/* 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-rtpsession * @short_description: an RTP session manager * @see_also: rtpjitterbuffer, rtpbin * * * * * Example pipelines * * * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink * * * * * Last reviewed on 2007-04-02 (0.10.6) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #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/Editor/Video", "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, LAST_SIGNAL }; enum { PROP_0 }; #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; }; /* 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 gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, gpointer user_data); static GstClockTime gst_rtp_session_get_time (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_clock_rate, gst_rtp_session_get_time }; /* 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 guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; 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); 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); 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); } static void gst_rtp_session_finalize (GObject * object) { GstRTPSession *rtpsession; rtpsession = GST_RTP_SESSION (object); 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) { 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) { default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void rtcp_thread (GstRTPSession * rtpsession) { GstClock *clock; GstClockID id; clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession)); if (clock == NULL) return; GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread"); GST_RTP_SESSION_LOCK (rtpsession); while (!rtpsession->priv->stop_thread) { gdouble timeout; GstClockTime target; GstClockReturn res; timeout = rtp_session_get_reporting_interval (rtpsession->priv->session); GST_DEBUG_OBJECT (rtpsession, "next RTCP timeout: %lf", timeout); target = gst_clock_get_time (clock); target += GST_SECOND * timeout; id = rtpsession->priv->id = gst_clock_new_single_shot_id (clock, target); GST_RTP_SESSION_UNLOCK (rtpsession); res = gst_clock_id_wait (id, NULL); if (res != GST_CLOCK_UNSCHEDULED) { GST_DEBUG_OBJECT (rtpsession, "got RTCP timeout"); /* make the session manager produce RTCP, we ignore the result. */ rtp_session_perform_reporting (rtpsession->priv->session); } else { GST_DEBUG_OBJECT (rtpsession, "got unscheduled"); } GST_RTP_SESSION_LOCK (rtpsession); gst_clock_id_unref (id); rtpsession->priv->id = NULL; } GST_RTP_SESSION_UNLOCK (rtpsession); gst_object_unref (clock); 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; 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); g_thread_join (rtpsession->priv->thread); } 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: stop_rtcp_thread (rtpsession); 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; } } /* 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) { result = gst_pad_push (rtpsession->recv_rtp_src, buffer); } else { 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; 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; GST_DEBUG_OBJECT (rtpsession, "sending RTCP"); if (rtpsession->send_rtcp_src) { result = gst_pad_push (rtpsession->send_rtcp_src, buffer); } else { gst_buffer_unref (buffer); result = GST_FLOW_OK; } return result; } /* called when the session manager needs the clock rate */ static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, gpointer user_data) { gint result = -1; GstRTPSession *rtpsession; GValue ret = { 0 }; GValue args[2] = { {0}, {0} }; GstCaps *caps; const GstStructure *caps_struct; rtpsession = GST_RTP_SESSION_CAST (user_data); 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; caps_struct = gst_caps_get_structure (caps, 0); if (!gst_structure_get_int (caps_struct, "clock-rate", &result)) goto no_clock_rate; GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result); return result; /* ERRORS */ no_caps: { GST_DEBUG_OBJECT (rtpsession, "could not get caps"); return -1; } no_clock_rate: { GST_DEBUG_OBJECT (rtpsession, "could not clock-rate from caps"); return -1; } } /* called when the session manager needs the time of clock */ static GstClockTime gst_rtp_session_get_time (RTPSession * sess, gpointer user_data) { GstClockTime result; GstRTPSession *rtpsession; GstClock *clock; rtpsession = GST_RTP_SESSION_CAST (user_data); clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession)); if (clock) { result = gst_clock_get_time (clock); gst_object_unref (clock); } else result = GST_CLOCK_TIME_NONE; return result; } 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)) { default: ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); break; } gst_object_unref (rtpsession); return ret; } /* 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; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "received RTP packet"); ret = rtp_session_process_rtp (priv->session, buffer); gst_object_unref (rtpsession); return ret; } 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: 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)) { default: ret = gst_pad_push_event (rtpsession->send_rtp_src, event); break; } gst_object_unref (rtpsession); return ret; } /* 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; rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); priv = rtpsession->priv; GST_DEBUG_OBJECT (rtpsession, "received RTP packet"); ret = rtp_session_send_rtp (priv->session, buffer); gst_object_unref (rtpsession); return ret; } /* 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, NULL); 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_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_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, NULL); 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_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, NULL); gst_pad_set_chain_function (rtpsession->send_rtp_sink, gst_rtp_session_chain_send_rtp); gst_pad_set_event_function (rtpsession->send_rtp_sink, gst_rtp_session_event_send_rtp_sink); gst_pad_set_active (rtpsession->send_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtcp_sink); rtpsession->send_rtp_src = gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template, NULL); 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, NULL); 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 ("rtpsession: this is not our template"); return NULL; } exists: { GST_RTP_SESSION_UNLOCK (rtpsession); g_warning ("rtpsession: pad already requested"); return NULL; } } static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad) { }