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 /**
21 * SECTION:element-rtpsession
22 * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
23 *
24 * The RTP session manager models participants with unique SSRC in an RTP
25 * session. This session can be used to send and receive RTP and RTCP packets.
26 * Based on what REQUEST pads are requested from the session manager, specific
27 * functionality can be activated.
28 *
29 * The session manager currently implements RFC 3550 including:
30 * <itemizedlist>
31 * <listitem>
32 * <para>RTP packet validation based on consecutive sequence numbers.</para>
33 * </listitem>
34 * <listitem>
35 * <para>Maintainance of the SSRC participant database.</para>
36 * </listitem>
37 * <listitem>
38 * <para>Keeping per participant statistics based on received RTCP packets.</para>
39 * </listitem>
40 * <listitem>
41 * <para>Scheduling of RR/SR RTCP packets.</para>
42 * </listitem>
43 * <listitem>
44 * <para>Support for multiple sender SSRC.</para>
45 * </listitem>
46 * </itemizedlist>
47 *
48 * The rtpsession will not demux packets based on SSRC or payload type, nor will
49 * it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
50 * #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
51 * perform these tasks. It is usually a good idea to use #GstRtpBin, which
52 * combines all these features in one element.
53 *
54 * To use #GstRtpSession as an RTP receiver, request a recv_rtp_sink pad, which will
55 * automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
56 * will be processed in the session and after being validated forwarded on the
57 * recv_rtp_src pad.
58 *
59 * To also use #GstRtpSession as an RTCP receiver, request a recv_rtcp_sink pad,
60 * which will automatically create a sync_src pad. Packets received on the RTCP
61 * pad will be used by the session manager to update the stats and database of
62 * the other participants. SR packets will be forwarded on the sync_src pad
63 * so that they can be used to perform inter-stream synchronisation when needed.
64 *
65 * If you want the session manager to generate and send RTCP packets, request
66 * the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
67 * that should be sent to all participants in the session.
68 *
69 * To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will
70 * automatically create a send_rtp_src pad. The session manager will
71 * forward the packets on the send_rtp_src pad after updating its internal state.
72 *
73 * The session manager needs the clock-rate of the payload types it is handling
74 * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
75 * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
76 * signal.
77 *
78 * <refsect2>
79 * <title>Example pipelines</title>
80 * |[
81 * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
82 * ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
83 * decoder and display. Note that the application/x-rtp caps on udpsrc should be
84 * configured based on some negotiation process such as RTSP for this pipeline
85 * to work correctly.
86 * |[
87 * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \
88 * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
89 * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
90 * ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
91 * decoder and display. Receive RTCP packets from port 5001 and process them in
92 * the session manager.
93 * Note that the application/x-rtp caps on udpsrc should be
94 * configured based on some negotiation process such as RTSP for this pipeline
95 * to work correctly.
96 * |[
97 * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000
98 * ]| Send theora RTP packets through the session manager and out on UDP port
99 * 5000.
100 * |[
101 * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \
102 * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
103 * ]| Send theora RTP packets through the session manager and out on UDP port
104 * 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
105 * correctly because the second udpsink will not preroll correctly (no RTCP
106 * packets are sent in the PAUSED state). Applications should manually set and
107 * keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
108 * </refsect2>
109 */
110
111 #ifdef HAVE_CONFIG_H
112 #include "config.h"
113 #endif
114
115 #include <gst/rtp/gstrtpbuffer.h>
116
117 #include <gst/glib-compat-private.h>
118
119 #include "gstrtpsession.h"
120 #include "rtpsession.h"
121
122 GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
123 #define GST_CAT_DEFAULT gst_rtp_session_debug
124
125 GType
gst_rtp_ntp_time_source_get_type(void)126 gst_rtp_ntp_time_source_get_type (void)
127 {
128 static GType type = 0;
129 static const GEnumValue values[] = {
130 {GST_RTP_NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
131 {GST_RTP_NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
132 {GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME,
133 "Running time based on pipeline clock",
134 "running-time"},
135 {GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
136 {0, NULL, NULL},
137 };
138
139 if (!type) {
140 type = g_enum_register_static ("GstRtpNtpTimeSource", values);
141 }
142 return type;
143 }
144
145 /* sink pads */
146 static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
147 GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
148 GST_PAD_SINK,
149 GST_PAD_REQUEST,
150 GST_STATIC_CAPS ("application/x-rtp")
151 );
152
153 static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
154 GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
155 GST_PAD_SINK,
156 GST_PAD_REQUEST,
157 GST_STATIC_CAPS ("application/x-rtcp")
158 );
159
160 static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
161 GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
162 GST_PAD_SINK,
163 GST_PAD_REQUEST,
164 GST_STATIC_CAPS ("application/x-rtp")
165 );
166
167 /* src pads */
168 static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
169 GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
170 GST_PAD_SRC,
171 GST_PAD_SOMETIMES,
172 GST_STATIC_CAPS ("application/x-rtp")
173 );
174
175 static GstStaticPadTemplate rtpsession_sync_src_template =
176 GST_STATIC_PAD_TEMPLATE ("sync_src",
177 GST_PAD_SRC,
178 GST_PAD_SOMETIMES,
179 GST_STATIC_CAPS ("application/x-rtcp")
180 );
181
182 static GstStaticPadTemplate rtpsession_send_rtp_src_template =
183 GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
184 GST_PAD_SRC,
185 GST_PAD_SOMETIMES,
186 GST_STATIC_CAPS ("application/x-rtp")
187 );
188
189 static GstStaticPadTemplate rtpsession_send_rtcp_src_template =
190 GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
191 GST_PAD_SRC,
192 GST_PAD_REQUEST,
193 GST_STATIC_CAPS ("application/x-rtcp")
194 );
195
196 /* signals and args */
197 enum
198 {
199 SIGNAL_REQUEST_PT_MAP,
200 SIGNAL_CLEAR_PT_MAP,
201
202 SIGNAL_ON_NEW_SSRC,
203 SIGNAL_ON_SSRC_COLLISION,
204 SIGNAL_ON_SSRC_VALIDATED,
205 SIGNAL_ON_SSRC_ACTIVE,
206 SIGNAL_ON_SSRC_SDES,
207 SIGNAL_ON_BYE_SSRC,
208 SIGNAL_ON_BYE_TIMEOUT,
209 SIGNAL_ON_TIMEOUT,
210 SIGNAL_ON_SENDER_TIMEOUT,
211 SIGNAL_ON_NEW_SENDER_SSRC,
212 SIGNAL_ON_SENDER_SSRC_ACTIVE,
213 LAST_SIGNAL
214 };
215
216 #define DEFAULT_BANDWIDTH 0
217 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
218 #define DEFAULT_RTCP_RR_BANDWIDTH -1
219 #define DEFAULT_RTCP_RS_BANDWIDTH -1
220 #define DEFAULT_SDES NULL
221 #define DEFAULT_NUM_SOURCES 0
222 #define DEFAULT_NUM_ACTIVE_SOURCES 0
223 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
224 #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
225 #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
226 #define DEFAULT_MAX_DROPOUT_TIME 60000
227 #define DEFAULT_MAX_MISORDER_TIME 2000
228 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
229 #define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
230 #define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
231
232 enum
233 {
234 PROP_0,
235 PROP_BANDWIDTH,
236 PROP_RTCP_FRACTION,
237 PROP_RTCP_RR_BANDWIDTH,
238 PROP_RTCP_RS_BANDWIDTH,
239 PROP_SDES,
240 PROP_NUM_SOURCES,
241 PROP_NUM_ACTIVE_SOURCES,
242 PROP_INTERNAL_SESSION,
243 PROP_USE_PIPELINE_CLOCK,
244 PROP_RTCP_MIN_INTERVAL,
245 PROP_PROBATION,
246 PROP_MAX_DROPOUT_TIME,
247 PROP_MAX_MISORDER_TIME,
248 PROP_STATS,
249 PROP_RTP_PROFILE,
250 PROP_NTP_TIME_SOURCE,
251 PROP_RTCP_SYNC_SEND_TIME
252 };
253
254 #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->priv->lock)
255 #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->priv->lock)
256
257 #define GST_RTP_SESSION_WAIT(sess) g_cond_wait (&(sess)->priv->cond, &(sess)->priv->lock)
258 #define GST_RTP_SESSION_SIGNAL(sess) g_cond_signal (&(sess)->priv->cond)
259
260 struct _GstRtpSessionPrivate
261 {
262 GMutex lock;
263 GCond cond;
264 GstClock *sysclock;
265
266 RTPSession *session;
267
268 /* thread for sending out RTCP */
269 GstClockID id;
270 gboolean stop_thread;
271 GThread *thread;
272 gboolean thread_stopped;
273 gboolean wait_send;
274
275 /* caps mapping */
276 GHashTable *ptmap;
277
278 GstClockTime send_latency;
279
280 gboolean use_pipeline_clock;
281 GstRtpNtpTimeSource ntp_time_source;
282 gboolean rtcp_sync_send_time;
283
284 guint recv_rtx_req_count;
285 guint sent_rtx_req_count;
286 };
287
288 /* callbacks to handle actions from the session manager */
289 static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
290 RTPSource * src, GstBuffer * buffer, gpointer user_data);
291 static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
292 RTPSource * src, gpointer data, gpointer user_data);
293 static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
294 RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
295 static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
296 GstBuffer * buffer, gpointer user_data);
297 static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
298 gpointer user_data);
299 static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
300 static void gst_rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
301 gboolean all_headers, gpointer user_data);
302 static GstClockTime gst_rtp_session_request_time (RTPSession * session,
303 gpointer user_data);
304 static void gst_rtp_session_notify_nack (RTPSession * sess,
305 guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data);
306 static void gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data);
307 static void gst_rtp_session_notify_early_rtcp (RTPSession * sess,
308 gpointer user_data);
309 static GstFlowReturn gst_rtp_session_chain_recv_rtp (GstPad * pad,
310 GstObject * parent, GstBuffer * buffer);
311 static GstFlowReturn gst_rtp_session_chain_recv_rtcp (GstPad * pad,
312 GstObject * parent, GstBuffer * buffer);
313 static GstFlowReturn gst_rtp_session_chain_send_rtp (GstPad * pad,
314 GstObject * parent, GstBuffer * buffer);
315 static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad,
316 GstObject * parent, GstBufferList * list);
317
318 static RTPSessionCallbacks callbacks = {
319 gst_rtp_session_process_rtp,
320 gst_rtp_session_send_rtp,
321 gst_rtp_session_sync_rtcp,
322 gst_rtp_session_send_rtcp,
323 gst_rtp_session_clock_rate,
324 gst_rtp_session_reconsider,
325 gst_rtp_session_request_key_unit,
326 gst_rtp_session_request_time,
327 gst_rtp_session_notify_nack,
328 gst_rtp_session_reconfigure,
329 gst_rtp_session_notify_early_rtcp
330 };
331
332 /* GObject vmethods */
333 static void gst_rtp_session_finalize (GObject * object);
334 static void gst_rtp_session_set_property (GObject * object, guint prop_id,
335 const GValue * value, GParamSpec * pspec);
336 static void gst_rtp_session_get_property (GObject * object, guint prop_id,
337 GValue * value, GParamSpec * pspec);
338
339 /* GstElement vmethods */
340 static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
341 GstStateChange transition);
342 static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
343 GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
344 static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
345
346 static gboolean gst_rtp_session_sink_setcaps (GstPad * pad,
347 GstRtpSession * rtpsession, GstCaps * caps);
348 static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad,
349 GstRtpSession * rtpsession, GstCaps * caps);
350
351 static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
352
353 static GstStructure *gst_rtp_session_create_stats (GstRtpSession * rtpsession);
354
355 static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
356
357 static void
on_new_ssrc(RTPSession * session,RTPSource * src,GstRtpSession * sess)358 on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
359 {
360 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0,
361 src->ssrc);
362 }
363
364 static void
on_ssrc_collision(RTPSession * session,RTPSource * src,GstRtpSession * sess)365 on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess)
366 {
367 GstPad *send_rtp_sink;
368
369 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
370 src->ssrc);
371
372 GST_RTP_SESSION_LOCK (sess);
373 if ((send_rtp_sink = sess->send_rtp_sink))
374 gst_object_ref (send_rtp_sink);
375 GST_RTP_SESSION_UNLOCK (sess);
376
377 if (send_rtp_sink) {
378 GstStructure *structure;
379 GstEvent *event;
380 RTPSource *internal_src;
381 guint32 suggested_ssrc;
382
383 structure = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT,
384 (guint) src->ssrc, NULL);
385
386 /* if there is no source using the suggested ssrc, most probably because
387 * this ssrc has just collided, suggest upstream to use it */
388 suggested_ssrc = rtp_session_suggest_ssrc (session, NULL);
389 internal_src = rtp_session_get_source_by_ssrc (session, suggested_ssrc);
390 if (!internal_src)
391 gst_structure_set (structure, "suggested-ssrc", G_TYPE_UINT,
392 (guint) suggested_ssrc, NULL);
393 else
394 g_object_unref (internal_src);
395
396 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
397 gst_pad_push_event (send_rtp_sink, event);
398 gst_object_unref (send_rtp_sink);
399 }
400 }
401
402 static void
on_ssrc_validated(RTPSession * session,RTPSource * src,GstRtpSession * sess)403 on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess)
404 {
405 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
406 src->ssrc);
407 }
408
409 static void
on_ssrc_active(RTPSession * session,RTPSource * src,GstRtpSession * sess)410 on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
411 {
412 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
413 src->ssrc);
414 }
415
416 static void
on_ssrc_sdes(RTPSession * session,RTPSource * src,GstRtpSession * sess)417 on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess)
418 {
419 GstStructure *s;
420 GstMessage *m;
421
422 /* convert the new SDES info into a message */
423 RTP_SESSION_LOCK (session);
424 g_object_get (src, "sdes", &s, NULL);
425 RTP_SESSION_UNLOCK (session);
426
427 m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s);
428 gst_element_post_message (GST_ELEMENT_CAST (sess), m);
429
430 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0,
431 src->ssrc);
432 }
433
434 static void
on_bye_ssrc(RTPSession * session,RTPSource * src,GstRtpSession * sess)435 on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
436 {
437 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0,
438 src->ssrc);
439 }
440
441 static void
on_bye_timeout(RTPSession * session,RTPSource * src,GstRtpSession * sess)442 on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
443 {
444 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
445 src->ssrc);
446 }
447
448 static void
on_timeout(RTPSession * session,RTPSource * src,GstRtpSession * sess)449 on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
450 {
451 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0,
452 src->ssrc);
453 }
454
455 static void
on_sender_timeout(RTPSession * session,RTPSource * src,GstRtpSession * sess)456 on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
457 {
458 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
459 src->ssrc);
460 }
461
462 static void
on_new_sender_ssrc(RTPSession * session,RTPSource * src,GstRtpSession * sess)463 on_new_sender_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
464 {
465 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
466 src->ssrc);
467 }
468
469 static void
on_sender_ssrc_active(RTPSession * session,RTPSource * src,GstRtpSession * sess)470 on_sender_ssrc_active (RTPSession * session, RTPSource * src,
471 GstRtpSession * sess)
472 {
473 g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
474 src->ssrc);
475 }
476
477 static void
on_notify_stats(RTPSession * session,GParamSpec * spec,GstRtpSession * rtpsession)478 on_notify_stats (RTPSession * session, GParamSpec * spec,
479 GstRtpSession * rtpsession)
480 {
481 g_object_notify (G_OBJECT (rtpsession), "stats");
482 }
483
484 #define gst_rtp_session_parent_class parent_class
485 G_DEFINE_TYPE_WITH_PRIVATE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
486
487 static void
gst_rtp_session_class_init(GstRtpSessionClass * klass)488 gst_rtp_session_class_init (GstRtpSessionClass * klass)
489 {
490 GObjectClass *gobject_class;
491 GstElementClass *gstelement_class;
492
493 gobject_class = (GObjectClass *) klass;
494 gstelement_class = (GstElementClass *) klass;
495
496 gobject_class->finalize = gst_rtp_session_finalize;
497 gobject_class->set_property = gst_rtp_session_set_property;
498 gobject_class->get_property = gst_rtp_session_get_property;
499
500 /**
501 * GstRtpSession::request-pt-map:
502 * @sess: the object which received the signal
503 * @pt: the pt
504 *
505 * Request the payload type as #GstCaps for @pt.
506 */
507 gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] =
508 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
509 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, request_pt_map),
510 NULL, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 1, G_TYPE_UINT);
511 /**
512 * GstRtpSession::clear-pt-map:
513 * @sess: the object which received the signal
514 *
515 * Clear the cached pt-maps requested with #GstRtpSession::request-pt-map.
516 */
517 gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
518 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
519 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
520 G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
521 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 0, G_TYPE_NONE);
522
523 /**
524 * GstRtpSession::on-new-ssrc:
525 * @sess: the object which received the signal
526 * @ssrc: the SSRC
527 *
528 * Notify of a new SSRC that entered @session.
529 */
530 gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
531 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
532 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
533 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
534 /**
535 * GstRtpSession::on-ssrc_collision:
536 * @sess: the object which received the signal
537 * @ssrc: the SSRC
538 *
539 * Notify when we have an SSRC collision
540 */
541 gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
542 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
543 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
544 on_ssrc_collision), NULL, NULL, g_cclosure_marshal_VOID__UINT,
545 G_TYPE_NONE, 1, G_TYPE_UINT);
546 /**
547 * GstRtpSession::on-ssrc_validated:
548 * @sess: the object which received the signal
549 * @ssrc: the SSRC
550 *
551 * Notify of a new SSRC that became validated.
552 */
553 gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
554 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
555 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
556 on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__UINT,
557 G_TYPE_NONE, 1, G_TYPE_UINT);
558 /**
559 * GstRtpSession::on-ssrc-active:
560 * @sess: the object which received the signal
561 * @ssrc: the SSRC
562 *
563 * Notify of a SSRC that is active, i.e., sending RTCP.
564 */
565 gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
566 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
567 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
568 on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT,
569 G_TYPE_NONE, 1, G_TYPE_UINT);
570 /**
571 * GstRtpSession::on-ssrc-sdes:
572 * @session: the object which received the signal
573 * @src: the SSRC
574 *
575 * Notify that a new SDES was received for SSRC.
576 */
577 gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
578 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
579 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_sdes),
580 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
581
582 /**
583 * GstRtpSession::on-bye-ssrc:
584 * @sess: the object which received the signal
585 * @ssrc: the SSRC
586 *
587 * Notify of an SSRC that became inactive because of a BYE packet.
588 */
589 gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
590 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
591 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_ssrc),
592 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
593 /**
594 * GstRtpSession::on-bye-timeout:
595 * @sess: the object which received the signal
596 * @ssrc: the SSRC
597 *
598 * Notify of an SSRC that has timed out because of BYE
599 */
600 gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
601 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
602 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_timeout),
603 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
604 /**
605 * GstRtpSession::on-timeout:
606 * @sess: the object which received the signal
607 * @ssrc: the SSRC
608 *
609 * Notify of an SSRC that has timed out
610 */
611 gst_rtp_session_signals[SIGNAL_ON_TIMEOUT] =
612 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
613 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout),
614 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
615 /**
616 * GstRtpSession::on-sender-timeout:
617 * @sess: the object which received the signal
618 * @ssrc: the SSRC
619 *
620 * Notify of a sender SSRC that has timed out and became a receiver
621 */
622 gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
623 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
624 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
625 on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
626 G_TYPE_NONE, 1, G_TYPE_UINT);
627
628 /**
629 * GstRtpSession::on-new-sender-ssrc:
630 * @sess: the object which received the signal
631 * @ssrc: the sender SSRC
632 *
633 * Notify of a new sender SSRC that entered @session.
634 *
635 * Since: 1.8
636 */
637 gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
638 g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
639 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
640 NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
641
642 /**
643 * GstRtpSession::on-sender-ssrc-active:
644 * @sess: the object which received the signal
645 * @ssrc: the sender SSRC
646 *
647 * Notify of a sender SSRC that is active, i.e., sending RTCP.
648 *
649 * Since: 1.8
650 */
651 gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
652 g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
653 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
654 on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT,
655 G_TYPE_NONE, 1, G_TYPE_UINT);
656
657 g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
658 g_param_spec_double ("bandwidth", "Bandwidth",
659 "The bandwidth of the session in bytes per second (0 for auto-discover)",
660 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
661 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
662
663 g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
664 g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
665 "The RTCP bandwidth of the session in bytes per second "
666 "(or as a real fraction of the RTP bandwidth if < 1.0)",
667 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
668 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
669
670 g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
671 g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
672 "The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
673 -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
674 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
675
676 g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
677 g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
678 "The RTCP bandwidth used for senders in bytes per second (-1 = default)",
679 -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
680 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
681
682 g_object_class_install_property (gobject_class, PROP_SDES,
683 g_param_spec_boxed ("sdes", "SDES",
684 "The SDES items of this session",
685 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
686
687 g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
688 g_param_spec_uint ("num-sources", "Num Sources",
689 "The number of sources in the session", 0, G_MAXUINT,
690 DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
691
692 g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
693 g_param_spec_uint ("num-active-sources", "Num Active Sources",
694 "The number of active sources in the session", 0, G_MAXUINT,
695 DEFAULT_NUM_ACTIVE_SOURCES,
696 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
697
698 g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION,
699 g_param_spec_object ("internal-session", "Internal Session",
700 "The internal RTPSession object", RTP_TYPE_SESSION,
701 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
702
703 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
704 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
705 "Use the pipeline running-time to set the NTP time in the RTCP SR messages "
706 "(DEPRECATED: Use ntp-time-source property)",
707 DEFAULT_USE_PIPELINE_CLOCK,
708 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
709
710 g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
711 g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
712 "Minimum interval between Regular RTCP packet (in ns)",
713 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
714 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
715
716 g_object_class_install_property (gobject_class, PROP_PROBATION,
717 g_param_spec_uint ("probation", "Number of probations",
718 "Consecutive packet sequence numbers to accept the source",
719 0, G_MAXUINT, DEFAULT_PROBATION,
720 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
721
722 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
723 g_param_spec_uint ("max-dropout-time", "Max dropout time",
724 "The maximum time (milliseconds) of missing packets tolerated.",
725 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
726 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
727
728 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
729 g_param_spec_uint ("max-misorder-time", "Max misorder time",
730 "The maximum time (milliseconds) of misordered packets tolerated.",
731 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
732 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
733
734 /**
735 * GstRtpSession::stats:
736 *
737 * Various session statistics. This property returns a GstStructure
738 * with name application/x-rtp-session-stats with the following fields:
739 *
740 * "recv-rtx-req-count G_TYPE_UINT The number of retransmission event
741 * received from downstream (in receiver mode) (Since 1.16)
742 * "sent-rtx-req-count" G_TYPE_UINT The number of retransmission event
743 * sent downstream (in sender mode) (Since 1.16)
744 * "rtx-count" G_TYPE_UINT DEPRECATED Since 1.16, same as
745 * "recv-rtx-req-count".
746 * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
747 * dropped (due to bandwidth constraints)
748 * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
749 * "recv-nack-count" G_TYPE_UINT Number of NACKs received
750 * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
751 * RTP sources (Since 1.8)
752 *
753 * Since: 1.4
754 */
755 g_object_class_install_property (gobject_class, PROP_STATS,
756 g_param_spec_boxed ("stats", "Statistics",
757 "Various statistics", GST_TYPE_STRUCTURE,
758 G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
759
760 g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
761 g_param_spec_enum ("rtp-profile", "RTP Profile",
762 "RTP profile to use", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
763 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
764
765 g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
766 g_param_spec_enum ("ntp-time-source", "NTP Time Source",
767 "NTP time source for RTCP packets",
768 gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
769 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
770
771 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
772 g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
773 "Use send time or capture time for RTCP sync "
774 "(TRUE = send time, FALSE = capture time)",
775 DEFAULT_RTCP_SYNC_SEND_TIME,
776 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
777
778 gstelement_class->change_state =
779 GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
780 gstelement_class->request_new_pad =
781 GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
782 gstelement_class->release_pad =
783 GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
784
785 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
786
787 /* sink pads */
788 gst_element_class_add_static_pad_template (gstelement_class,
789 &rtpsession_recv_rtp_sink_template);
790 gst_element_class_add_static_pad_template (gstelement_class,
791 &rtpsession_recv_rtcp_sink_template);
792 gst_element_class_add_static_pad_template (gstelement_class,
793 &rtpsession_send_rtp_sink_template);
794
795 /* src pads */
796 gst_element_class_add_static_pad_template (gstelement_class,
797 &rtpsession_recv_rtp_src_template);
798 gst_element_class_add_static_pad_template (gstelement_class,
799 &rtpsession_sync_src_template);
800 gst_element_class_add_static_pad_template (gstelement_class,
801 &rtpsession_send_rtp_src_template);
802 gst_element_class_add_static_pad_template (gstelement_class,
803 &rtpsession_send_rtcp_src_template);
804
805 gst_element_class_set_static_metadata (gstelement_class, "RTP Session",
806 "Filter/Network/RTP",
807 "Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>");
808
809 GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
810 "rtpsession", 0, "RTP Session");
811
812 GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp);
813 GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtcp);
814 GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp);
815 GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp_list);
816
817 }
818
819 static void
gst_rtp_session_init(GstRtpSession * rtpsession)820 gst_rtp_session_init (GstRtpSession * rtpsession)
821 {
822 rtpsession->priv = gst_rtp_session_get_instance_private (rtpsession);
823 g_mutex_init (&rtpsession->priv->lock);
824 g_cond_init (&rtpsession->priv->cond);
825 rtpsession->priv->sysclock = gst_system_clock_obtain ();
826 rtpsession->priv->session = rtp_session_new ();
827 rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
828 rtpsession->priv->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
829
830 /* configure callbacks */
831 rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
832 /* configure signals */
833 g_signal_connect (rtpsession->priv->session, "on-new-ssrc",
834 (GCallback) on_new_ssrc, rtpsession);
835 g_signal_connect (rtpsession->priv->session, "on-ssrc-collision",
836 (GCallback) on_ssrc_collision, rtpsession);
837 g_signal_connect (rtpsession->priv->session, "on-ssrc-validated",
838 (GCallback) on_ssrc_validated, rtpsession);
839 g_signal_connect (rtpsession->priv->session, "on-ssrc-active",
840 (GCallback) on_ssrc_active, rtpsession);
841 g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes",
842 (GCallback) on_ssrc_sdes, rtpsession);
843 g_signal_connect (rtpsession->priv->session, "on-bye-ssrc",
844 (GCallback) on_bye_ssrc, rtpsession);
845 g_signal_connect (rtpsession->priv->session, "on-bye-timeout",
846 (GCallback) on_bye_timeout, rtpsession);
847 g_signal_connect (rtpsession->priv->session, "on-timeout",
848 (GCallback) on_timeout, rtpsession);
849 g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
850 (GCallback) on_sender_timeout, rtpsession);
851 g_signal_connect (rtpsession->priv->session, "on-new-sender-ssrc",
852 (GCallback) on_new_sender_ssrc, rtpsession);
853 g_signal_connect (rtpsession->priv->session, "on-sender-ssrc-active",
854 (GCallback) on_sender_ssrc_active, rtpsession);
855 g_signal_connect (rtpsession->priv->session, "notify::stats",
856 (GCallback) on_notify_stats, rtpsession);
857 rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
858 (GDestroyNotify) gst_caps_unref);
859
860 rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
861
862 gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
863 gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
864
865 rtpsession->priv->thread_stopped = TRUE;
866
867 rtpsession->priv->recv_rtx_req_count = 0;
868 rtpsession->priv->sent_rtx_req_count = 0;
869
870 rtpsession->priv->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
871 }
872
873 static void
gst_rtp_session_finalize(GObject * object)874 gst_rtp_session_finalize (GObject * object)
875 {
876 GstRtpSession *rtpsession;
877
878 rtpsession = GST_RTP_SESSION (object);
879
880 g_hash_table_destroy (rtpsession->priv->ptmap);
881 g_mutex_clear (&rtpsession->priv->lock);
882 g_cond_clear (&rtpsession->priv->cond);
883 g_object_unref (rtpsession->priv->sysclock);
884 g_object_unref (rtpsession->priv->session);
885
886 G_OBJECT_CLASS (parent_class)->finalize (object);
887 }
888
889 static void
gst_rtp_session_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)890 gst_rtp_session_set_property (GObject * object, guint prop_id,
891 const GValue * value, GParamSpec * pspec)
892 {
893 GstRtpSession *rtpsession;
894 GstRtpSessionPrivate *priv;
895
896 rtpsession = GST_RTP_SESSION (object);
897 priv = rtpsession->priv;
898
899 switch (prop_id) {
900 case PROP_BANDWIDTH:
901 g_object_set_property (G_OBJECT (priv->session), "bandwidth", value);
902 break;
903 case PROP_RTCP_FRACTION:
904 g_object_set_property (G_OBJECT (priv->session), "rtcp-fraction", value);
905 break;
906 case PROP_RTCP_RR_BANDWIDTH:
907 g_object_set_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
908 value);
909 break;
910 case PROP_RTCP_RS_BANDWIDTH:
911 g_object_set_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
912 value);
913 break;
914 case PROP_SDES:
915 rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value));
916 break;
917 case PROP_USE_PIPELINE_CLOCK:
918 priv->use_pipeline_clock = g_value_get_boolean (value);
919 break;
920 case PROP_RTCP_MIN_INTERVAL:
921 g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval",
922 value);
923 break;
924 case PROP_PROBATION:
925 g_object_set_property (G_OBJECT (priv->session), "probation", value);
926 break;
927 case PROP_MAX_DROPOUT_TIME:
928 g_object_set_property (G_OBJECT (priv->session), "max-dropout-time",
929 value);
930 break;
931 case PROP_MAX_MISORDER_TIME:
932 g_object_set_property (G_OBJECT (priv->session), "max-misorder-time",
933 value);
934 break;
935 case PROP_RTP_PROFILE:
936 g_object_set_property (G_OBJECT (priv->session), "rtp-profile", value);
937 break;
938 case PROP_NTP_TIME_SOURCE:
939 priv->ntp_time_source = g_value_get_enum (value);
940 break;
941 case PROP_RTCP_SYNC_SEND_TIME:
942 priv->rtcp_sync_send_time = g_value_get_boolean (value);
943 break;
944 default:
945 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
946 break;
947 }
948 }
949
950 static void
gst_rtp_session_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)951 gst_rtp_session_get_property (GObject * object, guint prop_id,
952 GValue * value, GParamSpec * pspec)
953 {
954 GstRtpSession *rtpsession;
955 GstRtpSessionPrivate *priv;
956
957 rtpsession = GST_RTP_SESSION (object);
958 priv = rtpsession->priv;
959
960 switch (prop_id) {
961 case PROP_BANDWIDTH:
962 g_object_get_property (G_OBJECT (priv->session), "bandwidth", value);
963 break;
964 case PROP_RTCP_FRACTION:
965 g_object_get_property (G_OBJECT (priv->session), "rtcp-fraction", value);
966 break;
967 case PROP_RTCP_RR_BANDWIDTH:
968 g_object_get_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
969 value);
970 break;
971 case PROP_RTCP_RS_BANDWIDTH:
972 g_object_get_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
973 value);
974 break;
975 case PROP_SDES:
976 g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session));
977 break;
978 case PROP_NUM_SOURCES:
979 g_value_set_uint (value, rtp_session_get_num_sources (priv->session));
980 break;
981 case PROP_NUM_ACTIVE_SOURCES:
982 g_value_set_uint (value,
983 rtp_session_get_num_active_sources (priv->session));
984 break;
985 case PROP_INTERNAL_SESSION:
986 g_value_set_object (value, priv->session);
987 break;
988 case PROP_USE_PIPELINE_CLOCK:
989 g_value_set_boolean (value, priv->use_pipeline_clock);
990 break;
991 case PROP_RTCP_MIN_INTERVAL:
992 g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval",
993 value);
994 break;
995 case PROP_PROBATION:
996 g_object_get_property (G_OBJECT (priv->session), "probation", value);
997 break;
998 case PROP_MAX_DROPOUT_TIME:
999 g_object_get_property (G_OBJECT (priv->session), "max-dropout-time",
1000 value);
1001 break;
1002 case PROP_MAX_MISORDER_TIME:
1003 g_object_get_property (G_OBJECT (priv->session), "max-misorder-time",
1004 value);
1005 break;
1006 case PROP_STATS:
1007 g_value_take_boxed (value, gst_rtp_session_create_stats (rtpsession));
1008 break;
1009 case PROP_RTP_PROFILE:
1010 g_object_get_property (G_OBJECT (priv->session), "rtp-profile", value);
1011 break;
1012 case PROP_NTP_TIME_SOURCE:
1013 g_value_set_enum (value, priv->ntp_time_source);
1014 break;
1015 case PROP_RTCP_SYNC_SEND_TIME:
1016 g_value_set_boolean (value, priv->rtcp_sync_send_time);
1017 break;
1018 default:
1019 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1020 break;
1021 }
1022 }
1023
1024 static GstStructure *
gst_rtp_session_create_stats(GstRtpSession * rtpsession)1025 gst_rtp_session_create_stats (GstRtpSession * rtpsession)
1026 {
1027 GstStructure *s;
1028
1029 g_object_get (rtpsession->priv->session, "stats", &s, NULL);
1030 gst_structure_set (s, "rtx-count", G_TYPE_UINT,
1031 rtpsession->priv->recv_rtx_req_count, "recv-rtx-req-count", G_TYPE_UINT,
1032 rtpsession->priv->recv_rtx_req_count, "sent-rtx-req-count", G_TYPE_UINT,
1033 rtpsession->priv->sent_rtx_req_count, NULL);
1034
1035 return s;
1036 }
1037
1038 static void
get_current_times(GstRtpSession * rtpsession,GstClockTime * running_time,guint64 * ntpnstime)1039 get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time,
1040 guint64 * ntpnstime)
1041 {
1042 guint64 ntpns = -1;
1043 GstClock *clock;
1044 GstClockTime base_time, rt, clock_time;
1045
1046 GST_OBJECT_LOCK (rtpsession);
1047 if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
1048 base_time = GST_ELEMENT_CAST (rtpsession)->base_time;
1049 gst_object_ref (clock);
1050 GST_OBJECT_UNLOCK (rtpsession);
1051
1052 /* get current clock time and convert to running time */
1053 clock_time = gst_clock_get_time (clock);
1054 rt = clock_time - base_time;
1055
1056 if (rtpsession->priv->use_pipeline_clock) {
1057 ntpns = rt;
1058 /* add constant to convert from 1970 based time to 1900 based time */
1059 ntpns += (2208988800LL * GST_SECOND);
1060 } else {
1061 switch (rtpsession->priv->ntp_time_source) {
1062 case GST_RTP_NTP_TIME_SOURCE_NTP:
1063 case GST_RTP_NTP_TIME_SOURCE_UNIX:{
1064 GTimeVal current;
1065
1066 /* get current NTP time */
1067 g_get_current_time (¤t);
1068 ntpns = GST_TIMEVAL_TO_TIME (current);
1069
1070 /* add constant to convert from 1970 based time to 1900 based time */
1071 if (rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
1072 ntpns += (2208988800LL * GST_SECOND);
1073 break;
1074 }
1075 case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
1076 ntpns = rt;
1077 break;
1078 case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
1079 ntpns = clock_time;
1080 break;
1081 default:
1082 ntpns = -1;
1083 g_assert_not_reached ();
1084 break;
1085 }
1086 }
1087
1088 gst_object_unref (clock);
1089 } else {
1090 GST_OBJECT_UNLOCK (rtpsession);
1091 rt = -1;
1092 ntpns = -1;
1093 }
1094 if (running_time)
1095 *running_time = rt;
1096 if (ntpnstime)
1097 *ntpnstime = ntpns;
1098 }
1099
1100 /* must be called with GST_RTP_SESSION_LOCK */
1101 static void
signal_waiting_rtcp_thread_unlocked(GstRtpSession * rtpsession)1102 signal_waiting_rtcp_thread_unlocked (GstRtpSession * rtpsession)
1103 {
1104 if (rtpsession->priv->wait_send) {
1105 GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
1106 rtpsession->priv->wait_send = FALSE;
1107 GST_RTP_SESSION_SIGNAL (rtpsession);
1108 }
1109 }
1110
1111 static void
rtcp_thread(GstRtpSession * rtpsession)1112 rtcp_thread (GstRtpSession * rtpsession)
1113 {
1114 GstClockID id;
1115 GstClockTime current_time;
1116 GstClockTime next_timeout;
1117 guint64 ntpnstime;
1118 GstClockTime running_time;
1119 RTPSession *session;
1120 GstClock *sysclock;
1121
1122 GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
1123
1124 GST_RTP_SESSION_LOCK (rtpsession);
1125
1126 while (rtpsession->priv->wait_send) {
1127 GST_LOG_OBJECT (rtpsession, "waiting for getting started");
1128 GST_RTP_SESSION_WAIT (rtpsession);
1129 GST_LOG_OBJECT (rtpsession, "signaled...");
1130 }
1131
1132 sysclock = rtpsession->priv->sysclock;
1133 current_time = gst_clock_get_time (sysclock);
1134
1135 session = rtpsession->priv->session;
1136
1137 GST_DEBUG_OBJECT (rtpsession, "starting at %" GST_TIME_FORMAT,
1138 GST_TIME_ARGS (current_time));
1139 session->start_time = current_time;
1140
1141 while (!rtpsession->priv->stop_thread) {
1142 GstClockReturn res;
1143
1144 /* get initial estimate */
1145 next_timeout = rtp_session_next_timeout (session, current_time);
1146
1147 GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
1148 GST_TIME_ARGS (next_timeout));
1149
1150 /* leave if no more timeouts, the session ended */
1151 if (next_timeout == GST_CLOCK_TIME_NONE)
1152 break;
1153
1154 id = rtpsession->priv->id =
1155 gst_clock_new_single_shot_id (sysclock, next_timeout);
1156 GST_RTP_SESSION_UNLOCK (rtpsession);
1157
1158 res = gst_clock_id_wait (id, NULL);
1159
1160 GST_RTP_SESSION_LOCK (rtpsession);
1161 gst_clock_id_unref (id);
1162 rtpsession->priv->id = NULL;
1163
1164 if (rtpsession->priv->stop_thread)
1165 break;
1166
1167 /* update current time */
1168 current_time = gst_clock_get_time (sysclock);
1169
1170 /* get current NTP time */
1171 get_current_times (rtpsession, &running_time, &ntpnstime);
1172
1173 /* we get unlocked because we need to perform reconsideration, don't perform
1174 * the timeout but get a new reporting estimate. */
1175 GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
1176 res, GST_TIME_ARGS (current_time));
1177
1178 /* perform actions, we ignore result. Release lock because it might push. */
1179 GST_RTP_SESSION_UNLOCK (rtpsession);
1180 rtp_session_on_timeout (session, current_time, ntpnstime, running_time);
1181 GST_RTP_SESSION_LOCK (rtpsession);
1182 }
1183 /* mark the thread as stopped now */
1184 rtpsession->priv->thread_stopped = TRUE;
1185 GST_RTP_SESSION_UNLOCK (rtpsession);
1186
1187 GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
1188 }
1189
1190 static gboolean
start_rtcp_thread(GstRtpSession * rtpsession)1191 start_rtcp_thread (GstRtpSession * rtpsession)
1192 {
1193 GError *error = NULL;
1194 gboolean res;
1195
1196 GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread");
1197
1198 GST_RTP_SESSION_LOCK (rtpsession);
1199 rtpsession->priv->stop_thread = FALSE;
1200 if (rtpsession->priv->thread_stopped) {
1201 /* if the thread stopped, and we still have a handle to the thread, join it
1202 * now. We can safely join with the lock held, the thread will not take it
1203 * anymore. */
1204 if (rtpsession->priv->thread)
1205 g_thread_join (rtpsession->priv->thread);
1206 /* only create a new thread if the old one was stopped. Otherwise we can
1207 * just reuse the currently running one. */
1208 rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp-thread",
1209 (GThreadFunc) rtcp_thread, rtpsession, &error);
1210 rtpsession->priv->thread_stopped = FALSE;
1211 }
1212 GST_RTP_SESSION_UNLOCK (rtpsession);
1213
1214 if (error != NULL) {
1215 res = FALSE;
1216 GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message);
1217 g_error_free (error);
1218 } else {
1219 res = TRUE;
1220 }
1221 return res;
1222 }
1223
1224 static void
stop_rtcp_thread(GstRtpSession * rtpsession)1225 stop_rtcp_thread (GstRtpSession * rtpsession)
1226 {
1227 GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread");
1228
1229 GST_RTP_SESSION_LOCK (rtpsession);
1230 rtpsession->priv->stop_thread = TRUE;
1231 signal_waiting_rtcp_thread_unlocked (rtpsession);
1232 if (rtpsession->priv->id)
1233 gst_clock_id_unschedule (rtpsession->priv->id);
1234 GST_RTP_SESSION_UNLOCK (rtpsession);
1235 }
1236
1237 static void
join_rtcp_thread(GstRtpSession * rtpsession)1238 join_rtcp_thread (GstRtpSession * rtpsession)
1239 {
1240 GST_RTP_SESSION_LOCK (rtpsession);
1241 /* don't try to join when we have no thread */
1242 if (rtpsession->priv->thread != NULL) {
1243 GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread");
1244 GST_RTP_SESSION_UNLOCK (rtpsession);
1245
1246 g_thread_join (rtpsession->priv->thread);
1247
1248 GST_RTP_SESSION_LOCK (rtpsession);
1249 /* after the join, take the lock and clear the thread structure. The caller
1250 * is supposed to not concurrently call start and join. */
1251 rtpsession->priv->thread = NULL;
1252 }
1253 GST_RTP_SESSION_UNLOCK (rtpsession);
1254 }
1255
1256 static GstStateChangeReturn
gst_rtp_session_change_state(GstElement * element,GstStateChange transition)1257 gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
1258 {
1259 GstStateChangeReturn res;
1260 GstRtpSession *rtpsession;
1261
1262 rtpsession = GST_RTP_SESSION (element);
1263
1264 switch (transition) {
1265 case GST_STATE_CHANGE_NULL_TO_READY:
1266 break;
1267 case GST_STATE_CHANGE_READY_TO_PAUSED:
1268 GST_RTP_SESSION_LOCK (rtpsession);
1269 rtpsession->priv->wait_send = TRUE;
1270 GST_RTP_SESSION_UNLOCK (rtpsession);
1271 break;
1272 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1273 break;
1274 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1275 case GST_STATE_CHANGE_PAUSED_TO_READY:
1276 /* no need to join yet, we might want to continue later. Also, the
1277 * dataflow could block downstream so that a join could just block
1278 * forever. */
1279 stop_rtcp_thread (rtpsession);
1280 break;
1281 default:
1282 break;
1283 }
1284
1285 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1286
1287 switch (transition) {
1288 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1289 if (!start_rtcp_thread (rtpsession))
1290 goto failed_thread;
1291 break;
1292 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1293 break;
1294 case GST_STATE_CHANGE_PAUSED_TO_READY:
1295 /* downstream is now releasing the dataflow and we can join. */
1296 join_rtcp_thread (rtpsession);
1297 rtp_session_reset (rtpsession->priv->session);
1298 break;
1299 case GST_STATE_CHANGE_READY_TO_NULL:
1300 break;
1301 default:
1302 break;
1303 }
1304 return res;
1305
1306 /* ERRORS */
1307 failed_thread:
1308 {
1309 return GST_STATE_CHANGE_FAILURE;
1310 }
1311 }
1312
1313 static gboolean
return_true(gpointer key,gpointer value,gpointer user_data)1314 return_true (gpointer key, gpointer value, gpointer user_data)
1315 {
1316 return TRUE;
1317 }
1318
1319 static void
gst_rtp_session_clear_pt_map(GstRtpSession * rtpsession)1320 gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
1321 {
1322 GST_RTP_SESSION_LOCK (rtpsession);
1323 g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
1324 GST_RTP_SESSION_UNLOCK (rtpsession);
1325 }
1326
1327 /* called when the session manager has an RTP packet or a list of packets
1328 * ready for further processing */
1329 static GstFlowReturn
gst_rtp_session_process_rtp(RTPSession * sess,RTPSource * src,GstBuffer * buffer,gpointer user_data)1330 gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
1331 GstBuffer * buffer, gpointer user_data)
1332 {
1333 GstFlowReturn result;
1334 GstRtpSession *rtpsession;
1335 GstPad *rtp_src;
1336
1337 rtpsession = GST_RTP_SESSION (user_data);
1338
1339 GST_RTP_SESSION_LOCK (rtpsession);
1340 if ((rtp_src = rtpsession->recv_rtp_src))
1341 gst_object_ref (rtp_src);
1342 GST_RTP_SESSION_UNLOCK (rtpsession);
1343
1344 if (rtp_src) {
1345 GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
1346 result = gst_pad_push (rtp_src, buffer);
1347 gst_object_unref (rtp_src);
1348 } else {
1349 GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
1350 gst_buffer_unref (buffer);
1351 result = GST_FLOW_OK;
1352 }
1353 return result;
1354 }
1355
1356 /* called when the session manager has an RTP packet ready for further
1357 * sending */
1358 static GstFlowReturn
gst_rtp_session_send_rtp(RTPSession * sess,RTPSource * src,gpointer data,gpointer user_data)1359 gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
1360 gpointer data, gpointer user_data)
1361 {
1362 GstFlowReturn result;
1363 GstRtpSession *rtpsession;
1364 GstPad *rtp_src;
1365
1366 rtpsession = GST_RTP_SESSION (user_data);
1367
1368 GST_RTP_SESSION_LOCK (rtpsession);
1369 if ((rtp_src = rtpsession->send_rtp_src))
1370 gst_object_ref (rtp_src);
1371 signal_waiting_rtcp_thread_unlocked (rtpsession);
1372 GST_RTP_SESSION_UNLOCK (rtpsession);
1373
1374 if (rtp_src) {
1375 if (GST_IS_BUFFER (data)) {
1376 GST_LOG_OBJECT (rtpsession, "sending RTP packet");
1377 result = gst_pad_push (rtp_src, GST_BUFFER_CAST (data));
1378 } else {
1379 GST_LOG_OBJECT (rtpsession, "sending RTP list");
1380 result = gst_pad_push_list (rtp_src, GST_BUFFER_LIST_CAST (data));
1381 }
1382 gst_object_unref (rtp_src);
1383 } else {
1384 gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
1385 result = GST_FLOW_OK;
1386 }
1387 return result;
1388 }
1389
1390 static void
do_rtcp_events(GstRtpSession * rtpsession,GstPad * srcpad)1391 do_rtcp_events (GstRtpSession * rtpsession, GstPad * srcpad)
1392 {
1393 GstCaps *caps;
1394 GstSegment seg;
1395 GstEvent *event;
1396 gchar *stream_id;
1397 gboolean have_group_id;
1398 guint group_id;
1399
1400 stream_id =
1401 g_strdup_printf ("%08x%08x%08x%08x", g_random_int (), g_random_int (),
1402 g_random_int (), g_random_int ());
1403
1404 GST_RTP_SESSION_LOCK (rtpsession);
1405 if (rtpsession->recv_rtp_sink) {
1406 event =
1407 gst_pad_get_sticky_event (rtpsession->recv_rtp_sink,
1408 GST_EVENT_STREAM_START, 0);
1409 if (event) {
1410 if (gst_event_parse_group_id (event, &group_id))
1411 have_group_id = TRUE;
1412 else
1413 have_group_id = FALSE;
1414 gst_event_unref (event);
1415 } else {
1416 have_group_id = TRUE;
1417 group_id = gst_util_group_id_next ();
1418 }
1419 } else {
1420 have_group_id = TRUE;
1421 group_id = gst_util_group_id_next ();
1422 }
1423 GST_RTP_SESSION_UNLOCK (rtpsession);
1424
1425 event = gst_event_new_stream_start (stream_id);
1426 rtpsession->recv_rtcp_segment_seqnum = gst_event_get_seqnum (event);
1427 gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
1428 if (have_group_id)
1429 gst_event_set_group_id (event, group_id);
1430 gst_pad_push_event (srcpad, event);
1431 g_free (stream_id);
1432
1433 caps = gst_caps_new_empty_simple ("application/x-rtcp");
1434 gst_pad_set_caps (srcpad, caps);
1435 gst_caps_unref (caps);
1436
1437 gst_segment_init (&seg, GST_FORMAT_TIME);
1438 event = gst_event_new_segment (&seg);
1439 gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
1440 gst_pad_push_event (srcpad, event);
1441 }
1442
1443 /* called when the session manager has an RTCP packet ready for further
1444 * sending. The eos flag is set when an EOS event should be sent downstream as
1445 * well. */
1446 static GstFlowReturn
gst_rtp_session_send_rtcp(RTPSession * sess,RTPSource * src,GstBuffer * buffer,gboolean all_sources_bye,gpointer user_data)1447 gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
1448 GstBuffer * buffer, gboolean all_sources_bye, gpointer user_data)
1449 {
1450 GstFlowReturn result;
1451 GstRtpSession *rtpsession;
1452 GstPad *rtcp_src;
1453
1454 rtpsession = GST_RTP_SESSION (user_data);
1455
1456 GST_RTP_SESSION_LOCK (rtpsession);
1457 if (rtpsession->priv->stop_thread)
1458 goto stopping;
1459
1460 if ((rtcp_src = rtpsession->send_rtcp_src)) {
1461 gst_object_ref (rtcp_src);
1462 GST_RTP_SESSION_UNLOCK (rtpsession);
1463
1464 /* set rtcp caps on output pad */
1465 if (!gst_pad_has_current_caps (rtcp_src))
1466 do_rtcp_events (rtpsession, rtcp_src);
1467
1468 GST_LOG_OBJECT (rtpsession, "sending RTCP");
1469 result = gst_pad_push (rtcp_src, buffer);
1470
1471 /* Forward send an EOS on the RTCP sink if we received an EOS on the
1472 * send_rtp_sink. We don't need to check the recv_rtp_sink since in this
1473 * case the EOS event would already have been sent */
1474 if (all_sources_bye && rtpsession->send_rtp_sink &&
1475 GST_PAD_IS_EOS (rtpsession->send_rtp_sink)) {
1476 GstEvent *event;
1477
1478 GST_LOG_OBJECT (rtpsession, "sending EOS");
1479
1480 event = gst_event_new_eos ();
1481 gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
1482 gst_pad_push_event (rtcp_src, event);
1483 }
1484 gst_object_unref (rtcp_src);
1485 } else {
1486 GST_RTP_SESSION_UNLOCK (rtpsession);
1487
1488 GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
1489 gst_buffer_unref (buffer);
1490 result = GST_FLOW_OK;
1491 }
1492 return result;
1493
1494 /* ERRORS */
1495 stopping:
1496 {
1497 GST_DEBUG_OBJECT (rtpsession, "we are stopping");
1498 gst_buffer_unref (buffer);
1499 GST_RTP_SESSION_UNLOCK (rtpsession);
1500 return GST_FLOW_OK;
1501 }
1502 }
1503
1504 /* called when the session manager has an SR RTCP packet ready for handling
1505 * inter stream synchronisation */
1506 static GstFlowReturn
gst_rtp_session_sync_rtcp(RTPSession * sess,GstBuffer * buffer,gpointer user_data)1507 gst_rtp_session_sync_rtcp (RTPSession * sess,
1508 GstBuffer * buffer, gpointer user_data)
1509 {
1510 GstFlowReturn result;
1511 GstRtpSession *rtpsession;
1512 GstPad *sync_src;
1513
1514 rtpsession = GST_RTP_SESSION (user_data);
1515
1516 GST_RTP_SESSION_LOCK (rtpsession);
1517 if (rtpsession->priv->stop_thread)
1518 goto stopping;
1519
1520 if ((sync_src = rtpsession->sync_src)) {
1521 gst_object_ref (sync_src);
1522 GST_RTP_SESSION_UNLOCK (rtpsession);
1523
1524 /* set rtcp caps on output pad, this happens
1525 * when we receive RTCP muxed with RTP according
1526 * to RFC5761. Otherwise we would have forwarded
1527 * the events from the recv_rtcp_sink pad already
1528 */
1529 if (!gst_pad_has_current_caps (sync_src))
1530 do_rtcp_events (rtpsession, sync_src);
1531
1532 GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
1533 result = gst_pad_push (sync_src, buffer);
1534 gst_object_unref (sync_src);
1535 } else {
1536 GST_RTP_SESSION_UNLOCK (rtpsession);
1537
1538 GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
1539 gst_buffer_unref (buffer);
1540 result = GST_FLOW_OK;
1541 }
1542 return result;
1543
1544 /* ERRORS */
1545 stopping:
1546 {
1547 GST_DEBUG_OBJECT (rtpsession, "we are stopping");
1548 gst_buffer_unref (buffer);
1549 GST_RTP_SESSION_UNLOCK (rtpsession);
1550 return GST_FLOW_OK;
1551 }
1552 }
1553
1554 static void
gst_rtp_session_cache_caps(GstRtpSession * rtpsession,GstCaps * caps)1555 gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps)
1556 {
1557 GstRtpSessionPrivate *priv;
1558 const GstStructure *s;
1559 gint payload;
1560
1561 priv = rtpsession->priv;
1562
1563 GST_DEBUG_OBJECT (rtpsession, "parsing caps");
1564
1565 s = gst_caps_get_structure (caps, 0);
1566 if (!gst_structure_get_int (s, "payload", &payload))
1567 return;
1568
1569 if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload)))
1570 return;
1571
1572 g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload),
1573 gst_caps_ref (caps));
1574 }
1575
1576 static GstCaps *
gst_rtp_session_get_caps_for_pt(GstRtpSession * rtpsession,guint payload)1577 gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload)
1578 {
1579 GstCaps *caps = NULL;
1580 GValue args[2] = { {0}, {0} };
1581 GValue ret = { 0 };
1582
1583 GST_RTP_SESSION_LOCK (rtpsession);
1584 caps = g_hash_table_lookup (rtpsession->priv->ptmap,
1585 GINT_TO_POINTER (payload));
1586 if (caps) {
1587 gst_caps_ref (caps);
1588 goto done;
1589 }
1590
1591 /* not found in the cache, try to get it with a signal */
1592 g_value_init (&args[0], GST_TYPE_ELEMENT);
1593 g_value_set_object (&args[0], rtpsession);
1594 g_value_init (&args[1], G_TYPE_UINT);
1595 g_value_set_uint (&args[1], payload);
1596
1597 g_value_init (&ret, GST_TYPE_CAPS);
1598 g_value_set_boxed (&ret, NULL);
1599
1600 GST_RTP_SESSION_UNLOCK (rtpsession);
1601
1602 g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
1603 &ret);
1604
1605 GST_RTP_SESSION_LOCK (rtpsession);
1606
1607 g_value_unset (&args[0]);
1608 g_value_unset (&args[1]);
1609 caps = (GstCaps *) g_value_dup_boxed (&ret);
1610 g_value_unset (&ret);
1611 if (!caps)
1612 goto no_caps;
1613
1614 gst_rtp_session_cache_caps (rtpsession, caps);
1615
1616 done:
1617 GST_RTP_SESSION_UNLOCK (rtpsession);
1618
1619 return caps;
1620
1621 no_caps:
1622 {
1623 GST_DEBUG_OBJECT (rtpsession, "could not get caps");
1624 goto done;
1625 }
1626 }
1627
1628 /* called when the session manager needs the clock rate */
1629 static gint
gst_rtp_session_clock_rate(RTPSession * sess,guint8 payload,gpointer user_data)1630 gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
1631 gpointer user_data)
1632 {
1633 gint result = -1;
1634 GstRtpSession *rtpsession;
1635 GstCaps *caps;
1636 const GstStructure *s;
1637
1638 rtpsession = GST_RTP_SESSION_CAST (user_data);
1639
1640 caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
1641
1642 if (!caps)
1643 goto done;
1644
1645 s = gst_caps_get_structure (caps, 0);
1646 if (!gst_structure_get_int (s, "clock-rate", &result))
1647 goto no_clock_rate;
1648
1649 gst_caps_unref (caps);
1650
1651 GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
1652
1653 done:
1654
1655 return result;
1656
1657 /* ERRORS */
1658 no_clock_rate:
1659 {
1660 gst_caps_unref (caps);
1661 GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!");
1662 goto done;
1663 }
1664 }
1665
1666 /* called when the session manager asks us to reconsider the timeout */
1667 static void
gst_rtp_session_reconsider(RTPSession * sess,gpointer user_data)1668 gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data)
1669 {
1670 GstRtpSession *rtpsession;
1671
1672 rtpsession = GST_RTP_SESSION_CAST (user_data);
1673
1674 GST_RTP_SESSION_LOCK (rtpsession);
1675 GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration");
1676 if (rtpsession->priv->id)
1677 gst_clock_id_unschedule (rtpsession->priv->id);
1678 GST_RTP_SESSION_UNLOCK (rtpsession);
1679 }
1680
1681 static gboolean
gst_rtp_session_event_recv_rtp_sink(GstPad * pad,GstObject * parent,GstEvent * event)1682 gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstObject * parent,
1683 GstEvent * event)
1684 {
1685 GstRtpSession *rtpsession;
1686 gboolean ret = FALSE;
1687
1688 rtpsession = GST_RTP_SESSION (parent);
1689
1690 GST_DEBUG_OBJECT (rtpsession, "received event %s",
1691 GST_EVENT_TYPE_NAME (event));
1692
1693 switch (GST_EVENT_TYPE (event)) {
1694 case GST_EVENT_CAPS:
1695 {
1696 GstCaps *caps;
1697
1698 /* process */
1699 gst_event_parse_caps (event, &caps);
1700 gst_rtp_session_sink_setcaps (pad, rtpsession, caps);
1701 ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
1702 break;
1703 }
1704 case GST_EVENT_FLUSH_STOP:
1705 gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
1706 rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
1707 ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
1708 break;
1709 case GST_EVENT_SEGMENT:
1710 {
1711 GstSegment *segment, in_segment;
1712
1713 segment = &rtpsession->recv_rtp_seg;
1714
1715 /* the newsegment event is needed to convert the RTP timestamp to
1716 * running_time, which is needed to generate a mapping from RTP to NTP
1717 * timestamps in SR reports */
1718 gst_event_copy_segment (event, &in_segment);
1719 GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
1720 &in_segment);
1721
1722 /* accept upstream */
1723 gst_segment_copy_into (&in_segment, segment);
1724
1725 /* push event forward */
1726 ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
1727 break;
1728 }
1729 case GST_EVENT_EOS:
1730 {
1731 GstPad *rtcp_src;
1732
1733 ret =
1734 gst_pad_push_event (rtpsession->recv_rtp_src, gst_event_ref (event));
1735
1736 GST_RTP_SESSION_LOCK (rtpsession);
1737 if ((rtcp_src = rtpsession->send_rtcp_src))
1738 gst_object_ref (rtcp_src);
1739 GST_RTP_SESSION_UNLOCK (rtpsession);
1740
1741 gst_event_unref (event);
1742
1743 if (rtcp_src) {
1744 event = gst_event_new_eos ();
1745 if (rtpsession->recv_rtcp_segment_seqnum != GST_SEQNUM_INVALID)
1746 gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
1747 ret = gst_pad_push_event (rtcp_src, event);
1748 gst_object_unref (rtcp_src);
1749 } else {
1750 ret = TRUE;
1751 }
1752 break;
1753 }
1754 default:
1755 ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
1756 break;
1757 }
1758
1759 return ret;
1760
1761 }
1762
1763 static gboolean
gst_rtp_session_request_remote_key_unit(GstRtpSession * rtpsession,guint32 ssrc,guint payload,gboolean all_headers,gint count)1764 gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession,
1765 guint32 ssrc, guint payload, gboolean all_headers, gint count)
1766 {
1767 GstCaps *caps;
1768
1769 caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
1770
1771 if (caps) {
1772 const GstStructure *s = gst_caps_get_structure (caps, 0);
1773 gboolean pli;
1774 gboolean fir;
1775
1776 pli = gst_structure_has_field (s, "rtcp-fb-nack-pli");
1777 fir = gst_structure_has_field (s, "rtcp-fb-ccm-fir") && all_headers;
1778
1779 /* Google Talk uses FIR for repair, so send it even if we just want a
1780 * regular PLI */
1781 if (!pli &&
1782 gst_structure_has_field (s, "rtcp-fb-x-gstreamer-fir-as-repair"))
1783 fir = TRUE;
1784
1785 gst_caps_unref (caps);
1786
1787 if (pli || fir)
1788 return rtp_session_request_key_unit (rtpsession->priv->session, ssrc,
1789 fir, count);
1790 }
1791
1792 return FALSE;
1793 }
1794
1795 static gboolean
gst_rtp_session_event_recv_rtp_src(GstPad * pad,GstObject * parent,GstEvent * event)1796 gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstObject * parent,
1797 GstEvent * event)
1798 {
1799 GstRtpSession *rtpsession;
1800 gboolean forward = TRUE;
1801 gboolean ret = TRUE;
1802 const GstStructure *s;
1803 guint32 ssrc;
1804 guint pt;
1805
1806 rtpsession = GST_RTP_SESSION (parent);
1807
1808 switch (GST_EVENT_TYPE (event)) {
1809 case GST_EVENT_CUSTOM_UPSTREAM:
1810 s = gst_event_get_structure (event);
1811 if (gst_structure_has_name (s, "GstForceKeyUnit") &&
1812 gst_structure_get_uint (s, "ssrc", &ssrc) &&
1813 gst_structure_get_uint (s, "payload", &pt)) {
1814 gboolean all_headers = FALSE;
1815 gint count = -1;
1816
1817 gst_structure_get_boolean (s, "all-headers", &all_headers);
1818 if (gst_structure_get_int (s, "count", &count) && count < 0)
1819 count += G_MAXINT; /* Make sure count is positive if present */
1820 if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt,
1821 all_headers, count))
1822 forward = FALSE;
1823 } else if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
1824 guint seqnum, delay, deadline, max_delay, avg_rtt;
1825
1826 GST_RTP_SESSION_LOCK (rtpsession);
1827 rtpsession->priv->recv_rtx_req_count++;
1828 GST_RTP_SESSION_UNLOCK (rtpsession);
1829
1830 if (!gst_structure_get_uint (s, "ssrc", &ssrc))
1831 ssrc = -1;
1832 if (!gst_structure_get_uint (s, "seqnum", &seqnum))
1833 seqnum = -1;
1834 if (!gst_structure_get_uint (s, "delay", &delay))
1835 delay = 0;
1836 if (!gst_structure_get_uint (s, "deadline", &deadline))
1837 deadline = 100;
1838 if (!gst_structure_get_uint (s, "avg-rtt", &avg_rtt))
1839 avg_rtt = 40;
1840
1841 /* remaining time to receive the packet */
1842 max_delay = deadline;
1843 if (max_delay > delay)
1844 max_delay -= delay;
1845 /* estimated RTT */
1846 if (max_delay > avg_rtt)
1847 max_delay -= avg_rtt;
1848 else
1849 max_delay = 0;
1850
1851 if (rtp_session_request_nack (rtpsession->priv->session, ssrc, seqnum,
1852 max_delay * GST_MSECOND))
1853 forward = FALSE;
1854 }
1855 break;
1856 default:
1857 break;
1858 }
1859
1860 if (forward) {
1861 GstPad *recv_rtp_sink;
1862
1863 GST_RTP_SESSION_LOCK (rtpsession);
1864 if ((recv_rtp_sink = rtpsession->recv_rtp_sink))
1865 gst_object_ref (recv_rtp_sink);
1866 GST_RTP_SESSION_UNLOCK (rtpsession);
1867
1868 if (recv_rtp_sink) {
1869 ret = gst_pad_push_event (recv_rtp_sink, event);
1870 gst_object_unref (recv_rtp_sink);
1871 } else
1872 gst_event_unref (event);
1873 } else {
1874 gst_event_unref (event);
1875 }
1876
1877 return ret;
1878 }
1879
1880
1881 static GstIterator *
gst_rtp_session_iterate_internal_links(GstPad * pad,GstObject * parent)1882 gst_rtp_session_iterate_internal_links (GstPad * pad, GstObject * parent)
1883 {
1884 GstRtpSession *rtpsession;
1885 GstPad *otherpad = NULL;
1886 GstIterator *it = NULL;
1887
1888 rtpsession = GST_RTP_SESSION (parent);
1889
1890 GST_RTP_SESSION_LOCK (rtpsession);
1891 if (pad == rtpsession->recv_rtp_src) {
1892 otherpad = gst_object_ref (rtpsession->recv_rtp_sink);
1893 } else if (pad == rtpsession->recv_rtp_sink) {
1894 otherpad = gst_object_ref (rtpsession->recv_rtp_src);
1895 } else if (pad == rtpsession->send_rtp_src) {
1896 otherpad = gst_object_ref (rtpsession->send_rtp_sink);
1897 } else if (pad == rtpsession->send_rtp_sink) {
1898 otherpad = gst_object_ref (rtpsession->send_rtp_src);
1899 }
1900 GST_RTP_SESSION_UNLOCK (rtpsession);
1901
1902 if (otherpad) {
1903 GValue val = { 0, };
1904
1905 g_value_init (&val, GST_TYPE_PAD);
1906 g_value_set_object (&val, otherpad);
1907 it = gst_iterator_new_single (GST_TYPE_PAD, &val);
1908 g_value_unset (&val);
1909 gst_object_unref (otherpad);
1910 } else {
1911 it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
1912 }
1913
1914 return it;
1915 }
1916
1917 static gboolean
gst_rtp_session_sink_setcaps(GstPad * pad,GstRtpSession * rtpsession,GstCaps * caps)1918 gst_rtp_session_sink_setcaps (GstPad * pad, GstRtpSession * rtpsession,
1919 GstCaps * caps)
1920 {
1921 GST_RTP_SESSION_LOCK (rtpsession);
1922 gst_rtp_session_cache_caps (rtpsession, caps);
1923 GST_RTP_SESSION_UNLOCK (rtpsession);
1924
1925 return TRUE;
1926 }
1927
1928 /* receive a packet from a sender, send it to the RTP session manager and
1929 * forward the packet on the rtp_src pad
1930 */
1931 static GstFlowReturn
gst_rtp_session_chain_recv_rtp(GstPad * pad,GstObject * parent,GstBuffer * buffer)1932 gst_rtp_session_chain_recv_rtp (GstPad * pad, GstObject * parent,
1933 GstBuffer * buffer)
1934 {
1935 GstRtpSession *rtpsession;
1936 GstRtpSessionPrivate *priv;
1937 GstFlowReturn ret;
1938 GstClockTime current_time, running_time;
1939 GstClockTime timestamp;
1940 guint64 ntpnstime;
1941
1942 rtpsession = GST_RTP_SESSION (parent);
1943 priv = rtpsession->priv;
1944
1945 GST_LOG_OBJECT (rtpsession, "received RTP packet");
1946
1947 GST_RTP_SESSION_LOCK (rtpsession);
1948 signal_waiting_rtcp_thread_unlocked (rtpsession);
1949 GST_RTP_SESSION_UNLOCK (rtpsession);
1950
1951 /* get NTP time when this packet was captured, this depends on the timestamp. */
1952 timestamp = GST_BUFFER_PTS (buffer);
1953 if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
1954 /* convert to running time using the segment values */
1955 running_time =
1956 gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
1957 timestamp);
1958 ntpnstime = GST_CLOCK_TIME_NONE;
1959 } else {
1960 get_current_times (rtpsession, &running_time, &ntpnstime);
1961 }
1962 current_time = gst_clock_get_time (priv->sysclock);
1963
1964 ret = rtp_session_process_rtp (priv->session, buffer, current_time,
1965 running_time, ntpnstime);
1966 if (ret != GST_FLOW_OK)
1967 goto push_error;
1968
1969 done:
1970
1971 return ret;
1972
1973 /* ERRORS */
1974 push_error:
1975 {
1976 GST_DEBUG_OBJECT (rtpsession, "process returned %s",
1977 gst_flow_get_name (ret));
1978 goto done;
1979 }
1980 }
1981
1982 static gboolean
gst_rtp_session_event_recv_rtcp_sink(GstPad * pad,GstObject * parent,GstEvent * event)1983 gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstObject * parent,
1984 GstEvent * event)
1985 {
1986 GstRtpSession *rtpsession;
1987 gboolean ret = FALSE;
1988
1989 rtpsession = GST_RTP_SESSION (parent);
1990
1991 GST_DEBUG_OBJECT (rtpsession, "received event %s",
1992 GST_EVENT_TYPE_NAME (event));
1993
1994 switch (GST_EVENT_TYPE (event)) {
1995 case GST_EVENT_SEGMENT:
1996 /* Make sure that the sync_src pad has caps before the segment event.
1997 * Otherwise we might get a segment event before caps from the receive
1998 * RTCP pad, and then later when receiving RTCP packets will set caps.
1999 * This will results in a sticky event misordering warning
2000 */
2001 if (!gst_pad_has_current_caps (rtpsession->sync_src)) {
2002 GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
2003 gst_pad_set_caps (rtpsession->sync_src, caps);
2004 gst_caps_unref (caps);
2005 }
2006 /* fall through */
2007 default:
2008 ret = gst_pad_push_event (rtpsession->sync_src, event);
2009 break;
2010 }
2011
2012 return ret;
2013 }
2014
2015 /* Receive an RTCP packet from a sender, send it to the RTP session manager and
2016 * forward the SR packets to the sync_src pad.
2017 */
2018 static GstFlowReturn
gst_rtp_session_chain_recv_rtcp(GstPad * pad,GstObject * parent,GstBuffer * buffer)2019 gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstObject * parent,
2020 GstBuffer * buffer)
2021 {
2022 GstRtpSession *rtpsession;
2023 GstRtpSessionPrivate *priv;
2024 GstClockTime current_time;
2025 GstClockTime running_time;
2026 guint64 ntpnstime;
2027
2028 rtpsession = GST_RTP_SESSION (parent);
2029 priv = rtpsession->priv;
2030
2031 GST_LOG_OBJECT (rtpsession, "received RTCP packet");
2032
2033 GST_RTP_SESSION_LOCK (rtpsession);
2034 signal_waiting_rtcp_thread_unlocked (rtpsession);
2035 GST_RTP_SESSION_UNLOCK (rtpsession);
2036
2037 current_time = gst_clock_get_time (priv->sysclock);
2038 get_current_times (rtpsession, &running_time, &ntpnstime);
2039
2040 rtp_session_process_rtcp (priv->session, buffer, current_time, running_time,
2041 ntpnstime);
2042
2043 return GST_FLOW_OK; /* always return OK */
2044 }
2045
2046 static gboolean
gst_rtp_session_query_send_rtcp_src(GstPad * pad,GstObject * parent,GstQuery * query)2047 gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstObject * parent,
2048 GstQuery * query)
2049 {
2050 GstRtpSession *rtpsession;
2051 gboolean ret = FALSE;
2052
2053 rtpsession = GST_RTP_SESSION (parent);
2054
2055 GST_DEBUG_OBJECT (rtpsession, "received QUERY %s",
2056 GST_QUERY_TYPE_NAME (query));
2057
2058 switch (GST_QUERY_TYPE (query)) {
2059 case GST_QUERY_LATENCY:
2060 ret = TRUE;
2061 /* use the defaults for the latency query. */
2062 gst_query_set_latency (query, FALSE, 0, -1);
2063 break;
2064 default:
2065 /* other queries simply fail for now */
2066 break;
2067 }
2068
2069 return ret;
2070 }
2071
2072 static gboolean
gst_rtp_session_event_send_rtcp_src(GstPad * pad,GstObject * parent,GstEvent * event)2073 gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstObject * parent,
2074 GstEvent * event)
2075 {
2076 GstRtpSession *rtpsession;
2077 gboolean ret = TRUE;
2078
2079 rtpsession = GST_RTP_SESSION (parent);
2080 GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
2081 GST_EVENT_TYPE_NAME (event));
2082
2083 switch (GST_EVENT_TYPE (event)) {
2084 case GST_EVENT_SEEK:
2085 case GST_EVENT_LATENCY:
2086 gst_event_unref (event);
2087 ret = TRUE;
2088 break;
2089 default:
2090 /* other events simply fail for now */
2091 gst_event_unref (event);
2092 ret = FALSE;
2093 break;
2094 }
2095
2096 return ret;
2097 }
2098
2099
2100 static gboolean
gst_rtp_session_event_send_rtp_sink(GstPad * pad,GstObject * parent,GstEvent * event)2101 gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
2102 GstEvent * event)
2103 {
2104 GstRtpSession *rtpsession;
2105 gboolean ret = FALSE;
2106
2107 rtpsession = GST_RTP_SESSION (parent);
2108
2109 GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
2110 GST_EVENT_TYPE_NAME (event));
2111
2112 switch (GST_EVENT_TYPE (event)) {
2113 case GST_EVENT_CAPS:
2114 {
2115 GstCaps *caps;
2116
2117 /* process */
2118 gst_event_parse_caps (event, &caps);
2119 gst_rtp_session_setcaps_send_rtp (pad, rtpsession, caps);
2120 ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
2121 break;
2122 }
2123 case GST_EVENT_FLUSH_STOP:
2124 gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
2125 ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
2126 break;
2127 case GST_EVENT_SEGMENT:{
2128 GstSegment *segment, in_segment;
2129
2130 segment = &rtpsession->send_rtp_seg;
2131
2132 /* the newsegment event is needed to convert the RTP timestamp to
2133 * running_time, which is needed to generate a mapping from RTP to NTP
2134 * timestamps in SR reports */
2135 gst_event_copy_segment (event, &in_segment);
2136 GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
2137 &in_segment);
2138
2139 /* accept upstream */
2140 gst_segment_copy_into (&in_segment, segment);
2141
2142 /* push event forward */
2143 ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
2144 break;
2145 }
2146 case GST_EVENT_EOS:{
2147 GstClockTime current_time;
2148
2149 /* push downstream FIXME, we are not supposed to leave the session just
2150 * because we stop sending. */
2151 ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
2152 current_time = gst_clock_get_time (rtpsession->priv->sysclock);
2153
2154 GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
2155 rtp_session_mark_all_bye (rtpsession->priv->session, "End Of Stream");
2156 rtp_session_schedule_bye (rtpsession->priv->session, current_time);
2157 break;
2158 }
2159 default:{
2160 GstPad *send_rtp_src;
2161
2162 GST_RTP_SESSION_LOCK (rtpsession);
2163 if ((send_rtp_src = rtpsession->send_rtp_src))
2164 gst_object_ref (send_rtp_src);
2165 GST_RTP_SESSION_UNLOCK (rtpsession);
2166
2167 if (send_rtp_src) {
2168 ret = gst_pad_push_event (send_rtp_src, event);
2169 gst_object_unref (send_rtp_src);
2170 } else
2171 gst_event_unref (event);
2172
2173 break;
2174 }
2175 }
2176
2177 return ret;
2178 }
2179
2180 static gboolean
gst_rtp_session_event_send_rtp_src(GstPad * pad,GstObject * parent,GstEvent * event)2181 gst_rtp_session_event_send_rtp_src (GstPad * pad, GstObject * parent,
2182 GstEvent * event)
2183 {
2184 GstRtpSession *rtpsession;
2185 gboolean ret = FALSE;
2186
2187 rtpsession = GST_RTP_SESSION (parent);
2188
2189 GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
2190 GST_EVENT_TYPE_NAME (event));
2191
2192 switch (GST_EVENT_TYPE (event)) {
2193 case GST_EVENT_LATENCY:
2194 /* save the latency, we need this to know when an RTP packet will be
2195 * rendered by the sink */
2196 gst_event_parse_latency (event, &rtpsession->priv->send_latency);
2197
2198 ret = gst_pad_event_default (pad, parent, event);
2199 break;
2200 default:
2201 ret = gst_pad_event_default (pad, parent, event);
2202 break;
2203 }
2204 return ret;
2205 }
2206
2207 static GstCaps *
gst_rtp_session_getcaps_send_rtp(GstPad * pad,GstRtpSession * rtpsession,GstCaps * filter)2208 gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
2209 GstCaps * filter)
2210 {
2211 GstRtpSessionPrivate *priv;
2212 GstCaps *result;
2213 GstStructure *s1, *s2;
2214 guint ssrc;
2215 gboolean is_random;
2216
2217 priv = rtpsession->priv;
2218
2219 ssrc = rtp_session_suggest_ssrc (priv->session, &is_random);
2220
2221 /* we can basically accept anything but we prefer to receive packets with our
2222 * internal SSRC so that we don't have to patch it. Create a structure with
2223 * the SSRC and another one without.
2224 * Only do this if the session actually decided on an ssrc already,
2225 * otherwise we give upstream the opportunity to select an ssrc itself */
2226 if (!is_random) {
2227 s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc,
2228 NULL);
2229 s2 = gst_structure_new_empty ("application/x-rtp");
2230
2231 result = gst_caps_new_full (s1, s2, NULL);
2232 } else {
2233 result = gst_caps_new_empty_simple ("application/x-rtp");
2234 }
2235
2236 if (filter) {
2237 GstCaps *caps = result;
2238
2239 result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
2240 gst_caps_unref (caps);
2241 }
2242
2243 GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
2244
2245 return result;
2246 }
2247
2248 static gboolean
gst_rtp_session_query_send_rtp(GstPad * pad,GstObject * parent,GstQuery * query)2249 gst_rtp_session_query_send_rtp (GstPad * pad, GstObject * parent,
2250 GstQuery * query)
2251 {
2252 gboolean res = FALSE;
2253 GstRtpSession *rtpsession;
2254
2255 rtpsession = GST_RTP_SESSION (parent);
2256
2257 switch (GST_QUERY_TYPE (query)) {
2258 case GST_QUERY_CAPS:
2259 {
2260 GstCaps *filter, *caps;
2261
2262 gst_query_parse_caps (query, &filter);
2263 caps = gst_rtp_session_getcaps_send_rtp (pad, rtpsession, filter);
2264 gst_query_set_caps_result (query, caps);
2265 gst_caps_unref (caps);
2266 res = TRUE;
2267 break;
2268 }
2269 default:
2270 res = gst_pad_query_default (pad, parent, query);
2271 break;
2272 }
2273
2274 return res;
2275 }
2276
2277 static gboolean
gst_rtp_session_setcaps_send_rtp(GstPad * pad,GstRtpSession * rtpsession,GstCaps * caps)2278 gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
2279 GstCaps * caps)
2280 {
2281 GstRtpSessionPrivate *priv;
2282
2283 priv = rtpsession->priv;
2284
2285 rtp_session_update_send_caps (priv->session, caps);
2286
2287 return TRUE;
2288 }
2289
2290 /* Receive an RTP packet or a list of packets to be sent to the receivers,
2291 * send to RTP session manager and forward to send_rtp_src.
2292 */
2293 static GstFlowReturn
gst_rtp_session_chain_send_rtp_common(GstRtpSession * rtpsession,gpointer data,gboolean is_list)2294 gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
2295 gpointer data, gboolean is_list)
2296 {
2297 GstRtpSessionPrivate *priv;
2298 GstFlowReturn ret;
2299 GstClockTime timestamp, running_time;
2300 GstClockTime current_time;
2301
2302 priv = rtpsession->priv;
2303
2304 GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
2305
2306 /* get NTP time when this packet was captured, this depends on the timestamp. */
2307 if (is_list) {
2308 GstBuffer *buffer = NULL;
2309
2310 /* All buffers in a list have the same timestamp.
2311 * So, just take it from the first buffer. */
2312 buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0);
2313 if (buffer)
2314 timestamp = GST_BUFFER_PTS (buffer);
2315 else
2316 timestamp = -1;
2317 } else {
2318 timestamp = GST_BUFFER_PTS (GST_BUFFER_CAST (data));
2319 }
2320
2321 if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
2322 /* convert to running time using the segment start value. */
2323 running_time =
2324 gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
2325 timestamp);
2326 if (priv->rtcp_sync_send_time)
2327 running_time += priv->send_latency;
2328 } else {
2329 /* no timestamp. */
2330 running_time = -1;
2331 }
2332
2333 current_time = gst_clock_get_time (priv->sysclock);
2334 ret = rtp_session_send_rtp (priv->session, data, is_list, current_time,
2335 running_time);
2336 if (ret != GST_FLOW_OK)
2337 goto push_error;
2338
2339 done:
2340
2341 return ret;
2342
2343 /* ERRORS */
2344 push_error:
2345 {
2346 GST_DEBUG_OBJECT (rtpsession, "process returned %s",
2347 gst_flow_get_name (ret));
2348 goto done;
2349 }
2350 }
2351
2352 static GstFlowReturn
gst_rtp_session_chain_send_rtp(GstPad * pad,GstObject * parent,GstBuffer * buffer)2353 gst_rtp_session_chain_send_rtp (GstPad * pad, GstObject * parent,
2354 GstBuffer * buffer)
2355 {
2356 GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
2357
2358 return gst_rtp_session_chain_send_rtp_common (rtpsession, buffer, FALSE);
2359 }
2360
2361 static GstFlowReturn
gst_rtp_session_chain_send_rtp_list(GstPad * pad,GstObject * parent,GstBufferList * list)2362 gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstObject * parent,
2363 GstBufferList * list)
2364 {
2365 GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
2366
2367 return gst_rtp_session_chain_send_rtp_common (rtpsession, list, TRUE);
2368 }
2369
2370 /* Create sinkpad to receive RTP packets from senders. This will also create a
2371 * srcpad for the RTP packets.
2372 */
2373 static GstPad *
create_recv_rtp_sink(GstRtpSession * rtpsession)2374 create_recv_rtp_sink (GstRtpSession * rtpsession)
2375 {
2376 GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad");
2377
2378 rtpsession->recv_rtp_sink =
2379 gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
2380 "recv_rtp_sink");
2381 gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
2382 gst_rtp_session_chain_recv_rtp);
2383 gst_pad_set_event_function (rtpsession->recv_rtp_sink,
2384 gst_rtp_session_event_recv_rtp_sink);
2385 gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink,
2386 gst_rtp_session_iterate_internal_links);
2387 GST_PAD_SET_PROXY_ALLOCATION (rtpsession->recv_rtp_sink);
2388 gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
2389 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
2390 rtpsession->recv_rtp_sink);
2391
2392 GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad");
2393 rtpsession->recv_rtp_src =
2394 gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
2395 "recv_rtp_src");
2396 gst_pad_set_event_function (rtpsession->recv_rtp_src,
2397 gst_rtp_session_event_recv_rtp_src);
2398 gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src,
2399 gst_rtp_session_iterate_internal_links);
2400 gst_pad_use_fixed_caps (rtpsession->recv_rtp_src);
2401 gst_pad_set_active (rtpsession->recv_rtp_src, TRUE);
2402 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
2403
2404 return rtpsession->recv_rtp_sink;
2405 }
2406
2407 /* Remove sinkpad to receive RTP packets from senders. This will also remove
2408 * the srcpad for the RTP packets.
2409 */
2410 static void
remove_recv_rtp_sink(GstRtpSession * rtpsession)2411 remove_recv_rtp_sink (GstRtpSession * rtpsession)
2412 {
2413 GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad");
2414
2415 /* deactivate from source to sink */
2416 gst_pad_set_active (rtpsession->recv_rtp_src, FALSE);
2417 gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE);
2418
2419 /* remove pads */
2420 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2421 rtpsession->recv_rtp_sink);
2422 rtpsession->recv_rtp_sink = NULL;
2423
2424 GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad");
2425 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2426 rtpsession->recv_rtp_src);
2427 rtpsession->recv_rtp_src = NULL;
2428 }
2429
2430 /* Create a sinkpad to receive RTCP messages from senders, this will also create a
2431 * sync_src pad for the SR packets.
2432 */
2433 static GstPad *
create_recv_rtcp_sink(GstRtpSession * rtpsession)2434 create_recv_rtcp_sink (GstRtpSession * rtpsession)
2435 {
2436 GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad");
2437
2438 rtpsession->recv_rtcp_sink =
2439 gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
2440 "recv_rtcp_sink");
2441 gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
2442 gst_rtp_session_chain_recv_rtcp);
2443 gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
2444 gst_rtp_session_event_recv_rtcp_sink);
2445 gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtcp_sink,
2446 gst_rtp_session_iterate_internal_links);
2447 gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE);
2448 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
2449 rtpsession->recv_rtcp_sink);
2450
2451 GST_DEBUG_OBJECT (rtpsession, "creating sync src pad");
2452 rtpsession->sync_src =
2453 gst_pad_new_from_static_template (&rtpsession_sync_src_template,
2454 "sync_src");
2455 gst_pad_set_iterate_internal_links_function (rtpsession->sync_src,
2456 gst_rtp_session_iterate_internal_links);
2457 gst_pad_use_fixed_caps (rtpsession->sync_src);
2458 gst_pad_set_active (rtpsession->sync_src, TRUE);
2459 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
2460
2461 return rtpsession->recv_rtcp_sink;
2462 }
2463
2464 static void
remove_recv_rtcp_sink(GstRtpSession * rtpsession)2465 remove_recv_rtcp_sink (GstRtpSession * rtpsession)
2466 {
2467 GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad");
2468
2469 gst_pad_set_active (rtpsession->sync_src, FALSE);
2470 gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE);
2471
2472 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2473 rtpsession->recv_rtcp_sink);
2474 rtpsession->recv_rtcp_sink = NULL;
2475
2476 GST_DEBUG_OBJECT (rtpsession, "removing sync src pad");
2477 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
2478 rtpsession->sync_src = NULL;
2479 }
2480
2481 /* Create a sinkpad to receive RTP packets for receivers. This will also create a
2482 * send_rtp_src pad.
2483 */
2484 static GstPad *
create_send_rtp_sink(GstRtpSession * rtpsession)2485 create_send_rtp_sink (GstRtpSession * rtpsession)
2486 {
2487 GST_DEBUG_OBJECT (rtpsession, "creating pad");
2488
2489 rtpsession->send_rtp_sink =
2490 gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
2491 "send_rtp_sink");
2492 gst_pad_set_chain_function (rtpsession->send_rtp_sink,
2493 gst_rtp_session_chain_send_rtp);
2494 gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
2495 gst_rtp_session_chain_send_rtp_list);
2496 gst_pad_set_query_function (rtpsession->send_rtp_sink,
2497 gst_rtp_session_query_send_rtp);
2498 gst_pad_set_event_function (rtpsession->send_rtp_sink,
2499 gst_rtp_session_event_send_rtp_sink);
2500 gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink,
2501 gst_rtp_session_iterate_internal_links);
2502 GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_sink);
2503 GST_PAD_SET_PROXY_ALLOCATION (rtpsession->send_rtp_sink);
2504 gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
2505 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
2506 rtpsession->send_rtp_sink);
2507
2508 rtpsession->send_rtp_src =
2509 gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
2510 "send_rtp_src");
2511 gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src,
2512 gst_rtp_session_iterate_internal_links);
2513 gst_pad_set_event_function (rtpsession->send_rtp_src,
2514 gst_rtp_session_event_send_rtp_src);
2515 GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_src);
2516 gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
2517 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
2518
2519 return rtpsession->send_rtp_sink;
2520 }
2521
2522 static void
remove_send_rtp_sink(GstRtpSession * rtpsession)2523 remove_send_rtp_sink (GstRtpSession * rtpsession)
2524 {
2525 GST_DEBUG_OBJECT (rtpsession, "removing pad");
2526
2527 gst_pad_set_active (rtpsession->send_rtp_src, FALSE);
2528 gst_pad_set_active (rtpsession->send_rtp_sink, FALSE);
2529
2530 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2531 rtpsession->send_rtp_sink);
2532 rtpsession->send_rtp_sink = NULL;
2533
2534 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2535 rtpsession->send_rtp_src);
2536 rtpsession->send_rtp_src = NULL;
2537 }
2538
2539 /* Create a srcpad with the RTCP packets to send out.
2540 * This pad will be driven by the RTP session manager when it wants to send out
2541 * RTCP packets.
2542 */
2543 static GstPad *
create_send_rtcp_src(GstRtpSession * rtpsession)2544 create_send_rtcp_src (GstRtpSession * rtpsession)
2545 {
2546 GST_DEBUG_OBJECT (rtpsession, "creating pad");
2547
2548 rtpsession->send_rtcp_src =
2549 gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
2550 "send_rtcp_src");
2551 gst_pad_use_fixed_caps (rtpsession->send_rtcp_src);
2552 gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
2553 gst_pad_set_iterate_internal_links_function (rtpsession->send_rtcp_src,
2554 gst_rtp_session_iterate_internal_links);
2555 gst_pad_set_query_function (rtpsession->send_rtcp_src,
2556 gst_rtp_session_query_send_rtcp_src);
2557 gst_pad_set_event_function (rtpsession->send_rtcp_src,
2558 gst_rtp_session_event_send_rtcp_src);
2559 gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
2560 rtpsession->send_rtcp_src);
2561
2562 return rtpsession->send_rtcp_src;
2563 }
2564
2565 static void
remove_send_rtcp_src(GstRtpSession * rtpsession)2566 remove_send_rtcp_src (GstRtpSession * rtpsession)
2567 {
2568 GST_DEBUG_OBJECT (rtpsession, "removing pad");
2569
2570 gst_pad_set_active (rtpsession->send_rtcp_src, FALSE);
2571
2572 gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
2573 rtpsession->send_rtcp_src);
2574 rtpsession->send_rtcp_src = NULL;
2575 }
2576
2577 static GstPad *
gst_rtp_session_request_new_pad(GstElement * element,GstPadTemplate * templ,const gchar * name,const GstCaps * caps)2578 gst_rtp_session_request_new_pad (GstElement * element,
2579 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
2580 {
2581 GstRtpSession *rtpsession;
2582 GstElementClass *klass;
2583 GstPad *result;
2584
2585 g_return_val_if_fail (templ != NULL, NULL);
2586 g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
2587
2588 rtpsession = GST_RTP_SESSION (element);
2589 klass = GST_ELEMENT_GET_CLASS (element);
2590
2591 GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
2592
2593 GST_RTP_SESSION_LOCK (rtpsession);
2594
2595 /* figure out the template */
2596 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
2597 if (rtpsession->recv_rtp_sink != NULL)
2598 goto exists;
2599
2600 result = create_recv_rtp_sink (rtpsession);
2601 } else if (templ == gst_element_class_get_pad_template (klass,
2602 "recv_rtcp_sink")) {
2603 if (rtpsession->recv_rtcp_sink != NULL)
2604 goto exists;
2605
2606 result = create_recv_rtcp_sink (rtpsession);
2607 } else if (templ == gst_element_class_get_pad_template (klass,
2608 "send_rtp_sink")) {
2609 if (rtpsession->send_rtp_sink != NULL)
2610 goto exists;
2611
2612 result = create_send_rtp_sink (rtpsession);
2613 } else if (templ == gst_element_class_get_pad_template (klass,
2614 "send_rtcp_src")) {
2615 if (rtpsession->send_rtcp_src != NULL)
2616 goto exists;
2617
2618 result = create_send_rtcp_src (rtpsession);
2619 } else
2620 goto wrong_template;
2621
2622 GST_RTP_SESSION_UNLOCK (rtpsession);
2623
2624 return result;
2625
2626 /* ERRORS */
2627 wrong_template:
2628 {
2629 GST_RTP_SESSION_UNLOCK (rtpsession);
2630 g_warning ("rtpsession: this is not our template");
2631 return NULL;
2632 }
2633 exists:
2634 {
2635 GST_RTP_SESSION_UNLOCK (rtpsession);
2636 g_warning ("rtpsession: pad already requested");
2637 return NULL;
2638 }
2639 }
2640
2641 static void
gst_rtp_session_release_pad(GstElement * element,GstPad * pad)2642 gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
2643 {
2644 GstRtpSession *rtpsession;
2645
2646 g_return_if_fail (GST_IS_RTP_SESSION (element));
2647 g_return_if_fail (GST_IS_PAD (pad));
2648
2649 rtpsession = GST_RTP_SESSION (element);
2650
2651 GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
2652
2653 GST_RTP_SESSION_LOCK (rtpsession);
2654
2655 if (rtpsession->recv_rtp_sink == pad) {
2656 remove_recv_rtp_sink (rtpsession);
2657 } else if (rtpsession->recv_rtcp_sink == pad) {
2658 remove_recv_rtcp_sink (rtpsession);
2659 } else if (rtpsession->send_rtp_sink == pad) {
2660 remove_send_rtp_sink (rtpsession);
2661 } else if (rtpsession->send_rtcp_src == pad) {
2662 remove_send_rtcp_src (rtpsession);
2663 } else
2664 goto wrong_pad;
2665
2666 GST_RTP_SESSION_UNLOCK (rtpsession);
2667
2668 return;
2669
2670 /* ERRORS */
2671 wrong_pad:
2672 {
2673 GST_RTP_SESSION_UNLOCK (rtpsession);
2674 g_warning ("rtpsession: asked to release an unknown pad");
2675 return;
2676 }
2677 }
2678
2679 static void
gst_rtp_session_request_key_unit(RTPSession * sess,guint32 ssrc,gboolean all_headers,gpointer user_data)2680 gst_rtp_session_request_key_unit (RTPSession * sess,
2681 guint32 ssrc, gboolean all_headers, gpointer user_data)
2682 {
2683 GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
2684 GstEvent *event;
2685 GstPad *send_rtp_sink;
2686
2687 GST_RTP_SESSION_LOCK (rtpsession);
2688 if ((send_rtp_sink = rtpsession->send_rtp_sink))
2689 gst_object_ref (send_rtp_sink);
2690 GST_RTP_SESSION_UNLOCK (rtpsession);
2691
2692 if (send_rtp_sink) {
2693 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2694 gst_structure_new ("GstForceKeyUnit", "ssrc", G_TYPE_UINT, ssrc,
2695 "all-headers", G_TYPE_BOOLEAN, all_headers, NULL));
2696 gst_pad_push_event (send_rtp_sink, event);
2697 gst_object_unref (send_rtp_sink);
2698 }
2699 }
2700
2701 static GstClockTime
gst_rtp_session_request_time(RTPSession * session,gpointer user_data)2702 gst_rtp_session_request_time (RTPSession * session, gpointer user_data)
2703 {
2704 GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
2705
2706 return gst_clock_get_time (rtpsession->priv->sysclock);
2707 }
2708
2709 static void
gst_rtp_session_notify_nack(RTPSession * sess,guint16 seqnum,guint16 blp,guint32 ssrc,gpointer user_data)2710 gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum,
2711 guint16 blp, guint32 ssrc, gpointer user_data)
2712 {
2713 GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
2714 GstEvent *event;
2715 GstPad *send_rtp_sink;
2716
2717 GST_RTP_SESSION_LOCK (rtpsession);
2718 if ((send_rtp_sink = rtpsession->send_rtp_sink))
2719 gst_object_ref (send_rtp_sink);
2720 GST_RTP_SESSION_UNLOCK (rtpsession);
2721
2722 if (send_rtp_sink) {
2723 while (TRUE) {
2724 event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
2725 gst_structure_new ("GstRTPRetransmissionRequest",
2726 "seqnum", G_TYPE_UINT, (guint) seqnum,
2727 "ssrc", G_TYPE_UINT, (guint) ssrc, NULL));
2728 gst_pad_push_event (send_rtp_sink, event);
2729
2730 GST_RTP_SESSION_LOCK (rtpsession);
2731 rtpsession->priv->sent_rtx_req_count++;
2732 GST_RTP_SESSION_UNLOCK (rtpsession);
2733
2734 if (blp == 0)
2735 break;
2736
2737 seqnum++;
2738 while ((blp & 1) == 0) {
2739 seqnum++;
2740 blp >>= 1;
2741 }
2742 blp >>= 1;
2743 }
2744 gst_object_unref (send_rtp_sink);
2745 }
2746 }
2747
2748 static void
gst_rtp_session_reconfigure(RTPSession * sess,gpointer user_data)2749 gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data)
2750 {
2751 GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
2752 GstPad *send_rtp_sink;
2753
2754 GST_RTP_SESSION_LOCK (rtpsession);
2755 if ((send_rtp_sink = rtpsession->send_rtp_sink))
2756 gst_object_ref (send_rtp_sink);
2757 GST_RTP_SESSION_UNLOCK (rtpsession);
2758
2759 if (send_rtp_sink) {
2760 gst_pad_push_event (send_rtp_sink, gst_event_new_reconfigure ());
2761 gst_object_unref (send_rtp_sink);
2762 }
2763 }
2764
2765 static void
gst_rtp_session_notify_early_rtcp(RTPSession * sess,gpointer user_data)2766 gst_rtp_session_notify_early_rtcp (RTPSession * sess, gpointer user_data)
2767 {
2768 GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
2769
2770 GST_DEBUG_OBJECT (rtpsession, "Notified of early RTCP");
2771 /* with an early RTCP request, we might have to start the RTCP thread */
2772 GST_RTP_SESSION_LOCK (rtpsession);
2773 signal_waiting_rtcp_thread_unlocked (rtpsession);
2774 GST_RTP_SESSION_UNLOCK (rtpsession);
2775 }
2776