2 * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
17 * Boston, MA 02110-1301, USA.
21 * SECTION:element-rtpbin
22 * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
24 * RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
25 * #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
26 * RTP sessions that will be synchronized together using RTCP SR packets.
28 * #GstRtpBin is configured with a number of request pads that define the
29 * functionality that is activated, similar to the #GstRtpSession element.
31 * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%u pad. The session
32 * number must be specified in the pad name.
33 * Data received on the recv_rtp_sink_\%u pad will be processed in the #GstRtpSession
34 * manager and after being validated forwarded on #GstRtpSsrcDemux element. Each
35 * RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After
36 * the packets are released from the jitterbuffer, they will be forwarded to a
37 * #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
38 * on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
39 * rtpbin with the session number, SSRC and payload type respectively as the pad
42 * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The
43 * session number must be specified in the pad name.
45 * If you want the session manager to generate and send RTCP packets, request
46 * the send_rtcp_src_\%u pad with the session number in the pad name. Packet pushed
47 * on this pad contain SR/RR RTCP reports that should be sent to all participants
50 * To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will
51 * automatically create a send_rtp_src_\%u pad. If the session number is not provided,
52 * the pad from the lowest available session will be returned. The session manager will modify the
53 * SSRC in the RTP packets to its own SSRC and wil forward the packets on the
54 * send_rtp_src_\%u pad after updating its internal state.
56 * #GstRtpBin can also demultiplex incoming bundled streams. The first
57 * #GstRtpSession will have a #GstRtpSsrcDemux element splitting the streams
58 * based on their SSRC and potentially dispatched to a different #GstRtpSession.
59 * Because retransmission SSRCs need to be merged with the corresponding media
60 * stream the #GstRtpBin::on-bundled-ssrc signal is emitted so that the
61 * application can find out to which session the SSRC belongs.
63 * The session manager needs the clock-rate of the payload types it is handling
64 * and will signal the #GstRtpSession::request-pt-map signal when it needs such a
65 * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
68 * Access to the internal statistics of rtpbin is provided with the
69 * get-internal-session property. This action signal gives access to the
70 * RTPSession object which further provides action signals to retrieve the
71 * internal source and other sources.
73 * #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder,
74 * #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and
75 * #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders
76 * and decoders in order to support SRTP. The encoders must provide the pads
77 * rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for
78 * RTCP. The session number will be used in the pad name. The decoders must provide
79 * rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will
80 * be placed before the #GstRtpSession element, thus they must support SSRC demuxing
83 * #GstRtpBin has signals (#GstRtpBin::request-aux-sender and
84 * #GstRtpBin::request-aux-receiver to dynamically request an element that can be
85 * used to create or merge additional RTP streams. AUX elements are needed to
86 * implement FEC or retransmission (such as RFC 4588). An AUX sender must have one
87 * sink_\%u pad that matches the sessionid in the signal and it should have 1 or
88 * more src_\%u pads. For each src_%\u pad, a session will be made (if needed)
89 * and the pad will be linked to the session send_rtp_sink pad. Each session will
90 * then expose its source pad as send_rtp_src_\%u on #GstRtpBin.
91 * An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
92 * and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
93 * when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
96 * <title>Example pipelines</title>
98 * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
99 * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
100 * ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
102 * gst-launch-1.0 rtpbin name=rtpbin \
103 * v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
104 * rtpbin.send_rtp_src_0 ! udpsink port=5000 \
105 * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
106 * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
107 * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
108 * rtpbin.send_rtp_src_1 ! udpsink port=5002 \
109 * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
110 * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
111 * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
112 * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
113 * and the audio is sent to session 1. Video packets are sent on UDP port 5000
114 * and audio packets on port 5002. The video RTCP packets for session 0 are sent
115 * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
116 * RTCP packets for session 0 are received on port 5005 and RTCP for session 1
117 * is received on port 5007. Since RTCP packets from the sender should be sent
118 * as soon as possible and do not participate in preroll, sync=false and
119 * async=false is configured on udpsink
121 * gst-launch-1.0 -v rtpbin name=rtpbin \
122 * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
123 * port=5000 ! rtpbin.recv_rtp_sink_0 \
124 * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
125 * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
126 * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
127 * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
128 * port=5002 ! rtpbin.recv_rtp_sink_1 \
129 * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
130 * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
131 * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
132 * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
133 * decode and display the video.
134 * Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
135 * decode and play the audio.
136 * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
137 * session 1 on port 5003. These packets will be used for session management and
139 * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
150 #include <gst/rtp/gstrtpbuffer.h>
151 #include <gst/rtp/gstrtcpbuffer.h>
153 #include "gstrtpbin.h"
154 #include "rtpsession.h"
155 #include "gstrtpsession.h"
156 #include "gstrtpjitterbuffer.h"
158 #include <gst/glib-compat-private.h>
160 GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
161 #define GST_CAT_DEFAULT gst_rtp_bin_debug
164 static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
165 GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
168 GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
171 static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template =
172 GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
175 GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
178 static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
179 GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
182 GST_STATIC_CAPS ("application/x-rtp")
186 static GstStaticPadTemplate rtpbin_recv_rtp_src_template =
187 GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
190 GST_STATIC_CAPS ("application/x-rtp")
193 static GstStaticPadTemplate rtpbin_send_rtcp_src_template =
194 GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
197 GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
200 static GstStaticPadTemplate rtpbin_send_rtp_src_template =
201 GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
204 GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
207 #define GST_RTP_BIN_GET_PRIVATE(obj) \
208 (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
210 #define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
211 #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
213 /* lock to protect dynamic callbacks, like pad-added and new ssrc. */
214 #define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock)
215 #define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock)
217 /* lock for shutdown */
218 #define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
220 if (g_atomic_int_get (&bin->priv->shutdown)) \
222 GST_RTP_BIN_DYN_LOCK (bin); \
223 if (g_atomic_int_get (&bin->priv->shutdown)) { \
224 GST_RTP_BIN_DYN_UNLOCK (bin); \
229 /* unlock for shutdown */
230 #define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
231 GST_RTP_BIN_DYN_UNLOCK (bin); \
233 /* Minimum time offset to apply. This compensates for rounding errors in NTP to
234 * RTP timestamp conversions */
235 #define MIN_TS_OFFSET (4 * GST_MSECOND)
237 struct _GstRtpBinPrivate
241 /* lock protecting dynamic adding/removing */
244 /* if we are shutting down or not */
249 /* NTP time in ns of last SR sync used */
250 guint64 last_ntpnstime;
252 /* list of extra elements */
256 /* signals and args */
259 SIGNAL_REQUEST_PT_MAP,
260 SIGNAL_PAYLOAD_TYPE_CHANGE,
264 SIGNAL_GET_INTERNAL_SESSION,
266 SIGNAL_GET_INTERNAL_STORAGE,
269 SIGNAL_ON_SSRC_COLLISION,
270 SIGNAL_ON_SSRC_VALIDATED,
271 SIGNAL_ON_SSRC_ACTIVE,
274 SIGNAL_ON_BYE_TIMEOUT,
276 SIGNAL_ON_SENDER_TIMEOUT,
279 SIGNAL_REQUEST_RTP_ENCODER,
280 SIGNAL_REQUEST_RTP_DECODER,
281 SIGNAL_REQUEST_RTCP_ENCODER,
282 SIGNAL_REQUEST_RTCP_DECODER,
284 SIGNAL_REQUEST_FEC_DECODER,
285 SIGNAL_REQUEST_FEC_ENCODER,
287 SIGNAL_NEW_JITTERBUFFER,
290 SIGNAL_REQUEST_AUX_SENDER,
291 SIGNAL_REQUEST_AUX_RECEIVER,
293 SIGNAL_ON_NEW_SENDER_SSRC,
294 SIGNAL_ON_SENDER_SSRC_ACTIVE,
296 SIGNAL_ON_BUNDLED_SSRC,
301 #define DEFAULT_LATENCY_MS 200
302 #define DEFAULT_DROP_ON_LATENCY FALSE
303 #define DEFAULT_SDES NULL
304 #define DEFAULT_DO_LOST FALSE
305 #define DEFAULT_IGNORE_PT FALSE
306 #define DEFAULT_NTP_SYNC FALSE
307 #define DEFAULT_AUTOREMOVE FALSE
308 #define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
309 #define DEFAULT_USE_PIPELINE_CLOCK FALSE
310 #define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
311 #define DEFAULT_RTCP_SYNC_INTERVAL 0
312 #define DEFAULT_DO_SYNC_EVENT FALSE
313 #define DEFAULT_DO_RETRANSMISSION FALSE
314 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
315 #define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
316 #define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
317 #define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
318 #define DEFAULT_MAX_DROPOUT_TIME 60000
319 #define DEFAULT_MAX_MISORDER_TIME 2000
320 #define DEFAULT_RFC7273_SYNC FALSE
321 #define DEFAULT_MAX_STREAMS G_MAXUINT
322 #define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
323 #define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
329 PROP_DROP_ON_LATENCY,
335 PROP_RTCP_SYNC_INTERVAL,
338 PROP_USE_PIPELINE_CLOCK,
340 PROP_DO_RETRANSMISSION,
342 PROP_NTP_TIME_SOURCE,
343 PROP_RTCP_SYNC_SEND_TIME,
344 PROP_MAX_RTCP_RTP_TIME_DIFF,
345 PROP_MAX_DROPOUT_TIME,
346 PROP_MAX_MISORDER_TIME,
349 PROP_MAX_TS_OFFSET_ADJUSTMENT,
353 #define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
355 gst_rtp_bin_rtcp_sync_get_type (void)
357 static GType rtcp_sync_type = 0;
358 static const GEnumValue rtcp_sync_types[] = {
359 {GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
360 {GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
361 {GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
365 if (!rtcp_sync_type) {
366 rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
368 return rtcp_sync_type;
372 typedef struct _GstRtpBinSession GstRtpBinSession;
373 typedef struct _GstRtpBinStream GstRtpBinStream;
374 typedef struct _GstRtpBinClient GstRtpBinClient;
376 static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
378 static GstCaps *pt_map_requested (GstElement * element, guint pt,
379 GstRtpBinSession * session);
380 static void payload_type_change (GstElement * element, guint pt,
381 GstRtpBinSession * session);
382 static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
383 static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
384 static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
385 static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
386 static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
387 static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
388 static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
389 static GstPad *complete_session_sink (GstRtpBin * rtpbin,
390 GstRtpBinSession * session, gboolean bundle_demuxer_needed);
392 complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
394 static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
395 GstRtpBinSession * session, guint sessid, gboolean bundle_demuxer_needed);
397 /* Manages the RTP stream for one SSRC.
399 * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
400 * If we see an SDES RTCP packet that links multiple SSRCs together based on a
401 * common CNAME, we create a GstRtpBinClient structure to group the SSRCs
402 * together (see below).
404 struct _GstRtpBinStream
406 /* the SSRC of this stream */
412 /* the session this SSRC belongs to */
413 GstRtpBinSession *session;
415 /* the jitterbuffer of the SSRC */
417 gulong buffer_handlesync_sig;
418 gulong buffer_ptreq_sig;
419 gulong buffer_ntpstop_sig;
422 /* the PT demuxer of the SSRC */
424 gulong demux_newpad_sig;
425 gulong demux_padremoved_sig;
426 gulong demux_ptreq_sig;
427 gulong demux_ptchange_sig;
429 /* if we have calculated a valid rt_delta for this stream */
431 /* mapping to local RTP and NTP time */
434 /* base rtptime in gst time */
438 #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
439 #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
441 /* Manages the receiving end of the packets.
443 * There is one such structure for each RTP session (audio/video/...).
444 * We get the RTP/RTCP packets and stuff them into the session manager. From
445 * there they are pushed into an SSRC demuxer that splits the stream based on
446 * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
447 * the GstRtpBinStream above).
449 * Before the SSRC demuxer, a storage element may be inserted for the purpose
450 * of Forward Error Correction.
452 struct _GstRtpBinSession
458 /* the session element */
460 /* the SSRC demuxer */
462 gulong demux_newpad_sig;
463 gulong demux_padremoved_sig;
468 /* Bundling support */
469 GstElement *rtp_funnel;
470 GstElement *rtcp_funnel;
471 GstElement *bundle_demux;
472 gulong bundle_demux_newpad_sig;
476 /* list of GstRtpBinStream */
479 /* list of elements */
482 /* mapping of payload type to caps */
485 /* the pads of the session */
486 GstPad *recv_rtp_sink;
487 GstPad *recv_rtp_sink_ghost;
488 GstPad *recv_rtp_src;
489 GstPad *recv_rtcp_sink;
490 GstPad *recv_rtcp_sink_ghost;
492 GstPad *send_rtp_sink;
493 GstPad *send_rtp_sink_ghost;
494 GstPad *send_rtp_src_ghost;
495 GstPad *send_rtcp_src;
496 GstPad *send_rtcp_src_ghost;
499 /* Manages the RTP streams that come from one client and should therefore be
502 struct _GstRtpBinClient
504 /* the common CNAME for the streams */
513 /* find a session with the given id. Must be called with RTP_BIN_LOCK */
514 static GstRtpBinSession *
515 find_session_by_id (GstRtpBin * rtpbin, gint id)
519 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
520 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
528 /* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
529 static GstRtpBinSession *
530 find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
534 for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
535 GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
537 if ((sess->recv_rtp_sink_ghost == pad) ||
538 (sess->recv_rtcp_sink_ghost == pad) ||
539 (sess->send_rtp_sink_ghost == pad)
540 || (sess->send_rtcp_src_ghost == pad))
547 on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
549 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
554 on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
556 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
561 on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
563 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
568 on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
570 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
575 on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
577 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
582 on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
584 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
589 on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
591 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
594 if (sess->bin->priv->autoremove)
595 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
599 on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
601 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
604 if (sess->bin->priv->autoremove)
605 g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
609 on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
611 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
616 on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
618 g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
619 stream->session->id, stream->ssrc);
623 on_new_sender_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
625 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
630 on_sender_ssrc_active (GstElement * session, guint32 ssrc,
631 GstRtpBinSession * sess)
633 g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE],
637 /* must be called with the SESSION lock */
638 static GstRtpBinStream *
639 find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
643 for (walk = session->streams; walk; walk = g_slist_next (walk)) {
644 GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
646 if (stream->ssrc == ssrc)
653 ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
654 GstRtpBinSession * session)
656 GstRtpBinStream *stream = NULL;
659 rtpbin = session->bin;
661 GST_RTP_BIN_LOCK (rtpbin);
663 GST_RTP_SESSION_LOCK (session);
664 if ((stream = find_stream_by_ssrc (session, ssrc)))
665 session->streams = g_slist_remove (session->streams, stream);
666 GST_RTP_SESSION_UNLOCK (session);
669 free_stream (stream, rtpbin);
671 GST_RTP_BIN_UNLOCK (rtpbin);
675 new_bundled_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
676 GstRtpBinSession * session)
678 GValue result = G_VALUE_INIT;
679 GValue params[2] = { G_VALUE_INIT, G_VALUE_INIT };
680 guint session_id = 0;
681 GstRtpBinSession *target_session = NULL;
682 GstRtpBin *rtpbin = session->bin;
685 GstPad *recv_rtp_sink = NULL;
686 GstPad *recv_rtcp_sink = NULL;
687 GstPadLinkReturn ret;
689 GST_RTP_BIN_DYN_LOCK (rtpbin);
690 GST_DEBUG_OBJECT (rtpbin, "new bundled SSRC pad %08x, %s:%s", ssrc,
691 GST_DEBUG_PAD_NAME (pad));
693 g_value_init (&result, G_TYPE_UINT);
694 g_value_init (¶ms[0], GST_TYPE_ELEMENT);
695 g_value_set_object (¶ms[0], rtpbin);
696 g_value_init (¶ms[1], G_TYPE_UINT);
697 g_value_set_uint (¶ms[1], ssrc);
699 g_signal_emitv (params,
700 gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, &result);
701 g_value_unset (¶ms[0]);
703 session_id = g_value_get_uint (&result);
704 if (session_id == 0) {
705 target_session = session;
707 target_session = find_session_by_id (rtpbin, (gint) session_id);
708 if (!target_session) {
709 target_session = create_session (rtpbin, session_id);
711 if (!target_session) {
712 /* create_session() warned already */
713 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
717 if (!target_session->recv_rtp_sink) {
718 recv_rtp_sink = complete_session_sink (rtpbin, target_session, FALSE);
721 if (!target_session->recv_rtp_src)
722 complete_session_receiver (rtpbin, target_session, session_id);
724 if (!target_session->recv_rtcp_sink) {
726 complete_session_rtcp (rtpbin, target_session, session_id, FALSE);
730 GST_DEBUG_OBJECT (rtpbin, "Assigning bundled ssrc %u to session %u", ssrc,
733 if (!recv_rtp_sink) {
735 gst_element_get_request_pad (target_session->rtp_funnel, "sink_%u");
738 if (!recv_rtcp_sink) {
740 gst_element_get_request_pad (target_session->rtcp_funnel, "sink_%u");
743 name = g_strdup_printf ("src_%u", ssrc);
744 src_pad = gst_element_get_static_pad (element, name);
745 ret = gst_pad_link (src_pad, recv_rtp_sink);
747 gst_object_unref (src_pad);
748 gst_object_unref (recv_rtp_sink);
749 if (ret != GST_PAD_LINK_OK) {
751 ("rtpbin: failed to link bundle demuxer to receive rtp funnel for session %u",
755 name = g_strdup_printf ("rtcp_src_%u", ssrc);
756 src_pad = gst_element_get_static_pad (element, name);
757 gst_pad_link (src_pad, recv_rtcp_sink);
759 gst_object_unref (src_pad);
760 gst_object_unref (recv_rtcp_sink);
761 if (ret != GST_PAD_LINK_OK) {
763 ("rtpbin: failed to link bundle demuxer to receive rtcp sink pad for session %u",
767 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
770 /* create a session with the given id. Must be called with RTP_BIN_LOCK */
771 static GstRtpBinSession *
772 create_session (GstRtpBin * rtpbin, gint id)
774 GstRtpBinSession *sess;
775 GstElement *session, *demux;
776 GstElement *storage = NULL;
779 if (!(session = gst_element_factory_make ("rtpsession", NULL)))
782 if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
785 if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
788 g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
791 sess = g_new0 (GstRtpBinSession, 1);
792 g_mutex_init (&sess->lock);
795 sess->session = session;
797 sess->storage = storage;
799 sess->rtp_funnel = gst_element_factory_make ("funnel", NULL);
800 sess->rtcp_funnel = gst_element_factory_make ("funnel", NULL);
802 sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
803 (GDestroyNotify) gst_caps_unref);
804 rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
806 /* configure SDES items */
807 GST_OBJECT_LOCK (rtpbin);
808 g_object_set (session, "sdes", rtpbin->sdes, "rtp-profile",
809 rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
811 if (rtpbin->use_pipeline_clock)
812 g_object_set (session, "use-pipeline-clock", rtpbin->use_pipeline_clock,
815 g_object_set (session, "ntp-time-source", rtpbin->ntp_time_source, NULL);
817 g_object_set (session, "max-dropout-time", rtpbin->max_dropout_time,
818 "max-misorder-time", rtpbin->max_misorder_time, NULL);
819 GST_OBJECT_UNLOCK (rtpbin);
821 /* provide clock_rate to the session manager when needed */
822 g_signal_connect (session, "request-pt-map",
823 (GCallback) pt_map_requested, sess);
825 g_signal_connect (sess->session, "on-new-ssrc",
826 (GCallback) on_new_ssrc, sess);
827 g_signal_connect (sess->session, "on-ssrc-collision",
828 (GCallback) on_ssrc_collision, sess);
829 g_signal_connect (sess->session, "on-ssrc-validated",
830 (GCallback) on_ssrc_validated, sess);
831 g_signal_connect (sess->session, "on-ssrc-active",
832 (GCallback) on_ssrc_active, sess);
833 g_signal_connect (sess->session, "on-ssrc-sdes",
834 (GCallback) on_ssrc_sdes, sess);
835 g_signal_connect (sess->session, "on-bye-ssrc",
836 (GCallback) on_bye_ssrc, sess);
837 g_signal_connect (sess->session, "on-bye-timeout",
838 (GCallback) on_bye_timeout, sess);
839 g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
840 g_signal_connect (sess->session, "on-sender-timeout",
841 (GCallback) on_sender_timeout, sess);
842 g_signal_connect (sess->session, "on-new-sender-ssrc",
843 (GCallback) on_new_sender_ssrc, sess);
844 g_signal_connect (sess->session, "on-sender-ssrc-active",
845 (GCallback) on_sender_ssrc_active, sess);
847 gst_bin_add (GST_BIN_CAST (rtpbin), session);
848 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
849 gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtp_funnel);
850 gst_bin_add (GST_BIN_CAST (rtpbin), sess->rtcp_funnel);
851 gst_bin_add (GST_BIN_CAST (rtpbin), storage);
853 GST_OBJECT_LOCK (rtpbin);
854 target = GST_STATE_TARGET (rtpbin);
855 GST_OBJECT_UNLOCK (rtpbin);
857 /* change state only to what's needed */
858 gst_element_set_state (demux, target);
859 gst_element_set_state (session, target);
860 gst_element_set_state (sess->rtp_funnel, target);
861 gst_element_set_state (sess->rtcp_funnel, target);
862 gst_element_set_state (storage, target);
869 g_warning ("rtpbin: could not create rtpsession element");
874 gst_object_unref (session);
875 g_warning ("rtpbin: could not create rtpssrcdemux element");
880 gst_object_unref (session);
881 gst_object_unref (demux);
882 g_warning ("rtpbin: could not create rtpstorage element");
888 bin_manage_element (GstRtpBin * bin, GstElement * element)
890 GstRtpBinPrivate *priv = bin->priv;
892 if (g_list_find (priv->elements, element)) {
893 GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
895 GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
897 if (g_object_is_floating (element))
898 element = gst_object_ref_sink (element);
900 if (!gst_bin_add (GST_BIN_CAST (bin), element))
902 if (!gst_element_sync_state_with_parent (element))
903 GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin");
905 /* we add the element multiple times, each we need an equal number of
906 * removes to really remove the element from the bin */
907 priv->elements = g_list_prepend (priv->elements, element);
914 GST_WARNING_OBJECT (bin, "unable to add element");
915 gst_object_unref (element);
921 remove_bin_element (GstElement * element, GstRtpBin * bin)
923 GstRtpBinPrivate *priv = bin->priv;
926 find = g_list_find (priv->elements, element);
928 priv->elements = g_list_delete_link (priv->elements, find);
930 if (!g_list_find (priv->elements, element)) {
931 gst_element_set_locked_state (element, TRUE);
932 gst_bin_remove (GST_BIN_CAST (bin), element);
933 gst_element_set_state (element, GST_STATE_NULL);
936 gst_object_unref (element);
940 /* called with RTP_BIN_LOCK */
942 free_session (GstRtpBinSession * sess, GstRtpBin * bin)
944 GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
946 gst_element_set_locked_state (sess->demux, TRUE);
947 gst_element_set_locked_state (sess->session, TRUE);
949 gst_element_set_state (sess->demux, GST_STATE_NULL);
950 gst_element_set_state (sess->session, GST_STATE_NULL);
952 remove_recv_rtp (bin, sess);
953 remove_recv_rtcp (bin, sess);
954 remove_send_rtp (bin, sess);
955 remove_rtcp (bin, sess);
957 gst_bin_remove (GST_BIN_CAST (bin), sess->session);
958 gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
960 g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
961 g_slist_free (sess->elements);
963 g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
964 g_slist_free (sess->streams);
966 g_mutex_clear (&sess->lock);
967 g_hash_table_destroy (sess->ptmap);
972 /* get the payload type caps for the specific payload @pt in @session */
974 get_pt_map (GstRtpBinSession * session, guint pt)
976 GstCaps *caps = NULL;
979 GValue args[3] = { {0}, {0}, {0} };
981 GST_DEBUG ("searching pt %u in cache", pt);
983 GST_RTP_SESSION_LOCK (session);
985 /* first look in the cache */
986 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
994 GST_DEBUG ("emiting signal for pt %u in session %u", pt, session->id);
996 /* not in cache, send signal to request caps */
997 g_value_init (&args[0], GST_TYPE_ELEMENT);
998 g_value_set_object (&args[0], bin);
999 g_value_init (&args[1], G_TYPE_UINT);
1000 g_value_set_uint (&args[1], session->id);
1001 g_value_init (&args[2], G_TYPE_UINT);
1002 g_value_set_uint (&args[2], pt);
1004 g_value_init (&ret, GST_TYPE_CAPS);
1005 g_value_set_boxed (&ret, NULL);
1007 GST_RTP_SESSION_UNLOCK (session);
1009 g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
1011 GST_RTP_SESSION_LOCK (session);
1013 g_value_unset (&args[0]);
1014 g_value_unset (&args[1]);
1015 g_value_unset (&args[2]);
1017 /* look in the cache again because we let the lock go */
1018 caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
1020 gst_caps_ref (caps);
1021 g_value_unset (&ret);
1025 caps = (GstCaps *) g_value_dup_boxed (&ret);
1026 g_value_unset (&ret);
1030 GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
1032 /* store in cache, take additional ref */
1033 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
1034 gst_caps_ref (caps));
1037 GST_RTP_SESSION_UNLOCK (session);
1044 GST_RTP_SESSION_UNLOCK (session);
1045 GST_DEBUG ("no pt map could be obtained");
1051 return_true (gpointer key, gpointer value, gpointer user_data)
1057 gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
1059 GSList *clients, *streams;
1061 GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
1063 GST_RTP_BIN_LOCK (rtpbin);
1064 for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
1065 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
1067 /* reset sync on all streams for this client */
1068 for (streams = client->streams; streams; streams = g_slist_next (streams)) {
1069 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
1071 /* make use require a new SR packet for this stream before we attempt new
1073 stream->have_sync = FALSE;
1074 stream->rt_delta = 0;
1075 stream->rtp_delta = 0;
1076 stream->clock_base = -100 * GST_SECOND;
1079 GST_RTP_BIN_UNLOCK (rtpbin);
1083 gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
1085 GSList *sessions, *streams;
1087 GST_RTP_BIN_LOCK (bin);
1088 GST_DEBUG_OBJECT (bin, "clearing pt map");
1089 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
1090 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
1092 GST_DEBUG_OBJECT (bin, "clearing session %p", session);
1093 g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
1095 GST_RTP_SESSION_LOCK (session);
1096 g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
1098 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
1099 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
1101 GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
1102 g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
1104 g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
1106 GST_RTP_SESSION_UNLOCK (session);
1108 GST_RTP_BIN_UNLOCK (bin);
1110 /* reset sync too */
1111 gst_rtp_bin_reset_sync (bin);
1115 gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
1117 GstRtpBinSession *session;
1118 GstElement *ret = NULL;
1120 GST_RTP_BIN_LOCK (bin);
1121 GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
1122 session = find_session_by_id (bin, (gint) session_id);
1124 ret = gst_object_ref (session->session);
1126 GST_RTP_BIN_UNLOCK (bin);
1132 gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
1134 RTPSession *internal_session = NULL;
1135 GstRtpBinSession *session;
1137 GST_RTP_BIN_LOCK (bin);
1138 GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
1140 session = find_session_by_id (bin, (gint) session_id);
1142 g_object_get (session->session, "internal-session", &internal_session,
1145 GST_RTP_BIN_UNLOCK (bin);
1147 return internal_session;
1151 gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
1153 GstRtpBinSession *session;
1154 GstElement *res = NULL;
1156 GST_RTP_BIN_LOCK (bin);
1157 GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
1159 session = find_session_by_id (bin, (gint) session_id);
1160 if (session && session->storage) {
1161 res = gst_object_ref (session->storage);
1163 GST_RTP_BIN_UNLOCK (bin);
1169 gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
1171 GObject *internal_storage = NULL;
1172 GstRtpBinSession *session;
1174 GST_RTP_BIN_LOCK (bin);
1175 GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
1177 session = find_session_by_id (bin, (gint) session_id);
1178 if (session && session->storage) {
1179 g_object_get (session->storage, "internal-storage", &internal_storage,
1182 GST_RTP_BIN_UNLOCK (bin);
1184 return internal_storage;
1188 gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
1190 GST_DEBUG_OBJECT (bin, "return NULL encoder");
1195 gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
1197 GST_DEBUG_OBJECT (bin, "return NULL decoder");
1202 gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
1203 const gchar * name, const GValue * value)
1205 GSList *sessions, *streams;
1207 GST_RTP_BIN_LOCK (bin);
1208 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
1209 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
1211 GST_RTP_SESSION_LOCK (session);
1212 for (streams = session->streams; streams; streams = g_slist_next (streams)) {
1213 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
1215 g_object_set_property (G_OBJECT (stream->buffer), name, value);
1217 GST_RTP_SESSION_UNLOCK (session);
1219 GST_RTP_BIN_UNLOCK (bin);
1223 gst_rtp_bin_propagate_property_to_session (GstRtpBin * bin,
1224 const gchar * name, const GValue * value)
1228 GST_RTP_BIN_LOCK (bin);
1229 for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
1230 GstRtpBinSession *sess = (GstRtpBinSession *) sessions->data;
1232 g_object_set_property (G_OBJECT (sess->session), name, value);
1234 GST_RTP_BIN_UNLOCK (bin);
1237 /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
1238 static GstRtpBinClient *
1239 get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created)
1241 GstRtpBinClient *result = NULL;
1244 for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
1245 GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
1247 if (len != client->cname_len)
1250 if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
1251 GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
1258 /* nothing found, create one */
1259 if (result == NULL) {
1260 result = g_new0 (GstRtpBinClient, 1);
1261 result->cname = g_strndup ((gchar *) data, len);
1262 result->cname_len = len;
1263 bin->clients = g_slist_prepend (bin->clients, result);
1264 GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
1271 free_client (GstRtpBinClient * client, GstRtpBin * bin)
1273 GST_DEBUG_OBJECT (bin, "freeing client %p", client);
1274 g_slist_free (client->streams);
1275 g_free (client->cname);
1280 get_current_times (GstRtpBin * bin, GstClockTime * running_time,
1281 guint64 * ntpnstime)
1285 GstClockTime base_time, rt, clock_time;
1287 GST_OBJECT_LOCK (bin);
1288 if ((clock = GST_ELEMENT_CLOCK (bin))) {
1289 base_time = GST_ELEMENT_CAST (bin)->base_time;
1290 gst_object_ref (clock);
1291 GST_OBJECT_UNLOCK (bin);
1293 /* get current clock time and convert to running time */
1294 clock_time = gst_clock_get_time (clock);
1295 rt = clock_time - base_time;
1297 if (bin->use_pipeline_clock) {
1299 /* add constant to convert from 1970 based time to 1900 based time */
1300 ntpns += (2208988800LL * GST_SECOND);
1302 switch (bin->ntp_time_source) {
1303 case GST_RTP_NTP_TIME_SOURCE_NTP:
1304 case GST_RTP_NTP_TIME_SOURCE_UNIX:{
1307 /* get current NTP time */
1308 g_get_current_time (¤t);
1309 ntpns = GST_TIMEVAL_TO_TIME (current);
1311 /* add constant to convert from 1970 based time to 1900 based time */
1312 if (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
1313 ntpns += (2208988800LL * GST_SECOND);
1316 case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
1319 case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
1323 ntpns = -1; /* Fix uninited compiler warning */
1324 g_assert_not_reached ();
1329 gst_object_unref (clock);
1331 GST_OBJECT_UNLOCK (bin);
1342 stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
1343 gint64 ts_offset, gint64 max_ts_offset, gint64 min_ts_offset,
1344 gboolean allow_positive_ts_offset)
1346 gint64 prev_ts_offset;
1348 g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
1350 /* delta changed, see how much */
1351 if (prev_ts_offset != ts_offset) {
1354 diff = prev_ts_offset - ts_offset;
1356 GST_DEBUG_OBJECT (bin,
1357 "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
1358 ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
1360 /* ignore minor offsets */
1361 if (ABS (diff) < min_ts_offset) {
1362 GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
1366 /* sanity check offset */
1367 if (max_ts_offset > 0) {
1368 if (ts_offset > 0 && !allow_positive_ts_offset) {
1369 GST_DEBUG_OBJECT (bin,
1370 "offset is positive (clocks are out of sync), ignoring");
1373 if (ABS (ts_offset) > max_ts_offset) {
1374 GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
1379 g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
1381 GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
1382 stream->ssrc, ts_offset);
1386 gst_rtp_bin_send_sync_event (GstRtpBinStream * stream)
1388 if (stream->bin->send_sync_event) {
1392 GST_DEBUG_OBJECT (stream->bin,
1393 "sending GstRTCPSRReceived event downstream");
1395 event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
1396 gst_structure_new_empty ("GstRTCPSRReceived"));
1398 srcpad = gst_element_get_static_pad (stream->buffer, "src");
1399 gst_pad_push_event (srcpad, event);
1400 gst_object_unref (srcpad);
1404 /* associate a stream to the given CNAME. This will make sure all streams for
1405 * that CNAME are synchronized together.
1406 * Must be called with GST_RTP_BIN_LOCK */
1408 gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
1409 guint8 * data, guint64 ntptime, guint64 last_extrtptime,
1410 guint64 base_rtptime, guint64 base_time, guint clock_rate,
1411 gint64 rtp_clock_base)
1413 GstRtpBinClient *client;
1416 GstClockTime running_time, running_time_rtp;
1419 /* first find or create the CNAME */
1420 client = get_client (bin, len, data, &created);
1422 /* find stream in the client */
1423 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1424 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1426 if (ostream == stream)
1429 /* not found, add it to the list */
1431 GST_DEBUG_OBJECT (bin,
1432 "new association of SSRC %08x with client %p with CNAME %s",
1433 stream->ssrc, client, client->cname);
1434 client->streams = g_slist_prepend (client->streams, stream);
1437 GST_DEBUG_OBJECT (bin,
1438 "found association of SSRC %08x with client %p with CNAME %s",
1439 stream->ssrc, client, client->cname);
1442 if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
1443 GST_DEBUG_OBJECT (bin, "invalidated sync data");
1444 if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1445 /* we don't need that data, so carry on,
1446 * but make some values look saner */
1447 last_extrtptime = base_rtptime;
1449 /* nothing we can do with this data in this case */
1450 GST_DEBUG_OBJECT (bin, "bailing out");
1455 /* Take the extended rtptime we found in the SR packet and map it to the
1456 * local rtptime. The local rtp time is used to construct timestamps on the
1457 * buffers so we will calculate what running_time corresponds to the RTP
1458 * timestamp in the SR packet. */
1459 running_time_rtp = last_extrtptime - base_rtptime;
1461 GST_DEBUG_OBJECT (bin,
1462 "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
1463 ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
1464 "clock-base %" G_GINT64_FORMAT, base_rtptime,
1465 last_extrtptime, running_time_rtp, clock_rate, rtp_clock_base);
1467 /* calculate local RTP time in gstreamer timestamp, we essentially perform the
1468 * same conversion that a jitterbuffer would use to convert an rtp timestamp
1469 * into a corresponding gstreamer timestamp. Note that the base_time also
1470 * contains the drift between sender and receiver. */
1472 gst_util_uint64_scale_int (running_time_rtp, GST_SECOND, clock_rate);
1473 running_time += base_time;
1475 /* convert ntptime to nanoseconds */
1476 ntpnstime = gst_util_uint64_scale (ntptime, GST_SECOND,
1477 (G_GINT64_CONSTANT (1) << 32));
1479 stream->have_sync = TRUE;
1481 GST_DEBUG_OBJECT (bin,
1482 "SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
1483 running_time, ntpnstime);
1485 /* recalc inter stream playout offset, but only if there is more than one
1486 * stream or we're doing NTP sync. */
1487 if (bin->ntp_sync) {
1488 gint64 ntpdiff, rtdiff;
1489 guint64 local_ntpnstime;
1490 GstClockTime local_running_time;
1492 /* For NTP sync we need to first get a snapshot of running_time and NTP
1493 * time. We know at what running_time we play a certain RTP time, we also
1494 * calculated when we would play the RTP time in the SR packet. Now we need
1495 * to know how the running_time and the NTP time relate to eachother. */
1496 get_current_times (bin, &local_running_time, &local_ntpnstime);
1498 /* see how far away the NTP time is. This is the difference between the
1499 * current NTP time and the NTP time in the last SR packet. */
1500 ntpdiff = local_ntpnstime - ntpnstime;
1501 /* see how far away the running_time is. This is the difference between the
1502 * current running_time and the running_time of the RTP timestamp in the
1503 * last SR packet. */
1504 rtdiff = local_running_time - running_time;
1506 GST_DEBUG_OBJECT (bin,
1507 "local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
1508 local_ntpnstime, ntpnstime);
1509 GST_DEBUG_OBJECT (bin,
1510 "local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
1511 G_GUINT64_FORMAT, local_running_time, running_time);
1512 GST_DEBUG_OBJECT (bin,
1513 "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
1516 /* combine to get the final diff to apply to the running_time */
1517 stream->rt_delta = rtdiff - ntpdiff;
1519 stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
1522 gint64 min, rtp_min, clock_base = stream->clock_base;
1523 gboolean all_sync, use_rtp;
1524 gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
1526 /* calculate delta between server and receiver. ntpnstime is created by
1527 * converting the ntptime in the last SR packet to a gstreamer timestamp. This
1528 * delta expresses the difference to our timeline and the server timeline. The
1529 * difference in itself doesn't mean much but we can combine the delta of
1530 * multiple streams to create a stream specific offset. */
1531 stream->rt_delta = ntpnstime - running_time;
1533 /* calculate the min of all deltas, ignoring streams that did not yet have a
1534 * valid rt_delta because we did not yet receive an SR packet for those
1536 * We calculate the mininum because we would like to only apply positive
1537 * offsets to streams, delaying their playback instead of trying to speed up
1538 * other streams (which might be imposible when we have to create negative
1540 * The stream that has the smallest diff is selected as the reference stream,
1541 * all other streams will have a positive offset to this difference. */
1543 /* some alternative setting allow ignoring RTCP as much as possible,
1544 * for servers generating bogus ntp timeline */
1545 min = rtp_min = G_MAXINT64;
1547 if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
1551 /* signed version for convienience */
1552 clock_base = base_rtptime;
1553 /* deal with possible wrap-around */
1554 ext_base = base_rtptime;
1555 rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
1556 /* sanity check; base rtp and provided clock_base should be close */
1557 if (rtp_clock_base >= clock_base) {
1558 if (rtp_clock_base - clock_base < 10 * clock_rate) {
1559 rtp_clock_base = base_time +
1560 gst_util_uint64_scale_int (rtp_clock_base - clock_base,
1561 GST_SECOND, clock_rate);
1566 if (clock_base - rtp_clock_base < 10 * clock_rate) {
1567 rtp_clock_base = base_time -
1568 gst_util_uint64_scale_int (clock_base - rtp_clock_base,
1569 GST_SECOND, clock_rate);
1574 /* warn and bail for clarity out if no sane values */
1576 GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
1579 /* store to track changes */
1580 clock_base = rtp_clock_base;
1581 /* generate a fake as before,
1582 * now equating rtptime obtained from RTP-Info,
1583 * where the large time represent the otherwise irrelevant npt/ntp time */
1584 stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base;
1586 clock_base = rtp_clock_base;
1590 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1591 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1593 if (!ostream->have_sync) {
1598 /* change in current stream's base from previously init'ed value
1599 * leads to reset of all stream's base */
1600 if (stream != ostream && stream->clock_base >= 0 &&
1601 (stream->clock_base != clock_base)) {
1602 GST_DEBUG_OBJECT (bin, "reset upon clock base change");
1603 ostream->clock_base = -100 * GST_SECOND;
1604 ostream->rtp_delta = 0;
1607 if (ostream->rt_delta < min)
1608 min = ostream->rt_delta;
1609 if (ostream->rtp_delta < rtp_min)
1610 rtp_min = ostream->rtp_delta;
1613 /* arrange to re-sync for each stream upon significant change,
1615 all_sync = all_sync && (stream->clock_base == clock_base);
1616 stream->clock_base = clock_base;
1618 /* may need init performed above later on, but nothing more to do now */
1619 if (client->nstreams <= 1)
1622 GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
1623 " all sync %d", client, min, all_sync);
1624 GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
1626 switch (rtcp_sync) {
1627 case GST_RTP_BIN_RTCP_SYNC_RTP:
1630 GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
1631 "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
1633 case GST_RTP_BIN_RTCP_SYNC_INITIAL:
1634 /* if all have been synced already, do not bother further */
1636 GST_DEBUG_OBJECT (bin, "all streams already synced; done");
1644 /* bail out if we adjusted recently enough */
1645 if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
1646 bin->rtcp_sync_interval * GST_MSECOND) {
1647 GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
1648 "previous sender info too recent "
1649 "(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
1652 bin->priv->last_ntpnstime = ntpnstime;
1654 /* calculate offsets for each stream */
1655 for (walk = client->streams; walk; walk = g_slist_next (walk)) {
1656 GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
1659 /* ignore streams for which we didn't receive an SR packet yet, we
1660 * can't synchronize them yet. We can however sync other streams just
1662 if (!ostream->have_sync)
1665 /* calculate offset to our reference stream, this should always give a
1666 * positive number. */
1668 ts_offset = ostream->rtp_delta - rtp_min;
1670 ts_offset = ostream->rt_delta - min;
1672 stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
1673 MIN_TS_OFFSET, TRUE);
1676 gst_rtp_bin_send_sync_event (stream);
1681 #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
1682 for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
1683 (b) = gst_rtcp_packet_move_to_next ((packet)))
1685 #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
1686 for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
1687 (b) = gst_rtcp_packet_sdes_next_item ((packet)))
1689 #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
1690 for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
1691 (b) = gst_rtcp_packet_sdes_next_entry ((packet)))
1694 gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
1695 GstRtpBinStream * stream)
1698 GstRTCPPacket packet;
1701 gboolean have_sr, have_sdes;
1703 guint64 base_rtptime;
1709 GstRTCPBuffer rtcp = { NULL, };
1713 GST_DEBUG_OBJECT (bin, "sync handler called");
1715 /* get the last relation between the rtp timestamps and the gstreamer
1716 * timestamps. We get this info directly from the jitterbuffer which
1717 * constructs gstreamer timestamps from rtp timestamps and so it know exactly
1718 * what the current situation is. */
1720 g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
1721 base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
1722 clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
1723 clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
1725 g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
1726 buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
1731 gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
1733 GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
1734 /* first packet must be SR or RR or else the validate would have failed */
1735 switch (gst_rtcp_packet_get_type (&packet)) {
1736 case GST_RTCP_TYPE_SR:
1737 /* only parse first. There is only supposed to be one SR in the packet
1738 * but we will deal with malformed packets gracefully */
1741 /* get NTP and RTP times */
1742 gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
1745 GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
1746 /* ignore SR that is not ours */
1747 if (ssrc != stream->ssrc)
1752 case GST_RTCP_TYPE_SDES:
1754 gboolean more_items, more_entries;
1756 /* only deal with first SDES, there is only supposed to be one SDES in
1757 * the RTCP packet but we deal with bad packets gracefully. Also bail
1758 * out if we have not seen an SR item yet. */
1759 if (have_sdes || !have_sr)
1762 GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
1763 /* skip items that are not about the SSRC of the sender */
1764 if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
1767 /* find the CNAME entry */
1768 GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
1769 GstRTCPSDESType type;
1773 gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data);
1775 if (type == GST_RTCP_SDES_CNAME) {
1776 GST_RTP_BIN_LOCK (bin);
1777 /* associate the stream to CNAME */
1778 gst_rtp_bin_associate (bin, stream, len, data,
1779 ntptime, extrtptime, base_rtptime, base_time, clock_rate,
1781 GST_RTP_BIN_UNLOCK (bin);
1789 /* we can ignore these packets */
1793 gst_rtcp_buffer_unmap (&rtcp);
1796 /* create a new stream with @ssrc in @session. Must be called with
1797 * RTP_SESSION_LOCK. */
1798 static GstRtpBinStream *
1799 create_stream (GstRtpBinSession * session, guint32 ssrc)
1801 GstElement *buffer, *demux = NULL;
1802 GstRtpBinStream *stream;
1806 rtpbin = session->bin;
1808 if (g_slist_length (session->streams) >= rtpbin->max_streams)
1811 if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
1812 goto no_jitterbuffer;
1814 if (!rtpbin->ignore_pt) {
1815 if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
1819 stream = g_new0 (GstRtpBinStream, 1);
1820 stream->ssrc = ssrc;
1821 stream->bin = rtpbin;
1822 stream->session = session;
1823 stream->buffer = buffer;
1824 stream->demux = demux;
1826 stream->have_sync = FALSE;
1827 stream->rt_delta = 0;
1828 stream->rtp_delta = 0;
1829 stream->percent = 100;
1830 stream->clock_base = -100 * GST_SECOND;
1831 session->streams = g_slist_prepend (session->streams, stream);
1833 /* provide clock_rate to the jitterbuffer when needed */
1834 stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
1835 (GCallback) pt_map_requested, session);
1836 stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
1837 (GCallback) on_npt_stop, stream);
1839 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
1840 g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
1842 /* configure latency and packet lost */
1843 g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
1844 g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
1845 g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
1846 g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
1847 g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
1848 g_object_set (buffer, "max-rtcp-rtp-time-diff",
1849 rtpbin->max_rtcp_rtp_time_diff, NULL);
1850 g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time,
1851 "max-misorder-time", rtpbin->max_misorder_time, NULL);
1852 g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
1853 g_object_set (buffer, "max-ts-offset-adjustment",
1854 rtpbin->max_ts_offset_adjustment, NULL);
1856 g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
1857 buffer, session->id, ssrc);
1859 if (!rtpbin->ignore_pt)
1860 gst_bin_add (GST_BIN_CAST (rtpbin), demux);
1861 gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
1865 gst_element_link_pads_full (buffer, "src", demux, "sink",
1866 GST_PAD_LINK_CHECK_NOTHING);
1868 if (rtpbin->buffering) {
1871 GST_INFO_OBJECT (rtpbin,
1872 "bin is buffering, set jitterbuffer as not active");
1873 g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out);
1877 GST_OBJECT_LOCK (rtpbin);
1878 target = GST_STATE_TARGET (rtpbin);
1879 GST_OBJECT_UNLOCK (rtpbin);
1881 /* from sink to source */
1883 gst_element_set_state (demux, target);
1885 gst_element_set_state (buffer, target);
1892 GST_WARNING_OBJECT (rtpbin, "stream exeeds maximum (%d)",
1893 rtpbin->max_streams);
1898 g_warning ("rtpbin: could not create rtpjitterbuffer element");
1903 gst_object_unref (buffer);
1904 g_warning ("rtpbin: could not create rtpptdemux element");
1909 /* called with RTP_BIN_LOCK */
1911 free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
1913 GSList *clients, *next_client;
1915 GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
1917 if (stream->demux) {
1918 g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
1919 g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
1920 g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
1922 g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
1923 g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
1924 g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
1927 gst_element_set_locked_state (stream->demux, TRUE);
1928 gst_element_set_locked_state (stream->buffer, TRUE);
1931 gst_element_set_state (stream->demux, GST_STATE_NULL);
1932 gst_element_set_state (stream->buffer, GST_STATE_NULL);
1934 /* now remove this signal, we need this while going to NULL because it to
1935 * do some cleanups */
1937 g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
1939 gst_bin_remove (GST_BIN_CAST (bin), stream->buffer);
1941 gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
1943 for (clients = bin->clients; clients; clients = next_client) {
1944 GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
1945 GSList *streams, *next_stream;
1947 next_client = g_slist_next (clients);
1949 for (streams = client->streams; streams; streams = next_stream) {
1950 GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
1952 next_stream = g_slist_next (streams);
1954 if (ostream == stream) {
1955 client->streams = g_slist_delete_link (client->streams, streams);
1956 /* If this was the last stream belonging to this client,
1957 * clean up the client. */
1958 if (--client->nstreams == 0) {
1959 bin->clients = g_slist_delete_link (bin->clients, clients);
1960 free_client (client, bin);
1969 /* GObject vmethods */
1970 static void gst_rtp_bin_dispose (GObject * object);
1971 static void gst_rtp_bin_finalize (GObject * object);
1972 static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
1973 const GValue * value, GParamSpec * pspec);
1974 static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
1975 GValue * value, GParamSpec * pspec);
1977 /* GstElement vmethods */
1978 static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
1979 GstStateChange transition);
1980 static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
1981 GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
1982 static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
1983 static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
1985 #define gst_rtp_bin_parent_class parent_class
1986 G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
1989 _gst_element_accumulator (GSignalInvocationHint * ihint,
1990 GValue * return_accu, const GValue * handler_return, gpointer dummy)
1992 GstElement *element;
1994 element = g_value_get_object (handler_return);
1995 GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
1997 if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
1998 g_value_set_object (return_accu, element);
2000 /* stop emission if we have an element */
2001 return (element == NULL);
2005 _gst_caps_accumulator (GSignalInvocationHint * ihint,
2006 GValue * return_accu, const GValue * handler_return, gpointer dummy)
2010 caps = g_value_get_boxed (handler_return);
2011 GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
2013 if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP))
2014 g_value_set_boxed (return_accu, caps);
2016 /* stop emission if we have a caps */
2017 return (caps == NULL);
2021 gst_rtp_bin_class_init (GstRtpBinClass * klass)
2023 GObjectClass *gobject_class;
2024 GstElementClass *gstelement_class;
2025 GstBinClass *gstbin_class;
2027 gobject_class = (GObjectClass *) klass;
2028 gstelement_class = (GstElementClass *) klass;
2029 gstbin_class = (GstBinClass *) klass;
2031 g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
2033 gobject_class->dispose = gst_rtp_bin_dispose;
2034 gobject_class->finalize = gst_rtp_bin_finalize;
2035 gobject_class->set_property = gst_rtp_bin_set_property;
2036 gobject_class->get_property = gst_rtp_bin_get_property;
2038 g_object_class_install_property (gobject_class, PROP_LATENCY,
2039 g_param_spec_uint ("latency", "Buffer latency in ms",
2040 "Default amount of ms to buffer in the jitterbuffers", 0,
2041 G_MAXUINT, DEFAULT_LATENCY_MS,
2042 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2044 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
2045 g_param_spec_boolean ("drop-on-latency",
2046 "Drop buffers when maximum latency is reached",
2047 "Tells the jitterbuffer to never exceed the given latency in size",
2048 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2051 * GstRtpBin::request-pt-map:
2052 * @rtpbin: the object which received the signal
2053 * @session: the session
2056 * Request the payload type as #GstCaps for @pt in @session.
2058 gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
2059 g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
2060 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
2061 _gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS,
2062 2, G_TYPE_UINT, G_TYPE_UINT);
2065 * GstRtpBin::payload-type-change:
2066 * @rtpbin: the object which received the signal
2067 * @session: the session
2070 * Signal that the current payload type changed to @pt in @session.
2072 gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
2073 g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
2074 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
2075 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2079 * GstRtpBin::clear-pt-map:
2080 * @rtpbin: the object which received the signal
2082 * Clear all previously cached pt-mapping obtained with
2083 * #GstRtpBin::request-pt-map.
2085 gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] =
2086 g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
2087 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2088 clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
2092 * GstRtpBin::reset-sync:
2093 * @rtpbin: the object which received the signal
2095 * Reset all currently configured lip-sync parameters and require new SR
2096 * packets for all streams before lip-sync is attempted again.
2098 gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
2099 g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
2100 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2101 reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
2105 * GstRtpBin::get-session:
2106 * @rtpbin: the object which received the signal
2107 * @id: the session id
2109 * Request the related GstRtpSession as #GstElement related with session @id.
2113 gst_rtp_bin_signals[SIGNAL_GET_SESSION] =
2114 g_signal_new ("get-session", G_TYPE_FROM_CLASS (klass),
2115 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2116 get_session), NULL, NULL, g_cclosure_marshal_generic,
2117 GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2120 * GstRtpBin::get-internal-session:
2121 * @rtpbin: the object which received the signal
2122 * @id: the session id
2124 * Request the internal RTPSession object as #GObject in session @id.
2126 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
2127 g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
2128 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2129 get_internal_session), NULL, NULL, g_cclosure_marshal_generic,
2130 RTP_TYPE_SESSION, 1, G_TYPE_UINT);
2133 * GstRtpBin::get-internal-storage:
2134 * @rtpbin: the object which received the signal
2135 * @id: the session id
2137 * Request the internal RTPStorage object as #GObject in session @id.
2141 gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
2142 g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
2143 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2144 get_internal_storage), NULL, NULL, g_cclosure_marshal_generic,
2145 G_TYPE_OBJECT, 1, G_TYPE_UINT);
2148 * GstRtpBin::get-storage:
2149 * @rtpbin: the object which received the signal
2150 * @id: the session id
2152 * Request the RTPStorage element as #GObject in session @id.
2156 gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
2157 g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
2158 G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
2159 get_storage), NULL, NULL, g_cclosure_marshal_generic,
2160 GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2163 * GstRtpBin::on-new-ssrc:
2164 * @rtpbin: the object which received the signal
2165 * @session: the session
2168 * Notify of a new SSRC that entered @session.
2170 gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
2171 g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
2172 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
2173 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2176 * GstRtpBin::on-ssrc-collision:
2177 * @rtpbin: the object which received the signal
2178 * @session: the session
2181 * Notify when we have an SSRC collision
2183 gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
2184 g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
2185 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
2186 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2189 * GstRtpBin::on-ssrc-validated:
2190 * @rtpbin: the object which received the signal
2191 * @session: the session
2194 * Notify of a new SSRC that became validated.
2196 gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
2197 g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
2198 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
2199 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2202 * GstRtpBin::on-ssrc-active:
2203 * @rtpbin: the object which received the signal
2204 * @session: the session
2207 * Notify of a SSRC that is active, i.e., sending RTCP.
2209 gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
2210 g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
2211 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
2212 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2215 * GstRtpBin::on-ssrc-sdes:
2216 * @rtpbin: the object which received the signal
2217 * @session: the session
2220 * Notify of a SSRC that is active, i.e., sending RTCP.
2222 gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
2223 g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
2224 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
2225 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2229 * GstRtpBin::on-bye-ssrc:
2230 * @rtpbin: the object which received the signal
2231 * @session: the session
2234 * Notify of an SSRC that became inactive because of a BYE packet.
2236 gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
2237 g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
2238 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
2239 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2242 * GstRtpBin::on-bye-timeout:
2243 * @rtpbin: the object which received the signal
2244 * @session: the session
2247 * Notify of an SSRC that has timed out because of BYE
2249 gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
2250 g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
2251 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
2252 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2255 * GstRtpBin::on-timeout:
2256 * @rtpbin: the object which received the signal
2257 * @session: the session
2260 * Notify of an SSRC that has timed out
2262 gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
2263 g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
2264 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
2265 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2268 * GstRtpBin::on-sender-timeout:
2269 * @rtpbin: the object which received the signal
2270 * @session: the session
2273 * Notify of a sender SSRC that has timed out and became a receiver
2275 gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
2276 g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
2277 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
2278 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2282 * GstRtpBin::on-npt-stop:
2283 * @rtpbin: the object which received the signal
2284 * @session: the session
2287 * Notify that SSRC sender has sent data up to the configured NPT stop time.
2289 gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
2290 g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
2291 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
2292 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2296 * GstRtpBin::request-rtp-encoder:
2297 * @rtpbin: the object which received the signal
2298 * @session: the session
2300 * Request an RTP encoder element for the given @session. The encoder
2301 * element will be added to the bin if not previously added.
2303 * If no handler is connected, no encoder will be used.
2307 gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] =
2308 g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass),
2309 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2310 request_rtp_encoder), _gst_element_accumulator, NULL,
2311 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2314 * GstRtpBin::request-rtp-decoder:
2315 * @rtpbin: the object which received the signal
2316 * @session: the session
2318 * Request an RTP decoder element for the given @session. The decoder
2319 * element will be added to the bin if not previously added.
2321 * If no handler is connected, no encoder will be used.
2325 gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] =
2326 g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass),
2327 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2328 request_rtp_decoder), _gst_element_accumulator, NULL,
2329 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2332 * GstRtpBin::request-rtcp-encoder:
2333 * @rtpbin: the object which received the signal
2334 * @session: the session
2336 * Request an RTCP encoder element for the given @session. The encoder
2337 * element will be added to the bin if not previously added.
2339 * If no handler is connected, no encoder will be used.
2343 gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] =
2344 g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
2345 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2346 request_rtcp_encoder), _gst_element_accumulator, NULL,
2347 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2350 * GstRtpBin::request-rtcp-decoder:
2351 * @rtpbin: the object which received the signal
2352 * @session: the session
2354 * Request an RTCP decoder element for the given @session. The decoder
2355 * element will be added to the bin if not previously added.
2357 * If no handler is connected, no encoder will be used.
2361 gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] =
2362 g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
2363 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2364 request_rtcp_decoder), _gst_element_accumulator, NULL,
2365 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2368 * GstRtpBin::new-jitterbuffer:
2369 * @rtpbin: the object which received the signal
2370 * @jitterbuffer: the new jitterbuffer
2371 * @session: the session
2374 * Notify that a new @jitterbuffer was created for @session and @ssrc.
2375 * This signal can, for example, be used to configure @jitterbuffer.
2379 gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] =
2380 g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass),
2381 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2382 new_jitterbuffer), NULL, NULL, g_cclosure_marshal_generic,
2383 G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
2386 * GstRtpBin::new-storage:
2387 * @rtpbin: the object which received the signal
2388 * @storage: the new storage
2389 * @session: the session
2391 * Notify that a new @storage was created for @session.
2392 * This signal can, for example, be used to configure @storage.
2396 gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
2397 g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
2398 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2399 new_storage), NULL, NULL, g_cclosure_marshal_generic,
2400 G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
2403 * GstRtpBin::request-aux-sender:
2404 * @rtpbin: the object which received the signal
2405 * @session: the session
2407 * Request an AUX sender element for the given @session. The AUX
2408 * element will be added to the bin.
2410 * If no handler is connected, no AUX element will be used.
2414 gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] =
2415 g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
2416 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2417 request_aux_sender), _gst_element_accumulator, NULL,
2418 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2421 * GstRtpBin::request-aux-receiver:
2422 * @rtpbin: the object which received the signal
2423 * @session: the session
2425 * Request an AUX receiver element for the given @session. The AUX
2426 * element will be added to the bin.
2428 * If no handler is connected, no AUX element will be used.
2432 gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] =
2433 g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass),
2434 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2435 request_aux_receiver), _gst_element_accumulator, NULL,
2436 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2439 * GstRtpBin::request-fec-decoder:
2440 * @rtpbin: the object which received the signal
2441 * @session: the session index
2443 * Request a FEC decoder element for the given @session. The element
2444 * will be added to the bin after the pt demuxer.
2446 * If no handler is connected, no FEC decoder will be used.
2450 gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
2451 g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
2452 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2453 request_fec_decoder), _gst_element_accumulator, NULL,
2454 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2457 * GstRtpBin::request-fec-encoder:
2458 * @rtpbin: the object which received the signal
2459 * @session: the session index
2461 * Request a FEC encoder element for the given @session. The element
2462 * will be added to the bin after the RTPSession.
2464 * If no handler is connected, no FEC encoder will be used.
2468 gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
2469 g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
2470 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2471 request_fec_encoder), _gst_element_accumulator, NULL,
2472 g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
2475 * GstRtpBin::on-new-sender-ssrc:
2476 * @rtpbin: the object which received the signal
2477 * @session: the session
2478 * @ssrc: the sender SSRC
2480 * Notify of a new sender SSRC that entered @session.
2484 gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
2485 g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
2486 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_sender_ssrc),
2487 NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
2490 * GstRtpBin::on-sender-ssrc-active:
2491 * @rtpbin: the object which received the signal
2492 * @session: the session
2493 * @ssrc: the sender SSRC
2495 * Notify of a sender SSRC that is active, i.e., sending RTCP.
2499 gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
2500 g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
2501 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2502 on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_generic,
2503 G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
2507 * GstRtpBin::on-bundled-ssrc:
2508 * @rtpbin: the object which received the signal
2509 * @ssrc: the bundled SSRC
2511 * Notify of a new incoming bundled SSRC. If no handler is connected to the
2512 * signal then the #GstRtpSession created for the recv_rtp_sink_\%u
2513 * request pad will be managing this new SSRC. However if there is a handler
2514 * connected then the application can decided to dispatch this new stream to
2515 * another session by providing its ID as return value of the handler. This
2516 * can be particularly useful to keep retransmission SSRCs grouped with the
2517 * session for which they handle retransmission.
2521 gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC] =
2522 g_signal_new ("on-bundled-ssrc", G_TYPE_FROM_CLASS (klass),
2523 G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
2524 on_bundled_ssrc), NULL, NULL,
2525 g_cclosure_marshal_generic, G_TYPE_UINT, 1, G_TYPE_UINT);
2528 g_object_class_install_property (gobject_class, PROP_SDES,
2529 g_param_spec_boxed ("sdes", "SDES",
2530 "The SDES items of this session",
2531 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2533 g_object_class_install_property (gobject_class, PROP_DO_LOST,
2534 g_param_spec_boolean ("do-lost", "Do Lost",
2535 "Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
2536 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2538 g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
2539 g_param_spec_boolean ("autoremove", "Auto Remove",
2540 "Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
2541 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2543 g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
2544 g_param_spec_boolean ("ignore-pt", "Ignore PT",
2545 "Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
2546 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2548 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
2549 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
2550 "Use the pipeline running-time to set the NTP time in the RTCP SR messages "
2551 "(DEPRECATED: Use ntp-time-source property)",
2552 DEFAULT_USE_PIPELINE_CLOCK,
2553 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
2555 * GstRtpBin:buffer-mode:
2557 * Control the buffering and timestamping mode used by the jitterbuffer.
2559 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
2560 g_param_spec_enum ("buffer-mode", "Buffer Mode",
2561 "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
2562 DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2564 * GstRtpBin:ntp-sync:
2566 * Set the NTP time from the sender reports as the running-time on the
2567 * buffers. When both the sender and receiver have sychronized
2568 * running-time, i.e. when the clock and base-time is shared
2569 * between the receivers and the and the senders, this option can be
2570 * used to synchronize receivers on multiple machines.
2572 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
2573 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
2574 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
2575 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2578 * GstRtpBin:rtcp-sync:
2580 * If not synchronizing (directly) to the NTP clock, determines how to sync
2581 * the various streams.
2583 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
2584 g_param_spec_enum ("rtcp-sync", "RTCP Sync",
2585 "Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
2586 DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2589 * GstRtpBin:rtcp-sync-interval:
2591 * Determines how often to sync streams using RTCP data.
2593 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
2594 g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
2595 "RTCP SR interval synchronization (ms) (0 = always)",
2596 0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
2597 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2599 g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT,
2600 g_param_spec_boolean ("do-sync-event", "Do Sync Event",
2601 "Send event downstream when a stream is synchronized to the sender",
2602 DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2605 * GstRtpBin:do-retransmission:
2607 * Enables RTP retransmission on all streams. To control retransmission on
2608 * a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and
2609 * set the #GstRtpJitterBuffer::do-retransmission property on the
2610 * #GstRtpJitterBuffer object instead.
2612 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
2613 g_param_spec_boolean ("do-retransmission", "Do retransmission",
2614 "Enable retransmission on all streams",
2615 DEFAULT_DO_RETRANSMISSION,
2616 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2619 * GstRtpBin:rtp-profile:
2621 * Sets the default RTP profile of newly created RTP sessions. The
2622 * profile can be changed afterwards on a per-session basis.
2624 g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
2625 g_param_spec_enum ("rtp-profile", "RTP Profile",
2626 "Default RTP profile of newly created sessions",
2627 GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
2628 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2630 g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
2631 g_param_spec_enum ("ntp-time-source", "NTP Time Source",
2632 "NTP time source for RTCP packets",
2633 gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
2634 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2636 g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
2637 g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
2638 "Use send time or capture time for RTCP sync "
2639 "(TRUE = send time, FALSE = capture time)",
2640 DEFAULT_RTCP_SYNC_SEND_TIME,
2641 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2643 g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
2644 g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
2645 "Maximum amount of time in ms that the RTP time in RTCP SRs "
2646 "is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
2647 DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
2648 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2650 g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
2651 g_param_spec_uint ("max-dropout-time", "Max dropout time",
2652 "The maximum time (milliseconds) of missing packets tolerated.",
2653 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
2654 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2656 g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
2657 g_param_spec_uint ("max-misorder-time", "Max misorder time",
2658 "The maximum time (milliseconds) of misordered packets tolerated.",
2659 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
2660 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2662 g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
2663 g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
2664 "Synchronize received streams to the RFC7273 clock "
2665 "(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
2666 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2668 g_object_class_install_property (gobject_class, PROP_MAX_STREAMS,
2669 g_param_spec_uint ("max-streams", "Max Streams",
2670 "The maximum number of streams to create for one session",
2671 0, G_MAXUINT, DEFAULT_MAX_STREAMS,
2672 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2675 * GstRtpBin:max-ts-offset-adjustment:
2677 * Syncing time stamps to NTP time adds a time offset. This parameter
2678 * specifies the maximum number of nanoseconds per frame that this time offset
2679 * may be adjusted with. This is used to avoid sudden large changes to time
2684 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
2685 g_param_spec_uint64 ("max-ts-offset-adjustment",
2686 "Max Timestamp Offset Adjustment",
2687 "The maximum number of nanoseconds per frame that time stamp offsets "
2688 "may be adjusted (0 = no limit).", 0, G_MAXUINT64,
2689 DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
2690 G_PARAM_STATIC_STRINGS));
2693 * GstRtpBin:max-ts-offset:
2695 * Used to set an upper limit of how large a time offset may be. This
2696 * is used to protect against unrealistic values as a result of either
2697 * client,server or clock issues.
2701 g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
2702 g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
2703 "The maximum absolute value of the time offset in (nanoseconds). "
2704 "Note, if the ntp-sync parameter is set the default value is "
2705 "changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
2706 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
2708 gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
2709 gstelement_class->request_new_pad =
2710 GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
2711 gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
2714 gst_element_class_add_static_pad_template (gstelement_class,
2715 &rtpbin_recv_rtp_sink_template);
2716 gst_element_class_add_static_pad_template (gstelement_class,
2717 &rtpbin_recv_rtcp_sink_template);
2718 gst_element_class_add_static_pad_template (gstelement_class,
2719 &rtpbin_send_rtp_sink_template);
2722 gst_element_class_add_static_pad_template (gstelement_class,
2723 &rtpbin_recv_rtp_src_template);
2724 gst_element_class_add_static_pad_template (gstelement_class,
2725 &rtpbin_send_rtcp_src_template);
2726 gst_element_class_add_static_pad_template (gstelement_class,
2727 &rtpbin_send_rtp_src_template);
2729 gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
2730 "Filter/Network/RTP",
2731 "Real-Time Transport Protocol bin",
2732 "Wim Taymans <wim.taymans@gmail.com>");
2734 gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
2736 klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
2737 klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
2738 klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
2739 klass->get_internal_session =
2740 GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
2741 klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
2742 klass->get_internal_storage =
2743 GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
2744 klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
2745 klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
2746 klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
2747 klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
2749 GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
2753 gst_rtp_bin_init (GstRtpBin * rtpbin)
2757 rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin);
2758 g_mutex_init (&rtpbin->priv->bin_lock);
2759 g_mutex_init (&rtpbin->priv->dyn_lock);
2761 rtpbin->latency_ms = DEFAULT_LATENCY_MS;
2762 rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
2763 rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
2764 rtpbin->do_lost = DEFAULT_DO_LOST;
2765 rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
2766 rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
2767 rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC;
2768 rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
2769 rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
2770 rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
2771 rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
2772 rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT;
2773 rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION;
2774 rtpbin->rtp_profile = DEFAULT_RTP_PROFILE;
2775 rtpbin->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
2776 rtpbin->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
2777 rtpbin->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
2778 rtpbin->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
2779 rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
2780 rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
2781 rtpbin->max_streams = DEFAULT_MAX_STREAMS;
2782 rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
2783 rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
2784 rtpbin->max_ts_offset_is_set = FALSE;
2786 /* some default SDES entries */
2787 cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
2788 rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
2789 "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
2794 gst_rtp_bin_dispose (GObject * object)
2798 rtpbin = GST_RTP_BIN (object);
2800 GST_RTP_BIN_LOCK (rtpbin);
2801 GST_DEBUG_OBJECT (object, "freeing sessions");
2802 g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
2803 g_slist_free (rtpbin->sessions);
2804 rtpbin->sessions = NULL;
2805 GST_RTP_BIN_UNLOCK (rtpbin);
2807 G_OBJECT_CLASS (parent_class)->dispose (object);
2811 gst_rtp_bin_finalize (GObject * object)
2815 rtpbin = GST_RTP_BIN (object);
2818 gst_structure_free (rtpbin->sdes);
2820 g_mutex_clear (&rtpbin->priv->bin_lock);
2821 g_mutex_clear (&rtpbin->priv->dyn_lock);
2823 G_OBJECT_CLASS (parent_class)->finalize (object);
2828 gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
2835 GST_RTP_BIN_LOCK (bin);
2837 GST_OBJECT_LOCK (bin);
2839 gst_structure_free (bin->sdes);
2840 bin->sdes = gst_structure_copy (sdes);
2841 GST_OBJECT_UNLOCK (bin);
2843 /* store in all sessions */
2844 for (item = bin->sessions; item; item = g_slist_next (item)) {
2845 GstRtpBinSession *session = item->data;
2846 g_object_set (session->session, "sdes", sdes, NULL);
2849 GST_RTP_BIN_UNLOCK (bin);
2852 static GstStructure *
2853 gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
2855 GstStructure *result;
2857 GST_OBJECT_LOCK (bin);
2858 result = gst_structure_copy (bin->sdes);
2859 GST_OBJECT_UNLOCK (bin);
2865 gst_rtp_bin_set_property (GObject * object, guint prop_id,
2866 const GValue * value, GParamSpec * pspec)
2870 rtpbin = GST_RTP_BIN (object);
2874 GST_RTP_BIN_LOCK (rtpbin);
2875 rtpbin->latency_ms = g_value_get_uint (value);
2876 rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
2877 GST_RTP_BIN_UNLOCK (rtpbin);
2878 /* propagate the property down to the jitterbuffer */
2879 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
2881 case PROP_DROP_ON_LATENCY:
2882 GST_RTP_BIN_LOCK (rtpbin);
2883 rtpbin->drop_on_latency = g_value_get_boolean (value);
2884 GST_RTP_BIN_UNLOCK (rtpbin);
2885 /* propagate the property down to the jitterbuffer */
2886 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2887 "drop-on-latency", value);
2890 gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
2893 GST_RTP_BIN_LOCK (rtpbin);
2894 rtpbin->do_lost = g_value_get_boolean (value);
2895 GST_RTP_BIN_UNLOCK (rtpbin);
2896 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
2899 rtpbin->ntp_sync = g_value_get_boolean (value);
2900 /* The default value of max_ts_offset depends on ntp_sync. If user
2901 * hasn't set it then change default value */
2902 if (!rtpbin->max_ts_offset_is_set) {
2903 if (rtpbin->ntp_sync) {
2904 rtpbin->max_ts_offset = 0;
2906 rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
2910 case PROP_RTCP_SYNC:
2911 g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
2913 case PROP_RTCP_SYNC_INTERVAL:
2914 rtpbin->rtcp_sync_interval = g_value_get_uint (value);
2916 case PROP_IGNORE_PT:
2917 rtpbin->ignore_pt = g_value_get_boolean (value);
2919 case PROP_AUTOREMOVE:
2920 rtpbin->priv->autoremove = g_value_get_boolean (value);
2922 case PROP_USE_PIPELINE_CLOCK:
2925 GST_RTP_BIN_LOCK (rtpbin);
2926 rtpbin->use_pipeline_clock = g_value_get_boolean (value);
2927 for (sessions = rtpbin->sessions; sessions;
2928 sessions = g_slist_next (sessions)) {
2929 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2931 g_object_set (G_OBJECT (session->session),
2932 "use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
2934 GST_RTP_BIN_UNLOCK (rtpbin);
2937 case PROP_DO_SYNC_EVENT:
2938 rtpbin->send_sync_event = g_value_get_boolean (value);
2940 case PROP_BUFFER_MODE:
2941 GST_RTP_BIN_LOCK (rtpbin);
2942 rtpbin->buffer_mode = g_value_get_enum (value);
2943 GST_RTP_BIN_UNLOCK (rtpbin);
2944 /* propagate the property down to the jitterbuffer */
2945 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
2947 case PROP_DO_RETRANSMISSION:
2948 GST_RTP_BIN_LOCK (rtpbin);
2949 rtpbin->do_retransmission = g_value_get_boolean (value);
2950 GST_RTP_BIN_UNLOCK (rtpbin);
2951 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2952 "do-retransmission", value);
2954 case PROP_RTP_PROFILE:
2955 rtpbin->rtp_profile = g_value_get_enum (value);
2957 case PROP_NTP_TIME_SOURCE:{
2959 GST_RTP_BIN_LOCK (rtpbin);
2960 rtpbin->ntp_time_source = g_value_get_enum (value);
2961 for (sessions = rtpbin->sessions; sessions;
2962 sessions = g_slist_next (sessions)) {
2963 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2965 g_object_set (G_OBJECT (session->session),
2966 "ntp-time-source", rtpbin->ntp_time_source, NULL);
2968 GST_RTP_BIN_UNLOCK (rtpbin);
2971 case PROP_RTCP_SYNC_SEND_TIME:{
2973 GST_RTP_BIN_LOCK (rtpbin);
2974 rtpbin->rtcp_sync_send_time = g_value_get_boolean (value);
2975 for (sessions = rtpbin->sessions; sessions;
2976 sessions = g_slist_next (sessions)) {
2977 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
2979 g_object_set (G_OBJECT (session->session),
2980 "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time, NULL);
2982 GST_RTP_BIN_UNLOCK (rtpbin);
2985 case PROP_MAX_RTCP_RTP_TIME_DIFF:
2986 GST_RTP_BIN_LOCK (rtpbin);
2987 rtpbin->max_rtcp_rtp_time_diff = g_value_get_int (value);
2988 GST_RTP_BIN_UNLOCK (rtpbin);
2989 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2990 "max-rtcp-rtp-time-diff", value);
2992 case PROP_MAX_DROPOUT_TIME:
2993 GST_RTP_BIN_LOCK (rtpbin);
2994 rtpbin->max_dropout_time = g_value_get_uint (value);
2995 GST_RTP_BIN_UNLOCK (rtpbin);
2996 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
2997 "max-dropout-time", value);
2998 gst_rtp_bin_propagate_property_to_session (rtpbin, "max-dropout-time",
3001 case PROP_MAX_MISORDER_TIME:
3002 GST_RTP_BIN_LOCK (rtpbin);
3003 rtpbin->max_misorder_time = g_value_get_uint (value);
3004 GST_RTP_BIN_UNLOCK (rtpbin);
3005 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
3006 "max-misorder-time", value);
3007 gst_rtp_bin_propagate_property_to_session (rtpbin, "max-misorder-time",
3010 case PROP_RFC7273_SYNC:
3011 rtpbin->rfc7273_sync = g_value_get_boolean (value);
3012 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
3013 "rfc7273-sync", value);
3015 case PROP_MAX_STREAMS:
3016 rtpbin->max_streams = g_value_get_uint (value);
3018 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
3019 rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
3020 gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
3021 "max-ts-offset-adjustment", value);
3023 case PROP_MAX_TS_OFFSET:
3024 rtpbin->max_ts_offset = g_value_get_int64 (value);
3025 rtpbin->max_ts_offset_is_set = TRUE;
3028 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3034 gst_rtp_bin_get_property (GObject * object, guint prop_id,
3035 GValue * value, GParamSpec * pspec)
3039 rtpbin = GST_RTP_BIN (object);
3043 GST_RTP_BIN_LOCK (rtpbin);
3044 g_value_set_uint (value, rtpbin->latency_ms);
3045 GST_RTP_BIN_UNLOCK (rtpbin);
3047 case PROP_DROP_ON_LATENCY:
3048 GST_RTP_BIN_LOCK (rtpbin);
3049 g_value_set_boolean (value, rtpbin->drop_on_latency);
3050 GST_RTP_BIN_UNLOCK (rtpbin);
3053 g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
3056 GST_RTP_BIN_LOCK (rtpbin);
3057 g_value_set_boolean (value, rtpbin->do_lost);
3058 GST_RTP_BIN_UNLOCK (rtpbin);
3060 case PROP_IGNORE_PT:
3061 g_value_set_boolean (value, rtpbin->ignore_pt);
3064 g_value_set_boolean (value, rtpbin->ntp_sync);
3066 case PROP_RTCP_SYNC:
3067 g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
3069 case PROP_RTCP_SYNC_INTERVAL:
3070 g_value_set_uint (value, rtpbin->rtcp_sync_interval);
3072 case PROP_AUTOREMOVE:
3073 g_value_set_boolean (value, rtpbin->priv->autoremove);
3075 case PROP_BUFFER_MODE:
3076 g_value_set_enum (value, rtpbin->buffer_mode);
3078 case PROP_USE_PIPELINE_CLOCK:
3079 g_value_set_boolean (value, rtpbin->use_pipeline_clock);
3081 case PROP_DO_SYNC_EVENT:
3082 g_value_set_boolean (value, rtpbin->send_sync_event);
3084 case PROP_DO_RETRANSMISSION:
3085 GST_RTP_BIN_LOCK (rtpbin);
3086 g_value_set_boolean (value, rtpbin->do_retransmission);
3087 GST_RTP_BIN_UNLOCK (rtpbin);
3089 case PROP_RTP_PROFILE:
3090 g_value_set_enum (value, rtpbin->rtp_profile);
3092 case PROP_NTP_TIME_SOURCE:
3093 g_value_set_enum (value, rtpbin->ntp_time_source);
3095 case PROP_RTCP_SYNC_SEND_TIME:
3096 g_value_set_boolean (value, rtpbin->rtcp_sync_send_time);
3098 case PROP_MAX_RTCP_RTP_TIME_DIFF:
3099 GST_RTP_BIN_LOCK (rtpbin);
3100 g_value_set_int (value, rtpbin->max_rtcp_rtp_time_diff);
3101 GST_RTP_BIN_UNLOCK (rtpbin);
3103 case PROP_MAX_DROPOUT_TIME:
3104 g_value_set_uint (value, rtpbin->max_dropout_time);
3106 case PROP_MAX_MISORDER_TIME:
3107 g_value_set_uint (value, rtpbin->max_misorder_time);
3109 case PROP_RFC7273_SYNC:
3110 g_value_set_boolean (value, rtpbin->rfc7273_sync);
3112 case PROP_MAX_STREAMS:
3113 g_value_set_uint (value, rtpbin->max_streams);
3115 case PROP_MAX_TS_OFFSET_ADJUSTMENT:
3116 g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
3118 case PROP_MAX_TS_OFFSET:
3119 g_value_set_int64 (value, rtpbin->max_ts_offset);
3122 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
3128 gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
3132 rtpbin = GST_RTP_BIN (bin);
3134 switch (GST_MESSAGE_TYPE (message)) {
3135 case GST_MESSAGE_ELEMENT:
3137 const GstStructure *s = gst_message_get_structure (message);
3139 /* we change the structure name and add the session ID to it */
3140 if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
3141 GstRtpBinSession *sess;
3143 /* find the session we set it as object data */
3144 sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
3145 "GstRTPBin.session");
3147 if (G_LIKELY (sess)) {
3148 message = gst_message_make_writable (message);
3149 s = gst_message_get_structure (message);
3150 gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
3154 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3157 case GST_MESSAGE_BUFFERING:
3160 gint min_percent = 100;
3161 GSList *sessions, *streams;
3162 GstRtpBinStream *stream;
3163 gboolean change = FALSE, active = FALSE;
3164 GstClockTime min_out_time;
3165 GstBufferingMode mode;
3166 gint avg_in, avg_out;
3167 gint64 buffering_left;
3169 gst_message_parse_buffering (message, &percent);
3170 gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
3174 g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
3175 "GstRTPBin.stream");
3177 GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
3179 /* get the stream */
3180 if (G_LIKELY (stream)) {
3181 GST_RTP_BIN_LOCK (rtpbin);
3182 /* fill in the percent */
3183 stream->percent = percent;
3185 /* calculate the min value for all streams */
3186 for (sessions = rtpbin->sessions; sessions;
3187 sessions = g_slist_next (sessions)) {
3188 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
3190 GST_RTP_SESSION_LOCK (session);
3191 if (session->streams) {
3192 for (streams = session->streams; streams;
3193 streams = g_slist_next (streams)) {
3194 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
3196 GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
3199 /* find min percent */
3200 if (min_percent > stream->percent)
3201 min_percent = stream->percent;
3204 GST_INFO_OBJECT (bin,
3205 "session has no streams, setting min_percent to 0");
3208 GST_RTP_SESSION_UNLOCK (session);
3210 GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
3212 if (rtpbin->buffering) {
3213 if (min_percent == 100) {
3214 rtpbin->buffering = FALSE;
3219 if (min_percent < 100) {
3220 /* pause the streams */
3221 rtpbin->buffering = TRUE;
3226 GST_RTP_BIN_UNLOCK (rtpbin);
3228 gst_message_unref (message);
3230 /* make a new buffering message with the min value */
3232 gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
3233 gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
3236 if (G_UNLIKELY (change)) {
3238 guint64 running_time = 0;
3241 /* figure out the running time when we have a clock */
3242 if (G_LIKELY ((clock =
3243 gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
3244 guint64 now, base_time;
3246 now = gst_clock_get_time (clock);
3247 base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
3248 running_time = now - base_time;
3249 gst_object_unref (clock);
3251 GST_DEBUG_OBJECT (bin,
3252 "running time now %" GST_TIME_FORMAT,
3253 GST_TIME_ARGS (running_time));
3255 GST_RTP_BIN_LOCK (rtpbin);
3257 /* when we reactivate, calculate the offsets so that all streams have
3258 * an output time that is at least as big as the running_time */
3261 if (running_time > rtpbin->buffer_start) {
3262 offset = running_time - rtpbin->buffer_start;
3263 if (offset >= rtpbin->latency_ns)
3264 offset -= rtpbin->latency_ns;
3270 /* pause all streams */
3272 for (sessions = rtpbin->sessions; sessions;
3273 sessions = g_slist_next (sessions)) {
3274 GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
3276 GST_RTP_SESSION_LOCK (session);
3277 for (streams = session->streams; streams;
3278 streams = g_slist_next (streams)) {
3279 GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
3280 GstElement *element = stream->buffer;
3283 g_signal_emit_by_name (element, "set-active", active, offset,
3287 g_object_get (element, "percent", &stream->percent, NULL);
3291 if (min_out_time == -1 || last_out < min_out_time)
3292 min_out_time = last_out;
3295 GST_DEBUG_OBJECT (bin,
3296 "setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
3297 GST_TIME_FORMAT ", percent %d", element, active,
3298 GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
3301 GST_RTP_SESSION_UNLOCK (session);
3303 GST_DEBUG_OBJECT (bin,
3304 "min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
3306 /* the buffer_start is the min out time of all paused jitterbuffers */
3308 rtpbin->buffer_start = min_out_time;
3310 GST_RTP_BIN_UNLOCK (rtpbin);
3313 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3318 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
3324 static GstStateChangeReturn
3325 gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
3327 GstStateChangeReturn res;
3329 GstRtpBinPrivate *priv;
3331 rtpbin = GST_RTP_BIN (element);
3332 priv = rtpbin->priv;
3334 switch (transition) {
3335 case GST_STATE_CHANGE_NULL_TO_READY:
3337 case GST_STATE_CHANGE_READY_TO_PAUSED:
3338 priv->last_ntpnstime = 0;
3339 GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
3340 g_atomic_int_set (&priv->shutdown, 0);
3342 case GST_STATE_CHANGE_PAUSED_TO_READY:
3343 GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
3344 g_atomic_int_set (&priv->shutdown, 1);
3345 /* wait for all callbacks to end by taking the lock. No new callbacks will
3346 * be able to happen as we set the shutdown flag. */
3347 GST_RTP_BIN_DYN_LOCK (rtpbin);
3348 GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
3349 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
3355 res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
3357 switch (transition) {
3358 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
3360 case GST_STATE_CHANGE_PAUSED_TO_READY:
3362 case GST_STATE_CHANGE_READY_TO_NULL:
3371 session_request_element (GstRtpBinSession * session, guint signal)
3373 GstElement *element = NULL;
3374 GstRtpBin *bin = session->bin;
3376 g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element);
3379 if (!bin_manage_element (bin, element))
3381 session->elements = g_slist_prepend (session->elements, element);
3388 GST_WARNING_OBJECT (bin, "unable to manage element");
3389 gst_object_unref (element);
3395 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
3397 GstPad *gpad = GST_PAD_CAST (user_data);
3399 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
3400 gst_pad_store_sticky_event (gpad, *event);
3405 /* a new pad (SSRC) was created in @session. This signal is emitted from the
3406 * payload demuxer. */
3408 new_payload_found (GstElement * element, guint pt, GstPad * pad,
3409 GstRtpBinStream * stream)
3412 GstElementClass *klass;
3413 GstPadTemplate *templ;
3417 rtpbin = stream->bin;
3419 GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
3421 pad = gst_object_ref (pad);
3423 if (stream->session->storage) {
3424 GstElement *fec_decoder =
3425 session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
3428 GstPad *sinkpad, *srcpad;
3429 GstPadLinkReturn ret;
3431 sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
3434 goto fec_decoder_sink_failed;
3436 ret = gst_pad_link (pad, sinkpad);
3437 gst_object_unref (sinkpad);
3439 if (ret != GST_PAD_LINK_OK)
3440 goto fec_decoder_link_failed;
3442 srcpad = gst_element_get_static_pad (fec_decoder, "src");
3445 goto fec_decoder_src_failed;
3447 gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
3448 gst_object_unref (pad);
3453 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
3455 /* ghost the pad to the parent */
3456 klass = GST_ELEMENT_GET_CLASS (rtpbin);
3457 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
3458 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
3459 stream->session->id, stream->ssrc, pt);
3460 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
3462 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
3464 gst_pad_set_active (gpad, TRUE);
3465 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3467 gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
3468 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
3471 gst_object_unref (pad);
3477 GST_DEBUG ("ignoring, we are shutting down");
3480 fec_decoder_sink_failed:
3482 g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
3483 stream->session->id);
3486 fec_decoder_src_failed:
3488 g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
3489 stream->session->id);
3492 fec_decoder_link_failed:
3494 g_warning ("rtpbin: failed to link fec decoder for session %u",
3495 stream->session->id);
3501 payload_pad_removed (GstElement * element, GstPad * pad,
3502 GstRtpBinStream * stream)
3507 rtpbin = stream->bin;
3509 GST_DEBUG ("payload pad removed");
3511 GST_RTP_BIN_DYN_LOCK (rtpbin);
3512 if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
3513 g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
3515 gst_pad_set_active (gpad, FALSE);
3516 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
3518 GST_RTP_BIN_DYN_UNLOCK (rtpbin);
3522 pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
3527 rtpbin = session->bin;
3529 GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
3532 caps = get_pt_map (session, pt);
3541 GST_DEBUG_OBJECT (rtpbin, "could not get caps");
3547 ptdemux_pt_map_requested (GstElement * element, guint pt,
3548 GstRtpBinSession * session)
3550 GstCaps *ret = pt_map_requested (element, pt, session);
3552 if (ret && gst_caps_get_size (ret) == 1) {
3553 const GstStructure *s = gst_caps_get_structure (ret, 0);
3556 if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
3557 GValue v = G_VALUE_INIT;
3558 GValue v2 = G_VALUE_INIT;
3560 GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
3562 g_value_init (&v, GST_TYPE_ARRAY);
3563 g_value_init (&v2, G_TYPE_INT);
3564 g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
3565 g_value_set_int (&v2, pt);
3566 gst_value_array_append_value (&v, &v2);
3567 g_value_unset (&v2);
3568 g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
3577 payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
3579 GST_DEBUG_OBJECT (session->bin,
3580 "emiting signal for pt type changed to %u in session %u", pt,
3583 g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
3584 0, session->id, pt);
3587 /* emitted when caps changed for the session */
3589 caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
3594 const GstStructure *s;
3598 g_object_get (pad, "caps", &caps, NULL);
3603 GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
3605 s = gst_caps_get_structure (caps, 0);
3607 /* get payload, finish when it's not there */
3608 if (!gst_structure_get_int (s, "payload", &payload)) {
3609 gst_caps_unref (caps);
3613 GST_RTP_SESSION_LOCK (session);
3614 GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
3615 g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
3616 GST_RTP_SESSION_UNLOCK (session);
3619 /* a new pad (SSRC) was created in @session */
3621 new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
3622 GstRtpBinSession * session)
3625 GstRtpBinStream *stream;
3626 GstPad *sinkpad, *srcpad;
3629 rtpbin = session->bin;
3631 GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
3632 GST_DEBUG_PAD_NAME (pad));
3634 GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
3636 GST_RTP_SESSION_LOCK (session);
3638 /* create new stream */
3639 stream = create_stream (session, ssrc);
3643 /* get pad and link */
3644 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
3645 padname = g_strdup_printf ("src_%u", ssrc);
3646 srcpad = gst_element_get_static_pad (element, padname);
3648 sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
3649 gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
3650 gst_object_unref (sinkpad);
3651 gst_object_unref (srcpad);
3653 GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
3654 padname = g_strdup_printf ("rtcp_src_%u", ssrc);
3655 srcpad = gst_element_get_static_pad (element, padname);
3657 sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
3658 gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
3659 gst_object_unref (sinkpad);
3660 gst_object_unref (srcpad);
3662 /* connect to the RTCP sync signal from the jitterbuffer */
3663 GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
3664 stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
3665 "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
3667 if (stream->demux) {
3668 /* connect to the new-pad signal of the payload demuxer, this will expose the
3669 * new pad by ghosting it. */
3670 stream->demux_newpad_sig = g_signal_connect (stream->demux,
3671 "new-payload-type", (GCallback) new_payload_found, stream);
3672 stream->demux_padremoved_sig = g_signal_connect (stream->demux,
3673 "pad-removed", (GCallback) payload_pad_removed, stream);
3675 /* connect to the request-pt-map signal. This signal will be emitted by the
3676 * demuxer so that it can apply a proper caps on the buffers for the
3678 stream->demux_ptreq_sig = g_signal_connect (stream->demux,
3679 "request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
3680 /* connect to the signal so it can be forwarded. */
3681 stream->demux_ptchange_sig = g_signal_connect (stream->demux,
3682 "payload-type-change", (GCallback) payload_type_change, session);
3684 /* add rtpjitterbuffer src pad to pads */
3685 GstElementClass *klass;
3686 GstPadTemplate *templ;
3690 pad = gst_element_get_static_pad (stream->buffer, "src");
3692 /* ghost the pad to the parent */
3693 klass = GST_ELEMENT_GET_CLASS (rtpbin);
3694 templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
3695 padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
3696 stream->session->id, stream->ssrc, 255);
3697 gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
3700 gst_pad_set_active (gpad, TRUE);
3701 gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
3702 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
3704 gst_object_unref (pad);
3707 GST_RTP_SESSION_UNLOCK (session);
3708 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3715 GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
3720 GST_RTP_SESSION_UNLOCK (session);
3721 GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
3722 GST_DEBUG_OBJECT (rtpbin, "could not create stream");
3728 session_maybe_create_bundle_demuxer (GstRtpBinSession * session)
3732 if (session->bundle_demux)
3735 rtpbin = session->bin;
3736 if (g_signal_has_handler_pending (rtpbin,
3737 gst_rtp_bin_signals[SIGNAL_ON_BUNDLED_SSRC], 0, TRUE)) {
3738 GST_DEBUG_OBJECT (rtpbin, "Adding a bundle SSRC demuxer to session %u",
3740 session->bundle_demux = gst_element_factory_make ("rtpssrcdemux", NULL);
3741 session->bundle_demux_newpad_sig = g_signal_connect (session->bundle_demux,
3742 "new-ssrc-pad", (GCallback) new_bundled_ssrc_pad_found, session);
3744 gst_bin_add (GST_BIN_CAST (rtpbin), session->bundle_demux);
3745 gst_element_sync_state_with_parent (session->bundle_demux);
3747 GST_DEBUG_OBJECT (rtpbin,
3748 "No handler for the on-bundled-ssrc signal so no need for a bundle SSRC demuxer in session %u",
3754 complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session,
3755 gboolean bundle_demuxer_needed)
3757 guint sessid = session->id;
3758 GstPad *recv_rtp_sink;
3760 GstElement *decoder;
3762 g_assert (!session->recv_rtp_sink);
3764 /* get recv_rtp pad and store */
3765 session->recv_rtp_sink =
3766 gst_element_get_request_pad (session->session, "recv_rtp_sink");
3767 if (session->recv_rtp_sink == NULL)
3770 g_signal_connect (session->recv_rtp_sink, "notify::caps",
3771 (GCallback) caps_changed, session);
3773 if (bundle_demuxer_needed)
3774 session_maybe_create_bundle_demuxer (session);
3776 GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
3777 decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
3779 GstPad *decsrc, *decsink;
3780 GstPadLinkReturn ret;
3782 GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder");
3783 decsink = gst_element_get_static_pad (decoder, "rtp_sink");
3784 if (decsink == NULL)
3785 goto dec_sink_failed;
3787 recv_rtp_sink = decsink;
3789 decsrc = gst_element_get_static_pad (decoder, "rtp_src");
3791 goto dec_src_failed;
3793 if (session->bundle_demux) {
3795 demux_sink = gst_element_get_static_pad (session->bundle_demux, "sink");
3796 ret = gst_pad_link (decsrc, demux_sink);
3797 gst_object_unref (demux_sink);
3799 ret = gst_pad_link (decsrc, session->recv_rtp_sink);
3801 gst_object_unref (decsrc);
3803 if (ret != GST_PAD_LINK_OK)
3804 goto dec_link_failed;
3807 GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
3808 if (session->bundle_demux) {
3810 gst_element_get_static_pad (session->bundle_demux, "sink");
3813 gst_element_get_request_pad (session->rtp_funnel, "sink_%u");
3817 funnel_src = gst_element_get_static_pad (session->rtp_funnel, "src");
3818 gst_pad_link (funnel_src, session->recv_rtp_sink);
3819 gst_object_unref (funnel_src);
3821 return recv_rtp_sink;
3826 g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
3831 g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
3836 g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
3837 gst_object_unref (recv_rtp_sink);
3842 g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
3843 gst_object_unref (recv_rtp_sink);
3849 complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
3853 GstPad *recv_rtp_src;
3855 g_assert (!session->recv_rtp_src);
3857 session->recv_rtp_src =
3858 gst_element_get_static_pad (session->session, "recv_rtp_src");
3859 if (session->recv_rtp_src == NULL)
3862 /* find out if we need AUX elements */
3863 aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
3867 GstPadLinkReturn ret;
3869 GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
3871 pname = g_strdup_printf ("sink_%u", sessid);
3872 auxsink = gst_element_get_static_pad (aux, pname);
3874 if (auxsink == NULL)
3875 goto aux_sink_failed;
3877 ret = gst_pad_link (session->recv_rtp_src, auxsink);
3878 gst_object_unref (auxsink);
3879 if (ret != GST_PAD_LINK_OK)
3880 goto aux_link_failed;
3882 /* this can be NULL when this AUX element is not to be linked any further */
3883 pname = g_strdup_printf ("src_%u", sessid);
3884 recv_rtp_src = gst_element_get_static_pad (aux, pname);
3887 recv_rtp_src = gst_object_ref (session->recv_rtp_src);
3890 /* Add a storage element if needed */
3891 if (recv_rtp_src && session->storage) {
3892 GstPadLinkReturn ret;
3893 GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
3895 ret = gst_pad_link (recv_rtp_src, sinkpad);
3897 gst_object_unref (sinkpad);
3898 gst_object_unref (recv_rtp_src);
3900 if (ret != GST_PAD_LINK_OK)
3901 goto storage_link_failed;
3903 recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
3909 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
3910 sinkdpad = gst_element_get_static_pad (session->demux, "sink");
3911 GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
3912 gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
3913 gst_object_unref (sinkdpad);
3914 gst_object_unref (recv_rtp_src);
3916 /* connect to the new-ssrc-pad signal of the SSRC demuxer */
3917 session->demux_newpad_sig = g_signal_connect (session->demux,
3918 "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
3919 session->demux_padremoved_sig = g_signal_connect (session->demux,
3920 "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
3927 g_warning ("rtpbin: failed to get session recv_rtp_src pad");
3932 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
3937 g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
3940 storage_link_failed:
3942 g_warning ("rtpbin: failed to link storage");
3947 /* Create a pad for receiving RTP for the session in @name. Must be called with
3951 create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
3954 GstRtpBinSession *session;
3955 GstPad *recv_rtp_sink;
3957 /* first get the session number */
3958 if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
3961 GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
3963 /* get or create session */
3964 session = find_session_by_id (rtpbin, sessid);
3966 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
3967 /* create session now */
3968 session = create_session (rtpbin, sessid);
3969 if (session == NULL)
3973 /* check if pad was requested */
3974 if (session->recv_rtp_sink_ghost != NULL)
3975 return session->recv_rtp_sink_ghost;
3977 /* setup the session sink pad */
3978 recv_rtp_sink = complete_session_sink (rtpbin, session, TRUE);
3980 goto session_sink_failed;
3983 GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
3984 session->recv_rtp_sink_ghost =
3985 gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
3986 gst_object_unref (recv_rtp_sink);
3987 gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
3988 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
3990 complete_session_receiver (rtpbin, session, sessid);
3992 return session->recv_rtp_sink_ghost;
3997 g_warning ("rtpbin: invalid name given");
4002 /* create_session already warned */
4005 session_sink_failed:
4007 /* warning already done */
4013 remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4015 if (session->demux_newpad_sig) {
4016 g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
4017 session->demux_newpad_sig = 0;
4019 if (session->demux_padremoved_sig) {
4020 g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
4021 session->demux_padremoved_sig = 0;
4023 if (session->bundle_demux_newpad_sig) {
4024 g_signal_handler_disconnect (session->bundle_demux,
4025 session->bundle_demux_newpad_sig);
4026 session->bundle_demux_newpad_sig = 0;
4028 if (session->recv_rtp_src) {
4029 gst_object_unref (session->recv_rtp_src);
4030 session->recv_rtp_src = NULL;
4032 if (session->recv_rtp_sink) {
4033 gst_element_release_request_pad (session->session, session->recv_rtp_sink);
4034 gst_object_unref (session->recv_rtp_sink);
4035 session->recv_rtp_sink = NULL;
4037 if (session->recv_rtp_sink_ghost) {
4038 gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
4039 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4040 session->recv_rtp_sink_ghost);
4041 session->recv_rtp_sink_ghost = NULL;
4046 complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
4047 guint sessid, gboolean bundle_demuxer_needed)
4049 GstElement *decoder;
4051 GstPad *decsink = NULL;
4054 /* get recv_rtp pad and store */
4055 GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
4056 session->recv_rtcp_sink =
4057 gst_element_get_request_pad (session->session, "recv_rtcp_sink");
4058 if (session->recv_rtcp_sink == NULL)
4061 if (bundle_demuxer_needed)
4062 session_maybe_create_bundle_demuxer (session);
4064 GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
4065 decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
4068 GstPadLinkReturn ret;
4070 GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder");
4071 decsink = gst_element_get_static_pad (decoder, "rtcp_sink");
4072 decsrc = gst_element_get_static_pad (decoder, "rtcp_src");
4074 if (decsink == NULL)
4075 goto dec_sink_failed;
4078 goto dec_src_failed;
4080 if (session->bundle_demux) {
4083 gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
4084 ret = gst_pad_link (decsrc, demux_sink);
4085 gst_object_unref (demux_sink);
4087 ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
4089 gst_object_unref (decsrc);
4091 if (ret != GST_PAD_LINK_OK)
4092 goto dec_link_failed;
4094 GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
4095 if (session->bundle_demux) {
4096 decsink = gst_element_get_static_pad (session->bundle_demux, "rtcp_sink");
4098 decsink = gst_element_get_request_pad (session->rtcp_funnel, "sink_%u");
4102 /* get srcpad, link to SSRCDemux */
4103 GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
4104 session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
4105 if (session->sync_src == NULL)
4106 goto src_pad_failed;
4108 GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
4109 sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
4110 gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
4111 gst_object_unref (sinkdpad);
4113 funnel_src = gst_element_get_static_pad (session->rtcp_funnel, "src");
4114 gst_pad_link (funnel_src, session->recv_rtcp_sink);
4115 gst_object_unref (funnel_src);
4121 g_warning ("rtpbin: failed to get session rtcp_sink pad");
4126 g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
4131 g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
4136 g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
4141 g_warning ("rtpbin: failed to get session sync_src pad");
4145 gst_object_unref (decsink);
4149 /* Create a pad for receiving RTCP for the session in @name. Must be called with
4153 create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
4157 GstRtpBinSession *session;
4158 GstPad *decsink = NULL;
4160 /* first get the session number */
4161 if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
4164 GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
4166 /* get or create the session */
4167 session = find_session_by_id (rtpbin, sessid);
4169 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
4170 /* create session now */
4171 session = create_session (rtpbin, sessid);
4172 if (session == NULL)
4176 /* check if pad was requested */
4177 if (session->recv_rtcp_sink_ghost != NULL)
4178 return session->recv_rtcp_sink_ghost;
4180 decsink = complete_session_rtcp (rtpbin, session, sessid, TRUE);
4184 session->recv_rtcp_sink_ghost =
4185 gst_ghost_pad_new_from_template (name, decsink, templ);
4186 gst_object_unref (decsink);
4187 gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
4188 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
4189 session->recv_rtcp_sink_ghost);
4191 return session->recv_rtcp_sink_ghost;
4196 g_warning ("rtpbin: invalid name given");
4201 /* create_session already warned */
4207 remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4209 if (session->recv_rtcp_sink_ghost) {
4210 gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
4211 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4212 session->recv_rtcp_sink_ghost);
4213 session->recv_rtcp_sink_ghost = NULL;
4215 if (session->sync_src) {
4216 /* releasing the request pad should also unref the sync pad */
4217 gst_object_unref (session->sync_src);
4218 session->sync_src = NULL;
4220 if (session->recv_rtcp_sink) {
4221 gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
4222 gst_object_unref (session->recv_rtcp_sink);
4223 session->recv_rtcp_sink = NULL;
4228 complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
4231 guint sessid = session->id;
4232 GstPad *send_rtp_src;
4233 GstElement *encoder;
4234 GstElementClass *klass;
4235 GstPadTemplate *templ;
4236 gboolean ret = FALSE;
4239 send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
4241 if (send_rtp_src == NULL)
4244 GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
4245 encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER);
4248 GstPad *encsrc, *encsink;
4249 GstPadLinkReturn ret;
4251 GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
4252 ename = g_strdup_printf ("rtp_src_%u", sessid);
4253 encsrc = gst_element_get_static_pad (encoder, ename);
4257 goto enc_src_failed;
4259 ename = g_strdup_printf ("rtp_sink_%u", sessid);
4260 encsink = gst_element_get_static_pad (encoder, ename);
4262 if (encsink == NULL)
4263 goto enc_sink_failed;
4265 ret = gst_pad_link (send_rtp_src, encsink);
4266 gst_object_unref (encsink);
4267 gst_object_unref (send_rtp_src);
4269 send_rtp_src = encsrc;
4271 if (ret != GST_PAD_LINK_OK)
4272 goto enc_link_failed;
4274 GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
4277 /* ghost the new source pad */
4278 klass = GST_ELEMENT_GET_CLASS (rtpbin);
4279 gname = g_strdup_printf ("send_rtp_src_%u", sessid);
4280 templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
4281 session->send_rtp_src_ghost =
4282 gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
4283 gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
4284 gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
4285 session->send_rtp_src_ghost);
4286 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
4293 gst_object_unref (send_rtp_src);
4300 g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
4305 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4306 " src pad for session %u", encoder, sessid);
4311 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4312 " sink pad for session %u", encoder, sessid);
4317 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4324 setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data)
4329 GstRtpBinSession *session = user_data, *newsess;
4330 GstRtpBin *rtpbin = session->bin;
4331 GstPadLinkReturn ret;
4333 pad = g_value_get_object (item);
4334 name = gst_pad_get_name (pad);
4336 if (name == NULL || sscanf (name, "src_%u", &sessid) != 1)
4341 newsess = find_session_by_id (rtpbin, sessid);
4342 if (newsess == NULL) {
4343 /* create new session */
4344 newsess = create_session (rtpbin, sessid);
4345 if (newsess == NULL)
4347 } else if (newsess->send_rtp_sink != NULL)
4348 goto existing_session;
4350 /* get send_rtp pad and store */
4351 newsess->send_rtp_sink =
4352 gst_element_get_request_pad (newsess->session, "send_rtp_sink");
4353 if (newsess->send_rtp_sink == NULL)
4356 ret = gst_pad_link (pad, newsess->send_rtp_sink);
4357 if (ret != GST_PAD_LINK_OK)
4358 goto aux_link_failed;
4360 if (!complete_session_src (rtpbin, newsess))
4361 goto session_src_failed;
4368 GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name));
4374 /* create_session already warned */
4379 g_warning ("rtpbin: session %u is already a sender", sessid);
4384 g_warning ("rtpbin: failed to get session pad for session %u", sessid);
4389 g_warning ("rtpbin: failed to link AUX for session %u", sessid);
4394 g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
4400 setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session,
4404 GValue result = { 0, };
4405 GstIteratorResult res;
4407 it = gst_element_iterate_src_pads (aux);
4408 res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session);
4409 gst_iterator_free (it);
4411 return res == GST_ITERATOR_DONE;
4414 /* Create a pad for sending RTP for the session in @name. Must be called with
4418 create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
4422 GstPad *send_rtp_sink;
4424 GstElement *encoder;
4425 GstElement *prev = NULL;
4426 GstRtpBinSession *session;
4428 /* first get the session number */
4429 if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
4432 /* get or create session */
4433 session = find_session_by_id (rtpbin, sessid);
4435 /* create session now */
4436 session = create_session (rtpbin, sessid);
4437 if (session == NULL)
4441 /* check if pad was requested */
4442 if (session->send_rtp_sink_ghost != NULL)
4443 return session->send_rtp_sink_ghost;
4445 /* check if we are already using this session as a sender */
4446 if (session->send_rtp_sink != NULL)
4447 goto existing_session;
4449 encoder = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
4452 GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
4454 send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
4457 goto enc_sink_failed;
4462 GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
4463 aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
4466 GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
4467 if (!setup_aux_sender (rtpbin, session, aux))
4468 goto aux_session_failed;
4470 pname = g_strdup_printf ("sink_%u", sessid);
4471 sinkpad = gst_element_get_static_pad (aux, pname);
4474 if (sinkpad == NULL)
4475 goto aux_sink_failed;
4478 send_rtp_sink = sinkpad;
4480 GstPad *srcpad = gst_element_get_static_pad (prev, "src");
4481 GstPadLinkReturn ret;
4483 ret = gst_pad_link (srcpad, sinkpad);
4484 gst_object_unref (srcpad);
4485 if (ret != GST_PAD_LINK_OK) {
4486 goto aux_link_failed;
4491 /* get send_rtp pad and store */
4492 session->send_rtp_sink =
4493 gst_element_get_request_pad (session->session, "send_rtp_sink");
4494 if (session->send_rtp_sink == NULL)
4497 if (!complete_session_src (rtpbin, session))
4498 goto session_src_failed;
4501 send_rtp_sink = gst_object_ref (session->send_rtp_sink);
4503 GstPad *srcpad = gst_element_get_static_pad (prev, "src");
4504 GstPadLinkReturn ret;
4506 ret = gst_pad_link (srcpad, session->send_rtp_sink);
4507 gst_object_unref (srcpad);
4508 if (ret != GST_PAD_LINK_OK)
4509 goto session_link_failed;
4513 session->send_rtp_sink_ghost =
4514 gst_ghost_pad_new_from_template (name, send_rtp_sink, templ);
4515 gst_object_unref (send_rtp_sink);
4516 gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
4517 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
4519 return session->send_rtp_sink_ghost;
4524 g_warning ("rtpbin: invalid name given");
4529 /* create_session already warned */
4534 g_warning ("rtpbin: session %u is already in use", sessid);
4539 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
4544 g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
4549 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4555 g_warning ("rtpbin: failed to get session pad for session %u", sessid);
4560 g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
4563 session_link_failed:
4565 g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
4571 g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
4572 " sink pad for session %u", encoder, sessid);
4578 remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4580 if (session->send_rtp_src_ghost) {
4581 gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
4582 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4583 session->send_rtp_src_ghost);
4584 session->send_rtp_src_ghost = NULL;
4586 if (session->send_rtp_sink) {
4587 gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
4588 session->send_rtp_sink);
4589 gst_object_unref (session->send_rtp_sink);
4590 session->send_rtp_sink = NULL;
4592 if (session->send_rtp_sink_ghost) {
4593 gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
4594 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4595 session->send_rtp_sink_ghost);
4596 session->send_rtp_sink_ghost = NULL;
4600 /* Create a pad for sending RTCP for the session in @name. Must be called with
4604 create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
4609 GstElement *encoder;
4610 GstRtpBinSession *session;
4612 /* first get the session number */
4613 if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
4616 /* get or create session */
4617 session = find_session_by_id (rtpbin, sessid);
4619 GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
4620 /* create session now */
4621 session = create_session (rtpbin, sessid);
4622 if (session == NULL)
4626 /* check if pad was requested */
4627 if (session->send_rtcp_src_ghost != NULL)
4628 return session->send_rtcp_src_ghost;
4630 /* get rtcp_src pad and store */
4631 session->send_rtcp_src =
4632 gst_element_get_request_pad (session->session, "send_rtcp_src");
4633 if (session->send_rtcp_src == NULL)
4636 GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder");
4637 encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER);
4641 GstPadLinkReturn ret;
4643 GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
4645 ename = g_strdup_printf ("rtcp_src_%u", sessid);
4646 encsrc = gst_element_get_static_pad (encoder, ename);
4649 goto enc_src_failed;
4651 ename = g_strdup_printf ("rtcp_sink_%u", sessid);
4652 encsink = gst_element_get_static_pad (encoder, ename);
4654 if (encsink == NULL)
4655 goto enc_sink_failed;
4657 ret = gst_pad_link (session->send_rtcp_src, encsink);
4658 gst_object_unref (encsink);
4660 if (ret != GST_PAD_LINK_OK)
4661 goto enc_link_failed;
4663 GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given");
4664 encsrc = gst_object_ref (session->send_rtcp_src);
4667 session->send_rtcp_src_ghost =
4668 gst_ghost_pad_new_from_template (name, encsrc, templ);
4669 gst_object_unref (encsrc);
4670 gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
4671 gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
4673 return session->send_rtcp_src_ghost;
4678 g_warning ("rtpbin: invalid name given");
4683 /* create_session already warned */
4688 g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
4693 g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
4698 g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
4699 gst_object_unref (encsrc);
4704 g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
4705 gst_object_unref (encsrc);
4711 remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
4713 if (session->send_rtcp_src_ghost) {
4714 gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
4715 gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
4716 session->send_rtcp_src_ghost);
4717 session->send_rtcp_src_ghost = NULL;
4719 if (session->send_rtcp_src) {
4720 gst_element_release_request_pad (session->session, session->send_rtcp_src);
4721 gst_object_unref (session->send_rtcp_src);
4722 session->send_rtcp_src = NULL;
4726 /* If the requested name is NULL we should create a name with
4727 * the session number assuming we want the lowest posible session
4728 * with a free pad like the template */
4730 gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
4732 gboolean name_found = FALSE;
4734 GstIterator *pad_it = NULL;
4735 gchar *pad_name = NULL;
4736 GValue data = { 0, };
4738 GST_DEBUG_OBJECT (element, "find a free pad name for template");
4739 while (!name_found) {
4740 gboolean done = FALSE;
4743 pad_name = g_strdup_printf (templ->name_template, session++);
4744 pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
4747 switch (gst_iterator_next (pad_it, &data)) {
4748 case GST_ITERATOR_OK:
4753 pad = g_value_get_object (&data);
4754 name = gst_pad_get_name (pad);
4756 if (strcmp (name, pad_name) == 0) {
4761 g_value_reset (&data);
4764 case GST_ITERATOR_ERROR:
4765 case GST_ITERATOR_RESYNC:
4766 /* restart iteration */
4771 case GST_ITERATOR_DONE:
4776 g_value_unset (&data);
4777 gst_iterator_free (pad_it);
4780 GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
4787 gst_rtp_bin_request_new_pad (GstElement * element,
4788 GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
4791 GstElementClass *klass;
4794 gchar *pad_name = NULL;
4796 g_return_val_if_fail (templ != NULL, NULL);
4797 g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
4799 rtpbin = GST_RTP_BIN (element);
4800 klass = GST_ELEMENT_GET_CLASS (element);
4802 GST_RTP_BIN_LOCK (rtpbin);
4805 /* use a free pad name */
4806 pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
4808 /* use the provided name */
4809 pad_name = g_strdup (name);
4812 GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
4814 /* figure out the template */
4815 if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
4816 result = create_recv_rtp (rtpbin, templ, pad_name);
4817 } else if (templ == gst_element_class_get_pad_template (klass,
4818 "recv_rtcp_sink_%u")) {
4819 result = create_recv_rtcp (rtpbin, templ, pad_name);
4820 } else if (templ == gst_element_class_get_pad_template (klass,
4821 "send_rtp_sink_%u")) {
4822 result = create_send_rtp (rtpbin, templ, pad_name);
4823 } else if (templ == gst_element_class_get_pad_template (klass,
4824 "send_rtcp_src_%u")) {
4825 result = create_send_rtcp (rtpbin, templ, pad_name);
4827 goto wrong_template;
4830 GST_RTP_BIN_UNLOCK (rtpbin);
4838 GST_RTP_BIN_UNLOCK (rtpbin);
4839 g_warning ("rtpbin: this is not our template");
4845 gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
4847 GstRtpBinSession *session;
4850 g_return_if_fail (GST_IS_GHOST_PAD (pad));
4851 g_return_if_fail (GST_IS_RTP_BIN (element));
4853 rtpbin = GST_RTP_BIN (element);
4855 GST_RTP_BIN_LOCK (rtpbin);
4856 GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
4857 GST_DEBUG_PAD_NAME (pad));
4859 if (!(session = find_session_by_pad (rtpbin, pad)))
4862 if (session->recv_rtp_sink_ghost == pad) {
4863 remove_recv_rtp (rtpbin, session);
4864 } else if (session->recv_rtcp_sink_ghost == pad) {
4865 remove_recv_rtcp (rtpbin, session);
4866 } else if (session->send_rtp_sink_ghost == pad) {
4867 remove_send_rtp (rtpbin, session);
4868 } else if (session->send_rtcp_src_ghost == pad) {
4869 remove_rtcp (rtpbin, session);
4872 /* no more request pads, free the complete session */
4873 if (session->recv_rtp_sink_ghost == NULL
4874 && session->recv_rtcp_sink_ghost == NULL
4875 && session->send_rtp_sink_ghost == NULL
4876 && session->send_rtcp_src_ghost == NULL) {
4877 GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
4878 rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
4879 free_session (session, rtpbin);
4881 GST_RTP_BIN_UNLOCK (rtpbin);
4888 GST_RTP_BIN_UNLOCK (rtpbin);
4889 g_warning ("rtpbin: %s:%s is not one of our request pads",
4890 GST_DEBUG_PAD_NAME (pad));