From: kub <derkub@gmail.com>
Date: Fri, 24 Sep 2021 16:31:19 +0000 (+0200)
Subject: sound, fix psg audio
X-Git-Tag: v2.00~486
X-Git-Url: https://notaz.gp2x.de/cgi-bin/gitweb.cgi?a=commitdiff_plain;h=af01b1b16f4614c9f629ee4520c2bf28a985eb6a;p=picodrive.git

sound, fix psg audio
---

diff --git a/pico/sound/sn76496.c b/pico/sound/sn76496.c
index 4507507c..b4a8847f 100644
--- a/pico/sound/sn76496.c
+++ b/pico/sound/sn76496.c
@@ -22,7 +22,7 @@
 
 #include "sn76496.h"
 
-#define MAX_OUTPUT 0x47ff // was 0x7fff
+#define MAX_OUTPUT 0x4800 // was 0x7fff
 
 #define STEP 0x10000
 
@@ -31,22 +31,15 @@
 /* bit0 = output */
 
 /* 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 */
-//#define FB_PNOISE 0x10000 /* 16bit rorate */
-#define FB_PNOISE 0x08000   /* JH 981127 - fixes Do Run Run */
+#define FB_PNOISE_T 0x4000	/* 15bit rotate for TI */
+#define FB_PNOISE_S 0x8000	/* 16bit rotate for Sega PSG */
 
-/*
-0x08000 is definitely wrong. The Master System conversion of Marble Madness
-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
+#define FB_WNOISE FB_WNOISE_S	/* Sega */
+#define FB_PNOISE FB_PNOISE_S
 
 
 struct SN76496
@@ -58,7 +51,7 @@ struct SN76496
 	int Register[8];	/* registers */
 	int LastRegister;	/* last register written */
 	int Volume[4];		/* volume of voice 0-2 and noise */
-	unsigned int RNG;		/* noise generator      */
+	unsigned int RNG;	/* noise generator      */
 	int NoiseFB;		/* noise feedback mask */
 	int Period[4];
 	int Count[4];
@@ -97,6 +90,7 @@ void SN76496Write(int data)
 		case 4:	/* tone 2 : frequency */
 			R->Period[c] = R->UpdateStep * data;
 			if (R->Period[c] == 0) R->Period[c] = R->UpdateStep;
+			R->Count[c] = 0;
 			if (r == 4)
 			{
 				/* update noise shift frequency */
@@ -115,10 +109,11 @@ void SN76496Write(int data)
 			R->NoiseFB = (n & 4) ? FB_WNOISE : FB_PNOISE;
 			n &= 3;
 			/* 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 */
-			R->RNG = NG_PRESET;
+			R->RNG = FB_PNOISE;
 			R->Output[3] = R->RNG & 1;
 			break;
 	}
@@ -138,18 +133,6 @@ void SN76496Update(short *buffer, int length, int stereo)
 	int i;
 	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)
 	{
 		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 */
 			/* and vol[i] incremented only if the exit status of the square */
 			/* wave is 1. */
-			left = 0;
-			while (R->Count[i] <= 0)
+			if (R->Count[i] < -2*R->Period[i]) {
+				/* 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])
-					left+= 4, R->Count[i] += R->Period[i]*4;
-				else	left++,   R->Count[i] += R->Period[i];
-				if (R->Count[i] > 0)
+				R->Count[i] += R->Period[i];
+				if (R->Count[i] >= 0)
 				{
 					R->Output[i] ^= 1;
 					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];
 			}
 			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;
+		if (R->Output[3]) vol[3] += R->Count[3];
 		do
 		{
 			int nextevent;
@@ -202,27 +186,29 @@ void SN76496Update(short *buffer, int length, int stereo)
 			if (R->Count[3] < left) nextevent = R->Count[3];
 			else nextevent = left;
 
-			if (R->Output[3]) vol[3] += R->Count[3];
 			R->Count[3] -= nextevent;
 			if (R->Count[3] <= 0)
 			{
-				if (R->RNG & 1) R->RNG ^= R->NoiseFB;
-				R->RNG >>= 1;
 				R->Output[3] = R->RNG & 1;
+				R->RNG >>= 1;
+				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->Period[3];
 			}
-			if (R->Output[3]) vol[3] -= R->Count[3];
 
 			left -= nextevent;
 		} while (left > 0);
+		if (R->Output[3]) vol[3] -= R->Count[3];
 
 		out = vol[0] * R->Volume[0] + vol[1] * R->Volume[1] +
 				vol[2] * R->Volume[2] + vol[3] * R->Volume[3];
 
 		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;
 		if(stereo) buffer+=2; // only left for stereo, to be mixed to right later
 		else buffer++;
@@ -254,7 +240,7 @@ static void SN76496_set_gain(struct SN76496 *R,int gain)
 	gain &= 0xff;
 
 	/* increase max output basing on gain (0.2 dB per step) */
-	out = MAX_OUTPUT / 3;
+	out = MAX_OUTPUT / 4.0;
 	while (gain-- > 0)
 		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++)
 	{
 		/* 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;
 
 		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++)
 	{
-		R->Output[i] = 0;
-		R->Period[i] = R->Count[i] = R->UpdateStep;
+		R->Volume[i] = R->Output[i] = R->Count[i] = 0;
+		R->Period[i] = R->UpdateStep;
 	}
-	R->RNG = NG_PRESET;
+	R->RNG = FB_PNOISE;
 	R->Output[3] = R->RNG & 1;
 
 	// added