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,
51 SIGNAL_ON_FEEDBACK_RTCP,
53 SIGNAL_SEND_RTCP_FULL,
54 SIGNAL_ON_RECEIVING_RTCP,
55 SIGNAL_ON_NEW_SENDER_SSRC,
56 SIGNAL_ON_SENDER_SSRC_ACTIVE,
60 #define DEFAULT_INTERNAL_SOURCE NULL
61 #define DEFAULT_BANDWIDTH 0.0
62 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
63 #define DEFAULT_RTCP_RR_BANDWIDTH -1
64 #define DEFAULT_RTCP_RS_BANDWIDTH -1
65 #define DEFAULT_RTCP_MTU 1400
66 #define DEFAULT_SDES NULL
67 #define DEFAULT_NUM_SOURCES 0
68 #define DEFAULT_NUM_ACTIVE_SOURCES 0
69 #define DEFAULT_SOURCES NULL
70 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
71 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
72 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
73 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
74 #define DEFAULT_MAX_DROPOUT_TIME 60000
75 #define DEFAULT_MAX_MISORDER_TIME 2000
76 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
77 #define DEFAULT_RTCP_REDUCED_SIZE FALSE
86 PROP_RTCP_RR_BANDWIDTH,
87 PROP_RTCP_RS_BANDWIDTH,
91 PROP_NUM_ACTIVE_SOURCES,
94 PROP_RTCP_MIN_INTERVAL,
95 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
96 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
98 PROP_MAX_DROPOUT_TIME,
99 PROP_MAX_MISORDER_TIME,
102 PROP_RTCP_REDUCED_SIZE
105 /* update average packet size */
106 #define INIT_AVG(avg, val) \
108 #define UPDATE_AVG(avg, val) \
112 (avg) = ((val) + (15 * (avg))) >> 4;
115 /* GObject vmethods */
116 static void rtp_session_finalize (GObject * object);
117 static void rtp_session_set_property (GObject * object, guint prop_id,
118 const GValue * value, GParamSpec * pspec);
119 static void rtp_session_get_property (GObject * object, guint prop_id,
120 GValue * value, GParamSpec * pspec);
122 static gboolean rtp_session_send_rtcp (RTPSession * sess,
123 GstClockTime max_delay);
125 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
127 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
129 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
130 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
131 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
132 static RTPSource *obtain_internal_source (RTPSession * sess,
133 guint32 ssrc, gboolean * created, GstClockTime current_time);
134 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
135 GstClockTime current_time);
136 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
137 gboolean deterministic, gboolean first);
140 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
141 const GValue * handler_return, gpointer data)
143 if (g_value_get_boolean (handler_return))
144 g_value_set_boolean (return_accu, TRUE);
150 rtp_session_class_init (RTPSessionClass * klass)
152 GObjectClass *gobject_class;
154 gobject_class = (GObjectClass *) klass;
156 gobject_class->finalize = rtp_session_finalize;
157 gobject_class->set_property = rtp_session_set_property;
158 gobject_class->get_property = rtp_session_get_property;
161 * RTPSession::get-source-by-ssrc:
162 * @session: the object which received the signal
163 * @ssrc: the SSRC of the RTPSource
165 * Request the #RTPSource object with SSRC @ssrc in @session.
167 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
168 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
169 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
170 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
171 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
174 * RTPSession::on-new-ssrc:
175 * @session: the object which received the signal
176 * @src: the new RTPSource
178 * Notify of a new SSRC that entered @session.
180 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
181 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
182 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
183 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
186 * RTPSession::on-ssrc-collision:
187 * @session: the object which received the signal
188 * @src: the #RTPSource that caused a collision
190 * Notify when we have an SSRC collision
192 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
193 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
194 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
195 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
198 * RTPSession::on-ssrc-validated:
199 * @session: the object which received the signal
200 * @src: the new validated RTPSource
202 * Notify of a new SSRC that became validated.
204 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
205 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
206 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
207 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
210 * RTPSession::on-ssrc-active:
211 * @session: the object which received the signal
212 * @src: the active RTPSource
214 * Notify of a SSRC that is active, i.e., sending RTCP.
216 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
217 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
218 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
219 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
222 * RTPSession::on-ssrc-sdes:
223 * @session: the object which received the signal
224 * @src: the RTPSource
226 * Notify that a new SDES was received for SSRC.
228 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
229 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
230 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
231 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
234 * RTPSession::on-bye-ssrc:
235 * @session: the object which received the signal
236 * @src: the RTPSource that went away
238 * Notify of an SSRC that became inactive because of a BYE packet.
240 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
241 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
242 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
243 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
246 * RTPSession::on-bye-timeout:
247 * @session: the object which received the signal
248 * @src: the RTPSource that timed out
250 * Notify of an SSRC that has timed out because of BYE
252 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
253 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
254 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
255 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
258 * RTPSession::on-timeout:
259 * @session: the object which received the signal
260 * @src: the RTPSource that timed out
262 * Notify of an SSRC that has timed out
264 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
265 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
266 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
267 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
270 * RTPSession::on-sender-timeout:
271 * @session: the object which received the signal
272 * @src: the RTPSource that timed out
274 * Notify of an SSRC that was a sender but timed out and became a receiver.
276 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
277 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
278 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
279 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
283 * RTPSession::on-sending-rtcp
284 * @session: the object which received the signal
285 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
286 * @early: %TRUE if the packet is early, %FALSE if it is regular
288 * This signal is emitted before sending an RTCP packet, it can be used
289 * to add extra RTCP Packets.
291 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
292 * if suppressing it is acceptable
294 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
295 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
296 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
297 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
298 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
301 * RTPSession::on-app-rtcp:
302 * @session: the object which received the signal
303 * @subtype: The subtype of the packet
304 * @ssrc: The SSRC/CSRC of the packet
305 * @name: The name of the packet
306 * @data: a #GstBuffer with the application-dependant data or %NULL if
309 * Notify that a RTCP APP packet has been received
311 rtp_session_signals[SIGNAL_ON_APP_RTCP] =
312 g_signal_new ("on-app-rtcp", G_TYPE_FROM_CLASS (klass),
313 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_app_rtcp),
314 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 4,
315 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_STRING, GST_TYPE_BUFFER);
318 * RTPSession::on-feedback-rtcp:
319 * @session: the object which received the signal
320 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
321 * %GST_RTCP_TYPE_RTPFB
322 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
323 * @sender_ssrc: The SSRC of the sender
324 * @media_ssrc: The SSRC of the media this refers to
325 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
328 * Notify that a RTCP feedback packet has been received
330 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
331 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
332 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
333 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
334 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
337 * RTPSession::send-rtcp:
338 * @session: the object which received the signal
339 * @max_delay: The maximum delay after which the feedback will not be useful
342 * Requests that the #RTPSession initiate a new RTCP packet as soon as
343 * possible within the requested delay.
345 * This sets feedback to %TRUE if not already done before.
347 rtp_session_signals[SIGNAL_SEND_RTCP] =
348 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
349 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
350 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
351 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
354 * RTPSession::send-rtcp-full:
355 * @session: the object which received the signal
356 * @max_delay: The maximum delay after which the feedback will not be useful
359 * Requests that the #RTPSession initiate a new RTCP packet as soon as
360 * possible within the requested delay.
362 * This sets feedback to %TRUE if not already done before.
364 * Returns: TRUE if the new RTCP packet could be scheduled within the
365 * requested delay, FALSE otherwise.
369 rtp_session_signals[SIGNAL_SEND_RTCP_FULL] =
370 g_signal_new ("send-rtcp-full", G_TYPE_FROM_CLASS (klass),
371 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
372 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
373 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
376 * RTPSession::on-receiving-rtcp
377 * @session: the object which received the signal
378 * @buffer: the #GstBuffer containing the RTCP packet that was received
380 * This signal is emitted when receiving an RTCP packet before it is handled
381 * by the session. It can be used to extract custom information from RTCP packets.
385 rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP] =
386 g_signal_new ("on-receiving-rtcp", G_TYPE_FROM_CLASS (klass),
387 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_receiving_rtcp),
388 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
389 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
392 * RTPSession::on-new-sender-ssrc:
393 * @session: the object which received the signal
394 * @src: the new sender RTPSource
396 * Notify of a new sender SSRC that entered @session.
400 rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
401 g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
402 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_sender_ssrc),
403 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
407 * RTPSession::on-sender-ssrc-active:
408 * @session: the object which received the signal
409 * @src: the active sender RTPSource
411 * Notify of a sender SSRC that is active, i.e., sending RTCP.
415 rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
416 g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
417 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass,
418 on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__OBJECT,
419 G_TYPE_NONE, 1, RTP_TYPE_SOURCE);
421 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
422 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
423 "The internal SSRC used for the session (deprecated)",
424 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
426 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
427 g_param_spec_object ("internal-source", "Internal Source",
428 "The internal source element of the session (deprecated)",
429 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
431 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
432 g_param_spec_double ("bandwidth", "Bandwidth",
433 "The bandwidth of the session in bits per second (0 for auto-discover)",
434 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
435 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
437 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
438 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
439 "The fraction of the bandwidth used for RTCP in bits per second (or as a real fraction of the RTP bandwidth if < 1)",
440 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
441 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
443 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
444 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
445 "The RTCP bandwidth used for receivers in bits per second (-1 = default)",
446 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
447 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
449 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
450 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
451 "The RTCP bandwidth used for senders in bits per second (-1 = default)",
452 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
453 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
455 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
456 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
457 "The maximum size of the RTCP packets",
458 16, G_MAXINT16, DEFAULT_RTCP_MTU,
459 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
461 g_object_class_install_property (gobject_class, PROP_SDES,
462 g_param_spec_boxed ("sdes", "SDES",
463 "The SDES items of this session",
464 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
466 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
467 g_param_spec_uint ("num-sources", "Num Sources",
468 "The number of sources in the session", 0, G_MAXUINT,
469 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
471 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
472 g_param_spec_uint ("num-active-sources", "Num Active Sources",
473 "The number of active sources in the session", 0, G_MAXUINT,
474 DEFAULT_NUM_ACTIVE_SOURCES,
475 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
479 * Get a GValue Array of all sources in the session.
482 * <title>Getting the #RTPSources of a session
489 * g_object_get (sess, "sources", &arr, NULL);
491 * for (i = 0; i < arr->n_values; i++) {
494 * val = g_value_array_get_nth (arr, i);
495 * source = g_value_get_object (val);
497 * g_value_array_free (arr);
502 g_object_class_install_property (gobject_class, PROP_SOURCES,
503 g_param_spec_boxed ("sources", "Sources",
504 "An array of all known sources in the session",
505 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
507 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
508 g_param_spec_boolean ("favor-new", "Favor new sources",
509 "Resolve SSRC conflict in favor of new sources", FALSE,
510 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
512 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
513 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
514 "Minimum interval between Regular RTCP packet (in ns)",
515 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
516 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
518 g_object_class_install_property (gobject_class,
519 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
520 g_param_spec_uint64 ("rtcp-feedback-retention-window",
521 "RTCP Feedback retention window",
522 "Duration during which RTCP Feedback packets are retained (in ns)",
523 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
524 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
526 g_object_class_install_property (gobject_class,
527 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
528 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
529 "RTCP Immediate Feedback threshold",
530 "The maximum number of members of a RTP session for which immediate"
531 " feedback is used (DEPRECATED: has no effect and is not needed)",
532 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
533 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
535 g_object_class_install_property (gobject_class, PROP_PROBATION,
536 g_param_spec_uint ("probation", "Number of probations",
537 "Consecutive packet sequence numbers to accept the source",
538 0, G_MAXUINT, DEFAULT_PROBATION,
539 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
541 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
542 g_param_spec_uint ("max-dropout-time", "Max dropout time",
543 "The maximum time (milliseconds) of missing packets tolerated.",
544 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
545 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
547 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
548 g_param_spec_uint ("max-misorder-time", "Max misorder time",
549 "The maximum time (milliseconds) of misordered packets tolerated.",
550 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
551 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
556 * Various session statistics. This property returns a GstStructure
557 * with name application/x-rtp-session-stats with the following fields:
559 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
560 * dropped (due to bandwidth constraints)
561 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
562 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
563 * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
564 * RTP sources (Since 1.8)
568 g_object_class_install_property (gobject_class, PROP_STATS,
569 g_param_spec_boxed ("stats", "Statistics",
570 "Various statistics", GST_TYPE_STRUCTURE,
571 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
573 g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
574 g_param_spec_enum ("rtp-profile", "RTP Profile",
575 "RTP profile to use for this session", GST_TYPE_RTP_PROFILE,
576 DEFAULT_RTP_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
578 g_object_class_install_property (gobject_class, PROP_RTCP_REDUCED_SIZE,
579 g_param_spec_boolean ("rtcp-reduced-size", "RTCP Reduced Size",
580 "Use Reduced Size RTCP for feedback packets",
581 DEFAULT_RTCP_REDUCED_SIZE,
582 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
584 klass->get_source_by_ssrc =
585 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
586 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
588 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
592 rtp_session_init (RTPSession * sess)
597 g_mutex_init (&sess->lock);
598 sess->key = g_random_int ();
602 /* TODO: We currently only use the first hash table but this is the
603 * beginning of an implementation for RFC2762
604 for (i = 0; i < 32; i++) {
606 for (i = 0; i < 1; i++) {
608 g_hash_table_new_full (NULL, NULL, NULL,
609 (GDestroyNotify) g_object_unref);
612 rtp_stats_init_defaults (&sess->stats);
613 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
614 rtp_stats_set_min_interval (&sess->stats,
615 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
617 sess->recalc_bandwidth = TRUE;
618 sess->bandwidth = DEFAULT_BANDWIDTH;
619 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
620 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
621 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
623 /* default UDP header length */
624 sess->header_len = 28;
625 sess->mtu = DEFAULT_RTCP_MTU;
627 sess->probation = DEFAULT_PROBATION;
628 sess->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
629 sess->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
631 /* some default SDES entries */
632 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
634 /* we do not want to leak details like the username or hostname here */
635 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
636 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
640 /* we do not want to leak the user's real name here */
641 str = g_strdup_printf ("Anon%u", g_random_int ());
642 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
646 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
648 /* this is the SSRC we suggest */
649 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
650 sess->internal_ssrc_set = FALSE;
652 sess->first_rtcp = TRUE;
653 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
654 sess->last_rtcp_check_time = GST_CLOCK_TIME_NONE;
655 sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
656 sess->last_rtcp_interval = GST_CLOCK_TIME_NONE;
658 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
659 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
660 sess->rtcp_immediate_feedback_threshold =
661 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
662 sess->rtp_profile = DEFAULT_RTP_PROFILE;
663 sess->reduced_size_rtcp = DEFAULT_RTCP_REDUCED_SIZE;
665 sess->last_keyframe_request = GST_CLOCK_TIME_NONE;
667 sess->is_doing_ptp = TRUE;
671 rtp_session_finalize (GObject * object)
676 sess = RTP_SESSION_CAST (object);
678 gst_structure_free (sess->sdes);
680 g_list_free_full (sess->conflicting_addresses,
681 (GDestroyNotify) rtp_conflicting_address_free);
683 /* TODO: Change this again when implementing RFC 2762
684 * for (i = 0; i < 32; i++)
686 for (i = 0; i < 1; i++)
687 g_hash_table_destroy (sess->ssrcs[i]);
689 g_mutex_clear (&sess->lock);
691 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
695 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
697 GValue value = { 0 };
699 g_value_init (&value, RTP_TYPE_SOURCE);
700 g_value_take_object (&value, source);
701 /* copies the value */
702 g_value_array_append (arr, &value);
706 rtp_session_create_sources (RTPSession * sess)
711 RTP_SESSION_LOCK (sess);
712 /* get number of elements in the table */
713 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
714 /* create the result value array */
715 res = g_value_array_new (size);
717 /* and copy all values into the array */
718 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
719 RTP_SESSION_UNLOCK (sess);
725 create_source_stats (gpointer key, RTPSource * source, GValueArray * arr)
727 GValue value = G_VALUE_INIT;
730 g_object_get (source, "stats", &s, NULL);
732 g_value_init (&value, GST_TYPE_STRUCTURE);
733 gst_value_set_structure (&value, s);
734 g_value_array_append (arr, &value);
735 gst_structure_free (s);
736 g_value_unset (&value);
739 static GstStructure *
740 rtp_session_create_stats (RTPSession * sess)
743 GValueArray *source_stats;
744 GValue source_stats_v = G_VALUE_INIT;
747 RTP_SESSION_LOCK (sess);
748 s = gst_structure_new ("application/x-rtp-session-stats",
749 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
750 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
751 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
753 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
754 source_stats = g_value_array_new (size);
755 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
756 (GHFunc) create_source_stats, source_stats);
757 RTP_SESSION_UNLOCK (sess);
759 g_value_init (&source_stats_v, G_TYPE_VALUE_ARRAY);
760 g_value_take_boxed (&source_stats_v, source_stats);
761 gst_structure_take_value (s, "source-stats", &source_stats_v);
767 rtp_session_set_property (GObject * object, guint prop_id,
768 const GValue * value, GParamSpec * pspec)
772 sess = RTP_SESSION (object);
775 case PROP_INTERNAL_SSRC:
776 RTP_SESSION_LOCK (sess);
777 sess->suggested_ssrc = g_value_get_uint (value);
778 sess->internal_ssrc_set = TRUE;
779 sess->internal_ssrc_from_caps_or_property = TRUE;
780 RTP_SESSION_UNLOCK (sess);
781 if (sess->callbacks.reconfigure)
782 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
785 RTP_SESSION_LOCK (sess);
786 sess->bandwidth = g_value_get_double (value);
787 sess->recalc_bandwidth = TRUE;
788 RTP_SESSION_UNLOCK (sess);
790 case PROP_RTCP_FRACTION:
791 RTP_SESSION_LOCK (sess);
792 sess->rtcp_bandwidth = g_value_get_double (value);
793 sess->recalc_bandwidth = TRUE;
794 RTP_SESSION_UNLOCK (sess);
796 case PROP_RTCP_RR_BANDWIDTH:
797 RTP_SESSION_LOCK (sess);
798 sess->rtcp_rr_bandwidth = g_value_get_int (value);
799 sess->recalc_bandwidth = TRUE;
800 RTP_SESSION_UNLOCK (sess);
802 case PROP_RTCP_RS_BANDWIDTH:
803 RTP_SESSION_LOCK (sess);
804 sess->rtcp_rs_bandwidth = g_value_get_int (value);
805 sess->recalc_bandwidth = TRUE;
806 RTP_SESSION_UNLOCK (sess);
809 sess->mtu = g_value_get_uint (value);
812 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
815 sess->favor_new = g_value_get_boolean (value);
817 case PROP_RTCP_MIN_INTERVAL:
818 rtp_stats_set_min_interval (&sess->stats,
819 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
820 /* trigger reconsideration */
821 RTP_SESSION_LOCK (sess);
822 sess->next_rtcp_check_time = 0;
823 RTP_SESSION_UNLOCK (sess);
824 if (sess->callbacks.reconsider)
825 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
827 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
828 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
831 sess->probation = g_value_get_uint (value);
833 case PROP_MAX_DROPOUT_TIME:
834 sess->max_dropout_time = g_value_get_uint (value);
836 case PROP_MAX_MISORDER_TIME:
837 sess->max_misorder_time = g_value_get_uint (value);
839 case PROP_RTP_PROFILE:
840 sess->rtp_profile = g_value_get_enum (value);
841 /* trigger reconsideration */
842 RTP_SESSION_LOCK (sess);
843 sess->next_rtcp_check_time = 0;
844 RTP_SESSION_UNLOCK (sess);
845 if (sess->callbacks.reconsider)
846 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
848 case PROP_RTCP_REDUCED_SIZE:
849 sess->reduced_size_rtcp = g_value_get_boolean (value);
852 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
858 rtp_session_get_property (GObject * object, guint prop_id,
859 GValue * value, GParamSpec * pspec)
863 sess = RTP_SESSION (object);
866 case PROP_INTERNAL_SSRC:
867 g_value_set_uint (value, rtp_session_suggest_ssrc (sess, NULL));
869 case PROP_INTERNAL_SOURCE:
870 /* FIXME, return a random source */
871 g_value_set_object (value, NULL);
874 g_value_set_double (value, sess->bandwidth);
876 case PROP_RTCP_FRACTION:
877 g_value_set_double (value, sess->rtcp_bandwidth);
879 case PROP_RTCP_RR_BANDWIDTH:
880 g_value_set_int (value, sess->rtcp_rr_bandwidth);
882 case PROP_RTCP_RS_BANDWIDTH:
883 g_value_set_int (value, sess->rtcp_rs_bandwidth);
886 g_value_set_uint (value, sess->mtu);
889 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
891 case PROP_NUM_SOURCES:
892 g_value_set_uint (value, rtp_session_get_num_sources (sess));
894 case PROP_NUM_ACTIVE_SOURCES:
895 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
898 g_value_take_boxed (value, rtp_session_create_sources (sess));
901 g_value_set_boolean (value, sess->favor_new);
903 case PROP_RTCP_MIN_INTERVAL:
904 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
906 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
907 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
910 g_value_set_uint (value, sess->probation);
912 case PROP_MAX_DROPOUT_TIME:
913 g_value_set_uint (value, sess->max_dropout_time);
915 case PROP_MAX_MISORDER_TIME:
916 g_value_set_uint (value, sess->max_misorder_time);
919 g_value_take_boxed (value, rtp_session_create_stats (sess));
921 case PROP_RTP_PROFILE:
922 g_value_set_enum (value, sess->rtp_profile);
924 case PROP_RTCP_REDUCED_SIZE:
925 g_value_set_boolean (value, sess->reduced_size_rtcp);
928 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
934 on_new_ssrc (RTPSession * sess, RTPSource * source)
936 g_object_ref (source);
937 RTP_SESSION_UNLOCK (sess);
938 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
939 RTP_SESSION_LOCK (sess);
940 g_object_unref (source);
944 on_ssrc_collision (RTPSession * sess, RTPSource * source)
946 g_object_ref (source);
947 RTP_SESSION_UNLOCK (sess);
948 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
950 RTP_SESSION_LOCK (sess);
951 g_object_unref (source);
955 on_ssrc_validated (RTPSession * sess, RTPSource * source)
957 g_object_ref (source);
958 RTP_SESSION_UNLOCK (sess);
959 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
961 RTP_SESSION_LOCK (sess);
962 g_object_unref (source);
966 on_ssrc_active (RTPSession * sess, RTPSource * source)
968 g_object_ref (source);
969 RTP_SESSION_UNLOCK (sess);
970 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
971 RTP_SESSION_LOCK (sess);
972 g_object_unref (source);
976 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
978 g_object_ref (source);
979 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
980 RTP_SESSION_UNLOCK (sess);
981 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
982 RTP_SESSION_LOCK (sess);
983 g_object_unref (source);
987 on_bye_ssrc (RTPSession * sess, RTPSource * source)
989 g_object_ref (source);
990 RTP_SESSION_UNLOCK (sess);
991 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
992 RTP_SESSION_LOCK (sess);
993 g_object_unref (source);
997 on_bye_timeout (RTPSession * sess, RTPSource * source)
999 g_object_ref (source);
1000 RTP_SESSION_UNLOCK (sess);
1001 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
1002 RTP_SESSION_LOCK (sess);
1003 g_object_unref (source);
1007 on_timeout (RTPSession * sess, RTPSource * source)
1009 g_object_ref (source);
1010 RTP_SESSION_UNLOCK (sess);
1011 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
1012 RTP_SESSION_LOCK (sess);
1013 g_object_unref (source);
1017 on_sender_timeout (RTPSession * sess, RTPSource * source)
1019 g_object_ref (source);
1020 RTP_SESSION_UNLOCK (sess);
1021 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
1023 RTP_SESSION_LOCK (sess);
1024 g_object_unref (source);
1028 on_new_sender_ssrc (RTPSession * sess, RTPSource * source)
1030 g_object_ref (source);
1031 RTP_SESSION_UNLOCK (sess);
1032 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
1034 RTP_SESSION_LOCK (sess);
1035 g_object_unref (source);
1039 on_sender_ssrc_active (RTPSession * sess, RTPSource * source)
1041 g_object_ref (source);
1042 RTP_SESSION_UNLOCK (sess);
1043 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
1045 RTP_SESSION_LOCK (sess);
1046 g_object_unref (source);
1052 * Create a new session object.
1054 * Returns: a new #RTPSession. g_object_unref() after usage.
1057 rtp_session_new (void)
1061 sess = g_object_new (RTP_TYPE_SESSION, NULL);
1067 * rtp_session_set_callbacks:
1068 * @sess: an #RTPSession
1069 * @callbacks: callbacks to configure
1070 * @user_data: user data passed in the callbacks
1072 * Configure a set of callbacks to be notified of actions.
1075 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
1078 g_return_if_fail (RTP_IS_SESSION (sess));
1080 if (callbacks->process_rtp) {
1081 sess->callbacks.process_rtp = callbacks->process_rtp;
1082 sess->process_rtp_user_data = user_data;
1084 if (callbacks->send_rtp) {
1085 sess->callbacks.send_rtp = callbacks->send_rtp;
1086 sess->send_rtp_user_data = user_data;
1088 if (callbacks->send_rtcp) {
1089 sess->callbacks.send_rtcp = callbacks->send_rtcp;
1090 sess->send_rtcp_user_data = user_data;
1092 if (callbacks->sync_rtcp) {
1093 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
1094 sess->sync_rtcp_user_data = user_data;
1096 if (callbacks->clock_rate) {
1097 sess->callbacks.clock_rate = callbacks->clock_rate;
1098 sess->clock_rate_user_data = user_data;
1100 if (callbacks->reconsider) {
1101 sess->callbacks.reconsider = callbacks->reconsider;
1102 sess->reconsider_user_data = user_data;
1104 if (callbacks->request_key_unit) {
1105 sess->callbacks.request_key_unit = callbacks->request_key_unit;
1106 sess->request_key_unit_user_data = user_data;
1108 if (callbacks->request_time) {
1109 sess->callbacks.request_time = callbacks->request_time;
1110 sess->request_time_user_data = user_data;
1112 if (callbacks->notify_nack) {
1113 sess->callbacks.notify_nack = callbacks->notify_nack;
1114 sess->notify_nack_user_data = user_data;
1116 if (callbacks->reconfigure) {
1117 sess->callbacks.reconfigure = callbacks->reconfigure;
1118 sess->reconfigure_user_data = user_data;
1123 * rtp_session_set_process_rtp_callback:
1124 * @sess: an #RTPSession
1125 * @callback: callback to set
1126 * @user_data: user data passed in the callback
1128 * Configure only the process_rtp callback to be notified of the process_rtp action.
1131 rtp_session_set_process_rtp_callback (RTPSession * sess,
1132 RTPSessionProcessRTP callback, gpointer user_data)
1134 g_return_if_fail (RTP_IS_SESSION (sess));
1136 sess->callbacks.process_rtp = callback;
1137 sess->process_rtp_user_data = user_data;
1141 * rtp_session_set_send_rtp_callback:
1142 * @sess: an #RTPSession
1143 * @callback: callback to set
1144 * @user_data: user data passed in the callback
1146 * Configure only the send_rtp callback to be notified of the send_rtp action.
1149 rtp_session_set_send_rtp_callback (RTPSession * sess,
1150 RTPSessionSendRTP callback, gpointer user_data)
1152 g_return_if_fail (RTP_IS_SESSION (sess));
1154 sess->callbacks.send_rtp = callback;
1155 sess->send_rtp_user_data = user_data;
1159 * rtp_session_set_send_rtcp_callback:
1160 * @sess: an #RTPSession
1161 * @callback: callback to set
1162 * @user_data: user data passed in the callback
1164 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
1167 rtp_session_set_send_rtcp_callback (RTPSession * sess,
1168 RTPSessionSendRTCP callback, gpointer user_data)
1170 g_return_if_fail (RTP_IS_SESSION (sess));
1172 sess->callbacks.send_rtcp = callback;
1173 sess->send_rtcp_user_data = user_data;
1177 * rtp_session_set_sync_rtcp_callback:
1178 * @sess: an #RTPSession
1179 * @callback: callback to set
1180 * @user_data: user data passed in the callback
1182 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
1185 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
1186 RTPSessionSyncRTCP callback, gpointer user_data)
1188 g_return_if_fail (RTP_IS_SESSION (sess));
1190 sess->callbacks.sync_rtcp = callback;
1191 sess->sync_rtcp_user_data = user_data;
1195 * rtp_session_set_clock_rate_callback:
1196 * @sess: an #RTPSession
1197 * @callback: callback to set
1198 * @user_data: user data passed in the callback
1200 * Configure only the clock_rate callback to be notified of the clock_rate action.
1203 rtp_session_set_clock_rate_callback (RTPSession * sess,
1204 RTPSessionClockRate callback, gpointer user_data)
1206 g_return_if_fail (RTP_IS_SESSION (sess));
1208 sess->callbacks.clock_rate = callback;
1209 sess->clock_rate_user_data = user_data;
1213 * rtp_session_set_reconsider_callback:
1214 * @sess: an #RTPSession
1215 * @callback: callback to set
1216 * @user_data: user data passed in the callback
1218 * Configure only the reconsider callback to be notified of the reconsider action.
1221 rtp_session_set_reconsider_callback (RTPSession * sess,
1222 RTPSessionReconsider callback, gpointer user_data)
1224 g_return_if_fail (RTP_IS_SESSION (sess));
1226 sess->callbacks.reconsider = callback;
1227 sess->reconsider_user_data = user_data;
1231 * rtp_session_set_request_time_callback:
1232 * @sess: an #RTPSession
1233 * @callback: callback to set
1234 * @user_data: user data passed in the callback
1236 * Configure only the request_time callback
1239 rtp_session_set_request_time_callback (RTPSession * sess,
1240 RTPSessionRequestTime callback, gpointer user_data)
1242 g_return_if_fail (RTP_IS_SESSION (sess));
1244 sess->callbacks.request_time = callback;
1245 sess->request_time_user_data = user_data;
1249 * rtp_session_set_bandwidth:
1250 * @sess: an #RTPSession
1251 * @bandwidth: the bandwidth allocated
1253 * Set the session bandwidth in bits per second.
1256 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1258 g_return_if_fail (RTP_IS_SESSION (sess));
1260 RTP_SESSION_LOCK (sess);
1261 sess->stats.bandwidth = bandwidth;
1262 RTP_SESSION_UNLOCK (sess);
1266 * rtp_session_get_bandwidth:
1267 * @sess: an #RTPSession
1269 * Get the session bandwidth.
1271 * Returns: the session bandwidth.
1274 rtp_session_get_bandwidth (RTPSession * sess)
1278 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1280 RTP_SESSION_LOCK (sess);
1281 result = sess->stats.bandwidth;
1282 RTP_SESSION_UNLOCK (sess);
1288 * rtp_session_set_rtcp_fraction:
1289 * @sess: an #RTPSession
1290 * @bandwidth: the RTCP bandwidth
1292 * Set the bandwidth in bits per second that should be used for RTCP
1296 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1298 g_return_if_fail (RTP_IS_SESSION (sess));
1300 RTP_SESSION_LOCK (sess);
1301 sess->stats.rtcp_bandwidth = bandwidth;
1302 RTP_SESSION_UNLOCK (sess);
1306 * rtp_session_get_rtcp_fraction:
1307 * @sess: an #RTPSession
1309 * Get the session bandwidth used for RTCP.
1311 * Returns: The bandwidth used for RTCP messages.
1314 rtp_session_get_rtcp_fraction (RTPSession * sess)
1318 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1320 RTP_SESSION_LOCK (sess);
1321 result = sess->stats.rtcp_bandwidth;
1322 RTP_SESSION_UNLOCK (sess);
1328 * rtp_session_get_sdes_struct:
1329 * @sess: an #RTSPSession
1331 * Get the SDES data as a #GstStructure
1333 * Returns: a GstStructure with SDES items for @sess. This function returns a
1334 * copy of the SDES structure, use gst_structure_free() after usage.
1337 rtp_session_get_sdes_struct (RTPSession * sess)
1339 GstStructure *result = NULL;
1341 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1343 RTP_SESSION_LOCK (sess);
1345 result = gst_structure_copy (sess->sdes);
1346 RTP_SESSION_UNLOCK (sess);
1352 * rtp_session_set_sdes_struct:
1353 * @sess: an #RTSPSession
1354 * @sdes: a #GstStructure
1356 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1359 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1361 g_return_if_fail (sdes);
1362 g_return_if_fail (RTP_IS_SESSION (sess));
1364 RTP_SESSION_LOCK (sess);
1366 gst_structure_free (sess->sdes);
1367 sess->sdes = gst_structure_copy (sdes);
1368 RTP_SESSION_UNLOCK (sess);
1371 static GstFlowReturn
1372 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1374 GstFlowReturn result = GST_FLOW_OK;
1376 if (source->internal) {
1377 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1379 RTP_SESSION_UNLOCK (session);
1381 if (session->callbacks.send_rtp)
1383 session->callbacks.send_rtp (session, source, data,
1384 session->send_rtp_user_data);
1386 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1389 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1390 RTP_SESSION_UNLOCK (session);
1392 if (session->callbacks.process_rtp)
1394 session->callbacks.process_rtp (session, source,
1395 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1397 gst_buffer_unref (GST_BUFFER_CAST (data));
1399 RTP_SESSION_LOCK (session);
1405 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1409 RTP_SESSION_UNLOCK (session);
1411 if (session->callbacks.clock_rate)
1413 session->callbacks.clock_rate (session, pt,
1414 session->clock_rate_user_data);
1418 RTP_SESSION_LOCK (session);
1420 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1425 static RTPSourceCallbacks callbacks = {
1426 (RTPSourcePushRTP) source_push_rtp,
1427 (RTPSourceClockRate) source_clock_rate,
1432 * rtp_session_find_conflicting_address:
1433 * @session: The session the packet came in
1434 * @address: address to check for
1435 * @time: The time when the packet that is possibly in conflict arrived
1437 * Checks if an address which has a conflict is already known. If it is
1438 * a known conflict, remember the time
1440 * Returns: TRUE if it was a known conflict, FALSE otherwise
1443 rtp_session_find_conflicting_address (RTPSession * session,
1444 GSocketAddress * address, GstClockTime time)
1446 return find_conflicting_address (session->conflicting_addresses, address,
1451 * rtp_session_add_conflicting_address:
1452 * @session: The session the packet came in
1453 * @address: address to remember
1454 * @time: The time when the packet that is in conflict arrived
1456 * Adds a new conflict address
1459 rtp_session_add_conflicting_address (RTPSession * sess,
1460 GSocketAddress * address, GstClockTime time)
1462 sess->conflicting_addresses =
1463 add_conflicting_address (sess->conflicting_addresses, address, time);
1468 check_collision (RTPSession * sess, RTPSource * source,
1469 RTPPacketInfo * pinfo, gboolean rtp)
1473 /* If we have no pinfo address, we can't do collision checking */
1474 if (!pinfo->address)
1477 ssrc = rtp_source_get_ssrc (source);
1479 if (!source->internal) {
1480 GSocketAddress *from;
1482 /* This is not our local source, but lets check if two remote
1485 from = source->rtp_from;
1487 from = source->rtcp_from;
1491 if (__g_socket_address_equal (from, pinfo->address)) {
1492 /* Address is the same */
1495 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1496 if (sess->favor_new) {
1497 if (rtp_source_find_conflicting_address (source,
1498 pinfo->address, pinfo->current_time)) {
1501 buf1 = __g_socket_address_to_string (pinfo->address);
1502 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1510 /* Current address is not a known conflict, lets assume this is
1511 * a new source. Save old address in possible conflict list
1513 rtp_source_add_conflicting_address (source, from,
1514 pinfo->current_time);
1516 buf1 = __g_socket_address_to_string (from);
1517 buf2 = __g_socket_address_to_string (pinfo->address);
1519 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1520 " saving old as known conflict", ssrc, buf1, buf2);
1523 rtp_source_set_rtp_from (source, pinfo->address);
1525 rtp_source_set_rtcp_from (source, pinfo->address);
1533 /* Don't need to save old addresses, we ignore new sources */
1538 /* We don't already have a from address for RTP, just set it */
1540 rtp_source_set_rtp_from (source, pinfo->address);
1542 rtp_source_set_rtcp_from (source, pinfo->address);
1546 /* FIXME: Log 3rd party collision somehow
1547 * Maybe should be done in upper layer, only the SDES can tell us
1548 * if its a collision or a loop
1551 /* This is sending with our ssrc, is it an address we already know */
1552 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1553 pinfo->current_time)) {
1554 /* Its a known conflict, its probably a loop, not a collision
1555 * lets just drop the incoming packet
1557 GST_DEBUG ("Our packets are being looped back to us, dropping");
1559 /* Its a new collision, lets change our SSRC */
1560 rtp_session_add_conflicting_address (sess, pinfo->address,
1561 pinfo->current_time);
1563 GST_DEBUG ("Collision for SSRC %x", ssrc);
1564 /* mark the source BYE */
1565 rtp_source_mark_bye (source, "SSRC Collision");
1566 /* if we were suggesting this SSRC, change to something else */
1567 if (sess->suggested_ssrc == ssrc) {
1568 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1569 sess->internal_ssrc_set = TRUE;
1572 on_ssrc_collision (sess, source);
1574 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1583 gboolean is_doing_ptp;
1584 GSocketAddress *new_addr;
1587 /* check if the two given ip addr are the same (do not care about the port) */
1589 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1592 g_inet_address_equal (g_inet_socket_address_get_address
1593 (G_INET_SOCKET_ADDRESS (a)),
1594 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1598 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1599 CompareAddrData * data)
1601 /* only compare ip addr of remote sources which are also not closing */
1602 if (!source->internal && !source->closing && source->rtp_from) {
1603 /* look for the first rtp source */
1604 if (!data->new_addr)
1605 data->new_addr = source->rtp_from;
1606 /* compare current ip addr with the first one */
1608 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1613 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1614 CompareAddrData * data)
1616 /* only compare ip addr of remote sources which are also not closing */
1617 if (!source->internal && !source->closing && source->rtcp_from) {
1618 /* look for the first rtcp source */
1619 if (!data->new_addr)
1620 data->new_addr = source->rtcp_from;
1622 /* compare current ip addr with the first one */
1623 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1627 /* loop over our non-internal source to know if the session
1628 * is doing point-to-point */
1630 session_update_ptp (RTPSession * sess)
1632 /* to know if the session is doing point to point, the ip addr
1633 * of each non-internal (=remotes) source have to be compared
1636 gboolean is_doing_rtp_ptp;
1637 gboolean is_doing_rtcp_ptp;
1638 CompareAddrData data;
1640 /* compare the first remote source's ip addr that receive rtp packets
1641 * with other remote rtp source.
1642 * it's enough because the session just needs to know if they are all
1645 data.is_doing_ptp = TRUE;
1646 data.new_addr = NULL;
1647 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1648 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1649 is_doing_rtp_ptp = data.is_doing_ptp;
1651 /* same but about rtcp */
1652 data.is_doing_ptp = TRUE;
1653 data.new_addr = NULL;
1654 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1655 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1656 is_doing_rtcp_ptp = data.is_doing_ptp;
1658 /* the session is doing point-to-point if all rtp remote have the same
1659 * ip addr and if all rtcp remote sources have the same ip addr */
1660 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1662 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1666 add_source (RTPSession * sess, RTPSource * src)
1668 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1669 GINT_TO_POINTER (src->ssrc), src);
1670 /* report the new source ASAP */
1671 src->generation = sess->generation;
1672 /* we have one more source now */
1673 sess->total_sources++;
1674 if (RTP_SOURCE_IS_ACTIVE (src))
1675 sess->stats.active_sources++;
1676 if (src->internal) {
1677 sess->stats.internal_sources++;
1678 if (!sess->internal_ssrc_from_caps_or_property
1679 && sess->suggested_ssrc != src->ssrc) {
1680 sess->suggested_ssrc = src->ssrc;
1681 sess->internal_ssrc_set = TRUE;
1685 /* update point-to-point status */
1687 session_update_ptp (sess);
1691 find_source (RTPSession * sess, guint32 ssrc)
1693 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1694 GINT_TO_POINTER (ssrc));
1697 /* must be called with the session lock, the returned source needs to be
1698 * unreffed after usage. */
1700 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1701 RTPPacketInfo * pinfo, gboolean rtp)
1705 source = find_source (sess, ssrc);
1706 if (source == NULL) {
1707 /* make new Source in probation and insert */
1708 source = rtp_source_new (ssrc);
1710 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1712 /* for RTP packets we need to set the source in probation. Receiving RTCP
1713 * packets of an SSRC, on the other hand, is a strong indication that we
1714 * are dealing with a valid source. */
1715 g_object_set (source, "probation", rtp ? sess->probation : 0,
1716 "max-dropout-time", sess->max_dropout_time, "max-misorder-time",
1717 sess->max_misorder_time, NULL);
1719 /* store from address, if any */
1720 if (pinfo->address) {
1722 rtp_source_set_rtp_from (source, pinfo->address);
1724 rtp_source_set_rtcp_from (source, pinfo->address);
1727 /* configure a callback on the source */
1728 rtp_source_set_callbacks (source, &callbacks, sess);
1730 add_source (sess, source);
1734 /* check for collision, this updates the address when not previously set */
1735 if (check_collision (sess, source, pinfo, rtp)) {
1738 /* Receiving RTCP packets of an SSRC is a strong indication that we
1739 * are dealing with a valid source. */
1741 g_object_set (source, "probation", 0, NULL);
1743 /* update last activity */
1744 source->last_activity = pinfo->current_time;
1746 source->last_rtp_activity = pinfo->current_time;
1747 g_object_ref (source);
1752 /* must be called with the session lock, the returned source needs to be
1753 * unreffed after usage. */
1755 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1756 GstClockTime current_time)
1760 source = find_source (sess, ssrc);
1761 if (source == NULL) {
1762 /* make new internal Source and insert */
1763 source = rtp_source_new (ssrc);
1765 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1767 source->validated = TRUE;
1768 source->internal = TRUE;
1769 source->probation = FALSE;
1770 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1771 rtp_source_set_callbacks (source, &callbacks, sess);
1773 add_source (sess, source);
1778 /* update last activity */
1779 if (current_time != GST_CLOCK_TIME_NONE) {
1780 source->last_activity = current_time;
1781 source->last_rtp_activity = current_time;
1783 g_object_ref (source);
1789 * rtp_session_suggest_ssrc:
1790 * @sess: a #RTPSession
1791 * @is_random: if the suggested ssrc is random
1793 * Suggest an unused SSRC in @sess.
1795 * Returns: a free unused SSRC
1798 rtp_session_suggest_ssrc (RTPSession * sess, gboolean * is_random)
1802 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1804 RTP_SESSION_LOCK (sess);
1805 result = sess->suggested_ssrc;
1807 *is_random = !sess->internal_ssrc_set;
1808 RTP_SESSION_UNLOCK (sess);
1814 * rtp_session_add_source:
1815 * @sess: a #RTPSession
1816 * @src: #RTPSource to add
1818 * Add @src to @session.
1820 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1821 * existed in the session.
1824 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1826 gboolean result = FALSE;
1829 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1830 g_return_val_if_fail (src != NULL, FALSE);
1832 RTP_SESSION_LOCK (sess);
1833 find = find_source (sess, src->ssrc);
1835 add_source (sess, src);
1838 RTP_SESSION_UNLOCK (sess);
1844 * rtp_session_get_num_sources:
1845 * @sess: an #RTPSession
1847 * Get the number of sources in @sess.
1849 * Returns: The number of sources in @sess.
1852 rtp_session_get_num_sources (RTPSession * sess)
1856 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1858 RTP_SESSION_LOCK (sess);
1859 result = sess->total_sources;
1860 RTP_SESSION_UNLOCK (sess);
1866 * rtp_session_get_num_active_sources:
1867 * @sess: an #RTPSession
1869 * Get the number of active sources in @sess. A source is considered active when
1870 * it has been validated and has not yet received a BYE RTCP message.
1872 * Returns: The number of active sources in @sess.
1875 rtp_session_get_num_active_sources (RTPSession * sess)
1879 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1881 RTP_SESSION_LOCK (sess);
1882 result = sess->stats.active_sources;
1883 RTP_SESSION_UNLOCK (sess);
1889 * rtp_session_get_source_by_ssrc:
1890 * @sess: an #RTPSession
1893 * Find the source with @ssrc in @sess.
1895 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1896 * g_object_unref() after usage.
1899 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1903 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1905 RTP_SESSION_LOCK (sess);
1906 result = find_source (sess, ssrc);
1908 g_object_ref (result);
1909 RTP_SESSION_UNLOCK (sess);
1914 /* should be called with the SESSION lock */
1916 rtp_session_create_new_ssrc (RTPSession * sess)
1921 ssrc = g_random_int ();
1923 /* see if it exists in the session, we're done if it doesn't */
1924 if (find_source (sess, ssrc) == NULL)
1932 * rtp_session_create_source:
1933 * @sess: an #RTPSession
1935 * Create an #RTPSource for use in @sess. This function will create a source
1936 * with an ssrc that is currently not used by any participants in the session.
1938 * Returns: an #RTPSource.
1941 rtp_session_create_source (RTPSession * sess)
1946 RTP_SESSION_LOCK (sess);
1947 ssrc = rtp_session_create_new_ssrc (sess);
1948 source = rtp_source_new (ssrc);
1949 rtp_source_set_callbacks (source, &callbacks, sess);
1950 /* we need an additional ref for the source in the hashtable */
1951 g_object_ref (source);
1952 add_source (sess, source);
1953 RTP_SESSION_UNLOCK (sess);
1959 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1961 GstNetAddressMeta *meta;
1963 /* get packet size including header overhead */
1964 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1968 GstRTPBuffer rtp = { NULL };
1970 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1971 goto invalid_packet;
1973 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1977 /* only keep info for first buffer */
1978 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1979 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1980 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1981 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1982 /* copy available csrc */
1983 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1984 for (i = 0; i < pinfo->csrc_count; i++)
1985 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1987 gst_rtp_buffer_unmap (&rtp);
1991 /* for netbuffer we can store the IP address to check for collisions */
1992 meta = gst_buffer_get_net_address_meta (*buffer);
1994 g_object_unref (pinfo->address);
1996 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1998 pinfo->address = NULL;
2006 GST_DEBUG ("invalid RTP packet received");
2011 /* update the RTPPacketInfo structure with the current time and other bits
2012 * about the current buffer we are handling.
2013 * This function is typically called when a validated packet is received.
2014 * This function should be called with the SESSION_LOCK
2017 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
2018 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
2019 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2025 pinfo->is_list = is_list;
2027 pinfo->current_time = current_time;
2028 pinfo->running_time = running_time;
2029 pinfo->ntpnstime = ntpnstime;
2030 pinfo->header_len = sess->header_len;
2032 pinfo->payload_len = 0;
2036 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
2038 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
2041 GstBuffer *buffer = GST_BUFFER_CAST (data);
2042 res = update_packet (&buffer, 0, pinfo);
2048 clean_packet_info (RTPPacketInfo * pinfo)
2051 g_object_unref (pinfo->address);
2053 gst_mini_object_unref (pinfo->data);
2059 source_update_active (RTPSession * sess, RTPSource * source,
2060 gboolean prevactive)
2062 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
2063 guint32 ssrc = source->ssrc;
2065 if (prevactive == active)
2069 sess->stats.active_sources++;
2070 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
2071 sess->stats.active_sources);
2073 sess->stats.active_sources--;
2074 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2075 sess->stats.active_sources);
2081 source_update_sender (RTPSession * sess, RTPSource * source,
2082 gboolean prevsender)
2084 gboolean sender = RTP_SOURCE_IS_SENDER (source);
2085 guint32 ssrc = source->ssrc;
2087 if (prevsender == sender)
2091 sess->stats.sender_sources++;
2092 if (source->internal)
2093 sess->stats.internal_sender_sources++;
2094 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
2095 sess->stats.sender_sources);
2097 sess->stats.sender_sources--;
2098 if (source->internal)
2099 sess->stats.internal_sender_sources--;
2100 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2101 sess->stats.sender_sources);
2107 * rtp_session_process_rtp:
2108 * @sess: and #RTPSession
2109 * @buffer: an RTP buffer
2110 * @current_time: the current system time
2111 * @running_time: the running_time of @buffer
2113 * Process an RTP buffer in the session manager. This function takes ownership
2116 * Returns: a #GstFlowReturn.
2119 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
2120 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2122 GstFlowReturn result;
2126 gboolean prevsender, prevactive;
2127 RTPPacketInfo pinfo = { 0, };
2130 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2131 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2133 RTP_SESSION_LOCK (sess);
2135 /* update pinfo stats */
2136 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
2137 current_time, running_time, ntpnstime)) {
2138 GST_DEBUG ("invalid RTP packet received");
2139 RTP_SESSION_UNLOCK (sess);
2140 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
2145 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
2149 prevsender = RTP_SOURCE_IS_SENDER (source);
2150 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2151 oldrate = source->bitrate;
2153 /* let source process the packet */
2154 result = rtp_source_process_rtp (source, &pinfo);
2156 /* source became active */
2157 if (source_update_active (sess, source, prevactive))
2158 on_ssrc_validated (sess, source);
2160 source_update_sender (sess, source, prevsender);
2162 if (oldrate != source->bitrate)
2163 sess->recalc_bandwidth = TRUE;
2166 on_new_ssrc (sess, source);
2168 if (source->validated) {
2172 /* for validated sources, we add the CSRCs as well */
2173 for (i = 0; i < pinfo.csrc_count; i++) {
2175 RTPSource *csrc_src;
2177 csrc = pinfo.csrcs[i];
2180 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
2185 GST_DEBUG ("created new CSRC: %08x", csrc);
2186 rtp_source_set_as_csrc (csrc_src);
2187 source_update_active (sess, csrc_src, FALSE);
2188 on_new_ssrc (sess, csrc_src);
2190 g_object_unref (csrc_src);
2193 g_object_unref (source);
2195 RTP_SESSION_UNLOCK (sess);
2197 clean_packet_info (&pinfo);
2204 RTP_SESSION_UNLOCK (sess);
2205 clean_packet_info (&pinfo);
2206 GST_DEBUG ("ignoring packet because its collisioning");
2212 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
2213 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
2217 count = gst_rtcp_packet_get_rb_count (packet);
2218 for (i = 0; i < count; i++) {
2219 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
2220 guint8 fractionlost;
2224 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
2225 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2227 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2229 /* find our own source */
2230 src = find_source (sess, ssrc);
2234 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2235 /* only deal with report blocks for our session, we update the stats of
2236 * the sender of the RTCP message. We could also compare our stats against
2237 * the other sender to see if we are better or worse. */
2238 /* FIXME, need to keep track who the RB block is from */
2239 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2240 packetslost, exthighestseq, jitter, lsr, dlsr);
2243 on_ssrc_active (sess, source);
2246 /* A Sender report contains statistics about how the sender is doing. This
2247 * includes timing informataion such as the relation between RTP and NTP
2248 * timestamps and the number of packets/bytes it sent to us.
2250 * In this report is also included a set of report blocks related to how this
2251 * sender is receiving data (in case we (or somebody else) is also sending stuff
2252 * to it). This info includes the packet loss, jitter and seqnum. It also
2253 * contains information to calculate the round trip time (LSR/DLSR).
2256 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2257 RTPPacketInfo * pinfo, gboolean * do_sync)
2259 guint32 senderssrc, rtptime, packet_count, octet_count;
2262 gboolean created, prevsender;
2264 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2265 &packet_count, &octet_count);
2267 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2268 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2270 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2274 /* skip non-bye packets for sources that are marked BYE */
2275 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2278 /* don't try to do lip-sync for sources that sent a BYE */
2279 if (RTP_SOURCE_IS_MARKED_BYE (source))
2284 prevsender = RTP_SOURCE_IS_SENDER (source);
2286 /* first update the source */
2287 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2288 packet_count, octet_count);
2290 source_update_sender (sess, source, prevsender);
2293 on_new_ssrc (sess, source);
2295 rtp_session_process_rb (sess, source, packet, pinfo);
2298 g_object_unref (source);
2301 /* A receiver report contains statistics about how a receiver is doing. It
2302 * includes stuff like packet loss, jitter and the seqnum it received last. It
2303 * also contains info to calculate the round trip time.
2305 * We are only interested in how the sender of this report is doing wrt to us.
2308 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2309 RTPPacketInfo * pinfo)
2315 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2317 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2319 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2323 /* skip non-bye packets for sources that are marked BYE */
2324 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2328 on_new_ssrc (sess, source);
2330 rtp_session_process_rb (sess, source, packet, pinfo);
2333 g_object_unref (source);
2336 /* Get SDES items and store them in the SSRC */
2338 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2339 RTPPacketInfo * pinfo)
2342 gboolean more_items, more_entries;
2344 items = gst_rtcp_packet_sdes_get_item_count (packet);
2345 GST_DEBUG ("got SDES packet with %d items", items);
2347 more_items = gst_rtcp_packet_sdes_first_item (packet);
2349 while (more_items) {
2351 gboolean changed, created, prevactive;
2355 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2357 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2361 /* find src, no probation when dealing with RTCP */
2362 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2366 /* skip non-bye packets for sources that are marked BYE */
2367 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2370 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2372 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2374 while (more_entries) {
2375 GstRTCPSDESType type;
2381 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2383 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2386 if (type == GST_RTCP_SDES_PRIV) {
2387 name = g_strndup ((const gchar *) &data[1], data[0]);
2389 data += data[0] + 1;
2391 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2394 value = g_strndup ((const gchar *) data, len);
2396 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2401 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2405 /* takes ownership of sdes */
2406 changed = rtp_source_set_sdes_struct (source, sdes);
2408 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2409 source->validated = TRUE;
2412 on_new_ssrc (sess, source);
2414 /* source became active */
2415 if (source_update_active (sess, source, prevactive))
2416 on_ssrc_validated (sess, source);
2419 on_ssrc_sdes (sess, source);
2422 g_object_unref (source);
2424 more_items = gst_rtcp_packet_sdes_next_item (packet);
2429 /* BYE is sent when a client leaves the session
2432 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2433 RTPPacketInfo * pinfo)
2437 gboolean reconsider = FALSE;
2439 reason = gst_rtcp_packet_bye_get_reason (packet);
2440 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2442 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2443 for (i = 0; i < count; i++) {
2446 gboolean created, prevactive, prevsender;
2447 guint pmembers, members;
2449 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2450 GST_DEBUG ("SSRC: %08x", ssrc);
2452 /* find src and mark bye, no probation when dealing with RTCP */
2453 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2457 if (source->internal) {
2458 /* our own source, something weird with this packet */
2459 g_object_unref (source);
2463 /* store time for when we need to time out this source */
2464 source->bye_time = pinfo->current_time;
2466 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2467 prevsender = RTP_SOURCE_IS_SENDER (source);
2469 /* mark the source BYE */
2470 rtp_source_mark_bye (source, reason);
2472 pmembers = sess->stats.active_sources;
2474 source_update_active (sess, source, prevactive);
2475 source_update_sender (sess, source, prevsender);
2477 members = sess->stats.active_sources;
2479 if (!sess->scheduled_bye && members < pmembers) {
2480 /* some members went away since the previous timeout estimate.
2481 * Perform reverse reconsideration but only when we are not scheduling a
2483 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2484 pinfo->current_time < sess->next_rtcp_check_time) {
2485 GstClockTime time_remaining;
2487 /* Scale our next RTCP check time according to the change of numbers
2488 * of members. But only if a) this is the first RTCP, or b) this is not
2489 * a feedback session, or c) this is a feedback session but we schedule
2490 * for every RTCP interval (aka no t-rr-interval set).
2492 * FIXME: a) and b) are not great as we will possibly go below Tmin
2493 * for non-feedback profiles and in case of a) below
2494 * Tmin/t-rr-interval in any case.
2496 if (sess->last_rtcp_send_time == GST_CLOCK_TIME_NONE ||
2497 !(sess->rtp_profile == GST_RTP_PROFILE_AVPF
2498 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF) ||
2499 sess->next_rtcp_check_time - sess->last_rtcp_send_time ==
2500 sess->last_rtcp_interval) {
2501 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2502 sess->next_rtcp_check_time =
2503 gst_util_uint64_scale (time_remaining, members, pmembers);
2504 sess->next_rtcp_check_time += pinfo->current_time;
2506 sess->last_rtcp_interval =
2507 gst_util_uint64_scale (sess->last_rtcp_interval, members, pmembers);
2509 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2510 GST_TIME_ARGS (sess->next_rtcp_check_time));
2512 /* mark pending reconsider. We only want to signal the reconsideration
2513 * once after we handled all the source in the bye packet */
2519 on_new_ssrc (sess, source);
2521 on_bye_ssrc (sess, source);
2523 g_object_unref (source);
2526 RTP_SESSION_UNLOCK (sess);
2527 /* notify app of reconsideration */
2528 if (sess->callbacks.reconsider)
2529 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2530 RTP_SESSION_LOCK (sess);
2536 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2537 RTPPacketInfo * pinfo)
2539 GST_DEBUG ("received APP");
2541 if (g_signal_has_handler_pending (sess,
2542 rtp_session_signals[SIGNAL_ON_APP_RTCP], 0, TRUE)) {
2543 GstBuffer *data_buffer = NULL;
2544 guint16 data_length;
2547 data_length = gst_rtcp_packet_app_get_data_length (packet) * 4;
2548 if (data_length > 0) {
2549 guint8 *data = gst_rtcp_packet_app_get_data (packet);
2550 data_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2551 GST_BUFFER_COPY_MEMORY, data - packet->rtcp->map.data, data_length);
2552 GST_BUFFER_PTS (data_buffer) = pinfo->running_time;
2555 memcpy (name, gst_rtcp_packet_app_get_name (packet), 4);
2558 RTP_SESSION_UNLOCK (sess);
2559 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_APP_RTCP], 0,
2560 gst_rtcp_packet_app_get_subtype (packet),
2561 gst_rtcp_packet_app_get_ssrc (packet), name, data_buffer);
2562 RTP_SESSION_LOCK (sess);
2565 gst_buffer_unref (data_buffer);
2570 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2571 gboolean fir, GstClockTime current_time)
2573 guint32 round_trip = 0;
2575 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2577 if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2578 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2581 if (current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) {
2582 GST_DEBUG ("Ignoring %s request because one was send without one "
2583 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2584 fir ? "FIR" : "PLI",
2585 GST_TIME_ARGS (current_time - sess->last_keyframe_request),
2586 GST_TIME_ARGS (round_trip_in_ns));
2591 sess->last_keyframe_request = current_time;
2593 GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI",
2594 rtp_source_get_ssrc (src), sess->callbacks.process_rtp,
2595 sess->callbacks.request_key_unit);
2597 RTP_SESSION_UNLOCK (sess);
2598 sess->callbacks.request_key_unit (sess, fir,
2599 sess->request_key_unit_user_data);
2600 RTP_SESSION_LOCK (sess);
2606 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2607 guint32 media_ssrc, GstClockTime current_time)
2611 if (!sess->callbacks.request_key_unit)
2614 src = find_source (sess, sender_ssrc);
2618 rtp_session_request_local_key_unit (sess, src, FALSE, current_time);
2622 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2623 guint8 * fci_data, guint fci_length, GstClockTime current_time)
2628 gboolean our_request = FALSE;
2630 if (!sess->callbacks.request_key_unit)
2636 src = find_source (sess, sender_ssrc);
2638 /* Hack because Google fails to set the sender_ssrc correctly */
2639 if (!src && sender_ssrc == 1) {
2640 GHashTableIter iter;
2642 /* we can't find the source if there are multiple */
2643 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2646 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2647 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2648 if (!src->internal && rtp_source_is_sender (src))
2656 for (position = 0; position < fci_length; position += 8) {
2657 guint8 *data = fci_data + position;
2660 ssrc = GST_READ_UINT32_BE (data);
2662 own = find_source (sess, ssrc);
2666 if (own->internal) {
2674 rtp_session_request_local_key_unit (sess, src, TRUE, current_time);
2678 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2679 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2680 GstClockTime current_time)
2682 sess->stats.nacks_received++;
2684 if (!sess->callbacks.notify_nack)
2687 while (fci_length > 0) {
2688 guint16 seqnum, blp;
2690 seqnum = GST_READ_UINT16_BE (fci_data);
2691 blp = GST_READ_UINT16_BE (fci_data + 2);
2693 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2695 RTP_SESSION_UNLOCK (sess);
2696 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2697 sess->notify_nack_user_data);
2698 RTP_SESSION_LOCK (sess);
2706 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2707 RTPPacketInfo * pinfo, GstClockTime current_time)
2709 GstRTCPType type = gst_rtcp_packet_get_type (packet);
2710 GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
2711 guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2712 guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2713 guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet);
2714 guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet);
2717 src = find_source (sess, media_ssrc);
2719 /* skip non-bye packets for sources that are marked BYE */
2720 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2723 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2724 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2726 if (g_signal_has_handler_pending (sess,
2727 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2728 GstBuffer *fci_buffer = NULL;
2730 if (fci_length > 0) {
2731 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2732 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2734 GST_BUFFER_PTS (fci_buffer) = pinfo->running_time;
2737 RTP_SESSION_UNLOCK (sess);
2738 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2739 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2740 RTP_SESSION_LOCK (sess);
2743 gst_buffer_unref (fci_buffer);
2746 if (src && sess->rtcp_feedback_retention_window) {
2747 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2750 if ((src && src->internal) ||
2751 /* PSFB FIR puts the media ssrc inside the FCI */
2752 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2754 case GST_RTCP_TYPE_PSFB:
2756 case GST_RTCP_PSFB_TYPE_PLI:
2758 src->stats.recv_pli_count++;
2759 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2762 case GST_RTCP_PSFB_TYPE_FIR:
2764 src->stats.recv_fir_count++;
2765 rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length,
2772 case GST_RTCP_TYPE_RTPFB:
2774 case GST_RTCP_RTPFB_TYPE_NACK:
2775 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2776 fci_data, fci_length, current_time);
2788 * rtp_session_process_rtcp:
2789 * @sess: and #RTPSession
2790 * @buffer: an RTCP buffer
2791 * @current_time: the current system time
2792 * @ntpnstime: the current NTP time in nanoseconds
2794 * Process an RTCP buffer in the session manager. This function takes ownership
2797 * Returns: a #GstFlowReturn.
2800 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2801 GstClockTime current_time, guint64 ntpnstime)
2803 GstRTCPPacket packet;
2804 gboolean more, is_bye = FALSE, do_sync = FALSE;
2805 RTPPacketInfo pinfo = { 0, };
2806 GstFlowReturn result = GST_FLOW_OK;
2807 GstRTCPBuffer rtcp = { NULL, };
2809 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2810 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2812 if (!gst_rtcp_buffer_validate_reduced (buffer))
2813 goto invalid_packet;
2815 GST_DEBUG ("received RTCP packet");
2817 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
2820 RTP_SESSION_LOCK (sess);
2821 /* update pinfo stats */
2822 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2825 /* start processing the compound packet */
2826 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2827 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2831 type = gst_rtcp_packet_get_type (&packet);
2834 case GST_RTCP_TYPE_SR:
2835 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2837 case GST_RTCP_TYPE_RR:
2838 rtp_session_process_rr (sess, &packet, &pinfo);
2840 case GST_RTCP_TYPE_SDES:
2841 rtp_session_process_sdes (sess, &packet, &pinfo);
2843 case GST_RTCP_TYPE_BYE:
2845 /* don't try to attempt lip-sync anymore for streams with a BYE */
2847 rtp_session_process_bye (sess, &packet, &pinfo);
2849 case GST_RTCP_TYPE_APP:
2850 rtp_session_process_app (sess, &packet, &pinfo);
2852 case GST_RTCP_TYPE_RTPFB:
2853 case GST_RTCP_TYPE_PSFB:
2854 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2857 GST_WARNING ("got unknown RTCP packet");
2860 more = gst_rtcp_packet_move_to_next (&packet);
2863 gst_rtcp_buffer_unmap (&rtcp);
2865 /* if we are scheduling a BYE, we only want to count bye packets, else we
2866 * count everything */
2867 if (sess->scheduled_bye && is_bye) {
2868 sess->bye_stats.bye_members++;
2869 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2872 /* keep track of average packet size */
2873 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2875 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2876 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2877 RTP_SESSION_UNLOCK (sess);
2880 clean_packet_info (&pinfo);
2882 /* notify caller of sr packets in the callback */
2883 if (do_sync && sess->callbacks.sync_rtcp) {
2884 result = sess->callbacks.sync_rtcp (sess, buffer,
2885 sess->sync_rtcp_user_data);
2887 gst_buffer_unref (buffer);
2894 GST_DEBUG ("invalid RTCP packet received");
2895 gst_buffer_unref (buffer);
2901 * rtp_session_update_send_caps:
2902 * @sess: an #RTPSession
2905 * Update the caps of the sender in the rtp session.
2908 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2913 g_return_if_fail (RTP_IS_SESSION (sess));
2914 g_return_if_fail (GST_IS_CAPS (caps));
2916 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2918 s = gst_caps_get_structure (caps, 0);
2920 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2924 RTP_SESSION_LOCK (sess);
2925 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2926 sess->suggested_ssrc = ssrc;
2927 sess->internal_ssrc_set = TRUE;
2928 sess->internal_ssrc_from_caps_or_property = TRUE;
2930 rtp_source_update_caps (source, caps);
2933 on_new_sender_ssrc (sess, source);
2935 g_object_unref (source);
2938 if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
2940 obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2942 rtp_source_update_caps (source, caps);
2943 g_object_unref (source);
2946 RTP_SESSION_UNLOCK (sess);
2948 sess->internal_ssrc_from_caps_or_property = FALSE;
2953 * rtp_session_send_rtp:
2954 * @sess: an #RTPSession
2955 * @data: pointer to either an RTP buffer or a list of RTP buffers
2956 * @is_list: TRUE when @data is a buffer list
2957 * @current_time: the current system time
2958 * @running_time: the running time of @data
2960 * Send the RTP buffer in the session manager. This function takes ownership of
2963 * Returns: a #GstFlowReturn.
2966 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2967 GstClockTime current_time, GstClockTime running_time)
2969 GstFlowReturn result;
2971 gboolean prevsender;
2973 RTPPacketInfo pinfo = { 0, };
2976 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2977 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
2979 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
2981 RTP_SESSION_LOCK (sess);
2982 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
2983 current_time, running_time, -1))
2984 goto invalid_packet;
2986 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
2988 on_new_sender_ssrc (sess, source);
2990 prevsender = RTP_SOURCE_IS_SENDER (source);
2991 oldrate = source->bitrate;
2993 /* we use our own source to send */
2994 result = rtp_source_send_rtp (source, &pinfo);
2996 source_update_sender (sess, source, prevsender);
2998 if (oldrate != source->bitrate)
2999 sess->recalc_bandwidth = TRUE;
3000 RTP_SESSION_UNLOCK (sess);
3002 g_object_unref (source);
3003 clean_packet_info (&pinfo);
3009 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
3010 RTP_SESSION_UNLOCK (sess);
3011 GST_DEBUG ("invalid RTP packet received");
3017 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
3019 *bandwidth += source->bitrate;
3022 /* must be called with session lock */
3024 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
3027 GstClockTime result;
3028 RTPSessionStats *stats;
3030 /* recalculate bandwidth when it changed */
3031 if (sess->recalc_bandwidth) {
3034 if (sess->bandwidth > 0)
3035 bandwidth = sess->bandwidth;
3037 /* If it is <= 0, then try to estimate the actual bandwidth */
3040 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3041 (GHFunc) add_bitrates, &bandwidth);
3043 if (bandwidth < RTP_STATS_BANDWIDTH)
3044 bandwidth = RTP_STATS_BANDWIDTH;
3046 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
3047 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
3049 sess->recalc_bandwidth = FALSE;
3052 if (sess->scheduled_bye) {
3053 stats = &sess->bye_stats;
3054 result = rtp_stats_calculate_bye_interval (stats);
3056 session_update_ptp (sess);
3058 stats = &sess->stats;
3059 result = rtp_stats_calculate_rtcp_interval (stats,
3060 stats->internal_sender_sources > 0, sess->rtp_profile,
3061 sess->is_doing_ptp, first);
3064 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
3065 GST_TIME_ARGS (result), first);
3067 if (!deterministic && result != GST_CLOCK_TIME_NONE)
3068 result = rtp_stats_add_rtcp_jitter (stats, result);
3070 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
3076 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
3078 if (source->internal)
3079 rtp_source_mark_bye (source, reason);
3083 * rtp_session_mark_all_bye:
3084 * @sess: an #RTPSession
3087 * Mark all internal sources of the session as BYE with @reason.
3090 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
3092 g_return_if_fail (RTP_IS_SESSION (sess));
3094 RTP_SESSION_LOCK (sess);
3095 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3096 (GHFunc) source_mark_bye, (gpointer) reason);
3097 RTP_SESSION_UNLOCK (sess);
3100 /* Stop the current @sess and schedule a BYE message for the other members.
3101 * One must have the session lock to call this function
3103 static GstFlowReturn
3104 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
3106 GstFlowReturn result = GST_FLOW_OK;
3107 GstClockTime interval;
3109 /* nothing to do it we already scheduled bye */
3110 if (sess->scheduled_bye)
3113 /* we schedule BYE now */
3114 sess->scheduled_bye = TRUE;
3115 /* at least one member wants to send a BYE */
3116 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
3117 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
3118 sess->bye_stats.bye_members = 1;
3119 sess->first_rtcp = TRUE;
3121 /* reschedule transmission */
3122 sess->last_rtcp_send_time = current_time;
3123 sess->last_rtcp_check_time = current_time;
3124 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3126 if (interval != GST_CLOCK_TIME_NONE)
3127 sess->next_rtcp_check_time = current_time + interval;
3129 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
3130 sess->last_rtcp_interval = interval;
3132 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
3133 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
3135 RTP_SESSION_UNLOCK (sess);
3136 /* notify app of reconsideration */
3137 if (sess->callbacks.reconsider)
3138 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3139 RTP_SESSION_LOCK (sess);
3146 * rtp_session_schedule_bye:
3147 * @sess: an #RTPSession
3148 * @current_time: the current system time
3150 * Schedule a BYE message for all sources marked as BYE in @sess.
3152 * Returns: a #GstFlowReturn.
3155 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
3157 GstFlowReturn result;
3159 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3161 RTP_SESSION_LOCK (sess);
3162 result = rtp_session_schedule_bye_locked (sess, current_time);
3163 RTP_SESSION_UNLOCK (sess);
3169 * rtp_session_next_timeout:
3170 * @sess: an #RTPSession
3171 * @current_time: the current system time
3173 * Get the next time we should perform session maintenance tasks.
3175 * Returns: a time when rtp_session_on_timeout() should be called with the
3176 * current system time.
3179 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
3181 GstClockTime result, interval = 0;
3183 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
3185 RTP_SESSION_LOCK (sess);
3187 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3188 GST_DEBUG ("have early rtcp time");
3189 result = sess->next_early_rtcp_time;
3193 result = sess->next_rtcp_check_time;
3195 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3196 ", next time: %" GST_TIME_FORMAT,
3197 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3199 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
3200 GST_DEBUG ("take current time as base");
3201 /* our previous check time expired, start counting from the current time
3203 result = current_time;
3206 if (sess->scheduled_bye) {
3207 if (sess->bye_stats.active_sources >= 50) {
3208 GST_DEBUG ("reconsider BYE, more than 50 sources");
3209 /* reconsider BYE if members >= 50 */
3210 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3211 sess->last_rtcp_interval = interval;
3214 if (sess->first_rtcp) {
3215 GST_DEBUG ("first RTCP packet");
3216 /* we are called for the first time */
3217 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3218 sess->last_rtcp_interval = interval;
3219 } else if (sess->next_rtcp_check_time < current_time) {
3220 GST_DEBUG ("old check time expired, getting new timeout");
3221 /* get a new timeout when we need to */
3222 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
3223 sess->last_rtcp_interval = interval;
3225 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
3226 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
3227 && interval != GST_CLOCK_TIME_NONE) {
3228 /* Apply the rules from RFC 4585 section 3.5.3 */
3229 if (sess->stats.min_interval != 0) {
3230 GstClockTime T_rr_current_interval = g_random_double_range (0.5,
3231 1.5) * sess->stats.min_interval * GST_SECOND;
3233 if (T_rr_current_interval > interval) {
3234 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
3235 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
3236 GST_TIME_ARGS (interval));
3237 interval = T_rr_current_interval;
3244 if (interval != GST_CLOCK_TIME_NONE)
3247 result = GST_CLOCK_TIME_NONE;
3249 sess->next_rtcp_check_time = result;
3253 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3254 ", next time: %" GST_TIME_FORMAT,
3255 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3256 RTP_SESSION_UNLOCK (sess);
3270 GstRTCPBuffer rtcpbuf;
3273 guint num_to_report;
3278 GstClockTime current_time;
3280 GstClockTime running_time;
3281 GstClockTime interval;
3282 GstRTCPPacket packet;
3285 gboolean may_suppress;
3287 guint nacked_seqnums;
3291 session_start_rtcp (RTPSession * sess, ReportData * data)
3293 GstRTCPPacket *packet = &data->packet;
3294 RTPSource *own = data->source;
3295 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3297 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
3298 data->has_sdes = FALSE;
3300 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3302 if (data->is_early && sess->reduced_size_rtcp)
3305 if (RTP_SOURCE_IS_SENDER (own)) {
3308 guint32 packet_count, octet_count;
3310 /* we are a sender, create SR */
3311 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
3312 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
3314 /* get latest stats */
3315 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
3316 &ntptime, &rtptime, &packet_count, &octet_count);
3318 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
3319 packet_count, octet_count);
3321 /* fill in sender report info */
3322 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
3323 ntptime, rtptime, packet_count, octet_count);
3325 /* we are only receiver, create RR */
3326 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3327 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3328 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3332 /* construct a Sender or Receiver Report */
3334 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3336 RTPSession *sess = data->sess;
3337 GstRTCPPacket *packet = &data->packet;
3338 guint8 fractionlost;
3340 guint32 exthighestseq, jitter;
3343 /* don't report for sources in future generations */
3344 if (((gint16) (source->generation - sess->generation)) > 0) {
3345 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3346 source->generation, sess->generation);
3350 if (g_hash_table_contains (source->reported_in_sr_of,
3351 GUINT_TO_POINTER (data->source->ssrc))) {
3352 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3356 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3357 GST_DEBUG ("max RB count reached");
3361 /* only report about other sender */
3362 if (source == data->source)
3365 if (!RTP_SOURCE_IS_SENDER (source)) {
3366 GST_DEBUG ("source %08x not sender", source->ssrc);
3370 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3373 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3374 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3376 /* store last generated RR packet */
3377 source->last_rr.is_valid = TRUE;
3378 source->last_rr.fractionlost = fractionlost;
3379 source->last_rr.packetslost = packetslost;
3380 source->last_rr.exthighestseq = exthighestseq;
3381 source->last_rr.jitter = jitter;
3382 source->last_rr.lsr = lsr;
3383 source->last_rr.dlsr = dlsr;
3385 /* packet is not yet filled, add report block for this source. */
3386 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3387 exthighestseq, jitter, lsr, dlsr);
3390 g_hash_table_add (source->reported_in_sr_of,
3391 GUINT_TO_POINTER (data->source->ssrc));
3396 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3398 GstRTCPPacket *packet = &data->packet;
3402 if (!source->send_fir)
3405 len = gst_rtcp_packet_fb_get_fci_length (packet);
3406 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3407 /* exit because the packet is full, will put next request in a
3411 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3413 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3415 fci_data[0] = source->current_send_fir_seqnum;
3416 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3418 source->send_fir = FALSE;
3419 source->stats.sent_fir_count++;
3423 session_fir (RTPSession * sess, ReportData * data)
3425 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3426 GstRTCPPacket *packet = &data->packet;
3428 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3431 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3432 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3433 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3435 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3436 (GHFunc) session_add_fir, data);
3438 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3439 gst_rtcp_packet_remove (packet);
3441 data->may_suppress = FALSE;
3445 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3447 GstRTCPPacket packet;
3448 GstRTCPBuffer rtcp = { NULL, };
3449 gboolean ret = FALSE;
3451 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3453 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3454 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3455 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3459 gst_rtcp_buffer_unmap (&rtcp);
3466 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3468 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3469 GstRTCPPacket *packet = &data->packet;
3471 if (!source->send_pli)
3474 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3477 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3478 /* exit because the packet is full, will put next request in a
3482 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3483 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3484 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3486 source->send_pli = FALSE;
3487 data->may_suppress = FALSE;
3489 source->stats.sent_pli_count++;
3492 /* construct NACK */
3494 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3496 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3497 GstRTCPPacket *packet = &data->packet;
3502 if (!source->send_nack)
3505 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3506 /* exit because the packet is full, will put next request in a
3510 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3511 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3512 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3514 nacks = rtp_source_get_nacks (source, &n_nacks);
3515 GST_DEBUG ("%u NACKs", n_nacks);
3516 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3519 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3520 for (i = 0; i < n_nacks; i++) {
3521 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3523 data->nacked_seqnums++;
3526 rtp_source_clear_nacks (source);
3527 data->may_suppress = FALSE;
3530 /* perform cleanup of sources that timed out */
3532 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3534 gboolean remove = FALSE;
3535 gboolean byetimeout = FALSE;
3536 gboolean sendertimeout = FALSE;
3537 gboolean is_sender, is_active;
3538 RTPSession *sess = data->sess;
3539 GstClockTime interval, binterval;
3542 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3544 /* check for outdated collisions */
3545 if (source->internal) {
3546 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3547 rtp_source_timeout (source, data->current_time,
3548 data->running_time - sess->rtcp_feedback_retention_window);
3551 /* nothing else to do when without RTCP */
3552 if (data->interval == GST_CLOCK_TIME_NONE)
3555 is_sender = RTP_SOURCE_IS_SENDER (source);
3556 is_active = RTP_SOURCE_IS_ACTIVE (source);
3558 /* our own rtcp interval may have been forced low by secondary configuration,
3559 * while sender side may still operate with higher interval,
3560 * so do not just take our interval to decide on timing out sender,
3561 * but take (if data->interval <= 5 * GST_SECOND):
3562 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3563 * where sender_interval is difference between last 2 received RTCP reports
3565 if (data->interval >= 5 * GST_SECOND || source->internal) {
3566 binterval = data->interval;
3568 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3569 GST_TIME_ARGS (source->stats.prev_rtcptime),
3570 GST_TIME_ARGS (source->stats.last_rtcptime));
3571 /* if not received enough yet, fallback to larger default */
3572 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3573 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3575 binterval = 5 * GST_SECOND;
3576 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3578 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3579 GST_TIME_ARGS (binterval));
3581 if (!source->internal && source->marked_bye) {
3582 /* if we received a BYE from the source, remove the source after some
3584 if (data->current_time > source->bye_time &&
3585 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3586 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3592 if (source->internal && source->sent_bye) {
3593 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3597 /* sources that were inactive for more than 5 times the deterministic reporting
3598 * interval get timed out. the min timeout is 5 seconds. */
3599 /* mind old time that might pre-date last time going to PLAYING */
3600 btime = MAX (source->last_activity, sess->start_time);
3601 if (data->current_time > btime) {
3602 interval = MAX (binterval * 5, 5 * GST_SECOND);
3603 if (data->current_time - btime > interval) {
3604 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3605 source->ssrc, GST_TIME_ARGS (btime));
3606 if (source->internal) {
3607 /* this is an internal source that is not using our suggested ssrc.
3608 * since there must be another source using this ssrc, we can remove
3609 * this one instead of making it a receiver forever */
3610 if (source->ssrc != sess->suggested_ssrc) {
3611 rtp_source_mark_bye (source, "timed out");
3612 /* do not schedule bye here, since we are inside the RTCP timeout
3613 * processing and scheduling bye will interfere with SR/RR sending */
3621 /* senders that did not send for a long time become a receiver, this also
3622 * holds for our own sources. */
3624 /* mind old time that might pre-date last time going to PLAYING */
3625 btime = MAX (source->last_rtp_activity, sess->start_time);
3626 if (data->current_time > btime) {
3627 interval = MAX (binterval * 2, 5 * GST_SECOND);
3628 if (data->current_time - btime > interval) {
3629 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3630 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3631 sendertimeout = TRUE;
3637 sess->total_sources--;
3639 sess->stats.sender_sources--;
3640 if (source->internal)
3641 sess->stats.internal_sender_sources--;
3644 sess->stats.active_sources--;
3646 if (source->internal)
3647 sess->stats.internal_sources--;
3650 on_bye_timeout (sess, source);
3652 on_timeout (sess, source);
3654 if (sendertimeout) {
3655 source->is_sender = FALSE;
3656 sess->stats.sender_sources--;
3657 if (source->internal)
3658 sess->stats.internal_sender_sources--;
3660 on_sender_timeout (sess, source);
3662 /* count how many source to report in this generation */
3663 if (((gint16) (source->generation - sess->generation)) <= 0)
3664 data->num_to_report++;
3666 source->closing = remove;
3670 session_sdes (RTPSession * sess, ReportData * data)
3672 GstRTCPPacket *packet = &data->packet;
3673 const GstStructure *sdes;
3675 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3677 /* add SDES packet */
3678 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3680 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3682 sdes = rtp_source_get_sdes_struct (data->source);
3684 /* add all fields in the structure, the order is not important. */
3685 n_fields = gst_structure_n_fields (sdes);
3686 for (i = 0; i < n_fields; ++i) {
3689 GstRTCPSDESType type;
3691 field = gst_structure_nth_field_name (sdes, i);
3694 value = gst_structure_get_string (sdes, field);
3697 type = gst_rtcp_sdes_name_to_type (field);
3699 /* Early packets are minimal and only include the CNAME */
3700 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3703 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3704 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3705 (const guint8 *) value);
3706 } else if (type == GST_RTCP_SDES_PRIV) {
3712 /* don't accept entries that are too big */
3713 prefix_len = strlen (field);
3714 if (prefix_len > 255)
3716 value_len = strlen (value);
3717 if (value_len > 255)
3719 data_len = 1 + prefix_len + value_len;
3723 data[0] = prefix_len;
3724 memcpy (&data[1], field, prefix_len);
3725 memcpy (&data[1 + prefix_len], value, value_len);
3727 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3731 data->has_sdes = TRUE;
3734 /* schedule a BYE packet */
3736 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3738 GstRTCPPacket *packet = &data->packet;
3739 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3742 session_sdes (sess, data);
3743 /* add a BYE packet */
3744 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3745 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3746 if (source->bye_reason)
3747 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3749 /* we have a BYE packet now */
3750 source->sent_bye = TRUE;
3754 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3756 GstClockTime new_send_time;
3757 GstClockTime interval;
3758 RTPSessionStats *stats;
3760 if (sess->scheduled_bye)
3761 stats = &sess->bye_stats;
3763 stats = &sess->stats;
3765 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3766 data->is_early = TRUE;
3768 data->is_early = FALSE;
3770 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3771 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3772 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3773 GST_TIME_ARGS (current_time));
3774 } else if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3775 sess->next_rtcp_check_time > current_time) {
3776 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3777 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3778 GST_TIME_ARGS (current_time));
3782 /* take interval and add jitter */
3783 interval = data->interval;
3784 if (interval != GST_CLOCK_TIME_NONE)
3785 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3787 if (sess->last_rtcp_check_time != GST_CLOCK_TIME_NONE) {
3788 /* perform forward reconsideration */
3789 if (interval != GST_CLOCK_TIME_NONE) {
3790 GstClockTime elapsed;
3792 /* get elapsed time since we last reported */
3793 elapsed = current_time - sess->last_rtcp_check_time;
3795 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3796 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3797 new_send_time = interval + sess->last_rtcp_check_time;
3799 new_send_time = sess->last_rtcp_check_time;
3802 /* If this is the first RTCP packet, we can reconsider anything based
3803 * on the last RTCP send time because there was none.
3805 g_warn_if_fail (!data->is_early);
3806 data->is_early = FALSE;
3807 new_send_time = current_time;
3810 if (!data->is_early) {
3811 /* check if reconsideration */
3812 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3813 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3814 GST_TIME_ARGS (new_send_time));
3815 /* store new check time */
3816 sess->next_rtcp_check_time = new_send_time;
3817 sess->last_rtcp_interval = interval;
3821 sess->last_rtcp_interval = interval;
3822 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
3823 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
3824 && interval != GST_CLOCK_TIME_NONE) {
3825 /* Apply the rules from RFC 4585 section 3.5.3 */
3826 if (stats->min_interval != 0 && !sess->first_rtcp) {
3827 GstClockTime T_rr_current_interval =
3828 g_random_double_range (0.5, 1.5) * stats->min_interval * GST_SECOND;
3830 if (T_rr_current_interval > interval) {
3831 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
3832 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
3833 GST_TIME_ARGS (interval));
3834 interval = T_rr_current_interval;
3838 sess->next_rtcp_check_time = current_time + interval;
3842 GST_DEBUG ("can send RTCP now, next %" GST_TIME_FORMAT,
3843 GST_TIME_ARGS (sess->next_rtcp_check_time));
3849 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3851 g_hash_table_insert (hash_table, key, g_object_ref (source));
3855 remove_closing_sources (const gchar * key, RTPSource * source,
3858 if (source->closing)
3861 if (source->send_fir)
3862 data->have_fir = TRUE;
3863 if (source->send_pli)
3864 data->have_pli = TRUE;
3865 if (source->send_nack)
3866 data->have_nack = TRUE;
3872 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3874 RTPSession *sess = data->sess;
3875 gboolean is_bye = FALSE;
3876 ReportOutput *output;
3878 /* only generate RTCP for active internal sources */
3879 if (!source->internal || source->sent_bye)
3882 /* ignore other sources when we do the timeout after a scheduled BYE */
3883 if (sess->scheduled_bye && !source->marked_bye)
3886 data->source = source;
3889 session_start_rtcp (sess, data);
3891 if (source->marked_bye) {
3893 make_source_bye (sess, source, data);
3895 } else if (!data->is_early) {
3896 /* loop over all known sources and add report blocks. If we are early, we
3897 * just make a minimal RTCP packet and skip this step */
3898 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3899 (GHFunc) session_report_blocks, data);
3901 if (!data->has_sdes && (!data->is_early || !sess->reduced_size_rtcp))
3902 session_sdes (sess, data);
3905 session_fir (sess, data);
3908 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3909 (GHFunc) session_pli, data);
3911 if (data->have_nack)
3912 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3913 (GHFunc) session_nack, data);
3915 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3917 output = g_slice_new (ReportOutput);
3918 output->source = g_object_ref (source);
3919 output->is_bye = is_bye;
3920 output->buffer = data->rtcp;
3921 /* queue the RTCP packet to push later */
3922 g_queue_push_tail (&data->output, output);
3926 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3928 RTPSession *sess = data->sess;
3930 if (g_hash_table_size (source->reported_in_sr_of) >=
3931 sess->stats.internal_sources) {
3932 /* source is reported, move to next generation */
3933 source->generation = sess->generation + 1;
3934 g_hash_table_remove_all (source->reported_in_sr_of);
3936 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3937 source->generation);
3939 /* if we reported all sources in this generation, move to next */
3940 if (--data->num_to_report == 0) {
3942 GST_DEBUG ("all reported, generation now %u", sess->generation);
3948 * rtp_session_on_timeout:
3949 * @sess: an #RTPSession
3950 * @current_time: the current system time
3951 * @ntpnstime: the current NTP time in nanoseconds
3952 * @running_time: the current running_time of the pipeline
3954 * Perform maintenance actions after the timeout obtained with
3955 * rtp_session_next_timeout() expired.
3957 * This function will perform timeouts of receivers and senders, send a BYE
3958 * packet or generate RTCP packets with current session stats.
3960 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
3961 * times, for each packet that should be processed.
3963 * Returns: a #GstFlowReturn.
3966 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
3967 guint64 ntpnstime, GstClockTime running_time)
3969 GstFlowReturn result = GST_FLOW_OK;
3970 ReportData data = { GST_RTCP_BUFFER_INIT };
3971 GHashTable *table_copy;
3972 ReportOutput *output;
3974 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3976 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
3977 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
3978 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
3981 data.current_time = current_time;
3982 data.ntpnstime = ntpnstime;
3983 data.running_time = running_time;
3984 data.num_to_report = 0;
3985 data.may_suppress = FALSE;
3986 data.nacked_seqnums = 0;
3987 g_queue_init (&data.output);
3989 RTP_SESSION_LOCK (sess);
3990 /* get a new interval, we need this for various cleanups etc */
3991 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
3993 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
3995 /* we need an internal source now */
3996 if (sess->stats.internal_sources == 0) {
4000 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
4002 sess->internal_ssrc_set = TRUE;
4005 on_new_sender_ssrc (sess, source);
4007 g_object_unref (source);
4010 sess->conflicting_addresses =
4011 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
4013 /* Make a local copy of the hashtable. We need to do this because the
4014 * cleanup stage below releases the session lock. */
4015 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
4016 (GDestroyNotify) g_object_unref);
4017 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4018 (GHFunc) clone_ssrcs_hashtable, table_copy);
4020 /* Clean up the session, mark the source for removing, this might release the
4022 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
4023 g_hash_table_destroy (table_copy);
4025 /* Now remove the marked sources */
4026 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
4027 (GHRFunc) remove_closing_sources, &data);
4029 /* update point-to-point status */
4030 session_update_ptp (sess);
4032 /* see if we need to generate SR or RR packets */
4033 if (!is_rtcp_time (sess, current_time, &data))
4037 ("doing RTCP generation %u for %u sources, early %d, may suppress %d",
4038 sess->generation, data.num_to_report, data.is_early, data.may_suppress);
4040 /* generate RTCP for all internal sources */
4041 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4042 (GHFunc) generate_rtcp, &data);
4044 /* update the generation for all the sources that have been reported */
4045 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4046 (GHFunc) update_generation, &data);
4048 /* we keep track of the last report time in order to timeout inactive
4049 * receivers or senders */
4050 if (!data.is_early) {
4051 GST_DEBUG ("Time since last regular RTCP: %" GST_TIME_FORMAT " - %"
4052 GST_TIME_FORMAT " = %" GST_TIME_FORMAT,
4053 GST_TIME_ARGS (data.current_time),
4054 GST_TIME_ARGS (sess->last_rtcp_send_time),
4055 GST_TIME_ARGS (data.current_time - sess->last_rtcp_send_time));
4056 sess->last_rtcp_send_time = data.current_time;
4059 GST_DEBUG ("Time since last RTCP: %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT
4060 " = %" GST_TIME_FORMAT, GST_TIME_ARGS (data.current_time),
4061 GST_TIME_ARGS (sess->last_rtcp_send_time),
4062 GST_TIME_ARGS (data.current_time - sess->last_rtcp_check_time));
4063 sess->last_rtcp_check_time = data.current_time;
4064 sess->first_rtcp = FALSE;
4065 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
4066 sess->scheduled_bye = FALSE;
4069 RTP_SESSION_UNLOCK (sess);
4071 /* notify about updated statistics */
4072 g_object_notify (G_OBJECT (sess), "stats");
4074 /* push out the RTCP packets */
4075 while ((output = g_queue_pop_head (&data.output))) {
4076 gboolean do_not_suppress, empty_buffer;
4077 GstBuffer *buffer = output->buffer;
4078 RTPSource *source = output->source;
4080 /* Give the user a change to add its own packet */
4081 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
4082 buffer, data.is_early, &do_not_suppress);
4084 empty_buffer = gst_buffer_get_size (buffer) == 0;
4087 g_warning ("rtpsession: Trying to send an empty RTCP packet");
4089 if (sess->callbacks.send_rtcp &&
4090 !empty_buffer && (do_not_suppress || !data.may_suppress)) {
4093 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
4095 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
4096 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
4097 sess->stats.avg_rtcp_packet_size, packet_size);
4099 sess->callbacks.send_rtcp (sess, source, buffer, output->is_bye,
4100 sess->send_rtcp_user_data);
4101 sess->stats.nacks_sent += data.nacked_seqnums;
4103 RTP_SESSION_LOCK (sess);
4104 on_sender_ssrc_active (sess, source);
4105 RTP_SESSION_UNLOCK (sess);
4107 GST_DEBUG ("freeing packet callback: %p"
4108 " empty_buffer: %d, "
4109 " do_not_suppress: %d may_suppress: %d", sess->callbacks.send_rtcp,
4110 empty_buffer, do_not_suppress, data.may_suppress);
4112 sess->stats.nacks_dropped += data.nacked_seqnums;
4113 gst_buffer_unref (buffer);
4115 g_object_unref (source);
4116 g_slice_free (ReportOutput, output);
4122 * rtp_session_request_early_rtcp:
4123 * @sess: an #RTPSession
4124 * @current_time: the current system time
4125 * @max_delay: maximum delay
4127 * Request transmission of early RTCP
4129 * Returns: %TRUE if the related RTCP can be scheduled.
4132 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
4133 GstClockTime max_delay)
4135 GstClockTime T_dither_max, T_rr, offset = 0;
4137 gboolean allow_early;
4139 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
4141 RTP_SESSION_LOCK (sess);
4143 /* We assume a feedback profile if something is requesting RTCP
4145 sess->rtp_profile = GST_RTP_PROFILE_AVPF;
4147 /* Check if already requested */
4148 /* RFC 4585 section 3.5.2 step 2 */
4149 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
4150 GST_LOG_OBJECT (sess, "already have next early rtcp time");
4151 ret = (current_time + max_delay > sess->next_early_rtcp_time);
4155 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
4156 GST_LOG_OBJECT (sess, "no next RTCP check time");
4161 /* RFC 4585 section 3.5.3 step 1
4162 * If no regular RTCP packet has been sent before, then a regular
4163 * RTCP packet has to be scheduled first and FB messages might be
4166 if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
4167 GST_LOG_OBJECT (sess, "no RTCP sent yet");
4169 if (current_time + max_delay > sess->next_rtcp_check_time) {
4170 GST_LOG_OBJECT (sess,
4171 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4172 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4173 GST_TIME_ARGS (max_delay),
4174 GST_TIME_ARGS (sess->next_rtcp_check_time));
4177 GST_LOG_OBJECT (sess,
4178 "can't allow early feedback, next scheduled time is too late %"
4179 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4180 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4181 GST_TIME_ARGS (sess->next_rtcp_check_time));
4187 T_rr = sess->last_rtcp_interval;
4189 /* RFC 4585 section 3.5.2 step 2b */
4190 /* If the total sources is <=2, then there is only us and one peer */
4191 /* When there is one auxiliary stream the session can still do point
4194 if (sess->is_doing_ptp) {
4197 /* Divide by 2 because l = 0.5 */
4198 T_dither_max = T_rr;
4202 /* RFC 4585 section 3.5.2 step 3 */
4203 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
4204 GST_LOG_OBJECT (sess,
4205 "don't send because of dither, next scheduled time is too soon %"
4206 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
4207 GST_TIME_ARGS (current_time), GST_TIME_ARGS (T_dither_max),
4208 GST_TIME_ARGS (sess->next_rtcp_check_time));
4209 ret = T_dither_max <= max_delay;
4213 /* RFC 4585 section 3.5.2 step 4a and
4214 * RFC 4585 section 3.5.2 step 6 */
4215 allow_early = FALSE;
4216 if (sess->last_rtcp_check_time == sess->last_rtcp_send_time) {
4217 /* Last time we sent a full RTCP packet, we can now immediately
4218 * send an early one as allow_early was reset to TRUE */
4220 } else if (sess->last_rtcp_check_time + T_rr <= current_time + max_delay) {
4221 /* Last packet we sent was an early RTCP packet and more than
4222 * T_rr has passed since then, meaning we would have suppressed
4223 * a regular RTCP packet already and reset allow_early to TRUE */
4226 /* We have to offset a bit as T_rr has not passed yet, but will before
4228 if (sess->last_rtcp_check_time + T_rr > current_time)
4229 offset = (sess->last_rtcp_check_time + T_rr) - current_time;
4231 GST_DEBUG_OBJECT (sess,
4232 "can't allow early RTCP yet: last regular %" GST_TIME_FORMAT ", %"
4233 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT " + %"
4234 GST_TIME_FORMAT, GST_TIME_ARGS (sess->last_rtcp_send_time),
4235 GST_TIME_ARGS (sess->last_rtcp_check_time), GST_TIME_ARGS (T_rr),
4236 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay));
4240 /* Ignore the request a scheduled packet will be in time anyway */
4241 if (current_time + max_delay > sess->next_rtcp_check_time) {
4242 GST_LOG_OBJECT (sess,
4243 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4244 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4245 GST_TIME_ARGS (max_delay),
4246 GST_TIME_ARGS (sess->next_rtcp_check_time));
4249 GST_LOG_OBJECT (sess,
4250 "can't allow early feedback and next scheduled time is too late %"
4251 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4252 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4253 GST_TIME_ARGS (sess->next_rtcp_check_time));
4259 /* RFC 4585 section 3.5.2 step 4b */
4261 /* Schedule an early transmission later */
4262 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
4263 current_time + offset;
4265 /* If no dithering, schedule it for NOW */
4266 sess->next_early_rtcp_time = current_time + offset;
4269 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT
4270 ", next regular RTCP time %" GST_TIME_FORMAT,
4271 GST_TIME_ARGS (sess->next_early_rtcp_time),
4272 GST_TIME_ARGS (sess->next_rtcp_check_time));
4273 RTP_SESSION_UNLOCK (sess);
4275 /* notify app of need to send packet early
4276 * and therefore of timeout change */
4277 if (sess->callbacks.reconsider)
4278 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
4284 RTP_SESSION_UNLOCK (sess);
4290 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
4294 if (!sess->callbacks.send_rtcp)
4297 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
4299 return rtp_session_request_early_rtcp (sess, now, max_delay);
4303 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
4304 gboolean fir, gint count)
4308 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
4309 GST_DEBUG ("FIR/PLI not sent");
4313 RTP_SESSION_LOCK (sess);
4314 src = find_source (sess, ssrc);
4319 src->send_pli = FALSE;
4320 src->send_fir = TRUE;
4322 if (count == -1 || count != src->last_fir_count)
4323 src->current_send_fir_seqnum++;
4324 src->last_fir_count = count;
4325 } else if (!src->send_fir) {
4326 src->send_pli = TRUE;
4328 RTP_SESSION_UNLOCK (sess);
4335 RTP_SESSION_UNLOCK (sess);
4341 * rtp_session_request_nack:
4342 * @sess: a #RTPSession
4344 * @seqnum: the missing seqnum
4345 * @max_delay: max delay to request NACK
4347 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
4349 * Returns: %TRUE if the NACK feedback could be scheduled
4352 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
4353 GstClockTime max_delay)
4357 if (!rtp_session_send_rtcp (sess, max_delay)) {
4358 GST_DEBUG ("NACK not sent");
4362 RTP_SESSION_LOCK (sess);
4363 source = find_source (sess, ssrc);
4367 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
4368 rtp_source_register_nack (source, seqnum);
4369 RTP_SESSION_UNLOCK (sess);
4376 RTP_SESSION_UNLOCK (sess);