2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * SECTION:element-rtpbin
22 * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
24 * RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
25 * #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
26 * RTP sessions that will be synchronized together using RTCP SR packets.
28 * #GstRtpBin is configured with a number of request pads that define the
29 * functionality that is activated, similar to the #GstRtpSession element.
31 * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%u pad. The session
32 * number must be specified in the pad name.
33 * Data received on the recv_rtp_sink_\%u pad will be processed in the #GstRtpSession
34 * manager and after being validated forwarded on #GstRtpSsrcDemux element. Each
35 * RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After
36 * the packets are released from the jitterbuffer, they will be forwarded to a
37 * #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
38 * on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
39 * rtpbin with the session number, SSRC and payload type respectively as the pad
42 * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The
43 * session number must be specified in the pad name.
45 * If you want the session manager to generate and send RTCP packets, request
46 * the send_rtcp_src_\%u pad with the session number in the pad name. Packet pushed
47 * on this pad contain SR/RR RTCP reports that should be sent to all participants
50 * To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will
51 * automatically create a send_rtp_src_\%u pad. If the session number is not provided,
52 * the pad from the lowest available session will be returned. The session manager will modify the
53 * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
54 * send_rtp_src_\%u pad after updating its internal state.
56 * The session manager needs the clock-rate of the payload types it is handling
57 * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
58 * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
61 * Access to the internal statistics of rtpbin is provided with the
62 * get-internal-session property. This action signal gives access to the
63 * RTPSession object which further provides action signals to retrieve the
64 * internal source and other sources.
66 * #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder,
67 * #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and
68 * #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders
69 * and decoders in order to support SRTP. The encoders must provide the pads
70 * rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for
71 * RTCP. The session number will be used in the pad name. The decoders must provide
72 * rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will
73 * be placed before the #GstRtpSession element, thus they must support SSRC demuxing
76 * #GstRtpBin has signals (#GstRtpBin::request-aux-sender and
77 * #GstRtpBin::request-aux-receiver to dynamically request an element that can be
78 * used to create or merge additional RTP streams. AUX elements are needed to
79 * implement FEC or retransmission (such as RFC 4588). An AUX sender must have one
80 * sink_\%u pad that matches the sessionid in the signal and it should have 1 or
81 * more src_\%u pads. For each src_%\u pad, a session will be made (if needed)
82 * and the pad will be linked to the session send_rtp_sink pad. Each session will
83 * then expose its source pad as send_rtp_src_\%u on #GstRtpBin.
84 * An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
85 * and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
86 * when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
89 * <title>Example pipelines</title>
91 * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
92 * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
93 * ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
95 * gst-launch-1.0 rtpbin name=rtpbin \
96 * v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
97 * rtpbin.send_rtp_src_0 ! udpsink port=5000 \
98 * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
99 * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
100 * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
101 * rtpbin.send_rtp_src_1 ! udpsink port=5002 \
102 * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
103 * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
104 * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
105 * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
106 * and the audio is sent to session 1. Video packets are sent on UDP port 5000
107 * and audio packets on port 5002. The video RTCP packets for session 0 are sent
108 * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
109 * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
110 * is received on port 5007. Since RTCP packets from the sender should be sent
111 * as soon as possible and do not participate in preroll, sync=false and
112 * async=false is configured on udpsink
114 * gst-launch-1.0 -v rtpbin name=rtpbin \
115 * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
116 * port=5000 ! rtpbin.recv_rtp_sink_0 \
117 * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
118 * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
119 * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
120 * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
121 * port=5002 ! rtpbin.recv_rtp_sink_1 \
122 * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
123 * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
124 * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
125 * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
126 * decode and display the video.
127 * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
128 * decode and play the audio.
129 * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
130 * session 1 on port 5003. These packets will be used for session management and
132 * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
143 #include <gst/rtp/gstrtpbuffer.h>
144 #include <gst/rtp/gstrtcpbuffer.h>
146 #include "gstrtpbin.h"
147 #include "rtpsession.h"
148 #include "gstrtpsession.h"
149 #include "gstrtpjitterbuffer.h"
151 #include <gst/glib-compat-private.h>
153 GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
154 #define GST_CAT_DEFAULT gst_rtp_bin_debug
157 static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
158 GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
161 GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
164 static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
165 GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
168 GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
171 static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
172 GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
175 GST_STATIC_CAPS ("application/x-rtp")
179 static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
180 GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
183 GST_STATIC_CAPS ("application/x-rtp")
186 static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
187 GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
190 GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
193 static GstStaticPadTemplate rtpbin_send_rtp_src_template =
194 GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
197 GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
200 #define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
201 #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
203 /* lock to protect dynamic callbacks, like pad-added and new ssrc. */
204 #define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock)
205 #define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock)
207 /* lock for shutdown */
208 #define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
210 if (g_atomic_int_get (&bin->priv->shutdown)) \
212 GST_RTP_BIN_DYN_LOCK (bin); \
213 if (g_atomic_int_get (&bin->priv->shutdown)) { \
214 GST_RTP_BIN_DYN_UNLOCK (bin); \
219 /* unlock for shutdown */
220 #define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
221 GST_RTP_BIN_DYN_UNLOCK (bin); \
223 /* Minimum time offset to apply. This compensates for rounding errors in NTP to
224 * RTP timestamp conversions */
225 #define MIN_TS_OFFSET (4 * GST_MSECOND)
227 struct _GstRtpBinPrivate
231 /* lock protecting dynamic adding/removing */
234 /* if we are shutting down or not */
239 /* NTP time in ns of last SR sync used */
240 guint64 last_ntpnstime;
242 /* list of extra elements */
246 /* signals and args */
249 SIGNAL_REQUEST_PT_MAP,
250 SIGNAL_PAYLOAD_TYPE_CHANGE,
254 SIGNAL_GET_INTERNAL_SESSION,
256 SIGNAL_GET_INTERNAL_STORAGE,
259 SIGNAL_ON_SSRC_COLLISION,
260 SIGNAL_ON_SSRC_VALIDATED,
261 SIGNAL_ON_SSRC_ACTIVE,
264 SIGNAL_ON_BYE_TIMEOUT,
266 SIGNAL_ON_SENDER_TIMEOUT,
269 SIGNAL_REQUEST_RTP_ENCODER,
270 SIGNAL_REQUEST_RTP_DECODER,
271 SIGNAL_REQUEST_RTCP_ENCODER,
272 SIGNAL_REQUEST_RTCP_DECODER,
274 SIGNAL_REQUEST_FEC_DECODER,
275 SIGNAL_REQUEST_FEC_ENCODER,
277 SIGNAL_NEW_JITTERBUFFER,
280 SIGNAL_REQUEST_AUX_SENDER,
281 SIGNAL_REQUEST_AUX_RECEIVER,
283 SIGNAL_ON_NEW_SENDER_SSRC,
284 SIGNAL_ON_SENDER_SSRC_ACTIVE,
286 SIGNAL_ON_BUNDLED_SSRC,
291 #define DEFAULT_LATENCY_MS 200
292 #define DEFAULT_DROP_ON_LATENCY FALSE
293 #define DEFAULT_SDES NULL
294 #define DEFAULT_DO_LOST FALSE
295 #define DEFAULT_IGNORE_PT FALSE
296 #define DEFAULT_NTP_SYNC FALSE
297 #define DEFAULT_AUTOREMOVE FALSE
298 #define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
299 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
300 #define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
301 #define DEFAULT_RTCP_SYNC_INTERVAL 0
302 #define DEFAULT_DO_SYNC_EVENT FALSE
303 #define DEFAULT_DO_RETRANSMISSION FALSE
304 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
305 #define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
306 #define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
307 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
308 #define DEFAULT_MAX_DROPOUT_TIME 60000
309 #define DEFAULT_MAX_MISORDER_TIME 2000
310 #define DEFAULT_RFC7273_SYNC FALSE
311 #define DEFAULT_MAX_STREAMS G_MAXUINT
312 #define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
313 #define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
319 PROP_DROP_ON_LATENCY,
325 PROP_RTCP_SYNC_INTERVAL,
328 PROP_USE_PIPELINE_CLOCK,
330 PROP_DO_RETRANSMISSION,
332 PROP_NTP_TIME_SOURCE,
333 PROP_RTCP_SYNC_SEND_TIME,
334 PROP_MAX_RTCP_RTP_TIME_DIFF,
335 PROP_MAX_DROPOUT_TIME,
336 PROP_MAX_MISORDER_TIME,
339 PROP_MAX_TS_OFFSET_ADJUSTMENT,
343 #define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
345 gst_rtp_bin_rtcp_sync_get_type (void)
347 static GType rtcp_sync_type = 0;
348 static const GEnumValue rtcp_sync_types[] = {
349 {GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
350 {GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
351 {GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
355 if (!rtcp_sync_type) {
356 rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
358 return rtcp_sync_type;
362 typedef struct _GstRtpBinSession GstRtpBinSession;
363 typedef struct _GstRtpBinStream GstRtpBinStream;
364 typedef struct _GstRtpBinClient GstRtpBinClient;
366 static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
368 static GstCaps *pt_map_requested (GstElement * element, guint pt,
369 GstRtpBinSession * session);
370 static void payload_type_change (GstElement * element, guint pt,
371 GstRtpBinSession * session);
372 static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
373 static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
374 static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
375 static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
376 static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
377 static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
378 static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
379 static GstPad *complete_session_sink (GstRtpBin * rtpbin,
380 GstRtpBinSession * session);
382 complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
384 static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
385 GstRtpBinSession * session, guint sessid);
387 /* Manages the RTP stream for one SSRC.
389 * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
390 * If we see an SDES RTCP packet that links multiple SSRCs together based on a
391 * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
392 * together (see below).
394 struct _GstRtpBinStream
396 /* the SSRC of this stream */
402 /* the session this SSRC belongs to */
403 GstRtpBinSession *session;
405 /* the jitterbuffer of the SSRC */
407 gulong buffer_handlesync_sig;
408 gulong buffer_ptreq_sig;
409 gulong buffer_ntpstop_sig;
412 /* the PT demuxer of the SSRC */
414 gulong demux_newpad_sig;
415 gulong demux_padremoved_sig;
416 gulong demux_ptreq_sig;
417 gulong demux_ptchange_sig;
419 /* if we have calculated a valid rt_delta for this stream */
421 /* mapping to local RTP and NTP time */
424 /* base rtptime in gst time */
428 #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
429 #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
431 /* Manages the receiving end of the packets.
433 * There is one such structure for each RTP session (audio/video/...).
434 * We get the RTP/RTCP packets and stuff them into the session manager. From
435 * there they are pushed into an SSRC demuxer that splits the stream based on
436 * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
437 * the GstRtpBinStream above).
439 * Before the SSRC demuxer, a storage element may be inserted for the purpose
440 * of Forward Error Correction.
442 struct _GstRtpBinSession
448 /* the session element */
450 /* the SSRC demuxer */
452 gulong demux_newpad_sig;
453 gulong demux_padremoved_sig;
460 /* list of GstRtpBinStream */
463 /* list of elements */
466 /* mapping of payload type to caps */
469 /* the pads of the session */
470 GstPad *recv_rtp_sink;
471 GstPad *recv_rtp_sink_ghost;
472 GstPad *recv_rtp_src;
473 GstPad *recv_rtcp_sink;
474 GstPad *recv_rtcp_sink_ghost;
476 GstPad *send_rtp_sink;
477 GstPad *send_rtp_sink_ghost;
478 GstPad *send_rtp_src_ghost;
479 GstPad *send_rtcp_src;
480 GstPad *send_rtcp_src_ghost;
483 /* Manages the RTP streams that come from one client and should therefore be
486 struct _GstRtpBinClient
488 /* the common CNAME for the streams */
497 /* find a session with the given id. Must be called with RTP_BIN_LOCK */
498 static GstRtpBinSession *
499 find_session_by_id (GstRtpBin * rtpbin, gint id)
503 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
504 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
512 /* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
513 static GstRtpBinSession *
514 find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
518 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
519 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
521 if ((sess->recv_rtp_sink_ghost == pad) ||
522 (sess->recv_rtcp_sink_ghost == pad) ||
523 (sess->send_rtp_sink_ghost == pad)
524 || (sess->send_rtcp_src_ghost == pad))
531 on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
533 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
538 on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
540 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
545 on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
547 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
552 on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
554 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
559 on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
561 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
566 on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
568 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
573 on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
575 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
578 if (sess->bin->priv->autoremove)
579 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
583 on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
585 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
588 if (sess->bin->priv->autoremove)
589 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
593 on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
595 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
600 on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
602 g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
603 stream->session->id, stream->ssrc);
607 on_new_sender_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
609 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
614 on_sender_ssrc_active (GstElement * session, guint32 ssrc,
615 GstRtpBinSession * sess)
617 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE],
621 /* must be called with the SESSION lock */
622 static GstRtpBinStream *
623 find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
627 for (walk = session->streams; walk; walk = g_slist_next (walk)) {
628 GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
630 if (stream->ssrc == ssrc)
637 ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
638 GstRtpBinSession * session)
640 GstRtpBinStream *stream = NULL;
643 rtpbin = session->bin;
645 GST_RTP_BIN_LOCK (rtpbin);
647 GST_RTP_SESSION_LOCK (session);
648 if ((stream = find_stream_by_ssrc (session, ssrc)))
649 session->streams = g_slist_remove (session->streams, stream);
650 GST_RTP_SESSION_UNLOCK (session);
653 free_stream (stream, rtpbin);
655 GST_RTP_BIN_UNLOCK (rtpbin);
658 /* create a session with the given id. Must be called with RTP_BIN_LOCK */
659 static GstRtpBinSession *
660 create_session (GstRtpBin * rtpbin, gint id)
662 GstRtpBinSession *sess;
663 GstElement *session, *demux;
664 GstElement *storage = NULL;
667 if (!(session = gst_element_factory_make ("rtpsession", NULL)))
670 if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
673 if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
676 /* need to sink the storage or otherwise signal handlers from bindings will
677 * take ownership of it and we don't own it anymore */
678 gst_object_ref_sink (storage);
679 g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
682 sess = g_new0 (GstRtpBinSession, 1);
683 g_mutex_init (&sess->lock);
686 sess->session = session;
688 sess->storage = storage;
690 sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
691 (GDestroyNotify) gst_caps_unref);
692 rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
694 /* configure SDES items */
695 GST_OBJECT_LOCK (rtpbin);
696 g_object_set (session, "sdes", rtpbin->sdes, "rtp-profile",
697 rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
699 if (rtpbin->use_pipeline_clock)
700 g_object_set (session, "use-pipeline-clock", rtpbin->use_pipeline_clock,
703 g_object_set (session, "ntp-time-source", rtpbin->ntp_time_source, NULL);
705 g_object_set (session, "max-dropout-time", rtpbin->max_dropout_time,
706 "max-misorder-time", rtpbin->max_misorder_time, NULL);
707 GST_OBJECT_UNLOCK (rtpbin);
709 /* provide clock_rate to the session manager when needed */
710 g_signal_connect (session, "request-pt-map",
711 (GCallback) pt_map_requested, sess);
713 g_signal_connect (sess->session, "on-new-ssrc",
714 (GCallback) on_new_ssrc, sess);
715 g_signal_connect (sess->session, "on-ssrc-collision",
716 (GCallback) on_ssrc_collision, sess);
717 g_signal_connect (sess->session, "on-ssrc-validated",
718 (GCallback) on_ssrc_validated, sess);
719 g_signal_connect (sess->session, "on-ssrc-active",
720 (GCallback) on_ssrc_active, sess);
721 g_signal_connect (sess->session, "on-ssrc-sdes",
722 (GCallback) on_ssrc_sdes, sess);
723 g_signal_connect (sess->session, "on-bye-ssrc",
724 (GCallback) on_bye_ssrc, sess);
725 g_signal_connect (sess->session, "on-bye-timeout",
726 (GCallback) on_bye_timeout, sess);
727 g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
728 g_signal_connect (sess->session, "on-sender-timeout",
729 (GCallback) on_sender_timeout, sess);
730 g_signal_connect (sess->session, "on-new-sender-ssrc",
731 (GCallback) on_new_sender_ssrc, sess);
732 g_signal_connect (sess->session, "on-sender-ssrc-active",
733 (GCallback) on_sender_ssrc_active, sess);
735 gst_bin_add (GST_BIN_CAST (rtpbin), session);
736 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
737 gst_bin_add (GST_BIN_CAST (rtpbin), storage);
739 /* unref the storage again, the bin has a reference now and
740 * we don't need it anymore */
741 gst_object_unref (storage);
743 GST_OBJECT_LOCK (rtpbin);
744 target = GST_STATE_TARGET (rtpbin);
745 GST_OBJECT_UNLOCK (rtpbin);
747 /* change state only to what's needed */
748 gst_element_set_state (demux, target);
749 gst_element_set_state (session, target);
750 gst_element_set_state (storage, target);
757 g_warning ("rtpbin: could not create rtpsession element");
762 gst_object_unref (session);
763 g_warning ("rtpbin: could not create rtpssrcdemux element");
768 gst_object_unref (session);
769 gst_object_unref (demux);
770 g_warning ("rtpbin: could not create rtpstorage element");
776 bin_manage_element (GstRtpBin * bin, GstElement * element)
778 GstRtpBinPrivate *priv = bin->priv;
780 if (g_list_find (priv->elements, element)) {
781 GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
783 GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
785 if (g_object_is_floating (element))
786 element = gst_object_ref_sink (element);
788 if (!gst_bin_add (GST_BIN_CAST (bin), element))
790 if (!gst_element_sync_state_with_parent (element))
791 GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin");
793 /* we add the element multiple times, each we need an equal number of
794 * removes to really remove the element from the bin */
795 priv->elements = g_list_prepend (priv->elements, element);
802 GST_WARNING_OBJECT (bin, "unable to add element");
803 gst_object_unref (element);
809 remove_bin_element (GstElement * element, GstRtpBin * bin)
811 GstRtpBinPrivate *priv = bin->priv;
814 find = g_list_find (priv->elements, element);
816 priv->elements = g_list_delete_link (priv->elements, find);
818 if (!g_list_find (priv->elements, element)) {
819 gst_element_set_locked_state (element, TRUE);
820 gst_bin_remove (GST_BIN_CAST (bin), element);
821 gst_element_set_state (element, GST_STATE_NULL);
824 gst_object_unref (element);
828 /* called with RTP_BIN_LOCK */
830 free_session (GstRtpBinSession * sess, GstRtpBin * bin)
832 GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
834 gst_element_set_locked_state (sess->demux, TRUE);
835 gst_element_set_locked_state (sess->session, TRUE);
836 gst_element_set_locked_state (sess->storage, TRUE);
838 gst_element_set_state (sess->demux, GST_STATE_NULL);
839 gst_element_set_state (sess->session, GST_STATE_NULL);
840 gst_element_set_state (sess->storage, GST_STATE_NULL);
842 remove_recv_rtp (bin, sess);
843 remove_recv_rtcp (bin, sess);
844 remove_send_rtp (bin, sess);
845 remove_rtcp (bin, sess);
847 gst_bin_remove (GST_BIN_CAST (bin), sess->session);
848 gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
849 gst_bin_remove (GST_BIN_CAST (bin), sess->storage);
851 g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
852 g_slist_free (sess->elements);
854 g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
855 g_slist_free (sess->streams);
857 g_mutex_clear (&sess->lock);
858 g_hash_table_destroy (sess->ptmap);
863 /* get the payload type caps for the specific payload @pt in @session */
865 get_pt_map (GstRtpBinSession * session, guint pt)
867 GstCaps *caps = NULL;
870 GValue args[3] = { {0}, {0}, {0} };
872 GST_DEBUG ("searching pt %u in cache", pt);
874 GST_RTP_SESSION_LOCK (session);
876 /* first look in the cache */
877 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
885 GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
887 /* not in cache, send signal to request caps */
888 g_value_init (&args[0], GST_TYPE_ELEMENT);
889 g_value_set_object (&args[0], bin);
890 g_value_init (&args[1], G_TYPE_UINT);
891 g_value_set_uint (&args[1], session->id);
892 g_value_init (&args[2], G_TYPE_UINT);
893 g_value_set_uint (&args[2], pt);
895 g_value_init (&ret, GST_TYPE_CAPS);
896 g_value_set_boxed (&ret, NULL);
898 GST_RTP_SESSION_UNLOCK (session);
900 g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
902 GST_RTP_SESSION_LOCK (session);
904 g_value_unset (&args[0]);
905 g_value_unset (&args[1]);
906 g_value_unset (&args[2]);
908 /* look in the cache again because we let the lock go */
909 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
912 g_value_unset (&ret);
916 caps = (GstCaps *) g_value_dup_boxed (&ret);
917 g_value_unset (&ret);
921 GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
923 /* store in cache, take additional ref */
924 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
925 gst_caps_ref (caps));
928 GST_RTP_SESSION_UNLOCK (session);
935 GST_RTP_SESSION_UNLOCK (session);
936 GST_DEBUG ("no pt map could be obtained");
942 return_true (gpointer key, gpointer value, gpointer user_data)
948 gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
950 GSList *clients, *streams;
952 GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
954 GST_RTP_BIN_LOCK (rtpbin);
955 for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
956 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
958 /* reset sync on all streams for this client */
959 for (streams = client->streams; streams; streams = g_slist_next (streams)) {
960 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
962 /* make use require a new SR packet for this stream before we attempt new
964 stream->have_sync = FALSE;
965 stream->rt_delta = 0;
966 stream->rtp_delta = 0;
967 stream->clock_base = -100 * GST_SECOND;
970 GST_RTP_BIN_UNLOCK (rtpbin);
974 gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
976 GSList *sessions, *streams;
978 GST_RTP_BIN_LOCK (bin);
979 GST_DEBUG_OBJECT (bin, "clearing pt map");
980 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
981 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
983 GST_DEBUG_OBJECT (bin, "clearing session %p", session);
984 g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
986 GST_RTP_SESSION_LOCK (session);
987 g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
989 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
990 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
992 GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
993 g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
995 g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
997 GST_RTP_SESSION_UNLOCK (session);
999 GST_RTP_BIN_UNLOCK (bin);
1001 /* reset sync too */
1002 gst_rtp_bin_reset_sync (bin);
1006 gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
1008 GstRtpBinSession *session;
1009 GstElement *ret = NULL;
1011 GST_RTP_BIN_LOCK (bin);
1012 GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
1013 session = find_session_by_id (bin, (gint) session_id);
1015 ret = gst_object_ref (session->session);
1017 GST_RTP_BIN_UNLOCK (bin);
1023 gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
1025 RTPSession *internal_session = NULL;
1026 GstRtpBinSession *session;
1028 GST_RTP_BIN_LOCK (bin);
1029 GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
1031 session = find_session_by_id (bin, (gint) session_id);
1033 g_object_get (session->session, "internal-session", &internal_session,
1036 GST_RTP_BIN_UNLOCK (bin);
1038 return internal_session;
1042 gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
1044 GstRtpBinSession *session;
1045 GstElement *res = NULL;
1047 GST_RTP_BIN_LOCK (bin);
1048 GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
1050 session = find_session_by_id (bin, (gint) session_id);
1051 if (session && session->storage) {
1052 res = gst_object_ref (session->storage);
1054 GST_RTP_BIN_UNLOCK (bin);
1060 gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
1062 GObject *internal_storage = NULL;
1063 GstRtpBinSession *session;
1065 GST_RTP_BIN_LOCK (bin);
1066 GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
1068 session = find_session_by_id (bin, (gint) session_id);
1069 if (session && session->storage) {
1070 g_object_get (session->storage, "internal-storage", &internal_storage,
1073 GST_RTP_BIN_UNLOCK (bin);
1075 return internal_storage;
1079 gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
1081 GST_DEBUG_OBJECT (bin, "return NULL encoder");
1086 gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
1088 GST_DEBUG_OBJECT (bin, "return NULL decoder");
1093 gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
1094 const gchar * name, const GValue * value)
1096 GSList *sessions, *streams;
1098 GST_RTP_BIN_LOCK (bin);
1099 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
1100 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
1102 GST_RTP_SESSION_LOCK (session);
1103 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
1104 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
1106 g_object_set_property (G_OBJECT (stream->buffer), name, value);
1108 GST_RTP_SESSION_UNLOCK (session);
1110 GST_RTP_BIN_UNLOCK (bin);
1114 gst_rtp_bin_propagate_property_to_session (GstRtpBin * bin,
1115 const gchar * name, const GValue * value)
1119 GST_RTP_BIN_LOCK (bin);
1120 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
1121 GstRtpBinSession *sess = (GstRtpBinSession *) sessions->data;
1123 g_object_set_property (G_OBJECT (sess->session), name, value);
1125 GST_RTP_BIN_UNLOCK (bin);
1128 /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
1129 static GstRtpBinClient *
1130 get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
1132 GstRtpBinClient *result = NULL;
1135 for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
1136 GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
1138 if (len != client->cname_len)
1141 if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
1142 GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
1149 /* nothing found, create one */
1150 if (result == NULL) {
1151 result = g_new0 (GstRtpBinClient, 1);
1152 result->cname = g_strndup ((gchar *) data, len);
1153 result->cname_len = len;
1154 bin->clients = g_slist_prepend (bin->clients, result);
1155 GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
1162 free_client (GstRtpBinClient * client, GstRtpBin * bin)
1164 GST_DEBUG_OBJECT (bin, "freeing client %p", client);
1165 g_slist_free (client->streams);
1166 g_free (client->cname);
1171 get_current_times (GstRtpBin * bin, GstClockTime * running_time,
1172 guint64 * ntpnstime)
1176 GstClockTime base_time, rt, clock_time;
1178 GST_OBJECT_LOCK (bin);
1179 if ((clock = GST_ELEMENT_CLOCK (bin))) {
1180 base_time = GST_ELEMENT_CAST (bin)->base_time;
1181 gst_object_ref (clock);
1182 GST_OBJECT_UNLOCK (bin);
1184 /* get current clock time and convert to running time */
1185 clock_time = gst_clock_get_time (clock);
1186 rt = clock_time - base_time;
1188 if (bin->use_pipeline_clock) {
1190 /* add constant to convert from 1970 based time to 1900 based time */
1191 ntpns += (2208988800LL * GST_SECOND);
1193 switch (bin->ntp_time_source) {
1194 case GST_RTP_NTP_TIME_SOURCE_NTP:
1195 case GST_RTP_NTP_TIME_SOURCE_UNIX:{
1198 /* get current NTP time */
1199 g_get_current_time (¤t);
1200 ntpns = GST_TIMEVAL_TO_TIME (current);
1202 /* add constant to convert from 1970 based time to 1900 based time */
1203 if (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
1204 ntpns += (2208988800LL * GST_SECOND);
1207 case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
1210 case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
1214 ntpns = -1; /* Fix uninited compiler warning */
1215 g_assert_not_reached ();
1220 gst_object_unref (clock);
1222 GST_OBJECT_UNLOCK (bin);
1233 stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
1234 gint64 ts_offset, gint64 max_ts_offset, gint64 min_ts_offset,
1235 gboolean allow_positive_ts_offset)
1237 gint64 prev_ts_offset;
1239 g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
1241 /* delta changed, see how much */
1242 if (prev_ts_offset != ts_offset) {
1245 diff = prev_ts_offset - ts_offset;
1247 GST_DEBUG_OBJECT (bin,
1248 "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
1249 ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
1251 /* ignore minor offsets */
1252 if (ABS (diff) < min_ts_offset) {
1253 GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
1257 /* sanity check offset */
1258 if (max_ts_offset > 0) {
1259 if (ts_offset > 0 && !allow_positive_ts_offset) {
1260 GST_DEBUG_OBJECT (bin,
1261 "offset is positive (clocks are out of sync), ignoring");
1264 if (ABS (ts_offset) > max_ts_offset) {
1265 GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
1270 g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
1272 GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
1273 stream->ssrc, ts_offset);
1277 gst_rtp_bin_send_sync_event (GstRtpBinStream * stream)
1279 if (stream->bin->send_sync_event) {
1283 GST_DEBUG_OBJECT (stream->bin,
1284 "sending GstRTCPSRReceived event downstream");
1286 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1287 gst_structure_new_empty ("GstRTCPSRReceived"));
1289 srcpad = gst_element_get_static_pad (stream->buffer, "src");
1290 gst_pad_push_event (srcpad, event);
1291 gst_object_unref (srcpad);
1295 /* associate a stream to the given CNAME. This will make sure all streams for
1296 * that CNAME are synchronized together.
1297 * Must be called with GST_RTP_BIN_LOCK */
1299 gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
1300 guint8 * data, guint64 ntptime, guint64 last_extrtptime,
1301 guint64 base_rtptime, guint64 base_time, guint clock_rate,
1302 gint64 rtp_clock_base)
1304 GstRtpBinClient *client;
1307 GstClockTime running_time, running_time_rtp;
1310 /* first find or create the CNAME */
1311 client = get_client (bin, len, data, &created);
1313 /* find stream in the client */
1314 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1315 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1317 if (ostream == stream)
1320 /* not found, add it to the list */
1322 GST_DEBUG_OBJECT (bin,
1323 "new association of SSRC %08x with client %p with CNAME %s",
1324 stream->ssrc, client, client->cname);
1325 client->streams = g_slist_prepend (client->streams, stream);
1328 GST_DEBUG_OBJECT (bin,
1329 "found association of SSRC %08x with client %p with CNAME %s",
1330 stream->ssrc, client, client->cname);
1333 if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
1334 GST_DEBUG_OBJECT (bin, "invalidated sync data");
1335 if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1336 /* we don't need that data, so carry on,
1337 * but make some values look saner */
1338 last_extrtptime = base_rtptime;
1340 /* nothing we can do with this data in this case */
1341 GST_DEBUG_OBJECT (bin, "bailing out");
1346 /* Take the extended rtptime we found in the SR packet and map it to the
1347 * local rtptime. The local rtp time is used to construct timestamps on the
1348 * buffers so we will calculate what running_time corresponds to the RTP
1349 * timestamp in the SR packet. */
1350 running_time_rtp = last_extrtptime - base_rtptime;
1352 GST_DEBUG_OBJECT (bin,
1353 "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
1354 ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
1355 "clock-base %" G_GINT64_FORMAT, base_rtptime,
1356 last_extrtptime, running_time_rtp, clock_rate, rtp_clock_base);
1358 /* calculate local RTP time in gstreamer timestamp, we essentially perform the
1359 * same conversion that a jitterbuffer would use to convert an rtp timestamp
1360 * into a corresponding gstreamer timestamp. Note that the base_time also
1361 * contains the drift between sender and receiver. */
1363 gst_util_uint64_scale_int (running_time_rtp, GST_SECOND, clock_rate);
1364 running_time += base_time;
1366 /* convert ntptime to nanoseconds */
1367 ntpnstime = gst_util_uint64_scale (ntptime, GST_SECOND,
1368 (G_GINT64_CONSTANT (1) << 32));
1370 stream->have_sync = TRUE;
1372 GST_DEBUG_OBJECT (bin,
1373 "SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
1374 running_time, ntpnstime);
1376 /* recalc inter stream playout offset, but only if there is more than one
1377 * stream or we're doing NTP sync. */
1378 if (bin->ntp_sync) {
1379 gint64 ntpdiff, rtdiff;
1380 guint64 local_ntpnstime;
1381 GstClockTime local_running_time;
1383 /* For NTP sync we need to first get a snapshot of running_time and NTP
1384 * time. We know at what running_time we play a certain RTP time, we also
1385 * calculated when we would play the RTP time in the SR packet. Now we need
1386 * to know how the running_time and the NTP time relate to eachother. */
1387 get_current_times (bin, &local_running_time, &local_ntpnstime);
1389 /* see how far away the NTP time is. This is the difference between the
1390 * current NTP time and the NTP time in the last SR packet. */
1391 ntpdiff = local_ntpnstime - ntpnstime;
1392 /* see how far away the running_time is. This is the difference between the
1393 * current running_time and the running_time of the RTP timestamp in the
1394 * last SR packet. */
1395 rtdiff = local_running_time - running_time;
1397 GST_DEBUG_OBJECT (bin,
1398 "local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
1399 local_ntpnstime, ntpnstime);
1400 GST_DEBUG_OBJECT (bin,
1401 "local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
1402 G_GUINT64_FORMAT, local_running_time, running_time);
1403 GST_DEBUG_OBJECT (bin,
1404 "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
1407 /* combine to get the final diff to apply to the running_time */
1408 stream->rt_delta = rtdiff - ntpdiff;
1410 stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
1413 gint64 min, rtp_min, clock_base = stream->clock_base;
1414 gboolean all_sync, use_rtp;
1415 gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
1417 /* calculate delta between server and receiver. ntpnstime is created by
1418 * converting the ntptime in the last SR packet to a gstreamer timestamp. This
1419 * delta expresses the difference to our timeline and the server timeline. The
1420 * difference in itself doesn't mean much but we can combine the delta of
1421 * multiple streams to create a stream specific offset. */
1422 stream->rt_delta = ntpnstime - running_time;
1424 /* calculate the min of all deltas, ignoring streams that did not yet have a
1425 * valid rt_delta because we did not yet receive an SR packet for those
1427 * We calculate the mininum because we would like to only apply positive
1428 * offsets to streams, delaying their playback instead of trying to speed up
1429 * other streams (which might be imposible when we have to create negative
1431 * The stream that has the smallest diff is selected as the reference stream,
1432 * all other streams will have a positive offset to this difference. */
1434 /* some alternative setting allow ignoring RTCP as much as possible,
1435 * for servers generating bogus ntp timeline */
1436 min = rtp_min = G_MAXINT64;
1438 if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1442 /* signed version for convienience */
1443 clock_base = base_rtptime;
1444 /* deal with possible wrap-around */
1445 ext_base = base_rtptime;
1446 rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
1447 /* sanity check; base rtp and provided clock_base should be close */
1448 if (rtp_clock_base >= clock_base) {
1449 if (rtp_clock_base - clock_base < 10 * clock_rate) {
1450 rtp_clock_base = base_time +
1451 gst_util_uint64_scale_int (rtp_clock_base - clock_base,
1452 GST_SECOND, clock_rate);
1457 if (clock_base - rtp_clock_base < 10 * clock_rate) {
1458 rtp_clock_base = base_time -
1459 gst_util_uint64_scale_int (clock_base - rtp_clock_base,
1460 GST_SECOND, clock_rate);
1465 /* warn and bail for clarity out if no sane values */
1467 GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
1470 /* store to track changes */
1471 clock_base = rtp_clock_base;
1472 /* generate a fake as before,
1473 * now equating rtptime obtained from RTP-Info,
1474 * where the large time represent the otherwise irrelevant npt/ntp time */
1475 stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
1477 clock_base = rtp_clock_base;
1481 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1482 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1484 if (!ostream->have_sync) {
1489 /* change in current stream's base from previously init'ed value
1490 * leads to reset of all stream's base */
1491 if (stream != ostream && stream->clock_base >= 0 &&
1492 (stream->clock_base != clock_base)) {
1493 GST_DEBUG_OBJECT (bin, "reset upon clock base change");
1494 ostream->clock_base = -100 * GST_SECOND;
1495 ostream->rtp_delta = 0;
1498 if (ostream->rt_delta < min)
1499 min = ostream->rt_delta;
1500 if (ostream->rtp_delta < rtp_min)
1501 rtp_min = ostream->rtp_delta;
1504 /* arrange to re-sync for each stream upon significant change,
1506 all_sync = all_sync && (stream->clock_base == clock_base);
1507 stream->clock_base = clock_base;
1509 /* may need init performed above later on, but nothing more to do now */
1510 if (client->nstreams <= 1)
1513 GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
1514 " all sync %d", client, min, all_sync);
1515 GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
1517 switch (rtcp_sync) {
1518 case GST_RTP_BIN_RTCP_SYNC_RTP:
1521 GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
1522 "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
1524 case GST_RTP_BIN_RTCP_SYNC_INITIAL:
1525 /* if all have been synced already, do not bother further */
1527 GST_DEBUG_OBJECT (bin, "all streams already synced; done");
1535 /* bail out if we adjusted recently enough */
1536 if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
1537 bin->rtcp_sync_interval * GST_MSECOND) {
1538 GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
1539 "previous sender info too recent "
1540 "(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
1543 bin->priv->last_ntpnstime = ntpnstime;
1545 /* calculate offsets for each stream */
1546 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1547 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1550 /* ignore streams for which we didn't receive an SR packet yet, we
1551 * can't synchronize them yet. We can however sync other streams just
1553 if (!ostream->have_sync)
1556 /* calculate offset to our reference stream, this should always give a
1557 * positive number. */
1559 ts_offset = ostream->rtp_delta - rtp_min;
1561 ts_offset = ostream->rt_delta - min;
1563 stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
1564 MIN_TS_OFFSET, TRUE);
1567 gst_rtp_bin_send_sync_event (stream);
1572 #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
1573 for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
1574 (b) = gst_rtcp_packet_move_to_next ((packet)))
1576 #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
1577 for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
1578 (b) = gst_rtcp_packet_sdes_next_item ((packet)))
1580 #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
1581 for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
1582 (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
1585 gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
1586 GstRtpBinStream * stream)
1589 GstRTCPPacket packet;
1592 gboolean have_sr, have_sdes;
1594 guint64 base_rtptime;
1600 GstRTCPBuffer rtcp = { NULL, };
1604 GST_DEBUG_OBJECT (bin, "sync handler called");
1606 /* get the last relation between the rtp timestamps and the gstreamer
1607 * timestamps. We get this info directly from the jitterbuffer which
1608 * constructs gstreamer timestamps from rtp timestamps and so it know exactly
1609 * what the current situation is. */
1611 g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
1612 base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
1613 clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
1614 clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
1616 g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
1617 buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
1622 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
1624 GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
1625 /* first packet must be SR or RR or else the validate would have failed */
1626 switch (gst_rtcp_packet_get_type (&packet)) {
1627 case GST_RTCP_TYPE_SR:
1628 /* only parse first. There is only supposed to be one SR in the packet
1629 * but we will deal with malformed packets gracefully */
1632 /* get NTP and RTP times */
1633 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
1636 GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
1637 /* ignore SR that is not ours */
1638 if (ssrc != stream->ssrc)
1643 case GST_RTCP_TYPE_SDES:
1645 gboolean more_items, more_entries;
1647 /* only deal with first SDES, there is only supposed to be one SDES in
1648 * the RTCP packet but we deal with bad packets gracefully. Also bail
1649 * out if we have not seen an SR item yet. */
1650 if (have_sdes || !have_sr)
1653 GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
1654 /* skip items that are not about the SSRC of the sender */
1655 if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
1658 /* find the CNAME entry */
1659 GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
1660 GstRTCPSDESType type;
1664 gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
1666 if (type == GST_RTCP_SDES_CNAME) {
1667 GST_RTP_BIN_LOCK (bin);
1668 /* associate the stream to CNAME */
1669 gst_rtp_bin_associate (bin, stream, len, data,
1670 ntptime, extrtptime, base_rtptime, base_time, clock_rate,
1672 GST_RTP_BIN_UNLOCK (bin);
1680 /* we can ignore these packets */
1684 gst_rtcp_buffer_unmap (&rtcp);
1687 /* create a new stream with @ssrc in @session. Must be called with
1688 * RTP_SESSION_LOCK. */
1689 static GstRtpBinStream *
1690 create_stream (GstRtpBinSession * session, guint32 ssrc)
1692 GstElement *buffer, *demux = NULL;
1693 GstRtpBinStream *stream;
1697 rtpbin = session->bin;
1699 if (g_slist_length (session->streams) >= rtpbin->max_streams)
1702 if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
1703 goto no_jitterbuffer;
1705 if (!rtpbin->ignore_pt) {
1706 if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
1710 stream = g_new0 (GstRtpBinStream, 1);
1711 stream->ssrc = ssrc;
1712 stream->bin = rtpbin;
1713 stream->session = session;
1714 stream->buffer = buffer;
1715 stream->demux = demux;
1717 stream->have_sync = FALSE;
1718 stream->rt_delta = 0;
1719 stream->rtp_delta = 0;
1720 stream->percent = 100;
1721 stream->clock_base = -100 * GST_SECOND;
1722 session->streams = g_slist_prepend (session->streams, stream);
1724 /* provide clock_rate to the jitterbuffer when needed */
1725 stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
1726 (GCallback) pt_map_requested, session);
1727 stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
1728 (GCallback) on_npt_stop, stream);
1730 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
1731 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
1733 /* configure latency and packet lost */
1734 g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
1735 g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
1736 g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
1737 g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
1738 g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
1739 g_object_set (buffer, "max-rtcp-rtp-time-diff",
1740 rtpbin->max_rtcp_rtp_time_diff, NULL);
1741 g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
1742 "max-misorder-time", rtpbin->max_misorder_time, NULL);
1743 g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
1744 g_object_set (buffer, "max-ts-offset-adjustment",
1745 rtpbin->max_ts_offset_adjustment, NULL);
1747 /* need to sink the jitterbufer or otherwise signal handlers from bindings will
1748 * take ownership of it and we don't own it anymore */
1749 gst_object_ref_sink (buffer);
1750 g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
1751 buffer, session->id, ssrc);
1753 if (!rtpbin->ignore_pt)
1754 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
1755 gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
1757 /* unref the jitterbuffer again, the bin has a reference now and
1758 * we don't need it anymore */
1759 gst_object_unref (buffer);
1763 gst_element_link_pads_full (buffer, "src", demux, "sink",
1764 GST_PAD_LINK_CHECK_NOTHING);
1766 if (rtpbin->buffering) {
1769 GST_INFO_OBJECT (rtpbin,
1770 "bin is buffering, set jitterbuffer as not active");
1771 g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
1775 GST_OBJECT_LOCK (rtpbin);
1776 target = GST_STATE_TARGET (rtpbin);
1777 GST_OBJECT_UNLOCK (rtpbin);
1779 /* from sink to source */
1781 gst_element_set_state (demux, target);
1783 gst_element_set_state (buffer, target);
1790 GST_WARNING_OBJECT (rtpbin, "stream exeeds maximum (%d)",
1791 rtpbin->max_streams);
1796 g_warning ("rtpbin: could not create rtpjitterbuffer element");
1801 gst_object_unref (buffer);
1802 g_warning ("rtpbin: could not create rtpptdemux element");
1807 /* called with RTP_BIN_LOCK */
1809 free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
1811 GSList *clients, *next_client;
1813 GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
1815 if (stream->demux) {
1816 g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
1817 g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
1818 g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
1820 g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
1821 g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
1822 g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
1825 gst_element_set_locked_state (stream->demux, TRUE);
1826 gst_element_set_locked_state (stream->buffer, TRUE);
1829 gst_element_set_state (stream->demux, GST_STATE_NULL);
1830 gst_element_set_state (stream->buffer, GST_STATE_NULL);
1832 /* now remove this signal, we need this while going to NULL because it to
1833 * do some cleanups */
1835 g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
1837 gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
1839 gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
1841 for (clients = bin->clients; clients; clients = next_client) {
1842 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
1843 GSList *streams, *next_stream;
1845 next_client = g_slist_next (clients);
1847 for (streams = client->streams; streams; streams = next_stream) {
1848 GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
1850 next_stream = g_slist_next (streams);
1852 if (ostream == stream) {
1853 client->streams = g_slist_delete_link (client->streams, streams);
1854 /* If this was the last stream belonging to this client,
1855 * clean up the client. */
1856 if (--client->nstreams == 0) {
1857 bin->clients = g_slist_delete_link (bin->clients, clients);
1858 free_client (client, bin);
1867 /* GObject vmethods */
1868 static void gst_rtp_bin_dispose (GObject * object);
1869 static void gst_rtp_bin_finalize (GObject * object);
1870 static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
1871 const GValue * value, GParamSpec * pspec);
1872 static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
1873 GValue * value, GParamSpec * pspec);
1875 /* GstElement vmethods */
1876 static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
1877 GstStateChange transition);
1878 static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
1879 GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
1880 static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
1881 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
1883 #define gst_rtp_bin_parent_class parent_class
1884 G_DEFINE_TYPE_WITH_PRIVATE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
1887 _gst_element_accumulator (GSignalInvocationHint * ihint,
1888 GValue * return_accu, const GValue * handler_return, gpointer dummy)
1890 GstElement *element;
1892 element = g_value_get_object (handler_return);
1893 GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
1895 if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
1896 g_value_set_object (return_accu, element);
1898 /* stop emission if we have an element */
1899 return (element == NULL);
1903 _gst_caps_accumulator (GSignalInvocationHint * ihint,
1904 GValue * return_accu, const GValue * handler_return, gpointer dummy)
1908 caps = g_value_get_boxed (handler_return);
1909 GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
1911 if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
1912 g_value_set_boxed (return_accu, caps);
1914 /* stop emission if we have a caps */
1915 return (caps == NULL);
1919 gst_rtp_bin_class_init (GstRtpBinClass * klass)
1921 GObjectClass *gobject_class;
1922 GstElementClass *gstelement_class;
1923 GstBinClass *gstbin_class;
1925 gobject_class = (GObjectClass *) klass;
1926 gstelement_class = (GstElementClass *) klass;
1927 gstbin_class = (GstBinClass *) klass;
1929 gobject_class->dispose = gst_rtp_bin_dispose;
1930 gobject_class->finalize = gst_rtp_bin_finalize;
1931 gobject_class->set_property = gst_rtp_bin_set_property;
1932 gobject_class->get_property = gst_rtp_bin_get_property;
1934 g_object_class_install_property (gobject_class, PROP_LATENCY,
1935 g_param_spec_uint ("latency", "Buffer latency in ms",
1936 "Default amount of ms to buffer in the jitterbuffers", 0,
1937 G_MAXUINT, DEFAULT_LATENCY_MS,
1938 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1940 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
1941 g_param_spec_boolean ("drop-on-latency",
1942 "Drop buffers when maximum latency is reached",
1943 "Tells the jitterbuffer to never exceed the given latency in size",
1944 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1947 * GstRtpBin::request-pt-map:
1948 * @rtpbin: the object which received the signal
1949 * @session: the session
1952 * Request the payload type as #GstCaps for @pt in @session.
1954 gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
1955 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
1956 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
1957 _gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS,
1958 2, G_TYPE_UINT, G_TYPE_UINT);
1961 * GstRtpBin::payload-type-change:
1962 * @rtpbin: the object which received the signal
1963 * @session: the session
1966 * Signal that the current payload type changed to @pt in @session.
1968 gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
1969 g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
1970 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
1971 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
1975 * GstRtpBin::clear-pt-map:
1976 * @rtpbin: the object which received the signal
1978 * Clear all previously cached pt-mapping obtained with
1979 * #GstRtpBin::request-pt-map.
1981 gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
1982 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
1983 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1984 clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1988 * GstRtpBin::reset-sync:
1989 * @rtpbin: the object which received the signal
1991 * Reset all currently configured lip-sync parameters and require new SR
1992 * packets for all streams before lip-sync is attempted again.
1994 gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
1995 g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
1996 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1997 reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
2001 * GstRtpBin::get-session:
2002 * @rtpbin: the object which received the signal
2003 * @id: the session id
2005 * Request the related GstRtpSession as #GstElement related with session @id.
2009 gst_rtp_bin_signals[SIGNAL_GET_SESSION] =
2010 g_signal_new ("get-session", G_TYPE_FROM_CLASS (klass),
2011 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2012 get_session), NULL, NULL, g_cclosure_marshal_generic,
2013 GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2016 * GstRtpBin::get-internal-session:
2017 * @rtpbin: the object which received the signal
2018 * @id: the session id
2020 * Request the internal RTPSession object as #GObject in session @id.
2022 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
2023 g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
2024 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2025 get_internal_session), NULL, NULL, g_cclosure_marshal_generic,
2026 RTP_TYPE_SESSION, 1, G_TYPE_UINT);
2029 * GstRtpBin::get-internal-storage:
2030 * @rtpbin: the object which received the signal
2031 * @id: the session id
2033 * Request the internal RTPStorage object as #GObject in session @id.
2037 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
2038 g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
2039 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2040 get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
2041 G_TYPE_OBJECT, 1, G_TYPE_UINT);
2044 * GstRtpBin::get-storage:
2045 * @rtpbin: the object which received the signal
2046 * @id: the session id
2048 * Request the RTPStorage element as #GObject in session @id.
2052 gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
2053 g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
2054 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2055 get_storage), NULL, NULL, g_cclosure_marshal_generic,
2056 GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2059 * GstRtpBin::on-new-ssrc:
2060 * @rtpbin: the object which received the signal
2061 * @session: the session
2064 * Notify of a new SSRC that entered @session.
2066 gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
2067 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
2068 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
2069 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2072 * GstRtpBin::on-ssrc-collision:
2073 * @rtpbin: the object which received the signal
2074 * @session: the session
2077 * Notify when we have an SSRC collision
2079 gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
2080 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
2081 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
2082 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2085 * GstRtpBin::on-ssrc-validated:
2086 * @rtpbin: the object which received the signal
2087 * @session: the session
2090 * Notify of a new SSRC that became validated.
2092 gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
2093 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
2094 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
2095 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2098 * GstRtpBin::on-ssrc-active:
2099 * @rtpbin: the object which received the signal
2100 * @session: the session
2103 * Notify of a SSRC that is active, i.e., sending RTCP.
2105 gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
2106 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
2107 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
2108 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2111 * GstRtpBin::on-ssrc-sdes:
2112 * @rtpbin: the object which received the signal
2113 * @session: the session
2116 * Notify of a SSRC that is active, i.e., sending RTCP.
2118 gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
2119 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
2120 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
2121 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2125 * GstRtpBin::on-bye-ssrc:
2126 * @rtpbin: the object which received the signal
2127 * @session: the session
2130 * Notify of an SSRC that became inactive because of a BYE packet.
2132 gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
2133 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
2134 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
2135 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2138 * GstRtpBin::on-bye-timeout:
2139 * @rtpbin: the object which received the signal
2140 * @session: the session
2143 * Notify of an SSRC that has timed out because of BYE
2145 gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
2146 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
2147 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
2148 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2151 * GstRtpBin::on-timeout:
2152 * @rtpbin: the object which received the signal
2153 * @session: the session
2156 * Notify of an SSRC that has timed out
2158 gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
2159 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
2160 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
2161 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2164 * GstRtpBin::on-sender-timeout:
2165 * @rtpbin: the object which received the signal
2166 * @session: the session
2169 * Notify of a sender SSRC that has timed out and became a receiver
2171 gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
2172 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
2173 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
2174 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2178 * GstRtpBin::on-npt-stop:
2179 * @rtpbin: the object which received the signal
2180 * @session: the session
2183 * Notify that SSRC sender has sent data up to the configured NPT stop time.
2185 gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
2186 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
2187 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
2188 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2192 * GstRtpBin::request-rtp-encoder:
2193 * @rtpbin: the object which received the signal
2194 * @session: the session
2196 * Request an RTP encoder element for the given @session. The encoder
2197 * element will be added to the bin if not previously added.
2199 * If no handler is connected, no encoder will be used.
2203 gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] =
2204 g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass),
2205 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2206 request_rtp_encoder), _gst_element_accumulator, NULL,
2207 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2210 * GstRtpBin::request-rtp-decoder:
2211 * @rtpbin: the object which received the signal
2212 * @session: the session
2214 * Request an RTP decoder element for the given @session. The decoder
2215 * element will be added to the bin if not previously added.
2217 * If no handler is connected, no encoder will be used.
2221 gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] =
2222 g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass),
2223 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2224 request_rtp_decoder), _gst_element_accumulator, NULL,
2225 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2228 * GstRtpBin::request-rtcp-encoder:
2229 * @rtpbin: the object which received the signal
2230 * @session: the session
2232 * Request an RTCP encoder element for the given @session. The encoder
2233 * element will be added to the bin if not previously added.
2235 * If no handler is connected, no encoder will be used.
2239 gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] =
2240 g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
2241 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2242 request_rtcp_encoder), _gst_element_accumulator, NULL,
2243 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2246 * GstRtpBin::request-rtcp-decoder:
2247 * @rtpbin: the object which received the signal
2248 * @session: the session
2250 * Request an RTCP decoder element for the given @session. The decoder
2251 * element will be added to the bin if not previously added.
2253 * If no handler is connected, no encoder will be used.
2257 gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] =
2258 g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
2259 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2260 request_rtcp_decoder), _gst_element_accumulator, NULL,
2261 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2264 * GstRtpBin::new-jitterbuffer:
2265 * @rtpbin: the object which received the signal
2266 * @jitterbuffer: the new jitterbuffer
2267 * @session: the session
2270 * Notify that a new @jitterbuffer was created for @session and @ssrc.
2271 * This signal can, for example, be used to configure @jitterbuffer.
2275 gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] =
2276 g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass),
2277 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2278 new_jitterbuffer), NULL, NULL, g_cclosure_marshal_generic,
2279 G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
2282 * GstRtpBin::new-storage:
2283 * @rtpbin: the object which received the signal
2284 * @storage: the new storage
2285 * @session: the session
2287 * Notify that a new @storage was created for @session.
2288 * This signal can, for example, be used to configure @storage.
2292 gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
2293 g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
2294 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2295 new_storage), NULL, NULL, g_cclosure_marshal_generic,
2296 G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
2299 * GstRtpBin::request-aux-sender:
2300 * @rtpbin: the object which received the signal
2301 * @session: the session
2303 * Request an AUX sender element for the given @session. The AUX
2304 * element will be added to the bin.
2306 * If no handler is connected, no AUX element will be used.
2310 gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] =
2311 g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
2312 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2313 request_aux_sender), _gst_element_accumulator, NULL,
2314 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2317 * GstRtpBin::request-aux-receiver:
2318 * @rtpbin: the object which received the signal
2319 * @session: the session
2321 * Request an AUX receiver element for the given @session. The AUX
2322 * element will be added to the bin.
2324 * If no handler is connected, no AUX element will be used.
2328 gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] =
2329 g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass),
2330 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2331 request_aux_receiver), _gst_element_accumulator, NULL,
2332 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2335 * GstRtpBin::request-fec-decoder:
2336 * @rtpbin: the object which received the signal
2337 * @session: the session index
2339 * Request a FEC decoder element for the given @session. The element
2340 * will be added to the bin after the pt demuxer.
2342 * If no handler is connected, no FEC decoder will be used.
2346 gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
2347 g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
2348 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2349 request_fec_decoder), _gst_element_accumulator, NULL,
2350 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2353 * GstRtpBin::request-fec-encoder:
2354 * @rtpbin: the object which received the signal
2355 * @session: the session index
2357 * Request a FEC encoder element for the given @session. The element
2358 * will be added to the bin after the RTPSession.
2360 * If no handler is connected, no FEC encoder will be used.
2364 gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
2365 g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
2366 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2367 request_fec_encoder), _gst_element_accumulator, NULL,
2368 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2371 * GstRtpBin::on-new-sender-ssrc:
2372 * @rtpbin: the object which received the signal
2373 * @session: the session
2374 * @ssrc: the sender SSRC
2376 * Notify of a new sender SSRC that entered @session.
2380 gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
2381 g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
2382 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_sender_ssrc),
2383 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2386 * GstRtpBin::on-sender-ssrc-active:
2387 * @rtpbin: the object which received the signal
2388 * @session: the session
2389 * @ssrc: the sender SSRC
2391 * Notify of a sender SSRC that is active, i.e., sending RTCP.
2395 gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
2396 g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
2397 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2398 on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
2399 G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
2401 g_object_class_install_property (gobject_class, PROP_SDES,
2402 g_param_spec_boxed ("sdes", "SDES",
2403 "The SDES items of this session",
2404 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2406 g_object_class_install_property (gobject_class, PROP_DO_LOST,
2407 g_param_spec_boolean ("do-lost", "Do Lost",
2408 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
2409 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2411 g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
2412 g_param_spec_boolean ("autoremove", "Auto Remove",
2413 "Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
2414 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2416 g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
2417 g_param_spec_boolean ("ignore-pt", "Ignore PT",
2418 "Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
2419 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2421 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
2422 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
2423 "Use the pipeline running-time to set the NTP time in the RTCP SR messages "
2424 "(DEPRECATED: Use ntp-time-source property)",
2425 DEFAULT_USE_PIPELINE_CLOCK,
2426 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
2428 * GstRtpBin:buffer-mode:
2430 * Control the buffering and timestamping mode used by the jitterbuffer.
2432 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
2433 g_param_spec_enum ("buffer-mode", "Buffer Mode",
2434 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
2435 DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2437 * GstRtpBin:ntp-sync:
2439 * Set the NTP time from the sender reports as the running-time on the
2440 * buffers. When both the sender and receiver have sychronized
2441 * running-time, i.e. when the clock and base-time is shared
2442 * between the receivers and the and the senders, this option can be
2443 * used to synchronize receivers on multiple machines.
2445 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
2446 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
2447 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
2448 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2451 * GstRtpBin:rtcp-sync:
2453 * If not synchronizing (directly) to the NTP clock, determines how to sync
2454 * the various streams.
2456 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
2457 g_param_spec_enum ("rtcp-sync", "RTCP Sync",
2458 "Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
2459 DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2462 * GstRtpBin:rtcp-sync-interval:
2464 * Determines how often to sync streams using RTCP data.
2466 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
2467 g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
2468 "RTCP SR interval synchronization (ms) (0 = always)",
2469 0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
2470 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2472 g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT,
2473 g_param_spec_boolean ("do-sync-event", "Do Sync Event",
2474 "Send event downstream when a stream is synchronized to the sender",
2475 DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2478 * GstRtpBin:do-retransmission:
2480 * Enables RTP retransmission on all streams. To control retransmission on
2481 * a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and
2482 * set the #GstRtpJitterBuffer::do-retransmission property on the
2483 * #GstRtpJitterBuffer object instead.
2485 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
2486 g_param_spec_boolean ("do-retransmission", "Do retransmission",
2487 "Enable retransmission on all streams",
2488 DEFAULT_DO_RETRANSMISSION,
2489 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2492 * GstRtpBin:rtp-profile:
2494 * Sets the default RTP profile of newly created RTP sessions. The
2495 * profile can be changed afterwards on a per-session basis.
2497 g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
2498 g_param_spec_enum ("rtp-profile", "RTP Profile",
2499 "Default RTP profile of newly created sessions",
2500 GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
2501 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2503 g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
2504 g_param_spec_enum ("ntp-time-source", "NTP Time Source",
2505 "NTP time source for RTCP packets",
2506 gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
2507 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2509 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
2510 g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
2511 "Use send time or capture time for RTCP sync "
2512 "(TRUE = send time, FALSE = capture time)",
2513 DEFAULT_RTCP_SYNC_SEND_TIME,
2514 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2516 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
2517 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
2518 "Maximum amount of time in ms that the RTP time in RTCP SRs "
2519 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
2520 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
2521 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2523 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
2524 g_param_spec_uint ("max-dropout-time", "Max dropout time",
2525 "The maximum time (milliseconds) of missing packets tolerated.",
2526 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
2527 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2529 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
2530 g_param_spec_uint ("max-misorder-time", "Max misorder time",
2531 "The maximum time (milliseconds) of misordered packets tolerated.",
2532 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
2533 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2535 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
2536 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
2537 "Synchronize received streams to the RFC7273 clock "
2538 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
2539 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2541 g_object_class_install_property (gobject_class, PROP_MAX_STREAMS,
2542 g_param_spec_uint ("max-streams", "Max Streams",
2543 "The maximum number of streams to create for one session",
2544 0, G_MAXUINT, DEFAULT_MAX_STREAMS,
2545 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2548 * GstRtpBin:max-ts-offset-adjustment:
2550 * Syncing time stamps to NTP time adds a time offset. This parameter
2551 * specifies the maximum number of nanoseconds per frame that this time offset
2552 * may be adjusted with. This is used to avoid sudden large changes to time
2557 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
2558 g_param_spec_uint64 ("max-ts-offset-adjustment",
2559 "Max Timestamp Offset Adjustment",
2560 "The maximum number of nanoseconds per frame that time stamp offsets "
2561 "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
2562 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
2563 G_PARAM_STATIC_STRINGS));
2566 * GstRtpBin:max-ts-offset:
2568 * Used to set an upper limit of how large a time offset may be. This
2569 * is used to protect against unrealistic values as a result of either
2570 * client,server or clock issues.
2574 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
2575 g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
2576 "The maximum absolute value of the time offset in (nanoseconds). "
2577 "Note, if the ntp-sync parameter is set the default value is "
2578 "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
2579 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2581 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
2582 gstelement_class->request_new_pad =
2583 GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
2584 gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
2587 gst_element_class_add_static_pad_template (gstelement_class,
2588 &rtpbin_recv_rtp_sink_template);
2589 gst_element_class_add_static_pad_template (gstelement_class,
2590 &rtpbin_recv_rtcp_sink_template);
2591 gst_element_class_add_static_pad_template (gstelement_class,
2592 &rtpbin_send_rtp_sink_template);
2595 gst_element_class_add_static_pad_template (gstelement_class,
2596 &rtpbin_recv_rtp_src_template);
2597 gst_element_class_add_static_pad_template (gstelement_class,
2598 &rtpbin_send_rtcp_src_template);
2599 gst_element_class_add_static_pad_template (gstelement_class,
2600 &rtpbin_send_rtp_src_template);
2602 gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
2603 "Filter/Network/RTP",
2604 "Real-Time Transport Protocol bin",
2605 "Wim Taymans <wim.taymans@gmail.com>");
2607 gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
2609 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
2610 klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
2611 klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
2612 klass->get_internal_session =
2613 GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
2614 klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
2615 klass->get_internal_storage =
2616 GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
2617 klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
2618 klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
2619 klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
2620 klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
2622 GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
2626 gst_rtp_bin_init (GstRtpBin * rtpbin)
2630 rtpbin->priv = gst_rtp_bin_get_instance_private (rtpbin);
2631 g_mutex_init (&rtpbin->priv->bin_lock);
2632 g_mutex_init (&rtpbin->priv->dyn_lock);
2634 rtpbin->latency_ms = DEFAULT_LATENCY_MS;
2635 rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
2636 rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
2637 rtpbin->do_lost = DEFAULT_DO_LOST;
2638 rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
2639 rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
2640 rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC;
2641 rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
2642 rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
2643 rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
2644 rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
2645 rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT;
2646 rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION;
2647 rtpbin->rtp_profile = DEFAULT_RTP_PROFILE;
2648 rtpbin->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
2649 rtpbin->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
2650 rtpbin->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
2651 rtpbin->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
2652 rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
2653 rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
2654 rtpbin->max_streams = DEFAULT_MAX_STREAMS;
2655 rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
2656 rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
2657 rtpbin->max_ts_offset_is_set = FALSE;
2659 /* some default SDES entries */
2660 cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
2661 rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
2662 "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
2667 gst_rtp_bin_dispose (GObject * object)
2671 rtpbin = GST_RTP_BIN (object);
2673 GST_RTP_BIN_LOCK (rtpbin);
2674 GST_DEBUG_OBJECT (object, "freeing sessions");
2675 g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
2676 g_slist_free (rtpbin->sessions);
2677 rtpbin->sessions = NULL;
2678 GST_RTP_BIN_UNLOCK (rtpbin);
2680 G_OBJECT_CLASS (parent_class)->dispose (object);
2684 gst_rtp_bin_finalize (GObject * object)
2688 rtpbin = GST_RTP_BIN (object);
2691 gst_structure_free (rtpbin->sdes);
2693 g_mutex_clear (&rtpbin->priv->bin_lock);
2694 g_mutex_clear (&rtpbin->priv->dyn_lock);
2696 G_OBJECT_CLASS (parent_class)->finalize (object);
2701 gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
2708 GST_RTP_BIN_LOCK (bin);
2710 GST_OBJECT_LOCK (bin);
2712 gst_structure_free (bin->sdes);
2713 bin->sdes = gst_structure_copy (sdes);
2714 GST_OBJECT_UNLOCK (bin);
2716 /* store in all sessions */
2717 for (item = bin->sessions; item; item = g_slist_next (item)) {
2718 GstRtpBinSession *session = item->data;
2719 g_object_set (session->session, "sdes", sdes, NULL);
2722 GST_RTP_BIN_UNLOCK (bin);
2725 static GstStructure *
2726 gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
2728 GstStructure *result;
2730 GST_OBJECT_LOCK (bin);
2731 result = gst_structure_copy (bin->sdes);
2732 GST_OBJECT_UNLOCK (bin);
2738 gst_rtp_bin_set_property (GObject * object, guint prop_id,
2739 const GValue * value, GParamSpec * pspec)
2743 rtpbin = GST_RTP_BIN (object);
2747 GST_RTP_BIN_LOCK (rtpbin);
2748 rtpbin->latency_ms = g_value_get_uint (value);
2749 rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
2750 GST_RTP_BIN_UNLOCK (rtpbin);
2751 /* propagate the property down to the jitterbuffer */
2752 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
2754 case PROP_DROP_ON_LATENCY:
2755 GST_RTP_BIN_LOCK (rtpbin);
2756 rtpbin->drop_on_latency = g_value_get_boolean (value);
2757 GST_RTP_BIN_UNLOCK (rtpbin);
2758 /* propagate the property down to the jitterbuffer */
2759 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2760 "drop-on-latency", value);
2763 gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
2766 GST_RTP_BIN_LOCK (rtpbin);
2767 rtpbin->do_lost = g_value_get_boolean (value);
2768 GST_RTP_BIN_UNLOCK (rtpbin);
2769 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
2772 rtpbin->ntp_sync = g_value_get_boolean (value);
2773 /* The default value of max_ts_offset depends on ntp_sync. If user
2774 * hasn't set it then change default value */
2775 if (!rtpbin->max_ts_offset_is_set) {
2776 if (rtpbin->ntp_sync) {
2777 rtpbin->max_ts_offset = 0;
2779 rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
2783 case PROP_RTCP_SYNC:
2784 g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
2786 case PROP_RTCP_SYNC_INTERVAL:
2787 rtpbin->rtcp_sync_interval = g_value_get_uint (value);
2789 case PROP_IGNORE_PT:
2790 rtpbin->ignore_pt = g_value_get_boolean (value);
2792 case PROP_AUTOREMOVE:
2793 rtpbin->priv->autoremove = g_value_get_boolean (value);
2795 case PROP_USE_PIPELINE_CLOCK:
2798 GST_RTP_BIN_LOCK (rtpbin);
2799 rtpbin->use_pipeline_clock = g_value_get_boolean (value);
2800 for (sessions = rtpbin->sessions; sessions;
2801 sessions = g_slist_next (sessions)) {
2802 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2804 g_object_set (G_OBJECT (session->session),
2805 "use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
2807 GST_RTP_BIN_UNLOCK (rtpbin);
2810 case PROP_DO_SYNC_EVENT:
2811 rtpbin->send_sync_event = g_value_get_boolean (value);
2813 case PROP_BUFFER_MODE:
2814 GST_RTP_BIN_LOCK (rtpbin);
2815 rtpbin->buffer_mode = g_value_get_enum (value);
2816 GST_RTP_BIN_UNLOCK (rtpbin);
2817 /* propagate the property down to the jitterbuffer */
2818 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
2820 case PROP_DO_RETRANSMISSION:
2821 GST_RTP_BIN_LOCK (rtpbin);
2822 rtpbin->do_retransmission = g_value_get_boolean (value);
2823 GST_RTP_BIN_UNLOCK (rtpbin);
2824 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2825 "do-retransmission", value);
2827 case PROP_RTP_PROFILE:
2828 rtpbin->rtp_profile = g_value_get_enum (value);
2830 case PROP_NTP_TIME_SOURCE:{
2832 GST_RTP_BIN_LOCK (rtpbin);
2833 rtpbin->ntp_time_source = g_value_get_enum (value);
2834 for (sessions = rtpbin->sessions; sessions;
2835 sessions = g_slist_next (sessions)) {
2836 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2838 g_object_set (G_OBJECT (session->session),
2839 "ntp-time-source", rtpbin->ntp_time_source, NULL);
2841 GST_RTP_BIN_UNLOCK (rtpbin);
2844 case PROP_RTCP_SYNC_SEND_TIME:{
2846 GST_RTP_BIN_LOCK (rtpbin);
2847 rtpbin->rtcp_sync_send_time = g_value_get_boolean (value);
2848 for (sessions = rtpbin->sessions; sessions;
2849 sessions = g_slist_next (sessions)) {
2850 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2852 g_object_set (G_OBJECT (session->session),
2853 "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time, NULL);
2855 GST_RTP_BIN_UNLOCK (rtpbin);
2858 case PROP_MAX_RTCP_RTP_TIME_DIFF:
2859 GST_RTP_BIN_LOCK (rtpbin);
2860 rtpbin->max_rtcp_rtp_time_diff = g_value_get_int (value);
2861 GST_RTP_BIN_UNLOCK (rtpbin);
2862 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2863 "max-rtcp-rtp-time-diff", value);
2865 case PROP_MAX_DROPOUT_TIME:
2866 GST_RTP_BIN_LOCK (rtpbin);
2867 rtpbin->max_dropout_time = g_value_get_uint (value);
2868 GST_RTP_BIN_UNLOCK (rtpbin);
2869 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2870 "max-dropout-time", value);
2871 gst_rtp_bin_propagate_property_to_session (rtpbin, "max-dropout-time",
2874 case PROP_MAX_MISORDER_TIME:
2875 GST_RTP_BIN_LOCK (rtpbin);
2876 rtpbin->max_misorder_time = g_value_get_uint (value);
2877 GST_RTP_BIN_UNLOCK (rtpbin);
2878 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2879 "max-misorder-time", value);
2880 gst_rtp_bin_propagate_property_to_session (rtpbin, "max-misorder-time",
2883 case PROP_RFC7273_SYNC:
2884 rtpbin->rfc7273_sync = g_value_get_boolean (value);
2885 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2886 "rfc7273-sync", value);
2888 case PROP_MAX_STREAMS:
2889 rtpbin->max_streams = g_value_get_uint (value);
2891 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
2892 rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
2893 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2894 "max-ts-offset-adjustment", value);
2896 case PROP_MAX_TS_OFFSET:
2897 rtpbin->max_ts_offset = g_value_get_int64 (value);
2898 rtpbin->max_ts_offset_is_set = TRUE;
2901 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2907 gst_rtp_bin_get_property (GObject * object, guint prop_id,
2908 GValue * value, GParamSpec * pspec)
2912 rtpbin = GST_RTP_BIN (object);
2916 GST_RTP_BIN_LOCK (rtpbin);
2917 g_value_set_uint (value, rtpbin->latency_ms);
2918 GST_RTP_BIN_UNLOCK (rtpbin);
2920 case PROP_DROP_ON_LATENCY:
2921 GST_RTP_BIN_LOCK (rtpbin);
2922 g_value_set_boolean (value, rtpbin->drop_on_latency);
2923 GST_RTP_BIN_UNLOCK (rtpbin);
2926 g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
2929 GST_RTP_BIN_LOCK (rtpbin);
2930 g_value_set_boolean (value, rtpbin->do_lost);
2931 GST_RTP_BIN_UNLOCK (rtpbin);
2933 case PROP_IGNORE_PT:
2934 g_value_set_boolean (value, rtpbin->ignore_pt);
2937 g_value_set_boolean (value, rtpbin->ntp_sync);
2939 case PROP_RTCP_SYNC:
2940 g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
2942 case PROP_RTCP_SYNC_INTERVAL:
2943 g_value_set_uint (value, rtpbin->rtcp_sync_interval);
2945 case PROP_AUTOREMOVE:
2946 g_value_set_boolean (value, rtpbin->priv->autoremove);
2948 case PROP_BUFFER_MODE:
2949 g_value_set_enum (value, rtpbin->buffer_mode);
2951 case PROP_USE_PIPELINE_CLOCK:
2952 g_value_set_boolean (value, rtpbin->use_pipeline_clock);
2954 case PROP_DO_SYNC_EVENT:
2955 g_value_set_boolean (value, rtpbin->send_sync_event);
2957 case PROP_DO_RETRANSMISSION:
2958 GST_RTP_BIN_LOCK (rtpbin);
2959 g_value_set_boolean (value, rtpbin->do_retransmission);
2960 GST_RTP_BIN_UNLOCK (rtpbin);
2962 case PROP_RTP_PROFILE:
2963 g_value_set_enum (value, rtpbin->rtp_profile);
2965 case PROP_NTP_TIME_SOURCE:
2966 g_value_set_enum (value, rtpbin->ntp_time_source);
2968 case PROP_RTCP_SYNC_SEND_TIME:
2969 g_value_set_boolean (value, rtpbin->rtcp_sync_send_time);
2971 case PROP_MAX_RTCP_RTP_TIME_DIFF:
2972 GST_RTP_BIN_LOCK (rtpbin);
2973 g_value_set_int (value, rtpbin->max_rtcp_rtp_time_diff);
2974 GST_RTP_BIN_UNLOCK (rtpbin);
2976 case PROP_MAX_DROPOUT_TIME:
2977 g_value_set_uint (value, rtpbin->max_dropout_time);
2979 case PROP_MAX_MISORDER_TIME:
2980 g_value_set_uint (value, rtpbin->max_misorder_time);
2982 case PROP_RFC7273_SYNC:
2983 g_value_set_boolean (value, rtpbin->rfc7273_sync);
2985 case PROP_MAX_STREAMS:
2986 g_value_set_uint (value, rtpbin->max_streams);
2988 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
2989 g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
2991 case PROP_MAX_TS_OFFSET:
2992 g_value_set_int64 (value, rtpbin->max_ts_offset);
2995 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3001 gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
3005 rtpbin = GST_RTP_BIN (bin);
3007 switch (GST_MESSAGE_TYPE (message)) {
3008 case GST_MESSAGE_ELEMENT:
3010 const GstStructure *s = gst_message_get_structure (message);
3012 /* we change the structure name and add the session ID to it */
3013 if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
3014 GstRtpBinSession *sess;
3016 /* find the session we set it as object data */
3017 sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
3018 "GstRTPBin.session");
3020 if (G_LIKELY (sess)) {
3021 message = gst_message_make_writable (message);
3022 s = gst_message_get_structure (message);
3023 gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
3027 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3030 case GST_MESSAGE_BUFFERING:
3033 gint min_percent = 100;
3034 GSList *sessions, *streams;
3035 GstRtpBinStream *stream;
3036 gboolean change = FALSE, active = FALSE;
3037 GstClockTime min_out_time;
3038 GstBufferingMode mode;
3039 gint avg_in, avg_out;
3040 gint64 buffering_left;
3042 gst_message_parse_buffering (message, &percent);
3043 gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
3047 g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
3048 "GstRTPBin.stream");
3050 GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
3052 /* get the stream */
3053 if (G_LIKELY (stream)) {
3054 GST_RTP_BIN_LOCK (rtpbin);
3055 /* fill in the percent */
3056 stream->percent = percent;
3058 /* calculate the min value for all streams */
3059 for (sessions = rtpbin->sessions; sessions;
3060 sessions = g_slist_next (sessions)) {
3061 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
3063 GST_RTP_SESSION_LOCK (session);
3064 if (session->streams) {
3065 for (streams = session->streams; streams;
3066 streams = g_slist_next (streams)) {
3067 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
3069 GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
3072 /* find min percent */
3073 if (min_percent > stream->percent)
3074 min_percent = stream->percent;
3077 GST_INFO_OBJECT (bin,
3078 "session has no streams, setting min_percent to 0");
3081 GST_RTP_SESSION_UNLOCK (session);
3083 GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
3085 if (rtpbin->buffering) {
3086 if (min_percent == 100) {
3087 rtpbin->buffering = FALSE;
3092 if (min_percent < 100) {
3093 /* pause the streams */
3094 rtpbin->buffering = TRUE;
3099 GST_RTP_BIN_UNLOCK (rtpbin);
3101 gst_message_unref (message);
3103 /* make a new buffering message with the min value */
3105 gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
3106 gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
3109 if (G_UNLIKELY (change)) {
3111 guint64 running_time = 0;
3114 /* figure out the running time when we have a clock */
3115 if (G_LIKELY ((clock =
3116 gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
3117 guint64 now, base_time;
3119 now = gst_clock_get_time (clock);
3120 base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
3121 running_time = now - base_time;
3122 gst_object_unref (clock);
3124 GST_DEBUG_OBJECT (bin,
3125 "running time now %" GST_TIME_FORMAT,
3126 GST_TIME_ARGS (running_time));
3128 GST_RTP_BIN_LOCK (rtpbin);
3130 /* when we reactivate, calculate the offsets so that all streams have
3131 * an output time that is at least as big as the running_time */
3134 if (running_time > rtpbin->buffer_start) {
3135 offset = running_time - rtpbin->buffer_start;
3136 if (offset >= rtpbin->latency_ns)
3137 offset -= rtpbin->latency_ns;
3143 /* pause all streams */
3145 for (sessions = rtpbin->sessions; sessions;
3146 sessions = g_slist_next (sessions)) {
3147 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
3149 GST_RTP_SESSION_LOCK (session);
3150 for (streams = session->streams; streams;
3151 streams = g_slist_next (streams)) {
3152 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
3153 GstElement *element = stream->buffer;
3156 g_signal_emit_by_name (element, "set-active", active, offset,
3160 g_object_get (element, "percent", &stream->percent, NULL);
3164 if (min_out_time == -1 || last_out < min_out_time)
3165 min_out_time = last_out;
3168 GST_DEBUG_OBJECT (bin,
3169 "setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
3170 GST_TIME_FORMAT ", percent %d", element, active,
3171 GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
3174 GST_RTP_SESSION_UNLOCK (session);
3176 GST_DEBUG_OBJECT (bin,
3177 "min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
3179 /* the buffer_start is the min out time of all paused jitterbuffers */
3181 rtpbin->buffer_start = min_out_time;
3183 GST_RTP_BIN_UNLOCK (rtpbin);
3186 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3191 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3197 static GstStateChangeReturn
3198 gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
3200 GstStateChangeReturn res;
3202 GstRtpBinPrivate *priv;
3204 rtpbin = GST_RTP_BIN (element);
3205 priv = rtpbin->priv;
3207 switch (transition) {
3208 case GST_STATE_CHANGE_NULL_TO_READY:
3210 case GST_STATE_CHANGE_READY_TO_PAUSED:
3211 priv->last_ntpnstime = 0;
3212 GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
3213 g_atomic_int_set (&priv->shutdown, 0);
3215 case GST_STATE_CHANGE_PAUSED_TO_READY:
3216 GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
3217 g_atomic_int_set (&priv->shutdown, 1);
3218 /* wait for all callbacks to end by taking the lock. No new callbacks will
3219 * be able to happen as we set the shutdown flag. */
3220 GST_RTP_BIN_DYN_LOCK (rtpbin);
3221 GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
3222 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
3228 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
3230 switch (transition) {
3231 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
3233 case GST_STATE_CHANGE_PAUSED_TO_READY:
3235 case GST_STATE_CHANGE_READY_TO_NULL:
3244 session_request_element (GstRtpBinSession * session, guint signal)
3246 GstElement *element = NULL;
3247 GstRtpBin *bin = session->bin;
3249 g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element);
3252 if (!bin_manage_element (bin, element))
3254 session->elements = g_slist_prepend (session->elements, element);
3261 GST_WARNING_OBJECT (bin, "unable to manage element");
3262 gst_object_unref (element);
3268 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
3270 GstPad *gpad = GST_PAD_CAST (user_data);
3272 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
3273 gst_pad_store_sticky_event (gpad, *event);
3279 expose_recv_src_pad (GstRtpBin * rtpbin, GstPad * pad, GstRtpBinStream * stream,
3282 GstElementClass *klass;
3283 GstPadTemplate *templ;
3287 gst_object_ref (pad);
3289 if (stream->session->storage) {
3290 GstElement *fec_decoder =
3291 session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
3294 GstPad *sinkpad, *srcpad;
3295 GstPadLinkReturn ret;
3297 sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
3300 goto fec_decoder_sink_failed;
3302 ret = gst_pad_link (pad, sinkpad);
3303 gst_object_unref (sinkpad);
3305 if (ret != GST_PAD_LINK_OK)
3306 goto fec_decoder_link_failed;
3308 srcpad = gst_element_get_static_pad (fec_decoder, "src");
3311 goto fec_decoder_src_failed;
3313 gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
3314 gst_object_unref (pad);
3319 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
3321 /* ghost the pad to the parent */
3322 klass = GST_ELEMENT_GET_CLASS (rtpbin);
3323 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
3324 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
3325 stream->session->id, stream->ssrc, pt);
3326 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
3328 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
3330 gst_pad_set_active (gpad, TRUE);
3331 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3333 gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
3334 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
3337 gst_object_unref (pad);
3343 GST_DEBUG ("ignoring, we are shutting down");
3346 fec_decoder_sink_failed:
3348 g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
3349 stream->session->id);
3352 fec_decoder_src_failed:
3354 g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
3355 stream->session->id);
3358 fec_decoder_link_failed:
3360 g_warning ("rtpbin: failed to link fec decoder for session %u",
3361 stream->session->id);
3366 /* a new pad (SSRC) was created in @session. This signal is emited from the
3367 * payload demuxer. */
3369 new_payload_found (GstElement * element, guint pt, GstPad * pad,
3370 GstRtpBinStream * stream)
3374 rtpbin = stream->bin;
3376 GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
3378 expose_recv_src_pad (rtpbin, pad, stream, pt);
3382 payload_pad_removed (GstElement * element, GstPad * pad,
3383 GstRtpBinStream * stream)
3388 rtpbin = stream->bin;
3390 GST_DEBUG ("payload pad removed");
3392 GST_RTP_BIN_DYN_LOCK (rtpbin);
3393 if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
3394 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
3396 gst_pad_set_active (gpad, FALSE);
3397 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
3399 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
3403 pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
3408 rtpbin = session->bin;
3410 GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
3413 caps = get_pt_map (session, pt);
3422 GST_DEBUG_OBJECT (rtpbin, "could not get caps");
3428 ptdemux_pt_map_requested (GstElement * element, guint pt,
3429 GstRtpBinSession * session)
3431 GstCaps *ret = pt_map_requested (element, pt, session);
3433 if (ret && gst_caps_get_size (ret) == 1) {
3434 const GstStructure *s = gst_caps_get_structure (ret, 0);
3437 if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
3438 GValue v = G_VALUE_INIT;
3439 GValue v2 = G_VALUE_INIT;
3441 GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
3443 g_value_init (&v, GST_TYPE_ARRAY);
3444 g_value_init (&v2, G_TYPE_INT);
3445 g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
3446 g_value_set_int (&v2, pt);
3447 gst_value_array_append_value (&v, &v2);
3448 g_value_unset (&v2);
3449 g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
3458 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
3460 GST_DEBUG_OBJECT (session->bin,
3461 "emiting signal for pt type changed to %u in session %u", pt,
3464 g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
3465 0, session->id, pt);
3468 /* emitted when caps changed for the session */
3470 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
3475 const GstStructure *s;
3479 g_object_get (pad, "caps", &caps, NULL);
3484 GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
3486 s = gst_caps_get_structure (caps, 0);
3488 /* get payload, finish when it's not there */
3489 if (!gst_structure_get_int (s, "payload", &payload)) {
3490 gst_caps_unref (caps);
3494 GST_RTP_SESSION_LOCK (session);
3495 GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
3496 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
3497 GST_RTP_SESSION_UNLOCK (session);
3500 /* a new pad (SSRC) was created in @session */
3502 new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
3503 GstRtpBinSession * session)
3506 GstRtpBinStream *stream;
3507 GstPad *sinkpad, *srcpad;
3510 rtpbin = session->bin;
3512 GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
3513 GST_DEBUG_PAD_NAME (pad));
3515 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
3517 GST_RTP_SESSION_LOCK (session);
3519 /* create new stream */
3520 stream = create_stream (session, ssrc);
3524 /* get pad and link */
3525 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
3526 padname = g_strdup_printf ("src_%u", ssrc);
3527 srcpad = gst_element_get_static_pad (element, padname);
3529 sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
3530 gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
3531 gst_object_unref (sinkpad);
3532 gst_object_unref (srcpad);
3534 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
3535 padname = g_strdup_printf ("rtcp_src_%u", ssrc);
3536 srcpad = gst_element_get_static_pad (element, padname);
3538 sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
3539 gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
3540 gst_object_unref (sinkpad);
3541 gst_object_unref (srcpad);
3543 /* connect to the RTCP sync signal from the jitterbuffer */
3544 GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
3545 stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
3546 "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
3548 if (stream->demux) {
3549 /* connect to the new-pad signal of the payload demuxer, this will expose the
3550 * new pad by ghosting it. */
3551 stream->demux_newpad_sig = g_signal_connect (stream->demux,
3552 "new-payload-type", (GCallback) new_payload_found, stream);
3553 stream->demux_padremoved_sig = g_signal_connect (stream->demux,
3554 "pad-removed", (GCallback) payload_pad_removed, stream);
3556 /* connect to the request-pt-map signal. This signal will be emitted by the
3557 * demuxer so that it can apply a proper caps on the buffers for the
3559 stream->demux_ptreq_sig = g_signal_connect (stream->demux,
3560 "request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
3561 /* connect to the signal so it can be forwarded. */
3562 stream->demux_ptchange_sig = g_signal_connect (stream->demux,
3563 "payload-type-change", (GCallback) payload_type_change, session);
3565 GST_RTP_SESSION_UNLOCK (session);
3566 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3568 /* add rtpjitterbuffer src pad to pads */
3571 pad = gst_element_get_static_pad (stream->buffer, "src");
3573 GST_RTP_SESSION_UNLOCK (session);
3574 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3576 expose_recv_src_pad (rtpbin, pad, stream, 255);
3578 gst_object_unref (pad);
3586 GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
3591 GST_RTP_SESSION_UNLOCK (session);
3592 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3593 GST_DEBUG_OBJECT (rtpbin, "could not create stream");
3599 complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
3601 guint sessid = session->id;
3602 GstPad *recv_rtp_sink;
3603 GstElement *decoder;
3605 g_assert (!session->recv_rtp_sink);
3607 /* get recv_rtp pad and store */
3608 session->recv_rtp_sink =
3609 gst_element_get_request_pad (session->session, "recv_rtp_sink");
3610 if (session->recv_rtp_sink == NULL)
3613 g_signal_connect (session->recv_rtp_sink, "notify::caps",
3614 (GCallback) caps_changed, session);
3616 GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
3617 decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
3619 GstPad *decsrc, *decsink;
3620 GstPadLinkReturn ret;
3622 GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder");
3623 decsink = gst_element_get_static_pad (decoder, "rtp_sink");
3624 if (decsink == NULL)
3625 goto dec_sink_failed;
3627 recv_rtp_sink = decsink;
3629 decsrc = gst_element_get_static_pad (decoder, "rtp_src");
3631 goto dec_src_failed;
3633 ret = gst_pad_link (decsrc, session->recv_rtp_sink);
3635 gst_object_unref (decsrc);
3637 if (ret != GST_PAD_LINK_OK)
3638 goto dec_link_failed;
3641 GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
3642 recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
3645 return recv_rtp_sink;
3650 g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
3655 g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
3660 g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
3661 gst_object_unref (recv_rtp_sink);
3666 g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
3667 gst_object_unref (recv_rtp_sink);
3673 complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
3677 GstPad *recv_rtp_src;
3679 g_assert (!session->recv_rtp_src);
3681 session->recv_rtp_src =
3682 gst_element_get_static_pad (session->session, "recv_rtp_src");
3683 if (session->recv_rtp_src == NULL)
3686 /* find out if we need AUX elements */
3687 aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
3691 GstPadLinkReturn ret;
3693 GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
3695 pname = g_strdup_printf ("sink_%u", sessid);
3696 auxsink = gst_element_get_static_pad (aux, pname);
3698 if (auxsink == NULL)
3699 goto aux_sink_failed;
3701 ret = gst_pad_link (session->recv_rtp_src, auxsink);
3702 gst_object_unref (auxsink);
3703 if (ret != GST_PAD_LINK_OK)
3704 goto aux_link_failed;
3706 /* this can be NULL when this AUX element is not to be linked any further */
3707 pname = g_strdup_printf ("src_%u", sessid);
3708 recv_rtp_src = gst_element_get_static_pad (aux, pname);
3711 recv_rtp_src = gst_object_ref (session->recv_rtp_src);
3714 /* Add a storage element if needed */
3715 if (recv_rtp_src && session->storage) {
3716 GstPadLinkReturn ret;
3717 GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
3719 ret = gst_pad_link (recv_rtp_src, sinkpad);
3721 gst_object_unref (sinkpad);
3722 gst_object_unref (recv_rtp_src);
3724 if (ret != GST_PAD_LINK_OK)
3725 goto storage_link_failed;
3727 recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
3733 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
3734 sinkdpad = gst_element_get_static_pad (session->demux, "sink");
3735 GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
3736 gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
3737 gst_object_unref (sinkdpad);
3738 gst_object_unref (recv_rtp_src);
3740 /* connect to the new-ssrc-pad signal of the SSRC demuxer */
3741 session->demux_newpad_sig = g_signal_connect (session->demux,
3742 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
3743 session->demux_padremoved_sig = g_signal_connect (session->demux,
3744 "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
3751 g_warning ("rtpbin: failed to get session recv_rtp_src pad");
3756 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
3761 g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
3764 storage_link_failed:
3766 g_warning ("rtpbin: failed to link storage");
3771 /* Create a pad for receiving RTP for the session in @name. Must be called with
3775 create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
3778 GstRtpBinSession *session;
3779 GstPad *recv_rtp_sink;
3781 /* first get the session number */
3782 if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
3785 GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
3787 /* get or create session */
3788 session = find_session_by_id (rtpbin, sessid);
3790 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
3791 /* create session now */
3792 session = create_session (rtpbin, sessid);
3793 if (session == NULL)
3797 /* check if pad was requested */
3798 if (session->recv_rtp_sink_ghost != NULL)
3799 return session->recv_rtp_sink_ghost;
3801 /* setup the session sink pad */
3802 recv_rtp_sink = complete_session_sink (rtpbin, session);
3804 goto session_sink_failed;
3806 GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
3807 session->recv_rtp_sink_ghost =
3808 gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
3809 gst_object_unref (recv_rtp_sink);
3810 gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
3811 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
3813 complete_session_receiver (rtpbin, session, sessid);
3815 return session->recv_rtp_sink_ghost;
3820 g_warning ("rtpbin: invalid name given");
3825 /* create_session already warned */
3828 session_sink_failed:
3830 /* warning already done */
3836 remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
3838 if (session->demux_newpad_sig) {
3839 g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
3840 session->demux_newpad_sig = 0;
3842 if (session->demux_padremoved_sig) {
3843 g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
3844 session->demux_padremoved_sig = 0;
3846 if (session->recv_rtp_src) {
3847 gst_object_unref (session->recv_rtp_src);
3848 session->recv_rtp_src = NULL;
3850 if (session->recv_rtp_sink) {
3851 gst_element_release_request_pad (session->session, session->recv_rtp_sink);
3852 gst_object_unref (session->recv_rtp_sink);
3853 session->recv_rtp_sink = NULL;
3855 if (session->recv_rtp_sink_ghost) {
3856 gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
3857 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
3858 session->recv_rtp_sink_ghost);
3859 session->recv_rtp_sink_ghost = NULL;
3864 complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
3867 GstElement *decoder;
3869 GstPad *decsink = NULL;
3871 /* get recv_rtp pad and store */
3872 GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
3873 session->recv_rtcp_sink =
3874 gst_element_get_request_pad (session->session, "recv_rtcp_sink");
3875 if (session->recv_rtcp_sink == NULL)
3878 GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
3879 decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
3882 GstPadLinkReturn ret;
3884 GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder");
3885 decsink = gst_element_get_static_pad (decoder, "rtcp_sink");
3886 decsrc = gst_element_get_static_pad (decoder, "rtcp_src");
3888 if (decsink == NULL)
3889 goto dec_sink_failed;
3892 goto dec_src_failed;
3894 ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
3896 gst_object_unref (decsrc);
3898 if (ret != GST_PAD_LINK_OK)
3899 goto dec_link_failed;
3901 GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
3902 decsink = gst_object_ref (session->recv_rtcp_sink);
3905 /* get srcpad, link to SSRCDemux */
3906 GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
3907 session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
3908 if (session->sync_src == NULL)
3909 goto src_pad_failed;
3911 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
3912 sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
3913 gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
3914 gst_object_unref (sinkdpad);
3920 g_warning ("rtpbin: failed to get session rtcp_sink pad");
3925 g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
3930 g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
3935 g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
3940 g_warning ("rtpbin: failed to get session sync_src pad");
3944 gst_object_unref (decsink);
3948 /* Create a pad for receiving RTCP for the session in @name. Must be called with
3952 create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
3956 GstRtpBinSession *session;
3957 GstPad *decsink = NULL;
3959 /* first get the session number */
3960 if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
3963 GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
3965 /* get or create the session */
3966 session = find_session_by_id (rtpbin, sessid);
3968 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
3969 /* create session now */
3970 session = create_session (rtpbin, sessid);
3971 if (session == NULL)
3975 /* check if pad was requested */
3976 if (session->recv_rtcp_sink_ghost != NULL)
3977 return session->recv_rtcp_sink_ghost;
3979 decsink = complete_session_rtcp (rtpbin, session, sessid);
3983 session->recv_rtcp_sink_ghost =
3984 gst_ghost_pad_new_from_template (name, decsink, templ);
3985 gst_object_unref (decsink);
3986 gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
3987 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
3988 session->recv_rtcp_sink_ghost);
3990 return session->recv_rtcp_sink_ghost;
3995 g_warning ("rtpbin: invalid name given");
4000 /* create_session already warned */
4006 remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4008 if (session->recv_rtcp_sink_ghost) {
4009 gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
4010 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4011 session->recv_rtcp_sink_ghost);
4012 session->recv_rtcp_sink_ghost = NULL;
4014 if (session->sync_src) {
4015 /* releasing the request pad should also unref the sync pad */
4016 gst_object_unref (session->sync_src);
4017 session->sync_src = NULL;
4019 if (session->recv_rtcp_sink) {
4020 gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
4021 gst_object_unref (session->recv_rtcp_sink);
4022 session->recv_rtcp_sink = NULL;
4027 complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
4030 guint sessid = session->id;
4031 GstPad *send_rtp_src;
4032 GstElement *encoder;
4033 GstElementClass *klass;
4034 GstPadTemplate *templ;
4035 gboolean ret = FALSE;
4038 send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
4040 if (send_rtp_src == NULL)
4043 GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
4044 encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER);
4047 GstPad *encsrc, *encsink;
4048 GstPadLinkReturn ret;
4050 GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
4051 ename = g_strdup_printf ("rtp_src_%u", sessid);
4052 encsrc = gst_element_get_static_pad (encoder, ename);
4056 goto enc_src_failed;
4058 ename = g_strdup_printf ("rtp_sink_%u", sessid);
4059 encsink = gst_element_get_static_pad (encoder, ename);
4061 if (encsink == NULL)
4062 goto enc_sink_failed;
4064 ret = gst_pad_link (send_rtp_src, encsink);
4065 gst_object_unref (encsink);
4066 gst_object_unref (send_rtp_src);
4068 send_rtp_src = encsrc;
4070 if (ret != GST_PAD_LINK_OK)
4071 goto enc_link_failed;
4073 GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
4076 /* ghost the new source pad */
4077 klass = GST_ELEMENT_GET_CLASS (rtpbin);
4078 gname = g_strdup_printf ("send_rtp_src_%u", sessid);
4079 templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
4080 session->send_rtp_src_ghost =
4081 gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
4082 gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
4083 gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
4084 session->send_rtp_src_ghost);
4085 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
4092 gst_object_unref (send_rtp_src);
4099 g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
4104 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4105 " src pad for session %u", encoder, sessid);
4110 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4111 " sink pad for session %u", encoder, sessid);
4116 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4123 setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data)
4128 GstRtpBinSession *session = user_data, *newsess;
4129 GstRtpBin *rtpbin = session->bin;
4130 GstPadLinkReturn ret;
4132 pad = g_value_get_object (item);
4133 name = gst_pad_get_name (pad);
4135 if (name == NULL || sscanf (name, "src_%u", &sessid) != 1)
4140 newsess = find_session_by_id (rtpbin, sessid);
4141 if (newsess == NULL) {
4142 /* create new session */
4143 newsess = create_session (rtpbin, sessid);
4144 if (newsess == NULL)
4146 } else if (newsess->send_rtp_sink != NULL)
4147 goto existing_session;
4149 /* get send_rtp pad and store */
4150 newsess->send_rtp_sink =
4151 gst_element_get_request_pad (newsess->session, "send_rtp_sink");
4152 if (newsess->send_rtp_sink == NULL)
4155 ret = gst_pad_link (pad, newsess->send_rtp_sink);
4156 if (ret != GST_PAD_LINK_OK)
4157 goto aux_link_failed;
4159 if (!complete_session_src (rtpbin, newsess))
4160 goto session_src_failed;
4167 GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name));
4173 /* create_session already warned */
4178 GST_DEBUG_OBJECT (rtpbin,
4179 "skipping src_%i setup, since it is already configured.", sessid);
4184 g_warning ("rtpbin: failed to get session pad for session %u", sessid);
4189 g_warning ("rtpbin: failed to link AUX for session %u", sessid);
4194 g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
4200 setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session,
4204 GValue result = { 0, };
4205 GstIteratorResult res;
4207 it = gst_element_iterate_src_pads (aux);
4208 res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session);
4209 gst_iterator_free (it);
4211 return res == GST_ITERATOR_DONE;
4214 /* Create a pad for sending RTP for the session in @name. Must be called with
4218 create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
4222 GstPad *send_rtp_sink;
4224 GstElement *encoder;
4225 GstElement *prev = NULL;
4226 GstRtpBinSession *session;
4228 /* first get the session number */
4229 if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
4232 /* get or create session */
4233 session = find_session_by_id (rtpbin, sessid);
4235 /* create session now */
4236 session = create_session (rtpbin, sessid);
4237 if (session == NULL)
4241 /* check if pad was requested */
4242 if (session->send_rtp_sink_ghost != NULL)
4243 return session->send_rtp_sink_ghost;
4245 /* check if we are already using this session as a sender */
4246 if (session->send_rtp_sink != NULL)
4247 goto existing_session;
4249 encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
4252 GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
4254 send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
4257 goto enc_sink_failed;
4262 GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
4263 aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
4266 GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
4267 if (!setup_aux_sender (rtpbin, session, aux))
4268 goto aux_session_failed;
4270 pname = g_strdup_printf ("sink_%u", sessid);
4271 sinkpad = gst_element_get_static_pad (aux, pname);
4274 if (sinkpad == NULL)
4275 goto aux_sink_failed;
4278 send_rtp_sink = sinkpad;
4280 GstPad *srcpad = gst_element_get_static_pad (prev, "src");
4281 GstPadLinkReturn ret;
4283 ret = gst_pad_link (srcpad, sinkpad);
4284 gst_object_unref (srcpad);
4285 if (ret != GST_PAD_LINK_OK) {
4286 goto aux_link_failed;
4291 /* get send_rtp pad and store */
4292 session->send_rtp_sink =
4293 gst_element_get_request_pad (session->session, "send_rtp_sink");
4294 if (session->send_rtp_sink == NULL)
4297 if (!complete_session_src (rtpbin, session))
4298 goto session_src_failed;
4301 send_rtp_sink = gst_object_ref (session->send_rtp_sink);
4303 GstPad *srcpad = gst_element_get_static_pad (prev, "src");
4304 GstPadLinkReturn ret;
4306 ret = gst_pad_link (srcpad, session->send_rtp_sink);
4307 gst_object_unref (srcpad);
4308 if (ret != GST_PAD_LINK_OK)
4309 goto session_link_failed;
4313 session->send_rtp_sink_ghost =
4314 gst_ghost_pad_new_from_template (name, send_rtp_sink, templ);
4315 gst_object_unref (send_rtp_sink);
4316 gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
4317 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
4319 return session->send_rtp_sink_ghost;
4324 g_warning ("rtpbin: invalid name given");
4329 /* create_session already warned */
4334 g_warning ("rtpbin: session %u is already in use", sessid);
4339 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
4344 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
4349 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4355 g_warning ("rtpbin: failed to get session pad for session %u", sessid);
4360 g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
4363 session_link_failed:
4365 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4371 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4372 " sink pad for session %u", encoder, sessid);
4378 remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4380 if (session->send_rtp_src_ghost) {
4381 gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
4382 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4383 session->send_rtp_src_ghost);
4384 session->send_rtp_src_ghost = NULL;
4386 if (session->send_rtp_sink) {
4387 gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
4388 session->send_rtp_sink);
4389 gst_object_unref (session->send_rtp_sink);
4390 session->send_rtp_sink = NULL;
4392 if (session->send_rtp_sink_ghost) {
4393 gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
4394 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4395 session->send_rtp_sink_ghost);
4396 session->send_rtp_sink_ghost = NULL;
4400 /* Create a pad for sending RTCP for the session in @name. Must be called with
4404 create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
4409 GstElement *encoder;
4410 GstRtpBinSession *session;
4412 /* first get the session number */
4413 if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
4416 /* get or create session */
4417 session = find_session_by_id (rtpbin, sessid);
4419 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
4420 /* create session now */
4421 session = create_session (rtpbin, sessid);
4422 if (session == NULL)
4426 /* check if pad was requested */
4427 if (session->send_rtcp_src_ghost != NULL)
4428 return session->send_rtcp_src_ghost;
4430 /* get rtcp_src pad and store */
4431 session->send_rtcp_src =
4432 gst_element_get_request_pad (session->session, "send_rtcp_src");
4433 if (session->send_rtcp_src == NULL)
4436 GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder");
4437 encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER);
4441 GstPadLinkReturn ret;
4443 GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
4445 ename = g_strdup_printf ("rtcp_src_%u", sessid);
4446 encsrc = gst_element_get_static_pad (encoder, ename);
4449 goto enc_src_failed;
4451 ename = g_strdup_printf ("rtcp_sink_%u", sessid);
4452 encsink = gst_element_get_static_pad (encoder, ename);
4454 if (encsink == NULL)
4455 goto enc_sink_failed;
4457 ret = gst_pad_link (session->send_rtcp_src, encsink);
4458 gst_object_unref (encsink);
4460 if (ret != GST_PAD_LINK_OK)
4461 goto enc_link_failed;
4463 GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given");
4464 encsrc = gst_object_ref (session->send_rtcp_src);
4467 session->send_rtcp_src_ghost =
4468 gst_ghost_pad_new_from_template (name, encsrc, templ);
4469 gst_object_unref (encsrc);
4470 gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
4471 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
4473 return session->send_rtcp_src_ghost;
4478 g_warning ("rtpbin: invalid name given");
4483 /* create_session already warned */
4488 g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
4493 g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
4498 g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
4499 gst_object_unref (encsrc);
4504 g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
4505 gst_object_unref (encsrc);
4511 remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4513 if (session->send_rtcp_src_ghost) {
4514 gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
4515 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4516 session->send_rtcp_src_ghost);
4517 session->send_rtcp_src_ghost = NULL;
4519 if (session->send_rtcp_src) {
4520 gst_element_release_request_pad (session->session, session->send_rtcp_src);
4521 gst_object_unref (session->send_rtcp_src);
4522 session->send_rtcp_src = NULL;
4526 /* If the requested name is NULL we should create a name with
4527 * the session number assuming we want the lowest posible session
4528 * with a free pad like the template */
4530 gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
4532 gboolean name_found = FALSE;
4534 GstIterator *pad_it = NULL;
4535 gchar *pad_name = NULL;
4536 GValue data = { 0, };
4538 GST_DEBUG_OBJECT (element, "find a free pad name for template");
4539 while (!name_found) {
4540 gboolean done = FALSE;
4543 pad_name = g_strdup_printf (templ->name_template, session++);
4544 pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
4547 switch (gst_iterator_next (pad_it, &data)) {
4548 case GST_ITERATOR_OK:
4553 pad = g_value_get_object (&data);
4554 name = gst_pad_get_name (pad);
4556 if (strcmp (name, pad_name) == 0) {
4561 g_value_reset (&data);
4564 case GST_ITERATOR_ERROR:
4565 case GST_ITERATOR_RESYNC:
4566 /* restart iteration */
4571 case GST_ITERATOR_DONE:
4576 g_value_unset (&data);
4577 gst_iterator_free (pad_it);
4580 GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
4587 gst_rtp_bin_request_new_pad (GstElement * element,
4588 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
4591 GstElementClass *klass;
4594 gchar *pad_name = NULL;
4596 g_return_val_if_fail (templ != NULL, NULL);
4597 g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
4599 rtpbin = GST_RTP_BIN (element);
4600 klass = GST_ELEMENT_GET_CLASS (element);
4602 GST_RTP_BIN_LOCK (rtpbin);
4605 /* use a free pad name */
4606 pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
4608 /* use the provided name */
4609 pad_name = g_strdup (name);
4612 GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
4614 /* figure out the template */
4615 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
4616 result = create_recv_rtp (rtpbin, templ, pad_name);
4617 } else if (templ == gst_element_class_get_pad_template (klass,
4618 "recv_rtcp_sink_%u")) {
4619 result = create_recv_rtcp (rtpbin, templ, pad_name);
4620 } else if (templ == gst_element_class_get_pad_template (klass,
4621 "send_rtp_sink_%u")) {
4622 result = create_send_rtp (rtpbin, templ, pad_name);
4623 } else if (templ == gst_element_class_get_pad_template (klass,
4624 "send_rtcp_src_%u")) {
4625 result = create_send_rtcp (rtpbin, templ, pad_name);
4627 goto wrong_template;
4630 GST_RTP_BIN_UNLOCK (rtpbin);
4638 GST_RTP_BIN_UNLOCK (rtpbin);
4639 g_warning ("rtpbin: this is not our template");
4645 gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
4647 GstRtpBinSession *session;
4650 g_return_if_fail (GST_IS_GHOST_PAD (pad));
4651 g_return_if_fail (GST_IS_RTP_BIN (element));
4653 rtpbin = GST_RTP_BIN (element);
4655 GST_RTP_BIN_LOCK (rtpbin);
4656 GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
4657 GST_DEBUG_PAD_NAME (pad));
4659 if (!(session = find_session_by_pad (rtpbin, pad)))
4662 if (session->recv_rtp_sink_ghost == pad) {
4663 remove_recv_rtp (rtpbin, session);
4664 } else if (session->recv_rtcp_sink_ghost == pad) {
4665 remove_recv_rtcp (rtpbin, session);
4666 } else if (session->send_rtp_sink_ghost == pad) {
4667 remove_send_rtp (rtpbin, session);
4668 } else if (session->send_rtcp_src_ghost == pad) {
4669 remove_rtcp (rtpbin, session);
4672 /* no more request pads, free the complete session */
4673 if (session->recv_rtp_sink_ghost == NULL
4674 && session->recv_rtcp_sink_ghost == NULL
4675 && session->send_rtp_sink_ghost == NULL
4676 && session->send_rtcp_src_ghost == NULL) {
4677 GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
4678 rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
4679 free_session (session, rtpbin);
4681 GST_RTP_BIN_UNLOCK (rtpbin);
4688 GST_RTP_BIN_UNLOCK (rtpbin);
4689 g_warning ("rtpbin: %s:%s is not one of our request pads",
4690 GST_DEBUG_PAD_NAME (pad));