/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-gstrtpsession * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux * * The RTP session manager models one participant with a 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. * * * Maintainance of the SSRC participant database. * * * Keeping per participant statistics based on received RTCP packets. * * * Scheduling of RR/SR RTCP packets. * * * * The gstrtpsession 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 modify the * SSRC in the RTP packets to its own SSRC and wil 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 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .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 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession 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 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000 * ]| Send theora RTP packets through the session manager and out on UDP port * 5000. * |[ * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession 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. * * * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "gstrtpbin-marshal.h" #include "gstrtpsession.h" #include "rtpsession.h" GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug); #define GST_CAT_DEFAULT gst_rtp_session_debug /* 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, LAST_SIGNAL }; #define DEFAULT_NTP_NS_BASE 0 #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH #define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH #define DEFAULT_SDES NULL #define DEFAULT_NUM_SOURCES 0 #define DEFAULT_NUM_ACTIVE_SOURCES 0 enum { PROP_0, PROP_NTP_NS_BASE, PROP_BANDWIDTH, PROP_RTCP_FRACTION, PROP_SDES, PROP_NUM_SOURCES, PROP_NUM_ACTIVE_SOURCES, PROP_INTERNAL_SESSION, PROP_LAST }; #define GST_RTP_SESSION_GET_PRIVATE(obj) \ (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate)) #define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->priv->lock) #define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->priv->lock) struct _GstRtpSessionPrivate { GMutex *lock; GstClock *sysclock; RTPSession *session; /* thread for sending out RTCP */ GstClockID id; gboolean stop_thread; GThread *thread; gboolean thread_stopped; /* caps mapping */ GHashTable *ptmap; /* NTP base time */ guint64 ntpnsbase; }; /* 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, RTPSource * src, GstBuffer * buffer, gpointer user_data); static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, gpointer user_data); static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data); 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_clock_rate, gst_rtp_session_reconsider }; /* 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); static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad); static void gst_rtp_session_clear_pt_map (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) { g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0, src->ssrc); } 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); } GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT); static void gst_rtp_session_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); /* sink pads */ gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_send_rtp_sink_template)); /* src pads */ gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_recv_rtp_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_sync_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_send_rtp_src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&rtpsession_send_rtcp_src_template)); gst_element_class_set_details_simple (element_class, "RTP Session", "Filter/Network/RTP", "Implement an RTP session", "Wim Taymans "); } static void gst_rtp_session_class_init (GstRtpSessionClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; g_type_class_add_private (klass, sizeof (GstRtpSessionPrivate)); 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, gst_rtp_bin_marshal_BOXED__UINT, 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_ACTION, G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, 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, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE, g_param_spec_uint64 ("ntp-ns-base", "NTP base time", "The NTP base time corresponding to running_time 0 (deprecated)", 0, G_MAXUINT64, DEFAULT_NTP_NS_BASE, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, PROP_BANDWIDTH, g_param_spec_double ("bandwidth", "Bandwidth", "The bandwidth of the session", 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE)); g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION, g_param_spec_double ("rtcp-fraction", "RTCP Fraction", "The fraction of the bandwidth used for RTCP", 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE)); 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_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_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_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)); 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); GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug, "rtpsession", 0, "RTP Session"); } static void gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass) { rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession); rtpsession->priv->lock = g_mutex_new (); rtpsession->priv->sysclock = gst_system_clock_obtain (); rtpsession->priv->session = rtp_session_new (); /* 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); rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) gst_caps_unref); 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; } static void gst_rtp_session_finalize (GObject * object) { GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (object); if (rtpsession->recv_rtp_sink != NULL) gst_object_unref (rtpsession->recv_rtp_sink); if (rtpsession->recv_rtcp_sink != NULL) gst_object_unref (rtpsession->recv_rtcp_sink); if (rtpsession->send_rtp_sink != NULL) gst_object_unref (rtpsession->send_rtp_sink); if (rtpsession->send_rtcp_src != NULL) gst_object_unref (rtpsession->send_rtcp_src); g_hash_table_destroy (rtpsession->priv->ptmap); g_mutex_free (rtpsession->priv->lock); g_object_unref (rtpsession->priv->sysclock); g_object_unref (rtpsession->priv->session); 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_NTP_NS_BASE: GST_OBJECT_LOCK (rtpsession); priv->ntpnsbase = g_value_get_uint64 (value); GST_DEBUG_OBJECT (rtpsession, "setting NTP base to %" GST_TIME_FORMAT, GST_TIME_ARGS (priv->ntpnsbase)); GST_OBJECT_UNLOCK (rtpsession); break; case PROP_BANDWIDTH: rtp_session_set_bandwidth (priv->session, g_value_get_double (value)); break; case PROP_RTCP_FRACTION: rtp_session_set_rtcp_fraction (priv->session, g_value_get_double (value)); break; case PROP_SDES: rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (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_NTP_NS_BASE: GST_OBJECT_LOCK (rtpsession); g_value_set_uint64 (value, priv->ntpnsbase); GST_OBJECT_UNLOCK (rtpsession); break; case PROP_BANDWIDTH: g_value_set_double (value, rtp_session_get_bandwidth (priv->session)); break; case PROP_RTCP_FRACTION: g_value_set_double (value, rtp_session_get_rtcp_fraction (priv->session)); 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; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time, guint64 * ntpnstime) { guint64 ntpns; GstClock *clock; GstClockTime base_time, rt; GTimeVal current; 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 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 = gst_clock_get_time (clock) - base_time; gst_object_unref (clock); } else { GST_OBJECT_UNLOCK (rtpsession); rt = -1; ntpns = -1; } if (running_time) *running_time = rt; if (ntpnstime) *ntpnstime = ntpns; } 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); sysclock = rtpsession->priv->sysclock; current_time = gst_clock_get_time (sysclock); session = rtpsession->priv->session; 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_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &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; 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: break; case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: case GST_STATE_CHANGE_PAUSED_TO_READY: /* 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 = 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); 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) { g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL); } /* called when the session manager has an RTP packet or a list of packets * ready for further processing */ 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) { 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); 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; } /* 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 eos, 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)) { GstCaps *caps; /* set rtcp caps on output pad */ if (!(caps = GST_PAD_CAPS (rtcp_src))) { caps = gst_caps_new_simple ("application/x-rtcp", NULL); gst_pad_set_caps (rtcp_src, caps); } else gst_caps_ref (caps); gst_buffer_set_caps (buffer, caps); gst_caps_unref (caps); gst_object_ref (rtcp_src); GST_RTP_SESSION_UNLOCK (rtpsession); GST_LOG_OBJECT (rtpsession, "sending RTCP"); result = gst_pad_push (rtcp_src, buffer); /* we have to send EOS after this packet */ if (eos) { GST_LOG_OBJECT (rtpsession, "sending EOS"); gst_pad_push_event (rtcp_src, gst_event_new_eos ()); } 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, RTPSource * src, 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)) { GstCaps *caps; /* set rtcp caps on output pad */ if (!(caps = GST_PAD_CAPS (sync_src))) { caps = gst_caps_new_simple ("application/x-rtcp", NULL); gst_pad_set_caps (sync_src, caps); } else gst_caps_ref (caps); gst_buffer_set_caps (buffer, caps); gst_caps_unref (caps); gst_object_ref (sync_src); GST_RTP_SESSION_UNLOCK (rtpsession); 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; g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload), gst_caps_ref (caps)); } /* called when the session manager needs the clock rate */ static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, gpointer user_data) { gint ipayload, result = -1; GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GValue ret = { 0 }; GValue args[2] = { {0}, {0} }; GstCaps *caps; const GstStructure *s; rtpsession = GST_RTP_SESSION_CAST (user_data); priv = rtpsession->priv; GST_RTP_SESSION_LOCK (rtpsession); ipayload = payload; /* make compiler happy */ caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (ipayload)); if (caps) { gst_caps_ref (caps); goto found; } /* 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); g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); 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); found: s = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (s, "clock-rate", &result)) goto no_clock_rate; gst_caps_unref (caps); GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result); done: GST_RTP_SESSION_UNLOCK (rtpsession); return result; /* ERRORS */ no_caps: { GST_DEBUG_OBJECT (rtpsession, "could not get caps"); goto done; } no_clock_rate: { gst_caps_unref (caps); GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!"); goto done; } } /* 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, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (rtpsession, "received event %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { case GST_EVENT_FLUSH_STOP: gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; case GST_EVENT_NEWSEGMENT: { gboolean update; gdouble rate, arate; GstFormat format; gint64 start, stop, time; GstSegment *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_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); GST_DEBUG_OBJECT (rtpsession, "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " "format GST_FORMAT_TIME, " "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, update, rate, arate, GST_TIME_ARGS (segment->start), GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), GST_TIME_ARGS (segment->accum)); gst_segment_set_newsegment_full (segment, update, rate, arate, format, start, stop, time); /* push event forward */ ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } default: ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } gst_object_unref (rtpsession); return ret; } static GstIterator * gst_rtp_session_iterate_internal_links (GstPad * pad) { GstRtpSession *rtpsession; GstPad *otherpad = NULL; GstIterator *it; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); if (pad == rtpsession->recv_rtp_src) { otherpad = rtpsession->recv_rtp_sink; } else if (pad == rtpsession->recv_rtp_sink) { otherpad = rtpsession->recv_rtp_src; } else if (pad == rtpsession->send_rtp_src) { otherpad = rtpsession->send_rtp_sink; } else if (pad == rtpsession->send_rtp_sink) { otherpad = rtpsession->send_rtp_src; } it = gst_iterator_new_single (GST_TYPE_PAD, otherpad, (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref); gst_object_unref (rtpsession); return it; } static gboolean gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps) { GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_RTP_SESSION_LOCK (rtpsession); gst_rtp_session_cache_caps (rtpsession, caps); GST_RTP_SESSION_UNLOCK (rtpsession); gst_object_unref (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, GstBuffer * buffer) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstFlowReturn ret; GstClockTime current_time, running_time; GstClockTime timestamp; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_LOG_OBJECT (rtpsession, "received RTP packet"); /* get NTP time when this packet was captured, this depends on the timestamp. */ timestamp = GST_BUFFER_TIMESTAMP (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); } else { get_current_times (rtpsession, &running_time, NULL); } current_time = gst_clock_get_time (priv->sysclock); ret = rtp_session_process_rtp (priv->session, buffer, current_time, running_time); if (ret != GST_FLOW_OK) goto push_error; done: gst_object_unref (rtpsession); return ret; /* ERRORS */ push_error: { GST_DEBUG_OBJECT (rtpsession, "process returned %s", gst_flow_get_name (ret)); goto done; } } static gboolean gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (rtpsession, "received event %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { default: ret = gst_pad_push_event (rtpsession->sync_src, event); break; } gst_object_unref (rtpsession); 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, GstBuffer * buffer) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstClockTime current_time; GstFlowReturn ret; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_LOG_OBJECT (rtpsession, "received RTCP packet"); current_time = gst_clock_get_time (priv->sysclock); ret = rtp_session_process_rtcp (priv->session, buffer, current_time); gst_object_unref (rtpsession); return GST_FLOW_OK; } static gboolean gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstQuery * query) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (rtpsession, "received 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; } gst_object_unref (rtpsession); return ret; } static gboolean gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (rtpsession, "received 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; } gst_object_unref (rtpsession); return ret; } static gboolean gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); GST_DEBUG_OBJECT (rtpsession, "received event"); switch (GST_EVENT_TYPE (event)) { 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_NEWSEGMENT:{ gboolean update; gdouble rate, arate; GstFormat format; gint64 start, stop, time; GstSegment *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_parse_new_segment_full (event, &update, &rate, &arate, &format, &start, &stop, &time); GST_DEBUG_OBJECT (rtpsession, "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " "format GST_FORMAT_TIME, " "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, update, rate, arate, GST_TIME_ARGS (segment->start), GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), GST_TIME_ARGS (segment->accum)); gst_segment_set_newsegment_full (segment, update, rate, arate, format, start, stop, time); /* 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_DEBUG_OBJECT (rtpsession, "scheduling BYE message"); rtp_session_schedule_bye (rtpsession->priv->session, "End of stream", current_time); break; } default: ret = gst_pad_push_event (rtpsession->send_rtp_src, event); break; } gst_object_unref (rtpsession); return ret; } static GstCaps * gst_rtp_session_getcaps_send_rtp (GstPad * pad) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstCaps *result; GstStructure *s1, *s2; guint ssrc; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; ssrc = rtp_session_get_internal_ssrc (priv->session); /* 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. */ s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL); s2 = gst_structure_new ("application/x-rtp", NULL); result = gst_caps_new_full (s1, s2, NULL); GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result); gst_object_unref (rtpsession); return result; } static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstStructure *s = gst_caps_get_structure (caps, 0); guint ssrc; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; if (gst_structure_get_uint (s, "ssrc", &ssrc)) { GST_DEBUG_OBJECT (rtpsession, "setting internal SSRC to %08x", ssrc); rtp_session_set_internal_ssrc (priv->session, ssrc); } gst_object_unref (rtpsession); return TRUE; } /* Recieve an RTP packet or a list of packets to be send to the receivers, * send to RTP session manager and forward to send_rtp_src. */ static GstFlowReturn gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data, gboolean is_list) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstFlowReturn ret; GstClockTime timestamp, running_time; GstClockTime current_time; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); 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 groups in an list have the same timestamp. * So, just take it from the first group. */ buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0); if (buffer) timestamp = GST_BUFFER_TIMESTAMP (buffer); else timestamp = -1; } else { timestamp = GST_BUFFER_TIMESTAMP (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); } else { /* no timestamp. */ running_time = -1; } current_time = gst_clock_get_time (priv->sysclock); ret = rtp_session_send_rtp (priv->session, data, is_list, current_time, running_time); if (ret != GST_FLOW_OK) goto push_error; done: gst_object_unref (rtpsession); 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, GstBuffer * buffer) { return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE); } static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list) { return gst_rtp_session_chain_send_rtp_common (pad, 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_event_function (rtpsession->recv_rtp_sink, (GstPadEventFunction) gst_rtp_session_event_recv_rtp_sink); gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink, gst_rtp_session_sink_setcaps); gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink, gst_rtp_session_iterate_internal_links); 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_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, (GstPadEventFunction) 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_getcaps_function (rtpsession->send_rtp_sink, gst_rtp_session_getcaps_send_rtp); gst_pad_set_setcaps_function (rtpsession->send_rtp_sink, gst_rtp_session_setcaps_send_rtp); gst_pad_set_event_function (rtpsession->send_rtp_sink, (GstPadEventFunction) 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_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_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) { 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 ("gstrtpsession: this is not our template"); return NULL; } exists: { GST_RTP_SESSION_UNLOCK (rtpsession); g_warning ("gstrtpsession: 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 ("gstrtpsession: asked to release an unknown pad"); return; } }