mirror of
https://github.com/danog/libtgvoip.git
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113 lines
3.4 KiB
C++
113 lines
3.4 KiB
C++
//
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// libtgvoip is free and unencumbered public domain software.
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// For more information, see http://unlicense.org or the UNLICENSE file
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// you should have received with this source code distribution.
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//
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#ifndef LIBTGVOIP_JITTERBUFFER_H
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#define LIBTGVOIP_JITTERBUFFER_H
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#include <stdlib.h>
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#include <vector>
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#include <algorithm>
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#include <atomic>
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#include <stdio.h>
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#include "controller/media/MediaStreamItf.h"
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#include "tools/BlockingQueue.h"
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#include "tools/Buffers.h"
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#include "tools/threading.h"
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#include "tools/logging.h"
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#define JITTER_SLOT_COUNT 64
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#define JITTER_SLOT_SIZE 1024
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#define JR_OK 1
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#define JR_MISSING 2
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#define JR_BUFFERING 3
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namespace tgvoip
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{
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class JitterBuffer
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{
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public:
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JitterBuffer(uint32_t step);
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~JitterBuffer();
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void SetMinPacketCount(uint32_t count);
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int GetMinPacketCount();
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unsigned int GetCurrentDelay();
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double GetAverageDelay();
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void Reset();
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void HandleInput(unsigned char *data, size_t len, uint32_t timestamp, bool isEC);
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size_t HandleOutput(unsigned char *buffer, size_t len, bool advance, int &playbackScaledDuration, bool &isEC);
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void Tick();
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void GetAverageLateCount(double *out);
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int GetAndResetLostPacketCount();
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double GetLastMeasuredJitter();
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double GetLastMeasuredDelay();
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double GetTimeoutWindow();
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// Get minimum refetchable seq for (reverse) NACK logic.
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// Any sequence numbers smaller than this cannot possibly arrive in time for playing.
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inline uint32_t GetSeqTooLate(double rtt)
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{
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//LOGE("Next fetch timestamp: %ld, rtt %lf, step %d", nextFetchTimestamp.load(), rtt * 1000, step)
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// The absolute minimum time(stamp) that will (barely) be accepted by the jitter buffer in time + RTT time
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// Then convert timestamp into a seqno: remember, in protocol >= PROTOCOL_RELIABLE, seq = ts * step + 1
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return ((nextFetchTimestamp + (rtt * 1000)) / static_cast<uint64_t>(step) + 1) - lostCount; // Seqs start at 1
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}
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private:
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struct jitter_packet_t
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{
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Buffer buffer = Buffer();
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uint32_t timestamp = 0;
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size_t size = 0;
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bool isEC = 0;
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double recvTimeDiff = 0.0;
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};
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void PutInternal(const jitter_packet_t &pkt, bool overwriteExisting);
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int GetInternal(jitter_packet_t &pkt, bool advance);
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void Advance();
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BufferPool<JITTER_SLOT_SIZE, JITTER_SLOT_COUNT> bufferPool;
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Mutex mutex;
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uint32_t step;
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std::array<jitter_packet_t, JITTER_SLOT_COUNT> slots{0};
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std::atomic<int64_t> nextFetchTimestamp{0}; // What frame to read next (protected for GetSeqTooLate)
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std::atomic<double> minDelay{6};
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uint32_t minMinDelay;
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uint32_t maxMinDelay;
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uint32_t maxUsedSlots;
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uint32_t lastPutTimestamp;
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uint32_t lossesToReset;
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double resyncThreshold;
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unsigned int lostCount = 0;
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unsigned int lostSinceReset = 0;
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unsigned int gotSinceReset = 0;
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bool wasReset = true;
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bool needBuffering = true;
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HistoricBuffer<int, 64, double> delayHistory;
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HistoricBuffer<int, 64, double> lateHistory;
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bool adjustingDelay = false;
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unsigned int tickCount = 0;
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unsigned int latePacketCount = 0;
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unsigned int dontIncMinDelay = 0;
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unsigned int dontDecMinDelay = 0;
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int lostPackets = 0;
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double prevRecvTime = 0;
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double expectNextAtTime = 0;
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HistoricBuffer<double, 64> deviationHistory;
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double lastMeasuredJitter = 0;
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double lastMeasuredDelay = 0;
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int outstandingDelayChange = 0;
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unsigned int dontChangeDelay = 0;
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double avgDelay = 0;
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bool first = true;
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#ifdef TGVOIP_DUMP_JITTER_STATS
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FILE *dump;
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#endif
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};
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} // namespace tgvoip
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#endif //LIBTGVOIP_JITTERBUFFER_H
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