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#ifndef ALC_FILTER_H
#define ALC_FILTER_H
#include "AL/al.h"
#include "math_defs.h"
/* Filters implementation is based on the "Cookbook formulae for audio
* EQ biquad filter coefficients" by Robert Bristow-Johnson
* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
*/
/* Implementation note: For the shelf filters, the specified gain is for the
* reference frequency, which is the centerpoint of the transition band. This
* better matches EFX filter design. To set the gain for the shelf itself, use
* the square root of the desired linear gain (or halve the dB gain).
*/
typedef enum ALfilterType {
/** EFX-style low-pass filter, specifying a gain and reference frequency. */
ALfilterType_HighShelf,
/** EFX-style high-pass filter, specifying a gain and reference frequency. */
ALfilterType_LowShelf,
/** Peaking filter, specifying a gain and reference frequency. */
ALfilterType_Peaking,
/** Low-pass cut-off filter, specifying a cut-off frequency. */
ALfilterType_LowPass,
/** High-pass cut-off filter, specifying a cut-off frequency. */
ALfilterType_HighPass,
/** Band-pass filter, specifying a center frequency. */
ALfilterType_BandPass,
} ALfilterType;
typedef struct ALfilterState {
ALfloat x[2]; /* History of two last input samples */
ALfloat y[2]; /* History of two last output samples */
ALfloat b0, b1, b2; /* Transfer function coefficients "b" */
ALfloat a1, a2; /* Transfer function coefficients "a" (a0 is pre-applied) */
} ALfilterState;
/* Currently only a C-based filter process method is implemented. */
#define ALfilterState_process ALfilterState_processC
/**
* Calculates the rcpQ (i.e. 1/Q) coefficient for shelving filters, using the
* reference gain and shelf slope parameter.
* \param gain 0 < gain
* \param slope 0 < slope <= 1
*/
inline ALfloat calc_rcpQ_from_slope(ALfloat gain, ALfloat slope)
{
return sqrtf((gain + 1.0f/gain)*(1.0f/slope - 1.0f) + 2.0f);
}
/**
* Calculates the rcpQ (i.e. 1/Q) coefficient for filters, using the normalized
* reference frequency and bandwidth.
* \param f0norm 0 < f0norm < 0.5.
* \param bandwidth 0 < bandwidth
*/
inline ALfloat calc_rcpQ_from_bandwidth(ALfloat f0norm, ALfloat bandwidth)
{
ALfloat w0 = F_TAU * f0norm;
return 2.0f*sinhf(logf(2.0f)/2.0f*bandwidth*w0/sinf(w0));
}
inline void ALfilterState_clear(ALfilterState *filter)
{
filter->x[0] = 0.0f;
filter->x[1] = 0.0f;
filter->y[0] = 0.0f;
filter->y[1] = 0.0f;
}
/**
* Sets up the filter state for the specified filter type and its parameters.
*
* \param filter The filter object to prepare.
* \param type The type of filter for the object to apply.
* \param gain The gain for the reference frequency response. Only used by the
* Shelf and Peaking filter types.
* \param f0norm The normalized reference frequency (ref_freq / sample_rate).
* This is the center point for the Shelf, Peaking, and BandPass
* filter types, or the cutoff frequency for the LowPass and
* HighPass filter types.
* \param rcpQ The reciprocal of the Q coefficient for the filter's transition
* band. Can be generated from calc_rcpQ_from_slope or
* calc_rcpQ_from_bandwidth depending on the available data.
*/
void ALfilterState_setParams(ALfilterState *filter, ALfilterType type, ALfloat gain, ALfloat f0norm, ALfloat rcpQ);
inline void ALfilterState_copyParams(ALfilterState *restrict dst, const ALfilterState *restrict src)
{
dst->b0 = src->b0;
dst->b1 = src->b1;
dst->b2 = src->b2;
dst->a1 = src->a1;
dst->a2 = src->a2;
}
void ALfilterState_processC(ALfilterState *filter, ALfloat *restrict dst, const ALfloat *restrict src, ALsizei numsamples);
inline void ALfilterState_processPassthru(ALfilterState *filter, const ALfloat *restrict src, ALsizei numsamples)
{
if(numsamples >= 2)
{
filter->x[1] = src[numsamples-2];
filter->x[0] = src[numsamples-1];
filter->y[1] = src[numsamples-2];
filter->y[0] = src[numsamples-1];
}
else if(numsamples == 1)
{
filter->x[1] = filter->x[0];
filter->x[0] = src[0];
filter->y[1] = filter->y[0];
filter->y[0] = src[0];
}
}
#endif /* ALC_FILTER_H */
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