mirror of
https://github.com/danog/libtgvoip.git
synced 2024-12-11 16:49:52 +01:00
113 lines
3.4 KiB
C++
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, 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();
|
|
uint32_t timestamp = 0;
|
|
size_t size = 0;
|
|
bool isEC = 0;
|
|
double recvTimeDiff = 0.0;
|
|
};
|
|
void PutInternal(const jitter_packet_t &pkt, bool overwriteExisting);
|
|
int GetInternal(jitter_packet_t &pkt, 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{0}; // What frame to read next (protected for GetSeqTooLate)
|
|
std::atomic<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
|