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/*
* Copyright (c) 2012 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 "webrtc/modules/rtp_rtcp/source/rtcp_receiver_help.h"
#include <assert.h> // assert
#include <string.h> // memset
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
namespace webrtc {
using namespace RTCPHelp;
RTCPPacketInformation::RTCPPacketInformation()
: rtcpPacketTypeFlags(0),
remoteSSRC(0),
nackSequenceNumbers(),
applicationSubType(0),
applicationName(0),
applicationData(),
applicationLength(0),
rtt(0),
interArrivalJitter(0),
sliPictureId(0),
rpsiPictureId(0),
receiverEstimatedMaxBitrate(0),
ntp_secs(0),
ntp_frac(0),
rtp_timestamp(0),
xr_originator_ssrc(0),
xr_dlrr_item(false),
VoIPMetric(NULL) {
}
RTCPPacketInformation::~RTCPPacketInformation()
{
delete [] applicationData;
delete VoIPMetric;
}
void
RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
{
VoIPMetric = new RTCPVoIPMetric();
memcpy(VoIPMetric, metric, sizeof(RTCPVoIPMetric));
}
void RTCPPacketInformation::AddApplicationData(const uint8_t* data,
const uint16_t size) {
uint8_t* oldData = applicationData;
uint16_t oldLength = applicationLength;
// Don't copy more than kRtcpAppCode_DATA_SIZE bytes.
uint16_t copySize = size;
if (size > kRtcpAppCode_DATA_SIZE) {
copySize = kRtcpAppCode_DATA_SIZE;
}
applicationLength += copySize;
applicationData = new uint8_t[applicationLength];
if (oldData)
{
memcpy(applicationData, oldData, oldLength);
memcpy(applicationData+oldLength, data, copySize);
delete [] oldData;
} else
{
memcpy(applicationData, data, copySize);
}
}
void
RTCPPacketInformation::ResetNACKPacketIdArray()
{
nackSequenceNumbers.clear();
}
void
RTCPPacketInformation::AddNACKPacket(const uint16_t packetID)
{
if (nackSequenceNumbers.size() >= kSendSideNackListSizeSanity) {
return;
}
nackSequenceNumbers.push_back(packetID);
}
void
RTCPPacketInformation::AddReportInfo(
const RTCPReportBlockInformation& report_block_info)
{
this->rtt = report_block_info.RTT;
report_blocks.push_back(report_block_info.remoteReceiveBlock);
}
RTCPReportBlockInformation::RTCPReportBlockInformation():
remoteReceiveBlock(),
remoteMaxJitter(0),
RTT(0),
minRTT(0),
maxRTT(0),
avgRTT(0),
numAverageCalcs(0)
{
memset(&remoteReceiveBlock,0,sizeof(remoteReceiveBlock));
}
RTCPReportBlockInformation::~RTCPReportBlockInformation()
{
}
RTCPReceiveInformation::RTCPReceiveInformation()
: lastTimeReceived(0),
lastFIRSequenceNumber(-1),
lastFIRRequest(0),
readyForDelete(false) {
}
RTCPReceiveInformation::~RTCPReceiveInformation() {
}
// Increase size of TMMBRSet if needed, and also take care of
// the _tmmbrSetTimeouts vector.
void RTCPReceiveInformation::VerifyAndAllocateTMMBRSet(
const uint32_t minimumSize) {
if (minimumSize > TmmbrSet.sizeOfSet()) {
TmmbrSet.VerifyAndAllocateSetKeepingData(minimumSize);
// make sure that our buffers are big enough
_tmmbrSetTimeouts.reserve(minimumSize);
}
}
void RTCPReceiveInformation::InsertTMMBRItem(
const uint32_t senderSSRC,
const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
const int64_t currentTimeMS) {
// serach to see if we have it in our list
for (uint32_t i = 0; i < TmmbrSet.lengthOfSet(); i++) {
if (TmmbrSet.Ssrc(i) == senderSSRC) {
// we already have this SSRC in our list update it
TmmbrSet.SetEntry(i,
TMMBRItem.MaxTotalMediaBitRate,
TMMBRItem.MeasuredOverhead,
senderSSRC);
_tmmbrSetTimeouts[i] = currentTimeMS;
return;
}
}
VerifyAndAllocateTMMBRSet(TmmbrSet.lengthOfSet() + 1);
TmmbrSet.AddEntry(TMMBRItem.MaxTotalMediaBitRate,
TMMBRItem.MeasuredOverhead,
senderSSRC);
_tmmbrSetTimeouts.push_back(currentTimeMS);
}
int32_t RTCPReceiveInformation::GetTMMBRSet(
const uint32_t sourceIdx,
const uint32_t targetIdx,
TMMBRSet* candidateSet,
const int64_t currentTimeMS) {
if (sourceIdx >= TmmbrSet.lengthOfSet()) {
return -1;
}
if (targetIdx >= candidateSet->sizeOfSet()) {
return -1;
}
// use audio define since we don't know what interval the remote peer is using
if (currentTimeMS - _tmmbrSetTimeouts[sourceIdx] >
5 * RTCP_INTERVAL_AUDIO_MS) {
// value timed out
TmmbrSet.RemoveEntry(sourceIdx);
_tmmbrSetTimeouts.erase(_tmmbrSetTimeouts.begin() + sourceIdx);
return -1;
}
candidateSet->SetEntry(targetIdx,
TmmbrSet.Tmmbr(sourceIdx),
TmmbrSet.PacketOH(sourceIdx),
TmmbrSet.Ssrc(sourceIdx));
return 0;
}
void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(
const uint32_t minimumSize) {
TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
}
} // namespace webrtc
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