/* GStreamer * Copyright (C) <2007> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-rtpbin * @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux * * RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux, * #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple * RTP sessions that will be synchronized together using RTCP SR packets. * * #GstRtpBin is configured with a number of request pads that define the * functionality that is activated, similar to the #GstRtpSession element. * * To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%u pad. The session * number must be specified in the pad name. * Data received on the recv_rtp_sink_\%u pad will be processed in the #GstRtpSession * manager and after being validated forwarded on #GstRtpSsrcDemux element. Each * RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After * the packets are released from the jitterbuffer, they will be forwarded to a * #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based * on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on * rtpbin with the session number, SSRC and payload type respectively as the pad * name. * * To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The * session number must be specified in the pad name. * * If you want the session manager to generate and send RTCP packets, request * the send_rtcp_src_\%u pad with the session number in the pad name. Packet pushed * on this pad contain SR/RR RTCP reports that should be sent to all participants * in the session. * * To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will * automatically create a send_rtp_src_\%u pad. If the session number is not provided, * the pad from the lowest available session will be returned. The session manager will modify the * SSRC in the RTP packets to its own SSRC and wil forward the packets on the * send_rtp_src_\%u pad after updating its internal state. * * The session manager needs the clock-rate of the payload types it is handling * and will signal the #GstRtpSession::request-pt-map signal when it needs such a * mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map * signal. * * Access to the internal statistics of rtpbin is provided with the * get-internal-session property. This action signal gives access to the * RTPSession object which further provides action signals to retrieve the * internal source and other sources. * * #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder, * #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and * #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders * and decoders in order to support SRTP. The encoders must provide the pads * rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for * RTCP. The session number will be used in the pad name. The decoders must provide * rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will * be placed before the #GstRtpSession element, thus they must support SSRC demuxing * internally. * * #GstRtpBin has signals (#GstRtpBin::request-aux-sender and * #GstRtpBin::request-aux-receiver to dynamically request an element that can be * used to create or merge additional RTP streams. AUX elements are needed to * implement FEC or retransmission (such as RFC 4588). An AUX sender must have one * sink_\%u pad that matches the sessionid in the signal and it should have 1 or * more src_\%u pads. For each src_%\u pad, a session will be made (if needed) * and the pad will be linked to the session send_rtp_sink pad. Each session will * then expose its source pad as send_rtp_src_\%u on #GstRtpBin. * An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal * and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad * when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin. * * * Example pipelines * |[ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \ * rtpbin ! rtptheoradepay ! theoradec ! xvimagesink * ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin. * |[ * gst-launch-1.0 rtpbin name=rtpbin \ * v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \ * rtpbin.send_rtp_src_0 ! udpsink port=5000 \ * rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \ * udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \ * audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \ * rtpbin.send_rtp_src_1 ! udpsink port=5002 \ * rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \ * udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1 * ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR * audio generated from audiotestsrc. The video is sent to session 0 in rtpbin * and the audio is sent to session 1. Video packets are sent on UDP port 5000 * and audio packets on port 5002. The video RTCP packets for session 0 are sent * on port 5001 and the audio RTCP packets for session 0 are sent on port 5003. * RTCP packets for session 0 are received on port 5005 and RTCP for session 1 * is received on port 5007. Since RTCP packets from the sender should be sent * as soon as possible and do not participate in preroll, sync=false and * async=false is configured on udpsink * |[ * gst-launch-1.0 -v rtpbin name=rtpbin \ * udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \ * port=5000 ! rtpbin.recv_rtp_sink_0 \ * rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \ * udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \ * rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \ * udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \ * port=5002 ! rtpbin.recv_rtp_sink_1 \ * rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \ * udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \ * rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false * ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload, * decode and display the video. * Receive AMR on port 5002, send it through rtpbin in session 1, depayload, * decode and play the audio. * Receive server RTCP packets for session 0 on port 5001 and RTCP packets for * session 1 on port 5003. These packets will be used for session management and * synchronisation. * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 * on port 5007. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include "gstrtpbin.h" #include "rtpsession.h" #include "gstrtpsession.h" #include "gstrtpjitterbuffer.h" #include GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug); #define GST_CAT_DEFAULT gst_rtp_bin_debug /* sink pads */ static GstStaticPadTemplate rtpbin_recv_rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp;application/x-srtp") ); static GstStaticPadTemplate rtpbin_recv_rtcp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp") ); static GstStaticPadTemplate rtpbin_send_rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp") ); /* src pads */ static GstStaticPadTemplate rtpbin_recv_rtp_src_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate rtpbin_send_rtcp_src_template = GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u", GST_PAD_SRC, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp") ); static GstStaticPadTemplate rtpbin_send_rtp_src_template = GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp;application/x-srtp") ); #define GST_RTP_BIN_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate)) #define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock) #define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock) /* lock to protect dynamic callbacks, like pad-added and new ssrc. */ #define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock) #define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock) /* lock for shutdown */ #define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \ G_STMT_START { \ if (g_atomic_int_get (&bin->priv->shutdown)) \ goto label; \ GST_RTP_BIN_DYN_LOCK (bin); \ if (g_atomic_int_get (&bin->priv->shutdown)) { \ GST_RTP_BIN_DYN_UNLOCK (bin); \ goto label; \ } \ } G_STMT_END /* unlock for shutdown */ #define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \ GST_RTP_BIN_DYN_UNLOCK (bin); \ struct _GstRtpBinPrivate { GMutex bin_lock; /* lock protecting dynamic adding/removing */ GMutex dyn_lock; /* if we are shutting down or not */ gint shutdown; gboolean autoremove; /* UNIX (ntp) time of last SR sync used */ guint64 last_unix; /* list of extra elements */ GList *elements; }; /* signals and args */ enum { SIGNAL_REQUEST_PT_MAP, SIGNAL_PAYLOAD_TYPE_CHANGE, SIGNAL_CLEAR_PT_MAP, SIGNAL_RESET_SYNC, SIGNAL_GET_INTERNAL_SESSION, SIGNAL_ON_NEW_SSRC, SIGNAL_ON_SSRC_COLLISION, SIGNAL_ON_SSRC_VALIDATED, SIGNAL_ON_SSRC_ACTIVE, SIGNAL_ON_SSRC_SDES, SIGNAL_ON_BYE_SSRC, SIGNAL_ON_BYE_TIMEOUT, SIGNAL_ON_TIMEOUT, SIGNAL_ON_SENDER_TIMEOUT, SIGNAL_ON_NPT_STOP, SIGNAL_REQUEST_RTP_ENCODER, SIGNAL_REQUEST_RTP_DECODER, SIGNAL_REQUEST_RTCP_ENCODER, SIGNAL_REQUEST_RTCP_DECODER, SIGNAL_NEW_JITTERBUFFER, SIGNAL_REQUEST_AUX_SENDER, SIGNAL_REQUEST_AUX_RECEIVER, LAST_SIGNAL }; #define DEFAULT_LATENCY_MS 200 #define DEFAULT_DROP_ON_LATENCY FALSE #define DEFAULT_SDES NULL #define DEFAULT_DO_LOST FALSE #define DEFAULT_IGNORE_PT FALSE #define DEFAULT_NTP_SYNC FALSE #define DEFAULT_AUTOREMOVE FALSE #define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE #define DEFAULT_USE_PIPELINE_CLOCK FALSE #define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS #define DEFAULT_RTCP_SYNC_INTERVAL 0 #define DEFAULT_DO_SYNC_EVENT FALSE #define DEFAULT_DO_RETRANSMISSION FALSE #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVPF enum { PROP_0, PROP_LATENCY, PROP_DROP_ON_LATENCY, PROP_SDES, PROP_DO_LOST, PROP_IGNORE_PT, PROP_NTP_SYNC, PROP_RTCP_SYNC, PROP_RTCP_SYNC_INTERVAL, PROP_AUTOREMOVE, PROP_BUFFER_MODE, PROP_USE_PIPELINE_CLOCK, PROP_DO_SYNC_EVENT, PROP_DO_RETRANSMISSION, PROP_RTP_PROFILE }; #define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type()) static GType gst_rtp_bin_rtcp_sync_get_type (void) { static GType rtcp_sync_type = 0; static const GEnumValue rtcp_sync_types[] = { {GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"}, {GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"}, {GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"}, {0, NULL, NULL}, }; if (!rtcp_sync_type) { rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types); } return rtcp_sync_type; } /* helper objects */ typedef struct _GstRtpBinSession GstRtpBinSession; typedef struct _GstRtpBinStream GstRtpBinStream; typedef struct _GstRtpBinClient GstRtpBinClient; static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 }; static GstCaps *pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session); static void payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session); static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session); static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session); static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session); static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session); static void free_client (GstRtpBinClient * client, GstRtpBin * bin); static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin); /* Manages the RTP stream for one SSRC. * * We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer. * If we see an SDES RTCP packet that links multiple SSRCs together based on a * common CNAME, we create a GstRtpBinClient structure to group the SSRCs * together (see below). */ struct _GstRtpBinStream { /* the SSRC of this stream */ guint32 ssrc; /* parent bin */ GstRtpBin *bin; /* the session this SSRC belongs to */ GstRtpBinSession *session; /* the jitterbuffer of the SSRC */ GstElement *buffer; gulong buffer_handlesync_sig; gulong buffer_ptreq_sig; gulong buffer_ntpstop_sig; gint percent; /* the PT demuxer of the SSRC */ GstElement *demux; gulong demux_newpad_sig; gulong demux_padremoved_sig; gulong demux_ptreq_sig; gulong demux_ptchange_sig; /* if we have calculated a valid rt_delta for this stream */ gboolean have_sync; /* mapping to local RTP and NTP time */ gint64 rt_delta; gint64 rtp_delta; /* base rtptime in gst time */ gint64 clock_base; }; #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock) #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock) /* Manages the receiving end of the packets. * * There is one such structure for each RTP session (audio/video/...). * We get the RTP/RTCP packets and stuff them into the session manager. From * there they are pushed into an SSRC demuxer that splits the stream based on * SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with * the GstRtpBinStream above). */ struct _GstRtpBinSession { /* session id */ gint id; /* the parent bin */ GstRtpBin *bin; /* the session element */ GstElement *session; /* the SSRC demuxer */ GstElement *demux; gulong demux_newpad_sig; gulong demux_padremoved_sig; GMutex lock; /* list of GstRtpBinStream */ GSList *streams; /* list of elements */ GSList *elements; /* mapping of payload type to caps */ GHashTable *ptmap; /* the pads of the session */ GstPad *recv_rtp_sink; GstPad *recv_rtp_sink_ghost; GstPad *recv_rtp_src; GstPad *recv_rtcp_sink; GstPad *recv_rtcp_sink_ghost; GstPad *sync_src; GstPad *send_rtp_sink; GstPad *send_rtp_sink_ghost; GstPad *send_rtp_src; GstPad *send_rtp_src_ghost; GstPad *send_rtcp_src; GstPad *send_rtcp_src_ghost; }; /* Manages the RTP streams that come from one client and should therefore be * synchronized. */ struct _GstRtpBinClient { /* the common CNAME for the streams */ gchar *cname; guint cname_len; /* the streams */ guint nstreams; GSList *streams; }; /* find a session with the given id. Must be called with RTP_BIN_LOCK */ static GstRtpBinSession * find_session_by_id (GstRtpBin * rtpbin, gint id) { GSList *walk; for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) { GstRtpBinSession *sess = (GstRtpBinSession *) walk->data; if (sess->id == id) return sess; } return NULL; } /* find a session with the given request pad. Must be called with RTP_BIN_LOCK */ static GstRtpBinSession * find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad) { GSList *walk; for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) { GstRtpBinSession *sess = (GstRtpBinSession *) walk->data; if ((sess->recv_rtp_sink_ghost == pad) || (sess->recv_rtcp_sink_ghost == pad) || (sess->send_rtp_sink_ghost == pad) || (sess->send_rtcp_src_ghost == pad)) return sess; } return NULL; } static void on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0, sess->id, ssrc); } static void on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0, sess->id, ssrc); } static void on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0, sess->id, ssrc); } static void on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0, sess->id, ssrc); } static void on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0, sess->id, ssrc); } static void on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0, sess->id, ssrc); } static void on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0, sess->id, ssrc); if (sess->bin->priv->autoremove) g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL); } static void on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0, sess->id, ssrc); if (sess->bin->priv->autoremove) g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL); } static void on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess) { g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, sess->id, ssrc); } static void on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream) { g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0, stream->session->id, stream->ssrc); } /* must be called with the SESSION lock */ static GstRtpBinStream * find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc) { GSList *walk; for (walk = session->streams; walk; walk = g_slist_next (walk)) { GstRtpBinStream *stream = (GstRtpBinStream *) walk->data; if (stream->ssrc == ssrc) return stream; } return NULL; } static void ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad, GstRtpBinSession * session) { GstRtpBinStream *stream = NULL; GstRtpBin *rtpbin; rtpbin = session->bin; GST_RTP_BIN_LOCK (rtpbin); GST_RTP_SESSION_LOCK (session); if ((stream = find_stream_by_ssrc (session, ssrc))) session->streams = g_slist_remove (session->streams, stream); GST_RTP_SESSION_UNLOCK (session); if (stream) free_stream (stream, rtpbin); GST_RTP_BIN_UNLOCK (rtpbin); } /* create a session with the given id. Must be called with RTP_BIN_LOCK */ static GstRtpBinSession * create_session (GstRtpBin * rtpbin, gint id) { GstRtpBinSession *sess; GstElement *session, *demux; GstState target; if (!(session = gst_element_factory_make ("rtpsession", NULL))) goto no_session; if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL))) goto no_demux; sess = g_new0 (GstRtpBinSession, 1); g_mutex_init (&sess->lock); sess->id = id; sess->bin = rtpbin; sess->session = session; sess->demux = demux; sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) gst_caps_unref); rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess); /* configure SDES items */ GST_OBJECT_LOCK (rtpbin); g_object_set (session, "sdes", rtpbin->sdes, "use-pipeline-clock", rtpbin->use_pipeline_clock, "rtp-profile", rtpbin->rtp_profile, NULL); GST_OBJECT_UNLOCK (rtpbin); /* provide clock_rate to the session manager when needed */ g_signal_connect (session, "request-pt-map", (GCallback) pt_map_requested, sess); g_signal_connect (sess->session, "on-new-ssrc", (GCallback) on_new_ssrc, sess); g_signal_connect (sess->session, "on-ssrc-collision", (GCallback) on_ssrc_collision, sess); g_signal_connect (sess->session, "on-ssrc-validated", (GCallback) on_ssrc_validated, sess); g_signal_connect (sess->session, "on-ssrc-active", (GCallback) on_ssrc_active, sess); g_signal_connect (sess->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes, sess); g_signal_connect (sess->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, sess); g_signal_connect (sess->session, "on-bye-timeout", (GCallback) on_bye_timeout, sess); g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess); g_signal_connect (sess->session, "on-sender-timeout", (GCallback) on_sender_timeout, sess); gst_bin_add (GST_BIN_CAST (rtpbin), session); gst_bin_add (GST_BIN_CAST (rtpbin), demux); GST_OBJECT_LOCK (rtpbin); target = GST_STATE_TARGET (rtpbin); GST_OBJECT_UNLOCK (rtpbin); /* change state only to what's needed */ gst_element_set_state (demux, target); gst_element_set_state (session, target); return sess; /* ERRORS */ no_session: { g_warning ("rtpbin: could not create rtpsession element"); return NULL; } no_demux: { gst_object_unref (session); g_warning ("rtpbin: could not create rtpssrcdemux element"); return NULL; } } static gboolean bin_manage_element (GstRtpBin * bin, GstElement * element) { GstRtpBinPrivate *priv = bin->priv; if (g_list_find (priv->elements, element)) { GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element); } else { GST_DEBUG_OBJECT (bin, "adding requested element %p", element); if (!gst_bin_add (GST_BIN_CAST (bin), element)) goto add_failed; if (!gst_element_sync_state_with_parent (element)) GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin"); } /* we add the element multiple times, each we need an equal number of * removes to really remove the element from the bin */ priv->elements = g_list_prepend (priv->elements, element); return TRUE; /* ERRORS */ add_failed: { GST_WARNING_OBJECT (bin, "unable to add element"); return FALSE; } } static void remove_bin_element (GstElement * element, GstRtpBin * bin) { GstRtpBinPrivate *priv = bin->priv; GList *find; find = g_list_find (priv->elements, element); if (find) { priv->elements = g_list_delete_link (priv->elements, find); if (!g_list_find (priv->elements, element)) gst_bin_remove (GST_BIN_CAST (bin), element); else gst_object_unref (element); } } /* called with RTP_BIN_LOCK */ static void free_session (GstRtpBinSession * sess, GstRtpBin * bin) { GST_DEBUG_OBJECT (bin, "freeing session %p", sess); gst_element_set_locked_state (sess->demux, TRUE); gst_element_set_locked_state (sess->session, TRUE); gst_element_set_state (sess->demux, GST_STATE_NULL); gst_element_set_state (sess->session, GST_STATE_NULL); remove_recv_rtp (bin, sess); remove_recv_rtcp (bin, sess); remove_send_rtp (bin, sess); remove_rtcp (bin, sess); gst_bin_remove (GST_BIN_CAST (bin), sess->session); gst_bin_remove (GST_BIN_CAST (bin), sess->demux); g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin); g_slist_free (sess->elements); g_slist_foreach (sess->streams, (GFunc) free_stream, bin); g_slist_free (sess->streams); g_mutex_clear (&sess->lock); g_hash_table_destroy (sess->ptmap); g_free (sess); } /* get the payload type caps for the specific payload @pt in @session */ static GstCaps * get_pt_map (GstRtpBinSession * session, guint pt) { GstCaps *caps = NULL; GstRtpBin *bin; GValue ret = { 0 }; GValue args[3] = { {0}, {0}, {0} }; GST_DEBUG ("searching pt %d in cache", pt); GST_RTP_SESSION_LOCK (session); /* first look in the cache */ caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt)); if (caps) { gst_caps_ref (caps); goto done; } bin = session->bin; GST_DEBUG ("emiting signal for pt %d in session %d", pt, session->id); /* not in cache, send signal to request caps */ g_value_init (&args[0], GST_TYPE_ELEMENT); g_value_set_object (&args[0], bin); g_value_init (&args[1], G_TYPE_UINT); g_value_set_uint (&args[1], session->id); g_value_init (&args[2], G_TYPE_UINT); g_value_set_uint (&args[2], pt); g_value_init (&ret, GST_TYPE_CAPS); g_value_set_boxed (&ret, NULL); GST_RTP_SESSION_UNLOCK (session); g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); GST_RTP_SESSION_LOCK (session); g_value_unset (&args[0]); g_value_unset (&args[1]); g_value_unset (&args[2]); /* look in the cache again because we let the lock go */ caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt)); if (caps) { gst_caps_ref (caps); g_value_unset (&ret); goto done; } caps = (GstCaps *) g_value_dup_boxed (&ret); g_value_unset (&ret); if (!caps) goto no_caps; GST_DEBUG ("caching pt %d as %" GST_PTR_FORMAT, pt, caps); /* store in cache, take additional ref */ g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt), gst_caps_ref (caps)); done: GST_RTP_SESSION_UNLOCK (session); return caps; /* ERRORS */ no_caps: { GST_RTP_SESSION_UNLOCK (session); GST_DEBUG ("no pt map could be obtained"); return NULL; } } static gboolean return_true (gpointer key, gpointer value, gpointer user_data) { return TRUE; } static void gst_rtp_bin_reset_sync (GstRtpBin * rtpbin) { GSList *clients, *streams; GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients"); GST_RTP_BIN_LOCK (rtpbin); for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) { GstRtpBinClient *client = (GstRtpBinClient *) clients->data; /* reset sync on all streams for this client */ for (streams = client->streams; streams; streams = g_slist_next (streams)) { GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; /* make use require a new SR packet for this stream before we attempt new * lip-sync */ stream->have_sync = FALSE; stream->rt_delta = 0; stream->rtp_delta = 0; stream->clock_base = -100 * GST_SECOND; } } GST_RTP_BIN_UNLOCK (rtpbin); } static void gst_rtp_bin_clear_pt_map (GstRtpBin * bin) { GSList *sessions, *streams; GST_RTP_BIN_LOCK (bin); GST_DEBUG_OBJECT (bin, "clearing pt map"); for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) { GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; GST_DEBUG_OBJECT (bin, "clearing session %p", session); g_signal_emit_by_name (session->session, "clear-pt-map", NULL); GST_RTP_SESSION_LOCK (session); g_hash_table_foreach_remove (session->ptmap, return_true, NULL); for (streams = session->streams; streams; streams = g_slist_next (streams)) { GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; GST_DEBUG_OBJECT (bin, "clearing stream %p", stream); g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL); if (stream->demux) g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL); } GST_RTP_SESSION_UNLOCK (session); } GST_RTP_BIN_UNLOCK (bin); /* reset sync too */ gst_rtp_bin_reset_sync (bin); } static RTPSession * gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id) { RTPSession *internal_session = NULL; GstRtpBinSession *session; GST_RTP_BIN_LOCK (bin); GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %d", session_id); session = find_session_by_id (bin, (gint) session_id); if (session) { g_object_get (session->session, "internal-session", &internal_session, NULL); } GST_RTP_BIN_UNLOCK (bin); return internal_session; } static GstElement * gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id) { GST_DEBUG_OBJECT (bin, "return NULL encoder"); return NULL; } static GstElement * gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id) { GST_DEBUG_OBJECT (bin, "return NULL decoder"); return NULL; } static void gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin, const gchar * name, const GValue * value) { GSList *sessions, *streams; GST_RTP_BIN_LOCK (bin); for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) { GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; GST_RTP_SESSION_LOCK (session); for (streams = session->streams; streams; streams = g_slist_next (streams)) { GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; g_object_set_property (G_OBJECT (stream->buffer), name, value); } GST_RTP_SESSION_UNLOCK (session); } GST_RTP_BIN_UNLOCK (bin); } /* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */ static GstRtpBinClient * get_client (GstRtpBin * bin, guint8 len, guint8 * data, gboolean * created) { GstRtpBinClient *result = NULL; GSList *walk; for (walk = bin->clients; walk; walk = g_slist_next (walk)) { GstRtpBinClient *client = (GstRtpBinClient *) walk->data; if (len != client->cname_len) continue; if (!strncmp ((gchar *) data, client->cname, client->cname_len)) { GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client, client->cname); result = client; break; } } /* nothing found, create one */ if (result == NULL) { result = g_new0 (GstRtpBinClient, 1); result->cname = g_strndup ((gchar *) data, len); result->cname_len = len; bin->clients = g_slist_prepend (bin->clients, result); GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result, result->cname); } return result; } static void free_client (GstRtpBinClient * client, GstRtpBin * bin) { GST_DEBUG_OBJECT (bin, "freeing client %p", client); g_slist_free (client->streams); g_free (client->cname); g_free (client); } static void get_current_times (GstRtpBin * bin, GstClockTime * running_time, guint64 * ntpnstime) { guint64 ntpns; GstClock *clock; GstClockTime base_time, rt, clock_time; GST_OBJECT_LOCK (bin); if ((clock = GST_ELEMENT_CLOCK (bin))) { base_time = GST_ELEMENT_CAST (bin)->base_time; gst_object_ref (clock); GST_OBJECT_UNLOCK (bin); clock_time = gst_clock_get_time (clock); if (bin->use_pipeline_clock) { ntpns = clock_time - base_time; } else { GTimeVal current; /* get current NTP time */ g_get_current_time (¤t); ntpns = GST_TIMEVAL_TO_TIME (current); } /* add constant to convert from 1970 based time to 1900 based time */ ntpns += (2208988800LL * GST_SECOND); /* get current clock time and convert to running time */ rt = clock_time - base_time; gst_object_unref (clock); } else { GST_OBJECT_UNLOCK (bin); rt = -1; ntpns = -1; } if (running_time) *running_time = rt; if (ntpnstime) *ntpnstime = ntpns; } static void stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream, gint64 ts_offset, gboolean check) { gint64 prev_ts_offset; g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL); /* delta changed, see how much */ if (prev_ts_offset != ts_offset) { gint64 diff; diff = prev_ts_offset - ts_offset; GST_DEBUG_OBJECT (bin, "ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT ", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff); if (check) { /* only change diff when it changed more than 4 milliseconds. This * compensates for rounding errors in NTP to RTP timestamp * conversions */ if (ABS (diff) < 4 * GST_MSECOND) { GST_DEBUG_OBJECT (bin, "offset too small, ignoring"); return; } if (ABS (diff) > (3 * GST_SECOND)) { GST_WARNING_OBJECT (bin, "offset unusually large, ignoring"); return; } } g_object_set (stream->buffer, "ts-offset", ts_offset, NULL); } GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT, stream->ssrc, ts_offset); } static void gst_rtp_bin_send_sync_event (GstRtpBinStream * stream) { if (stream->bin->send_sync_event) { GstEvent *event; GstPad *srcpad; GST_DEBUG_OBJECT (stream->bin, "sending GstRTCPSRReceived event downstream"); event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM, gst_structure_new_empty ("GstRTCPSRReceived")); srcpad = gst_element_get_static_pad (stream->buffer, "src"); gst_pad_push_event (srcpad, event); gst_object_unref (srcpad); } } /* associate a stream to the given CNAME. This will make sure all streams for * that CNAME are synchronized together. * Must be called with GST_RTP_BIN_LOCK */ static void gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len, guint8 * data, guint64 ntptime, guint64 last_extrtptime, guint64 base_rtptime, guint64 base_time, guint clock_rate, gint64 rtp_clock_base) { GstRtpBinClient *client; gboolean created; GSList *walk; guint64 local_rt; guint64 local_rtp; GstClockTime running_time; guint64 ntpnstime; gint64 ntpdiff, rtdiff; guint64 last_unix; /* first find or create the CNAME */ client = get_client (bin, len, data, &created); /* find stream in the client */ for (walk = client->streams; walk; walk = g_slist_next (walk)) { GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; if (ostream == stream) break; } /* not found, add it to the list */ if (walk == NULL) { GST_DEBUG_OBJECT (bin, "new association of SSRC %08x with client %p with CNAME %s", stream->ssrc, client, client->cname); client->streams = g_slist_prepend (client->streams, stream); client->nstreams++; } else { GST_DEBUG_OBJECT (bin, "found association of SSRC %08x with client %p with CNAME %s", stream->ssrc, client, client->cname); } if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) { GST_DEBUG_OBJECT (bin, "invalidated sync data"); if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) { /* we don't need that data, so carry on, * but make some values look saner */ last_extrtptime = base_rtptime; } else { /* nothing we can do with this data in this case */ GST_DEBUG_OBJECT (bin, "bailing out"); return; } } /* Take the extended rtptime we found in the SR packet and map it to the * local rtptime. The local rtp time is used to construct timestamps on the * buffers so we will calculate what running_time corresponds to the RTP * timestamp in the SR packet. */ local_rtp = last_extrtptime - base_rtptime; GST_DEBUG_OBJECT (bin, "base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT ", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, " "clock-base %" G_GINT64_FORMAT, base_rtptime, last_extrtptime, local_rtp, clock_rate, rtp_clock_base); /* calculate local RTP time in gstreamer timestamp, we essentially perform the * same conversion that a jitterbuffer would use to convert an rtp timestamp * into a corresponding gstreamer timestamp. Note that the base_time also * contains the drift between sender and receiver. */ local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate); local_rt += base_time; /* convert ntptime to unix time since 1900 */ last_unix = gst_util_uint64_scale (ntptime, GST_SECOND, (G_GINT64_CONSTANT (1) << 32)); stream->have_sync = TRUE; GST_DEBUG_OBJECT (bin, "local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT, local_rt, last_unix); /* recalc inter stream playout offset, but only if there is more than one * stream or we're doing NTP sync. */ if (bin->ntp_sync) { /* For NTP sync we need to first get a snapshot of running_time and NTP * time. We know at what running_time we play a certain RTP time, we also * calculated when we would play the RTP time in the SR packet. Now we need * to know how the running_time and the NTP time relate to eachother. */ get_current_times (bin, &running_time, &ntpnstime); /* see how far away the NTP time is. This is the difference between the * current NTP time and the NTP time in the last SR packet. */ ntpdiff = ntpnstime - last_unix; /* see how far away the running_time is. This is the difference between the * current running_time and the running_time of the RTP timestamp in the * last SR packet. */ rtdiff = running_time - local_rt; GST_DEBUG_OBJECT (bin, "NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT, ntpnstime, last_unix); GST_DEBUG_OBJECT (bin, "NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff, rtdiff); /* combine to get the final diff to apply to the running_time */ stream->rt_delta = rtdiff - ntpdiff; stream_set_ts_offset (bin, stream, stream->rt_delta, FALSE); } else { gint64 min, rtp_min, clock_base = stream->clock_base; gboolean all_sync, use_rtp; gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync); /* calculate delta between server and receiver. last_unix is created by * converting the ntptime in the last SR packet to a gstreamer timestamp. This * delta expresses the difference to our timeline and the server timeline. The * difference in itself doesn't mean much but we can combine the delta of * multiple streams to create a stream specific offset. */ stream->rt_delta = last_unix - local_rt; /* calculate the min of all deltas, ignoring streams that did not yet have a * valid rt_delta because we did not yet receive an SR packet for those * streams. * We calculate the mininum because we would like to only apply positive * offsets to streams, delaying their playback instead of trying to speed up * other streams (which might be imposible when we have to create negative * latencies). * The stream that has the smallest diff is selected as the reference stream, * all other streams will have a positive offset to this difference. */ /* some alternative setting allow ignoring RTCP as much as possible, * for servers generating bogus ntp timeline */ min = rtp_min = G_MAXINT64; use_rtp = FALSE; if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) { guint64 ext_base; use_rtp = TRUE; /* signed version for convienience */ clock_base = base_rtptime; /* deal with possible wrap-around */ ext_base = base_rtptime; rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base); /* sanity check; base rtp and provided clock_base should be close */ if (rtp_clock_base >= clock_base) { if (rtp_clock_base - clock_base < 10 * clock_rate) { rtp_clock_base = base_time + gst_util_uint64_scale_int (rtp_clock_base - clock_base, GST_SECOND, clock_rate); } else { use_rtp = FALSE; } } else { if (clock_base - rtp_clock_base < 10 * clock_rate) { rtp_clock_base = base_time - gst_util_uint64_scale_int (clock_base - rtp_clock_base, GST_SECOND, clock_rate); } else { use_rtp = FALSE; } } /* warn and bail for clarity out if no sane values */ if (!use_rtp) { GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime"); return; } /* store to track changes */ clock_base = rtp_clock_base; /* generate a fake as before, * now equating rtptime obtained from RTP-Info, * where the large time represent the otherwise irrelevant npt/ntp time */ stream->rtp_delta = (GST_SECOND << 28) - rtp_clock_base; } else { clock_base = rtp_clock_base; } all_sync = TRUE; for (walk = client->streams; walk; walk = g_slist_next (walk)) { GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; if (!ostream->have_sync) { all_sync = FALSE; continue; } /* change in current stream's base from previously init'ed value * leads to reset of all stream's base */ if (stream != ostream && stream->clock_base >= 0 && (stream->clock_base != clock_base)) { GST_DEBUG_OBJECT (bin, "reset upon clock base change"); ostream->clock_base = -100 * GST_SECOND; ostream->rtp_delta = 0; } if (ostream->rt_delta < min) min = ostream->rt_delta; if (ostream->rtp_delta < rtp_min) rtp_min = ostream->rtp_delta; } /* arrange to re-sync for each stream upon significant change, * e.g. post-seek */ all_sync = all_sync && (stream->clock_base == clock_base); stream->clock_base = clock_base; /* may need init performed above later on, but nothing more to do now */ if (client->nstreams <= 1) return; GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT " all sync %d", client, min, all_sync); GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp); switch (rtcp_sync) { case GST_RTP_BIN_RTCP_SYNC_RTP: if (!use_rtp) break; GST_DEBUG_OBJECT (bin, "using rtp generated reports; " "client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min); /* fall-through */ case GST_RTP_BIN_RTCP_SYNC_INITIAL: /* if all have been synced already, do not bother further */ if (all_sync) { GST_DEBUG_OBJECT (bin, "all streams already synced; done"); return; } break; default: break; } /* bail out if we adjusted recently enough */ if (all_sync && (last_unix - bin->priv->last_unix) < bin->rtcp_sync_interval * GST_MSECOND) { GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; " "previous sender info too recent " "(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix); return; } bin->priv->last_unix = last_unix; /* calculate offsets for each stream */ for (walk = client->streams; walk; walk = g_slist_next (walk)) { GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data; gint64 ts_offset; /* ignore streams for which we didn't receive an SR packet yet, we * can't synchronize them yet. We can however sync other streams just * fine. */ if (!ostream->have_sync) continue; /* calculate offset to our reference stream, this should always give a * positive number. */ if (use_rtp) ts_offset = ostream->rtp_delta - rtp_min; else ts_offset = ostream->rt_delta - min; stream_set_ts_offset (bin, ostream, ts_offset, TRUE); } } gst_rtp_bin_send_sync_event (stream); return; } #define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \ for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \ (b) = gst_rtcp_packet_move_to_next ((packet))) #define GST_RTCP_SDES_FOR_ITEMS(b,packet) \ for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \ (b) = gst_rtcp_packet_sdes_next_item ((packet))) #define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \ for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \ (b) = gst_rtcp_packet_sdes_next_entry ((packet))) static void gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s, GstRtpBinStream * stream) { GstRtpBin *bin; GstRTCPPacket packet; guint32 ssrc; guint64 ntptime; gboolean have_sr, have_sdes; gboolean more; guint64 base_rtptime; guint64 base_time; guint clock_rate; guint64 clock_base; guint64 extrtptime; GstBuffer *buffer; GstRTCPBuffer rtcp = { NULL, }; bin = stream->bin; GST_DEBUG_OBJECT (bin, "sync handler called"); /* get the last relation between the rtp timestamps and the gstreamer * timestamps. We get this info directly from the jitterbuffer which * constructs gstreamer timestamps from rtp timestamps and so it know exactly * what the current situation is. */ base_rtptime = g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime")); base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time")); clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate")); clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base")); extrtptime = g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime")); buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer")); have_sr = FALSE; have_sdes = FALSE; gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp); GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) { /* first packet must be SR or RR or else the validate would have failed */ switch (gst_rtcp_packet_get_type (&packet)) { case GST_RTCP_TYPE_SR: /* only parse first. There is only supposed to be one SR in the packet * but we will deal with malformed packets gracefully */ if (have_sr) break; /* get NTP and RTP times */ gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL, NULL, NULL); GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc); /* ignore SR that is not ours */ if (ssrc != stream->ssrc) continue; have_sr = TRUE; break; case GST_RTCP_TYPE_SDES: { gboolean more_items, more_entries; /* only deal with first SDES, there is only supposed to be one SDES in * the RTCP packet but we deal with bad packets gracefully. Also bail * out if we have not seen an SR item yet. */ if (have_sdes || !have_sr) break; GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) { /* skip items that are not about the SSRC of the sender */ if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc) continue; /* find the CNAME entry */ GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) { GstRTCPSDESType type; guint8 len; guint8 *data; gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &data); if (type == GST_RTCP_SDES_CNAME) { GST_RTP_BIN_LOCK (bin); /* associate the stream to CNAME */ gst_rtp_bin_associate (bin, stream, len, data, ntptime, extrtptime, base_rtptime, base_time, clock_rate, clock_base); GST_RTP_BIN_UNLOCK (bin); } } } have_sdes = TRUE; break; } default: /* we can ignore these packets */ break; } } gst_rtcp_buffer_unmap (&rtcp); } /* create a new stream with @ssrc in @session. Must be called with * RTP_SESSION_LOCK. */ static GstRtpBinStream * create_stream (GstRtpBinSession * session, guint32 ssrc) { GstElement *buffer, *demux = NULL; GstRtpBinStream *stream; GstRtpBin *rtpbin; GstState target; rtpbin = session->bin; if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL))) goto no_jitterbuffer; if (!rtpbin->ignore_pt) if (!(demux = gst_element_factory_make ("rtpptdemux", NULL))) goto no_demux; stream = g_new0 (GstRtpBinStream, 1); stream->ssrc = ssrc; stream->bin = rtpbin; stream->session = session; stream->buffer = buffer; stream->demux = demux; stream->have_sync = FALSE; stream->rt_delta = 0; stream->rtp_delta = 0; stream->percent = 100; stream->clock_base = -100 * GST_SECOND; session->streams = g_slist_prepend (session->streams, stream); /* provide clock_rate to the jitterbuffer when needed */ stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map", (GCallback) pt_map_requested, session); stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop", (GCallback) on_npt_stop, stream); g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session); g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream); /* configure latency and packet lost */ g_object_set (buffer, "latency", rtpbin->latency_ms, NULL); g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL); g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL); g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL); g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL); g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0, buffer, session->id, ssrc); if (!rtpbin->ignore_pt) gst_bin_add (GST_BIN_CAST (rtpbin), demux); gst_bin_add (GST_BIN_CAST (rtpbin), buffer); /* link stuff */ if (demux) gst_element_link_pads_full (buffer, "src", demux, "sink", GST_PAD_LINK_CHECK_NOTHING); if (rtpbin->buffering) { guint64 last_out; GST_INFO_OBJECT (rtpbin, "bin is buffering, set jitterbuffer as not active"); g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0, &last_out); } GST_OBJECT_LOCK (rtpbin); target = GST_STATE_TARGET (rtpbin); GST_OBJECT_UNLOCK (rtpbin); /* from sink to source */ if (demux) gst_element_set_state (demux, target); gst_element_set_state (buffer, target); return stream; /* ERRORS */ no_jitterbuffer: { g_warning ("rtpbin: could not create rtpjitterbuffer element"); return NULL; } no_demux: { gst_object_unref (buffer); g_warning ("rtpbin: could not create rtpptdemux element"); return NULL; } } /* called with RTP_BIN_LOCK */ static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin) { GSList *clients, *next_client; GST_DEBUG_OBJECT (bin, "freeing stream %p", stream); if (stream->demux) { g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig); g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig); g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig); } g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig); g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig); g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig); if (stream->demux) gst_element_set_locked_state (stream->demux, TRUE); gst_element_set_locked_state (stream->buffer, TRUE); if (stream->demux) gst_element_set_state (stream->demux, GST_STATE_NULL); gst_element_set_state (stream->buffer, GST_STATE_NULL); /* now remove this signal, we need this while going to NULL because it to * do some cleanups */ if (stream->demux) g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig); gst_bin_remove (GST_BIN_CAST (bin), stream->buffer); if (stream->demux) gst_bin_remove (GST_BIN_CAST (bin), stream->demux); for (clients = bin->clients; clients; clients = next_client) { GstRtpBinClient *client = (GstRtpBinClient *) clients->data; GSList *streams, *next_stream; next_client = g_slist_next (clients); for (streams = client->streams; streams; streams = next_stream) { GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data; next_stream = g_slist_next (streams); if (ostream == stream) { client->streams = g_slist_delete_link (client->streams, streams); /* If this was the last stream belonging to this client, * clean up the client. */ if (--client->nstreams == 0) { bin->clients = g_slist_delete_link (bin->clients, clients); free_client (client, bin); break; } } } } g_free (stream); } /* GObject vmethods */ static void gst_rtp_bin_dispose (GObject * object); static void gst_rtp_bin_finalize (GObject * object); static void gst_rtp_bin_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_bin_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* GstElement vmethods */ static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element, GstStateChange transition); static GstPad *gst_rtp_bin_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps); static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad); static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message); #define gst_rtp_bin_parent_class parent_class G_DEFINE_TYPE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN); static gboolean _gst_element_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer dummy) { GstElement *element; element = g_value_get_object (handler_return); GST_DEBUG ("got element %" GST_PTR_FORMAT, element); if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP)) g_value_set_object (return_accu, element); /* stop emission if we have an element */ return (element == NULL); } static gboolean _gst_caps_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer dummy) { GstCaps *caps; caps = g_value_get_boxed (handler_return); GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps); if (!(ihint->run_type & G_SIGNAL_RUN_CLEANUP)) g_value_set_boxed (return_accu, caps); /* stop emission if we have a caps */ return (caps == NULL); } static void gst_rtp_bin_class_init (GstRtpBinClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstBinClass *gstbin_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gstbin_class = (GstBinClass *) klass; g_type_class_add_private (klass, sizeof (GstRtpBinPrivate)); gobject_class->dispose = gst_rtp_bin_dispose; gobject_class->finalize = gst_rtp_bin_finalize; gobject_class->set_property = gst_rtp_bin_set_property; gobject_class->get_property = gst_rtp_bin_get_property; g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Default amount of ms to buffer in the jitterbuffers", 0, G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY, g_param_spec_boolean ("drop-on-latency", "Drop buffers when maximum latency is reached", "Tells the jitterbuffer to never exceed the given latency in size", DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin::request-pt-map: * @rtpbin: the object which received the signal * @session: the session * @pt: the pt * * Request the payload type as #GstCaps for @pt in @session. */ gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] = g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map), _gst_caps_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::payload-type-change: * @rtpbin: the object which received the signal * @session: the session * @pt: the pt * * Signal that the current payload type changed to @pt in @session. */ gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] = g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::clear-pt-map: * @rtpbin: the object which received the signal * * Clear all previously cached pt-mapping obtained with * #GstRtpBin::request-pt-map. */ gst_rtp_bin_signals[SIGNAL_CLEAR_PT_MAP] = g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRtpBin::reset-sync: * @rtpbin: the object which received the signal * * Reset all currently configured lip-sync parameters and require new SR * packets for all streams before lip-sync is attempted again. */ gst_rtp_bin_signals[SIGNAL_RESET_SYNC] = g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, reset_sync), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRtpBin::get-internal-session: * @rtpbin: the object which received the signal * @id: the session id * * Request the internal RTPSession object as #GObject in session @id. */ gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] = g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass, get_internal_session), NULL, NULL, g_cclosure_marshal_generic, RTP_TYPE_SESSION, 1, G_TYPE_UINT); /** * GstRtpBin::on-new-ssrc: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a new SSRC that entered @session. */ gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] = g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-ssrc-collision: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify when we have an SSRC collision */ gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] = g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-ssrc-validated: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a new SSRC that became validated. */ gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] = g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-ssrc-active: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a SSRC that is active, i.e., sending RTCP. */ gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] = g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-ssrc-sdes: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a SSRC that is active, i.e., sending RTCP. */ gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] = g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-bye-ssrc: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that became inactive because of a BYE packet. */ gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] = g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-bye-timeout: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that has timed out because of BYE */ gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] = g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-timeout: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that has timed out */ gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] = g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-sender-timeout: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a sender SSRC that has timed out and became a receiver */ gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] = g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::on-npt-stop: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify that SSRC sender has sent data up to the configured NPT stop time. */ gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] = g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::request-rtp-encoder: * @rtpbin: the object which received the signal * @session: the session * * Request an RTP encoder element for the given @session. The encoder * element will be added to the bin if not previously added. * * If no handler is connected, no encoder will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] = g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_rtp_encoder), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); /** * GstRtpBin::request-rtp-decoder: * @rtpbin: the object which received the signal * @session: the session * * Request an RTP decoder element for the given @session. The decoder * element will be added to the bin if not previously added. * * If no handler is connected, no encoder will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] = g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_rtp_decoder), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); /** * GstRtpBin::request-rtcp-encoder: * @rtpbin: the object which received the signal * @session: the session * * Request an RTCP encoder element for the given @session. The encoder * element will be added to the bin if not previously added. * * If no handler is connected, no encoder will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] = g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_rtcp_encoder), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); /** * GstRtpBin::request-rtcp-decoder: * @rtpbin: the object which received the signal * @session: the session * * Request an RTCP decoder element for the given @session. The decoder * element will be added to the bin if not previously added. * * If no handler is connected, no encoder will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] = g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_rtcp_decoder), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); /** * GstRtpBin::new-jitterbuffer: * @rtpbin: the object which received the signal * @jitterbuffer: the new jitterbuffer * @session: the session * @ssrc: the SSRC * * Notify that a new @jitterbuffer was created for @session and @ssrc. * This signal can, for example, be used to configure @jitterbuffer. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] = g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, new_jitterbuffer), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT); /** * GstRtpBin::request-aux-sender: * @rtpbin: the object which received the signal * @session: the session * * Request an AUX sender element for the given @session. The AUX * element will be added to the bin. * * If no handler is connected, no AUX element will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] = g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_aux_sender), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); /** * GstRtpBin::request-aux-receiver: * @rtpbin: the object which received the signal * @session: the session * * Request an AUX receiver element for the given @session. The AUX * element will be added to the bin. * * If no handler is connected, no AUX element will be used. * * Since: 1.4 */ gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] = g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_aux_receiver), _gst_element_accumulator, NULL, g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT); g_object_class_install_property (gobject_class, PROP_SDES, g_param_spec_boxed ("sdes", "SDES", "The SDES items of this session", GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DO_LOST, g_param_spec_boolean ("do-lost", "Do Lost", "Send an event downstream when a packet is lost", DEFAULT_DO_LOST, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_AUTOREMOVE, g_param_spec_boolean ("autoremove", "Auto Remove", "Automatically remove timed out sources", DEFAULT_AUTOREMOVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_IGNORE_PT, g_param_spec_boolean ("ignore-pt", "Ignore PT", "Do not demultiplex based on PT values", DEFAULT_IGNORE_PT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK, g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock", "Use the pipeline running-time to set the NTP time in the RTCP SR messages", DEFAULT_USE_PIPELINE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:buffer-mode: * * Control the buffering and timestamping mode used by the jitterbuffer. */ g_object_class_install_property (gobject_class, PROP_BUFFER_MODE, g_param_spec_enum ("buffer-mode", "Buffer Mode", "Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE, DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:ntp-sync: * * Set the NTP time from the sender reports as the running-time on the * buffers. When both the sender and receiver have sychronized * running-time, i.e. when the clock and base-time is shared * between the receivers and the and the senders, this option can be * used to synchronize receivers on multiple machines. */ g_object_class_install_property (gobject_class, PROP_NTP_SYNC, g_param_spec_boolean ("ntp-sync", "Sync on NTP clock", "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:rtcp-sync: * * If not synchronizing (directly) to the NTP clock, determines how to sync * the various streams. */ g_object_class_install_property (gobject_class, PROP_RTCP_SYNC, g_param_spec_enum ("rtcp-sync", "RTCP Sync", "Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE, DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:rtcp-sync-interval: * * Determines how often to sync streams using RTCP data. */ g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL, g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval", "RTCP SR interval synchronization (ms) (0 = always)", 0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT, g_param_spec_boolean ("do-sync-event", "Do Sync Event", "Send event downstream when a stream is synchronized to the sender", DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:do-retransmission: * * Enables RTP retransmission on all streams. To control retransmission on * a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and * set the #GstRtpJitterBuffer::do-retransmission property on the * #GstRtpJitterBuffer object instead. */ g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION, g_param_spec_boolean ("do-retransmission", "Do retransmission", "Enable retransmission on all streams", DEFAULT_DO_RETRANSMISSION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpBin:rtp-profile: * * Sets the default RTP profile of newly created RTP sessions. The * profile can be changed afterwards on a per-session basis. */ g_object_class_install_property (gobject_class, PROP_RTP_PROFILE, g_param_spec_enum ("rtp-profile", "RTP Profile", "Default RTP profile of newly created sessions", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad); /* sink pads */ gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_send_rtp_sink_template)); /* src pads */ gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_recv_rtp_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_send_rtcp_src_template)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&rtpbin_send_rtp_src_template)); gst_element_class_set_static_metadata (gstelement_class, "RTP Bin", "Filter/Network/RTP", "Real-Time Transport Protocol bin", "Wim Taymans "); gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message); klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map); klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync); klass->get_internal_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session); klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder); klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder); klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder); klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder); GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin"); } static void gst_rtp_bin_init (GstRtpBin * rtpbin) { gchar *cname; rtpbin->priv = GST_RTP_BIN_GET_PRIVATE (rtpbin); g_mutex_init (&rtpbin->priv->bin_lock); g_mutex_init (&rtpbin->priv->dyn_lock); rtpbin->latency_ms = DEFAULT_LATENCY_MS; rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND; rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY; rtpbin->do_lost = DEFAULT_DO_LOST; rtpbin->ignore_pt = DEFAULT_IGNORE_PT; rtpbin->ntp_sync = DEFAULT_NTP_SYNC; rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC; rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL; rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE; rtpbin->buffer_mode = DEFAULT_BUFFER_MODE; rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK; rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT; rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION; rtpbin->rtp_profile = DEFAULT_RTP_PROFILE; /* some default SDES entries */ cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ()); rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes", "cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL); g_free (cname); } static void gst_rtp_bin_dispose (GObject * object) { GstRtpBin *rtpbin; rtpbin = GST_RTP_BIN (object); GST_RTP_BIN_LOCK (rtpbin); GST_DEBUG_OBJECT (object, "freeing sessions"); g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin); g_slist_free (rtpbin->sessions); rtpbin->sessions = NULL; GST_RTP_BIN_UNLOCK (rtpbin); G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_rtp_bin_finalize (GObject * object) { GstRtpBin *rtpbin; rtpbin = GST_RTP_BIN (object); if (rtpbin->sdes) gst_structure_free (rtpbin->sdes); g_mutex_clear (&rtpbin->priv->bin_lock); g_mutex_clear (&rtpbin->priv->dyn_lock); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes) { GSList *item; if (sdes == NULL) return; GST_RTP_BIN_LOCK (bin); GST_OBJECT_LOCK (bin); if (bin->sdes) gst_structure_free (bin->sdes); bin->sdes = gst_structure_copy (sdes); GST_OBJECT_UNLOCK (bin); /* store in all sessions */ for (item = bin->sessions; item; item = g_slist_next (item)) { GstRtpBinSession *session = item->data; g_object_set (session->session, "sdes", sdes, NULL); } GST_RTP_BIN_UNLOCK (bin); } static GstStructure * gst_rtp_bin_get_sdes_struct (GstRtpBin * bin) { GstStructure *result; GST_OBJECT_LOCK (bin); result = gst_structure_copy (bin->sdes); GST_OBJECT_UNLOCK (bin); return result; } static void gst_rtp_bin_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpBin *rtpbin; rtpbin = GST_RTP_BIN (object); switch (prop_id) { case PROP_LATENCY: GST_RTP_BIN_LOCK (rtpbin); rtpbin->latency_ms = g_value_get_uint (value); rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND; GST_RTP_BIN_UNLOCK (rtpbin); /* propagate the property down to the jitterbuffer */ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value); break; case PROP_DROP_ON_LATENCY: GST_RTP_BIN_LOCK (rtpbin); rtpbin->drop_on_latency = g_value_get_boolean (value); GST_RTP_BIN_UNLOCK (rtpbin); /* propagate the property down to the jitterbuffer */ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "drop-on-latency", value); break; case PROP_SDES: gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value)); break; case PROP_DO_LOST: GST_RTP_BIN_LOCK (rtpbin); rtpbin->do_lost = g_value_get_boolean (value); GST_RTP_BIN_UNLOCK (rtpbin); gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value); break; case PROP_NTP_SYNC: rtpbin->ntp_sync = g_value_get_boolean (value); break; case PROP_RTCP_SYNC: g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value)); break; case PROP_RTCP_SYNC_INTERVAL: rtpbin->rtcp_sync_interval = g_value_get_uint (value); break; case PROP_IGNORE_PT: rtpbin->ignore_pt = g_value_get_boolean (value); break; case PROP_AUTOREMOVE: rtpbin->priv->autoremove = g_value_get_boolean (value); break; case PROP_USE_PIPELINE_CLOCK: { GSList *sessions; GST_RTP_BIN_LOCK (rtpbin); rtpbin->use_pipeline_clock = g_value_get_boolean (value); for (sessions = rtpbin->sessions; sessions; sessions = g_slist_next (sessions)) { GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; g_object_set (G_OBJECT (session->session), "use-pipeline-clock", rtpbin->use_pipeline_clock, NULL); } GST_RTP_BIN_UNLOCK (rtpbin); } break; case PROP_DO_SYNC_EVENT: rtpbin->send_sync_event = g_value_get_boolean (value); break; case PROP_BUFFER_MODE: GST_RTP_BIN_LOCK (rtpbin); rtpbin->buffer_mode = g_value_get_enum (value); GST_RTP_BIN_UNLOCK (rtpbin); /* propagate the property down to the jitterbuffer */ gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value); break; case PROP_DO_RETRANSMISSION: GST_RTP_BIN_LOCK (rtpbin); rtpbin->do_retransmission = g_value_get_boolean (value); GST_RTP_BIN_UNLOCK (rtpbin); gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-retransmission", value); break; case PROP_RTP_PROFILE: rtpbin->rtp_profile = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_bin_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpBin *rtpbin; rtpbin = GST_RTP_BIN (object); switch (prop_id) { case PROP_LATENCY: GST_RTP_BIN_LOCK (rtpbin); g_value_set_uint (value, rtpbin->latency_ms); GST_RTP_BIN_UNLOCK (rtpbin); break; case PROP_DROP_ON_LATENCY: GST_RTP_BIN_LOCK (rtpbin); g_value_set_boolean (value, rtpbin->drop_on_latency); GST_RTP_BIN_UNLOCK (rtpbin); break; case PROP_SDES: g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin)); break; case PROP_DO_LOST: GST_RTP_BIN_LOCK (rtpbin); g_value_set_boolean (value, rtpbin->do_lost); GST_RTP_BIN_UNLOCK (rtpbin); break; case PROP_IGNORE_PT: g_value_set_boolean (value, rtpbin->ignore_pt); break; case PROP_NTP_SYNC: g_value_set_boolean (value, rtpbin->ntp_sync); break; case PROP_RTCP_SYNC: g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync)); break; case PROP_RTCP_SYNC_INTERVAL: g_value_set_uint (value, rtpbin->rtcp_sync_interval); break; case PROP_AUTOREMOVE: g_value_set_boolean (value, rtpbin->priv->autoremove); break; case PROP_BUFFER_MODE: g_value_set_enum (value, rtpbin->buffer_mode); break; case PROP_USE_PIPELINE_CLOCK: g_value_set_boolean (value, rtpbin->use_pipeline_clock); break; case PROP_DO_SYNC_EVENT: g_value_set_boolean (value, rtpbin->send_sync_event); break; case PROP_DO_RETRANSMISSION: GST_RTP_BIN_LOCK (rtpbin); g_value_set_boolean (value, rtpbin->do_retransmission); GST_RTP_BIN_UNLOCK (rtpbin); break; case PROP_RTP_PROFILE: g_value_set_enum (value, rtpbin->rtp_profile); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message) { GstRtpBin *rtpbin; rtpbin = GST_RTP_BIN (bin); switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ELEMENT: { const GstStructure *s = gst_message_get_structure (message); /* we change the structure name and add the session ID to it */ if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) { GstRtpBinSession *sess; /* find the session we set it as object data */ sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)), "GstRTPBin.session"); if (G_LIKELY (sess)) { message = gst_message_make_writable (message); s = gst_message_get_structure (message); gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT, sess->id, NULL); } } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } case GST_MESSAGE_BUFFERING: { gint percent; gint min_percent = 100; GSList *sessions, *streams; GstRtpBinStream *stream; gboolean change = FALSE, active = FALSE; GstClockTime min_out_time; GstBufferingMode mode; gint avg_in, avg_out; gint64 buffering_left; gst_message_parse_buffering (message, &percent); gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out, &buffering_left); stream = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)), "GstRTPBin.stream"); GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream); /* get the stream */ if (G_LIKELY (stream)) { GST_RTP_BIN_LOCK (rtpbin); /* fill in the percent */ stream->percent = percent; /* calculate the min value for all streams */ for (sessions = rtpbin->sessions; sessions; sessions = g_slist_next (sessions)) { GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; GST_RTP_SESSION_LOCK (session); if (session->streams) { for (streams = session->streams; streams; streams = g_slist_next (streams)) { GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream, stream->percent); /* find min percent */ if (min_percent > stream->percent) min_percent = stream->percent; } } else { GST_INFO_OBJECT (bin, "session has no streams, setting min_percent to 0"); min_percent = 0; } GST_RTP_SESSION_UNLOCK (session); } GST_DEBUG_OBJECT (bin, "min percent %d", min_percent); if (rtpbin->buffering) { if (min_percent == 100) { rtpbin->buffering = FALSE; active = TRUE; change = TRUE; } } else { if (min_percent < 100) { /* pause the streams */ rtpbin->buffering = TRUE; active = FALSE; change = TRUE; } } GST_RTP_BIN_UNLOCK (rtpbin); gst_message_unref (message); /* make a new buffering message with the min value */ message = gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent); gst_message_set_buffering_stats (message, mode, avg_in, avg_out, buffering_left); if (G_UNLIKELY (change)) { GstClock *clock; guint64 running_time = 0; guint64 offset = 0; /* figure out the running time when we have a clock */ if (G_LIKELY ((clock = gst_element_get_clock (GST_ELEMENT_CAST (bin))))) { guint64 now, base_time; now = gst_clock_get_time (clock); base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin)); running_time = now - base_time; gst_object_unref (clock); } GST_DEBUG_OBJECT (bin, "running time now %" GST_TIME_FORMAT, GST_TIME_ARGS (running_time)); GST_RTP_BIN_LOCK (rtpbin); /* when we reactivate, calculate the offsets so that all streams have * an output time that is at least as big as the running_time */ offset = 0; if (active) { if (running_time > rtpbin->buffer_start) { offset = running_time - rtpbin->buffer_start; if (offset >= rtpbin->latency_ns) offset -= rtpbin->latency_ns; else offset = 0; } } /* pause all streams */ min_out_time = -1; for (sessions = rtpbin->sessions; sessions; sessions = g_slist_next (sessions)) { GstRtpBinSession *session = (GstRtpBinSession *) sessions->data; GST_RTP_SESSION_LOCK (session); for (streams = session->streams; streams; streams = g_slist_next (streams)) { GstRtpBinStream *stream = (GstRtpBinStream *) streams->data; GstElement *element = stream->buffer; guint64 last_out; g_signal_emit_by_name (element, "set-active", active, offset, &last_out); if (!active) { g_object_get (element, "percent", &stream->percent, NULL); if (last_out == -1) last_out = 0; if (min_out_time == -1 || last_out < min_out_time) min_out_time = last_out; } GST_DEBUG_OBJECT (bin, "setting %p to %d, offset %" GST_TIME_FORMAT ", last %" GST_TIME_FORMAT ", percent %d", element, active, GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out), stream->percent); } GST_RTP_SESSION_UNLOCK (session); } GST_DEBUG_OBJECT (bin, "min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time)); /* the buffer_start is the min out time of all paused jitterbuffers */ if (!active) rtpbin->buffer_start = min_out_time; GST_RTP_BIN_UNLOCK (rtpbin); } } GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } default: { GST_BIN_CLASS (parent_class)->handle_message (bin, message); break; } } } static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn res; GstRtpBin *rtpbin; GstRtpBinPrivate *priv; rtpbin = GST_RTP_BIN (element); priv = rtpbin->priv; switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: priv->last_unix = 0; GST_LOG_OBJECT (rtpbin, "clearing shutdown flag"); g_atomic_int_set (&priv->shutdown, 0); break; case GST_STATE_CHANGE_PAUSED_TO_READY: GST_LOG_OBJECT (rtpbin, "setting shutdown flag"); g_atomic_int_set (&priv->shutdown, 1); /* wait for all callbacks to end by taking the lock. No new callbacks will * be able to happen as we set the shutdown flag. */ GST_RTP_BIN_DYN_LOCK (rtpbin); GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown"); GST_RTP_BIN_DYN_UNLOCK (rtpbin); break; default: break; } res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; } static GstElement * session_request_element (GstRtpBinSession * session, guint signal) { GstElement *element = NULL; GstRtpBin *bin = session->bin; g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element); if (element) { if (!bin_manage_element (bin, element)) goto manage_failed; session->elements = g_slist_prepend (session->elements, element); } return element; /* ERRORS */ manage_failed: { GST_WARNING_OBJECT (bin, "unable to manage element"); gst_object_unref (element); return NULL; } } static gboolean copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data) { GstPad *gpad = GST_PAD_CAST (user_data); GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event); gst_pad_store_sticky_event (gpad, *event); return TRUE; } /* a new pad (SSRC) was created in @session. This signal is emited from the * payload demuxer. */ static void new_payload_found (GstElement * element, guint pt, GstPad * pad, GstRtpBinStream * stream) { GstRtpBin *rtpbin; GstElementClass *klass; GstPadTemplate *templ; gchar *padname; GstPad *gpad; rtpbin = stream->bin; GST_DEBUG ("new payload pad %d", pt); GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown); /* ghost the pad to the parent */ klass = GST_ELEMENT_GET_CLASS (rtpbin); templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u"); padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u", stream->session->id, stream->ssrc, pt); gpad = gst_ghost_pad_new_from_template (padname, pad, templ); g_free (padname); g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad); gst_pad_set_active (gpad, TRUE); GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad); return; shutdown: { GST_DEBUG ("ignoring, we are shutting down"); return; } } static void payload_pad_removed (GstElement * element, GstPad * pad, GstRtpBinStream * stream) { GstRtpBin *rtpbin; GstPad *gpad; rtpbin = stream->bin; GST_DEBUG ("payload pad removed"); GST_RTP_BIN_DYN_LOCK (rtpbin); if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) { g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL); gst_pad_set_active (gpad, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad); } GST_RTP_BIN_DYN_UNLOCK (rtpbin); } static GstCaps * pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session) { GstRtpBin *rtpbin; GstCaps *caps; rtpbin = session->bin; GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %d in session %d", pt, session->id); caps = get_pt_map (session, pt); if (!caps) goto no_caps; return caps; /* ERRORS */ no_caps: { GST_DEBUG_OBJECT (rtpbin, "could not get caps"); return NULL; } } static void payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session) { GST_DEBUG_OBJECT (session->bin, "emiting signal for pt type changed to %d in session %d", pt, session->id); g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE], 0, session->id, pt); } /* emited when caps changed for the session */ static void caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session) { GstRtpBin *bin; GstCaps *caps; gint payload; const GstStructure *s; bin = session->bin; g_object_get (pad, "caps", &caps, NULL); if (caps == NULL) return; GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps); s = gst_caps_get_structure (caps, 0); /* get payload, finish when it's not there */ if (!gst_structure_get_int (s, "payload", &payload)) { gst_caps_unref (caps); return; } GST_RTP_SESSION_LOCK (session); GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload); g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps); GST_RTP_SESSION_UNLOCK (session); } /* a new pad (SSRC) was created in @session */ static void new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad, GstRtpBinSession * session) { GstRtpBin *rtpbin; GstRtpBinStream *stream; GstPad *sinkpad, *srcpad; gchar *padname; rtpbin = session->bin; GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc, GST_DEBUG_PAD_NAME (pad)); GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown); GST_RTP_SESSION_LOCK (session); /* create new stream */ stream = create_stream (session, ssrc); if (!stream) goto no_stream; /* get pad and link */ GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP"); padname = g_strdup_printf ("src_%u", ssrc); srcpad = gst_element_get_static_pad (element, padname); g_free (padname); sinkpad = gst_element_get_static_pad (stream->buffer, "sink"); gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (sinkpad); gst_object_unref (srcpad); GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP"); padname = g_strdup_printf ("rtcp_src_%u", ssrc); srcpad = gst_element_get_static_pad (element, padname); g_free (padname); sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp"); gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (sinkpad); gst_object_unref (srcpad); /* connect to the RTCP sync signal from the jitterbuffer */ GST_DEBUG_OBJECT (rtpbin, "connecting sync signal"); stream->buffer_handlesync_sig = g_signal_connect (stream->buffer, "handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream); if (stream->demux) { /* connect to the new-pad signal of the payload demuxer, this will expose the * new pad by ghosting it. */ stream->demux_newpad_sig = g_signal_connect (stream->demux, "new-payload-type", (GCallback) new_payload_found, stream); stream->demux_padremoved_sig = g_signal_connect (stream->demux, "pad-removed", (GCallback) payload_pad_removed, stream); /* connect to the request-pt-map signal. This signal will be emited by the * demuxer so that it can apply a proper caps on the buffers for the * depayloaders. */ stream->demux_ptreq_sig = g_signal_connect (stream->demux, "request-pt-map", (GCallback) pt_map_requested, session); /* connect to the signal so it can be forwarded. */ stream->demux_ptchange_sig = g_signal_connect (stream->demux, "payload-type-change", (GCallback) payload_type_change, session); } else { /* add rtpjitterbuffer src pad to pads */ GstElementClass *klass; GstPadTemplate *templ; gchar *padname; GstPad *gpad, *pad; pad = gst_element_get_static_pad (stream->buffer, "src"); /* ghost the pad to the parent */ klass = GST_ELEMENT_GET_CLASS (rtpbin); templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u"); padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u", stream->session->id, stream->ssrc, 255); gpad = gst_ghost_pad_new_from_template (padname, pad, templ); g_free (padname); gst_pad_set_active (gpad, TRUE); gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad); gst_object_unref (pad); } GST_RTP_SESSION_UNLOCK (session); GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); return; /* ERRORS */ shutdown: { GST_DEBUG_OBJECT (rtpbin, "we are shutting down"); return; } no_stream: { GST_RTP_SESSION_UNLOCK (session); GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin); GST_DEBUG_OBJECT (rtpbin, "could not create stream"); return; } } static gboolean complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session) { gchar *gname; guint sessid = session->id; GstPad *recv_rtp_sink; GstElement *decoder; GstElementClass *klass; GstPadTemplate *templ; /* get recv_rtp pad and store */ session->recv_rtp_sink = gst_element_get_request_pad (session->session, "recv_rtp_sink"); if (session->recv_rtp_sink == NULL) goto pad_failed; g_signal_connect (session->recv_rtp_sink, "notify::caps", (GCallback) caps_changed, session); GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder"); decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER); if (decoder) { GstPad *decsrc, *decsink; GstPadLinkReturn ret; GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder"); decsink = gst_element_get_static_pad (decoder, "rtp_sink"); if (decsink == NULL) goto dec_sink_failed; recv_rtp_sink = decsink; decsrc = gst_element_get_static_pad (decoder, "rtp_src"); if (decsrc == NULL) goto dec_src_failed; ret = gst_pad_link (decsrc, session->recv_rtp_sink); gst_object_unref (decsrc); if (ret != GST_PAD_LINK_OK) goto dec_link_failed; } else { GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given"); recv_rtp_sink = gst_object_ref (session->recv_rtp_sink); } GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad"); klass = GST_ELEMENT_GET_CLASS (rtpbin); gname = g_strdup_printf ("recv_rtp_sink_%u", sessid); templ = gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u"); session->recv_rtp_sink_ghost = gst_ghost_pad_new_from_template (gname, recv_rtp_sink, templ); gst_object_unref (recv_rtp_sink); gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost); g_free (gname); return TRUE; /* ERRORS */ pad_failed: { g_warning ("rtpbin: failed to get session recv_rtp_sink pad"); return FALSE; } dec_sink_failed: { g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid); return FALSE; } dec_src_failed: { g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid); gst_object_unref (recv_rtp_sink); return FALSE; } dec_link_failed: { g_warning ("rtpbin: failed to link rtp decoder for session %d", sessid); gst_object_unref (recv_rtp_sink); return FALSE; } } /* Create a pad for receiving RTP for the session in @name. Must be called with * RTP_BIN_LOCK. */ static GstPad * create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) { guint sessid; GstElement *aux; GstPad *recv_rtp_src; GstRtpBinSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1) goto no_name; GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid); /* get or create session */ session = find_session_by_id (rtpbin, sessid); if (!session) { GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid); /* create session now */ session = create_session (rtpbin, sessid); if (session == NULL) goto create_error; } /* check if pad was requested */ if (session->recv_rtp_sink_ghost != NULL) return session->recv_rtp_sink_ghost; /* setup the session sink pad */ if (!complete_session_sink (rtpbin, session)) goto session_sink_failed; session->recv_rtp_src = gst_element_get_static_pad (session->session, "recv_rtp_src"); if (session->recv_rtp_src == NULL) goto pad_failed; /* find out if we need AUX elements or if we can go into the SSRC demuxer * directly */ aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER); if (aux) { gchar *pname; GstPad *auxsink; GstPadLinkReturn ret; GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver"); pname = g_strdup_printf ("sink_%d", sessid); auxsink = gst_element_get_static_pad (aux, pname); g_free (pname); if (auxsink == NULL) goto aux_sink_failed; ret = gst_pad_link (session->recv_rtp_src, auxsink); gst_object_unref (auxsink); if (ret != GST_PAD_LINK_OK) goto aux_link_failed; /* this can be NULL when this AUX element is not to be linked to * an SSRC demuxer */ pname = g_strdup_printf ("src_%d", sessid); recv_rtp_src = gst_element_get_static_pad (aux, pname); g_free (pname); } else { recv_rtp_src = gst_object_ref (session->recv_rtp_src); } if (recv_rtp_src) { GstPad *sinkdpad; GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad"); sinkdpad = gst_element_get_static_pad (session->demux, "sink"); GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad"); gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (recv_rtp_src); gst_object_unref (sinkdpad); /* connect to the new-ssrc-pad signal of the SSRC demuxer */ session->demux_newpad_sig = g_signal_connect (session->demux, "new-ssrc-pad", (GCallback) new_ssrc_pad_found, session); session->demux_padremoved_sig = g_signal_connect (session->demux, "removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session); } return session->recv_rtp_sink_ghost; /* ERRORS */ no_name: { g_warning ("rtpbin: invalid name given"); return NULL; } create_error: { /* create_session already warned */ return NULL; } session_sink_failed: { /* warning already done */ return NULL; } pad_failed: { g_warning ("rtpbin: failed to get session recv_rtp_src pad"); return NULL; } aux_sink_failed: { g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid); return NULL; } aux_link_failed: { g_warning ("rtpbin: failed to link AUX pad to session %d", sessid); return NULL; } } static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session) { if (session->demux_newpad_sig) { g_signal_handler_disconnect (session->demux, session->demux_newpad_sig); session->demux_newpad_sig = 0; } if (session->demux_padremoved_sig) { g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig); session->demux_padremoved_sig = 0; } if (session->recv_rtp_src) { gst_object_unref (session->recv_rtp_src); session->recv_rtp_src = NULL; } if (session->recv_rtp_sink) { gst_element_release_request_pad (session->session, session->recv_rtp_sink); gst_object_unref (session->recv_rtp_sink); session->recv_rtp_sink = NULL; } if (session->recv_rtp_sink_ghost) { gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost); session->recv_rtp_sink_ghost = NULL; } } /* Create a pad for receiving RTCP for the session in @name. Must be called with * RTP_BIN_LOCK. */ static GstPad * create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) { guint sessid; GstElement *decoder; GstRtpBinSession *session; GstPad *sinkdpad, *decsink; /* first get the session number */ if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1) goto no_name; GST_DEBUG_OBJECT (rtpbin, "finding session %d", sessid); /* get or create the session */ session = find_session_by_id (rtpbin, sessid); if (!session) { GST_DEBUG_OBJECT (rtpbin, "creating session %d", sessid); /* create session now */ session = create_session (rtpbin, sessid); if (session == NULL) goto create_error; } /* check if pad was requested */ if (session->recv_rtcp_sink_ghost != NULL) return session->recv_rtcp_sink_ghost; /* get recv_rtp pad and store */ GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad"); session->recv_rtcp_sink = gst_element_get_request_pad (session->session, "recv_rtcp_sink"); if (session->recv_rtcp_sink == NULL) goto pad_failed; GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder"); decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER); if (decoder) { GstPad *decsrc; GstPadLinkReturn ret; GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder"); decsink = gst_element_get_static_pad (decoder, "rtcp_sink"); decsrc = gst_element_get_static_pad (decoder, "rtcp_src"); if (decsink == NULL) goto dec_sink_failed; if (decsrc == NULL) goto dec_src_failed; ret = gst_pad_link (decsrc, session->recv_rtcp_sink); gst_object_unref (decsrc); if (ret != GST_PAD_LINK_OK) goto dec_link_failed; } else { GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given"); decsink = gst_object_ref (session->recv_rtcp_sink); } /* get srcpad, link to SSRCDemux */ GST_DEBUG_OBJECT (rtpbin, "getting sync src pad"); session->sync_src = gst_element_get_static_pad (session->session, "sync_src"); if (session->sync_src == NULL) goto src_pad_failed; GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad"); sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink"); gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING); gst_object_unref (sinkdpad); session->recv_rtcp_sink_ghost = gst_ghost_pad_new_from_template (name, decsink, templ); gst_object_unref (decsink); gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtcp_sink_ghost); return session->recv_rtcp_sink_ghost; /* ERRORS */ no_name: { g_warning ("rtpbin: invalid name given"); return NULL; } create_error: { /* create_session already warned */ return NULL; } pad_failed: { g_warning ("rtpbin: failed to get session rtcp_sink pad"); return NULL; } dec_sink_failed: { g_warning ("rtpbin: failed to get decoder sink pad for session %d", sessid); return NULL; } dec_src_failed: { g_warning ("rtpbin: failed to get decoder src pad for session %d", sessid); gst_object_unref (decsink); return NULL; } dec_link_failed: { g_warning ("rtpbin: failed to link rtcp decoder for session %d", sessid); gst_object_unref (decsink); return NULL; } src_pad_failed: { g_warning ("rtpbin: failed to get session sync_src pad"); gst_object_unref (decsink); return NULL; } } static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session) { if (session->recv_rtcp_sink_ghost) { gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtcp_sink_ghost); session->recv_rtcp_sink_ghost = NULL; } if (session->sync_src) { /* releasing the request pad should also unref the sync pad */ gst_object_unref (session->sync_src); session->sync_src = NULL; } if (session->recv_rtcp_sink) { gst_element_release_request_pad (session->session, session->recv_rtcp_sink); gst_object_unref (session->recv_rtcp_sink); session->recv_rtcp_sink = NULL; } } static gboolean complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session) { gchar *gname; guint sessid = session->id; GstPad *send_rtp_src; GstElement *encoder; GstElementClass *klass; GstPadTemplate *templ; /* get srcpad */ session->send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src"); if (session->send_rtp_src == NULL) goto no_srcpad; GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder"); encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER); if (encoder) { gchar *ename; GstPad *encsrc, *encsink; GstPadLinkReturn ret; GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder"); ename = g_strdup_printf ("rtp_src_%d", sessid); encsrc = gst_element_get_static_pad (encoder, ename); g_free (ename); if (encsrc == NULL) goto enc_src_failed; send_rtp_src = encsrc; ename = g_strdup_printf ("rtp_sink_%d", sessid); encsink = gst_element_get_static_pad (encoder, ename); g_free (ename); if (encsink == NULL) goto enc_sink_failed; ret = gst_pad_link (session->send_rtp_src, encsink); gst_object_unref (encsink); if (ret != GST_PAD_LINK_OK) goto enc_link_failed; } else { GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given"); send_rtp_src = gst_object_ref (session->send_rtp_src); } /* ghost the new source pad */ klass = GST_ELEMENT_GET_CLASS (rtpbin); gname = g_strdup_printf ("send_rtp_src_%u", sessid); templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u"); session->send_rtp_src_ghost = gst_ghost_pad_new_from_template (gname, send_rtp_src, templ); gst_object_unref (send_rtp_src); gst_pad_set_active (session->send_rtp_src_ghost, TRUE); gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events, session->send_rtp_src_ghost); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost); g_free (gname); return TRUE; /* ERRORS */ no_srcpad: { g_warning ("rtpbin: failed to get rtp source pad for session %d", sessid); return FALSE; } enc_src_failed: { g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid); return FALSE; } enc_sink_failed: { g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid); gst_object_unref (send_rtp_src); return FALSE; } enc_link_failed: { g_warning ("rtpbin: failed to link rtp encoder for session %d", sessid); gst_object_unref (send_rtp_src); return FALSE; } } static gboolean setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data) { GstPad *pad; gchar *name; guint sessid; GstRtpBinSession *session = user_data, *newsess; GstRtpBin *rtpbin = session->bin; GstPadLinkReturn ret; pad = g_value_get_object (item); name = gst_pad_get_name (pad); if (name == NULL || sscanf (name, "src_%u", &sessid) != 1) goto no_name; g_free (name); newsess = find_session_by_id (rtpbin, sessid); if (newsess == NULL) { /* create new session */ newsess = create_session (rtpbin, sessid); if (newsess == NULL) goto create_error; } else if (newsess->send_rtp_sink != NULL) goto existing_session; /* get send_rtp pad and store */ newsess->send_rtp_sink = gst_element_get_request_pad (newsess->session, "send_rtp_sink"); if (newsess->send_rtp_sink == NULL) goto pad_failed; ret = gst_pad_link (pad, newsess->send_rtp_sink); if (ret != GST_PAD_LINK_OK) goto aux_link_failed; if (!complete_session_src (rtpbin, newsess)) goto session_src_failed; return TRUE; /* ERRORS */ no_name: { GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name)); g_free (name); return TRUE; } create_error: { /* create_session already warned */ return FALSE; } existing_session: { g_warning ("rtpbin: session %d is already a sender", sessid); return FALSE; } pad_failed: { g_warning ("rtpbin: failed to get session pad for session %d", sessid); return FALSE; } aux_link_failed: { g_warning ("rtpbin: failed to link AUX for session %d", sessid); return FALSE; } session_src_failed: { g_warning ("rtpbin: failed to complete AUX for session %d", sessid); return FALSE; } } static gboolean setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session, GstElement * aux) { GstIterator *it; GValue result = { 0, }; GstIteratorResult res; it = gst_element_iterate_src_pads (aux); res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session); gst_iterator_free (it); return res == GST_ITERATOR_DONE; } /* Create a pad for sending RTP for the session in @name. Must be called with * RTP_BIN_LOCK. */ static GstPad * create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) { gchar *pname; guint sessid; GstPad *send_rtp_sink; GstElement *aux; GstRtpBinSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1) goto no_name; /* get or create session */ session = find_session_by_id (rtpbin, sessid); if (!session) { /* create session now */ session = create_session (rtpbin, sessid); if (session == NULL) goto create_error; } /* check if pad was requested */ if (session->send_rtp_sink_ghost != NULL) return session->send_rtp_sink_ghost; /* check if we are already using this session as a sender */ if (session->send_rtp_sink != NULL) goto existing_session; GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender"); aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER); if (aux) { GST_DEBUG_OBJECT (rtpbin, "linking AUX sender"); if (!setup_aux_sender (rtpbin, session, aux)) goto aux_session_failed; pname = g_strdup_printf ("sink_%d", sessid); send_rtp_sink = gst_element_get_static_pad (aux, pname); g_free (pname); if (send_rtp_sink == NULL) goto aux_sink_failed; } else { /* get send_rtp pad and store */ session->send_rtp_sink = gst_element_get_request_pad (session->session, "send_rtp_sink"); if (session->send_rtp_sink == NULL) goto pad_failed; if (!complete_session_src (rtpbin, session)) goto session_src_failed; send_rtp_sink = gst_object_ref (session->send_rtp_sink); } session->send_rtp_sink_ghost = gst_ghost_pad_new_from_template (name, send_rtp_sink, templ); gst_object_unref (send_rtp_sink); gst_pad_set_active (session->send_rtp_sink_ghost, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost); return session->send_rtp_sink_ghost; /* ERRORS */ no_name: { g_warning ("rtpbin: invalid name given"); return NULL; } create_error: { /* create_session already warned */ return NULL; } existing_session: { g_warning ("rtpbin: session %d is already in use", sessid); return NULL; } aux_session_failed: { g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid); return NULL; } aux_sink_failed: { g_warning ("rtpbin: failed to get AUX sink pad for session %d", sessid); return NULL; } pad_failed: { g_warning ("rtpbin: failed to get session pad for session %d", sessid); return NULL; } session_src_failed: { g_warning ("rtpbin: failed to setup source pads for session %d", sessid); return NULL; } } static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session) { if (session->send_rtp_src_ghost) { gst_pad_set_active (session->send_rtp_src_ghost, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost); session->send_rtp_src_ghost = NULL; } if (session->send_rtp_src) { gst_object_unref (session->send_rtp_src); session->send_rtp_src = NULL; } if (session->send_rtp_sink) { gst_element_release_request_pad (GST_ELEMENT_CAST (session->session), session->send_rtp_sink); gst_object_unref (session->send_rtp_sink); session->send_rtp_sink = NULL; } if (session->send_rtp_sink_ghost) { gst_pad_set_active (session->send_rtp_sink_ghost, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost); session->send_rtp_sink_ghost = NULL; } } /* Create a pad for sending RTCP for the session in @name. Must be called with * RTP_BIN_LOCK. */ static GstPad * create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name) { guint sessid; GstPad *encsrc; GstElement *encoder; GstRtpBinSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1) goto no_name; /* get or create session */ session = find_session_by_id (rtpbin, sessid); if (!session) goto no_session; /* check if pad was requested */ if (session->send_rtcp_src_ghost != NULL) return session->send_rtcp_src_ghost; /* get rtcp_src pad and store */ session->send_rtcp_src = gst_element_get_request_pad (session->session, "send_rtcp_src"); if (session->send_rtcp_src == NULL) goto pad_failed; GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder"); encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER); if (encoder) { gchar *ename; GstPad *encsink; GstPadLinkReturn ret; GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder"); ename = g_strdup_printf ("rtcp_src_%d", sessid); encsrc = gst_element_get_static_pad (encoder, ename); g_free (ename); if (encsrc == NULL) goto enc_src_failed; ename = g_strdup_printf ("rtcp_sink_%d", sessid); encsink = gst_element_get_static_pad (encoder, ename); g_free (ename); if (encsink == NULL) goto enc_sink_failed; ret = gst_pad_link (session->send_rtcp_src, encsink); gst_object_unref (encsink); if (ret != GST_PAD_LINK_OK) goto enc_link_failed; } else { GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given"); encsrc = gst_object_ref (session->send_rtcp_src); } session->send_rtcp_src_ghost = gst_ghost_pad_new_from_template (name, encsrc, templ); gst_object_unref (encsrc); gst_pad_set_active (session->send_rtcp_src_ghost, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost); return session->send_rtcp_src_ghost; /* ERRORS */ no_name: { g_warning ("rtpbin: invalid name given"); return NULL; } no_session: { g_warning ("rtpbin: session with id %d does not exist", sessid); return NULL; } pad_failed: { g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid); return NULL; } enc_src_failed: { g_warning ("rtpbin: failed to get encoder src pad for session %d", sessid); return NULL; } enc_sink_failed: { g_warning ("rtpbin: failed to get encoder sink pad for session %d", sessid); gst_object_unref (encsrc); return NULL; } enc_link_failed: { g_warning ("rtpbin: failed to link rtcp encoder for session %d", sessid); gst_object_unref (encsrc); return NULL; } } static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session) { if (session->send_rtcp_src_ghost) { gst_pad_set_active (session->send_rtcp_src_ghost, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost); session->send_rtcp_src_ghost = NULL; } if (session->send_rtcp_src) { gst_element_release_request_pad (session->session, session->send_rtcp_src); gst_object_unref (session->send_rtcp_src); session->send_rtcp_src = NULL; } } /* If the requested name is NULL we should create a name with * the session number assuming we want the lowest posible session * with a free pad like the template */ static gchar * gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ) { gboolean name_found = FALSE; gint session = 0; GstIterator *pad_it = NULL; gchar *pad_name = NULL; GValue data = { 0, }; GST_DEBUG_OBJECT (element, "find a free pad name for template"); while (!name_found) { gboolean done = FALSE; g_free (pad_name); pad_name = g_strdup_printf (templ->name_template, session++); pad_it = gst_element_iterate_pads (GST_ELEMENT (element)); name_found = TRUE; while (!done) { switch (gst_iterator_next (pad_it, &data)) { case GST_ITERATOR_OK: { GstPad *pad; gchar *name; pad = g_value_get_object (&data); name = gst_pad_get_name (pad); if (strcmp (name, pad_name) == 0) { done = TRUE; name_found = FALSE; } g_free (name); g_value_reset (&data); break; } case GST_ITERATOR_ERROR: case GST_ITERATOR_RESYNC: /* restart iteration */ done = TRUE; name_found = FALSE; session = 0; break; case GST_ITERATOR_DONE: done = TRUE; break; } } g_value_unset (&data); gst_iterator_free (pad_it); } GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name); return pad_name; } /* */ static GstPad * gst_rtp_bin_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps) { GstRtpBin *rtpbin; GstElementClass *klass; GstPad *result; gchar *pad_name = NULL; g_return_val_if_fail (templ != NULL, NULL); g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL); rtpbin = GST_RTP_BIN (element); klass = GST_ELEMENT_GET_CLASS (element); GST_RTP_BIN_LOCK (rtpbin); if (name == NULL) { /* use a free pad name */ pad_name = gst_rtp_bin_get_free_pad_name (element, templ); } else { /* use the provided name */ pad_name = g_strdup (name); } GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name); /* figure out the template */ if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) { result = create_recv_rtp (rtpbin, templ, pad_name); } else if (templ == gst_element_class_get_pad_template (klass, "recv_rtcp_sink_%u")) { result = create_recv_rtcp (rtpbin, templ, pad_name); } else if (templ == gst_element_class_get_pad_template (klass, "send_rtp_sink_%u")) { result = create_send_rtp (rtpbin, templ, pad_name); } else if (templ == gst_element_class_get_pad_template (klass, "send_rtcp_src_%u")) { result = create_rtcp (rtpbin, templ, pad_name); } else goto wrong_template; g_free (pad_name); GST_RTP_BIN_UNLOCK (rtpbin); return result; /* ERRORS */ wrong_template: { g_free (pad_name); GST_RTP_BIN_UNLOCK (rtpbin); g_warning ("rtpbin: this is not our template"); return NULL; } } static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad) { GstRtpBinSession *session; GstRtpBin *rtpbin; g_return_if_fail (GST_IS_GHOST_PAD (pad)); g_return_if_fail (GST_IS_RTP_BIN (element)); rtpbin = GST_RTP_BIN (element); GST_RTP_BIN_LOCK (rtpbin); GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s", GST_DEBUG_PAD_NAME (pad)); if (!(session = find_session_by_pad (rtpbin, pad))) goto unknown_pad; if (session->recv_rtp_sink_ghost == pad) { remove_recv_rtp (rtpbin, session); } else if (session->recv_rtcp_sink_ghost == pad) { remove_recv_rtcp (rtpbin, session); } else if (session->send_rtp_sink_ghost == pad) { remove_send_rtp (rtpbin, session); } else if (session->send_rtcp_src_ghost == pad) { remove_rtcp (rtpbin, session); } /* no more request pads, free the complete session */ if (session->recv_rtp_sink_ghost == NULL && session->recv_rtcp_sink_ghost == NULL && session->send_rtp_sink_ghost == NULL && session->send_rtcp_src_ghost == NULL) { GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session); rtpbin->sessions = g_slist_remove (rtpbin->sessions, session); free_session (session, rtpbin); } GST_RTP_BIN_UNLOCK (rtpbin); return; /* ERROR */ unknown_pad: { GST_RTP_BIN_UNLOCK (rtpbin); g_warning ("rtpbin: %s:%s is not one of our request pads", GST_DEBUG_PAD_NAME (pad)); return; } }