/* RTP Retransmission receiver element for GStreamer * * gstrtprtxreceive.c: * * Copyright (C) 2013 Collabora Ltd. * @author Julien Isorce * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-rtprtxreceive * @see_also: rtprtxsend, rtpsession, rtpjitterbuffer * * The receiver will listen to the custom retransmission events from the * downstream jitterbuffer and will remember the SSRC1 of the stream and * seqnum that was requested. When it sees a packet with one of the stored * seqnum, it associates the SSRC2 of the stream with the SSRC1 of the * master stream. From then it knows that SSRC2 is the retransmission * stream of SSRC1. This algorithm is stated in RFC 4588. For this * algorithm to work, RFC4588 also states that no two pending retransmission * requests can exist for the same seqnum and different SSRCs or else it * would be impossible to associate the retransmission with the original * requester SSRC. * When the RTX receiver has associated the retransmission packets, * it can depayload and forward them to the source pad of the element. * RTX is SSRC-multiplexed. See #GstRtpRtxSend * * * Example pipelines * |[ * gst-launch-1.0 rtpsession name=rtpsession \ * audiotestsrc ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 ! \ * identity drop-probability=0.1 ! rtpsession.send_rtp_sink \ * rtpsession.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \ * udpsrc port=5001 ! rtpsession.recv_rtcp_sink \ * rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false * ]| Send audio stream through port 5000. (5001 and 5002 are just the rtcp link with the receiver) * |[ * gst-launch-1.0 rtpsession name=rtpsession \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1" ! \ * rtpsession.recv_rtp_sink \ * rtpsession.recv_rtp_src ! rtprtxreceive rtx-payload-types="99" ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! \ * speexdec ! audioconvert ! autoaudiosink \ * rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink sync=fakse async=false * ]| Receive audio stream from port 5000. (5001 and 5002 are just the rtcp link with the sender) * On sender side make sure to use a different payload type for the stream and * its associated retransmission stream (see #GstRtpRtxSend). Note that several retransmission streams can * have the same payload type so this is not deterministic. Actually the * rtprtxreceiver element does the association using seqnum values. * On receiver side set all the retransmission payload types (Those informations are retrieve * through SDP). * You should still hear a clear sound when setting drop-probability to something greater than 0. * The rtpjitterbuffer will generate a custom upstream event GstRTPRetransmissionRequest when * it assumes that one packet is missing. Then this request is translated to a FB NACK in the rtcp link * Finally the rtpsession of the sender side re-convert it in a GstRTPRetransmissionRequest that will * be handle by rtprtxsend. * When increasing this value it may be possible that even the retransmission stream would be dropped * so the receiver will ask to resend the packets again and again until it actually receive them. * If the value is too high the rtprtxsend will not be able to retrieve the packet in its list of * stored packets. For learning purpose you could try to increase the max-size-packets or max-size-time * rtprtxsender's properties. * Also note that you should use rtprtxsend through rtpbin and its set-aux-send property. See #GstRtpBin. * |[ * gst-launch-1.0 rtpsession name=rtpsession0 \ * audiotestsrc wave=0 ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 seqnum-offset=1 ! \ * identity drop-probability=0.1 ! rtpsession0.send_rtp_sink \ * rtpsession0.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \ * udpsrc port=5001 ! rtpsession0.recv_rtcp_sink \ * rtpsession0.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false \ * rtpsession name=rtpsession1 \ * audiotestsrc wave=0 ! speexenc ! rtpspeexpay pt=97 ! rtprtxsend rtx-payload-type=99 seqnum-offset=10 ! \ * identity drop-probability=0.1 ! rtpsession1.send_rtp_sink \ * rtpsession1.send_rtp_src ! udpsink host="127.0.0.1" port=5000 \ * udpsrc port=5004 ! rtpsession1.recv_rtcp_sink \ * rtpsession1.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 sync=false async=false * ]| Send two audio streams to port 5000. * |[ * gst-launch-1.0 rtpsession name=rtpsession * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)SPEEX,encoding-params=(string)1,octet-align=(string)1" ! \ * rtpsession.recv_rtp_sink \ * rtpsession.recv_rtp_src ! rtprtxreceive rtx-payload-types="99" ! rtpssrcdemux name=demux \ * demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink \ * demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpspeexdepay ! speexdec ! audioconvert ! autoaudiosink \ * rtpsession.send_rtcp_src ! ! tee name=t ! queue ! udpsink host="127.0.0.1" port=5001 t. ! queue ! udpsink host="127.0.0.1" port=5004 \ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink sync=fakse async=false * ]| Receive audio stream from port 5000. * On sender side the two streams have the same payload type for master streams, Same about retransmission streams. * The streams are sent to the network through two distincts sessions. * But we need to set a different seqnum-offset to make sure their seqnum navigate at a different rate like in concrete cases. * We could also choose the same seqnum offset but we would require to set a different initial seqnum value. * This is also why the rtprtxreceive can succeed to do the association between master and retransmission stream. * On receiver side the same session is used to receive the two streams. So the rtpssrcdemux is here to demultiplex * those two streams. The rtprtxreceive is responsible for reconstructing the original packets from the two retransmission streams. * You can play with the drop-probability value for one or both streams. * You should hear a clear sound. (after a few seconds the two streams wave feel synchronized) * * * Last reviewed on 2013-11-08 (1.x) */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include "gstrtprtxreceive.h" GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_receive_debug); #define GST_CAT_DEFAULT gst_rtp_rtx_receive_debug enum { PROP_0, PROP_PAYLOAD_TYPE_MAP, PROP_NUM_RTX_REQUESTS, PROP_NUM_RTX_PACKETS, PROP_NUM_RTX_ASSOC_PACKETS, PROP_LAST }; static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp") ); static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("application/x-rtp") ); static gboolean gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent, GstEvent * event); static GstFlowReturn gst_rtp_rtx_receive_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer); static GstStateChangeReturn gst_rtp_rtx_receive_change_state (GstElement * element, GstStateChange transition); static void gst_rtp_rtx_receive_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_rtp_rtx_receive_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static void gst_rtp_rtx_receive_finalize (GObject * object); G_DEFINE_TYPE (GstRtpRtxReceive, gst_rtp_rtx_receive, GST_TYPE_ELEMENT); static void gst_rtp_rtx_receive_class_init (GstRtpRtxReceiveClass * klass) { GObjectClass *gobject_class; GstElementClass *gstelement_class; gobject_class = (GObjectClass *) klass; gstelement_class = (GstElementClass *) klass; gobject_class->get_property = gst_rtp_rtx_receive_get_property; gobject_class->set_property = gst_rtp_rtx_receive_set_property; gobject_class->finalize = gst_rtp_rtx_receive_finalize; g_object_class_install_property (gobject_class, PROP_PAYLOAD_TYPE_MAP, g_param_spec_boxed ("payload-type-map", "Payload Type Map", "Map of original payload types to their retransmission payload types", GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_NUM_RTX_REQUESTS, g_param_spec_uint ("num-rtx-requests", "Num RTX Requests", "Number of retransmission events received", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_NUM_RTX_PACKETS, g_param_spec_uint ("num-rtx-packets", "Num RTX Packets", " Number of retransmission packets received", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_NUM_RTX_ASSOC_PACKETS, g_param_spec_uint ("num-rtx-assoc-packets", "Num RTX Associated Packets", "Number of retransmission packets " "correctly associated with retransmission requests", 0, G_MAXUINT, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&src_factory)); gst_element_class_add_pad_template (gstelement_class, gst_static_pad_template_get (&sink_factory)); gst_element_class_set_static_metadata (gstelement_class, "RTP Retransmission receiver", "Codec", "Receive retransmitted RTP packets according to RFC4588", "Julien Isorce "); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_change_state); } static void gst_rtp_rtx_receive_reset (GstRtpRtxReceive * rtx) { GST_OBJECT_LOCK (rtx); g_hash_table_remove_all (rtx->ssrc2_ssrc1_map); g_hash_table_remove_all (rtx->seqnum_ssrc1_map); rtx->num_rtx_requests = 0; rtx->num_rtx_packets = 0; rtx->num_rtx_assoc_packets = 0; GST_OBJECT_UNLOCK (rtx); } static void gst_rtp_rtx_receive_finalize (GObject * object) { GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object); g_hash_table_unref (rtx->ssrc2_ssrc1_map); g_hash_table_unref (rtx->seqnum_ssrc1_map); g_hash_table_unref (rtx->rtx_pt_map); if (rtx->rtx_pt_map_structure) gst_structure_free (rtx->rtx_pt_map_structure); G_OBJECT_CLASS (gst_rtp_rtx_receive_parent_class)->finalize (object); } static void gst_rtp_rtx_receive_init (GstRtpRtxReceive * rtx) { GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx); rtx->srcpad = gst_pad_new_from_template (gst_element_class_get_pad_template (klass, "src"), "src"); GST_PAD_SET_PROXY_CAPS (rtx->srcpad); GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad); gst_pad_set_event_function (rtx->srcpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_src_event)); gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad); rtx->sinkpad = gst_pad_new_from_template (gst_element_class_get_pad_template (klass, "sink"), "sink"); GST_PAD_SET_PROXY_CAPS (rtx->sinkpad); GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad); gst_pad_set_chain_function (rtx->sinkpad, GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_chain)); gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad); rtx->ssrc2_ssrc1_map = g_hash_table_new (g_direct_hash, g_direct_equal); rtx->seqnum_ssrc1_map = g_hash_table_new (g_direct_hash, g_direct_equal); rtx->rtx_pt_map = g_hash_table_new (g_direct_hash, g_direct_equal); } static gboolean gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent, GstEvent * event) { GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (parent); gboolean res; switch (GST_EVENT_TYPE (event)) { case GST_EVENT_CUSTOM_UPSTREAM: { const GstStructure *s = gst_event_get_structure (event); /* This event usually comes from the downstream gstrtpjitterbuffer */ if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) { guint seqnum = 0; guint ssrc = 0; gpointer ssrc1 = 0; gpointer ssrc2 = 0; /* retrieve seqnum of the packet that need to be restransmisted */ if (!gst_structure_get_uint (s, "seqnum", &seqnum)) seqnum = -1; /* retrieve ssrc of the packet that need to be restransmisted * it's usefull when reconstructing the original packet from the rtx packet */ if (!gst_structure_get_uint (s, "ssrc", &ssrc)) ssrc = -1; GST_DEBUG_OBJECT (rtx, "request seqnum: %" G_GUINT16_FORMAT ", ssrc: %" G_GUINT32_FORMAT, seqnum, ssrc); GST_OBJECT_LOCK (rtx); /* increase number of seen requests for our statistics */ ++rtx->num_rtx_requests; /* First, we lookup in our map to see if we have already associate this * master stream ssrc with its retransmisted stream. * Every ssrc are unique so we can use the same hash table * for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1 */ if (g_hash_table_lookup_extended (rtx->ssrc2_ssrc1_map, GUINT_TO_POINTER (ssrc), NULL, &ssrc2) && GPOINTER_TO_UINT (ssrc2) != GPOINTER_TO_UINT (ssrc)) { GST_DEBUG ("Retransmited stream %" G_GUINT32_FORMAT " already associated to its master", GPOINTER_TO_UINT (ssrc2)); } else { /* not already associated but also we have to check that we have not * already considered this request. */ if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (seqnum), NULL, &ssrc1)) { if (GPOINTER_TO_UINT (ssrc1) == ssrc) { /* do nothing because we have already considered this request * The jitter may be too impatient of the rtx packet has been * lost too. * It does not mean we reject the event, we still want to forward * the request to the gstrtpsession to be translater into a FB NACK */ GST_DEBUG ("Duplicated request seqnum: %" G_GUINT16_FORMAT ", ssrc1: %" G_GUINT32_FORMAT, seqnum, ssrc); } else { /* From RFC 4588: * the receiver MUST NOT have two outstanding requests for the * same packet sequence number in two different original streams * before the association is resolved. Otherwise it's impossible * to associate a rtx stream and its master stream */ GST_DEBUG ("reject request for seqnum %" G_GUINT16_FORMAT "of master stream %" G_GUINT32_FORMAT, seqnum, ssrc); res = TRUE; /* remove seqnum in order to reuse the spot */ g_hash_table_remove (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (seqnum)); /* do not forward the event as we are rejecting this request */ GST_OBJECT_UNLOCK (rtx); gst_event_unref (event); return res; } } else { /* the request has not been already considered * insert it for the first time */ GST_DEBUG ("packet number %" G_GUINT16_FORMAT " of master stream %" G_GUINT32_FORMAT " needs to be retransmited", seqnum, ssrc); g_hash_table_insert (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (seqnum), GUINT_TO_POINTER (ssrc)); } } GST_OBJECT_UNLOCK (rtx); } /* Transfer event upstream so that the request can acutally by translated * through gstrtpsession through the network */ res = gst_pad_event_default (pad, parent, event); break; } default: res = gst_pad_event_default (pad, parent, event); break; } return res; } /* Copy fixed header and extension. Replace current ssrc by ssrc1, * remove OSN and replace current seq num by OSN. * Copy memory to avoid to manually copy each rtp buffer field. */ static GstBuffer * _gst_rtp_buffer_new_from_rtx (GstRTPBuffer * rtp, guint32 ssrc1, guint16 orign_seqnum, guint8 origin_payload_type) { GstMemory *mem = NULL; GstRTPBuffer new_rtp = GST_RTP_BUFFER_INIT; GstBuffer *new_buffer = gst_buffer_new (); GstMapInfo map; guint payload_len = 0; /* copy fixed header */ mem = gst_memory_copy (rtp->map[0].memory, 0, rtp->size[0]); gst_buffer_append_memory (new_buffer, mem); /* copy extension if any */ if (rtp->size[1]) { mem = gst_memory_copy (rtp->map[1].memory, 0, rtp->size[1]); gst_buffer_append_memory (new_buffer, mem); } /* copy payload and remove OSN */ payload_len = rtp->size[2] - 2; mem = gst_allocator_alloc (NULL, payload_len, NULL); gst_memory_map (mem, &map, GST_MAP_WRITE); if (rtp->size[2]) memcpy (map.data, (guint8 *) rtp->data[2] + 2, payload_len); gst_memory_unmap (mem, &map); gst_buffer_append_memory (new_buffer, mem); /* the sender always constructs rtx packets without padding, * But the receiver can still receive rtx packets with padding. * So just copy it. */ if (rtp->size[3]) { guint pad_len = rtp->size[3]; mem = gst_allocator_alloc (NULL, pad_len, NULL); gst_memory_map (mem, &map, GST_MAP_WRITE); map.data[pad_len - 1] = pad_len; gst_memory_unmap (mem, &map); gst_buffer_append_memory (new_buffer, mem); } /* set ssrc and seq num */ gst_rtp_buffer_map (new_buffer, GST_MAP_WRITE, &new_rtp); gst_rtp_buffer_set_ssrc (&new_rtp, ssrc1); gst_rtp_buffer_set_seq (&new_rtp, orign_seqnum); gst_rtp_buffer_set_payload_type (&new_rtp, origin_payload_type); gst_rtp_buffer_unmap (&new_rtp); return new_buffer; } static GstFlowReturn gst_rtp_rtx_receive_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer) { GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (parent); GstRTPBuffer rtp = GST_RTP_BUFFER_INIT; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *new_buffer = NULL; guint32 ssrc = 0; gpointer ssrc1 = 0; guint32 ssrc2 = 0; guint16 seqnum = 0; guint16 orign_seqnum = 0; guint8 payload_type = 0; guint8 origin_payload_type = 0; gboolean is_rtx = FALSE; gboolean drop = FALSE; /* map current rtp packet to parse its header */ gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp); ssrc = gst_rtp_buffer_get_ssrc (&rtp); seqnum = gst_rtp_buffer_get_seq (&rtp); payload_type = gst_rtp_buffer_get_payload_type (&rtp); /* check if we have a retransmission packet (this information comes from SDP) */ GST_OBJECT_LOCK (rtx); is_rtx = g_hash_table_lookup_extended (rtx->rtx_pt_map, GUINT_TO_POINTER (payload_type), NULL, NULL); GST_OBJECT_UNLOCK (rtx); if (is_rtx) { /* read OSN in the rtx payload */ orign_seqnum = GST_READ_UINT16_BE (gst_rtp_buffer_get_payload (&rtp)); origin_payload_type = GPOINTER_TO_UINT (g_hash_table_lookup (rtx->rtx_pt_map, GUINT_TO_POINTER (payload_type))); } GST_OBJECT_LOCK (rtx); /* if the current packet is from a retransmission stream */ if (is_rtx) { /* increase our statistic */ ++rtx->num_rtx_packets; /* first we check if we already have associated this retransmission stream * to a master stream */ if (g_hash_table_lookup_extended (rtx->ssrc2_ssrc1_map, GUINT_TO_POINTER (ssrc), NULL, &ssrc1)) { GST_DEBUG ("packet is from retransmission stream %" G_GUINT32_FORMAT " already associated to master stream %" G_GUINT32_FORMAT, ssrc, GPOINTER_TO_UINT (ssrc1)); ssrc2 = ssrc; } else { /* the current retransmisted packet has its rtx stream not already * associated to a master stream, so retrieve it from our request * history */ if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (orign_seqnum), NULL, &ssrc1)) { GST_DEBUG ("associate retransmisted stream %" G_GUINT32_FORMAT " to master stream %" G_GUINT32_FORMAT " thanks to packet %" G_GUINT16_FORMAT "", ssrc, GPOINTER_TO_UINT (ssrc1), orign_seqnum); ssrc2 = ssrc; /* free the spot so that this seqnum can be used to do another * association */ g_hash_table_remove (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (orign_seqnum)); /* actually do the association between rtx stream and master stream */ g_hash_table_insert (rtx->ssrc2_ssrc1_map, GUINT_TO_POINTER (ssrc2), ssrc1); /* just put a guard */ if (GPOINTER_TO_UINT (ssrc1) == ssrc2) g_warning ("RTX receiver ssrc2_ssrc1_map bad state, ssrc %" G_GUINT32_FORMAT " are the same\n", ssrc); /* also do the association between master stream and rtx stream * every ssrc are unique so we can use the same hash table * for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1 */ g_hash_table_insert (rtx->ssrc2_ssrc1_map, ssrc1, GUINT_TO_POINTER (ssrc2)); } else { /* we are not able to associate this rtx packet with a master stream */ GST_DEBUG ("drop rtx packet because its orign_seqnum %" G_GUINT16_FORMAT " is not in pending retransmission requests", orign_seqnum); drop = TRUE; } } } /* if not dropped the packet was successfully associated */ if (is_rtx && !drop) ++rtx->num_rtx_assoc_packets; GST_OBJECT_UNLOCK (rtx); /* just drop the packet if the association could not have been made */ if (drop) { gst_rtp_buffer_unmap (&rtp); gst_buffer_unref (buffer); return GST_FLOW_OK; } /* create the retransmission packet */ if (is_rtx) new_buffer = _gst_rtp_buffer_new_from_rtx (&rtp, GPOINTER_TO_UINT (ssrc1), orign_seqnum, origin_payload_type); gst_rtp_buffer_unmap (&rtp); /* push the packet */ if (is_rtx) { gst_buffer_unref (buffer); GST_LOG_OBJECT (rtx, "push packet seqnum:%" G_GUINT16_FORMAT " from a restransmission stream ssrc2:%" G_GUINT32_FORMAT " (src %" G_GUINT32_FORMAT ")", orign_seqnum, ssrc2, GPOINTER_TO_UINT (ssrc1)); ret = gst_pad_push (rtx->srcpad, new_buffer); } else { GST_LOG_OBJECT (rtx, "push packet seqnum:%" G_GUINT16_FORMAT " from a master stream ssrc: %" G_GUINT32_FORMAT, seqnum, ssrc); ret = gst_pad_push (rtx->srcpad, buffer); } return ret; } static void gst_rtp_rtx_receive_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object); switch (prop_id) { case PROP_PAYLOAD_TYPE_MAP: GST_OBJECT_LOCK (rtx); g_value_set_boxed (value, rtx->rtx_pt_map_structure); GST_OBJECT_UNLOCK (rtx); break; case PROP_NUM_RTX_REQUESTS: GST_OBJECT_LOCK (rtx); g_value_set_uint (value, rtx->num_rtx_requests); GST_OBJECT_UNLOCK (rtx); break; case PROP_NUM_RTX_PACKETS: GST_OBJECT_LOCK (rtx); g_value_set_uint (value, rtx->num_rtx_packets); GST_OBJECT_UNLOCK (rtx); break; case PROP_NUM_RTX_ASSOC_PACKETS: GST_OBJECT_LOCK (rtx); g_value_set_uint (value, rtx->num_rtx_assoc_packets); GST_OBJECT_UNLOCK (rtx); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean structure_to_hash_table_inv (GQuark field_id, const GValue * value, gpointer hash) { const gchar *field_str; guint field_uint; guint value_uint; field_str = g_quark_to_string (field_id); field_uint = atoi (field_str); value_uint = g_value_get_uint (value); g_hash_table_insert ((GHashTable *) hash, GUINT_TO_POINTER (value_uint), GUINT_TO_POINTER (field_uint)); return TRUE; } static void gst_rtp_rtx_receive_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object); switch (prop_id) { case PROP_PAYLOAD_TYPE_MAP: GST_OBJECT_LOCK (rtx); if (rtx->rtx_pt_map_structure) gst_structure_free (rtx->rtx_pt_map_structure); rtx->rtx_pt_map_structure = g_value_dup_boxed (value); g_hash_table_remove_all (rtx->rtx_pt_map); gst_structure_foreach (rtx->rtx_pt_map_structure, structure_to_hash_table_inv, rtx->rtx_pt_map); GST_OBJECT_UNLOCK (rtx); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static GstStateChangeReturn gst_rtp_rtx_receive_change_state (GstElement * element, GstStateChange transition) { GstStateChangeReturn ret; GstRtpRtxReceive *rtx; rtx = GST_RTP_RTX_RECEIVE (element); switch (transition) { default: break; } ret = GST_ELEMENT_CLASS (gst_rtp_rtx_receive_parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_PAUSED_TO_READY: gst_rtp_rtx_receive_reset (rtx); break; default: break; } return ret; } gboolean gst_rtp_rtx_receive_plugin_init (GstPlugin * plugin) { GST_DEBUG_CATEGORY_INIT (gst_rtp_rtx_receive_debug, "rtprtxreceive", 0, "rtp retransmission receiver"); return gst_element_register (plugin, "rtprtxreceive", GST_RANK_NONE, GST_TYPE_RTP_RTX_RECEIVE); }