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 * @short_description: handle media from one RTP bin
23 * @see_also: gstrtpjitterbuffer, gstrtpsession, gstrtpptdemux, gstrtpssrcdemux
27 * RTP bin combines the functions of gstrtpsession, gstrtpssrcdemux, gstrtpjitterbuffer
28 * and gstrtpptdemux in one element. It allows for multiple RTP sessions that will
29 * be synchronized together using RTCP SR packets.
32 * gstrtpbin is configured with a number of request pads that define the
33 * functionality that is activated, similar to the gstrtpsession element.
36 * To use gstrtpbin as an RTP receiver, request a recv_rtp_sink_%%d pad. The session
37 * number must be specified in the pad name.
38 * Data received on the recv_rtp_sink_%%d pad will be processed in the gstrtpsession
39 * manager and after being validated forwarded on gstrtpssrcdemuxer element. Each
40 * RTP stream is demuxed based on the SSRC and send to a gstrtpjitterbuffer. After
41 * the packets are released from the jitterbuffer, they will be forwarded to a
42 * gstrtpptdemuxer element. The gstrtpptdemuxer element will demux the packets based
43 * on the payload type and will create a unique pad recv_rtp_src_%%d_%%d_%%d on
44 * gstrtpbin with the session number, SSRC and payload type respectively as the pad
48 * To also use gstrtpbin as an RTCP receiver, request a recv_rtcp_sink_%%d pad. The
49 * session number must be specified in the pad name.
52 * If you want the session manager to generate and send RTCP packets, request
53 * the send_rtcp_src_%%d pad with the session number in the pad name. Packet pushed
54 * on this pad contain SR/RR RTCP reports that should be sent to all participants
58 * To use gstrtpbin as a sender, request a send_rtp_sink_%%d pad, which will
59 * automatically create a send_rtp_src_%%d pad. If the session number is not provided,
60 * the pad from the lowest available session will be returned. The session manager will modify the
61 * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
62 * send_rtp_src_%%d pad after updating its internal state.
65 * The session manager needs the clock-rate of the payload types it is handling
66 * and will signal the GstRtpSession::request-pt-map signal when it needs such a
67 * mapping. One can clear the cached values with the GstRtpSession::clear-pt-map
70 * <title>Example pipelines</title>
73 * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
74 * gstrtpbin ! rtptheoradepay ! theoradec ! xvimagesink
76 * Receive RTP data from port 5000 and send to the session 0 in gstrtpbin.
80 * gst-launch gstrtpbin name=rtpbin \
81 * v4l2src ! ffmpegcolorspace ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
82 * rtpbin.send_rtp_src_0 ! udpsink port=5000 \
83 * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
84 * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
85 * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
86 * rtpbin.send_rtp_src_1 ! udpsink port=5002 \
87 * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
88 * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
90 * Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
91 * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
92 * and the audio is sent to session 1. Video packets are sent on UDP port 5000
93 * and audio packets on port 5002. The video RTCP packets for session 0 are sent
94 * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
95 * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
96 * is received on port 5007. Since RTCP packets from the sender should be sent
97 * as soon as possible and do not participate in preroll, sync=false and
98 * async=false is configured on udpsink
102 * gst-launch -v gstrtpbin name=rtpbin \
103 * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
104 * port=5000 ! rtpbin.recv_rtp_sink_0 \
105 * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
106 * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
107 * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
108 * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
109 * port=5002 ! rtpbin.recv_rtp_sink_1 \
110 * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
111 * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
112 * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
114 * Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
115 * decode and display the video.
116 * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
117 * decode and play the audio.
118 * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
119 * session 1 on port 5003. These packets will be used for session management and
121 * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
126 * Last reviewed on 2007-08-30 (0.10.6)
134 #include <gst/rtp/gstrtpbuffer.h>
135 #include <gst/rtp/gstrtcpbuffer.h>
137 #include "gstrtpbin-marshal.h"
138 #include "gstrtpbin.h"
140 GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
141 #define GST_CAT_DEFAULT gst_rtp_bin_debug
143 /* elementfactory information */
144 static const GstElementDetails rtpbin_details = GST_ELEMENT_DETAILS ("RTP Bin",
145 "Filter/Network/RTP",
146 "Implement an RTP bin",
147 "Wim Taymans <wim.taymans@gmail.com>");
150 static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
151 GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%d",
154 GST_STATIC_CAPS ("application/x-rtp")
157 static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
158 GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%d",
161 GST_STATIC_CAPS ("application/x-rtcp")
164 static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
165 GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%d",
168 GST_STATIC_CAPS ("application/x-rtp")
172 static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
173 GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%d_%d_%d",
176 GST_STATIC_CAPS ("application/x-rtp")
179 static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
180 GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%d",
183 GST_STATIC_CAPS ("application/x-rtcp")
186 static GstStaticPadTemplate rtpbin_send_rtp_src_template =
187 GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%d",
190 GST_STATIC_CAPS ("application/x-rtp")
193 /* padtemplate for the internal pad */
194 static GstStaticPadTemplate rtpbin_sync_sink_template =
195 GST_STATIC_PAD_TEMPLATE ("sink_%d",
198 GST_STATIC_CAPS ("application/x-rtcp")
201 #define GST_RTP_BIN_GET_PRIVATE(obj) \
202 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
204 #define GST_RTP_BIN_LOCK(bin) g_mutex_lock ((bin)->priv->bin_lock)
205 #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock ((bin)->priv->bin_lock)
207 struct _GstRtpBinPrivate
211 GstClockTime ntp_ns_base;
214 /* signals and args */
217 SIGNAL_REQUEST_PT_MAP,
221 SIGNAL_ON_SSRC_COLLISION,
222 SIGNAL_ON_SSRC_VALIDATED,
223 SIGNAL_ON_SSRC_ACTIVE,
226 SIGNAL_ON_BYE_TIMEOUT,
231 #define DEFAULT_LATENCY_MS 200
232 #define DEFAULT_SDES_CNAME NULL
233 #define DEFAULT_SDES_NAME NULL
234 #define DEFAULT_SDES_EMAIL NULL
235 #define DEFAULT_SDES_PHONE NULL
236 #define DEFAULT_SDES_LOCATION NULL
237 #define DEFAULT_SDES_TOOL NULL
238 #define DEFAULT_SDES_NOTE NULL
239 #define DEFAULT_DO_LOST FALSE
257 typedef struct _GstRtpBinSession GstRtpBinSession;
258 typedef struct _GstRtpBinStream GstRtpBinStream;
259 typedef struct _GstRtpBinClient GstRtpBinClient;
261 static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
263 static GstCaps *pt_map_requested (GstElement * element, guint pt,
264 GstRtpBinSession * session);
265 static const gchar *sdes_type_to_name (GstRTCPSDESType type);
266 static void gst_rtp_bin_set_sdes_string (GstRtpBin * bin,
267 GstRTCPSDESType type, const gchar * data);
269 static void free_stream (GstRtpBinStream * stream);
271 /* Manages the RTP stream for one SSRC.
273 * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
274 * If we see an SDES RTCP packet that links multiple SSRCs together based on a
275 * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
276 * together (see below).
278 struct _GstRtpBinStream
280 /* the SSRC of this stream */
286 /* the session this SSRC belongs to */
287 GstRtpBinSession *session;
289 /* the jitterbuffer of the SSRC */
292 /* the PT demuxer of the SSRC */
294 gulong demux_newpad_sig;
295 gulong demux_ptreq_sig;
296 gulong demux_pt_change_sig;
298 /* the internal pad we use to get RTCP sync messages */
302 guint64 last_extrtptime;
304 /* mapping to local RTP and NTP time */
311 guint64 clock_base_time;
314 gint64 prev_ts_offset;
318 #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->lock)
319 #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->lock)
321 /* Manages the receiving end of the packets.
323 * There is one such structure for each RTP session (audio/video/...).
324 * We get the RTP/RTCP packets and stuff them into the session manager. From
325 * there they are pushed into an SSRC demuxer that splits the stream based on
326 * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
327 * the GstRtpBinStream above).
329 struct _GstRtpBinSession
335 /* the session element */
337 /* the SSRC demuxer */
339 gulong demux_newpad_sig;
343 /* list of GstRtpBinStream */
346 /* mapping of payload type to caps */
349 /* the pads of the session */
350 GstPad *recv_rtp_sink;
351 GstPad *recv_rtp_src;
352 GstPad *recv_rtcp_sink;
354 GstPad *send_rtp_sink;
355 GstPad *send_rtp_src;
356 GstPad *send_rtcp_src;
359 /* Manages the RTP streams that come from one client and should therefore be
362 struct _GstRtpBinClient
364 /* the common CNAME for the streams */
375 /* find a session with the given id. Must be called with RTP_BIN_LOCK */
376 static GstRtpBinSession *
377 find_session_by_id (GstRtpBin * rtpbin, gint id)
381 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
382 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
391 on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
393 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
398 on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
400 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
405 on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
407 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
412 on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
414 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
419 on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
421 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
426 on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
428 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
433 on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
435 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
440 on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
442 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
446 /* create a session with the given id. Must be called with RTP_BIN_LOCK */
447 static GstRtpBinSession *
448 create_session (GstRtpBin * rtpbin, gint id)
450 GstRtpBinSession *sess;
451 GstElement *session, *demux;
454 if (!(session = gst_element_factory_make ("gstrtpsession", NULL)))
457 if (!(demux = gst_element_factory_make ("gstrtpssrcdemux", NULL)))
460 sess = g_new0 (GstRtpBinSession, 1);
461 sess->lock = g_mutex_new ();
464 sess->session = session;
466 sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
467 (GDestroyNotify) gst_caps_unref);
468 rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
470 /* set NTP base or new session */
471 g_object_set (session, "ntp-ns-base", rtpbin->priv->ntp_ns_base, NULL);
472 /* configure SDES items */
473 GST_OBJECT_LOCK (rtpbin);
474 for (i = GST_RTCP_SDES_CNAME; i < GST_RTCP_SDES_PRIV; i++) {
475 g_object_set (session, sdes_type_to_name (i), rtpbin->sdes[i], NULL);
477 GST_OBJECT_UNLOCK (rtpbin);
479 /* provide clock_rate to the session manager when needed */
480 g_signal_connect (session, "request-pt-map",
481 (GCallback) pt_map_requested, sess);
483 g_signal_connect (sess->session, "on-new-ssrc",
484 (GCallback) on_new_ssrc, sess);
485 g_signal_connect (sess->session, "on-ssrc-collision",
486 (GCallback) on_ssrc_collision, sess);
487 g_signal_connect (sess->session, "on-ssrc-validated",
488 (GCallback) on_ssrc_validated, sess);
489 g_signal_connect (sess->session, "on-ssrc-active",
490 (GCallback) on_ssrc_active, sess);
491 g_signal_connect (sess->session, "on-ssrc-sdes",
492 (GCallback) on_ssrc_sdes, sess);
493 g_signal_connect (sess->session, "on-bye-ssrc",
494 (GCallback) on_bye_ssrc, sess);
495 g_signal_connect (sess->session, "on-bye-timeout",
496 (GCallback) on_bye_timeout, sess);
497 g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
499 /* FIXME, change state only to what's needed */
500 gst_bin_add (GST_BIN_CAST (rtpbin), session);
501 gst_element_set_state (session, GST_STATE_PLAYING);
502 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
503 gst_element_set_state (demux, GST_STATE_PLAYING);
510 g_warning ("gstrtpbin: could not create gstrtpsession element");
515 gst_object_unref (session);
516 g_warning ("gstrtpbin: could not create gstrtpssrcdemux element");
522 free_session (GstRtpBinSession * sess)
528 gst_element_set_state (sess->session, GST_STATE_NULL);
529 gst_element_set_state (sess->demux, GST_STATE_NULL);
531 if (sess->recv_rtp_sink != NULL)
532 gst_element_release_request_pad (sess->session, sess->recv_rtp_sink);
533 if (sess->recv_rtp_src != NULL)
534 gst_object_unref (sess->recv_rtp_src);
535 if (sess->recv_rtcp_sink != NULL)
536 gst_element_release_request_pad (sess->session, sess->recv_rtcp_sink);
537 if (sess->sync_src != NULL)
538 gst_object_unref (sess->sync_src);
539 if (sess->send_rtp_sink != NULL)
540 gst_element_release_request_pad (sess->session, sess->send_rtp_sink);
541 if (sess->send_rtp_src != NULL)
542 gst_object_unref (sess->send_rtp_src);
543 if (sess->send_rtcp_src != NULL)
544 gst_element_release_request_pad (sess->session, sess->send_rtcp_src);
546 gst_bin_remove (GST_BIN_CAST (bin), sess->session);
547 gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
549 g_slist_foreach (sess->streams, (GFunc) free_stream, NULL);
550 g_slist_free (sess->streams);
552 g_mutex_free (sess->lock);
553 g_hash_table_destroy (sess->ptmap);
555 bin->sessions = g_slist_remove (bin->sessions, sess);
561 static GstRtpBinStream *
562 find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
566 for (walk = session->streams; walk; walk = g_slist_next (walk)) {
567 GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
569 if (stream->ssrc == ssrc)
576 /* get the payload type caps for the specific payload @pt in @session */
578 get_pt_map (GstRtpBinSession * session, guint pt)
580 GstCaps *caps = NULL;
583 GValue args[3] = { {0}, {0}, {0} };
585 GST_DEBUG ("searching pt %d in cache", pt);
587 GST_RTP_SESSION_LOCK (session);
589 /* first look in the cache */
590 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
598 GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id);
600 /* not in cache, send signal to request caps */
601 g_value_init (&args[0], GST_TYPE_ELEMENT);
602 g_value_set_object (&args[0], bin);
603 g_value_init (&args[1], G_TYPE_UINT);
604 g_value_set_uint (&args[1], session->id);
605 g_value_init (&args[2], G_TYPE_UINT);
606 g_value_set_uint (&args[2], pt);
608 g_value_init (&ret, GST_TYPE_CAPS);
609 g_value_set_boxed (&ret, NULL);
611 g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
613 g_value_unset (&args[0]);
614 g_value_unset (&args[1]);
615 g_value_unset (&args[2]);
616 caps = (GstCaps *) g_value_dup_boxed (&ret);
617 g_value_unset (&ret);
621 GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps);
623 /* store in cache, take additional ref */
624 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
625 gst_caps_ref (caps));
628 GST_RTP_SESSION_UNLOCK (session);
635 GST_RTP_SESSION_UNLOCK (session);
636 GST_DEBUG ("no pt map could be obtained");
642 return_true (gpointer key, gpointer value, gpointer user_data)
648 gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
650 GSList *sessions, *streams;
652 GST_RTP_BIN_LOCK (bin);
653 GST_DEBUG_OBJECT (bin, "clearing pt map");
654 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
655 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
657 GST_DEBUG_OBJECT (bin, "clearing session %p", session);
658 g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
660 GST_RTP_SESSION_LOCK (session);
661 g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
663 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
664 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
666 GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
667 g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
668 g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
670 GST_RTP_SESSION_UNLOCK (session);
672 GST_RTP_BIN_UNLOCK (bin);
676 gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
677 const gchar * name, const GValue * value)
679 GSList *sessions, *streams;
681 GST_RTP_BIN_LOCK (bin);
682 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
683 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
685 GST_RTP_SESSION_LOCK (session);
686 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
687 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
689 g_object_set_property (G_OBJECT (stream->buffer), name, value);
691 GST_RTP_SESSION_UNLOCK (session);
693 GST_RTP_BIN_UNLOCK (bin);
696 /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
697 static GstRtpBinClient *
698 get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
700 GstRtpBinClient *result = NULL;
703 for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
704 GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
706 if (len != client->cname_len)
709 if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
710 GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
717 /* nothing found, create one */
718 if (result == NULL) {
719 result = g_new0 (GstRtpBinClient, 1);
720 result->cname = g_strndup ((gchar *) data, len);
721 result->cname_len = len;
722 result->min_delta = G_MAXINT64;
723 bin->clients = g_slist_prepend (bin->clients, result);
724 GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
731 free_client (GstRtpBinClient * client)
733 g_slist_free (client->streams);
734 g_free (client->cname);
738 /* associate a stream to the given CNAME. This will make sure all streams for
739 * that CNAME are synchronized together. */
741 gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
744 GstRtpBinClient *client;
748 /* first find or create the CNAME */
749 GST_RTP_BIN_LOCK (bin);
750 client = get_client (bin, len, data, &created);
752 /* find stream in the client */
753 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
754 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
756 if (ostream == stream)
759 /* not found, add it to the list */
761 GST_DEBUG_OBJECT (bin,
762 "new association of SSRC %08x with client %p with CNAME %s",
763 stream->ssrc, client, client->cname);
764 client->streams = g_slist_prepend (client->streams, stream);
767 GST_DEBUG_OBJECT (bin,
768 "found association of SSRC %08x with client %p with CNAME %s",
769 stream->ssrc, client, client->cname);
772 /* we can only continue if we know the local clock-base and clock-rate */
773 if (stream->clock_base == -1)
776 if (stream->clock_rate <= 0) {
778 GstCaps *caps = NULL;
779 GstStructure *s = NULL;
781 GST_RTP_SESSION_LOCK (stream->session);
782 pt = stream->last_pt;
783 GST_RTP_SESSION_UNLOCK (stream->session);
788 caps = get_pt_map (stream->session, pt);
792 s = gst_caps_get_structure (caps, 0);
793 gst_structure_get_int (s, "clock-rate", &stream->clock_rate);
794 gst_caps_unref (caps);
796 if (stream->clock_rate <= 0)
800 /* map last RTP time to local timeline using our clock-base */
801 stream->local_rtp = stream->last_extrtptime - stream->clock_base;
803 GST_DEBUG_OBJECT (bin,
804 "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
805 ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d", stream->clock_base,
806 stream->last_extrtptime, stream->local_rtp, stream->clock_rate);
808 /* calculate local NTP time in gstreamer timestamp */
810 gst_util_uint64_scale_int (stream->local_rtp, GST_SECOND,
812 stream->local_unix += stream->clock_base_time;
813 /* calculate delta between server and receiver */
814 stream->unix_delta = stream->last_unix - stream->local_unix;
816 GST_DEBUG_OBJECT (bin,
817 "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT
818 ", delta %" G_GINT64_FORMAT, stream->local_unix, stream->last_unix,
821 /* recalc inter stream playout offset, but only if there are more than one
823 if (client->nstreams > 1) {
826 /* calculate the min of all deltas */
828 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
829 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
831 if (ostream->unix_delta && ostream->unix_delta < min)
832 min = ostream->unix_delta;
835 GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT, client,
838 /* calculate offsets for each stream */
839 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
840 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
842 if (ostream->unix_delta == 0)
845 ostream->ts_offset = ostream->unix_delta - min;
847 /* delta changed, see how much */
848 if (ostream->prev_ts_offset != ostream->ts_offset) {
851 if (ostream->prev_ts_offset > ostream->ts_offset)
852 diff = ostream->prev_ts_offset - ostream->ts_offset;
854 diff = ostream->ts_offset - ostream->prev_ts_offset;
856 /* only change diff when it changed more than 1 millisecond. This
857 * compensates for rounding errors in NTP to RTP timestamp
859 if (diff > GST_MSECOND)
860 g_object_set (ostream->buffer, "ts-offset", ostream->ts_offset, NULL);
862 ostream->prev_ts_offset = ostream->ts_offset;
864 GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
865 ostream->ssrc, ostream->ts_offset);
868 GST_RTP_BIN_UNLOCK (bin);
873 GST_WARNING_OBJECT (bin, "we have no clock-base");
874 GST_RTP_BIN_UNLOCK (bin);
879 GST_WARNING_OBJECT (bin, "we have no clock-rate");
880 GST_RTP_BIN_UNLOCK (bin);
885 #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
886 for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
887 (b) = gst_rtcp_packet_move_to_next ((packet)))
889 #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
890 for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
891 (b) = gst_rtcp_packet_sdes_next_item ((packet)))
893 #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
894 for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
895 (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
898 gst_rtp_bin_sync_chain (GstPad * pad, GstBuffer * buffer)
900 GstFlowReturn ret = GST_FLOW_OK;
901 GstRtpBinStream *stream;
903 GstRTCPPacket packet;
907 gboolean have_sr, have_sdes;
910 stream = gst_pad_get_element_private (pad);
913 GST_DEBUG_OBJECT (bin, "received sync packet");
915 if (!gst_rtcp_buffer_validate (buffer))
920 GST_RTCP_BUFFER_FOR_PACKETS (more, buffer, &packet) {
921 /* first packet must be SR or RR or else the validate would have failed */
922 switch (gst_rtcp_packet_get_type (&packet)) {
923 case GST_RTCP_TYPE_SR:
924 /* only parse first. There is only supposed to be one SR in the packet
925 * but we will deal with malformed packets gracefully */
928 /* get NTP and RTP times */
929 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
932 GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
933 /* ignore SR that is not ours */
934 if (ssrc != stream->ssrc)
939 /* store values in the stream */
940 stream->have_sync = TRUE;
941 stream->last_unix = gst_rtcp_ntp_to_unix (ntptime);
942 /* use extended timestamp */
943 gst_rtp_buffer_ext_timestamp (&stream->last_extrtptime, rtptime);
945 case GST_RTCP_TYPE_SDES:
947 gboolean more_items, more_entries;
949 /* only deal with first SDES, there is only supposed to be one SDES in
950 * the RTCP packet but we deal with bad packets gracefully. Also bail
951 * out if we have not seen an SR item yet. */
952 if (have_sdes || !have_sr)
955 GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
956 /* skip items that are not about the SSRC of the sender */
957 if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
960 /* find the CNAME entry */
961 GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
962 GstRTCPSDESType type;
966 gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
968 if (type == GST_RTCP_SDES_CNAME) {
969 stream->clock_base = GST_BUFFER_OFFSET (buffer);
970 stream->clock_base_time = GST_BUFFER_OFFSET_END (buffer);
971 /* associate the stream to CNAME */
972 gst_rtp_bin_associate (bin, stream, len, data);
980 /* we can ignore these packets */
985 gst_buffer_unref (buffer);
992 /* this is fatal and should be filtered earlier */
993 GST_ELEMENT_ERROR (bin, STREAM, DECODE, (NULL),
994 ("invalid RTCP packet received"));
995 gst_buffer_unref (buffer);
996 return GST_FLOW_ERROR;
1000 /* create a new stream with @ssrc in @session. Must be called with
1001 * RTP_SESSION_LOCK. */
1002 static GstRtpBinStream *
1003 create_stream (GstRtpBinSession * session, guint32 ssrc)
1005 GstElement *buffer, *demux;
1006 GstRtpBinStream *stream;
1007 GstPadTemplate *templ;
1010 if (!(buffer = gst_element_factory_make ("gstrtpjitterbuffer", NULL)))
1011 goto no_jitterbuffer;
1013 if (!(demux = gst_element_factory_make ("gstrtpptdemux", NULL)))
1016 stream = g_new0 (GstRtpBinStream, 1);
1017 stream->ssrc = ssrc;
1018 stream->bin = session->bin;
1019 stream->session = session;
1020 stream->buffer = buffer;
1021 stream->demux = demux;
1022 stream->last_extrtptime = -1;
1023 stream->last_pt = -1;
1024 stream->have_sync = FALSE;
1025 session->streams = g_slist_prepend (session->streams, stream);
1027 /* make an internal sinkpad for RTCP sync packets. Take ownership of the
1028 * pad. We will link this pad later. */
1029 padname = g_strdup_printf ("sync_%d", ssrc);
1030 templ = gst_static_pad_template_get (&rtpbin_sync_sink_template);
1031 stream->sync_pad = gst_pad_new_from_template (templ, padname);
1032 gst_object_unref (templ);
1034 gst_object_ref (stream->sync_pad);
1035 gst_object_sink (stream->sync_pad);
1036 gst_pad_set_element_private (stream->sync_pad, stream);
1037 gst_pad_set_chain_function (stream->sync_pad, gst_rtp_bin_sync_chain);
1038 gst_pad_set_active (stream->sync_pad, TRUE);
1040 /* provide clock_rate to the jitterbuffer when needed */
1041 g_signal_connect (buffer, "request-pt-map",
1042 (GCallback) pt_map_requested, session);
1044 /* configure latency and packet lost */
1045 g_object_set (buffer, "latency", session->bin->latency, NULL);
1046 g_object_set (buffer, "do-lost", session->bin->do_lost, NULL);
1048 gst_bin_add (GST_BIN_CAST (session->bin), buffer);
1049 gst_element_set_state (buffer, GST_STATE_PLAYING);
1050 gst_bin_add (GST_BIN_CAST (session->bin), demux);
1051 gst_element_set_state (demux, GST_STATE_PLAYING);
1054 gst_element_link (buffer, demux);
1061 g_warning ("gstrtpbin: could not create gstrtpjitterbuffer element");
1066 gst_object_unref (buffer);
1067 g_warning ("gstrtpbin: could not create gstrtpptdemux element");
1073 free_stream (GstRtpBinStream * stream)
1075 GstRtpBinSession *session;
1077 session = stream->session;
1079 gst_element_set_state (stream->buffer, GST_STATE_NULL);
1080 gst_element_set_state (stream->demux, GST_STATE_NULL);
1082 gst_bin_remove (GST_BIN_CAST (session->bin), stream->buffer);
1083 gst_bin_remove (GST_BIN_CAST (session->bin), stream->demux);
1085 gst_object_unref (stream->sync_pad);
1087 session->streams = g_slist_remove (session->streams, stream);
1092 /* GObject vmethods */
1093 static void gst_rtp_bin_dispose (GObject * object);
1094 static void gst_rtp_bin_finalize (GObject * object);
1095 static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
1096 const GValue * value, GParamSpec * pspec);
1097 static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
1098 GValue * value, GParamSpec * pspec);
1100 /* GstElement vmethods */
1101 static GstClock *gst_rtp_bin_provide_clock (GstElement * element);
1102 static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
1103 GstStateChange transition);
1104 static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
1105 GstPadTemplate * templ, const gchar * name);
1106 static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
1107 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
1108 static void gst_rtp_bin_clear_pt_map (GstRtpBin * bin);
1110 GST_BOILERPLATE (GstRtpBin, gst_rtp_bin, GstBin, GST_TYPE_BIN);
1113 gst_rtp_bin_base_init (gpointer klass)
1115 GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
1118 gst_element_class_add_pad_template (element_class,
1119 gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
1120 gst_element_class_add_pad_template (element_class,
1121 gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
1122 gst_element_class_add_pad_template (element_class,
1123 gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
1126 gst_element_class_add_pad_template (element_class,
1127 gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
1128 gst_element_class_add_pad_template (element_class,
1129 gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
1130 gst_element_class_add_pad_template (element_class,
1131 gst_static_pad_template_get (&rtpbin_send_rtp_src_template));
1133 gst_element_class_set_details (element_class, &rtpbin_details);
1137 gst_rtp_bin_class_init (GstRtpBinClass * klass)
1139 GObjectClass *gobject_class;
1140 GstElementClass *gstelement_class;
1141 GstBinClass *gstbin_class;
1143 gobject_class = (GObjectClass *) klass;
1144 gstelement_class = (GstElementClass *) klass;
1145 gstbin_class = (GstBinClass *) klass;
1147 g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
1149 gobject_class->dispose = gst_rtp_bin_dispose;
1150 gobject_class->finalize = gst_rtp_bin_finalize;
1151 gobject_class->set_property = gst_rtp_bin_set_property;
1152 gobject_class->get_property = gst_rtp_bin_get_property;
1154 g_object_class_install_property (gobject_class, PROP_LATENCY,
1155 g_param_spec_uint ("latency", "Buffer latency in ms",
1156 "Default amount of ms to buffer in the jitterbuffers", 0,
1157 G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE));
1160 * GstRtpBin::request-pt-map:
1161 * @rtpbin: the object which received the signal
1162 * @session: the session
1165 * Request the payload type as #GstCaps for @pt in @session.
1167 gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
1168 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
1169 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
1170 NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, GST_TYPE_CAPS, 2,
1171 G_TYPE_UINT, G_TYPE_UINT);
1173 * GstRtpBin::clear-pt-map:
1174 * @rtpbin: the object which received the signal
1176 * Clear all previously cached pt-mapping obtained with
1177 * GstRtpBin::request-pt-map.
1179 gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
1180 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
1181 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
1182 clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
1186 * GstRtpBin::on-new-ssrc:
1187 * @rtpbin: the object which received the signal
1188 * @session: the session
1191 * Notify of a new SSRC that entered @session.
1193 gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
1194 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
1195 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
1196 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1197 G_TYPE_UINT, G_TYPE_UINT);
1199 * GstRtpBin::on-ssrc-collision:
1200 * @rtpbin: the object which received the signal
1201 * @session: the session
1204 * Notify when we have an SSRC collision
1206 gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
1207 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
1208 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
1209 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1210 G_TYPE_UINT, G_TYPE_UINT);
1212 * GstRtpBin::on-ssrc-validated:
1213 * @rtpbin: the object which received the signal
1214 * @session: the session
1217 * Notify of a new SSRC that became validated.
1219 gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
1220 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
1221 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
1222 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1223 G_TYPE_UINT, G_TYPE_UINT);
1225 * GstRtpBin::on-ssrc-active:
1226 * @rtpbin: the object which received the signal
1227 * @session: the session
1230 * Notify of a SSRC that is active, i.e., sending RTCP.
1232 gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
1233 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
1234 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
1235 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1236 G_TYPE_UINT, G_TYPE_UINT);
1238 * GstRtpBin::on-ssrc-sdes:
1239 * @rtpbin: the object which received the signal
1240 * @session: the session
1243 * Notify of a SSRC that is active, i.e., sending RTCP.
1245 gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
1246 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
1247 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
1248 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1249 G_TYPE_UINT, G_TYPE_UINT);
1252 * GstRtpBin::on-bye-ssrc:
1253 * @rtpbin: the object which received the signal
1254 * @session: the session
1257 * Notify of an SSRC that became inactive because of a BYE packet.
1259 gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
1260 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
1261 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
1262 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1263 G_TYPE_UINT, G_TYPE_UINT);
1265 * GstRtpBin::on-bye-timeout:
1266 * @rtpbin: the object which received the signal
1267 * @session: the session
1270 * Notify of an SSRC that has timed out because of BYE
1272 gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
1273 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
1274 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
1275 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1276 G_TYPE_UINT, G_TYPE_UINT);
1278 * GstRtpBin::on-timeout:
1279 * @rtpbin: the object which received the signal
1280 * @session: the session
1283 * Notify of an SSRC that has timed out
1285 gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
1286 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
1287 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
1288 NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT, G_TYPE_NONE, 2,
1289 G_TYPE_UINT, G_TYPE_UINT);
1291 g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
1292 g_param_spec_string ("sdes-cname", "SDES CNAME",
1293 "The CNAME to put in SDES messages of this session",
1294 DEFAULT_SDES_CNAME, G_PARAM_READWRITE));
1296 g_object_class_install_property (gobject_class, PROP_SDES_NAME,
1297 g_param_spec_string ("sdes-name", "SDES NAME",
1298 "The NAME to put in SDES messages of this session",
1299 DEFAULT_SDES_NAME, G_PARAM_READWRITE));
1301 g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
1302 g_param_spec_string ("sdes-email", "SDES EMAIL",
1303 "The EMAIL to put in SDES messages of this session",
1304 DEFAULT_SDES_EMAIL, G_PARAM_READWRITE));
1306 g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
1307 g_param_spec_string ("sdes-phone", "SDES PHONE",
1308 "The PHONE to put in SDES messages of this session",
1309 DEFAULT_SDES_PHONE, G_PARAM_READWRITE));
1311 g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
1312 g_param_spec_string ("sdes-location", "SDES LOCATION",
1313 "The LOCATION to put in SDES messages of this session",
1314 DEFAULT_SDES_LOCATION, G_PARAM_READWRITE));
1316 g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
1317 g_param_spec_string ("sdes-tool", "SDES TOOL",
1318 "The TOOL to put in SDES messages of this session",
1319 DEFAULT_SDES_TOOL, G_PARAM_READWRITE));
1321 g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
1322 g_param_spec_string ("sdes-note", "SDES NOTE",
1323 "The NOTE to put in SDES messages of this session",
1324 DEFAULT_SDES_NOTE, G_PARAM_READWRITE));
1326 g_object_class_install_property (gobject_class, PROP_DO_LOST,
1327 g_param_spec_boolean ("do-lost", "Do Lost",
1328 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
1329 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
1331 gstelement_class->provide_clock =
1332 GST_DEBUG_FUNCPTR (gst_rtp_bin_provide_clock);
1333 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
1334 gstelement_class->request_new_pad =
1335 GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
1336 gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
1338 gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
1340 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
1342 GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
1346 gst_rtp_bin_init (GstRtpBin * rtpbin, GstRtpBinClass * klass)
1350 rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
1351 rtpbin->priv->bin_lock = g_mutex_new ();
1352 rtpbin->provided_clock = gst_system_clock_obtain ();
1354 rtpbin->latency = DEFAULT_LATENCY_MS;
1355 rtpbin->do_lost = DEFAULT_DO_LOST;
1357 /* some default SDES entries */
1358 str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
1359 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_CNAME, str);
1362 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NAME, g_get_real_name ());
1363 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_TOOL, "GStreamer");
1367 gst_rtp_bin_dispose (GObject * object)
1371 rtpbin = GST_RTP_BIN (object);
1373 g_slist_foreach (rtpbin->sessions, (GFunc) free_session, NULL);
1374 g_slist_free (rtpbin->sessions);
1375 rtpbin->sessions = NULL;
1376 g_slist_foreach (rtpbin->clients, (GFunc) free_client, NULL);
1377 g_slist_free (rtpbin->clients);
1378 rtpbin->clients = NULL;
1380 G_OBJECT_CLASS (parent_class)->dispose (object);
1384 gst_rtp_bin_finalize (GObject * object)
1389 rtpbin = GST_RTP_BIN (object);
1391 for (i = 0; i < 9; i++)
1392 g_free (rtpbin->sdes[i]);
1394 g_mutex_free (rtpbin->priv->bin_lock);
1395 gst_object_unref (rtpbin->provided_clock);
1397 G_OBJECT_CLASS (parent_class)->finalize (object);
1400 static const gchar *
1401 sdes_type_to_name (GstRTCPSDESType type)
1403 const gchar *result;
1406 case GST_RTCP_SDES_CNAME:
1407 result = "sdes-cname";
1409 case GST_RTCP_SDES_NAME:
1410 result = "sdes-name";
1412 case GST_RTCP_SDES_EMAIL:
1413 result = "sdes-email";
1415 case GST_RTCP_SDES_PHONE:
1416 result = "sdes-phone";
1418 case GST_RTCP_SDES_LOC:
1419 result = "sdes-location";
1421 case GST_RTCP_SDES_TOOL:
1422 result = "sdes-tool";
1424 case GST_RTCP_SDES_NOTE:
1425 result = "sdes-note";
1427 case GST_RTCP_SDES_PRIV:
1428 result = "sdes-priv";
1438 gst_rtp_bin_set_sdes_string (GstRtpBin * bin, GstRTCPSDESType type,
1444 if (type < 0 || type > 8)
1447 GST_OBJECT_LOCK (bin);
1448 g_free (bin->sdes[type]);
1449 bin->sdes[type] = g_strdup (data);
1450 name = sdes_type_to_name (type);
1451 /* store in all sessions */
1452 for (item = bin->sessions; item; item = g_slist_next (item))
1453 g_object_set (item->data, name, bin->sdes[type], NULL);
1454 GST_OBJECT_UNLOCK (bin);
1458 gst_rtp_bin_get_sdes_string (GstRtpBin * bin, GstRTCPSDESType type)
1462 if (type < 0 || type > 8)
1465 GST_OBJECT_LOCK (bin);
1466 result = g_strdup (bin->sdes[type]);
1467 GST_OBJECT_UNLOCK (bin);
1473 gst_rtp_bin_set_property (GObject * object, guint prop_id,
1474 const GValue * value, GParamSpec * pspec)
1478 rtpbin = GST_RTP_BIN (object);
1482 GST_RTP_BIN_LOCK (rtpbin);
1483 rtpbin->latency = g_value_get_uint (value);
1484 GST_RTP_BIN_UNLOCK (rtpbin);
1485 /* propegate the property down to the jitterbuffer */
1486 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
1488 case PROP_SDES_CNAME:
1489 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_CNAME,
1490 g_value_get_string (value));
1492 case PROP_SDES_NAME:
1493 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NAME,
1494 g_value_get_string (value));
1496 case PROP_SDES_EMAIL:
1497 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_EMAIL,
1498 g_value_get_string (value));
1500 case PROP_SDES_PHONE:
1501 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_PHONE,
1502 g_value_get_string (value));
1504 case PROP_SDES_LOCATION:
1505 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_LOC,
1506 g_value_get_string (value));
1508 case PROP_SDES_TOOL:
1509 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_TOOL,
1510 g_value_get_string (value));
1512 case PROP_SDES_NOTE:
1513 gst_rtp_bin_set_sdes_string (rtpbin, GST_RTCP_SDES_NOTE,
1514 g_value_get_string (value));
1517 GST_RTP_BIN_LOCK (rtpbin);
1518 rtpbin->do_lost = g_value_get_boolean (value);
1519 GST_RTP_BIN_UNLOCK (rtpbin);
1520 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
1523 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1529 gst_rtp_bin_get_property (GObject * object, guint prop_id,
1530 GValue * value, GParamSpec * pspec)
1534 rtpbin = GST_RTP_BIN (object);
1538 GST_RTP_BIN_LOCK (rtpbin);
1539 g_value_set_uint (value, rtpbin->latency);
1540 GST_RTP_BIN_UNLOCK (rtpbin);
1542 case PROP_SDES_CNAME:
1543 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1544 GST_RTCP_SDES_CNAME));
1546 case PROP_SDES_NAME:
1547 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1548 GST_RTCP_SDES_NAME));
1550 case PROP_SDES_EMAIL:
1551 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1552 GST_RTCP_SDES_EMAIL));
1554 case PROP_SDES_PHONE:
1555 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1556 GST_RTCP_SDES_PHONE));
1558 case PROP_SDES_LOCATION:
1559 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1560 GST_RTCP_SDES_LOC));
1562 case PROP_SDES_TOOL:
1563 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1564 GST_RTCP_SDES_TOOL));
1566 case PROP_SDES_NOTE:
1567 g_value_take_string (value, gst_rtp_bin_get_sdes_string (rtpbin,
1568 GST_RTCP_SDES_NOTE));
1571 GST_RTP_BIN_LOCK (rtpbin);
1572 g_value_set_boolean (value, rtpbin->do_lost);
1573 GST_RTP_BIN_UNLOCK (rtpbin);
1576 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1582 gst_rtp_bin_provide_clock (GstElement * element)
1586 rtpbin = GST_RTP_BIN (element);
1588 return GST_CLOCK_CAST (gst_object_ref (rtpbin->provided_clock));
1592 gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
1596 rtpbin = GST_RTP_BIN (bin);
1598 switch (GST_MESSAGE_TYPE (message)) {
1599 case GST_MESSAGE_ELEMENT:
1601 const GstStructure *s = gst_message_get_structure (message);
1603 /* we change the structure name and add the session ID to it */
1604 if (gst_structure_has_name (s, "GstRTPSessionSDES")) {
1607 /* find the session, the message source has it */
1608 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
1609 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
1611 /* if we found the session, change message. else we exit the loop and
1612 * leave the message unchanged */
1613 if (GST_OBJECT_CAST (sess->session) == GST_MESSAGE_SRC (message)) {
1614 message = gst_message_make_writable (message);
1615 s = gst_message_get_structure (message);
1617 gst_structure_set_name ((GstStructure *) s, "GstRTPBinSDES");
1619 gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
1625 /* fallthrough to forward the modified message to the parent */
1629 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
1636 calc_ntp_ns_base (GstRtpBin * bin)
1642 /* get the current time and convert it to NTP time in nanoseconds */
1643 g_get_current_time (¤t);
1644 now = GST_TIMEVAL_TO_TIME (current);
1645 now += (2208988800LL * GST_SECOND);
1647 GST_RTP_BIN_LOCK (bin);
1648 bin->priv->ntp_ns_base = now;
1649 for (walk = bin->sessions; walk; walk = g_slist_next (walk)) {
1650 GstRtpBinSession *session = (GstRtpBinSession *) walk->data;
1652 g_object_set (session->session, "ntp-ns-base", now, NULL);
1654 GST_RTP_BIN_UNLOCK (bin);
1659 static GstStateChangeReturn
1660 gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
1662 GstStateChangeReturn res;
1665 rtpbin = GST_RTP_BIN (element);
1667 switch (transition) {
1668 case GST_STATE_CHANGE_NULL_TO_READY:
1670 case GST_STATE_CHANGE_READY_TO_PAUSED:
1672 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
1673 calc_ntp_ns_base (rtpbin);
1679 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
1681 switch (transition) {
1682 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
1684 case GST_STATE_CHANGE_PAUSED_TO_READY:
1686 case GST_STATE_CHANGE_READY_TO_NULL:
1694 /* a new pad (SSRC) was created in @session */
1696 new_payload_found (GstElement * element, guint pt, GstPad * pad,
1697 GstRtpBinStream * stream)
1700 GstElementClass *klass;
1701 GstPadTemplate *templ;
1705 rtpbin = stream->bin;
1707 GST_DEBUG ("new payload pad %d", pt);
1709 /* ghost the pad to the parent */
1710 klass = GST_ELEMENT_GET_CLASS (rtpbin);
1711 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%d_%d_%d");
1712 padname = g_strdup_printf ("recv_rtp_src_%d_%u_%d",
1713 stream->session->id, stream->ssrc, pt);
1714 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
1717 gst_pad_set_caps (gpad, GST_PAD_CAPS (pad));
1718 gst_pad_set_active (gpad, TRUE);
1719 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
1723 pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
1728 rtpbin = session->bin;
1730 GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt,
1733 caps = get_pt_map (session, pt);
1742 GST_DEBUG_OBJECT (rtpbin, "could not get caps");
1747 /* emited when caps changed for the session */
1749 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
1754 const GstStructure *s;
1758 g_object_get (pad, "caps", &caps, NULL);
1763 GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
1765 s = gst_caps_get_structure (caps, 0);
1767 /* get payload, finish when it's not there */
1768 if (!gst_structure_get_int (s, "payload", &payload))
1771 GST_RTP_SESSION_LOCK (session);
1772 GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
1773 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
1774 GST_RTP_SESSION_UNLOCK (session);
1777 /* Stores the last payload type received on a particular stream */
1779 payload_type_change (GstElement * element, guint pt, GstRtpBinStream * stream)
1781 GST_RTP_SESSION_LOCK (stream->session);
1782 stream->last_pt = pt;
1783 GST_RTP_SESSION_UNLOCK (stream->session);
1786 /* a new pad (SSRC) was created in @session */
1788 new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
1789 GstRtpBinSession * session)
1791 GstRtpBinStream *stream;
1792 GstPad *sinkpad, *srcpad;
1796 GST_DEBUG_OBJECT (session->bin, "new SSRC pad %08x", ssrc);
1798 GST_RTP_SESSION_LOCK (session);
1800 /* create new stream */
1801 stream = create_stream (session, ssrc);
1805 /* get the caps of the pad, we need the clock-rate and base_time if any. */
1806 if ((caps = gst_pad_get_caps (pad))) {
1807 const GstStructure *s;
1810 GST_DEBUG_OBJECT (session->bin, "pad has caps %" GST_PTR_FORMAT, caps);
1812 s = gst_caps_get_structure (caps, 0);
1814 if (!gst_structure_get_int (s, "clock-rate", &stream->clock_rate)) {
1815 stream->clock_rate = -1;
1817 GST_WARNING_OBJECT (session->bin,
1818 "Caps have no clock rate %s from pad %s:%s",
1819 gst_caps_to_string (caps), GST_DEBUG_PAD_NAME (pad));
1822 if (gst_structure_get_uint (s, "clock-base", &val))
1823 stream->clock_base = val;
1825 stream->clock_base = -1;
1827 gst_caps_unref (caps);
1830 /* get pad and link */
1831 GST_DEBUG_OBJECT (session->bin, "linking jitterbuffer");
1832 padname = g_strdup_printf ("src_%d", ssrc);
1833 srcpad = gst_element_get_static_pad (element, padname);
1835 sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
1836 gst_pad_link (srcpad, sinkpad);
1837 gst_object_unref (sinkpad);
1838 gst_object_unref (srcpad);
1840 /* get the RTCP sync pad */
1841 GST_DEBUG_OBJECT (session->bin, "linking sync pad");
1842 padname = g_strdup_printf ("rtcp_src_%d", ssrc);
1843 srcpad = gst_element_get_static_pad (element, padname);
1845 gst_pad_link (srcpad, stream->sync_pad);
1846 gst_object_unref (srcpad);
1848 /* connect to the new-pad signal of the payload demuxer, this will expose the
1849 * new pad by ghosting it. */
1850 stream->demux_newpad_sig = g_signal_connect (stream->demux,
1851 "new-payload-type", (GCallback) new_payload_found, stream);
1852 /* connect to the request-pt-map signal. This signal will be emited by the
1853 * demuxer so that it can apply a proper caps on the buffers for the
1855 stream->demux_ptreq_sig = g_signal_connect (stream->demux,
1856 "request-pt-map", (GCallback) pt_map_requested, session);
1857 /* connect to the payload-type-change signal so that we can know which
1858 * PT is the current PT so that the jitterbuffer can be matched to the right
1860 stream->demux_pt_change_sig = g_signal_connect (stream->demux,
1861 "payload-type-change", (GCallback) payload_type_change, stream);
1863 GST_RTP_SESSION_UNLOCK (session);
1870 GST_RTP_SESSION_UNLOCK (session);
1871 GST_DEBUG_OBJECT (session->bin, "could not create stream");
1876 /* Create a pad for receiving RTP for the session in @name. Must be called with
1880 create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
1882 GstPad *result, *sinkdpad;
1884 GstRtpBinSession *session;
1885 GstPadLinkReturn lres;
1887 /* first get the session number */
1888 if (name == NULL || sscanf (name, "recv_rtp_sink_%d", &sessid) != 1)
1891 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
1893 /* get or create session */
1894 session = find_session_by_id (rtpbin, sessid);
1896 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
1897 /* create session now */
1898 session = create_session (rtpbin, sessid);
1899 if (session == NULL)
1903 /* check if pad was requested */
1904 if (session->recv_rtp_sink != NULL)
1907 GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
1908 /* get recv_rtp pad and store */
1909 session->recv_rtp_sink =
1910 gst_element_get_request_pad (session->session, "recv_rtp_sink");
1911 if (session->recv_rtp_sink == NULL)
1914 g_signal_connect (session->recv_rtp_sink, "notify::caps",
1915 (GCallback) caps_changed, session);
1917 GST_DEBUG_OBJECT (rtpbin, "getting RTP src pad");
1918 /* get srcpad, link to SSRCDemux */
1919 session->recv_rtp_src =
1920 gst_element_get_static_pad (session->session, "recv_rtp_src");
1921 if (session->recv_rtp_src == NULL)
1924 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
1925 sinkdpad = gst_element_get_static_pad (session->demux, "sink");
1926 GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
1927 lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
1928 gst_object_unref (sinkdpad);
1929 if (lres != GST_PAD_LINK_OK)
1932 /* connect to the new-ssrc-pad signal of the SSRC demuxer */
1933 session->demux_newpad_sig = g_signal_connect (session->demux,
1934 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
1936 GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
1938 gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
1939 gst_pad_set_active (result, TRUE);
1940 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
1947 g_warning ("gstrtpbin: invalid name given");
1952 /* create_session already warned */
1957 g_warning ("gstrtpbin: recv_rtp pad already requested for session %d",
1963 g_warning ("gstrtpbin: failed to get session pad");
1968 g_warning ("gstrtpbin: failed to link pads");
1973 /* Create a pad for receiving RTCP for the session in @name. Must be called with
1977 create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
1982 GstRtpBinSession *session;
1984 GstPadLinkReturn lres;
1986 /* first get the session number */
1987 if (name == NULL || sscanf (name, "recv_rtcp_sink_%d", &sessid) != 1)
1990 GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid);
1992 /* get or create the session */
1993 session = find_session_by_id (rtpbin, sessid);
1995 GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid);
1996 /* create session now */
1997 session = create_session (rtpbin, sessid);
1998 if (session == NULL)
2002 /* check if pad was requested */
2003 if (session->recv_rtcp_sink != NULL)
2006 /* get recv_rtp pad and store */
2007 GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
2008 session->recv_rtcp_sink =
2009 gst_element_get_request_pad (session->session, "recv_rtcp_sink");
2010 if (session->recv_rtcp_sink == NULL)
2013 /* get srcpad, link to SSRCDemux */
2014 GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
2015 session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
2016 if (session->sync_src == NULL)
2019 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
2020 sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
2021 lres = gst_pad_link (session->sync_src, sinkdpad);
2022 gst_object_unref (sinkdpad);
2023 if (lres != GST_PAD_LINK_OK)
2027 gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
2028 gst_pad_set_active (result, TRUE);
2029 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
2036 g_warning ("gstrtpbin: invalid name given");
2041 /* create_session already warned */
2046 g_warning ("gstrtpbin: recv_rtcp pad already requested for session %d",
2052 g_warning ("gstrtpbin: failed to get session pad");
2057 g_warning ("gstrtpbin: failed to link pads");
2062 /* Create a pad for sending RTP for the session in @name. Must be called with
2066 create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2068 GstPad *result, *srcghost;
2071 GstRtpBinSession *session;
2072 GstElementClass *klass;
2074 /* first get the session number */
2075 if (name == NULL || sscanf (name, "send_rtp_sink_%d", &sessid) != 1)
2078 /* get or create session */
2079 session = find_session_by_id (rtpbin, sessid);
2081 /* create session now */
2082 session = create_session (rtpbin, sessid);
2083 if (session == NULL)
2087 /* check if pad was requested */
2088 if (session->send_rtp_sink != NULL)
2091 /* get send_rtp pad and store */
2092 session->send_rtp_sink =
2093 gst_element_get_request_pad (session->session, "send_rtp_sink");
2094 if (session->send_rtp_sink == NULL)
2098 gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
2099 gst_pad_set_active (result, TRUE);
2100 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
2103 session->send_rtp_src =
2104 gst_element_get_static_pad (session->session, "send_rtp_src");
2105 if (session->send_rtp_src == NULL)
2108 /* ghost the new source pad */
2109 klass = GST_ELEMENT_GET_CLASS (rtpbin);
2110 gname = g_strdup_printf ("send_rtp_src_%d", sessid);
2111 templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%d");
2113 gst_ghost_pad_new_from_template (gname, session->send_rtp_src, templ);
2114 gst_pad_set_active (srcghost, TRUE);
2115 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), srcghost);
2123 g_warning ("gstrtpbin: invalid name given");
2128 /* create_session already warned */
2133 g_warning ("gstrtpbin: send_rtp pad already requested for session %d",
2139 g_warning ("gstrtpbin: failed to get session pad for session %d", sessid);
2144 g_warning ("gstrtpbin: failed to get rtp source pad for session %d",
2150 /* Create a pad for sending RTCP for the session in @name. Must be called with
2154 create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
2158 GstRtpBinSession *session;
2160 /* first get the session number */
2161 if (name == NULL || sscanf (name, "send_rtcp_src_%d", &sessid) != 1)
2164 /* get or create session */
2165 session = find_session_by_id (rtpbin, sessid);
2169 /* check if pad was requested */
2170 if (session->send_rtcp_src != NULL)
2173 /* get rtcp_src pad and store */
2174 session->send_rtcp_src =
2175 gst_element_get_request_pad (session->session, "send_rtcp_src");
2176 if (session->send_rtcp_src == NULL)
2180 gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
2181 gst_pad_set_active (result, TRUE);
2182 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
2189 g_warning ("gstrtpbin: invalid name given");
2194 g_warning ("gstrtpbin: session with id %d does not exist", sessid);
2199 g_warning ("gstrtpbin: send_rtcp_src pad already requested for session %d",
2205 g_warning ("gstrtpbin: failed to get rtcp pad for session %d", sessid);
2210 /* If the requested name is NULL we should create a name with
2211 * the session number assuming we want the lowest posible session
2212 * with a free pad like the template */
2214 gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
2216 gboolean name_found = FALSE;
2219 GstIterator *pad_it = NULL;
2220 gchar *pad_name = NULL;
2222 GST_DEBUG_OBJECT (element, "find a free pad name for template");
2223 while (!name_found) {
2225 pad_name = g_strdup_printf (templ->name_template, session++);
2226 pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
2228 while (gst_iterator_next (pad_it, (gpointer) & pad) == GST_ITERATOR_OK) {
2231 name = gst_pad_get_name (pad);
2232 if (strcmp (name, pad_name) == 0)
2236 gst_iterator_free (pad_it);
2239 GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
2246 gst_rtp_bin_request_new_pad (GstElement * element,
2247 GstPadTemplate * templ, const gchar * name)
2250 GstElementClass *klass;
2252 gchar *pad_name = NULL;
2254 g_return_val_if_fail (templ != NULL, NULL);
2255 g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
2257 rtpbin = GST_RTP_BIN (element);
2258 klass = GST_ELEMENT_GET_CLASS (element);
2260 GST_RTP_BIN_LOCK (rtpbin);
2263 /* use a free pad name */
2264 pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
2266 /* use the provided name */
2267 pad_name = g_strdup (name);
2270 GST_DEBUG ("Trying to request a pad with name %s", pad_name);
2272 /* figure out the template */
2273 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%d")) {
2274 result = create_recv_rtp (rtpbin, templ, pad_name);
2275 } else if (templ == gst_element_class_get_pad_template (klass,
2276 "recv_rtcp_sink_%d")) {
2277 result = create_recv_rtcp (rtpbin, templ, pad_name);
2278 } else if (templ == gst_element_class_get_pad_template (klass,
2279 "send_rtp_sink_%d")) {
2280 result = create_send_rtp (rtpbin, templ, pad_name);
2281 } else if (templ == gst_element_class_get_pad_template (klass,
2282 "send_rtcp_src_%d")) {
2283 result = create_rtcp (rtpbin, templ, pad_name);
2285 goto wrong_template;
2288 GST_RTP_BIN_UNLOCK (rtpbin);
2296 GST_RTP_BIN_UNLOCK (rtpbin);
2297 g_warning ("gstrtpbin: this is not our template");
2303 gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)