| /* |
| * 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_sender.h" |
| |
| #include <string.h> // memcpy |
| #include <cassert> // assert |
| #include <cstdlib> // rand |
| |
| #include "trace.h" |
| #include "common_types.h" |
| #include "critical_section_wrapper.h" |
| |
| #include "rtp_rtcp_impl.h" |
| |
| namespace webrtc { |
| |
| using RTCPUtility::RTCPCnameInformation; |
| |
| RTCPSender::RTCPSender(const WebRtc_Word32 id, |
| const bool audio, |
| RtpRtcpClock* clock, |
| ModuleRtpRtcpImpl* owner) : |
| _id(id), |
| _audio(audio), |
| _clock(*clock), |
| _method(kRtcpOff), |
| _rtpRtcp(*owner), |
| _criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()), |
| _cbTransport(NULL), |
| |
| _criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()), |
| _usingNack(false), |
| _sending(false), |
| _sendTMMBN(false), |
| _REMB(false), |
| _sendREMB(false), |
| _TMMBR(false), |
| _IJ(false), |
| _nextTimeToSendRTCP(0), |
| _SSRC(0), |
| _remoteSSRC(0), |
| _CNAME(), |
| _reportBlocks(), |
| _csrcCNAMEs(), |
| |
| _cameraDelayMS(0), |
| |
| _lastSendReport(), |
| _lastRTCPTime(), |
| |
| _CSRCs(0), |
| _CSRC(), |
| _includeCSRCs(true), |
| |
| _sequenceNumberFIR(0), |
| _lastTimeFIR(0), |
| |
| _lengthRembSSRC(0), |
| _sizeRembSSRC(0), |
| _rembSSRC(NULL), |
| _rembBitrate(0), |
| _bitrate_observer(NULL), |
| |
| _tmmbrHelp(audio), |
| _tmmbr_Send(0), |
| _packetOH_Send(0), |
| _remoteRateControl(), |
| |
| _appSend(false), |
| _appSubType(0), |
| _appName(), |
| _appData(NULL), |
| _appLength(0), |
| _xrSendVoIPMetric(false), |
| _xrVoIPMetric() |
| { |
| memset(_CNAME, 0, sizeof(_CNAME)); |
| memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__); |
| } |
| |
| RTCPSender::~RTCPSender() { |
| delete [] _rembSSRC; |
| delete [] _appData; |
| |
| while (!_reportBlocks.empty()) { |
| std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| _reportBlocks.begin(); |
| delete it->second; |
| _reportBlocks.erase(it); |
| } |
| while (!_csrcCNAMEs.empty()) { |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it = |
| _csrcCNAMEs.begin(); |
| delete it->second; |
| _csrcCNAMEs.erase(it); |
| } |
| delete _criticalSectionTransport; |
| delete _criticalSectionRTCPSender; |
| |
| WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__); |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::Init() |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| _method = kRtcpOff; |
| _cbTransport = NULL; |
| _usingNack = false; |
| _sending = false; |
| _sendTMMBN = false; |
| _TMMBR = false; |
| _IJ = false; |
| _REMB = false; |
| _sendREMB = false; |
| _SSRC = 0; |
| _remoteSSRC = 0; |
| _cameraDelayMS = 0; |
| _sequenceNumberFIR = 0; |
| _tmmbr_Send = 0; |
| _packetOH_Send = 0; |
| _remoteRateControl.Reset(); |
| _nextTimeToSendRTCP = 0; |
| _CSRCs = 0; |
| _appSend = false; |
| _appSubType = 0; |
| |
| if(_appData) |
| { |
| delete [] _appData; |
| _appData = NULL; |
| } |
| _appLength = 0; |
| |
| _xrSendVoIPMetric = false; |
| |
| memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric)); |
| memset(_CNAME, 0, sizeof(_CNAME)); |
| memset(_lastSendReport, 0, sizeof(_lastSendReport)); |
| memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime)); |
| return 0; |
| } |
| |
| void |
| RTCPSender::ChangeUniqueId(const WebRtc_Word32 id) |
| { |
| _id = id; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::RegisterSendTransport(Transport* outgoingTransport) |
| { |
| CriticalSectionScoped lock(_criticalSectionTransport); |
| _cbTransport = outgoingTransport; |
| return 0; |
| } |
| |
| RTCPMethod |
| RTCPSender::Status() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _method; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetRTCPStatus(const RTCPMethod method) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if(method != kRtcpOff) |
| { |
| if(_audio) |
| { |
| _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_AUDIO_MS/2); |
| } else |
| { |
| _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_VIDEO_MS/2); |
| } |
| } |
| _method = method; |
| return 0; |
| } |
| |
| bool |
| RTCPSender::Sending() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _sending; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetSendingStatus(const bool sending) |
| { |
| bool sendRTCPBye = false; |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if(_method != kRtcpOff) |
| { |
| if(sending == false && _sending == true) |
| { |
| // Trigger RTCP bye |
| sendRTCPBye = true; |
| } |
| } |
| _sending = sending; |
| } |
| if(sendRTCPBye) |
| { |
| return SendRTCP(kRtcpBye); |
| } |
| return 0; |
| } |
| |
| bool |
| RTCPSender::REMB() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _REMB; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetREMBStatus(const bool enable) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _REMB = enable; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetREMBData(const WebRtc_UWord32 bitrate, |
| const WebRtc_UWord8 numberOfSSRC, |
| const WebRtc_UWord32* SSRC) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _rembBitrate = bitrate; |
| |
| if(_sizeRembSSRC < numberOfSSRC) |
| { |
| delete [] _rembSSRC; |
| _rembSSRC = new WebRtc_UWord32[numberOfSSRC]; |
| _sizeRembSSRC = numberOfSSRC; |
| } |
| |
| _lengthRembSSRC = numberOfSSRC; |
| for (int i = 0; i < numberOfSSRC; i++) |
| { |
| _rembSSRC[i] = SSRC[i]; |
| } |
| _sendREMB = true; |
| return 0; |
| } |
| |
| bool RTCPSender::SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer) { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if (observer && _bitrate_observer) { |
| return false; |
| } |
| _bitrate_observer = observer; |
| return true; |
| } |
| |
| void RTCPSender::UpdateRemoteBitrateEstimate(unsigned int target_bitrate) { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if (_bitrate_observer) { |
| _bitrate_observer->OnReceiveBitrateChanged(_remoteSSRC, target_bitrate); |
| } |
| } |
| |
| bool |
| RTCPSender::TMMBR() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _TMMBR; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetTMMBRStatus(const bool enable) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _TMMBR = enable; |
| return 0; |
| } |
| |
| bool |
| RTCPSender::IJ() const |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _IJ; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetIJStatus(const bool enable) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _IJ = enable; |
| return 0; |
| } |
| |
| void |
| RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if(_SSRC != 0) |
| { |
| // not first SetSSRC, probably due to a collision |
| // schedule a new RTCP report |
| // make sure that we send a RTP packet |
| _nextTimeToSendRTCP = _clock.GetTimeInMS() + 100; |
| } |
| _SSRC = ssrc; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _remoteSSRC = ssrc; |
| _remoteRateControl.Reset(); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetCameraDelay(const WebRtc_Word32 delayMS) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if(delayMS > 1000 || delayMS < -1000) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__); |
| return -1; |
| } |
| _cameraDelayMS = delayMS; |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::CNAME(char cName[RTCP_CNAME_SIZE]) { |
| assert(cName); |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| cName[RTCP_CNAME_SIZE - 1] = 0; |
| strncpy(cName, _CNAME, RTCP_CNAME_SIZE - 1); |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) { |
| assert(cName); |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| _CNAME[RTCP_CNAME_SIZE - 1] = 0; |
| strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1); |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC, |
| const char cName[RTCP_CNAME_SIZE]) { |
| assert(cName); |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if (_csrcCNAMEs.size() >= kRtpCsrcSize) { |
| return -1; |
| } |
| RTCPCnameInformation* ptr = new RTCPCnameInformation(); |
| ptr->name[RTCP_CNAME_SIZE - 1] = 0; |
| strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1); |
| _csrcCNAMEs[SSRC] = ptr; |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it = |
| _csrcCNAMEs.find(SSRC); |
| |
| if (it == _csrcCNAMEs.end()) { |
| return -1; |
| } |
| delete it->second; |
| _csrcCNAMEs.erase(it); |
| return 0; |
| } |
| |
| bool |
| RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const |
| { |
| /* |
| For audio we use a fix 5 sec interval |
| |
| For video we use 1 sec interval fo a BW smaller than 360 kbit/s, |
| technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare |
| |
| |
| From RFC 3550 |
| |
| MAX RTCP BW is 5% if the session BW |
| A send report is approximately 65 bytes inc CNAME |
| A report report is approximately 28 bytes |
| |
| The RECOMMENDED value for the reduced minimum in seconds is 360 |
| divided by the session bandwidth in kilobits/second. This minimum |
| is smaller than 5 seconds for bandwidths greater than 72 kb/s. |
| |
| If the participant has not yet sent an RTCP packet (the variable |
| initial is true), the constant Tmin is set to 2.5 seconds, else it |
| is set to 5 seconds. |
| |
| The interval between RTCP packets is varied randomly over the |
| range [0.5,1.5] times the calculated interval to avoid unintended |
| synchronization of all participants |
| |
| if we send |
| If the participant is a sender (we_sent true), the constant C is |
| set to the average RTCP packet size (avg_rtcp_size) divided by 25% |
| of the RTCP bandwidth (rtcp_bw), and the constant n is set to the |
| number of senders. |
| |
| if we receive only |
| If we_sent is not true, the constant C is set |
| to the average RTCP packet size divided by 75% of the RTCP |
| bandwidth. The constant n is set to the number of receivers |
| (members - senders). If the number of senders is greater than |
| 25%, senders and receivers are treated together. |
| |
| reconsideration NOT required for peer-to-peer |
| "timer reconsideration" is |
| employed. This algorithm implements a simple back-off mechanism |
| which causes users to hold back RTCP packet transmission if the |
| group sizes are increasing. |
| |
| n = number of members |
| C = avg_size/(rtcpBW/4) |
| |
| 3. The deterministic calculated interval Td is set to max(Tmin, n*C). |
| |
| 4. The calculated interval T is set to a number uniformly distributed |
| between 0.5 and 1.5 times the deterministic calculated interval. |
| |
| 5. The resulting value of T is divided by e-3/2=1.21828 to compensate |
| for the fact that the timer reconsideration algorithm converges to |
| a value of the RTCP bandwidth below the intended average |
| */ |
| |
| WebRtc_UWord32 now = _clock.GetTimeInMS(); |
| |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if(_method == kRtcpOff) |
| { |
| return false; |
| } |
| |
| if(!_audio && sendKeyframeBeforeRTP) |
| { |
| // for video key-frames we want to send the RTCP before the large key-frame |
| // if we have a 100 ms margin |
| now += RTCP_SEND_BEFORE_KEY_FRAME_MS; |
| } |
| |
| if(now > _nextTimeToSendRTCP) |
| { |
| return true; |
| |
| } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin |
| { |
| // wrap |
| return true; |
| } |
| return false; |
| } |
| |
| WebRtc_UWord32 |
| RTCPSender::LastSendReport( WebRtc_UWord32& lastRTCPTime) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| lastRTCPTime = _lastRTCPTime[0]; |
| return _lastSendReport[0]; |
| } |
| |
| WebRtc_UWord32 |
| RTCPSender::SendTimeOfSendReport(const WebRtc_UWord32 sendReport) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| // This is only saved when we are the sender |
| if((_lastSendReport[0] == 0) || (sendReport == 0)) |
| { |
| return 0; // will be ignored |
| } else |
| { |
| for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i) |
| { |
| if( _lastSendReport[i] == sendReport) |
| { |
| return _lastRTCPTime[i]; |
| } |
| } |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::AddReportBlock(const WebRtc_UWord32 SSRC, |
| const RTCPReportBlock* reportBlock) { |
| if (reportBlock == NULL) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if (_reportBlocks.size() >= RTCP_MAX_REPORT_BLOCKS) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| RTCPReportBlock* copyReportBlock = new RTCPReportBlock(); |
| memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock)); |
| _reportBlocks[SSRC] = copyReportBlock; |
| return 0; |
| } |
| |
| WebRtc_Word32 RTCPSender::RemoveReportBlock(const WebRtc_UWord32 SSRC) { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| _reportBlocks.find(SSRC); |
| |
| if (it == _reportBlocks.end()) { |
| return -1; |
| } |
| delete it->second; |
| _reportBlocks.erase(it); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| const WebRtc_UWord32 NTPsec, |
| const WebRtc_UWord32 NTPfrac, |
| const RTCPReportBlock* received) |
| { |
| // sanity |
| if(pos + 52 >= IP_PACKET_SIZE) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -2; |
| } |
| WebRtc_UWord32 RTPtime; |
| WebRtc_UWord32 BackTimedNTPsec; |
| WebRtc_UWord32 BackTimedNTPfrac; |
| |
| WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| |
| // Sender report |
| rtcpbuffer[pos++]=(WebRtc_UWord8)200; |
| |
| for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--) |
| { |
| // shift old |
| _lastSendReport[i+1] = _lastSendReport[i]; |
| _lastRTCPTime[i+1] =_lastRTCPTime[i]; |
| } |
| |
| _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); // before video cam compensation |
| |
| if(_cameraDelayMS >= 0) |
| { |
| // fraction of a second as an unsigned word32 4.294 967 296E9 |
| WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)_cameraDelayMS* 4294967; // note camera delay can't be larger than +/-1000ms |
| if(NTPfrac > cameraDelayFixFrac) |
| { |
| // no problem just reduce the fraction part |
| BackTimedNTPfrac = NTPfrac - cameraDelayFixFrac; |
| BackTimedNTPsec = NTPsec; |
| } else |
| { |
| // we need to reduce the sec and add that sec to the frac |
| BackTimedNTPsec = NTPsec - 1; |
| BackTimedNTPfrac = 0xffffffff - (cameraDelayFixFrac - NTPfrac); |
| } |
| } else |
| { |
| // fraction of a second as an unsigned word32 4.294 967 296E9 |
| WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)(-_cameraDelayMS)* 4294967; // note camera delay can't be larger than +/-1000ms |
| if(NTPfrac > 0xffffffff - cameraDelayFixFrac) |
| { |
| // we need to add the sec and add that sec to the frac |
| BackTimedNTPsec = NTPsec + 1; |
| BackTimedNTPfrac = cameraDelayFixFrac + NTPfrac; // this will wrap but that is ok |
| } else |
| { |
| // no problem just add the fraction part |
| BackTimedNTPsec = NTPsec; |
| BackTimedNTPfrac = NTPfrac + cameraDelayFixFrac; |
| } |
| } |
| _lastSendReport[0] = (BackTimedNTPsec <<16) + (BackTimedNTPfrac >> 16); |
| |
| // RTP timestamp |
| // This should have a ramdom start value added |
| // RTP is counted from NTP not the acctual RTP |
| // This reflects the perfect RTP time |
| // we solve this by initiating RTP to our NTP :) |
| |
| WebRtc_UWord32 freqHz = 90000; // For video |
| if(_audio) |
| { |
| freqHz = _rtpRtcp.CurrentSendFrequencyHz(); |
| RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freqHz); |
| } |
| else // video |
| { |
| // used to be (WebRtc_UWord32)(((float)BackTimedNTPfrac/(float)FRAC)* 90000) |
| WebRtc_UWord32 tmp = 9*(BackTimedNTPfrac/429496); |
| RTPtime = BackTimedNTPsec*freqHz + tmp; |
| } |
| |
| |
| |
| |
| // Add sender data |
| // Save for our length field |
| pos++; |
| pos++; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| // NTP |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPsec); |
| pos += 4; |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPfrac); |
| pos += 4; |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime); |
| pos += 4; |
| |
| //sender's packet count |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.PacketCountSent()); |
| pos += 4; |
| |
| //sender's octet count |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.ByteCountSent()); |
| pos += 4; |
| |
| WebRtc_UWord8 numberOfReportBlocks = 0; |
| WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| if(retVal < 0) |
| { |
| // |
| return retVal ; |
| } |
| rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| |
| WebRtc_UWord16 len = WebRtc_UWord16((pos/4) -1); |
| ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| return 0; |
| } |
| |
| |
| WebRtc_Word32 RTCPSender::BuildSDEC(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos) { |
| size_t lengthCname = strlen(_CNAME); |
| assert(lengthCname < RTCP_CNAME_SIZE); |
| |
| // sanity |
| if(pos + 12 + lengthCname >= IP_PACKET_SIZE) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -2; |
| } |
| // SDEC Source Description |
| |
| // We always need to add SDES CNAME |
| rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(0x80 + 1 + _csrcCNAMEs.size()); |
| rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(202); |
| |
| // handle SDES length later on |
| WebRtc_UWord32 SDESLengthPos = pos; |
| pos++; |
| pos++; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // CNAME = 1 |
| rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1); |
| |
| // |
| rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(lengthCname); |
| |
| WebRtc_UWord16 SDESLength = 10; |
| |
| memcpy(&rtcpbuffer[pos], _CNAME, lengthCname); |
| pos += lengthCname; |
| SDESLength += (WebRtc_UWord16)lengthCname; |
| |
| WebRtc_UWord16 padding = 0; |
| // We must have a zero field even if we have an even multiple of 4 bytes |
| if ((pos % 4) == 0) { |
| padding++; |
| rtcpbuffer[pos++]=0; |
| } |
| while ((pos % 4) != 0) { |
| padding++; |
| rtcpbuffer[pos++]=0; |
| } |
| SDESLength += padding; |
| |
| std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>::iterator it = |
| _csrcCNAMEs.begin(); |
| |
| for(; it != _csrcCNAMEs.end(); it++) { |
| RTCPCnameInformation* cname = it->second; |
| WebRtc_UWord32 SSRC = it->first; |
| |
| // Add SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC); |
| pos += 4; |
| |
| // CNAME = 1 |
| rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1); |
| |
| size_t length = strlen(cname->name); |
| assert(length < RTCP_CNAME_SIZE); |
| |
| rtcpbuffer[pos++]= static_cast<WebRtc_UWord8>(length); |
| SDESLength += 6; |
| |
| memcpy(&rtcpbuffer[pos],cname->name, length); |
| |
| pos += length; |
| SDESLength += length; |
| WebRtc_UWord16 padding = 0; |
| |
| // We must have a zero field even if we have an even multiple of 4 bytes |
| if((pos % 4) == 0){ |
| padding++; |
| rtcpbuffer[pos++]=0; |
| } |
| while((pos % 4) != 0){ |
| padding++; |
| rtcpbuffer[pos++] = 0; |
| } |
| SDESLength += padding; |
| } |
| // in 32-bit words minus one and we don't count the header |
| WebRtc_UWord16 buffer_length = (SDESLength / 4) - 1; |
| ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos, |
| buffer_length); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildRR(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| const WebRtc_UWord32 NTPsec, |
| const WebRtc_UWord32 NTPfrac, |
| const RTCPReportBlock* received) |
| { |
| // sanity one block |
| if(pos + 32 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| WebRtc_UWord32 posNumberOfReportBlocks = pos; |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)201; |
| |
| // Save for our length field |
| pos++; |
| pos++; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| WebRtc_UWord8 numberOfReportBlocks = 0; |
| WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac); |
| if(retVal < 0) |
| { |
| return retVal; |
| } |
| rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks; |
| |
| WebRtc_UWord16 len = WebRtc_UWord16((pos)/4 -1); |
| ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len); |
| return 0; |
| } |
| |
| // From RFC 5450: Transmission Time Offsets in RTP Streams. |
| // 0 1 2 3 |
| // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // hdr |V=2|P| RC | PT=IJ=195 | length | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // | inter-arrival jitter | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // . . |
| // . . |
| // . . |
| // | inter-arrival jitter | |
| // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| // |
| // If present, this RTCP packet must be placed after a receiver report |
| // (inside a compound RTCP packet), and MUST have the same value for RC |
| // (reception report count) as the receiver report. |
| |
| WebRtc_Word32 |
| RTCPSender::BuildExtendedJitterReport( |
| WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| const WebRtc_UWord32 jitterTransmissionTimeOffset) |
| { |
| if (_reportBlocks.size() > 0) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented."); |
| return 0; |
| } |
| |
| // sanity |
| if(pos + 8 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add picture loss indicator |
| WebRtc_UWord8 RC = 1; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + RC; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)195; |
| |
| // Used fixed length of 2 |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(1); |
| |
| // Add inter-arrival jitter |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, |
| jitterTransmissionTimeOffset); |
| pos += 4; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // sanity |
| if(pos + 12 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add picture loss indicator |
| WebRtc_UWord8 FMT = 1; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| |
| //Used fixed length of 2 |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(2); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add the remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildFIR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord32 RTT) |
| { |
| bool firRepeat = false; |
| WebRtc_UWord32 diff = _clock.GetTimeInMS() - _lastTimeFIR; |
| if(diff < RTT + 3) // 3 is processing jitter |
| { |
| // we have recently sent a FIR |
| // don't send another |
| return 0; |
| |
| } else |
| { |
| if(diff < (RTT*2 + RTCP_MIN_FRAME_LENGTH_MS)) |
| { |
| // send a FIR_REPEAT instead of a FIR |
| firRepeat = true; |
| } |
| } |
| _lastTimeFIR = _clock.GetTimeInMS(); |
| if(!firRepeat) |
| { |
| _sequenceNumberFIR++; // do not increase if repetition |
| } |
| |
| // sanity |
| if(pos + 20 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| |
| // add full intra request indicator |
| WebRtc_UWord8 FMT = 4; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| |
| //Length of 4 |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(4); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // RFC 5104 4.3.1.2. Semantics |
| |
| // SSRC of media source |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| |
| // Additional Feedback Control Information (FCI) |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(_sequenceNumberFIR); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| return 0; |
| } |
| |
| /* |
| 0 1 2 3 |
| 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | First | Number | PictureID | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| */ |
| WebRtc_Word32 |
| RTCPSender::BuildSLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord8 pictureID) |
| { |
| // sanity |
| if(pos + 16 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add slice loss indicator |
| WebRtc_UWord8 FMT = 2; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| |
| //Used fixed length of 3 |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(3); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add the remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| // Add first, number & picture ID 6 bits |
| // first = 0, 13 - bits |
| // number = 0x1fff, 13 - bits only ones for now |
| WebRtc_UWord32 sliField = (0x1fff << 6)+ (0x3f & pictureID); |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField); |
| pos += 4; |
| return 0; |
| } |
| |
| /* |
| 0 1 2 3 |
| 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | PB |0| Payload Type| Native RPSI bit string | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| | defined per codec ... | Padding (0) | |
| +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
| */ |
| /* |
| * Note: not generic made for VP8 |
| */ |
| WebRtc_Word32 |
| RTCPSender::BuildRPSI(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| const WebRtc_UWord64 pictureID, |
| const WebRtc_UWord8 payloadType) |
| { |
| // sanity |
| if(pos + 24 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add Reference Picture Selection Indication |
| WebRtc_UWord8 FMT = 3; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| |
| // calc length |
| WebRtc_UWord32 bitsRequired = 7; |
| WebRtc_UWord8 bytesRequired = 1; |
| while((pictureID>>bitsRequired) > 0) |
| { |
| bitsRequired += 7; |
| bytesRequired++; |
| } |
| |
| WebRtc_UWord8 size = 3; |
| if(bytesRequired > 6) |
| { |
| size = 5; |
| } else if(bytesRequired > 2) |
| { |
| size = 4; |
| } |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=size; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add the remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| // calc padding length |
| WebRtc_UWord8 paddingBytes = 4-((2+bytesRequired)%4); |
| if(paddingBytes == 4) |
| { |
| paddingBytes = 0; |
| } |
| // add padding length in bits |
| rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24 |
| pos++; |
| |
| // add payload type |
| rtcpbuffer[pos] = payloadType; |
| pos++; |
| |
| // add picture ID |
| for(int i = bytesRequired-1; i > 0; i--) |
| { |
| rtcpbuffer[pos] = 0x80 | WebRtc_UWord8(pictureID >> (i*7)); |
| pos++; |
| } |
| // add last byte of picture ID |
| rtcpbuffer[pos] = WebRtc_UWord8(pictureID & 0x7f); |
| pos++; |
| |
| // add padding |
| for(int j = 0; j <paddingBytes; j++) |
| { |
| rtcpbuffer[pos] = 0; |
| pos++; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // sanity |
| if(pos + 20 + 4 * _lengthRembSSRC >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add application layer feedback |
| WebRtc_UWord8 FMT = 15; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)206; |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=_lengthRembSSRC + 4; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Remote SSRC must be 0 |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0); |
| pos += 4; |
| |
| rtcpbuffer[pos++]='R'; |
| rtcpbuffer[pos++]='E'; |
| rtcpbuffer[pos++]='M'; |
| rtcpbuffer[pos++]='B'; |
| |
| rtcpbuffer[pos++] = _lengthRembSSRC; |
| // 6 bit Exp |
| // 18 bit mantissa |
| WebRtc_UWord8 brExp = 0; |
| for(WebRtc_UWord32 i=0; i<64; i++) |
| { |
| if(_rembBitrate <= ((WebRtc_UWord32)262143 << i)) |
| { |
| brExp = i; |
| break; |
| } |
| } |
| const WebRtc_UWord32 brMantissa = (_rembBitrate >> brExp); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)((brExp << 2) + ((brMantissa >> 16) & 0x03)); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa >> 8); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa); |
| |
| for (int i = 0; i < _lengthRembSSRC; i++) |
| { |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]); |
| pos += 4; |
| } |
| return 0; |
| } |
| |
| WebRtc_UWord32 |
| RTCPSender::CalculateNewTargetBitrate(WebRtc_UWord32 RTT) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| WebRtc_UWord32 target_bitrate = |
| _remoteRateControl.TargetBitRate(RTT, _clock.GetTimeInMS()); |
| _tmmbr_Send = target_bitrate / 1000; |
| return target_bitrate; |
| } |
| |
| bool |
| RTCPSender::ValidBitrateEstimate() { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _remoteRateControl.ValidEstimate(); |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate |
| // If the sender is an owner of the TMMBN -> send TMMBR |
| // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR |
| |
| // get current bounding set from RTCP receiver |
| bool tmmbrOwner = false; |
| TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet(); // store in candidateSet, allocates one extra slot |
| |
| // holding _criticalSectionRTCPSender while calling RTCPreceiver which will accuire _criticalSectionRTCPReceiver |
| // is a potental deadlock but since RTCPreceiver is not doing the revese we should be fine |
| WebRtc_Word32 lengthOfBoundingSet = _rtpRtcp.BoundingSet(tmmbrOwner, candidateSet); |
| |
| if(lengthOfBoundingSet > 0) |
| { |
| for (WebRtc_Word32 i = 0; i < lengthOfBoundingSet; i++) |
| { |
| if( candidateSet->ptrTmmbrSet[i] == _tmmbr_Send && |
| candidateSet->ptrPacketOHSet[i] == _packetOH_Send) |
| { |
| // do not send the same tuple |
| return 0; |
| } |
| } |
| if(!tmmbrOwner) |
| { |
| // use received bounding set as candidate set |
| // add current tuple |
| candidateSet->ptrTmmbrSet[lengthOfBoundingSet] = _tmmbr_Send; |
| candidateSet->ptrPacketOHSet[lengthOfBoundingSet] = _packetOH_Send; |
| candidateSet->ptrSsrcSet[lengthOfBoundingSet] = _SSRC; |
| int numCandidates = lengthOfBoundingSet+ 1; |
| |
| // find bounding set |
| TMMBRSet* boundingSet = NULL; |
| int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet); |
| if(numBoundingSet > 0 || numBoundingSet <= numCandidates) |
| { |
| tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet); |
| } |
| if(!tmmbrOwner) |
| { |
| // did not enter bounding set, no meaning to send this request |
| return 0; |
| } |
| } |
| } |
| |
| if(_tmmbr_Send) |
| { |
| // sanity |
| if(pos + 20 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| // add TMMBR indicator |
| WebRtc_UWord8 FMT = 3; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| |
| //Length of 4 |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(4); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // RFC 5104 4.2.1.2. Semantics |
| |
| // SSRC of media source |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| |
| // Additional Feedback Control Information (FCI) |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| WebRtc_UWord32 bitRate = _tmmbr_Send*1000; |
| WebRtc_UWord32 mmbrExp = 0; |
| for(WebRtc_UWord32 i=0;i<64;i++) |
| { |
| if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| { |
| mmbrExp = i; |
| break; |
| } |
| } |
| WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01)); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(_packetOH_Send); |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend(); |
| if(boundingSet == NULL) |
| { |
| return -1; |
| } |
| // sanity |
| if(pos + 12 + boundingSet->lengthOfSet*8 >= IP_PACKET_SIZE) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -2; |
| } |
| WebRtc_UWord8 FMT = 4; |
| // add TMMBN indicator |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| |
| //Add length later |
| int posLength = pos; |
| pos++; |
| pos++; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // RFC 5104 4.2.2.2. Semantics |
| |
| // SSRC of media source |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| |
| // Additional Feedback Control Information (FCI) |
| int numBoundingSet = 0; |
| for(WebRtc_UWord32 n=0; n< boundingSet->lengthOfSet; n++) |
| { |
| if (boundingSet->ptrTmmbrSet[n] > 0) |
| { |
| WebRtc_UWord32 tmmbrSSRC = boundingSet->ptrSsrcSet[n]; |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC); |
| pos += 4; |
| |
| WebRtc_UWord32 bitRate = boundingSet->ptrTmmbrSet[n] * 1000; |
| WebRtc_UWord32 mmbrExp = 0; |
| for(int i=0; i<64; i++) |
| { |
| if(bitRate <= ((WebRtc_UWord32)131071 << i)) |
| { |
| mmbrExp = i; |
| break; |
| } |
| } |
| WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp); |
| WebRtc_UWord32 measuredOH = boundingSet->ptrPacketOHSet[n]; |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03)); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01)); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(measuredOH); |
| numBoundingSet++; |
| } |
| } |
| WebRtc_UWord16 length= (WebRtc_UWord16)(2+2*numBoundingSet); |
| rtcpbuffer[posLength++]=(WebRtc_UWord8)(length>>8); |
| rtcpbuffer[posLength]=(WebRtc_UWord8)(length); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // sanity |
| if(_appData == NULL) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__); |
| return -1; |
| } |
| if(pos + 12 + _appLength >= IP_PACKET_SIZE) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -2; |
| } |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + _appSubType; |
| |
| // Add APP ID |
| rtcpbuffer[pos++]=(WebRtc_UWord8)204; |
| |
| WebRtc_UWord16 length = (_appLength>>2) + 2; // include SSRC and name |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(length>>8); |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(length); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add our application name |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName); |
| pos += 4; |
| |
| // Add the data |
| memcpy(rtcpbuffer +pos, _appData,_appLength); |
| pos += _appLength; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| const WebRtc_Word32 nackSize, |
| const WebRtc_UWord16* nackList) |
| { |
| // sanity |
| if(pos + 16 >= IP_PACKET_SIZE) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -2; |
| } |
| |
| // int size, WebRtc_UWord16* nackList |
| // add nack list |
| WebRtc_UWord8 FMT = 1; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)205; |
| |
| rtcpbuffer[pos++]=(WebRtc_UWord8) 0; |
| int nackSizePos = pos; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(3); //setting it to one kNACK signal as default |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add the remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| // add the list |
| int i = 0; |
| int numOfNackFields = 0; |
| while(nackSize > i && numOfNackFields < 253) |
| { |
| WebRtc_UWord16 nack = nackList[i]; |
| // put dow our sequence number |
| ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack); |
| pos += 2; |
| |
| i++; |
| numOfNackFields++; |
| if(nackSize > i) |
| { |
| bool moreThan16Away = (WebRtc_UWord16(nack+16) < nackList[i])?true: false; |
| if(!moreThan16Away) |
| { |
| // check for a wrap |
| if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| { |
| // wrap |
| moreThan16Away = true; |
| } |
| } |
| if(moreThan16Away) |
| { |
| // next is more than 16 away |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| } else |
| { |
| // build our bitmask |
| WebRtc_UWord16 bitmask = 0; |
| |
| bool within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| if(within16Away) |
| { |
| // check for a wrap |
| if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| { |
| // wrap |
| within16Away = false; |
| } |
| } |
| |
| while( nackSize > i && within16Away) |
| { |
| WebRtc_Word16 shift = (nackList[i]-nack)-1; |
| assert(!(shift > 15) && !(shift < 0)); |
| |
| bitmask += (1<< shift); |
| i++; |
| if(nackSize > i) |
| { |
| within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false; |
| if(within16Away) |
| { |
| // check for a wrap |
| if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff) |
| { |
| // wrap |
| within16Away = false; |
| } |
| } |
| } |
| } |
| ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, bitmask); |
| pos += 2; |
| } |
| // sanity do we have room from one more 4 byte block? |
| if(pos + 4 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| } else |
| { |
| // no more in the list |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| } |
| } |
| rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // sanity |
| if(pos + 8 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| if(_includeCSRCs) |
| { |
| // Add a bye packet |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs |
| rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| |
| // length |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)(1 + _CSRCs); |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // add CSRCs |
| for(int i = 0; i < _CSRCs; i++) |
| { |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]); |
| pos += 4; |
| } |
| } else |
| { |
| // Add a bye packet |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1; // number of SSRC+CSRCs |
| rtcpbuffer[pos++]=(WebRtc_UWord8)203; |
| |
| // length |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)1; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| } |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos) |
| { |
| // sanity |
| if(pos + 44 >= IP_PACKET_SIZE) |
| { |
| return -2; |
| } |
| |
| // Add XR header |
| rtcpbuffer[pos++]=(WebRtc_UWord8)0x80; |
| rtcpbuffer[pos++]=(WebRtc_UWord8)207; |
| |
| WebRtc_UWord32 XRLengthPos = pos; |
| |
| // handle length later on |
| pos++; |
| pos++; |
| |
| // Add our own SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC); |
| pos += 4; |
| |
| // Add a VoIP metrics block |
| rtcpbuffer[pos++]=7; |
| rtcpbuffer[pos++]=0; |
| rtcpbuffer[pos++]=0; |
| rtcpbuffer[pos++]=8; |
| |
| // Add the remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| rtcpbuffer[pos++] = _xrVoIPMetric.lossRate; |
| rtcpbuffer[pos++] = _xrVoIPMetric.discardRate; |
| rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity; |
| rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity; |
| |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration); |
| |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay); |
| |
| rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel; |
| rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel; |
| rtcpbuffer[pos++] = _xrVoIPMetric.RERL; |
| rtcpbuffer[pos++] = _xrVoIPMetric.Gmin; |
| |
| rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor; |
| rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor; |
| rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ; |
| rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ; |
| |
| rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig; |
| rtcpbuffer[pos++] = 0; // reserved |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal); |
| |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax >> 8); |
| rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax); |
| |
| rtcpbuffer[XRLengthPos]=(WebRtc_UWord8)(0); |
| rtcpbuffer[XRLengthPos+1]=(WebRtc_UWord8)(10); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags, |
| const WebRtc_Word32 nackSize, // NACK |
| const WebRtc_UWord16* nackList, // NACK |
| const WebRtc_UWord32 RTT, // FIR |
| const WebRtc_UWord64 pictureID) // SLI & RPSI |
| { |
| WebRtc_UWord32 rtcpPacketTypeFlags = packetTypeFlags; |
| WebRtc_UWord32 pos = 0; |
| WebRtc_UWord8 rtcpbuffer[IP_PACKET_SIZE]; |
| |
| do // only to be able to use break :) (and the critsect must be inside its own scope) |
| { |
| // collect the received information |
| RTCPReportBlock received; |
| bool hasReceived = false; |
| WebRtc_UWord32 NTPsec = 0; |
| WebRtc_UWord32 NTPfrac = 0; |
| bool rtcpCompound = false; |
| WebRtc_UWord32 jitterTransmissionOffset = 0; |
| |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if(_method == kRtcpOff) |
| { |
| WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, |
| "%s invalid state", __FUNCTION__); |
| return -1; |
| } |
| rtcpCompound = (_method == kRtcpCompound) ? true : false; |
| } |
| |
| if (rtcpCompound || |
| rtcpPacketTypeFlags & kRtcpReport || |
| rtcpPacketTypeFlags & kRtcpSr || |
| rtcpPacketTypeFlags & kRtcpRr) |
| { |
| // get statistics from our RTPreceiver outside critsect |
| if(_rtpRtcp.ReportBlockStatistics(&received.fractionLost, |
| &received.cumulativeLost, |
| &received.extendedHighSeqNum, |
| &received.jitter, |
| &jitterTransmissionOffset) == 0) |
| { |
| hasReceived = true; |
| |
| WebRtc_UWord32 lastReceivedRRNTPsecs = 0; |
| WebRtc_UWord32 lastReceivedRRNTPfrac = 0; |
| WebRtc_UWord32 remoteSR = 0; |
| |
| // ok even if we have not received a SR, we will send 0 in that case |
| _rtpRtcp.LastReceivedNTP(lastReceivedRRNTPsecs, |
| lastReceivedRRNTPfrac, |
| remoteSR); |
| |
| // get our NTP as late as possible to avoid a race |
| _clock.CurrentNTP(NTPsec, NTPfrac); |
| |
| // Delay since last received report |
| WebRtc_UWord32 delaySinceLastReceivedSR = 0; |
| if((lastReceivedRRNTPsecs !=0) || (lastReceivedRRNTPfrac !=0)) |
| { |
| // get the 16 lowest bits of seconds and the 16 higest bits of fractions |
| WebRtc_UWord32 now=NTPsec&0x0000FFFF; |
| now <<=16; |
| now += (NTPfrac&0xffff0000)>>16; |
| |
| WebRtc_UWord32 receiveTime = lastReceivedRRNTPsecs&0x0000FFFF; |
| receiveTime <<=16; |
| receiveTime += (lastReceivedRRNTPfrac&0xffff0000)>>16; |
| |
| delaySinceLastReceivedSR = now-receiveTime; |
| } |
| received.delaySinceLastSR = delaySinceLastReceivedSR; |
| received.lastSR = remoteSR; |
| } else |
| { |
| // we need to send our NTP even if we dont have received any reports |
| _clock.CurrentNTP(NTPsec, NTPfrac); |
| } |
| } |
| |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if(_TMMBR ) // attach TMMBR to send and receive reports |
| { |
| rtcpPacketTypeFlags |= kRtcpTmmbr; |
| } |
| if(_appSend) |
| { |
| rtcpPacketTypeFlags |= kRtcpApp; |
| _appSend = false; |
| } |
| if(_REMB && _sendREMB) |
| { |
| // Always attach REMB to SR if that is configured. Note that REMB is |
| // only sent on one of the RTP modules in the REMB group. |
| rtcpPacketTypeFlags |= kRtcpRemb; |
| } |
| if(_xrSendVoIPMetric) |
| { |
| rtcpPacketTypeFlags |= kRtcpXrVoipMetric; |
| _xrSendVoIPMetric = false; |
| } |
| if(_sendTMMBN) // set when having received a TMMBR |
| { |
| rtcpPacketTypeFlags |= kRtcpTmmbn; |
| _sendTMMBN = false; |
| } |
| |
| if(_method == kRtcpCompound) |
| { |
| if(_sending) |
| { |
| rtcpPacketTypeFlags |= kRtcpSr; |
| } else |
| { |
| rtcpPacketTypeFlags |= kRtcpRr; |
| } |
| if (_IJ && hasReceived) |
| { |
| rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset; |
| } |
| } else if(_method == kRtcpNonCompound) |
| { |
| if(rtcpPacketTypeFlags & kRtcpReport) |
| { |
| if(_sending) |
| { |
| rtcpPacketTypeFlags |= kRtcpSr; |
| } else |
| { |
| rtcpPacketTypeFlags |= kRtcpRr; |
| } |
| } |
| } |
| if( rtcpPacketTypeFlags & kRtcpRr || |
| rtcpPacketTypeFlags & kRtcpSr) |
| { |
| // generate next time to send a RTCP report |
| // seeded from RTP constructor |
| WebRtc_Word32 random = rand() % 1000; |
| WebRtc_Word32 timeToNext = RTCP_INTERVAL_AUDIO_MS; |
| |
| if(_audio) |
| { |
| timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) + (RTCP_INTERVAL_AUDIO_MS*random/1000); |
| }else |
| { |
| WebRtc_UWord32 minIntervalMs = RTCP_INTERVAL_AUDIO_MS; |
| if(_sending) |
| { |
| // calc bw for video 360/sendBW in kbit/s |
| WebRtc_UWord32 sendBitrateKbit = 0; |
| WebRtc_UWord32 videoRate = 0; |
| WebRtc_UWord32 fecRate = 0; |
| WebRtc_UWord32 nackRate = 0; |
| _rtpRtcp.BitrateSent(&sendBitrateKbit, |
| &videoRate, |
| &fecRate, |
| &nackRate); |
| sendBitrateKbit /= 1000; |
| if(sendBitrateKbit != 0) |
| { |
| minIntervalMs = 360000/sendBitrateKbit; |
| } |
| } |
| if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS) |
| { |
| minIntervalMs = RTCP_INTERVAL_VIDEO_MS; |
| } |
| timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000); |
| } |
| _nextTimeToSendRTCP = _clock.GetTimeInMS() + timeToNext; |
| } |
| |
| // if the data does not fitt in the packet we fill it as much as possible |
| WebRtc_Word32 buildVal = 0; |
| |
| if(rtcpPacketTypeFlags & kRtcpSr) |
| { |
| if(hasReceived) |
| { |
| buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac, &received); |
| } else |
| { |
| buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| } |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| buildVal = BuildSDEC(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| |
| }else if(rtcpPacketTypeFlags & kRtcpRr) |
| { |
| if(hasReceived) |
| { |
| buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac,&received); |
| }else |
| { |
| buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac); |
| } |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| // only of set |
| if(_CNAME[0] != 0) |
| { |
| buildVal = BuildSDEC(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| } |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset) |
| { |
| // If present, this RTCP packet must be placed after a |
| // receiver report. |
| buildVal = BuildExtendedJitterReport(rtcpbuffer, |
| pos, |
| jitterTransmissionOffset); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| } |
| else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpPli) |
| { |
| buildVal = BuildPLI(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpFir) |
| { |
| buildVal = BuildFIR(rtcpbuffer, pos, RTT); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpSli) |
| { |
| buildVal = BuildSLI(rtcpbuffer, pos, (WebRtc_UWord8)pictureID); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpRpsi) |
| { |
| const WebRtc_Word8 payloadType = _rtpRtcp.SendPayloadType(); |
| if(payloadType == -1) |
| { |
| return -1; |
| } |
| buildVal = BuildRPSI(rtcpbuffer, pos, pictureID, (WebRtc_UWord8)payloadType); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpRemb) |
| { |
| buildVal = BuildREMB(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpBye) |
| { |
| buildVal = BuildBYE(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpApp) |
| { |
| buildVal = BuildAPP(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpTmmbr) |
| { |
| buildVal = BuildTMMBR(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpTmmbn) |
| { |
| buildVal = BuildTMMBN(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpNack) |
| { |
| buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| if(rtcpPacketTypeFlags & kRtcpXrVoipMetric) |
| { |
| buildVal = BuildVoIPMetric(rtcpbuffer, pos); |
| if(buildVal == -1) |
| { |
| return -1; // error |
| |
| }else if(buildVal == -2) |
| { |
| break; // out of buffer |
| } |
| } |
| }while (false); |
| // Sanity don't send empty packets. |
| if (pos == 0) |
| { |
| return -1; |
| } |
| return SendToNetwork(rtcpbuffer, (WebRtc_UWord16)pos); |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SendToNetwork(const WebRtc_UWord8* dataBuffer, |
| const WebRtc_UWord16 length) |
| { |
| CriticalSectionScoped lock(_criticalSectionTransport); |
| if(_cbTransport) |
| { |
| if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0) |
| { |
| return 0; |
| } |
| } |
| return -1; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetCSRCStatus(const bool include) |
| { |
| _includeCSRCs = include; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], |
| const WebRtc_UWord8 arrLength) |
| { |
| if(arrLength > kRtpCsrcSize) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| assert(false); |
| return -1; |
| } |
| |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| for(int i = 0; i < arrLength;i++) |
| { |
| _CSRC[i] = arrOfCSRC[i]; |
| } |
| _CSRCs = arrLength; |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetApplicationSpecificData(const WebRtc_UWord8 subType, |
| const WebRtc_UWord32 name, |
| const WebRtc_UWord8* data, |
| const WebRtc_UWord16 length) |
| { |
| if(length %4 != 0) |
| { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if(_appData) |
| { |
| delete [] _appData; |
| } |
| |
| _appSend = true; |
| _appSubType = subType; |
| _appName = name; |
| _appData = new WebRtc_UWord8[length]; |
| _appLength = length; |
| memcpy(_appData, data, length); |
| return 0; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric)); |
| |
| _xrSendVoIPMetric = true; |
| return 0; |
| } |
| |
| // called under critsect _criticalSectionRTCPSender |
| WebRtc_Word32 RTCPSender::AddReportBlocks(WebRtc_UWord8* rtcpbuffer, |
| WebRtc_UWord32& pos, |
| WebRtc_UWord8& numberOfReportBlocks, |
| const RTCPReportBlock* received, |
| const WebRtc_UWord32 NTPsec, |
| const WebRtc_UWord32 NTPfrac) { |
| // sanity one block |
| if(pos + 24 >= IP_PACKET_SIZE) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| numberOfReportBlocks = _reportBlocks.size(); |
| if (received) { |
| // add our multiple RR to numberOfReportBlocks |
| numberOfReportBlocks++; |
| } |
| if (received) { |
| // answer to the one that sends to me |
| _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); |
| |
| // Remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC); |
| pos += 4; |
| |
| // fraction lost |
| rtcpbuffer[pos++]=received->fractionLost; |
| |
| // cumulative loss |
| ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, |
| received->cumulativeLost); |
| pos += 3; |
| // extended highest seq_no, contain the highest sequence number received |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| received->extendedHighSeqNum); |
| pos += 4; |
| |
| //Jitter |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->jitter); |
| pos += 4; |
| |
| // Last SR timestamp, our NTP time when we received the last report |
| // This is the value that we read from the send report packet not when we |
| // received it... |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->lastSR); |
| pos += 4; |
| |
| // Delay since last received report,time since we received the report |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| received->delaySinceLastSR); |
| pos += 4; |
| } |
| if ((pos + _reportBlocks.size() * 24) >= IP_PACKET_SIZE) { |
| WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, |
| "%s invalid argument", __FUNCTION__); |
| return -1; |
| } |
| std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it = |
| _reportBlocks.begin(); |
| |
| for (; it != _reportBlocks.end(); it++) { |
| // we can have multiple report block in a conference |
| WebRtc_UWord32 remoteSSRC = it->first; |
| RTCPReportBlock* reportBlock = it->second; |
| if (reportBlock) { |
| // Remote SSRC |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSSRC); |
| pos += 4; |
| |
| // fraction lost |
| rtcpbuffer[pos++] = reportBlock->fractionLost; |
| |
| // cumulative loss |
| ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos, |
| reportBlock->cumulativeLost); |
| pos += 3; |
| |
| // extended highest seq_no, contain the highest sequence number received |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| reportBlock->extendedHighSeqNum); |
| pos += 4; |
| |
| //Jitter |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| reportBlock->jitter); |
| pos += 4; |
| |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| reportBlock->lastSR); |
| pos += 4; |
| |
| ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, |
| reportBlock->delaySinceLastSR); |
| pos += 4; |
| } |
| } |
| return pos; |
| } |
| |
| // no callbacks allowed inside this function |
| WebRtc_Word32 |
| RTCPSender::SetTMMBN(const TMMBRSet* boundingSet, |
| const WebRtc_UWord32 maxBitrateKbit) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| |
| if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit)) |
| { |
| _sendTMMBN = true; |
| return 0; |
| } |
| return -1; |
| } |
| |
| WebRtc_Word32 |
| RTCPSender::RequestTMMBR(WebRtc_UWord32 estimatedBW, WebRtc_UWord32 packetOH) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| if(_TMMBR) |
| { |
| _tmmbr_Send = estimatedBW; |
| _packetOH_Send = packetOH; |
| |
| return 0; |
| } |
| return -1; |
| } |
| |
| RateControlRegion |
| RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse) |
| { |
| CriticalSectionScoped lock(_criticalSectionRTCPSender); |
| return _remoteRateControl.Update(rateControlInput, firstOverUse, |
| _clock.GetTimeInMS()); |
| } |
| } // namespace webrtc |
