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AdLib Gold changes of the day (September 23rd, 2025) (#6218)

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1. Make sure the check to the Surround module is properly placed when disabled/enabled.
2. Replace local adgold_buffer with opl_buffer from its struct to improve the audio output and less clipping.
This commit is contained in:
TC1995
2025-09-23 19:49:15 +02:00
committed by GitHub
parent 9f41857b5c
commit a87bcd410e
1 changed files with 30 additions and 41 deletions
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71
src/sound/snd_adlibgold.c
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@@ -291,7 +291,8 @@ adgold_write(uint16_t addr, uint8_t val, void *priv)
case 0x18: /*Surround*/
adgold->adgold_38x_regs[0x18] = val;
ym7128_write(&adgold->ym7128, val);
if (adgold->surround_enabled)
ym7128_write(&adgold->ym7128, val);
break;
default:
@@ -782,34 +783,30 @@ static void
adgold_get_buffer(int32_t *buffer, int len, void *priv)
{
adgold_t *adgold = (adgold_t *) priv;
int16_t *adgold_buffer = malloc(sizeof(int16_t) * len * 2);
if (adgold_buffer == NULL)
fatal("adgold_buffer = NULL");
int c;
adgold_update(adgold);
for (c = 0; c < len * 2; c += 2) {
adgold_buffer[c] = ((adgold->mma_buffer[0][c >> 1] * adgold->samp_vol_l) >> 7) / 4;
adgold_buffer[c + 1] = ((adgold->mma_buffer[1][c >> 1] * adgold->samp_vol_r) >> 7) / 4;
adgold->opl_buffer[c] = ((adgold->mma_buffer[0][c >> 1] * adgold->samp_vol_l) >> 7) / 4;
adgold->opl_buffer[c + 1] = ((adgold->mma_buffer[1][c >> 1] * adgold->samp_vol_r) >> 7) / 4;
}
if (adgold->surround_enabled)
ym7128_apply(&adgold->ym7128, adgold_buffer, len);
ym7128_apply(&adgold->ym7128, adgold->opl_buffer, len);
switch (adgold->adgold_38x_regs[0x8] & 6) {
case 0:
for (c = 0; c < len * 2; c++)
adgold_buffer[c] = 0;
adgold->opl_buffer[c] = 0;
break;
case 2: /*Left channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c + 1] = adgold_buffer[c];
adgold->opl_buffer[c + 1] = adgold->opl_buffer[c];
break;
case 4: /*Right channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] = adgold_buffer[c + 1];
adgold->opl_buffer[c] = adgold->opl_buffer[c + 1];
break;
case 6: /*Left and right channels*/
break;
@@ -821,7 +818,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
switch (adgold->adgold_38x_regs[0x8] & 0x18) {
case 0x00: /*Forced mono*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] = adgold_buffer[c + 1] = ((int32_t) adgold_buffer[c] + (int32_t) adgold_buffer[c + 1]) / 2;
adgold->opl_buffer[c] = adgold->opl_buffer[c + 1] = ((int32_t) adgold->opl_buffer[c] + (int32_t) adgold->opl_buffer[c + 1]) / 2;
break;
case 0x08: /*Linear stereo*/
break;
@@ -829,17 +826,17 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
/*Filter left channel, leave right channel unchanged*/
/*Filter cutoff is largely a guess*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] += adgold_pseudo_stereo_iir(0, adgold_buffer[c]);
adgold->opl_buffer[c] += adgold_pseudo_stereo_iir(0, adgold->opl_buffer[c]);
break;
case 0x18: /*Spatial stereo*/
/*Quite probably wrong, I only have the diagram in the TDA8425 datasheet
and a very vague understanding of how op-amps work to go on*/
for (c = 0; c < len * 2; c += 2) {
int16_t l = adgold_buffer[c];
int16_t r = adgold_buffer[c + 1];
int16_t l = adgold->opl_buffer[c];
int16_t r = adgold->opl_buffer[c + 1];
adgold_buffer[c] += (r / 3) + ((l * 2) / 3);
adgold_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
adgold->opl_buffer[c] += (r / 3) + ((l * 2) / 3);
adgold->opl_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
}
break;
@@ -853,7 +850,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
int32_t highpass;
/*Output is deliberately halved to avoid clipping*/
temp = ((int32_t) adgold_buffer[c] * adgold->vol_l) >> 18;
temp = ((int32_t) adgold->opl_buffer[c] * adgold->vol_l) >> 18;
lowpass = adgold_lowpass_iir(0, 0, temp);
highpass = adgold_highpass_iir(0, 0, temp);
if (adgold->bass > 6)
@@ -870,7 +867,7 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
temp = 32767;
buffer[c] += temp;
temp = ((int32_t) adgold_buffer[c + 1] * adgold->vol_r) >> 18;
temp = ((int32_t) adgold->opl_buffer[c + 1] * adgold->vol_r) >> 18;
lowpass = adgold_lowpass_iir(0, 1, temp);
highpass = adgold_highpass_iir(0, 1, temp);
if (adgold->bass > 6)
@@ -889,42 +886,36 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
}
adgold->pos = 0;
free(adgold_buffer);
}
static void
adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
{
adgold_t *adgold = (adgold_t *) priv;
int16_t *adgold_buffer = malloc(sizeof(int16_t) * len * 2);
if (adgold_buffer == NULL)
fatal("adgold_buffer = NULL");
int c;
const int32_t *opl_buf = adgold->opl.update(adgold->opl.priv);
for (c = 0; c < len * 2; c += 2) {
adgold_buffer[c] = ((opl_buf[c] * adgold->fm_vol_l) >> 7) / 2;
adgold_buffer[c + 1] = ((opl_buf[c + 1] * adgold->fm_vol_r) >> 7) / 2;
adgold->opl_buffer[c] = ((opl_buf[c] * adgold->fm_vol_l) >> 7) / 2;
adgold->opl_buffer[c + 1] = ((opl_buf[c + 1] * adgold->fm_vol_r) >> 7) / 2;
}
if (adgold->surround_enabled)
ym7128_apply(&adgold->ym7128, adgold_buffer, len);
ym7128_apply(&adgold->ym7128, adgold->opl_buffer, len);
switch (adgold->adgold_38x_regs[0x8] & 6) {
case 0:
for (c = 0; c < len * 2; c++)
adgold_buffer[c] = 0;
adgold->opl_buffer[c] = 0;
break;
case 2: /*Left channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c + 1] = adgold_buffer[c];
adgold->opl_buffer[c + 1] = adgold->opl_buffer[c];
break;
case 4: /*Right channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] = adgold_buffer[c + 1];
adgold->opl_buffer[c] = adgold->opl_buffer[c + 1];
break;
case 6: /*Left and right channels*/
break;
@@ -936,7 +927,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
switch (adgold->adgold_38x_regs[0x8] & 0x18) {
case 0x00: /*Forced mono*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] = adgold_buffer[c + 1] = ((int32_t) adgold_buffer[c] + (int32_t) adgold_buffer[c + 1]) / 2;
adgold->opl_buffer[c] = adgold->opl_buffer[c + 1] = ((int32_t) adgold->opl_buffer[c] + (int32_t) adgold->opl_buffer[c + 1]) / 2;
break;
case 0x08: /*Linear stereo*/
break;
@@ -944,17 +935,17 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
/*Filter left channel, leave right channel unchanged*/
/*Filter cutoff is largely a guess*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] += adgold_pseudo_stereo_iir(1, adgold_buffer[c]);
adgold->opl_buffer[c] += adgold_pseudo_stereo_iir(1, adgold->opl_buffer[c]);
break;
case 0x18: /*Spatial stereo*/
/*Quite probably wrong, I only have the diagram in the TDA8425 datasheet
and a very vague understanding of how op-amps work to go on*/
for (c = 0; c < len * 2; c += 2) {
int16_t l = adgold_buffer[c];
int16_t r = adgold_buffer[c + 1];
int16_t l = adgold->opl_buffer[c];
int16_t r = adgold->opl_buffer[c + 1];
adgold_buffer[c] += (r / 3) + ((l * 2) / 3);
adgold_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
adgold->opl_buffer[c] += (r / 3) + ((l * 2) / 3);
adgold->opl_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
}
break;
@@ -968,7 +959,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
int32_t highpass;
/*Output is deliberately halved to avoid clipping*/
temp = ((int32_t) adgold_buffer[c] * adgold->vol_l) >> 18;
temp = ((int32_t) adgold->opl_buffer[c] * adgold->vol_l) >> 18;
lowpass = adgold_lowpass_iir(1, 0, temp);
highpass = adgold_highpass_iir(1, 0, temp);
if (adgold->bass > 6)
@@ -985,7 +976,7 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
temp = 32767;
buffer[c] += temp;
temp = ((int32_t) adgold_buffer[c + 1] * adgold->vol_r) >> 18;
temp = ((int32_t) adgold->opl_buffer[c + 1] * adgold->vol_r) >> 18;
lowpass = adgold_lowpass_iir(1, 1, temp);
highpass = adgold_highpass_iir(1, 1, temp);
if (adgold->bass > 6)
@@ -1004,8 +995,6 @@ adgold_get_music_buffer(int32_t *buffer, int len, void *priv)
}
adgold->opl.reset_buffer(adgold->opl.priv);
free(adgold_buffer);
}
static void