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., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
21 * SECTION:element-gstrtpbin
22 * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
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 * gstrtpbin 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 gstrtpbin 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 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
70 * gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
71 * ]| Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
73 * gst-launch gstrtpbin name=rtpbin \
74 * v4l2src ! ffmpegcolorspace ! 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 -v gstrtpbin 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);
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;
549 GST_RTP_SESSION_LOCK (session);
550 if ((stream = find_stream_by_ssrc (session, ssrc)))
551 session->streams = g_slist_remove (session->streams, stream);
552 GST_RTP_SESSION_UNLOCK (session);
555 free_stream (stream);
558 /* create a session with the given id. Must be called with RTP_BIN_LOCK */
559 static GstRtpBinSession *
560 create_session (GstRtpBin * rtpbin, gint id)
562 GstRtpBinSession *sess;
563 GstElement *session, *demux;
566 if (!(session = gst_element_factory_make ("rtpsession", NULL)))
569 if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
572 sess = g_new0 (GstRtpBinSession, 1);
573 g_mutex_init (&sess->lock);
576 sess->session = session;
578 sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
579 (GDestroyNotify) gst_caps_unref);
580 rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
582 /* configure SDES items */
583 GST_OBJECT_LOCK (rtpbin);
584 g_object_set (session, "sdes", rtpbin->sdes, "use-pipeline-clock",
585 rtpbin->use_pipeline_clock, NULL);
586 GST_OBJECT_UNLOCK (rtpbin);
588 /* provide clock_rate to the session manager when needed */
589 g_signal_connect (session, "request-pt-map",
590 (GCallback) pt_map_requested, sess);
592 g_signal_connect (sess->session, "on-new-ssrc",
593 (GCallback) on_new_ssrc, sess);
594 g_signal_connect (sess->session, "on-ssrc-collision",
595 (GCallback) on_ssrc_collision, sess);
596 g_signal_connect (sess->session, "on-ssrc-validated",
597 (GCallback) on_ssrc_validated, sess);
598 g_signal_connect (sess->session, "on-ssrc-active",
599 (GCallback) on_ssrc_active, sess);
600 g_signal_connect (sess->session, "on-ssrc-sdes",
601 (GCallback) on_ssrc_sdes, sess);
602 g_signal_connect (sess->session, "on-bye-ssrc",
603 (GCallback) on_bye_ssrc, sess);
604 g_signal_connect (sess->session, "on-bye-timeout",
605 (GCallback) on_bye_timeout, sess);
606 g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
607 g_signal_connect (sess->session, "on-sender-timeout",
608 (GCallback) on_sender_timeout, sess);
610 gst_bin_add (GST_BIN_CAST (rtpbin), session);
611 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
613 GST_OBJECT_LOCK (rtpbin);
614 target = GST_STATE_TARGET (rtpbin);
615 GST_OBJECT_UNLOCK (rtpbin);
617 /* change state only to what's needed */
618 gst_element_set_state (demux, target);
619 gst_element_set_state (session, target);
626 g_warning ("rtpbin: could not create gstrtpsession element");
631 gst_object_unref (session);
632 g_warning ("rtpbin: could not create gstrtpssrcdemux element");
637 /* called with RTP_BIN_LOCK */
639 free_session (GstRtpBinSession * sess, GstRtpBin * bin)
643 GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
645 gst_element_set_locked_state (sess->demux, TRUE);
646 gst_element_set_locked_state (sess->session, TRUE);
648 gst_element_set_state (sess->demux, GST_STATE_NULL);
649 gst_element_set_state (sess->session, GST_STATE_NULL);
651 remove_recv_rtp (bin, sess);
652 remove_recv_rtcp (bin, sess);
653 remove_send_rtp (bin, sess);
654 remove_rtcp (bin, sess);
656 gst_bin_remove (GST_BIN_CAST (bin), sess->session);
657 gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
659 /* remove any references in bin->clients to the streams in sess->streams */
660 client_walk = bin->clients;
661 while (client_walk) {
662 GSList *client_node = client_walk;
663 GstRtpBinClient *client = (GstRtpBinClient *) client_node->data;
664 GSList *stream_walk = client->streams;
666 while (stream_walk) {
667 GSList *stream_node = stream_walk;
668 GstRtpBinStream *stream = (GstRtpBinStream *) stream_node->data;
671 stream_walk = g_slist_next (stream_walk);
673 for (inner_walk = sess->streams; inner_walk;
674 inner_walk = g_slist_next (inner_walk)) {
675 if ((GstRtpBinStream *) inner_walk->data == stream) {
676 client->streams = g_slist_delete_link (client->streams, stream_node);
682 client_walk = g_slist_next (client_walk);
684 g_assert ((client->streams && client->nstreams > 0) || (!client->streams
685 && client->streams == 0));
686 if (client->nstreams == 0) {
687 free_client (client, bin);
688 bin->clients = g_slist_delete_link (bin->clients, client_node);
692 g_slist_foreach (sess->streams, (GFunc) free_stream, NULL);
693 g_slist_free (sess->streams);
695 g_mutex_clear (&sess->lock);
696 g_hash_table_destroy (sess->ptmap);
701 /* get the payload type caps for the specific payload @pt in @session */
703 get_pt_map (GstRtpBinSession * session, guint pt)
705 GstCaps *caps = NULL;
708 GValue args[3] = { {0}, {0}, {0} };
710 GST_DEBUG ("searching pt %d in cache", pt);
712 GST_RTP_SESSION_LOCK (session);
714 /* first look in the cache */
715 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
723 GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
725 /* not in cache, send signal to request caps */
726 g_value_init (&args[0], GST_TYPE_ELEMENT);
727 g_value_set_object (&args[0], bin);
728 g_value_init (&args[1], G_TYPE_UINT);
729 g_value_set_uint (&args[1], session->id);
730 g_value_init (&args[2], G_TYPE_UINT);
731 g_value_set_uint (&args[2], pt);
733 g_value_init (&ret, GST_TYPE_CAPS);
734 g_value_set_boxed (&ret, NULL);
736 GST_RTP_SESSION_UNLOCK (session);
738 g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
740 GST_RTP_SESSION_LOCK (session);
742 g_value_unset (&args[0]);
743 g_value_unset (&args[1]);
744 g_value_unset (&args[2]);
746 /* look in the cache again because we let the lock go */
747 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
750 g_value_unset (&ret);
754 caps = (GstCaps *) g_value_dup_boxed (&ret);
755 g_value_unset (&ret);
759 GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
761 /* store in cache, take additional ref */
762 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
763 gst_caps_ref (caps));
766 GST_RTP_SESSION_UNLOCK (session);
773 GST_RTP_SESSION_UNLOCK (session);
774 GST_DEBUG ("no pt map could be obtained");
780 return_true (gpointer key, gpointer value, gpointer user_data)
786 gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
788 GSList *clients, *streams;
790 GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
792 GST_RTP_BIN_LOCK (rtpbin);
793 for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
794 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
796 /* reset sync on all streams for this client */
797 for (streams = client->streams; streams; streams = g_slist_next (streams)) {
798 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
800 /* make use require a new SR packet for this stream before we attempt new
802 stream->have_sync = FALSE;
803 stream->rt_delta = 0;
804 stream->rtp_delta = 0;
805 stream->clock_base = -100 * GST_SECOND;
808 GST_RTP_BIN_UNLOCK (rtpbin);
812 gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
814 GSList *sessions, *streams;
816 GST_RTP_BIN_LOCK (bin);
817 GST_DEBUG_OBJECT (bin, "clearing pt map");
818 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
819 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
821 GST_DEBUG_OBJECT (bin, "clearing session %p", session);
822 g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
824 GST_RTP_SESSION_LOCK (session);
825 g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
827 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
828 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
830 GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
831 g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
833 g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
835 GST_RTP_SESSION_UNLOCK (session);
837 GST_RTP_BIN_UNLOCK (bin);
840 gst_rtp_bin_reset_sync (bin);
844 gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
846 RTPSession *internal_session = NULL;
847 GstRtpBinSession *session;
849 GST_RTP_BIN_LOCK (bin);
850 GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d",
852 session = find_session_by_id (bin, (gint) session_id);
854 g_object_get (session->session, "internal-session", &internal_session,
857 GST_RTP_BIN_UNLOCK (bin);
859 return internal_session;
863 gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
864 const gchar * name, const GValue * value)
866 GSList *sessions, *streams;
868 GST_RTP_BIN_LOCK (bin);
869 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
870 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
872 GST_RTP_SESSION_LOCK (session);
873 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
874 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
876 g_object_set_property (G_OBJECT (stream->buffer), name, value);
878 GST_RTP_SESSION_UNLOCK (session);
880 GST_RTP_BIN_UNLOCK (bin);
883 /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
884 static GstRtpBinClient *
885 get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
887 GstRtpBinClient *result = NULL;
890 for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
891 GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
893 if (len != client->cname_len)
896 if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
897 GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
904 /* nothing found, create one */
905 if (result == NULL) {
906 result = g_new0 (GstRtpBinClient, 1);
907 result->cname = g_strndup ((gchar *) data, len);
908 result->cname_len = len;
909 bin->clients = g_slist_prepend (bin->clients, result);
910 GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
917 free_client (GstRtpBinClient * client, GstRtpBin * bin)
919 GST_DEBUG_OBJECT (bin, "freeing client %p", client);
920 g_slist_free (client->streams);
921 g_free (client->cname);
926 get_current_times (GstRtpBin * bin, GstClockTime * running_time,
931 GstClockTime base_time, rt, clock_time;
933 GST_OBJECT_LOCK (bin);
934 if ((clock = GST_ELEMENT_CLOCK (bin))) {
935 base_time = GST_ELEMENT_CAST (bin)->base_time;
936 gst_object_ref (clock);
937 GST_OBJECT_UNLOCK (bin);
939 clock_time = gst_clock_get_time (clock);
941 if (bin->use_pipeline_clock) {
946 /* get current NTP time */
947 g_get_current_time (¤t);
948 ntpns = GST_TIMEVAL_TO_TIME (current);
951 /* add constant to convert from 1970 based time to 1900 based time */
952 ntpns += (2208988800LL * GST_SECOND);
954 /* get current clock time and convert to running time */
955 rt = clock_time - base_time;
957 gst_object_unref (clock);
959 GST_OBJECT_UNLOCK (bin);
970 stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
973 gint64 prev_ts_offset;
975 g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
977 /* delta changed, see how much */
978 if (prev_ts_offset != ts_offset) {
981 diff = prev_ts_offset - ts_offset;
983 GST_DEBUG_OBJECT (bin,
984 "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
985 ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
987 /* only change diff when it changed more than 4 milliseconds. This
988 * compensates for rounding errors in NTP to RTP timestamp
990 if (ABS (diff) > 4 * GST_MSECOND) {
991 if (ABS (diff) < (3 * GST_SECOND)) {
992 g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
994 GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
997 GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
1000 GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
1001 stream->ssrc, ts_offset);
1004 /* associate a stream to the given CNAME. This will make sure all streams for
1005 * that CNAME are synchronized together.
1006 * Must be called with GST_RTP_BIN_LOCK */
1008 gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
1009 guint8 * data, guint64 ntptime, guint64 last_extrtptime,
1010 guint64 base_rtptime, guint64 base_time, guint clock_rate,
1011 gint64 rtp_clock_base)
1013 GstRtpBinClient *client;
1018 GstClockTime running_time;
1020 gint64 ntpdiff, rtdiff;
1023 /* first find or create the CNAME */
1024 client = get_client (bin, len, data, &created);
1026 /* find stream in the client */
1027 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1028 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1030 if (ostream == stream)
1033 /* not found, add it to the list */
1035 GST_DEBUG_OBJECT (bin,
1036 "new association of SSRC %08x with client %p with CNAME %s",
1037 stream->ssrc, client, client->cname);
1038 client->streams = g_slist_prepend (client->streams, stream);
1041 GST_DEBUG_OBJECT (bin,
1042 "found association of SSRC %08x with client %p with CNAME %s",
1043 stream->ssrc, client, client->cname);
1046 if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
1047 GST_DEBUG_OBJECT (bin, "invalidated sync data");
1048 if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1049 /* we don't need that data, so carry on,
1050 * but make some values look saner */
1051 last_extrtptime = base_rtptime;
1053 /* nothing we can do with this data in this case */
1054 GST_DEBUG_OBJECT (bin, "bailing out");
1059 /* Take the extended rtptime we found in the SR packet and map it to the
1060 * local rtptime. The local rtp time is used to construct timestamps on the
1061 * buffers so we will calculate what running_time corresponds to the RTP
1062 * timestamp in the SR packet. */
1063 local_rtp = last_extrtptime - base_rtptime;
1065 GST_DEBUG_OBJECT (bin,
1066 "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
1067 ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
1068 "clock-base %" G_GINT64_FORMAT, base_rtptime,
1069 last_extrtptime, local_rtp, clock_rate, rtp_clock_base);
1071 /* calculate local RTP time in gstreamer timestamp, we essentially perform the
1072 * same conversion that a jitterbuffer would use to convert an rtp timestamp
1073 * into a corresponding gstreamer timestamp. Note that the base_time also
1074 * contains the drift between sender and receiver. */
1075 local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
1076 local_rt += base_time;
1078 /* convert ntptime to unix time since 1900 */
1079 last_unix = gst_util_uint64_scale (ntptime, GST_SECOND,
1080 (G_GINT64_CONSTANT (1) << 32));
1082 stream->have_sync = TRUE;
1084 GST_DEBUG_OBJECT (bin,
1085 "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT,
1086 local_rt, last_unix);
1088 /* recalc inter stream playout offset, but only if there is more than one
1089 * stream or we're doing NTP sync. */
1090 if (bin->ntp_sync) {
1091 /* For NTP sync we need to first get a snapshot of running_time and NTP
1092 * time. We know at what running_time we play a certain RTP time, we also
1093 * calculated when we would play the RTP time in the SR packet. Now we need
1094 * to know how the running_time and the NTP time relate to eachother. */
1095 get_current_times (bin, &running_time, &ntpnstime);
1097 /* see how far away the NTP time is. This is the difference between the
1098 * current NTP time and the NTP time in the last SR packet. */
1099 ntpdiff = ntpnstime - last_unix;
1100 /* see how far away the running_time is. This is the difference between the
1101 * current running_time and the running_time of the RTP timestamp in the
1102 * last SR packet. */
1103 rtdiff = running_time - local_rt;
1105 GST_DEBUG_OBJECT (bin,
1106 "NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT,
1107 ntpnstime, last_unix);
1108 GST_DEBUG_OBJECT (bin,
1109 "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
1112 /* combine to get the final diff to apply to the running_time */
1113 stream->rt_delta = rtdiff - ntpdiff;
1115 stream_set_ts_offset (bin, stream, stream->rt_delta);
1117 gint64 min, rtp_min, clock_base = stream->clock_base;
1118 gboolean all_sync, use_rtp;
1119 gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
1121 /* calculate delta between server and receiver. last_unix is created by
1122 * converting the ntptime in the last SR packet to a gstreamer timestamp. This
1123 * delta expresses the difference to our timeline and the server timeline. The
1124 * difference in itself doesn't mean much but we can combine the delta of
1125 * multiple streams to create a stream specific offset. */
1126 stream->rt_delta = last_unix - local_rt;
1128 /* calculate the min of all deltas, ignoring streams that did not yet have a
1129 * valid rt_delta because we did not yet receive an SR packet for those
1131 * We calculate the mininum because we would like to only apply positive
1132 * offsets to streams, delaying their playback instead of trying to speed up
1133 * other streams (which might be imposible when we have to create negative
1135 * The stream that has the smallest diff is selected as the reference stream,
1136 * all other streams will have a positive offset to this difference. */
1138 /* some alternative setting allow ignoring RTCP as much as possible,
1139 * for servers generating bogus ntp timeline */
1140 min = rtp_min = G_MAXINT64;
1142 if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1146 /* signed version for convienience */
1147 clock_base = base_rtptime;
1148 /* deal with possible wrap-around */
1149 ext_base = base_rtptime;
1150 rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
1151 /* sanity check; base rtp and provided clock_base should be close */
1152 if (rtp_clock_base >= clock_base) {
1153 if (rtp_clock_base - clock_base < 10 * clock_rate) {
1154 rtp_clock_base = base_time +
1155 gst_util_uint64_scale_int (rtp_clock_base - clock_base,
1156 GST_SECOND, clock_rate);
1161 if (clock_base - rtp_clock_base < 10 * clock_rate) {
1162 rtp_clock_base = base_time -
1163 gst_util_uint64_scale_int (clock_base - rtp_clock_base,
1164 GST_SECOND, clock_rate);
1169 /* warn and bail for clarity out if no sane values */
1171 GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
1174 /* store to track changes */
1175 clock_base = rtp_clock_base;
1176 /* generate a fake as before,
1177 * now equating rtptime obtained from RTP-Info,
1178 * where the large time represent the otherwise irrelevant npt/ntp time */
1179 stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
1181 clock_base = rtp_clock_base;
1185 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1186 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1188 if (!ostream->have_sync) {
1193 /* change in current stream's base from previously init'ed value
1194 * leads to reset of all stream's base */
1195 if (stream != ostream && stream->clock_base >= 0 &&
1196 (stream->clock_base != clock_base)) {
1197 GST_DEBUG_OBJECT (bin, "reset upon clock base change");
1198 ostream->clock_base = -100 * GST_SECOND;
1199 ostream->rtp_delta = 0;
1202 if (ostream->rt_delta < min)
1203 min = ostream->rt_delta;
1204 if (ostream->rtp_delta < rtp_min)
1205 rtp_min = ostream->rtp_delta;
1208 /* arrange to re-sync for each stream upon significant change,
1210 all_sync = all_sync && (stream->clock_base == clock_base);
1211 stream->clock_base = clock_base;
1213 /* may need init performed above later on, but nothing more to do now */
1214 if (client->nstreams <= 1)
1217 GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
1218 " all sync %d", client, min, all_sync);
1219 GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
1221 switch (rtcp_sync) {
1222 case GST_RTP_BIN_RTCP_SYNC_RTP:
1225 GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
1226 "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
1228 case GST_RTP_BIN_RTCP_SYNC_INITIAL:
1229 /* if all have been synced already, do not bother further */
1231 GST_DEBUG_OBJECT (bin, "all streams already synced; done");
1239 /* bail out if we adjusted recently enough */
1240 if (all_sync && (last_unix - bin->priv->last_unix) <
1241 bin->rtcp_sync_interval * GST_MSECOND) {
1242 GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
1243 "previous sender info too recent "
1244 "(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix);
1247 bin->priv->last_unix = last_unix;
1249 /* calculate offsets for each stream */
1250 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1251 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1254 /* ignore streams for which we didn't receive an SR packet yet, we
1255 * can't synchronize them yet. We can however sync other streams just
1257 if (!ostream->have_sync)
1260 /* calculate offset to our reference stream, this should always give a
1261 * positive number. */
1263 ts_offset = ostream->rtp_delta - rtp_min;
1265 ts_offset = ostream->rt_delta - min;
1267 stream_set_ts_offset (bin, ostream, ts_offset);
1273 #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
1274 for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
1275 (b) = gst_rtcp_packet_move_to_next ((packet)))
1277 #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
1278 for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
1279 (b) = gst_rtcp_packet_sdes_next_item ((packet)))
1281 #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
1282 for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
1283 (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
1286 gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
1287 GstRtpBinStream * stream)
1290 GstRTCPPacket packet;
1293 gboolean have_sr, have_sdes;
1295 guint64 base_rtptime;
1301 GstRTCPBuffer rtcp = { NULL, };
1305 GST_DEBUG_OBJECT (bin, "sync handler called");
1307 /* get the last relation between the rtp timestamps and the gstreamer
1308 * timestamps. We get this info directly from the jitterbuffer which
1309 * constructs gstreamer timestamps from rtp timestamps and so it know exactly
1310 * what the current situation is. */
1312 g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
1313 base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
1314 clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
1315 clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
1317 g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
1318 buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
1323 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
1325 GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
1326 /* first packet must be SR or RR or else the validate would have failed */
1327 switch (gst_rtcp_packet_get_type (&packet)) {
1328 case GST_RTCP_TYPE_SR:
1329 /* only parse first. There is only supposed to be one SR in the packet
1330 * but we will deal with malformed packets gracefully */
1333 /* get NTP and RTP times */
1334 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
1337 GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
1338 /* ignore SR that is not ours */
1339 if (ssrc != stream->ssrc)
1344 case GST_RTCP_TYPE_SDES:
1346 gboolean more_items, more_entries;
1348 /* only deal with first SDES, there is only supposed to be one SDES in
1349 * the RTCP packet but we deal with bad packets gracefully. Also bail
1350 * out if we have not seen an SR item yet. */
1351 if (have_sdes || !have_sr)
1354 GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
1355 /* skip items that are not about the SSRC of the sender */
1356 if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
1359 /* find the CNAME entry */
1360 GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
1361 GstRTCPSDESType type;
1365 gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
1367 if (type == GST_RTCP_SDES_CNAME) {
1368 GST_RTP_BIN_LOCK (bin);
1369 /* associate the stream to CNAME */
1370 gst_rtp_bin_associate (bin, stream, len, data,
1371 ntptime, extrtptime, base_rtptime, base_time, clock_rate,
1373 GST_RTP_BIN_UNLOCK (bin);
1381 /* we can ignore these packets */
1385 gst_rtcp_buffer_unmap (&rtcp);
1388 /* create a new stream with @ssrc in @session. Must be called with
1389 * RTP_SESSION_LOCK. */
1390 static GstRtpBinStream *
1391 create_stream (GstRtpBinSession * session, guint32 ssrc)
1393 GstElement *buffer, *demux = NULL;
1394 GstRtpBinStream *stream;
1398 rtpbin = session->bin;
1400 if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
1401 goto no_jitterbuffer;
1403 if (!rtpbin->ignore_pt)
1404 if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
1408 stream = g_new0 (GstRtpBinStream, 1);
1409 stream->ssrc = ssrc;
1410 stream->bin = rtpbin;
1411 stream->session = session;
1412 stream->buffer = buffer;
1413 stream->demux = demux;
1415 stream->have_sync = FALSE;
1416 stream->rt_delta = 0;
1417 stream->rtp_delta = 0;
1418 stream->percent = 100;
1419 stream->clock_base = -100 * GST_SECOND;
1420 session->streams = g_slist_prepend (session->streams, stream);
1422 /* provide clock_rate to the jitterbuffer when needed */
1423 stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
1424 (GCallback) pt_map_requested, session);
1425 stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
1426 (GCallback) on_npt_stop, stream);
1428 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
1429 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
1431 /* configure latency and packet lost */
1432 g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
1433 g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
1434 g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
1435 g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
1437 if (!rtpbin->ignore_pt)
1438 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
1439 gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
1443 gst_element_link (buffer, demux);
1445 if (rtpbin->buffering) {
1448 GST_INFO_OBJECT (rtpbin,
1449 "bin is buffering, set jitterbuffer as not active");
1450 g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
1454 GST_OBJECT_LOCK (rtpbin);
1455 target = GST_STATE_TARGET (rtpbin);
1456 GST_OBJECT_UNLOCK (rtpbin);
1458 /* from sink to source */
1460 gst_element_set_state (demux, target);
1462 gst_element_set_state (buffer, target);
1469 g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
1474 gst_object_unref (buffer);
1475 g_warning ("rtpbin: could not create gstrtpptdemux element");
1481 free_stream (GstRtpBinStream * stream)
1483 GstRtpBinSession *session;
1485 session = stream->session;
1487 if (stream->demux) {
1488 g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
1489 g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
1490 g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
1492 g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
1493 g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
1494 g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
1497 gst_element_set_locked_state (stream->demux, TRUE);
1498 gst_element_set_locked_state (stream->buffer, TRUE);
1501 gst_element_set_state (stream->demux, GST_STATE_NULL);
1502 gst_element_set_state (stream->buffer, GST_STATE_NULL);
1504 /* now remove this signal, we need this while going to NULL because it to
1505 * do some cleanups */
1507 g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
1509 gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer);
1511 gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux);
1516 /* GObject vmethods */
1517 static void gst_rtp_bin_dispose (GObject * object);
1518 static void gst_rtp_bin_finalize (GObject * object);
1519 static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
1520 const GValue * value, GParamSpec * pspec);
1521 static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
1522 GValue * value, GParamSpec * pspec);
1524 /* GstElement vmethods */
1525 static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
1526 GstStateChange transition);
1527 static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
1528 GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
1529 static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
1530 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
1532 #define gst_rtp_bin_parent_class parent_class
1533 G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
1536 gst_rtp_bin_class_init (GstRtpBinClass * klass)
1538 GObjectClass *gobject_class;
1539 GstElementClass *gstelement_class;
1540 GstBinClass *gstbin_class;
1542 gobject_class = (GObjectClass *) klass;
1543 gstelement_class = (GstElementClass *) klass;
1544 gstbin_class = (GstBinClass *) klass;
1546 g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
1548 gobject_class->dispose = gst_rtp_bin_dispose;
1549 gobject_class->finalize = gst_rtp_bin_finalize;
1550 gobject_class->set_property = gst_rtp_bin_set_property;
1551 gobject_class->get_property = gst_rtp_bin_get_property;
1553 g_object_class_install_property (gobject_class, PROP_LATENCY,
1554 g_param_spec_uint ("latency", "Buffer latency in ms",
1555 "Default amount of ms to buffer in the jitterbuffers", 0,
1556 G_MAXUINT, DEFAULT_LATENCY_MS,
1557 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1559 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
1560 g_param_spec_boolean ("drop-on-latency",
1561 "Drop buffers when maximum latency is reached",
1562 "Tells the jitterbuffer to never exceed the given latency in size",
1563 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1566 * GstRtpBin::request-pt-map:
1567 * @rtpbin: the object which received the signal
1568 * @session: the session
1571 * Request the payload type as #GstCaps for @pt in @session.
1573 gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
1574 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
1575 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
1576 NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
1577 G_TYPE_UINT, G_TYPE_UINT);
1580 * GstRtpBin::payload-type-change:
1581 * @rtpbin: the object which received the signal
1582 * @session: the session
1585 * Signal that the current payload type changed to @pt in @session.
1589 gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
1590 g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
1591 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
1592 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1593 G_TYPE_UINT, G_TYPE_UINT);
1596 * GstRtpBin::clear-pt-map:
1597 * @rtpbin: the object which received the signal
1599 * Clear all previously cached pt-mapping obtained with
1600 * #GstRtpBin::request-pt-map.
1602 gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
1603 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
1604 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1605 clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1609 * GstRtpBin::reset-sync:
1610 * @rtpbin: the object which received the signal
1612 * Reset all currently configured lip-sync parameters and require new SR
1613 * packets for all streams before lip-sync is attempted again.
1615 gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
1616 g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
1617 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1618 reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1622 * GstRtpBin::get-internal-session:
1623 * @rtpbin: the object which received the signal
1624 * @id: the session id
1626 * Request the internal RTPSession object as #GObject in session @id.
1628 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
1629 g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
1630 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1631 get_internal_session), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
1632 RTP_TYPE_SESSION, 1, G_TYPE_UINT);
1635 * GstRtpBin::on-new-ssrc:
1636 * @rtpbin: the object which received the signal
1637 * @session: the session
1640 * Notify of a new SSRC that entered @session.
1642 gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
1643 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
1644 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
1645 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1646 G_TYPE_UINT, G_TYPE_UINT);
1648 * GstRtpBin::on-ssrc-collision:
1649 * @rtpbin: the object which received the signal
1650 * @session: the session
1653 * Notify when we have an SSRC collision
1655 gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
1656 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
1657 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
1658 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1659 G_TYPE_UINT, G_TYPE_UINT);
1661 * GstRtpBin::on-ssrc-validated:
1662 * @rtpbin: the object which received the signal
1663 * @session: the session
1666 * Notify of a new SSRC that became validated.
1668 gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
1669 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
1670 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
1671 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1672 G_TYPE_UINT, G_TYPE_UINT);
1674 * GstRtpBin::on-ssrc-active:
1675 * @rtpbin: the object which received the signal
1676 * @session: the session
1679 * Notify of a SSRC that is active, i.e., sending RTCP.
1681 gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
1682 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
1683 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
1684 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1685 G_TYPE_UINT, G_TYPE_UINT);
1687 * GstRtpBin::on-ssrc-sdes:
1688 * @rtpbin: the object which received the signal
1689 * @session: the session
1692 * Notify of a SSRC that is active, i.e., sending RTCP.
1694 gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
1695 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
1696 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
1697 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1698 G_TYPE_UINT, G_TYPE_UINT);
1701 * GstRtpBin::on-bye-ssrc:
1702 * @rtpbin: the object which received the signal
1703 * @session: the session
1706 * Notify of an SSRC that became inactive because of a BYE packet.
1708 gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
1709 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
1710 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
1711 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1712 G_TYPE_UINT, G_TYPE_UINT);
1714 * GstRtpBin::on-bye-timeout:
1715 * @rtpbin: the object which received the signal
1716 * @session: the session
1719 * Notify of an SSRC that has timed out because of BYE
1721 gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
1722 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
1723 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
1724 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1725 G_TYPE_UINT, G_TYPE_UINT);
1727 * GstRtpBin::on-timeout:
1728 * @rtpbin: the object which received the signal
1729 * @session: the session
1732 * Notify of an SSRC that has timed out
1734 gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
1735 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
1736 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
1737 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1738 G_TYPE_UINT, G_TYPE_UINT);
1740 * GstRtpBin::on-sender-timeout:
1741 * @rtpbin: the object which received the signal
1742 * @session: the session
1745 * Notify of a sender SSRC that has timed out and became a receiver
1747 gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
1748 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
1749 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
1750 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1751 G_TYPE_UINT, G_TYPE_UINT);
1754 * GstRtpBin::on-npt-stop:
1755 * @rtpbin: the object which received the signal
1756 * @session: the session
1759 * Notify that SSRC sender has sent data up to the configured NPT stop time.
1761 gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
1762 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
1763 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
1764 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1765 G_TYPE_UINT, G_TYPE_UINT);
1767 g_object_class_install_property (gobject_class, PROP_SDES,
1768 g_param_spec_boxed ("sdes", "SDES",
1769 "The SDES items of this session",
1770 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1772 g_object_class_install_property (gobject_class, PROP_DO_LOST,
1773 g_param_spec_boolean ("do-lost", "Do Lost",
1774 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
1775 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1777 g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
1778 g_param_spec_boolean ("autoremove", "Auto Remove",
1779 "Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
1780 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1782 g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
1783 g_param_spec_boolean ("ignore-pt", "Ignore PT",
1784 "Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
1785 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1787 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
1788 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
1789 "Use the pipeline clock to set the NTP time in the RTCP SR messages",
1790 DEFAULT_USE_PIPELINE_CLOCK,
1791 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1793 * GstRtpBin::buffer-mode:
1795 * Control the buffering and timestamping mode used by the jitterbuffer.
1799 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
1800 g_param_spec_enum ("buffer-mode", "Buffer Mode",
1801 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
1802 DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1804 * GstRtpBin::ntp-sync:
1806 * Synchronize received streams to the NTP clock. When the NTP clock is shared
1807 * between the receivers and the senders (such as when using ntpd) this option
1808 * can be 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_DEBUG_OBJECT (object, "freeing clients");
1920 g_slist_foreach (rtpbin->clients, (GFunc) free_client, rtpbin);
1921 g_slist_free (rtpbin->clients);
1922 rtpbin->clients = NULL;
1923 GST_RTP_BIN_UNLOCK (rtpbin);
1925 G_OBJECT_CLASS (parent_class)->dispose (object);
1929 gst_rtp_bin_finalize (GObject * object)
1933 rtpbin = GST_RTP_BIN (object);
1936 gst_structure_free (rtpbin->sdes);
1938 g_mutex_clear (&rtpbin->priv->bin_lock);
1939 g_mutex_clear (&rtpbin->priv->dyn_lock);
1941 G_OBJECT_CLASS (parent_class)->finalize (object);
1946 gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
1953 GST_RTP_BIN_LOCK (bin);
1955 GST_OBJECT_LOCK (bin);
1957 gst_structure_free (bin->sdes);
1958 bin->sdes = gst_structure_copy (sdes);
1959 GST_OBJECT_UNLOCK (bin);
1961 /* store in all sessions */
1962 for (item = bin->sessions; item; item = g_slist_next (item)) {
1963 GstRtpBinSession *session = item->data;
1964 g_object_set (session->session, "sdes", sdes, NULL);
1967 GST_RTP_BIN_UNLOCK (bin);
1970 static GstStructure *
1971 gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
1973 GstStructure *result;
1975 GST_OBJECT_LOCK (bin);
1976 result = gst_structure_copy (bin->sdes);
1977 GST_OBJECT_UNLOCK (bin);
1983 gst_rtp_bin_set_property (GObject * object, guint prop_id,
1984 const GValue * value, GParamSpec * pspec)
1988 rtpbin = GST_RTP_BIN (object);
1992 GST_RTP_BIN_LOCK (rtpbin);
1993 rtpbin->latency_ms = g_value_get_uint (value);
1994 rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
1995 GST_RTP_BIN_UNLOCK (rtpbin);
1996 /* propagate the property down to the jitterbuffer */
1997 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
1999 case PROP_DROP_ON_LATENCY:
2000 GST_RTP_BIN_LOCK (rtpbin);
2001 rtpbin->drop_on_latency = g_value_get_boolean (value);
2002 GST_RTP_BIN_UNLOCK (rtpbin);
2003 /* propagate the property down to the jitterbuffer */
2004 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2005 "drop-on-latency", value);
2008 gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
2011 GST_RTP_BIN_LOCK (rtpbin);
2012 rtpbin->do_lost = g_value_get_boolean (value);
2013 GST_RTP_BIN_UNLOCK (rtpbin);
2014 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
2017 rtpbin->ntp_sync = g_value_get_boolean (value);
2019 case PROP_RTCP_SYNC:
2020 g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
2022 case PROP_RTCP_SYNC_INTERVAL:
2023 rtpbin->rtcp_sync_interval = g_value_get_uint (value);
2025 case PROP_IGNORE_PT:
2026 rtpbin->ignore_pt = g_value_get_boolean (value);
2028 case PROP_AUTOREMOVE:
2029 rtpbin->priv->autoremove = g_value_get_boolean (value);
2031 case PROP_USE_PIPELINE_CLOCK:
2034 GST_RTP_BIN_LOCK (rtpbin);
2035 rtpbin->use_pipeline_clock = g_value_get_boolean (value);
2036 for (sessions = rtpbin->sessions; sessions;
2037 sessions = g_slist_next (sessions)) {
2038 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2040 g_object_set (G_OBJECT (session->session),
2041 "use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
2043 GST_RTP_BIN_UNLOCK (rtpbin);
2046 case PROP_BUFFER_MODE:
2047 GST_RTP_BIN_LOCK (rtpbin);
2048 rtpbin->buffer_mode = g_value_get_enum (value);
2049 GST_RTP_BIN_UNLOCK (rtpbin);
2050 /* propagate the property down to the jitterbuffer */
2051 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
2054 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2060 gst_rtp_bin_get_property (GObject * object, guint prop_id,
2061 GValue * value, GParamSpec * pspec)
2065 rtpbin = GST_RTP_BIN (object);
2069 GST_RTP_BIN_LOCK (rtpbin);
2070 g_value_set_uint (value, rtpbin->latency_ms);
2071 GST_RTP_BIN_UNLOCK (rtpbin);
2073 case PROP_DROP_ON_LATENCY:
2074 GST_RTP_BIN_LOCK (rtpbin);
2075 g_value_set_boolean (value, rtpbin->drop_on_latency);
2076 GST_RTP_BIN_UNLOCK (rtpbin);
2079 g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
2082 GST_RTP_BIN_LOCK (rtpbin);
2083 g_value_set_boolean (value, rtpbin->do_lost);
2084 GST_RTP_BIN_UNLOCK (rtpbin);
2086 case PROP_IGNORE_PT:
2087 g_value_set_boolean (value, rtpbin->ignore_pt);
2090 g_value_set_boolean (value, rtpbin->ntp_sync);
2092 case PROP_RTCP_SYNC:
2093 g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
2095 case PROP_RTCP_SYNC_INTERVAL:
2096 g_value_set_uint (value, rtpbin->rtcp_sync_interval);
2098 case PROP_AUTOREMOVE:
2099 g_value_set_boolean (value, rtpbin->priv->autoremove);
2101 case PROP_BUFFER_MODE:
2102 g_value_set_enum (value, rtpbin->buffer_mode);
2104 case PROP_USE_PIPELINE_CLOCK:
2105 g_value_set_boolean (value, rtpbin->use_pipeline_clock);
2108 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
2114 gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
2118 rtpbin = GST_RTP_BIN (bin);
2120 switch (GST_MESSAGE_TYPE (message)) {
2121 case GST_MESSAGE_ELEMENT:
2123 const GstStructure *s = gst_message_get_structure (message);
2125 /* we change the structure name and add the session ID to it */
2126 if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
2127 GstRtpBinSession *sess;
2129 /* find the session we set it as object data */
2130 sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
2131 "GstRTPBin.session");
2133 if (G_LIKELY (sess)) {
2134 message = gst_message_make_writable (message);
2135 s = gst_message_get_structure (message);
2136 gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
2140 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2143 case GST_MESSAGE_BUFFERING:
2146 gint min_percent = 100;
2147 GSList *sessions, *streams;
2148 GstRtpBinStream *stream;
2149 gboolean change = FALSE, active = FALSE;
2150 GstClockTime min_out_time;
2151 GstBufferingMode mode;
2152 gint avg_in, avg_out;
2153 gint64 buffering_left;
2155 gst_message_parse_buffering (message, &percent);
2156 gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
2160 g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
2161 "GstRTPBin.stream");
2163 GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
2165 /* get the stream */
2166 if (G_LIKELY (stream)) {
2167 GST_RTP_BIN_LOCK (rtpbin);
2168 /* fill in the percent */
2169 stream->percent = percent;
2171 /* calculate the min value for all streams */
2172 for (sessions = rtpbin->sessions; sessions;
2173 sessions = g_slist_next (sessions)) {
2174 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2176 GST_RTP_SESSION_LOCK (session);
2177 if (session->streams) {
2178 for (streams = session->streams; streams;
2179 streams = g_slist_next (streams)) {
2180 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
2182 GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
2185 /* find min percent */
2186 if (min_percent > stream->percent)
2187 min_percent = stream->percent;
2190 GST_INFO_OBJECT (bin,
2191 "session has no streams, setting min_percent to 0");
2194 GST_RTP_SESSION_UNLOCK (session);
2196 GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
2198 if (rtpbin->buffering) {
2199 if (min_percent == 100) {
2200 rtpbin->buffering = FALSE;
2205 if (min_percent < 100) {
2206 /* pause the streams */
2207 rtpbin->buffering = TRUE;
2212 GST_RTP_BIN_UNLOCK (rtpbin);
2214 gst_message_unref (message);
2216 /* make a new buffering message with the min value */
2218 gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
2219 gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
2222 if (G_UNLIKELY (change)) {
2224 guint64 running_time = 0;
2227 /* figure out the running time when we have a clock */
2228 if (G_LIKELY ((clock =
2229 gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
2230 guint64 now, base_time;
2232 now = gst_clock_get_time (clock);
2233 base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
2234 running_time = now - base_time;
2235 gst_object_unref (clock);
2237 GST_DEBUG_OBJECT (bin,
2238 "running time now %" GST_TIME_FORMAT,
2239 GST_TIME_ARGS (running_time));
2241 GST_RTP_BIN_LOCK (rtpbin);
2243 /* when we reactivate, calculate the offsets so that all streams have
2244 * an output time that is at least as big as the running_time */
2247 if (running_time > rtpbin->buffer_start) {
2248 offset = running_time - rtpbin->buffer_start;
2249 if (offset >= rtpbin->latency_ns)
2250 offset -= rtpbin->latency_ns;
2256 /* pause all streams */
2258 for (sessions = rtpbin->sessions; sessions;
2259 sessions = g_slist_next (sessions)) {
2260 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2262 GST_RTP_SESSION_LOCK (session);
2263 for (streams = session->streams; streams;
2264 streams = g_slist_next (streams)) {
2265 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
2266 GstElement *element = stream->buffer;
2269 g_signal_emit_by_name (element, "set-active", active, offset,
2273 g_object_get (element, "percent", &stream->percent, NULL);
2277 if (min_out_time == -1 || last_out < min_out_time)
2278 min_out_time = last_out;
2281 GST_DEBUG_OBJECT (bin,
2282 "setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
2283 GST_TIME_FORMAT ", percent %d", element, active,
2284 GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
2287 GST_RTP_SESSION_UNLOCK (session);
2289 GST_DEBUG_OBJECT (bin,
2290 "min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
2292 /* the buffer_start is the min out time of all paused jitterbuffers */
2294 rtpbin->buffer_start = min_out_time;
2296 GST_RTP_BIN_UNLOCK (rtpbin);
2299 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2304 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
2310 static GstStateChangeReturn
2311 gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
2313 GstStateChangeReturn res;
2315 GstRtpBinPrivate *priv;
2317 rtpbin = GST_RTP_BIN (element);
2318 priv = rtpbin->priv;
2320 switch (transition) {
2321 case GST_STATE_CHANGE_NULL_TO_READY:
2323 case GST_STATE_CHANGE_READY_TO_PAUSED:
2324 priv->last_unix = 0;
2325 GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
2326 g_atomic_int_set (&priv->shutdown, 0);
2328 case GST_STATE_CHANGE_PAUSED_TO_READY:
2329 GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
2330 g_atomic_int_set (&priv->shutdown, 1);
2331 /* wait for all callbacks to end by taking the lock. No new callbacks will
2332 * be able to happen as we set the shutdown flag. */
2333 GST_RTP_BIN_DYN_LOCK (rtpbin);
2334 GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
2335 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
2341 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
2343 switch (transition) {
2344 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
2346 case GST_STATE_CHANGE_PAUSED_TO_READY:
2348 case GST_STATE_CHANGE_READY_TO_NULL:
2356 /* a new pad (SSRC) was created in @session. This signal is emited from the
2357 * payload demuxer. */
2359 new_payload_found (GstElement * element, guint pt, GstPad * pad,
2360 GstRtpBinStream * stream)
2363 GstElementClass *klass;
2364 GstPadTemplate *templ;
2368 rtpbin = stream->bin;
2370 GST_DEBUG ("new payload pad %d", pt);
2372 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
2374 /* ghost the pad to the parent */
2375 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2376 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
2377 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
2378 stream->session->id, stream->ssrc, pt);
2379 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
2381 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
2383 gst_pad_set_active (gpad, TRUE);
2384 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2386 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2392 GST_DEBUG ("ignoring, we are shutting down");
2398 payload_pad_removed (GstElement * element, GstPad * pad,
2399 GstRtpBinStream * stream)
2404 rtpbin = stream->bin;
2406 GST_DEBUG ("payload pad removed");
2408 GST_RTP_BIN_DYN_LOCK (rtpbin);
2409 if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
2410 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
2412 gst_pad_set_active (gpad, FALSE);
2413 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2415 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
2419 pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
2424 rtpbin = session->bin;
2426 GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
2429 caps = get_pt_map (session, pt);
2438 GST_DEBUG_OBJECT (rtpbin, "could not get caps");
2444 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
2446 GST_DEBUG_OBJECT (session->bin,
2447 "emiting signal for pt type changed to %d in session %d", pt,
2450 g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
2451 0, session->id, pt);
2454 /* emited when caps changed for the session */
2456 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
2461 const GstStructure *s;
2465 g_object_get (pad, "caps", &caps, NULL);
2470 GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
2472 s = gst_caps_get_structure (caps, 0);
2474 /* get payload, finish when it's not there */
2475 if (!gst_structure_get_int (s, "payload", &payload))
2478 GST_RTP_SESSION_LOCK (session);
2479 GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
2480 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
2481 GST_RTP_SESSION_UNLOCK (session);
2484 /* a new pad (SSRC) was created in @session */
2486 new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
2487 GstRtpBinSession * session)
2490 GstRtpBinStream *stream;
2491 GstPad *sinkpad, *srcpad;
2494 rtpbin = session->bin;
2496 GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
2497 GST_DEBUG_PAD_NAME (pad));
2499 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
2501 GST_RTP_SESSION_LOCK (session);
2503 /* create new stream */
2504 stream = create_stream (session, ssrc);
2508 /* get pad and link */
2509 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
2510 padname = g_strdup_printf ("src_%u", ssrc);
2511 srcpad = gst_element_get_static_pad (element, padname);
2513 sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
2514 gst_pad_link (srcpad, sinkpad);
2515 gst_object_unref (sinkpad);
2516 gst_object_unref (srcpad);
2518 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
2519 padname = g_strdup_printf ("rtcp_src_%u", ssrc);
2520 srcpad = gst_element_get_static_pad (element, padname);
2522 sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
2523 gst_pad_link (srcpad, sinkpad);
2524 gst_object_unref (sinkpad);
2525 gst_object_unref (srcpad);
2527 /* connect to the RTCP sync signal from the jitterbuffer */
2528 GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
2529 stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
2530 "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
2532 if (stream->demux) {
2533 /* connect to the new-pad signal of the payload demuxer, this will expose the
2534 * new pad by ghosting it. */
2535 stream->demux_newpad_sig = g_signal_connect (stream->demux,
2536 "new-payload-type", (GCallback) new_payload_found, stream);
2537 stream->demux_padremoved_sig = g_signal_connect (stream->demux,
2538 "pad-removed", (GCallback) payload_pad_removed, stream);
2540 /* connect to the request-pt-map signal. This signal will be emited by the
2541 * demuxer so that it can apply a proper caps on the buffers for the
2543 stream->demux_ptreq_sig = g_signal_connect (stream->demux,
2544 "request-pt-map", (GCallback) pt_map_requested, session);
2545 /* connect to the signal so it can be forwarded. */
2546 stream->demux_ptchange_sig = g_signal_connect (stream->demux,
2547 "payload-type-change", (GCallback) payload_type_change, session);
2549 /* add gstrtpjitterbuffer src pad to pads */
2550 GstElementClass *klass;
2551 GstPadTemplate *templ;
2555 pad = gst_element_get_static_pad (stream->buffer, "src");
2557 /* ghost the pad to the parent */
2558 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2559 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
2560 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
2561 stream->session->id, stream->ssrc, 255);
2562 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
2565 gst_pad_set_active (gpad, TRUE);
2566 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
2568 gst_object_unref (pad);
2571 GST_RTP_SESSION_UNLOCK (session);
2572 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2579 GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
2584 GST_RTP_SESSION_UNLOCK (session);
2585 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
2586 GST_DEBUG_OBJECT (rtpbin, "could not create stream");
2591 /* Create a pad for receiving RTP for the session in @name. Must be called with
2595 create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2599 GstRtpBinSession *session;
2600 GstPadLinkReturn lres;
2602 /* first get the session number */
2603 if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
2606 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
2608 /* get or create session */
2609 session = find_session_by_id (rtpbin, sessid);
2611 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
2612 /* create session now */
2613 session = create_session (rtpbin, sessid);
2614 if (session == NULL)
2618 /* check if pad was requested */
2619 if (session->recv_rtp_sink_ghost != NULL)
2620 return session->recv_rtp_sink_ghost;
2622 GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
2623 /* get recv_rtp pad and store */
2624 session->recv_rtp_sink =
2625 gst_element_get_request_pad (session->session, "recv_rtp_sink");
2626 if (session->recv_rtp_sink == NULL)
2629 g_signal_connect (session->recv_rtp_sink, "notify::caps",
2630 (GCallback) caps_changed, session);
2632 GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
2633 /* get srcpad, link to SSRCDemux */
2634 session->recv_rtp_src =
2635 gst_element_get_static_pad (session->session, "recv_rtp_src");
2636 if (session->recv_rtp_src == NULL)
2639 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
2640 sinkdpad = gst_element_get_static_pad (session->demux, "sink");
2641 GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
2642 lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
2643 gst_object_unref (sinkdpad);
2644 if (lres != GST_PAD_LINK_OK)
2647 /* connect to the new-ssrc-pad signal of the SSRC demuxer */
2648 session->demux_newpad_sig = g_signal_connect (session->demux,
2649 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
2650 session->demux_padremoved_sig = g_signal_connect (session->demux,
2651 "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
2653 GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
2654 session->recv_rtp_sink_ghost =
2655 gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
2656 gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
2657 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
2659 return session->recv_rtp_sink_ghost;
2664 g_warning ("rtpbin: invalid name given");
2669 /* create_session already warned */
2674 g_warning ("rtpbin: failed to get session pad");
2679 g_warning ("rtpbin: failed to link pads");
2685 remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2687 if (session->demux_newpad_sig) {
2688 g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
2689 session->demux_newpad_sig = 0;
2691 if (session->demux_padremoved_sig) {
2692 g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
2693 session->demux_padremoved_sig = 0;
2695 if (session->recv_rtp_src) {
2696 gst_object_unref (session->recv_rtp_src);
2697 session->recv_rtp_src = NULL;
2699 if (session->recv_rtp_sink) {
2700 gst_element_release_request_pad (session->session, session->recv_rtp_sink);
2701 gst_object_unref (session->recv_rtp_sink);
2702 session->recv_rtp_sink = NULL;
2704 if (session->recv_rtp_sink_ghost) {
2705 gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
2706 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2707 session->recv_rtp_sink_ghost);
2708 session->recv_rtp_sink_ghost = NULL;
2712 /* Create a pad for receiving RTCP for the session in @name. Must be called with
2716 create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
2720 GstRtpBinSession *session;
2722 GstPadLinkReturn lres;
2724 /* first get the session number */
2725 if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
2728 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
2730 /* get or create the session */
2731 session = find_session_by_id (rtpbin, sessid);
2733 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
2734 /* create session now */
2735 session = create_session (rtpbin, sessid);
2736 if (session == NULL)
2740 /* check if pad was requested */
2741 if (session->recv_rtcp_sink_ghost != NULL)
2742 return session->recv_rtcp_sink_ghost;
2744 /* get recv_rtp pad and store */
2745 GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
2746 session->recv_rtcp_sink =
2747 gst_element_get_request_pad (session->session, "recv_rtcp_sink");
2748 if (session->recv_rtcp_sink == NULL)
2751 /* get srcpad, link to SSRCDemux */
2752 GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
2753 session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
2754 if (session->sync_src == NULL)
2757 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
2758 sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
2759 lres = gst_pad_link (session->sync_src, sinkdpad);
2760 gst_object_unref (sinkdpad);
2761 if (lres != GST_PAD_LINK_OK)
2764 session->recv_rtcp_sink_ghost =
2765 gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
2766 gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
2767 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
2768 session->recv_rtcp_sink_ghost);
2770 return session->recv_rtcp_sink_ghost;
2775 g_warning ("rtpbin: invalid name given");
2780 /* create_session already warned */
2785 g_warning ("rtpbin: failed to get session pad");
2790 g_warning ("rtpbin: failed to link pads");
2796 remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2798 if (session->recv_rtcp_sink_ghost) {
2799 gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
2800 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2801 session->recv_rtcp_sink_ghost);
2802 session->recv_rtcp_sink_ghost = NULL;
2804 if (session->sync_src) {
2805 /* releasing the request pad should also unref the sync pad */
2806 gst_object_unref (session->sync_src);
2807 session->sync_src = NULL;
2809 if (session->recv_rtcp_sink) {
2810 gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
2811 gst_object_unref (session->recv_rtcp_sink);
2812 session->recv_rtcp_sink = NULL;
2816 /* Create a pad for sending RTP for the session in @name. Must be called with
2820 create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2824 GstRtpBinSession *session;
2825 GstElementClass *klass;
2827 /* first get the session number */
2828 if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
2831 /* get or create session */
2832 session = find_session_by_id (rtpbin, sessid);
2834 /* create session now */
2835 session = create_session (rtpbin, sessid);
2836 if (session == NULL)
2840 /* check if pad was requested */
2841 if (session->send_rtp_sink_ghost != NULL)
2842 return session->send_rtp_sink_ghost;
2844 /* get send_rtp pad and store */
2845 session->send_rtp_sink =
2846 gst_element_get_request_pad (session->session, "send_rtp_sink");
2847 if (session->send_rtp_sink == NULL)
2850 session->send_rtp_sink_ghost =
2851 gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
2852 gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
2853 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
2856 session->send_rtp_src =
2857 gst_element_get_static_pad (session->session, "send_rtp_src");
2858 if (session->send_rtp_src == NULL)
2861 /* ghost the new source pad */
2862 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2863 gname = g_strdup_printf ("send_rtp_src_%u", sessid);
2864 templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
2865 session->send_rtp_src_ghost =
2866 gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
2867 gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
2868 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
2871 return session->send_rtp_sink_ghost;
2876 g_warning ("rtpbin: invalid name given");
2881 /* create_session already warned */
2886 g_warning ("rtpbin: failed to get session pad for session %d", sessid);
2891 g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid);
2897 remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2899 if (session->send_rtp_src_ghost) {
2900 gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
2901 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2902 session->send_rtp_src_ghost);
2903 session->send_rtp_src_ghost = NULL;
2905 if (session->send_rtp_src) {
2906 gst_object_unref (session->send_rtp_src);
2907 session->send_rtp_src = NULL;
2909 if (session->send_rtp_sink) {
2910 gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
2911 session->send_rtp_sink);
2912 gst_object_unref (session->send_rtp_sink);
2913 session->send_rtp_sink = NULL;
2915 if (session->send_rtp_sink_ghost) {
2916 gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
2917 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2918 session->send_rtp_sink_ghost);
2919 session->send_rtp_sink_ghost = NULL;
2923 /* Create a pad for sending RTCP for the session in @name. Must be called with
2927 create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2930 GstRtpBinSession *session;
2932 /* first get the session number */
2933 if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
2936 /* get or create session */
2937 session = find_session_by_id (rtpbin, sessid);
2941 /* check if pad was requested */
2942 if (session->send_rtcp_src_ghost != NULL)
2943 return session->send_rtcp_src_ghost;
2945 /* get rtcp_src pad and store */
2946 session->send_rtcp_src =
2947 gst_element_get_request_pad (session->session, "send_rtcp_src");
2948 if (session->send_rtcp_src == NULL)
2951 session->send_rtcp_src_ghost =
2952 gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
2953 gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
2954 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
2956 return session->send_rtcp_src_ghost;
2961 g_warning ("rtpbin: invalid name given");
2966 g_warning ("rtpbin: session with id %d does not exist", sessid);
2971 g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
2977 remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
2979 if (session->send_rtcp_src_ghost) {
2980 gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
2981 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
2982 session->send_rtcp_src_ghost);
2983 session->send_rtcp_src_ghost = NULL;
2985 if (session->send_rtcp_src) {
2986 gst_element_release_request_pad (session->session, session->send_rtcp_src);
2987 gst_object_unref (session->send_rtcp_src);
2988 session->send_rtcp_src = NULL;
2992 /* If the requested name is NULL we should create a name with
2993 * the session number assuming we want the lowest posible session
2994 * with a free pad like the template */
2996 gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
2998 gboolean name_found = FALSE;
3000 GstIterator *pad_it = NULL;
3001 gchar *pad_name = NULL;
3002 GValue data = { 0, };
3004 GST_DEBUG_OBJECT (element, "find a free pad name for template");
3005 while (!name_found) {
3006 gboolean done = FALSE;
3009 pad_name = g_strdup_printf (templ->name_template, session++);
3010 pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
3013 switch (gst_iterator_next (pad_it, &data)) {
3014 case GST_ITERATOR_OK:
3019 pad = g_value_get_object (&data);
3020 name = gst_pad_get_name (pad);
3022 if (strcmp (name, pad_name) == 0) {
3027 g_value_reset (&data);
3030 case GST_ITERATOR_ERROR:
3031 case GST_ITERATOR_RESYNC:
3032 /* restart iteration */
3037 case GST_ITERATOR_DONE:
3042 g_value_unset (&data);
3043 gst_iterator_free (pad_it);
3046 GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
3053 gst_rtp_bin_request_new_pad (GstElement * element,
3054 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
3057 GstElementClass *klass;
3060 gchar *pad_name = NULL;
3062 g_return_val_if_fail (templ != NULL, NULL);
3063 g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
3065 rtpbin = GST_RTP_BIN (element);
3066 klass = GST_ELEMENT_GET_CLASS (element);
3068 GST_RTP_BIN_LOCK (rtpbin);
3071 /* use a free pad name */
3072 pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
3074 /* use the provided name */
3075 pad_name = g_strdup (name);
3078 GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
3080 /* figure out the template */
3081 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
3082 result = create_recv_rtp (rtpbin, templ, pad_name);
3083 } else if (templ == gst_element_class_get_pad_template (klass,
3084 "recv_rtcp_sink_%u")) {
3085 result = create_recv_rtcp (rtpbin, templ, pad_name);
3086 } else if (templ == gst_element_class_get_pad_template (klass,
3087 "send_rtp_sink_%u")) {
3088 result = create_send_rtp (rtpbin, templ, pad_name);
3089 } else if (templ == gst_element_class_get_pad_template (klass,
3090 "send_rtcp_src_%u")) {
3091 result = create_rtcp (rtpbin, templ, pad_name);
3093 goto wrong_template;
3096 GST_RTP_BIN_UNLOCK (rtpbin);
3104 GST_RTP_BIN_UNLOCK (rtpbin);
3105 g_warning ("rtpbin: this is not our template");
3111 gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
3113 GstRtpBinSession *session;
3116 g_return_if_fail (GST_IS_GHOST_PAD (pad));
3117 g_return_if_fail (GST_IS_RTP_BIN (element));
3119 rtpbin = GST_RTP_BIN (element);
3121 GST_RTP_BIN_LOCK (rtpbin);
3122 GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
3123 GST_DEBUG_PAD_NAME (pad));
3125 if (!(session = find_session_by_pad (rtpbin, pad)))
3128 if (session->recv_rtp_sink_ghost == pad) {
3129 remove_recv_rtp (rtpbin, session);
3130 } else if (session->recv_rtcp_sink_ghost == pad) {
3131 remove_recv_rtcp (rtpbin, session);
3132 } else if (session->send_rtp_sink_ghost == pad) {
3133 remove_send_rtp (rtpbin, session);
3134 } else if (session->send_rtcp_src_ghost == pad) {
3135 remove_rtcp (rtpbin, session);
3138 /* no more request pads, free the complete session */
3139 if (session->recv_rtp_sink_ghost == NULL
3140 && session->recv_rtcp_sink_ghost == NULL
3141 && session->send_rtp_sink_ghost == NULL
3142 && session->send_rtcp_src_ghost == NULL) {
3143 GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
3144 rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
3145 free_session (session, rtpbin);
3147 GST_RTP_BIN_UNLOCK (rtpbin);
3154 GST_RTP_BIN_UNLOCK (rtpbin);
3155 g_warning ("rtpbin: %s:%s is not one of our request pads",
3156 GST_DEBUG_PAD_NAME (pad));