2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "gstrtpbin-marshal.h"
32 #include "rtpsession.h"
34 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
35 #define GST_CAT_DEFAULT rtp_session_debug
37 /* signals and args */
40 SIGNAL_GET_SOURCE_BY_SSRC,
42 SIGNAL_ON_SSRC_COLLISION,
43 SIGNAL_ON_SSRC_VALIDATED,
44 SIGNAL_ON_SSRC_ACTIVE,
47 SIGNAL_ON_BYE_TIMEOUT,
49 SIGNAL_ON_SENDER_TIMEOUT,
50 SIGNAL_ON_SENDING_RTCP,
51 SIGNAL_ON_FEEDBACK_RTCP,
56 #define DEFAULT_INTERNAL_SOURCE NULL
57 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
58 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
59 #define DEFAULT_RTCP_RR_BANDWIDTH -1
60 #define DEFAULT_RTCP_RS_BANDWIDTH -1
61 #define DEFAULT_RTCP_MTU 1400
62 #define DEFAULT_SDES NULL
63 #define DEFAULT_NUM_SOURCES 0
64 #define DEFAULT_NUM_ACTIVE_SOURCES 0
65 #define DEFAULT_SOURCES NULL
66 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
67 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
68 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
69 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
78 PROP_RTCP_RR_BANDWIDTH,
79 PROP_RTCP_RS_BANDWIDTH,
83 PROP_NUM_ACTIVE_SOURCES,
86 PROP_RTCP_MIN_INTERVAL,
87 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
88 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
93 /* update average packet size */
94 #define INIT_AVG(avg, val) \
96 #define UPDATE_AVG(avg, val) \
100 (avg) = ((val) + (15 * (avg))) >> 4;
103 /* The number RTCP intervals after which to timeout entries in the
106 #define RTCP_INTERVAL_COLLISION_TIMEOUT 10
108 /* GObject vmethods */
109 static void rtp_session_finalize (GObject * object);
110 static void rtp_session_set_property (GObject * object, guint prop_id,
111 const GValue * value, GParamSpec * pspec);
112 static void rtp_session_get_property (GObject * object, guint prop_id,
113 GValue * value, GParamSpec * pspec);
115 static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
116 GstBuffer * buffer, gboolean early);
117 static void rtp_session_send_rtcp (RTPSession * sess,
118 GstClockTimeDiff max_delay);
121 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
123 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
125 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
126 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
127 gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
128 static RTPSource *obtain_internal_source (RTPSession * sess,
129 guint32 ssrc, gboolean * created);
130 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
131 GstClockTime current_time);
132 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
133 gboolean deterministic, gboolean first);
136 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
137 const GValue * handler_return, gpointer data)
139 if (g_value_get_boolean (handler_return))
140 g_value_set_boolean (return_accu, TRUE);
146 rtp_session_class_init (RTPSessionClass * klass)
148 GObjectClass *gobject_class;
150 gobject_class = (GObjectClass *) klass;
152 gobject_class->finalize = rtp_session_finalize;
153 gobject_class->set_property = rtp_session_set_property;
154 gobject_class->get_property = rtp_session_get_property;
157 * RTPSession::get-source-by-ssrc:
158 * @session: the object which received the signal
159 * @ssrc: the SSRC of the RTPSource
161 * Request the #RTPSource object with SSRC @ssrc in @session.
163 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
164 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
165 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
166 get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
167 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
170 * RTPSession::on-new-ssrc:
171 * @session: the object which received the signal
172 * @src: the new RTPSource
174 * Notify of a new SSRC that entered @session.
176 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
177 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
178 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
179 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
182 * RTPSession::on-ssrc-collision:
183 * @session: the object which received the signal
184 * @src: the #RTPSource that caused a collision
186 * Notify when we have an SSRC collision
188 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
189 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
190 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
191 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
194 * RTPSession::on-ssrc-validated:
195 * @session: the object which received the signal
196 * @src: the new validated RTPSource
198 * Notify of a new SSRC that became validated.
200 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
201 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
202 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
203 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
206 * RTPSession::on-ssrc-active:
207 * @session: the object which received the signal
208 * @src: the active RTPSource
210 * Notify of a SSRC that is active, i.e., sending RTCP.
212 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
213 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
214 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
215 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
218 * RTPSession::on-ssrc-sdes:
219 * @session: the object which received the signal
220 * @src: the RTPSource
222 * Notify that a new SDES was received for SSRC.
224 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
225 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
226 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
227 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
230 * RTPSession::on-bye-ssrc:
231 * @session: the object which received the signal
232 * @src: the RTPSource that went away
234 * Notify of an SSRC that became inactive because of a BYE packet.
236 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
237 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
238 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
239 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
242 * RTPSession::on-bye-timeout:
243 * @session: the object which received the signal
244 * @src: the RTPSource that timed out
246 * Notify of an SSRC that has timed out because of BYE
248 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
249 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
250 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
251 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
254 * RTPSession::on-timeout:
255 * @session: the object which received the signal
256 * @src: the RTPSource that timed out
258 * Notify of an SSRC that has timed out
260 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
261 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
262 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
263 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
266 * RTPSession::on-sender-timeout:
267 * @session: the object which received the signal
268 * @src: the RTPSource that timed out
270 * Notify of an SSRC that was a sender but timed out and became a receiver.
272 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
273 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
274 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
275 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
279 * RTPSession::on-sending-rtcp
280 * @session: the object which received the signal
281 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
282 * @early: %TRUE if the packet is early, %FALSE if it is regular
284 * This signal is emitted before sending an RTCP packet, it can be used
285 * to add extra RTCP Packets.
287 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
288 * if suppressing it is acceptable
290 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
291 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
292 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
293 accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__BOXED_BOOLEAN,
294 G_TYPE_BOOLEAN, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE,
298 * RTPSession::on-feedback-rtcp:
299 * @session: the object which received the signal
300 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
301 * %GST_RTCP_TYPE_RTPFB
302 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
303 * @sender_ssrc: The SSRC of the sender
304 * @media_ssrc: The SSRC of the media this refers to
305 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
308 * Notify that a RTCP feedback packet has been received
310 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
311 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
312 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
313 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_BOXED,
314 G_TYPE_NONE, 5, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
318 * RTPSession::send-rtcp:
319 * @session: the object which received the signal
320 * @max_delay: The maximum delay after which the feedback will not be useful
323 * Requests that the #RTPSession initiate a new RTCP packet as soon as
324 * possible within the requested delay.
327 rtp_session_signals[SIGNAL_SEND_RTCP] =
328 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
329 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
330 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
331 gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64);
333 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
334 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
335 "The internal SSRC used for the session",
336 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
338 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
339 g_param_spec_object ("internal-source", "Internal Source",
340 "The internal source element of the session",
341 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
343 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
344 g_param_spec_double ("bandwidth", "Bandwidth",
345 "The bandwidth of the session (0 for auto-discover)",
346 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
347 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
349 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
350 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
351 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
352 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
353 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
355 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
356 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
357 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
358 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
359 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
361 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
362 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
363 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
364 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
365 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
367 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
368 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
369 "The maximum size of the RTCP packets",
370 16, G_MAXINT16, DEFAULT_RTCP_MTU,
371 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
373 g_object_class_install_property (gobject_class, PROP_SDES,
374 g_param_spec_boxed ("sdes", "SDES",
375 "The SDES items of this session",
376 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
378 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
379 g_param_spec_uint ("num-sources", "Num Sources",
380 "The number of sources in the session", 0, G_MAXUINT,
381 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
383 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
384 g_param_spec_uint ("num-active-sources", "Num Active Sources",
385 "The number of active sources in the session", 0, G_MAXUINT,
386 DEFAULT_NUM_ACTIVE_SOURCES,
387 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
391 * Get a GValue Array of all sources in the session.
394 * <title>Getting the #RTPSources of a session
401 * g_object_get (sess, "sources", &arr, NULL);
403 * for (i = 0; i < arr->n_values; i++) {
406 * val = g_value_array_get_nth (arr, i);
407 * source = g_value_get_object (val);
409 * g_value_array_free (arr);
414 g_object_class_install_property (gobject_class, PROP_SOURCES,
415 g_param_spec_boxed ("sources", "Sources",
416 "An array of all known sources in the session",
417 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
419 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
420 g_param_spec_boolean ("favor-new", "Favor new sources",
421 "Resolve SSRC conflict in favor of new sources", FALSE,
422 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
424 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
425 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
426 "Minimum interval between Regular RTCP packet (in ns)",
427 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
428 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
430 g_object_class_install_property (gobject_class,
431 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
432 g_param_spec_uint64 ("rtcp-feedback-retention-window",
433 "RTCP Feedback retention window",
434 "Duration during which RTCP Feedback packets are retained (in ns)",
435 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
436 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
438 g_object_class_install_property (gobject_class,
439 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
440 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
441 "RTCP Immediate Feedback threshold",
442 "The maximum number of members of a RTP session for which immediate"
444 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
445 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
447 g_object_class_install_property (gobject_class, PROP_PROBATION,
448 g_param_spec_uint ("probation", "Number of probations",
449 "Consecutive packet sequence numbers to accept the source",
450 0, G_MAXUINT, DEFAULT_PROBATION,
451 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
453 klass->get_source_by_ssrc =
454 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
455 klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
456 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
458 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
462 rtp_session_init (RTPSession * sess)
470 g_mutex_init (&sess->lock);
471 sess->key = g_random_int ();
475 for (i = 0; i < 32; i++) {
477 g_hash_table_new_full (NULL, NULL, NULL,
478 (GDestroyNotify) g_object_unref);
481 rtp_stats_init_defaults (&sess->stats);
482 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
483 rtp_stats_set_min_interval (&sess->stats,
484 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
486 sess->recalc_bandwidth = TRUE;
487 sess->bandwidth = DEFAULT_BANDWIDTH;
488 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
489 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
490 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
492 /* default UDP header length */
493 sess->header_len = 28;
494 sess->mtu = DEFAULT_RTCP_MTU;
496 sess->probation = DEFAULT_PROBATION;
498 /* some default SDES entries */
499 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
501 /* we do not want to leak details like the username or hostname here */
502 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
503 gst_structure_set (sdes, "cname", G_TYPE_STRING, str, NULL);
507 /* we do not want to leak the user's real name here */
508 str = g_strdup_printf ("Anon%u", g_random_int ());
509 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
513 gst_structure_set (sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
515 /* create an active SSRC for this session manager */
516 ssrc = rtp_session_create_new_ssrc (sess);
517 sess->source = obtain_internal_source (sess, ssrc, &created);
518 /* and configure sdes in the source */
519 rtp_source_set_sdes_struct (sess->source, sdes);
521 sess->first_rtcp = TRUE;
522 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
524 sess->allow_early = TRUE;
525 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
526 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
527 sess->rtcp_immediate_feedback_threshold =
528 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
530 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
534 rtp_session_finalize (GObject * object)
539 sess = RTP_SESSION_CAST (object);
541 g_mutex_clear (&sess->lock);
543 for (i = 0; i < 32; i++)
544 g_hash_table_destroy (sess->ssrcs[i]);
546 g_object_unref (sess->source);
548 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
552 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
554 GValue value = { 0 };
556 g_value_init (&value, RTP_TYPE_SOURCE);
557 g_value_take_object (&value, source);
558 /* copies the value */
559 g_value_array_append (arr, &value);
563 rtp_session_create_sources (RTPSession * sess)
568 RTP_SESSION_LOCK (sess);
569 /* get number of elements in the table */
570 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
571 /* create the result value array */
572 res = g_value_array_new (size);
574 /* and copy all values into the array */
575 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
576 RTP_SESSION_UNLOCK (sess);
582 rtp_session_set_property (GObject * object, guint prop_id,
583 const GValue * value, GParamSpec * pspec)
587 sess = RTP_SESSION (object);
590 case PROP_INTERNAL_SSRC:
591 rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
594 RTP_SESSION_LOCK (sess);
595 sess->bandwidth = g_value_get_double (value);
596 sess->recalc_bandwidth = TRUE;
597 RTP_SESSION_UNLOCK (sess);
599 case PROP_RTCP_FRACTION:
600 RTP_SESSION_LOCK (sess);
601 sess->rtcp_bandwidth = g_value_get_double (value);
602 sess->recalc_bandwidth = TRUE;
603 RTP_SESSION_UNLOCK (sess);
605 case PROP_RTCP_RR_BANDWIDTH:
606 RTP_SESSION_LOCK (sess);
607 sess->rtcp_rr_bandwidth = g_value_get_int (value);
608 sess->recalc_bandwidth = TRUE;
609 RTP_SESSION_UNLOCK (sess);
611 case PROP_RTCP_RS_BANDWIDTH:
612 RTP_SESSION_LOCK (sess);
613 sess->rtcp_rs_bandwidth = g_value_get_int (value);
614 sess->recalc_bandwidth = TRUE;
615 RTP_SESSION_UNLOCK (sess);
618 sess->mtu = g_value_get_uint (value);
621 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
624 sess->favor_new = g_value_get_boolean (value);
626 case PROP_RTCP_MIN_INTERVAL:
627 rtp_stats_set_min_interval (&sess->stats,
628 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
629 /* trigger reconsideration */
630 RTP_SESSION_LOCK (sess);
631 sess->next_rtcp_check_time = 0;
632 RTP_SESSION_UNLOCK (sess);
633 if (sess->callbacks.reconsider)
634 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
636 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
637 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
640 sess->probation = g_value_get_uint (value);
641 g_object_set_property (G_OBJECT (sess->source), "probation", value);
644 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
650 rtp_session_get_property (GObject * object, guint prop_id,
651 GValue * value, GParamSpec * pspec)
655 sess = RTP_SESSION (object);
658 case PROP_INTERNAL_SSRC:
659 g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
661 case PROP_INTERNAL_SOURCE:
662 g_value_take_object (value, rtp_session_get_internal_source (sess));
665 g_value_set_double (value, sess->bandwidth);
667 case PROP_RTCP_FRACTION:
668 g_value_set_double (value, sess->rtcp_bandwidth);
670 case PROP_RTCP_RR_BANDWIDTH:
671 g_value_set_int (value, sess->rtcp_rr_bandwidth);
673 case PROP_RTCP_RS_BANDWIDTH:
674 g_value_set_int (value, sess->rtcp_rs_bandwidth);
677 g_value_set_uint (value, sess->mtu);
680 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
682 case PROP_NUM_SOURCES:
683 g_value_set_uint (value, rtp_session_get_num_sources (sess));
685 case PROP_NUM_ACTIVE_SOURCES:
686 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
689 g_value_take_boxed (value, rtp_session_create_sources (sess));
692 g_value_set_boolean (value, sess->favor_new);
694 case PROP_RTCP_MIN_INTERVAL:
695 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
697 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
698 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
701 g_value_set_uint (value, sess->probation);
702 g_object_get_property (G_OBJECT (sess->source), "probation", value);
705 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
711 on_new_ssrc (RTPSession * sess, RTPSource * source)
713 g_object_ref (source);
714 RTP_SESSION_UNLOCK (sess);
715 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
716 RTP_SESSION_LOCK (sess);
717 g_object_unref (source);
721 on_ssrc_collision (RTPSession * sess, RTPSource * source)
723 g_object_ref (source);
724 RTP_SESSION_UNLOCK (sess);
725 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
727 RTP_SESSION_LOCK (sess);
728 g_object_unref (source);
732 on_ssrc_validated (RTPSession * sess, RTPSource * source)
734 g_object_ref (source);
735 RTP_SESSION_UNLOCK (sess);
736 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
738 RTP_SESSION_LOCK (sess);
739 g_object_unref (source);
743 on_ssrc_active (RTPSession * sess, RTPSource * source)
745 g_object_ref (source);
746 RTP_SESSION_UNLOCK (sess);
747 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
748 RTP_SESSION_LOCK (sess);
749 g_object_unref (source);
753 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
755 g_object_ref (source);
756 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
757 RTP_SESSION_UNLOCK (sess);
758 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
759 RTP_SESSION_LOCK (sess);
760 g_object_unref (source);
764 on_bye_ssrc (RTPSession * sess, RTPSource * source)
766 g_object_ref (source);
767 RTP_SESSION_UNLOCK (sess);
768 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
769 RTP_SESSION_LOCK (sess);
770 g_object_unref (source);
774 on_bye_timeout (RTPSession * sess, RTPSource * source)
776 g_object_ref (source);
777 RTP_SESSION_UNLOCK (sess);
778 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
779 RTP_SESSION_LOCK (sess);
780 g_object_unref (source);
784 on_timeout (RTPSession * sess, RTPSource * source)
786 g_object_ref (source);
787 RTP_SESSION_UNLOCK (sess);
788 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
789 RTP_SESSION_LOCK (sess);
790 g_object_unref (source);
794 on_sender_timeout (RTPSession * sess, RTPSource * source)
796 g_object_ref (source);
797 RTP_SESSION_UNLOCK (sess);
798 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
800 RTP_SESSION_LOCK (sess);
801 g_object_unref (source);
807 * Create a new session object.
809 * Returns: a new #RTPSession. g_object_unref() after usage.
812 rtp_session_new (void)
816 sess = g_object_new (RTP_TYPE_SESSION, NULL);
822 * rtp_session_set_callbacks:
823 * @sess: an #RTPSession
824 * @callbacks: callbacks to configure
825 * @user_data: user data passed in the callbacks
827 * Configure a set of callbacks to be notified of actions.
830 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
833 g_return_if_fail (RTP_IS_SESSION (sess));
835 if (callbacks->process_rtp) {
836 sess->callbacks.process_rtp = callbacks->process_rtp;
837 sess->process_rtp_user_data = user_data;
839 if (callbacks->send_rtp) {
840 sess->callbacks.send_rtp = callbacks->send_rtp;
841 sess->send_rtp_user_data = user_data;
843 if (callbacks->send_rtcp) {
844 sess->callbacks.send_rtcp = callbacks->send_rtcp;
845 sess->send_rtcp_user_data = user_data;
847 if (callbacks->sync_rtcp) {
848 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
849 sess->sync_rtcp_user_data = user_data;
851 if (callbacks->clock_rate) {
852 sess->callbacks.clock_rate = callbacks->clock_rate;
853 sess->clock_rate_user_data = user_data;
855 if (callbacks->reconsider) {
856 sess->callbacks.reconsider = callbacks->reconsider;
857 sess->reconsider_user_data = user_data;
859 if (callbacks->request_key_unit) {
860 sess->callbacks.request_key_unit = callbacks->request_key_unit;
861 sess->request_key_unit_user_data = user_data;
863 if (callbacks->request_time) {
864 sess->callbacks.request_time = callbacks->request_time;
865 sess->request_time_user_data = user_data;
870 * rtp_session_set_process_rtp_callback:
871 * @sess: an #RTPSession
872 * @callback: callback to set
873 * @user_data: user data passed in the callback
875 * Configure only the process_rtp callback to be notified of the process_rtp action.
878 rtp_session_set_process_rtp_callback (RTPSession * sess,
879 RTPSessionProcessRTP callback, gpointer user_data)
881 g_return_if_fail (RTP_IS_SESSION (sess));
883 sess->callbacks.process_rtp = callback;
884 sess->process_rtp_user_data = user_data;
888 * rtp_session_set_send_rtp_callback:
889 * @sess: an #RTPSession
890 * @callback: callback to set
891 * @user_data: user data passed in the callback
893 * Configure only the send_rtp callback to be notified of the send_rtp action.
896 rtp_session_set_send_rtp_callback (RTPSession * sess,
897 RTPSessionSendRTP callback, gpointer user_data)
899 g_return_if_fail (RTP_IS_SESSION (sess));
901 sess->callbacks.send_rtp = callback;
902 sess->send_rtp_user_data = user_data;
906 * rtp_session_set_send_rtcp_callback:
907 * @sess: an #RTPSession
908 * @callback: callback to set
909 * @user_data: user data passed in the callback
911 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
914 rtp_session_set_send_rtcp_callback (RTPSession * sess,
915 RTPSessionSendRTCP callback, gpointer user_data)
917 g_return_if_fail (RTP_IS_SESSION (sess));
919 sess->callbacks.send_rtcp = callback;
920 sess->send_rtcp_user_data = user_data;
924 * rtp_session_set_sync_rtcp_callback:
925 * @sess: an #RTPSession
926 * @callback: callback to set
927 * @user_data: user data passed in the callback
929 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
932 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
933 RTPSessionSyncRTCP callback, gpointer user_data)
935 g_return_if_fail (RTP_IS_SESSION (sess));
937 sess->callbacks.sync_rtcp = callback;
938 sess->sync_rtcp_user_data = user_data;
942 * rtp_session_set_clock_rate_callback:
943 * @sess: an #RTPSession
944 * @callback: callback to set
945 * @user_data: user data passed in the callback
947 * Configure only the clock_rate callback to be notified of the clock_rate action.
950 rtp_session_set_clock_rate_callback (RTPSession * sess,
951 RTPSessionClockRate callback, gpointer user_data)
953 g_return_if_fail (RTP_IS_SESSION (sess));
955 sess->callbacks.clock_rate = callback;
956 sess->clock_rate_user_data = user_data;
960 * rtp_session_set_reconsider_callback:
961 * @sess: an #RTPSession
962 * @callback: callback to set
963 * @user_data: user data passed in the callback
965 * Configure only the reconsider callback to be notified of the reconsider action.
968 rtp_session_set_reconsider_callback (RTPSession * sess,
969 RTPSessionReconsider callback, gpointer user_data)
971 g_return_if_fail (RTP_IS_SESSION (sess));
973 sess->callbacks.reconsider = callback;
974 sess->reconsider_user_data = user_data;
978 * rtp_session_set_request_time_callback:
979 * @sess: an #RTPSession
980 * @callback: callback to set
981 * @user_data: user data passed in the callback
983 * Configure only the request_time callback
986 rtp_session_set_request_time_callback (RTPSession * sess,
987 RTPSessionRequestTime callback, gpointer user_data)
989 g_return_if_fail (RTP_IS_SESSION (sess));
991 sess->callbacks.request_time = callback;
992 sess->request_time_user_data = user_data;
996 * rtp_session_set_bandwidth:
997 * @sess: an #RTPSession
998 * @bandwidth: the bandwidth allocated
1000 * Set the session bandwidth in bytes per second.
1003 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1005 g_return_if_fail (RTP_IS_SESSION (sess));
1007 RTP_SESSION_LOCK (sess);
1008 sess->stats.bandwidth = bandwidth;
1009 RTP_SESSION_UNLOCK (sess);
1013 * rtp_session_get_bandwidth:
1014 * @sess: an #RTPSession
1016 * Get the session bandwidth.
1018 * Returns: the session bandwidth.
1021 rtp_session_get_bandwidth (RTPSession * sess)
1025 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1027 RTP_SESSION_LOCK (sess);
1028 result = sess->stats.bandwidth;
1029 RTP_SESSION_UNLOCK (sess);
1035 * rtp_session_set_rtcp_fraction:
1036 * @sess: an #RTPSession
1037 * @bandwidth: the RTCP bandwidth
1039 * Set the bandwidth in bytes per second that should be used for RTCP
1043 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1045 g_return_if_fail (RTP_IS_SESSION (sess));
1047 RTP_SESSION_LOCK (sess);
1048 sess->stats.rtcp_bandwidth = bandwidth;
1049 RTP_SESSION_UNLOCK (sess);
1053 * rtp_session_get_rtcp_fraction:
1054 * @sess: an #RTPSession
1056 * Get the session bandwidth used for RTCP.
1058 * Returns: The bandwidth used for RTCP messages.
1061 rtp_session_get_rtcp_fraction (RTPSession * sess)
1065 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1067 RTP_SESSION_LOCK (sess);
1068 result = sess->stats.rtcp_bandwidth;
1069 RTP_SESSION_UNLOCK (sess);
1075 * rtp_session_get_sdes_struct:
1076 * @sess: an #RTSPSession
1078 * Get the SDES data as a #GstStructure
1080 * Returns: a GstStructure with SDES items for @sess. This function returns a
1081 * copy of the SDES structure, use gst_structure_free() after usage.
1084 rtp_session_get_sdes_struct (RTPSession * sess)
1086 const GstStructure *sdes;
1087 GstStructure *result = NULL;
1089 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1091 RTP_SESSION_LOCK (sess);
1092 sdes = rtp_source_get_sdes_struct (sess->source);
1094 result = gst_structure_copy (sdes);
1095 RTP_SESSION_UNLOCK (sess);
1101 * rtp_session_set_sdes_struct:
1102 * @sess: an #RTSPSession
1103 * @sdes: a #GstStructure
1105 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1108 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1110 g_return_if_fail (sdes);
1111 g_return_if_fail (RTP_IS_SESSION (sess));
1113 RTP_SESSION_LOCK (sess);
1114 rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
1115 RTP_SESSION_UNLOCK (sess);
1118 static GstFlowReturn
1119 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1121 GstFlowReturn result = GST_FLOW_OK;
1123 if (source->internal) {
1124 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1126 RTP_SESSION_UNLOCK (session);
1128 if (session->callbacks.send_rtp)
1130 session->callbacks.send_rtp (session, source, data,
1131 session->send_rtp_user_data);
1133 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1136 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1137 RTP_SESSION_UNLOCK (session);
1139 if (session->callbacks.process_rtp)
1141 session->callbacks.process_rtp (session, source,
1142 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1144 gst_buffer_unref (GST_BUFFER_CAST (data));
1146 RTP_SESSION_LOCK (session);
1152 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1156 RTP_SESSION_UNLOCK (session);
1158 if (session->callbacks.clock_rate)
1160 session->callbacks.clock_rate (session, pt,
1161 session->clock_rate_user_data);
1165 RTP_SESSION_LOCK (session);
1167 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1172 static RTPSourceCallbacks callbacks = {
1173 (RTPSourcePushRTP) source_push_rtp,
1174 (RTPSourceClockRate) source_clock_rate,
1178 check_collision (RTPSession * sess, RTPSource * source,
1179 RTPArrivalStats * arrival, gboolean rtp)
1181 /* If we have no arrival address, we can't do collision checking */
1182 if (!arrival->address)
1185 if (!source->internal) {
1186 GSocketAddress *from;
1188 /* This is not our local source, but lets check if two remote
1191 from = source->rtp_from;
1193 from = source->rtcp_from;
1197 if (__g_socket_address_equal (from, arrival->address)) {
1198 /* Address is the same */
1201 GST_LOG ("we have a third-party collision or loop ssrc:%x",
1202 rtp_source_get_ssrc (source));
1203 if (sess->favor_new) {
1204 if (rtp_source_find_conflicting_address (source,
1205 arrival->address, arrival->current_time)) {
1208 buf1 = __g_socket_address_to_string (arrival->address);
1209 GST_LOG ("Known conflict on %x for %s, dropping packet",
1210 rtp_source_get_ssrc (source), buf1);
1217 /* Current address is not a known conflict, lets assume this is
1218 * a new source. Save old address in possible conflict list
1220 rtp_source_add_conflicting_address (source, from,
1221 arrival->current_time);
1223 buf1 = __g_socket_address_to_string (from);
1224 buf2 = __g_socket_address_to_string (arrival->address);
1226 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1227 " saving old as known conflict",
1228 rtp_source_get_ssrc (source), buf1, buf2);
1231 rtp_source_set_rtp_from (source, arrival->address);
1233 rtp_source_set_rtcp_from (source, arrival->address);
1241 /* Don't need to save old addresses, we ignore new sources */
1246 /* We don't already have a from address for RTP, just set it */
1248 rtp_source_set_rtp_from (source, arrival->address);
1250 rtp_source_set_rtcp_from (source, arrival->address);
1254 /* FIXME: Log 3rd party collision somehow
1255 * Maybe should be done in upper layer, only the SDES can tell us
1256 * if its a collision or a loop
1259 /* This is sending with our ssrc, is it an address we already know */
1261 if (rtp_source_find_conflicting_address (source, arrival->address,
1262 arrival->current_time)) {
1263 /* Its a known conflict, its probably a loop, not a collision
1264 * lets just drop the incoming packet
1266 GST_DEBUG ("Our packets are being looped back to us, dropping");
1268 /* Its a new collision, lets change our SSRC */
1270 rtp_source_add_conflicting_address (source, arrival->address,
1271 arrival->current_time);
1273 GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1274 on_ssrc_collision (sess, source);
1276 sess->change_ssrc = TRUE;
1278 rtp_source_mark_bye (source, "SSRC Collision");
1279 rtp_session_schedule_bye_locked (sess, arrival->current_time);
1287 find_source (RTPSession * sess, guint32 ssrc)
1289 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1290 GINT_TO_POINTER (ssrc));
1294 add_source (RTPSession * sess, RTPSource * src)
1296 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1297 GINT_TO_POINTER (src->ssrc), src);
1298 /* we have one more source now */
1299 sess->total_sources++;
1300 if (RTP_SOURCE_IS_ACTIVE (src))
1301 sess->stats.active_sources++;
1303 sess->stats.internal_sources++;
1306 /* must be called with the session lock, the returned source needs to be
1307 * unreffed after usage. */
1309 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1310 RTPArrivalStats * arrival, gboolean rtp)
1314 source = find_source (sess, ssrc);
1315 if (source == NULL) {
1316 /* make new Source in probation and insert */
1317 source = rtp_source_new (ssrc);
1319 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1321 /* for RTP packets we need to set the source in probation. Receiving RTCP
1322 * packets of an SSRC, on the other hand, is a strong indication that we
1323 * are dealing with a valid source. */
1325 g_object_set (source, "probation", sess->probation, NULL);
1327 g_object_set (source, "probation", 0, NULL);
1329 /* store from address, if any */
1330 if (arrival->address) {
1332 rtp_source_set_rtp_from (source, arrival->address);
1334 rtp_source_set_rtcp_from (source, arrival->address);
1337 /* configure a callback on the source */
1338 rtp_source_set_callbacks (source, &callbacks, sess);
1340 add_source (sess, source);
1344 /* check for collision, this updates the address when not previously set */
1345 if (check_collision (sess, source, arrival, rtp)) {
1348 /* Receiving RTCP packets of an SSRC is a strong indication that we
1349 * are dealing with a valid source. */
1351 g_object_set (source, "probation", 0, NULL);
1353 /* update last activity */
1354 source->last_activity = arrival->current_time;
1356 source->last_rtp_activity = arrival->current_time;
1357 g_object_ref (source);
1362 /* must be called with the session lock, the returned source needs to be
1363 * unreffed after usage. */
1365 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
1369 source = find_source (sess, ssrc);
1370 if (source == NULL) {
1371 /* make new internal Source and insert */
1372 source = rtp_source_new (ssrc);
1374 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1376 source->validated = TRUE;
1377 source->internal = TRUE;
1378 rtp_source_set_callbacks (source, &callbacks, sess);
1380 add_source (sess, source);
1385 g_object_ref (source);
1391 * rtp_session_get_internal_source:
1392 * @sess: a #RTPSession
1394 * Get the internal #RTPSource of @sess.
1396 * Returns: The internal #RTPSource. g_object_unref() after usage.
1399 rtp_session_get_internal_source (RTPSession * sess)
1403 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1405 result = g_object_ref (sess->source);
1411 * rtp_session_set_internal_ssrc:
1412 * @sess: a #RTPSession
1415 * Set the SSRC of @sess to @ssrc.
1418 rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1420 RTP_SESSION_LOCK (sess);
1421 if (ssrc != sess->source->ssrc) {
1422 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1423 GINT_TO_POINTER (sess->source->ssrc));
1425 GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1426 /* After this call, any receiver of the old SSRC either in RTP or RTCP
1427 * packets will timeout on the old SSRC, we could potentially schedule a
1428 * BYE RTCP for the old SSRC... */
1429 sess->source->ssrc = ssrc;
1430 rtp_source_reset (sess->source);
1432 /* rehash with the new SSRC */
1433 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1434 GINT_TO_POINTER (sess->source->ssrc), sess->source);
1436 RTP_SESSION_UNLOCK (sess);
1438 g_object_notify (G_OBJECT (sess), "internal-ssrc");
1442 * rtp_session_get_internal_ssrc:
1443 * @sess: a #RTPSession
1445 * Get the internal SSRC of @sess.
1447 * Returns: The SSRC of the session.
1450 rtp_session_get_internal_ssrc (RTPSession * sess)
1454 RTP_SESSION_LOCK (sess);
1455 ssrc = sess->source->ssrc;
1456 RTP_SESSION_UNLOCK (sess);
1462 * rtp_session_add_source:
1463 * @sess: a #RTPSession
1464 * @src: #RTPSource to add
1466 * Add @src to @session.
1468 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1469 * existed in the session.
1472 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1474 gboolean result = FALSE;
1477 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1478 g_return_val_if_fail (src != NULL, FALSE);
1480 RTP_SESSION_LOCK (sess);
1481 find = find_source (sess, src->ssrc);
1483 add_source (sess, src);
1486 RTP_SESSION_UNLOCK (sess);
1492 * rtp_session_get_num_sources:
1493 * @sess: an #RTPSession
1495 * Get the number of sources in @sess.
1497 * Returns: The number of sources in @sess.
1500 rtp_session_get_num_sources (RTPSession * sess)
1504 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1506 RTP_SESSION_LOCK (sess);
1507 result = sess->total_sources;
1508 RTP_SESSION_UNLOCK (sess);
1514 * rtp_session_get_num_active_sources:
1515 * @sess: an #RTPSession
1517 * Get the number of active sources in @sess. A source is considered active when
1518 * it has been validated and has not yet received a BYE RTCP message.
1520 * Returns: The number of active sources in @sess.
1523 rtp_session_get_num_active_sources (RTPSession * sess)
1527 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1529 RTP_SESSION_LOCK (sess);
1530 result = sess->stats.active_sources;
1531 RTP_SESSION_UNLOCK (sess);
1537 * rtp_session_get_source_by_ssrc:
1538 * @sess: an #RTPSession
1541 * Find the source with @ssrc in @sess.
1543 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1544 * g_object_unref() after usage.
1547 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1551 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1553 RTP_SESSION_LOCK (sess);
1554 result = find_source (sess, ssrc);
1556 g_object_ref (result);
1557 RTP_SESSION_UNLOCK (sess);
1562 /* should be called with the SESSION lock */
1564 rtp_session_create_new_ssrc (RTPSession * sess)
1569 ssrc = g_random_int ();
1571 /* see if it exists in the session, we're done if it doesn't */
1572 if (find_source (sess, 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 add_source (sess, source);
1601 RTP_SESSION_UNLOCK (sess);
1606 /* update the RTPArrivalStats structure with the current time and other bits
1607 * about the current buffer we are handling.
1608 * This function is typically called when a validated packet is received.
1609 * This function should be called with the SESSION_LOCK
1612 update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1613 gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1614 GstClockTime running_time, guint64 ntpnstime)
1616 GstNetAddressMeta *meta;
1617 GstRTPBuffer rtpb = { NULL };
1619 /* get time of arrival */
1620 arrival->current_time = current_time;
1621 arrival->running_time = running_time;
1622 arrival->ntpnstime = ntpnstime;
1624 /* get packet size including header overhead */
1625 arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len;
1628 gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb);
1629 arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb);
1630 gst_rtp_buffer_unmap (&rtpb);
1632 arrival->payload_len = 0;
1635 /* for netbuffer we can store the IP address to check for collisions */
1636 meta = gst_buffer_get_net_address_meta (buffer);
1637 if (arrival->address)
1638 g_object_unref (arrival->address);
1640 arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1642 arrival->address = NULL;
1647 clean_arrival_stats (RTPArrivalStats * arrival)
1649 if (arrival->address)
1650 g_object_unref (arrival->address);
1654 * rtp_session_process_rtp:
1655 * @sess: and #RTPSession
1656 * @buffer: an RTP buffer
1657 * @current_time: the current system time
1658 * @running_time: the running_time of @buffer
1660 * Process an RTP buffer in the session manager. This function takes ownership
1663 * Returns: a #GstFlowReturn.
1666 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1667 GstClockTime current_time, GstClockTime running_time)
1669 GstFlowReturn result;
1673 gboolean prevsender, prevactive;
1674 RTPArrivalStats arrival = { NULL, };
1678 GstRTPBuffer rtp = { NULL };
1680 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1681 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1683 if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
1684 goto invalid_packet;
1686 /* get SSRC to look up in session database */
1687 ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1688 /* copy available csrc for later */
1689 count = gst_rtp_buffer_get_csrc_count (&rtp);
1690 /* make sure to not overflow our array. An RTP buffer can maximally contain
1692 count = MIN (count, 16);
1694 for (i = 0; i < count; i++)
1695 csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1697 gst_rtp_buffer_unmap (&rtp);
1699 RTP_SESSION_LOCK (sess);
1700 /* ignore more RTP packets when we left the session */
1701 if (sess->source->marked_bye)
1704 /* update arrival stats */
1705 update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1708 source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1712 prevsender = RTP_SOURCE_IS_SENDER (source);
1713 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1714 oldrate = source->bitrate;
1716 /* let source process the packet */
1717 result = rtp_source_process_rtp (source, buffer, &arrival);
1719 /* source became active */
1720 if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1721 sess->stats.active_sources++;
1722 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1723 sess->stats.active_sources);
1724 on_ssrc_validated (sess, source);
1726 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1727 sess->stats.sender_sources++;
1728 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1729 sess->stats.sender_sources);
1731 if (oldrate != source->bitrate)
1732 sess->recalc_bandwidth = TRUE;
1735 on_new_ssrc (sess, source);
1737 if (source->validated) {
1740 /* for validated sources, we add the CSRCs as well */
1741 for (i = 0; i < count; i++) {
1743 RTPSource *csrc_src;
1748 csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1753 GST_DEBUG ("created new CSRC: %08x", csrc);
1754 rtp_source_set_as_csrc (csrc_src);
1755 if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1756 sess->stats.active_sources++;
1757 on_new_ssrc (sess, csrc_src);
1759 g_object_unref (csrc_src);
1762 g_object_unref (source);
1764 RTP_SESSION_UNLOCK (sess);
1766 clean_arrival_stats (&arrival);
1773 gst_buffer_unref (buffer);
1774 GST_DEBUG ("invalid RTP packet received");
1779 RTP_SESSION_UNLOCK (sess);
1780 gst_buffer_unref (buffer);
1781 GST_DEBUG ("ignoring RTP packet because we are leaving");
1786 RTP_SESSION_UNLOCK (sess);
1787 gst_buffer_unref (buffer);
1788 clean_arrival_stats (&arrival);
1789 GST_DEBUG ("ignoring packet because its collisioning");
1795 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1796 GstRTCPPacket * packet, RTPArrivalStats * arrival)
1800 count = gst_rtcp_packet_get_rb_count (packet);
1801 for (i = 0; i < count; i++) {
1802 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1803 guint8 fractionlost;
1806 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1807 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1809 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1811 if (ssrc == sess->source->ssrc) {
1812 /* only deal with report blocks for our session, we update the stats of
1813 * the sender of the RTCP message. We could also compare our stats against
1814 * the other sender to see if we are better or worse. */
1815 rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
1816 packetslost, exthighestseq, jitter, lsr, dlsr);
1819 on_ssrc_active (sess, source);
1822 /* A Sender report contains statistics about how the sender is doing. This
1823 * includes timing informataion such as the relation between RTP and NTP
1824 * timestamps and the number of packets/bytes it sent to us.
1826 * In this report is also included a set of report blocks related to how this
1827 * sender is receiving data (in case we (or somebody else) is also sending stuff
1828 * to it). This info includes the packet loss, jitter and seqnum. It also
1829 * contains information to calculate the round trip time (LSR/DLSR).
1832 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1833 RTPArrivalStats * arrival, gboolean * do_sync)
1835 guint32 senderssrc, rtptime, packet_count, octet_count;
1838 gboolean created, prevsender;
1840 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1841 &packet_count, &octet_count);
1843 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1844 senderssrc, GST_TIME_ARGS (arrival->current_time));
1846 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1850 /* don't try to do lip-sync for sources that sent a BYE */
1851 if (RTP_SOURCE_IS_MARKED_BYE (source))
1856 prevsender = RTP_SOURCE_IS_SENDER (source);
1858 /* first update the source */
1859 rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
1860 packet_count, octet_count);
1862 if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1863 sess->stats.sender_sources++;
1864 GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1865 sess->stats.sender_sources);
1869 on_new_ssrc (sess, source);
1871 rtp_session_process_rb (sess, source, packet, arrival);
1872 g_object_unref (source);
1875 /* A receiver report contains statistics about how a receiver is doing. It
1876 * includes stuff like packet loss, jitter and the seqnum it received last. It
1877 * also contains info to calculate the round trip time.
1879 * We are only interested in how the sender of this report is doing wrt to us.
1882 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1883 RTPArrivalStats * arrival)
1889 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1891 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1893 source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1898 on_new_ssrc (sess, source);
1900 rtp_session_process_rb (sess, source, packet, arrival);
1901 g_object_unref (source);
1904 /* Get SDES items and store them in the SSRC */
1906 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1907 RTPArrivalStats * arrival)
1910 gboolean more_items, more_entries;
1912 items = gst_rtcp_packet_sdes_get_item_count (packet);
1913 GST_DEBUG ("got SDES packet with %d items", items);
1915 more_items = gst_rtcp_packet_sdes_first_item (packet);
1917 while (more_items) {
1919 gboolean changed, created, validated;
1923 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1925 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1929 /* find src, no probation when dealing with RTCP */
1930 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1934 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
1936 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1938 while (more_entries) {
1939 GstRTCPSDESType type;
1945 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1947 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1950 if (type == GST_RTCP_SDES_PRIV) {
1951 name = g_strndup ((const gchar *) &data[1], data[0]);
1953 data += data[0] + 1;
1955 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
1958 value = g_strndup ((const gchar *) data, len);
1960 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
1965 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1969 /* takes ownership of sdes */
1970 changed = rtp_source_set_sdes_struct (source, sdes);
1972 validated = !RTP_SOURCE_IS_ACTIVE (source);
1973 source->validated = TRUE;
1976 on_new_ssrc (sess, source);
1978 /* source became active */
1980 sess->stats.active_sources++;
1981 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1982 sess->stats.active_sources);
1983 on_ssrc_validated (sess, source);
1987 on_ssrc_sdes (sess, source);
1989 g_object_unref (source);
1991 more_items = gst_rtcp_packet_sdes_next_item (packet);
1996 /* BYE is sent when a client leaves the session
1999 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2000 RTPArrivalStats * arrival)
2004 gboolean reconsider = FALSE;
2006 reason = gst_rtcp_packet_bye_get_reason (packet);
2007 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2009 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2010 for (i = 0; i < count; i++) {
2013 gboolean created, prevactive, prevsender;
2014 guint pmembers, members;
2016 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2017 GST_DEBUG ("SSRC: %08x", ssrc);
2019 /* find src and mark bye, no probation when dealing with RTCP */
2020 source = obtain_source (sess, ssrc, &created, arrival, FALSE);
2024 if (source->internal) {
2025 /* our own source, something weird with this packet */
2026 g_object_unref (source);
2030 /* store time for when we need to time out this source */
2031 source->bye_time = arrival->current_time;
2033 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2034 prevsender = RTP_SOURCE_IS_SENDER (source);
2036 /* mark the source BYE */
2037 rtp_source_mark_bye (source, reason);
2039 pmembers = sess->stats.active_sources;
2041 if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
2042 sess->stats.active_sources--;
2043 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2044 sess->stats.active_sources);
2046 if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
2047 sess->stats.sender_sources--;
2048 if (source->internal)
2049 sess->stats.internal_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->scheduled_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 (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2060 arrival->current_time < sess->next_rtcp_check_time) {
2061 GstClockTime time_remaining;
2063 time_remaining = sess->next_rtcp_check_time - arrival->current_time;
2064 sess->next_rtcp_check_time =
2065 gst_util_uint64_scale (time_remaining, members, pmembers);
2067 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2068 GST_TIME_ARGS (sess->next_rtcp_check_time));
2070 sess->next_rtcp_check_time += arrival->current_time;
2072 /* mark pending reconsider. We only want to signal the reconsideration
2073 * once after we handled all the source in the bye packet */
2079 on_new_ssrc (sess, source);
2081 on_bye_ssrc (sess, source);
2083 g_object_unref (source);
2086 RTP_SESSION_UNLOCK (sess);
2087 /* notify app of reconsideration */
2088 if (sess->callbacks.reconsider)
2089 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2090 RTP_SESSION_LOCK (sess);
2096 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2097 RTPArrivalStats * arrival)
2099 GST_DEBUG ("received APP");
2103 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2104 gboolean fir, GstClockTime current_time)
2106 guint32 round_trip = 0;
2108 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2110 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2111 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2114 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
2115 current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2116 GST_DEBUG ("Ignoring %s request because one was send without one "
2117 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2118 fir ? "FIR" : "PLI",
2119 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2120 GST_TIME_ARGS (round_trip_in_ns));;
2125 sess->last_keyframe_request = current_time;
2127 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2128 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2129 sess->callbacks.request_key_unit);
2131 RTP_SESSION_UNLOCK (sess);
2132 sess->callbacks.request_key_unit (sess, fir,
2133 sess->request_key_unit_user_data);
2134 RTP_SESSION_LOCK (sess);
2140 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2141 guint32 media_ssrc, GstClockTime current_time)
2145 if (!sess->callbacks.request_key_unit)
2148 src = find_source (sess, 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 = find_source (sess, sender_ssrc);
2172 /* Hack because Google fails to set the sender_ssrc correctly */
2173 if (!src && sender_ssrc == 1) {
2174 GHashTableIter iter;
2176 if (sess->stats.sender_sources >
2177 RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1)
2180 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2182 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2183 if (src != sess->source && rtp_source_is_sender (src))
2192 for (position = 0; position < fci_length; position += 8) {
2193 guint8 *data = fci_data + position;
2196 ssrc = GST_READ_UINT32_BE (data);
2198 own = find_source (sess, ssrc);
2199 if (own->internal) {
2207 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2211 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2212 RTPArrivalStats * arrival, GstClockTime current_time)
2214 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2215 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2216 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2217 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2218 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2219 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2222 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2223 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2225 if (g_signal_has_handler_pending (sess,
2226 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2227 GstBuffer *fci_buffer = NULL;
2229 if (fci_length > 0) {
2230 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2231 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2233 GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time;
2236 RTP_SESSION_UNLOCK (sess);
2237 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2238 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2239 RTP_SESSION_LOCK (sess);
2242 gst_buffer_unref (fci_buffer);
2245 src = find_source (sess, media_ssrc);
2249 if (sess->rtcp_feedback_retention_window) {
2250 rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
2253 if (src->internal ||
2254 /* PSFB FIR puts the media ssrc inside the FCI */
2255 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2257 case GST_RTCP_TYPE_PSFB:
2259 case GST_RTCP_PSFB_TYPE_PLI:
2260 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2263 case GST_RTCP_PSFB_TYPE_FIR:
2264 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2271 case GST_RTCP_TYPE_RTPFB:
2279 * rtp_session_process_rtcp:
2280 * @sess: and #RTPSession
2281 * @buffer: an RTCP buffer
2282 * @current_time: the current system time
2283 * @ntpnstime: the current NTP time in nanoseconds
2285 * Process an RTCP buffer in the session manager. This function takes ownership
2288 * Returns: a #GstFlowReturn.
2291 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2292 GstClockTime current_time, guint64 ntpnstime)
2294 GstRTCPPacket packet;
2295 gboolean more, is_bye = FALSE, do_sync = FALSE;
2296 RTPArrivalStats arrival = { NULL, };
2297 GstFlowReturn result = GST_FLOW_OK;
2298 GstRTCPBuffer rtcp = { NULL, };
2300 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2301 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2303 if (!gst_rtcp_buffer_validate (buffer))
2304 goto invalid_packet;
2306 GST_DEBUG ("received RTCP packet");
2308 RTP_SESSION_LOCK (sess);
2309 /* update arrival stats */
2310 update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
2313 if (sess->source->sent_bye)
2316 /* start processing the compound packet */
2317 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2318 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2322 type = gst_rtcp_packet_get_type (&packet);
2324 /* when we are leaving the session, we should ignore all non-BYE messages */
2325 if (sess->scheduled_bye && type != GST_RTCP_TYPE_BYE) {
2326 GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
2331 case GST_RTCP_TYPE_SR:
2332 rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
2334 case GST_RTCP_TYPE_RR:
2335 rtp_session_process_rr (sess, &packet, &arrival);
2337 case GST_RTCP_TYPE_SDES:
2338 rtp_session_process_sdes (sess, &packet, &arrival);
2340 case GST_RTCP_TYPE_BYE:
2342 /* don't try to attempt lip-sync anymore for streams with a BYE */
2344 rtp_session_process_bye (sess, &packet, &arrival);
2346 case GST_RTCP_TYPE_APP:
2347 rtp_session_process_app (sess, &packet, &arrival);
2349 case GST_RTCP_TYPE_RTPFB:
2350 case GST_RTCP_TYPE_PSFB:
2351 rtp_session_process_feedback (sess, &packet, &arrival, current_time);
2354 GST_WARNING ("got unknown RTCP packet");
2358 more = gst_rtcp_packet_move_to_next (&packet);
2361 gst_rtcp_buffer_unmap (&rtcp);
2363 /* if we are scheduling a BYE, we only want to count bye packets, else we
2364 * count everything */
2365 if (sess->scheduled_bye) {
2367 sess->stats.bye_members++;
2368 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2371 /* keep track of average packet size */
2372 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
2374 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2375 sess->stats.avg_rtcp_packet_size, arrival.bytes);
2376 RTP_SESSION_UNLOCK (sess);
2378 clean_arrival_stats (&arrival);
2380 /* notify caller of sr packets in the callback */
2381 if (do_sync && sess->callbacks.sync_rtcp) {
2382 /* make writable, we might want to change the buffer */
2383 buffer = gst_buffer_make_writable (buffer);
2385 result = sess->callbacks.sync_rtcp (sess, buffer,
2386 sess->sync_rtcp_user_data);
2388 gst_buffer_unref (buffer);
2395 GST_DEBUG ("invalid RTCP packet received");
2396 gst_buffer_unref (buffer);
2401 RTP_SESSION_UNLOCK (sess);
2402 gst_buffer_unref (buffer);
2403 clean_arrival_stats (&arrival);
2404 GST_DEBUG ("ignoring RTCP packet because we left");
2410 * rtp_session_update_send_caps:
2411 * @sess: an #RTPSession
2414 * Update the caps of the sender in the rtp session.
2417 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2419 g_return_if_fail (RTP_IS_SESSION (sess));
2420 g_return_if_fail (GST_IS_CAPS (caps));
2422 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2424 RTP_SESSION_LOCK (sess);
2425 rtp_source_update_caps (sess->source, caps);
2426 RTP_SESSION_UNLOCK (sess);
2430 * rtp_session_send_rtp:
2431 * @sess: an #RTPSession
2432 * @data: pointer to either an RTP buffer or a list of RTP buffers
2433 * @is_list: TRUE when @data is a buffer list
2434 * @current_time: the current system time
2435 * @running_time: the running time of @data
2437 * Send the RTP buffer in the session manager. This function takes ownership of
2440 * Returns: a #GstFlowReturn.
2443 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2444 GstClockTime current_time, GstClockTime running_time)
2446 GstFlowReturn result;
2448 gboolean prevsender;
2451 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2452 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2454 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2456 RTP_SESSION_LOCK (sess);
2457 source = sess->source;
2459 /* update last activity */
2460 source->last_rtp_activity = current_time;
2462 prevsender = RTP_SOURCE_IS_SENDER (source);
2463 oldrate = source->bitrate;
2465 /* we use our own source to send */
2466 result = rtp_source_send_rtp (source, data, is_list, running_time);
2468 if (RTP_SOURCE_IS_SENDER (source) && !prevsender) {
2469 sess->stats.sender_sources++;
2470 sess->stats.internal_sender_sources++;
2472 if (oldrate != source->bitrate)
2473 sess->recalc_bandwidth = TRUE;
2474 RTP_SESSION_UNLOCK (sess);
2480 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2482 *bandwidth += source->bitrate;
2485 /* must be called with session lock */
2487 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2490 GstClockTime result;
2492 /* recalculate bandwidth when it changed */
2493 if (sess->recalc_bandwidth) {
2496 if (sess->bandwidth > 0)
2497 bandwidth = sess->bandwidth;
2499 /* If it is <= 0, then try to estimate the actual bandwidth */
2502 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2503 (GHFunc) add_bitrates, &bandwidth);
2506 if (bandwidth < 8000)
2507 bandwidth = RTP_STATS_BANDWIDTH;
2509 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2510 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2512 sess->recalc_bandwidth = FALSE;
2515 if (sess->scheduled_bye) {
2516 result = rtp_stats_calculate_bye_interval (&sess->stats);
2518 result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2519 sess->stats.internal_sender_sources > 0, first);
2522 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2523 GST_TIME_ARGS (result), first);
2525 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2526 result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2528 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2534 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2536 if (source->internal)
2537 rtp_source_mark_bye (source, reason);
2541 * rtp_session_mark_all_bye:
2542 * @sess: an #RTPSession
2545 * Mark all internal sources of the session as BYE with @reason.
2548 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2550 g_return_if_fail (RTP_IS_SESSION (sess));
2552 RTP_SESSION_LOCK (sess);
2553 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2554 (GHFunc) source_mark_bye, (gpointer) reason);
2555 RTP_SESSION_UNLOCK (sess);
2558 /* Stop the current @sess and schedule a BYE message for the other members.
2559 * One must have the session lock to call this function
2561 static GstFlowReturn
2562 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2564 GstFlowReturn result = GST_FLOW_OK;
2565 GstClockTime interval;
2567 /* nothing to do it we already scheduled bye */
2568 if (sess->scheduled_bye)
2571 /* we schedule BYE now */
2572 sess->scheduled_bye = TRUE;
2573 /* at least one member wants to send a BYE */
2574 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
2575 sess->stats.bye_members = 1;
2576 sess->first_rtcp = TRUE;
2577 sess->allow_early = TRUE;
2579 /* reschedule transmission */
2580 sess->last_rtcp_send_time = current_time;
2581 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2583 if (interval != GST_CLOCK_TIME_NONE)
2584 sess->next_rtcp_check_time = current_time + interval;
2586 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2588 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2589 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2591 RTP_SESSION_UNLOCK (sess);
2592 /* notify app of reconsideration */
2593 if (sess->callbacks.reconsider)
2594 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2595 RTP_SESSION_LOCK (sess);
2602 * rtp_session_schedule_bye:
2603 * @sess: an #RTPSession
2604 * @current_time: the current system time
2606 * Schedule a BYE message for all sources marked as BYE in @sess.
2608 * Returns: a #GstFlowReturn.
2611 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2613 GstFlowReturn result = GST_FLOW_OK;
2615 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2617 RTP_SESSION_LOCK (sess);
2618 result = rtp_session_schedule_bye_locked (sess, current_time);
2619 RTP_SESSION_UNLOCK (sess);
2625 * rtp_session_next_timeout:
2626 * @sess: an #RTPSession
2627 * @current_time: the current system time
2629 * Get the next time we should perform session maintenance tasks.
2631 * Returns: a time when rtp_session_on_timeout() should be called with the
2632 * current system time.
2635 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2637 GstClockTime result, interval = 0;
2639 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2641 RTP_SESSION_LOCK (sess);
2643 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2644 result = sess->next_early_rtcp_time;
2648 result = sess->next_rtcp_check_time;
2650 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2651 ", next time: %" GST_TIME_FORMAT,
2652 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2654 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2655 GST_DEBUG ("take current time as base");
2656 /* our previous check time expired, start counting from the current time
2658 result = current_time;
2661 if (sess->scheduled_bye) {
2662 if (sess->source->sent_bye) {
2663 GST_DEBUG ("we sent BYE already");
2664 interval = GST_CLOCK_TIME_NONE;
2665 } else if (sess->stats.active_sources >= 50) {
2666 GST_DEBUG ("reconsider BYE, more than 50 sources");
2667 /* reconsider BYE if members >= 50 */
2668 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2671 if (sess->first_rtcp) {
2672 GST_DEBUG ("first RTCP packet");
2673 /* we are called for the first time */
2674 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2675 } else if (sess->next_rtcp_check_time < current_time) {
2676 GST_DEBUG ("old check time expired, getting new timeout");
2677 /* get a new timeout when we need to */
2678 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2682 if (interval != GST_CLOCK_TIME_NONE)
2685 result = GST_CLOCK_TIME_NONE;
2687 sess->next_rtcp_check_time = result;
2691 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2692 ", next time: %" GST_TIME_FORMAT,
2693 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2694 RTP_SESSION_UNLOCK (sess);
2701 GstRTCPBuffer rtcpbuf;
2705 GstClockTime current_time;
2707 GstClockTime running_time;
2708 GstClockTime interval;
2709 GstRTCPPacket packet;
2713 gboolean may_suppress;
2718 session_start_rtcp (RTPSession * sess, ReportData * data)
2720 GstRTCPPacket *packet = &data->packet;
2721 RTPSource *own = data->source;
2722 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2724 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2725 data->is_bye = FALSE;
2726 data->has_sdes = FALSE;
2728 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
2730 if (RTP_SOURCE_IS_SENDER (own)) {
2733 guint32 packet_count, octet_count;
2735 /* we are a sender, create SR */
2736 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2737 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
2739 /* get latest stats */
2740 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
2741 &ntptime, &rtptime, &packet_count, &octet_count);
2743 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2744 packet_count, octet_count);
2746 /* fill in sender report info */
2747 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2748 ntptime, rtptime, packet_count, octet_count);
2750 /* we are only receiver, create RR */
2751 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2752 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
2753 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2757 /* construct a Sender or Receiver Report */
2759 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2761 GstRTCPPacket *packet = &data->packet;
2763 if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2764 /* only report about other sender sources */
2765 if (source != data->source && RTP_SOURCE_IS_SENDER (source)) {
2766 guint8 fractionlost;
2768 guint32 exthighestseq, jitter;
2772 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2773 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2775 /* store last generated RR packet */
2776 source->last_rr.is_valid = TRUE;
2777 source->last_rr.fractionlost = fractionlost;
2778 source->last_rr.packetslost = packetslost;
2779 source->last_rr.exthighestseq = exthighestseq;
2780 source->last_rr.jitter = jitter;
2781 source->last_rr.lsr = lsr;
2782 source->last_rr.dlsr = dlsr;
2784 /* packet is not yet filled, add report block for this source. */
2785 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2786 exthighestseq, jitter, lsr, dlsr);
2791 /* perform cleanup of sources that timed out */
2793 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2795 gboolean remove = FALSE;
2796 gboolean byetimeout = FALSE;
2797 gboolean sendertimeout = FALSE;
2798 gboolean is_sender, is_active;
2799 RTPSession *sess = data->sess;
2800 GstClockTime interval, binterval;
2803 /* check for outdated collisions */
2804 if (source->internal) {
2805 GST_DEBUG ("Timing out collisions");
2806 rtp_source_timeout (source, data->current_time,
2807 /* "a relatively long time" -- RFC 3550 section 8.2 */
2808 RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
2809 data->running_time - sess->rtcp_feedback_retention_window);
2812 /* nothing else to do when without RTCP */
2813 if (data->interval == GST_CLOCK_TIME_NONE)
2816 is_sender = RTP_SOURCE_IS_SENDER (source);
2817 is_active = RTP_SOURCE_IS_ACTIVE (source);
2819 /* our own rtcp interval may have been forced low by secondary configuration,
2820 * while sender side may still operate with higher interval,
2821 * so do not just take our interval to decide on timing out sender,
2822 * but take (if data->interval <= 5 * GST_SECOND):
2823 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
2824 * where sender_interval is difference between last 2 received RTCP reports
2826 if (data->interval >= 5 * GST_SECOND || source->internal) {
2827 binterval = data->interval;
2829 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
2830 GST_TIME_ARGS (source->stats.prev_rtcptime),
2831 GST_TIME_ARGS (source->stats.last_rtcptime));
2832 /* if not received enough yet, fallback to larger default */
2833 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
2834 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
2836 binterval = 5 * GST_SECOND;
2837 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
2839 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
2840 GST_TIME_ARGS (binterval));
2842 /* check for our own source, we don't want to delete our own source. */
2843 if (!source->internal) {
2844 if (source->marked_bye) {
2845 /* if we received a BYE from the source, remove the source after some
2847 if (data->current_time > source->bye_time &&
2848 data->current_time - source->bye_time > sess->stats.bye_timeout) {
2849 GST_DEBUG ("removing BYE source %08x", source->ssrc);
2854 /* sources that were inactive for more than 5 times the deterministic reporting
2855 * interval get timed out. the min timeout is 5 seconds. */
2856 /* mind old time that might pre-date last time going to PLAYING */
2857 btime = MAX (source->last_activity, sess->start_time);
2858 if (data->current_time > btime) {
2859 interval = MAX (binterval * 5, 5 * GST_SECOND);
2860 if (data->current_time - btime > interval) {
2861 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2862 source->ssrc, GST_TIME_ARGS (btime));
2868 /* senders that did not send for a long time become a receiver, this also
2869 * holds for our own sources. */
2871 /* mind old time that might pre-date last time going to PLAYING */
2872 btime = MAX (source->last_rtp_activity, sess->start_time);
2873 if (data->current_time > btime) {
2874 interval = MAX (binterval * 2, 5 * GST_SECOND);
2875 if (data->current_time - btime > interval) {
2876 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2877 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
2878 source->is_sender = FALSE;
2879 sess->stats.sender_sources--;
2880 if (source->internal)
2881 sess->stats.internal_sender_sources--;
2882 sendertimeout = TRUE;
2888 sess->total_sources--;
2890 sess->stats.sender_sources--;
2891 if (source->internal)
2892 sess->stats.internal_sender_sources--;
2895 sess->stats.active_sources--;
2897 if (source->internal)
2898 sess->stats.internal_sources--;
2901 on_bye_timeout (sess, source);
2903 on_timeout (sess, source);
2906 on_sender_timeout (sess, source);
2909 source->closing = remove;
2913 session_sdes (RTPSession * sess, ReportData * data)
2915 GstRTCPPacket *packet = &data->packet;
2916 const GstStructure *sdes;
2918 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2920 /* add SDES packet */
2921 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
2923 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
2925 sdes = rtp_source_get_sdes_struct (data->source);
2927 /* add all fields in the structure, the order is not important. */
2928 n_fields = gst_structure_n_fields (sdes);
2929 for (i = 0; i < n_fields; ++i) {
2932 GstRTCPSDESType type;
2934 field = gst_structure_nth_field_name (sdes, i);
2937 value = gst_structure_get_string (sdes, field);
2940 type = gst_rtcp_sdes_name_to_type (field);
2942 /* Early packets are minimal and only include the CNAME */
2943 if (data->is_early && type != GST_RTCP_SDES_CNAME)
2946 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
2947 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
2948 (const guint8 *) value);
2949 } else if (type == GST_RTCP_SDES_PRIV) {
2955 /* don't accept entries that are too big */
2956 prefix_len = strlen (field);
2957 if (prefix_len > 255)
2959 value_len = strlen (value);
2960 if (value_len > 255)
2962 data_len = 1 + prefix_len + value_len;
2966 data[0] = prefix_len;
2967 memcpy (&data[1], field, prefix_len);
2968 memcpy (&data[1 + prefix_len], value, value_len);
2970 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
2974 data->has_sdes = TRUE;
2977 /* schedule a BYE packet */
2979 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
2981 GstRTCPPacket *packet = &data->packet;
2982 GstRTCPBuffer *rtcp = &data->rtcpbuf;
2985 session_sdes (sess, data);
2986 /* add a BYE packet */
2987 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
2988 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
2989 if (source->bye_reason)
2990 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
2992 /* we have a BYE packet now */
2993 data->is_bye = TRUE;
2994 source->sent_bye = TRUE;
2998 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3000 GstClockTime new_send_time, elapsed;
3002 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3003 data->is_early = TRUE;
3005 data->is_early = FALSE;
3007 if (data->is_early && sess->next_early_rtcp_time < current_time)
3010 /* no need to check yet */
3011 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3012 sess->next_rtcp_check_time > current_time) {
3013 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3014 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3015 GST_TIME_ARGS (current_time));
3019 /* get elapsed time since we last reported */
3020 elapsed = current_time - sess->last_rtcp_send_time;
3022 new_send_time = data->interval;
3023 /* perform forward reconsideration */
3024 if (new_send_time != GST_CLOCK_TIME_NONE) {
3025 new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
3027 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3028 GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time),
3029 GST_TIME_ARGS (elapsed));
3031 new_send_time += sess->last_rtcp_send_time;
3034 /* check if reconsideration */
3035 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3036 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3037 GST_TIME_ARGS (new_send_time));
3038 /* store new check time */
3039 sess->next_rtcp_check_time = new_send_time;
3045 new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
3047 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3048 GST_TIME_ARGS (new_send_time));
3050 sess->next_rtcp_check_time = new_send_time;
3051 if (new_send_time != GST_CLOCK_TIME_NONE) {
3052 sess->next_rtcp_check_time += current_time;
3054 /* Apply the rules from RFC 4585 section 3.5.3 */
3055 if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
3056 GstClockTimeDiff T_rr_current_interval =
3057 g_random_double_range (0.5, 1.5) * sess->stats.min_interval;
3059 /* This will caused the RTCP to be suppressed if no FB packets are added */
3060 if (sess->last_rtcp_send_time + T_rr_current_interval >
3061 sess->next_rtcp_check_time) {
3062 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3063 " last: %" GST_TIME_FORMAT
3064 " + T_rr_current_interval: %" GST_TIME_FORMAT
3065 " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
3066 GST_TIME_ARGS (sess->stats.min_interval),
3067 GST_TIME_ARGS (sess->last_rtcp_send_time),
3068 GST_TIME_ARGS (T_rr_current_interval),
3069 GST_TIME_ARGS (sess->next_rtcp_check_time));
3070 data->may_suppress = TRUE;
3079 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3081 g_hash_table_insert (hash_table, key, g_object_ref (source));
3085 remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
3087 return source->closing;
3091 generate_rtcp (RTPSource * source, ReportData * data)
3093 RTPSession *sess = data->sess;
3095 /* only generate RTCP for active internal sources */
3096 if (!source->internal || source->sent_bye)
3099 data->source = source;
3102 session_start_rtcp (sess, data);
3104 if (source->marked_bye) {
3106 make_source_bye (sess, source, data);
3107 } else if (!data->is_early) {
3108 /* loop over all known sources and add report blocks. If we are ealy, we
3109 * just make a minimal RTCP packet and skip this step */
3110 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3111 (GHFunc) session_report_blocks, data);
3113 if (!data->has_sdes)
3114 session_sdes (sess, data);
3116 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3118 if (sess->change_ssrc) {
3119 GST_DEBUG ("need to change our SSRC (%08x)", source->ssrc);
3120 g_hash_table_steal (sess->ssrcs[sess->mask_idx],
3121 GINT_TO_POINTER (source->ssrc));
3123 source->ssrc = rtp_session_create_new_ssrc (sess);
3124 rtp_source_reset (source);
3126 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
3127 GINT_TO_POINTER (source->ssrc), source);
3129 sess->change_ssrc = FALSE;
3130 data->notify = TRUE;
3131 GST_DEBUG ("changed our SSRC to %08x", source->ssrc);
3136 * rtp_session_on_timeout:
3137 * @sess: an #RTPSession
3138 * @current_time: the current system time
3139 * @ntpnstime: the current NTP time in nanoseconds
3140 * @running_time: the current running_time of the pipeline
3142 * Perform maintenance actions after the timeout obtained with
3143 * rtp_session_next_timeout() expired.
3145 * This function will perform timeouts of receivers and senders, send a BYE
3146 * packet or generate RTCP packets with current session stats.
3148 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3149 * times, for each packet that should be processed.
3151 * Returns: a #GstFlowReturn.
3154 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3155 guint64 ntpnstime, GstClockTime running_time)
3157 GstFlowReturn result = GST_FLOW_OK;
3158 ReportData data = { GST_RTCP_BUFFER_INIT };
3160 GHashTable *table_copy;
3162 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3164 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3165 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3166 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3169 data.current_time = current_time;
3170 data.ntpnstime = ntpnstime;
3171 data.running_time = running_time;
3172 data.may_suppress = FALSE;
3173 data.notify = FALSE;
3177 RTP_SESSION_LOCK (sess);
3178 /* get a new interval, we need this for various cleanups etc */
3179 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3181 /* Make a local copy of the hashtable. We need to do this because the
3182 * cleanup stage below releases the session lock. */
3183 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3184 (GDestroyNotify) g_object_unref);
3185 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3186 (GHFunc) clone_ssrcs_hashtable, table_copy);
3188 /* Clean up the session, mark the source for removing, this might release the
3190 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3191 g_hash_table_destroy (table_copy);
3193 /* Now remove the marked sources */
3194 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3195 (GHRFunc) remove_closing_sources, NULL);
3197 /* see if we need to generate SR or RR packets */
3198 if (!is_rtcp_time (sess, current_time, &data))
3201 generate_rtcp (own, &data);
3203 /* we keep track of the last report time in order to timeout inactive
3204 * receivers or senders */
3205 if (!data.is_early && !data.may_suppress)
3206 sess->last_rtcp_send_time = data.current_time;
3207 sess->first_rtcp = FALSE;
3208 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3211 RTP_SESSION_UNLOCK (sess);
3214 g_object_notify (G_OBJECT (sess), "internal-ssrc");
3216 /* push out the RTCP packet */
3218 gboolean do_not_suppress;
3219 GstBuffer *buffer = data.rtcp;
3221 /* Give the user a change to add its own packet */
3222 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3223 buffer, data.is_early, &do_not_suppress);
3225 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3228 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3230 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3231 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3232 sess->stats.avg_rtcp_packet_size, packet_size);
3234 sess->callbacks.send_rtcp (sess, own, buffer, data.is_bye,
3235 sess->send_rtcp_user_data);
3237 GST_DEBUG ("freeing packet callback: %p"
3238 " do_not_suppress: %d may_suppress: %d",
3239 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3240 gst_buffer_unref (buffer);
3248 * rtp_session_request_early_rtcp:
3249 * @sess: an #RTPSession
3250 * @current_time: the current system time
3251 * @max_delay: maximum delay
3253 * Request transmission of early RTCP
3256 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3257 GstClockTimeDiff max_delay)
3259 GstClockTime T_dither_max;
3261 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3263 RTP_SESSION_LOCK (sess);
3265 /* Check if already requested */
3266 /* RFC 4585 section 3.5.2 step 2 */
3267 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3270 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time))
3273 /* Ignore the request a scheduled packet will be in time anyway */
3274 if (current_time + max_delay > sess->next_rtcp_check_time)
3277 /* RFC 4585 section 3.5.2 step 2b */
3278 /* If the total sources is <=2, then there is only us and one peer */
3279 if (sess->total_sources <= 2) {
3282 /* Divide by 2 because l = 0.5 */
3283 T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3287 /* RFC 4585 section 3.5.2 step 3 */
3288 if (current_time + T_dither_max > sess->next_rtcp_check_time)
3291 /* RFC 4585 section 3.5.2 step 4
3292 * Don't send if allow_early is FALSE, but not if we are in
3293 * immediate mode, meaning we are part of a group of at most the
3294 * application-specific threshold.
3296 if (sess->total_sources > sess->rtcp_immediate_feedback_threshold &&
3297 sess->allow_early == FALSE)
3301 /* Schedule an early transmission later */
3302 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3305 /* If no dithering, schedule it for NOW */
3306 sess->next_early_rtcp_time = current_time;
3309 RTP_SESSION_UNLOCK (sess);
3311 /* notify app of need to send packet early
3312 * and therefore of timeout change */
3313 if (sess->callbacks.reconsider)
3314 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3320 RTP_SESSION_UNLOCK (sess);
3324 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now,
3325 gboolean fir, gint count)
3327 RTPSource *src = find_source (sess, ssrc);
3333 src->send_pli = FALSE;
3334 src->send_fir = TRUE;
3336 if (count == -1 || count != src->last_fir_count)
3337 src->current_send_fir_seqnum++;
3338 src->last_fir_count = count;
3339 } else if (!src->send_fir) {
3340 src->send_pli = TRUE;
3343 rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND);
3349 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3351 GstRTCPPacket packet;
3352 GstRTCPBuffer rtcp = { NULL, };
3353 gboolean ret = FALSE;
3355 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3357 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3358 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3359 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3363 gst_rtcp_buffer_unmap (&rtcp);
3369 rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
3372 gboolean ret = FALSE;
3373 GHashTableIter iter;
3374 gpointer key, value;
3375 gboolean started_fir = FALSE;
3376 GstRTCPPacket fir_rtcppacket;
3377 GstRTCPPacket packet;
3378 GstRTCPBuffer rtcp = { NULL, };
3381 gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp);
3383 gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
3384 switch (gst_rtcp_packet_get_type (&packet)) {
3385 case GST_RTCP_TYPE_SR:
3386 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc,
3387 NULL, NULL, NULL, NULL);
3389 case GST_RTCP_TYPE_RR:
3390 ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
3396 RTP_SESSION_LOCK (sess);
3397 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3398 while (g_hash_table_iter_next (&iter, &key, &value)) {
3399 guint media_ssrc = GPOINTER_TO_UINT (key);
3400 RTPSource *media_src = value;
3403 if (media_src->send_fir) {
3405 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3408 gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR);
3409 gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket, ssrc);
3410 gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0);
3412 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) {
3413 gst_rtcp_packet_remove (&fir_rtcppacket);
3419 if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket,
3420 !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2))
3424 fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) -
3425 ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4);
3427 GST_WRITE_UINT32_BE (fci_data, media_ssrc);
3429 fci_data[0] = media_src->current_send_fir_seqnum;
3430 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3431 media_src->send_fir = FALSE;
3435 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3436 while (g_hash_table_iter_next (&iter, &key, &value)) {
3437 guint media_ssrc = GPOINTER_TO_UINT (key);
3438 RTPSource *media_src = value;
3439 GstRTCPPacket pli_rtcppacket;
3441 if (media_src->send_pli && !rtp_source_has_retained (media_src,
3442 has_pli_compare_func, NULL)) {
3443 if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB,
3445 /* Break because the packet is full, will put next request in a
3448 gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
3449 gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket, ssrc);
3450 gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc);
3453 media_src->send_pli = FALSE;
3455 RTP_SESSION_UNLOCK (sess);
3458 gst_rtcp_buffer_unmap (&rtcp);
3464 rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
3468 if (!sess->callbacks.send_rtcp)
3471 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3473 rtp_session_request_early_rtcp (sess, now, max_delay);