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- /*
- * 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.
- */
- #include <string.h>
- #define MAXOUT (+32767)
- #define MINOUT (-32768)
- /* limitter */
- #define Limit16(val) \
- val -= val >> 3; /* reduce level to avoid clipping */ \
- if ((short)val != val) val = (val < 0 ? MINOUT : MAXOUT)
- int mix_32_to_16l_level;
- static struct iir {
- int alpha; // alpha for EMA low pass
- int y[2]; // filter intermediates
- } 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 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])*(1-1/8192) (corner approx. 1Hz, gain 1)
- static inline int filter_exp(struct iir *fi2, int x)
- {
- fi2->y[1] += ((x << QB) - fi2->y[1]) >> QB;
- return x - (fi2->y[1] >> QB);
- }
- // unfiltered (for testing)
- 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(&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);
- }
- void mix_32_to_16l_stereo(short *dest, int *src, int count)
- {
- 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(&lf, l);
- Limit16(l);
- *dest++ = l;
- }
- lfi2 = lf;
- }
- void mix_16h_to_32(int *dest_buf, short *mp3_buf, int count)
- {
- while (count--)
- {
- *dest_buf++ += *mp3_buf++ >> 1;
- }
- }
- void mix_16h_to_32_s1(int *dest_buf, short *mp3_buf, int count)
- {
- count >>= 1;
- while (count--)
- {
- *dest_buf++ += *mp3_buf++ >> 1;
- *dest_buf++ += *mp3_buf++ >> 1;
- mp3_buf += 1*2;
- }
- }
- void mix_16h_to_32_s2(int *dest_buf, short *mp3_buf, int count)
- {
- count >>= 1;
- while (count--)
- {
- *dest_buf++ += *mp3_buf++ >> 1;
- *dest_buf++ += *mp3_buf++ >> 1;
- mp3_buf += 3*2;
- }
- }
- 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
- }
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