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https://github.com/RaySollium99/picodrive.git
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sound, fix psg audio
This commit is contained in:
kub
parent
b7697ca351
commit
af01b1b16f
1 changed files with 35 additions and 49 deletions
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@ -22,7 +22,7 @@
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#include "sn76496.h"
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#define MAX_OUTPUT 0x47ff // was 0x7fff
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#define MAX_OUTPUT 0x4800 // was 0x7fff
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#define STEP 0x10000
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@ -31,22 +31,15 @@
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/* bit0 = output */
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/* noise feedback for white noise mode (verified on real SN76489 by John Kortink) */
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#define FB_WNOISE 0x14002 /* (16bits) bit16 = bit0(out) ^ bit2 ^ bit15 */
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#define FB_WNOISE_T 0x3000 /* (15bits) bit15 = bit1 ^ bit2, TI */
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#define FB_WNOISE_S 0x9000 /* (16bits) bit16 = bit0 ^ bit3, Sega PSG */
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/* noise feedback for periodic noise mode */
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//#define FB_PNOISE 0x10000 /* 16bit rorate */
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#define FB_PNOISE 0x08000 /* JH 981127 - fixes Do Run Run */
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#define FB_PNOISE_T 0x4000 /* 15bit rotate for TI */
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#define FB_PNOISE_S 0x8000 /* 16bit rotate for Sega PSG */
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/*
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0x08000 is definitely wrong. The Master System conversion of Marble Madness
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uses periodic noise as a baseline. With a 15-bit rotate, the bassline is
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out of tune.
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The 16-bit rotate has been confirmed against a real PAL Sega Master System 2.
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Hope that helps the System E stuff, more news on the PSG as and when!
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*/
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/* noise generator start preset (for periodic noise) */
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#define NG_PRESET 0x0f35
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#define FB_WNOISE FB_WNOISE_S /* Sega */
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#define FB_PNOISE FB_PNOISE_S
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struct SN76496
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@ -58,7 +51,7 @@ struct SN76496
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int Register[8]; /* registers */
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int LastRegister; /* last register written */
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int Volume[4]; /* volume of voice 0-2 and noise */
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unsigned int RNG; /* noise generator */
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unsigned int RNG; /* noise generator */
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int NoiseFB; /* noise feedback mask */
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int Period[4];
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int Count[4];
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@ -97,6 +90,7 @@ void SN76496Write(int data)
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case 4: /* tone 2 : frequency */
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R->Period[c] = R->UpdateStep * data;
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if (R->Period[c] == 0) R->Period[c] = R->UpdateStep;
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R->Count[c] = 0;
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if (r == 4)
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{
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/* update noise shift frequency */
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@ -115,10 +109,11 @@ void SN76496Write(int data)
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R->NoiseFB = (n & 4) ? FB_WNOISE : FB_PNOISE;
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n &= 3;
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/* N/512,N/1024,N/2048,Tone #3 output */
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R->Period[3] = (n == 3) ? 2 * R->Period[2] : (R->UpdateStep << (5 + n));
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R->Period[3] = (n == 3) ? 2 * R->Period[2] : (R->UpdateStep << (4 + n));
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R->Count[3] = 0;
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/* reset noise shifter */
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R->RNG = NG_PRESET;
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R->RNG = FB_PNOISE;
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R->Output[3] = R->RNG & 1;
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break;
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}
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@ -138,18 +133,6 @@ void SN76496Update(short *buffer, int length, int stereo)
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int i;
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struct SN76496 *R = &ono_sn;
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/* If the volume is 0, increase the counter */
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for (i = 0;i < 4;i++)
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{
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if (R->Volume[i] == 0)
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{
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/* note that I do count += length, NOT count = length + 1. You might think */
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/* it's the same since the volume is 0, but doing the latter could cause */
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/* interferencies when the program is rapidly modulating the volume. */
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if (R->Count[i] <= length*STEP) R->Count[i] += length*STEP;
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}
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}
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while (length > 0)
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{
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int vol[4];
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@ -173,13 +156,16 @@ void SN76496Update(short *buffer, int length, int stereo)
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/* If we exit the loop in the middle, Output[i] has to be inverted */
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/* and vol[i] incremented only if the exit status of the square */
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/* wave is 1. */
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left = 0;
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while (R->Count[i] <= 0)
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if (R->Count[i] < -2*R->Period[i]) {
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/* Cut of anything above the Nyquist freqency */
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/* It will only create aliasing anyway */
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vol[i] += STEP/2; // mean value
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R->Count[i] = R->Output[i] = 0;
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}
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while (R->Count[i] < 0)
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{
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if (R->Count[i] + R->Period[i]*4 < R->Period[i])
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left+= 4, R->Count[i] += R->Period[i]*4;
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else left++, R->Count[i] += R->Period[i];
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if (R->Count[i] > 0)
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R->Count[i] += R->Period[i];
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if (R->Count[i] >= 0)
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{
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R->Output[i] ^= 1;
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if (R->Output[i]) vol[i] += R->Period[i];
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@ -189,12 +175,10 @@ void SN76496Update(short *buffer, int length, int stereo)
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vol[i] += R->Period[i];
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}
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if (R->Output[i]) vol[i] -= R->Count[i];
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/* Cut of anything above the sample freqency. It will only create */
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/* aliasing and hearable distortions anyway. */
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if (left > 1) vol[i] = STEP/2;
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}
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left = STEP;
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if (R->Output[3]) vol[3] += R->Count[3];
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do
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{
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int nextevent;
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@ -202,27 +186,29 @@ void SN76496Update(short *buffer, int length, int stereo)
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if (R->Count[3] < left) nextevent = R->Count[3];
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else nextevent = left;
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if (R->Output[3]) vol[3] += R->Count[3];
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R->Count[3] -= nextevent;
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if (R->Count[3] <= 0)
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{
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if (R->RNG & 1) R->RNG ^= R->NoiseFB;
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R->RNG >>= 1;
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R->Output[3] = R->RNG & 1;
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R->RNG >>= 1;
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if (R->Output[3])
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{
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R->RNG ^= R->NoiseFB;
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vol[3] += R->Period[3];
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}
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R->Count[3] += R->Period[3];
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if (R->Output[3]) vol[3] += R->Period[3];
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}
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if (R->Output[3]) vol[3] -= R->Count[3];
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left -= nextevent;
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} while (left > 0);
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if (R->Output[3]) vol[3] -= R->Count[3];
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out = vol[0] * R->Volume[0] + vol[1] * R->Volume[1] +
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vol[2] * R->Volume[2] + vol[3] * R->Volume[3];
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if (out > MAX_OUTPUT * STEP) out = MAX_OUTPUT * STEP;
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if ((out /= STEP)) // will be optimized to shift; max 0x47ff = 18431
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if ((out /= STEP)) // will be optimized to shift; max 0x4800 = 18432
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*buffer += out;
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if(stereo) buffer+=2; // only left for stereo, to be mixed to right later
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else buffer++;
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@ -254,7 +240,7 @@ static void SN76496_set_gain(struct SN76496 *R,int gain)
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gain &= 0xff;
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/* increase max output basing on gain (0.2 dB per step) */
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out = MAX_OUTPUT / 3;
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out = MAX_OUTPUT / 4.0;
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while (gain-- > 0)
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out *= 1.023292992; /* = (10 ^ (0.2/20)) */
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@ -262,7 +248,7 @@ static void SN76496_set_gain(struct SN76496 *R,int gain)
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for (i = 0;i < 15;i++)
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{
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/* limit volume to avoid clipping */
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if (out > MAX_OUTPUT / 3) R->VolTable[i] = MAX_OUTPUT / 3;
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if (out > MAX_OUTPUT / 4) R->VolTable[i] = MAX_OUTPUT / 4;
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else R->VolTable[i] = out;
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out /= 1.258925412; /* = 10 ^ (2/20) = 2dB */
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@ -294,10 +280,10 @@ int SN76496_init(int clock,int sample_rate)
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for (i = 0;i < 4;i++)
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{
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R->Output[i] = 0;
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R->Period[i] = R->Count[i] = R->UpdateStep;
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R->Volume[i] = R->Output[i] = R->Count[i] = 0;
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R->Period[i] = R->UpdateStep;
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}
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R->RNG = NG_PRESET;
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R->RNG = FB_PNOISE;
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R->Output[3] = R->RNG & 1;
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// added
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