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add DC filter to sound mixer to remove potential PCM DC offset

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This commit is contained in:
kub 2019-12-31 10:55:40 +01:00
parent 9090dc0f22
commit 2a942f0d41
4 changed files with 121 additions and 35 deletions
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75
pico/sound/mix.c
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@ -6,41 +6,72 @@
* See COPYING file in the top-level directory.
*/
#include "string.h"
#define MAXOUT (+32767)
#define MINOUT (-32768)
/* limitter */
#define Limit(val, max,min) { \
if ( val > max ) val = max; \
else if ( val < min ) val = min; \
#define Limit16(val) { \
val -= (val >> 2); \
if ((short)val != val) val = (val < 0 ? MINOUT : MAXOUT); \
}
int mix_32_to_16l_level;
void mix_32_to_16l_stereo_core(short *dest, int *src, int count, int level)
{
int l, r;
static struct iir2 { // 2-pole IIR
int x[2]; // sample buffer
int y[2]; // filter intermediates
} lfi2, rfi2;
for (; count > 0; count--)
{
l = r = *dest;
l += *src++ >> level;
r += *src++ >> level;
Limit( l, MAXOUT, MINOUT );
Limit( r, MAXOUT, MINOUT );
*dest++ = l;
*dest++ = r;
}
// 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.
#define QB 12
// 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)
{
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;
}
// unfiltered (for testing)
static inline int filter_null(struct iir2 *fi2, int x)
{
return x;
}
#define mix_32_to_16l_stereo_core(dest, src, count, lv, fl) { \
int l, r; \
\
for (; count > 0; count--) \
{ \
l = r = *dest; \
l += *src++ >> lv; \
r += *src++ >> lv; \
l = fl(&lfi2, l); \
r = fl(&rfi2, r); \
Limit16(l); \
Limit16(r); \
*dest++ = l; \
*dest++ = r; \
} \
}
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);
mix_32_to_16l_stereo_core(dest, src, count, mix_32_to_16l_level, filter_exp);
}
void mix_32_to_16l_stereo(short *dest, int *src, int count)
{
mix_32_to_16l_stereo_core(dest, src, count, 0);
mix_32_to_16l_stereo_core(dest, src, count, 0, filter_exp);
}
void mix_32_to_16_mono(short *dest, int *src, int count)
@ -51,7 +82,8 @@ void mix_32_to_16_mono(short *dest, int *src, int count)
{
l = *dest;
l += *src++;
Limit( l, MAXOUT, MINOUT );
l = filter_exp(&lfi2, l);
Limit16(l);
*dest++ = l;
}
}
@ -87,3 +119,8 @@ void mix_16h_to_32_s2(int *dest_buf, short *mp3_buf, int count)
}
}
void mix_reset(void)
{
memset(&lfi2, 0, sizeof(lfi2));
memset(&rfi2, 0, sizeof(rfi2));
}