claunia/86Box
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Moved OPL2 and OPL3 to a new 49716 Hz source so resampling is no longer needed, also fixed SB OPL and PC Speaker filtering (OPL was being downsampled to the selected DSP sample rate, which is incorrect, and the PC Speaker filter was using the wrong filter index in some liens).

Browse Source
This commit is contained in:
OBattler
2024-03-01 06:52:48 +01:00
parent 71ecdc1b55
commit e0d80aefb4
18 changed files with 683 additions and 193 deletions
Download Patch File Download Diff File Expand all files Collapse all files

130
src/sound/snd_adlibgold.c
View File

@@ -788,13 +788,10 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
int c;
const int32_t *opl_buf = adgold->opl.update(adgold->opl.priv);
adgold_update(adgold);
for (c = 0; c < len * 2; c += 2) {
adgold_buffer[c] = ((opl_buf[c] * adgold->fm_vol_l) >> 7) / 2;
adgold_buffer[c] += ((adgold->mma_buffer[0][c >> 1] * adgold->samp_vol_l) >> 7) / 4;
adgold_buffer[c + 1] = ((opl_buf[c + 1] * adgold->fm_vol_r) >> 7) / 2;
adgold_buffer[c + 1] += ((adgold->mma_buffer[1][c >> 1] * adgold->samp_vol_r) >> 7) / 4;
}
@@ -857,8 +854,8 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
/*Output is deliberately halved to avoid clipping*/
temp = ((int32_t) adgold_buffer[c] * adgold->vol_l) >> 17;
lowpass = adgold_lowpass_iir(0, temp);
highpass = adgold_highpass_iir(0, temp);
lowpass = adgold_lowpass_iir(0, 0, temp);
highpass = adgold_highpass_iir(0, 0, temp);
if (adgold->bass > 6)
temp += (lowpass * bass_attenuation[adgold->bass]) >> 14;
else if (adgold->bass < 6)
@@ -874,8 +871,124 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
buffer[c] += temp;
temp = ((int32_t) adgold_buffer[c + 1] * adgold->vol_r) >> 17;
lowpass = adgold_lowpass_iir(1, temp);
highpass = adgold_highpass_iir(1, temp);
lowpass = adgold_lowpass_iir(0, 1, temp);
highpass = adgold_highpass_iir(0, 1, temp);
if (adgold->bass > 6)
temp += (lowpass * bass_attenuation[adgold->bass]) >> 14;
else if (adgold->bass < 6)
temp = highpass + ((temp * bass_cut[adgold->bass]) >> 14);
if (adgold->treble > 6)
temp += (highpass * treble_attenuation[adgold->treble]) >> 14;
else if (adgold->treble < 6)
temp = lowpass + ((temp * treble_cut[adgold->treble]) >> 14);
if (temp < -32768)
temp = -32768;
if (temp > 32767)
temp = 32767;
buffer[c + 1] += temp;
}
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);
adgold_update(adgold);
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;
}
if (adgold->surround_enabled)
ym7128_apply(&adgold->ym7128, adgold_buffer, len);
switch (adgold->adgold_38x_regs[0x8] & 6) {
case 0:
for (c = 0; c < len * 2; c++)
adgold_buffer[c] = 0;
break;
case 2: /*Left channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c + 1] = adgold_buffer[c];
break;
case 4: /*Right channel only*/
for (c = 0; c < len * 2; c += 2)
adgold_buffer[c] = adgold_buffer[c + 1];
break;
case 6: /*Left and right channels*/
break;
default:
break;
}
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;
break;
case 0x08: /*Linear stereo*/
break;
case 0x10: /*Pseudo stereo*/
/*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(adgold_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];
adgold_buffer[c] += (r / 3) + ((l * 2) / 3);
adgold_buffer[c + 1] += (l / 3) + ((r * 2) / 3);
}
break;
default:
break;
}
for (c = 0; c < len * 2; c += 2) {
int32_t temp;
int32_t lowpass;
int32_t highpass;
/*Output is deliberately halved to avoid clipping*/
temp = ((int32_t) adgold_buffer[c] * adgold->vol_l) >> 17;
lowpass = adgold_lowpass_iir(1, 0, temp);
highpass = adgold_highpass_iir(1, 0, temp);
if (adgold->bass > 6)
temp += (lowpass * bass_attenuation[adgold->bass]) >> 14;
else if (adgold->bass < 6)
temp = highpass + ((temp * bass_cut[adgold->bass]) >> 14);
if (adgold->treble > 6)
temp += (highpass * treble_attenuation[adgold->treble]) >> 14;
else if (adgold->treble < 6)
temp = lowpass + ((temp * treble_cut[adgold->treble]) >> 14);
if (temp < -32768)
temp = -32768;
if (temp > 32767)
temp = 32767;
buffer[c] += temp;
temp = ((int32_t) adgold_buffer[c + 1] * adgold->vol_r) >> 17;
lowpass = adgold_lowpass_iir(1, 1, temp);
highpass = adgold_highpass_iir(1, 1, temp);
if (adgold->bass > 6)
temp += (lowpass * bass_attenuation[adgold->bass]) >> 14;
else if (adgold->bass < 6)
@@ -892,7 +1005,6 @@ adgold_get_buffer(int32_t *buffer, int len, void *priv)
}
adgold->opl.reset_buffer(adgold->opl.priv);
adgold->pos = 0;
free(adgold_buffer);
}
@@ -1054,6 +1166,8 @@ adgold_init(UNUSED(const device_t *info))
timer_add(&adgold->adgold_mma_timer_count, adgold_timer_poll, adgold, 1);
sound_add_handler(adgold_get_buffer, adgold);
music_add_handler(adgold_get_music_buffer, adgold);
sound_set_cd_audio_filter(adgold_filter_cd_audio, adgold);
if (device_get_config_int("receive_input"))