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Mixer functions; recording (incomplete/commented out); set default IRQ to 5

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Kagamiin~
2024-03-12 18:12:57 -03:00
parent 7ad48f8d29
commit 0ed203cbd5
2 changed files with 188 additions and 39 deletions
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2
src/include/86box/snd_sb.h
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@@ -112,6 +112,8 @@ typedef struct sb_ct1745_mixer_t {
#define INPUT_LINE_L 16
#define INPUT_MIDI_R 32
#define INPUT_MIDI_L 64
#define INPUT_MIXER_L 128
#define INPUT_MIXER_R 256
int mic_agc;

203
src/sound/snd_ess.c
View File

@@ -21,6 +21,7 @@
* Copyright 2024 Cacodemon345
* Copyright 2024 Kagamiin~
*/
#include <stdarg.h>
#include <stdint.h>
#include <stdio.h>
@@ -48,6 +49,8 @@
#include <86box/snd_sb.h>
#include <86box/plat_unused.h>
static const double sb_att_4dbstep_3bits[] = {
164.0, 2067.0, 3276.0, 5193.0, 8230.0, 13045.0, 20675.0, 32767.0
};
@@ -56,6 +59,11 @@ static const double sb_att_7dbstep_2bits[] = {
164.0, 6537.0, 14637.0, 32767.0
};
static const double sb_att_1p4dbstep_4bits[] = {
164.0, 3431.0, 4031.0, 4736.0, 5565.0, 6537.0, 7681.0, 9025.0,
10603.0, 12458.0, 14637.0, 17196.0, 20204.0, 23738.0, 27889.0, 32767.0
};
/* SB PRO */
typedef struct ess_mixer_t {
double master_l;
@@ -68,7 +76,10 @@ typedef struct ess_mixer_t {
double cd_r;
double line_l;
double line_r;
double mic;
double mic_l;
double mic_r;
double auxb_l;
double auxb_r;
/*see sb_ct1745_mixer for values for input selector*/
int32_t input_selector;
@@ -84,6 +95,12 @@ typedef struct ess_mixer_t {
uint8_t ess_id_str[256];
uint8_t ess_id_str_pos;
#if 0
int record_pos_write_cd;
double record_pos_write_cd_sigma;
int record_pos_write_music;
#endif
} ess_mixer_t;
typedef struct ess_t {
@@ -167,10 +184,56 @@ ess_mixer_write(uint16_t addr, uint8_t val, void *priv)
mixer->regs[mixer->index + 0x20] = ((val & 0xe) << 4) | (val & 0xe);
break;
case 0x0A:
{
uint8_t mic_vol_2bit = (mixer->regs[0x0a] >> 1) & 0x3;
mixer->mic_l = mixer->mic_r = sb_att_7dbstep_2bits[mic_vol_2bit] / 32768.0;
mixer->regs[0x1A] = mic_vol_2bit | (mic_vol_2bit << 2);
break;
}
case 0x0C:
switch (mixer->regs[0x0C] & 6) {
case 2:
mixer->input_selector = INPUT_CD_L | INPUT_CD_R;
break;
case 6:
mixer->input_selector = INPUT_LINE_L | INPUT_LINE_R;
break;
default:
mixer->input_selector = INPUT_MIC;
break;
}
break;
case 0x14:
mixer->regs[0x4] = val & 0xee;
break;
case 0x1A:
mixer->mic_l = sb_att_1p4dbstep_4bits[(mixer->regs[0x1A] >> 4) & 0xF];
mixer->mic_r = sb_att_1p4dbstep_4bits[mixer->regs[0x1A] & 0xF];
break;
case 0x1C:
if ((mixer->regs[0x1C] & 0x07) == 0x07)
{
mixer->input_selector = INPUT_MIXER_L | INPUT_MIXER_R;
}
else if ((mixer->regs[0x1C] & 0x07) == 0x06)
{
mixer->input_selector = INPUT_LINE_L | INPUT_LINE_R;
}
else if ((mixer->regs[0x1C] & 0x06) == 0x02)
{
mixer->input_selector = INPUT_CD_L | INPUT_CD_R;
}
else if ((mixer->regs[0x1C] & 0x02) == 0)
{
mixer->input_selector = INPUT_MIC;
}
break;
case 0x22:
case 0x26:
case 0x28:
@@ -192,10 +255,13 @@ ess_mixer_write(uint16_t addr, uint8_t val, void *priv)
mixer->regs[mixer->index - 0x10] = (val & 0xee);
break;
case 0x3a:
break;
case 0x00:
case 0x04:
case 0x0a:
case 0x0c:
break;
case 0x0e:
break;
@@ -268,18 +334,18 @@ ess_mixer_write(uint16_t addr, uint8_t val, void *priv)
}
}
mixer->voice_l = ess_mixer_get_vol_4bit(mixer->regs[0x14]);
mixer->voice_r = ess_mixer_get_vol_4bit(mixer->regs[0x14] >> 4);
mixer->master_l = ess_mixer_get_vol_4bit(mixer->regs[0x32]);
mixer->master_r = ess_mixer_get_vol_4bit(mixer->regs[0x32] >> 4);
mixer->fm_l = ess_mixer_get_vol_4bit(mixer->regs[0x36]);
mixer->fm_r = ess_mixer_get_vol_4bit(mixer->regs[0x36] >> 4);
mixer->cd_l = ess_mixer_get_vol_4bit(mixer->regs[0x38]);
mixer->cd_r = ess_mixer_get_vol_4bit(mixer->regs[0x38] >> 4);
mixer->line_l = ess_mixer_get_vol_4bit(mixer->regs[0x3e]);
mixer->line_r = ess_mixer_get_vol_4bit(mixer->regs[0x3e] >> 4);
mixer->mic = sb_att_7dbstep_2bits[(mixer->regs[0x0a] >> 1) & 0x3] / 32768.0;
mixer->voice_l = ess_mixer_get_vol_4bit(mixer->regs[0x14] >> 4);
mixer->voice_r = ess_mixer_get_vol_4bit(mixer->regs[0x14]);
mixer->master_l = ess_mixer_get_vol_4bit(mixer->regs[0x32] >> 4);
mixer->master_r = ess_mixer_get_vol_4bit(mixer->regs[0x32]);
mixer->fm_l = ess_mixer_get_vol_4bit(mixer->regs[0x36] >> 4);
mixer->fm_r = ess_mixer_get_vol_4bit(mixer->regs[0x36]);
mixer->cd_l = ess_mixer_get_vol_4bit(mixer->regs[0x38] >> 4);
mixer->cd_r = ess_mixer_get_vol_4bit(mixer->regs[0x38]);
mixer->line_l = ess_mixer_get_vol_4bit(mixer->regs[0x3e] >> 4);
mixer->line_r = ess_mixer_get_vol_4bit(mixer->regs[0x3e]);
mixer->auxb_l = ess_mixer_get_vol_4bit(mixer->regs[0x3a] >> 4);
mixer->auxb_r = ess_mixer_get_vol_4bit(mixer->regs[0x3a]);
mixer->output_filter = !(mixer->regs[0xe] & 0x20);
mixer->input_filter = !(mixer->regs[0xc] & 0x20);
@@ -288,18 +354,6 @@ ess_mixer_write(uint16_t addr, uint8_t val, void *priv)
if (mixer->index == 0xe)
sb_dsp_set_stereo(&ess->dsp, val & 2);
switch (mixer->regs[0xc] & 6) {
case 2:
mixer->input_selector = INPUT_CD_L | INPUT_CD_R;
break;
case 6:
mixer->input_selector = INPUT_LINE_L | INPUT_LINE_R;
break;
default:
mixer->input_selector = INPUT_MIC;
break;
}
/* TODO: pcspeaker volume? Or is it not worth? */
}
}
@@ -407,6 +461,12 @@ ess_get_music_buffer_sbpro(int32_t *buffer, int len, void *priv)
{
ess_t *ess = (ess_t *) priv;
const ess_mixer_t *mixer = &ess->mixer_sbpro;
#if 0
int rec_pos = ess->mixer_sbpro.record_pos_write_music;
int c_record;
int32_t in_l;
int32_t in_r;
#endif
double out_l = 0.0;
double out_r = 0.0;
const int32_t *opl_buf = NULL;
@@ -430,10 +490,52 @@ ess_get_music_buffer_sbpro(int32_t *buffer, int len, void *priv)
out_l *= mixer->master_l;
out_r *= mixer->master_r;
#if 0
// Pull input after applying mixer's master volume scaling
in_l = (mixer->input_selector & INPUT_MIXER_L) ? ((int32_t) out_l) : 0;
in_r = (mixer->input_selector & INPUT_MIXER_R) ? ((int32_t) out_l) : 0;
if (ess->dsp.sb_enable_i) {
// NOTE: this is nearest-neighbor sampling. This is gonna generate aliasing like HECK. Is this what the real card does?
c_record = rec_pos + ((c * ess->dsp.sb_freq) / MUSIC_FREQ);
ess->dsp.record_buffer[c_record & 0xfffe] += in_l;
ess->dsp.record_buffer[(c_record & 0xfffe) + 1] += in_r;
if (ess->dsp.record_buffer[c_record & 0xfffe] < -32768)
{
ess->dsp.record_buffer[c_record & 0xfffe] = -32768;
}
else if (ess->dsp.record_buffer[c_record & 0xfffe] > 32767)
{
ess->dsp.record_buffer[c_record & 0xfffe] = 32767;
}
if (ess->dsp.record_buffer[(c_record & 0xfffe) + 1] < -32768)
{
ess->dsp.record_buffer[(c_record & 0xfffe) + 1] = -32768;
}
else if (ess->dsp.record_buffer[(c_record & 0xfffe) + 1] > 32767)
{
ess->dsp.record_buffer[(c_record & 0xfffe) + 1] = 32767;
}
}
buffer[c] += (int32_t) out_l;
buffer[c + 1] += (int32_t) out_r;
#endif
}
#if 0
ess->mixer_sbpro.record_pos_write_music += ((len * 2 * ess->dsp.sb_freq) / MUSIC_FREQ);
ess->mixer_sbpro.record_pos_write_music &= 0xfffe;
if (ess->mixer_sbpro.record_pos_write_music < ess->mixer_sbpro.record_pos_write_cd)
{
ess->dsp.record_pos_write = ess->mixer_sbpro.record_pos_write_music;
}
#endif
ess->opl.reset_buffer(ess->opl.priv);
}
@@ -443,11 +545,51 @@ ess_filter_cd_audio(int channel, double *buffer, void *priv)
const ess_t *ess = (ess_t *) priv;
const ess_mixer_t *mixer = &ess->mixer_sbpro;
double c;
#if 0
double rec_pos = ess->mixer_sbpro.record_pos_write_cd;
double rec_pos_sigma = ess->mixer_sbpro.record_pos_write_cd_sigma;
int c_record;
int selector = channel ? INPUT_MIXER_R : INPUT_MIXER_L;
int rec_buf_pos;
int32_t in;
#endif
double cd = channel ? mixer->cd_r : mixer->cd_l;
double master = channel ? mixer->master_r : mixer->master_l;
c = (*buffer * cd) / 3.0;
*buffer = c * master;
#if 0
in = (mixer->input_selector & selector) ? (int32_t)(c * master) : 0;
if (ess->dsp.sb_enable_i)
{
// NOTE: this is nearest-neighbor sampling. This is gonna generate aliasing like HECK. Is this what the real card does?
c_record = (int)(rec_pos + rec_pos_sigma);
rec_buf_pos = channel ? ((c_record & 0xfffe) + 1) : (c_record & 0xfffe);
ess->dsp.record_buffer[rec_buf_pos] += in;
if (ess->dsp.record_buffer[rec_buf_pos] < -32768)
{
ess->dsp.record_buffer[rec_buf_pos] = -32768;
}
else if (ess->dsp.record_buffer[rec_buf_pos] > 32767)
{
ess->dsp.record_buffer[rec_buf_pos] = 32767;
}
}
ess->mixer_sbpro.record_pos_write_cd += ((2 * (double)ess->dsp.sb_freq) / MUSIC_FREQ) + rec_pos_sigma;
ess->mixer_sbpro.record_pos_write_cd &= ~1;
ess->mixer_sbpro.record_pos_write_cd_sigma = (double)rec_pos + ((2 * (double)ess->dsp.sb_freq) / MUSIC_FREQ) + rec_pos_sigma - ess->mixer_sbpro.record_pos_write_cd;
ess->mixer_sbpro.record_pos_write_cd &= 0xfffe;
if (ess->mixer_sbpro.record_pos_write_cd < ess->mixer_sbpro.record_pos_write_music)
{
ess->dsp.record_pos_write = ess->mixer_sbpro.record_pos_write_cd;
}
#endif
}
static void *
@@ -507,6 +649,11 @@ ess_1688_init(UNUSED(const device_t *info))
ess->mixer_sbpro.ess_id_str[3] = addr & 0xff;
}
#if 0
ess->mixer_sbpro.record_pos_write_cd = ess->dsp.record_pos_write;
ess->mixer_sbpro.record_pos_write_music = ess->dsp.record_pos_write;
#endif
ess->mpu = (mpu_t *) calloc(1, sizeof(mpu_t));
mpu401_init(ess->mpu, 0, -1, M_UART, 1);
sb_dsp_set_mpu(&ess->dsp, ess->mpu);
@@ -561,7 +708,7 @@ static const device_config_t ess_config[] = {
.description = "IRQ",
.type = CONFIG_SELECTION,
.default_string = "",
.default_int = 7,
.default_int = 5,
.file_filter = "",
.spinner = { 0 },
.selection = {