/* 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-rtpsession * @title: rtpsession * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux * * The RTP session manager models participants with unique SSRC in an RTP * session. This session can be used to send and receive RTP and RTCP packets. * Based on what REQUEST pads are requested from the session manager, specific * functionality can be activated. * * The session manager currently implements RFC 3550 including: * * * RTP packet validation based on consecutive sequence numbers. * * * Maintenance of the SSRC participant database. * * * Keeping per participant statistics based on received RTCP packets. * * * Scheduling of RR/SR RTCP packets. * * * Support for multiple sender SSRC. * * The rtpsession will not demux packets based on SSRC or payload type, nor will * it correct for packet reordering and jitter. Use #GstRtpSsrcDemux, * #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to * perform these tasks. It is usually a good idea to use #GstRtpBin, which * combines all these features in one element. * * To use #GstRtpSession as an RTP receiver, request a recv_rtp_sink pad, which will * automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad * will be processed in the session and after being validated forwarded on the * recv_rtp_src pad. * * To also use #GstRtpSession as an RTCP receiver, request a recv_rtcp_sink pad, * which will automatically create a sync_src pad. Packets received on the RTCP * pad will be used by the session manager to update the stats and database of * the other participants. SR packets will be forwarded on the sync_src pad * so that they can be used to perform inter-stream synchronisation when needed. * * If you want the session manager to generate and send RTCP packets, request * the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports * that should be sent to all participants in the session. * * To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will * automatically create a send_rtp_src pad. The session manager will * forward the packets on the send_rtp_src 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. * * ## Example pipelines * |[ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink * ]| Receive theora RTP packets from port 5000 and send them to the depayloader, * decoder and display. Note that the application/x-rtp caps on udpsrc should be * configured based on some negotiation process such as RTSP for this pipeline * to work correctly. * |[ * gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \ * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \ * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink * ]| Receive theora RTP packets from port 5000 and send them to the depayloader, * decoder and display. Receive RTCP packets from port 5001 and process them in * the session manager. * Note that the application/x-rtp caps on udpsrc should be * configured based on some negotiation process such as RTSP for this pipeline * to work correctly. * |[ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000 * ]| Send theora RTP packets through the session manager and out on UDP port * 5000. * |[ * gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \ * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001 * ]| Send theora RTP packets through the session manager and out on UDP port * 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll * correctly because the second udpsink will not preroll correctly (no RTCP * packets are sent in the PAUSED state). Applications should manually set and * keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include "gstrtpsession.h" #include "rtpsession.h" #include "gstrtputils.h" GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug); #define GST_CAT_DEFAULT gst_rtp_session_debug #define GST_TYPE_RTP_NTP_TIME_SOURCE (gst_rtp_ntp_time_source_get_type ()) GType gst_rtp_ntp_time_source_get_type (void) { static GType type = 0; static const GEnumValue values[] = { {GST_RTP_NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"}, {GST_RTP_NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"}, {GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME, "Running time based on pipeline clock", "running-time"}, {GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"}, {0, NULL, NULL}, }; if (!type) { type = g_enum_register_static ("GstRtpNtpTimeSource", values); } return type; } /* sink pads */ static GstStaticPadTemplate rtpsession_recv_rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp") ); static GstStaticPadTemplate rtpsession_send_rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("send_rtp_sink", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp") ); /* src pads */ static GstStaticPadTemplate rtpsession_recv_rtp_src_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_src", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate rtpsession_sync_src_template = GST_STATIC_PAD_TEMPLATE ("sync_src", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtcp") ); static GstStaticPadTemplate rtpsession_send_rtp_src_template = GST_STATIC_PAD_TEMPLATE ("send_rtp_src", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate rtpsession_send_rtcp_src_template = GST_STATIC_PAD_TEMPLATE ("send_rtcp_src", GST_PAD_SRC, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp") ); /* signals and args */ enum { SIGNAL_REQUEST_PT_MAP, SIGNAL_CLEAR_PT_MAP, 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_NEW_SENDER_SSRC, SIGNAL_ON_SENDER_SSRC_ACTIVE, LAST_SIGNAL }; #define DEFAULT_BANDWIDTH 0 #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION #define DEFAULT_RTCP_RR_BANDWIDTH -1 #define DEFAULT_RTCP_RS_BANDWIDTH -1 #define DEFAULT_SDES NULL #define DEFAULT_NUM_SOURCES 0 #define DEFAULT_NUM_ACTIVE_SOURCES 0 #define DEFAULT_USE_PIPELINE_CLOCK FALSE #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION #define DEFAULT_MAX_DROPOUT_TIME 60000 #define DEFAULT_MAX_MISORDER_TIME 2000 #define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP #define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP #define DEFAULT_RTCP_SYNC_SEND_TIME TRUE #define DEFAULT_UPDATE_NTP64_HEADER_EXT TRUE #define DEFAULT_TIMEOUT_INACTIVE_SOURCES TRUE enum { PROP_0, PROP_BANDWIDTH, PROP_RTCP_FRACTION, PROP_RTCP_RR_BANDWIDTH, PROP_RTCP_RS_BANDWIDTH, PROP_SDES, PROP_NUM_SOURCES, PROP_NUM_ACTIVE_SOURCES, PROP_INTERNAL_SESSION, PROP_USE_PIPELINE_CLOCK, PROP_RTCP_MIN_INTERVAL, PROP_PROBATION, PROP_MAX_DROPOUT_TIME, PROP_MAX_MISORDER_TIME, PROP_STATS, PROP_TWCC_STATS, PROP_RTP_PROFILE, PROP_NTP_TIME_SOURCE, PROP_RTCP_SYNC_SEND_TIME, PROP_UPDATE_NTP64_HEADER_EXT, PROP_TIMEOUT_INACTIVE_SOURCES, }; #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->priv->lock) #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->priv->lock) #define GST_RTP_SESSION_WAIT(sess) g_cond_wait (&(sess)->priv->cond, &(sess)->priv->lock) #define GST_RTP_SESSION_SIGNAL(sess) g_cond_signal (&(sess)->priv->cond) struct _GstRtpSessionPrivate { GMutex lock; GCond cond; GstClock *sysclock; RTPSession *session; /* thread for sending out RTCP */ GstClockID id; gboolean stop_thread; GThread *thread; gboolean thread_stopped; gboolean wait_send; /* caps mapping */ GHashTable *ptmap; GstClockTime send_latency; /* Set if we warned once already that no latency is configured yet but we * need it to calculate correct send running time of the packets */ gboolean warned_latency_once; gboolean use_pipeline_clock; GstRtpNtpTimeSource ntp_time_source; gboolean rtcp_sync_send_time; guint recv_rtx_req_count; guint sent_rtx_req_count; GstStructure *last_twcc_stats; /* * This is the list of processed packets in the receive path when upstream * pushed a buffer list. */ GstBufferList *processed_list; gboolean send_rtp_sink_eos; }; /* callbacks to handle actions from the session manager */ static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gpointer user_data); static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src, gpointer data, gpointer user_data); static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data); static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess, GstBuffer * buffer, gpointer user_data); static GstCaps *gst_rtp_session_caps (RTPSession * sess, guint8 payload, gpointer user_data); static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data); static void gst_rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean all_headers, gpointer user_data); static GstClockTime gst_rtp_session_request_time (RTPSession * session, gpointer user_data); static void gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data); static void gst_rtp_session_notify_twcc (RTPSession * sess, GstStructure * twcc_packets, GstStructure * twcc_stats, gpointer user_data); static void gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data); static void gst_rtp_session_notify_early_rtcp (RTPSession * sess, gpointer user_data); static GstFlowReturn gst_rtp_session_chain_recv_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_session_chain_recv_rtp_list (GstPad * pad, GstObject * parent, GstBufferList * list); static GstFlowReturn gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_session_chain_send_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstObject * parent, GstBufferList * list); static RTPSessionCallbacks callbacks = { gst_rtp_session_process_rtp, gst_rtp_session_send_rtp, gst_rtp_session_sync_rtcp, gst_rtp_session_send_rtcp, gst_rtp_session_caps, gst_rtp_session_reconsider, gst_rtp_session_request_key_unit, gst_rtp_session_request_time, gst_rtp_session_notify_nack, gst_rtp_session_notify_twcc, gst_rtp_session_reconfigure, gst_rtp_session_notify_early_rtcp }; /* GObject vmethods */ static void gst_rtp_session_finalize (GObject * object); static void gst_rtp_session_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); /* GstElement vmethods */ static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element, GstStateChange transition); static GstPad *gst_rtp_session_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps); static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad); static gboolean gst_rtp_session_setcaps_recv_rtp (GstPad * pad, GstRtpSession * rtpsession, GstCaps * caps); static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession, GstCaps * caps); static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession); static GstStructure *gst_rtp_session_create_stats (GstRtpSession * rtpsession); static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; static void on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, src->ssrc); } static void on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess) { GstPad *send_rtp_sink; g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0, src->ssrc); GST_RTP_SESSION_LOCK (sess); if ((send_rtp_sink = sess->send_rtp_sink)) gst_object_ref (send_rtp_sink); GST_RTP_SESSION_UNLOCK (sess); if (send_rtp_sink) { GstStructure *structure; GstEvent *event; RTPSource *internal_src; guint32 suggested_ssrc; structure = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT, (guint) src->ssrc, NULL); /* if there is no source using the suggested ssrc, most probably because * this ssrc has just collided, suggest upstream to use it */ suggested_ssrc = rtp_session_suggest_ssrc (session, NULL); internal_src = rtp_session_get_source_by_ssrc (session, suggested_ssrc); if (!internal_src) gst_structure_set (structure, "suggested-ssrc", G_TYPE_UINT, (guint) suggested_ssrc, NULL); else g_object_unref (internal_src); event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure); gst_pad_push_event (send_rtp_sink, event); gst_object_unref (send_rtp_sink); } } static void on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0, src->ssrc); } static void on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, src->ssrc); } static void on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess) { GstStructure *s; GstMessage *m; /* convert the new SDES info into a message */ RTP_SESSION_LOCK (session); g_object_get (src, "sdes", &s, NULL); RTP_SESSION_UNLOCK (session); m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s); gst_element_post_message (GST_ELEMENT_CAST (sess), m); g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, src->ssrc); } static void on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, src->ssrc); } static void on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, src->ssrc); } static void on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, src->ssrc); } static void on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, src->ssrc); } static void on_new_sender_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0, src->ssrc); } static void on_sender_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0, src->ssrc); } static void on_notify_stats (RTPSession * session, GParamSpec * spec, GstRtpSession * rtpsession) { g_object_notify (G_OBJECT (rtpsession), "stats"); } #define gst_rtp_session_parent_class parent_class G_DEFINE_TYPE_WITH_PRIVATE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT); GST_ELEMENT_REGISTER_DEFINE (rtpsession, "rtpsession", GST_RANK_NONE, GST_TYPE_RTP_SESSION); static void gst_rtp_session_class_init (GstRtpSessionClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->finalize = gst_rtp_session_finalize; gobject_class->set_property = gst_rtp_session_set_property; gobject_class->get_property = gst_rtp_session_get_property; /** * GstRtpSession::request-pt-map: * @sess: the object which received the signal * @pt: the pt * * Request the payload type as #GstCaps for @pt. */ gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] = g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, request_pt_map), NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT); /** * GstRtpSession::clear-pt-map: * @sess: the object which received the signal * * Clear the cached pt-maps requested with #GstRtpSession::request-pt-map. */ gst_rtp_session_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 (GstRtpSessionClass, clear_pt_map), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRtpSession::on-new-ssrc: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of a new SSRC that entered @session. */ gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC] = g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-ssrc_collision: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify when we have an SSRC collision */ gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] = g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_collision), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-ssrc_validated: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of a new SSRC that became validated. */ gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] = g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_validated), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-ssrc-active: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of a SSRC that is active, i.e., sending RTCP. */ gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] = g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_active), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-ssrc-sdes: * @session: the object which received the signal * @src: the SSRC * * Notify that a new SDES was received for SSRC. */ gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES] = g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_sdes), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-bye-ssrc: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of an SSRC that became inactive because of a BYE packet. */ gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC] = g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_ssrc), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-bye-timeout: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of an SSRC that has timed out because of BYE */ gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] = g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_timeout), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-timeout: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of an SSRC that has timed out */ gst_rtp_session_signals[SIGNAL_ON_TIMEOUT] = g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-sender-timeout: * @sess: the object which received the signal * @ssrc: the SSRC * * Notify of a sender SSRC that has timed out and became a receiver */ gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] = g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_sender_timeout), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-new-sender-ssrc: * @sess: the object which received the signal * @ssrc: the sender SSRC * * Notify of a new sender SSRC that entered @session. * * Since: 1.8 */ gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] = g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); /** * GstRtpSession::on-sender-ssrc-active: * @sess: the object which received the signal * @ssrc: the sender SSRC * * Notify of a sender SSRC that is active, i.e., sending RTCP. * * Since: 1.8 */ gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] = g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_active), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT); g_object_class_install_property (gobject_class, PROP_BANDWIDTH, g_param_spec_double ("bandwidth", "Bandwidth", "The bandwidth of the session in bytes per second (0 for auto-discover)", 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION, g_param_spec_double ("rtcp-fraction", "RTCP Fraction", "The RTCP bandwidth of the session in bytes per second " "(or as a real fraction of the RTP bandwidth if < 1.0)", 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH, g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth", "The RTCP bandwidth used for receivers in bytes per second (-1 = default)", -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH, g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth", "The RTCP bandwidth used for senders in bytes per second (-1 = default)", -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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 | GST_PARAM_DOC_SHOW_DEFAULT)); g_object_class_install_property (gobject_class, PROP_NUM_SOURCES, g_param_spec_uint ("num-sources", "Num Sources", "The number of sources in the session", 0, G_MAXUINT, DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES, g_param_spec_uint ("num-active-sources", "Num Active Sources", "The number of active sources in the session", 0, G_MAXUINT, DEFAULT_NUM_ACTIVE_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION, g_param_spec_object ("internal-session", "Internal Session", "The internal RTPSession object", RTP_TYPE_SESSION, G_PARAM_READABLE | 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 " "(DEPRECATED: Use ntp-time-source property)", DEFAULT_USE_PIPELINE_CLOCK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED)); g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL, g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval", "Minimum interval between Regular RTCP packet (in ns)", 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROBATION, g_param_spec_uint ("probation", "Number of probations", "Consecutive packet sequence numbers to accept the source", 0, G_MAXUINT, DEFAULT_PROBATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME, g_param_spec_uint ("max-dropout-time", "Max dropout time", "The maximum time (milliseconds) of missing packets tolerated.", 0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME, g_param_spec_uint ("max-misorder-time", "Max misorder time", "The maximum time (milliseconds) of misordered packets tolerated.", 0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSession:stats: * * Various session statistics. This property returns a #GstStructure * with name `application/x-rtp-session-stats` with the following fields: * * * "recv-rtx-req-count" G_TYPE_UINT The number of retransmission events * received from downstream (in receiver mode) (Since 1.16) * * "sent-rtx-req-count" G_TYPE_UINT The number of retransmission events * sent downstream (in sender mode) (Since 1.16) * * "rtx-count" G_TYPE_UINT DEPRECATED Since 1.16, same as * "recv-rtx-req-count". * * "rtx-drop-count" G_TYPE_UINT The number of retransmission events * dropped (due to bandwidth constraints) * * "sent-nack-count" G_TYPE_UINT Number of NACKs sent * * "recv-nack-count" G_TYPE_UINT Number of NACKs received * * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource:stats for all * RTP sources (Since 1.8) * * Since: 1.4 */ g_object_class_install_property (gobject_class, PROP_STATS, g_param_spec_boxed ("stats", "Statistics", "Various statistics", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSession:twcc-stats: * * Various statistics derived from TWCC. This property returns a GstStructure * with name RTPTWCCStats with the following fields: * * "bitrate-sent" G_TYPE_UINT The actual sent bitrate of TWCC packets * "bitrate-recv" G_TYPE_UINT The estimated bitrate for the receiver. * "packets-sent" G_TYPE_UINT Number of packets sent * "packets-recv" G_TYPE_UINT Number of packets reported recevied * "packet-loss-pct" G_TYPE_DOUBLE Packetloss percentage, based on * packets reported as lost from the receiver. Note: depending on the * implementation of the receiver and due to the nature of the TWCC * RRs being sent with high frequency, out of order packets may not * be fully accounted for and this number could be higher than other * measurement sources of packet loss. * "avg-delta-of-delta", G_TYPE_INT64 In nanoseconds, a moving window * average of the difference in inter-packet spacing between * sender and receiver. A sudden increase in this number can indicate * network congestion. * * Since: 1.18 */ g_object_class_install_property (gobject_class, PROP_TWCC_STATS, g_param_spec_boxed ("twcc-stats", "TWCC Statistics", "Various statistics from TWCC", GST_TYPE_STRUCTURE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTP_PROFILE, g_param_spec_enum ("rtp-profile", "RTP Profile", "RTP profile to use", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE, g_param_spec_enum ("ntp-time-source", "NTP Time Source", "NTP time source for RTCP packets", GST_TYPE_RTP_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME, g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time", "Use send time or capture time for RTCP sync " "(TRUE = send time, FALSE = capture time)", DEFAULT_RTCP_SYNC_SEND_TIME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSession:update-ntp64-header-ext: * * Whether RTP NTP header extension should be updated with actual * NTP time. If not, use the NTP time from buffer timestamp metadata * * Since: 1.22 */ g_object_class_install_property (gobject_class, PROP_UPDATE_NTP64_HEADER_EXT, g_param_spec_boolean ("update-ntp64-header-ext", "Update NTP-64 RTP Header Extension", "Whether RTP NTP header extension should be updated with actual NTP time", DEFAULT_UPDATE_NTP64_HEADER_EXT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRtpSession:timeout-inactive-sources: * * Whether inactive sources should be timed out * * Since: 1.24 */ g_object_class_install_property (gobject_class, PROP_TIMEOUT_INACTIVE_SOURCES, g_param_spec_boolean ("timeout-inactive-sources", "Time out inactive sources", "Whether sources that don't receive RTP or RTCP packets for longer " "than 5x RTCP interval should be removed", DEFAULT_TIMEOUT_INACTIVE_SOURCES, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_session_change_state); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad); klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map); /* sink pads */ gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_recv_rtp_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_recv_rtcp_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_send_rtp_sink_template); /* src pads */ gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_recv_rtp_src_template); gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_sync_src_template); gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_send_rtp_src_template); gst_element_class_add_static_pad_template (gstelement_class, &rtpsession_send_rtcp_src_template); gst_element_class_set_static_metadata (gstelement_class, "RTP Session", "Filter/Network/RTP", "Implement an RTP session", "Wim Taymans "); GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug, "rtpsession", 0, "RTP Session"); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp_list); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtcp); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp); GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp_list); gst_type_mark_as_plugin_api (GST_TYPE_RTP_NTP_TIME_SOURCE, 0); gst_type_mark_as_plugin_api (RTP_TYPE_SESSION, 0); gst_type_mark_as_plugin_api (RTP_TYPE_SOURCE, 0); } static void gst_rtp_session_init (GstRtpSession * rtpsession) { rtpsession->priv = gst_rtp_session_get_instance_private (rtpsession); g_mutex_init (&rtpsession->priv->lock); g_cond_init (&rtpsession->priv->cond); rtpsession->priv->sysclock = gst_system_clock_obtain (); rtpsession->priv->session = rtp_session_new (); rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK; rtpsession->priv->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME; /* configure callbacks */ rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession); /* configure signals */ g_signal_connect (rtpsession->priv->session, "on-new-ssrc", (GCallback) on_new_ssrc, rtpsession); g_signal_connect (rtpsession->priv->session, "on-ssrc-collision", (GCallback) on_ssrc_collision, rtpsession); g_signal_connect (rtpsession->priv->session, "on-ssrc-validated", (GCallback) on_ssrc_validated, rtpsession); g_signal_connect (rtpsession->priv->session, "on-ssrc-active", (GCallback) on_ssrc_active, rtpsession); g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes", (GCallback) on_ssrc_sdes, rtpsession); g_signal_connect (rtpsession->priv->session, "on-bye-ssrc", (GCallback) on_bye_ssrc, rtpsession); g_signal_connect (rtpsession->priv->session, "on-bye-timeout", (GCallback) on_bye_timeout, rtpsession); g_signal_connect (rtpsession->priv->session, "on-timeout", (GCallback) on_timeout, rtpsession); g_signal_connect (rtpsession->priv->session, "on-sender-timeout", (GCallback) on_sender_timeout, rtpsession); g_signal_connect (rtpsession->priv->session, "on-new-sender-ssrc", (GCallback) on_new_sender_ssrc, rtpsession); g_signal_connect (rtpsession->priv->session, "on-sender-ssrc-active", (GCallback) on_sender_ssrc_active, rtpsession); g_signal_connect (rtpsession->priv->session, "notify::stats", (GCallback) on_notify_stats, rtpsession); rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) gst_caps_unref); rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID; gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED); rtpsession->priv->thread_stopped = TRUE; rtpsession->priv->recv_rtx_req_count = 0; rtpsession->priv->sent_rtx_req_count = 0; rtpsession->priv->ntp_time_source = DEFAULT_NTP_TIME_SOURCE; rtpsession->priv->send_rtp_sink_eos = FALSE; } static void gst_rtp_session_finalize (GObject * object) { GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (object); g_hash_table_destroy (rtpsession->priv->ptmap); g_mutex_clear (&rtpsession->priv->lock); g_cond_clear (&rtpsession->priv->cond); g_object_unref (rtpsession->priv->sysclock); g_object_unref (rtpsession->priv->session); if (rtpsession->priv->last_twcc_stats) gst_structure_free (rtpsession->priv->last_twcc_stats); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_rtp_session_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (object); priv = rtpsession->priv; switch (prop_id) { case PROP_BANDWIDTH: g_object_set_property (G_OBJECT (priv->session), "bandwidth", value); break; case PROP_RTCP_FRACTION: g_object_set_property (G_OBJECT (priv->session), "rtcp-fraction", value); break; case PROP_RTCP_RR_BANDWIDTH: g_object_set_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth", value); break; case PROP_RTCP_RS_BANDWIDTH: g_object_set_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth", value); break; case PROP_SDES: rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value)); break; case PROP_USE_PIPELINE_CLOCK: priv->use_pipeline_clock = g_value_get_boolean (value); break; case PROP_RTCP_MIN_INTERVAL: g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval", value); break; case PROP_PROBATION: g_object_set_property (G_OBJECT (priv->session), "probation", value); break; case PROP_MAX_DROPOUT_TIME: g_object_set_property (G_OBJECT (priv->session), "max-dropout-time", value); break; case PROP_MAX_MISORDER_TIME: g_object_set_property (G_OBJECT (priv->session), "max-misorder-time", value); break; case PROP_RTP_PROFILE: g_object_set_property (G_OBJECT (priv->session), "rtp-profile", value); break; case PROP_NTP_TIME_SOURCE: priv->ntp_time_source = g_value_get_enum (value); break; case PROP_RTCP_SYNC_SEND_TIME: priv->rtcp_sync_send_time = g_value_get_boolean (value); break; case PROP_UPDATE_NTP64_HEADER_EXT: g_object_set_property (G_OBJECT (priv->session), "update-ntp64-header-ext", value); break; case PROP_TIMEOUT_INACTIVE_SOURCES: g_object_set_property (G_OBJECT (priv->session), "timeout-inactive-sources", value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (object); priv = rtpsession->priv; switch (prop_id) { case PROP_BANDWIDTH: g_object_get_property (G_OBJECT (priv->session), "bandwidth", value); break; case PROP_RTCP_FRACTION: g_object_get_property (G_OBJECT (priv->session), "rtcp-fraction", value); break; case PROP_RTCP_RR_BANDWIDTH: g_object_get_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth", value); break; case PROP_RTCP_RS_BANDWIDTH: g_object_get_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth", value); break; case PROP_SDES: g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session)); break; case PROP_NUM_SOURCES: g_value_set_uint (value, rtp_session_get_num_sources (priv->session)); break; case PROP_NUM_ACTIVE_SOURCES: g_value_set_uint (value, rtp_session_get_num_active_sources (priv->session)); break; case PROP_INTERNAL_SESSION: g_value_set_object (value, priv->session); break; case PROP_USE_PIPELINE_CLOCK: g_value_set_boolean (value, priv->use_pipeline_clock); break; case PROP_RTCP_MIN_INTERVAL: g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval", value); break; case PROP_PROBATION: g_object_get_property (G_OBJECT (priv->session), "probation", value); break; case PROP_MAX_DROPOUT_TIME: g_object_get_property (G_OBJECT (priv->session), "max-dropout-time", value); break; case PROP_MAX_MISORDER_TIME: g_object_get_property (G_OBJECT (priv->session), "max-misorder-time", value); break; case PROP_STATS: g_value_take_boxed (value, gst_rtp_session_create_stats (rtpsession)); break; case PROP_TWCC_STATS: GST_RTP_SESSION_LOCK (rtpsession); g_value_set_boxed (value, priv->last_twcc_stats); GST_RTP_SESSION_UNLOCK (rtpsession); break; case PROP_RTP_PROFILE: g_object_get_property (G_OBJECT (priv->session), "rtp-profile", value); break; case PROP_NTP_TIME_SOURCE: g_value_set_enum (value, priv->ntp_time_source); break; case PROP_RTCP_SYNC_SEND_TIME: g_value_set_boolean (value, priv->rtcp_sync_send_time); break; case PROP_UPDATE_NTP64_HEADER_EXT: g_object_get_property (G_OBJECT (priv->session), "update-ntp64-header-ext", value); break; case PROP_TIMEOUT_INACTIVE_SOURCES: g_object_get_property (G_OBJECT (priv->session), "timeout-inactive-sources", value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStructure * gst_rtp_session_create_stats (GstRtpSession * rtpsession) { GstStructure *s; g_object_get (rtpsession->priv->session, "stats", &s, NULL); gst_structure_set (s, "rtx-count", G_TYPE_UINT, rtpsession->priv->recv_rtx_req_count, "recv-rtx-req-count", G_TYPE_UINT, rtpsession->priv->recv_rtx_req_count, "sent-rtx-req-count", G_TYPE_UINT, rtpsession->priv->sent_rtx_req_count, NULL); return s; } static void get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time, guint64 * ntpnstime) { guint64 ntpns = -1; GstClock *clock; GstClockTime base_time, rt, clock_time; GST_OBJECT_LOCK (rtpsession); if ((clock = GST_ELEMENT_CLOCK (rtpsession))) { base_time = GST_ELEMENT_CAST (rtpsession)->base_time; gst_object_ref (clock); GST_OBJECT_UNLOCK (rtpsession); /* get current clock time and convert to running time */ clock_time = gst_clock_get_time (clock); rt = clock_time - base_time; if (rtpsession->priv->use_pipeline_clock) { ntpns = rt; /* add constant to convert from 1970 based time to 1900 based time */ ntpns += (GST_RTP_NTP_UNIX_OFFSET * GST_SECOND); } else { switch (rtpsession->priv->ntp_time_source) { case GST_RTP_NTP_TIME_SOURCE_NTP: case GST_RTP_NTP_TIME_SOURCE_UNIX:{ /* get current NTP time */ ntpns = g_get_real_time () * GST_USECOND; /* add constant to convert from 1970 based time to 1900 based time */ if (rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP) ntpns += (GST_RTP_NTP_UNIX_OFFSET * GST_SECOND); break; } case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME: ntpns = rt; break; case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME: ntpns = clock_time; break; default: ntpns = -1; g_assert_not_reached (); break; } } gst_object_unref (clock); } else { GST_OBJECT_UNLOCK (rtpsession); rt = -1; ntpns = -1; } if (running_time) *running_time = rt; if (ntpnstime) *ntpnstime = ntpns; } /* must be called with GST_RTP_SESSION_LOCK */ static void signal_waiting_rtcp_thread_unlocked (GstRtpSession * rtpsession) { if (rtpsession->priv->wait_send) { GST_LOG_OBJECT (rtpsession, "signal RTCP thread"); rtpsession->priv->wait_send = FALSE; GST_RTP_SESSION_SIGNAL (rtpsession); } } static void rtcp_thread (GstRtpSession * rtpsession) { GstClockID id; GstClockTime current_time; GstClockTime next_timeout; guint64 ntpnstime; GstClockTime running_time; RTPSession *session; GstClock *sysclock; GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread"); GST_RTP_SESSION_LOCK (rtpsession); while (rtpsession->priv->wait_send) { GST_LOG_OBJECT (rtpsession, "waiting for getting started"); GST_RTP_SESSION_WAIT (rtpsession); GST_LOG_OBJECT (rtpsession, "signaled..."); } sysclock = rtpsession->priv->sysclock; current_time = gst_clock_get_time (sysclock); session = rtpsession->priv->session; GST_DEBUG_OBJECT (rtpsession, "starting at %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time)); session->start_time = current_time; while (!rtpsession->priv->stop_thread) { GstClockReturn res; /* get initial estimate */ next_timeout = rtp_session_next_timeout (session, current_time); GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT, GST_TIME_ARGS (next_timeout)); /* leave if no more timeouts, the session ended */ if (next_timeout == GST_CLOCK_TIME_NONE) break; id = rtpsession->priv->id = gst_clock_new_single_shot_id (sysclock, next_timeout); GST_RTP_SESSION_UNLOCK (rtpsession); res = gst_clock_id_wait (id, NULL); GST_RTP_SESSION_LOCK (rtpsession); gst_clock_id_unref (id); rtpsession->priv->id = NULL; if (rtpsession->priv->stop_thread) break; /* update current time */ current_time = gst_clock_get_time (sysclock); /* get current NTP time */ get_current_times (rtpsession, &running_time, &ntpnstime); /* we get unlocked because we need to perform reconsideration, don't perform * the timeout but get a new reporting estimate. */ GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT, res, GST_TIME_ARGS (current_time)); /* perform actions, we ignore result. Release lock because it might push. */ GST_RTP_SESSION_UNLOCK (rtpsession); rtp_session_on_timeout (session, current_time, ntpnstime, running_time); GST_RTP_SESSION_LOCK (rtpsession); } /* mark the thread as stopped now */ rtpsession->priv->thread_stopped = TRUE; GST_RTP_SESSION_UNLOCK (rtpsession); GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread"); } static gboolean start_rtcp_thread (GstRtpSession * rtpsession) { GError *error = NULL; gboolean res; GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread"); GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->stop_thread = FALSE; if (rtpsession->priv->thread_stopped) { /* if the thread stopped, and we still have a handle to the thread, join it * now. We can safely join with the lock held, the thread will not take it * anymore. */ if (rtpsession->priv->thread) g_thread_join (rtpsession->priv->thread); /* only create a new thread if the old one was stopped. Otherwise we can * just reuse the currently running one. */ rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp", (GThreadFunc) rtcp_thread, rtpsession, &error); rtpsession->priv->thread_stopped = FALSE; } GST_RTP_SESSION_UNLOCK (rtpsession); if (error != NULL) { res = FALSE; GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message); g_error_free (error); } else { res = TRUE; } return res; } static void stop_rtcp_thread (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread"); GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->stop_thread = TRUE; signal_waiting_rtcp_thread_unlocked (rtpsession); if (rtpsession->priv->id) gst_clock_id_unschedule (rtpsession->priv->id); GST_RTP_SESSION_UNLOCK (rtpsession); } static void join_rtcp_thread (GstRtpSession * rtpsession) { GST_RTP_SESSION_LOCK (rtpsession); /* don't try to join when we have no thread */ if (rtpsession->priv->thread != NULL) { GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread"); GST_RTP_SESSION_UNLOCK (rtpsession); g_thread_join (rtpsession->priv->thread); GST_RTP_SESSION_LOCK (rtpsession); /* after the join, take the lock and clear the thread structure. The caller * is supposed to not concurrently call start and join. */ rtpsession->priv->thread = NULL; } GST_RTP_SESSION_UNLOCK (rtpsession); } static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn res; GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: break; case GST_STATE_CHANGE_READY_TO_PAUSED: GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->wait_send = TRUE; rtpsession->priv->send_latency = GST_CLOCK_TIME_NONE; rtpsession->priv->warned_latency_once = FALSE; GST_RTP_SESSION_UNLOCK (rtpsession); break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PAUSED_TO_READY: GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->send_rtp_sink_eos = FALSE; GST_RTP_SESSION_UNLOCK (rtpsession); /* no need to join yet, we might want to continue later. Also, the * dataflow could block downstream so that a join could just block * forever. */ stop_rtcp_thread (rtpsession); break; default: break; } res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_PLAYING: if (!start_rtcp_thread (rtpsession)) goto failed_thread; break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: /* downstream is now releasing the dataflow and we can join. */ join_rtcp_thread (rtpsession); rtp_session_reset (rtpsession->priv->session); break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; /* ERRORS */ failed_thread: { return GST_STATE_CHANGE_FAILURE; } } static gboolean return_true (gpointer key, gpointer value, gpointer user_data) { return TRUE; } static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession) { GST_RTP_SESSION_LOCK (rtpsession); g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL); GST_RTP_SESSION_UNLOCK (rtpsession); } /* called when the session manager has an RTP packet ready to be pushed */ static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstPad *rtp_src; rtpsession = GST_RTP_SESSION (user_data); GST_RTP_SESSION_LOCK (rtpsession); if ((rtp_src = rtpsession->recv_rtp_src)) gst_object_ref (rtp_src); GST_RTP_SESSION_UNLOCK (rtpsession); if (rtp_src) { if (rtpsession->priv->processed_list) { GST_LOG_OBJECT (rtpsession, "queueing received RTP packet"); gst_buffer_list_add (rtpsession->priv->processed_list, buffer); result = GST_FLOW_OK; } else { GST_LOG_OBJECT (rtpsession, "pushing received RTP packet"); result = gst_pad_push (rtp_src, buffer); } gst_object_unref (rtp_src); } else { GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; } /* called when the session manager has an RTP packet ready for further * sending */ static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src, gpointer data, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstPad *rtp_src; rtpsession = GST_RTP_SESSION (user_data); GST_RTP_SESSION_LOCK (rtpsession); if ((rtp_src = rtpsession->send_rtp_src)) gst_object_ref (rtp_src); signal_waiting_rtcp_thread_unlocked (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession); if (rtp_src) { if (GST_IS_BUFFER (data)) { GST_LOG_OBJECT (rtpsession, "sending RTP packet"); result = gst_pad_push (rtp_src, GST_BUFFER_CAST (data)); } else { GST_LOG_OBJECT (rtpsession, "sending RTP list"); result = gst_pad_push_list (rtp_src, GST_BUFFER_LIST_CAST (data)); } gst_object_unref (rtp_src); } else { gst_mini_object_unref (GST_MINI_OBJECT_CAST (data)); result = GST_FLOW_OK; } return result; } static void do_rtcp_events (GstRtpSession * rtpsession, GstPad * srcpad) { GstCaps *caps; GstSegment seg; GstEvent *event; gchar *stream_id; gboolean have_group_id; guint group_id; stream_id = g_strdup_printf ("%08x%08x%08x%08x", g_random_int (), g_random_int (), g_random_int (), g_random_int ()); GST_RTP_SESSION_LOCK (rtpsession); if (rtpsession->recv_rtp_sink) { event = gst_pad_get_sticky_event (rtpsession->recv_rtp_sink, GST_EVENT_STREAM_START, 0); if (event) { if (gst_event_parse_group_id (event, &group_id)) have_group_id = TRUE; else have_group_id = FALSE; gst_event_unref (event); } else { have_group_id = TRUE; group_id = gst_util_group_id_next (); } } else { have_group_id = TRUE; group_id = gst_util_group_id_next (); } GST_RTP_SESSION_UNLOCK (rtpsession); event = gst_event_new_stream_start (stream_id); rtpsession->recv_rtcp_segment_seqnum = gst_event_get_seqnum (event); gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum); if (have_group_id) gst_event_set_group_id (event, group_id); gst_pad_push_event (srcpad, event); g_free (stream_id); caps = gst_caps_new_empty_simple ("application/x-rtcp"); gst_pad_set_caps (srcpad, caps); gst_caps_unref (caps); gst_segment_init (&seg, GST_FORMAT_TIME); event = gst_event_new_segment (&seg); gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum); gst_pad_push_event (srcpad, event); } /* called when the session manager has an RTCP packet ready for further * sending. The eos flag is set when an EOS event should be sent downstream as * well. */ static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gboolean all_sources_bye, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstPad *rtcp_src; rtpsession = GST_RTP_SESSION (user_data); GST_RTP_SESSION_LOCK (rtpsession); if (rtpsession->priv->stop_thread) goto stopping; if ((rtcp_src = rtpsession->send_rtcp_src)) { gst_object_ref (rtcp_src); GST_RTP_SESSION_UNLOCK (rtpsession); /* set rtcp caps on output pad */ if (!gst_pad_has_current_caps (rtcp_src)) do_rtcp_events (rtpsession, rtcp_src); GST_LOG_OBJECT (rtpsession, "sending RTCP"); result = gst_pad_push (rtcp_src, buffer); /* Forward send an EOS on the RTCP sink if we received an EOS on the * send_rtp_sink. We don't need to check the recv_rtp_sink since in this * case the EOS event would already have been sent. Also, prevent a * race condition between the EOS event handling and rtcp send * function/thread by using send_rtp_sink_eos directly instead of * GST_PAD_IS_EOS*/ GST_RTP_SESSION_LOCK (rtpsession); if (all_sources_bye && rtpsession->priv->send_rtp_sink_eos) { GstEvent *event; GST_LOG_OBJECT (rtpsession, "sending EOS"); event = gst_event_new_eos (); gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum); gst_pad_push_event (rtcp_src, event); } GST_RTP_SESSION_UNLOCK (rtpsession); gst_object_unref (rtcp_src); } else { GST_RTP_SESSION_UNLOCK (rtpsession); GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; /* ERRORS */ stopping: { GST_DEBUG_OBJECT (rtpsession, "we are stopping"); gst_buffer_unref (buffer); GST_RTP_SESSION_UNLOCK (rtpsession); return GST_FLOW_OK; } } /* called when the session manager has an SR RTCP packet ready for handling * inter stream synchronisation */ static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess, GstBuffer * buffer, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstPad *sync_src; rtpsession = GST_RTP_SESSION (user_data); GST_RTP_SESSION_LOCK (rtpsession); if (rtpsession->priv->stop_thread) goto stopping; if ((sync_src = rtpsession->sync_src)) { gst_object_ref (sync_src); GST_RTP_SESSION_UNLOCK (rtpsession); /* set rtcp caps on output pad, this happens * when we receive RTCP muxed with RTP according * to RFC5761. Otherwise we would have forwarded * the events from the recv_rtcp_sink pad already */ if (!gst_pad_has_current_caps (sync_src)) do_rtcp_events (rtpsession, sync_src); GST_LOG_OBJECT (rtpsession, "sending Sync RTCP"); result = gst_pad_push (sync_src, buffer); gst_object_unref (sync_src); } else { GST_RTP_SESSION_UNLOCK (rtpsession); GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; /* ERRORS */ stopping: { GST_DEBUG_OBJECT (rtpsession, "we are stopping"); gst_buffer_unref (buffer); GST_RTP_SESSION_UNLOCK (rtpsession); return GST_FLOW_OK; } } static void gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps) { GstRtpSessionPrivate *priv; const GstStructure *s; gint payload; priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "parsing caps"); s = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (s, "payload", &payload)) return; if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload))) return; rtp_session_update_recv_caps_structure (rtpsession->priv->session, s); g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload), gst_caps_ref (caps)); } static GstCaps * gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload) { GstCaps *caps = NULL; GValue args[2] = { {0}, {0} }; GValue ret = { 0 }; GST_RTP_SESSION_LOCK (rtpsession); caps = g_hash_table_lookup (rtpsession->priv->ptmap, GINT_TO_POINTER (payload)); if (caps) { gst_caps_ref (caps); goto done; } /* not found in the cache, try to get it with a signal */ g_value_init (&args[0], GST_TYPE_ELEMENT); g_value_set_object (&args[0], rtpsession); g_value_init (&args[1], G_TYPE_UINT); g_value_set_uint (&args[1], payload); g_value_init (&ret, GST_TYPE_CAPS); g_value_set_boxed (&ret, NULL); GST_RTP_SESSION_UNLOCK (rtpsession); g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); GST_RTP_SESSION_LOCK (rtpsession); g_value_unset (&args[0]); g_value_unset (&args[1]); caps = (GstCaps *) g_value_dup_boxed (&ret); g_value_unset (&ret); if (!caps) goto no_caps; gst_rtp_session_cache_caps (rtpsession, caps); done: GST_RTP_SESSION_UNLOCK (rtpsession); return caps; no_caps: { GST_DEBUG_OBJECT (rtpsession, "could not get caps"); goto done; } } /* called when the session manager needs the clock rate */ static GstCaps * gst_rtp_session_caps (RTPSession * sess, guint8 payload, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION_CAST (user_data); return gst_rtp_session_get_caps_for_pt (rtpsession, payload); } /* called when the session manager asks us to reconsider the timeout */ static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data) { GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION_CAST (user_data); GST_RTP_SESSION_LOCK (rtpsession); GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration"); if (rtpsession->priv->id) gst_clock_id_unschedule (rtpsession->priv->id); GST_RTP_SESSION_UNLOCK (rtpsession); } static gboolean gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received event %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; /* process */ gst_event_parse_caps (event, &caps); gst_rtp_session_setcaps_recv_rtp (pad, rtpsession, caps); ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } case GST_EVENT_FLUSH_STOP: gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID; ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->send_rtp_sink_eos = FALSE; GST_RTP_SESSION_UNLOCK (rtpsession); break; case GST_EVENT_SEGMENT: { GstSegment *segment, in_segment; segment = &rtpsession->recv_rtp_seg; /* the newsegment event is needed to convert the RTP timestamp to * running_time, which is needed to generate a mapping from RTP to NTP * timestamps in SR reports */ gst_event_copy_segment (event, &in_segment); GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT, &in_segment); /* accept upstream */ gst_segment_copy_into (&in_segment, segment); /* push event forward */ ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } case GST_EVENT_EOS: { GstPad *rtcp_src; ret = gst_pad_push_event (rtpsession->recv_rtp_src, gst_event_ref (event)); GST_RTP_SESSION_LOCK (rtpsession); if ((rtcp_src = rtpsession->send_rtcp_src)) gst_object_ref (rtcp_src); GST_RTP_SESSION_UNLOCK (rtpsession); gst_event_unref (event); if (rtcp_src) { event = gst_event_new_eos (); if (rtpsession->recv_rtcp_segment_seqnum != GST_SEQNUM_INVALID) gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum); ret = gst_pad_push_event (rtcp_src, event); gst_object_unref (rtcp_src); } else { ret = TRUE; } break; } default: ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } return ret; } static gboolean gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession, guint32 ssrc, guint payload, gboolean all_headers, gint count) { GstCaps *caps; caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload); if (caps) { const GstStructure *s = gst_caps_get_structure (caps, 0); gboolean pli; gboolean fir; pli = gst_structure_has_field (s, "rtcp-fb-nack-pli"); fir = gst_structure_has_field (s, "rtcp-fb-ccm-fir") && all_headers; /* Google Talk uses FIR for repair, so send it even if we just want a * regular PLI */ if (!pli && gst_structure_has_field (s, "rtcp-fb-x-gstreamer-fir-as-repair")) fir = TRUE; gst_caps_unref (caps); if (pli || fir) return rtp_session_request_key_unit (rtpsession->priv->session, ssrc, fir, count); } return FALSE; } static gboolean gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean forward = TRUE; gboolean ret = TRUE; const GstStructure *s; guint32 ssrc; guint pt; rtpsession = GST_RTP_SESSION (parent); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CUSTOM_UPSTREAM: s = gst_event_get_structure (event); if (gst_structure_has_name (s, "GstForceKeyUnit") && gst_structure_get_uint (s, "ssrc", &ssrc) && gst_structure_get_uint (s, "payload", &pt)) { gboolean all_headers = FALSE; gint count = -1; gst_structure_get_boolean (s, "all-headers", &all_headers); if (gst_structure_get_int (s, "count", &count) && count < 0) count += G_MAXINT; /* Make sure count is positive if present */ if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt, all_headers, count)) forward = FALSE; } else if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) { guint seqnum, delay, deadline, max_delay, avg_rtt; GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->recv_rtx_req_count++; GST_RTP_SESSION_UNLOCK (rtpsession); if (!gst_structure_get_uint (s, "ssrc", &ssrc)) ssrc = -1; if (!gst_structure_get_uint (s, "seqnum", &seqnum)) seqnum = -1; if (!gst_structure_get_uint (s, "delay", &delay)) delay = 0; if (!gst_structure_get_uint (s, "deadline", &deadline)) deadline = 100; if (!gst_structure_get_uint (s, "avg-rtt", &avg_rtt)) avg_rtt = 40; /* remaining time to receive the packet */ max_delay = deadline; if (max_delay > delay) max_delay -= delay; /* estimated RTT */ if (max_delay > avg_rtt) max_delay -= avg_rtt; else max_delay = 0; if (rtp_session_request_nack (rtpsession->priv->session, ssrc, seqnum, max_delay * GST_MSECOND)) forward = FALSE; } break; default: break; } if (forward) { GstPad *recv_rtp_sink; GST_RTP_SESSION_LOCK (rtpsession); if ((recv_rtp_sink = rtpsession->recv_rtp_sink)) gst_object_ref (recv_rtp_sink); GST_RTP_SESSION_UNLOCK (rtpsession); if (recv_rtp_sink) { ret = gst_pad_push_event (recv_rtp_sink, event); gst_object_unref (recv_rtp_sink); } else gst_event_unref (event); } else { gst_event_unref (event); } return ret; } static GstIterator * gst_rtp_session_iterate_internal_links (GstPad * pad, GstObject * parent) { GstRtpSession *rtpsession; GstPad *otherpad = NULL; GstIterator *it = NULL; rtpsession = GST_RTP_SESSION (parent); GST_RTP_SESSION_LOCK (rtpsession); if (pad == rtpsession->recv_rtp_src) { otherpad = gst_object_ref (rtpsession->recv_rtp_sink); } else if (pad == rtpsession->recv_rtp_sink) { otherpad = gst_object_ref (rtpsession->recv_rtp_src); } else if (pad == rtpsession->send_rtp_src) { otherpad = gst_object_ref (rtpsession->send_rtp_sink); } else if (pad == rtpsession->send_rtp_sink) { otherpad = gst_object_ref (rtpsession->send_rtp_src); } GST_RTP_SESSION_UNLOCK (rtpsession); if (otherpad) { GValue val = { 0, }; g_value_init (&val, GST_TYPE_PAD); g_value_set_object (&val, otherpad); it = gst_iterator_new_single (GST_TYPE_PAD, &val); g_value_unset (&val); gst_object_unref (otherpad); } else { it = gst_iterator_new_single (GST_TYPE_PAD, NULL); } return it; } static gboolean gst_rtp_session_setcaps_recv_rtp (GstPad * pad, GstRtpSession * rtpsession, GstCaps * caps) { GST_RTP_SESSION_LOCK (rtpsession); gst_rtp_session_cache_caps (rtpsession, caps); GST_RTP_SESSION_UNLOCK (rtpsession); return TRUE; } /* receive a packet from a sender, send it to the RTP session manager and * forward the packet on the rtp_src pad */ static GstFlowReturn gst_rtp_session_chain_recv_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstFlowReturn ret; GstClockTime current_time, running_time; GstClockTime timestamp; guint64 ntpnstime; rtpsession = GST_RTP_SESSION (parent); priv = rtpsession->priv; GST_LOG_OBJECT (rtpsession, "received RTP packet"); GST_RTP_SESSION_LOCK (rtpsession); signal_waiting_rtcp_thread_unlocked (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession); /* get NTP time when this packet was captured, this depends on the timestamp. */ timestamp = GST_BUFFER_PTS (buffer); if (GST_CLOCK_TIME_IS_VALID (timestamp)) { /* convert to running time using the segment values */ running_time = gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME, timestamp); ntpnstime = GST_CLOCK_TIME_NONE; } else { get_current_times (rtpsession, &running_time, &ntpnstime); } current_time = gst_clock_get_time (priv->sysclock); ret = rtp_session_process_rtp (priv->session, buffer, current_time, running_time, ntpnstime); if (ret != GST_FLOW_OK) goto push_error; done: return ret; /* ERRORS */ push_error: { GST_DEBUG_OBJECT (rtpsession, "process returned %s", gst_flow_get_name (ret)); goto done; } } static gboolean process_received_buffer_in_list (GstBuffer ** buffer, guint idx, gpointer data) { gint ret; ret = gst_rtp_session_chain_recv_rtp (NULL, data, *buffer); if (ret != GST_FLOW_OK) GST_ERROR ("Processing individual buffer in a list failed"); /* * The buffer has been processed, remove it from the original list, if it was * a valid RTP buffer it has been added to the "processed" list in * gst_rtp_session_process_rtp(). */ *buffer = NULL; return TRUE; } static GstFlowReturn gst_rtp_session_chain_recv_rtp_list (GstPad * pad, GstObject * parent, GstBufferList * list) { GstRtpSession *rtpsession = GST_RTP_SESSION (parent); GstBufferList *processed_list; processed_list = gst_buffer_list_new (); /* Set some private data to detect that a buffer list is being pushed. */ rtpsession->priv->processed_list = processed_list; /* * Individually process the buffers from the incoming buffer list as the * incoming RTP packets in the list can be mixed in all sorts of ways: * - different frames, * - different sources, * - different types (RTP or RTCP) */ gst_buffer_list_foreach (list, (GstBufferListFunc) process_received_buffer_in_list, parent); gst_buffer_list_unref (list); /* Clean up private data in case the next push does not use a buffer list. */ rtpsession->priv->processed_list = NULL; if (gst_buffer_list_length (processed_list) == 0 || !rtpsession->recv_rtp_src) { gst_buffer_list_unref (processed_list); return GST_FLOW_OK; } GST_LOG_OBJECT (rtpsession, "pushing received RTP list"); return gst_pad_push_list (rtpsession->recv_rtp_src, processed_list); } static gboolean gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received event %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEGMENT: /* Make sure that the sync_src pad has caps before the segment event. * Otherwise we might get a segment event before caps from the receive * RTCP pad, and then later when receiving RTCP packets will set caps. * This will results in a sticky event misordering warning */ if (!gst_pad_has_current_caps (rtpsession->sync_src)) { GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp"); gst_pad_set_caps (rtpsession->sync_src, caps); gst_caps_unref (caps); } /* fall through */ default: ret = gst_pad_push_event (rtpsession->sync_src, event); break; } return ret; } /* Receive an RTCP packet from a sender, send it to the RTP session manager and * forward the SR packets to the sync_src pad. */ static GstFlowReturn gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstClockTime current_time; GstClockTime running_time; guint64 ntpnstime; rtpsession = GST_RTP_SESSION (parent); priv = rtpsession->priv; GST_LOG_OBJECT (rtpsession, "received RTCP packet"); GST_RTP_SESSION_LOCK (rtpsession); signal_waiting_rtcp_thread_unlocked (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession); current_time = gst_clock_get_time (priv->sysclock); get_current_times (rtpsession, &running_time, &ntpnstime); rtp_session_process_rtcp (priv->session, buffer, current_time, running_time, ntpnstime); return GST_FLOW_OK; /* always return OK */ } static gboolean gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstObject * parent, GstQuery * query) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received QUERY %s", GST_QUERY_TYPE_NAME (query)); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: ret = TRUE; /* use the defaults for the latency query. */ gst_query_set_latency (query, FALSE, 0, -1); break; default: /* other queries simply fail for now */ break; } return ret; } static gboolean gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = TRUE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received EVENT %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_SEEK: case GST_EVENT_LATENCY: gst_event_unref (event); ret = TRUE; break; default: /* other events simply fail for now */ gst_event_unref (event); ret = FALSE; break; } return ret; } static gboolean gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received EVENT %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CAPS: { GstCaps *caps; /* process */ gst_event_parse_caps (event, &caps); gst_rtp_session_setcaps_send_rtp (pad, rtpsession, caps); ret = gst_pad_push_event (rtpsession->send_rtp_src, event); break; } case GST_EVENT_FLUSH_STOP: gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED); ret = gst_pad_push_event (rtpsession->send_rtp_src, event); break; case GST_EVENT_SEGMENT:{ GstSegment *segment, in_segment; segment = &rtpsession->send_rtp_seg; /* the newsegment event is needed to convert the RTP timestamp to * running_time, which is needed to generate a mapping from RTP to NTP * timestamps in SR reports */ gst_event_copy_segment (event, &in_segment); GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT, &in_segment); /* accept upstream */ gst_segment_copy_into (&in_segment, segment); /* push event forward */ ret = gst_pad_push_event (rtpsession->send_rtp_src, event); break; } case GST_EVENT_EOS:{ GstClockTime current_time; /* push downstream FIXME, we are not supposed to leave the session just * because we stop sending. */ ret = gst_pad_push_event (rtpsession->send_rtp_src, event); current_time = gst_clock_get_time (rtpsession->priv->sysclock); GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->send_rtp_sink_eos = TRUE; GST_RTP_SESSION_UNLOCK (rtpsession); GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message"); rtp_session_mark_all_bye (rtpsession->priv->session, "End Of Stream"); rtp_session_schedule_bye (rtpsession->priv->session, current_time); break; } default:{ GstPad *send_rtp_src; GST_RTP_SESSION_LOCK (rtpsession); if ((send_rtp_src = rtpsession->send_rtp_src)) gst_object_ref (send_rtp_src); GST_RTP_SESSION_UNLOCK (rtpsession); if (send_rtp_src) { ret = gst_pad_push_event (send_rtp_src, event); gst_object_unref (send_rtp_src); } else gst_event_unref (event); break; } } return ret; } static gboolean gst_rtp_session_event_send_rtp_src (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (parent); GST_DEBUG_OBJECT (rtpsession, "received EVENT %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_LATENCY: /* save the latency, we need this to know when an RTP packet will be * rendered by the sink */ gst_event_parse_latency (event, &rtpsession->priv->send_latency); ret = gst_pad_event_default (pad, parent, event); break; default: ret = gst_pad_event_default (pad, parent, event); break; } return ret; } static GstCaps * gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession, GstCaps * filter) { GstRtpSessionPrivate *priv; GstCaps *result; GstStructure *s1, *s2; guint ssrc; gboolean is_random; priv = rtpsession->priv; ssrc = rtp_session_suggest_ssrc (priv->session, &is_random); /* we can basically accept anything but we prefer to receive packets with our * internal SSRC so that we don't have to patch it. Create a structure with * the SSRC and another one without. * Only do this if the session actually decided on an ssrc already, * otherwise we give upstream the opportunity to select an ssrc itself */ if (!is_random) { s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL); s2 = gst_structure_new_empty ("application/x-rtp"); result = gst_caps_new_full (s1, s2, NULL); } else { result = gst_caps_new_empty_simple ("application/x-rtp"); } if (filter) { GstCaps *caps = result; result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); } GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result); return result; } static gboolean gst_rtp_session_query_send_rtp (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res = FALSE; GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (parent); switch (GST_QUERY_TYPE (query)) { case GST_QUERY_CAPS: { GstCaps *filter, *caps; gst_query_parse_caps (query, &filter); caps = gst_rtp_session_getcaps_send_rtp (pad, rtpsession, filter); gst_query_set_caps_result (query, caps); gst_caps_unref (caps); res = TRUE; break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession, GstCaps * caps) { GstRtpSessionPrivate *priv; priv = rtpsession->priv; rtp_session_update_send_caps (priv->session, caps); return TRUE; } /* Receive an RTP packet or a list of packets to be sent to the receivers, * send to RTP session manager and forward to send_rtp_src. */ static GstFlowReturn gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession, gpointer data, gboolean is_list) { GstRtpSessionPrivate *priv; GstFlowReturn ret; GstClockTime timestamp, running_time; GstClockTime current_time; guint64 ntpnstime; GstClock *clock; priv = rtpsession->priv; GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet"); /* get NTP time when this packet was captured, this depends on the timestamp. */ if (is_list) { GstBuffer *buffer = NULL; /* All buffers in a list have the same timestamp. * So, just take it from the first buffer. */ buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0); if (buffer) timestamp = GST_BUFFER_PTS (buffer); else timestamp = -1; } else { timestamp = GST_BUFFER_PTS (GST_BUFFER_CAST (data)); } if (GST_CLOCK_TIME_IS_VALID (timestamp)) { /* convert to running time using the segment start value. */ running_time = gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, timestamp); if (priv->rtcp_sync_send_time) { if (priv->send_latency != GST_CLOCK_TIME_NONE) { running_time += priv->send_latency; } else { if (!priv->warned_latency_once) { priv->warned_latency_once = TRUE; GST_WARNING_OBJECT (rtpsession, "Can't determine running time for this packet without knowing configured latency"); } else { GST_LOG_OBJECT (rtpsession, "Can't determine running time for this packet without knowing configured latency"); } running_time = -1; } } } else { /* no timestamp. */ running_time = -1; } current_time = gst_clock_get_time (priv->sysclock); /* Calculate the NTP time of this packet based on the session configuration * and the running time from above */ GST_OBJECT_LOCK (rtpsession); if (running_time != -1 && (clock = GST_ELEMENT_CLOCK (rtpsession))) { GstClockTime base_time; base_time = GST_ELEMENT_CAST (rtpsession)->base_time; gst_object_ref (clock); GST_OBJECT_UNLOCK (rtpsession); if (rtpsession->priv->use_pipeline_clock) { ntpnstime = running_time; /* add constant to convert from 1970 based time to 1900 based time */ ntpnstime += (GST_RTP_NTP_UNIX_OFFSET * GST_SECOND); } else { switch (rtpsession->priv->ntp_time_source) { case GST_RTP_NTP_TIME_SOURCE_NTP: case GST_RTP_NTP_TIME_SOURCE_UNIX:{ GstClockTime wallclock_now, pipeline_now; /* pipeline clock time for this packet */ ntpnstime = running_time + base_time; /* get current wallclock and pipeline clock time */ wallclock_now = g_get_real_time () * GST_USECOND; pipeline_now = gst_clock_get_time (clock); /* adjust pipeline clock time by the current diff. * Note that this will include some jitter for each packet */ if (wallclock_now > pipeline_now) { GstClockTime diff = wallclock_now - pipeline_now; ntpnstime += diff; } else { GstClockTime diff = pipeline_now - wallclock_now; if (diff > ntpnstime) { /* This can't really happen unless the clock configuration is * broken */ ntpnstime = GST_CLOCK_TIME_NONE; } else { ntpnstime -= diff; } } /* add constant to convert from 1970 based time to 1900 based time */ if (ntpnstime != GST_CLOCK_TIME_NONE && rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP) ntpnstime += (GST_RTP_NTP_UNIX_OFFSET * GST_SECOND); break; } case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME: ntpnstime = running_time; break; case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME: ntpnstime = running_time + base_time; break; default: ntpnstime = -1; g_assert_not_reached (); break; } } gst_object_unref (clock); } else { if (!GST_ELEMENT_CLOCK (rtpsession)) { GST_WARNING_OBJECT (rtpsession, "Don't have a clock yet and can't determine NTP time for this packet"); } GST_OBJECT_UNLOCK (rtpsession); ntpnstime = GST_CLOCK_TIME_NONE; } ret = rtp_session_send_rtp (priv->session, data, is_list, current_time, running_time, ntpnstime); if (ret != GST_FLOW_OK) goto push_error; done: return ret; /* ERRORS */ push_error: { GST_DEBUG_OBJECT (rtpsession, "process returned %s", gst_flow_get_name (ret)); goto done; } } static GstFlowReturn gst_rtp_session_chain_send_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpSession *rtpsession = GST_RTP_SESSION (parent); return gst_rtp_session_chain_send_rtp_common (rtpsession, buffer, FALSE); } static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstObject * parent, GstBufferList * list) { GstRtpSession *rtpsession = GST_RTP_SESSION (parent); return gst_rtp_session_chain_send_rtp_common (rtpsession, list, TRUE); } /* Create sinkpad to receive RTP packets from senders. This will also create a * srcpad for the RTP packets. */ static GstPad * create_recv_rtp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad"); rtpsession->recv_rtp_sink = gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template, "recv_rtp_sink"); gst_pad_set_chain_function (rtpsession->recv_rtp_sink, gst_rtp_session_chain_recv_rtp); gst_pad_set_chain_list_function (rtpsession->recv_rtp_sink, gst_rtp_session_chain_recv_rtp_list); gst_pad_set_event_function (rtpsession->recv_rtp_sink, gst_rtp_session_event_recv_rtp_sink); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink, gst_rtp_session_iterate_internal_links); GST_PAD_SET_PROXY_ALLOCATION (rtpsession->recv_rtp_sink); gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_sink); GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad"); rtpsession->recv_rtp_src = gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template, "recv_rtp_src"); gst_pad_set_event_function (rtpsession->recv_rtp_src, gst_rtp_session_event_recv_rtp_src); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src, gst_rtp_session_iterate_internal_links); gst_pad_use_fixed_caps (rtpsession->recv_rtp_src); gst_pad_set_active (rtpsession->recv_rtp_src, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src); return rtpsession->recv_rtp_sink; } /* Remove sinkpad to receive RTP packets from senders. This will also remove * the srcpad for the RTP packets. */ static void remove_recv_rtp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad"); /* deactivate from source to sink */ gst_pad_set_active (rtpsession->recv_rtp_src, FALSE); gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE); /* remove pads */ gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_sink); rtpsession->recv_rtp_sink = NULL; GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad"); gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src); rtpsession->recv_rtp_src = NULL; } /* Create a sinkpad to receive RTCP messages from senders, this will also create a * sync_src pad for the SR packets. */ static GstPad * create_recv_rtcp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad"); rtpsession->recv_rtcp_sink = gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template, "recv_rtcp_sink"); gst_pad_set_chain_function (rtpsession->recv_rtcp_sink, gst_rtp_session_chain_recv_rtcp); gst_pad_set_event_function (rtpsession->recv_rtcp_sink, gst_rtp_session_event_recv_rtcp_sink); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtcp_sink, gst_rtp_session_iterate_internal_links); gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtcp_sink); GST_DEBUG_OBJECT (rtpsession, "creating sync src pad"); rtpsession->sync_src = gst_pad_new_from_static_template (&rtpsession_sync_src_template, "sync_src"); gst_pad_set_iterate_internal_links_function (rtpsession->sync_src, gst_rtp_session_iterate_internal_links); gst_pad_use_fixed_caps (rtpsession->sync_src); gst_pad_set_active (rtpsession->sync_src, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src); return rtpsession->recv_rtcp_sink; } static void remove_recv_rtcp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad"); gst_pad_set_active (rtpsession->sync_src, FALSE); gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtcp_sink); rtpsession->recv_rtcp_sink = NULL; GST_DEBUG_OBJECT (rtpsession, "removing sync src pad"); gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src); rtpsession->sync_src = NULL; } /* Create a sinkpad to receive RTP packets for receivers. This will also create a * send_rtp_src pad. */ static GstPad * create_send_rtp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "creating pad"); rtpsession->send_rtp_sink = gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template, "send_rtp_sink"); gst_pad_set_chain_function (rtpsession->send_rtp_sink, gst_rtp_session_chain_send_rtp); gst_pad_set_chain_list_function (rtpsession->send_rtp_sink, gst_rtp_session_chain_send_rtp_list); gst_pad_set_query_function (rtpsession->send_rtp_sink, gst_rtp_session_query_send_rtp); gst_pad_set_event_function (rtpsession->send_rtp_sink, gst_rtp_session_event_send_rtp_sink); gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink, gst_rtp_session_iterate_internal_links); GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_sink); GST_PAD_SET_PROXY_ALLOCATION (rtpsession->send_rtp_sink); gst_pad_set_active (rtpsession->send_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_sink); rtpsession->send_rtp_src = gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template, "send_rtp_src"); gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src, gst_rtp_session_iterate_internal_links); gst_pad_set_event_function (rtpsession->send_rtp_src, gst_rtp_session_event_send_rtp_src); GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_src); gst_pad_set_active (rtpsession->send_rtp_src, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src); return rtpsession->send_rtp_sink; } static void remove_send_rtp_sink (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "removing pad"); gst_pad_set_active (rtpsession->send_rtp_src, FALSE); gst_pad_set_active (rtpsession->send_rtp_sink, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_sink); rtpsession->send_rtp_sink = NULL; gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src); rtpsession->send_rtp_src = NULL; } /* Create a srcpad with the RTCP packets to send out. * This pad will be driven by the RTP session manager when it wants to send out * RTCP packets. */ static GstPad * create_send_rtcp_src (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "creating pad"); rtpsession->send_rtcp_src = gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template, "send_rtcp_src"); gst_pad_use_fixed_caps (rtpsession->send_rtcp_src); gst_pad_set_active (rtpsession->send_rtcp_src, TRUE); gst_pad_set_iterate_internal_links_function (rtpsession->send_rtcp_src, gst_rtp_session_iterate_internal_links); gst_pad_set_query_function (rtpsession->send_rtcp_src, gst_rtp_session_query_send_rtcp_src); gst_pad_set_event_function (rtpsession->send_rtcp_src, gst_rtp_session_event_send_rtcp_src); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtcp_src); return rtpsession->send_rtcp_src; } static void remove_send_rtcp_src (GstRtpSession * rtpsession) { GST_DEBUG_OBJECT (rtpsession, "removing pad"); gst_pad_set_active (rtpsession->send_rtcp_src, FALSE); gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtcp_src); rtpsession->send_rtcp_src = NULL; } static GstPad * gst_rtp_session_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps) { GstRtpSession *rtpsession; GstElementClass *klass; GstPad *result; g_return_val_if_fail (templ != NULL, NULL); g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL); rtpsession = GST_RTP_SESSION (element); klass = GST_ELEMENT_GET_CLASS (element); GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name)); GST_RTP_SESSION_LOCK (rtpsession); /* figure out the template */ if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) { if (rtpsession->recv_rtp_sink != NULL) goto exists; result = create_recv_rtp_sink (rtpsession); } else if (templ == gst_element_class_get_pad_template (klass, "recv_rtcp_sink")) { if (rtpsession->recv_rtcp_sink != NULL) goto exists; result = create_recv_rtcp_sink (rtpsession); } else if (templ == gst_element_class_get_pad_template (klass, "send_rtp_sink")) { if (rtpsession->send_rtp_sink != NULL) goto exists; result = create_send_rtp_sink (rtpsession); } else if (templ == gst_element_class_get_pad_template (klass, "send_rtcp_src")) { if (rtpsession->send_rtcp_src != NULL) goto exists; result = create_send_rtcp_src (rtpsession); } else goto wrong_template; GST_RTP_SESSION_UNLOCK (rtpsession); return result; /* ERRORS */ wrong_template: { GST_RTP_SESSION_UNLOCK (rtpsession); g_warning ("rtpsession: this is not our template"); return NULL; } exists: { GST_RTP_SESSION_UNLOCK (rtpsession); g_warning ("rtpsession: pad already requested"); return NULL; } } static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad) { GstRtpSession *rtpsession; g_return_if_fail (GST_IS_RTP_SESSION (element)); g_return_if_fail (GST_IS_PAD (pad)); rtpsession = GST_RTP_SESSION (element); GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad)); GST_RTP_SESSION_LOCK (rtpsession); if (rtpsession->recv_rtp_sink == pad) { remove_recv_rtp_sink (rtpsession); } else if (rtpsession->recv_rtcp_sink == pad) { remove_recv_rtcp_sink (rtpsession); } else if (rtpsession->send_rtp_sink == pad) { remove_send_rtp_sink (rtpsession); } else if (rtpsession->send_rtcp_src == pad) { remove_send_rtcp_src (rtpsession); } else goto wrong_pad; GST_RTP_SESSION_UNLOCK (rtpsession); return; /* ERRORS */ wrong_pad: { GST_RTP_SESSION_UNLOCK (rtpsession); g_warning ("rtpsession: asked to release an unknown pad"); return; } } static void gst_rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean all_headers, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); GstEvent *event; GstPad *send_rtp_sink; GST_RTP_SESSION_LOCK (rtpsession); if ((send_rtp_sink = rtpsession->send_rtp_sink)) gst_object_ref (send_rtp_sink); GST_RTP_SESSION_UNLOCK (rtpsession); if (send_rtp_sink) { event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, gst_structure_new ("GstForceKeyUnit", "ssrc", G_TYPE_UINT, ssrc, "all-headers", G_TYPE_BOOLEAN, all_headers, NULL)); gst_pad_push_event (send_rtp_sink, event); gst_object_unref (send_rtp_sink); } } static GstClockTime gst_rtp_session_request_time (RTPSession * session, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); return gst_clock_get_time (rtpsession->priv->sysclock); } static void gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); GstEvent *event; GstPad *send_rtp_sink; GST_RTP_SESSION_LOCK (rtpsession); if ((send_rtp_sink = rtpsession->send_rtp_sink)) gst_object_ref (send_rtp_sink); GST_RTP_SESSION_UNLOCK (rtpsession); if (send_rtp_sink) { while (TRUE) { event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, gst_structure_new ("GstRTPRetransmissionRequest", "seqnum", G_TYPE_UINT, (guint) seqnum, "ssrc", G_TYPE_UINT, (guint) ssrc, NULL)); gst_pad_push_event (send_rtp_sink, event); GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->sent_rtx_req_count++; GST_RTP_SESSION_UNLOCK (rtpsession); if (blp == 0) break; seqnum++; while ((blp & 1) == 0) { seqnum++; blp >>= 1; } blp >>= 1; } gst_object_unref (send_rtp_sink); } } static void gst_rtp_session_notify_twcc (RTPSession * sess, GstStructure * twcc_packets, GstStructure * twcc_stats, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); GstEvent *event; GstPad *send_rtp_sink; GST_RTP_SESSION_LOCK (rtpsession); if ((send_rtp_sink = rtpsession->send_rtp_sink)) gst_object_ref (send_rtp_sink); if (rtpsession->priv->last_twcc_stats) gst_structure_free (rtpsession->priv->last_twcc_stats); rtpsession->priv->last_twcc_stats = twcc_stats; GST_RTP_SESSION_UNLOCK (rtpsession); if (send_rtp_sink) { event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, twcc_packets); gst_pad_push_event (send_rtp_sink, event); gst_object_unref (send_rtp_sink); } else { gst_structure_free (twcc_packets); } g_object_notify (G_OBJECT (rtpsession), "twcc-stats"); } static void gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); GstPad *send_rtp_sink; GST_RTP_SESSION_LOCK (rtpsession); if ((send_rtp_sink = rtpsession->send_rtp_sink)) gst_object_ref (send_rtp_sink); GST_RTP_SESSION_UNLOCK (rtpsession); if (send_rtp_sink) { gst_pad_push_event (send_rtp_sink, gst_event_new_reconfigure ()); gst_object_unref (send_rtp_sink); } } static void gst_rtp_session_notify_early_rtcp (RTPSession * sess, gpointer user_data) { GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); GST_DEBUG_OBJECT (rtpsession, "Notified of early RTCP"); /* with an early RTCP request, we might have to start the RTCP thread */ GST_RTP_SESSION_LOCK (rtpsession); signal_waiting_rtcp_thread_unlocked (rtpsession); GST_RTP_SESSION_UNLOCK (rtpsession); }