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#pragma once
#include <cstdint>
/*
A 32.32 fixed point number. Th*is is used for expressing fractional
positions in a buffer.
When used for converting from sample rate f_s to f_d, the timing
error at time t is bounded by t * 2^-33 * r_d / r_s. For a
pessimistic conversion ratio, 8 kHz -> 192 kHz, this means that it
will take at least five days to accumulate one millisecond of error.
*/
using lfr_fixed_t = std::int64_t;
/* ========================================
Sample formats
======================================== */
/*
Audio sample formats.
*/
enum lfr_fmt_t {
LFR_FMT_U8,
LFR_FMT_S16BE,
LFR_FMT_S16LE,
LFR_FMT_S24BE,
LFR_FMT_S24LE,
LFR_FMT_F32BE,
LFR_FMT_F32LE,
LFR_FMT_COUNT
};
/*
Specialized resampling function, designed to work with a specific
sample format and filter type. Do not attempt to reuse a function
for a different filter.
Note that lengths are still measured in frames, but since the
function is specialized for a given format and number of channels,
there is no need to specify the format or number of channels.
*/
using lfr_resample_func_t = void (*)( lfr_fixed_t *pos, lfr_fixed_t inv_ratio, unsigned *dither, void *out, int outlen, const void *in, int inlen, const struct lfr_filter *filter);
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