| /* |
| * 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 "rtcp_receiver.h" |
| |
| #include <string.h> //memset |
| #include <cassert> //assert |
| |
| #include "trace.h" |
| #include "critical_section_wrapper.h" |
| #include "rtcp_utility.h" |
| #include "rtp_rtcp_impl.h" |
| |
| namespace |
| { |
| const float FRAC = 4.294967296E9; |
| } |
| |
| namespace webrtc { |
| using namespace RTCPUtility; |
| using namespace RTCPHelp; |
| |
| RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id, |
| RtpRtcpClock* clock, |
| ModuleRtpRtcpImpl* owner) : |
| _id(id), |
| _clock(*clock), |
| _method(kRtcpOff), |
| _lastReceived(0), |
| _rtpRtcp(*owner), |
| _criticalSectionFeedbacks(CriticalSectionWrapper::CreateCriticalSection()), |
| _cbRtcpFeedback(NULL), |
| _cbVideoFeedback(NULL), |
| _criticalSectionRTCPReceiver( |
| CriticalSectionWrapper::CreateCriticalSection()), |
| _SSRC(0), |
| _remoteSSRC(0), |
| _remoteSenderInfo(), |
| _lastReceivedSRNTPsecs(0), |
| _lastReceivedSRNTPfrac(0), |
| _receivedInfoMap(), |
| _packetTimeOutMS(0) |
| { |
| memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo)); |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); |
| } |
| |
| RTCPReceiver::~RTCPReceiver() { |
| delete _criticalSectionRTCPReceiver; |
| delete _criticalSectionFeedbacks; |
| |
| while (!_receivedReportBlockMap.empty()) { |
| std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator first = |
| _receivedReportBlockMap.begin(); |
| delete first->second; |
| _receivedReportBlockMap.erase(first); |
| } |
| while (!_receivedInfoMap.empty()) { |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator first = |
| _receivedInfoMap.begin(); |
| delete first->second; |
| _receivedInfoMap.erase(first); |
| } |
| while (!_receivedCnameMap.empty()) { |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator first = |
| _receivedCnameMap.begin(); |
| delete first->second; |
| _receivedCnameMap.erase(first); |
| } |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, |
| "%s deleted", __FUNCTION__); |
| } |
| |
| void |
| RTCPReceiver::ChangeUniqueId(const WebRtc_Word32 id) |
| { |
| _id = id; |
| } |
| |
| RTCPMethod |
| RTCPReceiver::Status() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| return _method; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::SetRTCPStatus(const RTCPMethod method) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| _method = method; |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTCPReceiver::LastReceived() |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| return _lastReceived; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::SetRemoteSSRC( const WebRtc_UWord32 ssrc) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| // new SSRC reset old reports |
| memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo)); |
| _lastReceivedSRNTPsecs = 0; |
| _lastReceivedSRNTPfrac = 0; |
| |
| _remoteSSRC = ssrc; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback) |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| _cbRtcpFeedback = incomingMessagesCallback; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback) |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| _cbVideoFeedback = incomingMessagesCallback; |
| return 0; |
| } |
| |
| void |
| RTCPReceiver::SetSSRC( const WebRtc_UWord32 ssrc) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| _SSRC = ssrc; |
| } |
| |
| WebRtc_Word32 RTCPReceiver::ResetRTT(const WebRtc_UWord32 remoteSSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| RTCPReportBlockInformation* reportBlock = |
| GetReportBlockInformation(remoteSSRC); |
| if (reportBlock == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "\tfailed to GetReportBlockInformation(%u)", remoteSSRC); |
| return -1; |
| } |
| reportBlock->RTT = 0; |
| reportBlock->avgRTT = 0; |
| reportBlock->minRTT = 0; |
| reportBlock->maxRTT = 0; |
| |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPReceiver::RTT(const WebRtc_UWord32 remoteSSRC, |
| WebRtc_UWord16* RTT, |
| WebRtc_UWord16* avgRTT, |
| WebRtc_UWord16* minRTT, |
| WebRtc_UWord16* maxRTT) const { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| RTCPReportBlockInformation* reportBlock = |
| GetReportBlockInformation(remoteSSRC); |
| |
| if (reportBlock == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "\tfailed to GetReportBlockInformation(%u)", |
| remoteSSRC); |
| return -1; |
| } |
| if (RTT) { |
| *RTT = reportBlock->RTT; |
| } |
| if (avgRTT) { |
| *avgRTT = reportBlock->avgRTT; |
| } |
| if (minRTT) { |
| *minRTT = reportBlock->minRTT; |
| } |
| if (maxRTT) { |
| *maxRTT = reportBlock->maxRTT; |
| } |
| return 0; |
| } |
| |
| void |
| RTCPReceiver::UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| if(_cbRtcpFeedback) |
| { |
| _cbRtcpFeedback->OnLipSyncUpdate(_id,audioVideoOffset); |
| } |
| }; |
| |
| WebRtc_Word32 |
| RTCPReceiver::NTP(WebRtc_UWord32 *ReceivedNTPsecs, |
| WebRtc_UWord32 *ReceivedNTPfrac, |
| WebRtc_UWord32 *RTCPArrivalTimeSecs, |
| WebRtc_UWord32 *RTCPArrivalTimeFrac) const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| if(ReceivedNTPsecs) |
| { |
| *ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport |
| } |
| if(ReceivedNTPfrac) |
| { |
| *ReceivedNTPfrac = _remoteSenderInfo.NTPfraction; |
| } |
| if(RTCPArrivalTimeFrac) |
| { |
| *RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block |
| } |
| if(RTCPArrivalTimeSecs) |
| { |
| *RTCPArrivalTimeSecs = _lastReceivedSRNTPsecs; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const |
| { |
| if(senderInfo == NULL) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| if(_lastReceivedSRNTPsecs == 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__); |
| return -1; |
| } |
| memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo)); |
| return 0; |
| } |
| |
| // statistics |
| // we can get multiple receive reports when we receive the report from a CE |
| WebRtc_Word32 RTCPReceiver::StatisticsReceived( |
| std::vector<RTCPReportBlock>* receiveBlocks) const { |
| assert(receiveBlocks); |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it = |
| _receivedReportBlockMap.begin(); |
| |
| while (it != _receivedReportBlockMap.end()) { |
| receiveBlocks->push_back(it->second->remoteReceiveBlock); |
| it++; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation, |
| RTCPUtility::RTCPParserV2* rtcpParser) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| _lastReceived = _clock.GetTimeInMS(); |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin(); |
| while (pktType != RTCPUtility::kRtcpNotValidCode) |
| { |
| // Each "case" is responsible for iterate the parser to the |
| // next top level packet. |
| switch (pktType) |
| { |
| case RTCPUtility::kRtcpSrCode: |
| case RTCPUtility::kRtcpRrCode: |
| HandleSenderReceiverReport(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpSdesCode: |
| HandleSDES(*rtcpParser); |
| break; |
| case RTCPUtility::kRtcpXrVoipMetricCode: |
| HandleXRVOIPMetric(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpByeCode: |
| HandleBYE(*rtcpParser); |
| break; |
| case RTCPUtility::kRtcpRtpfbNackCode: |
| HandleNACK(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpRtpfbTmmbrCode: |
| HandleTMMBR(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpRtpfbTmmbnCode: |
| HandleTMMBN(*rtcpParser); |
| break; |
| case RTCPUtility::kRtcpRtpfbSrReqCode: |
| HandleSR_REQ(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpPsfbPliCode: |
| HandlePLI(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpPsfbSliCode: |
| HandleSLI(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpPsfbRpsiCode: |
| HandleRPSI(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpExtendedIjCode: |
| HandleIJ(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpPsfbFirCode: |
| HandleFIR(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpPsfbAppCode: |
| HandlePsfbApp(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpAppCode: |
| // generic application messages |
| HandleAPP(*rtcpParser, rtcpPacketInformation); |
| break; |
| case RTCPUtility::kRtcpAppItemCode: |
| // generic application messages |
| HandleAPPItem(*rtcpParser, rtcpPacketInformation); |
| break; |
| default: |
| rtcpParser->Iterate(); |
| break; |
| } |
| pktType = rtcpParser->PacketType(); |
| } |
| return 0; |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| RTCPUtility::RTCPPacketTypes rtcpPacketType = rtcpParser.PacketType(); |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| assert((rtcpPacketType == RTCPUtility::kRtcpRrCode) || (rtcpPacketType == RTCPUtility::kRtcpSrCode)); |
| |
| // SR.SenderSSRC |
| // The synchronization source identifier for the originator of this SR packet |
| |
| // rtcpPacket.RR.SenderSSRC |
| // The source of the packet sender, same as of SR? or is this a CE? |
| |
| const WebRtc_UWord32 remoteSSRC = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.SenderSSRC:rtcpPacket.SR.SenderSSRC; |
| const WebRtc_UWord8 numberOfReportBlocks = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.NumberOfReportBlocks:rtcpPacket.SR.NumberOfReportBlocks; |
| |
| rtcpPacketInformation.remoteSSRC = remoteSSRC; |
| |
| RTCPReceiveInformation* ptrReceiveInfo = CreateReceiveInformation(remoteSSRC); |
| if (!ptrReceiveInfo) |
| { |
| rtcpParser.Iterate(); |
| return; |
| } |
| |
| if (rtcpPacketType == RTCPUtility::kRtcpSrCode) |
| { |
| WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id, |
| "Received SR(%d). SSRC:0x%x, from SSRC:0x%x, to us %d.", _id, _SSRC, remoteSSRC, (_remoteSSRC == remoteSSRC)?1:0); |
| |
| if (_remoteSSRC == remoteSSRC) // have I received RTP packets from this party |
| { |
| // only signal that we have received a SR when we accept one |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSr; |
| |
| // We will only store the send report from one source, but |
| // we will store all the receive block |
| |
| // Save the NTP time of this report |
| _remoteSenderInfo.NTPseconds = rtcpPacket.SR.NTPMostSignificant; |
| _remoteSenderInfo.NTPfraction = rtcpPacket.SR.NTPLeastSignificant; |
| _remoteSenderInfo.RTPtimeStamp = rtcpPacket.SR.RTPTimestamp; |
| _remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount; |
| _remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount; |
| |
| _clock.CurrentNTP(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac); |
| } |
| else |
| { |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr; |
| } |
| } else |
| { |
| WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id, |
| "Received RR(%d). SSRC:0x%x, from SSRC:0x%x", _id, _SSRC, remoteSSRC); |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr; |
| } |
| UpdateReceiveInformation(*ptrReceiveInfo); |
| |
| rtcpPacketType = rtcpParser.Iterate(); |
| |
| while (rtcpPacketType == RTCPUtility::kRtcpReportBlockItemCode) |
| { |
| HandleReportBlock(rtcpPacket, rtcpPacketInformation, remoteSSRC, numberOfReportBlocks); |
| rtcpPacketType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation, |
| const WebRtc_UWord32 remoteSSRC, |
| const WebRtc_UWord8 numberOfReportBlocks) { |
| // This will be called once per report block in the RTCP packet. |
| // We filter out all report blocks that are not for us. |
| // Each packet has max 31 RR blocks. |
| // |
| // We can calc RTT if we send a send report and get a report block back. |
| |
| // |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to |
| // which the information in this reception report block pertains. |
| |
| // Filter out all report blocks that are not for us. |
| if (rtcpPacket.ReportBlockItem.SSRC != _SSRC) { |
| // This block is not for us ignore it. |
| return; |
| } |
| |
| // To avoid problem with acquiring _criticalSectionRTCPSender while holding |
| // _criticalSectionRTCPReceiver. |
| _criticalSectionRTCPReceiver->Leave(); |
| WebRtc_UWord32 sendTimeMS = |
| _rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR); |
| _criticalSectionRTCPReceiver->Enter(); |
| |
| RTCPReportBlockInformation* reportBlock = |
| CreateReportBlockInformation(remoteSSRC); |
| if (reportBlock == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC); |
| return; |
| } |
| const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem; |
| reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC; |
| reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC; |
| reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost; |
| reportBlock->remoteReceiveBlock.cumulativeLost = |
| rb.CumulativeNumOfPacketsLost; |
| reportBlock->remoteReceiveBlock.extendedHighSeqNum = |
| rb.ExtendedHighestSequenceNumber; |
| reportBlock->remoteReceiveBlock.jitter = rb.Jitter; |
| reportBlock->remoteReceiveBlock.delaySinceLastSR = rb.DelayLastSR; |
| reportBlock->remoteReceiveBlock.lastSR = rb.LastSR; |
| |
| if (rtcpPacket.ReportBlockItem.Jitter > reportBlock->remoteMaxJitter) { |
| reportBlock->remoteMaxJitter = rtcpPacket.ReportBlockItem.Jitter; |
| } |
| |
| WebRtc_UWord32 delaySinceLastSendReport = |
| rtcpPacket.ReportBlockItem.DelayLastSR; |
| |
| // local NTP time when we received this |
| WebRtc_UWord32 lastReceivedRRNTPsecs = 0; |
| WebRtc_UWord32 lastReceivedRRNTPfrac = 0; |
| |
| _clock.CurrentNTP(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); |
| |
| // time when we received this in MS |
| WebRtc_UWord32 receiveTimeMS = ModuleRTPUtility::ConvertNTPTimeToMS( |
| lastReceivedRRNTPsecs, lastReceivedRRNTPfrac); |
| |
| // Estimate RTT |
| WebRtc_UWord32 d = (delaySinceLastSendReport & 0x0000ffff) * 1000; |
| d /= 65536; |
| d += ((delaySinceLastSendReport & 0xffff0000) >> 16) * 1000; |
| |
| WebRtc_Word32 RTT = 0; |
| |
| if (sendTimeMS > 0) { |
| RTT = receiveTimeMS - d - sendTimeMS; |
| if (RTT <= 0) { |
| RTT = 1; |
| } |
| if (RTT > reportBlock->maxRTT) { |
| // store max RTT |
| reportBlock->maxRTT = (WebRtc_UWord16) RTT; |
| } |
| if (reportBlock->minRTT == 0) { |
| // first RTT |
| reportBlock->minRTT = (WebRtc_UWord16) RTT; |
| } else if (RTT < reportBlock->minRTT) { |
| // Store min RTT |
| reportBlock->minRTT = (WebRtc_UWord16) RTT; |
| } |
| // store last RTT |
| reportBlock->RTT = (WebRtc_UWord16) RTT; |
| |
| // store average RTT |
| if (reportBlock->numAverageCalcs != 0) { |
| float ac = static_cast<float> (reportBlock->numAverageCalcs); |
| float newAverage = ((ac / (ac + 1)) * reportBlock->avgRTT) |
| + ((1 / (ac + 1)) * RTT); |
| reportBlock->avgRTT = static_cast<int> (newAverage + 0.5f); |
| } else { |
| // first RTT |
| reportBlock->avgRTT = (WebRtc_UWord16) RTT; |
| } |
| reportBlock->numAverageCalcs++; |
| } |
| |
| WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id, |
| " -> Received report block(%d), from SSRC:0x%x, RTT:%d, loss:%d", |
| _id, remoteSSRC, RTT, rtcpPacket.ReportBlockItem.FractionLost); |
| |
| // rtcpPacketInformation |
| rtcpPacketInformation.AddReportInfo( |
| reportBlock->remoteReceiveBlock.fractionLost, (WebRtc_UWord16) RTT, |
| reportBlock->remoteReceiveBlock.extendedHighSeqNum, |
| reportBlock->remoteReceiveBlock.jitter); |
| } |
| |
| RTCPReportBlockInformation* |
| RTCPReceiver::CreateReportBlockInformation(WebRtc_UWord32 remoteSSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator it = |
| _receivedReportBlockMap.find(remoteSSRC); |
| |
| RTCPReportBlockInformation* ptrReportBlockInfo = NULL; |
| if (it != _receivedReportBlockMap.end()) { |
| ptrReportBlockInfo = it->second; |
| } else { |
| ptrReportBlockInfo = new RTCPReportBlockInformation; |
| _receivedReportBlockMap[remoteSSRC] = ptrReportBlockInfo; |
| } |
| return ptrReportBlockInfo; |
| } |
| |
| RTCPReportBlockInformation* |
| RTCPReceiver::GetReportBlockInformation(WebRtc_UWord32 remoteSSRC) const { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it = |
| _receivedReportBlockMap.find(remoteSSRC); |
| |
| if (it == _receivedReportBlockMap.end()) { |
| return NULL; |
| } |
| return it->second; |
| } |
| |
| RTCPCnameInformation* |
| RTCPReceiver::CreateCnameInformation(WebRtc_UWord32 remoteSSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it = |
| _receivedCnameMap.find(remoteSSRC); |
| |
| if (it != _receivedCnameMap.end()) { |
| return it->second; |
| } |
| RTCPCnameInformation* cnameInfo = new RTCPCnameInformation; |
| memset(cnameInfo->name, 0, RTCP_CNAME_SIZE); |
| _receivedCnameMap[remoteSSRC] = cnameInfo; |
| return cnameInfo; |
| } |
| |
| RTCPCnameInformation* |
| RTCPReceiver::GetCnameInformation(WebRtc_UWord32 remoteSSRC) const { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::const_iterator it = |
| _receivedCnameMap.find(remoteSSRC); |
| |
| if (it == _receivedCnameMap.end()) { |
| return NULL; |
| } |
| return it->second; |
| } |
| |
| RTCPReceiveInformation* |
| RTCPReceiver::CreateReceiveInformation(WebRtc_UWord32 remoteSSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it = |
| _receivedInfoMap.find(remoteSSRC); |
| |
| if (it != _receivedInfoMap.end()) { |
| return it->second; |
| } |
| RTCPReceiveInformation* receiveInfo = new RTCPReceiveInformation; |
| _receivedInfoMap[remoteSSRC] = receiveInfo; |
| return receiveInfo; |
| } |
| |
| RTCPReceiveInformation* |
| RTCPReceiver::GetReceiveInformation(WebRtc_UWord32 remoteSSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it = |
| _receivedInfoMap.find(remoteSSRC); |
| if (it == _receivedInfoMap.end()) { |
| return NULL; |
| } |
| return it->second; |
| } |
| |
| void RTCPReceiver::UpdateReceiveInformation( |
| RTCPReceiveInformation& receiveInformation) { |
| // Update that this remote is alive |
| receiveInformation.lastTimeReceived = _clock.GetTimeInMS(); |
| } |
| |
| bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| bool updateBoundingSet = false; |
| WebRtc_UWord32 timeNow = _clock.GetTimeInMS(); |
| |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt = |
| _receivedInfoMap.begin(); |
| |
| while (receiveInfoIt != _receivedInfoMap.end()) { |
| RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; |
| if (receiveInfo == NULL) { |
| return updateBoundingSet; |
| } |
| // time since last received rtcp packet |
| // when we dont have a lastTimeReceived and the object is marked |
| // readyForDelete it's removed from the map |
| if (receiveInfo->lastTimeReceived) { |
| /// use audio define since we don't know what interval the remote peer is |
| // using |
| if ((timeNow - receiveInfo->lastTimeReceived) > |
| 5 * RTCP_INTERVAL_AUDIO_MS) { |
| // no rtcp packet for the last five regular intervals, reset limitations |
| receiveInfo->TmmbrSet.lengthOfSet = 0; |
| // prevent that we call this over and over again |
| receiveInfo->lastTimeReceived = 0; |
| // send new TMMBN to all channels using the default codec |
| updateBoundingSet = true; |
| } |
| receiveInfoIt++; |
| } else if (receiveInfo->readyForDelete) { |
| // store our current receiveInfoItem |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator |
| receiveInfoItemToBeErased = receiveInfoIt; |
| receiveInfoIt++; |
| delete receiveInfoItemToBeErased->second; |
| _receivedInfoMap.erase(receiveInfoItemToBeErased); |
| } else { |
| receiveInfoIt++; |
| } |
| } |
| return updateBoundingSet; |
| } |
| |
| WebRtc_Word32 RTCPReceiver::BoundingSet(bool &tmmbrOwner, |
| TMMBRSet*& boundingSetRec) { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt = |
| _receivedInfoMap.find(_remoteSSRC); |
| |
| if (receiveInfoIt == _receivedInfoMap.end()) { |
| return -1; |
| } |
| RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; |
| if (receiveInfo == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s failed to get RTCPReceiveInformation", |
| __FUNCTION__); |
| return -1; |
| } |
| if (receiveInfo->TmmbnBoundingSet.lengthOfSet > 0) { |
| boundingSetRec->VerifyAndAllocateSet( |
| receiveInfo->TmmbnBoundingSet.lengthOfSet + 1); |
| for(WebRtc_UWord32 i=0; i< receiveInfo->TmmbnBoundingSet.lengthOfSet; i++) { |
| if(receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i] == _SSRC) { |
| // owner of bounding set |
| tmmbrOwner = true; |
| } |
| boundingSetRec->ptrTmmbrSet[i] = |
| receiveInfo->TmmbnBoundingSet.ptrTmmbrSet[i]; |
| boundingSetRec->ptrPacketOHSet[i] = |
| receiveInfo->TmmbnBoundingSet.ptrPacketOHSet[i]; |
| boundingSetRec->ptrSsrcSet[i] = |
| receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i]; |
| } |
| } |
| return receiveInfo->TmmbnBoundingSet.lengthOfSet; |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser) |
| { |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpSdesChunkCode) |
| { |
| HandleSDESChunk(rtcpParser); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| RTCPCnameInformation* cnameInfo = |
| CreateCnameInformation(rtcpPacket.CName.SenderSSRC); |
| assert(cnameInfo); |
| |
| cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0; |
| strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1); |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.NACK.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| if (_SSRC != rtcpPacket.NACK.MediaSSRC) |
| { |
| // Not to us. |
| rtcpParser.Iterate(); |
| return; |
| } |
| |
| rtcpPacketInformation.ResetNACKPacketIdArray(); |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpRtpfbNackItemCode) |
| { |
| HandleNACKItem(rtcpPacket, rtcpPacketInformation); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID); |
| |
| WebRtc_UWord16 bitMask = rtcpPacket.NACKItem.BitMask; |
| if(bitMask) |
| { |
| for(int i=1; i <= 16; ++i) |
| { |
| if(bitMask & 0x01) |
| { |
| rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID + i); |
| } |
| bitMask = bitMask >>1; |
| } |
| } |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpNack; |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void RTCPReceiver::HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser) { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| // clear our lists |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator |
| reportBlockInfoIt = _receivedReportBlockMap.find( |
| rtcpPacket.BYE.SenderSSRC); |
| |
| if (reportBlockInfoIt != _receivedReportBlockMap.end()) { |
| delete reportBlockInfoIt->second; |
| _receivedReportBlockMap.erase(reportBlockInfoIt); |
| } |
| // we can't delete it due to TMMBR |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt = |
| _receivedInfoMap.find(rtcpPacket.BYE.SenderSSRC); |
| |
| if (receiveInfoIt != _receivedInfoMap.end()) { |
| receiveInfoIt->second->readyForDelete = true; |
| } |
| |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator cnameInfoIt = |
| _receivedCnameMap.find(rtcpPacket.BYE.SenderSSRC); |
| |
| if (cnameInfoIt != _receivedCnameMap.end()) { |
| delete cnameInfoIt->second; |
| _receivedCnameMap.erase(cnameInfoIt); |
| } |
| rtcpParser.Iterate(); |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| if(rtcpPacket.XRVOIPMetricItem.SSRC == _SSRC) |
| { |
| // Store VoIP metrics block if it's about me |
| // from OriginatorSSRC do we filter it? |
| // rtcpPacket.XR.OriginatorSSRC; |
| |
| RTCPVoIPMetric receivedVoIPMetrics; |
| receivedVoIPMetrics.burstDensity = rtcpPacket.XRVOIPMetricItem.burstDensity; |
| receivedVoIPMetrics.burstDuration = rtcpPacket.XRVOIPMetricItem.burstDuration; |
| receivedVoIPMetrics.discardRate = rtcpPacket.XRVOIPMetricItem.discardRate; |
| receivedVoIPMetrics.endSystemDelay = rtcpPacket.XRVOIPMetricItem.endSystemDelay; |
| receivedVoIPMetrics.extRfactor = rtcpPacket.XRVOIPMetricItem.extRfactor; |
| receivedVoIPMetrics.gapDensity = rtcpPacket.XRVOIPMetricItem.gapDensity; |
| receivedVoIPMetrics.gapDuration = rtcpPacket.XRVOIPMetricItem.gapDuration; |
| receivedVoIPMetrics.Gmin = rtcpPacket.XRVOIPMetricItem.Gmin; |
| receivedVoIPMetrics.JBabsMax = rtcpPacket.XRVOIPMetricItem.JBabsMax; |
| receivedVoIPMetrics.JBmax = rtcpPacket.XRVOIPMetricItem.JBmax; |
| receivedVoIPMetrics.JBnominal = rtcpPacket.XRVOIPMetricItem.JBnominal; |
| receivedVoIPMetrics.lossRate = rtcpPacket.XRVOIPMetricItem.lossRate; |
| receivedVoIPMetrics.MOSCQ = rtcpPacket.XRVOIPMetricItem.MOSCQ; |
| receivedVoIPMetrics.MOSLQ = rtcpPacket.XRVOIPMetricItem.MOSLQ; |
| receivedVoIPMetrics.noiseLevel = rtcpPacket.XRVOIPMetricItem.noiseLevel; |
| receivedVoIPMetrics.RERL = rtcpPacket.XRVOIPMetricItem.RERL; |
| receivedVoIPMetrics.Rfactor = rtcpPacket.XRVOIPMetricItem.Rfactor; |
| receivedVoIPMetrics.roundTripDelay = rtcpPacket.XRVOIPMetricItem.roundTripDelay; |
| receivedVoIPMetrics.RXconfig = rtcpPacket.XRVOIPMetricItem.RXconfig; |
| receivedVoIPMetrics.signalLevel = rtcpPacket.XRVOIPMetricItem.signalLevel; |
| |
| rtcpPacketInformation.AddVoIPMetric(&receivedVoIPMetrics); |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrVoipMetric; // received signal |
| } |
| rtcpParser.Iterate(); |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.PLI.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| if (_SSRC != rtcpPacket.PLI.MediaSSRC) |
| { |
| // Not to us. |
| rtcpParser.Iterate(); |
| return; |
| } |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli; // received signal that we need to send a new key frame |
| rtcpParser.Iterate(); |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| WebRtc_UWord32 senderSSRC = rtcpPacket.TMMBR.SenderSSRC; |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(senderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| if(rtcpPacket.TMMBR.MediaSSRC) |
| { |
| // rtcpPacket.TMMBR.MediaSSRC SHOULD be 0 if same as SenderSSRC |
| // in relay mode this is a valid number |
| senderSSRC = rtcpPacket.TMMBR.MediaSSRC; |
| } |
| |
| // Use packet length to calc max number of TMMBR blocks |
| // each TMMBR block is 8 bytes |
| ptrdiff_t maxNumOfTMMBRBlocks = rtcpParser.LengthLeft() / 8; |
| |
| // sanity |
| if(maxNumOfTMMBRBlocks > 200) // we can't have more than what's in one packet |
| { |
| assert(false); |
| rtcpParser.Iterate(); |
| return; |
| } |
| ptrReceiveInfo->VerifyAndAllocateTMMBRSet((WebRtc_UWord32)maxNumOfTMMBRBlocks); |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpRtpfbTmmbrItemCode) |
| { |
| HandleTMMBRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation, senderSSRC); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo, |
| const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation, |
| const WebRtc_UWord32 senderSSRC) |
| { |
| if (_SSRC == rtcpPacket.TMMBRItem.SSRC && |
| rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0) |
| { |
| receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem, |
| _clock.GetTimeInMS()); |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr; |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.TMMBN.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| // Use packet length to calc max number of TMMBN blocks |
| // each TMMBN block is 8 bytes |
| ptrdiff_t maxNumOfTMMBNBlocks = rtcpParser.LengthLeft() / 8; |
| |
| // sanity |
| if(maxNumOfTMMBNBlocks > 200) // we cant have more than what's in one packet |
| { |
| assert(false); |
| rtcpParser.Iterate(); |
| return; |
| } |
| |
| ptrReceiveInfo->VerifyAndAllocateBoundingSet((WebRtc_UWord32)maxNumOfTMMBNBlocks); |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode) |
| { |
| HandleTMMBNItem(*ptrReceiveInfo, rtcpPacket); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSrReq; |
| rtcpParser.Iterate(); |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo, |
| const RTCPUtility::RTCPPacket& rtcpPacket) |
| { |
| const unsigned int idx = receiveInfo.TmmbnBoundingSet.lengthOfSet; |
| |
| receiveInfo.TmmbnBoundingSet.ptrTmmbrSet[idx] = rtcpPacket.TMMBNItem.MaxTotalMediaBitRate; |
| receiveInfo.TmmbnBoundingSet.ptrPacketOHSet[idx] = rtcpPacket.TMMBNItem.MeasuredOverhead; |
| receiveInfo.TmmbnBoundingSet.ptrSsrcSet[idx] = rtcpPacket.TMMBNItem.SSRC; |
| |
| ++receiveInfo.TmmbnBoundingSet.lengthOfSet; |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.SLI.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpPsfbSliItemCode) |
| { |
| HandleSLIItem(rtcpPacket, rtcpPacketInformation); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| // in theory there could be multiple slices lost |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSli; // received signal that we need to refresh a slice |
| rtcpPacketInformation.sliPictureId = rtcpPacket.SLIItem.PictureId; |
| } |
| |
| void |
| RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPHelp::RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.RPSI.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| if(pktType == RTCPUtility::kRtcpPsfbRpsiCode) |
| { |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRpsi; // received signal that we have a confirmed reference picture |
| if(rtcpPacket.RPSI.NumberOfValidBits%8 != 0) |
| { |
| // to us unknown |
| // continue |
| rtcpParser.Iterate(); |
| return; |
| } |
| rtcpPacketInformation.rpsiPictureId = 0; |
| |
| // convert NativeBitString to rpsiPictureId |
| WebRtc_UWord8 numberOfBytes = rtcpPacket.RPSI.NumberOfValidBits /8; |
| for(WebRtc_UWord8 n = 0; n < (numberOfBytes-1); n++) |
| { |
| rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[n] & 0x7f); |
| rtcpPacketInformation.rpsiPictureId <<= 7; // prepare next |
| } |
| rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[numberOfBytes-1] & 0x7f); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| if (pktType == RTCPUtility::kRtcpPsfbRembItemCode) |
| { |
| HandleREMBItem(rtcpParser, rtcpPacketInformation); |
| rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpExtendedIjItemCode) |
| { |
| HandleIJItem(rtcpPacket, rtcpPacketInformation); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| void |
| RTCPReceiver::HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset; |
| rtcpPacketInformation.interArrivalJitter = |
| rtcpPacket.ExtendedJitterReportItem.Jitter; |
| } |
| |
| void |
| RTCPReceiver::HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| rtcpParser.Iterate(); |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb; |
| rtcpPacketInformation.receiverEstimatedMaxBitrate = rtcpPacket.REMB.BitRate; |
| // TODO(pwestin) send up SSRCs and do a sanity check |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.FIR.SenderSSRC); |
| if (ptrReceiveInfo == NULL) |
| { |
| // This remote SSRC must be saved before. |
| rtcpParser.Iterate(); |
| return; |
| } |
| |
| RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate(); |
| while (pktType == RTCPUtility::kRtcpPsfbFirItemCode) |
| { |
| HandleFIRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation); |
| pktType = rtcpParser.Iterate(); |
| } |
| } |
| |
| // no need for critsect we have _criticalSectionRTCPReceiver |
| void |
| RTCPReceiver::HandleFIRItem(RTCPReceiveInformation& receiveInfo, |
| const RTCPUtility::RTCPPacket& rtcpPacket, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| if (_SSRC == rtcpPacket.FIRItem.SSRC) // is it our sender that is requested to generate a new keyframe |
| { |
| // rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it |
| // we don't know who this originate from |
| |
| // check if we have reported this FIRSequenceNumber before |
| if (rtcpPacket.FIRItem.CommandSequenceNumber != receiveInfo.lastFIRSequenceNumber) |
| { |
| // |
| WebRtc_UWord32 now = _clock.GetTimeInMS(); |
| |
| // extra sanity don't go crazy with the callbacks |
| if( (now - receiveInfo.lastFIRRequest) > RTCP_MIN_FRAME_LENGTH_MS) |
| { |
| receiveInfo.lastFIRRequest = now; |
| receiveInfo.lastFIRSequenceNumber = rtcpPacket.FIRItem.CommandSequenceNumber; |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir; // received signal that we need to send a new key frame |
| } |
| } |
| } |
| } |
| |
| void |
| RTCPReceiver::HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpApp; |
| rtcpPacketInformation.applicationSubType = rtcpPacket.APP.SubType; |
| rtcpPacketInformation.applicationName = rtcpPacket.APP.Name; |
| |
| rtcpParser.Iterate(); |
| } |
| |
| void |
| RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser, |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); |
| |
| rtcpPacketInformation.AddApplicationData(rtcpPacket.APP.Data, rtcpPacket.APP.Size); |
| |
| rtcpParser.Iterate(); |
| } |
| |
| void |
| RTCPReceiver::OnReceivedIntraFrameRequest(const FrameType frameType, |
| const WebRtc_UWord8 streamIdx) const |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| |
| if(_cbVideoFeedback) |
| { |
| _cbVideoFeedback->OnReceivedIntraFrameRequest(_id, frameType, streamIdx); |
| } |
| } |
| |
| void |
| RTCPReceiver::OnReceivedSliceLossIndication(const WebRtc_UWord8 pitureID) const |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| |
| if(_cbRtcpFeedback) |
| { |
| _cbRtcpFeedback->OnSLIReceived(_id, pitureID); |
| } |
| } |
| |
| void |
| RTCPReceiver::OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64 pitureID) const |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| |
| if(_cbRtcpFeedback) |
| { |
| _cbRtcpFeedback->OnRPSIReceived(_id, pitureID); |
| } |
| } |
| |
| // Holding no Critical section |
| void RTCPReceiver::TriggerCallbacksFromRTCPPacket( |
| RTCPPacketInformation& rtcpPacketInformation) |
| { |
| // Process TMMBR and REMB first to avoid multiple callbacks |
| // to OnNetworkChanged. |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr) |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, |
| "SIG [RTCP] Incoming TMMBR to id:%d", _id); |
| |
| // Might trigger a OnReceivedBandwidthEstimateUpdate. |
| _rtpRtcp.OnReceivedTMMBR(); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, |
| "SIG [RTCP] Incoming REMB to id:%d", _id); |
| |
| // We need to bounce this to the default channel. |
| _rtpRtcp.OnReceivedEstimatedMaxBitrate( |
| rtcpPacketInformation.receiverEstimatedMaxBitrate); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr || |
| rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr) |
| { |
| if (rtcpPacketInformation.reportBlock) |
| { |
| _rtpRtcp.OnPacketLossStatisticsUpdate( |
| rtcpPacketInformation.fractionLost, |
| rtcpPacketInformation.roundTripTime, |
| rtcpPacketInformation.lastReceivedExtendedHighSeqNum); |
| } |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) |
| { |
| _rtpRtcp.OnReceivedNTP(); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq) |
| { |
| _rtpRtcp.OnRequestSendReport(); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) |
| { |
| if (rtcpPacketInformation.nackSequenceNumbersLength > 0) |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, |
| "SIG [RTCP] Incoming NACK to id:%d", _id); |
| _rtpRtcp.OnReceivedNACK( |
| rtcpPacketInformation.nackSequenceNumbersLength, |
| rtcpPacketInformation.nackSequenceNumbers); |
| } |
| } |
| if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) || |
| (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) |
| { |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, |
| "SIG [RTCP] Incoming PLI to id:%d", _id); |
| } else |
| { |
| WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, |
| "SIG [RTCP] Incoming FIR to id:%d", _id); |
| } |
| _rtpRtcp.OnReceivedIntraFrameRequest(&_rtpRtcp); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSli) |
| { |
| // we need use a bounce it up to handle default channel |
| _rtpRtcp.OnReceivedSliceLossIndication( |
| rtcpPacketInformation.sliPictureId); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi) |
| { |
| // we need use a bounce it up to handle default channel |
| _rtpRtcp.OnReceivedReferencePictureSelectionIndication( |
| rtcpPacketInformation.rpsiPictureId); |
| } |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| |
| // we need a feedback that we have received a report block(s) so that we can generate a new packet |
| // in a conference relay scenario, one received report can generate several RTCP packets, based |
| // on number relayed/mixed |
| // a send report block should go out to all receivers |
| if(_cbRtcpFeedback) |
| { |
| if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) |
| { |
| _cbRtcpFeedback->OnSendReportReceived(_id, rtcpPacketInformation.remoteSSRC); |
| } else |
| { |
| _cbRtcpFeedback->OnReceiveReportReceived(_id, rtcpPacketInformation.remoteSSRC); |
| } |
| if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) |
| { |
| _cbRtcpFeedback->OnReceiverEstimatedMaxBitrateReceived(_id, |
| rtcpPacketInformation.receiverEstimatedMaxBitrate); |
| } |
| if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric) |
| { |
| WebRtc_Word8 VoIPmetricBuffer[7*4]; |
| VoIPmetricBuffer[0] = rtcpPacketInformation.VoIPMetric->lossRate; |
| VoIPmetricBuffer[1] = rtcpPacketInformation.VoIPMetric->discardRate; |
| VoIPmetricBuffer[2] = rtcpPacketInformation.VoIPMetric->burstDensity; |
| VoIPmetricBuffer[3] = rtcpPacketInformation.VoIPMetric->gapDensity; |
| |
| VoIPmetricBuffer[4] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration >> 8); |
| VoIPmetricBuffer[5] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration); |
| VoIPmetricBuffer[6] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration >> 8); |
| VoIPmetricBuffer[7] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration); |
| |
| VoIPmetricBuffer[8] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay >> 8); |
| VoIPmetricBuffer[9] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay); |
| VoIPmetricBuffer[10] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay >> 8); |
| VoIPmetricBuffer[11] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay); |
| |
| VoIPmetricBuffer[12] = rtcpPacketInformation.VoIPMetric->signalLevel; |
| VoIPmetricBuffer[13] = rtcpPacketInformation.VoIPMetric->noiseLevel; |
| VoIPmetricBuffer[14] = rtcpPacketInformation.VoIPMetric->RERL; |
| VoIPmetricBuffer[15] = rtcpPacketInformation.VoIPMetric->Gmin; |
| |
| VoIPmetricBuffer[16] = rtcpPacketInformation.VoIPMetric->Rfactor; |
| VoIPmetricBuffer[17] = rtcpPacketInformation.VoIPMetric->extRfactor; |
| VoIPmetricBuffer[18] = rtcpPacketInformation.VoIPMetric->MOSLQ; |
| VoIPmetricBuffer[19] = rtcpPacketInformation.VoIPMetric->MOSCQ; |
| |
| VoIPmetricBuffer[20] = rtcpPacketInformation.VoIPMetric->RXconfig; |
| VoIPmetricBuffer[21] = 0; // reserved |
| VoIPmetricBuffer[22] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal >> 8); |
| VoIPmetricBuffer[23] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal); |
| |
| VoIPmetricBuffer[24] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax >> 8); |
| VoIPmetricBuffer[25] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax); |
| VoIPmetricBuffer[26] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax >> 8); |
| VoIPmetricBuffer[27] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax); |
| |
| _cbRtcpFeedback->OnXRVoIPMetricReceived(_id, rtcpPacketInformation.VoIPMetric, VoIPmetricBuffer); |
| } |
| if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp) |
| { |
| _cbRtcpFeedback->OnApplicationDataReceived(_id, |
| rtcpPacketInformation.applicationSubType, |
| rtcpPacketInformation.applicationName, |
| rtcpPacketInformation.applicationLength, |
| rtcpPacketInformation.applicationData); |
| } |
| } |
| } |
| } |
| |
| void |
| RTCPReceiver::UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit) |
| { |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| |
| if(_cbRtcpFeedback) |
| { |
| _cbRtcpFeedback->OnTMMBRReceived(_id, bwEstimateKbit); |
| } |
| |
| } |
| |
| WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC, |
| char cName[RTCP_CNAME_SIZE]) const { |
| if (cName == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC); |
| assert(cnameInfo); |
| |
| cName[RTCP_CNAME_SIZE - 1] = 0; |
| strncpy(cName, cnameInfo->name, RTCP_CNAME_SIZE - 1); |
| return 0; |
| } |
| |
| // no callbacks allowed inside this function |
| WebRtc_Word32 RTCPReceiver::TMMBRReceived(const WebRtc_UWord32 size, |
| const WebRtc_UWord32 accNumCandidates, |
| TMMBRSet* candidateSet) const { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| |
| std::map<WebRtc_UWord32, RTCPReceiveInformation*>::const_iterator |
| receiveInfoIt = _receivedInfoMap.begin(); |
| if (receiveInfoIt == _receivedInfoMap.end()) { |
| return -1; |
| } |
| WebRtc_UWord32 num = accNumCandidates; |
| if (candidateSet) { |
| while( num < size && receiveInfoIt != _receivedInfoMap.end()) { |
| RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; |
| if (receiveInfo == NULL) { |
| return 0; |
| } |
| for (WebRtc_UWord32 i = 0; |
| (num < size) && (i < receiveInfo->TmmbrSet.lengthOfSet); i++) { |
| if (receiveInfo->GetTMMBRSet(i, num, candidateSet, |
| _clock.GetTimeInMS()) == 0) { |
| num++; |
| } |
| } |
| receiveInfoIt++; |
| } |
| } else { |
| while (receiveInfoIt != _receivedInfoMap.end()) { |
| RTCPReceiveInformation* receiveInfo = receiveInfoIt->second; |
| if(receiveInfo == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s failed to get RTCPReceiveInformation", |
| __FUNCTION__); |
| return -1; |
| } |
| num += receiveInfo->TmmbrSet.lengthOfSet; |
| receiveInfoIt++; |
| } |
| } |
| return num; |
| } |
| |
| WebRtc_Word32 |
| RTCPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| _packetTimeOutMS = timeoutMS; |
| return 0; |
| } |
| |
| void RTCPReceiver::PacketTimeout() |
| { |
| if(_packetTimeOutMS == 0) |
| { |
| // not configured |
| return; |
| } |
| |
| bool packetTimeOut = false; |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPReceiver); |
| if(_lastReceived == 0) |
| { |
| // not active |
| return; |
| } |
| |
| WebRtc_UWord32 now = _clock.GetTimeInMS(); |
| |
| if(now - _lastReceived > _packetTimeOutMS) |
| { |
| packetTimeOut = true; |
| _lastReceived = 0; // only one callback |
| } |
| } |
| CriticalSectionScoped lock(_criticalSectionFeedbacks); |
| if(packetTimeOut && _cbRtcpFeedback) |
| { |
| _cbRtcpFeedback->OnRTCPPacketTimeout(_id); |
| } |
| } |
| } // namespace webrtc |
