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sound, improved and optimized reimplementation of libretro lowpass filter

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This commit is contained in:
kub 2020-12-21 23:22:00 +01:00
parent b437951ade
commit 30969671e5
5 changed files with 141 additions and 123 deletions
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55
pico/sound/mix.c
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@ -1,6 +1,7 @@
/*
* some code for sample mixing
* (C) notaz, 2006,2007
* (C) kub, 2019,2020 added filtering
*
* This work is licensed under the terms of MAME license.
* See COPYING file in the top-level directory.
@ -13,78 +14,97 @@
/* limitter */
#define Limit16(val) \
val -= val >> 2; /* reduce level to avoid clipping */ \
if ((short)val != val) val = (val < 0 ? MINOUT : MAXOUT)
int mix_32_to_16l_level;
static struct iir2 { // 2-pole IIR
int x[2]; // sample buffer
static struct iir {
int alpha; // alpha for EMA low pass
int y[2]; // filter intermediates
int i;
} lfi2, rfi2;
// NB ">>" rounds to -infinity, "/" to 0. To compensate the effect possibly use
// "-(-y>>n)" (round to +infinity) instead of "y>>n" in places.
// NB uses Q12 fixpoint; samples mustn't have more than 20 bits for this.
// NB uses fixpoint; samples mustn't have more than (32-QB) bits. Adding the
// outputs of the sound sources together yields a max. of 18 bits, restricting
// QB to a maximum of 14.
#define QB 12
// NB alpha for DC filtering shouldn't be smaller than 1/(1<<QB) to avoid loss.
// exponential moving average combined DC filter and lowpass filter
// y0[n] = (x[n]-y0[n-1])*alpha+y0[n-1], y1[n] = (y0[n] - y1[n-1])*(1-1/8192)
static inline int filter_band(struct iir *fi2, int x)
{
// low pass. alpha is Q8 to avoid loss by 32 bit overflow.
// fi2->y[0] += ((x<<(QB-8)) - (fi2->y[0]>>8)) * fi2->alpha;
fi2->y[0] += (x - (fi2->y[0]>>QB)) * fi2->alpha;
// DC filter. for alpha=1-1/8192 cutoff ~1HZ, for 1-1/1024 ~7Hz
fi2->y[1] += (fi2->y[0] - fi2->y[1]) >> QB;
return (fi2->y[0] - fi2->y[1]) >> QB;
}
// exponential moving average filter for DC filtering
// y[n] = (x[n]-y[n-1])*(1/8192) (corner approx. 20Hz, gain 1)
static inline int filter_exp(struct iir2 *fi2, int x)
// y[n] = (x[n]-y[n-1])*(1-1/8192) (corner approx. 1Hz, gain 1)
static inline int filter_exp(struct iir *fi2, int x)
{
int xf = (x<<QB) - fi2->y[0];
fi2->y[0] += xf >> 13;
xf -= xf >> 2; // level reduction to avoid clipping from overshoot
return xf>>QB;
fi2->y[1] += ((x << QB) - fi2->y[1]) >> QB;
return x - (fi2->y[1] >> QB);
}
// unfiltered (for testing)
static inline int filter_null(struct iir2 *fi2, int x)
static inline int filter_null(struct iir *fi2, int x)
{
return x;
}
#define filter filter_band
#define mix_32_to_16l_stereo_core(dest, src, count, lv, fl) { \
int l, r; \
struct iir lf = lfi2, rf = rfi2; \
\
for (; count > 0; count--) \
{ \
l = r = *dest; \
l += *src++ >> lv; \
r += *src++ >> lv; \
l = fl(&lfi2, l); \
r = fl(&rfi2, r); \
l = fl(&lf, l); \
r = fl(&rf, r); \
Limit16(l); \
Limit16(r); \
*dest++ = l; \
*dest++ = r; \
} \
lfi2 = lf, rfi2 = rf; \
}
void mix_32_to_16l_stereo_lvl(short *dest, int *src, int count)
{
mix_32_to_16l_stereo_core(dest, src, count, mix_32_to_16l_level, filter_exp);
mix_32_to_16l_stereo_core(dest, src, count, mix_32_to_16l_level, filter);
}
void mix_32_to_16l_stereo(short *dest, int *src, int count)
{
mix_32_to_16l_stereo_core(dest, src, count, 0, filter_exp);
mix_32_to_16l_stereo_core(dest, src, count, 0, filter);
}
void mix_32_to_16_mono(short *dest, int *src, int count)
{
int l;
struct iir lf = lfi2;
for (; count > 0; count--)
{
l = *dest;
l += *src++;
l = filter_exp(&lfi2, l);
l = filter(&lf, l);
Limit16(l);
*dest++ = l;
}
lfi2 = lf;
}
@ -118,8 +138,9 @@ void mix_16h_to_32_s2(int *dest_buf, short *mp3_buf, int count)
}
}
void mix_reset(void)
void mix_reset(int alpha_q16)
{
memset(&lfi2, 0, sizeof(lfi2));
memset(&rfi2, 0, sizeof(rfi2));
lfi2.alpha = rfi2.alpha = (0x10000-alpha_q16) >> 4; // filter alpha, Q12
}