1 /* GStreamer
2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
3 *
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.
8 *
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.
13 *
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.
18 */
19
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
23
24 #include <string.h>
25
26 #include <gst/rtp/gstrtpbuffer.h>
27 #include <gst/rtp/gstrtcpbuffer.h>
28
29 #include <gst/glib-compat-private.h>
30
31 #include "rtpsession.h"
32
33 GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
34 #define GST_CAT_DEFAULT rtp_session_debug
35
36 /* signals and args */
37 enum
38 {
39 SIGNAL_GET_SOURCE_BY_SSRC,
40 SIGNAL_ON_NEW_SSRC,
41 SIGNAL_ON_SSRC_COLLISION,
42 SIGNAL_ON_SSRC_VALIDATED,
43 SIGNAL_ON_SSRC_ACTIVE,
44 SIGNAL_ON_SSRC_SDES,
45 SIGNAL_ON_BYE_SSRC,
46 SIGNAL_ON_BYE_TIMEOUT,
47 SIGNAL_ON_TIMEOUT,
48 SIGNAL_ON_SENDER_TIMEOUT,
49 SIGNAL_ON_SENDING_RTCP,
50 SIGNAL_ON_APP_RTCP,
51 SIGNAL_ON_FEEDBACK_RTCP,
52 SIGNAL_SEND_RTCP,
53 SIGNAL_SEND_RTCP_FULL,
54 SIGNAL_ON_RECEIVING_RTCP,
55 SIGNAL_ON_NEW_SENDER_SSRC,
56 SIGNAL_ON_SENDER_SSRC_ACTIVE,
57 SIGNAL_ON_SENDING_NACKS,
58 LAST_SIGNAL
59 };
60
61 #define DEFAULT_INTERNAL_SOURCE NULL
62 #define DEFAULT_BANDWIDTH 0.0
63 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
64 #define DEFAULT_RTCP_RR_BANDWIDTH -1
65 #define DEFAULT_RTCP_RS_BANDWIDTH -1
66 #define DEFAULT_RTCP_MTU 1400
67 #define DEFAULT_SDES NULL
68 #define DEFAULT_NUM_SOURCES 0
69 #define DEFAULT_NUM_ACTIVE_SOURCES 0
70 #define DEFAULT_SOURCES NULL
71 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
72 #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
73 #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
74 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
75 #define DEFAULT_MAX_DROPOUT_TIME 60000
76 #define DEFAULT_MAX_MISORDER_TIME 2000
77 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
78 #define DEFAULT_RTCP_REDUCED_SIZE FALSE
79 #define DEFAULT_RTCP_DISABLE_SR_TIMESTAMP FALSE
80
81 enum
82 {
83 PROP_0,
84 PROP_INTERNAL_SSRC,
85 PROP_INTERNAL_SOURCE,
86 PROP_BANDWIDTH,
87 PROP_RTCP_FRACTION,
88 PROP_RTCP_RR_BANDWIDTH,
89 PROP_RTCP_RS_BANDWIDTH,
90 PROP_RTCP_MTU,
91 PROP_SDES,
92 PROP_NUM_SOURCES,
93 PROP_NUM_ACTIVE_SOURCES,
94 PROP_SOURCES,
95 PROP_FAVOR_NEW,
96 PROP_RTCP_MIN_INTERVAL,
97 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
98 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
99 PROP_PROBATION,
100 PROP_MAX_DROPOUT_TIME,
101 PROP_MAX_MISORDER_TIME,
102 PROP_STATS,
103 PROP_RTP_PROFILE,
104 PROP_RTCP_REDUCED_SIZE,
105 PROP_RTCP_DISABLE_SR_TIMESTAMP
106 };
107
108 /* update average packet size */
109 #define INIT_AVG(avg, val) \
110 (avg) = (val);
111 #define UPDATE_AVG(avg, val) \
112 if ((avg) == 0) \
113 (avg) = (val); \
114 else \
115 (avg) = ((val) + (15 * (avg))) >> 4;
116
117
118 /* GObject vmethods */
119 static void rtp_session_finalize (GObject * object);
120 static void rtp_session_set_property (GObject * object, guint prop_id,
121 const GValue * value, GParamSpec * pspec);
122 static void rtp_session_get_property (GObject * object, guint prop_id,
123 GValue * value, GParamSpec * pspec);
124
125 static gboolean rtp_session_send_rtcp (RTPSession * sess,
126 GstClockTime max_delay);
127 static gboolean rtp_session_send_rtcp_with_deadline (RTPSession * sess,
128 GstClockTime deadline);
129
130 static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
131
132 G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
133
134 static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
135 static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
136 gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
137 static RTPSource *obtain_internal_source (RTPSession * sess,
138 guint32 ssrc, gboolean * created, GstClockTime current_time);
139 static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
140 GstClockTime current_time);
141 static GstClockTime calculate_rtcp_interval (RTPSession * sess,
142 gboolean deterministic, gboolean first);
143
144 static gboolean
accumulate_trues(GSignalInvocationHint * ihint,GValue * return_accu,const GValue * handler_return,gpointer data)145 accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
146 const GValue * handler_return, gpointer data)
147 {
148 if (g_value_get_boolean (handler_return))
149 g_value_set_boolean (return_accu, TRUE);
150
151 return TRUE;
152 }
153
154 static void
rtp_session_class_init(RTPSessionClass * klass)155 rtp_session_class_init (RTPSessionClass * klass)
156 {
157 GObjectClass *gobject_class;
158
159 gobject_class = (GObjectClass *) klass;
160
161 gobject_class->finalize = rtp_session_finalize;
162 gobject_class->set_property = rtp_session_set_property;
163 gobject_class->get_property = rtp_session_get_property;
164
165 /**
166 * RTPSession::get-source-by-ssrc:
167 * @session: the object which received the signal
168 * @ssrc: the SSRC of the RTPSource
169 *
170 * Request the #RTPSource object with SSRC @ssrc in @session.
171 */
172 rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
173 g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
174 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
175 get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
176 RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
177
178 /**
179 * RTPSession::on-new-ssrc:
180 * @session: the object which received the signal
181 * @src: the new RTPSource
182 *
183 * Notify of a new SSRC that entered @session.
184 */
185 rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
186 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
188 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
189 RTP_TYPE_SOURCE);
190 /**
191 * RTPSession::on-ssrc-collision:
192 * @session: the object which received the signal
193 * @src: the #RTPSource that caused a collision
194 *
195 * Notify when we have an SSRC collision
196 */
197 rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
198 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
199 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
200 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
201 RTP_TYPE_SOURCE);
202 /**
203 * RTPSession::on-ssrc-validated:
204 * @session: the object which received the signal
205 * @src: the new validated RTPSource
206 *
207 * Notify of a new SSRC that became validated.
208 */
209 rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
210 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
212 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
213 RTP_TYPE_SOURCE);
214 /**
215 * RTPSession::on-ssrc-active:
216 * @session: the object which received the signal
217 * @src: the active RTPSource
218 *
219 * Notify of a SSRC that is active, i.e., sending RTCP.
220 */
221 rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
222 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
224 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
225 RTP_TYPE_SOURCE);
226 /**
227 * RTPSession::on-ssrc-sdes:
228 * @session: the object which received the signal
229 * @src: the RTPSource
230 *
231 * Notify that a new SDES was received for SSRC.
232 */
233 rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
234 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
235 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
236 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
237 RTP_TYPE_SOURCE);
238 /**
239 * RTPSession::on-bye-ssrc:
240 * @session: the object which received the signal
241 * @src: the RTPSource that went away
242 *
243 * Notify of an SSRC that became inactive because of a BYE packet.
244 */
245 rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
246 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
248 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
249 RTP_TYPE_SOURCE);
250 /**
251 * RTPSession::on-bye-timeout:
252 * @session: the object which received the signal
253 * @src: the RTPSource that timed out
254 *
255 * Notify of an SSRC that has timed out because of BYE
256 */
257 rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
258 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
259 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
260 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
261 RTP_TYPE_SOURCE);
262 /**
263 * RTPSession::on-timeout:
264 * @session: the object which received the signal
265 * @src: the RTPSource that timed out
266 *
267 * Notify of an SSRC that has timed out
268 */
269 rtp_session_signals[SIGNAL_ON_TIMEOUT] =
270 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
271 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
272 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
273 RTP_TYPE_SOURCE);
274 /**
275 * RTPSession::on-sender-timeout:
276 * @session: the object which received the signal
277 * @src: the RTPSource that timed out
278 *
279 * Notify of an SSRC that was a sender but timed out and became a receiver.
280 */
281 rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
282 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
283 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
284 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
285 RTP_TYPE_SOURCE);
286
287 /**
288 * RTPSession::on-sending-rtcp
289 * @session: the object which received the signal
290 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
291 * @early: %TRUE if the packet is early, %FALSE if it is regular
292 *
293 * This signal is emitted before sending an RTCP packet, it can be used
294 * to add extra RTCP Packets.
295 *
296 * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
297 * if suppressing it is acceptable
298 */
299 rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
300 g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
301 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
302 accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
303 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
304
305 /**
306 * RTPSession::on-app-rtcp:
307 * @session: the object which received the signal
308 * @subtype: The subtype of the packet
309 * @ssrc: The SSRC/CSRC of the packet
310 * @name: The name of the packet
311 * @data: a #GstBuffer with the application-dependant data or %NULL if
312 * there was no data
313 *
314 * Notify that a RTCP APP packet has been received
315 */
316 rtp_session_signals[SIGNAL_ON_APP_RTCP] =
317 g_signal_new ("on-app-rtcp", G_TYPE_FROM_CLASS (klass),
318 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_app_rtcp),
319 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 4,
320 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_STRING, GST_TYPE_BUFFER);
321
322 /**
323 * RTPSession::on-feedback-rtcp:
324 * @session: the object which received the signal
325 * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
326 * %GST_RTCP_TYPE_RTPFB
327 * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
328 * @sender_ssrc: The SSRC of the sender
329 * @media_ssrc: The SSRC of the media this refers to
330 * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
331 * there was no FCI
332 *
333 * Notify that a RTCP feedback packet has been received
334 */
335 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
336 g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
337 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
338 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
339 G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
340
341 /**
342 * RTPSession::send-rtcp:
343 * @session: the object which received the signal
344 * @max_delay: The maximum delay after which the feedback will not be useful
345 * anymore
346 *
347 * Requests that the #RTPSession initiate a new RTCP packet as soon as
348 * possible within the requested delay.
349 *
350 * This sets feedback to %TRUE if not already done before.
351 */
352 rtp_session_signals[SIGNAL_SEND_RTCP] =
353 g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
354 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
355 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
356 g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
357
358 /**
359 * RTPSession::send-rtcp-full:
360 * @session: the object which received the signal
361 * @max_delay: The maximum delay after which the feedback will not be useful
362 * anymore
363 *
364 * Requests that the #RTPSession initiate a new RTCP packet as soon as
365 * possible within the requested delay.
366 *
367 * This sets feedback to %TRUE if not already done before.
368 *
369 * Returns: TRUE if the new RTCP packet could be scheduled within the
370 * requested delay, FALSE otherwise.
371 *
372 * Since: 1.6
373 */
374 rtp_session_signals[SIGNAL_SEND_RTCP_FULL] =
375 g_signal_new ("send-rtcp-full", G_TYPE_FROM_CLASS (klass),
376 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
377 G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
378 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
379
380 /**
381 * RTPSession::on-receiving-rtcp
382 * @session: the object which received the signal
383 * @buffer: the #GstBuffer containing the RTCP packet that was received
384 *
385 * This signal is emitted when receiving an RTCP packet before it is handled
386 * by the session. It can be used to extract custom information from RTCP packets.
387 *
388 * Since: 1.6
389 */
390 rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP] =
391 g_signal_new ("on-receiving-rtcp", G_TYPE_FROM_CLASS (klass),
392 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_receiving_rtcp),
393 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
394 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
395
396 /**
397 * RTPSession::on-new-sender-ssrc:
398 * @session: the object which received the signal
399 * @src: the new sender RTPSource
400 *
401 * Notify of a new sender SSRC that entered @session.
402 *
403 * Since: 1.8
404 */
405 rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
406 g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
407 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_sender_ssrc),
408 NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
409 RTP_TYPE_SOURCE);
410
411 /**
412 * RTPSession::on-sender-ssrc-active:
413 * @session: the object which received the signal
414 * @src: the active sender RTPSource
415 *
416 * Notify of a sender SSRC that is active, i.e., sending RTCP.
417 *
418 * Since: 1.8
419 */
420 rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
421 g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
422 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass,
423 on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__OBJECT,
424 G_TYPE_NONE, 1, RTP_TYPE_SOURCE);
425
426 /**
427 * RTPSession::on-sending-nack
428 * @session: the object which received the signal
429 * @sender_ssrc: the sender ssrc
430 * @media_ssrc: the media ssrc
431 * @nacks: (element-type guint16): the list of seqnum to be nacked
432 * @buffer: the #GstBuffer containing the RTCP packet about to be sent
433 *
434 * This signal is emitted before NACK packets are added into the RTCP
435 * packet. This signal can be used to override the conversion of the NACK
436 * seqnum array into packets. This can be used if your protocol uses
437 * different type of NACK (e.g. based on RTCP APP).
438 *
439 * The handler should transform the seqnum from @nacks array into packets.
440 * @nacks seqnum must be consumed from the start. The remaining will be
441 * rescheduled for later base on bandwidth. Only one handler will be
442 * signalled.
443 *
444 * A handler may return 0 to signal that generic NACKs should be created
445 * for this set. This can be useful if the signal is used for other purpose
446 * or if the other type of NACK would use more space.
447 *
448 * Returns: the number of NACK seqnum that was consumed from @nacks.
449 *
450 * Since: 1.16
451 */
452 rtp_session_signals[SIGNAL_ON_SENDING_NACKS] =
453 g_signal_new ("on-sending-nacks", G_TYPE_FROM_CLASS (klass),
454 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_nacks),
455 g_signal_accumulator_first_wins, NULL, g_cclosure_marshal_generic,
456 G_TYPE_UINT, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_ARRAY,
457 GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
458
459 g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
460 g_param_spec_uint ("internal-ssrc", "Internal SSRC",
461 "The internal SSRC used for the session (deprecated)",
462 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
463
464 g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
465 g_param_spec_object ("internal-source", "Internal Source",
466 "The internal source element of the session (deprecated)",
467 RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
468
469 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
470 g_param_spec_double ("bandwidth", "Bandwidth",
471 "The bandwidth of the session in bits per second (0 for auto-discover)",
472 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
473 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
474
475 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
476 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
477 "The fraction of the bandwidth used for RTCP in bits per second (or as a real fraction of the RTP bandwidth if < 1)",
478 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
479 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
480
481 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
482 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
483 "The RTCP bandwidth used for receivers in bits per second (-1 = default)",
484 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
485 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
486
487 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
488 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
489 "The RTCP bandwidth used for senders in bits per second (-1 = default)",
490 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
491 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
492
493 g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
494 g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
495 "The maximum size of the RTCP packets",
496 16, G_MAXINT16, DEFAULT_RTCP_MTU,
497 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
498
499 g_object_class_install_property (gobject_class, PROP_SDES,
500 g_param_spec_boxed ("sdes", "SDES",
501 "The SDES items of this session",
502 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
503
504 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
505 g_param_spec_uint ("num-sources", "Num Sources",
506 "The number of sources in the session", 0, G_MAXUINT,
507 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
508
509 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
510 g_param_spec_uint ("num-active-sources", "Num Active Sources",
511 "The number of active sources in the session", 0, G_MAXUINT,
512 DEFAULT_NUM_ACTIVE_SOURCES,
513 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
514 /**
515 * RTPSource::sources
516 *
517 * Get a GValue Array of all sources in the session.
518 *
519 * <example>
520 * <title>Getting the #RTPSources of a session
521 * <programlisting>
522 * {
523 * GValueArray *arr;
524 * GValue *val;
525 * guint i;
526 *
527 * g_object_get (sess, "sources", &arr, NULL);
528 *
529 * for (i = 0; i < arr->n_values; i++) {
530 * RTPSource *source;
531 *
532 * val = g_value_array_get_nth (arr, i);
533 * source = g_value_get_object (val);
534 * }
535 * g_value_array_free (arr);
536 * }
537 * </programlisting>
538 * </example>
539 */
540 g_object_class_install_property (gobject_class, PROP_SOURCES,
541 g_param_spec_boxed ("sources", "Sources",
542 "An array of all known sources in the session",
543 G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
544
545 g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
546 g_param_spec_boolean ("favor-new", "Favor new sources",
547 "Resolve SSRC conflict in favor of new sources", FALSE,
548 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
549
550 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
551 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
552 "Minimum interval between Regular RTCP packet (in ns)",
553 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
554 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
555
556 g_object_class_install_property (gobject_class,
557 PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
558 g_param_spec_uint64 ("rtcp-feedback-retention-window",
559 "RTCP Feedback retention window",
560 "Duration during which RTCP Feedback packets are retained (in ns)",
561 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
562 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
563
564 g_object_class_install_property (gobject_class,
565 PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
566 g_param_spec_uint ("rtcp-immediate-feedback-threshold",
567 "RTCP Immediate Feedback threshold",
568 "The maximum number of members of a RTP session for which immediate"
569 " feedback is used (DEPRECATED: has no effect and is not needed)",
570 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
571 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
572
573 g_object_class_install_property (gobject_class, PROP_PROBATION,
574 g_param_spec_uint ("probation", "Number of probations",
575 "Consecutive packet sequence numbers to accept the source",
576 0, G_MAXUINT, DEFAULT_PROBATION,
577 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
578
579 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
580 g_param_spec_uint ("max-dropout-time", "Max dropout time",
581 "The maximum time (milliseconds) of missing packets tolerated.",
582 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
583 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
584
585 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
586 g_param_spec_uint ("max-misorder-time", "Max misorder time",
587 "The maximum time (milliseconds) of misordered packets tolerated.",
588 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
589 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
590
591 /**
592 * RTPSession::stats:
593 *
594 * Various session statistics. This property returns a GstStructure
595 * with name application/x-rtp-session-stats with the following fields:
596 *
597 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
598 * dropped (due to bandwidth constraints)
599 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
600 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
601 * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
602 * RTP sources (Since 1.8)
603 *
604 * Since: 1.4
605 */
606 g_object_class_install_property (gobject_class, PROP_STATS,
607 g_param_spec_boxed ("stats", "Statistics",
608 "Various statistics", GST_TYPE_STRUCTURE,
609 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
610
611 g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
612 g_param_spec_enum ("rtp-profile", "RTP Profile",
613 "RTP profile to use for this session", GST_TYPE_RTP_PROFILE,
614 DEFAULT_RTP_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
615
616 g_object_class_install_property (gobject_class, PROP_RTCP_REDUCED_SIZE,
617 g_param_spec_boolean ("rtcp-reduced-size", "RTCP Reduced Size",
618 "Use Reduced Size RTCP for feedback packets",
619 DEFAULT_RTCP_REDUCED_SIZE,
620 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
621
622 /**
623 * RTPSession::disable-sr-timestamp:
624 *
625 * Whether sender reports should be timestamped.
626 *
627 * Since: 1.16
628 */
629 g_object_class_install_property (gobject_class,
630 PROP_RTCP_DISABLE_SR_TIMESTAMP,
631 g_param_spec_boolean ("disable-sr-timestamp",
632 "Disable Sender Report Timestamp",
633 "Whether sender reports should be timestamped",
634 DEFAULT_RTCP_DISABLE_SR_TIMESTAMP,
635 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
636
637 klass->get_source_by_ssrc =
638 GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
639 klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
640
641 GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
642 }
643
644 static void
rtp_session_init(RTPSession * sess)645 rtp_session_init (RTPSession * sess)
646 {
647 gint i;
648 gchar *str;
649
650 g_mutex_init (&sess->lock);
651 sess->key = g_random_int ();
652 sess->mask_idx = 0;
653 sess->mask = 0;
654
655 /* TODO: We currently only use the first hash table but this is the
656 * beginning of an implementation for RFC2762
657 for (i = 0; i < 32; i++) {
658 */
659 for (i = 0; i < 1; i++) {
660 sess->ssrcs[i] =
661 g_hash_table_new_full (NULL, NULL, NULL,
662 (GDestroyNotify) g_object_unref);
663 }
664
665 rtp_stats_init_defaults (&sess->stats);
666 INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
667 rtp_stats_set_min_interval (&sess->stats,
668 (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
669
670 sess->recalc_bandwidth = TRUE;
671 sess->bandwidth = DEFAULT_BANDWIDTH;
672 sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
673 sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
674 sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
675
676 /* default UDP header length */
677 sess->header_len = 28;
678 sess->mtu = DEFAULT_RTCP_MTU;
679
680 sess->probation = DEFAULT_PROBATION;
681 sess->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
682 sess->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
683
684 /* some default SDES entries */
685 sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
686
687 /* we do not want to leak details like the username or hostname here */
688 str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
689 gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
690 g_free (str);
691
692 #if 0
693 /* we do not want to leak the user's real name here */
694 str = g_strdup_printf ("Anon%u", g_random_int ());
695 gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
696 g_free (str);
697 #endif
698
699 gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
700
701 /* this is the SSRC we suggest */
702 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
703 sess->internal_ssrc_set = FALSE;
704
705 sess->first_rtcp = TRUE;
706 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
707 sess->last_rtcp_check_time = GST_CLOCK_TIME_NONE;
708 sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
709 sess->last_rtcp_interval = GST_CLOCK_TIME_NONE;
710
711 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
712 sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
713 sess->rtcp_immediate_feedback_threshold =
714 DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD;
715 sess->rtp_profile = DEFAULT_RTP_PROFILE;
716 sess->reduced_size_rtcp = DEFAULT_RTCP_REDUCED_SIZE;
717 sess->timestamp_sender_reports = !DEFAULT_RTCP_DISABLE_SR_TIMESTAMP;
718
719 sess->is_doing_ptp = TRUE;
720 }
721
722 static void
rtp_session_finalize(GObject * object)723 rtp_session_finalize (GObject * object)
724 {
725 RTPSession *sess;
726 gint i;
727
728 sess = RTP_SESSION_CAST (object);
729
730 gst_structure_free (sess->sdes);
731
732 g_list_free_full (sess->conflicting_addresses,
733 (GDestroyNotify) rtp_conflicting_address_free);
734
735 /* TODO: Change this again when implementing RFC 2762
736 * for (i = 0; i < 32; i++)
737 */
738 for (i = 0; i < 1; i++)
739 g_hash_table_destroy (sess->ssrcs[i]);
740
741 g_mutex_clear (&sess->lock);
742
743 G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
744 }
745
746 static void
copy_source(gpointer key,RTPSource * source,GValueArray * arr)747 copy_source (gpointer key, RTPSource * source, GValueArray * arr)
748 {
749 GValue value = { 0 };
750
751 g_value_init (&value, RTP_TYPE_SOURCE);
752 g_value_take_object (&value, source);
753 /* copies the value */
754 g_value_array_append (arr, &value);
755 }
756
757 static GValueArray *
rtp_session_create_sources(RTPSession * sess)758 rtp_session_create_sources (RTPSession * sess)
759 {
760 GValueArray *res;
761 guint size;
762
763 RTP_SESSION_LOCK (sess);
764 /* get number of elements in the table */
765 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
766 /* create the result value array */
767 res = g_value_array_new (size);
768
769 /* and copy all values into the array */
770 g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
771 RTP_SESSION_UNLOCK (sess);
772
773 return res;
774 }
775
776 static void
create_source_stats(gpointer key,RTPSource * source,GValueArray * arr)777 create_source_stats (gpointer key, RTPSource * source, GValueArray * arr)
778 {
779 GValue *value;
780 GstStructure *s;
781
782 g_object_get (source, "stats", &s, NULL);
783
784 g_value_array_append (arr, NULL);
785 value = g_value_array_get_nth (arr, arr->n_values - 1);
786 g_value_init (value, GST_TYPE_STRUCTURE);
787 g_value_take_boxed (value, s);
788 }
789
790 static GstStructure *
rtp_session_create_stats(RTPSession * sess)791 rtp_session_create_stats (RTPSession * sess)
792 {
793 GstStructure *s;
794 GValueArray *source_stats;
795 GValue source_stats_v = G_VALUE_INIT;
796 guint size;
797
798 RTP_SESSION_LOCK (sess);
799 s = gst_structure_new ("application/x-rtp-session-stats",
800 "rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
801 "sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
802 "recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
803
804 size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
805 source_stats = g_value_array_new (size);
806 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
807 (GHFunc) create_source_stats, source_stats);
808 RTP_SESSION_UNLOCK (sess);
809
810 g_value_init (&source_stats_v, G_TYPE_VALUE_ARRAY);
811 g_value_take_boxed (&source_stats_v, source_stats);
812 gst_structure_take_value (s, "source-stats", &source_stats_v);
813
814 return s;
815 }
816
817 static void
rtp_session_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)818 rtp_session_set_property (GObject * object, guint prop_id,
819 const GValue * value, GParamSpec * pspec)
820 {
821 RTPSession *sess;
822
823 sess = RTP_SESSION (object);
824
825 switch (prop_id) {
826 case PROP_INTERNAL_SSRC:
827 RTP_SESSION_LOCK (sess);
828 sess->suggested_ssrc = g_value_get_uint (value);
829 sess->internal_ssrc_set = TRUE;
830 sess->internal_ssrc_from_caps_or_property = TRUE;
831 RTP_SESSION_UNLOCK (sess);
832 if (sess->callbacks.reconfigure)
833 sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
834 break;
835 case PROP_BANDWIDTH:
836 RTP_SESSION_LOCK (sess);
837 sess->bandwidth = g_value_get_double (value);
838 sess->recalc_bandwidth = TRUE;
839 RTP_SESSION_UNLOCK (sess);
840 break;
841 case PROP_RTCP_FRACTION:
842 RTP_SESSION_LOCK (sess);
843 sess->rtcp_bandwidth = g_value_get_double (value);
844 sess->recalc_bandwidth = TRUE;
845 RTP_SESSION_UNLOCK (sess);
846 break;
847 case PROP_RTCP_RR_BANDWIDTH:
848 RTP_SESSION_LOCK (sess);
849 sess->rtcp_rr_bandwidth = g_value_get_int (value);
850 sess->recalc_bandwidth = TRUE;
851 RTP_SESSION_UNLOCK (sess);
852 break;
853 case PROP_RTCP_RS_BANDWIDTH:
854 RTP_SESSION_LOCK (sess);
855 sess->rtcp_rs_bandwidth = g_value_get_int (value);
856 sess->recalc_bandwidth = TRUE;
857 RTP_SESSION_UNLOCK (sess);
858 break;
859 case PROP_RTCP_MTU:
860 sess->mtu = g_value_get_uint (value);
861 break;
862 case PROP_SDES:
863 rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
864 break;
865 case PROP_FAVOR_NEW:
866 sess->favor_new = g_value_get_boolean (value);
867 break;
868 case PROP_RTCP_MIN_INTERVAL:
869 rtp_stats_set_min_interval (&sess->stats,
870 (gdouble) g_value_get_uint64 (value) / GST_SECOND);
871 /* trigger reconsideration */
872 RTP_SESSION_LOCK (sess);
873 sess->next_rtcp_check_time = 0;
874 RTP_SESSION_UNLOCK (sess);
875 if (sess->callbacks.reconsider)
876 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
877 break;
878 case PROP_RTCP_FEEDBACK_RETENTION_WINDOW:
879 sess->rtcp_feedback_retention_window = g_value_get_uint64 (value);
880 break;
881 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
882 sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value);
883 break;
884 case PROP_PROBATION:
885 sess->probation = g_value_get_uint (value);
886 break;
887 case PROP_MAX_DROPOUT_TIME:
888 sess->max_dropout_time = g_value_get_uint (value);
889 break;
890 case PROP_MAX_MISORDER_TIME:
891 sess->max_misorder_time = g_value_get_uint (value);
892 break;
893 case PROP_RTP_PROFILE:
894 sess->rtp_profile = g_value_get_enum (value);
895 /* trigger reconsideration */
896 RTP_SESSION_LOCK (sess);
897 sess->next_rtcp_check_time = 0;
898 RTP_SESSION_UNLOCK (sess);
899 if (sess->callbacks.reconsider)
900 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
901 break;
902 case PROP_RTCP_REDUCED_SIZE:
903 sess->reduced_size_rtcp = g_value_get_boolean (value);
904 break;
905 case PROP_RTCP_DISABLE_SR_TIMESTAMP:
906 sess->timestamp_sender_reports = !g_value_get_boolean (value);
907 break;
908 default:
909 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
910 break;
911 }
912 }
913
914 static void
rtp_session_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)915 rtp_session_get_property (GObject * object, guint prop_id,
916 GValue * value, GParamSpec * pspec)
917 {
918 RTPSession *sess;
919
920 sess = RTP_SESSION (object);
921
922 switch (prop_id) {
923 case PROP_INTERNAL_SSRC:
924 g_value_set_uint (value, rtp_session_suggest_ssrc (sess, NULL));
925 break;
926 case PROP_INTERNAL_SOURCE:
927 /* FIXME, return a random source */
928 g_value_set_object (value, NULL);
929 break;
930 case PROP_BANDWIDTH:
931 g_value_set_double (value, sess->bandwidth);
932 break;
933 case PROP_RTCP_FRACTION:
934 g_value_set_double (value, sess->rtcp_bandwidth);
935 break;
936 case PROP_RTCP_RR_BANDWIDTH:
937 g_value_set_int (value, sess->rtcp_rr_bandwidth);
938 break;
939 case PROP_RTCP_RS_BANDWIDTH:
940 g_value_set_int (value, sess->rtcp_rs_bandwidth);
941 break;
942 case PROP_RTCP_MTU:
943 g_value_set_uint (value, sess->mtu);
944 break;
945 case PROP_SDES:
946 g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
947 break;
948 case PROP_NUM_SOURCES:
949 g_value_set_uint (value, rtp_session_get_num_sources (sess));
950 break;
951 case PROP_NUM_ACTIVE_SOURCES:
952 g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
953 break;
954 case PROP_SOURCES:
955 g_value_take_boxed (value, rtp_session_create_sources (sess));
956 break;
957 case PROP_FAVOR_NEW:
958 g_value_set_boolean (value, sess->favor_new);
959 break;
960 case PROP_RTCP_MIN_INTERVAL:
961 g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
962 break;
963 case PROP_RTCP_FEEDBACK_RETENTION_WINDOW:
964 g_value_set_uint64 (value, sess->rtcp_feedback_retention_window);
965 break;
966 case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD:
967 g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold);
968 break;
969 case PROP_PROBATION:
970 g_value_set_uint (value, sess->probation);
971 break;
972 case PROP_MAX_DROPOUT_TIME:
973 g_value_set_uint (value, sess->max_dropout_time);
974 break;
975 case PROP_MAX_MISORDER_TIME:
976 g_value_set_uint (value, sess->max_misorder_time);
977 break;
978 case PROP_STATS:
979 g_value_take_boxed (value, rtp_session_create_stats (sess));
980 break;
981 case PROP_RTP_PROFILE:
982 g_value_set_enum (value, sess->rtp_profile);
983 break;
984 case PROP_RTCP_REDUCED_SIZE:
985 g_value_set_boolean (value, sess->reduced_size_rtcp);
986 break;
987 case PROP_RTCP_DISABLE_SR_TIMESTAMP:
988 g_value_set_boolean (value, !sess->timestamp_sender_reports);
989 break;
990 default:
991 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
992 break;
993 }
994 }
995
996 static void
on_new_ssrc(RTPSession * sess,RTPSource * source)997 on_new_ssrc (RTPSession * sess, RTPSource * source)
998 {
999 g_object_ref (source);
1000 RTP_SESSION_UNLOCK (sess);
1001 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
1002 RTP_SESSION_LOCK (sess);
1003 g_object_unref (source);
1004 }
1005
1006 static void
on_ssrc_collision(RTPSession * sess,RTPSource * source)1007 on_ssrc_collision (RTPSession * sess, RTPSource * source)
1008 {
1009 g_object_ref (source);
1010 RTP_SESSION_UNLOCK (sess);
1011 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
1012 source);
1013 RTP_SESSION_LOCK (sess);
1014 g_object_unref (source);
1015 }
1016
1017 static void
on_ssrc_validated(RTPSession * sess,RTPSource * source)1018 on_ssrc_validated (RTPSession * sess, RTPSource * source)
1019 {
1020 g_object_ref (source);
1021 RTP_SESSION_UNLOCK (sess);
1022 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
1023 source);
1024 RTP_SESSION_LOCK (sess);
1025 g_object_unref (source);
1026 }
1027
1028 static void
on_ssrc_active(RTPSession * sess,RTPSource * source)1029 on_ssrc_active (RTPSession * sess, RTPSource * source)
1030 {
1031 g_object_ref (source);
1032 RTP_SESSION_UNLOCK (sess);
1033 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
1034 RTP_SESSION_LOCK (sess);
1035 g_object_unref (source);
1036 }
1037
1038 static void
on_ssrc_sdes(RTPSession * sess,RTPSource * source)1039 on_ssrc_sdes (RTPSession * sess, RTPSource * source)
1040 {
1041 g_object_ref (source);
1042 GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
1043 RTP_SESSION_UNLOCK (sess);
1044 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
1045 RTP_SESSION_LOCK (sess);
1046 g_object_unref (source);
1047 }
1048
1049 static void
on_bye_ssrc(RTPSession * sess,RTPSource * source)1050 on_bye_ssrc (RTPSession * sess, RTPSource * source)
1051 {
1052 g_object_ref (source);
1053 RTP_SESSION_UNLOCK (sess);
1054 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
1055 RTP_SESSION_LOCK (sess);
1056 g_object_unref (source);
1057 }
1058
1059 static void
on_bye_timeout(RTPSession * sess,RTPSource * source)1060 on_bye_timeout (RTPSession * sess, RTPSource * source)
1061 {
1062 g_object_ref (source);
1063 RTP_SESSION_UNLOCK (sess);
1064 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
1065 RTP_SESSION_LOCK (sess);
1066 g_object_unref (source);
1067 }
1068
1069 static void
on_timeout(RTPSession * sess,RTPSource * source)1070 on_timeout (RTPSession * sess, RTPSource * source)
1071 {
1072 g_object_ref (source);
1073 RTP_SESSION_UNLOCK (sess);
1074 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
1075 RTP_SESSION_LOCK (sess);
1076 g_object_unref (source);
1077 }
1078
1079 static void
on_sender_timeout(RTPSession * sess,RTPSource * source)1080 on_sender_timeout (RTPSession * sess, RTPSource * source)
1081 {
1082 g_object_ref (source);
1083 RTP_SESSION_UNLOCK (sess);
1084 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
1085 source);
1086 RTP_SESSION_LOCK (sess);
1087 g_object_unref (source);
1088 }
1089
1090 static void
on_new_sender_ssrc(RTPSession * sess,RTPSource * source)1091 on_new_sender_ssrc (RTPSession * sess, RTPSource * source)
1092 {
1093 g_object_ref (source);
1094 RTP_SESSION_UNLOCK (sess);
1095 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
1096 source);
1097 RTP_SESSION_LOCK (sess);
1098 g_object_unref (source);
1099 }
1100
1101 static void
on_sender_ssrc_active(RTPSession * sess,RTPSource * source)1102 on_sender_ssrc_active (RTPSession * sess, RTPSource * source)
1103 {
1104 g_object_ref (source);
1105 RTP_SESSION_UNLOCK (sess);
1106 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
1107 source);
1108 RTP_SESSION_LOCK (sess);
1109 g_object_unref (source);
1110 }
1111
1112 /**
1113 * rtp_session_new:
1114 *
1115 * Create a new session object.
1116 *
1117 * Returns: a new #RTPSession. g_object_unref() after usage.
1118 */
1119 RTPSession *
rtp_session_new(void)1120 rtp_session_new (void)
1121 {
1122 RTPSession *sess;
1123
1124 sess = g_object_new (RTP_TYPE_SESSION, NULL);
1125
1126 return sess;
1127 }
1128
1129 /**
1130 * rtp_session_reset:
1131 * @sess: an #RTPSession
1132 *
1133 * Reset the sources of @sess.
1134 */
1135 void
rtp_session_reset(RTPSession * sess)1136 rtp_session_reset (RTPSession * sess)
1137 {
1138 g_return_if_fail (RTP_IS_SESSION (sess));
1139
1140 /* remove all sources */
1141 g_hash_table_remove_all (sess->ssrcs[sess->mask_idx]);
1142 sess->total_sources = 0;
1143 sess->stats.sender_sources = 0;
1144 sess->stats.internal_sender_sources = 0;
1145 sess->stats.internal_sources = 0;
1146 sess->stats.active_sources = 0;
1147
1148 sess->generation = 0;
1149 sess->first_rtcp = TRUE;
1150 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
1151 sess->last_rtcp_check_time = GST_CLOCK_TIME_NONE;
1152 sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
1153 sess->last_rtcp_interval = GST_CLOCK_TIME_NONE;
1154 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
1155 sess->scheduled_bye = FALSE;
1156
1157 /* reset session stats */
1158 sess->stats.bye_members = 0;
1159 sess->stats.nacks_dropped = 0;
1160 sess->stats.nacks_sent = 0;
1161 sess->stats.nacks_received = 0;
1162
1163 sess->is_doing_ptp = TRUE;
1164
1165 g_list_free_full (sess->conflicting_addresses,
1166 (GDestroyNotify) rtp_conflicting_address_free);
1167 sess->conflicting_addresses = NULL;
1168 }
1169
1170 /**
1171 * rtp_session_set_callbacks:
1172 * @sess: an #RTPSession
1173 * @callbacks: callbacks to configure
1174 * @user_data: user data passed in the callbacks
1175 *
1176 * Configure a set of callbacks to be notified of actions.
1177 */
1178 void
rtp_session_set_callbacks(RTPSession * sess,RTPSessionCallbacks * callbacks,gpointer user_data)1179 rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
1180 gpointer user_data)
1181 {
1182 g_return_if_fail (RTP_IS_SESSION (sess));
1183
1184 if (callbacks->process_rtp) {
1185 sess->callbacks.process_rtp = callbacks->process_rtp;
1186 sess->process_rtp_user_data = user_data;
1187 }
1188 if (callbacks->send_rtp) {
1189 sess->callbacks.send_rtp = callbacks->send_rtp;
1190 sess->send_rtp_user_data = user_data;
1191 }
1192 if (callbacks->send_rtcp) {
1193 sess->callbacks.send_rtcp = callbacks->send_rtcp;
1194 sess->send_rtcp_user_data = user_data;
1195 }
1196 if (callbacks->sync_rtcp) {
1197 sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
1198 sess->sync_rtcp_user_data = user_data;
1199 }
1200 if (callbacks->clock_rate) {
1201 sess->callbacks.clock_rate = callbacks->clock_rate;
1202 sess->clock_rate_user_data = user_data;
1203 }
1204 if (callbacks->reconsider) {
1205 sess->callbacks.reconsider = callbacks->reconsider;
1206 sess->reconsider_user_data = user_data;
1207 }
1208 if (callbacks->request_key_unit) {
1209 sess->callbacks.request_key_unit = callbacks->request_key_unit;
1210 sess->request_key_unit_user_data = user_data;
1211 }
1212 if (callbacks->request_time) {
1213 sess->callbacks.request_time = callbacks->request_time;
1214 sess->request_time_user_data = user_data;
1215 }
1216 if (callbacks->notify_nack) {
1217 sess->callbacks.notify_nack = callbacks->notify_nack;
1218 sess->notify_nack_user_data = user_data;
1219 }
1220 if (callbacks->reconfigure) {
1221 sess->callbacks.reconfigure = callbacks->reconfigure;
1222 sess->reconfigure_user_data = user_data;
1223 }
1224 if (callbacks->notify_early_rtcp) {
1225 sess->callbacks.notify_early_rtcp = callbacks->notify_early_rtcp;
1226 sess->notify_early_rtcp_user_data = user_data;
1227 }
1228 }
1229
1230 /**
1231 * rtp_session_set_process_rtp_callback:
1232 * @sess: an #RTPSession
1233 * @callback: callback to set
1234 * @user_data: user data passed in the callback
1235 *
1236 * Configure only the process_rtp callback to be notified of the process_rtp action.
1237 */
1238 void
rtp_session_set_process_rtp_callback(RTPSession * sess,RTPSessionProcessRTP callback,gpointer user_data)1239 rtp_session_set_process_rtp_callback (RTPSession * sess,
1240 RTPSessionProcessRTP callback, gpointer user_data)
1241 {
1242 g_return_if_fail (RTP_IS_SESSION (sess));
1243
1244 sess->callbacks.process_rtp = callback;
1245 sess->process_rtp_user_data = user_data;
1246 }
1247
1248 /**
1249 * rtp_session_set_send_rtp_callback:
1250 * @sess: an #RTPSession
1251 * @callback: callback to set
1252 * @user_data: user data passed in the callback
1253 *
1254 * Configure only the send_rtp callback to be notified of the send_rtp action.
1255 */
1256 void
rtp_session_set_send_rtp_callback(RTPSession * sess,RTPSessionSendRTP callback,gpointer user_data)1257 rtp_session_set_send_rtp_callback (RTPSession * sess,
1258 RTPSessionSendRTP callback, gpointer user_data)
1259 {
1260 g_return_if_fail (RTP_IS_SESSION (sess));
1261
1262 sess->callbacks.send_rtp = callback;
1263 sess->send_rtp_user_data = user_data;
1264 }
1265
1266 /**
1267 * rtp_session_set_send_rtcp_callback:
1268 * @sess: an #RTPSession
1269 * @callback: callback to set
1270 * @user_data: user data passed in the callback
1271 *
1272 * Configure only the send_rtcp callback to be notified of the send_rtcp action.
1273 */
1274 void
rtp_session_set_send_rtcp_callback(RTPSession * sess,RTPSessionSendRTCP callback,gpointer user_data)1275 rtp_session_set_send_rtcp_callback (RTPSession * sess,
1276 RTPSessionSendRTCP callback, gpointer user_data)
1277 {
1278 g_return_if_fail (RTP_IS_SESSION (sess));
1279
1280 sess->callbacks.send_rtcp = callback;
1281 sess->send_rtcp_user_data = user_data;
1282 }
1283
1284 /**
1285 * rtp_session_set_sync_rtcp_callback:
1286 * @sess: an #RTPSession
1287 * @callback: callback to set
1288 * @user_data: user data passed in the callback
1289 *
1290 * Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
1291 */
1292 void
rtp_session_set_sync_rtcp_callback(RTPSession * sess,RTPSessionSyncRTCP callback,gpointer user_data)1293 rtp_session_set_sync_rtcp_callback (RTPSession * sess,
1294 RTPSessionSyncRTCP callback, gpointer user_data)
1295 {
1296 g_return_if_fail (RTP_IS_SESSION (sess));
1297
1298 sess->callbacks.sync_rtcp = callback;
1299 sess->sync_rtcp_user_data = user_data;
1300 }
1301
1302 /**
1303 * rtp_session_set_clock_rate_callback:
1304 * @sess: an #RTPSession
1305 * @callback: callback to set
1306 * @user_data: user data passed in the callback
1307 *
1308 * Configure only the clock_rate callback to be notified of the clock_rate action.
1309 */
1310 void
rtp_session_set_clock_rate_callback(RTPSession * sess,RTPSessionClockRate callback,gpointer user_data)1311 rtp_session_set_clock_rate_callback (RTPSession * sess,
1312 RTPSessionClockRate callback, gpointer user_data)
1313 {
1314 g_return_if_fail (RTP_IS_SESSION (sess));
1315
1316 sess->callbacks.clock_rate = callback;
1317 sess->clock_rate_user_data = user_data;
1318 }
1319
1320 /**
1321 * rtp_session_set_reconsider_callback:
1322 * @sess: an #RTPSession
1323 * @callback: callback to set
1324 * @user_data: user data passed in the callback
1325 *
1326 * Configure only the reconsider callback to be notified of the reconsider action.
1327 */
1328 void
rtp_session_set_reconsider_callback(RTPSession * sess,RTPSessionReconsider callback,gpointer user_data)1329 rtp_session_set_reconsider_callback (RTPSession * sess,
1330 RTPSessionReconsider callback, gpointer user_data)
1331 {
1332 g_return_if_fail (RTP_IS_SESSION (sess));
1333
1334 sess->callbacks.reconsider = callback;
1335 sess->reconsider_user_data = user_data;
1336 }
1337
1338 /**
1339 * rtp_session_set_request_time_callback:
1340 * @sess: an #RTPSession
1341 * @callback: callback to set
1342 * @user_data: user data passed in the callback
1343 *
1344 * Configure only the request_time callback
1345 */
1346 void
rtp_session_set_request_time_callback(RTPSession * sess,RTPSessionRequestTime callback,gpointer user_data)1347 rtp_session_set_request_time_callback (RTPSession * sess,
1348 RTPSessionRequestTime callback, gpointer user_data)
1349 {
1350 g_return_if_fail (RTP_IS_SESSION (sess));
1351
1352 sess->callbacks.request_time = callback;
1353 sess->request_time_user_data = user_data;
1354 }
1355
1356 /**
1357 * rtp_session_set_bandwidth:
1358 * @sess: an #RTPSession
1359 * @bandwidth: the bandwidth allocated
1360 *
1361 * Set the session bandwidth in bits per second.
1362 */
1363 void
rtp_session_set_bandwidth(RTPSession * sess,gdouble bandwidth)1364 rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
1365 {
1366 g_return_if_fail (RTP_IS_SESSION (sess));
1367
1368 RTP_SESSION_LOCK (sess);
1369 sess->stats.bandwidth = bandwidth;
1370 RTP_SESSION_UNLOCK (sess);
1371 }
1372
1373 /**
1374 * rtp_session_get_bandwidth:
1375 * @sess: an #RTPSession
1376 *
1377 * Get the session bandwidth.
1378 *
1379 * Returns: the session bandwidth.
1380 */
1381 gdouble
rtp_session_get_bandwidth(RTPSession * sess)1382 rtp_session_get_bandwidth (RTPSession * sess)
1383 {
1384 gdouble result;
1385
1386 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1387
1388 RTP_SESSION_LOCK (sess);
1389 result = sess->stats.bandwidth;
1390 RTP_SESSION_UNLOCK (sess);
1391
1392 return result;
1393 }
1394
1395 /**
1396 * rtp_session_set_rtcp_fraction:
1397 * @sess: an #RTPSession
1398 * @bandwidth: the RTCP bandwidth
1399 *
1400 * Set the bandwidth in bits per second that should be used for RTCP
1401 * messages.
1402 */
1403 void
rtp_session_set_rtcp_fraction(RTPSession * sess,gdouble bandwidth)1404 rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
1405 {
1406 g_return_if_fail (RTP_IS_SESSION (sess));
1407
1408 RTP_SESSION_LOCK (sess);
1409 sess->stats.rtcp_bandwidth = bandwidth;
1410 RTP_SESSION_UNLOCK (sess);
1411 }
1412
1413 /**
1414 * rtp_session_get_rtcp_fraction:
1415 * @sess: an #RTPSession
1416 *
1417 * Get the session bandwidth used for RTCP.
1418 *
1419 * Returns: The bandwidth used for RTCP messages.
1420 */
1421 gdouble
rtp_session_get_rtcp_fraction(RTPSession * sess)1422 rtp_session_get_rtcp_fraction (RTPSession * sess)
1423 {
1424 gdouble result;
1425
1426 g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
1427
1428 RTP_SESSION_LOCK (sess);
1429 result = sess->stats.rtcp_bandwidth;
1430 RTP_SESSION_UNLOCK (sess);
1431
1432 return result;
1433 }
1434
1435 /**
1436 * rtp_session_get_sdes_struct:
1437 * @sess: an #RTSPSession
1438 *
1439 * Get the SDES data as a #GstStructure
1440 *
1441 * Returns: a GstStructure with SDES items for @sess. This function returns a
1442 * copy of the SDES structure, use gst_structure_free() after usage.
1443 */
1444 GstStructure *
rtp_session_get_sdes_struct(RTPSession * sess)1445 rtp_session_get_sdes_struct (RTPSession * sess)
1446 {
1447 GstStructure *result = NULL;
1448
1449 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1450
1451 RTP_SESSION_LOCK (sess);
1452 if (sess->sdes)
1453 result = gst_structure_copy (sess->sdes);
1454 RTP_SESSION_UNLOCK (sess);
1455
1456 return result;
1457 }
1458
1459 static void
source_set_sdes(const gchar * key,RTPSource * source,GstStructure * sdes)1460 source_set_sdes (const gchar * key, RTPSource * source, GstStructure * sdes)
1461 {
1462 rtp_source_set_sdes_struct (source, gst_structure_copy (sdes));
1463 }
1464
1465 /**
1466 * rtp_session_set_sdes_struct:
1467 * @sess: an #RTSPSession
1468 * @sdes: a #GstStructure
1469 *
1470 * Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
1471 */
1472 void
rtp_session_set_sdes_struct(RTPSession * sess,const GstStructure * sdes)1473 rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
1474 {
1475 g_return_if_fail (sdes);
1476 g_return_if_fail (RTP_IS_SESSION (sess));
1477
1478 RTP_SESSION_LOCK (sess);
1479 if (sess->sdes)
1480 gst_structure_free (sess->sdes);
1481 sess->sdes = gst_structure_copy (sdes);
1482
1483 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1484 (GHFunc) source_set_sdes, sess->sdes);
1485 RTP_SESSION_UNLOCK (sess);
1486 }
1487
1488 static GstFlowReturn
source_push_rtp(RTPSource * source,gpointer data,RTPSession * session)1489 source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
1490 {
1491 GstFlowReturn result = GST_FLOW_OK;
1492
1493 if (source->internal) {
1494 GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
1495
1496 RTP_SESSION_UNLOCK (session);
1497
1498 if (session->callbacks.send_rtp)
1499 result =
1500 session->callbacks.send_rtp (session, source, data,
1501 session->send_rtp_user_data);
1502 else {
1503 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1504 }
1505 } else {
1506 GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
1507 RTP_SESSION_UNLOCK (session);
1508
1509 if (session->callbacks.process_rtp)
1510 result =
1511 session->callbacks.process_rtp (session, source,
1512 GST_BUFFER_CAST (data), session->process_rtp_user_data);
1513 else
1514 gst_buffer_unref (GST_BUFFER_CAST (data));
1515 }
1516 RTP_SESSION_LOCK (session);
1517
1518 return result;
1519 }
1520
1521 static gint
source_clock_rate(RTPSource * source,guint8 pt,RTPSession * session)1522 source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
1523 {
1524 gint result;
1525
1526 RTP_SESSION_UNLOCK (session);
1527
1528 if (session->callbacks.clock_rate)
1529 result =
1530 session->callbacks.clock_rate (session, pt,
1531 session->clock_rate_user_data);
1532 else
1533 result = -1;
1534
1535 RTP_SESSION_LOCK (session);
1536
1537 GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1538
1539 return result;
1540 }
1541
1542 static RTPSourceCallbacks callbacks = {
1543 (RTPSourcePushRTP) source_push_rtp,
1544 (RTPSourceClockRate) source_clock_rate,
1545 };
1546
1547
1548 /**
1549 * rtp_session_find_conflicting_address:
1550 * @session: The session the packet came in
1551 * @address: address to check for
1552 * @time: The time when the packet that is possibly in conflict arrived
1553 *
1554 * Checks if an address which has a conflict is already known. If it is
1555 * a known conflict, remember the time
1556 *
1557 * Returns: TRUE if it was a known conflict, FALSE otherwise
1558 */
1559 static gboolean
rtp_session_find_conflicting_address(RTPSession * session,GSocketAddress * address,GstClockTime time)1560 rtp_session_find_conflicting_address (RTPSession * session,
1561 GSocketAddress * address, GstClockTime time)
1562 {
1563 return find_conflicting_address (session->conflicting_addresses, address,
1564 time);
1565 }
1566
1567 /**
1568 * rtp_session_add_conflicting_address:
1569 * @session: The session the packet came in
1570 * @address: address to remember
1571 * @time: The time when the packet that is in conflict arrived
1572 *
1573 * Adds a new conflict address
1574 */
1575 static void
rtp_session_add_conflicting_address(RTPSession * sess,GSocketAddress * address,GstClockTime time)1576 rtp_session_add_conflicting_address (RTPSession * sess,
1577 GSocketAddress * address, GstClockTime time)
1578 {
1579 sess->conflicting_addresses =
1580 add_conflicting_address (sess->conflicting_addresses, address, time);
1581 }
1582
1583
1584 static gboolean
check_collision(RTPSession * sess,RTPSource * source,RTPPacketInfo * pinfo,gboolean rtp)1585 check_collision (RTPSession * sess, RTPSource * source,
1586 RTPPacketInfo * pinfo, gboolean rtp)
1587 {
1588 guint32 ssrc;
1589
1590 /* If we have no pinfo address, we can't do collision checking */
1591 if (!pinfo->address)
1592 return FALSE;
1593
1594 ssrc = rtp_source_get_ssrc (source);
1595
1596 if (!source->internal) {
1597 GSocketAddress *from;
1598
1599 /* This is not our local source, but lets check if two remote
1600 * source collide */
1601 if (rtp) {
1602 from = source->rtp_from;
1603 } else {
1604 from = source->rtcp_from;
1605 }
1606
1607 if (from) {
1608 if (__g_socket_address_equal (from, pinfo->address)) {
1609 /* Address is the same */
1610 return FALSE;
1611 } else {
1612 GST_LOG ("we have a third-party collision or loop ssrc:%x", ssrc);
1613 if (sess->favor_new) {
1614 if (rtp_source_find_conflicting_address (source,
1615 pinfo->address, pinfo->current_time)) {
1616 gchar *buf1;
1617
1618 buf1 = __g_socket_address_to_string (pinfo->address);
1619 GST_LOG ("Known conflict on %x for %s, dropping packet", ssrc,
1620 buf1);
1621 g_free (buf1);
1622
1623 return TRUE;
1624 } else {
1625 gchar *buf1, *buf2;
1626
1627 /* Current address is not a known conflict, lets assume this is
1628 * a new source. Save old address in possible conflict list
1629 */
1630 rtp_source_add_conflicting_address (source, from,
1631 pinfo->current_time);
1632
1633 buf1 = __g_socket_address_to_string (from);
1634 buf2 = __g_socket_address_to_string (pinfo->address);
1635
1636 GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
1637 " saving old as known conflict", ssrc, buf1, buf2);
1638
1639 if (rtp)
1640 rtp_source_set_rtp_from (source, pinfo->address);
1641 else
1642 rtp_source_set_rtcp_from (source, pinfo->address);
1643
1644 g_free (buf1);
1645 g_free (buf2);
1646
1647 return FALSE;
1648 }
1649 } else {
1650 /* Don't need to save old addresses, we ignore new sources */
1651 return TRUE;
1652 }
1653 }
1654 } else {
1655 /* We don't already have a from address for RTP, just set it */
1656 if (rtp)
1657 rtp_source_set_rtp_from (source, pinfo->address);
1658 else
1659 rtp_source_set_rtcp_from (source, pinfo->address);
1660 return FALSE;
1661 }
1662
1663 /* FIXME: Log 3rd party collision somehow
1664 * Maybe should be done in upper layer, only the SDES can tell us
1665 * if its a collision or a loop
1666 */
1667 } else {
1668 /* This is sending with our ssrc, is it an address we already know */
1669 if (rtp_session_find_conflicting_address (sess, pinfo->address,
1670 pinfo->current_time)) {
1671 /* Its a known conflict, its probably a loop, not a collision
1672 * lets just drop the incoming packet
1673 */
1674 GST_DEBUG ("Our packets are being looped back to us, dropping");
1675 } else {
1676 /* Its a new collision, lets change our SSRC */
1677 rtp_session_add_conflicting_address (sess, pinfo->address,
1678 pinfo->current_time);
1679
1680 GST_DEBUG ("Collision for SSRC %x", ssrc);
1681 /* mark the source BYE */
1682 rtp_source_mark_bye (source, "SSRC Collision");
1683 /* if we were suggesting this SSRC, change to something else */
1684 if (sess->suggested_ssrc == ssrc) {
1685 sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
1686 sess->internal_ssrc_set = TRUE;
1687 }
1688
1689 on_ssrc_collision (sess, source);
1690
1691 rtp_session_schedule_bye_locked (sess, pinfo->current_time);
1692 }
1693 }
1694
1695 return TRUE;
1696 }
1697
1698 typedef struct
1699 {
1700 gboolean is_doing_ptp;
1701 GSocketAddress *new_addr;
1702 } CompareAddrData;
1703
1704 /* check if the two given ip addr are the same (do not care about the port) */
1705 static gboolean
ip_addr_equal(GSocketAddress * a,GSocketAddress * b)1706 ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
1707 {
1708 return
1709 g_inet_address_equal (g_inet_socket_address_get_address
1710 (G_INET_SOCKET_ADDRESS (a)),
1711 g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
1712 }
1713
1714 static void
compare_rtp_source_addr(const gchar * key,RTPSource * source,CompareAddrData * data)1715 compare_rtp_source_addr (const gchar * key, RTPSource * source,
1716 CompareAddrData * data)
1717 {
1718 /* only compare ip addr of remote sources which are also not closing */
1719 if (!source->internal && !source->closing && source->rtp_from) {
1720 /* look for the first rtp source */
1721 if (!data->new_addr)
1722 data->new_addr = source->rtp_from;
1723 /* compare current ip addr with the first one */
1724 else
1725 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
1726 }
1727 }
1728
1729 static void
compare_rtcp_source_addr(const gchar * key,RTPSource * source,CompareAddrData * data)1730 compare_rtcp_source_addr (const gchar * key, RTPSource * source,
1731 CompareAddrData * data)
1732 {
1733 /* only compare ip addr of remote sources which are also not closing */
1734 if (!source->internal && !source->closing && source->rtcp_from) {
1735 /* look for the first rtcp source */
1736 if (!data->new_addr)
1737 data->new_addr = source->rtcp_from;
1738 else
1739 /* compare current ip addr with the first one */
1740 data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
1741 }
1742 }
1743
1744 /* loop over our non-internal source to know if the session
1745 * is doing point-to-point */
1746 static void
session_update_ptp(RTPSession * sess)1747 session_update_ptp (RTPSession * sess)
1748 {
1749 /* to know if the session is doing point to point, the ip addr
1750 * of each non-internal (=remotes) source have to be compared
1751 * to each other.
1752 */
1753 gboolean is_doing_rtp_ptp;
1754 gboolean is_doing_rtcp_ptp;
1755 CompareAddrData data;
1756
1757 /* compare the first remote source's ip addr that receive rtp packets
1758 * with other remote rtp source.
1759 * it's enough because the session just needs to know if they are all
1760 * equals or not
1761 */
1762 data.is_doing_ptp = TRUE;
1763 data.new_addr = NULL;
1764 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1765 (GHFunc) compare_rtp_source_addr, (gpointer) & data);
1766 is_doing_rtp_ptp = data.is_doing_ptp;
1767
1768 /* same but about rtcp */
1769 data.is_doing_ptp = TRUE;
1770 data.new_addr = NULL;
1771 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
1772 (GHFunc) compare_rtcp_source_addr, (gpointer) & data);
1773 is_doing_rtcp_ptp = data.is_doing_ptp;
1774
1775 /* the session is doing point-to-point if all rtp remote have the same
1776 * ip addr and if all rtcp remote sources have the same ip addr */
1777 sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
1778
1779 GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
1780 }
1781
1782 static void
add_source(RTPSession * sess,RTPSource * src)1783 add_source (RTPSession * sess, RTPSource * src)
1784 {
1785 g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1786 GINT_TO_POINTER (src->ssrc), src);
1787 /* report the new source ASAP */
1788 src->generation = sess->generation;
1789 /* we have one more source now */
1790 sess->total_sources++;
1791 if (RTP_SOURCE_IS_ACTIVE (src))
1792 sess->stats.active_sources++;
1793 if (src->internal) {
1794 sess->stats.internal_sources++;
1795 if (!sess->internal_ssrc_from_caps_or_property
1796 && sess->suggested_ssrc != src->ssrc) {
1797 sess->suggested_ssrc = src->ssrc;
1798 sess->internal_ssrc_set = TRUE;
1799 }
1800 }
1801
1802 /* update point-to-point status */
1803 if (!src->internal)
1804 session_update_ptp (sess);
1805 }
1806
1807 static RTPSource *
find_source(RTPSession * sess,guint32 ssrc)1808 find_source (RTPSession * sess, guint32 ssrc)
1809 {
1810 return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1811 GINT_TO_POINTER (ssrc));
1812 }
1813
1814 /* must be called with the session lock, the returned source needs to be
1815 * unreffed after usage. */
1816 static RTPSource *
obtain_source(RTPSession * sess,guint32 ssrc,gboolean * created,RTPPacketInfo * pinfo,gboolean rtp)1817 obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1818 RTPPacketInfo * pinfo, gboolean rtp)
1819 {
1820 RTPSource *source;
1821
1822 source = find_source (sess, ssrc);
1823 if (source == NULL) {
1824 /* make new Source in probation and insert */
1825 source = rtp_source_new (ssrc);
1826
1827 GST_DEBUG ("creating new source %08x %p", ssrc, source);
1828
1829 /* for RTP packets we need to set the source in probation. Receiving RTCP
1830 * packets of an SSRC, on the other hand, is a strong indication that we
1831 * are dealing with a valid source. */
1832 g_object_set (source, "probation", rtp ? sess->probation : 0,
1833 "max-dropout-time", sess->max_dropout_time, "max-misorder-time",
1834 sess->max_misorder_time, NULL);
1835
1836 /* store from address, if any */
1837 if (pinfo->address) {
1838 if (rtp)
1839 rtp_source_set_rtp_from (source, pinfo->address);
1840 else
1841 rtp_source_set_rtcp_from (source, pinfo->address);
1842 }
1843
1844 /* configure a callback on the source */
1845 rtp_source_set_callbacks (source, &callbacks, sess);
1846
1847 add_source (sess, source);
1848 *created = TRUE;
1849 } else {
1850 *created = FALSE;
1851 /* check for collision, this updates the address when not previously set */
1852 if (check_collision (sess, source, pinfo, rtp)) {
1853 return NULL;
1854 }
1855 /* Receiving RTCP packets of an SSRC is a strong indication that we
1856 * are dealing with a valid source. */
1857 if (!rtp)
1858 g_object_set (source, "probation", 0, NULL);
1859 }
1860 /* update last activity */
1861 source->last_activity = pinfo->current_time;
1862 if (rtp)
1863 source->last_rtp_activity = pinfo->current_time;
1864 g_object_ref (source);
1865
1866 return source;
1867 }
1868
1869 /* must be called with the session lock, the returned source needs to be
1870 * unreffed after usage. */
1871 static RTPSource *
obtain_internal_source(RTPSession * sess,guint32 ssrc,gboolean * created,GstClockTime current_time)1872 obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1873 GstClockTime current_time)
1874 {
1875 RTPSource *source;
1876
1877 source = find_source (sess, ssrc);
1878 if (source == NULL) {
1879 /* make new internal Source and insert */
1880 source = rtp_source_new (ssrc);
1881
1882 GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
1883
1884 source->validated = TRUE;
1885 source->internal = TRUE;
1886 source->probation = FALSE;
1887 rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
1888 rtp_source_set_callbacks (source, &callbacks, sess);
1889
1890 add_source (sess, source);
1891 *created = TRUE;
1892 } else {
1893 *created = FALSE;
1894 }
1895 /* update last activity */
1896 if (current_time != GST_CLOCK_TIME_NONE) {
1897 source->last_activity = current_time;
1898 source->last_rtp_activity = current_time;
1899 }
1900 g_object_ref (source);
1901
1902 return source;
1903 }
1904
1905 /**
1906 * rtp_session_suggest_ssrc:
1907 * @sess: a #RTPSession
1908 * @is_random: if the suggested ssrc is random
1909 *
1910 * Suggest an unused SSRC in @sess.
1911 *
1912 * Returns: a free unused SSRC
1913 */
1914 guint32
rtp_session_suggest_ssrc(RTPSession * sess,gboolean * is_random)1915 rtp_session_suggest_ssrc (RTPSession * sess, gboolean * is_random)
1916 {
1917 guint32 result;
1918
1919 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1920
1921 RTP_SESSION_LOCK (sess);
1922 result = sess->suggested_ssrc;
1923 if (is_random)
1924 *is_random = !sess->internal_ssrc_set;
1925 RTP_SESSION_UNLOCK (sess);
1926
1927 return result;
1928 }
1929
1930 /**
1931 * rtp_session_add_source:
1932 * @sess: a #RTPSession
1933 * @src: #RTPSource to add
1934 *
1935 * Add @src to @session.
1936 *
1937 * Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1938 * existed in the session.
1939 */
1940 gboolean
rtp_session_add_source(RTPSession * sess,RTPSource * src)1941 rtp_session_add_source (RTPSession * sess, RTPSource * src)
1942 {
1943 gboolean result = FALSE;
1944 RTPSource *find;
1945
1946 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1947 g_return_val_if_fail (src != NULL, FALSE);
1948
1949 RTP_SESSION_LOCK (sess);
1950 find = find_source (sess, src->ssrc);
1951 if (find == NULL) {
1952 add_source (sess, src);
1953 result = TRUE;
1954 }
1955 RTP_SESSION_UNLOCK (sess);
1956
1957 return result;
1958 }
1959
1960 /**
1961 * rtp_session_get_num_sources:
1962 * @sess: an #RTPSession
1963 *
1964 * Get the number of sources in @sess.
1965 *
1966 * Returns: The number of sources in @sess.
1967 */
1968 guint
rtp_session_get_num_sources(RTPSession * sess)1969 rtp_session_get_num_sources (RTPSession * sess)
1970 {
1971 guint result;
1972
1973 g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1974
1975 RTP_SESSION_LOCK (sess);
1976 result = sess->total_sources;
1977 RTP_SESSION_UNLOCK (sess);
1978
1979 return result;
1980 }
1981
1982 /**
1983 * rtp_session_get_num_active_sources:
1984 * @sess: an #RTPSession
1985 *
1986 * Get the number of active sources in @sess. A source is considered active when
1987 * it has been validated and has not yet received a BYE RTCP message.
1988 *
1989 * Returns: The number of active sources in @sess.
1990 */
1991 guint
rtp_session_get_num_active_sources(RTPSession * sess)1992 rtp_session_get_num_active_sources (RTPSession * sess)
1993 {
1994 guint result;
1995
1996 g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1997
1998 RTP_SESSION_LOCK (sess);
1999 result = sess->stats.active_sources;
2000 RTP_SESSION_UNLOCK (sess);
2001
2002 return result;
2003 }
2004
2005 /**
2006 * rtp_session_get_source_by_ssrc:
2007 * @sess: an #RTPSession
2008 * @ssrc: an SSRC
2009 *
2010 * Find the source with @ssrc in @sess.
2011 *
2012 * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
2013 * g_object_unref() after usage.
2014 */
2015 RTPSource *
rtp_session_get_source_by_ssrc(RTPSession * sess,guint32 ssrc)2016 rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
2017 {
2018 RTPSource *result;
2019
2020 g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
2021
2022 RTP_SESSION_LOCK (sess);
2023 result = find_source (sess, ssrc);
2024 if (result != NULL)
2025 g_object_ref (result);
2026 RTP_SESSION_UNLOCK (sess);
2027
2028 return result;
2029 }
2030
2031 /* should be called with the SESSION lock */
2032 static guint32
rtp_session_create_new_ssrc(RTPSession * sess)2033 rtp_session_create_new_ssrc (RTPSession * sess)
2034 {
2035 guint32 ssrc;
2036
2037 while (TRUE) {
2038 ssrc = g_random_int ();
2039
2040 /* see if it exists in the session, we're done if it doesn't */
2041 if (find_source (sess, ssrc) == NULL)
2042 break;
2043 }
2044 return ssrc;
2045 }
2046
2047 static gboolean
update_packet(GstBuffer ** buffer,guint idx,RTPPacketInfo * pinfo)2048 update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
2049 {
2050 GstNetAddressMeta *meta;
2051
2052 /* get packet size including header overhead */
2053 pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
2054 pinfo->packets++;
2055
2056 if (pinfo->rtp) {
2057 GstRTPBuffer rtp = { NULL };
2058
2059 if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
2060 goto invalid_packet;
2061
2062 pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
2063 if (idx == 0) {
2064 gint i;
2065
2066 /* only keep info for first buffer */
2067 pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
2068 pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
2069 pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
2070 pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
2071 /* copy available csrc */
2072 pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
2073 for (i = 0; i < pinfo->csrc_count; i++)
2074 pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
2075 }
2076 gst_rtp_buffer_unmap (&rtp);
2077 }
2078
2079 if (idx == 0) {
2080 /* for netbuffer we can store the IP address to check for collisions */
2081 meta = gst_buffer_get_net_address_meta (*buffer);
2082 if (pinfo->address)
2083 g_object_unref (pinfo->address);
2084 if (meta) {
2085 pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
2086 } else {
2087 pinfo->address = NULL;
2088 }
2089 }
2090 return TRUE;
2091
2092 /* ERRORS */
2093 invalid_packet:
2094 {
2095 GST_DEBUG ("invalid RTP packet received");
2096 return FALSE;
2097 }
2098 }
2099
2100 /* update the RTPPacketInfo structure with the current time and other bits
2101 * about the current buffer we are handling.
2102 * This function is typically called when a validated packet is received.
2103 * This function should be called with the RTP_SESSION_LOCK
2104 */
2105 static gboolean
update_packet_info(RTPSession * sess,RTPPacketInfo * pinfo,gboolean send,gboolean rtp,gboolean is_list,gpointer data,GstClockTime current_time,GstClockTime running_time,guint64 ntpnstime)2106 update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
2107 gboolean send, gboolean rtp, gboolean is_list, gpointer data,
2108 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2109 {
2110 gboolean res;
2111
2112 pinfo->send = send;
2113 pinfo->rtp = rtp;
2114 pinfo->is_list = is_list;
2115 pinfo->data = data;
2116 pinfo->current_time = current_time;
2117 pinfo->running_time = running_time;
2118 pinfo->ntpnstime = ntpnstime;
2119 pinfo->header_len = sess->header_len;
2120 pinfo->bytes = 0;
2121 pinfo->payload_len = 0;
2122 pinfo->packets = 0;
2123
2124 if (is_list) {
2125 GstBufferList *list = GST_BUFFER_LIST_CAST (data);
2126 res =
2127 gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
2128 pinfo);
2129 } else {
2130 GstBuffer *buffer = GST_BUFFER_CAST (data);
2131 res = update_packet (&buffer, 0, pinfo);
2132 }
2133 return res;
2134 }
2135
2136 static void
clean_packet_info(RTPPacketInfo * pinfo)2137 clean_packet_info (RTPPacketInfo * pinfo)
2138 {
2139 if (pinfo->address)
2140 g_object_unref (pinfo->address);
2141 if (pinfo->data) {
2142 gst_mini_object_unref (pinfo->data);
2143 pinfo->data = NULL;
2144 }
2145 }
2146
2147 static gboolean
source_update_active(RTPSession * sess,RTPSource * source,gboolean prevactive)2148 source_update_active (RTPSession * sess, RTPSource * source,
2149 gboolean prevactive)
2150 {
2151 gboolean active = RTP_SOURCE_IS_ACTIVE (source);
2152 guint32 ssrc = source->ssrc;
2153
2154 if (prevactive == active)
2155 return FALSE;
2156
2157 if (active) {
2158 sess->stats.active_sources++;
2159 GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
2160 sess->stats.active_sources);
2161 } else {
2162 sess->stats.active_sources--;
2163 GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
2164 sess->stats.active_sources);
2165 }
2166 return TRUE;
2167 }
2168
2169 static gboolean
source_update_sender(RTPSession * sess,RTPSource * source,gboolean prevsender)2170 source_update_sender (RTPSession * sess, RTPSource * source,
2171 gboolean prevsender)
2172 {
2173 gboolean sender = RTP_SOURCE_IS_SENDER (source);
2174 guint32 ssrc = source->ssrc;
2175
2176 if (prevsender == sender)
2177 return FALSE;
2178
2179 if (sender) {
2180 sess->stats.sender_sources++;
2181 if (source->internal)
2182 sess->stats.internal_sender_sources++;
2183 GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
2184 sess->stats.sender_sources);
2185 } else {
2186 sess->stats.sender_sources--;
2187 if (source->internal)
2188 sess->stats.internal_sender_sources--;
2189 GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
2190 sess->stats.sender_sources);
2191 }
2192 return TRUE;
2193 }
2194
2195 /**
2196 * rtp_session_process_rtp:
2197 * @sess: and #RTPSession
2198 * @buffer: an RTP buffer
2199 * @current_time: the current system time
2200 * @running_time: the running_time of @buffer
2201 *
2202 * Process an RTP buffer in the session manager. This function takes ownership
2203 * of @buffer.
2204 *
2205 * Returns: a #GstFlowReturn.
2206 */
2207 GstFlowReturn
rtp_session_process_rtp(RTPSession * sess,GstBuffer * buffer,GstClockTime current_time,GstClockTime running_time,guint64 ntpnstime)2208 rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
2209 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2210 {
2211 GstFlowReturn result;
2212 guint32 ssrc;
2213 RTPSource *source;
2214 gboolean created;
2215 gboolean prevsender, prevactive;
2216 RTPPacketInfo pinfo = { 0, };
2217 guint64 oldrate;
2218
2219 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2220 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2221
2222 RTP_SESSION_LOCK (sess);
2223
2224 /* update pinfo stats */
2225 if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
2226 current_time, running_time, ntpnstime)) {
2227 GST_DEBUG ("invalid RTP packet received");
2228 RTP_SESSION_UNLOCK (sess);
2229 return rtp_session_process_rtcp (sess, buffer, current_time, running_time,
2230 ntpnstime);
2231 }
2232
2233 ssrc = pinfo.ssrc;
2234
2235 source = obtain_source (sess, ssrc, &created, &pinfo, TRUE);
2236 if (!source)
2237 goto collision;
2238
2239 prevsender = RTP_SOURCE_IS_SENDER (source);
2240 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2241 oldrate = source->bitrate;
2242
2243 if (created)
2244 on_new_ssrc (sess, source);
2245
2246 /* let source process the packet */
2247 result = rtp_source_process_rtp (source, &pinfo);
2248
2249 /* source became active */
2250 if (source_update_active (sess, source, prevactive))
2251 on_ssrc_validated (sess, source);
2252
2253 source_update_sender (sess, source, prevsender);
2254
2255 if (oldrate != source->bitrate)
2256 sess->recalc_bandwidth = TRUE;
2257
2258
2259 if (source->validated) {
2260 gboolean created;
2261 gint i;
2262
2263 /* for validated sources, we add the CSRCs as well */
2264 for (i = 0; i < pinfo.csrc_count; i++) {
2265 guint32 csrc;
2266 RTPSource *csrc_src;
2267
2268 csrc = pinfo.csrcs[i];
2269
2270 /* get source */
2271 csrc_src = obtain_source (sess, csrc, &created, &pinfo, TRUE);
2272 if (!csrc_src)
2273 continue;
2274
2275 if (created) {
2276 GST_DEBUG ("created new CSRC: %08x", csrc);
2277 rtp_source_set_as_csrc (csrc_src);
2278 source_update_active (sess, csrc_src, FALSE);
2279 on_new_ssrc (sess, csrc_src);
2280 }
2281 g_object_unref (csrc_src);
2282 }
2283 }
2284 g_object_unref (source);
2285
2286 RTP_SESSION_UNLOCK (sess);
2287
2288 clean_packet_info (&pinfo);
2289
2290 return result;
2291
2292 /* ERRORS */
2293 collision:
2294 {
2295 RTP_SESSION_UNLOCK (sess);
2296 clean_packet_info (&pinfo);
2297 GST_DEBUG ("ignoring packet because its collisioning");
2298 return GST_FLOW_OK;
2299 }
2300 }
2301
2302 static void
rtp_session_process_rb(RTPSession * sess,RTPSource * source,GstRTCPPacket * packet,RTPPacketInfo * pinfo)2303 rtp_session_process_rb (RTPSession * sess, RTPSource * source,
2304 GstRTCPPacket * packet, RTPPacketInfo * pinfo)
2305 {
2306 guint count, i;
2307
2308 count = gst_rtcp_packet_get_rb_count (packet);
2309 for (i = 0; i < count; i++) {
2310 guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
2311 guint8 fractionlost;
2312 gint32 packetslost;
2313 RTPSource *src;
2314
2315 gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
2316 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2317
2318 GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
2319
2320 /* find our own source */
2321 src = find_source (sess, ssrc);
2322 if (src == NULL)
2323 continue;
2324
2325 if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
2326 /* only deal with report blocks for our session, we update the stats of
2327 * the sender of the RTCP message. We could also compare our stats against
2328 * the other sender to see if we are better or worse. */
2329 /* FIXME, need to keep track who the RB block is from */
2330 rtp_source_process_rb (source, pinfo->ntpnstime, fractionlost,
2331 packetslost, exthighestseq, jitter, lsr, dlsr);
2332 }
2333 }
2334 on_ssrc_active (sess, source);
2335 }
2336
2337 /* A Sender report contains statistics about how the sender is doing. This
2338 * includes timing informataion such as the relation between RTP and NTP
2339 * timestamps and the number of packets/bytes it sent to us.
2340 *
2341 * In this report is also included a set of report blocks related to how this
2342 * sender is receiving data (in case we (or somebody else) is also sending stuff
2343 * to it). This info includes the packet loss, jitter and seqnum. It also
2344 * contains information to calculate the round trip time (LSR/DLSR).
2345 */
2346 static void
rtp_session_process_sr(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo,gboolean * do_sync)2347 rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
2348 RTPPacketInfo * pinfo, gboolean * do_sync)
2349 {
2350 guint32 senderssrc, rtptime, packet_count, octet_count;
2351 guint64 ntptime;
2352 RTPSource *source;
2353 gboolean created, prevsender;
2354
2355 gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
2356 &packet_count, &octet_count);
2357
2358 GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
2359 senderssrc, GST_TIME_ARGS (pinfo->current_time));
2360
2361 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2362 if (!source)
2363 return;
2364
2365 /* skip non-bye packets for sources that are marked BYE */
2366 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2367 goto out;
2368
2369 /* don't try to do lip-sync for sources that sent a BYE */
2370 if (RTP_SOURCE_IS_MARKED_BYE (source))
2371 *do_sync = FALSE;
2372 else
2373 *do_sync = TRUE;
2374
2375 prevsender = RTP_SOURCE_IS_SENDER (source);
2376
2377 /* first update the source */
2378 rtp_source_process_sr (source, pinfo->current_time, ntptime, rtptime,
2379 packet_count, octet_count);
2380
2381 source_update_sender (sess, source, prevsender);
2382
2383 if (created)
2384 on_new_ssrc (sess, source);
2385
2386 rtp_session_process_rb (sess, source, packet, pinfo);
2387
2388 out:
2389 g_object_unref (source);
2390 }
2391
2392 /* A receiver report contains statistics about how a receiver is doing. It
2393 * includes stuff like packet loss, jitter and the seqnum it received last. It
2394 * also contains info to calculate the round trip time.
2395 *
2396 * We are only interested in how the sender of this report is doing wrt to us.
2397 */
2398 static void
rtp_session_process_rr(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo)2399 rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
2400 RTPPacketInfo * pinfo)
2401 {
2402 guint32 senderssrc;
2403 RTPSource *source;
2404 gboolean created;
2405
2406 senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
2407
2408 GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
2409
2410 source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
2411 if (!source)
2412 return;
2413
2414 /* skip non-bye packets for sources that are marked BYE */
2415 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2416 goto out;
2417
2418 if (created)
2419 on_new_ssrc (sess, source);
2420
2421 rtp_session_process_rb (sess, source, packet, pinfo);
2422
2423 out:
2424 g_object_unref (source);
2425 }
2426
2427 /* Get SDES items and store them in the SSRC */
2428 static void
rtp_session_process_sdes(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo)2429 rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
2430 RTPPacketInfo * pinfo)
2431 {
2432 guint items, i, j;
2433 gboolean more_items, more_entries;
2434
2435 items = gst_rtcp_packet_sdes_get_item_count (packet);
2436 GST_DEBUG ("got SDES packet with %d items", items);
2437
2438 more_items = gst_rtcp_packet_sdes_first_item (packet);
2439 i = 0;
2440 while (more_items) {
2441 guint32 ssrc;
2442 gboolean changed, created, prevactive;
2443 RTPSource *source;
2444 GstStructure *sdes;
2445
2446 ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
2447
2448 GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
2449
2450 changed = FALSE;
2451
2452 /* find src, no probation when dealing with RTCP */
2453 source = obtain_source (sess, ssrc, &created, pinfo, FALSE);
2454 if (!source)
2455 return;
2456
2457 /* skip non-bye packets for sources that are marked BYE */
2458 if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
2459 goto next;
2460
2461 sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
2462
2463 more_entries = gst_rtcp_packet_sdes_first_entry (packet);
2464 j = 0;
2465 while (more_entries) {
2466 GstRTCPSDESType type;
2467 guint8 len;
2468 guint8 *data;
2469 gchar *name;
2470 gchar *value;
2471
2472 gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
2473
2474 GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
2475 data);
2476
2477 if (type == GST_RTCP_SDES_PRIV) {
2478 name = g_strndup ((const gchar *) &data[1], data[0]);
2479 len -= data[0] + 1;
2480 data += data[0] + 1;
2481 } else {
2482 name = g_strdup (gst_rtcp_sdes_type_to_name (type));
2483 }
2484
2485 value = g_strndup ((const gchar *) data, len);
2486
2487 if (g_utf8_validate (value, -1, NULL)) {
2488 gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
2489 } else {
2490 GST_WARNING ("ignore SDES field %s with non-utf8 data %s", name, value);
2491 }
2492
2493 g_free (name);
2494 g_free (value);
2495
2496 more_entries = gst_rtcp_packet_sdes_next_entry (packet);
2497 j++;
2498 }
2499
2500 /* takes ownership of sdes */
2501 changed = rtp_source_set_sdes_struct (source, sdes);
2502
2503 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2504 source->validated = TRUE;
2505
2506 if (created)
2507 on_new_ssrc (sess, source);
2508
2509 /* source became active */
2510 if (source_update_active (sess, source, prevactive))
2511 on_ssrc_validated (sess, source);
2512
2513 if (changed)
2514 on_ssrc_sdes (sess, source);
2515
2516 next:
2517 g_object_unref (source);
2518
2519 more_items = gst_rtcp_packet_sdes_next_item (packet);
2520 i++;
2521 }
2522 }
2523
2524 /* BYE is sent when a client leaves the session
2525 */
2526 static void
rtp_session_process_bye(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo)2527 rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
2528 RTPPacketInfo * pinfo)
2529 {
2530 guint count, i;
2531 gchar *reason;
2532 gboolean reconsider = FALSE;
2533
2534 reason = gst_rtcp_packet_bye_get_reason (packet);
2535 GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
2536
2537 count = gst_rtcp_packet_bye_get_ssrc_count (packet);
2538 for (i = 0; i < count; i++) {
2539 guint32 ssrc;
2540 RTPSource *source;
2541 gboolean prevactive, prevsender;
2542 guint pmembers, members;
2543
2544 ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
2545 GST_DEBUG ("SSRC: %08x", ssrc);
2546
2547 /* find src and mark bye, no probation when dealing with RTCP */
2548 source = find_source (sess, ssrc);
2549 if (!source || source->internal) {
2550 GST_DEBUG ("Ignoring suspicious BYE packet (reason: %s)",
2551 !source ? "can't find source" : "has internal source SSRC");
2552 break;
2553 }
2554
2555 /* store time for when we need to time out this source */
2556 source->bye_time = pinfo->current_time;
2557
2558 prevactive = RTP_SOURCE_IS_ACTIVE (source);
2559 prevsender = RTP_SOURCE_IS_SENDER (source);
2560
2561 /* mark the source BYE */
2562 rtp_source_mark_bye (source, reason);
2563
2564 pmembers = sess->stats.active_sources;
2565
2566 source_update_active (sess, source, prevactive);
2567 source_update_sender (sess, source, prevsender);
2568
2569 members = sess->stats.active_sources;
2570
2571 if (!sess->scheduled_bye && members < pmembers) {
2572 /* some members went away since the previous timeout estimate.
2573 * Perform reverse reconsideration but only when we are not scheduling a
2574 * BYE ourselves. */
2575 if (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
2576 pinfo->current_time < sess->next_rtcp_check_time) {
2577 GstClockTime time_remaining;
2578
2579 /* Scale our next RTCP check time according to the change of numbers
2580 * of members. But only if a) this is the first RTCP, or b) this is not
2581 * a feedback session, or c) this is a feedback session but we schedule
2582 * for every RTCP interval (aka no t-rr-interval set).
2583 *
2584 * FIXME: a) and b) are not great as we will possibly go below Tmin
2585 * for non-feedback profiles and in case of a) below
2586 * Tmin/t-rr-interval in any case.
2587 */
2588 if (sess->last_rtcp_send_time == GST_CLOCK_TIME_NONE ||
2589 !(sess->rtp_profile == GST_RTP_PROFILE_AVPF
2590 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF) ||
2591 sess->next_rtcp_check_time - sess->last_rtcp_send_time ==
2592 sess->last_rtcp_interval) {
2593 time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
2594 sess->next_rtcp_check_time =
2595 gst_util_uint64_scale (time_remaining, members, pmembers);
2596 sess->next_rtcp_check_time += pinfo->current_time;
2597 }
2598 sess->last_rtcp_interval =
2599 gst_util_uint64_scale (sess->last_rtcp_interval, members, pmembers);
2600
2601 GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
2602 GST_TIME_ARGS (sess->next_rtcp_check_time));
2603
2604 /* mark pending reconsider. We only want to signal the reconsideration
2605 * once after we handled all the source in the bye packet */
2606 reconsider = TRUE;
2607 }
2608 }
2609
2610 on_bye_ssrc (sess, source);
2611 }
2612 if (reconsider) {
2613 RTP_SESSION_UNLOCK (sess);
2614 /* notify app of reconsideration */
2615 if (sess->callbacks.reconsider)
2616 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2617 RTP_SESSION_LOCK (sess);
2618 }
2619
2620 g_free (reason);
2621 }
2622
2623 static void
rtp_session_process_app(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo)2624 rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
2625 RTPPacketInfo * pinfo)
2626 {
2627 GST_DEBUG ("received APP");
2628
2629 if (g_signal_has_handler_pending (sess,
2630 rtp_session_signals[SIGNAL_ON_APP_RTCP], 0, TRUE)) {
2631 GstBuffer *data_buffer = NULL;
2632 guint16 data_length;
2633 gchar name[5];
2634
2635 data_length = gst_rtcp_packet_app_get_data_length (packet) * 4;
2636 if (data_length > 0) {
2637 guint8 *data = gst_rtcp_packet_app_get_data (packet);
2638 data_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2639 GST_BUFFER_COPY_MEMORY, data - packet->rtcp->map.data, data_length);
2640 GST_BUFFER_PTS (data_buffer) = pinfo->running_time;
2641 }
2642
2643 memcpy (name, gst_rtcp_packet_app_get_name (packet), 4);
2644 name[4] = '\0';
2645
2646 RTP_SESSION_UNLOCK (sess);
2647 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_APP_RTCP], 0,
2648 gst_rtcp_packet_app_get_subtype (packet),
2649 gst_rtcp_packet_app_get_ssrc (packet), name, data_buffer);
2650 RTP_SESSION_LOCK (sess);
2651
2652 if (data_buffer)
2653 gst_buffer_unref (data_buffer);
2654 }
2655 }
2656
2657 static gboolean
rtp_session_request_local_key_unit(RTPSession * sess,RTPSource * src,guint32 media_ssrc,gboolean fir,GstClockTime current_time)2658 rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
2659 guint32 media_ssrc, gboolean fir, GstClockTime current_time)
2660 {
2661 guint32 round_trip = 0;
2662
2663 rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
2664
2665 if (src->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
2666 GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
2667 GST_SECOND, 65536);
2668
2669 /* Sanity check to avoid always ignoring PLI/FIR if we receive RTCP
2670 * packets with erroneous values resulting in crazy high RTT. */
2671 if (round_trip_in_ns > 5 * GST_SECOND)
2672 round_trip_in_ns = GST_SECOND / 2;
2673
2674 if (current_time - src->last_keyframe_request < 2 * round_trip_in_ns) {
2675 GST_DEBUG ("Ignoring %s request from %X because one was send without one "
2676 "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
2677 fir ? "FIR" : "PLI", rtp_source_get_ssrc (src),
2678 GST_TIME_ARGS (current_time - src->last_keyframe_request),
2679 GST_TIME_ARGS (round_trip_in_ns));
2680 return FALSE;
2681 }
2682 }
2683
2684 src->last_keyframe_request = current_time;
2685
2686 GST_LOG ("received %s request from %X about %X %p(%p)", fir ? "FIR" : "PLI",
2687 rtp_source_get_ssrc (src), media_ssrc, sess->callbacks.process_rtp,
2688 sess->callbacks.request_key_unit);
2689
2690 RTP_SESSION_UNLOCK (sess);
2691 sess->callbacks.request_key_unit (sess, media_ssrc, fir,
2692 sess->request_key_unit_user_data);
2693 RTP_SESSION_LOCK (sess);
2694
2695 return TRUE;
2696 }
2697
2698 static void
rtp_session_process_pli(RTPSession * sess,guint32 sender_ssrc,guint32 media_ssrc,GstClockTime current_time)2699 rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
2700 guint32 media_ssrc, GstClockTime current_time)
2701 {
2702 RTPSource *src;
2703
2704 if (!sess->callbacks.request_key_unit)
2705 return;
2706
2707 src = find_source (sess, sender_ssrc);
2708 if (src == NULL) {
2709 /* try to find a src with media_ssrc instead */
2710 src = find_source (sess, media_ssrc);
2711 if (src == NULL)
2712 return;
2713 }
2714
2715 rtp_session_request_local_key_unit (sess, src, media_ssrc, FALSE,
2716 current_time);
2717 }
2718
2719 static void
rtp_session_process_fir(RTPSession * sess,guint32 sender_ssrc,guint32 media_ssrc,guint8 * fci_data,guint fci_length,GstClockTime current_time)2720 rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
2721 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2722 GstClockTime current_time)
2723 {
2724 RTPSource *src;
2725 guint32 ssrc;
2726 guint position = 0;
2727 gboolean our_request = FALSE;
2728
2729 if (!sess->callbacks.request_key_unit)
2730 return;
2731
2732 if (fci_length < 8)
2733 return;
2734
2735 src = find_source (sess, sender_ssrc);
2736
2737 /* Hack because Google fails to set the sender_ssrc correctly */
2738 if (!src && sender_ssrc == 1) {
2739 GHashTableIter iter;
2740
2741 /* we can't find the source if there are multiple */
2742 if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 1)
2743 return;
2744
2745 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
2746 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
2747 if (!src->internal && rtp_source_is_sender (src))
2748 break;
2749 src = NULL;
2750 }
2751 }
2752 if (!src)
2753 return;
2754
2755 for (position = 0; position < fci_length; position += 8) {
2756 guint8 *data = fci_data + position;
2757 RTPSource *own;
2758
2759 ssrc = GST_READ_UINT32_BE (data);
2760
2761 own = find_source (sess, ssrc);
2762 if (own == NULL)
2763 continue;
2764
2765 if (own->internal) {
2766 our_request = TRUE;
2767 break;
2768 }
2769 }
2770 if (!our_request)
2771 return;
2772
2773 rtp_session_request_local_key_unit (sess, src, media_ssrc, TRUE,
2774 current_time);
2775 }
2776
2777 static void
rtp_session_process_nack(RTPSession * sess,guint32 sender_ssrc,guint32 media_ssrc,guint8 * fci_data,guint fci_length,GstClockTime current_time)2778 rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
2779 guint32 media_ssrc, guint8 * fci_data, guint fci_length,
2780 GstClockTime current_time)
2781 {
2782 sess->stats.nacks_received++;
2783
2784 if (!sess->callbacks.notify_nack)
2785 return;
2786
2787 while (fci_length > 0) {
2788 guint16 seqnum, blp;
2789
2790 seqnum = GST_READ_UINT16_BE (fci_data);
2791 blp = GST_READ_UINT16_BE (fci_data + 2);
2792
2793 GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
2794
2795 RTP_SESSION_UNLOCK (sess);
2796 sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
2797 sess->notify_nack_user_data);
2798 RTP_SESSION_LOCK (sess);
2799
2800 fci_data += 4;
2801 fci_length -= 4;
2802 }
2803 }
2804
2805 static void
rtp_session_process_feedback(RTPSession * sess,GstRTCPPacket * packet,RTPPacketInfo * pinfo,GstClockTime current_time)2806 rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
2807 RTPPacketInfo * pinfo, GstClockTime current_time)
2808 {
2809 GstRTCPType type;
2810 GstRTCPFBType fbtype;
2811 guint32 sender_ssrc, media_ssrc;
2812 guint8 *fci_data;
2813 guint fci_length;
2814 RTPSource *src;
2815
2816 /* The feedback packet must include both sender SSRC and media SSRC */
2817 if (packet->length < 2)
2818 return;
2819
2820 type = gst_rtcp_packet_get_type (packet);
2821 fbtype = gst_rtcp_packet_fb_get_type (packet);
2822 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
2823 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
2824
2825 src = find_source (sess, media_ssrc);
2826
2827 /* skip non-bye packets for sources that are marked BYE */
2828 if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
2829 return;
2830
2831 if (src)
2832 g_object_ref (src);
2833
2834 fci_data = gst_rtcp_packet_fb_get_fci (packet);
2835 fci_length = gst_rtcp_packet_fb_get_fci_length (packet) * sizeof (guint32);
2836
2837 GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of "
2838 "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length);
2839
2840 if (g_signal_has_handler_pending (sess,
2841 rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
2842 GstBuffer *fci_buffer = NULL;
2843
2844 if (fci_length > 0) {
2845 fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
2846 GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data,
2847 fci_length);
2848 GST_BUFFER_PTS (fci_buffer) = pinfo->running_time;
2849 }
2850
2851 RTP_SESSION_UNLOCK (sess);
2852 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
2853 type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
2854 RTP_SESSION_LOCK (sess);
2855
2856 if (fci_buffer)
2857 gst_buffer_unref (fci_buffer);
2858 }
2859
2860 if (src && sess->rtcp_feedback_retention_window != GST_CLOCK_TIME_NONE) {
2861 rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
2862 }
2863
2864 if ((src && src->internal) ||
2865 /* PSFB FIR puts the media ssrc inside the FCI */
2866 (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) {
2867 switch (type) {
2868 case GST_RTCP_TYPE_PSFB:
2869 switch (fbtype) {
2870 case GST_RTCP_PSFB_TYPE_PLI:
2871 if (src)
2872 src->stats.recv_pli_count++;
2873 rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
2874 current_time);
2875 break;
2876 case GST_RTCP_PSFB_TYPE_FIR:
2877 if (src)
2878 src->stats.recv_fir_count++;
2879 rtp_session_process_fir (sess, sender_ssrc, media_ssrc, fci_data,
2880 fci_length, current_time);
2881 break;
2882 default:
2883 break;
2884 }
2885 break;
2886 case GST_RTCP_TYPE_RTPFB:
2887 switch (fbtype) {
2888 case GST_RTCP_RTPFB_TYPE_NACK:
2889 if (src)
2890 src->stats.recv_nack_count++;
2891 rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
2892 fci_data, fci_length, current_time);
2893 break;
2894 default:
2895 break;
2896 }
2897 default:
2898 break;
2899 }
2900 }
2901
2902 if (src)
2903 g_object_unref (src);
2904 }
2905
2906 /**
2907 * rtp_session_process_rtcp:
2908 * @sess: and #RTPSession
2909 * @buffer: an RTCP buffer
2910 * @current_time: the current system time
2911 * @ntpnstime: the current NTP time in nanoseconds
2912 *
2913 * Process an RTCP buffer in the session manager. This function takes ownership
2914 * of @buffer.
2915 *
2916 * Returns: a #GstFlowReturn.
2917 */
2918 GstFlowReturn
rtp_session_process_rtcp(RTPSession * sess,GstBuffer * buffer,GstClockTime current_time,GstClockTime running_time,guint64 ntpnstime)2919 rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
2920 GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
2921 {
2922 GstRTCPPacket packet;
2923 gboolean more, is_bye = FALSE, do_sync = FALSE;
2924 RTPPacketInfo pinfo = { 0, };
2925 GstFlowReturn result = GST_FLOW_OK;
2926 GstRTCPBuffer rtcp = { NULL, };
2927
2928 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2929 g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
2930
2931 if (!gst_rtcp_buffer_validate_reduced (buffer))
2932 goto invalid_packet;
2933
2934 GST_DEBUG ("received RTCP packet");
2935
2936 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
2937 buffer);
2938
2939 RTP_SESSION_LOCK (sess);
2940 /* update pinfo stats */
2941 update_packet_info (sess, &pinfo, FALSE, FALSE, FALSE, buffer, current_time,
2942 running_time, ntpnstime);
2943
2944 /* start processing the compound packet */
2945 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
2946 more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet);
2947 while (more) {
2948 GstRTCPType type;
2949
2950 type = gst_rtcp_packet_get_type (&packet);
2951
2952 switch (type) {
2953 case GST_RTCP_TYPE_SR:
2954 rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
2955 break;
2956 case GST_RTCP_TYPE_RR:
2957 rtp_session_process_rr (sess, &packet, &pinfo);
2958 break;
2959 case GST_RTCP_TYPE_SDES:
2960 rtp_session_process_sdes (sess, &packet, &pinfo);
2961 break;
2962 case GST_RTCP_TYPE_BYE:
2963 is_bye = TRUE;
2964 /* don't try to attempt lip-sync anymore for streams with a BYE */
2965 do_sync = FALSE;
2966 rtp_session_process_bye (sess, &packet, &pinfo);
2967 break;
2968 case GST_RTCP_TYPE_APP:
2969 rtp_session_process_app (sess, &packet, &pinfo);
2970 break;
2971 case GST_RTCP_TYPE_RTPFB:
2972 case GST_RTCP_TYPE_PSFB:
2973 rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
2974 break;
2975 case GST_RTCP_TYPE_XR:
2976 /* FIXME: This block is added to downgrade warning level.
2977 * Once the parser is implemented, it should be replaced with
2978 * a proper process function. */
2979 GST_DEBUG ("got RTCP XR packet, but ignored");
2980 break;
2981 default:
2982 GST_WARNING ("got unknown RTCP packet type: %d", type);
2983 break;
2984 }
2985 more = gst_rtcp_packet_move_to_next (&packet);
2986 }
2987
2988 gst_rtcp_buffer_unmap (&rtcp);
2989
2990 /* if we are scheduling a BYE, we only want to count bye packets, else we
2991 * count everything */
2992 if (sess->scheduled_bye && is_bye) {
2993 sess->bye_stats.bye_members++;
2994 UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
2995 }
2996
2997 /* keep track of average packet size */
2998 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
2999
3000 GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
3001 sess->stats.avg_rtcp_packet_size, pinfo.bytes);
3002 RTP_SESSION_UNLOCK (sess);
3003
3004 pinfo.data = NULL;
3005 clean_packet_info (&pinfo);
3006
3007 /* notify caller of sr packets in the callback */
3008 if (do_sync && sess->callbacks.sync_rtcp) {
3009 result = sess->callbacks.sync_rtcp (sess, buffer,
3010 sess->sync_rtcp_user_data);
3011 } else
3012 gst_buffer_unref (buffer);
3013
3014 return result;
3015
3016 /* ERRORS */
3017 invalid_packet:
3018 {
3019 GST_DEBUG ("invalid RTCP packet received");
3020 gst_buffer_unref (buffer);
3021 return GST_FLOW_OK;
3022 }
3023 }
3024
3025 /**
3026 * rtp_session_update_send_caps:
3027 * @sess: an #RTPSession
3028 * @caps: a #GstCaps
3029 *
3030 * Update the caps of the sender in the rtp session.
3031 */
3032 void
rtp_session_update_send_caps(RTPSession * sess,GstCaps * caps)3033 rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps)
3034 {
3035 GstStructure *s;
3036 guint ssrc;
3037
3038 g_return_if_fail (RTP_IS_SESSION (sess));
3039 g_return_if_fail (GST_IS_CAPS (caps));
3040
3041 GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
3042
3043 s = gst_caps_get_structure (caps, 0);
3044
3045 if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
3046 RTPSource *source;
3047 gboolean created;
3048
3049 RTP_SESSION_LOCK (sess);
3050 source = obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
3051 sess->suggested_ssrc = ssrc;
3052 sess->internal_ssrc_set = TRUE;
3053 sess->internal_ssrc_from_caps_or_property = TRUE;
3054 if (source) {
3055 rtp_source_update_caps (source, caps);
3056
3057 if (created)
3058 on_new_sender_ssrc (sess, source);
3059
3060 g_object_unref (source);
3061 }
3062
3063 if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
3064 source =
3065 obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
3066 if (source) {
3067 rtp_source_update_caps (source, caps);
3068
3069 if (created)
3070 on_new_sender_ssrc (sess, source);
3071
3072 g_object_unref (source);
3073 }
3074 }
3075 RTP_SESSION_UNLOCK (sess);
3076 } else {
3077 sess->internal_ssrc_from_caps_or_property = FALSE;
3078 }
3079 }
3080
3081 /**
3082 * rtp_session_send_rtp:
3083 * @sess: an #RTPSession
3084 * @data: pointer to either an RTP buffer or a list of RTP buffers
3085 * @is_list: TRUE when @data is a buffer list
3086 * @current_time: the current system time
3087 * @running_time: the running time of @data
3088 *
3089 * Send the RTP data (a buffer or buffer list) in the session manager. This
3090 * function takes ownership of @data.
3091 *
3092 * Returns: a #GstFlowReturn.
3093 */
3094 GstFlowReturn
rtp_session_send_rtp(RTPSession * sess,gpointer data,gboolean is_list,GstClockTime current_time,GstClockTime running_time)3095 rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
3096 GstClockTime current_time, GstClockTime running_time)
3097 {
3098 GstFlowReturn result;
3099 RTPSource *source;
3100 gboolean prevsender;
3101 guint64 oldrate;
3102 RTPPacketInfo pinfo = { 0, };
3103 gboolean created;
3104
3105 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3106 g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
3107
3108 GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
3109
3110 RTP_SESSION_LOCK (sess);
3111 if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
3112 current_time, running_time, -1))
3113 goto invalid_packet;
3114
3115 source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
3116 if (created)
3117 on_new_sender_ssrc (sess, source);
3118
3119 if (!source->internal)
3120 /* FIXME: Send GstRTPCollision upstream */
3121 goto collision;
3122
3123 prevsender = RTP_SOURCE_IS_SENDER (source);
3124 oldrate = source->bitrate;
3125
3126 /* we use our own source to send */
3127 result = rtp_source_send_rtp (source, &pinfo);
3128
3129 source_update_sender (sess, source, prevsender);
3130
3131 if (oldrate != source->bitrate)
3132 sess->recalc_bandwidth = TRUE;
3133 RTP_SESSION_UNLOCK (sess);
3134
3135 g_object_unref (source);
3136 clean_packet_info (&pinfo);
3137
3138 return result;
3139
3140 invalid_packet:
3141 {
3142 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
3143 RTP_SESSION_UNLOCK (sess);
3144 GST_DEBUG ("invalid RTP packet received");
3145 return GST_FLOW_OK;
3146 }
3147 collision:
3148 {
3149 g_object_unref (source);
3150 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
3151 RTP_SESSION_UNLOCK (sess);
3152 GST_WARNING ("non-internal source with same ssrc %08x, drop packet",
3153 pinfo.ssrc);
3154 return GST_FLOW_OK;
3155 }
3156 }
3157
3158 static void
add_bitrates(gpointer key,RTPSource * source,gdouble * bandwidth)3159 add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
3160 {
3161 *bandwidth += source->bitrate;
3162 }
3163
3164 /* must be called with session lock */
3165 static GstClockTime
calculate_rtcp_interval(RTPSession * sess,gboolean deterministic,gboolean first)3166 calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
3167 gboolean first)
3168 {
3169 GstClockTime result;
3170 RTPSessionStats *stats;
3171
3172 /* recalculate bandwidth when it changed */
3173 if (sess->recalc_bandwidth) {
3174 gdouble bandwidth;
3175
3176 if (sess->bandwidth > 0)
3177 bandwidth = sess->bandwidth;
3178 else {
3179 /* If it is <= 0, then try to estimate the actual bandwidth */
3180 bandwidth = 0;
3181
3182 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3183 (GHFunc) add_bitrates, &bandwidth);
3184 }
3185 if (bandwidth < RTP_STATS_BANDWIDTH)
3186 bandwidth = RTP_STATS_BANDWIDTH;
3187
3188 rtp_stats_set_bandwidths (&sess->stats, bandwidth,
3189 sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
3190
3191 sess->recalc_bandwidth = FALSE;
3192 }
3193
3194 if (sess->scheduled_bye) {
3195 stats = &sess->bye_stats;
3196 result = rtp_stats_calculate_bye_interval (stats);
3197 } else {
3198 session_update_ptp (sess);
3199
3200 stats = &sess->stats;
3201 result = rtp_stats_calculate_rtcp_interval (stats,
3202 stats->internal_sender_sources > 0, sess->rtp_profile,
3203 sess->is_doing_ptp, first);
3204 }
3205
3206 GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
3207 GST_TIME_ARGS (result), first);
3208
3209 if (!deterministic && result != GST_CLOCK_TIME_NONE)
3210 result = rtp_stats_add_rtcp_jitter (stats, result);
3211
3212 GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
3213
3214 return result;
3215 }
3216
3217 static void
source_mark_bye(const gchar * key,RTPSource * source,const gchar * reason)3218 source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
3219 {
3220 if (source->internal)
3221 rtp_source_mark_bye (source, reason);
3222 }
3223
3224 /**
3225 * rtp_session_mark_all_bye:
3226 * @sess: an #RTPSession
3227 * @reason: a reason
3228 *
3229 * Mark all internal sources of the session as BYE with @reason.
3230 */
3231 void
rtp_session_mark_all_bye(RTPSession * sess,const gchar * reason)3232 rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
3233 {
3234 g_return_if_fail (RTP_IS_SESSION (sess));
3235
3236 RTP_SESSION_LOCK (sess);
3237 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3238 (GHFunc) source_mark_bye, (gpointer) reason);
3239 RTP_SESSION_UNLOCK (sess);
3240 }
3241
3242 /* Stop the current @sess and schedule a BYE message for the other members.
3243 * One must have the session lock to call this function
3244 */
3245 static GstFlowReturn
rtp_session_schedule_bye_locked(RTPSession * sess,GstClockTime current_time)3246 rtp_session_schedule_bye_locked (RTPSession * sess, GstClockTime current_time)
3247 {
3248 GstFlowReturn result = GST_FLOW_OK;
3249 GstClockTime interval;
3250
3251 /* nothing to do it we already scheduled bye */
3252 if (sess->scheduled_bye)
3253 goto done;
3254
3255 /* we schedule BYE now */
3256 sess->scheduled_bye = TRUE;
3257 /* at least one member wants to send a BYE */
3258 memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
3259 INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
3260 sess->bye_stats.bye_members = 1;
3261 sess->first_rtcp = TRUE;
3262
3263 /* reschedule transmission */
3264 sess->last_rtcp_send_time = current_time;
3265 sess->last_rtcp_check_time = current_time;
3266 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3267
3268 if (interval != GST_CLOCK_TIME_NONE)
3269 sess->next_rtcp_check_time = current_time + interval;
3270 else
3271 sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
3272 sess->last_rtcp_interval = interval;
3273
3274 GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
3275 GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
3276
3277 RTP_SESSION_UNLOCK (sess);
3278 /* notify app of reconsideration */
3279 if (sess->callbacks.reconsider)
3280 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
3281 RTP_SESSION_LOCK (sess);
3282 done:
3283
3284 return result;
3285 }
3286
3287 /**
3288 * rtp_session_schedule_bye:
3289 * @sess: an #RTPSession
3290 * @current_time: the current system time
3291 *
3292 * Schedule a BYE message for all sources marked as BYE in @sess.
3293 *
3294 * Returns: a #GstFlowReturn.
3295 */
3296 GstFlowReturn
rtp_session_schedule_bye(RTPSession * sess,GstClockTime current_time)3297 rtp_session_schedule_bye (RTPSession * sess, GstClockTime current_time)
3298 {
3299 GstFlowReturn result;
3300
3301 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
3302
3303 RTP_SESSION_LOCK (sess);
3304 result = rtp_session_schedule_bye_locked (sess, current_time);
3305 RTP_SESSION_UNLOCK (sess);
3306
3307 return result;
3308 }
3309
3310 /**
3311 * rtp_session_next_timeout:
3312 * @sess: an #RTPSession
3313 * @current_time: the current system time
3314 *
3315 * Get the next time we should perform session maintenance tasks.
3316 *
3317 * Returns: a time when rtp_session_on_timeout() should be called with the
3318 * current system time.
3319 */
3320 GstClockTime
rtp_session_next_timeout(RTPSession * sess,GstClockTime current_time)3321 rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
3322 {
3323 GstClockTime result, interval = 0;
3324
3325 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
3326
3327 RTP_SESSION_LOCK (sess);
3328
3329 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
3330 GST_DEBUG ("have early rtcp time");
3331 result = sess->next_early_rtcp_time;
3332 goto early_exit;
3333 }
3334
3335 result = sess->next_rtcp_check_time;
3336
3337 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3338 ", next time: %" GST_TIME_FORMAT,
3339 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3340
3341 if (result == GST_CLOCK_TIME_NONE || result < current_time) {
3342 GST_DEBUG ("take current time as base");
3343 /* our previous check time expired, start counting from the current time
3344 * again. */
3345 result = current_time;
3346 }
3347
3348 if (sess->scheduled_bye) {
3349 if (sess->bye_stats.active_sources >= 50) {
3350 GST_DEBUG ("reconsider BYE, more than 50 sources");
3351 /* reconsider BYE if members >= 50 */
3352 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3353 sess->last_rtcp_interval = interval;
3354 }
3355 } else {
3356 if (sess->first_rtcp) {
3357 GST_DEBUG ("first RTCP packet");
3358 /* we are called for the first time */
3359 interval = calculate_rtcp_interval (sess, FALSE, TRUE);
3360 sess->last_rtcp_interval = interval;
3361 } else if (sess->next_rtcp_check_time < current_time) {
3362 GST_DEBUG ("old check time expired, getting new timeout");
3363 /* get a new timeout when we need to */
3364 interval = calculate_rtcp_interval (sess, FALSE, FALSE);
3365 sess->last_rtcp_interval = interval;
3366
3367 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
3368 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
3369 && interval != GST_CLOCK_TIME_NONE) {
3370 /* Apply the rules from RFC 4585 section 3.5.3 */
3371 if (sess->stats.min_interval != 0) {
3372 GstClockTime T_rr_current_interval = g_random_double_range (0.5,
3373 1.5) * sess->stats.min_interval * GST_SECOND;
3374
3375 if (T_rr_current_interval > interval) {
3376 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
3377 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
3378 GST_TIME_ARGS (interval));
3379 interval = T_rr_current_interval;
3380 }
3381 }
3382 }
3383 }
3384 }
3385
3386 if (interval != GST_CLOCK_TIME_NONE)
3387 result += interval;
3388 else
3389 result = GST_CLOCK_TIME_NONE;
3390
3391 sess->next_rtcp_check_time = result;
3392
3393 early_exit:
3394
3395 GST_DEBUG ("current time: %" GST_TIME_FORMAT
3396 ", next time: %" GST_TIME_FORMAT,
3397 GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
3398 RTP_SESSION_UNLOCK (sess);
3399
3400 return result;
3401 }
3402
3403 typedef struct
3404 {
3405 RTPSource *source;
3406 gboolean is_bye;
3407 GstBuffer *buffer;
3408 } ReportOutput;
3409
3410 typedef struct
3411 {
3412 GstRTCPBuffer rtcpbuf;
3413 RTPSession *sess;
3414 RTPSource *source;
3415 guint num_to_report;
3416 gboolean have_fir;
3417 gboolean have_pli;
3418 gboolean have_nack;
3419 GstBuffer *rtcp;
3420 GstClockTime current_time;
3421 guint64 ntpnstime;
3422 GstClockTime running_time;
3423 GstClockTime interval;
3424 GstRTCPPacket packet;
3425 gboolean has_sdes;
3426 gboolean is_early;
3427 gboolean may_suppress;
3428 GQueue output;
3429 guint nacked_seqnums;
3430 } ReportData;
3431
3432 static void
session_start_rtcp(RTPSession * sess,ReportData * data)3433 session_start_rtcp (RTPSession * sess, ReportData * data)
3434 {
3435 GstRTCPPacket *packet = &data->packet;
3436 RTPSource *own = data->source;
3437 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3438
3439 data->rtcp = gst_rtcp_buffer_new (sess->mtu);
3440 data->has_sdes = FALSE;
3441
3442 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3443
3444 if (data->is_early && sess->reduced_size_rtcp)
3445 return;
3446
3447 if (RTP_SOURCE_IS_SENDER (own)) {
3448 guint64 ntptime;
3449 guint32 rtptime;
3450 guint32 packet_count, octet_count;
3451
3452 /* we are a sender, create SR */
3453 GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
3454 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet);
3455
3456 /* get latest stats */
3457 rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
3458 &ntptime, &rtptime, &packet_count, &octet_count);
3459 /* store stats */
3460 rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
3461 packet_count, octet_count);
3462
3463 /* fill in sender report info */
3464 gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
3465 sess->timestamp_sender_reports ? ntptime : 0,
3466 sess->timestamp_sender_reports ? rtptime : 0,
3467 packet_count, octet_count);
3468 } else {
3469 /* we are only receiver, create RR */
3470 GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
3471 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet);
3472 gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
3473 }
3474 }
3475
3476 /* construct a Sender or Receiver Report */
3477 static void
session_report_blocks(const gchar * key,RTPSource * source,ReportData * data)3478 session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
3479 {
3480 RTPSession *sess = data->sess;
3481 GstRTCPPacket *packet = &data->packet;
3482 guint8 fractionlost;
3483 gint32 packetslost;
3484 guint32 exthighestseq, jitter;
3485 guint32 lsr, dlsr;
3486
3487 /* don't report for sources in future generations */
3488 if (((gint16) (source->generation - sess->generation)) > 0) {
3489 GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
3490 source->generation, sess->generation);
3491 return;
3492 }
3493
3494 if (g_hash_table_contains (source->reported_in_sr_of,
3495 GUINT_TO_POINTER (data->source->ssrc))) {
3496 GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
3497 return;
3498 }
3499
3500 if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
3501 GST_DEBUG ("max RB count reached");
3502 return;
3503 }
3504
3505 /* only report about remote sources */
3506 if (source->internal)
3507 goto reported;
3508
3509 if (!RTP_SOURCE_IS_SENDER (source)) {
3510 GST_DEBUG ("source %08x not sender", source->ssrc);
3511 goto reported;
3512 }
3513
3514 if (source->disable_rtcp) {
3515 GST_DEBUG ("source %08x has RTCP disabled", source->ssrc);
3516 goto reported;
3517 }
3518
3519 GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
3520
3521 /* get new stats */
3522 rtp_source_get_new_rb (source, data->current_time, &fractionlost,
3523 &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
3524
3525 /* store last generated RR packet */
3526 source->last_rr.is_valid = TRUE;
3527 source->last_rr.fractionlost = fractionlost;
3528 source->last_rr.packetslost = packetslost;
3529 source->last_rr.exthighestseq = exthighestseq;
3530 source->last_rr.jitter = jitter;
3531 source->last_rr.lsr = lsr;
3532 source->last_rr.dlsr = dlsr;
3533
3534 /* packet is not yet filled, add report block for this source. */
3535 gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
3536 exthighestseq, jitter, lsr, dlsr);
3537
3538 reported:
3539 g_hash_table_add (source->reported_in_sr_of,
3540 GUINT_TO_POINTER (data->source->ssrc));
3541 }
3542
3543 /* construct FIR */
3544 static void
session_add_fir(const gchar * key,RTPSource * source,ReportData * data)3545 session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
3546 {
3547 GstRTCPPacket *packet = &data->packet;
3548 guint16 len;
3549 guint8 *fci_data;
3550
3551 if (!source->send_fir)
3552 return;
3553
3554 len = gst_rtcp_packet_fb_get_fci_length (packet);
3555 if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
3556 /* exit because the packet is full, will put next request in a
3557 * further packet */
3558 return;
3559
3560 fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
3561
3562 GST_WRITE_UINT32_BE (fci_data, source->ssrc);
3563 fci_data += 4;
3564 fci_data[0] = source->current_send_fir_seqnum;
3565 fci_data[1] = fci_data[2] = fci_data[3] = 0;
3566
3567 source->send_fir = FALSE;
3568 source->stats.sent_fir_count++;
3569 }
3570
3571 static void
session_fir(RTPSession * sess,ReportData * data)3572 session_fir (RTPSession * sess, ReportData * data)
3573 {
3574 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3575 GstRTCPPacket *packet = &data->packet;
3576
3577 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3578 return;
3579
3580 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
3581 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3582 gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
3583
3584 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
3585 (GHFunc) session_add_fir, data);
3586
3587 if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
3588 gst_rtcp_packet_remove (packet);
3589 else
3590 data->may_suppress = FALSE;
3591 }
3592
3593 static gboolean
has_pli_compare_func(gconstpointer a,gconstpointer ignored)3594 has_pli_compare_func (gconstpointer a, gconstpointer ignored)
3595 {
3596 GstRTCPPacket packet;
3597 GstRTCPBuffer rtcp = { NULL, };
3598 gboolean ret = FALSE;
3599
3600 gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp);
3601
3602 if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
3603 if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
3604 gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
3605 ret = TRUE;
3606 }
3607
3608 gst_rtcp_buffer_unmap (&rtcp);
3609
3610 return ret;
3611 }
3612
3613 /* construct PLI */
3614 static void
session_pli(const gchar * key,RTPSource * source,ReportData * data)3615 session_pli (const gchar * key, RTPSource * source, ReportData * data)
3616 {
3617 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3618 GstRTCPPacket *packet = &data->packet;
3619
3620 if (!source->send_pli)
3621 return;
3622
3623 if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
3624 return;
3625
3626 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
3627 /* exit because the packet is full, will put next request in a
3628 * further packet */
3629 return;
3630
3631 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
3632 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3633 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3634
3635 source->send_pli = FALSE;
3636 data->may_suppress = FALSE;
3637
3638 source->stats.sent_pli_count++;
3639 }
3640
3641 /* construct NACK */
3642 static void
session_nack(const gchar * key,RTPSource * source,ReportData * data)3643 session_nack (const gchar * key, RTPSource * source, ReportData * data)
3644 {
3645 RTPSession *sess = data->sess;
3646 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3647 GstRTCPPacket *packet = &data->packet;
3648 guint16 *nacks;
3649 GstClockTime *nack_deadlines;
3650 guint n_nacks, i = 0;
3651 guint nacked_seqnums = 0;
3652 guint16 n_fb_nacks = 0;
3653 guint8 *fci_data;
3654
3655 if (!source->send_nack)
3656 return;
3657
3658 nacks = rtp_source_get_nacks (source, &n_nacks);
3659 nack_deadlines = rtp_source_get_nack_deadlines (source, NULL);
3660 GST_DEBUG ("%u NACKs current time %" GST_TIME_FORMAT, n_nacks,
3661 GST_TIME_ARGS (data->current_time));
3662
3663 /* cleanup expired nacks */
3664 for (i = 0; i < n_nacks; i++) {
3665 GST_DEBUG ("#%u deadline %" GST_TIME_FORMAT, nacks[i],
3666 GST_TIME_ARGS (nack_deadlines[i]));
3667 if (nack_deadlines[i] >= data->current_time)
3668 break;
3669 }
3670
3671 if (data->is_early) {
3672 /* don't remove them all if this is an early RTCP packet. It may happen
3673 * that the NACKs are late due to high RTT, not sending NACKs at all would
3674 * keep the RTX RTT stats high and maintain a dropping state. */
3675 i = MIN (n_nacks - 1, i);
3676 }
3677
3678 if (i) {
3679 GST_WARNING ("Removing %u expired NACKS", i);
3680 rtp_source_clear_nacks (source, i);
3681 n_nacks -= i;
3682 if (n_nacks == 0)
3683 return;
3684 }
3685
3686 /* allow overriding NACK to packet conversion */
3687 if (g_signal_has_handler_pending (sess,
3688 rtp_session_signals[SIGNAL_ON_SENDING_NACKS], 0, TRUE)) {
3689 /* this is needed as it will actually resize the buffer */
3690 gst_rtcp_buffer_unmap (rtcp);
3691
3692 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_NACKS], 0,
3693 data->source->ssrc, source->ssrc, source->nacks, data->rtcp,
3694 &nacked_seqnums);
3695
3696 /* and now remap for the remaining work */
3697 gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
3698
3699 if (nacked_seqnums > 0)
3700 goto done;
3701 }
3702
3703 if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
3704 /* exit because the packet is full, will put next request in a
3705 * further packet */
3706 return;
3707
3708 gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
3709 gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
3710 gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
3711
3712 if (!gst_rtcp_packet_fb_set_fci_length (packet, 1)) {
3713 gst_rtcp_packet_remove (packet);
3714 GST_WARNING ("no nacks fit in the packet");
3715 return;
3716 }
3717
3718 fci_data = gst_rtcp_packet_fb_get_fci (packet);
3719 for (i = 0; i < n_nacks; i = nacked_seqnums) {
3720 guint16 seqnum = nacks[i];
3721 guint16 blp = 0;
3722 guint j;
3723
3724 if (!gst_rtcp_packet_fb_set_fci_length (packet, n_fb_nacks + 1))
3725 break;
3726
3727 n_fb_nacks++;
3728 nacked_seqnums++;
3729
3730 for (j = i + 1; j < n_nacks; j++) {
3731 gint diff;
3732
3733 diff = gst_rtp_buffer_compare_seqnum (seqnum, nacks[j]);
3734 GST_TRACE ("[%u][%u] %u %u diff %i", i, j, seqnum, nacks[j], diff);
3735 if (diff > 16)
3736 break;
3737
3738 blp |= 1 << (diff - 1);
3739 nacked_seqnums++;
3740 }
3741
3742 GST_WRITE_UINT32_BE (fci_data, seqnum << 16 | blp);
3743 fci_data += 4;
3744 }
3745
3746 GST_DEBUG ("Sent %u seqnums into %u FB NACKs", nacked_seqnums, n_fb_nacks);
3747 source->stats.sent_nack_count += n_fb_nacks;
3748
3749 done:
3750 data->nacked_seqnums += nacked_seqnums;
3751 rtp_source_clear_nacks (source, nacked_seqnums);
3752 data->may_suppress = FALSE;
3753 }
3754
3755 /* perform cleanup of sources that timed out */
3756 static void
session_cleanup(const gchar * key,RTPSource * source,ReportData * data)3757 session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
3758 {
3759 gboolean remove = FALSE;
3760 gboolean byetimeout = FALSE;
3761 gboolean sendertimeout = FALSE;
3762 gboolean is_sender, is_active;
3763 RTPSession *sess = data->sess;
3764 GstClockTime interval, binterval;
3765 GstClockTime btime;
3766
3767 GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
3768
3769 /* check for outdated collisions */
3770 if (source->internal) {
3771 GST_DEBUG ("Timing out collisions for %x", source->ssrc);
3772 rtp_source_timeout (source, data->current_time, data->running_time,
3773 sess->rtcp_feedback_retention_window);
3774 }
3775
3776 /* nothing else to do when without RTCP */
3777 if (data->interval == GST_CLOCK_TIME_NONE)
3778 return;
3779
3780 is_sender = RTP_SOURCE_IS_SENDER (source);
3781 is_active = RTP_SOURCE_IS_ACTIVE (source);
3782
3783 /* our own rtcp interval may have been forced low by secondary configuration,
3784 * while sender side may still operate with higher interval,
3785 * so do not just take our interval to decide on timing out sender,
3786 * but take (if data->interval <= 5 * GST_SECOND):
3787 * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND)
3788 * where sender_interval is difference between last 2 received RTCP reports
3789 */
3790 if (data->interval >= 5 * GST_SECOND || source->internal) {
3791 binterval = data->interval;
3792 } else {
3793 GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT,
3794 GST_TIME_ARGS (source->stats.prev_rtcptime),
3795 GST_TIME_ARGS (source->stats.last_rtcptime));
3796 /* if not received enough yet, fallback to larger default */
3797 if (source->stats.last_rtcptime > source->stats.prev_rtcptime)
3798 binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime;
3799 else
3800 binterval = 5 * GST_SECOND;
3801 binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND);
3802 }
3803 GST_LOG ("timeout base interval %" GST_TIME_FORMAT,
3804 GST_TIME_ARGS (binterval));
3805
3806 if (!source->internal && source->marked_bye) {
3807 /* if we received a BYE from the source, remove the source after some
3808 * time. */
3809 if (data->current_time > source->bye_time &&
3810 data->current_time - source->bye_time > sess->stats.bye_timeout) {
3811 GST_DEBUG ("removing BYE source %08x", source->ssrc);
3812 remove = TRUE;
3813 byetimeout = TRUE;
3814 }
3815 }
3816
3817 if (source->internal && source->sent_bye) {
3818 GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
3819 remove = TRUE;
3820 }
3821
3822 /* sources that were inactive for more than 5 times the deterministic reporting
3823 * interval get timed out. the min timeout is 5 seconds. */
3824 /* mind old time that might pre-date last time going to PLAYING */
3825 btime = MAX (source->last_activity, sess->start_time);
3826 if (data->current_time > btime) {
3827 interval = MAX (binterval * 5, 5 * GST_SECOND);
3828 if (data->current_time - btime > interval) {
3829 GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
3830 source->ssrc, GST_TIME_ARGS (btime));
3831 if (source->internal) {
3832 /* this is an internal source that is not using our suggested ssrc.
3833 * since there must be another source using this ssrc, we can remove
3834 * this one instead of making it a receiver forever */
3835 if (source->ssrc != sess->suggested_ssrc) {
3836 rtp_source_mark_bye (source, "timed out");
3837 /* do not schedule bye here, since we are inside the RTCP timeout
3838 * processing and scheduling bye will interfere with SR/RR sending */
3839 }
3840 } else {
3841 remove = TRUE;
3842 }
3843 }
3844 }
3845
3846 /* senders that did not send for a long time become a receiver, this also
3847 * holds for our own sources. */
3848 if (is_sender) {
3849 /* mind old time that might pre-date last time going to PLAYING */
3850 btime = MAX (source->last_rtp_activity, sess->start_time);
3851 if (data->current_time > btime) {
3852 interval = MAX (binterval * 2, 5 * GST_SECOND);
3853 if (data->current_time - btime > interval) {
3854 GST_DEBUG ("sender source %08x timed out and became receiver, last %"
3855 GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
3856 sendertimeout = TRUE;
3857 }
3858 }
3859 }
3860
3861 if (remove) {
3862 sess->total_sources--;
3863 if (is_sender) {
3864 sess->stats.sender_sources--;
3865 if (source->internal)
3866 sess->stats.internal_sender_sources--;
3867 }
3868 if (is_active)
3869 sess->stats.active_sources--;
3870
3871 if (source->internal)
3872 sess->stats.internal_sources--;
3873
3874 if (byetimeout)
3875 on_bye_timeout (sess, source);
3876 else
3877 on_timeout (sess, source);
3878 } else {
3879 if (sendertimeout) {
3880 source->is_sender = FALSE;
3881 sess->stats.sender_sources--;
3882 if (source->internal)
3883 sess->stats.internal_sender_sources--;
3884
3885 on_sender_timeout (sess, source);
3886 }
3887 /* count how many source to report in this generation */
3888 if (((gint16) (source->generation - sess->generation)) <= 0)
3889 data->num_to_report++;
3890 }
3891 source->closing = remove;
3892 }
3893
3894 static void
session_sdes(RTPSession * sess,ReportData * data)3895 session_sdes (RTPSession * sess, ReportData * data)
3896 {
3897 GstRTCPPacket *packet = &data->packet;
3898 const GstStructure *sdes;
3899 gint i, n_fields;
3900 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3901
3902 /* add SDES packet */
3903 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet);
3904
3905 gst_rtcp_packet_sdes_add_item (packet, data->source->ssrc);
3906
3907 sdes = rtp_source_get_sdes_struct (data->source);
3908
3909 /* add all fields in the structure, the order is not important. */
3910 n_fields = gst_structure_n_fields (sdes);
3911 for (i = 0; i < n_fields; ++i) {
3912 const gchar *field;
3913 const gchar *value;
3914 GstRTCPSDESType type;
3915
3916 field = gst_structure_nth_field_name (sdes, i);
3917 if (field == NULL)
3918 continue;
3919 value = gst_structure_get_string (sdes, field);
3920 if (value == NULL)
3921 continue;
3922 type = gst_rtcp_sdes_name_to_type (field);
3923
3924 /* Early packets are minimal and only include the CNAME */
3925 if (data->is_early && type != GST_RTCP_SDES_CNAME)
3926 continue;
3927
3928 if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
3929 gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
3930 (const guint8 *) value);
3931 } else if (type == GST_RTCP_SDES_PRIV) {
3932 gsize prefix_len;
3933 gsize value_len;
3934 gsize data_len;
3935 guint8 data[256];
3936
3937 /* don't accept entries that are too big */
3938 prefix_len = strlen (field);
3939 if (prefix_len > 255)
3940 continue;
3941 value_len = strlen (value);
3942 if (value_len > 255)
3943 continue;
3944 data_len = 1 + prefix_len + value_len;
3945 if (data_len > 255)
3946 continue;
3947
3948 data[0] = prefix_len;
3949 memcpy (&data[1], field, prefix_len);
3950 memcpy (&data[1 + prefix_len], value, value_len);
3951
3952 gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
3953 }
3954 }
3955
3956 data->has_sdes = TRUE;
3957 }
3958
3959 /* schedule a BYE packet */
3960 static void
make_source_bye(RTPSession * sess,RTPSource * source,ReportData * data)3961 make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
3962 {
3963 GstRTCPPacket *packet = &data->packet;
3964 GstRTCPBuffer *rtcp = &data->rtcpbuf;
3965
3966 /* add SDES */
3967 session_sdes (sess, data);
3968 /* add a BYE packet */
3969 gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet);
3970 gst_rtcp_packet_bye_add_ssrc (packet, source->ssrc);
3971 if (source->bye_reason)
3972 gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
3973
3974 /* we have a BYE packet now */
3975 source->sent_bye = TRUE;
3976 }
3977
3978 static gboolean
is_rtcp_time(RTPSession * sess,GstClockTime current_time,ReportData * data)3979 is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
3980 {
3981 GstClockTime new_send_time;
3982 GstClockTime interval;
3983 RTPSessionStats *stats;
3984
3985 if (sess->scheduled_bye)
3986 stats = &sess->bye_stats;
3987 else
3988 stats = &sess->stats;
3989
3990 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
3991 data->is_early = TRUE;
3992 else
3993 data->is_early = FALSE;
3994
3995 if (data->is_early && sess->next_early_rtcp_time <= current_time) {
3996 GST_DEBUG ("early feedback %" GST_TIME_FORMAT " <= now %"
3997 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
3998 GST_TIME_ARGS (current_time));
3999 } else if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
4000 sess->next_rtcp_check_time > current_time) {
4001 GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
4002 GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
4003 GST_TIME_ARGS (current_time));
4004 return FALSE;
4005 }
4006
4007 /* take interval and add jitter */
4008 interval = data->interval;
4009 if (interval != GST_CLOCK_TIME_NONE)
4010 interval = rtp_stats_add_rtcp_jitter (stats, interval);
4011
4012 if (sess->last_rtcp_check_time != GST_CLOCK_TIME_NONE) {
4013 /* perform forward reconsideration */
4014 if (interval != GST_CLOCK_TIME_NONE) {
4015 GstClockTime elapsed;
4016
4017 /* get elapsed time since we last reported */
4018 elapsed = current_time - sess->last_rtcp_check_time;
4019
4020 GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
4021 GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
4022 new_send_time = interval + sess->last_rtcp_check_time;
4023 } else {
4024 new_send_time = sess->last_rtcp_check_time;
4025 }
4026 } else {
4027 /* If this is the first RTCP packet, we can reconsider anything based
4028 * on the last RTCP send time because there was none.
4029 */
4030 g_warn_if_fail (!data->is_early);
4031 data->is_early = FALSE;
4032 new_send_time = current_time;
4033 }
4034
4035 if (!data->is_early) {
4036 /* check if reconsideration */
4037 if (new_send_time == GST_CLOCK_TIME_NONE || current_time < new_send_time) {
4038 GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
4039 GST_TIME_ARGS (new_send_time));
4040 /* store new check time */
4041 sess->next_rtcp_check_time = new_send_time;
4042 sess->last_rtcp_interval = interval;
4043 return FALSE;
4044 }
4045
4046 sess->last_rtcp_interval = interval;
4047 if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
4048 || sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
4049 && interval != GST_CLOCK_TIME_NONE) {
4050 /* Apply the rules from RFC 4585 section 3.5.3 */
4051 if (stats->min_interval != 0 && !sess->first_rtcp) {
4052 GstClockTime T_rr_current_interval =
4053 g_random_double_range (0.5, 1.5) * stats->min_interval * GST_SECOND;
4054
4055 if (T_rr_current_interval > interval) {
4056 GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
4057 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
4058 GST_TIME_ARGS (interval));
4059 interval = T_rr_current_interval;
4060 }
4061 }
4062 }
4063 sess->next_rtcp_check_time = current_time + interval;
4064 }
4065
4066
4067 GST_DEBUG ("can send RTCP now, next %" GST_TIME_FORMAT,
4068 GST_TIME_ARGS (sess->next_rtcp_check_time));
4069
4070 return TRUE;
4071 }
4072
4073 static void
clone_ssrcs_hashtable(gchar * key,RTPSource * source,GHashTable * hash_table)4074 clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
4075 {
4076 g_hash_table_insert (hash_table, key, g_object_ref (source));
4077 }
4078
4079 static gboolean
remove_closing_sources(const gchar * key,RTPSource * source,ReportData * data)4080 remove_closing_sources (const gchar * key, RTPSource * source,
4081 ReportData * data)
4082 {
4083 if (source->closing)
4084 return TRUE;
4085
4086 if (source->send_fir)
4087 data->have_fir = TRUE;
4088 if (source->send_pli)
4089 data->have_pli = TRUE;
4090 if (source->send_nack)
4091 data->have_nack = TRUE;
4092
4093 return FALSE;
4094 }
4095
4096 static void
generate_rtcp(const gchar * key,RTPSource * source,ReportData * data)4097 generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
4098 {
4099 RTPSession *sess = data->sess;
4100 gboolean is_bye = FALSE;
4101 ReportOutput *output;
4102
4103 /* only generate RTCP for active internal sources */
4104 if (!source->internal || source->sent_bye)
4105 return;
4106
4107 /* ignore other sources when we do the timeout after a scheduled BYE */
4108 if (sess->scheduled_bye && !source->marked_bye)
4109 return;
4110
4111 /* skip if RTCP is disabled */
4112 if (source->disable_rtcp) {
4113 GST_DEBUG ("source %08x has RTCP disabled", source->ssrc);
4114 return;
4115 }
4116
4117 data->source = source;
4118
4119 /* open packet */
4120 session_start_rtcp (sess, data);
4121
4122 if (source->marked_bye) {
4123 /* send BYE */
4124 make_source_bye (sess, source, data);
4125 is_bye = TRUE;
4126 } else if (!data->is_early) {
4127 /* loop over all known sources and add report blocks. If we are early, we
4128 * just make a minimal RTCP packet and skip this step */
4129 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4130 (GHFunc) session_report_blocks, data);
4131 }
4132 if (!data->has_sdes && (!data->is_early || !sess->reduced_size_rtcp))
4133 session_sdes (sess, data);
4134
4135 if (data->have_fir)
4136 session_fir (sess, data);
4137
4138 if (data->have_pli)
4139 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4140 (GHFunc) session_pli, data);
4141
4142 if (data->have_nack)
4143 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4144 (GHFunc) session_nack, data);
4145
4146 gst_rtcp_buffer_unmap (&data->rtcpbuf);
4147
4148 output = g_slice_new (ReportOutput);
4149 output->source = g_object_ref (source);
4150 output->is_bye = is_bye;
4151 output->buffer = data->rtcp;
4152 /* queue the RTCP packet to push later */
4153 g_queue_push_tail (&data->output, output);
4154 }
4155
4156 static void
update_generation(const gchar * key,RTPSource * source,ReportData * data)4157 update_generation (const gchar * key, RTPSource * source, ReportData * data)
4158 {
4159 RTPSession *sess = data->sess;
4160
4161 if (g_hash_table_size (source->reported_in_sr_of) >=
4162 sess->stats.internal_sources) {
4163 /* source is reported, move to next generation */
4164 source->generation = sess->generation + 1;
4165 g_hash_table_remove_all (source->reported_in_sr_of);
4166
4167 GST_LOG ("reported source %x, new generation: %d", source->ssrc,
4168 source->generation);
4169
4170 /* if we reported all sources in this generation, move to next */
4171 if (--data->num_to_report == 0) {
4172 sess->generation++;
4173 GST_DEBUG ("all reported, generation now %u", sess->generation);
4174 }
4175 }
4176 }
4177
4178 static void
schedule_remaining_nacks(const gchar * key,RTPSource * source,ReportData * data)4179 schedule_remaining_nacks (const gchar * key, RTPSource * source,
4180 ReportData * data)
4181 {
4182 RTPSession *sess = data->sess;
4183 GstClockTime *nack_deadlines;
4184 GstClockTime deadline;
4185 guint n_nacks;
4186
4187 if (!source->send_nack)
4188 return;
4189
4190 /* the scheduling is entirely based on available bandwidth, just take the
4191 * biggest seqnum, which will have the largest deadline to request early
4192 * RTCP. */
4193 nack_deadlines = rtp_source_get_nack_deadlines (source, &n_nacks);
4194 deadline = nack_deadlines[n_nacks - 1];
4195 RTP_SESSION_UNLOCK (sess);
4196 rtp_session_send_rtcp_with_deadline (sess, deadline);
4197 RTP_SESSION_LOCK (sess);
4198 }
4199
4200 static gboolean
rtp_session_are_all_sources_bye(RTPSession * sess)4201 rtp_session_are_all_sources_bye (RTPSession * sess)
4202 {
4203 GHashTableIter iter;
4204 RTPSource *src;
4205
4206 RTP_SESSION_LOCK (sess);
4207 g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
4208 while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
4209 if (src->internal && !src->sent_bye) {
4210 RTP_SESSION_UNLOCK (sess);
4211 return FALSE;
4212 }
4213 }
4214 RTP_SESSION_UNLOCK (sess);
4215
4216 return TRUE;
4217 }
4218
4219 /**
4220 * rtp_session_on_timeout:
4221 * @sess: an #RTPSession
4222 * @current_time: the current system time
4223 * @ntpnstime: the current NTP time in nanoseconds
4224 * @running_time: the current running_time of the pipeline
4225 *
4226 * Perform maintenance actions after the timeout obtained with
4227 * rtp_session_next_timeout() expired.
4228 *
4229 * This function will perform timeouts of receivers and senders, send a BYE
4230 * packet or generate RTCP packets with current session stats.
4231 *
4232 * This function can call the #RTPSessionSendRTCP callback, possibly multiple
4233 * times, for each packet that should be processed.
4234 *
4235 * Returns: a #GstFlowReturn.
4236 */
4237 GstFlowReturn
rtp_session_on_timeout(RTPSession * sess,GstClockTime current_time,guint64 ntpnstime,GstClockTime running_time)4238 rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
4239 guint64 ntpnstime, GstClockTime running_time)
4240 {
4241 GstFlowReturn result = GST_FLOW_OK;
4242 ReportData data = { GST_RTCP_BUFFER_INIT };
4243 GHashTable *table_copy;
4244 ReportOutput *output;
4245 gboolean all_empty = FALSE;
4246
4247 g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
4248
4249 GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
4250 ", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4251 GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
4252
4253 data.sess = sess;
4254 data.current_time = current_time;
4255 data.ntpnstime = ntpnstime;
4256 data.running_time = running_time;
4257 data.num_to_report = 0;
4258 data.may_suppress = FALSE;
4259 data.nacked_seqnums = 0;
4260 g_queue_init (&data.output);
4261
4262 RTP_SESSION_LOCK (sess);
4263 /* get a new interval, we need this for various cleanups etc */
4264 data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
4265
4266 GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
4267
4268 /* we need an internal source now */
4269 if (sess->stats.internal_sources == 0) {
4270 RTPSource *source;
4271 gboolean created;
4272
4273 source = obtain_internal_source (sess, sess->suggested_ssrc, &created,
4274 current_time);
4275 sess->internal_ssrc_set = TRUE;
4276
4277 if (created)
4278 on_new_sender_ssrc (sess, source);
4279
4280 g_object_unref (source);
4281 }
4282
4283 sess->conflicting_addresses =
4284 timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
4285
4286 /* Make a local copy of the hashtable. We need to do this because the
4287 * cleanup stage below releases the session lock. */
4288 table_copy = g_hash_table_new_full (NULL, NULL, NULL,
4289 (GDestroyNotify) g_object_unref);
4290 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4291 (GHFunc) clone_ssrcs_hashtable, table_copy);
4292
4293 /* Clean up the session, mark the source for removing, this might release the
4294 * session lock. */
4295 g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
4296 g_hash_table_destroy (table_copy);
4297
4298 /* Now remove the marked sources */
4299 g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
4300 (GHRFunc) remove_closing_sources, &data);
4301
4302 /* update point-to-point status */
4303 session_update_ptp (sess);
4304
4305 /* see if we need to generate SR or RR packets */
4306 if (!is_rtcp_time (sess, current_time, &data))
4307 goto done;
4308
4309 /* check if all the buffers are empty afer generation */
4310 all_empty = TRUE;
4311
4312 GST_DEBUG
4313 ("doing RTCP generation %u for %u sources, early %d, may suppress %d",
4314 sess->generation, data.num_to_report, data.is_early, data.may_suppress);
4315
4316 /* generate RTCP for all internal sources */
4317 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4318 (GHFunc) generate_rtcp, &data);
4319
4320 /* update the generation for all the sources that have been reported */
4321 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4322 (GHFunc) update_generation, &data);
4323
4324 /* we keep track of the last report time in order to timeout inactive
4325 * receivers or senders */
4326 if (!data.is_early) {
4327 GST_DEBUG ("Time since last regular RTCP: %" GST_TIME_FORMAT " - %"
4328 GST_TIME_FORMAT " = %" GST_TIME_FORMAT,
4329 GST_TIME_ARGS (data.current_time),
4330 GST_TIME_ARGS (sess->last_rtcp_send_time),
4331 GST_TIME_ARGS (data.current_time - sess->last_rtcp_send_time));
4332 sess->last_rtcp_send_time = data.current_time;
4333 }
4334
4335 GST_DEBUG ("Time since last RTCP: %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT
4336 " = %" GST_TIME_FORMAT, GST_TIME_ARGS (data.current_time),
4337 GST_TIME_ARGS (sess->last_rtcp_check_time),
4338 GST_TIME_ARGS (data.current_time - sess->last_rtcp_check_time));
4339 sess->last_rtcp_check_time = data.current_time;
4340 sess->first_rtcp = FALSE;
4341 sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
4342 sess->scheduled_bye = FALSE;
4343
4344 done:
4345 RTP_SESSION_UNLOCK (sess);
4346
4347 /* notify about updated statistics */
4348 g_object_notify (G_OBJECT (sess), "stats");
4349
4350 /* push out the RTCP packets */
4351 while ((output = g_queue_pop_head (&data.output))) {
4352 gboolean do_not_suppress, empty_buffer;
4353 GstBuffer *buffer = output->buffer;
4354 RTPSource *source = output->source;
4355
4356 /* Give the user a change to add its own packet */
4357 g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
4358 buffer, data.is_early, &do_not_suppress);
4359
4360 empty_buffer = gst_buffer_get_size (buffer) == 0;
4361
4362 if (!empty_buffer)
4363 all_empty = FALSE;
4364
4365 if (sess->callbacks.send_rtcp &&
4366 !empty_buffer && (do_not_suppress || !data.may_suppress)) {
4367 guint packet_size;
4368
4369 packet_size = gst_buffer_get_size (buffer) + sess->header_len;
4370
4371 UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
4372 GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
4373 sess->stats.avg_rtcp_packet_size, packet_size);
4374 result =
4375 sess->callbacks.send_rtcp (sess, source, buffer,
4376 rtp_session_are_all_sources_bye (sess), sess->send_rtcp_user_data);
4377
4378 RTP_SESSION_LOCK (sess);
4379 sess->stats.nacks_sent += data.nacked_seqnums;
4380 on_sender_ssrc_active (sess, source);
4381 RTP_SESSION_UNLOCK (sess);
4382 } else {
4383 GST_DEBUG ("freeing packet callback: %p"
4384 " empty_buffer: %d, "
4385 " do_not_suppress: %d may_suppress: %d", sess->callbacks.send_rtcp,
4386 empty_buffer, do_not_suppress, data.may_suppress);
4387 if (!empty_buffer) {
4388 RTP_SESSION_LOCK (sess);
4389 sess->stats.nacks_dropped += data.nacked_seqnums;
4390 RTP_SESSION_UNLOCK (sess);
4391 }
4392 gst_buffer_unref (buffer);
4393 }
4394 g_object_unref (source);
4395 g_slice_free (ReportOutput, output);
4396 }
4397
4398 if (all_empty)
4399 GST_ERROR ("generated empty RTCP messages for all the sources");
4400
4401 /* schedule remaining nacks */
4402 RTP_SESSION_LOCK (sess);
4403 g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
4404 (GHFunc) schedule_remaining_nacks, &data);
4405 RTP_SESSION_UNLOCK (sess);
4406
4407 return result;
4408 }
4409
4410 /**
4411 * rtp_session_request_early_rtcp:
4412 * @sess: an #RTPSession
4413 * @current_time: the current system time
4414 * @max_delay: maximum delay
4415 *
4416 * Request transmission of early RTCP
4417 *
4418 * Returns: %TRUE if the related RTCP can be scheduled.
4419 */
4420 gboolean
rtp_session_request_early_rtcp(RTPSession * sess,GstClockTime current_time,GstClockTime max_delay)4421 rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
4422 GstClockTime max_delay)
4423 {
4424 GstClockTime T_dither_max, T_rr, offset = 0;
4425 gboolean ret;
4426 gboolean allow_early;
4427
4428 /* Implements the algorithm described in RFC 4585 section 3.5.2 */
4429
4430 RTP_SESSION_LOCK (sess);
4431
4432 /* We assume a feedback profile if something is requesting RTCP
4433 * to be sent */
4434 sess->rtp_profile = GST_RTP_PROFILE_AVPF;
4435
4436 /* Check if already requested */
4437 /* RFC 4585 section 3.5.2 step 2 */
4438 if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
4439 GST_LOG_OBJECT (sess, "already have next early rtcp time");
4440 ret = (current_time + max_delay > sess->next_early_rtcp_time);
4441 goto end;
4442 }
4443
4444 if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
4445 GST_LOG_OBJECT (sess, "no next RTCP check time");
4446 ret = FALSE;
4447 goto end;
4448 }
4449
4450 /* RFC 4585 section 3.5.3 step 1
4451 * If no regular RTCP packet has been sent before, then a regular
4452 * RTCP packet has to be scheduled first and FB messages might be
4453 * included there
4454 */
4455 if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
4456 GST_LOG_OBJECT (sess, "no RTCP sent yet");
4457
4458 if (current_time + max_delay > sess->next_rtcp_check_time) {
4459 GST_LOG_OBJECT (sess,
4460 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4461 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4462 GST_TIME_ARGS (max_delay),
4463 GST_TIME_ARGS (sess->next_rtcp_check_time));
4464 ret = TRUE;
4465 } else {
4466 GST_LOG_OBJECT (sess,
4467 "can't allow early feedback, next scheduled time is too late %"
4468 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4469 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4470 GST_TIME_ARGS (sess->next_rtcp_check_time));
4471 ret = FALSE;
4472 }
4473 goto end;
4474 }
4475
4476 T_rr = sess->last_rtcp_interval;
4477
4478 /* RFC 4585 section 3.5.2 step 2b */
4479 /* If the total sources is <=2, then there is only us and one peer */
4480 /* When there is one auxiliary stream the session can still do point
4481 * to point.
4482 */
4483 if (sess->is_doing_ptp) {
4484 T_dither_max = 0;
4485 } else {
4486 /* Divide by 2 because l = 0.5 */
4487 T_dither_max = T_rr;
4488 T_dither_max /= 2;
4489 }
4490
4491 /* RFC 4585 section 3.5.2 step 3 */
4492 if (current_time + T_dither_max > sess->next_rtcp_check_time) {
4493 GST_LOG_OBJECT (sess,
4494 "don't send because of dither, next scheduled time is too soon %"
4495 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
4496 GST_TIME_ARGS (current_time), GST_TIME_ARGS (T_dither_max),
4497 GST_TIME_ARGS (sess->next_rtcp_check_time));
4498 ret = T_dither_max <= max_delay;
4499 goto end;
4500 }
4501
4502 /* RFC 4585 section 3.5.2 step 4a and
4503 * RFC 4585 section 3.5.2 step 6 */
4504 allow_early = FALSE;
4505 if (sess->last_rtcp_check_time == sess->last_rtcp_send_time) {
4506 /* Last time we sent a full RTCP packet, we can now immediately
4507 * send an early one as allow_early was reset to TRUE */
4508 allow_early = TRUE;
4509 } else if (sess->last_rtcp_check_time + T_rr <= current_time + max_delay) {
4510 /* Last packet we sent was an early RTCP packet and more than
4511 * T_rr has passed since then, meaning we would have suppressed
4512 * a regular RTCP packet already and reset allow_early to TRUE */
4513 allow_early = TRUE;
4514
4515 /* We have to offset a bit as T_rr has not passed yet, but will before
4516 * max_delay */
4517 if (sess->last_rtcp_check_time + T_rr > current_time)
4518 offset = (sess->last_rtcp_check_time + T_rr) - current_time;
4519 } else {
4520 GST_DEBUG_OBJECT (sess,
4521 "can't allow early RTCP yet: last regular %" GST_TIME_FORMAT ", %"
4522 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT " + %"
4523 GST_TIME_FORMAT, GST_TIME_ARGS (sess->last_rtcp_send_time),
4524 GST_TIME_ARGS (sess->last_rtcp_check_time), GST_TIME_ARGS (T_rr),
4525 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay));
4526 }
4527
4528 if (!allow_early) {
4529 /* Ignore the request a scheduled packet will be in time anyway */
4530 if (current_time + max_delay > sess->next_rtcp_check_time) {
4531 GST_LOG_OBJECT (sess,
4532 "next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
4533 " > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
4534 GST_TIME_ARGS (max_delay),
4535 GST_TIME_ARGS (sess->next_rtcp_check_time));
4536 ret = TRUE;
4537 } else {
4538 GST_LOG_OBJECT (sess,
4539 "can't allow early feedback and next scheduled time is too late %"
4540 GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
4541 GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
4542 GST_TIME_ARGS (sess->next_rtcp_check_time));
4543 ret = FALSE;
4544 }
4545 goto end;
4546 }
4547
4548 /* RFC 4585 section 3.5.2 step 4b */
4549 if (T_dither_max) {
4550 /* Schedule an early transmission later */
4551 sess->next_early_rtcp_time = g_random_double () * T_dither_max +
4552 current_time + offset;
4553 } else {
4554 /* If no dithering, schedule it for NOW */
4555 sess->next_early_rtcp_time = current_time + offset;
4556 }
4557
4558 GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT
4559 ", next regular RTCP time %" GST_TIME_FORMAT,
4560 GST_TIME_ARGS (sess->next_early_rtcp_time),
4561 GST_TIME_ARGS (sess->next_rtcp_check_time));
4562 RTP_SESSION_UNLOCK (sess);
4563
4564 /* notify app of need to send packet early
4565 * and therefore of timeout change */
4566 if (sess->callbacks.reconsider)
4567 sess->callbacks.reconsider (sess, sess->reconsider_user_data);
4568
4569 return TRUE;
4570
4571 end:
4572
4573 RTP_SESSION_UNLOCK (sess);
4574
4575 return ret;
4576 }
4577
4578 static gboolean
rtp_session_send_rtcp_internal(RTPSession * sess,GstClockTime now,GstClockTime max_delay)4579 rtp_session_send_rtcp_internal (RTPSession * sess, GstClockTime now,
4580 GstClockTime max_delay)
4581 {
4582 /* notify the application that we intend to send early RTCP */
4583 if (sess->callbacks.notify_early_rtcp)
4584 sess->callbacks.notify_early_rtcp (sess, sess->notify_early_rtcp_user_data);
4585
4586 return rtp_session_request_early_rtcp (sess, now, max_delay);
4587 }
4588
4589 static gboolean
rtp_session_send_rtcp_with_deadline(RTPSession * sess,GstClockTime deadline)4590 rtp_session_send_rtcp_with_deadline (RTPSession * sess, GstClockTime deadline)
4591 {
4592 GstClockTime now, max_delay;
4593
4594 if (!sess->callbacks.send_rtcp)
4595 return FALSE;
4596
4597 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
4598
4599 if (deadline < now)
4600 return FALSE;
4601
4602 max_delay = deadline - now;
4603
4604 return rtp_session_send_rtcp_internal (sess, now, max_delay);
4605 }
4606
4607 static gboolean
rtp_session_send_rtcp(RTPSession * sess,GstClockTime max_delay)4608 rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
4609 {
4610 GstClockTime now;
4611
4612 if (!sess->callbacks.send_rtcp)
4613 return FALSE;
4614
4615 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
4616
4617 return rtp_session_send_rtcp_internal (sess, now, max_delay);
4618 }
4619
4620 gboolean
rtp_session_request_key_unit(RTPSession * sess,guint32 ssrc,gboolean fir,gint count)4621 rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
4622 gboolean fir, gint count)
4623 {
4624 RTPSource *src;
4625
4626 RTP_SESSION_LOCK (sess);
4627 src = find_source (sess, ssrc);
4628 if (src == NULL)
4629 goto no_source;
4630
4631 if (fir) {
4632 src->send_pli = FALSE;
4633 src->send_fir = TRUE;
4634
4635 if (count == -1 || count != src->last_fir_count)
4636 src->current_send_fir_seqnum++;
4637 src->last_fir_count = count;
4638 } else if (!src->send_fir) {
4639 src->send_pli = TRUE;
4640 }
4641 RTP_SESSION_UNLOCK (sess);
4642
4643 if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
4644 GST_DEBUG ("FIR/PLI not sent early, sending with next regular RTCP");
4645 }
4646
4647 return TRUE;
4648
4649 /* ERRORS */
4650 no_source:
4651 {
4652 RTP_SESSION_UNLOCK (sess);
4653 return FALSE;
4654 }
4655 }
4656
4657 /**
4658 * rtp_session_request_nack:
4659 * @sess: a #RTPSession
4660 * @ssrc: the SSRC
4661 * @seqnum: the missing seqnum
4662 * @max_delay: max delay to request NACK
4663 *
4664 * Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
4665 *
4666 * Returns: %TRUE if the NACK feedback could be scheduled
4667 */
4668 gboolean
rtp_session_request_nack(RTPSession * sess,guint32 ssrc,guint16 seqnum,GstClockTime max_delay)4669 rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
4670 GstClockTime max_delay)
4671 {
4672 RTPSource *source;
4673 GstClockTime now;
4674
4675 if (!sess->callbacks.send_rtcp)
4676 return FALSE;
4677
4678 now = sess->callbacks.request_time (sess, sess->request_time_user_data);
4679
4680 RTP_SESSION_LOCK (sess);
4681 source = find_source (sess, ssrc);
4682 if (source == NULL)
4683 goto no_source;
4684
4685 GST_DEBUG ("request NACK for SSRC %08x, #%u, deadline %" GST_TIME_FORMAT,
4686 ssrc, seqnum, GST_TIME_ARGS (now + max_delay));
4687 rtp_source_register_nack (source, seqnum, now + max_delay);
4688 RTP_SESSION_UNLOCK (sess);
4689
4690 if (!rtp_session_send_rtcp_internal (sess, now, max_delay)) {
4691 GST_DEBUG ("NACK not sent early, sending with next regular RTCP");
4692 }
4693
4694 return TRUE;
4695
4696 /* ERRORS */
4697 no_source:
4698 {
4699 RTP_SESSION_UNLOCK (sess);
4700 return FALSE;
4701 }
4702 }
4703