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.
67 * <title>Example pipelines</title>
69 * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
70 * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
71 * ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
73 * gst-launch-1.0 rtpbin name=rtpbin \
74 * v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
75 * rtpbin.send_rtp_src_0 ! udpsink port=5000 \
76 * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
77 * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
78 * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
79 * rtpbin.send_rtp_src_1 ! udpsink port=5002 \
80 * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
81 * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
82 * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
83 * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
84 * and the audio is sent to session 1. Video packets are sent on UDP port 5000
85 * and audio packets on port 5002. The video RTCP packets for session 0 are sent
86 * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
87 * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
88 * is received on port 5007. Since RTCP packets from the sender should be sent
89 * as soon as possible and do not participate in preroll, sync=false and
90 * async=false is configured on udpsink
92 * gst-launch-1.0 -v rtpbin name=rtpbin \
93 * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
94 * port=5000 ! rtpbin.recv_rtp_sink_0 \
95 * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
96 * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
97 * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
98 * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
99 * port=5002 ! rtpbin.recv_rtp_sink_1 \
100 * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
101 * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
102 * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
103 * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
104 * decode and display the video.
105 * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
106 * decode and play the audio.
107 * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
108 * session 1 on port 5003. These packets will be used for session management and
110 * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
114 * Last reviewed on 2007-08-30 (0.10.6)
123 #include <gst/rtp/gstrtpbuffer.h>
124 #include <gst/rtp/gstrtcpbuffer.h>
126 #include "gstrtpbin-marshal.h"
127 #include "gstrtpbin.h"
128 #include "rtpsession.h"
129 #include "gstrtpsession.h"
130 #include "gstrtpjitterbuffer.h"
132 #include <gst/glib-compat-private.h>
134 GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
135 #define GST_CAT_DEFAULT gst_rtp_bin_debug
138 static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
139 GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
142 GST_STATIC_CAPS ("application/x-rtp")
145 static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
146 GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
149 GST_STATIC_CAPS ("application/x-rtcp")
152 static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
153 GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
156 GST_STATIC_CAPS ("application/x-rtp")
160 static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
161 GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
164 GST_STATIC_CAPS ("application/x-rtp")
167 static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
168 GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
171 GST_STATIC_CAPS ("application/x-rtcp")
174 static GstStaticPadTemplate rtpbin_send_rtp_src_template =
175 GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
178 GST_STATIC_CAPS ("application/x-rtp")
181 #define GST_RTP_BIN_GET_PRIVATE(obj) \
182 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
184 #define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
185 #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
187 /* lock to protect dynamic callbacks, like pad-added and new ssrc. */
188 #define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock)
189 #define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock)
191 /* lock for shutdown */
192 #define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
194 if (g_atomic_int_get (&bin->priv->shutdown)) \
196 GST_RTP_BIN_DYN_LOCK (bin); \
197 if (g_atomic_int_get (&bin->priv->shutdown)) { \
198 GST_RTP_BIN_DYN_UNLOCK (bin); \
203 /* unlock for shutdown */
204 #define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
205 GST_RTP_BIN_DYN_UNLOCK (bin); \
207 struct _GstRtpBinPrivate
211 /* lock protecting dynamic adding/removing */
214 /* if we are shutting down or not */
219 /* UNIX (ntp) time of last SR sync used */
223 /* signals and args */
226 SIGNAL_REQUEST_PT_MAP,
227 SIGNAL_PAYLOAD_TYPE_CHANGE,
230 SIGNAL_GET_INTERNAL_SESSION,
233 SIGNAL_ON_SSRC_COLLISION,
234 SIGNAL_ON_SSRC_VALIDATED,
235 SIGNAL_ON_SSRC_ACTIVE,
238 SIGNAL_ON_BYE_TIMEOUT,
240 SIGNAL_ON_SENDER_TIMEOUT,
245 #define DEFAULT_LATENCY_MS 200
246 #define DEFAULT_DROP_ON_LATENCY FALSE
247 #define DEFAULT_SDES NULL
248 #define DEFAULT_DO_LOST FALSE
249 #define DEFAULT_IGNORE_PT FALSE
250 #define DEFAULT_NTP_SYNC FALSE
251 #define DEFAULT_AUTOREMOVE FALSE
252 #define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
253 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
254 #define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
255 #define DEFAULT_RTCP_SYNC_INTERVAL 0
261 PROP_DROP_ON_LATENCY,
267 PROP_RTCP_SYNC_INTERVAL,
270 PROP_USE_PIPELINE_CLOCK,
276 GST_RTP_BIN_RTCP_SYNC_ALWAYS,
277 GST_RTP_BIN_RTCP_SYNC_INITIAL,
278 GST_RTP_BIN_RTCP_SYNC_RTP
281 #define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
283 gst_rtp_bin_rtcp_sync_get_type (void)
285 static GType rtcp_sync_type = 0;
286 static const GEnumValue rtcp_sync_types[] = {
287 {GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
288 {GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
289 {GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
293 if (!rtcp_sync_type) {
294 rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
296 return rtcp_sync_type;
300 typedef struct _GstRtpBinSession GstRtpBinSession;
301 typedef struct _GstRtpBinStream GstRtpBinStream;
302 typedef struct _GstRtpBinClient GstRtpBinClient;
304 static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
306 static GstCaps *pt_map_requested (GstElement * element, guint pt,
307 GstRtpBinSession * session);
308 static void payload_type_change (GstElement * element, guint pt,
309 GstRtpBinSession * session);
310 static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
311 static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
312 static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
313 static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
314 static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
315 static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
317 /* Manages the RTP stream for one SSRC.
319 * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
320 * If we see an SDES RTCP packet that links multiple SSRCs together based on a
321 * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
322 * together (see below).
324 struct _GstRtpBinStream
326 /* the SSRC of this stream */
332 /* the session this SSRC belongs to */
333 GstRtpBinSession *session;
335 /* the jitterbuffer of the SSRC */
337 gulong buffer_handlesync_sig;
338 gulong buffer_ptreq_sig;
339 gulong buffer_ntpstop_sig;
342 /* the PT demuxer of the SSRC */
344 gulong demux_newpad_sig;
345 gulong demux_padremoved_sig;
346 gulong demux_ptreq_sig;
347 gulong demux_ptchange_sig;
349 /* if we have calculated a valid rt_delta for this stream */
351 /* mapping to local RTP and NTP time */
354 /* base rtptime in gst time */
358 #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
359 #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
361 /* Manages the receiving end of the packets.
363 * There is one such structure for each RTP session (audio/video/...).
364 * We get the RTP/RTCP packets and stuff them into the session manager. From
365 * there they are pushed into an SSRC demuxer that splits the stream based on
366 * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
367 * the GstRtpBinStream above).
369 struct _GstRtpBinSession
375 /* the session element */
377 /* the SSRC demuxer */
379 gulong demux_newpad_sig;
380 gulong demux_padremoved_sig;
384 /* list of GstRtpBinStream */
387 /* mapping of payload type to caps */
390 /* the pads of the session */
391 GstPad *recv_rtp_sink;
392 GstPad *recv_rtp_sink_ghost;
393 GstPad *recv_rtp_src;
394 GstPad *recv_rtcp_sink;
395 GstPad *recv_rtcp_sink_ghost;
397 GstPad *send_rtp_sink;
398 GstPad *send_rtp_sink_ghost;
399 GstPad *send_rtp_src;
400 GstPad *send_rtp_src_ghost;
401 GstPad *send_rtcp_src;
402 GstPad *send_rtcp_src_ghost;
405 /* Manages the RTP streams that come from one client and should therefore be
408 struct _GstRtpBinClient
410 /* the common CNAME for the streams */
419 /* find a session with the given id. Must be called with RTP_BIN_LOCK */
420 static GstRtpBinSession *
421 find_session_by_id (GstRtpBin * rtpbin, gint id)
425 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
426 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
434 /* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
435 static GstRtpBinSession *
436 find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
440 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
441 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
443 if ((sess->recv_rtp_sink_ghost == pad) ||
444 (sess->recv_rtcp_sink_ghost == pad) ||
445 (sess->send_rtp_sink_ghost == pad)
446 || (sess->send_rtcp_src_ghost == pad))
453 on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
455 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
460 on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
462 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
467 on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
469 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
474 on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
476 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
481 on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
483 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
488 on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
490 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
495 on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
497 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
500 if (sess->bin->priv->autoremove)
501 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
505 on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
507 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
510 if (sess->bin->priv->autoremove)
511 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
515 on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
517 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
522 on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
524 g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
525 stream->session->id, stream->ssrc);
528 /* must be called with the SESSION lock */
529 static GstRtpBinStream *
530 find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
534 for (walk = session->streams; walk; walk = g_slist_next (walk)) {
535 GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
537 if (stream->ssrc == ssrc)
544 ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
545 GstRtpBinSession * session)
547 GstRtpBinStream *stream = NULL;
550 rtpbin = session->bin;
552 GST_RTP_BIN_LOCK (rtpbin);
554 GST_RTP_SESSION_LOCK (session);
555 if ((stream = find_stream_by_ssrc (session, ssrc)))
556 session->streams = g_slist_remove (session->streams, stream);
557 GST_RTP_SESSION_UNLOCK (session);
560 free_stream (stream, rtpbin);
562 GST_RTP_BIN_UNLOCK (rtpbin);
565 /* create a session with the given id. Must be called with RTP_BIN_LOCK */
566 static GstRtpBinSession *
567 create_session (GstRtpBin * rtpbin, gint id)
569 GstRtpBinSession *sess;
570 GstElement *session, *demux;
573 if (!(session = gst_element_factory_make ("rtpsession", NULL)))
576 if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
579 sess = g_new0 (GstRtpBinSession, 1);
580 g_mutex_init (&sess->lock);
583 sess->session = session;
585 sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
586 (GDestroyNotify) gst_caps_unref);
587 rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
589 /* configure SDES items */
590 GST_OBJECT_LOCK (rtpbin);
591 g_object_set (session, "sdes", rtpbin->sdes, "use-pipeline-clock",
592 rtpbin->use_pipeline_clock, NULL);
593 GST_OBJECT_UNLOCK (rtpbin);
595 /* provide clock_rate to the session manager when needed */
596 g_signal_connect (session, "request-pt-map",
597 (GCallback) pt_map_requested, sess);
599 g_signal_connect (sess->session, "on-new-ssrc",
600 (GCallback) on_new_ssrc, sess);
601 g_signal_connect (sess->session, "on-ssrc-collision",
602 (GCallback) on_ssrc_collision, sess);
603 g_signal_connect (sess->session, "on-ssrc-validated",
604 (GCallback) on_ssrc_validated, sess);
605 g_signal_connect (sess->session, "on-ssrc-active",
606 (GCallback) on_ssrc_active, sess);
607 g_signal_connect (sess->session, "on-ssrc-sdes",
608 (GCallback) on_ssrc_sdes, sess);
609 g_signal_connect (sess->session, "on-bye-ssrc",
610 (GCallback) on_bye_ssrc, sess);
611 g_signal_connect (sess->session, "on-bye-timeout",
612 (GCallback) on_bye_timeout, sess);
613 g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
614 g_signal_connect (sess->session, "on-sender-timeout",
615 (GCallback) on_sender_timeout, sess);
617 gst_bin_add (GST_BIN_CAST (rtpbin), session);
618 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
620 GST_OBJECT_LOCK (rtpbin);
621 target = GST_STATE_TARGET (rtpbin);
622 GST_OBJECT_UNLOCK (rtpbin);
624 /* change state only to what's needed */
625 gst_element_set_state (demux, target);
626 gst_element_set_state (session, target);
633 g_warning ("rtpbin: could not create gstrtpsession element");
638 gst_object_unref (session);
639 g_warning ("rtpbin: could not create gstrtpssrcdemux element");
644 /* called with RTP_BIN_LOCK */
646 free_session (GstRtpBinSession * sess, GstRtpBin * bin)
648 GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
650 gst_element_set_locked_state (sess->demux, TRUE);
651 gst_element_set_locked_state (sess->session, TRUE);
653 gst_element_set_state (sess->demux, GST_STATE_NULL);
654 gst_element_set_state (sess->session, GST_STATE_NULL);
656 remove_recv_rtp (bin, sess);
657 remove_recv_rtcp (bin, sess);
658 remove_send_rtp (bin, sess);
659 remove_rtcp (bin, sess);
661 gst_bin_remove (GST_BIN_CAST (bin), sess->session);
662 gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
664 g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
665 g_slist_free (sess->streams);
667 g_mutex_clear (&sess->lock);
668 g_hash_table_destroy (sess->ptmap);
673 /* get the payload type caps for the specific payload @pt in @session */
675 get_pt_map (GstRtpBinSession * session, guint pt)
677 GstCaps *caps = NULL;
680 GValue args[3] = { {0}, {0}, {0} };
682 GST_DEBUG ("searching pt %d in cache", pt);
684 GST_RTP_SESSION_LOCK (session);
686 /* first look in the cache */
687 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
695 GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
697 /* not in cache, send signal to request caps */
698 g_value_init (&args[0], GST_TYPE_ELEMENT);
699 g_value_set_object (&args[0], bin);
700 g_value_init (&args[1], G_TYPE_UINT);
701 g_value_set_uint (&args[1], session->id);
702 g_value_init (&args[2], G_TYPE_UINT);
703 g_value_set_uint (&args[2], pt);
705 g_value_init (&ret, GST_TYPE_CAPS);
706 g_value_set_boxed (&ret, NULL);
708 GST_RTP_SESSION_UNLOCK (session);
710 g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
712 GST_RTP_SESSION_LOCK (session);
714 g_value_unset (&args[0]);
715 g_value_unset (&args[1]);
716 g_value_unset (&args[2]);
718 /* look in the cache again because we let the lock go */
719 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
722 g_value_unset (&ret);
726 caps = (GstCaps *) g_value_dup_boxed (&ret);
727 g_value_unset (&ret);
731 GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
733 /* store in cache, take additional ref */
734 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
735 gst_caps_ref (caps));
738 GST_RTP_SESSION_UNLOCK (session);
745 GST_RTP_SESSION_UNLOCK (session);
746 GST_DEBUG ("no pt map could be obtained");
752 return_true (gpointer key, gpointer value, gpointer user_data)
758 gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
760 GSList *clients, *streams;
762 GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
764 GST_RTP_BIN_LOCK (rtpbin);
765 for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
766 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
768 /* reset sync on all streams for this client */
769 for (streams = client->streams; streams; streams = g_slist_next (streams)) {
770 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
772 /* make use require a new SR packet for this stream before we attempt new
774 stream->have_sync = FALSE;
775 stream->rt_delta = 0;
776 stream->rtp_delta = 0;
777 stream->clock_base = -100 * GST_SECOND;
780 GST_RTP_BIN_UNLOCK (rtpbin);
784 gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
786 GSList *sessions, *streams;
788 GST_RTP_BIN_LOCK (bin);
789 GST_DEBUG_OBJECT (bin, "clearing pt map");
790 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
791 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
793 GST_DEBUG_OBJECT (bin, "clearing session %p", session);
794 g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
796 GST_RTP_SESSION_LOCK (session);
797 g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
799 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
800 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
802 GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
803 g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
805 g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
807 GST_RTP_SESSION_UNLOCK (session);
809 GST_RTP_BIN_UNLOCK (bin);
812 gst_rtp_bin_reset_sync (bin);
816 gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
818 RTPSession *internal_session = NULL;
819 GstRtpBinSession *session;
821 GST_RTP_BIN_LOCK (bin);
822 GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
824 session = find_session_by_id (bin, (gint) session_id);
826 g_object_get (session->session, "internal-session", &internal_session,
829 GST_RTP_BIN_UNLOCK (bin);
831 return internal_session;
835 gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
836 const gchar * name, const GValue * value)
838 GSList *sessions, *streams;
840 GST_RTP_BIN_LOCK (bin);
841 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
842 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
844 GST_RTP_SESSION_LOCK (session);
845 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
846 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
848 g_object_set_property (G_OBJECT (stream->buffer), name, value);
850 GST_RTP_SESSION_UNLOCK (session);
852 GST_RTP_BIN_UNLOCK (bin);
855 /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
856 static GstRtpBinClient *
857 get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
859 GstRtpBinClient *result = NULL;
862 for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
863 GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
865 if (len != client->cname_len)
868 if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
869 GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
876 /* nothing found, create one */
877 if (result == NULL) {
878 result = g_new0 (GstRtpBinClient, 1);
879 result->cname = g_strndup ((gchar *) data, len);
880 result->cname_len = len;
881 bin->clients = g_slist_prepend (bin->clients, result);
882 GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
889 free_client (GstRtpBinClient * client, GstRtpBin * bin)
891 GST_DEBUG_OBJECT (bin, "freeing client %p", client);
892 g_slist_free (client->streams);
893 g_free (client->cname);
898 get_current_times (GstRtpBin * bin, GstClockTime * running_time,
903 GstClockTime base_time, rt, clock_time;
905 GST_OBJECT_LOCK (bin);
906 if ((clock = GST_ELEMENT_CLOCK (bin))) {
907 base_time = GST_ELEMENT_CAST (bin)->base_time;
908 gst_object_ref (clock);
909 GST_OBJECT_UNLOCK (bin);
911 clock_time = gst_clock_get_time (clock);
913 if (bin->use_pipeline_clock) {
914 ntpns = clock_time - base_time;
918 /* get current NTP time */
919 g_get_current_time (¤t);
920 ntpns = GST_TIMEVAL_TO_TIME (current);
923 /* add constant to convert from 1970 based time to 1900 based time */
924 ntpns += (2208988800LL * GST_SECOND);
926 /* get current clock time and convert to running time */
927 rt = clock_time - base_time;
929 gst_object_unref (clock);
931 GST_OBJECT_UNLOCK (bin);
942 stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
943 gint64 ts_offset, gboolean check)
945 gint64 prev_ts_offset;
947 g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
949 /* delta changed, see how much */
950 if (prev_ts_offset != ts_offset) {
953 diff = prev_ts_offset - ts_offset;
955 GST_DEBUG_OBJECT (bin,
956 "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
957 ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
960 /* only change diff when it changed more than 4 milliseconds. This
961 * compensates for rounding errors in NTP to RTP timestamp
963 if (ABS (diff) < 4 * GST_MSECOND) {
964 GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
967 if (ABS (diff) > (3 * GST_SECOND)) {
968 GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
972 g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
974 GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
975 stream->ssrc, ts_offset);
978 /* associate a stream to the given CNAME. This will make sure all streams for
979 * that CNAME are synchronized together.
980 * Must be called with GST_RTP_BIN_LOCK */
982 gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
983 guint8 * data, guint64 ntptime, guint64 last_extrtptime,
984 guint64 base_rtptime, guint64 base_time, guint clock_rate,
985 gint64 rtp_clock_base)
987 GstRtpBinClient *client;
992 GstClockTime running_time;
994 gint64 ntpdiff, rtdiff;
997 /* first find or create the CNAME */
998 client = get_client (bin, len, data, &created);
1000 /* find stream in the client */
1001 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1002 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1004 if (ostream == stream)
1007 /* not found, add it to the list */
1009 GST_DEBUG_OBJECT (bin,
1010 "new association of SSRC %08x with client %p with CNAME %s",
1011 stream->ssrc, client, client->cname);
1012 client->streams = g_slist_prepend (client->streams, stream);
1015 GST_DEBUG_OBJECT (bin,
1016 "found association of SSRC %08x with client %p with CNAME %s",
1017 stream->ssrc, client, client->cname);
1020 if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
1021 GST_DEBUG_OBJECT (bin, "invalidated sync data");
1022 if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1023 /* we don't need that data, so carry on,
1024 * but make some values look saner */
1025 last_extrtptime = base_rtptime;
1027 /* nothing we can do with this data in this case */
1028 GST_DEBUG_OBJECT (bin, "bailing out");
1033 /* Take the extended rtptime we found in the SR packet and map it to the
1034 * local rtptime. The local rtp time is used to construct timestamps on the
1035 * buffers so we will calculate what running_time corresponds to the RTP
1036 * timestamp in the SR packet. */
1037 local_rtp = last_extrtptime - base_rtptime;
1039 GST_DEBUG_OBJECT (bin,
1040 "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
1041 ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
1042 "clock-base %" G_GINT64_FORMAT, base_rtptime,
1043 last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
1045 /* calculate local RTP time in gstreamer timestamp, we essentially perform the
1046 * same conversion that a jitterbuffer would use to convert an rtp timestamp
1047 * into a corresponding gstreamer timestamp. Note that the base_time also
1048 * contains the drift between sender and receiver. */
1049 local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
1050 local_rt += base_time;
1052 /* convert ntptime to unix time since 1900 */
1053 last_unix = gst_util_uint64_scale (ntptime, GST_SECOND,
1054 (G_GINT64_CONSTANT (1) << 32));
1056 stream->have_sync = TRUE;
1058 GST_DEBUG_OBJECT (bin,
1059 "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT,
1060 local_rt, last_unix);
1062 /* recalc inter stream playout offset, but only if there is more than one
1063 * stream or we're doing NTP sync. */
1064 if (bin->ntp_sync) {
1065 /* For NTP sync we need to first get a snapshot of running_time and NTP
1066 * time. We know at what running_time we play a certain RTP time, we also
1067 * calculated when we would play the RTP time in the SR packet. Now we need
1068 * to know how the running_time and the NTP time relate to eachother. */
1069 get_current_times (bin, &running_time, &ntpnstime);
1071 /* see how far away the NTP time is. This is the difference between the
1072 * current NTP time and the NTP time in the last SR packet. */
1073 ntpdiff = ntpnstime - last_unix;
1074 /* see how far away the running_time is. This is the difference between the
1075 * current running_time and the running_time of the RTP timestamp in the
1076 * last SR packet. */
1077 rtdiff = running_time - local_rt;
1079 GST_DEBUG_OBJECT (bin,
1080 "NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT,
1081 ntpnstime, last_unix);
1082 GST_DEBUG_OBJECT (bin,
1083 "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
1086 /* combine to get the final diff to apply to the running_time */
1087 stream->rt_delta = rtdiff - ntpdiff;
1089 stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE);
1091 gint64 min, rtp_min, clock_base = stream->clock_base;
1092 gboolean all_sync, use_rtp;
1093 gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
1095 /* calculate delta between server and receiver. last_unix is created by
1096 * converting the ntptime in the last SR packet to a gstreamer timestamp. This
1097 * delta expresses the difference to our timeline and the server timeline. The
1098 * difference in itself doesn't mean much but we can combine the delta of
1099 * multiple streams to create a stream specific offset. */
1100 stream->rt_delta = last_unix - local_rt;
1102 /* calculate the min of all deltas, ignoring streams that did not yet have a
1103 * valid rt_delta because we did not yet receive an SR packet for those
1105 * We calculate the mininum because we would like to only apply positive
1106 * offsets to streams, delaying their playback instead of trying to speed up
1107 * other streams (which might be imposible when we have to create negative
1109 * The stream that has the smallest diff is selected as the reference stream,
1110 * all other streams will have a positive offset to this difference. */
1112 /* some alternative setting allow ignoring RTCP as much as possible,
1113 * for servers generating bogus ntp timeline */
1114 min = rtp_min = G_MAXINT64;
1116 if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1120 /* signed version for convienience */
1121 clock_base = base_rtptime;
1122 /* deal with possible wrap-around */
1123 ext_base = base_rtptime;
1124 rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
1125 /* sanity check; base rtp and provided clock_base should be close */
1126 if (rtp_clock_base >= clock_base) {
1127 if (rtp_clock_base - clock_base < 10 * clock_rate) {
1128 rtp_clock_base = base_time +
1129 gst_util_uint64_scale_int (rtp_clock_base - clock_base,
1130 GST_SECOND, clock_rate);
1135 if (clock_base - rtp_clock_base < 10 * clock_rate) {
1136 rtp_clock_base = base_time -
1137 gst_util_uint64_scale_int (clock_base - rtp_clock_base,
1138 GST_SECOND, clock_rate);
1143 /* warn and bail for clarity out if no sane values */
1145 GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
1148 /* store to track changes */
1149 clock_base = rtp_clock_base;
1150 /* generate a fake as before,
1151 * now equating rtptime obtained from RTP-Info,
1152 * where the large time represent the otherwise irrelevant npt/ntp time */
1153 stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
1155 clock_base = rtp_clock_base;
1159 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1160 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1162 if (!ostream->have_sync) {
1167 /* change in current stream's base from previously init'ed value
1168 * leads to reset of all stream's base */
1169 if (stream != ostream && stream->clock_base >= 0 &&
1170 (stream->clock_base != clock_base)) {
1171 GST_DEBUG_OBJECT (bin, "reset upon clock base change");
1172 ostream->clock_base = -100 * GST_SECOND;
1173 ostream->rtp_delta = 0;
1176 if (ostream->rt_delta < min)
1177 min = ostream->rt_delta;
1178 if (ostream->rtp_delta < rtp_min)
1179 rtp_min = ostream->rtp_delta;
1182 /* arrange to re-sync for each stream upon significant change,
1184 all_sync = all_sync && (stream->clock_base == clock_base);
1185 stream->clock_base = clock_base;
1187 /* may need init performed above later on, but nothing more to do now */
1188 if (client->nstreams <= 1)
1191 GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
1192 " all sync %d", client, min, all_sync);
1193 GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
1195 switch (rtcp_sync) {
1196 case GST_RTP_BIN_RTCP_SYNC_RTP:
1199 GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
1200 "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
1202 case GST_RTP_BIN_RTCP_SYNC_INITIAL:
1203 /* if all have been synced already, do not bother further */
1205 GST_DEBUG_OBJECT (bin, "all streams already synced; done");
1213 /* bail out if we adjusted recently enough */
1214 if (all_sync && (last_unix - bin->priv->last_unix) <
1215 bin->rtcp_sync_interval * GST_MSECOND) {
1216 GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
1217 "previous sender info too recent "
1218 "(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix);
1221 bin->priv->last_unix = last_unix;
1223 /* calculate offsets for each stream */
1224 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1225 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1228 /* ignore streams for which we didn't receive an SR packet yet, we
1229 * can't synchronize them yet. We can however sync other streams just
1231 if (!ostream->have_sync)
1234 /* calculate offset to our reference stream, this should always give a
1235 * positive number. */
1237 ts_offset = ostream->rtp_delta - rtp_min;
1239 ts_offset = ostream->rt_delta - min;
1241 stream_set_ts_offset (bin, ostream, ts_offset, TRUE);
1247 #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
1248 for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
1249 (b) = gst_rtcp_packet_move_to_next ((packet)))
1251 #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
1252 for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
1253 (b) = gst_rtcp_packet_sdes_next_item ((packet)))
1255 #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
1256 for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
1257 (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
1260 gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
1261 GstRtpBinStream * stream)
1264 GstRTCPPacket packet;
1267 gboolean have_sr, have_sdes;
1269 guint64 base_rtptime;
1275 GstRTCPBuffer rtcp = { NULL, };
1279 GST_DEBUG_OBJECT (bin, "sync handler called");
1281 /* get the last relation between the rtp timestamps and the gstreamer
1282 * timestamps. We get this info directly from the jitterbuffer which
1283 * constructs gstreamer timestamps from rtp timestamps and so it know exactly
1284 * what the current situation is. */
1286 g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
1287 base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
1288 clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
1289 clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
1291 g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
1292 buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
1297 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
1299 GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
1300 /* first packet must be SR or RR or else the validate would have failed */
1301 switch (gst_rtcp_packet_get_type (&packet)) {
1302 case GST_RTCP_TYPE_SR:
1303 /* only parse first. There is only supposed to be one SR in the packet
1304 * but we will deal with malformed packets gracefully */
1307 /* get NTP and RTP times */
1308 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
1311 GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
1312 /* ignore SR that is not ours */
1313 if (ssrc != stream->ssrc)
1318 case GST_RTCP_TYPE_SDES:
1320 gboolean more_items, more_entries;
1322 /* only deal with first SDES, there is only supposed to be one SDES in
1323 * the RTCP packet but we deal with bad packets gracefully. Also bail
1324 * out if we have not seen an SR item yet. */
1325 if (have_sdes || !have_sr)
1328 GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
1329 /* skip items that are not about the SSRC of the sender */
1330 if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
1333 /* find the CNAME entry */
1334 GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
1335 GstRTCPSDESType type;
1339 gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
1341 if (type == GST_RTCP_SDES_CNAME) {
1342 GST_RTP_BIN_LOCK (bin);
1343 /* associate the stream to CNAME */
1344 gst_rtp_bin_associate (bin, stream, len, data,
1345 ntptime, extrtptime, base_rtptime, base_time, clock_rate,
1347 GST_RTP_BIN_UNLOCK (bin);
1355 /* we can ignore these packets */
1359 gst_rtcp_buffer_unmap (&rtcp);
1362 /* create a new stream with @ssrc in @session. Must be called with
1363 * RTP_SESSION_LOCK. */
1364 static GstRtpBinStream *
1365 create_stream (GstRtpBinSession * session, guint32 ssrc)
1367 GstElement *buffer, *demux = NULL;
1368 GstRtpBinStream *stream;
1372 rtpbin = session->bin;
1374 if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
1375 goto no_jitterbuffer;
1377 if (!rtpbin->ignore_pt)
1378 if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
1382 stream = g_new0 (GstRtpBinStream, 1);
1383 stream->ssrc = ssrc;
1384 stream->bin = rtpbin;
1385 stream->session = session;
1386 stream->buffer = buffer;
1387 stream->demux = demux;
1389 stream->have_sync = FALSE;
1390 stream->rt_delta = 0;
1391 stream->rtp_delta = 0;
1392 stream->percent = 100;
1393 stream->clock_base = -100 * GST_SECOND;
1394 session->streams = g_slist_prepend (session->streams, stream);
1396 /* provide clock_rate to the jitterbuffer when needed */
1397 stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
1398 (GCallback) pt_map_requested, session);
1399 stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
1400 (GCallback) on_npt_stop, stream);
1402 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
1403 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
1405 /* configure latency and packet lost */
1406 g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
1407 g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
1408 g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
1409 g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
1411 if (!rtpbin->ignore_pt)
1412 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
1413 gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
1417 gst_element_link (buffer, demux);
1419 if (rtpbin->buffering) {
1422 GST_INFO_OBJECT (rtpbin,
1423 "bin is buffering, set jitterbuffer as not active");
1424 g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
1428 GST_OBJECT_LOCK (rtpbin);
1429 target = GST_STATE_TARGET (rtpbin);
1430 GST_OBJECT_UNLOCK (rtpbin);
1432 /* from sink to source */
1434 gst_element_set_state (demux, target);
1436 gst_element_set_state (buffer, target);
1443 g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
1448 gst_object_unref (buffer);
1449 g_warning ("rtpbin: could not create gstrtpptdemux element");
1454 /* called with RTP_BIN_LOCK */
1456 free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
1458 GSList *clients, *next_client;
1460 GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
1462 if (stream->demux) {
1463 g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
1464 g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
1465 g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
1467 g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
1468 g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
1469 g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
1472 gst_element_set_locked_state (stream->demux, TRUE);
1473 gst_element_set_locked_state (stream->buffer, TRUE);
1476 gst_element_set_state (stream->demux, GST_STATE_NULL);
1477 gst_element_set_state (stream->buffer, GST_STATE_NULL);
1479 /* now remove this signal, we need this while going to NULL because it to
1480 * do some cleanups */
1482 g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
1484 gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
1486 gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
1488 for (clients = bin->clients; clients; clients = next_client) {
1489 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
1490 GSList *streams, *next_stream;
1492 next_client = g_slist_next (clients);
1494 for (streams = client->streams; streams; streams = next_stream) {
1495 GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
1497 next_stream = g_slist_next (streams);
1499 if (ostream == stream) {
1500 client->streams = g_slist_delete_link (client->streams, streams);
1501 /* If this was the last stream belonging to this client,
1502 * clean up the client. */
1503 if (--client->nstreams == 0) {
1504 bin->clients = g_slist_delete_link (bin->clients, clients);
1505 free_client (client, bin);
1514 /* GObject vmethods */
1515 static void gst_rtp_bin_dispose (GObject * object);
1516 static void gst_rtp_bin_finalize (GObject * object);
1517 static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
1518 const GValue * value, GParamSpec * pspec);
1519 static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
1520 GValue * value, GParamSpec * pspec);
1522 /* GstElement vmethods */
1523 static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
1524 GstStateChange transition);
1525 static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
1526 GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
1527 static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
1528 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
1530 #define gst_rtp_bin_parent_class parent_class
1531 G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
1534 gst_rtp_bin_class_init (GstRtpBinClass * klass)
1536 GObjectClass *gobject_class;
1537 GstElementClass *gstelement_class;
1538 GstBinClass *gstbin_class;
1540 gobject_class = (GObjectClass *) klass;
1541 gstelement_class = (GstElementClass *) klass;
1542 gstbin_class = (GstBinClass *) klass;
1544 g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
1546 gobject_class->dispose = gst_rtp_bin_dispose;
1547 gobject_class->finalize = gst_rtp_bin_finalize;
1548 gobject_class->set_property = gst_rtp_bin_set_property;
1549 gobject_class->get_property = gst_rtp_bin_get_property;
1551 g_object_class_install_property (gobject_class, PROP_LATENCY,
1552 g_param_spec_uint ("latency", "Buffer latency in ms",
1553 "Default amount of ms to buffer in the jitterbuffers", 0,
1554 G_MAXUINT, DEFAULT_LATENCY_MS,
1555 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1557 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
1558 g_param_spec_boolean ("drop-on-latency",
1559 "Drop buffers when maximum latency is reached",
1560 "Tells the jitterbuffer to never exceed the given latency in size",
1561 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1564 * GstRtpBin::request-pt-map:
1565 * @rtpbin: the object which received the signal
1566 * @session: the session
1569 * Request the payload type as #GstCaps for @pt in @session.
1571 gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
1572 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
1573 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
1574 NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
1575 G_TYPE_UINT, G_TYPE_UINT);
1578 * GstRtpBin::payload-type-change:
1579 * @rtpbin: the object which received the signal
1580 * @session: the session
1583 * Signal that the current payload type changed to @pt in @session.
1587 gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
1588 g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
1589 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
1590 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1591 G_TYPE_UINT, G_TYPE_UINT);
1594 * GstRtpBin::clear-pt-map:
1595 * @rtpbin: the object which received the signal
1597 * Clear all previously cached pt-mapping obtained with
1598 * #GstRtpBin::request-pt-map.
1600 gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
1601 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
1602 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1603 clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1607 * GstRtpBin::reset-sync:
1608 * @rtpbin: the object which received the signal
1610 * Reset all currently configured lip-sync parameters and require new SR
1611 * packets for all streams before lip-sync is attempted again.
1613 gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
1614 g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
1615 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1616 reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1620 * GstRtpBin::get-internal-session:
1621 * @rtpbin: the object which received the signal
1622 * @id: the session id
1624 * Request the internal RTPSession object as #GObject in session @id.
1626 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
1627 g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
1628 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1629 get_internal_session), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
1630 RTP_TYPE_SESSION, 1, G_TYPE_UINT);
1633 * GstRtpBin::on-new-ssrc:
1634 * @rtpbin: the object which received the signal
1635 * @session: the session
1638 * Notify of a new SSRC that entered @session.
1640 gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
1641 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
1642 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
1643 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1644 G_TYPE_UINT, G_TYPE_UINT);
1646 * GstRtpBin::on-ssrc-collision:
1647 * @rtpbin: the object which received the signal
1648 * @session: the session
1651 * Notify when we have an SSRC collision
1653 gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
1654 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
1655 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
1656 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1657 G_TYPE_UINT, G_TYPE_UINT);
1659 * GstRtpBin::on-ssrc-validated:
1660 * @rtpbin: the object which received the signal
1661 * @session: the session
1664 * Notify of a new SSRC that became validated.
1666 gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
1667 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
1668 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
1669 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1670 G_TYPE_UINT, G_TYPE_UINT);
1672 * GstRtpBin::on-ssrc-active:
1673 * @rtpbin: the object which received the signal
1674 * @session: the session
1677 * Notify of a SSRC that is active, i.e., sending RTCP.
1679 gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
1680 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
1681 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
1682 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1683 G_TYPE_UINT, G_TYPE_UINT);
1685 * GstRtpBin::on-ssrc-sdes:
1686 * @rtpbin: the object which received the signal
1687 * @session: the session
1690 * Notify of a SSRC that is active, i.e., sending RTCP.
1692 gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
1693 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
1694 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
1695 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1696 G_TYPE_UINT, G_TYPE_UINT);
1699 * GstRtpBin::on-bye-ssrc:
1700 * @rtpbin: the object which received the signal
1701 * @session: the session
1704 * Notify of an SSRC that became inactive because of a BYE packet.
1706 gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
1707 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
1708 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
1709 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1710 G_TYPE_UINT, G_TYPE_UINT);
1712 * GstRtpBin::on-bye-timeout:
1713 * @rtpbin: the object which received the signal
1714 * @session: the session
1717 * Notify of an SSRC that has timed out because of BYE
1719 gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
1720 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
1721 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
1722 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1723 G_TYPE_UINT, G_TYPE_UINT);
1725 * GstRtpBin::on-timeout:
1726 * @rtpbin: the object which received the signal
1727 * @session: the session
1730 * Notify of an SSRC that has timed out
1732 gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
1733 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
1734 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
1735 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1736 G_TYPE_UINT, G_TYPE_UINT);
1738 * GstRtpBin::on-sender-timeout:
1739 * @rtpbin: the object which received the signal
1740 * @session: the session
1743 * Notify of a sender SSRC that has timed out and became a receiver
1745 gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
1746 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
1747 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
1748 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1749 G_TYPE_UINT, G_TYPE_UINT);
1752 * GstRtpBin::on-npt-stop:
1753 * @rtpbin: the object which received the signal
1754 * @session: the session
1757 * Notify that SSRC sender has sent data up to the configured NPT stop time.
1759 gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
1760 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
1761 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
1762 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1763 G_TYPE_UINT, G_TYPE_UINT);
1765 g_object_class_install_property (gobject_class, PROP_SDES,
1766 g_param_spec_boxed ("sdes", "SDES",
1767 "The SDES items of this session",
1768 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1770 g_object_class_install_property (gobject_class, PROP_DO_LOST,
1771 g_param_spec_boolean ("do-lost", "Do Lost",
1772 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
1773 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1775 g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
1776 g_param_spec_boolean ("autoremove", "Auto Remove",
1777 "Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
1778 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1780 g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
1781 g_param_spec_boolean ("ignore-pt", "Ignore PT",
1782 "Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
1783 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1785 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
1786 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
1787 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
1788 DEFAULT_USE_PIPELINE_CLOCK,
1789 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1791 * GstRtpBin::buffer-mode:
1793 * Control the buffering and timestamping mode used by the jitterbuffer.
1797 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
1798 g_param_spec_enum ("buffer-mode", "Buffer Mode",
1799 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
1800 DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1802 * GstRtpBin::ntp-sync:
1804 * Set the NTP time from the sender reports as the running-time on the
1805 * buffers. When both the sender and receiver have sychronized
1806 * running-time, i.e. when the clock and base-time is shared
1807 * between the receivers and the and the senders, this option can be
1808 * used to synchronize receivers on multiple machines.
1812 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
1813 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
1814 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
1815 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1818 * GstRtpBin::rtcp-sync:
1820 * If not synchronizing (directly) to the NTP clock, determines how to sync
1821 * the various streams.
1825 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
1826 g_param_spec_enum ("rtcp-sync", "RTCP Sync",
1827 "Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
1828 DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1831 * GstRtpBin::rtcp-sync-interval:
1833 * Determines how often to sync streams using RTCP data.
1837 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
1838 g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
1839 "RTCP SR interval synchronization (ms) (0 = always)",
1840 0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
1841 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1843 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
1844 gstelement_class->request_new_pad =
1845 GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
1846 gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
1849 gst_element_class_add_pad_template (gstelement_class,
1850 gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
1851 gst_element_class_add_pad_template (gstelement_class,
1852 gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
1853 gst_element_class_add_pad_template (gstelement_class,
1854 gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
1857 gst_element_class_add_pad_template (gstelement_class,
1858 gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
1859 gst_element_class_add_pad_template (gstelement_class,
1860 gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
1861 gst_element_class_add_pad_template (gstelement_class,
1862 gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
1864 gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
1865 "Filter/Network/RTP",
1866 "Real-Time Transport Protocol bin",
1867 "Wim Taymans <wim.taymans@gmail.com>");
1869 gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
1871 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
1872 klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
1873 klass->get_internal_session =
1874 GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
1876 GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
1880 gst_rtp_bin_init (GstRtpBin * rtpbin)
1884 rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
1885 g_mutex_init (&rtpbin->priv->bin_lock);
1886 g_mutex_init (&rtpbin->priv->dyn_lock);
1888 rtpbin->latency_ms = DEFAULT_LATENCY_MS;
1889 rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
1890 rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
1891 rtpbin->do_lost = DEFAULT_DO_LOST;
1892 rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
1893 rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
1894 rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC;
1895 rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
1896 rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
1897 rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
1898 rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
1900 /* some default SDES entries */
1901 cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
1902 rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
1903 "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
1908 gst_rtp_bin_dispose (GObject * object)
1912 rtpbin = GST_RTP_BIN (object);
1914 GST_RTP_BIN_LOCK (rtpbin);
1915 GST_DEBUG_OBJECT (object, "freeing sessions");
1916 g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
1917 g_slist_free (rtpbin->sessions);
1918 rtpbin->sessions = NULL;
1919 GST_RTP_BIN_UNLOCK (rtpbin);
1921 G_OBJECT_CLASS (parent_class)->dispose (object);
1925 gst_rtp_bin_finalize (GObject * object)
1929 rtpbin = GST_RTP_BIN (object);
1932 gst_structure_free (rtpbin->sdes);
1934 g_mutex_clear (&rtpbin->priv->bin_lock);
1935 g_mutex_clear (&rtpbin->priv->dyn_lock);
1937 G_OBJECT_CLASS (parent_class)->finalize (object);
1942 gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
1949 GST_RTP_BIN_LOCK (bin);
1951 GST_OBJECT_LOCK (bin);
1953 gst_structure_free (bin->sdes);
1954 bin->sdes = gst_structure_copy (sdes);
1955 GST_OBJECT_UNLOCK (bin);
1957 /* store in all sessions */
1958 for (item = bin->sessions; item; item = g_slist_next (item)) {
1959 GstRtpBinSession *session = item->data;
1960 g_object_set (session->session, "sdes", sdes, NULL);
1963 GST_RTP_BIN_UNLOCK (bin);
1966 static GstStructure *
1967 gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
1969 GstStructure *result;
1971 GST_OBJECT_LOCK (bin);
1972 result = gst_structure_copy (bin->sdes);
1973 GST_OBJECT_UNLOCK (bin);
1979 gst_rtp_bin_set_property (GObject * object, guint prop_id,
1980 const GValue * value, GParamSpec * pspec)
1984 rtpbin = GST_RTP_BIN (object);
1988 GST_RTP_BIN_LOCK (rtpbin);
1989 rtpbin->latency_ms = g_value_get_uint (value);
1990 rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
1991 GST_RTP_BIN_UNLOCK (rtpbin);
1992 /* propagate the property down to the jitterbuffer */
1993 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
1995 case PROP_DROP_ON_LATENCY:
1996 GST_RTP_BIN_LOCK (rtpbin);
1997 rtpbin->drop_on_latency = g_value_get_boolean (value);
1998 GST_RTP_BIN_UNLOCK (rtpbin);
1999 /* propagate the property down to the jitterbuffer */
2000 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2001 "drop-on-latency", value);
2004 gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
2007 GST_RTP_BIN_LOCK (rtpbin);
2008 rtpbin->do_lost = g_value_get_boolean (value);
2009 GST_RTP_BIN_UNLOCK (rtpbin);
2010 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
2013 rtpbin->ntp_sync = g_value_get_boolean (value);
2015 case PROP_RTCP_SYNC:
2016 g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
2018 case PROP_RTCP_SYNC_INTERVAL:
2019 rtpbin->rtcp_sync_interval = g_value_get_uint (value);
2021 case PROP_IGNORE_PT:
2022 rtpbin->ignore_pt = g_value_get_boolean (value);
2024 case PROP_AUTOREMOVE:
2025 rtpbin->priv->autoremove = g_value_get_boolean (value);
2027 case PROP_USE_PIPELINE_CLOCK:
2030 GST_RTP_BIN_LOCK (rtpbin);
2031 rtpbin->use_pipeline_clock = g_value_get_boolean (value);
2032 for (sessions = rtpbin->sessions; sessions;
2033 sessions = g_slist_next (sessions)) {
2034 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2036 g_object_set (G_OBJECT (session->session),
2037 "use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
2039 GST_RTP_BIN_UNLOCK (rtpbin);
2042 case PROP_BUFFER_MODE:
2043 GST_RTP_BIN_LOCK (rtpbin);
2044 rtpbin->buffer_mode = g_value_get_enum (value);
2045 GST_RTP_BIN_UNLOCK (rtpbin);
2046 /* propagate the property down to the jitterbuffer */
2047 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
2050 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2056 gst_rtp_bin_get_property (GObject * object, guint prop_id,
2057 GValue * value, GParamSpec * pspec)
2061 rtpbin = GST_RTP_BIN (object);
2065 GST_RTP_BIN_LOCK (rtpbin);
2066 g_value_set_uint (value, rtpbin->latency_ms);
2067 GST_RTP_BIN_UNLOCK (rtpbin);
2069 case PROP_DROP_ON_LATENCY:
2070 GST_RTP_BIN_LOCK (rtpbin);
2071 g_value_set_boolean (value, rtpbin->drop_on_latency);
2072 GST_RTP_BIN_UNLOCK (rtpbin);
2075 g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
2078 GST_RTP_BIN_LOCK (rtpbin);
2079 g_value_set_boolean (value, rtpbin->do_lost);
2080 GST_RTP_BIN_UNLOCK (rtpbin);
2082 case PROP_IGNORE_PT:
2083 g_value_set_boolean (value, rtpbin->ignore_pt);
2086 g_value_set_boolean (value, rtpbin->ntp_sync);
2088 case PROP_RTCP_SYNC:
2089 g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
2091 case PROP_RTCP_SYNC_INTERVAL:
2092 g_value_set_uint (value, rtpbin->rtcp_sync_interval);
2094 case PROP_AUTOREMOVE:
2095 g_value_set_boolean (value, rtpbin->priv->autoremove);
2097 case PROP_BUFFER_MODE:
2098 g_value_set_enum (value, rtpbin->buffer_mode);
2100 case PROP_USE_PIPELINE_CLOCK:
2101 g_value_set_boolean (value, rtpbin->use_pipeline_clock);
2104 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2110 gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
2114 rtpbin = GST_RTP_BIN (bin);
2116 switch (GST_MESSAGE_TYPE (message)) {
2117 case GST_MESSAGE_ELEMENT:
2119 const GstStructure *s = gst_message_get_structure (message);
2121 /* we change the structure name and add the session ID to it */
2122 if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
2123 GstRtpBinSession *sess;
2125 /* find the session we set it as object data */
2126 sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
2127 "GstRTPBin.session");
2129 if (G_LIKELY (sess)) {
2130 message = gst_message_make_writable (message);
2131 s = gst_message_get_structure (message);
2132 gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
2136 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2139 case GST_MESSAGE_BUFFERING:
2142 gint min_percent = 100;
2143 GSList *sessions, *streams;
2144 GstRtpBinStream *stream;
2145 gboolean change = FALSE, active = FALSE;
2146 GstClockTime min_out_time;
2147 GstBufferingMode mode;
2148 gint avg_in, avg_out;
2149 gint64 buffering_left;
2151 gst_message_parse_buffering (message, &percent);
2152 gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
2156 g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
2157 "GstRTPBin.stream");
2159 GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
2161 /* get the stream */
2162 if (G_LIKELY (stream)) {
2163 GST_RTP_BIN_LOCK (rtpbin);
2164 /* fill in the percent */
2165 stream->percent = percent;
2167 /* calculate the min value for all streams */
2168 for (sessions = rtpbin->sessions; sessions;
2169 sessions = g_slist_next (sessions)) {
2170 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2172 GST_RTP_SESSION_LOCK (session);
2173 if (session->streams) {
2174 for (streams = session->streams; streams;
2175 streams = g_slist_next (streams)) {
2176 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
2178 GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
2181 /* find min percent */
2182 if (min_percent > stream->percent)
2183 min_percent = stream->percent;
2186 GST_INFO_OBJECT (bin,
2187 "session has no streams, setting min_percent to 0");
2190 GST_RTP_SESSION_UNLOCK (session);
2192 GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
2194 if (rtpbin->buffering) {
2195 if (min_percent == 100) {
2196 rtpbin->buffering = FALSE;
2201 if (min_percent < 100) {
2202 /* pause the streams */
2203 rtpbin->buffering = TRUE;
2208 GST_RTP_BIN_UNLOCK (rtpbin);
2210 gst_message_unref (message);
2212 /* make a new buffering message with the min value */
2214 gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
2215 gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
2218 if (G_UNLIKELY (change)) {
2220 guint64 running_time = 0;
2223 /* figure out the running time when we have a clock */
2224 if (G_LIKELY ((clock =
2225 gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
2226 guint64 now, base_time;
2228 now = gst_clock_get_time (clock);
2229 base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
2230 running_time = now - base_time;
2231 gst_object_unref (clock);
2233 GST_DEBUG_OBJECT (bin,
2234 "running time now %" GST_TIME_FORMAT,
2235 GST_TIME_ARGS (running_time));
2237 GST_RTP_BIN_LOCK (rtpbin);
2239 /* when we reactivate, calculate the offsets so that all streams have
2240 * an output time that is at least as big as the running_time */
2243 if (running_time > rtpbin->buffer_start) {
2244 offset = running_time - rtpbin->buffer_start;
2245 if (offset >= rtpbin->latency_ns)
2246 offset -= rtpbin->latency_ns;
2252 /* pause all streams */
2254 for (sessions = rtpbin->sessions; sessions;
2255 sessions = g_slist_next (sessions)) {
2256 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2258 GST_RTP_SESSION_LOCK (session);
2259 for (streams = session->streams; streams;
2260 streams = g_slist_next (streams)) {
2261 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
2262 GstElement *element = stream->buffer;
2265 g_signal_emit_by_name (element, "set-active", active, offset,
2269 g_object_get (element, "percent", &stream->percent, NULL);
2273 if (min_out_time == -1 || last_out < min_out_time)
2274 min_out_time = last_out;
2277 GST_DEBUG_OBJECT (bin,
2278 "setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
2279 GST_TIME_FORMAT ", percent %d", element, active,
2280 GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
2283 GST_RTP_SESSION_UNLOCK (session);
2285 GST_DEBUG_OBJECT (bin,
2286 "min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
2288 /* the buffer_start is the min out time of all paused jitterbuffers */
2290 rtpbin->buffer_start = min_out_time;
2292 GST_RTP_BIN_UNLOCK (rtpbin);
2295 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2300 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2306 static GstStateChangeReturn
2307 gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
2309 GstStateChangeReturn res;
2311 GstRtpBinPrivate *priv;
2313 rtpbin = GST_RTP_BIN (element);
2314 priv = rtpbin->priv;
2316 switch (transition) {
2317 case GST_STATE_CHANGE_NULL_TO_READY:
2319 case GST_STATE_CHANGE_READY_TO_PAUSED:
2320 priv->last_unix = 0;
2321 GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
2322 g_atomic_int_set (&priv->shutdown, 0);
2324 case GST_STATE_CHANGE_PAUSED_TO_READY:
2325 GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
2326 g_atomic_int_set (&priv->shutdown, 1);
2327 /* wait for all callbacks to end by taking the lock. No new callbacks will
2328 * be able to happen as we set the shutdown flag. */
2329 GST_RTP_BIN_DYN_LOCK (rtpbin);
2330 GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
2331 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
2337 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2339 switch (transition) {
2340 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2342 case GST_STATE_CHANGE_PAUSED_TO_READY:
2344 case GST_STATE_CHANGE_READY_TO_NULL:
2352 /* a new pad (SSRC) was created in @session. This signal is emited from the
2353 * payload demuxer. */
2355 new_payload_found (GstElement * element, guint pt, GstPad * pad,
2356 GstRtpBinStream * stream)
2359 GstElementClass *klass;
2360 GstPadTemplate *templ;
2364 rtpbin = stream->bin;
2366 GST_DEBUG ("new payload pad %d", pt);
2368 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
2370 /* ghost the pad to the parent */
2371 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2372 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
2373 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
2374 stream->session->id, stream->ssrc, pt);
2375 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
2377 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
2379 gst_pad_set_active (gpad, TRUE);
2380 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2382 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2388 GST_DEBUG ("ignoring, we are shutting down");
2394 payload_pad_removed (GstElement * element, GstPad * pad,
2395 GstRtpBinStream * stream)
2400 rtpbin = stream->bin;
2402 GST_DEBUG ("payload pad removed");
2404 GST_RTP_BIN_DYN_LOCK (rtpbin);
2405 if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
2406 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
2408 gst_pad_set_active (gpad, FALSE);
2409 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2411 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
2415 pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
2420 rtpbin = session->bin;
2422 GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
2425 caps = get_pt_map (session, pt);
2434 GST_DEBUG_OBJECT (rtpbin, "could not get caps");
2440 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
2442 GST_DEBUG_OBJECT (session->bin,
2443 "emiting signal for pt type changed to %d in session %d", pt,
2446 g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
2447 0, session->id, pt);
2450 /* emited when caps changed for the session */
2452 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
2457 const GstStructure *s;
2461 g_object_get (pad, "caps", &caps, NULL);
2466 GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
2468 s = gst_caps_get_structure (caps, 0);
2470 /* get payload, finish when it's not there */
2471 if (!gst_structure_get_int (s, "payload", &payload))
2474 GST_RTP_SESSION_LOCK (session);
2475 GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
2476 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
2477 GST_RTP_SESSION_UNLOCK (session);
2480 /* a new pad (SSRC) was created in @session */
2482 new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
2483 GstRtpBinSession * session)
2486 GstRtpBinStream *stream;
2487 GstPad *sinkpad, *srcpad;
2490 rtpbin = session->bin;
2492 GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
2493 GST_DEBUG_PAD_NAME (pad));
2495 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
2497 GST_RTP_SESSION_LOCK (session);
2499 /* create new stream */
2500 stream = create_stream (session, ssrc);
2504 /* get pad and link */
2505 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
2506 padname = g_strdup_printf ("src_%u", ssrc);
2507 srcpad = gst_element_get_static_pad (element, padname);
2509 sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
2510 gst_pad_link (srcpad, sinkpad);
2511 gst_object_unref (sinkpad);
2512 gst_object_unref (srcpad);
2514 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
2515 padname = g_strdup_printf ("rtcp_src_%u", ssrc);
2516 srcpad = gst_element_get_static_pad (element, padname);
2518 sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
2519 gst_pad_link (srcpad, sinkpad);
2520 gst_object_unref (sinkpad);
2521 gst_object_unref (srcpad);
2523 /* connect to the RTCP sync signal from the jitterbuffer */
2524 GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
2525 stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
2526 "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
2528 if (stream->demux) {
2529 /* connect to the new-pad signal of the payload demuxer, this will expose the
2530 * new pad by ghosting it. */
2531 stream->demux_newpad_sig = g_signal_connect (stream->demux,
2532 "new-payload-type", (GCallback) new_payload_found, stream);
2533 stream->demux_padremoved_sig = g_signal_connect (stream->demux,
2534 "pad-removed", (GCallback) payload_pad_removed, stream);
2536 /* connect to the request-pt-map signal. This signal will be emited by the
2537 * demuxer so that it can apply a proper caps on the buffers for the
2539 stream->demux_ptreq_sig = g_signal_connect (stream->demux,
2540 "request-pt-map", (GCallback) pt_map_requested, session);
2541 /* connect to the signal so it can be forwarded. */
2542 stream->demux_ptchange_sig = g_signal_connect (stream->demux,
2543 "payload-type-change", (GCallback) payload_type_change, session);
2545 /* add gstrtpjitterbuffer src pad to pads */
2546 GstElementClass *klass;
2547 GstPadTemplate *templ;
2551 pad = gst_element_get_static_pad (stream->buffer, "src");
2553 /* ghost the pad to the parent */
2554 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2555 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
2556 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
2557 stream->session->id, stream->ssrc, 255);
2558 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
2561 gst_pad_set_active (gpad, TRUE);
2562 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2564 gst_object_unref (pad);
2567 GST_RTP_SESSION_UNLOCK (session);
2568 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2575 GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
2580 GST_RTP_SESSION_UNLOCK (session);
2581 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2582 GST_DEBUG_OBJECT (rtpbin, "could not create stream");
2587 /* Create a pad for receiving RTP for the session in @name. Must be called with
2591 create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2595 GstRtpBinSession *session;
2596 GstPadLinkReturn lres;
2598 /* first get the session number */
2599 if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
2602 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
2604 /* get or create session */
2605 session = find_session_by_id (rtpbin, sessid);
2607 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
2608 /* create session now */
2609 session = create_session (rtpbin, sessid);
2610 if (session == NULL)
2614 /* check if pad was requested */
2615 if (session->recv_rtp_sink_ghost != NULL)
2616 return session->recv_rtp_sink_ghost;
2618 GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
2619 /* get recv_rtp pad and store */
2620 session->recv_rtp_sink =
2621 gst_element_get_request_pad (session->session, "recv_rtp_sink");
2622 if (session->recv_rtp_sink == NULL)
2625 g_signal_connect (session->recv_rtp_sink, "notify::caps",
2626 (GCallback) caps_changed, session);
2628 GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
2629 /* get srcpad, link to SSRCDemux */
2630 session->recv_rtp_src =
2631 gst_element_get_static_pad (session->session, "recv_rtp_src");
2632 if (session->recv_rtp_src == NULL)
2635 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
2636 sinkdpad = gst_element_get_static_pad (session->demux, "sink");
2637 GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
2638 lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
2639 gst_object_unref (sinkdpad);
2640 if (lres != GST_PAD_LINK_OK)
2643 /* connect to the new-ssrc-pad signal of the SSRC demuxer */
2644 session->demux_newpad_sig = g_signal_connect (session->demux,
2645 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
2646 session->demux_padremoved_sig = g_signal_connect (session->demux,
2647 "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
2649 GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
2650 session->recv_rtp_sink_ghost =
2651 gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
2652 gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
2653 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
2655 return session->recv_rtp_sink_ghost;
2660 g_warning ("rtpbin: invalid name given");
2665 /* create_session already warned */
2670 g_warning ("rtpbin: failed to get session pad");
2675 g_warning ("rtpbin: failed to link pads");
2681 remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2683 if (session->demux_newpad_sig) {
2684 g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
2685 session->demux_newpad_sig = 0;
2687 if (session->demux_padremoved_sig) {
2688 g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
2689 session->demux_padremoved_sig = 0;
2691 if (session->recv_rtp_src) {
2692 gst_object_unref (session->recv_rtp_src);
2693 session->recv_rtp_src = NULL;
2695 if (session->recv_rtp_sink) {
2696 gst_element_release_request_pad (session->session, session->recv_rtp_sink);
2697 gst_object_unref (session->recv_rtp_sink);
2698 session->recv_rtp_sink = NULL;
2700 if (session->recv_rtp_sink_ghost) {
2701 gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
2702 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2703 session->recv_rtp_sink_ghost);
2704 session->recv_rtp_sink_ghost = NULL;
2708 /* Create a pad for receiving RTCP for the session in @name. Must be called with
2712 create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
2716 GstRtpBinSession *session;
2718 GstPadLinkReturn lres;
2720 /* first get the session number */
2721 if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
2724 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
2726 /* get or create the session */
2727 session = find_session_by_id (rtpbin, sessid);
2729 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
2730 /* create session now */
2731 session = create_session (rtpbin, sessid);
2732 if (session == NULL)
2736 /* check if pad was requested */
2737 if (session->recv_rtcp_sink_ghost != NULL)
2738 return session->recv_rtcp_sink_ghost;
2740 /* get recv_rtp pad and store */
2741 GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
2742 session->recv_rtcp_sink =
2743 gst_element_get_request_pad (session->session, "recv_rtcp_sink");
2744 if (session->recv_rtcp_sink == NULL)
2747 /* get srcpad, link to SSRCDemux */
2748 GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
2749 session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
2750 if (session->sync_src == NULL)
2753 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
2754 sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
2755 lres = gst_pad_link (session->sync_src, sinkdpad);
2756 gst_object_unref (sinkdpad);
2757 if (lres != GST_PAD_LINK_OK)
2760 session->recv_rtcp_sink_ghost =
2761 gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
2762 gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
2763 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
2764 session->recv_rtcp_sink_ghost);
2766 return session->recv_rtcp_sink_ghost;
2771 g_warning ("rtpbin: invalid name given");
2776 /* create_session already warned */
2781 g_warning ("rtpbin: failed to get session pad");
2786 g_warning ("rtpbin: failed to link pads");
2792 remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2794 if (session->recv_rtcp_sink_ghost) {
2795 gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
2796 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2797 session->recv_rtcp_sink_ghost);
2798 session->recv_rtcp_sink_ghost = NULL;
2800 if (session->sync_src) {
2801 /* releasing the request pad should also unref the sync pad */
2802 gst_object_unref (session->sync_src);
2803 session->sync_src = NULL;
2805 if (session->recv_rtcp_sink) {
2806 gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
2807 gst_object_unref (session->recv_rtcp_sink);
2808 session->recv_rtcp_sink = NULL;
2812 /* Create a pad for sending RTP for the session in @name. Must be called with
2816 create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2820 GstRtpBinSession *session;
2821 GstElementClass *klass;
2823 /* first get the session number */
2824 if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
2827 /* get or create session */
2828 session = find_session_by_id (rtpbin, sessid);
2830 /* create session now */
2831 session = create_session (rtpbin, sessid);
2832 if (session == NULL)
2836 /* check if pad was requested */
2837 if (session->send_rtp_sink_ghost != NULL)
2838 return session->send_rtp_sink_ghost;
2840 /* get send_rtp pad and store */
2841 session->send_rtp_sink =
2842 gst_element_get_request_pad (session->session, "send_rtp_sink");
2843 if (session->send_rtp_sink == NULL)
2846 session->send_rtp_sink_ghost =
2847 gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
2848 gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
2849 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
2852 session->send_rtp_src =
2853 gst_element_get_static_pad (session->session, "send_rtp_src");
2854 if (session->send_rtp_src == NULL)
2857 /* ghost the new source pad */
2858 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2859 gname = g_strdup_printf ("send_rtp_src_%u", sessid);
2860 templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
2861 session->send_rtp_src_ghost =
2862 gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
2863 gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
2864 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
2867 return session->send_rtp_sink_ghost;
2872 g_warning ("rtpbin: invalid name given");
2877 /* create_session already warned */
2882 g_warning ("rtpbin: failed to get session pad for session %d", sessid);
2887 g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
2893 remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2895 if (session->send_rtp_src_ghost) {
2896 gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
2897 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2898 session->send_rtp_src_ghost);
2899 session->send_rtp_src_ghost = NULL;
2901 if (session->send_rtp_src) {
2902 gst_object_unref (session->send_rtp_src);
2903 session->send_rtp_src = NULL;
2905 if (session->send_rtp_sink) {
2906 gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
2907 session->send_rtp_sink);
2908 gst_object_unref (session->send_rtp_sink);
2909 session->send_rtp_sink = NULL;
2911 if (session->send_rtp_sink_ghost) {
2912 gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
2913 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2914 session->send_rtp_sink_ghost);
2915 session->send_rtp_sink_ghost = NULL;
2919 /* Create a pad for sending RTCP for the session in @name. Must be called with
2923 create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2926 GstRtpBinSession *session;
2928 /* first get the session number */
2929 if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
2932 /* get or create session */
2933 session = find_session_by_id (rtpbin, sessid);
2937 /* check if pad was requested */
2938 if (session->send_rtcp_src_ghost != NULL)
2939 return session->send_rtcp_src_ghost;
2941 /* get rtcp_src pad and store */
2942 session->send_rtcp_src =
2943 gst_element_get_request_pad (session->session, "send_rtcp_src");
2944 if (session->send_rtcp_src == NULL)
2947 session->send_rtcp_src_ghost =
2948 gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
2949 gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
2950 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
2952 return session->send_rtcp_src_ghost;
2957 g_warning ("rtpbin: invalid name given");
2962 g_warning ("rtpbin: session with id %d does not exist", sessid);
2967 g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
2973 remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2975 if (session->send_rtcp_src_ghost) {
2976 gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
2977 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2978 session->send_rtcp_src_ghost);
2979 session->send_rtcp_src_ghost = NULL;
2981 if (session->send_rtcp_src) {
2982 gst_element_release_request_pad (session->session, session->send_rtcp_src);
2983 gst_object_unref (session->send_rtcp_src);
2984 session->send_rtcp_src = NULL;
2988 /* If the requested name is NULL we should create a name with
2989 * the session number assuming we want the lowest posible session
2990 * with a free pad like the template */
2992 gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
2994 gboolean name_found = FALSE;
2996 GstIterator *pad_it = NULL;
2997 gchar *pad_name = NULL;
2998 GValue data = { 0, };
3000 GST_DEBUG_OBJECT (element, "find a free pad name for template");
3001 while (!name_found) {
3002 gboolean done = FALSE;
3005 pad_name = g_strdup_printf (templ->name_template, session++);
3006 pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
3009 switch (gst_iterator_next (pad_it, &data)) {
3010 case GST_ITERATOR_OK:
3015 pad = g_value_get_object (&data);
3016 name = gst_pad_get_name (pad);
3018 if (strcmp (name, pad_name) == 0) {
3023 g_value_reset (&data);
3026 case GST_ITERATOR_ERROR:
3027 case GST_ITERATOR_RESYNC:
3028 /* restart iteration */
3033 case GST_ITERATOR_DONE:
3038 g_value_unset (&data);
3039 gst_iterator_free (pad_it);
3042 GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
3049 gst_rtp_bin_request_new_pad (GstElement * element,
3050 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
3053 GstElementClass *klass;
3056 gchar *pad_name = NULL;
3058 g_return_val_if_fail (templ != NULL, NULL);
3059 g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
3061 rtpbin = GST_RTP_BIN (element);
3062 klass = GST_ELEMENT_GET_CLASS (element);
3064 GST_RTP_BIN_LOCK (rtpbin);
3067 /* use a free pad name */
3068 pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
3070 /* use the provided name */
3071 pad_name = g_strdup (name);
3074 GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
3076 /* figure out the template */
3077 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
3078 result = create_recv_rtp (rtpbin, templ, pad_name);
3079 } else if (templ == gst_element_class_get_pad_template (klass,
3080 "recv_rtcp_sink_%u")) {
3081 result = create_recv_rtcp (rtpbin, templ, pad_name);
3082 } else if (templ == gst_element_class_get_pad_template (klass,
3083 "send_rtp_sink_%u")) {
3084 result = create_send_rtp (rtpbin, templ, pad_name);
3085 } else if (templ == gst_element_class_get_pad_template (klass,
3086 "send_rtcp_src_%u")) {
3087 result = create_rtcp (rtpbin, templ, pad_name);
3089 goto wrong_template;
3092 GST_RTP_BIN_UNLOCK (rtpbin);
3100 GST_RTP_BIN_UNLOCK (rtpbin);
3101 g_warning ("rtpbin: this is not our template");
3107 gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
3109 GstRtpBinSession *session;
3112 g_return_if_fail (GST_IS_GHOST_PAD (pad));
3113 g_return_if_fail (GST_IS_RTP_BIN (element));
3115 rtpbin = GST_RTP_BIN (element);
3117 GST_RTP_BIN_LOCK (rtpbin);
3118 GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
3119 GST_DEBUG_PAD_NAME (pad));
3121 if (!(session = find_session_by_pad (rtpbin, pad)))
3124 if (session->recv_rtp_sink_ghost == pad) {
3125 remove_recv_rtp (rtpbin, session);
3126 } else if (session->recv_rtcp_sink_ghost == pad) {
3127 remove_recv_rtcp (rtpbin, session);
3128 } else if (session->send_rtp_sink_ghost == pad) {
3129 remove_send_rtp (rtpbin, session);
3130 } else if (session->send_rtcp_src_ghost == pad) {
3131 remove_rtcp (rtpbin, session);
3134 /* no more request pads, free the complete session */
3135 if (session->recv_rtp_sink_ghost == NULL
3136 && session->recv_rtcp_sink_ghost == NULL
3137 && session->send_rtp_sink_ghost == NULL
3138 && session->send_rtcp_src_ghost == NULL) {
3139 GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
3140 rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
3141 free_session (session, rtpbin);
3143 GST_RTP_BIN_UNLOCK (rtpbin);
3150 GST_RTP_BIN_UNLOCK (rtpbin);
3151 g_warning ("rtpbin: %s:%s is not one of our request pads",
3152 GST_DEBUG_PAD_NAME (pad));