| /** |
| * Copyright (C) 2022 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| #ifndef RTCP_XR_ANALYZER_H_INCLUDED |
| #define RTCP_XR_ANALYZER_H_INCLUDED |
| |
| #include <ImsMediaDefine.h> |
| #include <ImsMediaBitWriter.h> |
| #include <list> |
| #include <vector> |
| |
| #define MAX_BLOCK_LENGTH 220 |
| #define BLOCK_LENGTH_STATISTICS 40 |
| #define BLOCK_LENGTH_VOIP_METRICS 36 |
| #define G_MIN_THRESHOLD 16 |
| |
| struct tLossReport |
| { |
| tLossReport() : |
| beginSeq(-1), |
| endSeq(-1), |
| numLostPackets(-1), |
| numPacketsReceived(-1), |
| lossRate(-1) |
| { |
| } |
| int16_t beginSeq; |
| int16_t endSeq; |
| int32_t numLostPackets; |
| int32_t numPacketsReceived; |
| int32_t lossRate; |
| }; |
| |
| struct tTTLReport |
| { |
| tTTLReport() : |
| beginSeq(-1), |
| endSeq(-1), |
| ipVersion(-1), |
| minTTL(-1), |
| meanTTL(-1), |
| maxTTL(-1), |
| devTTL(-1) |
| { |
| } |
| int16_t beginSeq; |
| int16_t endSeq; |
| uint8_t ipVersion; |
| int32_t minTTL; |
| int32_t meanTTL; |
| int32_t maxTTL; |
| int32_t devTTL; |
| }; |
| |
| struct tDuplicateReport |
| { |
| tDuplicateReport() : |
| beginSeq(-1), |
| endSeq(-1), |
| numDuplicatedPackets(-1), |
| numPacketsReceived(-1) |
| { |
| } |
| int16_t beginSeq; |
| int16_t endSeq; |
| int32_t numDuplicatedPackets; |
| int32_t numPacketsReceived; |
| }; |
| |
| struct tJitterReport |
| { |
| tJitterReport() : |
| beginSeq(-1), |
| endSeq(-1), |
| minJitter(-1), |
| meanJitter(-1), |
| maxJitter(-1), |
| devJitter(-1) |
| { |
| } |
| int16_t beginSeq; |
| int16_t endSeq; |
| int32_t minJitter; |
| int32_t meanJitter; |
| int32_t maxJitter; |
| int32_t devJitter; |
| }; |
| |
| struct tVoIPMatricReport |
| { |
| tVoIPMatricReport() : |
| ssrc(0), |
| lossRate(0), |
| discardRate(0), |
| burstDensity(0), |
| gapDensity(0), |
| burstDuration(0), |
| gapDuration(0), |
| roundTripDelay(0), |
| endSystemDelay(0), |
| signalLevel(0), |
| noiseLevel(0), |
| rerl(0), |
| gMin(0), |
| rFactor(0), |
| extRFactor(0), |
| rxConfig(0), |
| jitterBufferNominal(0), |
| jitterBufferMaximum(0), |
| jitterBufferAbsMaximum(0) |
| { |
| } |
| uint32_t ssrc; |
| uint32_t lossRate; |
| uint32_t discardRate; |
| uint32_t burstDensity; |
| uint32_t gapDensity; |
| uint32_t burstDuration; |
| uint32_t gapDuration; |
| uint32_t roundTripDelay; |
| uint32_t endSystemDelay; |
| uint32_t signalLevel; |
| uint32_t noiseLevel; |
| uint32_t rerl; |
| uint32_t gMin; |
| uint32_t rFactor; |
| uint32_t extRFactor; |
| uint8_t rxConfig; |
| uint32_t jitterBufferNominal; |
| uint32_t jitterBufferMaximum; |
| uint32_t jitterBufferAbsMaximum; |
| }; |
| |
| class RtcpXrEncoder |
| { |
| public: |
| RtcpXrEncoder(); |
| virtual ~RtcpXrEncoder(); |
| |
| /** |
| * @brief Set the receiving stream ssrc |
| */ |
| void setSsrc(const uint32_t ssrc); |
| |
| /** |
| * @brief Set the sampling rate of audio codec in kHz unit |
| */ |
| void setSamplingRate(const uint32_t rate); |
| |
| /** |
| * @brief Set the round trip delay in milliseconds unit |
| */ |
| void setRoundTripDelay(const uint32_t delay); |
| |
| /** |
| * @brief Stack receiving rtp status |
| * |
| * @param status The status of the audio frame how it is processed in jitter buffer |
| * @param delay The delay of the audio frame from stacked in jitter buffer to play to the audio |
| * codec |
| */ |
| void stackRxRtpStatus(const int32_t status, const uint32_t delay); |
| |
| /** |
| * @brief Set the jitter buffer size |
| * |
| * @param current The current jitter buffer size |
| * @param max The maximum jitter buffer size in worst condition |
| */ |
| void setJitterBufferStatus(const uint32_t current, const uint32_t max); |
| |
| /** |
| * @brief Create a rtcp-xr report blocks |
| * |
| * @param nRtcpXrReport The bitmask of the enabled report block types |
| * @param packets The list of the packets received to use in statistic summary report block |
| * @param lostPackets The list of the lost packet counted to use in statistic summary report |
| * block |
| * @param beginSeq The beginning of the rtp sequence number to generate statistics summery |
| * report block |
| * @param endSeq The end of the rtp sequence number to generate statistics summery report block |
| * @param data The data buffer to store the report block |
| * @param size The size of the data buffer to stored |
| * @return true The report block generated without error |
| * @return false The bitmask of the report block types are zero or data buffer is null |
| */ |
| bool createRtcpXrReport(const uint32_t nRtcpXrReport, std::list<RtpPacket*>* packets, |
| std::list<LostPacket*>* lostPackets, uint16_t beginSeq, uint16_t endSeq, uint8_t* data, |
| uint32_t& size); |
| |
| private: |
| tLossReport* createLossAnalysisReport(std::list<RtpPacket*>* packets, |
| std::list<LostPacket*>* lostPackets, uint16_t beginSeq, uint16_t endSeq); |
| tJitterReport* createJitterAnalysisReport( |
| std::list<RtpPacket*>* packets, uint16_t beginSeq, uint16_t endSeq); |
| tTTLReport* createTTLAnalysisReport( |
| std::list<RtpPacket*>* packets, uint16_t beginSeq, uint16_t endSeq); |
| tDuplicateReport* createDuplicateAnalysisReport( |
| std::list<RtpPacket*>* packets, uint16_t beginSeq, uint16_t endSeq); |
| tVoIPMatricReport* createVoIPMatricReport(); |
| void encodeStatisticSummeryReport(tLossReport* lossReport, tJitterReport* jitterReport, |
| tTTLReport* ttlReport, tDuplicateReport* duplicateReport, uint8_t* data); |
| void encodeVoipMetricReport(tVoIPMatricReport* report, uint8_t* data); |
| |
| uint32_t mSsrc; |
| uint32_t mSamplingRate; |
| uint32_t mRoundTripDelay; |
| uint32_t mVoipLossCount; |
| uint32_t mVoipDiscardedCount; |
| uint32_t mVoipPktCount; |
| uint32_t mVoipLostCountInBurst; |
| uint32_t mJitterBufferNominal; |
| uint32_t mJitterBufferMax; |
| uint32_t mJitterBufferAbsMax; |
| |
| /** |
| * state 1 = received a packet during a gap |
| * state 2 = received a packet during a burst |
| * state 3 = lost a packet during a burst |
| * state 4 = lost an isolated packet during a gap |
| */ |
| // calculated when rtp packet received |
| uint32_t mVoipC11; |
| uint32_t mVoipC13; |
| uint32_t mVoipC14; |
| uint32_t mVoipC22; |
| uint32_t mVoipC23; |
| uint32_t mVoipC33; |
| // calculate when create the voip report |
| uint32_t mVoipC31; |
| uint32_t mVoipC32; |
| ImsMediaBitWriter mBitWriter; |
| }; |
| |
| #endif |
