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797 lines
28 KiB
C
797 lines
28 KiB
C
/* RTP Retransmission receiver element for GStreamer
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*
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* gstrtprtxreceive.c:
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*
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* Copyright (C) 2013 Collabora Ltd.
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* @author Julien Isorce <julien.isorce@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtprtxreceive
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* @title: rtprtxreceive
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* @see_also: rtprtxsend, rtpsession, rtpjitterbuffer
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*
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* rtprtxreceive listens to the retransmission events from the
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* downstream rtpjitterbuffer and remembers the SSRC (ssrc1) of the stream and
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* the sequence number that was requested. When it receives a packet with
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* a sequence number equal to one of the ones stored and with a different SSRC,
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* it identifies the new SSRC (ssrc2) as the retransmission stream of ssrc1.
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* From this point on, it replaces ssrc2 with ssrc1 in all packets of the
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* ssrc2 stream and flags them as retransmissions, so that rtpjitterbuffer
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* can reconstruct the original stream.
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*
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* This algorithm is implemented as specified in RFC 4588.
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*
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* This element is meant to be used with rtprtxsend on the sender side.
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* See #GstRtpRtxSend
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*
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* Below you can see some examples that illustrate how rtprtxreceive and
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* rtprtxsend fit among the other rtp elements and how they work internally.
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* Normally, hoewever, you should avoid using such pipelines and use
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* rtpbin instead, with its #GstRtpBin::request-aux-sender and
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* #GstRtpBin::request-aux-receiver signals. See #GstRtpBin.
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*
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=96 ! \
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* rtprtxsend payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
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* rtpsession.send_rtp_sink \
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* rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
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* udpsink host="127.0.0.1" port=5000 \
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* udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
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* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 \
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* sync=false async=false
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* ]| Send audio stream through port 5000 (5001 and 5002 are just the rtcp
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* link with the receiver)
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*
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* |[
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* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)96" ! \
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* rtpsession.recv_rtp_sink \
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* rtpsession.recv_rtp_src ! \
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* rtprtxreceive payload-type-map="application/x-rtp-pt-map,96=(uint)97" ! \
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* rtpssrcdemux ! rtpjitterbuffer do-retransmission=true ! \
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* rtpopusdepay ! opusdec ! audioconvert ! audioresample ! autoaudiosink \
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* rtpsession.send_rtcp_src ! \
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* udpsink host="127.0.0.1" port=5001 sync=false async=false \
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* udpsrc port=5002 ! rtpsession.recv_rtcp_sink
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* ]|
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* Receive audio stream from port 5000 (5001 and 5002 are just the rtcp
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* link with the sender)
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*
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* In this example we can see a simple streaming of an OPUS stream with some
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* of the packets being artificially dropped by the identity element.
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* Thanks to retransmission, you should still hear a clear sound when setting
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* drop-probability to something greater than 0.
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*
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* Internally, the rtpjitterbuffer will generate a custom upstream event,
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* GstRTPRetransmissionRequest, when it detects that one packet is missing.
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* Then this request is translated to a FB NACK in the rtcp link by rtpsession.
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* Finally the rtpsession of the sender side will re-convert it in a
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* GstRTPRetransmissionRequest that will be handled by rtprtxsend. rtprtxsend
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* will then re-send the missing packet with a new srrc and a different payload
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* type (here, 97), but with the same original sequence number. On the receiver
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* side, rtprtxreceive will associate this new stream with the original and
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* forward the retransmission packets to rtpjitterbuffer with the original
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* ssrc and payload type.
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*
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* |[
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* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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* audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 seqnum-offset=1 ! \
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* rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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* funnel name=f ! rtpsession.send_rtp_sink \
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* audiotestsrc freq=660.0 is-live=true ! opusenc ! \
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* rtpopuspay pt=97 seqnum-offset=100 ! \
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* rtprtxsend payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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* f. \
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* rtpsession.send_rtp_src ! identity drop-probability=0.01 ! \
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* udpsink host="127.0.0.1" port=5000 \
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* udpsrc port=5001 ! rtpsession.recv_rtcp_sink \
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* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5002 \
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* sync=false async=false
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* ]|
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* Send two audio streams to port 5000.
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* |[
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* gst-launch-1.0 rtpsession name=rtpsession rtp-profile=avpf \
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* udpsrc port=5000 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS,payload=(int)97" ! \
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* rtpsession.recv_rtp_sink \
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* rtpsession.recv_rtp_src ! \
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* rtprtxreceive payload-type-map="application/x-rtp-pt-map,97=(uint)99" ! \
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* rtpssrcdemux name=demux \
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* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
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* opusdec ! audioconvert ! autoaudiosink \
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* demux. ! queue ! rtpjitterbuffer do-retransmission=true ! rtpopusdepay ! \
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* opusdec ! audioconvert ! autoaudiosink \
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* udpsrc port=5002 ! rtpsession.recv_rtcp_sink \
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* rtpsession.send_rtcp_src ! udpsink host="127.0.0.1" port=5001 \
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* sync=false async=false
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* ]|
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* Receive two audio streams from port 5000.
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*
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* In this example we are streaming two streams of the same type through the
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* same port. They, however, are using a different SSRC (ssrc is randomly
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* generated on each payloader - rtpopuspay in this example), so they can be
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* identified and demultiplexed by rtpssrcdemux on the receiver side. This is
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* an example of SSRC-multiplexing.
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*
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* It is important here to use a different starting sequence number
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* (seqnum-offset), since this is the only means of identification that
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* rtprtxreceive uses the very first time to identify retransmission streams.
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* It is an error, according to RFC4588 to have two retransmission requests for
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* packets belonging to two different streams but with the same sequence number.
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* Note that the default seqnum-offset value (-1, which means random) would
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* work just fine, but it is overridden here for illustration purposes.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <string.h>
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#include <stdlib.h>
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#include "gstrtprtxreceive.h"
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#define ASSOC_TIMEOUT (GST_SECOND)
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_rtx_receive_debug);
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#define GST_CAT_DEFAULT gst_rtp_rtx_receive_debug
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enum
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{
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PROP_0,
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PROP_PAYLOAD_TYPE_MAP,
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PROP_NUM_RTX_REQUESTS,
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PROP_NUM_RTX_PACKETS,
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PROP_NUM_RTX_ASSOC_PACKETS
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};
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static gboolean gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event);
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static GstFlowReturn gst_rtp_rtx_receive_chain (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static GstStateChangeReturn gst_rtp_rtx_receive_change_state (GstElement *
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element, GstStateChange transition);
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static void gst_rtp_rtx_receive_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_rtx_receive_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static void gst_rtp_rtx_receive_finalize (GObject * object);
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G_DEFINE_TYPE (GstRtpRtxReceive, gst_rtp_rtx_receive, GST_TYPE_ELEMENT);
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static void
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gst_rtp_rtx_receive_class_init (GstRtpRtxReceiveClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gobject_class->get_property = gst_rtp_rtx_receive_get_property;
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gobject_class->set_property = gst_rtp_rtx_receive_set_property;
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gobject_class->finalize = gst_rtp_rtx_receive_finalize;
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g_object_class_install_property (gobject_class, PROP_PAYLOAD_TYPE_MAP,
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g_param_spec_boxed ("payload-type-map", "Payload Type Map",
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"Map of original payload types to their retransmission payload types",
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GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_RTX_REQUESTS,
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g_param_spec_uint ("num-rtx-requests", "Num RTX Requests",
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"Number of retransmission events received", 0, G_MAXUINT,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_RTX_PACKETS,
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g_param_spec_uint ("num-rtx-packets", "Num RTX Packets",
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" Number of retransmission packets received", 0, G_MAXUINT,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_RTX_ASSOC_PACKETS,
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g_param_spec_uint ("num-rtx-assoc-packets",
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"Num RTX Associated Packets", "Number of retransmission packets "
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"correctly associated with retransmission requests", 0, G_MAXUINT,
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0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_static_pad_template (gstelement_class, &src_factory);
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gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Retransmission receiver", "Codec",
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"Receive retransmitted RTP packets according to RFC4588",
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"Julien Isorce <julien.isorce@collabora.co.uk>");
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gstelement_class->change_state =
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_change_state);
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}
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static void
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gst_rtp_rtx_receive_reset (GstRtpRtxReceive * rtx)
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{
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GST_OBJECT_LOCK (rtx);
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g_hash_table_remove_all (rtx->ssrc2_ssrc1_map);
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g_hash_table_remove_all (rtx->seqnum_ssrc1_map);
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rtx->num_rtx_requests = 0;
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rtx->num_rtx_packets = 0;
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rtx->num_rtx_assoc_packets = 0;
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GST_OBJECT_UNLOCK (rtx);
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}
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static void
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gst_rtp_rtx_receive_finalize (GObject * object)
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{
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GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (object);
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g_hash_table_unref (rtx->ssrc2_ssrc1_map);
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g_hash_table_unref (rtx->seqnum_ssrc1_map);
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g_hash_table_unref (rtx->rtx_pt_map);
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if (rtx->rtx_pt_map_structure)
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gst_structure_free (rtx->rtx_pt_map_structure);
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G_OBJECT_CLASS (gst_rtp_rtx_receive_parent_class)->finalize (object);
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}
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typedef struct
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{
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guint32 ssrc;
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GstClockTime time;
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} SsrcAssoc;
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static SsrcAssoc *
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ssrc_assoc_new (guint32 ssrc, GstClockTime time)
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{
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SsrcAssoc *assoc = g_slice_new (SsrcAssoc);
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assoc->ssrc = ssrc;
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assoc->time = time;
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return assoc;
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}
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static void
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ssrc_assoc_free (SsrcAssoc * assoc)
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{
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g_slice_free (SsrcAssoc, assoc);
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}
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static void
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gst_rtp_rtx_receive_init (GstRtpRtxReceive * rtx)
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{
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GstElementClass *klass = GST_ELEMENT_GET_CLASS (rtx);
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rtx->srcpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"src"), "src");
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GST_PAD_SET_PROXY_CAPS (rtx->srcpad);
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GST_PAD_SET_PROXY_ALLOCATION (rtx->srcpad);
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gst_pad_set_event_function (rtx->srcpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_src_event));
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gst_element_add_pad (GST_ELEMENT (rtx), rtx->srcpad);
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rtx->sinkpad =
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gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
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"sink"), "sink");
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GST_PAD_SET_PROXY_CAPS (rtx->sinkpad);
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GST_PAD_SET_PROXY_ALLOCATION (rtx->sinkpad);
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gst_pad_set_chain_function (rtx->sinkpad,
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GST_DEBUG_FUNCPTR (gst_rtp_rtx_receive_chain));
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gst_element_add_pad (GST_ELEMENT (rtx), rtx->sinkpad);
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rtx->ssrc2_ssrc1_map = g_hash_table_new (g_direct_hash, g_direct_equal);
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rtx->seqnum_ssrc1_map = g_hash_table_new_full (g_direct_hash, g_direct_equal,
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NULL, (GDestroyNotify) ssrc_assoc_free);
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rtx->rtx_pt_map = g_hash_table_new (g_direct_hash, g_direct_equal);
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}
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static gboolean
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gst_rtp_rtx_receive_src_event (GstPad * pad, GstObject * parent,
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GstEvent * event)
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{
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GstRtpRtxReceive *rtx = GST_RTP_RTX_RECEIVE (parent);
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gboolean res;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_CUSTOM_UPSTREAM:
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{
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const GstStructure *s = gst_event_get_structure (event);
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/* This event usually comes from the downstream gstrtpjitterbuffer */
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if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
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guint seqnum = 0;
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guint ssrc = 0;
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gpointer ssrc2 = 0;
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/* retrieve seqnum of the packet that need to be retransmitted */
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if (!gst_structure_get_uint (s, "seqnum", &seqnum))
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seqnum = -1;
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/* retrieve ssrc of the packet that need to be retransmitted
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* it's useful when reconstructing the original packet from the rtx packet */
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if (!gst_structure_get_uint (s, "ssrc", &ssrc))
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ssrc = -1;
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GST_DEBUG_OBJECT (rtx, "got rtx request for seqnum: %u, ssrc: %X",
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seqnum, ssrc);
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GST_OBJECT_LOCK (rtx);
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/* increase number of seen requests for our statistics */
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++rtx->num_rtx_requests;
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/* First, we lookup in our map to see if we have already associate this
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* master stream ssrc with its retransmitted stream.
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* Every ssrc are unique so we can use the same hash table
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* for both retrieving the ssrc1 from ssrc2 and also ssrc2 from ssrc1
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*/
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if (g_hash_table_lookup_extended (rtx->ssrc2_ssrc1_map,
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GUINT_TO_POINTER (ssrc), NULL, &ssrc2)
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&& GPOINTER_TO_UINT (ssrc2) != GPOINTER_TO_UINT (ssrc)) {
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GST_TRACE_OBJECT (rtx, "Retransmitted stream %X already associated "
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"to its master, %X", GPOINTER_TO_UINT (ssrc2), ssrc);
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} else {
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SsrcAssoc *assoc;
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/* not already associated but also we have to check that we have not
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* already considered this request.
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*/
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if (g_hash_table_lookup_extended (rtx->seqnum_ssrc1_map,
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GUINT_TO_POINTER (seqnum), NULL, (gpointer *) & assoc)) {
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if (assoc->ssrc == ssrc) {
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/* same seqnum, same ssrc */
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/* do nothing because we have already considered this request
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* The jitter may be too impatient of the rtx packet has been
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* lost too.
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* It does not mean we reject the event, we still want to forward
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* the request to the gstrtpsession to be translator into a FB NACK
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*/
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GST_LOG_OBJECT (rtx, "Duplicate request: seqnum: %u, ssrc: %X",
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seqnum, ssrc);
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} else {
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/* same seqnum, different ssrc */
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/* If the association attempt is larger than ASSOC_TIMEOUT,
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* then we give up on it, and try this one.
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*/
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if (!GST_CLOCK_TIME_IS_VALID (rtx->last_time) ||
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!GST_CLOCK_TIME_IS_VALID (assoc->time) ||
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assoc->time + ASSOC_TIMEOUT < rtx->last_time) {
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/* From RFC 4588:
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* the receiver MUST NOT have two outstanding requests for the
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* same packet sequence number in two different original streams
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* before the association is resolved. Otherwise it's impossible
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* to associate a rtx stream and its master stream
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*/
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/* remove seqnum in order to reuse the spot */
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g_hash_table_remove (rtx->seqnum_ssrc1_map,
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GUINT_TO_POINTER (seqnum));
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goto retransmit;
|
|
} else {
|
|
GST_INFO_OBJECT (rtx, "rejecting request for seqnum %u"
|
|
" of master stream %X; there is already a pending request "
|
|
"for the same seqnum on ssrc %X that has not expired",
|
|
seqnum, ssrc, assoc->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 %u of master stream %X"
|
|
" needs to be retransmitted", seqnum, ssrc);
|
|
|
|
GST_OBJECT_UNLOCK (rtx);
|
|
}
|
|
|
|
/* Transfer event upstream so that the request can actually 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)));
|
|
|
|
GST_DEBUG_OBJECT (rtx, "Got rtx packet: rtx seqnum %u, rtx ssrc %X, "
|
|
"rtx pt %u, orig seqnum %u, orig pt %u", seqnum, ssrc, payload_type,
|
|
orign_seqnum, origin_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_TRACE_OBJECT (rtx,
|
|
"packet is from retransmission stream %X already associated to "
|
|
"master stream %X", 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_LOG_OBJECT (rtx,
|
|
"associating retransmitted stream %X to master stream %X thanks "
|
|
"to rtx packet %u (orig seqnum %u)", ssrc, assoc->ssrc, seqnum,
|
|
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, "
|
|
"master and rtx SSRCs are the same (%X)\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_INFO_OBJECT (rtx,
|
|
"dropping rtx packet %u because its orig seqnum (%u) is not in our"
|
|
" pending retransmission requests", seqnum, 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, "pushing packet seqnum:%u from restransmission "
|
|
"stream ssrc: %X (master ssrc %X)", orign_seqnum, ssrc2,
|
|
GPOINTER_TO_UINT (ssrc1));
|
|
ret = gst_pad_push (rtx->srcpad, new_buffer);
|
|
} else {
|
|
GST_TRACE_OBJECT (rtx, "pushing packet seqnum:%u from master stream "
|
|
"ssrc: %X", 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);
|
|
}
|