webrtc/modules/rtp_rtcp/source/receiver_fec.cc

/*
 *  Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
 *
 *  Use of this source code is governed by a BSD-style license
 *  that can be found in the LICENSE file in the root of the source
 *  tree. An additional intellectual property rights grant can be found
 *  in the file PATENTS.  All contributing project authors may
 *  be found in the AUTHORS file in the root of the source tree.
 */

#include <cassert>

#include "receiver_fec.h"
#include "rtp_receiver_video.h"
#include "forward_error_correction.h"
#include "rtp_utility.h"

// RFC 5109
namespace webrtc {
ReceiverFEC::ReceiverFEC(const WebRtc_Word32 id, RTPReceiverVideo* owner) :
    _owner(owner),
    _fec(new ForwardErrorCorrection(id)),
    _payloadTypeFEC(-1),
    _lastFECSeqNum(0),
    _frameComplete(true)
{
}

ReceiverFEC::~ReceiverFEC()
{
    // Clean up DecodeFEC()
    while (_receivedPacketList.First() != NULL)
    {
        ForwardErrorCorrection::ReceivedPacket* receivedPacket =
            static_cast<ForwardErrorCorrection::ReceivedPacket*>(
            _receivedPacketList.First()->GetItem());
        delete receivedPacket->pkt;
        delete receivedPacket;
        receivedPacket = NULL;
        _receivedPacketList.PopFront();
    }
    assert(_receivedPacketList.Empty());

    if (_fec != NULL)
    {
        bool frameComplete = true;
        _fec->DecodeFEC(_receivedPacketList, _recoveredPacketList,_lastFECSeqNum, frameComplete);
        delete _fec;
    }
}

void
ReceiverFEC::SetPayloadTypeFEC(const WebRtc_Word8 payloadType)
{
    _payloadTypeFEC = payloadType;
}

/*
    0                   1                    2                   3
    0 1 2 3 4 5 6 7 8 9 0 1 2 3  4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
   |F|   block PT  |  timestamp offset         |   block length    |
   +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+


RFC 2198          RTP Payload for Redundant Audio Data    September 1997

   The bits in the header are specified as follows:

   F: 1 bit First bit in header indicates whether another header block
       follows.  If 1 further header blocks follow, if 0 this is the
       last header block.
       If 0 there is only 1 byte RED header

   block PT: 7 bits RTP payload type for this block.

   timestamp offset:  14 bits Unsigned offset of timestamp of this block
       relative to timestamp given in RTP header.  The use of an unsigned
       offset implies that redundant data must be sent after the primary
       data, and is hence a time to be subtracted from the current
       timestamp to determine the timestamp of the data for which this
       block is the redundancy.

   block length:  10 bits Length in bytes of the corresponding data
       block excluding header.
*/

WebRtc_Word32
ReceiverFEC::AddReceivedFECPacket(const WebRtcRTPHeader* rtpHeader,
                                  const WebRtc_UWord8* incomingRtpPacket,
                                  const WebRtc_UWord16 payloadDataLength,
                                  bool& FECpacket )
{
    if (_payloadTypeFEC == -1)
    {
        return -1;
    }

    WebRtc_UWord8 REDHeaderLength = 1;

    // Add to list without RED header, aka a virtual RTP packet
    // we remove the RED header

    ForwardErrorCorrection::ReceivedPacket* receivedPacket = new ForwardErrorCorrection::ReceivedPacket;
    receivedPacket->pkt = new ForwardErrorCorrection::Packet;

    // get payload type from RED header
    WebRtc_UWord8 payloadType = incomingRtpPacket[rtpHeader->header.headerLength] & 0x7f;

    // use the payloadType to decide if it's FEC or coded data
    if(_payloadTypeFEC == payloadType)
    {
        receivedPacket->isFec = true;
        FECpacket = true;
    } else
    {
        receivedPacket->isFec = false;
        FECpacket = false;
    }
    receivedPacket->lastMediaPktInFrame = rtpHeader->header.markerBit;
    receivedPacket->seqNum = rtpHeader->header.sequenceNumber;

    WebRtc_UWord16 blockLength = 0;
    if(incomingRtpPacket[rtpHeader->header.headerLength] & 0x80)
    {
        // f bit set in RED header
        REDHeaderLength = 4;
        WebRtc_UWord16 timestampOffset = (incomingRtpPacket[rtpHeader->header.headerLength + 1]) << 8;
        timestampOffset += incomingRtpPacket[rtpHeader->header.headerLength+2];
        timestampOffset = timestampOffset >> 2;
        if(timestampOffset != 0)
        {
            // sanity timestampOffset must be 0
            assert(false);
            return -1;
        }
        blockLength = (0x03 & incomingRtpPacket[rtpHeader->header.headerLength + 2]) << 8;
        blockLength += (incomingRtpPacket[rtpHeader->header.headerLength + 3]);

        // check next RED header
        if(incomingRtpPacket[rtpHeader->header.headerLength+4] & 0x80)
        {
            // more than 2 blocks in packet not supported
            assert(false);
            return -1;
        }
        if(blockLength > payloadDataLength - REDHeaderLength)
        {
            // block length longer than packet
            assert(false);
            return -1;
        }
    }

    ForwardErrorCorrection::ReceivedPacket* secondReceivedPacket = NULL;
    if(blockLength > 0)
    {
        // handle block length, split into 2 packets
        REDHeaderLength = 5;

        // copy the RTP header
        memcpy(receivedPacket->pkt->data,
               incomingRtpPacket,
               rtpHeader->header.headerLength);

        // replace the RED payload type
        receivedPacket->pkt->data[1] &= 0x80;         // reset the payload
        receivedPacket->pkt->data[1] += payloadType;  // set the media payload type

        // copy the payload data
        memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
              incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
              blockLength);

        receivedPacket->pkt->length = blockLength;

        secondReceivedPacket = new ForwardErrorCorrection::ReceivedPacket;
        secondReceivedPacket->pkt = new ForwardErrorCorrection::Packet;

        secondReceivedPacket->isFec = true;
        secondReceivedPacket->lastMediaPktInFrame = false;
        secondReceivedPacket->seqNum = rtpHeader->header.sequenceNumber;

        // copy the FEC payload data
        memcpy(secondReceivedPacket->pkt->data,
               incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength +
               blockLength, payloadDataLength - REDHeaderLength - blockLength);

        secondReceivedPacket->pkt->length = payloadDataLength - REDHeaderLength -
            blockLength;

    } else if(receivedPacket->isFec)
    {
        // everything behind the RED header
        memcpy(receivedPacket->pkt->data,
               incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
               payloadDataLength - REDHeaderLength);
        receivedPacket->pkt->length = payloadDataLength - REDHeaderLength;
        receivedPacket->ssrc = ModuleRTPUtility::BufferToUWord32(&incomingRtpPacket[8]);

    }else
    {
        // copy the RTP header
        memcpy(receivedPacket->pkt->data,
               incomingRtpPacket,
               rtpHeader->header.headerLength);

        // replace the RED payload type
        receivedPacket->pkt->data[1] &= 0x80;         // reset the payload
        receivedPacket->pkt->data[1] += payloadType;  // set the media payload type

        // copy the media payload data
        memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
               incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
               payloadDataLength - REDHeaderLength);

        receivedPacket->pkt->length = rtpHeader->header.headerLength +
            payloadDataLength - REDHeaderLength;
    }

    if(receivedPacket->isFec)
    {
        AddReceivedFECInfo(rtpHeader, NULL, FECpacket);
    }

    if(receivedPacket->pkt->length == 0)
    {
        delete receivedPacket->pkt;
        delete receivedPacket;
        return 0;
    }
    _receivedPacketList.PushBack(receivedPacket);
    if (secondReceivedPacket)
    {
        _receivedPacketList.PushBack(secondReceivedPacket);
    }
    return 0;
}

void
ReceiverFEC::AddReceivedFECInfo(const WebRtcRTPHeader* rtpHeader,
                                const WebRtc_UWord8* incomingRtpPacket,
                                bool& FECpacket)
{
    // store the highest FEC seq num received
    if(_lastFECSeqNum >= rtpHeader->header.sequenceNumber)
    {
        if(_lastFECSeqNum > 0xff00 && rtpHeader->header.sequenceNumber < 0x0ff ) //detect wrap around
        {
            // wrap
            _lastFECSeqNum = rtpHeader->header.sequenceNumber;
        } else
        {
            // old seqNum
        }
    }else
    {
        // check for a wrap
        if(rtpHeader->header.sequenceNumber > 0xff00 && _lastFECSeqNum < 0x0ff ) //detect wrap around
        {
            // old seqNum
        }else
        {
            _lastFECSeqNum = rtpHeader->header.sequenceNumber;
        }
    }

    if(incomingRtpPacket)
    {
        // get payload type from RED header
        WebRtc_UWord8 payloadType = incomingRtpPacket[rtpHeader->header.headerLength] & 0x7f;

        // use the payloadType to decide if it's FEC or coded data
        if(_payloadTypeFEC == payloadType)
        {
            FECpacket = true;
        } else
        {
            FECpacket = false;
        }
    }
}

WebRtc_Word32
ReceiverFEC::ProcessReceivedFEC(const bool forceFrameDecode)
{
    if (!_receivedPacketList.Empty())
    {
        if (_fec->DecodeFEC(_receivedPacketList,
                            _recoveredPacketList,
                            _lastFECSeqNum,
                            _frameComplete) != 0)
        {
            return -1;
        }
        assert(_receivedPacketList.Empty());
    }
    if (forceFrameDecode)
    {
        _frameComplete = true;
    }
    if (_frameComplete)
    {
        while (_recoveredPacketList.First() != NULL)
        {
            ForwardErrorCorrection::RecoveredPacket* recoveredPacket =
                static_cast<ForwardErrorCorrection::RecoveredPacket*>(_recoveredPacketList.First()->GetItem());

            WebRtcRTPHeader rtpHeader;
            memset(&rtpHeader, 0, sizeof(rtpHeader));

            ModuleRTPUtility::RTPHeaderParser rtpHeaderParser(recoveredPacket->pkt->data,
                                                              recoveredPacket->pkt->length);

            if (!rtpHeaderParser.Parse(rtpHeader))
            {
                return -1;
            }
            if (_owner->ReceiveRecoveredPacketCallback(&rtpHeader,
                                               &recoveredPacket->pkt->data[rtpHeader.header.headerLength],
                                               recoveredPacket->pkt->length - rtpHeader.header.headerLength) != 0)
            {
                return -1;
            }

            delete recoveredPacket->pkt;
            delete recoveredPacket;
            recoveredPacket = NULL;
            _recoveredPacketList.PopFront();
        }
        assert(_recoveredPacketList.Empty());
    }

    return 0;
}
} // namespace webrtc