/* 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 * * rtprtxreceive listens to the retransmission events from the * downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and * the sequence number that was requested. When it receives a packet with * a sequence number equal to one of the ones stored and with a different SSRC, * it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1. * From this point on, it replaces ssrc2 with ssrc1 in all packets of the * ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer * can reconstruct the original stream. * * This algorithm is implemented as specified in RFC 4588. * * This element is meant to be used with rtprtxsend on the sender side. * See #GstRtpRtxSend * * Below you can see some examples that illustrate how rtprtxreceive and * rtprtxsend fit among the other rtp elements and how they work internally. * Normally, hoewever, you should avoid using such pipelines and use * rtpbin instead, with its #GstRtpBin::request-aux-sender and * #GstRtpBin::request-aux-receiver signals. See #GstRtpBin. * * # Example pipelines * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! \ * rtprtxsend payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \ * rtpsession.send_rtp_sink \ * rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \ * 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 rtp-profile=avpf \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \ * rtpsession.recv_rtp_sink \ * rtpsession.recv_rtp_src ! \ * rtprtxreceive payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \ * rtpssrcdemux ! rtpjitterbuffer do-retransmission=true ! \ * rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \ * rtpsession.send_rtcp_src ! \ * udpsink host="127.0.0.1" port=5001 sync=false async=false \ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink * ]| Receive audio stream from port 5000 (5001 and 5002 are just the rtcp * link with the sender) * * In this example we can see a simple streaming of an OPUS stream with some * of the packets being artificially dropped by the identity element. * Thanks to retransmission, you should still hear a clear sound when setting * drop-probability to something greater than 0. * * Internally, the rtpjitterbuffer will generate a custom upstream event, * GstRTPRetransmissionRequest, when it detects that one packet is missing. * Then this request is translated to a FB NACK in the rtcp link by rtpsession. * Finally the rtpsession of the sender side will re-convert it in a * GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend * will then re-send the missing packet with a new srrc and a different payload * type (here, 97), but with the same original sequence number. On the receiver * side, rtprtxreceive will associate this new stream with the original and * forward the retransmission packets to rtpjitterbuffer with the original * ssrc and payload type. * * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 seqnum-offset=1 ! \ * rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \ * funnel name=f ! rtpsession.send_rtp_sink \ * audiotestsrc freq=660.0 is-live=true ! opusenc ! \ * rtpopuspay pt=97 seqnum-offset=100 ! \ * rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \ * f. \ * rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \ * 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 two audio streams to port 5000. * |[ * gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \ * udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97" ! \ * rtpsession.recv_rtp_sink \ * rtpsession.recv_rtp_src ! \ * rtprtxreceive payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \ * rtpssrcdemux name=demux \ * demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \ * opusdec ! audioconvert ! autoaudiosink \ * demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \ * opusdec ! audioconvert ! autoaudiosink \ * udpsrc port=5002 ! rtpsession.recv_rtcp_sink \ * rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \ * sync=false async=false * ]| Receive two audio streams from port 5000. * * In this example we are streaming two streams of the same type through the * same port. They, however, are using a different SSRC (ssrc is randomly * generated on each payloader - rtpopuspay in this example), so they can be * identified and demultiplexed by rtpssrcdemux on the receiver side. This is * an example of SSRC-multiplexing. * * It is important here to use a different starting sequence number * (seqnum-offset), since this is the only means of identification that * rtprtxreceive uses the very first time to identify retransmission streams. * It is an error, according to RFC4588 to have two retransmission requests for * packets belonging to two different streams but with the same sequence number. * Note that the default seqnum-offset value (-1, which means random) would * work just fine, but it is overriden here for illustration purposes. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include #include #include #include "gstrtprtxreceive.h" #define ASSOC_TIMEOUT (GST_SECOND) 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 }; 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_static_pad_template (gstelement_class, &src_factory); gst_element_class_add_static_pad_template (gstelement_class, &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); } typedef struct { guint32 ssrc; GstClockTime time; } SsrcAssoc; static SsrcAssoc * ssrc_assoc_new (guint32 ssrc, GstClockTime time) { SsrcAssoc *assoc = g_slice_new (SsrcAssoc); assoc->ssrc = ssrc; assoc->time = time; return assoc; } static void ssrc_assoc_free (SsrcAssoc * assoc) { g_slice_free (SsrcAssoc, assoc); } 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_full (g_direct_hash, g_direct_equal, NULL, (GDestroyNotify) ssrc_assoc_free); 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 ssrc2 = 0; /* retrieve seqnum of the packet that need to be retransmitted */ if (!gst_structure_get_uint (s, "seqnum", &seqnum)) seqnum = -1; /* retrieve ssrc of the packet that need to be retransmitted * it's useful 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_GUINT32_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 retransmitted 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_OBJECT (rtx, "Retransmited stream %" G_GUINT32_FORMAT " already associated to its master", GPOINTER_TO_UINT (ssrc2)); } else { SsrcAssoc *assoc; /* 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, (gpointer *) & assoc)) { if (assoc->ssrc == 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_OBJECT (rtx, "Duplicated request seqnum: %" G_GUINT32_FORMAT ", ssrc1: %" G_GUINT32_FORMAT, seqnum, ssrc); } else { /* If the association attempt is larger than ASSOC_TIMEOUT, * then we give up on it, and try this one. */ if (!GST_CLOCK_TIME_IS_VALID (rtx->last_time) || !GST_CLOCK_TIME_IS_VALID (assoc->time) || assoc->time + ASSOC_TIMEOUT < rtx->last_time) { /* 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 */ /* remove seqnum in order to reuse the spot */ g_hash_table_remove (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (seqnum)); goto retransmit; } else { GST_DEBUG_OBJECT (rtx, "reject request for seqnum %" G_GUINT32_FORMAT " of master stream %" G_GUINT32_FORMAT, seqnum, ssrc); /* do not forward the event as we are rejecting this request */ GST_OBJECT_UNLOCK (rtx); gst_event_unref (event); return TRUE; } } } else { retransmit: /* the request has not been already considered * insert it for the first time */ g_hash_table_insert (rtx->seqnum_ssrc1_map, GUINT_TO_POINTER (seqnum), ssrc_assoc_new (ssrc, rtx->last_time)); } } GST_DEBUG_OBJECT (rtx, "packet number %" G_GUINT32_FORMAT " of master stream %" G_GUINT32_FORMAT " needs to be retransmitted", seqnum, 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, (guint8 *) rtp->data[0] - rtp->map[0].data, 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, (guint8 *) rtp->data[1] - rtp->map[1].data, 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); gst_buffer_copy_into (new_buffer, rtp->buffer, GST_BUFFER_COPY_FLAGS | GST_BUFFER_COPY_TIMESTAMPS, 0, -1); GST_BUFFER_FLAG_SET (new_buffer, GST_RTP_BUFFER_FLAG_RETRANSMISSION); 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; gpointer payload = NULL; guint8 origin_payload_type = 0; gboolean is_rtx; gboolean drop = FALSE; /* map current rtp packet to parse its header */ if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) goto invalid_buffer; 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); if (is_rtx) { payload = gst_rtp_buffer_get_payload (&rtp); if (!payload || gst_rtp_buffer_get_payload_len (&rtp) < 2) { GST_OBJECT_UNLOCK (rtx); gst_rtp_buffer_unmap (&rtp); goto invalid_buffer; } } rtx->last_time = GST_BUFFER_PTS (buffer); if (g_hash_table_size (rtx->seqnum_ssrc1_map) > 0) { GHashTableIter iter; gpointer key, value; g_hash_table_iter_init (&iter, rtx->seqnum_ssrc1_map); while (g_hash_table_iter_next (&iter, &key, &value)) { SsrcAssoc *assoc = value; /* remove association request if it is too old */ if (GST_CLOCK_TIME_IS_VALID (rtx->last_time) && GST_CLOCK_TIME_IS_VALID (assoc->time) && assoc->time + ASSOC_TIMEOUT < rtx->last_time) { g_hash_table_iter_remove (&iter); } } } /* if the current packet is from a retransmission stream */ if (is_rtx) { /* increase our statistic */ ++rtx->num_rtx_packets; /* 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))); /* 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_OBJECT (rtx, "packet is from retransmission stream %" G_GUINT32_FORMAT " already associated to master stream %" G_GUINT32_FORMAT, ssrc, GPOINTER_TO_UINT (ssrc1)); ssrc2 = ssrc; } else { SsrcAssoc *assoc; /* the current retransmitted 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, (gpointer *) & assoc)) { GST_DEBUG_OBJECT (rtx, "associate retransmitted stream %" G_GUINT32_FORMAT " to master stream %" G_GUINT32_FORMAT " thanks to packet %" G_GUINT16_FORMAT "", ssrc, assoc->ssrc, orign_seqnum); ssrc1 = GUINT_TO_POINTER (assoc->ssrc); ssrc2 = ssrc; /* just put a guard */ if (GPOINTER_TO_UINT (ssrc1) == ssrc2) GST_WARNING_OBJECT (rtx, "RTX receiver ssrc2_ssrc1_map bad state, " "ssrc %" G_GUINT32_FORMAT " are the same\n", 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); /* 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_OBJECT (rtx, "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; invalid_buffer: { GST_ELEMENT_WARNING (rtx, STREAM, DECODE, (NULL), ("Received invalid RTP payload, dropping")); gst_buffer_unref (buffer); return GST_FLOW_OK; } } 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); }