/* GStreamer * Copyright (C) <2005,2006> Wim Taymans * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /* * Unless otherwise indicated, Source Code is licensed under MIT license. * See further explanation attached in License Statement (distributed in the file * LICENSE). * * Permission is hereby granted, free of charge, to any person obtaining a copy of * this software and associated documentation files (the "Software"), to deal in * the Software without restriction, including without limitation the rights to * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies * of the Software, and to permit persons to whom the Software is furnished to do * so, subject to the following conditions: * * The above copyright notice and this permission notice shall be included in all * copies or substantial portions of the Software. * * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE * SOFTWARE. */ /* Element-Checklist-Version: 5 */ /** * SECTION:element-rtpdec * @title: rtpdec * * A simple RTP session manager used internally by rtspsrc. */ /* #define HAVE_RTCP */ #include #ifdef HAVE_RTCP #include #endif #include "gstrtpdec.h" #include GST_DEBUG_CATEGORY_STATIC (rtpdec_debug); #define GST_CAT_DEFAULT (rtpdec_debug) /* GstRTPDec 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_LATENCY_MS 200 enum { PROP_0, PROP_LATENCY }; static GstStaticPadTemplate gst_rtp_dec_recv_rtp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate gst_rtp_dec_recv_rtcp_sink_template = GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u", GST_PAD_SINK, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp") ); static GstStaticPadTemplate gst_rtp_dec_recv_rtp_src_template = GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u", GST_PAD_SRC, GST_PAD_SOMETIMES, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate gst_rtp_dec_rtcp_src_template = GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u", GST_PAD_SRC, GST_PAD_REQUEST, GST_STATIC_CAPS ("application/x-rtcp") ); static void gst_rtp_dec_finalize (GObject * object); static void gst_rtp_dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstClock *gst_rtp_dec_provide_clock (GstElement * element); static GstStateChangeReturn gst_rtp_dec_change_state (GstElement * element, GstStateChange transition); static GstPad *gst_rtp_dec_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps); static void gst_rtp_dec_release_pad (GstElement * element, GstPad * pad); static GstFlowReturn gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstFlowReturn gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer); /* Manages the receiving end of the packets. * * There is one such structure for each RTP session (audio/video/...). * We get the RTP/RTCP packets and stuff them into the session manager. */ struct _GstRTPDecSession { /* session id */ gint id; /* the parent bin */ GstRTPDec *dec; gboolean active; /* we only support one ssrc and one pt */ guint32 ssrc; guint8 pt; GstCaps *caps; /* the pads of the session */ GstPad *recv_rtp_sink; GstPad *recv_rtp_src; GstPad *recv_rtcp_sink; GstPad *rtcp_src; }; /* find a session with the given id */ static GstRTPDecSession * find_session_by_id (GstRTPDec * rtpdec, gint id) { GSList *walk; for (walk = rtpdec->sessions; walk; walk = g_slist_next (walk)) { GstRTPDecSession *sess = (GstRTPDecSession *) walk->data; if (sess->id == id) return sess; } return NULL; } /* create a session with the given id */ static GstRTPDecSession * create_session (GstRTPDec * rtpdec, gint id) { GstRTPDecSession *sess; sess = g_new0 (GstRTPDecSession, 1); sess->id = id; sess->dec = rtpdec; rtpdec->sessions = g_slist_prepend (rtpdec->sessions, sess); return sess; } static void free_session (GstRTPDecSession * session) { g_free (session); } static guint gst_rtp_dec_signals[LAST_SIGNAL] = { 0 }; #define gst_rtp_dec_parent_class parent_class G_DEFINE_TYPE (GstRTPDec, gst_rtp_dec, GST_TYPE_ELEMENT); static void gst_rtp_dec_class_init (GstRTPDecClass * g_class) { GObjectClass *gobject_class; GstElementClass *gstelement_class; GstRTPDecClass *klass; klass = (GstRTPDecClass *) g_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder"); gobject_class->finalize = gst_rtp_dec_finalize; gobject_class->set_property = gst_rtp_dec_set_property; gobject_class->get_property = gst_rtp_dec_get_property; g_object_class_install_property (gobject_class, PROP_LATENCY, g_param_spec_uint ("latency", "Buffer latency in ms", "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); /** * GstRTPDec::request-pt-map: * @rtpdec: the object which received the signal * @session: the session * @pt: the pt * * Request the payload type as #GstCaps for @pt in @session. */ gst_rtp_dec_signals[SIGNAL_REQUEST_PT_MAP] = g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, request_pt_map), NULL, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 2, G_TYPE_UINT, G_TYPE_UINT); gst_rtp_dec_signals[SIGNAL_CLEAR_PT_MAP] = g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, clear_pt_map), NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); /** * GstRTPDec::on-new-ssrc: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a new SSRC that entered @session. */ gst_rtp_dec_signals[SIGNAL_ON_NEW_SSRC] = g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_new_ssrc), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRTPDec::on-ssrc_collision: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify when we have an SSRC collision */ gst_rtp_dec_signals[SIGNAL_ON_SSRC_COLLISION] = g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_collision), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRTPDec::on-ssrc_validated: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of a new SSRC that became validated. */ gst_rtp_dec_signals[SIGNAL_ON_SSRC_VALIDATED] = g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_validated), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRTPDec::on-bye-ssrc: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that became inactive because of a BYE packet. */ gst_rtp_dec_signals[SIGNAL_ON_BYE_SSRC] = g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_ssrc), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRTPDec::on-bye-timeout: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that has timed out because of BYE */ gst_rtp_dec_signals[SIGNAL_ON_BYE_TIMEOUT] = g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_timeout), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); /** * GstRTPDec::on-timeout: * @rtpbin: the object which received the signal * @session: the session * @ssrc: the SSRC * * Notify of an SSRC that has timed out */ gst_rtp_dec_signals[SIGNAL_ON_TIMEOUT] = g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_timeout), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT); gstelement_class->provide_clock = GST_DEBUG_FUNCPTR (gst_rtp_dec_provide_clock); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_dec_change_state); gstelement_class->request_new_pad = GST_DEBUG_FUNCPTR (gst_rtp_dec_request_new_pad); gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_dec_release_pad); /* sink pads */ gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dec_recv_rtp_sink_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dec_recv_rtcp_sink_template); /* src pads */ gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dec_recv_rtp_src_template); gst_element_class_add_static_pad_template (gstelement_class, &gst_rtp_dec_rtcp_src_template); gst_element_class_set_static_metadata (gstelement_class, "RTP Decoder", "Codec/Parser/Network", "Accepts raw RTP and RTCP packets and sends them forward", "Wim Taymans "); } static void gst_rtp_dec_init (GstRTPDec * rtpdec) { rtpdec->provided_clock = gst_system_clock_obtain (); rtpdec->latency = DEFAULT_LATENCY_MS; GST_OBJECT_FLAG_SET (rtpdec, GST_ELEMENT_FLAG_PROVIDE_CLOCK); } static void gst_rtp_dec_finalize (GObject * object) { GstRTPDec *rtpdec; rtpdec = GST_RTP_DEC (object); gst_object_unref (rtpdec->provided_clock); g_slist_foreach (rtpdec->sessions, (GFunc) free_session, NULL); g_slist_free (rtpdec->sessions); G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_rtp_dec_query_src (GstPad * pad, GstObject * parent, GstQuery * query) { gboolean res; switch (GST_QUERY_TYPE (query)) { case GST_QUERY_LATENCY: { /* we pretend to be live with a 3 second latency */ /* FIXME: Do we really have infinite maximum latency? */ gst_query_set_latency (query, TRUE, 3 * GST_SECOND, -1); res = TRUE; break; } default: res = gst_pad_query_default (pad, parent, query); break; } return res; } static GstFlowReturn gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstFlowReturn res; GstRTPDec *rtpdec; GstRTPDecSession *session; guint32 ssrc; guint8 pt; GstRTPBuffer rtp = { NULL, }; rtpdec = GST_RTP_DEC (parent); GST_DEBUG_OBJECT (rtpdec, "got rtp packet"); if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) goto bad_packet; ssrc = gst_rtp_buffer_get_ssrc (&rtp); pt = gst_rtp_buffer_get_payload_type (&rtp); gst_rtp_buffer_unmap (&rtp); GST_DEBUG_OBJECT (rtpdec, "SSRC %08x, PT %d", ssrc, pt); /* find session */ session = gst_pad_get_element_private (pad); /* see if we have the pad */ if (!session->active) { GstPadTemplate *templ; GstElementClass *klass; gchar *name; GstCaps *caps; GValue ret = { 0 }; GValue args[3] = { {0} , {0} , {0} }; GST_DEBUG_OBJECT (rtpdec, "creating stream"); session->ssrc = ssrc; session->pt = pt; /* get pt map */ g_value_init (&args[0], GST_TYPE_ELEMENT); g_value_set_object (&args[0], rtpdec); g_value_init (&args[1], G_TYPE_UINT); g_value_set_uint (&args[1], session->id); g_value_init (&args[2], G_TYPE_UINT); g_value_set_uint (&args[2], pt); g_value_init (&ret, GST_TYPE_CAPS); g_value_set_boxed (&ret, NULL); g_signal_emitv (args, gst_rtp_dec_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret); caps = (GstCaps *) g_value_get_boxed (&ret); name = g_strdup_printf ("recv_rtp_src_%u_%u_%u", session->id, ssrc, pt); klass = GST_ELEMENT_GET_CLASS (rtpdec); templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u"); session->recv_rtp_src = gst_pad_new_from_template (templ, name); g_free (name); gst_pad_set_caps (session->recv_rtp_src, caps); gst_pad_set_element_private (session->recv_rtp_src, session); gst_pad_set_query_function (session->recv_rtp_src, gst_rtp_dec_query_src); gst_pad_set_active (session->recv_rtp_src, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_src); session->active = TRUE; } res = gst_pad_push (session->recv_rtp_src, buffer); return res; bad_packet: { GST_ELEMENT_WARNING (rtpdec, STREAM, DECODE, (NULL), ("RTP packet did not validate, dropping")); gst_buffer_unref (buffer); return GST_FLOW_OK; } } static GstFlowReturn gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRTPDec *src; #ifdef HAVE_RTCP gboolean valid; GstRTCPPacket packet; gboolean more; #endif src = GST_RTP_DEC (parent); GST_DEBUG_OBJECT (src, "got rtcp packet"); #ifdef HAVE_RTCP valid = gst_rtcp_buffer_validate (buffer); if (!valid) goto bad_packet; /* position on first packet */ more = gst_rtcp_buffer_get_first_packet (buffer, &packet); while (more) { switch (gst_rtcp_packet_get_type (&packet)) { case GST_RTCP_TYPE_SR: { guint32 ssrc, rtptime, packet_count, octet_count; guint64 ntptime; guint count, i; gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime, &packet_count, &octet_count); GST_DEBUG_OBJECT (src, "got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT ", RTP %u, PC %u, OC %u", ssrc, ntptime, rtptime, packet_count, octet_count); count = gst_rtcp_packet_get_rb_count (&packet); for (i = 0; i < count; i++) { guint32 ssrc, exthighestseq, jitter, lsr, dlsr; guint8 fractionlost; gint32 packetslost; gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u" ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } break; } case GST_RTCP_TYPE_RR: { guint32 ssrc; guint count, i; ssrc = gst_rtcp_packet_rr_get_ssrc (&packet); GST_DEBUG_OBJECT (src, "got RR packet: SSRC %08x", ssrc); count = gst_rtcp_packet_get_rb_count (&packet); for (i = 0; i < count; i++) { guint32 ssrc, exthighestseq, jitter, lsr, dlsr; guint8 fractionlost; gint32 packetslost; gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u" ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } break; } case GST_RTCP_TYPE_SDES: { guint chunks, i, j; gboolean more_chunks, more_items; chunks = gst_rtcp_packet_sdes_get_chunk_count (&packet); GST_DEBUG_OBJECT (src, "got SDES packet with %d chunks", chunks); more_chunks = gst_rtcp_packet_sdes_first_chunk (&packet); i = 0; while (more_chunks) { guint32 ssrc; ssrc = gst_rtcp_packet_sdes_get_ssrc (&packet); GST_DEBUG_OBJECT (src, "chunk %d, SSRC %08x", i, ssrc); more_items = gst_rtcp_packet_sdes_first_item (&packet); j = 0; while (more_items) { GstRTCPSDESType type; guint8 len; gchar *data; gst_rtcp_packet_sdes_get_item (&packet, &type, &len, &data); GST_DEBUG_OBJECT (src, "item %d, type %d, len %d, data %s", j, type, len, data); more_items = gst_rtcp_packet_sdes_next_item (&packet); j++; } more_chunks = gst_rtcp_packet_sdes_next_chunk (&packet); i++; } break; } case GST_RTCP_TYPE_BYE: { guint count, i; gchar *reason; reason = gst_rtcp_packet_bye_get_reason (&packet); GST_DEBUG_OBJECT (src, "got BYE packet (reason: %s)", GST_STR_NULL (reason)); g_free (reason); count = gst_rtcp_packet_bye_get_ssrc_count (&packet); for (i = 0; i < count; i++) { guint32 ssrc; ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, i); GST_DEBUG_OBJECT (src, "SSRC: %08x", ssrc); } break; } case GST_RTCP_TYPE_APP: GST_DEBUG_OBJECT (src, "got APP packet"); break; default: GST_WARNING_OBJECT (src, "got unknown RTCP packet"); break; } more = gst_rtcp_packet_move_to_next (&packet); } gst_buffer_unref (buffer); return GST_FLOW_OK; bad_packet: { GST_WARNING_OBJECT (src, "got invalid RTCP packet"); gst_buffer_unref (buffer); return GST_FLOW_OK; } #else gst_buffer_unref (buffer); return GST_FLOW_OK; #endif } static void gst_rtp_dec_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRTPDec *src; src = GST_RTP_DEC (object); switch (prop_id) { case PROP_LATENCY: src->latency = g_value_get_uint (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_rtp_dec_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRTPDec *src; src = GST_RTP_DEC (object); switch (prop_id) { case PROP_LATENCY: g_value_set_uint (value, src->latency); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstClock * gst_rtp_dec_provide_clock (GstElement * element) { GstRTPDec *rtpdec; rtpdec = GST_RTP_DEC (element); return GST_CLOCK_CAST (gst_object_ref (rtpdec->provided_clock)); } static GstStateChangeReturn gst_rtp_dec_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_PAUSED: case GST_STATE_CHANGE_PLAYING_TO_PAUSED: /* we're NO_PREROLL when going to PAUSED */ ret = GST_STATE_CHANGE_NO_PREROLL; break; default: break; } return ret; } /* Create a pad for receiving RTP for the session in @name */ static GstPad * create_recv_rtp (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name) { guint sessid; GstRTPDecSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1) goto no_name; GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid); /* get or create session */ session = find_session_by_id (rtpdec, sessid); if (!session) { GST_DEBUG_OBJECT (rtpdec, "creating session %d", sessid); /* create session now */ session = create_session (rtpdec, sessid); if (session == NULL) goto create_error; } /* check if pad was requested */ if (session->recv_rtp_sink != NULL) goto existed; GST_DEBUG_OBJECT (rtpdec, "getting RTP sink pad"); session->recv_rtp_sink = gst_pad_new_from_template (templ, name); gst_pad_set_element_private (session->recv_rtp_sink, session); gst_pad_set_chain_function (session->recv_rtp_sink, gst_rtp_dec_chain_rtp); gst_pad_set_active (session->recv_rtp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_sink); return session->recv_rtp_sink; /* ERRORS */ no_name: { g_warning ("rtpdec: invalid name given"); return NULL; } create_error: { /* create_session already warned */ return NULL; } existed: { g_warning ("rtpdec: recv_rtp pad already requested for session %d", sessid); return NULL; } } /* Create a pad for receiving RTCP for the session in @name */ static GstPad * create_recv_rtcp (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name) { guint sessid; GstRTPDecSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1) goto no_name; GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid); /* get the session, it must exist or we error */ session = find_session_by_id (rtpdec, sessid); if (!session) goto no_session; /* check if pad was requested */ if (session->recv_rtcp_sink != NULL) goto existed; GST_DEBUG_OBJECT (rtpdec, "getting RTCP sink pad"); session->recv_rtcp_sink = gst_pad_new_from_template (templ, name); gst_pad_set_element_private (session->recv_rtp_sink, session); gst_pad_set_chain_function (session->recv_rtcp_sink, gst_rtp_dec_chain_rtcp); gst_pad_set_active (session->recv_rtcp_sink, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtcp_sink); return session->recv_rtcp_sink; /* ERRORS */ no_name: { g_warning ("rtpdec: invalid name given"); return NULL; } no_session: { g_warning ("rtpdec: no session with id %d", sessid); return NULL; } existed: { g_warning ("rtpdec: recv_rtcp pad already requested for session %d", sessid); return NULL; } } /* Create a pad for sending RTCP for the session in @name */ static GstPad * create_rtcp (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name) { guint sessid; GstRTPDecSession *session; /* first get the session number */ if (name == NULL || sscanf (name, "rtcp_src_%u", &sessid) != 1) goto no_name; /* get or create session */ session = find_session_by_id (rtpdec, sessid); if (!session) goto no_session; /* check if pad was requested */ if (session->rtcp_src != NULL) goto existed; session->rtcp_src = gst_pad_new_from_template (templ, name); gst_pad_set_active (session->rtcp_src, TRUE); gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->rtcp_src); return session->rtcp_src; /* ERRORS */ no_name: { g_warning ("rtpdec: invalid name given"); return NULL; } no_session: { g_warning ("rtpdec: session with id %d does not exist", sessid); return NULL; } existed: { g_warning ("rtpdec: rtcp_src pad already requested for session %d", sessid); return NULL; } } /* */ static GstPad * gst_rtp_dec_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name, const GstCaps * caps) { GstRTPDec *rtpdec; GstElementClass *klass; GstPad *result; g_return_val_if_fail (templ != NULL, NULL); g_return_val_if_fail (GST_IS_RTP_DEC (element), NULL); rtpdec = GST_RTP_DEC (element); klass = GST_ELEMENT_GET_CLASS (element); /* figure out the template */ if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) { result = create_recv_rtp (rtpdec, templ, name); } else if (templ == gst_element_class_get_pad_template (klass, "recv_rtcp_sink_%u")) { result = create_recv_rtcp (rtpdec, templ, name); } else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%u")) { result = create_rtcp (rtpdec, templ, name); } else goto wrong_template; return result; /* ERRORS */ wrong_template: { g_warning ("rtpdec: this is not our template"); return NULL; } } static void gst_rtp_dec_release_pad (GstElement * element, GstPad * pad) { }