mirror of
https://github.com/danog/libtgvoip.git
synced 2024-11-27 04:34:42 +01:00
3216b76349
Better jitter buffer with packet rescaling Tried to fix some issues on Linux (telegramdesktop/tdesktop#3413) Fixes for Windows Phone
96 lines
2.4 KiB
C++
96 lines
2.4 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 <stdio.h>
|
|
#include "MediaStreamItf.h"
|
|
#include "BlockingQueue.h"
|
|
#include "BufferPool.h"
|
|
#include "threading.h"
|
|
|
|
#define JITTER_SLOT_COUNT 64
|
|
#define JITTER_SLOT_SIZE 1024
|
|
#define JR_OK 1
|
|
#define JR_MISSING 2
|
|
#define JR_BUFFERING 3
|
|
|
|
struct jitter_packet_t{
|
|
unsigned char* buffer;
|
|
size_t size;
|
|
uint32_t timestamp;
|
|
double recvTimeDiff;
|
|
};
|
|
typedef struct jitter_packet_t jitter_packet_t;
|
|
|
|
namespace tgvoip{
|
|
class JitterBuffer{
|
|
public:
|
|
JitterBuffer(MediaStreamItf* out, uint32_t step);
|
|
~JitterBuffer();
|
|
void SetMinPacketCount(uint32_t count);
|
|
int GetMinPacketCount();
|
|
int GetCurrentDelay();
|
|
double GetAverageDelay();
|
|
void Reset();
|
|
void HandleInput(unsigned char* data, size_t len, uint32_t timestamp);
|
|
size_t HandleOutput(unsigned char* buffer, size_t len, int offsetInSteps, int* playbackScaledDuration);
|
|
void Tick();
|
|
void GetAverageLateCount(double* out);
|
|
int GetAndResetLostPacketCount();
|
|
double GetLastMeasuredJitter();
|
|
double GetLastMeasuredDelay();
|
|
|
|
private:
|
|
static size_t CallbackIn(unsigned char* data, size_t len, void* param);
|
|
static size_t CallbackOut(unsigned char* data, size_t len, void* param);
|
|
void PutInternal(jitter_packet_t* pkt);
|
|
int GetInternal(jitter_packet_t* pkt, int offset);
|
|
void Advance();
|
|
|
|
BufferPool bufferPool;
|
|
tgvoip_mutex_t mutex;
|
|
jitter_packet_t slots[JITTER_SLOT_COUNT];
|
|
int64_t nextTimestamp;
|
|
uint32_t step;
|
|
uint32_t minDelay;
|
|
uint32_t minMinDelay;
|
|
uint32_t maxMinDelay;
|
|
uint32_t maxUsedSlots;
|
|
uint32_t lastPutTimestamp;
|
|
uint32_t lossesToReset;
|
|
double resyncThreshold;
|
|
int lostCount;
|
|
int lostSinceReset;
|
|
int gotSinceReset;
|
|
bool wasReset;
|
|
bool needBuffering;
|
|
int delayHistory[64];
|
|
int lateHistory[64];
|
|
bool adjustingDelay;
|
|
unsigned int tickCount;
|
|
unsigned int latePacketCount;
|
|
unsigned int dontIncMinDelay;
|
|
unsigned int dontDecMinDelay;
|
|
int lostPackets;
|
|
double prevRecvTime;
|
|
double expectNextAtTime;
|
|
double deviationHistory[64];
|
|
int deviationPtr;
|
|
double lastMeasuredJitter;
|
|
double lastMeasuredDelay;
|
|
int outstandingDelayChange;
|
|
unsigned int dontChangeDelay;
|
|
double avgDelay;
|
|
//FILE* dump;
|
|
};
|
|
}
|
|
|
|
#endif //LIBTGVOIP_JITTERBUFFER_H
|