git-svn-id: http://webrtc.googlecode.com/svn/trunk@4 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
niklase@google.com
248
modules/rtp_rtcp/source/rtcp_receiver_help.cc
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248
modules/rtp_rtcp/source/rtcp_receiver_help.cc
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/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "rtcp_receiver_help.h"
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#include "rtp_utility.h"
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#include <string.h> //memset
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#include <cassert> //assert
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namespace webrtc {
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using namespace RTCPHelp;
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RTCPPacketInformation::RTCPPacketInformation() :
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rtcpPacketTypeFlags(0),
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nackSequenceNumbers(0),
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nackSequenceNumbersLength(0),
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applicationSubType(0),
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applicationName(0),
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applicationData(),
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applicationLength(0),
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reportBlock(false),
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fractionLost(0),
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roundTripTime(0),
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lastReceivedExtendedHighSeqNum(0),
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jitter(0),
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sliPictureId(0),
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rpsiPictureId(0),
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VoIPMetric(NULL)
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{
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}
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RTCPPacketInformation::~RTCPPacketInformation()
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{
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delete [] nackSequenceNumbers;
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delete [] applicationData;
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delete VoIPMetric;
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}
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void
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RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
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{
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VoIPMetric = new RTCPVoIPMetric();
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memcpy(VoIPMetric, metric, sizeof(RTCPVoIPMetric));
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}
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void
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RTCPPacketInformation::AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size)
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{
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WebRtc_UWord8* oldData = applicationData;
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WebRtc_UWord16 oldLength = applicationLength;
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applicationLength += size;
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applicationData = new WebRtc_UWord8[applicationLength];
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if(oldData)
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{
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memcpy(applicationData, oldData, oldLength);
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memcpy(applicationData+oldLength, data, size);
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delete [] oldData;
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} else
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{
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memcpy(applicationData, data, size);
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}
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}
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void
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RTCPPacketInformation::ResetNACKPacketIdArray()
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{
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if(NULL == nackSequenceNumbers)
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{
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nackSequenceNumbers = new WebRtc_UWord16[NACK_PACKETS_MAX_SIZE];
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}
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nackSequenceNumbersLength = 0;
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}
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void
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RTCPPacketInformation::AddNACKPacket(const WebRtc_UWord16 packetID)
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{
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assert(nackSequenceNumbers);
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WebRtc_UWord16& idx = nackSequenceNumbersLength;
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if (idx < NACK_PACKETS_MAX_SIZE)
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{
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nackSequenceNumbers[idx++] = packetID;
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}
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}
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void
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RTCPPacketInformation::AddReportInfo(const WebRtc_UWord8 fraction,
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const WebRtc_UWord16 rtt,
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const WebRtc_UWord32 extendedHighSeqNum,
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const WebRtc_UWord32 j)
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{
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reportBlock = true;
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fractionLost = fraction;
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roundTripTime = rtt;
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jitter = j;
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lastReceivedExtendedHighSeqNum = extendedHighSeqNum;
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}
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RTCPReportBlockInformation::RTCPReportBlockInformation():
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remoteReceiveBlock(),
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remoteMaxJitter(0),
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RTT(0),
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minRTT(0),
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maxRTT(0),
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avgRTT(0),
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numAverageCalcs(0)
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{
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memset(&remoteReceiveBlock,0,sizeof(remoteReceiveBlock));
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}
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RTCPReportBlockInformation::~RTCPReportBlockInformation()
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{
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}
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RTCPReceiveInformation::RTCPReceiveInformation() :
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lastTimeReceived(0),
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lastFIRSequenceNumber(-1),
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lastFIRRequest(0),
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readyForDelete(false),
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_tmmbrSetTimeouts(NULL)
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{
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}
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RTCPReceiveInformation::~RTCPReceiveInformation()
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{
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if(_tmmbrSetTimeouts)
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{
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delete [] _tmmbrSetTimeouts;
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}
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}
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// don't use TmmbrSet.VerifyAndAllocate this version keeps the data
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void
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RTCPReceiveInformation::VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize)
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{
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if(minimumSize > TmmbrSet.sizeOfSet)
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{
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// make sure that our buffers are big enough
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WebRtc_UWord32* ptrTmmbrSet = new WebRtc_UWord32[minimumSize];
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WebRtc_UWord32* ptrTmmbrPacketOHSet = new WebRtc_UWord32[minimumSize];
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WebRtc_UWord32* ptrTmmbrSsrcSet = new WebRtc_UWord32[minimumSize];
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WebRtc_UWord32* tmmbrSetTimeouts = new WebRtc_UWord32[minimumSize];
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if(TmmbrSet.lengthOfSet > 0)
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{
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// copy old values
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memcpy(ptrTmmbrSet, TmmbrSet.ptrTmmbrSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
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memcpy(ptrTmmbrPacketOHSet, TmmbrSet.ptrPacketOHSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
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memcpy(ptrTmmbrSsrcSet, TmmbrSet.ptrSsrcSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
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memcpy(tmmbrSetTimeouts, _tmmbrSetTimeouts, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
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}
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if(TmmbrSet.ptrTmmbrSet)
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{
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delete [] TmmbrSet.ptrTmmbrSet;
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delete [] TmmbrSet.ptrPacketOHSet;
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delete [] TmmbrSet.ptrSsrcSet;
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}
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if(_tmmbrSetTimeouts)
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{
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delete [] _tmmbrSetTimeouts;
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}
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TmmbrSet.ptrTmmbrSet = ptrTmmbrSet;
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TmmbrSet.ptrPacketOHSet = ptrTmmbrPacketOHSet;
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TmmbrSet.ptrSsrcSet = ptrTmmbrSsrcSet;
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TmmbrSet.sizeOfSet = minimumSize;
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_tmmbrSetTimeouts = tmmbrSetTimeouts;
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}
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}
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void
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RTCPReceiveInformation::InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
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const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem)
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{
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// serach to see if we have it in our list
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for(WebRtc_UWord32 i = 0; i < TmmbrSet.lengthOfSet; i++)
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{
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if(TmmbrSet.ptrSsrcSet[i] == senderSSRC)
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{
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// we already have this SSRC in our list
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// update it
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TmmbrSet.ptrPacketOHSet[i] = TMMBRItem.MeasuredOverhead;
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TmmbrSet.ptrTmmbrSet[i] = TMMBRItem.MaxTotalMediaBitRate;
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_tmmbrSetTimeouts[i] = ModuleRTPUtility::GetTimeInMS();
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return;
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}
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}
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VerifyAndAllocateTMMBRSet(TmmbrSet.lengthOfSet+1);
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const WebRtc_UWord32 idx = TmmbrSet.lengthOfSet;
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TmmbrSet.ptrPacketOHSet[idx] = TMMBRItem.MeasuredOverhead;
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TmmbrSet.ptrTmmbrSet[idx] = TMMBRItem.MaxTotalMediaBitRate;
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TmmbrSet.ptrSsrcSet[idx] = senderSSRC;
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_tmmbrSetTimeouts[idx] = ModuleRTPUtility::GetTimeInMS();
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TmmbrSet.lengthOfSet++;
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}
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WebRtc_Word32
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RTCPReceiveInformation::GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
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const WebRtc_UWord32 targetIdx,
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TMMBRSet* candidateSet)
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{
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if(sourceIdx >= TmmbrSet.lengthOfSet)
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{
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return -1;
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}
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if(targetIdx >= candidateSet->sizeOfSet)
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{
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return -1;
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}
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WebRtc_UWord32 timeNow = ModuleRTPUtility::GetTimeInMS();
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// use audio define since we don't know what interval the remote peer is using
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if(timeNow - _tmmbrSetTimeouts[sourceIdx] > 5*RTCP_INTERVAL_AUDIO_MS)
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{
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// value timed out
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const WebRtc_UWord32 move = TmmbrSet.lengthOfSet - (sourceIdx + 1);
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if(move > 0)
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{
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memmove(&(TmmbrSet.ptrTmmbrSet[sourceIdx]), &(TmmbrSet.ptrTmmbrSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
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memmove(&(TmmbrSet.ptrPacketOHSet[sourceIdx]),&(TmmbrSet.ptrPacketOHSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
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memmove(&(TmmbrSet.ptrSsrcSet[sourceIdx]),&(TmmbrSet.ptrSsrcSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
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memmove(&(_tmmbrSetTimeouts[sourceIdx]),&(_tmmbrSetTimeouts[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
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}
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TmmbrSet.lengthOfSet--;
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return -1;
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}
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candidateSet->ptrTmmbrSet[targetIdx] = TmmbrSet.ptrTmmbrSet[sourceIdx];
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candidateSet->ptrPacketOHSet[targetIdx] = TmmbrSet.ptrPacketOHSet[sourceIdx];
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candidateSet->ptrSsrcSet[targetIdx] = TmmbrSet.ptrSsrcSet[sourceIdx];
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return 0;
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}
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void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize)
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{
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TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
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}
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} // namespace webrtc
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