ref: 0d1a2cf15c023a03f0ee62017133b7035e142a56
dir: /src/pt2_filters.h/
/* These are second variants of low-pass/high-pass filters that are better than ** the ones used in the main audio mixer. The reason we use a different one for ** the main audio mixer is because it makes it sound closer to real Amigas. ** ** These ones are used for filtering samples when loading samples, or with the ** FILTERS toolbox in the Sample Editor. */ #pragma once #include <stdio.h> #include <stdbool.h> /* 8bitbubsy: Before we downsample a loaded WAV/AIFF (>22kHz) sample by 2x, we low-pass ** filter it. ** *** I think this value ought to be 4.0 (nyquist freq. / 2), but it cuts off too much in ** my opinion! The improvement is only noticable on samples that has quite a bit of high ** frequencies in them to begin with. ** ** This is probably not how to do it, so if someone with a bit more knowledge can do this ** in a proper way without using an external resampler library, that would be neato! */ #define DOWNSAMPLE_CUTOFF_FACTOR 4.0 bool lowPassSample8Bit(int8_t *buffer, int32_t length, int32_t sampleFrequency, double cutoff); bool lowPassSample8BitUnsigned(uint8_t *buffer, int32_t length, int32_t sampleFrequency, double cutoff); bool lowPassSample16Bit(int16_t *buffer, int32_t length, int32_t sampleFrequency, double cutoff); bool lowPassSample32Bit(int32_t *buffer, int32_t length, int32_t sampleFrequency, double cutoff); bool lowPassSampleFloat(float *buffer, int32_t length, int32_t sampleFrequency, double cutoff); bool lowPassSampleDouble(double *buffer, int32_t length, int32_t sampleFrequency, double cutoff);