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sound, fix psg audio

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This commit is contained in:
kub 2021-09-24 18:31:19 +02:00
parent b7697ca351
commit af01b1b16f
1 changed files with 35 additions and 49 deletions
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84
pico/sound/sn76496.c
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@ -22,7 +22,7 @@
#include "sn76496.h" #include "sn76496.h"
#define MAX_OUTPUT 0x47ff // was 0x7fff #define MAX_OUTPUT 0x4800 // was 0x7fff
#define STEP 0x10000 #define STEP 0x10000
@ -31,22 +31,15 @@
/* bit0 = output */ /* bit0 = output */
/* noise feedback for white noise mode (verified on real SN76489 by John Kortink) */ /* noise feedback for white noise mode (verified on real SN76489 by John Kortink) */
#define FB_WNOISE 0x14002 /* (16bits) bit16 = bit0(out) ^ bit2 ^ bit15 */ #define FB_WNOISE_T 0x3000 /* (15bits) bit15 = bit1 ^ bit2, TI */
#define FB_WNOISE_S 0x9000 /* (16bits) bit16 = bit0 ^ bit3, Sega PSG */
/* noise feedback for periodic noise mode */ /* noise feedback for periodic noise mode */
//#define FB_PNOISE 0x10000 /* 16bit rorate */ #define FB_PNOISE_T 0x4000 /* 15bit rotate for TI */
#define FB_PNOISE 0x08000 /* JH 981127 - fixes Do Run Run */ #define FB_PNOISE_S 0x8000 /* 16bit rotate for Sega PSG */
/* #define FB_WNOISE FB_WNOISE_S /* Sega */
0x08000 is definitely wrong. The Master System conversion of Marble Madness #define FB_PNOISE FB_PNOISE_S
uses periodic noise as a baseline. With a 15-bit rotate, the bassline is
out of tune.
The 16-bit rotate has been confirmed against a real PAL Sega Master System 2.
Hope that helps the System E stuff, more news on the PSG as and when!
*/
/* noise generator start preset (for periodic noise) */
#define NG_PRESET 0x0f35
struct SN76496 struct SN76496
@ -97,6 +90,7 @@ void SN76496Write(int data)
case 4: /* tone 2 : frequency */ case 4: /* tone 2 : frequency */
R->Period[c] = R->UpdateStep * data; R->Period[c] = R->UpdateStep * data;
if (R->Period[c] == 0) R->Period[c] = R->UpdateStep; if (R->Period[c] == 0) R->Period[c] = R->UpdateStep;
R->Count[c] = 0;
if (r == 4) if (r == 4)
{ {
/* update noise shift frequency */ /* update noise shift frequency */
@ -115,10 +109,11 @@ void SN76496Write(int data)
R->NoiseFB = (n & 4) ? FB_WNOISE : FB_PNOISE; R->NoiseFB = (n & 4) ? FB_WNOISE : FB_PNOISE;
n &= 3; n &= 3;
/* N/512,N/1024,N/2048,Tone #3 output */ /* N/512,N/1024,N/2048,Tone #3 output */
R->Period[3] = (n == 3) ? 2 * R->Period[2] : (R->UpdateStep << (5 + n)); R->Period[3] = (n == 3) ? 2 * R->Period[2] : (R->UpdateStep << (4 + n));
R->Count[3] = 0;
/* reset noise shifter */ /* reset noise shifter */
R->RNG = NG_PRESET; R->RNG = FB_PNOISE;
R->Output[3] = R->RNG & 1; R->Output[3] = R->RNG & 1;
break; break;
} }
@ -138,18 +133,6 @@ void SN76496Update(short *buffer, int length, int stereo)
int i; int i;
struct SN76496 *R = &ono_sn; struct SN76496 *R = &ono_sn;
/* If the volume is 0, increase the counter */
for (i = 0;i < 4;i++)
{
if (R->Volume[i] == 0)
{
/* note that I do count += length, NOT count = length + 1. You might think */
/* it's the same since the volume is 0, but doing the latter could cause */
/* interferencies when the program is rapidly modulating the volume. */
if (R->Count[i] <= length*STEP) R->Count[i] += length*STEP;
}
}
while (length > 0) while (length > 0)
{ {
int vol[4]; int vol[4];
@ -173,13 +156,16 @@ void SN76496Update(short *buffer, int length, int stereo)
/* If we exit the loop in the middle, Output[i] has to be inverted */ /* If we exit the loop in the middle, Output[i] has to be inverted */
/* and vol[i] incremented only if the exit status of the square */ /* and vol[i] incremented only if the exit status of the square */
/* wave is 1. */ /* wave is 1. */
left = 0; if (R->Count[i] < -2*R->Period[i]) {
while (R->Count[i] <= 0) /* Cut of anything above the Nyquist freqency */
/* It will only create aliasing anyway */
vol[i] += STEP/2; // mean value
R->Count[i] = R->Output[i] = 0;
}
while (R->Count[i] < 0)
{ {
if (R->Count[i] + R->Period[i]*4 < R->Period[i]) R->Count[i] += R->Period[i];
left+= 4, R->Count[i] += R->Period[i]*4; if (R->Count[i] >= 0)
else left++, R->Count[i] += R->Period[i];
if (R->Count[i] > 0)
{ {
R->Output[i] ^= 1; R->Output[i] ^= 1;
if (R->Output[i]) vol[i] += R->Period[i]; if (R->Output[i]) vol[i] += R->Period[i];
@ -189,12 +175,10 @@ void SN76496Update(short *buffer, int length, int stereo)
vol[i] += R->Period[i]; vol[i] += R->Period[i];
} }
if (R->Output[i]) vol[i] -= R->Count[i]; if (R->Output[i]) vol[i] -= R->Count[i];
/* Cut of anything above the sample freqency. It will only create */
/* aliasing and hearable distortions anyway. */
if (left > 1) vol[i] = STEP/2;
} }
left = STEP; left = STEP;
if (R->Output[3]) vol[3] += R->Count[3];
do do
{ {
int nextevent; int nextevent;
@ -202,27 +186,29 @@ void SN76496Update(short *buffer, int length, int stereo)
if (R->Count[3] < left) nextevent = R->Count[3]; if (R->Count[3] < left) nextevent = R->Count[3];
else nextevent = left; else nextevent = left;
if (R->Output[3]) vol[3] += R->Count[3];
R->Count[3] -= nextevent; R->Count[3] -= nextevent;
if (R->Count[3] <= 0) if (R->Count[3] <= 0)
{ {
if (R->RNG & 1) R->RNG ^= R->NoiseFB;
R->RNG >>= 1;
R->Output[3] = R->RNG & 1; R->Output[3] = R->RNG & 1;
R->Count[3] += R->Period[3]; R->RNG >>= 1;
if (R->Output[3]) vol[3] += R->Period[3]; if (R->Output[3])
{
R->RNG ^= R->NoiseFB;
vol[3] += R->Period[3];
}
R->Count[3] += R->Period[3];
} }
if (R->Output[3]) vol[3] -= R->Count[3];
left -= nextevent; left -= nextevent;
} while (left > 0); } while (left > 0);
if (R->Output[3]) vol[3] -= R->Count[3];
out = vol[0] * R->Volume[0] + vol[1] * R->Volume[1] + out = vol[0] * R->Volume[0] + vol[1] * R->Volume[1] +
vol[2] * R->Volume[2] + vol[3] * R->Volume[3]; vol[2] * R->Volume[2] + vol[3] * R->Volume[3];
if (out > MAX_OUTPUT * STEP) out = MAX_OUTPUT * STEP; if (out > MAX_OUTPUT * STEP) out = MAX_OUTPUT * STEP;
if ((out /= STEP)) // will be optimized to shift; max 0x47ff = 18431 if ((out /= STEP)) // will be optimized to shift; max 0x4800 = 18432
*buffer += out; *buffer += out;
if(stereo) buffer+=2; // only left for stereo, to be mixed to right later if(stereo) buffer+=2; // only left for stereo, to be mixed to right later
else buffer++; else buffer++;
@ -254,7 +240,7 @@ static void SN76496_set_gain(struct SN76496 *R,int gain)
gain &= 0xff; gain &= 0xff;
/* increase max output basing on gain (0.2 dB per step) */ /* increase max output basing on gain (0.2 dB per step) */
out = MAX_OUTPUT / 3; out = MAX_OUTPUT / 4.0;
while (gain-- > 0) while (gain-- > 0)
out *= 1.023292992; /* = (10 ^ (0.2/20)) */ out *= 1.023292992; /* = (10 ^ (0.2/20)) */
@ -262,7 +248,7 @@ static void SN76496_set_gain(struct SN76496 *R,int gain)
for (i = 0;i < 15;i++) for (i = 0;i < 15;i++)
{ {
/* limit volume to avoid clipping */ /* limit volume to avoid clipping */
if (out > MAX_OUTPUT / 3) R->VolTable[i] = MAX_OUTPUT / 3; if (out > MAX_OUTPUT / 4) R->VolTable[i] = MAX_OUTPUT / 4;
else R->VolTable[i] = out; else R->VolTable[i] = out;
out /= 1.258925412; /* = 10 ^ (2/20) = 2dB */ out /= 1.258925412; /* = 10 ^ (2/20) = 2dB */
@ -294,10 +280,10 @@ int SN76496_init(int clock,int sample_rate)
for (i = 0;i < 4;i++) for (i = 0;i < 4;i++)
{ {
R->Output[i] = 0; R->Volume[i] = R->Output[i] = R->Count[i] = 0;
R->Period[i] = R->Count[i] = R->UpdateStep; R->Period[i] = R->UpdateStep;
} }
R->RNG = NG_PRESET; R->RNG = FB_PNOISE;
R->Output[3] = R->RNG & 1; R->Output[3] = R->RNG & 1;
// added // added