2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
20 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
21 * with newer GLib versions (>= 2.31.0) */
22 #define GLIB_DISABLE_DEPRECATION_WARNINGS
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
29 #include <gst/glib-compat-private.h>
31 #include "rtpsession.h"
33 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
34 #define GST_CAT_DEFAULT rtp_session_debug
36 /* signals and args */
39 SIGNAL_GET_SOURCE_BY_SSRC,
41 SIGNAL_ON_SSRC_COLLISION,
42 SIGNAL_ON_SSRC_VALIDATED,
43 SIGNAL_ON_SSRC_ACTIVE,
46 SIGNAL_ON_BYE_TIMEOUT,
48 SIGNAL_ON_SENDER_TIMEOUT,
49 SIGNAL_ON_SENDING_RTCP,
50 SIGNAL_ON_FEEDBACK_RTCP,
52 SIGNAL_SEND_RTCP_FULL,
53 SIGNAL_ON_RECEIVING_RTCP,
57 #define DEFAULT_INTERNAL_SOURCE NULL
58 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
59 #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
60 #define DEFAULT_RTCP_RR_BANDWIDTH -1
61 #define DEFAULT_RTCP_RS_BANDWIDTH -1
62 #define DEFAULT_RTCP_MTU 1400
63 #define DEFAULT_SDES NULL
64 #define DEFAULT_NUM_SOURCES 0
65 #define DEFAULT_NUM_ACTIVE_SOURCES 0
66 #define DEFAULT_SOURCES NULL
67 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
68 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
69 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
70 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
79 PROP_RTCP_RR_BANDWIDTH,
80 PROP_RTCP_RS_BANDWIDTH,
84 PROP_NUM_ACTIVE_SOURCES,
87 PROP_RTCP_MIN_INTERVAL,
88 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
89 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
95 /* update average packet size */
96 #define INIT_AVG(avg, val) \
98 #define UPDATE_AVG(avg, val) \
102 (avg) = ((val) + (15 * (avg))) >> 4;
105 /* GObject vmethods */
106 static void rtp_session_finalize (GObject * object);
107 static void rtp_session_set_property (GObject * object, guint prop_id,
108 const GValue * value, GParamSpec * pspec);
109 static void rtp_session_get_property (GObject * object, guint prop_id,
110 GValue * value, GParamSpec * pspec);
112 static gboolean rtp_session_send_rtcp (RTPSession * sess,
113 GstClockTime max_delay);
115 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
117 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
119 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
120 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
121 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
122 static RTPSource *obtain_internal_source (RTPSession * sess,
123 guint32 ssrc, gboolean * created, GstClockTime current_time);
124 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
125 GstClockTime current_time);
126 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
127 gboolean deterministic, gboolean first);
130 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
131 const GValue * handler_return, gpointer data)
133 if (g_value_get_boolean (handler_return))
134 g_value_set_boolean (return_accu, TRUE);
140 rtp_session_class_init (RTPSessionClass * klass)
142 GObjectClass *gobject_class;
144 gobject_class = (GObjectClass *) klass;
146 gobject_class->finalize = rtp_session_finalize;
147 gobject_class->set_property = rtp_session_set_property;
148 gobject_class->get_property = rtp_session_get_property;
151 * RTPSession::get-source-by-ssrc:
152 * @session: the object which received the signal
153 * @ssrc: the SSRC of the RTPSource
155 * Request the #RTPSource object with SSRC @ssrc in @session.
157 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
158 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
159 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
160 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
161 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
164 * RTPSession::on-new-ssrc:
165 * @session: the object which received the signal
166 * @src: the new RTPSource
168 * Notify of a new SSRC that entered @session.
170 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
171 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
172 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
173 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
176 * RTPSession::on-ssrc-collision:
177 * @session: the object which received the signal
178 * @src: the #RTPSource that caused a collision
180 * Notify when we have an SSRC collision
182 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
183 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
184 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
185 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
188 * RTPSession::on-ssrc-validated:
189 * @session: the object which received the signal
190 * @src: the new validated RTPSource
192 * Notify of a new SSRC that became validated.
194 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
195 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
196 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
197 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
200 * RTPSession::on-ssrc-active:
201 * @session: the object which received the signal
202 * @src: the active RTPSource
204 * Notify of a SSRC that is active, i.e., sending RTCP.
206 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
207 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
208 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
209 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
212 * RTPSession::on-ssrc-sdes:
213 * @session: the object which received the signal
214 * @src: the RTPSource
216 * Notify that a new SDES was received for SSRC.
218 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
219 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
220 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
221 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
224 * RTPSession::on-bye-ssrc:
225 * @session: the object which received the signal
226 * @src: the RTPSource that went away
228 * Notify of an SSRC that became inactive because of a BYE packet.
230 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
231 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
232 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
233 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
236 * RTPSession::on-bye-timeout:
237 * @session: the object which received the signal
238 * @src: the RTPSource that timed out
240 * Notify of an SSRC that has timed out because of BYE
242 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
243 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
244 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
245 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
248 * RTPSession::on-timeout:
249 * @session: the object which received the signal
250 * @src: the RTPSource that timed out
252 * Notify of an SSRC that has timed out
254 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
255 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
256 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
257 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
260 * RTPSession::on-sender-timeout:
261 * @session: the object which received the signal
262 * @src: the RTPSource that timed out
264 * Notify of an SSRC that was a sender but timed out and became a receiver.
266 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
267 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
268 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
269 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
273 * RTPSession::on-sending-rtcp
274 * @session: the object which received the signal
275 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
276 * @early: %TRUE if the packet is early, %FALSE if it is regular
278 * This signal is emitted before sending an RTCP packet, it can be used
279 * to add extra RTCP Packets.
281 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
282 * if suppressing it is acceptable
284 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
285 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
286 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
287 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
288 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
291 * RTPSession::on-feedback-rtcp:
292 * @session: the object which received the signal
293 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
294 * %GST_RTCP_TYPE_RTPFB
295 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
296 * @sender_ssrc: The SSRC of the sender
297 * @media_ssrc: The SSRC of the media this refers to
298 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
301 * Notify that a RTCP feedback packet has been received
303 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
304 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
305 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
306 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
307 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
310 * RTPSession::send-rtcp:
311 * @session: the object which received the signal
312 * @max_delay: The maximum delay after which the feedback will not be useful
315 * Requests that the #RTPSession initiate a new RTCP packet as soon as
316 * possible within the requested delay.
318 rtp_session_signals[SIGNAL_SEND_RTCP] =
319 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
320 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
321 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
322 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
325 * RTPSession::send-rtcp-full:
326 * @session: the object which received the signal
327 * @max_delay: The maximum delay after which the feedback will not be useful
330 * Requests that the #RTPSession initiate a new RTCP packet as soon as
331 * possible within the requested delay.
333 * Returns: TRUE if the new RTCP packet could be scheduled within the
334 * requested delay, FALSE otherwise.
338 rtp_session_signals[SIGNAL_SEND_RTCP_FULL] =
339 g_signal_new ("send-rtcp-full", G_TYPE_FROM_CLASS (klass),
340 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
341 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
342 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
345 * RTPSession::on-receiving-rtcp
346 * @session: the object which received the signal
347 * @buffer: the #GstBuffer containing the RTCP packet that was received
349 * This signal is emitted when receiving an RTCP packet before it is handled
350 * by the session. It can be used to extract custom information from RTCP packets.
354 rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP] =
355 g_signal_new ("on-receiving-rtcp", G_TYPE_FROM_CLASS (klass),
356 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_receiving_rtcp),
357 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
358 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
360 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
361 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
362 "The internal SSRC used for the session (deprecated)",
363 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
365 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
366 g_param_spec_object ("internal-source", "Internal Source",
367 "The internal source element of the session (deprecated)",
368 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
370 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
371 g_param_spec_double ("bandwidth", "Bandwidth",
372 "The bandwidth of the session (0 for auto-discover)",
373 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
374 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
376 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
377 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
378 "The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
379 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
380 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
382 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
383 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
384 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
385 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
386 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
388 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
389 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
390 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
391 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
392 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
394 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
395 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
396 "The maximum size of the RTCP packets",
397 16, G_MAXINT16, DEFAULT_RTCP_MTU,
398 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
400 g_object_class_install_property (gobject_class, PROP_SDES,
401 g_param_spec_boxed ("sdes", "SDES",
402 "The SDES items of this session",
403 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
405 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
406 g_param_spec_uint ("num-sources", "Num Sources",
407 "The number of sources in the session", 0, G_MAXUINT,
408 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
410 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
411 g_param_spec_uint ("num-active-sources", "Num Active Sources",
412 "The number of active sources in the session", 0, G_MAXUINT,
413 DEFAULT_NUM_ACTIVE_SOURCES,
414 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
418 * Get a GValue Array of all sources in the session.
421 * <title>Getting the #RTPSources of a session
428 * g_object_get (sess, "sources", &arr, NULL);
430 * for (i = 0; i < arr->n_values; i++) {
433 * val = g_value_array_get_nth (arr, i);
434 * source = g_value_get_object (val);
436 * g_value_array_free (arr);
441 g_object_class_install_property (gobject_class, PROP_SOURCES,
442 g_param_spec_boxed ("sources", "Sources",
443 "An array of all known sources in the session",
444 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
446 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
447 g_param_spec_boolean ("favor-new", "Favor new sources",
448 "Resolve SSRC conflict in favor of new sources", FALSE,
449 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
451 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
452 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
453 "Minimum interval between Regular RTCP packet (in ns)",
454 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
455 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
457 g_object_class_install_property (gobject_class,
458 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
459 g_param_spec_uint64 ("rtcp-feedback-retention-window",
460 "RTCP Feedback retention window",
461 "Duration during which RTCP Feedback packets are retained (in ns)",
462 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
463 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
465 g_object_class_install_property (gobject_class,
466 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
467 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
468 "RTCP Immediate Feedback threshold",
469 "The maximum number of members of a RTP session for which immediate"
470 " feedback is used (DEPRECATED: has no effect and is not needed)",
471 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
472 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
474 g_object_class_install_property (gobject_class, PROP_PROBATION,
475 g_param_spec_uint ("probation", "Number of probations",
476 "Consecutive packet sequence numbers to accept the source",
477 0, G_MAXUINT, DEFAULT_PROBATION,
478 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
483 * Various session statistics. This property returns a GstStructure
484 * with name application/x-rtp-session-stats with the following fields:
486 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
487 * dropped (due to bandwidth constraints)
488 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
489 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
493 g_object_class_install_property (gobject_class, PROP_STATS,
494 g_param_spec_boxed ("stats", "Statistics",
495 "Various statistics", GST_TYPE_STRUCTURE,
496 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
498 klass->get_source_by_ssrc =
499 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
500 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
502 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
506 rtp_session_init (RTPSession * sess)
511 g_mutex_init (&sess->lock);
512 sess->key = g_random_int ();
516 for (i = 0; i < 32; i++) {
518 g_hash_table_new_full (NULL, NULL, NULL,
519 (GDestroyNotify) g_object_unref);
522 rtp_stats_init_defaults (&sess->stats);
523 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
524 rtp_stats_set_min_interval (&sess->stats,
525 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
527 sess->recalc_bandwidth = TRUE;
528 sess->bandwidth = DEFAULT_BANDWIDTH;
529 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
530 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
531 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
533 /* default UDP header length */
534 sess->header_len = 28;
535 sess->mtu = DEFAULT_RTCP_MTU;
537 sess->probation = DEFAULT_PROBATION;
539 /* some default SDES entries */
540 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
542 /* we do not want to leak details like the username or hostname here */
543 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
544 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
548 /* we do not want to leak the user's real name here */
549 str = g_strdup_printf ("Anon%u", g_random_int ());
550 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
554 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
556 /* this is the SSRC we suggest */
557 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
559 sess->first_rtcp = TRUE;
560 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
561 sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
563 sess->allow_early = TRUE;
564 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
565 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
566 sess->rtcp_immediate_feedback_threshold =
567 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
569 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
571 sess->is_doing_ptp = TRUE;
575 rtp_session_finalize (GObject * object)
580 sess = RTP_SESSION_CAST (object);
582 gst_structure_free (sess->sdes);
584 g_list_free_full (sess->conflicting_addresses,
585 (GDestroyNotify) rtp_conflicting_address_free);
587 for (i = 0; i < 32; i++)
588 g_hash_table_destroy (sess->ssrcs[i]);
590 g_mutex_clear (&sess->lock);
592 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
596 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
598 GValue value = { 0 };
600 g_value_init (&value, RTP_TYPE_SOURCE);
601 g_value_take_object (&value, source);
602 /* copies the value */
603 g_value_array_append (arr, &value);
607 rtp_session_create_sources (RTPSession * sess)
612 RTP_SESSION_LOCK (sess);
613 /* get number of elements in the table */
614 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
615 /* create the result value array */
616 res = g_value_array_new (size);
618 /* and copy all values into the array */
619 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
620 RTP_SESSION_UNLOCK (sess);
625 static GstStructure *
626 rtp_session_create_stats (RTPSession * sess)
630 s = gst_structure_new ("application/x-rtp-session-stats",
631 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
632 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
633 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
639 rtp_session_set_property (GObject * object, guint prop_id,
640 const GValue * value, GParamSpec * pspec)
644 sess = RTP_SESSION (object);
647 case PROP_INTERNAL_SSRC:
648 RTP_SESSION_LOCK (sess);
649 sess->suggested_ssrc = g_value_get_uint (value);
650 RTP_SESSION_UNLOCK (sess);
651 if (sess->callbacks.reconfigure)
652 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
655 RTP_SESSION_LOCK (sess);
656 sess->bandwidth = g_value_get_double (value);
657 sess->recalc_bandwidth = TRUE;
658 RTP_SESSION_UNLOCK (sess);
660 case PROP_RTCP_FRACTION:
661 RTP_SESSION_LOCK (sess);
662 sess->rtcp_bandwidth = g_value_get_double (value);
663 sess->recalc_bandwidth = TRUE;
664 RTP_SESSION_UNLOCK (sess);
666 case PROP_RTCP_RR_BANDWIDTH:
667 RTP_SESSION_LOCK (sess);
668 sess->rtcp_rr_bandwidth = g_value_get_int (value);
669 sess->recalc_bandwidth = TRUE;
670 RTP_SESSION_UNLOCK (sess);
672 case PROP_RTCP_RS_BANDWIDTH:
673 RTP_SESSION_LOCK (sess);
674 sess->rtcp_rs_bandwidth = g_value_get_int (value);
675 sess->recalc_bandwidth = TRUE;
676 RTP_SESSION_UNLOCK (sess);
679 sess->mtu = g_value_get_uint (value);
682 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
685 sess->favor_new = g_value_get_boolean (value);
687 case PROP_RTCP_MIN_INTERVAL:
688 rtp_stats_set_min_interval (&sess->stats,
689 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
690 /* trigger reconsideration */
691 RTP_SESSION_LOCK (sess);
692 sess->next_rtcp_check_time = 0;
693 RTP_SESSION_UNLOCK (sess);
694 if (sess->callbacks.reconsider)
695 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
697 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
698 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
701 sess->probation = g_value_get_uint (value);
704 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
710 rtp_session_get_property (GObject * object, guint prop_id,
711 GValue * value, GParamSpec * pspec)
715 sess = RTP_SESSION (object);
718 case PROP_INTERNAL_SSRC:
719 g_value_set_uint (value, rtp_session_suggest_ssrc (sess));
721 case PROP_INTERNAL_SOURCE:
722 /* FIXME, return a random source */
723 g_value_set_object (value, NULL);
726 g_value_set_double (value, sess->bandwidth);
728 case PROP_RTCP_FRACTION:
729 g_value_set_double (value, sess->rtcp_bandwidth);
731 case PROP_RTCP_RR_BANDWIDTH:
732 g_value_set_int (value, sess->rtcp_rr_bandwidth);
734 case PROP_RTCP_RS_BANDWIDTH:
735 g_value_set_int (value, sess->rtcp_rs_bandwidth);
738 g_value_set_uint (value, sess->mtu);
741 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
743 case PROP_NUM_SOURCES:
744 g_value_set_uint (value, rtp_session_get_num_sources (sess));
746 case PROP_NUM_ACTIVE_SOURCES:
747 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
750 g_value_take_boxed (value, rtp_session_create_sources (sess));
753 g_value_set_boolean (value, sess->favor_new);
755 case PROP_RTCP_MIN_INTERVAL:
756 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
758 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
759 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
762 g_value_set_uint (value, sess->probation);
765 g_value_take_boxed (value, rtp_session_create_stats (sess));
768 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
774 on_new_ssrc (RTPSession * sess, RTPSource * source)
776 g_object_ref (source);
777 RTP_SESSION_UNLOCK (sess);
778 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
779 RTP_SESSION_LOCK (sess);
780 g_object_unref (source);
784 on_ssrc_collision (RTPSession * sess, RTPSource * source)
786 g_object_ref (source);
787 RTP_SESSION_UNLOCK (sess);
788 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
790 RTP_SESSION_LOCK (sess);
791 g_object_unref (source);
795 on_ssrc_validated (RTPSession * sess, RTPSource * source)
797 g_object_ref (source);
798 RTP_SESSION_UNLOCK (sess);
799 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
801 RTP_SESSION_LOCK (sess);
802 g_object_unref (source);
806 on_ssrc_active (RTPSession * sess, RTPSource * source)
808 g_object_ref (source);
809 RTP_SESSION_UNLOCK (sess);
810 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
811 RTP_SESSION_LOCK (sess);
812 g_object_unref (source);
816 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
818 g_object_ref (source);
819 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
820 RTP_SESSION_UNLOCK (sess);
821 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
822 RTP_SESSION_LOCK (sess);
823 g_object_unref (source);
827 on_bye_ssrc (RTPSession * sess, RTPSource * source)
829 g_object_ref (source);
830 RTP_SESSION_UNLOCK (sess);
831 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
832 RTP_SESSION_LOCK (sess);
833 g_object_unref (source);
837 on_bye_timeout (RTPSession * sess, RTPSource * source)
839 g_object_ref (source);
840 RTP_SESSION_UNLOCK (sess);
841 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
842 RTP_SESSION_LOCK (sess);
843 g_object_unref (source);
847 on_timeout (RTPSession * sess, RTPSource * source)
849 g_object_ref (source);
850 RTP_SESSION_UNLOCK (sess);
851 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
852 RTP_SESSION_LOCK (sess);
853 g_object_unref (source);
857 on_sender_timeout (RTPSession * sess, RTPSource * source)
859 g_object_ref (source);
860 RTP_SESSION_UNLOCK (sess);
861 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
863 RTP_SESSION_LOCK (sess);
864 g_object_unref (source);
870 * Create a new session object.
872 * Returns: a new #RTPSession. g_object_unref() after usage.
875 rtp_session_new (void)
879 sess = g_object_new (RTP_TYPE_SESSION, NULL);
885 * rtp_session_set_callbacks:
886 * @sess: an #RTPSession
887 * @callbacks: callbacks to configure
888 * @user_data: user data passed in the callbacks
890 * Configure a set of callbacks to be notified of actions.
893 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
896 g_return_if_fail (RTP_IS_SESSION (sess));
898 if (callbacks->process_rtp) {
899 sess->callbacks.process_rtp = callbacks->process_rtp;
900 sess->process_rtp_user_data = user_data;
902 if (callbacks->send_rtp) {
903 sess->callbacks.send_rtp = callbacks->send_rtp;
904 sess->send_rtp_user_data = user_data;
906 if (callbacks->send_rtcp) {
907 sess->callbacks.send_rtcp = callbacks->send_rtcp;
908 sess->send_rtcp_user_data = user_data;
910 if (callbacks->sync_rtcp) {
911 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
912 sess->sync_rtcp_user_data = user_data;
914 if (callbacks->clock_rate) {
915 sess->callbacks.clock_rate = callbacks->clock_rate;
916 sess->clock_rate_user_data = user_data;
918 if (callbacks->reconsider) {
919 sess->callbacks.reconsider = callbacks->reconsider;
920 sess->reconsider_user_data = user_data;
922 if (callbacks->request_key_unit) {
923 sess->callbacks.request_key_unit = callbacks->request_key_unit;
924 sess->request_key_unit_user_data = user_data;
926 if (callbacks->request_time) {
927 sess->callbacks.request_time = callbacks->request_time;
928 sess->request_time_user_data = user_data;
930 if (callbacks->notify_nack) {
931 sess->callbacks.notify_nack = callbacks->notify_nack;
932 sess->notify_nack_user_data = user_data;
934 if (callbacks->reconfigure) {
935 sess->callbacks.reconfigure = callbacks->reconfigure;
936 sess->reconfigure_user_data = user_data;
941 * rtp_session_set_process_rtp_callback:
942 * @sess: an #RTPSession
943 * @callback: callback to set
944 * @user_data: user data passed in the callback
946 * Configure only the process_rtp callback to be notified of the process_rtp action.
949 rtp_session_set_process_rtp_callback (RTPSession * sess,
950 RTPSessionProcessRTP callback, gpointer user_data)
952 g_return_if_fail (RTP_IS_SESSION (sess));
954 sess->callbacks.process_rtp = callback;
955 sess->process_rtp_user_data = user_data;
959 * rtp_session_set_send_rtp_callback:
960 * @sess: an #RTPSession
961 * @callback: callback to set
962 * @user_data: user data passed in the callback
964 * Configure only the send_rtp callback to be notified of the send_rtp action.
967 rtp_session_set_send_rtp_callback (RTPSession * sess,
968 RTPSessionSendRTP callback, gpointer user_data)
970 g_return_if_fail (RTP_IS_SESSION (sess));
972 sess->callbacks.send_rtp = callback;
973 sess->send_rtp_user_data = user_data;
977 * rtp_session_set_send_rtcp_callback:
978 * @sess: an #RTPSession
979 * @callback: callback to set
980 * @user_data: user data passed in the callback
982 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
985 rtp_session_set_send_rtcp_callback (RTPSession * sess,
986 RTPSessionSendRTCP callback, gpointer user_data)
988 g_return_if_fail (RTP_IS_SESSION (sess));
990 sess->callbacks.send_rtcp = callback;
991 sess->send_rtcp_user_data = user_data;
995 * rtp_session_set_sync_rtcp_callback:
996 * @sess: an #RTPSession
997 * @callback: callback to set
998 * @user_data: user data passed in the callback
1000 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
1003 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
1004 RTPSessionSyncRTCP callback, gpointer user_data)
1006 g_return_if_fail (RTP_IS_SESSION (sess));
1008 sess->callbacks.sync_rtcp = callback;
1009 sess->sync_rtcp_user_data = user_data;
1013 * rtp_session_set_clock_rate_callback:
1014 * @sess: an #RTPSession
1015 * @callback: callback to set
1016 * @user_data: user data passed in the callback
1018 * Configure only the clock_rate callback to be notified of the clock_rate action.
1021 rtp_session_set_clock_rate_callback (RTPSession * sess,
1022 RTPSessionClockRate callback, gpointer user_data)
1024 g_return_if_fail (RTP_IS_SESSION (sess));
1026 sess->callbacks.clock_rate = callback;
1027 sess->clock_rate_user_data = user_data;
1031 * rtp_session_set_reconsider_callback:
1032 * @sess: an #RTPSession
1033 * @callback: callback to set
1034 * @user_data: user data passed in the callback
1036 * Configure only the reconsider callback to be notified of the reconsider action.
1039 rtp_session_set_reconsider_callback (RTPSession * sess,
1040 RTPSessionReconsider callback, gpointer user_data)
1042 g_return_if_fail (RTP_IS_SESSION (sess));
1044 sess->callbacks.reconsider = callback;
1045 sess->reconsider_user_data = user_data;
1049 * rtp_session_set_request_time_callback:
1050 * @sess: an #RTPSession
1051 * @callback: callback to set
1052 * @user_data: user data passed in the callback
1054 * Configure only the request_time callback
1057 rtp_session_set_request_time_callback (RTPSession * sess,
1058 RTPSessionRequestTime callback, gpointer user_data)
1060 g_return_if_fail (RTP_IS_SESSION (sess));
1062 sess->callbacks.request_time = callback;
1063 sess->request_time_user_data = user_data;
1067 * rtp_session_set_bandwidth:
1068 * @sess: an #RTPSession
1069 * @bandwidth: the bandwidth allocated
1071 * Set the session bandwidth in bytes per second.
1074 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1076 g_return_if_fail (RTP_IS_SESSION (sess));
1078 RTP_SESSION_LOCK (sess);
1079 sess->stats.bandwidth = bandwidth;
1080 RTP_SESSION_UNLOCK (sess);
1084 * rtp_session_get_bandwidth:
1085 * @sess: an #RTPSession
1087 * Get the session bandwidth.
1089 * Returns: the session bandwidth.
1092 rtp_session_get_bandwidth (RTPSession * sess)
1096 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1098 RTP_SESSION_LOCK (sess);
1099 result = sess->stats.bandwidth;
1100 RTP_SESSION_UNLOCK (sess);
1106 * rtp_session_set_rtcp_fraction:
1107 * @sess: an #RTPSession
1108 * @bandwidth: the RTCP bandwidth
1110 * Set the bandwidth in bytes per second that should be used for RTCP
1114 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1116 g_return_if_fail (RTP_IS_SESSION (sess));
1118 RTP_SESSION_LOCK (sess);
1119 sess->stats.rtcp_bandwidth = bandwidth;
1120 RTP_SESSION_UNLOCK (sess);
1124 * rtp_session_get_rtcp_fraction:
1125 * @sess: an #RTPSession
1127 * Get the session bandwidth used for RTCP.
1129 * Returns: The bandwidth used for RTCP messages.
1132 rtp_session_get_rtcp_fraction (RTPSession * sess)
1136 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1138 RTP_SESSION_LOCK (sess);
1139 result = sess->stats.rtcp_bandwidth;
1140 RTP_SESSION_UNLOCK (sess);
1146 * rtp_session_get_sdes_struct:
1147 * @sess: an #RTSPSession
1149 * Get the SDES data as a #GstStructure
1151 * Returns: a GstStructure with SDES items for @sess. This function returns a
1152 * copy of the SDES structure, use gst_structure_free() after usage.
1155 rtp_session_get_sdes_struct (RTPSession * sess)
1157 GstStructure *result = NULL;
1159 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1161 RTP_SESSION_LOCK (sess);
1163 result = gst_structure_copy (sess->sdes);
1164 RTP_SESSION_UNLOCK (sess);
1170 * rtp_session_set_sdes_struct:
1171 * @sess: an #RTSPSession
1172 * @sdes: a #GstStructure
1174 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1177 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1179 g_return_if_fail (sdes);
1180 g_return_if_fail (RTP_IS_SESSION (sess));
1182 RTP_SESSION_LOCK (sess);
1184 gst_structure_free (sess->sdes);
1185 sess->sdes = gst_structure_copy (sdes);
1186 RTP_SESSION_UNLOCK (sess);
1189 static GstFlowReturn
1190 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1192 GstFlowReturn result = GST_FLOW_OK;
1194 if (source->internal) {
1195 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1197 RTP_SESSION_UNLOCK (session);
1199 if (session->callbacks.send_rtp)
1201 session->callbacks.send_rtp (session, source, data,
1202 session->send_rtp_user_data);
1204 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1207 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1208 RTP_SESSION_UNLOCK (session);
1210 if (session->callbacks.process_rtp)
1212 session->callbacks.process_rtp (session, source,
1213 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1215 gst_buffer_unref (GST_BUFFER_CAST (data));
1217 RTP_SESSION_LOCK (session);
1223 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1227 RTP_SESSION_UNLOCK (session);
1229 if (session->callbacks.clock_rate)
1231 session->callbacks.clock_rate (session, pt,
1232 session->clock_rate_user_data);
1236 RTP_SESSION_LOCK (session);
1238 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1243 static RTPSourceCallbacks callbacks = {
1244 (RTPSourcePushRTP) source_push_rtp,
1245 (RTPSourceClockRate) source_clock_rate,
1250 * rtp_session_find_conflicting_address:
1251 * @session: The session the packet came in
1252 * @address: address to check for
1253 * @time: The time when the packet that is possibly in conflict arrived
1255 * Checks if an address which has a conflict is already known. If it is
1256 * a known conflict, remember the time
1258 * Returns: TRUE if it was a known conflict, FALSE otherwise
1261 rtp_session_find_conflicting_address (RTPSession * session,
1262 GSocketAddress * address, GstClockTime time)
1264 return find_conflicting_address (session->conflicting_addresses, address,
1269 * rtp_session_add_conflicting_address:
1270 * @session: The session the packet came in
1271 * @address: address to remember
1272 * @time: The time when the packet that is in conflict arrived
1274 * Adds a new conflict address
1277 rtp_session_add_conflicting_address (RTPSession * sess,
1278 GSocketAddress * address, GstClockTime time)
1280 sess->conflicting_addresses =
1281 add_conflicting_address (sess->conflicting_addresses, address, time);
1286 check_collision (RTPSession * sess, RTPSource * source,
1287 RTPPacketInfo * pinfo, gboolean rtp)
1291 /* If we have no pinfo address, we can't do collision checking */
1292 if (!pinfo->address)
1295 ssrc = rtp_source_get_ssrc (source);
1297 if (!source->internal) {
1298 GSocketAddress *from;
1300 /* This is not our local source, but lets check if two remote
1303 from = source->rtp_from;
1305 from = source->rtcp_from;
1309 if (__g_socket_address_equal (from, pinfo->address)) {
1310 /* Address is the same */
1313 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1314 if (sess->favor_new) {
1315 if (rtp_source_find_conflicting_address (source,
1316 pinfo->address, pinfo->current_time)) {
1319 buf1 = __g_socket_address_to_string (pinfo->address);
1320 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1328 /* Current address is not a known conflict, lets assume this is
1329 * a new source. Save old address in possible conflict list
1331 rtp_source_add_conflicting_address (source, from,
1332 pinfo->current_time);
1334 buf1 = __g_socket_address_to_string (from);
1335 buf2 = __g_socket_address_to_string (pinfo->address);
1337 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1338 " saving old as known conflict", ssrc, buf1, buf2);
1341 rtp_source_set_rtp_from (source, pinfo->address);
1343 rtp_source_set_rtcp_from (source, pinfo->address);
1351 /* Don't need to save old addresses, we ignore new sources */
1356 /* We don't already have a from address for RTP, just set it */
1358 rtp_source_set_rtp_from (source, pinfo->address);
1360 rtp_source_set_rtcp_from (source, pinfo->address);
1364 /* FIXME: Log 3rd party collision somehow
1365 * Maybe should be done in upper layer, only the SDES can tell us
1366 * if its a collision or a loop
1369 /* This is sending with our ssrc, is it an address we already know */
1370 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1371 pinfo->current_time)) {
1372 /* Its a known conflict, its probably a loop, not a collision
1373 * lets just drop the incoming packet
1375 GST_DEBUG ("Our packets are being looped back to us, dropping");
1377 /* Its a new collision, lets change our SSRC */
1378 rtp_session_add_conflicting_address (sess, pinfo->address,
1379 pinfo->current_time);
1381 GST_DEBUG ("Collision for SSRC %x", ssrc);
1382 /* mark the source BYE */
1383 rtp_source_mark_bye (source, "SSRC Collision");
1384 /* if we were suggesting this SSRC, change to something else */
1385 if (sess->suggested_ssrc == ssrc)
1386 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1388 on_ssrc_collision (sess, source);
1390 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1399 gboolean is_doing_ptp;
1400 GSocketAddress *new_addr;
1403 /* check if the two given ip addr are the same (do not care about the port) */
1405 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1408 g_inet_address_equal (g_inet_socket_address_get_address
1409 (G_INET_SOCKET_ADDRESS (a)),
1410 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1414 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1415 CompareAddrData * data)
1417 /* only compare ip addr of remote sources which are also not closing */
1418 if (!source->internal && !source->closing && source->rtp_from) {
1419 /* look for the first rtp source */
1420 if (!data->new_addr)
1421 data->new_addr = source->rtp_from;
1422 /* compare current ip addr with the first one */
1424 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1429 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1430 CompareAddrData * data)
1432 /* only compare ip addr of remote sources which are also not closing */
1433 if (!source->internal && !source->closing && source->rtcp_from) {
1434 /* look for the first rtcp source */
1435 if (!data->new_addr)
1436 data->new_addr = source->rtcp_from;
1438 /* compare current ip addr with the first one */
1439 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1443 /* loop over our non-internal source to know if the session
1444 * is doing point-to-point */
1446 session_update_ptp (RTPSession * sess)
1448 /* to know if the session is doing point to point, the ip addr
1449 * of each non-internal (=remotes) source have to be compared
1452 gboolean is_doing_rtp_ptp;
1453 gboolean is_doing_rtcp_ptp;
1454 CompareAddrData data;
1456 /* compare the first remote source's ip addr that receive rtp packets
1457 * with other remote rtp source.
1458 * it's enough because the session just needs to know if they are all
1461 data.is_doing_ptp = TRUE;
1462 data.new_addr = NULL;
1463 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1464 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1465 is_doing_rtp_ptp = data.is_doing_ptp;
1467 /* same but about rtcp */
1468 data.is_doing_ptp = TRUE;
1469 data.new_addr = NULL;
1470 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1471 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1472 is_doing_rtcp_ptp = data.is_doing_ptp;
1474 /* the session is doing point-to-point if all rtp remote have the same
1475 * ip addr and if all rtcp remote sources have the same ip addr */
1476 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1478 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1482 add_source (RTPSession * sess, RTPSource * src)
1484 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1485 GINT_TO_POINTER (src->ssrc), src);
1486 /* report the new source ASAP */
1487 src->generation = sess->generation;
1488 /* we have one more source now */
1489 sess->total_sources++;
1490 if (RTP_SOURCE_IS_ACTIVE (src))
1491 sess->stats.active_sources++;
1492 if (src->internal) {
1493 sess->stats.internal_sources++;
1494 if (sess->suggested_ssrc != src->ssrc)
1495 sess->suggested_ssrc = src->ssrc;
1498 /* update point-to-point status */
1500 session_update_ptp (sess);
1504 find_source (RTPSession * sess, guint32 ssrc)
1506 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1507 GINT_TO_POINTER (ssrc));
1510 /* must be called with the session lock, the returned source needs to be
1511 * unreffed after usage. */
1513 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1514 RTPPacketInfo * pinfo, gboolean rtp)
1518 source = find_source (sess, ssrc);
1519 if (source == NULL) {
1520 /* make new Source in probation and insert */
1521 source = rtp_source_new (ssrc);
1523 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1525 /* for RTP packets we need to set the source in probation. Receiving RTCP
1526 * packets of an SSRC, on the other hand, is a strong indication that we
1527 * are dealing with a valid source. */
1529 g_object_set (source, "probation", sess->probation, NULL);
1531 g_object_set (source, "probation", 0, NULL);
1533 /* store from address, if any */
1534 if (pinfo->address) {
1536 rtp_source_set_rtp_from (source, pinfo->address);
1538 rtp_source_set_rtcp_from (source, pinfo->address);
1541 /* configure a callback on the source */
1542 rtp_source_set_callbacks (source, &callbacks, sess);
1544 add_source (sess, source);
1548 /* check for collision, this updates the address when not previously set */
1549 if (check_collision (sess, source, pinfo, rtp)) {
1552 /* Receiving RTCP packets of an SSRC is a strong indication that we
1553 * are dealing with a valid source. */
1555 g_object_set (source, "probation", 0, NULL);
1557 /* update last activity */
1558 source->last_activity = pinfo->current_time;
1560 source->last_rtp_activity = pinfo->current_time;
1561 g_object_ref (source);
1566 /* must be called with the session lock, the returned source needs to be
1567 * unreffed after usage. */
1569 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1570 GstClockTime current_time)
1574 source = find_source (sess, ssrc);
1575 if (source == NULL) {
1576 /* make new internal Source and insert */
1577 source = rtp_source_new (ssrc);
1579 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1581 source->validated = TRUE;
1582 source->internal = TRUE;
1583 source->probation = FALSE;
1584 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1585 rtp_source_set_callbacks (source, &callbacks, sess);
1587 add_source (sess, source);
1592 /* update last activity */
1593 if (current_time != GST_CLOCK_TIME_NONE) {
1594 source->last_activity = current_time;
1595 source->last_rtp_activity = current_time;
1597 g_object_ref (source);
1603 * rtp_session_suggest_ssrc:
1604 * @sess: a #RTPSession
1606 * Suggest an unused SSRC in @sess.
1608 * Returns: a free unused SSRC
1611 rtp_session_suggest_ssrc (RTPSession * sess)
1615 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1617 RTP_SESSION_LOCK (sess);
1618 result = sess->suggested_ssrc;
1619 RTP_SESSION_UNLOCK (sess);
1625 * rtp_session_add_source:
1626 * @sess: a #RTPSession
1627 * @src: #RTPSource to add
1629 * Add @src to @session.
1631 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1632 * existed in the session.
1635 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1637 gboolean result = FALSE;
1640 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1641 g_return_val_if_fail (src != NULL, FALSE);
1643 RTP_SESSION_LOCK (sess);
1644 find = find_source (sess, src->ssrc);
1646 add_source (sess, src);
1649 RTP_SESSION_UNLOCK (sess);
1655 * rtp_session_get_num_sources:
1656 * @sess: an #RTPSession
1658 * Get the number of sources in @sess.
1660 * Returns: The number of sources in @sess.
1663 rtp_session_get_num_sources (RTPSession * sess)
1667 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1669 RTP_SESSION_LOCK (sess);
1670 result = sess->total_sources;
1671 RTP_SESSION_UNLOCK (sess);
1677 * rtp_session_get_num_active_sources:
1678 * @sess: an #RTPSession
1680 * Get the number of active sources in @sess. A source is considered active when
1681 * it has been validated and has not yet received a BYE RTCP message.
1683 * Returns: The number of active sources in @sess.
1686 rtp_session_get_num_active_sources (RTPSession * sess)
1690 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1692 RTP_SESSION_LOCK (sess);
1693 result = sess->stats.active_sources;
1694 RTP_SESSION_UNLOCK (sess);
1700 * rtp_session_get_source_by_ssrc:
1701 * @sess: an #RTPSession
1704 * Find the source with @ssrc in @sess.
1706 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1707 * g_object_unref() after usage.
1710 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1714 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1716 RTP_SESSION_LOCK (sess);
1717 result = find_source (sess, ssrc);
1719 g_object_ref (result);
1720 RTP_SESSION_UNLOCK (sess);
1725 /* should be called with the SESSION lock */
1727 rtp_session_create_new_ssrc (RTPSession * sess)
1732 ssrc = g_random_int ();
1734 /* see if it exists in the session, we're done if it doesn't */
1735 if (find_source (sess, ssrc) == NULL)
1743 * rtp_session_create_source:
1744 * @sess: an #RTPSession
1746 * Create an #RTPSource for use in @sess. This function will create a source
1747 * with an ssrc that is currently not used by any participants in the session.
1749 * Returns: an #RTPSource.
1752 rtp_session_create_source (RTPSession * sess)
1757 RTP_SESSION_LOCK (sess);
1758 ssrc = rtp_session_create_new_ssrc (sess);
1759 source = rtp_source_new (ssrc);
1760 rtp_source_set_callbacks (source, &callbacks, sess);
1761 /* we need an additional ref for the source in the hashtable */
1762 g_object_ref (source);
1763 add_source (sess, source);
1764 RTP_SESSION_UNLOCK (sess);
1770 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1772 GstNetAddressMeta *meta;
1774 /* get packet size including header overhead */
1775 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1779 GstRTPBuffer rtp = { NULL };
1781 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1782 goto invalid_packet;
1784 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1788 /* only keep info for first buffer */
1789 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1790 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1791 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1792 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1793 /* copy available csrc */
1794 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1795 for (i = 0; i < pinfo->csrc_count; i++)
1796 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1798 gst_rtp_buffer_unmap (&rtp);
1802 /* for netbuffer we can store the IP address to check for collisions */
1803 meta = gst_buffer_get_net_address_meta (*buffer);
1805 g_object_unref (pinfo->address);
1807 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1809 pinfo->address = NULL;
1817 GST_DEBUG ("invalid RTP packet received");
1822 /* update the RTPPacketInfo structure with the current time and other bits
1823 * about the current buffer we are handling.
1824 * This function is typically called when a validated packet is received.
1825 * This function should be called with the SESSION_LOCK
1828 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
1829 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
1830 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1836 pinfo->is_list = is_list;
1838 pinfo->current_time = current_time;
1839 pinfo->running_time = running_time;
1840 pinfo->ntpnstime = ntpnstime;
1841 pinfo->header_len = sess->header_len;
1843 pinfo->payload_len = 0;
1847 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
1849 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
1852 GstBuffer *buffer = GST_BUFFER_CAST (data);
1853 res = update_packet (&buffer, 0, pinfo);
1859 clean_packet_info (RTPPacketInfo * pinfo)
1862 g_object_unref (pinfo->address);
1864 gst_mini_object_unref (pinfo->data);
1870 source_update_active (RTPSession * sess, RTPSource * source,
1871 gboolean prevactive)
1873 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
1874 guint32 ssrc = source->ssrc;
1876 if (prevactive == active)
1880 sess->stats.active_sources++;
1881 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1882 sess->stats.active_sources);
1884 sess->stats.active_sources--;
1885 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1886 sess->stats.active_sources);
1892 source_update_sender (RTPSession * sess, RTPSource * source,
1893 gboolean prevsender)
1895 gboolean sender = RTP_SOURCE_IS_SENDER (source);
1896 guint32 ssrc = source->ssrc;
1898 if (prevsender == sender)
1902 sess->stats.sender_sources++;
1903 if (source->internal)
1904 sess->stats.internal_sender_sources++;
1905 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1906 sess->stats.sender_sources);
1908 sess->stats.sender_sources--;
1909 if (source->internal)
1910 sess->stats.internal_sender_sources--;
1911 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1912 sess->stats.sender_sources);
1918 * rtp_session_process_rtp:
1919 * @sess: and #RTPSession
1920 * @buffer: an RTP buffer
1921 * @current_time: the current system time
1922 * @running_time: the running_time of @buffer
1924 * Process an RTP buffer in the session manager. This function takes ownership
1927 * Returns: a #GstFlowReturn.
1930 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1931 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1933 GstFlowReturn result;
1937 gboolean prevsender, prevactive;
1938 RTPPacketInfo pinfo = { 0, };
1941 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1942 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1944 RTP_SESSION_LOCK (sess);
1946 /* update pinfo stats */
1947 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
1948 current_time, running_time, ntpnstime)) {
1949 GST_DEBUG ("invalid RTP packet received");
1950 RTP_SESSION_UNLOCK (sess);
1951 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
1956 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
1960 prevsender = RTP_SOURCE_IS_SENDER (source);
1961 prevactive = RTP_SOURCE_IS_ACTIVE (source);
1962 oldrate = source->bitrate;
1964 /* let source process the packet */
1965 result = rtp_source_process_rtp (source, &pinfo);
1967 /* source became active */
1968 if (source_update_active (sess, source, prevactive))
1969 on_ssrc_validated (sess, source);
1971 source_update_sender (sess, source, prevsender);
1973 if (oldrate != source->bitrate)
1974 sess->recalc_bandwidth = TRUE;
1977 on_new_ssrc (sess, source);
1979 if (source->validated) {
1983 /* for validated sources, we add the CSRCs as well */
1984 for (i = 0; i < pinfo.csrc_count; i++) {
1986 RTPSource *csrc_src;
1988 csrc = pinfo.csrcs[i];
1991 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
1996 GST_DEBUG ("created new CSRC: %08x", csrc);
1997 rtp_source_set_as_csrc (csrc_src);
1998 source_update_active (sess, csrc_src, FALSE);
1999 on_new_ssrc (sess, csrc_src);
2001 g_object_unref (csrc_src);
2004 g_object_unref (source);
2006 RTP_SESSION_UNLOCK (sess);
2008 clean_packet_info (&pinfo);
2015 RTP_SESSION_UNLOCK (sess);
2016 clean_packet_info (&pinfo);
2017 GST_DEBUG ("ignoring packet because its collisioning");
2023 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
2024 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
2028 count = gst_rtcp_packet_get_rb_count (packet);
2029 for (i = 0; i < count; i++) {
2030 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
2031 guint8 fractionlost;
2035 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
2036 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2038 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2040 /* find our own source */
2041 src = find_source (sess, ssrc);
2045 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2046 /* only deal with report blocks for our session, we update the stats of
2047 * the sender of the RTCP message. We could also compare our stats against
2048 * the other sender to see if we are better or worse. */
2049 /* FIXME, need to keep track who the RB block is from */
2050 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2051 packetslost, exthighestseq, jitter, lsr, dlsr);
2054 on_ssrc_active (sess, source);
2057 /* A Sender report contains statistics about how the sender is doing. This
2058 * includes timing informataion such as the relation between RTP and NTP
2059 * timestamps and the number of packets/bytes it sent to us.
2061 * In this report is also included a set of report blocks related to how this
2062 * sender is receiving data (in case we (or somebody else) is also sending stuff
2063 * to it). This info includes the packet loss, jitter and seqnum. It also
2064 * contains information to calculate the round trip time (LSR/DLSR).
2067 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2068 RTPPacketInfo * pinfo, gboolean * do_sync)
2070 guint32 senderssrc, rtptime, packet_count, octet_count;
2073 gboolean created, prevsender;
2075 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2076 &packet_count, &octet_count);
2078 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2079 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2081 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2085 /* skip non-bye packets for sources that are marked BYE */
2086 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2089 /* don't try to do lip-sync for sources that sent a BYE */
2090 if (RTP_SOURCE_IS_MARKED_BYE (source))
2095 prevsender = RTP_SOURCE_IS_SENDER (source);
2097 /* first update the source */
2098 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2099 packet_count, octet_count);
2101 source_update_sender (sess, source, prevsender);
2104 on_new_ssrc (sess, source);
2106 rtp_session_process_rb (sess, source, packet, pinfo);
2109 g_object_unref (source);
2112 /* A receiver report contains statistics about how a receiver is doing. It
2113 * includes stuff like packet loss, jitter and the seqnum it received last. It
2114 * also contains info to calculate the round trip time.
2116 * We are only interested in how the sender of this report is doing wrt to us.
2119 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2120 RTPPacketInfo * pinfo)
2126 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2128 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2130 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2134 /* skip non-bye packets for sources that are marked BYE */
2135 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2139 on_new_ssrc (sess, source);
2141 rtp_session_process_rb (sess, source, packet, pinfo);
2144 g_object_unref (source);
2147 /* Get SDES items and store them in the SSRC */
2149 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2150 RTPPacketInfo * pinfo)
2153 gboolean more_items, more_entries;
2155 items = gst_rtcp_packet_sdes_get_item_count (packet);
2156 GST_DEBUG ("got SDES packet with %d items", items);
2158 more_items = gst_rtcp_packet_sdes_first_item (packet);
2160 while (more_items) {
2162 gboolean changed, created, prevactive;
2166 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2168 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2172 /* find src, no probation when dealing with RTCP */
2173 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2177 /* skip non-bye packets for sources that are marked BYE */
2178 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2181 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2183 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2185 while (more_entries) {
2186 GstRTCPSDESType type;
2192 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2194 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2197 if (type == GST_RTCP_SDES_PRIV) {
2198 name = g_strndup ((const gchar *) &data[1], data[0]);
2200 data += data[0] + 1;
2202 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2205 value = g_strndup ((const gchar *) data, len);
2207 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2212 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2216 /* takes ownership of sdes */
2217 changed = rtp_source_set_sdes_struct (source, sdes);
2219 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2220 source->validated = TRUE;
2223 on_new_ssrc (sess, source);
2225 /* source became active */
2226 if (source_update_active (sess, source, prevactive))
2227 on_ssrc_validated (sess, source);
2230 on_ssrc_sdes (sess, source);
2233 g_object_unref (source);
2235 more_items = gst_rtcp_packet_sdes_next_item (packet);
2240 /* BYE is sent when a client leaves the session
2243 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2244 RTPPacketInfo * pinfo)
2248 gboolean reconsider = FALSE;
2250 reason = gst_rtcp_packet_bye_get_reason (packet);
2251 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2253 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2254 for (i = 0; i < count; i++) {
2257 gboolean created, prevactive, prevsender;
2258 guint pmembers, members;
2260 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2261 GST_DEBUG ("SSRC: %08x", ssrc);
2263 /* find src and mark bye, no probation when dealing with RTCP */
2264 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2268 if (source->internal) {
2269 /* our own source, something weird with this packet */
2270 g_object_unref (source);
2274 /* store time for when we need to time out this source */
2275 source->bye_time = pinfo->current_time;
2277 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2278 prevsender = RTP_SOURCE_IS_SENDER (source);
2280 /* mark the source BYE */
2281 rtp_source_mark_bye (source, reason);
2283 pmembers = sess->stats.active_sources;
2285 source_update_active (sess, source, prevactive);
2286 source_update_sender (sess, source, prevsender);
2288 members = sess->stats.active_sources;
2290 if (!sess->scheduled_bye && members < pmembers) {
2291 /* some members went away since the previous timeout estimate.
2292 * Perform reverse reconsideration but only when we are not scheduling a
2294 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2295 pinfo->current_time < sess->next_rtcp_check_time) {
2296 GstClockTime time_remaining;
2298 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2299 sess->next_rtcp_check_time =
2300 gst_util_uint64_scale (time_remaining, members, pmembers);
2302 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2303 GST_TIME_ARGS (sess->next_rtcp_check_time));
2305 sess->next_rtcp_check_time += pinfo->current_time;
2307 /* mark pending reconsider. We only want to signal the reconsideration
2308 * once after we handled all the source in the bye packet */
2314 on_new_ssrc (sess, source);
2316 on_bye_ssrc (sess, source);
2318 g_object_unref (source);
2321 RTP_SESSION_UNLOCK (sess);
2322 /* notify app of reconsideration */
2323 if (sess->callbacks.reconsider)
2324 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2325 RTP_SESSION_LOCK (sess);
2331 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2332 RTPPacketInfo * pinfo)
2334 GST_DEBUG ("received APP");
2338 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2339 gboolean fir, GstClockTime current_time)
2341 guint32 round_trip = 0;
2343 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2345 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2346 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2349 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2350 GST_DEBUG ("Ignoring %s request because one was send without one "
2351 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2352 fir ? "FIR" : "PLI",
2353 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2354 GST_TIME_ARGS (round_trip_in_ns));;
2359 sess->last_keyframe_request = current_time;
2361 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2362 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2363 sess->callbacks.request_key_unit);
2365 RTP_SESSION_UNLOCK (sess);
2366 sess->callbacks.request_key_unit (sess, fir,
2367 sess->request_key_unit_user_data);
2368 RTP_SESSION_LOCK (sess);
2374 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2375 guint32 media_ssrc, GstClockTime current_time)
2379 if (!sess->callbacks.request_key_unit)
2382 src = find_source (sess, sender_ssrc);
2386 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2390 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2391 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2396 gboolean our_request = FALSE;
2398 if (!sess->callbacks.request_key_unit)
2404 src = find_source (sess, sender_ssrc);
2406 /* Hack because Google fails to set the sender_ssrc correctly */
2407 if (!src && sender_ssrc == 1) {
2408 GHashTableIter iter;
2410 /* we can't find the source if there are multiple */
2411 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2414 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2415 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2416 if (!src->internal && rtp_source_is_sender (src))
2424 for (position = 0; position < fci_length; position += 8) {
2425 guint8 *data = fci_data + position;
2428 ssrc = GST_READ_UINT32_BE (data);
2430 own = find_source (sess, ssrc);
2434 if (own->internal) {
2442 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2446 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2447 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2448 GstClockTime current_time)
2450 sess->stats.nacks_received++;
2452 if (!sess->callbacks.notify_nack)
2455 while (fci_length > 0) {
2456 guint16 seqnum, blp;
2458 seqnum = GST_READ_UINT16_BE (fci_data);
2459 blp = GST_READ_UINT16_BE (fci_data + 2);
2461 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2463 RTP_SESSION_UNLOCK (sess);
2464 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2465 sess->notify_nack_user_data);
2466 RTP_SESSION_LOCK (sess);
2474 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2475 RTPPacketInfo * pinfo, GstClockTime current_time)
2477 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2478 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2479 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2480 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2481 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2482 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2485 src = find_source (sess, media_ssrc);
2487 /* skip non-bye packets for sources that are marked BYE */
2488 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2491 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2492 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2494 if (g_signal_has_handler_pending (sess,
2495 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2496 GstBuffer *fci_buffer = NULL;
2498 if (fci_length > 0) {
2499 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2500 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2502 GST_BUFFER_TIMESTAMP (fci_buffer) = pinfo->running_time;
2505 RTP_SESSION_UNLOCK (sess);
2506 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2507 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2508 RTP_SESSION_LOCK (sess);
2511 gst_buffer_unref (fci_buffer);
2514 if (src && sess->rtcp_feedback_retention_window) {
2515 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2518 if ((src && src->internal) ||
2519 /* PSFB FIR puts the media ssrc inside the FCI */
2520 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2522 case GST_RTCP_TYPE_PSFB:
2524 case GST_RTCP_PSFB_TYPE_PLI:
2525 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2528 case GST_RTCP_PSFB_TYPE_FIR:
2529 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2536 case GST_RTCP_TYPE_RTPFB:
2538 case GST_RTCP_RTPFB_TYPE_NACK:
2539 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2540 fci_data, fci_length, current_time);
2552 * rtp_session_process_rtcp:
2553 * @sess: and #RTPSession
2554 * @buffer: an RTCP buffer
2555 * @current_time: the current system time
2556 * @ntpnstime: the current NTP time in nanoseconds
2558 * Process an RTCP buffer in the session manager. This function takes ownership
2561 * Returns: a #GstFlowReturn.
2564 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2565 GstClockTime current_time, guint64 ntpnstime)
2567 GstRTCPPacket packet;
2568 gboolean more, is_bye = FALSE, do_sync = FALSE;
2569 RTPPacketInfo pinfo = { 0, };
2570 GstFlowReturn result = GST_FLOW_OK;
2571 GstRTCPBuffer rtcp = { NULL, };
2573 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2574 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2576 if (!gst_rtcp_buffer_validate (buffer))
2577 goto invalid_packet;
2579 GST_DEBUG ("received RTCP packet");
2581 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
2584 RTP_SESSION_LOCK (sess);
2585 /* update pinfo stats */
2586 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2589 /* start processing the compound packet */
2590 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2591 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2595 type = gst_rtcp_packet_get_type (&packet);
2598 case GST_RTCP_TYPE_SR:
2599 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2601 case GST_RTCP_TYPE_RR:
2602 rtp_session_process_rr (sess, &packet, &pinfo);
2604 case GST_RTCP_TYPE_SDES:
2605 rtp_session_process_sdes (sess, &packet, &pinfo);
2607 case GST_RTCP_TYPE_BYE:
2609 /* don't try to attempt lip-sync anymore for streams with a BYE */
2611 rtp_session_process_bye (sess, &packet, &pinfo);
2613 case GST_RTCP_TYPE_APP:
2614 rtp_session_process_app (sess, &packet, &pinfo);
2616 case GST_RTCP_TYPE_RTPFB:
2617 case GST_RTCP_TYPE_PSFB:
2618 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2621 GST_WARNING ("got unknown RTCP packet");
2624 more = gst_rtcp_packet_move_to_next (&packet);
2627 gst_rtcp_buffer_unmap (&rtcp);
2629 /* if we are scheduling a BYE, we only want to count bye packets, else we
2630 * count everything */
2631 if (sess->scheduled_bye && is_bye) {
2632 sess->bye_stats.bye_members++;
2633 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2636 /* keep track of average packet size */
2637 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2639 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2640 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2641 RTP_SESSION_UNLOCK (sess);
2644 clean_packet_info (&pinfo);
2646 /* notify caller of sr packets in the callback */
2647 if (do_sync && sess->callbacks.sync_rtcp) {
2648 result = sess->callbacks.sync_rtcp (sess, buffer,
2649 sess->sync_rtcp_user_data);
2651 gst_buffer_unref (buffer);
2658 GST_DEBUG ("invalid RTCP packet received");
2659 gst_buffer_unref (buffer);
2665 * rtp_session_update_send_caps:
2666 * @sess: an #RTPSession
2669 * Update the caps of the sender in the rtp session.
2672 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2677 g_return_if_fail (RTP_IS_SESSION (sess));
2678 g_return_if_fail (GST_IS_CAPS (caps));
2680 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2682 s = gst_caps_get_structure (caps, 0);
2684 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2688 RTP_SESSION_LOCK (sess);
2689 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2691 rtp_source_update_caps (source, caps);
2692 g_object_unref (source);
2694 RTP_SESSION_UNLOCK (sess);
2699 * rtp_session_send_rtp:
2700 * @sess: an #RTPSession
2701 * @data: pointer to either an RTP buffer or a list of RTP buffers
2702 * @is_list: TRUE when @data is a buffer list
2703 * @current_time: the current system time
2704 * @running_time: the running time of @data
2706 * Send the RTP buffer in the session manager. This function takes ownership of
2709 * Returns: a #GstFlowReturn.
2712 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2713 GstClockTime current_time, GstClockTime running_time)
2715 GstFlowReturn result;
2717 gboolean prevsender;
2719 RTPPacketInfo pinfo = { 0, };
2722 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2723 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2725 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2727 RTP_SESSION_LOCK (sess);
2728 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2729 current_time, running_time, -1))
2730 goto invalid_packet;
2732 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
2734 prevsender = RTP_SOURCE_IS_SENDER (source);
2735 oldrate = source->bitrate;
2737 /* we use our own source to send */
2738 result = rtp_source_send_rtp (source, &pinfo);
2740 source_update_sender (sess, source, prevsender);
2742 if (oldrate != source->bitrate)
2743 sess->recalc_bandwidth = TRUE;
2744 RTP_SESSION_UNLOCK (sess);
2746 g_object_unref (source);
2747 clean_packet_info (&pinfo);
2753 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
2754 RTP_SESSION_UNLOCK (sess);
2755 GST_DEBUG ("invalid RTP packet received");
2761 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
2763 *bandwidth += source->bitrate;
2766 /* must be called with session lock */
2768 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2771 GstClockTime result;
2772 RTPSessionStats *stats;
2774 /* recalculate bandwidth when it changed */
2775 if (sess->recalc_bandwidth) {
2778 if (sess->bandwidth > 0)
2779 bandwidth = sess->bandwidth;
2781 /* If it is <= 0, then try to estimate the actual bandwidth */
2784 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2785 (GHFunc) add_bitrates, &bandwidth);
2788 if (bandwidth < 8000)
2789 bandwidth = RTP_STATS_BANDWIDTH;
2791 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
2792 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
2794 sess->recalc_bandwidth = FALSE;
2797 if (sess->scheduled_bye) {
2798 stats = &sess->bye_stats;
2799 result = rtp_stats_calculate_bye_interval (stats);
2801 stats = &sess->stats;
2802 result = rtp_stats_calculate_rtcp_interval (stats,
2803 stats->internal_sender_sources > 0, first);
2806 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2807 GST_TIME_ARGS (result), first);
2809 if (!deterministic && result != GST_CLOCK_TIME_NONE)
2810 result = rtp_stats_add_rtcp_jitter (stats, result);
2812 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2818 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
2820 if (source->internal)
2821 rtp_source_mark_bye (source, reason);
2825 * rtp_session_mark_all_bye:
2826 * @sess: an #RTPSession
2829 * Mark all internal sources of the session as BYE with @reason.
2832 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
2834 g_return_if_fail (RTP_IS_SESSION (sess));
2836 RTP_SESSION_LOCK (sess);
2837 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2838 (GHFunc) source_mark_bye, (gpointer) reason);
2839 RTP_SESSION_UNLOCK (sess);
2842 /* Stop the current @sess and schedule a BYE message for the other members.
2843 * One must have the session lock to call this function
2845 static GstFlowReturn
2846 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
2848 GstFlowReturn result = GST_FLOW_OK;
2849 GstClockTime interval;
2851 /* nothing to do it we already scheduled bye */
2852 if (sess->scheduled_bye)
2855 /* we schedule BYE now */
2856 sess->scheduled_bye = TRUE;
2857 /* at least one member wants to send a BYE */
2858 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
2859 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
2860 sess->bye_stats.bye_members = 1;
2861 sess->first_rtcp = TRUE;
2862 sess->allow_early = TRUE;
2864 /* reschedule transmission */
2865 sess->last_rtcp_send_time = current_time;
2866 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2868 if (interval != GST_CLOCK_TIME_NONE)
2869 sess->next_rtcp_check_time = current_time + interval;
2871 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
2873 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2874 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2876 RTP_SESSION_UNLOCK (sess);
2877 /* notify app of reconsideration */
2878 if (sess->callbacks.reconsider)
2879 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2880 RTP_SESSION_LOCK (sess);
2887 * rtp_session_schedule_bye:
2888 * @sess: an #RTPSession
2889 * @current_time: the current system time
2891 * Schedule a BYE message for all sources marked as BYE in @sess.
2893 * Returns: a #GstFlowReturn.
2896 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
2898 GstFlowReturn result;
2900 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2902 RTP_SESSION_LOCK (sess);
2903 result = rtp_session_schedule_bye_locked (sess, current_time);
2904 RTP_SESSION_UNLOCK (sess);
2910 * rtp_session_next_timeout:
2911 * @sess: an #RTPSession
2912 * @current_time: the current system time
2914 * Get the next time we should perform session maintenance tasks.
2916 * Returns: a time when rtp_session_on_timeout() should be called with the
2917 * current system time.
2920 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2922 GstClockTime result, interval = 0;
2924 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
2926 RTP_SESSION_LOCK (sess);
2928 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
2929 GST_DEBUG ("have early rtcp time");
2930 result = sess->next_early_rtcp_time;
2934 result = sess->next_rtcp_check_time;
2936 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2937 ", next time: %" GST_TIME_FORMAT,
2938 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2940 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
2941 GST_DEBUG ("take current time as base");
2942 /* our previous check time expired, start counting from the current time
2944 result = current_time;
2947 if (sess->scheduled_bye) {
2948 if (sess->bye_stats.active_sources >= 50) {
2949 GST_DEBUG ("reconsider BYE, more than 50 sources");
2950 /* reconsider BYE if members >= 50 */
2951 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2954 if (sess->first_rtcp) {
2955 GST_DEBUG ("first RTCP packet");
2956 /* we are called for the first time */
2957 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2958 } else if (sess->next_rtcp_check_time < current_time) {
2959 GST_DEBUG ("old check time expired, getting new timeout");
2960 /* get a new timeout when we need to */
2961 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
2965 if (interval != GST_CLOCK_TIME_NONE)
2968 result = GST_CLOCK_TIME_NONE;
2970 sess->next_rtcp_check_time = result;
2974 GST_DEBUG ("current time: %" GST_TIME_FORMAT
2975 ", next time: %" GST_TIME_FORMAT,
2976 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2977 RTP_SESSION_UNLOCK (sess);
2991 GstRTCPBuffer rtcpbuf;
2994 guint num_to_report;
2999 GstClockTime current_time;
3001 GstClockTime running_time;
3002 GstClockTime interval;
3003 GstRTCPPacket packet;
3006 gboolean may_suppress;
3008 guint nacked_seqnums;
3012 session_start_rtcp (RTPSession * sess, ReportData * data)
3014 GstRTCPPacket *packet = &data->packet;
3015 RTPSource *own = data->source;
3016 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3018 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
3019 data->has_sdes = FALSE;
3021 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3023 if (RTP_SOURCE_IS_SENDER (own)) {
3026 guint32 packet_count, octet_count;
3028 /* we are a sender, create SR */
3029 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
3030 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
3032 /* get latest stats */
3033 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
3034 &ntptime, &rtptime, &packet_count, &octet_count);
3036 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
3037 packet_count, octet_count);
3039 /* fill in sender report info */
3040 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
3041 ntptime, rtptime, packet_count, octet_count);
3043 /* we are only receiver, create RR */
3044 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3045 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3046 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3050 /* construct a Sender or Receiver Report */
3052 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3054 RTPSession *sess = data->sess;
3055 GstRTCPPacket *packet = &data->packet;
3056 guint8 fractionlost;
3058 guint32 exthighestseq, jitter;
3061 /* don't report for sources in future generations */
3062 if (((gint16) (source->generation - sess->generation)) > 0) {
3063 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3064 source->generation, sess->generation);
3068 if (g_hash_table_contains (source->reported_in_sr_of,
3069 GUINT_TO_POINTER (data->source->ssrc))) {
3070 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3074 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3075 GST_DEBUG ("max RB count reached");
3079 /* only report about other sender */
3080 if (source == data->source)
3083 if (!RTP_SOURCE_IS_SENDER (source)) {
3084 GST_DEBUG ("source %08x not sender", source->ssrc);
3088 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3091 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3092 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3094 /* store last generated RR packet */
3095 source->last_rr.is_valid = TRUE;
3096 source->last_rr.fractionlost = fractionlost;
3097 source->last_rr.packetslost = packetslost;
3098 source->last_rr.exthighestseq = exthighestseq;
3099 source->last_rr.jitter = jitter;
3100 source->last_rr.lsr = lsr;
3101 source->last_rr.dlsr = dlsr;
3103 /* packet is not yet filled, add report block for this source. */
3104 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3105 exthighestseq, jitter, lsr, dlsr);
3108 g_hash_table_add (source->reported_in_sr_of,
3109 GUINT_TO_POINTER (data->source->ssrc));
3114 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3116 GstRTCPPacket *packet = &data->packet;
3120 if (!source->send_fir)
3123 len = gst_rtcp_packet_fb_get_fci_length (packet);
3124 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3125 /* exit because the packet is full, will put next request in a
3129 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3131 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3133 fci_data[0] = source->current_send_fir_seqnum;
3134 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3136 source->send_fir = FALSE;
3140 session_fir (RTPSession * sess, ReportData * data)
3142 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3143 GstRTCPPacket *packet = &data->packet;
3145 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3148 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3149 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3150 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3152 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3153 (GHFunc) session_add_fir, data);
3155 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3156 gst_rtcp_packet_remove (packet);
3158 data->may_suppress = FALSE;
3162 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3164 GstRTCPPacket packet;
3165 GstRTCPBuffer rtcp = { NULL, };
3166 gboolean ret = FALSE;
3168 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3170 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3171 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3172 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3176 gst_rtcp_buffer_unmap (&rtcp);
3183 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3185 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3186 GstRTCPPacket *packet = &data->packet;
3188 if (!source->send_pli)
3191 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3194 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3195 /* exit because the packet is full, will put next request in a
3199 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3200 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3201 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3203 source->send_pli = FALSE;
3204 data->may_suppress = FALSE;
3207 /* construct NACK */
3209 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3211 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3212 GstRTCPPacket *packet = &data->packet;
3217 if (!source->send_nack)
3220 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3221 /* exit because the packet is full, will put next request in a
3225 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3226 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3227 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3229 nacks = rtp_source_get_nacks (source, &n_nacks);
3230 GST_DEBUG ("%u NACKs", n_nacks);
3231 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3234 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3235 for (i = 0; i < n_nacks; i++) {
3236 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3238 data->nacked_seqnums++;
3241 rtp_source_clear_nacks (source);
3242 data->may_suppress = FALSE;
3245 /* perform cleanup of sources that timed out */
3247 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3249 gboolean remove = FALSE;
3250 gboolean byetimeout = FALSE;
3251 gboolean sendertimeout = FALSE;
3252 gboolean is_sender, is_active;
3253 RTPSession *sess = data->sess;
3254 GstClockTime interval, binterval;
3257 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3259 /* check for outdated collisions */
3260 if (source->internal) {
3261 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3262 rtp_source_timeout (source, data->current_time,
3263 data->running_time - sess->rtcp_feedback_retention_window);
3266 /* nothing else to do when without RTCP */
3267 if (data->interval == GST_CLOCK_TIME_NONE)
3270 is_sender = RTP_SOURCE_IS_SENDER (source);
3271 is_active = RTP_SOURCE_IS_ACTIVE (source);
3273 /* our own rtcp interval may have been forced low by secondary configuration,
3274 * while sender side may still operate with higher interval,
3275 * so do not just take our interval to decide on timing out sender,
3276 * but take (if data->interval <= 5 * GST_SECOND):
3277 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3278 * where sender_interval is difference between last 2 received RTCP reports
3280 if (data->interval >= 5 * GST_SECOND || source->internal) {
3281 binterval = data->interval;
3283 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3284 GST_TIME_ARGS (source->stats.prev_rtcptime),
3285 GST_TIME_ARGS (source->stats.last_rtcptime));
3286 /* if not received enough yet, fallback to larger default */
3287 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3288 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3290 binterval = 5 * GST_SECOND;
3291 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3293 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3294 GST_TIME_ARGS (binterval));
3296 if (!source->internal && source->marked_bye) {
3297 /* if we received a BYE from the source, remove the source after some
3299 if (data->current_time > source->bye_time &&
3300 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3301 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3307 if (source->internal && source->sent_bye) {
3308 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3312 /* sources that were inactive for more than 5 times the deterministic reporting
3313 * interval get timed out. the min timeout is 5 seconds. */
3314 /* mind old time that might pre-date last time going to PLAYING */
3315 btime = MAX (source->last_activity, sess->start_time);
3316 if (data->current_time > btime) {
3317 interval = MAX (binterval * 5, 5 * GST_SECOND);
3318 if (data->current_time - btime > interval) {
3319 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3320 source->ssrc, GST_TIME_ARGS (btime));
3321 if (source->internal) {
3322 /* this is an internal source that is not using our suggested ssrc.
3323 * since there must be another source using this ssrc, we can remove
3324 * this one instead of making it a receiver forever */
3325 if (source->ssrc != sess->suggested_ssrc) {
3326 rtp_source_mark_bye (source, "timed out");
3327 /* do not schedule bye here, since we are inside the RTCP timeout
3328 * processing and scheduling bye will interfere with SR/RR sending */
3336 /* senders that did not send for a long time become a receiver, this also
3337 * holds for our own sources. */
3339 /* mind old time that might pre-date last time going to PLAYING */
3340 btime = MAX (source->last_rtp_activity, sess->start_time);
3341 if (data->current_time > btime) {
3342 interval = MAX (binterval * 2, 5 * GST_SECOND);
3343 if (data->current_time - btime > interval) {
3344 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3345 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3346 sendertimeout = TRUE;
3352 sess->total_sources--;
3354 sess->stats.sender_sources--;
3355 if (source->internal)
3356 sess->stats.internal_sender_sources--;
3359 sess->stats.active_sources--;
3361 if (source->internal)
3362 sess->stats.internal_sources--;
3365 on_bye_timeout (sess, source);
3367 on_timeout (sess, source);
3369 if (sendertimeout) {
3370 source->is_sender = FALSE;
3371 sess->stats.sender_sources--;
3372 if (source->internal)
3373 sess->stats.internal_sender_sources--;
3375 on_sender_timeout (sess, source);
3377 /* count how many source to report in this generation */
3378 if (((gint16) (source->generation - sess->generation)) <= 0)
3379 data->num_to_report++;
3381 source->closing = remove;
3385 session_sdes (RTPSession * sess, ReportData * data)
3387 GstRTCPPacket *packet = &data->packet;
3388 const GstStructure *sdes;
3390 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3392 /* add SDES packet */
3393 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3395 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3397 sdes = rtp_source_get_sdes_struct (data->source);
3399 /* add all fields in the structure, the order is not important. */
3400 n_fields = gst_structure_n_fields (sdes);
3401 for (i = 0; i < n_fields; ++i) {
3404 GstRTCPSDESType type;
3406 field = gst_structure_nth_field_name (sdes, i);
3409 value = gst_structure_get_string (sdes, field);
3412 type = gst_rtcp_sdes_name_to_type (field);
3414 /* Early packets are minimal and only include the CNAME */
3415 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3418 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3419 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3420 (const guint8 *) value);
3421 } else if (type == GST_RTCP_SDES_PRIV) {
3427 /* don't accept entries that are too big */
3428 prefix_len = strlen (field);
3429 if (prefix_len > 255)
3431 value_len = strlen (value);
3432 if (value_len > 255)
3434 data_len = 1 + prefix_len + value_len;
3438 data[0] = prefix_len;
3439 memcpy (&data[1], field, prefix_len);
3440 memcpy (&data[1 + prefix_len], value, value_len);
3442 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3446 data->has_sdes = TRUE;
3449 /* schedule a BYE packet */
3451 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3453 GstRTCPPacket *packet = &data->packet;
3454 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3457 session_sdes (sess, data);
3458 /* add a BYE packet */
3459 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3460 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3461 if (source->bye_reason)
3462 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3464 /* we have a BYE packet now */
3465 source->sent_bye = TRUE;
3469 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3471 GstClockTime new_send_time;
3472 GstClockTime interval;
3473 RTPSessionStats *stats;
3475 if (sess->scheduled_bye)
3476 stats = &sess->bye_stats;
3478 stats = &sess->stats;
3480 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3481 data->is_early = TRUE;
3483 data->is_early = FALSE;
3485 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3486 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3487 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3488 GST_TIME_ARGS (current_time));
3492 /* no need to check yet */
3493 if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3494 sess->next_rtcp_check_time > current_time) {
3495 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3496 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3497 GST_TIME_ARGS (current_time));
3503 /* take interval and add jitter */
3504 interval = data->interval;
3505 if (interval != GST_CLOCK_TIME_NONE)
3506 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3508 if (sess->last_rtcp_send_time != GST_CLOCK_TIME_NONE) {
3509 /* perform forward reconsideration */
3510 if (interval != GST_CLOCK_TIME_NONE) {
3511 GstClockTime elapsed;
3513 /* get elapsed time since we last reported */
3514 elapsed = current_time - sess->last_rtcp_send_time;
3516 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3517 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3518 new_send_time = interval + sess->last_rtcp_send_time;
3520 new_send_time = sess->last_rtcp_send_time;
3523 /* If this is the first RTCP packet, we can reconsider anything based
3524 * on the last RTCP send time because there was none.
3526 g_warn_if_fail (!data->is_early);
3527 data->is_early = FALSE;
3528 new_send_time = current_time;
3531 if (!data->is_early) {
3532 /* check if reconsideration */
3533 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3534 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3535 GST_TIME_ARGS (new_send_time));
3536 /* store new check time */
3537 sess->next_rtcp_check_time = new_send_time;
3540 sess->next_rtcp_check_time = current_time + interval;
3541 } else if (interval != GST_CLOCK_TIME_NONE) {
3542 /* Apply the rules from RFC 4585 section 3.5.3 */
3543 if (stats->min_interval != 0 && !sess->first_rtcp) {
3544 GstClockTime T_rr_current_interval =
3545 g_random_double_range (0.5, 1.5) * stats->min_interval;
3547 /* This will caused the RTCP to be suppressed if no FB packets are added */
3548 if (sess->last_rtcp_send_time + T_rr_current_interval > new_send_time) {
3549 GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
3550 " last: %" GST_TIME_FORMAT
3551 " + T_rr_current_interval: %" GST_TIME_FORMAT
3552 " > new_send_time: %" GST_TIME_FORMAT,
3553 GST_TIME_ARGS (stats->min_interval),
3554 GST_TIME_ARGS (sess->last_rtcp_send_time),
3555 GST_TIME_ARGS (T_rr_current_interval),
3556 GST_TIME_ARGS (new_send_time));
3557 data->may_suppress = TRUE;
3562 GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
3563 GST_TIME_ARGS (new_send_time));
3569 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3571 g_hash_table_insert (hash_table, key, g_object_ref (source));
3575 remove_closing_sources (const gchar * key, RTPSource * source,
3578 if (source->closing)
3581 if (source->send_fir)
3582 data->have_fir = TRUE;
3583 if (source->send_pli)
3584 data->have_pli = TRUE;
3585 if (source->send_nack)
3586 data->have_nack = TRUE;
3592 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3594 RTPSession *sess = data->sess;
3595 gboolean is_bye = FALSE;
3596 ReportOutput *output;
3598 /* only generate RTCP for active internal sources */
3599 if (!source->internal || source->sent_bye)
3602 /* ignore other sources when we do the timeout after a scheduled BYE */
3603 if (sess->scheduled_bye && !source->marked_bye)
3606 data->source = source;
3609 session_start_rtcp (sess, data);
3611 if (source->marked_bye) {
3613 make_source_bye (sess, source, data);
3615 } else if (!data->is_early) {
3616 /* loop over all known sources and add report blocks. If we are early, we
3617 * just make a minimal RTCP packet and skip this step */
3618 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3619 (GHFunc) session_report_blocks, data);
3621 if (!data->has_sdes)
3622 session_sdes (sess, data);
3625 session_fir (sess, data);
3628 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3629 (GHFunc) session_pli, data);
3631 if (data->have_nack)
3632 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3633 (GHFunc) session_nack, data);
3635 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3637 output = g_slice_new (ReportOutput);
3638 output->source = g_object_ref (source);
3639 output->is_bye = is_bye;
3640 output->buffer = data->rtcp;
3641 /* queue the RTCP packet to push later */
3642 g_queue_push_tail (&data->output, output);
3646 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3648 RTPSession *sess = data->sess;
3650 if (g_hash_table_size (source->reported_in_sr_of) >=
3651 sess->stats.internal_sources) {
3652 /* source is reported, move to next generation */
3653 source->generation = sess->generation + 1;
3654 g_hash_table_remove_all (source->reported_in_sr_of);
3656 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3657 source->generation);
3659 /* if we reported all sources in this generation, move to next */
3660 if (--data->num_to_report == 0) {
3662 GST_DEBUG ("all reported, generation now %u", sess->generation);
3668 * rtp_session_on_timeout:
3669 * @sess: an #RTPSession
3670 * @current_time: the current system time
3671 * @ntpnstime: the current NTP time in nanoseconds
3672 * @running_time: the current running_time of the pipeline
3674 * Perform maintenance actions after the timeout obtained with
3675 * rtp_session_next_timeout() expired.
3677 * This function will perform timeouts of receivers and senders, send a BYE
3678 * packet or generate RTCP packets with current session stats.
3680 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3681 * times, for each packet that should be processed.
3683 * Returns: a #GstFlowReturn.
3686 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3687 guint64 ntpnstime, GstClockTime running_time)
3689 GstFlowReturn result = GST_FLOW_OK;
3690 ReportData data = { GST_RTCP_BUFFER_INIT };
3691 GHashTable *table_copy;
3692 ReportOutput *output;
3694 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3696 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3697 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3698 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3701 data.current_time = current_time;
3702 data.ntpnstime = ntpnstime;
3703 data.running_time = running_time;
3704 data.num_to_report = 0;
3705 data.may_suppress = FALSE;
3706 data.nacked_seqnums = 0;
3707 g_queue_init (&data.output);
3709 RTP_SESSION_LOCK (sess);
3710 /* get a new interval, we need this for various cleanups etc */
3711 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3713 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3715 /* we need an internal source now */
3716 if (sess->stats.internal_sources == 0) {
3720 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
3722 g_object_unref (source);
3725 sess->conflicting_addresses =
3726 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
3728 /* Make a local copy of the hashtable. We need to do this because the
3729 * cleanup stage below releases the session lock. */
3730 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
3731 (GDestroyNotify) g_object_unref);
3732 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3733 (GHFunc) clone_ssrcs_hashtable, table_copy);
3735 /* Clean up the session, mark the source for removing, this might release the
3737 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
3738 g_hash_table_destroy (table_copy);
3740 /* Now remove the marked sources */
3741 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
3742 (GHRFunc) remove_closing_sources, &data);
3744 /* update point-to-point status */
3745 session_update_ptp (sess);
3747 /* see if we need to generate SR or RR packets */
3748 if (!is_rtcp_time (sess, current_time, &data))
3751 GST_DEBUG ("doing RTCP generation %u for %u sources, early %d",
3752 sess->generation, data.num_to_report, data.is_early);
3754 /* generate RTCP for all internal sources */
3755 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3756 (GHFunc) generate_rtcp, &data);
3758 /* update the generation for all the sources that have been reported */
3759 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3760 (GHFunc) update_generation, &data);
3762 /* we keep track of the last report time in order to timeout inactive
3763 * receivers or senders */
3764 if (!data.is_early && !data.may_suppress)
3765 sess->last_rtcp_send_time = data.current_time;
3766 sess->first_rtcp = FALSE;
3767 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
3768 sess->scheduled_bye = FALSE;
3770 /* RFC 4585 section 3.5.2 step 6 */
3771 if (!data.is_early) {
3772 sess->allow_early = TRUE;
3776 RTP_SESSION_UNLOCK (sess);
3778 /* push out the RTCP packets */
3779 while ((output = g_queue_pop_head (&data.output))) {
3780 gboolean do_not_suppress;
3781 GstBuffer *buffer = output->buffer;
3782 RTPSource *source = output->source;
3784 /* Give the user a change to add its own packet */
3785 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
3786 buffer, data.is_early, &do_not_suppress);
3788 if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
3791 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
3793 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
3794 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
3795 sess->stats.avg_rtcp_packet_size, packet_size);
3797 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
3798 sess->send_rtcp_user_data);
3799 sess->stats.nacks_sent += data.nacked_seqnums;
3801 GST_DEBUG ("freeing packet callback: %p"
3802 " do_not_suppress: %d may_suppress: %d",
3803 sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
3804 sess->stats.nacks_dropped += data.nacked_seqnums;
3805 gst_buffer_unref (buffer);
3807 g_object_unref (source);
3808 g_slice_free (ReportOutput, output);
3814 * rtp_session_request_early_rtcp:
3815 * @sess: an #RTPSession
3816 * @current_time: the current system time
3817 * @max_delay: maximum delay
3819 * Request transmission of early RTCP
3821 * Returns: %TRUE if the related RTCP can be scheduled.
3824 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
3825 GstClockTime max_delay)
3827 GstClockTime T_dither_max, T_rr;
3830 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
3832 RTP_SESSION_LOCK (sess);
3834 /* Check if already requested */
3835 /* RFC 4585 section 3.5.2 step 2 */
3836 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3837 GST_LOG_OBJECT (sess, "already have next early rtcp time");
3842 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
3843 GST_LOG_OBJECT (sess, "no next RTCP check time");
3848 /* RFC 4585 section 3.5.3 step 1
3849 * If no regular RTCP packet has been sent before, then a regular
3850 * RTCP packet has to be scheduled first and FB messages might be
3853 if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
3854 GST_LOG_OBJECT (sess, "no RTCP sent yet");
3856 if (current_time + max_delay > sess->next_rtcp_check_time) {
3857 GST_LOG_OBJECT (sess,
3858 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
3859 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3860 GST_TIME_ARGS (max_delay),
3861 GST_TIME_ARGS (sess->next_rtcp_check_time));
3869 T_rr = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
3871 /* RFC 4585 section 3.5.2 step 2b */
3872 /* If the total sources is <=2, then there is only us and one peer */
3873 /* When there is one auxiliary stream the session can still do point
3876 if (sess->is_doing_ptp) {
3879 /* Divide by 2 because l = 0.5 */
3880 T_dither_max = T_rr;
3884 /* RFC 4585 section 3.5.2 step 3 */
3885 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
3886 GST_LOG_OBJECT (sess, "don't send because of dither");
3891 /* RFC 4585 section 3.5.2 step 4a */
3892 if (sess->allow_early == FALSE) {
3893 /* Ignore the request a scheduled packet will be in time anyway */
3894 if (current_time + max_delay > sess->next_rtcp_check_time) {
3895 GST_LOG_OBJECT (sess,
3896 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
3897 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3898 GST_TIME_ARGS (max_delay),
3899 GST_TIME_ARGS (sess->next_rtcp_check_time));
3902 GST_LOG_OBJECT (sess, "can't allow early feedback");
3908 /* RFC 4585 section 3.5.2 step 4b */
3910 /* Schedule an early transmission later */
3911 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
3914 /* If no dithering, schedule it for NOW */
3915 sess->next_early_rtcp_time = current_time;
3918 /* RFC 4585 section 3.5.2 step 6 */
3919 sess->allow_early = FALSE;
3920 /* Delay next regular RTCP packet to not exceed the short-term
3921 * RTCP bandwidth when using early feedback as compared to
3923 sess->next_rtcp_check_time = sess->last_rtcp_send_time + 2 * T_rr;
3924 sess->last_rtcp_send_time += T_rr;
3926 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT,
3927 GST_TIME_ARGS (sess->next_early_rtcp_time));
3928 RTP_SESSION_UNLOCK (sess);
3930 /* notify app of need to send packet early
3931 * and therefore of timeout change */
3932 if (sess->callbacks.reconsider)
3933 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3939 RTP_SESSION_UNLOCK (sess);
3945 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
3949 if (!sess->callbacks.send_rtcp)
3952 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
3954 return rtp_session_request_early_rtcp (sess, now, max_delay);
3958 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
3959 gboolean fir, gint count)
3963 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
3964 GST_DEBUG ("FIR/PLI not sent");
3968 RTP_SESSION_LOCK (sess);
3969 src = find_source (sess, ssrc);
3974 src->send_pli = FALSE;
3975 src->send_fir = TRUE;
3977 if (count == -1 || count != src->last_fir_count)
3978 src->current_send_fir_seqnum++;
3979 src->last_fir_count = count;
3980 } else if (!src->send_fir) {
3981 src->send_pli = TRUE;
3983 RTP_SESSION_UNLOCK (sess);
3990 RTP_SESSION_UNLOCK (sess);
3996 * rtp_session_request_nack:
3997 * @sess: a #RTPSession
3999 * @seqnum: the missing seqnum
4000 * @max_delay: max delay to request NACK
4002 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
4004 * Returns: %TRUE if the NACK feedback could be scheduled
4007 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
4008 GstClockTime max_delay)
4012 if (!rtp_session_send_rtcp (sess, max_delay)) {
4013 GST_DEBUG ("NACK not sent");
4017 RTP_SESSION_LOCK (sess);
4018 source = find_source (sess, ssrc);
4022 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
4023 rtp_source_register_nack (source, seqnum);
4024 RTP_SESSION_UNLOCK (sess);
4031 RTP_SESSION_UNLOCK (sess);