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libtgvoip/controller/net/JitterBuffer.h

113 lines
3.4 KiB
C++

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