/* * 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 //memset #include //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; // The number of RTCP time intervals needed to trigger a timeout. const int kRrTimeoutIntervals = 3; RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock, ModuleRtpRtcpImpl* owner) : TMMBRHelp(), _id(id), _clock(*clock), _method(kRtcpOff), _lastReceived(0), _rtpRtcp(*owner), _criticalSectionFeedbacks( CriticalSectionWrapper::CreateCriticalSection()), _cbRtcpFeedback(NULL), _cbRtcpBandwidthObserver(NULL), _cbRtcpIntraFrameObserver(NULL), _criticalSectionRTCPReceiver( CriticalSectionWrapper::CreateCriticalSection()), _SSRC(0), _remoteSSRC(0), _remoteSenderInfo(), _lastReceivedSRNTPsecs(0), _lastReceivedSRNTPfrac(0), _receivedInfoMap(), _packetTimeOutMS(0), _lastReceivedRrMs(0), _lastIncreasedSequenceNumberMs(0), _rtt(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::iterator first = _receivedReportBlockMap.begin(); delete first->second; _receivedReportBlockMap.erase(first); } while (!_receivedInfoMap.empty()) { std::map::iterator first = _receivedInfoMap.begin(); delete first->second; _receivedInfoMap.erase(first); } while (!_receivedCnameMap.empty()) { std::map::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_Word64 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; } void RTCPReceiver::RegisterRtcpObservers( RtcpIntraFrameObserver* intra_frame_callback, RtcpBandwidthObserver* bandwidth_callback, RtcpFeedback* feedback_callback) { CriticalSectionScoped lock(_criticalSectionFeedbacks); _cbRtcpIntraFrameObserver = intra_frame_callback; _cbRtcpBandwidthObserver = bandwidth_callback; _cbRtcpFeedback = feedback_callback; } void RTCPReceiver::SetSSRC(const WebRtc_UWord32 ssrc) { WebRtc_UWord32 old_ssrc = 0; { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); old_ssrc = _SSRC; _SSRC = ssrc; } { CriticalSectionScoped lock(_criticalSectionFeedbacks); if (_cbRtcpIntraFrameObserver && old_ssrc != ssrc) { _cbRtcpIntraFrameObserver->OnLocalSsrcChanged(old_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) { return -1; } if (RTT) { *RTT = reportBlock->RTT; } if (avgRTT) { *avgRTT = reportBlock->avgRTT; } if (minRTT) { *minRTT = reportBlock->minRTT; } if (maxRTT) { *maxRTT = reportBlock->maxRTT; } return 0; } WebRtc_UWord16 RTCPReceiver::RTT() const { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if (!_receivedReportBlockMap.empty()) { return 0; } return _rtt; } int RTCPReceiver::SetRTT(WebRtc_UWord16 rtt) { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if (!_receivedReportBlockMap.empty()) { return -1; } _rtt = rtt; return 0; } WebRtc_Word32 RTCPReceiver::NTP(WebRtc_UWord32 *ReceivedNTPsecs, WebRtc_UWord32 *ReceivedNTPfrac, WebRtc_UWord32 *RTCPArrivalTimeSecs, WebRtc_UWord32 *RTCPArrivalTimeFrac, WebRtc_UWord32 *rtcp_timestamp) 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; } if (rtcp_timestamp) { *rtcp_timestamp = _remoteSenderInfo.RTPtimeStamp; } 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* receiveBlocks) const { assert(receiveBlocks); CriticalSectionScoped lock(_criticalSectionRTCPReceiver); std::map::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, rtcpPacketInformation); 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; rtcpPacketInformation.ntp_secs = rtcpPacket.SR.NTPMostSignificant; rtcpPacketInformation.ntp_frac = rtcpPacket.SR.NTPLeastSignificant; rtcpPacketInformation.rtp_timestamp = rtcpPacket.SR.RTPTimestamp; // 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; } _lastReceivedRrMs = _clock.GetTimeInMS(); const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem; reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC; reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC; reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost; reportBlock->remoteReceiveBlock.cumulativeLost = rb.CumulativeNumOfPacketsLost; if (rb.ExtendedHighestSequenceNumber > reportBlock->remoteReceiveBlock.extendedHighSeqNum) { // We have successfully delivered new RTP packets to the remote side after // the last RR was sent from the remote side. _lastIncreasedSequenceNumberMs = _lastReceivedRrMs; } 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 (reportBlock->numAverageCalcs); float newAverage = ((ac / (ac + 1)) * reportBlock->avgRTT) + ((1 / (ac + 1)) * RTT); reportBlock->avgRTT = static_cast (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::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::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::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::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::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::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::RtcpRrTimeout(int64_t rtcp_interval_ms) { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if (_lastReceivedRrMs == 0) return false; int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms; if (_clock.GetTimeInMS() > _lastReceivedRrMs + time_out_ms) { // Reset the timer to only trigger one log. _lastReceivedRrMs = 0; return true; } return false; } bool RTCPReceiver::RtcpRrSequenceNumberTimeout(int64_t rtcp_interval_ms) { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); if (_lastIncreasedSequenceNumberMs == 0) return false; int64_t time_out_ms = kRrTimeoutIntervals * rtcp_interval_ms; if (_clock.GetTimeInMS() > _lastIncreasedSequenceNumberMs + time_out_ms) { // Reset the timer to only trigger one log. _lastIncreasedSequenceNumberMs = 0; return true; } return false; } bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() { CriticalSectionScoped lock(_criticalSectionRTCPReceiver); bool updateBoundingSet = false; WebRtc_Word64 timeNow = _clock.GetTimeInMS(); std::map::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.clearSet(); // 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::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::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.Ssrc(i) == _SSRC) { // owner of bounding set tmmbrOwner = true; } boundingSetRec->SetEntry(i, receiveInfo->TmmbnBoundingSet.Tmmbr(i), receiveInfo->TmmbnBoundingSet.PacketOH(i), receiveInfo->TmmbnBoundingSet.Ssrc(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(); 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::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::iterator receiveInfoIt = _receivedInfoMap.find(rtcpPacket.BYE.SenderSSRC); if (receiveInfoIt != _receivedInfoMap.end()) { receiveInfoIt->second->readyForDelete = true; } std::map::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(); if (_SSRC == rtcpPacket.PLI.MediaSSRC) { // Received a signal that we need to send a new key frame. rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli; } 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, RTCPPacketInformation& rtcpPacketInformation) { 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; } rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbn; // 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) { receiveInfo.TmmbnBoundingSet.AddEntry( rtcpPacket.TMMBNItem.MaxTotalMediaBitRate, rtcpPacket.TMMBNItem.MeasuredOverhead, rtcpPacket.TMMBNItem.SSRC); } // no need for critsect we have _criticalSectionRTCPReceiver void RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser, RTCPPacketInformation& rtcpPacketInformation) { const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); 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(); 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::kRtcpPsfbRembCode) { 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) { const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet(); rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb; rtcpPacketInformation.receiverEstimatedMaxBitrate = rtcpPacket.REMBItem.BitRate; } // 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); 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) { // Is it our sender that is requested to generate a new keyframe if (_SSRC != rtcpPacket.FIRItem.SSRC) { return; } // rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it // we don't know who this originate from if (receiveInfo) { // check if we have reported this FIRSequenceNumber before if (rtcpPacket.FIRItem.CommandSequenceNumber != receiveInfo->lastFIRSequenceNumber) { WebRtc_Word64 now = _clock.GetTimeInMS(); // 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; // received signal that we need to send a new key frame rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir; } } } else { // received signal that we need to send a new key frame rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir; } } 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(); } WebRtc_Word32 RTCPReceiver::UpdateTMMBR() { WebRtc_Word32 numBoundingSet = 0; WebRtc_UWord32 bitrate = 0; WebRtc_UWord32 accNumCandidates = 0; WebRtc_Word32 size = TMMBRReceived(0, 0, NULL); if (size > 0) { TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size); // Get candidate set from receiver. accNumCandidates = TMMBRReceived(size, accNumCandidates, candidateSet); } else { // Candidate set empty. VerifyAndAllocateCandidateSet(0); // resets candidate set } // Find bounding set TMMBRSet* boundingSet = NULL; numBoundingSet = FindTMMBRBoundingSet(boundingSet); if (numBoundingSet == -1) { WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Failed to find TMMBR bounding set."); return -1; } // Set bounding set // Inform remote clients about the new bandwidth // inform the remote client _rtpRtcp.SetTMMBN(boundingSet); // might trigger a TMMBN if (numBoundingSet == 0) { // owner of max bitrate request has timed out // empty bounding set has been sent return 0; } // Get net bitrate from bounding set depending on sent packet rate if (CalcMinBitRate(&bitrate)) { // we have a new bandwidth estimate on this channel CriticalSectionScoped lock(_criticalSectionFeedbacks); if (_cbRtcpBandwidthObserver) { _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate(bitrate * 1000); WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "Set TMMBR request:%d kbps", bitrate); } } return 0; } // 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. UpdateTMMBR(); } unsigned int local_ssrc = 0; { // We don't want to hold this critsect when triggering the callbacks below. CriticalSectionScoped lock(_criticalSectionRTCPReceiver); local_ssrc = _SSRC; } if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq) { _rtpRtcp.OnRequestSendReport(); } if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack) { if (rtcpPacketInformation.nackSequenceNumbersLength > 0) { WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming NACK length:%d", rtcpPacketInformation.nackSequenceNumbersLength); _rtpRtcp.OnReceivedNACK( rtcpPacketInformation.nackSequenceNumbersLength, rtcpPacketInformation.nackSequenceNumbers); } } { CriticalSectionScoped lock(_criticalSectionFeedbacks); // We need 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 (_cbRtcpIntraFrameObserver) { if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) || (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir)) { if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) { WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming PLI from SSRC:0x%x", rtcpPacketInformation.remoteSSRC); } else { WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming FIR from SSRC:0x%x", rtcpPacketInformation.remoteSSRC); } _cbRtcpIntraFrameObserver->OnReceivedIntraFrameRequest(local_ssrc); } if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSli) { _cbRtcpIntraFrameObserver->OnReceivedSLI( local_ssrc, rtcpPacketInformation.sliPictureId); } if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi) { _cbRtcpIntraFrameObserver->OnReceivedRPSI( local_ssrc, rtcpPacketInformation.rpsiPictureId); } } if (_cbRtcpBandwidthObserver) { if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb) { WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id, "SIG [RTCP] Incoming REMB:%d", rtcpPacketInformation.receiverEstimatedMaxBitrate); _cbRtcpBandwidthObserver->OnReceivedEstimatedBitrate( rtcpPacketInformation.receiverEstimatedMaxBitrate); } if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr || rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr) && rtcpPacketInformation.reportBlock) { WebRtc_Word64 now = _clock.GetTimeInMS(); _cbRtcpBandwidthObserver->OnReceivedRtcpReceiverReport( rtcpPacketInformation.remoteSSRC, rtcpPacketInformation.fractionLost, rtcpPacketInformation.roundTripTime, rtcpPacketInformation.lastReceivedExtendedHighSeqNum, now); } } if(_cbRtcpFeedback) { if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr) { _cbRtcpFeedback->OnSendReportReceived(_id, rtcpPacketInformation.remoteSSRC, rtcpPacketInformation.ntp_secs, rtcpPacketInformation.ntp_frac, rtcpPacketInformation.rtp_timestamp); } else { _cbRtcpFeedback->OnReceiveReportReceived(_id, rtcpPacketInformation.remoteSSRC); } if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric) { _cbRtcpFeedback->OnXRVoIPMetricReceived(_id, rtcpPacketInformation.VoIPMetric); } if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp) { _cbRtcpFeedback->OnApplicationDataReceived(_id, rtcpPacketInformation.applicationSubType, rtcpPacketInformation.applicationName, rtcpPacketInformation.applicationLength, rtcpPacketInformation.applicationData); } } } } WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC, char cName[RTCP_CNAME_SIZE]) const { assert(cName); CriticalSectionScoped lock(_criticalSectionRTCPReceiver); RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC); if (cnameInfo == NULL) { return -1; } 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::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_Word64 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