chromium / chromiumos / third_party / espeak-ng / cd20ac3b30963450080f20b0dbd1cc8767fa6a1d / . / emscripten / third_party / libresample / src / filterkit.c

/********************************************************************** | |

resamplesubs.c | |

Real-time library interface by Dominic Mazzoni | |

Based on resample-1.7: | |

http://www-ccrma.stanford.edu/~jos/resample/ | |

Dual-licensed as LGPL and BSD; see README.md and LICENSE* files. | |

This file provides Kaiser-windowed low-pass filter support, | |

including a function to create the filter coefficients, and | |

two functions to apply the filter at a particular point. | |

**********************************************************************/ | |

/* Definitions */ | |

#include "resample_defs.h" | |

#include "filterkit.h" | |

#include <stdlib.h> | |

#include <string.h> | |

#include <stdio.h> | |

#include <math.h> | |

/* LpFilter() | |

* | |

* reference: "Digital Filters, 2nd edition" | |

* R.W. Hamming, pp. 178-179 | |

* | |

* Izero() computes the 0th order modified bessel function of the first kind. | |

* (Needed to compute Kaiser window). | |

* | |

* LpFilter() computes the coeffs of a Kaiser-windowed low pass filter with | |

* the following characteristics: | |

* | |

* c[] = array in which to store computed coeffs | |

* frq = roll-off frequency of filter | |

* N = Half the window length in number of coeffs | |

* Beta = parameter of Kaiser window | |

* Num = number of coeffs before 1/frq | |

* | |

* Beta trades the rejection of the lowpass filter against the transition | |

* width from passband to stopband. Larger Beta means a slower | |

* transition and greater stopband rejection. See Rabiner and Gold | |

* (Theory and Application of DSP) under Kaiser windows for more about | |

* Beta. The following table from Rabiner and Gold gives some feel | |

* for the effect of Beta: | |

* | |

* All ripples in dB, width of transition band = D*N where N = window length | |

* | |

* BETA D PB RIP SB RIP | |

* 2.120 1.50 +-0.27 -30 | |

* 3.384 2.23 0.0864 -40 | |

* 4.538 2.93 0.0274 -50 | |

* 5.658 3.62 0.00868 -60 | |

* 6.764 4.32 0.00275 -70 | |

* 7.865 5.0 0.000868 -80 | |

* 8.960 5.7 0.000275 -90 | |

* 10.056 6.4 0.000087 -100 | |

*/ | |

#define IzeroEPSILON 1E-21 /* Max error acceptable in Izero */ | |

static double Izero(double x) | |

{ | |

double sum, u, halfx, temp; | |

int n; | |

sum = u = n = 1; | |

halfx = x/2.0; | |

do { | |

temp = halfx/(double)n; | |

n += 1; | |

temp *= temp; | |

u *= temp; | |

sum += u; | |

} while (u >= IzeroEPSILON*sum); | |

return(sum); | |

} | |

void lrsLpFilter(double c[], int N, double frq, double Beta, int Num) | |

{ | |

double IBeta, temp, temp1, inm1; | |

int i; | |

/* Calculate ideal lowpass filter impulse response coefficients: */ | |

c[0] = 2.0*frq; | |

for (i=1; i<N; i++) { | |

temp = PI*(double)i/(double)Num; | |

c[i] = sin(2.0*temp*frq)/temp; /* Analog sinc function, cutoff = frq */ | |

} | |

/* | |

* Calculate and Apply Kaiser window to ideal lowpass filter. | |

* Note: last window value is IBeta which is NOT zero. | |

* You're supposed to really truncate the window here, not ramp | |

* it to zero. This helps reduce the first sidelobe. | |

*/ | |

IBeta = 1.0/Izero(Beta); | |

inm1 = 1.0/((double)(N-1)); | |

for (i=1; i<N; i++) { | |

temp = (double)i * inm1; | |

temp1 = 1.0 - temp*temp; | |

temp1 = (temp1<0? 0: temp1); /* make sure it's not negative since | |

we're taking the square root - this | |

happens on Pentium 4's due to tiny | |

roundoff errors */ | |

c[i] *= Izero(Beta*sqrt(temp1)) * IBeta; | |

} | |

} | |

float lrsFilterUp(float Imp[], /* impulse response */ | |

float ImpD[], /* impulse response deltas */ | |

UWORD Nwing, /* len of one wing of filter */ | |

BOOL Interp, /* Interpolate coefs using deltas? */ | |

float *Xp, /* Current sample */ | |

double Ph, /* Phase */ | |

int Inc) /* increment (1 for right wing or -1 for left) */ | |

{ | |

float *Hp, *Hdp = NULL, *End; | |

double a = 0; | |

float v, t; | |

Ph *= Npc; /* Npc is number of values per 1/delta in impulse response */ | |

v = 0.0; /* The output value */ | |

Hp = &Imp[(int)Ph]; | |

End = &Imp[Nwing]; | |

if (Interp) { | |

Hdp = &ImpD[(int)Ph]; | |

a = Ph - floor(Ph); /* fractional part of Phase */ | |

} | |

if (Inc == 1) /* If doing right wing... */ | |

{ /* ...drop extra coeff, so when Ph is */ | |

End--; /* 0.5, we don't do too many mult's */ | |

if (Ph == 0) /* If the phase is zero... */ | |

{ /* ...then we've already skipped the */ | |

Hp += Npc; /* first sample, so we must also */ | |

Hdp += Npc; /* skip ahead in Imp[] and ImpD[] */ | |

} | |

} | |

if (Interp) | |

while (Hp < End) { | |

t = *Hp; /* Get filter coeff */ | |

t += (*Hdp)*a; /* t is now interp'd filter coeff */ | |

Hdp += Npc; /* Filter coeff differences step */ | |

t *= *Xp; /* Mult coeff by input sample */ | |

v += t; /* The filter output */ | |

Hp += Npc; /* Filter coeff step */ | |

Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */ | |

} | |

else | |

while (Hp < End) { | |

t = *Hp; /* Get filter coeff */ | |

t *= *Xp; /* Mult coeff by input sample */ | |

v += t; /* The filter output */ | |

Hp += Npc; /* Filter coeff step */ | |

Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */ | |

} | |

return v; | |

} | |

float lrsFilterUD(float Imp[], /* impulse response */ | |

float ImpD[], /* impulse response deltas */ | |

UWORD Nwing, /* len of one wing of filter */ | |

BOOL Interp, /* Interpolate coefs using deltas? */ | |

float *Xp, /* Current sample */ | |

double Ph, /* Phase */ | |

int Inc, /* increment (1 for right wing or -1 for left) */ | |

double dhb) /* filter sampling period */ | |

{ | |

float a; | |

float *Hp, *Hdp, *End; | |

float v, t; | |

double Ho; | |

v = 0.0; /* The output value */ | |

Ho = Ph*dhb; | |

End = &Imp[Nwing]; | |

if (Inc == 1) /* If doing right wing... */ | |

{ /* ...drop extra coeff, so when Ph is */ | |

End--; /* 0.5, we don't do too many mult's */ | |

if (Ph == 0) /* If the phase is zero... */ | |

Ho += dhb; /* ...then we've already skipped the */ | |

} /* first sample, so we must also */ | |

/* skip ahead in Imp[] and ImpD[] */ | |

if (Interp) | |

while ((Hp = &Imp[(int)Ho]) < End) { | |

t = *Hp; /* Get IR sample */ | |

Hdp = &ImpD[(int)Ho]; /* get interp bits from diff table*/ | |

a = Ho - floor(Ho); /* a is logically between 0 and 1 */ | |

t += (*Hdp)*a; /* t is now interp'd filter coeff */ | |

t *= *Xp; /* Mult coeff by input sample */ | |

v += t; /* The filter output */ | |

Ho += dhb; /* IR step */ | |

Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */ | |

} | |

else | |

while ((Hp = &Imp[(int)Ho]) < End) { | |

t = *Hp; /* Get IR sample */ | |

t *= *Xp; /* Mult coeff by input sample */ | |

v += t; /* The filter output */ | |

Ho += dhb; /* IR step */ | |

Xp += Inc; /* Input signal step. NO CHECK ON BOUNDS */ | |

} | |

return v; | |

} |