2041 pmembers = sess->stats.active_sources;
2043 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2044 sess->stats.active_sources--;
2045 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2046 sess->stats.active_sources);
2048 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2049 sess->stats.sender_sources--;
2050 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2051 sess->stats.sender_sources);
2053 members = sess->stats.active_sources;
2055 if (!sess->source->received_bye && members < pmembers) {
2056 /* some members went away since the previous timeout estimate.
2057 * Perform reverse reconsideration but only when we are not scheduling a
2059 if (arrival->current_time < sess->next_rtcp_check_time) {
2060 GstClockTime time_remaining;
2062 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2063 sess->next_rtcp_check_time =
2064 gst_util_uint64_scale (time_remaining, members, pmembers);
2066 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2067 GST_TIME_ARGS (sess->next_rtcp_check_time));
2069 sess->next_rtcp_check_time += arrival->current_time;
2071 /* mark pending reconsider. We only want to signal the reconsideration
2072 * once after we handled all the source in the bye packet */
2078 on_new_ssrc (sess, source);
2080 on_bye_ssrc (sess, source);
2082 g_object_unref (source);
2085 RTP_SESSION_UNLOCK (sess);
2086 /* notify app of reconsideration */
2087 if (sess->callbacks.reconsider)
2088 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2089 RTP_SESSION_LOCK (sess);
2095 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2096 RTPArrivalStats * arrival)
2098 GST_DEBUG ("received APP");
2102 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2103 gboolean fir, GstClockTime current_time)
2105 guint32 round_trip = 0;
2107 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2109 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2110 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2113 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2114 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2115 GST_DEBUG ("Ignoring %s request because one was send without one "
2116 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2117 fir ? "FIR" : "PLI",
2118 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2119 GST_TIME_ARGS (round_trip_in_ns));;
2124 sess->last_keyframe_request = current_time;
2126 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2127 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2128 sess->callbacks.request_key_unit);
2130 RTP_SESSION_UNLOCK (sess);
2131 sess->callbacks.request_key_unit (sess, fir,
2132 sess->request_key_unit_user_data);
2133 RTP_SESSION_LOCK (sess);
2139 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2140 guint32 media_ssrc, GstClockTime current_time)
2144 if (!sess->callbacks.request_key_unit)
2147 src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
2148 GINT_TO_POINTER (sender_ssrc));
2152 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2156 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2157 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2162 gboolean our_request = FALSE;
2164 if (!sess->callbacks.request_key_unit)
2170 src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
2171 GINT_TO_POINTER (sender_ssrc));
2173 /* Hack because Google fails to set the sender_ssrc correctly */
2174 if (!src && sender_ssrc == 1) {
2175 GHashTableIter iter;
2177 if (sess->stats.sender_sources >
2178 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2181 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2183 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2184 if (src != sess->source && rtp_source_is_sender (src))
2193 for (position = 0; position < fci_length; position += 8) {
2194 guint8 *data = fci_data + position;
2196 ssrc = GST_READ_UINT32_BE (data);
2198 if (ssrc == rtp_source_get_ssrc (sess->source)) {
2206 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2210 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2211 RTPArrivalStats * arrival, GstClockTime current_time)
2213 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2214 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2215 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2216 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2217 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2218 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2220 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2221 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2223 if (g_signal_has_handler_pending (sess,
2224 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2225 GstBuffer *fci_buffer = NULL;
2227 if (fci_length > 0) {
2228 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2229 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2231 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2234 RTP_SESSION_UNLOCK (sess);
2235 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2236 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2237 RTP_SESSION_LOCK (sess);
2240 gst_buffer_unref (fci_buffer);
2243 if (sess->rtcp_feedback_retention_window) {
2244 RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
2245 GINT_TO_POINTER (media_ssrc));
2248 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2251 if (rtp_source_get_ssrc (sess->source) == media_ssrc ||
2252 /* PSFB FIR puts the media ssrc inside the FCI */
2253 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2255 case GST_RTCP_TYPE_PSFB:
2257 case GST_RTCP_PSFB_TYPE_PLI:
2258 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2261 case GST_RTCP_PSFB_TYPE_FIR:
2262 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2269 case GST_RTCP_TYPE_RTPFB:
2277 * rtp_session_process_rtcp:
2278 * @sess: and #RTPSession
2279 * @buffer: an RTCP buffer
2280 * @current_time: the current system time
2281 * @ntpnstime: the current NTP time in nanoseconds
2283 * Process an RTCP buffer in the session manager. This function takes ownership
2286 * Returns: a #GstFlowReturn.
2289 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2290 GstClockTime current_time, guint64 ntpnstime)
2292 GstRTCPPacket packet;
2293 gboolean more, is_bye = FALSE, do_sync = FALSE;
2294 RTPArrivalStats arrival = { NULL, };
2295 GstFlowReturn result = GST_FLOW_OK;
2296 GstRTCPBuffer rtcp = { NULL, };
2298 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2299 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2301 if (!gst_rtcp_buffer_validate (buffer))
2302 goto invalid_packet;
2304 GST_DEBUG ("received RTCP packet");
2306 RTP_SESSION_LOCK (sess);
2307 /* update arrival stats */
2308 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2314 /* start processing the compound packet */
2315 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2316 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2320 type = gst_rtcp_packet_get_type (&packet);
2322 /* when we are leaving the session, we should ignore all non-BYE messages */
2323 if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
2324 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2329 case GST_RTCP_TYPE_SR:
2330 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2332 case GST_RTCP_TYPE_RR:
2333 rtp_session_process_rr (sess, &packet, &arrival);
2335 case GST_RTCP_TYPE_SDES:
2336 rtp_session_process_sdes (sess, &packet, &arrival);
2338 case GST_RTCP_TYPE_BYE:
2340 /* don't try to attempt lip-sync anymore for streams with a BYE */
2342 rtp_session_process_bye (sess, &packet, &arrival);
2344 case GST_RTCP_TYPE_APP:
2345 rtp_session_process_app (sess, &packet, &arrival);
2347 case GST_RTCP_TYPE_RTPFB:
2348 case GST_RTCP_TYPE_PSFB:
2349 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2352 GST_WARNING ("got unknown RTCP packet");
2356 more = gst_rtcp_packet_move_to_next (&packet);
2359 gst_rtcp_buffer_unmap (&rtcp);
2361 /* if we are scheduling a BYE, we only want to count bye packets, else we
2362 * count everything */
2363 if (sess->source->received_bye) {
2365 sess->stats.bye_members++;
2366 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2369 /* keep track of average packet size */
2370 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2372 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2373 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2374 RTP_SESSION_UNLOCK (sess);
2376 clean_arrival_stats (&arrival);
2378 /* notify caller of sr packets in the callback */
2379 if (do_sync && sess->callbacks.sync_rtcp) {
2380 /* make writable, we might want to change the buffer */
2381 buffer = gst_buffer_make_writable (buffer);
2383 result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
2384 sess->sync_rtcp_user_data);
2386 gst_buffer_unref (buffer);
2393 GST_DEBUG ("invalid RTCP packet received");
2394 gst_buffer_unref (buffer);
2399 RTP_SESSION_UNLOCK (sess);
2400 gst_buffer_unref (buffer);
2401 clean_arrival_stats (&arrival);
2402 GST_DEBUG ("ignoring RTCP packet because we left");
2408 * rtp_session_update_send_caps:
2409 * @sess: an #RTPSession
2412 * Update the caps of the sender in the rtp session.
2415 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2417 g_return_if_fail (RTP_IS_SESSION (sess));
2418 g_return_if_fail (GST_IS_CAPS (caps));
2420 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2422 RTP_SESSION_LOCK (sess);
2423 rtp_source_update_caps (sess->source, caps);
2424 RTP_SESSION_UNLOCK (sess);
2428 * rtp_session_send_rtp:
2429 * @sess: an #RTPSession
2430 * @data: pointer to either an RTP buffer or a list of RTP buffers
2431 * @is_list: TRUE when @data is a buffer list
2432 * @current_time: the current system time
2433 * @running_time: the running time of @data
2435 * Send the RTP buffer in the session manager. This function takes ownership of
2438 * Returns: a #GstFlowReturn.
2441 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2442 GstClockTime current_time, GstClockTime running_time)
2444 GstFlowReturn result;
2446 gboolean prevsender;
2449 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2450 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2452 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2454 RTP_SESSION_LOCK (sess);
2455 source = sess->source;
2457 /* update last activity */
2458 source->last_rtp_activity = current_time;
2460 prevsender = RTP_SOURCE_IS_SENDER (source);
2461 oldrate = source->bitrate;
2463 /* we use our own source to send */
2464 result = rtp_source_send_rtp (source, data, is_list, running_time);
2466 if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2467 sess->stats.sender_sources++;
2468 if (oldrate != source->bitrate)
2469 sess->recalc_bandwidth = TRUE;
2470 RTP_SESSION_UNLOCK (sess);
2476 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2478 *bandwidth += source->bitrate;
2482 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2485 GstClockTime result;
2487 /* recalculate bandwidth when it changed */
2488 if (sess->recalc_bandwidth) {
2491 if (sess->bandwidth > 0)
2492 bandwidth = sess->bandwidth;
2494 /* If it is <= 0, then try to estimate the actual bandwidth */
2495 bandwidth = sess->source->bitrate;
2497 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2498 (GHFunc) add_bitrates, &bandwidth);
2501 if (bandwidth < 8000)
2502 bandwidth = RTP_STATS_BANDWIDTH;
2504 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2505 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2507 sess->recalc_bandwidth = FALSE;
2510 if (sess->source->received_bye) {
2511 result = rtp_stats_calculate_bye_interval (&sess->stats);
2513 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2514 RTP_SOURCE_IS_SENDER (sess->source), first);
2517 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2518 GST_TIME_ARGS (result), first);
2520 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2521 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2523 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2528 /* Stop the current @sess and schedule a BYE message for the other members.
2529 * One must have the session lock to call this function
2531 static GstFlowReturn
2532 rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2533 GstClockTime current_time)
2535 GstFlowReturn result = GST_FLOW_OK;
2537 GstClockTime interval;
2539 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2541 source = sess->source;
2543 /* ignore more BYEs */
2544 if (source->received_bye)
2547 /* we have BYE now */
2548 source->received_bye = TRUE;
2549 /* at least one member wants to send a BYE */
2550 g_free (sess->bye_reason);
2551 sess->bye_reason = g_strdup (reason);
2552 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2553 sess->stats.bye_members = 1;
2554 sess->first_rtcp = TRUE;
2555 sess->sent_bye = FALSE;
2556 sess->allow_early = TRUE;
2558 /* reschedule transmission */
2559 sess->last_rtcp_send_time = current_time;
2560 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2561 sess->next_rtcp_check_time = current_time + interval;
2563 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2564 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2566 RTP_SESSION_UNLOCK (sess);
2567 /* notify app of reconsideration */
2568 if (sess->callbacks.reconsider)
2569 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2570 RTP_SESSION_LOCK (sess);
2577 * rtp_session_schedule_bye:
2578 * @sess: an #RTPSession
2579 * @reason: a reason or NULL
2580 * @current_time: the current system time
2582 * Stop the current @sess and schedule a BYE message for the other members.
2584 * Returns: a #GstFlowReturn.
2587 rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2588 GstClockTime current_time)
2590 GstFlowReturn result = GST_FLOW_OK;
2592 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2594 RTP_SESSION_LOCK (sess);
2595 result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2596 RTP_SESSION_UNLOCK (sess);
2602 * rtp_session_next_timeout:
2603 * @sess: an #RTPSession
2604 * @current_time: the current system time
2606 * Get the next time we should perform session maintenance tasks.
2608 * Returns: a time when rtp_session_on_timeout() should be called with the
2609 * current system time.
2612 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2614 GstClockTime result, interval = 0;
2616 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2618 RTP_SESSION_LOCK (sess);
2620 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2621 result = sess->next_early_rtcp_time;
2625 result = sess->next_rtcp_check_time;
2627 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2628 ", next time: %" GST_TIME_FORMAT,
2629 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2631 if (result < current_time) {
2632 GST_DEBUG ("take current time as base");
2633 /* our previous check time expired, start counting from the current time
2635 result = current_time;
2638 if (sess->source->received_bye) {
2639 if (sess->sent_bye) {
2640 GST_DEBUG ("we sent BYE already");
2641 interval = GST_CLOCK_TIME_NONE;
2642 } else if (sess->stats.active_sources >= 50) {
2643 GST_DEBUG ("reconsider BYE, more than 50 sources");
2644 /* reconsider BYE if members >= 50 */
2645 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2648 if (sess->first_rtcp) {
2649 GST_DEBUG ("first RTCP packet");
2650 /* we are called for the first time */
2651 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2652 } else if (sess->next_rtcp_check_time < current_time) {
2653 GST_DEBUG ("old check time expired, getting new timeout");
2654 /* get a new timeout when we need to */
2655 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2659 if (interval != GST_CLOCK_TIME_NONE)
2662 result = GST_CLOCK_TIME_NONE;
2664 sess->next_rtcp_check_time = result;
2668 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2669 ", next time: %" GST_TIME_FORMAT,
2670 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2671 RTP_SESSION_UNLOCK (sess);
2678 GstRTCPBuffer rtcpbuf;
2681 GstClockTime current_time;
2683 GstClockTime running_time;
2684 GstClockTime interval;
2685 GstRTCPPacket packet;
2689 gboolean may_suppress;
2693 session_start_rtcp (RTPSession * sess, ReportData * data)
2695 GstRTCPPacket *packet = &data->packet;
2696 RTPSource *own = sess->source;
2697 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2699 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2701 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2703 if (RTP_SOURCE_IS_SENDER (own)) {
2706 guint32 packet_count, octet_count;
2708 /* we are a sender, create SR */
2709 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2710 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2712 /* get latest stats */
2713 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2714 &ntptime, &rtptime, &packet_count, &octet_count);
2716 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2717 packet_count, octet_count);
2719 /* fill in sender report info */
2720 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2721 ntptime, rtptime, packet_count, octet_count);
2723 /* we are only receiver, create RR */
2724 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2725 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2726 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2730 /* construct a Sender or Receiver Report */
2732 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2734 RTPSession *sess = data->sess;
2735 GstRTCPPacket *packet = &data->packet;
2737 /* create a new buffer if needed */
2738 if (data->rtcp == NULL) {
2739 session_start_rtcp (sess, data);
2740 } else if (data->is_early) {
2741 /* Put a single RR or SR in minimal compound packets */
2744 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2745 /* only report about other sender sources */
2746 if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2747 guint8 fractionlost;
2749 guint32 exthighestseq, jitter;
2753 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2754 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2756 /* store last generated RR packet */
2757 source->last_rr.is_valid = TRUE;
2758 source->last_rr.fractionlost = fractionlost;
2759 source->last_rr.packetslost = packetslost;
2760 source->last_rr.exthighestseq = exthighestseq;
2761 source->last_rr.jitter = jitter;
2762 source->last_rr.lsr = lsr;
2763 source->last_rr.dlsr = dlsr;
2765 /* packet is not yet filled, add report block for this source. */
2766 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2767 exthighestseq, jitter, lsr, dlsr);
2772 /* perform cleanup of sources that timed out */
2774 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2776 gboolean remove = FALSE;
2777 gboolean byetimeout = FALSE;
2778 gboolean sendertimeout = FALSE;
2779 gboolean is_sender, is_active;
2780 RTPSession *sess = data->sess;
2781 GstClockTime interval, binterval;
2784 is_sender = RTP_SOURCE_IS_SENDER (source);
2785 is_active = RTP_SOURCE_IS_ACTIVE (source);
2787 /* our own rtcp interval may have been forced low by secondary configuration,
2788 * while sender side may still operate with higher interval,
2789 * so do not just take our interval to decide on timing out sender,
2790 * but take (if data->interval <= 5 * GST_SECOND):
2791 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2792 * where sender_interval is difference between last 2 received RTCP reports
2794 if (data->interval >= 5 * GST_SECOND || (source == sess->source)) {
2795 binterval = data->interval;
2797 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2798 GST_TIME_ARGS (source->stats.prev_rtcptime),
2799 GST_TIME_ARGS (source->stats.last_rtcptime));
2800 /* if not received enough yet, fallback to larger default */
2801 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2802 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2804 binterval = 5 * GST_SECOND;
2805 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2807 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2808 GST_TIME_ARGS (binterval));
2810 /* check for our own source, we don't want to delete our own source. */
2811 if (!(source == sess->source)) {
2812 if (source->received_bye) {
2813 /* if we received a BYE from the source, remove the source after some
2815 if (data->current_time > source->bye_time &&
2816 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2817 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2822 /* sources that were inactive for more than 5 times the deterministic reporting
2823 * interval get timed out. the min timeout is 5 seconds. */
2824 /* mind old time that might pre-date last time going to PLAYING */
2825 btime = MAX (source->last_activity, sess->start_time);
2826 if (data->current_time > btime) {
2827 interval = MAX (binterval * 5, 5 * GST_SECOND);
2828 if (data->current_time - btime > interval) {
2829 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2830 source->ssrc, GST_TIME_ARGS (btime));
2836 /* senders that did not send for a long time become a receiver, this also
2837 * holds for our own source. */
2839 /* mind old time that might pre-date last time going to PLAYING */
2840 btime = MAX (source->last_rtp_activity, sess->start_time);
2841 if (data->current_time > btime) {
2842 interval = MAX (binterval * 2, 5 * GST_SECOND);
2843 if (data->current_time - btime > interval) {
2844 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2845 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2846 source->is_sender = FALSE;
2847 sess->stats.sender_sources--;
2848 sendertimeout = TRUE;
2854 sess->total_sources--;
2856 sess->stats.sender_sources--;
2858 sess->stats.active_sources--;
2861 on_bye_timeout (sess, source);
2863 on_timeout (sess, source);
2866 on_sender_timeout (sess, source);
2869 source->closing = remove;
2873 session_sdes (RTPSession * sess, ReportData * data)
2875 GstRTCPPacket *packet = &data->packet;
2876 const GstStructure *sdes;
2878 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2880 /* add SDES packet */
2881 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2883 gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2885 sdes = rtp_source_get_sdes_struct (sess->source);
2887 /* add all fields in the structure, the order is not important. */
2888 n_fields = gst_structure_n_fields (sdes);
2889 for (i = 0; i < n_fields; ++i) {
2892 GstRTCPSDESType type;
2894 field = gst_structure_nth_field_name (sdes, i);
2897 value = gst_structure_get_string (sdes, field);
2900 type = gst_rtcp_sdes_name_to_type (field);
2902 /* Early packets are minimal and only include the CNAME */
2903 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2906 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2907 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2908 (const guint8 *) value);
2909 } else if (type == GST_RTCP_SDES_PRIV) {
2915 /* don't accept entries that are too big */
2916 prefix_len = strlen (field);
2917 if (prefix_len > 255)
2919 value_len = strlen (value);
2920 if (value_len > 255)
2922 data_len = 1 + prefix_len + value_len;
2926 data[0] = prefix_len;
2927 memcpy (&data[1], field, prefix_len);
2928 memcpy (&data[1 + prefix_len], value, value_len);
2930 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2934 data->has_sdes = TRUE;
2937 /* schedule a BYE packet */
2939 session_bye (RTPSession * sess, ReportData * data)
2941 GstRTCPPacket *packet = &data->packet;
2942 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2945 session_start_rtcp (sess, data);
2948 session_sdes (sess, data);
2950 /* add a BYE packet */
2951 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
2952 gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2953 if (sess->bye_reason)
2954 gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2956 /* we have a BYE packet now */
2957 data->is_bye = TRUE;
2961 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2963 GstClockTime new_send_time, elapsed;
2965 if (data->is_early && sess->next_early_rtcp_time < current_time)
2968 /* no need to check yet */
2969 if (sess->next_rtcp_check_time > current_time) {
2970 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2971 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2972 GST_TIME_ARGS (current_time));
2976 /* get elapsed time since we last reported */
2977 elapsed = current_time - sess->last_rtcp_send_time;
2979 /* perform forward reconsideration */
2980 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2982 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2983 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2985 new_send_time += sess->last_rtcp_send_time;
2987 /* check if reconsideration */
2988 if (current_time < new_send_time) {
2989 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2990 GST_TIME_ARGS (new_send_time));
2991 /* store new check time */
2992 sess->next_rtcp_check_time = new_send_time;
2998 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
3000 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3001 GST_TIME_ARGS (new_send_time));
3002 sess->next_rtcp_check_time = current_time + new_send_time;
3004 /* Apply the rules from RFC 4585 section 3.5.3 */
3005 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
3006 GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) *
3007 sess->stats.min_interval;
3009 /* This will caused the RTCP to be suppressed if no FB packets are added */
3010 if (sess->last_rtcp_send_time + T_rr_current_interval >
3011 sess->next_rtcp_check_time) {
3012 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3013 " last: %" GST_TIME_FORMAT
3014 " + T_rr_current_interval: %" GST_TIME_FORMAT
3015 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3016 GST_TIME_ARGS (sess->stats.min_interval),
3017 GST_TIME_ARGS (sess->last_rtcp_send_time),
3018 GST_TIME_ARGS (T_rr_current_interval),
3019 GST_TIME_ARGS (sess->next_rtcp_check_time));
3020 data->may_suppress = TRUE;
3028 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3030 g_hash_table_insert (hash_table, key, g_object_ref (source));
3034 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3036 return source->closing;
3040 * rtp_session_on_timeout:
3041 * @sess: an #RTPSession
3042 * @current_time: the current system time
3043 * @ntpnstime: the current NTP time in nanoseconds
3044 * @running_time: the current running_time of the pipeline
3046 * Perform maintenance actions after the timeout obtained with
3047 * rtp_session_next_timeout() expired.
3049 * This function will perform timeouts of receivers and senders, send a BYE
3050 * packet or generate RTCP packets with current session stats.
3052 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3053 * times, for each packet that should be processed.
3055 * Returns: a #GstFlowReturn.
3058 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3059 guint64 ntpnstime, GstClockTime running_time)
3061 GstFlowReturn result = GST_FLOW_OK;
3062 ReportData data = { GST_RTCP_BUFFER_INIT };
3064 GHashTable *table_copy;
3065 gboolean notify = FALSE;
3067 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3069 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3070 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3071 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3075 data.current_time = current_time;
3076 data.ntpnstime = ntpnstime;
3077 data.is_bye = FALSE;
3078 data.has_sdes = FALSE;
3079 data.may_suppress = FALSE;
3080 data.running_time = running_time;
3084 RTP_SESSION_LOCK (sess);
3085 /* get a new interval, we need this for various cleanups etc */
3086 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3088 /* Make a local copy of the hashtable. We need to do this because the
3089 * cleanup stage below releases the session lock. */
3090 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3091 (GDestroyNotify) g_object_unref);
3092 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3093 (GHFunc) clone_ssrcs_hashtable, table_copy);
3095 /* Clean up the session, mark the source for removing, this might release the
3097 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3098 g_hash_table_destroy (table_copy);
3100 /* Now remove the marked sources */
3101 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3102 (GHRFunc) remove_closing_sources, NULL);
3104 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3105 data.is_early = TRUE;
3107 data.is_early = FALSE;
3109 /* see if we need to generate SR or RR packets */
3110 if (is_rtcp_time (sess, current_time, &data)) {
3111 if (own->received_bye) {
3112 /* generate BYE instead */
3113 GST_DEBUG ("generating BYE message");
3114 session_bye (sess, &data);
3115 sess->sent_bye = TRUE;
3117 /* loop over all known sources and do something */
3118 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3119 (GHFunc) session_report_blocks, &data);
3124 /* we keep track of the last report time in order to timeout inactive
3125 * receivers or senders */
3126 if (!data.is_early && !data.may_suppress)
3127 sess->last_rtcp_send_time = data.current_time;
3128 sess->first_rtcp = FALSE;
3129 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3131 /* add SDES for this source when not already added */
3133 session_sdes (sess, &data);
3136 /* check for outdated collisions */
3137 GST_DEBUG ("Timing out collisions");
3138 rtp_source_timeout (sess->source, current_time,
3139 /* "a relatively long time" -- RFC 3550 section 8.2 */
3140 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
3141 running_time - sess->rtcp_feedback_retention_window);
3143 if (sess->change_ssrc) {
3144 GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
3145 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3146 GINT_TO_POINTER (own->ssrc));
3148 own->ssrc = rtp_session_create_new_ssrc (sess);
3149 rtp_source_reset (own);
3151 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3152 GINT_TO_POINTER (own->ssrc), own);
3154 g_free (sess->bye_reason);
3155 sess->bye_reason = NULL;
3156 sess->sent_bye = FALSE;
3157 sess->change_ssrc = FALSE;
3159 GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
3162 sess->allow_early = TRUE;
3164 RTP_SESSION_UNLOCK (sess);
3167 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3169 /* push out the RTCP packet */
3171 gboolean do_not_suppress;
3173 gst_rtcp_buffer_unmap (&data.rtcpbuf);
3175 /* Give the user a change to add its own packet */
3176 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3177 data.rtcp, data.is_early, &do_not_suppress);
3179 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3182 packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len;
3184 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3185 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3186 sess->stats.avg_rtcp_packet_size, packet_size);
3188 sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye,
3189 sess->send_rtcp_user_data);
3191 GST_DEBUG ("freeing packet callback: %p"
3192 " do_not_suppress: %d may_suppress: %d",
3193 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3194 gst_buffer_unref (data.rtcp);
3202 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3203 GstClockTimeDiff max_delay)
3205 GstClockTime T_dither_max;
3207 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3209 RTP_SESSION_LOCK (sess);
3211 /* Check if already requested */
3212 /* RFC 4585 section 3.5.2 step 2 */
3213 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3216 /* Ignore the request a scheduled packet will be in time anyway */
3217 if (current_time + max_delay > sess->next_rtcp_check_time)
3220 /* RFC 4585 section 3.5.2 step 2b */
3221 /* If the total sources is <=2, then there is only us and one peer */
3222 if (sess->total_sources <= 2) {
3225 /* Divide by 2 because l = 0.5 */
3226 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3230 /* RFC 4585 section 3.5.2 step 3 */
3231 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3234 /* RFC 4585 section 3.5.2 step 4
3235 * Don't send if allow_early is FALSE, but not if we are in
3236 * immediate mode, meaning we are part of a group of at most the
3237 * application-specific threshold.
3239 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3240 sess->allow_early == FALSE)
3244 /* Schedule an early transmission later */
3245 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3248 /* If no dithering, schedule it for NOW */
3249 sess->next_early_rtcp_time = current_time;
3252 RTP_SESSION_UNLOCK (sess);
3254 /* notify app of need to send packet early
3255 * and therefore of timeout change */
3256 if (sess->callbacks.reconsider)
3257 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3263 RTP_SESSION_UNLOCK (sess);
3267 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3268 gboolean fir, gint count)
3270 RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
3271 GUINT_TO_POINTER (ssrc));
3277 src->send_pli = FALSE;
3278 src->send_fir = TRUE;
3280 if (count == -1 || count != src->last_fir_count)
3281 src->current_send_fir_seqnum++;
3282 src->last_fir_count = count;
3283 } else if (!src->send_fir) {
3284 src->send_pli = TRUE;
3287 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3293 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3295 GstRTCPPacket packet;
3296 GstRTCPBuffer rtcp = { NULL, };
3297 gboolean ret = FALSE;
3299 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3301 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3302 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3303 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3307 gst_rtcp_buffer_unmap (&rtcp);
3313 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3316 gboolean ret = FALSE;
3317 GHashTableIter iter;
3318 gpointer key, value;
3319 gboolean started_fir = FALSE;
3320 GstRTCPPacket fir_rtcppacket;
3321 GstRTCPBuffer rtcp = { NULL, };
3323 RTP_SESSION_LOCK (sess);
3325 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3327 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3328 while (g_hash_table_iter_next (&iter, &key, &value)) {
3329 guint media_ssrc = GPOINTER_TO_UINT (key);
3330 RTPSource *media_src = value;
3333 if (media_src->send_fir) {
3335 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3338 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3339 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket,
3340 rtp_source_get_ssrc (sess->source));
3341 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3343 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3344 gst_rtcp_packet_remove (&fir_rtcppacket);
3350 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3351 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3355 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3356 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3358 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3360 fci_data[0] = media_src->current_send_fir_seqnum;
3361 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3362 media_src->send_fir = FALSE;
3366 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3367 while (g_hash_table_iter_next (&iter, &key, &value)) {
3368 guint media_ssrc = GPOINTER_TO_UINT (key);
3369 RTPSource *media_src = value;
3370 GstRTCPPacket pli_rtcppacket;
3372 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3373 has_pli_compare_func, NULL)) {
3374 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3376 /* Break because the packet is full, will put next request in a
3379 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3380 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket,
3381 rtp_source_get_ssrc (sess->source));
3382 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3385 media_src->send_pli = FALSE;
3387 gst_rtcp_buffer_unmap (&rtcp);
3389 RTP_SESSION_UNLOCK (sess);
3395 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3399 if (!sess->callbacks.send_rtcp)
3402 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3404 rtp_session_request_early_rtcp (sess, now, max_delay);