/* 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 * @short_description: an RTP session manager * @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 /* elementfactory information */ static const GstElementDetails rtpsession_details = GST_ELEMENT_DETAILS ("RTP Session", "Filter/Network/RTP", "Implement an RTP session", "Wim Taymans "); /* 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_BYE_SSRC, SIGNAL_ON_BYE_TIMEOUT, SIGNAL_ON_TIMEOUT, LAST_SIGNAL }; #define DEFAULT_NTP_NS_BASE 0 enum { PROP_0, PROP_NTP_NS_BASE }; #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; RTPSession *session; /* thread for sending out RTCP */ GstClockID id; gboolean stop_thread; GThread *thread; /* 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, GstBuffer * buffer, gpointer user_data); static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, 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_send_rtcp, gst_rtp_session_sync_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_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); } 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 (element_class, &rtpsession_details); } 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-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); 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", 0, G_MAXUINT64, DEFAULT_NTP_NS_BASE, G_PARAM_READWRITE)); 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->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-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); rtpsession->priv->ptmap = g_hash_table_new (NULL, NULL); gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED); } static void gst_rtp_session_finalize (GObject * object) { GstRtpSession *rtpsession; rtpsession = GST_RTP_SESSION (object); g_hash_table_destroy (rtpsession->priv->ptmap); g_mutex_free (rtpsession->priv->lock); 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; rtpsession = GST_RTP_SESSION (object); switch (prop_id) { case PROP_NTP_NS_BASE: GST_OBJECT_LOCK (rtpsession); rtpsession->priv->ntpnsbase = g_value_get_uint64 (value); GST_DEBUG_OBJECT (rtpsession, "setting NTP base to %" GST_TIME_FORMAT, GST_TIME_ARGS (rtpsession->priv->ntpnsbase)); GST_OBJECT_UNLOCK (rtpsession); 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; rtpsession = GST_RTP_SESSION (object); switch (prop_id) { case PROP_NTP_NS_BASE: GST_OBJECT_LOCK (rtpsession); g_value_set_uint64 (value, rtpsession->priv->ntpnsbase); GST_OBJECT_UNLOCK (rtpsession); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static guint64 get_current_ntp_ns_time (GstRtpSession * rtpsession) { guint64 ntpnstime; GstClock *clock; GstClockTime base_time, ntpnsbase; GST_OBJECT_LOCK (rtpsession); if ((clock = GST_ELEMENT_CLOCK (rtpsession))) { base_time = GST_ELEMENT_CAST (rtpsession)->base_time; ntpnsbase = rtpsession->priv->ntpnsbase; gst_object_ref (clock); GST_OBJECT_UNLOCK (rtpsession); /* get current NTP time */ ntpnstime = gst_clock_get_time (clock); /* convert to running time */ ntpnstime -= base_time; /* add NTP base offset */ ntpnstime += ntpnsbase; gst_object_unref (clock); } else { GST_OBJECT_UNLOCK (rtpsession); ntpnstime = -1; } return ntpnstime; } static void rtcp_thread (GstRtpSession * rtpsession) { GstClock *sysclock; GstClockID id; GstClockTime current_time; GstClockTime next_timeout; guint64 ntpnstime; /* for RTCP timeouts we use the system clock */ sysclock = gst_system_clock_obtain (); if (sysclock == NULL) goto no_sysclock; current_time = gst_clock_get_time (sysclock); GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread"); GST_RTP_SESSION_LOCK (rtpsession); while (!rtpsession->priv->stop_thread) { GstClockReturn res; /* get initial estimate */ next_timeout = rtp_session_next_timeout (rtpsession->priv->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 */ ntpnstime = get_current_ntp_ns_time (rtpsession); /* 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 (rtpsession->priv->session, current_time, ntpnstime); GST_RTP_SESSION_LOCK (rtpsession); } GST_RTP_SESSION_UNLOCK (rtpsession); gst_object_unref (sysclock); GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread"); return; /* ERRORS */ no_sysclock: { GST_ELEMENT_ERROR (rtpsession, CORE, CLOCK, (NULL), ("Could not get system clock")); return; } } 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; rtpsession->priv->thread = g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error); 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); /* FIXME, can deadlock because the thread might be blocked in a push */ g_thread_join (rtpsession->priv->thread); } static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn res; GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (element); priv = rtpsession->priv; 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: 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: break; case GST_STATE_CHANGE_READY_TO_NULL: break; default: break; } return res; /* ERRORS */ failed_thread: { return GST_STATE_CHANGE_FAILURE; } } static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession) { /* FIXME, do something */ } /* called when the session manager has an RTP packet ready for further * processing */ static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (user_data); priv = rtpsession->priv; if (rtpsession->recv_rtp_src) { GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet"); result = gst_pad_push (rtpsession->recv_rtp_src, buffer); } 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, GstBuffer * buffer, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (user_data); priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "sending RTP packet"); if (rtpsession->send_rtp_src) { result = gst_pad_push (rtpsession->send_rtp_src, buffer); } else { gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; } /* called when the session manager has an RTCP packet ready for further * sending */ static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, GstBuffer * buffer, gpointer user_data) { GstFlowReturn result; GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (user_data); priv = rtpsession->priv; if (rtpsession->send_rtcp_src) { GstCaps *caps; /* set rtcp caps on output pad */ if (!(caps = GST_PAD_CAPS (rtpsession->send_rtcp_src))) { caps = gst_caps_new_simple ("application/x-rtcp", NULL); gst_pad_set_caps (rtpsession->send_rtcp_src, caps); gst_caps_unref (caps); } gst_buffer_set_caps (buffer, caps); GST_DEBUG_OBJECT (rtpsession, "sending RTCP"); result = gst_pad_push (rtpsession->send_rtcp_src, buffer); } else { GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; } /* 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; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (user_data); priv = rtpsession->priv; if (rtpsession->sync_src) { GstCaps *caps; /* set rtcp caps on output pad */ if (!(caps = GST_PAD_CAPS (rtpsession->sync_src))) { caps = gst_caps_new_simple ("application/x-rtcp", NULL); gst_pad_set_caps (rtpsession->sync_src, caps); gst_caps_unref (caps); } gst_buffer_set_caps (buffer, caps); GST_DEBUG_OBJECT (rtpsession, "sending Sync RTCP"); result = gst_pad_push (rtpsession->sync_src, buffer); } else { GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad"); gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; } 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; caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload)); if (caps) return; g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload), 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) goto done; 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); caps = (GstCaps *) g_value_get_boxed (&ret); if (!caps) goto no_caps; gst_rtp_session_cache_caps (rtpsession, caps); s = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (s, "clock-rate", &result)) goto no_clock_rate; 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_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 GstFlowReturn gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; 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); 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 GList * gst_rtp_session_internal_links (GstPad * pad) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GList *res = NULL; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; if (pad == rtpsession->recv_rtp_src) { res = g_list_prepend (res, rtpsession->recv_rtp_sink); } else if (pad == rtpsession->recv_rtp_sink) { res = g_list_prepend (res, rtpsession->recv_rtp_src); } else if (pad == rtpsession->send_rtp_src) { res = g_list_prepend (res, rtpsession->send_rtp_sink); } else if (pad == rtpsession->send_rtp_sink) { res = g_list_prepend (res, rtpsession->send_rtp_src); } gst_object_unref (rtpsession); return res; } static gboolean gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; 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; guint64 ntpnstime; GstClockTime timestamp; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_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 */ ntpnstime = gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME, timestamp); /* add constant to convert running time to NTP time */ ntpnstime += priv->ntpnsbase; } else { ntpnstime = get_current_ntp_ns_time (rtpsession); } ret = rtp_session_process_rtp (priv->session, buffer, ntpnstime); 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_event_recv_rtcp_sink (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "received event %s", GST_EVENT_TYPE_NAME (event)); switch (GST_EVENT_TYPE (event)) { default: if (rtpsession->send_rtcp_src) { gst_event_ref (event); ret = gst_pad_push_event (rtpsession->send_rtcp_src, event); } 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; GstFlowReturn ret; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "received RTCP packet"); ret = rtp_session_process_rtcp (priv->session, buffer); gst_object_unref (rtpsession); return GST_FLOW_OK; } static GstFlowReturn gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; gboolean ret = FALSE; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; 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); 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; } 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; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; /* 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, priv->session->source->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; } /* Recieve an RTP packet 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 (GstPad * pad, GstBuffer * buffer) { GstRtpSession *rtpsession; GstRtpSessionPrivate *priv; GstFlowReturn ret; GstClockTime timestamp; guint64 ntpnstime; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_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 start value. */ ntpnstime = gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, timestamp); /* convert to NTP time by adding the NTP base */ ntpnstime += priv->ntpnsbase; } else { /* no timestamp, we could take the current running_time and convert it to * NTP time. */ ntpnstime = -1; } ret = rtp_session_send_rtp (priv->session, buffer, ntpnstime); 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; } } /* 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, gst_rtp_session_event_recv_rtp_sink); gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink, gst_rtp_session_sink_setcaps); gst_pad_set_internal_link_function (rtpsession->recv_rtp_sink, gst_rtp_session_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_internal_link_function (rtpsession->recv_rtp_src, gst_rtp_session_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; } /* 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_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_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; } /* 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_getcaps_function (rtpsession->send_rtp_sink, gst_rtp_session_getcaps_send_rtp); gst_pad_set_event_function (rtpsession->send_rtp_sink, gst_rtp_session_event_send_rtp_sink); gst_pad_set_internal_link_function (rtpsession->send_rtp_sink, gst_rtp_session_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_use_fixed_caps (rtpsession->send_rtp_src); gst_pad_set_internal_link_function (rtpsession->send_rtp_src, gst_rtp_session_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; } /* 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_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtcp_src); return rtpsession->send_rtcp_src; } 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) { }