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->is_doing_ptp = TRUE;
669 rtp_session_finalize (GObject * object)
674 sess = RTP_SESSION_CAST (object);
676 gst_structure_free (sess->sdes);
678 g_list_free_full (sess->conflicting_addresses,
679 (GDestroyNotify) rtp_conflicting_address_free);
681 /* TODO: Change this again when implementing RFC 2762
682 * for (i = 0; i < 32; i++)
684 for (i = 0; i < 1; i++)
685 g_hash_table_destroy (sess->ssrcs[i]);
687 g_mutex_clear (&sess->lock);
689 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
693 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
695 GValue value = { 0 };
697 g_value_init (&value, RTP_TYPE_SOURCE);
698 g_value_take_object (&value, source);
699 /* copies the value */
700 g_value_array_append (arr, &value);
704 rtp_session_create_sources (RTPSession * sess)
709 RTP_SESSION_LOCK (sess);
710 /* get number of elements in the table */
711 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
712 /* create the result value array */
713 res = g_value_array_new (size);
715 /* and copy all values into the array */
716 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
717 RTP_SESSION_UNLOCK (sess);
723 create_source_stats (gpointer key, RTPSource * source, GValueArray * arr)
725 GValue value = G_VALUE_INIT;
728 g_object_get (source, "stats", &s, NULL);
730 g_value_init (&value, GST_TYPE_STRUCTURE);
731 gst_value_set_structure (&value, s);
732 g_value_array_append (arr, &value);
733 gst_structure_free (s);
734 g_value_unset (&value);
737 static GstStructure *
738 rtp_session_create_stats (RTPSession * sess)
741 GValueArray *source_stats;
742 GValue source_stats_v = G_VALUE_INIT;
745 RTP_SESSION_LOCK (sess);
746 s = gst_structure_new ("application/x-rtp-session-stats",
747 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
748 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
749 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
751 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
752 source_stats = g_value_array_new (size);
753 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
754 (GHFunc) create_source_stats, source_stats);
755 RTP_SESSION_UNLOCK (sess);
757 g_value_init (&source_stats_v, G_TYPE_VALUE_ARRAY);
758 g_value_take_boxed (&source_stats_v, source_stats);
759 gst_structure_take_value (s, "source-stats", &source_stats_v);
765 rtp_session_set_property (GObject * object, guint prop_id,
766 const GValue * value, GParamSpec * pspec)
770 sess = RTP_SESSION (object);
773 case PROP_INTERNAL_SSRC:
774 RTP_SESSION_LOCK (sess);
775 sess->suggested_ssrc = g_value_get_uint (value);
776 sess->internal_ssrc_set = TRUE;
777 sess->internal_ssrc_from_caps_or_property = TRUE;
778 RTP_SESSION_UNLOCK (sess);
779 if (sess->callbacks.reconfigure)
780 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
783 RTP_SESSION_LOCK (sess);
784 sess->bandwidth = g_value_get_double (value);
785 sess->recalc_bandwidth = TRUE;
786 RTP_SESSION_UNLOCK (sess);
788 case PROP_RTCP_FRACTION:
789 RTP_SESSION_LOCK (sess);
790 sess->rtcp_bandwidth = g_value_get_double (value);
791 sess->recalc_bandwidth = TRUE;
792 RTP_SESSION_UNLOCK (sess);
794 case PROP_RTCP_RR_BANDWIDTH:
795 RTP_SESSION_LOCK (sess);
796 sess->rtcp_rr_bandwidth = g_value_get_int (value);
797 sess->recalc_bandwidth = TRUE;
798 RTP_SESSION_UNLOCK (sess);
800 case PROP_RTCP_RS_BANDWIDTH:
801 RTP_SESSION_LOCK (sess);
802 sess->rtcp_rs_bandwidth = g_value_get_int (value);
803 sess->recalc_bandwidth = TRUE;
804 RTP_SESSION_UNLOCK (sess);
807 sess->mtu = g_value_get_uint (value);
810 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
813 sess->favor_new = g_value_get_boolean (value);
815 case PROP_RTCP_MIN_INTERVAL:
816 rtp_stats_set_min_interval (&sess->stats,
817 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
818 /* trigger reconsideration */
819 RTP_SESSION_LOCK (sess);
820 sess->next_rtcp_check_time = 0;
821 RTP_SESSION_UNLOCK (sess);
822 if (sess->callbacks.reconsider)
823 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
825 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
826 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
829 sess->probation = g_value_get_uint (value);
831 case PROP_MAX_DROPOUT_TIME:
832 sess->max_dropout_time = g_value_get_uint (value);
834 case PROP_MAX_MISORDER_TIME:
835 sess->max_misorder_time = g_value_get_uint (value);
837 case PROP_RTP_PROFILE:
838 sess->rtp_profile = g_value_get_enum (value);
839 /* trigger reconsideration */
840 RTP_SESSION_LOCK (sess);
841 sess->next_rtcp_check_time = 0;
842 RTP_SESSION_UNLOCK (sess);
843 if (sess->callbacks.reconsider)
844 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
846 case PROP_RTCP_REDUCED_SIZE:
847 sess->reduced_size_rtcp = g_value_get_boolean (value);
850 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
856 rtp_session_get_property (GObject * object, guint prop_id,
857 GValue * value, GParamSpec * pspec)
861 sess = RTP_SESSION (object);
864 case PROP_INTERNAL_SSRC:
865 g_value_set_uint (value, rtp_session_suggest_ssrc (sess, NULL));
867 case PROP_INTERNAL_SOURCE:
868 /* FIXME, return a random source */
869 g_value_set_object (value, NULL);
872 g_value_set_double (value, sess->bandwidth);
874 case PROP_RTCP_FRACTION:
875 g_value_set_double (value, sess->rtcp_bandwidth);
877 case PROP_RTCP_RR_BANDWIDTH:
878 g_value_set_int (value, sess->rtcp_rr_bandwidth);
880 case PROP_RTCP_RS_BANDWIDTH:
881 g_value_set_int (value, sess->rtcp_rs_bandwidth);
884 g_value_set_uint (value, sess->mtu);
887 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
889 case PROP_NUM_SOURCES:
890 g_value_set_uint (value, rtp_session_get_num_sources (sess));
892 case PROP_NUM_ACTIVE_SOURCES:
893 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
896 g_value_take_boxed (value, rtp_session_create_sources (sess));
899 g_value_set_boolean (value, sess->favor_new);
901 case PROP_RTCP_MIN_INTERVAL:
902 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
904 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
905 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
908 g_value_set_uint (value, sess->probation);
910 case PROP_MAX_DROPOUT_TIME:
911 g_value_set_uint (value, sess->max_dropout_time);
913 case PROP_MAX_MISORDER_TIME:
914 g_value_set_uint (value, sess->max_misorder_time);
917 g_value_take_boxed (value, rtp_session_create_stats (sess));
919 case PROP_RTP_PROFILE:
920 g_value_set_enum (value, sess->rtp_profile);
922 case PROP_RTCP_REDUCED_SIZE:
923 g_value_set_boolean (value, sess->reduced_size_rtcp);
926 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
932 on_new_ssrc (RTPSession * sess, RTPSource * source)
934 g_object_ref (source);
935 RTP_SESSION_UNLOCK (sess);
936 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
937 RTP_SESSION_LOCK (sess);
938 g_object_unref (source);
942 on_ssrc_collision (RTPSession * sess, RTPSource * source)
944 g_object_ref (source);
945 RTP_SESSION_UNLOCK (sess);
946 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
948 RTP_SESSION_LOCK (sess);
949 g_object_unref (source);
953 on_ssrc_validated (RTPSession * sess, RTPSource * source)
955 g_object_ref (source);
956 RTP_SESSION_UNLOCK (sess);
957 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
959 RTP_SESSION_LOCK (sess);
960 g_object_unref (source);
964 on_ssrc_active (RTPSession * sess, RTPSource * source)
966 g_object_ref (source);
967 RTP_SESSION_UNLOCK (sess);
968 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
969 RTP_SESSION_LOCK (sess);
970 g_object_unref (source);
974 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
976 g_object_ref (source);
977 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
978 RTP_SESSION_UNLOCK (sess);
979 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
980 RTP_SESSION_LOCK (sess);
981 g_object_unref (source);
985 on_bye_ssrc (RTPSession * sess, RTPSource * source)
987 g_object_ref (source);
988 RTP_SESSION_UNLOCK (sess);
989 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
990 RTP_SESSION_LOCK (sess);
991 g_object_unref (source);
995 on_bye_timeout (RTPSession * sess, RTPSource * source)
997 g_object_ref (source);
998 RTP_SESSION_UNLOCK (sess);
999 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
1000 RTP_SESSION_LOCK (sess);
1001 g_object_unref (source);
1005 on_timeout (RTPSession * sess, RTPSource * source)
1007 g_object_ref (source);
1008 RTP_SESSION_UNLOCK (sess);
1009 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
1010 RTP_SESSION_LOCK (sess);
1011 g_object_unref (source);
1015 on_sender_timeout (RTPSession * sess, RTPSource * source)
1017 g_object_ref (source);
1018 RTP_SESSION_UNLOCK (sess);
1019 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
1021 RTP_SESSION_LOCK (sess);
1022 g_object_unref (source);
1026 on_new_sender_ssrc (RTPSession * sess, RTPSource * source)
1028 g_object_ref (source);
1029 RTP_SESSION_UNLOCK (sess);
1030 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
1032 RTP_SESSION_LOCK (sess);
1033 g_object_unref (source);
1037 on_sender_ssrc_active (RTPSession * sess, RTPSource * source)
1039 g_object_ref (source);
1040 RTP_SESSION_UNLOCK (sess);
1041 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
1043 RTP_SESSION_LOCK (sess);
1044 g_object_unref (source);
1050 * Create a new session object.
1052 * Returns: a new #RTPSession. g_object_unref() after usage.
1055 rtp_session_new (void)
1059 sess = g_object_new (RTP_TYPE_SESSION, NULL);
1065 * rtp_session_set_callbacks:
1066 * @sess: an #RTPSession
1067 * @callbacks: callbacks to configure
1068 * @user_data: user data passed in the callbacks
1070 * Configure a set of callbacks to be notified of actions.
1073 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
1076 g_return_if_fail (RTP_IS_SESSION (sess));
1078 if (callbacks->process_rtp) {
1079 sess->callbacks.process_rtp = callbacks->process_rtp;
1080 sess->process_rtp_user_data = user_data;
1082 if (callbacks->send_rtp) {
1083 sess->callbacks.send_rtp = callbacks->send_rtp;
1084 sess->send_rtp_user_data = user_data;
1086 if (callbacks->send_rtcp) {
1087 sess->callbacks.send_rtcp = callbacks->send_rtcp;
1088 sess->send_rtcp_user_data = user_data;
1090 if (callbacks->sync_rtcp) {
1091 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
1092 sess->sync_rtcp_user_data = user_data;
1094 if (callbacks->clock_rate) {
1095 sess->callbacks.clock_rate = callbacks->clock_rate;
1096 sess->clock_rate_user_data = user_data;
1098 if (callbacks->reconsider) {
1099 sess->callbacks.reconsider = callbacks->reconsider;
1100 sess->reconsider_user_data = user_data;
1102 if (callbacks->request_key_unit) {
1103 sess->callbacks.request_key_unit = callbacks->request_key_unit;
1104 sess->request_key_unit_user_data = user_data;
1106 if (callbacks->request_time) {
1107 sess->callbacks.request_time = callbacks->request_time;
1108 sess->request_time_user_data = user_data;
1110 if (callbacks->notify_nack) {
1111 sess->callbacks.notify_nack = callbacks->notify_nack;
1112 sess->notify_nack_user_data = user_data;
1114 if (callbacks->reconfigure) {
1115 sess->callbacks.reconfigure = callbacks->reconfigure;
1116 sess->reconfigure_user_data = user_data;
1121 * rtp_session_set_process_rtp_callback:
1122 * @sess: an #RTPSession
1123 * @callback: callback to set
1124 * @user_data: user data passed in the callback
1126 * Configure only the process_rtp callback to be notified of the process_rtp action.
1129 rtp_session_set_process_rtp_callback (RTPSession * sess,
1130 RTPSessionProcessRTP callback, gpointer user_data)
1132 g_return_if_fail (RTP_IS_SESSION (sess));
1134 sess->callbacks.process_rtp = callback;
1135 sess->process_rtp_user_data = user_data;
1139 * rtp_session_set_send_rtp_callback:
1140 * @sess: an #RTPSession
1141 * @callback: callback to set
1142 * @user_data: user data passed in the callback
1144 * Configure only the send_rtp callback to be notified of the send_rtp action.
1147 rtp_session_set_send_rtp_callback (RTPSession * sess,
1148 RTPSessionSendRTP callback, gpointer user_data)
1150 g_return_if_fail (RTP_IS_SESSION (sess));
1152 sess->callbacks.send_rtp = callback;
1153 sess->send_rtp_user_data = user_data;
1157 * rtp_session_set_send_rtcp_callback:
1158 * @sess: an #RTPSession
1159 * @callback: callback to set
1160 * @user_data: user data passed in the callback
1162 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
1165 rtp_session_set_send_rtcp_callback (RTPSession * sess,
1166 RTPSessionSendRTCP callback, gpointer user_data)
1168 g_return_if_fail (RTP_IS_SESSION (sess));
1170 sess->callbacks.send_rtcp = callback;
1171 sess->send_rtcp_user_data = user_data;
1175 * rtp_session_set_sync_rtcp_callback:
1176 * @sess: an #RTPSession
1177 * @callback: callback to set
1178 * @user_data: user data passed in the callback
1180 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
1183 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
1184 RTPSessionSyncRTCP callback, gpointer user_data)
1186 g_return_if_fail (RTP_IS_SESSION (sess));
1188 sess->callbacks.sync_rtcp = callback;
1189 sess->sync_rtcp_user_data = user_data;
1193 * rtp_session_set_clock_rate_callback:
1194 * @sess: an #RTPSession
1195 * @callback: callback to set
1196 * @user_data: user data passed in the callback
1198 * Configure only the clock_rate callback to be notified of the clock_rate action.
1201 rtp_session_set_clock_rate_callback (RTPSession * sess,
1202 RTPSessionClockRate callback, gpointer user_data)
1204 g_return_if_fail (RTP_IS_SESSION (sess));
1206 sess->callbacks.clock_rate = callback;
1207 sess->clock_rate_user_data = user_data;
1211 * rtp_session_set_reconsider_callback:
1212 * @sess: an #RTPSession
1213 * @callback: callback to set
1214 * @user_data: user data passed in the callback
1216 * Configure only the reconsider callback to be notified of the reconsider action.
1219 rtp_session_set_reconsider_callback (RTPSession * sess,
1220 RTPSessionReconsider callback, gpointer user_data)
1222 g_return_if_fail (RTP_IS_SESSION (sess));
1224 sess->callbacks.reconsider = callback;
1225 sess->reconsider_user_data = user_data;
1229 * rtp_session_set_request_time_callback:
1230 * @sess: an #RTPSession
1231 * @callback: callback to set
1232 * @user_data: user data passed in the callback
1234 * Configure only the request_time callback
1237 rtp_session_set_request_time_callback (RTPSession * sess,
1238 RTPSessionRequestTime callback, gpointer user_data)
1240 g_return_if_fail (RTP_IS_SESSION (sess));
1242 sess->callbacks.request_time = callback;
1243 sess->request_time_user_data = user_data;
1247 * rtp_session_set_bandwidth:
1248 * @sess: an #RTPSession
1249 * @bandwidth: the bandwidth allocated
1251 * Set the session bandwidth in bits per second.
1254 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1256 g_return_if_fail (RTP_IS_SESSION (sess));
1258 RTP_SESSION_LOCK (sess);
1259 sess->stats.bandwidth = bandwidth;
1260 RTP_SESSION_UNLOCK (sess);
1264 * rtp_session_get_bandwidth:
1265 * @sess: an #RTPSession
1267 * Get the session bandwidth.
1269 * Returns: the session bandwidth.
1272 rtp_session_get_bandwidth (RTPSession * sess)
1276 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1278 RTP_SESSION_LOCK (sess);
1279 result = sess->stats.bandwidth;
1280 RTP_SESSION_UNLOCK (sess);
1286 * rtp_session_set_rtcp_fraction:
1287 * @sess: an #RTPSession
1288 * @bandwidth: the RTCP bandwidth
1290 * Set the bandwidth in bits per second that should be used for RTCP
1294 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1296 g_return_if_fail (RTP_IS_SESSION (sess));
1298 RTP_SESSION_LOCK (sess);
1299 sess->stats.rtcp_bandwidth = bandwidth;
1300 RTP_SESSION_UNLOCK (sess);
1304 * rtp_session_get_rtcp_fraction:
1305 * @sess: an #RTPSession
1307 * Get the session bandwidth used for RTCP.
1309 * Returns: The bandwidth used for RTCP messages.
1312 rtp_session_get_rtcp_fraction (RTPSession * sess)
1316 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1318 RTP_SESSION_LOCK (sess);
1319 result = sess->stats.rtcp_bandwidth;
1320 RTP_SESSION_UNLOCK (sess);
1326 * rtp_session_get_sdes_struct:
1327 * @sess: an #RTSPSession
1329 * Get the SDES data as a #GstStructure
1331 * Returns: a GstStructure with SDES items for @sess. This function returns a
1332 * copy of the SDES structure, use gst_structure_free() after usage.
1335 rtp_session_get_sdes_struct (RTPSession * sess)
1337 GstStructure *result = NULL;
1339 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1341 RTP_SESSION_LOCK (sess);
1343 result = gst_structure_copy (sess->sdes);
1344 RTP_SESSION_UNLOCK (sess);
1350 * rtp_session_set_sdes_struct:
1351 * @sess: an #RTSPSession
1352 * @sdes: a #GstStructure
1354 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1357 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1359 g_return_if_fail (sdes);
1360 g_return_if_fail (RTP_IS_SESSION (sess));
1362 RTP_SESSION_LOCK (sess);
1364 gst_structure_free (sess->sdes);
1365 sess->sdes = gst_structure_copy (sdes);
1366 RTP_SESSION_UNLOCK (sess);
1369 static GstFlowReturn
1370 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1372 GstFlowReturn result = GST_FLOW_OK;
1374 if (source->internal) {
1375 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1377 RTP_SESSION_UNLOCK (session);
1379 if (session->callbacks.send_rtp)
1381 session->callbacks.send_rtp (session, source, data,
1382 session->send_rtp_user_data);
1384 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1387 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1388 RTP_SESSION_UNLOCK (session);
1390 if (session->callbacks.process_rtp)
1392 session->callbacks.process_rtp (session, source,
1393 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1395 gst_buffer_unref (GST_BUFFER_CAST (data));
1397 RTP_SESSION_LOCK (session);
1403 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1407 RTP_SESSION_UNLOCK (session);
1409 if (session->callbacks.clock_rate)
1411 session->callbacks.clock_rate (session, pt,
1412 session->clock_rate_user_data);
1416 RTP_SESSION_LOCK (session);
1418 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1423 static RTPSourceCallbacks callbacks = {
1424 (RTPSourcePushRTP) source_push_rtp,
1425 (RTPSourceClockRate) source_clock_rate,
1430 * rtp_session_find_conflicting_address:
1431 * @session: The session the packet came in
1432 * @address: address to check for
1433 * @time: The time when the packet that is possibly in conflict arrived
1435 * Checks if an address which has a conflict is already known. If it is
1436 * a known conflict, remember the time
1438 * Returns: TRUE if it was a known conflict, FALSE otherwise
1441 rtp_session_find_conflicting_address (RTPSession * session,
1442 GSocketAddress * address, GstClockTime time)
1444 return find_conflicting_address (session->conflicting_addresses, address,
1449 * rtp_session_add_conflicting_address:
1450 * @session: The session the packet came in
1451 * @address: address to remember
1452 * @time: The time when the packet that is in conflict arrived
1454 * Adds a new conflict address
1457 rtp_session_add_conflicting_address (RTPSession * sess,
1458 GSocketAddress * address, GstClockTime time)
1460 sess->conflicting_addresses =
1461 add_conflicting_address (sess->conflicting_addresses, address, time);
1466 check_collision (RTPSession * sess, RTPSource * source,
1467 RTPPacketInfo * pinfo, gboolean rtp)
1471 /* If we have no pinfo address, we can't do collision checking */
1472 if (!pinfo->address)
1475 ssrc = rtp_source_get_ssrc (source);
1477 if (!source->internal) {
1478 GSocketAddress *from;
1480 /* This is not our local source, but lets check if two remote
1483 from = source->rtp_from;
1485 from = source->rtcp_from;
1489 if (__g_socket_address_equal (from, pinfo->address)) {
1490 /* Address is the same */
1493 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1494 if (sess->favor_new) {
1495 if (rtp_source_find_conflicting_address (source,
1496 pinfo->address, pinfo->current_time)) {
1499 buf1 = __g_socket_address_to_string (pinfo->address);
1500 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1508 /* Current address is not a known conflict, lets assume this is
1509 * a new source. Save old address in possible conflict list
1511 rtp_source_add_conflicting_address (source, from,
1512 pinfo->current_time);
1514 buf1 = __g_socket_address_to_string (from);
1515 buf2 = __g_socket_address_to_string (pinfo->address);
1517 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1518 " saving old as known conflict", ssrc, buf1, buf2);
1521 rtp_source_set_rtp_from (source, pinfo->address);
1523 rtp_source_set_rtcp_from (source, pinfo->address);
1531 /* Don't need to save old addresses, we ignore new sources */
1536 /* We don't already have a from address for RTP, just set it */
1538 rtp_source_set_rtp_from (source, pinfo->address);
1540 rtp_source_set_rtcp_from (source, pinfo->address);
1544 /* FIXME: Log 3rd party collision somehow
1545 * Maybe should be done in upper layer, only the SDES can tell us
1546 * if its a collision or a loop
1549 /* This is sending with our ssrc, is it an address we already know */
1550 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1551 pinfo->current_time)) {
1552 /* Its a known conflict, its probably a loop, not a collision
1553 * lets just drop the incoming packet
1555 GST_DEBUG ("Our packets are being looped back to us, dropping");
1557 /* Its a new collision, lets change our SSRC */
1558 rtp_session_add_conflicting_address (sess, pinfo->address,
1559 pinfo->current_time);
1561 GST_DEBUG ("Collision for SSRC %x", ssrc);
1562 /* mark the source BYE */
1563 rtp_source_mark_bye (source, "SSRC Collision");
1564 /* if we were suggesting this SSRC, change to something else */
1565 if (sess->suggested_ssrc == ssrc) {
1566 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1567 sess->internal_ssrc_set = TRUE;
1570 on_ssrc_collision (sess, source);
1572 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1581 gboolean is_doing_ptp;
1582 GSocketAddress *new_addr;
1585 /* check if the two given ip addr are the same (do not care about the port) */
1587 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1590 g_inet_address_equal (g_inet_socket_address_get_address
1591 (G_INET_SOCKET_ADDRESS (a)),
1592 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1596 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1597 CompareAddrData * data)
1599 /* only compare ip addr of remote sources which are also not closing */
1600 if (!source->internal && !source->closing && source->rtp_from) {
1601 /* look for the first rtp source */
1602 if (!data->new_addr)
1603 data->new_addr = source->rtp_from;
1604 /* compare current ip addr with the first one */
1606 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1611 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1612 CompareAddrData * data)
1614 /* only compare ip addr of remote sources which are also not closing */
1615 if (!source->internal && !source->closing && source->rtcp_from) {
1616 /* look for the first rtcp source */
1617 if (!data->new_addr)
1618 data->new_addr = source->rtcp_from;
1620 /* compare current ip addr with the first one */
1621 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1625 /* loop over our non-internal source to know if the session
1626 * is doing point-to-point */
1628 session_update_ptp (RTPSession * sess)
1630 /* to know if the session is doing point to point, the ip addr
1631 * of each non-internal (=remotes) source have to be compared
1634 gboolean is_doing_rtp_ptp;
1635 gboolean is_doing_rtcp_ptp;
1636 CompareAddrData data;
1638 /* compare the first remote source's ip addr that receive rtp packets
1639 * with other remote rtp source.
1640 * it's enough because the session just needs to know if they are all
1643 data.is_doing_ptp = TRUE;
1644 data.new_addr = NULL;
1645 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1646 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1647 is_doing_rtp_ptp = data.is_doing_ptp;
1649 /* same but about rtcp */
1650 data.is_doing_ptp = TRUE;
1651 data.new_addr = NULL;
1652 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1653 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1654 is_doing_rtcp_ptp = data.is_doing_ptp;
1656 /* the session is doing point-to-point if all rtp remote have the same
1657 * ip addr and if all rtcp remote sources have the same ip addr */
1658 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1660 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1664 add_source (RTPSession * sess, RTPSource * src)
1666 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1667 GINT_TO_POINTER (src->ssrc), src);
1668 /* report the new source ASAP */
1669 src->generation = sess->generation;
1670 /* we have one more source now */
1671 sess->total_sources++;
1672 if (RTP_SOURCE_IS_ACTIVE (src))
1673 sess->stats.active_sources++;
1674 if (src->internal) {
1675 sess->stats.internal_sources++;
1676 if (!sess->internal_ssrc_from_caps_or_property
1677 && sess->suggested_ssrc != src->ssrc) {
1678 sess->suggested_ssrc = src->ssrc;
1679 sess->internal_ssrc_set = TRUE;
1683 /* update point-to-point status */
1685 session_update_ptp (sess);
1689 find_source (RTPSession * sess, guint32 ssrc)
1691 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1692 GINT_TO_POINTER (ssrc));
1695 /* must be called with the session lock, the returned source needs to be
1696 * unreffed after usage. */
1698 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1699 RTPPacketInfo * pinfo, gboolean rtp)
1703 source = find_source (sess, ssrc);
1704 if (source == NULL) {
1705 /* make new Source in probation and insert */
1706 source = rtp_source_new (ssrc);
1708 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1710 /* for RTP packets we need to set the source in probation. Receiving RTCP
1711 * packets of an SSRC, on the other hand, is a strong indication that we
1712 * are dealing with a valid source. */
1713 g_object_set (source, "probation", rtp ? sess->probation : 0,
1714 "max-dropout-time", sess->max_dropout_time, "max-misorder-time",
1715 sess->max_misorder_time, NULL);
1717 /* store from address, if any */
1718 if (pinfo->address) {
1720 rtp_source_set_rtp_from (source, pinfo->address);
1722 rtp_source_set_rtcp_from (source, pinfo->address);
1725 /* configure a callback on the source */
1726 rtp_source_set_callbacks (source, &callbacks, sess);
1728 add_source (sess, source);
1732 /* check for collision, this updates the address when not previously set */
1733 if (check_collision (sess, source, pinfo, rtp)) {
1736 /* Receiving RTCP packets of an SSRC is a strong indication that we
1737 * are dealing with a valid source. */
1739 g_object_set (source, "probation", 0, NULL);
1741 /* update last activity */
1742 source->last_activity = pinfo->current_time;
1744 source->last_rtp_activity = pinfo->current_time;
1745 g_object_ref (source);
1750 /* must be called with the session lock, the returned source needs to be
1751 * unreffed after usage. */
1753 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1754 GstClockTime current_time)
1758 source = find_source (sess, ssrc);
1759 if (source == NULL) {
1760 /* make new internal Source and insert */
1761 source = rtp_source_new (ssrc);
1763 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1765 source->validated = TRUE;
1766 source->internal = TRUE;
1767 source->probation = FALSE;
1768 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1769 rtp_source_set_callbacks (source, &callbacks, sess);
1771 add_source (sess, source);
1776 /* update last activity */
1777 if (current_time != GST_CLOCK_TIME_NONE) {
1778 source->last_activity = current_time;
1779 source->last_rtp_activity = current_time;
1781 g_object_ref (source);
1787 * rtp_session_suggest_ssrc:
1788 * @sess: a #RTPSession
1789 * @is_random: if the suggested ssrc is random
1791 * Suggest an unused SSRC in @sess.
1793 * Returns: a free unused SSRC
1796 rtp_session_suggest_ssrc (RTPSession * sess, gboolean * is_random)
1800 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1802 RTP_SESSION_LOCK (sess);
1803 result = sess->suggested_ssrc;
1805 *is_random = !sess->internal_ssrc_set;
1806 RTP_SESSION_UNLOCK (sess);
1812 * rtp_session_add_source:
1813 * @sess: a #RTPSession
1814 * @src: #RTPSource to add
1816 * Add @src to @session.
1818 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1819 * existed in the session.
1822 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1824 gboolean result = FALSE;
1827 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1828 g_return_val_if_fail (src != NULL, FALSE);
1830 RTP_SESSION_LOCK (sess);
1831 find = find_source (sess, src->ssrc);
1833 add_source (sess, src);
1836 RTP_SESSION_UNLOCK (sess);
1842 * rtp_session_get_num_sources:
1843 * @sess: an #RTPSession
1845 * Get the number of sources in @sess.
1847 * Returns: The number of sources in @sess.
1850 rtp_session_get_num_sources (RTPSession * sess)
1854 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1856 RTP_SESSION_LOCK (sess);
1857 result = sess->total_sources;
1858 RTP_SESSION_UNLOCK (sess);
1864 * rtp_session_get_num_active_sources:
1865 * @sess: an #RTPSession
1867 * Get the number of active sources in @sess. A source is considered active when
1868 * it has been validated and has not yet received a BYE RTCP message.
1870 * Returns: The number of active sources in @sess.
1873 rtp_session_get_num_active_sources (RTPSession * sess)
1877 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1879 RTP_SESSION_LOCK (sess);
1880 result = sess->stats.active_sources;
1881 RTP_SESSION_UNLOCK (sess);
1887 * rtp_session_get_source_by_ssrc:
1888 * @sess: an #RTPSession
1891 * Find the source with @ssrc in @sess.
1893 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1894 * g_object_unref() after usage.
1897 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1901 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1903 RTP_SESSION_LOCK (sess);
1904 result = find_source (sess, ssrc);
1906 g_object_ref (result);
1907 RTP_SESSION_UNLOCK (sess);
1912 /* should be called with the SESSION lock */
1914 rtp_session_create_new_ssrc (RTPSession * sess)
1919 ssrc = g_random_int ();
1921 /* see if it exists in the session, we're done if it doesn't */
1922 if (find_source (sess, ssrc) == NULL)
1930 * rtp_session_create_source:
1931 * @sess: an #RTPSession
1933 * Create an #RTPSource for use in @sess. This function will create a source
1934 * with an ssrc that is currently not used by any participants in the session.
1936 * Returns: an #RTPSource.
1939 rtp_session_create_source (RTPSession * sess)
1944 RTP_SESSION_LOCK (sess);
1945 ssrc = rtp_session_create_new_ssrc (sess);
1946 source = rtp_source_new (ssrc);
1947 rtp_source_set_callbacks (source, &callbacks, sess);
1948 /* we need an additional ref for the source in the hashtable */
1949 g_object_ref (source);
1950 add_source (sess, source);
1951 RTP_SESSION_UNLOCK (sess);
1957 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
1959 GstNetAddressMeta *meta;
1961 /* get packet size including header overhead */
1962 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
1966 GstRTPBuffer rtp = { NULL };
1968 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
1969 goto invalid_packet;
1971 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
1975 /* only keep info for first buffer */
1976 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
1977 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
1978 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
1979 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
1980 /* copy available csrc */
1981 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
1982 for (i = 0; i < pinfo->csrc_count; i++)
1983 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
1985 gst_rtp_buffer_unmap (&rtp);
1989 /* for netbuffer we can store the IP address to check for collisions */
1990 meta = gst_buffer_get_net_address_meta (*buffer);
1992 g_object_unref (pinfo->address);
1994 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
1996 pinfo->address = NULL;
2004 GST_DEBUG ("invalid RTP packet received");
2009 /* update the RTPPacketInfo structure with the current time and other bits
2010 * about the current buffer we are handling.
2011 * This function is typically called when a validated packet is received.
2012 * This function should be called with the SESSION_LOCK
2015 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
2016 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
2017 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2023 pinfo->is_list = is_list;
2025 pinfo->current_time = current_time;
2026 pinfo->running_time = running_time;
2027 pinfo->ntpnstime = ntpnstime;
2028 pinfo->header_len = sess->header_len;
2030 pinfo->payload_len = 0;
2034 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
2036 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
2039 GstBuffer *buffer = GST_BUFFER_CAST (data);
2040 res = update_packet (&buffer, 0, pinfo);
2046 clean_packet_info (RTPPacketInfo * pinfo)
2049 g_object_unref (pinfo->address);
2051 gst_mini_object_unref (pinfo->data);
2057 source_update_active (RTPSession * sess, RTPSource * source,
2058 gboolean prevactive)
2060 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
2061 guint32 ssrc = source->ssrc;
2063 if (prevactive == active)
2067 sess->stats.active_sources++;
2068 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
2069 sess->stats.active_sources);
2071 sess->stats.active_sources--;
2072 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2073 sess->stats.active_sources);
2079 source_update_sender (RTPSession * sess, RTPSource * source,
2080 gboolean prevsender)
2082 gboolean sender = RTP_SOURCE_IS_SENDER (source);
2083 guint32 ssrc = source->ssrc;
2085 if (prevsender == sender)
2089 sess->stats.sender_sources++;
2090 if (source->internal)
2091 sess->stats.internal_sender_sources++;
2092 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
2093 sess->stats.sender_sources);
2095 sess->stats.sender_sources--;
2096 if (source->internal)
2097 sess->stats.internal_sender_sources--;
2098 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2099 sess->stats.sender_sources);
2105 * rtp_session_process_rtp:
2106 * @sess: and #RTPSession
2107 * @buffer: an RTP buffer
2108 * @current_time: the current system time
2109 * @running_time: the running_time of @buffer
2111 * Process an RTP buffer in the session manager. This function takes ownership
2114 * Returns: a #GstFlowReturn.
2117 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
2118 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2120 GstFlowReturn result;
2124 gboolean prevsender, prevactive;
2125 RTPPacketInfo pinfo = { 0, };
2128 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2129 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2131 RTP_SESSION_LOCK (sess);
2133 /* update pinfo stats */
2134 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
2135 current_time, running_time, ntpnstime)) {
2136 GST_DEBUG ("invalid RTP packet received");
2137 RTP_SESSION_UNLOCK (sess);
2138 return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
2143 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
2147 prevsender = RTP_SOURCE_IS_SENDER (source);
2148 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2149 oldrate = source->bitrate;
2151 /* let source process the packet */
2152 result = rtp_source_process_rtp (source, &pinfo);
2154 /* source became active */
2155 if (source_update_active (sess, source, prevactive))
2156 on_ssrc_validated (sess, source);
2158 source_update_sender (sess, source, prevsender);
2160 if (oldrate != source->bitrate)
2161 sess->recalc_bandwidth = TRUE;
2164 on_new_ssrc (sess, source);
2166 if (source->validated) {
2170 /* for validated sources, we add the CSRCs as well */
2171 for (i = 0; i < pinfo.csrc_count; i++) {
2173 RTPSource *csrc_src;
2175 csrc = pinfo.csrcs[i];
2178 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
2183 GST_DEBUG ("created new CSRC: %08x", csrc);
2184 rtp_source_set_as_csrc (csrc_src);
2185 source_update_active (sess, csrc_src, FALSE);
2186 on_new_ssrc (sess, csrc_src);
2188 g_object_unref (csrc_src);
2191 g_object_unref (source);
2193 RTP_SESSION_UNLOCK (sess);
2195 clean_packet_info (&pinfo);
2202 RTP_SESSION_UNLOCK (sess);
2203 clean_packet_info (&pinfo);
2204 GST_DEBUG ("ignoring packet because its collisioning");
2210 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
2211 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
2215 count = gst_rtcp_packet_get_rb_count (packet);
2216 for (i = 0; i < count; i++) {
2217 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
2218 guint8 fractionlost;
2222 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
2223 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2225 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2227 /* find our own source */
2228 src = find_source (sess, ssrc);
2232 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2233 /* only deal with report blocks for our session, we update the stats of
2234 * the sender of the RTCP message. We could also compare our stats against
2235 * the other sender to see if we are better or worse. */
2236 /* FIXME, need to keep track who the RB block is from */
2237 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2238 packetslost, exthighestseq, jitter, lsr, dlsr);
2241 on_ssrc_active (sess, source);
2244 /* A Sender report contains statistics about how the sender is doing. This
2245 * includes timing informataion such as the relation between RTP and NTP
2246 * timestamps and the number of packets/bytes it sent to us.
2248 * In this report is also included a set of report blocks related to how this
2249 * sender is receiving data (in case we (or somebody else) is also sending stuff
2250 * to it). This info includes the packet loss, jitter and seqnum. It also
2251 * contains information to calculate the round trip time (LSR/DLSR).
2254 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2255 RTPPacketInfo * pinfo, gboolean * do_sync)
2257 guint32 senderssrc, rtptime, packet_count, octet_count;
2260 gboolean created, prevsender;
2262 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2263 &packet_count, &octet_count);
2265 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2266 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2268 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2272 /* skip non-bye packets for sources that are marked BYE */
2273 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2276 /* don't try to do lip-sync for sources that sent a BYE */
2277 if (RTP_SOURCE_IS_MARKED_BYE (source))
2282 prevsender = RTP_SOURCE_IS_SENDER (source);
2284 /* first update the source */
2285 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2286 packet_count, octet_count);
2288 source_update_sender (sess, source, prevsender);
2291 on_new_ssrc (sess, source);
2293 rtp_session_process_rb (sess, source, packet, pinfo);
2296 g_object_unref (source);
2299 /* A receiver report contains statistics about how a receiver is doing. It
2300 * includes stuff like packet loss, jitter and the seqnum it received last. It
2301 * also contains info to calculate the round trip time.
2303 * We are only interested in how the sender of this report is doing wrt to us.
2306 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2307 RTPPacketInfo * pinfo)
2313 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2315 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2317 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2321 /* skip non-bye packets for sources that are marked BYE */
2322 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2326 on_new_ssrc (sess, source);
2328 rtp_session_process_rb (sess, source, packet, pinfo);
2331 g_object_unref (source);
2334 /* Get SDES items and store them in the SSRC */
2336 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2337 RTPPacketInfo * pinfo)
2340 gboolean more_items, more_entries;
2342 items = gst_rtcp_packet_sdes_get_item_count (packet);
2343 GST_DEBUG ("got SDES packet with %d items", items);
2345 more_items = gst_rtcp_packet_sdes_first_item (packet);
2347 while (more_items) {
2349 gboolean changed, created, prevactive;
2353 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2355 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2359 /* find src, no probation when dealing with RTCP */
2360 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2364 /* skip non-bye packets for sources that are marked BYE */
2365 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2368 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2370 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2372 while (more_entries) {
2373 GstRTCPSDESType type;
2379 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2381 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2384 if (type == GST_RTCP_SDES_PRIV) {
2385 name = g_strndup ((const gchar *) &data[1], data[0]);
2387 data += data[0] + 1;
2389 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2392 value = g_strndup ((const gchar *) data, len);
2394 if (g_utf8_validate (value, -1, NULL)) {
2395 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2397 GST_WARNING ("ignore SDES field %s with non-utf8 data %s", name, value);
2403 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2407 /* takes ownership of sdes */
2408 changed = rtp_source_set_sdes_struct (source, sdes);
2410 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2411 source->validated = TRUE;
2414 on_new_ssrc (sess, source);
2416 /* source became active */
2417 if (source_update_active (sess, source, prevactive))
2418 on_ssrc_validated (sess, source);
2421 on_ssrc_sdes (sess, source);
2424 g_object_unref (source);
2426 more_items = gst_rtcp_packet_sdes_next_item (packet);
2431 /* BYE is sent when a client leaves the session
2434 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2435 RTPPacketInfo * pinfo)
2439 gboolean reconsider = FALSE;
2441 reason = gst_rtcp_packet_bye_get_reason (packet);
2442 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2444 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2445 for (i = 0; i < count; i++) {
2448 gboolean prevactive, prevsender;
2449 guint pmembers, members;
2451 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2452 GST_DEBUG ("SSRC: %08x", ssrc);
2454 /* find src and mark bye, no probation when dealing with RTCP */
2455 source = find_source (sess, ssrc);
2456 if (!source || source->internal) {
2457 GST_DEBUG ("Ignoring suspicious BYE packet (reason: %s)",
2458 !source ? "can't find source" : "has internal source SSRC");
2462 /* store time for when we need to time out this source */
2463 source->bye_time = pinfo->current_time;
2465 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2466 prevsender = RTP_SOURCE_IS_SENDER (source);
2468 /* mark the source BYE */
2469 rtp_source_mark_bye (source, reason);
2471 pmembers = sess->stats.active_sources;
2473 source_update_active (sess, source, prevactive);
2474 source_update_sender (sess, source, prevsender);
2476 members = sess->stats.active_sources;
2478 if (!sess->scheduled_bye && members < pmembers) {
2479 /* some members went away since the previous timeout estimate.
2480 * Perform reverse reconsideration but only when we are not scheduling a
2482 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2483 pinfo->current_time < sess->next_rtcp_check_time) {
2484 GstClockTime time_remaining;
2486 /* Scale our next RTCP check time according to the change of numbers
2487 * of members. But only if a) this is the first RTCP, or b) this is not
2488 * a feedback session, or c) this is a feedback session but we schedule
2489 * for every RTCP interval (aka no t-rr-interval set).
2491 * FIXME: a) and b) are not great as we will possibly go below Tmin
2492 * for non-feedback profiles and in case of a) below
2493 * Tmin/t-rr-interval in any case.
2495 if (sess->last_rtcp_send_time == GST_CLOCK_TIME_NONE ||
2496 !(sess->rtp_profile == GST_RTP_PROFILE_AVPF
2497 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF) ||
2498 sess->next_rtcp_check_time - sess->last_rtcp_send_time ==
2499 sess->last_rtcp_interval) {
2500 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2501 sess->next_rtcp_check_time =
2502 gst_util_uint64_scale (time_remaining, members, pmembers);
2503 sess->next_rtcp_check_time += pinfo->current_time;
2505 sess->last_rtcp_interval =
2506 gst_util_uint64_scale (sess->last_rtcp_interval, members, pmembers);
2508 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2509 GST_TIME_ARGS (sess->next_rtcp_check_time));
2511 /* mark pending reconsider. We only want to signal the reconsideration
2512 * once after we handled all the source in the bye packet */
2517 on_bye_ssrc (sess, source);
2520 RTP_SESSION_UNLOCK (sess);
2521 /* notify app of reconsideration */
2522 if (sess->callbacks.reconsider)
2523 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2524 RTP_SESSION_LOCK (sess);
2531 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2532 RTPPacketInfo * pinfo)
2534 GST_DEBUG ("received APP");
2536 if (g_signal_has_handler_pending (sess,
2537 rtp_session_signals[SIGNAL_ON_APP_RTCP], 0, TRUE)) {
2538 GstBuffer *data_buffer = NULL;
2539 guint16 data_length;
2542 data_length = gst_rtcp_packet_app_get_data_length (packet) * 4;
2543 if (data_length > 0) {
2544 guint8 *data = gst_rtcp_packet_app_get_data (packet);
2545 data_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2546 GST_BUFFER_COPY_MEMORY, data - packet->rtcp->map.data, data_length);
2547 GST_BUFFER_PTS (data_buffer) = pinfo->running_time;
2550 memcpy (name, gst_rtcp_packet_app_get_name (packet), 4);
2553 RTP_SESSION_UNLOCK (sess);
2554 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_APP_RTCP], 0,
2555 gst_rtcp_packet_app_get_subtype (packet),
2556 gst_rtcp_packet_app_get_ssrc (packet), name, data_buffer);
2557 RTP_SESSION_LOCK (sess);
2560 gst_buffer_unref (data_buffer);
2565 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2566 guint32 media_ssrc, gboolean fir, GstClockTime current_time)
2568 guint32 round_trip = 0;
2570 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2572 if (src->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2573 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2576 /* Sanity check to avoid always ignoring PLI/FIR if we receive RTCP
2577 * packets with erroneous values resulting in crazy high RTT. */
2578 if (round_trip_in_ns > 5 * GST_SECOND)
2579 round_trip_in_ns = GST_SECOND / 2;
2581 if (current_time - src->last_keyframe_request < 2 * round_trip_in_ns) {
2582 GST_DEBUG ("Ignoring %s request from %X because one was send without one "
2583 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2584 fir ? "FIR" : "PLI", rtp_source_get_ssrc (src),
2585 GST_TIME_ARGS (current_time - src->last_keyframe_request),
2586 GST_TIME_ARGS (round_trip_in_ns));
2591 src->last_keyframe_request = current_time;
2593 GST_LOG ("received %s request from %X about %X %p(%p)", fir ? "FIR" : "PLI",
2594 rtp_source_get_ssrc (src), media_ssrc, sess->callbacks.process_rtp,
2595 sess->callbacks.request_key_unit);
2597 RTP_SESSION_UNLOCK (sess);
2598 sess->callbacks.request_key_unit (sess, media_ssrc, 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);
2616 /* try to find a src with media_ssrc instead */
2617 src = find_source (sess, media_ssrc);
2622 rtp_session_request_local_key_unit (sess, src, media_ssrc, FALSE,
2627 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2628 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2629 GstClockTime current_time)
2634 gboolean our_request = FALSE;
2636 if (!sess->callbacks.request_key_unit)
2642 src = find_source (sess, sender_ssrc);
2644 /* Hack because Google fails to set the sender_ssrc correctly */
2645 if (!src && sender_ssrc == 1) {
2646 GHashTableIter iter;
2648 /* we can't find the source if there are multiple */
2649 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2652 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2653 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2654 if (!src->internal && rtp_source_is_sender (src))
2662 for (position = 0; position < fci_length; position += 8) {
2663 guint8 *data = fci_data + position;
2666 ssrc = GST_READ_UINT32_BE (data);
2668 own = find_source (sess, ssrc);
2672 if (own->internal) {
2680 rtp_session_request_local_key_unit (sess, src, media_ssrc, TRUE,
2685 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2686 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2687 GstClockTime current_time)
2689 sess->stats.nacks_received++;
2691 if (!sess->callbacks.notify_nack)
2694 while (fci_length > 0) {
2695 guint16 seqnum, blp;
2697 seqnum = GST_READ_UINT16_BE (fci_data);
2698 blp = GST_READ_UINT16_BE (fci_data + 2);
2700 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2702 RTP_SESSION_UNLOCK (sess);
2703 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2704 sess->notify_nack_user_data);
2705 RTP_SESSION_LOCK (sess);
2713 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2714 RTPPacketInfo * pinfo, GstClockTime current_time)
2717 GstRTCPFBType fbtype;
2718 guint32 sender_ssrc, media_ssrc;
2723 /* The feedback packet must include both sender SSRC and media SSRC */
2724 if (packet->length < 2)
2727 type = gst_rtcp_packet_get_type (packet);
2728 fbtype = gst_rtcp_packet_fb_get_type (packet);
2729 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2730 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2732 src = find_source (sess, media_ssrc);
2734 /* skip non-bye packets for sources that are marked BYE */
2735 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2741 fci_data = gst_rtcp_packet_fb_get_fci (packet);
2742 fci_length = gst_rtcp_packet_fb_get_fci_length (packet) * sizeof (guint32);
2744 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2745 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2747 if (g_signal_has_handler_pending (sess,
2748 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2749 GstBuffer *fci_buffer = NULL;
2751 if (fci_length > 0) {
2752 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2753 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2755 GST_BUFFER_PTS (fci_buffer) = pinfo->running_time;
2758 RTP_SESSION_UNLOCK (sess);
2759 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2760 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2761 RTP_SESSION_LOCK (sess);
2764 gst_buffer_unref (fci_buffer);
2767 if (src && sess->rtcp_feedback_retention_window) {
2768 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2771 if ((src && src->internal) ||
2772 /* PSFB FIR puts the media ssrc inside the FCI */
2773 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2775 case GST_RTCP_TYPE_PSFB:
2777 case GST_RTCP_PSFB_TYPE_PLI:
2779 src->stats.recv_pli_count++;
2780 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2783 case GST_RTCP_PSFB_TYPE_FIR:
2785 src->stats.recv_fir_count++;
2786 rtp_session_process_fir (sess, sender_ssrc, media_ssrc, fci_data,
2787 fci_length, current_time);
2793 case GST_RTCP_TYPE_RTPFB:
2795 case GST_RTCP_RTPFB_TYPE_NACK:
2797 src->stats.recv_nack_count++;
2798 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2799 fci_data, fci_length, current_time);
2810 g_object_unref (src);
2814 * rtp_session_process_rtcp:
2815 * @sess: and #RTPSession
2816 * @buffer: an RTCP buffer
2817 * @current_time: the current system time
2818 * @ntpnstime: the current NTP time in nanoseconds
2820 * Process an RTCP buffer in the session manager. This function takes ownership
2823 * Returns: a #GstFlowReturn.
2826 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2827 GstClockTime current_time, guint64 ntpnstime)
2829 GstRTCPPacket packet;
2830 gboolean more, is_bye = FALSE, do_sync = FALSE;
2831 RTPPacketInfo pinfo = { 0, };
2832 GstFlowReturn result = GST_FLOW_OK;
2833 GstRTCPBuffer rtcp = { NULL, };
2835 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2836 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2838 if (!gst_rtcp_buffer_validate_reduced (buffer))
2839 goto invalid_packet;
2841 GST_DEBUG ("received RTCP packet");
2843 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
2846 RTP_SESSION_LOCK (sess);
2847 /* update pinfo stats */
2848 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2851 /* start processing the compound packet */
2852 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2853 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2857 type = gst_rtcp_packet_get_type (&packet);
2860 case GST_RTCP_TYPE_SR:
2861 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2863 case GST_RTCP_TYPE_RR:
2864 rtp_session_process_rr (sess, &packet, &pinfo);
2866 case GST_RTCP_TYPE_SDES:
2867 rtp_session_process_sdes (sess, &packet, &pinfo);
2869 case GST_RTCP_TYPE_BYE:
2871 /* don't try to attempt lip-sync anymore for streams with a BYE */
2873 rtp_session_process_bye (sess, &packet, &pinfo);
2875 case GST_RTCP_TYPE_APP:
2876 rtp_session_process_app (sess, &packet, &pinfo);
2878 case GST_RTCP_TYPE_RTPFB:
2879 case GST_RTCP_TYPE_PSFB:
2880 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2882 case GST_RTCP_TYPE_XR:
2883 /* FIXME: This block is added to downgrade warning level.
2884 * Once the parser is implemented, it should be replaced with
2885 * a proper process function. */
2886 GST_DEBUG ("got RTCP XR packet, but ignored");
2889 GST_WARNING ("got unknown RTCP packet type: %d", type);
2892 more = gst_rtcp_packet_move_to_next (&packet);
2895 gst_rtcp_buffer_unmap (&rtcp);
2897 /* if we are scheduling a BYE, we only want to count bye packets, else we
2898 * count everything */
2899 if (sess->scheduled_bye && is_bye) {
2900 sess->bye_stats.bye_members++;
2901 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2904 /* keep track of average packet size */
2905 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2907 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
2908 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2909 RTP_SESSION_UNLOCK (sess);
2912 clean_packet_info (&pinfo);
2914 /* notify caller of sr packets in the callback */
2915 if (do_sync && sess->callbacks.sync_rtcp) {
2916 result = sess->callbacks.sync_rtcp (sess, buffer,
2917 sess->sync_rtcp_user_data);
2919 gst_buffer_unref (buffer);
2926 GST_DEBUG ("invalid RTCP packet received");
2927 gst_buffer_unref (buffer);
2933 * rtp_session_update_send_caps:
2934 * @sess: an #RTPSession
2937 * Update the caps of the sender in the rtp session.
2940 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
2945 g_return_if_fail (RTP_IS_SESSION (sess));
2946 g_return_if_fail (GST_IS_CAPS (caps));
2948 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
2950 s = gst_caps_get_structure (caps, 0);
2952 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
2956 RTP_SESSION_LOCK (sess);
2957 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2958 sess->suggested_ssrc = ssrc;
2959 sess->internal_ssrc_set = TRUE;
2960 sess->internal_ssrc_from_caps_or_property = TRUE;
2962 rtp_source_update_caps (source, caps);
2965 on_new_sender_ssrc (sess, source);
2967 g_object_unref (source);
2970 if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
2972 obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
2974 rtp_source_update_caps (source, caps);
2975 g_object_unref (source);
2978 RTP_SESSION_UNLOCK (sess);
2980 sess->internal_ssrc_from_caps_or_property = FALSE;
2985 * rtp_session_send_rtp:
2986 * @sess: an #RTPSession
2987 * @data: pointer to either an RTP buffer or a list of RTP buffers
2988 * @is_list: TRUE when @data is a buffer list
2989 * @current_time: the current system time
2990 * @running_time: the running time of @data
2992 * Send the RTP buffer in the session manager. This function takes ownership of
2995 * Returns: a #GstFlowReturn.
2998 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
2999 GstClockTime current_time, GstClockTime running_time)
3001 GstFlowReturn result;
3003 gboolean prevsender;
3005 RTPPacketInfo pinfo = { 0, };
3008 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3009 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
3011 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
3013 RTP_SESSION_LOCK (sess);
3014 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
3015 current_time, running_time, -1))
3016 goto invalid_packet;
3018 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
3020 on_new_sender_ssrc (sess, source);
3022 prevsender = RTP_SOURCE_IS_SENDER (source);
3023 oldrate = source->bitrate;
3025 /* we use our own source to send */
3026 result = rtp_source_send_rtp (source, &pinfo);
3028 source_update_sender (sess, source, prevsender);
3030 if (oldrate != source->bitrate)
3031 sess->recalc_bandwidth = TRUE;
3032 RTP_SESSION_UNLOCK (sess);
3034 g_object_unref (source);
3035 clean_packet_info (&pinfo);
3041 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
3042 RTP_SESSION_UNLOCK (sess);
3043 GST_DEBUG ("invalid RTP packet received");
3049 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
3051 *bandwidth += source->bitrate;
3054 /* must be called with session lock */
3056 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
3059 GstClockTime result;
3060 RTPSessionStats *stats;
3062 /* recalculate bandwidth when it changed */
3063 if (sess->recalc_bandwidth) {
3066 if (sess->bandwidth > 0)
3067 bandwidth = sess->bandwidth;
3069 /* If it is <= 0, then try to estimate the actual bandwidth */
3072 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3073 (GHFunc) add_bitrates, &bandwidth);
3075 if (bandwidth < RTP_STATS_BANDWIDTH)
3076 bandwidth = RTP_STATS_BANDWIDTH;
3078 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
3079 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
3081 sess->recalc_bandwidth = FALSE;
3084 if (sess->scheduled_bye) {
3085 stats = &sess->bye_stats;
3086 result = rtp_stats_calculate_bye_interval (stats);
3088 session_update_ptp (sess);
3090 stats = &sess->stats;
3091 result = rtp_stats_calculate_rtcp_interval (stats,
3092 stats->internal_sender_sources > 0, sess->rtp_profile,
3093 sess->is_doing_ptp, first);
3096 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
3097 GST_TIME_ARGS (result), first);
3099 if (!deterministic && result != GST_CLOCK_TIME_NONE)
3100 result = rtp_stats_add_rtcp_jitter (stats, result);
3102 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
3108 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
3110 if (source->internal)
3111 rtp_source_mark_bye (source, reason);
3115 * rtp_session_mark_all_bye:
3116 * @sess: an #RTPSession
3119 * Mark all internal sources of the session as BYE with @reason.
3122 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
3124 g_return_if_fail (RTP_IS_SESSION (sess));
3126 RTP_SESSION_LOCK (sess);
3127 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3128 (GHFunc) source_mark_bye, (gpointer) reason);
3129 RTP_SESSION_UNLOCK (sess);
3132 /* Stop the current @sess and schedule a BYE message for the other members.
3133 * One must have the session lock to call this function
3135 static GstFlowReturn
3136 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
3138 GstFlowReturn result = GST_FLOW_OK;
3139 GstClockTime interval;
3141 /* nothing to do it we already scheduled bye */
3142 if (sess->scheduled_bye)
3145 /* we schedule BYE now */
3146 sess->scheduled_bye = TRUE;
3147 /* at least one member wants to send a BYE */
3148 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
3149 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
3150 sess->bye_stats.bye_members = 1;
3151 sess->first_rtcp = TRUE;
3153 /* reschedule transmission */
3154 sess->last_rtcp_send_time = current_time;
3155 sess->last_rtcp_check_time = current_time;
3156 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3158 if (interval != GST_CLOCK_TIME_NONE)
3159 sess->next_rtcp_check_time = current_time + interval;
3161 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
3162 sess->last_rtcp_interval = interval;
3164 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
3165 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
3167 RTP_SESSION_UNLOCK (sess);
3168 /* notify app of reconsideration */
3169 if (sess->callbacks.reconsider)
3170 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3171 RTP_SESSION_LOCK (sess);
3178 * rtp_session_schedule_bye:
3179 * @sess: an #RTPSession
3180 * @current_time: the current system time
3182 * Schedule a BYE message for all sources marked as BYE in @sess.
3184 * Returns: a #GstFlowReturn.
3187 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
3189 GstFlowReturn result;
3191 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3193 RTP_SESSION_LOCK (sess);
3194 result = rtp_session_schedule_bye_locked (sess, current_time);
3195 RTP_SESSION_UNLOCK (sess);
3201 * rtp_session_next_timeout:
3202 * @sess: an #RTPSession
3203 * @current_time: the current system time
3205 * Get the next time we should perform session maintenance tasks.
3207 * Returns: a time when rtp_session_on_timeout() should be called with the
3208 * current system time.
3211 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
3213 GstClockTime result, interval = 0;
3215 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
3217 RTP_SESSION_LOCK (sess);
3219 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3220 GST_DEBUG ("have early rtcp time");
3221 result = sess->next_early_rtcp_time;
3225 result = sess->next_rtcp_check_time;
3227 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3228 ", next time: %" GST_TIME_FORMAT,
3229 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3231 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
3232 GST_DEBUG ("take current time as base");
3233 /* our previous check time expired, start counting from the current time
3235 result = current_time;
3238 if (sess->scheduled_bye) {
3239 if (sess->bye_stats.active_sources >= 50) {
3240 GST_DEBUG ("reconsider BYE, more than 50 sources");
3241 /* reconsider BYE if members >= 50 */
3242 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3243 sess->last_rtcp_interval = interval;
3246 if (sess->first_rtcp) {
3247 GST_DEBUG ("first RTCP packet");
3248 /* we are called for the first time */
3249 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3250 sess->last_rtcp_interval = interval;
3251 } else if (sess->next_rtcp_check_time < current_time) {
3252 GST_DEBUG ("old check time expired, getting new timeout");
3253 /* get a new timeout when we need to */
3254 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
3255 sess->last_rtcp_interval = interval;
3257 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
3258 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
3259 && interval != GST_CLOCK_TIME_NONE) {
3260 /* Apply the rules from RFC 4585 section 3.5.3 */
3261 if (sess->stats.min_interval != 0) {
3262 GstClockTime T_rr_current_interval = g_random_double_range (0.5,
3263 1.5) * sess->stats.min_interval * GST_SECOND;
3265 if (T_rr_current_interval > interval) {
3266 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
3267 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
3268 GST_TIME_ARGS (interval));
3269 interval = T_rr_current_interval;
3276 if (interval != GST_CLOCK_TIME_NONE)
3279 result = GST_CLOCK_TIME_NONE;
3281 sess->next_rtcp_check_time = result;
3285 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3286 ", next time: %" GST_TIME_FORMAT,
3287 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3288 RTP_SESSION_UNLOCK (sess);
3302 GstRTCPBuffer rtcpbuf;
3305 guint num_to_report;
3310 GstClockTime current_time;
3312 GstClockTime running_time;
3313 GstClockTime interval;
3314 GstRTCPPacket packet;
3317 gboolean may_suppress;
3319 guint nacked_seqnums;
3323 session_start_rtcp (RTPSession * sess, ReportData * data)
3325 GstRTCPPacket *packet = &data->packet;
3326 RTPSource *own = data->source;
3327 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3329 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
3330 data->has_sdes = FALSE;
3332 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3334 if (data->is_early && sess->reduced_size_rtcp)
3337 if (RTP_SOURCE_IS_SENDER (own)) {
3340 guint32 packet_count, octet_count;
3342 /* we are a sender, create SR */
3343 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
3344 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
3346 /* get latest stats */
3347 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
3348 &ntptime, &rtptime, &packet_count, &octet_count);
3350 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
3351 packet_count, octet_count);
3353 /* fill in sender report info */
3354 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
3355 ntptime, rtptime, packet_count, octet_count);
3357 /* we are only receiver, create RR */
3358 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3359 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3360 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3364 /* construct a Sender or Receiver Report */
3366 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3368 RTPSession *sess = data->sess;
3369 GstRTCPPacket *packet = &data->packet;
3370 guint8 fractionlost;
3372 guint32 exthighestseq, jitter;
3375 /* don't report for sources in future generations */
3376 if (((gint16) (source->generation - sess->generation)) > 0) {
3377 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3378 source->generation, sess->generation);
3382 if (g_hash_table_contains (source->reported_in_sr_of,
3383 GUINT_TO_POINTER (data->source->ssrc))) {
3384 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3388 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3389 GST_DEBUG ("max RB count reached");
3393 /* only report about remote sources */
3394 if (source->internal)
3397 if (!RTP_SOURCE_IS_SENDER (source)) {
3398 GST_DEBUG ("source %08x not sender", source->ssrc);
3402 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3405 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3406 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3408 /* store last generated RR packet */
3409 source->last_rr.is_valid = TRUE;
3410 source->last_rr.fractionlost = fractionlost;
3411 source->last_rr.packetslost = packetslost;
3412 source->last_rr.exthighestseq = exthighestseq;
3413 source->last_rr.jitter = jitter;
3414 source->last_rr.lsr = lsr;
3415 source->last_rr.dlsr = dlsr;
3417 /* packet is not yet filled, add report block for this source. */
3418 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3419 exthighestseq, jitter, lsr, dlsr);
3422 g_hash_table_add (source->reported_in_sr_of,
3423 GUINT_TO_POINTER (data->source->ssrc));
3428 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3430 GstRTCPPacket *packet = &data->packet;
3434 if (!source->send_fir)
3437 len = gst_rtcp_packet_fb_get_fci_length (packet);
3438 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3439 /* exit because the packet is full, will put next request in a
3443 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3445 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3447 fci_data[0] = source->current_send_fir_seqnum;
3448 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3450 source->send_fir = FALSE;
3451 source->stats.sent_fir_count++;
3455 session_fir (RTPSession * sess, ReportData * data)
3457 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3458 GstRTCPPacket *packet = &data->packet;
3460 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3463 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3464 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3465 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3467 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3468 (GHFunc) session_add_fir, data);
3470 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3471 gst_rtcp_packet_remove (packet);
3473 data->may_suppress = FALSE;
3477 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3479 GstRTCPPacket packet;
3480 GstRTCPBuffer rtcp = { NULL, };
3481 gboolean ret = FALSE;
3483 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3485 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3486 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3487 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3491 gst_rtcp_buffer_unmap (&rtcp);
3498 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3500 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3501 GstRTCPPacket *packet = &data->packet;
3503 if (!source->send_pli)
3506 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3509 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3510 /* exit because the packet is full, will put next request in a
3514 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3515 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3516 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3518 source->send_pli = FALSE;
3519 data->may_suppress = FALSE;
3521 source->stats.sent_pli_count++;
3524 /* construct NACK */
3526 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3528 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3529 GstRTCPPacket *packet = &data->packet;
3534 if (!source->send_nack)
3537 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3538 /* exit because the packet is full, will put next request in a
3542 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3543 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3544 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3546 nacks = rtp_source_get_nacks (source, &n_nacks);
3547 GST_DEBUG ("%u NACKs", n_nacks);
3548 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
3551 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3552 for (i = 0; i < n_nacks; i++) {
3553 GST_WRITE_UINT32_BE (fci_data, nacks[i]);
3555 data->nacked_seqnums++;
3558 rtp_source_clear_nacks (source);
3559 data->may_suppress = FALSE;
3560 source->stats.sent_nack_count += n_nacks;
3563 /* perform cleanup of sources that timed out */
3565 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3567 gboolean remove = FALSE;
3568 gboolean byetimeout = FALSE;
3569 gboolean sendertimeout = FALSE;
3570 gboolean is_sender, is_active;
3571 RTPSession *sess = data->sess;
3572 GstClockTime interval, binterval;
3575 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3577 /* check for outdated collisions */
3578 if (source->internal) {
3579 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3580 rtp_source_timeout (source, data->current_time,
3581 data->running_time - sess->rtcp_feedback_retention_window);
3584 /* nothing else to do when without RTCP */
3585 if (data->interval == GST_CLOCK_TIME_NONE)
3588 is_sender = RTP_SOURCE_IS_SENDER (source);
3589 is_active = RTP_SOURCE_IS_ACTIVE (source);
3591 /* our own rtcp interval may have been forced low by secondary configuration,
3592 * while sender side may still operate with higher interval,
3593 * so do not just take our interval to decide on timing out sender,
3594 * but take (if data->interval <= 5 * GST_SECOND):
3595 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3596 * where sender_interval is difference between last 2 received RTCP reports
3598 if (data->interval >= 5 * GST_SECOND || source->internal) {
3599 binterval = data->interval;
3601 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3602 GST_TIME_ARGS (source->stats.prev_rtcptime),
3603 GST_TIME_ARGS (source->stats.last_rtcptime));
3604 /* if not received enough yet, fallback to larger default */
3605 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3606 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3608 binterval = 5 * GST_SECOND;
3609 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3611 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3612 GST_TIME_ARGS (binterval));
3614 if (!source->internal && source->marked_bye) {
3615 /* if we received a BYE from the source, remove the source after some
3617 if (data->current_time > source->bye_time &&
3618 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3619 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3625 if (source->internal && source->sent_bye) {
3626 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3630 /* sources that were inactive for more than 5 times the deterministic reporting
3631 * interval get timed out. the min timeout is 5 seconds. */
3632 /* mind old time that might pre-date last time going to PLAYING */
3633 btime = MAX (source->last_activity, sess->start_time);
3634 if (data->current_time > btime) {
3635 interval = MAX (binterval * 5, 5 * GST_SECOND);
3636 if (data->current_time - btime > interval) {
3637 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3638 source->ssrc, GST_TIME_ARGS (btime));
3639 if (source->internal) {
3640 /* this is an internal source that is not using our suggested ssrc.
3641 * since there must be another source using this ssrc, we can remove
3642 * this one instead of making it a receiver forever */
3643 if (source->ssrc != sess->suggested_ssrc) {
3644 rtp_source_mark_bye (source, "timed out");
3645 /* do not schedule bye here, since we are inside the RTCP timeout
3646 * processing and scheduling bye will interfere with SR/RR sending */
3654 /* senders that did not send for a long time become a receiver, this also
3655 * holds for our own sources. */
3657 /* mind old time that might pre-date last time going to PLAYING */
3658 btime = MAX (source->last_rtp_activity, sess->start_time);
3659 if (data->current_time > btime) {
3660 interval = MAX (binterval * 2, 5 * GST_SECOND);
3661 if (data->current_time - btime > interval) {
3662 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3663 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3664 sendertimeout = TRUE;
3670 sess->total_sources--;
3672 sess->stats.sender_sources--;
3673 if (source->internal)
3674 sess->stats.internal_sender_sources--;
3677 sess->stats.active_sources--;
3679 if (source->internal)
3680 sess->stats.internal_sources--;
3683 on_bye_timeout (sess, source);
3685 on_timeout (sess, source);
3687 if (sendertimeout) {
3688 source->is_sender = FALSE;
3689 sess->stats.sender_sources--;
3690 if (source->internal)
3691 sess->stats.internal_sender_sources--;
3693 on_sender_timeout (sess, source);
3695 /* count how many source to report in this generation */
3696 if (((gint16) (source->generation - sess->generation)) <= 0)
3697 data->num_to_report++;
3699 source->closing = remove;
3703 session_sdes (RTPSession * sess, ReportData * data)
3705 GstRTCPPacket *packet = &data->packet;
3706 const GstStructure *sdes;
3708 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3710 /* add SDES packet */
3711 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3713 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3715 sdes = rtp_source_get_sdes_struct (data->source);
3717 /* add all fields in the structure, the order is not important. */
3718 n_fields = gst_structure_n_fields (sdes);
3719 for (i = 0; i < n_fields; ++i) {
3722 GstRTCPSDESType type;
3724 field = gst_structure_nth_field_name (sdes, i);
3727 value = gst_structure_get_string (sdes, field);
3730 type = gst_rtcp_sdes_name_to_type (field);
3732 /* Early packets are minimal and only include the CNAME */
3733 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3736 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3737 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3738 (const guint8 *) value);
3739 } else if (type == GST_RTCP_SDES_PRIV) {
3745 /* don't accept entries that are too big */
3746 prefix_len = strlen (field);
3747 if (prefix_len > 255)
3749 value_len = strlen (value);
3750 if (value_len > 255)
3752 data_len = 1 + prefix_len + value_len;
3756 data[0] = prefix_len;
3757 memcpy (&data[1], field, prefix_len);
3758 memcpy (&data[1 + prefix_len], value, value_len);
3760 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3764 data->has_sdes = TRUE;
3767 /* schedule a BYE packet */
3769 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3771 GstRTCPPacket *packet = &data->packet;
3772 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3775 session_sdes (sess, data);
3776 /* add a BYE packet */
3777 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3778 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3779 if (source->bye_reason)
3780 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3782 /* we have a BYE packet now */
3783 source->sent_bye = TRUE;
3787 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3789 GstClockTime new_send_time;
3790 GstClockTime interval;
3791 RTPSessionStats *stats;
3793 if (sess->scheduled_bye)
3794 stats = &sess->bye_stats;
3796 stats = &sess->stats;
3798 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3799 data->is_early = TRUE;
3801 data->is_early = FALSE;
3803 if (data->is_early && sess->next_early_rtcp_time < current_time) {
3804 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
3805 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3806 GST_TIME_ARGS (current_time));
3807 } else if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
3808 sess->next_rtcp_check_time > current_time) {
3809 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
3810 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
3811 GST_TIME_ARGS (current_time));
3815 /* take interval and add jitter */
3816 interval = data->interval;
3817 if (interval != GST_CLOCK_TIME_NONE)
3818 interval = rtp_stats_add_rtcp_jitter (stats, interval);
3820 if (sess->last_rtcp_check_time != GST_CLOCK_TIME_NONE) {
3821 /* perform forward reconsideration */
3822 if (interval != GST_CLOCK_TIME_NONE) {
3823 GstClockTime elapsed;
3825 /* get elapsed time since we last reported */
3826 elapsed = current_time - sess->last_rtcp_check_time;
3828 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
3829 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
3830 new_send_time = interval + sess->last_rtcp_check_time;
3832 new_send_time = sess->last_rtcp_check_time;
3835 /* If this is the first RTCP packet, we can reconsider anything based
3836 * on the last RTCP send time because there was none.
3838 g_warn_if_fail (!data->is_early);
3839 data->is_early = FALSE;
3840 new_send_time = current_time;
3843 if (!data->is_early) {
3844 /* check if reconsideration */
3845 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
3846 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
3847 GST_TIME_ARGS (new_send_time));
3848 /* store new check time */
3849 sess->next_rtcp_check_time = new_send_time;
3850 sess->last_rtcp_interval = interval;
3854 sess->last_rtcp_interval = interval;
3855 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
3856 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
3857 && interval != GST_CLOCK_TIME_NONE) {
3858 /* Apply the rules from RFC 4585 section 3.5.3 */
3859 if (stats->min_interval != 0 && !sess->first_rtcp) {
3860 GstClockTime T_rr_current_interval =
3861 g_random_double_range (0.5, 1.5) * stats->min_interval * GST_SECOND;
3863 if (T_rr_current_interval > interval) {
3864 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
3865 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
3866 GST_TIME_ARGS (interval));
3867 interval = T_rr_current_interval;
3871 sess->next_rtcp_check_time = current_time + interval;
3875 GST_DEBUG ("can send RTCP now, next %" GST_TIME_FORMAT,
3876 GST_TIME_ARGS (sess->next_rtcp_check_time));
3882 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
3884 g_hash_table_insert (hash_table, key, g_object_ref (source));
3888 remove_closing_sources (const gchar * key, RTPSource * source,
3891 if (source->closing)
3894 if (source->send_fir)
3895 data->have_fir = TRUE;
3896 if (source->send_pli)
3897 data->have_pli = TRUE;
3898 if (source->send_nack)
3899 data->have_nack = TRUE;
3905 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
3907 RTPSession *sess = data->sess;
3908 gboolean is_bye = FALSE;
3909 ReportOutput *output;
3911 /* only generate RTCP for active internal sources */
3912 if (!source->internal || source->sent_bye)
3915 /* ignore other sources when we do the timeout after a scheduled BYE */
3916 if (sess->scheduled_bye && !source->marked_bye)
3919 data->source = source;
3922 session_start_rtcp (sess, data);
3924 if (source->marked_bye) {
3926 make_source_bye (sess, source, data);
3928 } else if (!data->is_early) {
3929 /* loop over all known sources and add report blocks. If we are early, we
3930 * just make a minimal RTCP packet and skip this step */
3931 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3932 (GHFunc) session_report_blocks, data);
3934 if (!data->has_sdes && (!data->is_early || !sess->reduced_size_rtcp))
3935 session_sdes (sess, data);
3938 session_fir (sess, data);
3941 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3942 (GHFunc) session_pli, data);
3944 if (data->have_nack)
3945 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3946 (GHFunc) session_nack, data);
3948 gst_rtcp_buffer_unmap (&data->rtcpbuf);
3950 output = g_slice_new (ReportOutput);
3951 output->source = g_object_ref (source);
3952 output->is_bye = is_bye;
3953 output->buffer = data->rtcp;
3954 /* queue the RTCP packet to push later */
3955 g_queue_push_tail (&data->output, output);
3959 update_generation (const gchar * key, RTPSource * source, ReportData * data)
3961 RTPSession *sess = data->sess;
3963 if (g_hash_table_size (source->reported_in_sr_of) >=
3964 sess->stats.internal_sources) {
3965 /* source is reported, move to next generation */
3966 source->generation = sess->generation + 1;
3967 g_hash_table_remove_all (source->reported_in_sr_of);
3969 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
3970 source->generation);
3972 /* if we reported all sources in this generation, move to next */
3973 if (--data->num_to_report == 0) {
3975 GST_DEBUG ("all reported, generation now %u", sess->generation);
3981 rtp_session_are_all_sources_bye (RTPSession * sess)
3983 GHashTableIter iter;
3986 RTP_SESSION_LOCK (sess);
3987 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
3988 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
3989 if (src->internal && !src->sent_bye) {
3990 RTP_SESSION_UNLOCK (sess);
3994 RTP_SESSION_UNLOCK (sess);
4000 * rtp_session_on_timeout:
4001 * @sess: an #RTPSession
4002 * @current_time: the current system time
4003 * @ntpnstime: the current NTP time in nanoseconds
4004 * @running_time: the current running_time of the pipeline
4006 * Perform maintenance actions after the timeout obtained with
4007 * rtp_session_next_timeout() expired.
4009 * This function will perform timeouts of receivers and senders, send a BYE
4010 * packet or generate RTCP packets with current session stats.
4012 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
4013 * times, for each packet that should be processed.
4015 * Returns: a #GstFlowReturn.
4018 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
4019 guint64 ntpnstime, GstClockTime running_time)
4021 GstFlowReturn result = GST_FLOW_OK;
4022 ReportData data = { GST_RTCP_BUFFER_INIT };
4023 GHashTable *table_copy;
4024 ReportOutput *output;
4025 gboolean all_empty = FALSE;
4027 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
4029 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
4030 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4031 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
4034 data.current_time = current_time;
4035 data.ntpnstime = ntpnstime;
4036 data.running_time = running_time;
4037 data.num_to_report = 0;
4038 data.may_suppress = FALSE;
4039 data.nacked_seqnums = 0;
4040 g_queue_init (&data.output);
4042 RTP_SESSION_LOCK (sess);
4043 /* get a new interval, we need this for various cleanups etc */
4044 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
4046 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
4048 /* we need an internal source now */
4049 if (sess->stats.internal_sources == 0) {
4053 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
4055 sess->internal_ssrc_set = TRUE;
4058 on_new_sender_ssrc (sess, source);
4060 g_object_unref (source);
4063 sess->conflicting_addresses =
4064 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
4066 /* Make a local copy of the hashtable. We need to do this because the
4067 * cleanup stage below releases the session lock. */
4068 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
4069 (GDestroyNotify) g_object_unref);
4070 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4071 (GHFunc) clone_ssrcs_hashtable, table_copy);
4073 /* Clean up the session, mark the source for removing, this might release the
4075 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
4076 g_hash_table_destroy (table_copy);
4078 /* Now remove the marked sources */
4079 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
4080 (GHRFunc) remove_closing_sources, &data);
4082 /* update point-to-point status */
4083 session_update_ptp (sess);
4085 /* see if we need to generate SR or RR packets */
4086 if (!is_rtcp_time (sess, current_time, &data))
4089 /* check if all the buffers are empty afer generation */
4093 ("doing RTCP generation %u for %u sources, early %d, may suppress %d",
4094 sess->generation, data.num_to_report, data.is_early, data.may_suppress);
4096 /* generate RTCP for all internal sources */
4097 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4098 (GHFunc) generate_rtcp, &data);
4100 /* update the generation for all the sources that have been reported */
4101 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4102 (GHFunc) update_generation, &data);
4104 /* we keep track of the last report time in order to timeout inactive
4105 * receivers or senders */
4106 if (!data.is_early) {
4107 GST_DEBUG ("Time since last regular RTCP: %" GST_TIME_FORMAT " - %"
4108 GST_TIME_FORMAT " = %" GST_TIME_FORMAT,
4109 GST_TIME_ARGS (data.current_time),
4110 GST_TIME_ARGS (sess->last_rtcp_send_time),
4111 GST_TIME_ARGS (data.current_time - sess->last_rtcp_send_time));
4112 sess->last_rtcp_send_time = data.current_time;
4115 GST_DEBUG ("Time since last RTCP: %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT
4116 " = %" GST_TIME_FORMAT, GST_TIME_ARGS (data.current_time),
4117 GST_TIME_ARGS (sess->last_rtcp_check_time),
4118 GST_TIME_ARGS (data.current_time - sess->last_rtcp_check_time));
4119 sess->last_rtcp_check_time = data.current_time;
4120 sess->first_rtcp = FALSE;
4121 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
4122 sess->scheduled_bye = FALSE;
4125 RTP_SESSION_UNLOCK (sess);
4127 /* notify about updated statistics */
4128 g_object_notify (G_OBJECT (sess), "stats");
4130 /* push out the RTCP packets */
4131 while ((output = g_queue_pop_head (&data.output))) {
4132 gboolean do_not_suppress, empty_buffer;
4133 GstBuffer *buffer = output->buffer;
4134 RTPSource *source = output->source;
4136 /* Give the user a change to add its own packet */
4137 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
4138 buffer, data.is_early, &do_not_suppress);
4140 empty_buffer = gst_buffer_get_size (buffer) == 0;
4145 if (sess->callbacks.send_rtcp &&
4146 !empty_buffer && (do_not_suppress || !data.may_suppress)) {
4149 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
4151 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
4152 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
4153 sess->stats.avg_rtcp_packet_size, packet_size);
4155 sess->callbacks.send_rtcp (sess, source, buffer,
4156 rtp_session_are_all_sources_bye (sess), sess->send_rtcp_user_data);
4158 RTP_SESSION_LOCK (sess);
4159 sess->stats.nacks_sent += data.nacked_seqnums;
4160 on_sender_ssrc_active (sess, source);
4161 RTP_SESSION_UNLOCK (sess);
4163 GST_DEBUG ("freeing packet callback: %p"
4164 " empty_buffer: %d, "
4165 " do_not_suppress: %d may_suppress: %d", sess->callbacks.send_rtcp,
4166 empty_buffer, do_not_suppress, data.may_suppress);
4167 if (!empty_buffer) {
4168 RTP_SESSION_LOCK (sess);
4169 sess->stats.nacks_dropped += data.nacked_seqnums;
4170 RTP_SESSION_UNLOCK (sess);
4172 gst_buffer_unref (buffer);
4174 g_object_unref (source);
4175 g_slice_free (ReportOutput, output);
4179 GST_ERROR ("generated empty RTCP messages for all the sources");
4185 * rtp_session_request_early_rtcp:
4186 * @sess: an #RTPSession
4187 * @current_time: the current system time
4188 * @max_delay: maximum delay
4190 * Request transmission of early RTCP
4192 * Returns: %TRUE if the related RTCP can be scheduled.
4195 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
4196 GstClockTime max_delay)
4198 GstClockTime T_dither_max, T_rr, offset = 0;
4200 gboolean allow_early;
4202 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
4204 RTP_SESSION_LOCK (sess);
4206 /* We assume a feedback profile if something is requesting RTCP
4208 sess->rtp_profile = GST_RTP_PROFILE_AVPF;
4210 /* Check if already requested */
4211 /* RFC 4585 section 3.5.2 step 2 */
4212 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
4213 GST_LOG_OBJECT (sess, "already have next early rtcp time");
4214 ret = (current_time + max_delay > sess->next_early_rtcp_time);
4218 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
4219 GST_LOG_OBJECT (sess, "no next RTCP check time");
4224 /* RFC 4585 section 3.5.3 step 1
4225 * If no regular RTCP packet has been sent before, then a regular
4226 * RTCP packet has to be scheduled first and FB messages might be
4229 if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
4230 GST_LOG_OBJECT (sess, "no RTCP sent yet");
4232 if (current_time + max_delay > sess->next_rtcp_check_time) {
4233 GST_LOG_OBJECT (sess,
4234 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4235 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4236 GST_TIME_ARGS (max_delay),
4237 GST_TIME_ARGS (sess->next_rtcp_check_time));
4240 GST_LOG_OBJECT (sess,
4241 "can't allow early feedback, next scheduled time is too late %"
4242 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4243 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4244 GST_TIME_ARGS (sess->next_rtcp_check_time));
4250 T_rr = sess->last_rtcp_interval;
4252 /* RFC 4585 section 3.5.2 step 2b */
4253 /* If the total sources is <=2, then there is only us and one peer */
4254 /* When there is one auxiliary stream the session can still do point
4257 if (sess->is_doing_ptp) {
4260 /* Divide by 2 because l = 0.5 */
4261 T_dither_max = T_rr;
4265 /* RFC 4585 section 3.5.2 step 3 */
4266 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
4267 GST_LOG_OBJECT (sess,
4268 "don't send because of dither, next scheduled time is too soon %"
4269 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
4270 GST_TIME_ARGS (current_time), GST_TIME_ARGS (T_dither_max),
4271 GST_TIME_ARGS (sess->next_rtcp_check_time));
4272 ret = T_dither_max <= max_delay;
4276 /* RFC 4585 section 3.5.2 step 4a and
4277 * RFC 4585 section 3.5.2 step 6 */
4278 allow_early = FALSE;
4279 if (sess->last_rtcp_check_time == sess->last_rtcp_send_time) {
4280 /* Last time we sent a full RTCP packet, we can now immediately
4281 * send an early one as allow_early was reset to TRUE */
4283 } else if (sess->last_rtcp_check_time + T_rr <= current_time + max_delay) {
4284 /* Last packet we sent was an early RTCP packet and more than
4285 * T_rr has passed since then, meaning we would have suppressed
4286 * a regular RTCP packet already and reset allow_early to TRUE */
4289 /* We have to offset a bit as T_rr has not passed yet, but will before
4291 if (sess->last_rtcp_check_time + T_rr > current_time)
4292 offset = (sess->last_rtcp_check_time + T_rr) - current_time;
4294 GST_DEBUG_OBJECT (sess,
4295 "can't allow early RTCP yet: last regular %" GST_TIME_FORMAT ", %"
4296 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT " + %"
4297 GST_TIME_FORMAT, GST_TIME_ARGS (sess->last_rtcp_send_time),
4298 GST_TIME_ARGS (sess->last_rtcp_check_time), GST_TIME_ARGS (T_rr),
4299 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay));
4303 /* Ignore the request a scheduled packet will be in time anyway */
4304 if (current_time + max_delay > sess->next_rtcp_check_time) {
4305 GST_LOG_OBJECT (sess,
4306 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4307 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4308 GST_TIME_ARGS (max_delay),
4309 GST_TIME_ARGS (sess->next_rtcp_check_time));
4312 GST_LOG_OBJECT (sess,
4313 "can't allow early feedback and next scheduled time is too late %"
4314 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4315 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4316 GST_TIME_ARGS (sess->next_rtcp_check_time));
4322 /* RFC 4585 section 3.5.2 step 4b */
4324 /* Schedule an early transmission later */
4325 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
4326 current_time + offset;
4328 /* If no dithering, schedule it for NOW */
4329 sess->next_early_rtcp_time = current_time + offset;
4332 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT
4333 ", next regular RTCP time %" GST_TIME_FORMAT,
4334 GST_TIME_ARGS (sess->next_early_rtcp_time),
4335 GST_TIME_ARGS (sess->next_rtcp_check_time));
4336 RTP_SESSION_UNLOCK (sess);
4338 /* notify app of need to send packet early
4339 * and therefore of timeout change */
4340 if (sess->callbacks.reconsider)
4341 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
4347 RTP_SESSION_UNLOCK (sess);
4353 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
4357 if (!sess->callbacks.send_rtcp)
4360 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
4362 return rtp_session_request_early_rtcp (sess, now, max_delay);
4366 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
4367 gboolean fir, gint count)
4371 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
4372 GST_DEBUG ("FIR/PLI not sent");
4376 RTP_SESSION_LOCK (sess);
4377 src = find_source (sess, ssrc);
4382 src->send_pli = FALSE;
4383 src->send_fir = TRUE;
4385 if (count == -1 || count != src->last_fir_count)
4386 src->current_send_fir_seqnum++;
4387 src->last_fir_count = count;
4388 } else if (!src->send_fir) {
4389 src->send_pli = TRUE;
4391 RTP_SESSION_UNLOCK (sess);
4398 RTP_SESSION_UNLOCK (sess);
4404 * rtp_session_request_nack:
4405 * @sess: a #RTPSession
4407 * @seqnum: the missing seqnum
4408 * @max_delay: max delay to request NACK
4410 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
4412 * Returns: %TRUE if the NACK feedback could be scheduled
4415 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
4416 GstClockTime max_delay)
4420 if (!rtp_session_send_rtcp (sess, max_delay)) {
4421 GST_DEBUG ("NACK not sent");
4425 RTP_SESSION_LOCK (sess);
4426 source = find_source (sess, ssrc);
4430 GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
4431 rtp_source_register_nack (source, seqnum);
4432 RTP_SESSION_UNLOCK (sess);
4439 RTP_SESSION_UNLOCK (sess);