ref: 653e85a92f58a1aab94793e209e12fde6a8d734b
dir: /src/pt2_audio.h/
#pragma once #include <stdint.h> #include <stdbool.h> // for the low-pass/high-pass filters in the SAMPLER screen #define FILTERS_BASE_FREQ (PAULA_PAL_CLK / 214.0) enum { AUDIO_NO_OVERSAMPLING = 0, AUDIO_2X_OVERSAMPLING = 1 }; typedef struct audio_t { volatile bool locked, isSampling; bool ledFilterEnabled, oversamplingFlag; uint32_t outputRate, audioBufferSize; int64_t tickSampleCounter64, samplesPerTick64; int64_t samplesPerTickTable[256-32]; // 32.32 fixed-point // for audio sampling bool rescanAudioDevicesSupported; // for audio/video syncing bool resetSyncTickTimeFlag; uint64_t tickLengthTable[224]; } audio_t; void updateReplayerTimingMode(void); void setSyncTickTimeLen(uint32_t timeLen, uint32_t timeLenFrac); void resetAudioDithering(void); void generateBpmTable(double dAudioFreq, bool vblankTimingFlag); uint16_t get16BitPeak(int16_t *sampleData, uint32_t sampleLength); uint32_t get32BitPeak(int32_t *sampleData, uint32_t sampleLength); float getFloatPeak(float *fSampleData, uint32_t sampleLength); double getDoublePeak(double *dSampleData, uint32_t sampleLength); void normalize16BitTo8Bit(int16_t *sampleData, uint32_t sampleLength); void normalize32BitTo8Bit(int32_t *sampleData, uint32_t sampleLength); void normalizeFloatTo8Bit(float *fSampleData, uint32_t sampleLength); void normalizeDoubleTo8Bit(double *dSampleData, uint32_t sampleLength); void toggleAmigaPanMode(void); void lockAudio(void); void unlockAudio(void); void audioSetStereoSeparation(uint8_t percentage); void outputAudio(int16_t *target, int32_t numSamples); bool setupAudio(void); void audioClose(void); extern audio_t audio; // pt2_audio.c