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Merge branch 'cps2' of https://github.com/superctr/furnace into superctr-cps2

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tildearrow 2022-02-22 18:21:57 -05:00
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/*
Capcom DL-1425 QSound emulator
==============================
by superctr (Ian Karlsson)
with thanks to Valley Bell
2018-05-12 - 2018-05-15
*/
#include <string.h>
#include <stdint.h>
#include <stdlib.h>
#include <math.h>
#include "qsound.h"
#define CLAMP(x, low, high) (((x) > (high)) ? (high) : (((x) < (low)) ? (low) : (x)))
// ============================================================================
static const int16_t qsound_dry_mix_table[33] = {
-16384,-16384,-16384,-16384,-16384,-16384,-16384,-16384,
-16384,-16384,-16384,-16384,-16384,-16384,-16384,-16384,
-16384,-14746,-13107,-11633,-10486,-9175,-8520,-7209,
-6226,-5226,-4588,-3768,-3277,-2703,-2130,-1802,
0
};
static const int16_t qsound_wet_mix_table[33] = {
0,-1638,-1966,-2458,-2949,-3441,-4096,-4669,
-4915,-5120,-5489,-6144,-7537,-8831,-9339,-9830,
-10240,-10322,-10486,-10568,-10650,-11796,-12288,-12288,
-12534,-12648,-12780,-12829,-12943,-13107,-13418,-14090,
-16384
};
static const int16_t qsound_linear_mix_table[33] = {
-16379,-16338,-16257,-16135,-15973,-15772,-15531,-15251,
-14934,-14580,-14189,-13763,-13303,-12810,-12284,-11729,
-11729,-11144,-10531,-9893,-9229,-8543,-7836,-7109,
-6364,-5604,-4829,-4043,-3246,-2442,-1631,-817,
0
};
static const int16_t qsound_filter_data[5][95] = {
{ // d53 - 0
0,0,0,6,44,-24,-53,-10,59,-40,-27,1,39,-27,56,127,174,36,-13,49,
212,142,143,-73,-20,66,-108,-117,-399,-265,-392,-569,-473,-71,95,-319,-218,-230,331,638,
449,477,-180,532,1107,750,9899,3828,-2418,1071,-176,191,-431,64,117,-150,-274,-97,-238,165,
166,250,-19,4,37,204,186,-6,140,-77,-1,1,18,-10,-151,-149,-103,-9,55,23,
-102,-97,-11,13,-48,-27,5,18,-61,-30,64,72,0,0,0,
},
{ // db2 - 1 - default left filter
0,0,0,85,24,-76,-123,-86,-29,-14,-20,-7,6,-28,-87,-89,-5,100,154,160,
150,118,41,-48,-78,-23,59,83,-2,-176,-333,-344,-203,-66,-39,2,224,495,495,280,
432,1340,2483,5377,1905,658,0,97,347,285,35,-95,-78,-82,-151,-192,-171,-149,-147,-113,
-22,71,118,129,127,110,71,31,20,36,46,23,-27,-63,-53,-21,-19,-60,-92,-69,
-12,25,29,30,40,41,29,30,46,39,-15,-74,0,0,0,
},
{ // e11 - 2 - default right filter
0,0,0,23,42,47,29,10,2,-14,-54,-92,-93,-70,-64,-77,-57,18,94,113,
87,69,67,50,25,29,58,62,24,-39,-131,-256,-325,-234,-45,58,78,223,485,496,
127,6,857,2283,2683,4928,1328,132,79,314,189,-80,-90,35,-21,-186,-195,-99,-136,-258,
-189,82,257,185,53,41,84,68,38,63,77,14,-60,-71,-71,-120,-151,-84,14,29,
-8,7,66,69,12,-3,54,92,52,-6,-15,-2,0,0,0,
},
{ // e70 - 3
0,0,0,2,-28,-37,-17,0,-9,-22,-3,35,52,39,20,7,-6,2,55,121,
129,67,8,1,9,-6,-16,16,66,96,118,130,75,-47,-92,43,223,239,151,219,
440,475,226,206,940,2100,2663,4980,865,49,-33,186,231,103,42,114,191,184,116,29,
-47,-72,-21,60,96,68,31,32,63,87,76,39,7,14,55,85,67,18,-12,-3,
21,34,29,6,-27,-49,-37,-2,16,0,-21,-16,0,0,0,
},
{ // ecf - 4
0,0,0,48,7,-22,-29,-10,24,54,59,29,-36,-117,-185,-213,-185,-99,13,90,
83,24,-5,23,53,47,38,56,67,57,75,107,16,-242,-440,-355,-120,-33,-47,152,
501,472,-57,-292,544,1937,2277,6145,1240,153,47,200,152,36,64,134,74,-82,-208,-266,
-268,-188,-42,65,74,56,89,133,114,44,-3,-1,17,29,29,-2,-76,-156,-187,-151,
-85,-31,-5,7,20,32,24,-5,-20,6,48,62,0,0,0,
}
};
static const int16_t qsound_filter_data2[209] = {
// f2e - following 95 values used for "disable output" filter
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,
// f73 - following 45 values used for "mode 2" filter (overlaps with f2e)
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,
-371,-196,-268,-512,-303,-315,-184,-76,276,-256,298,196,990,236,1114,-126,4377,6549,791,
// fa0 - filtering disabled (for 95-taps) (use fa3 or fa4 for mode2 filters)
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,-16384,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,
0,0,0,0,0,0,0,0,0,0,0,0,0,0,0
};
static const int16_t adpcm_step_table[16] = {
154, 154, 128, 102, 77, 58, 58, 58,
58, 58, 58, 58, 77, 102, 128, 154
};
// DSP states
enum {
STATE_INIT1 = 0x288,
STATE_INIT2 = 0x61a,
STATE_REFRESH1 = 0x039,
STATE_REFRESH2 = 0x04f,
STATE_NORMAL1 = 0x314,
STATE_NORMAL2 = 0x6b2,
};
enum {
PANTBL_LEFT = 0,
PANTBL_RIGHT = 1,
PANTBL_DRY = 0,
PANTBL_WET = 1,
};
static void init_pan_tables(struct qsound_chip *chip);
static void init_register_map(struct qsound_chip *chip);
static void state_init(struct qsound_chip *chip);
static void state_refresh_filter_1(struct qsound_chip *chip);
static void state_refresh_filter_2(struct qsound_chip *chip);
static void state_normal_update(struct qsound_chip *chip);
static inline int16_t get_sample(struct qsound_chip *chip, uint16_t bank,uint16_t address);
static inline int16_t* get_filter_table(struct qsound_chip *chip, uint16_t offset);
static inline int16_t pcm_update(struct qsound_chip *chip, int voice_no, int32_t *echo_out);
static inline void adpcm_update(struct qsound_chip *chip, int voice_no, int nibble);
static inline int16_t echo(struct qsound_echo *r,int32_t input);
static inline int32_t fir(struct qsound_fir *f, int16_t input);
static inline int32_t delay(struct qsound_delay *d, int32_t input);
static inline void delay_update(struct qsound_delay *d);
// ============================================================================
long qsound_start(struct qsound_chip *chip, int clock)
{
memset(chip,0,sizeof(*chip));
init_pan_tables(chip);
init_register_map(chip);
return clock / 2 / 1248; // DSP program uses 1248 machine cycles per iteration
}
void qsound_reset(struct qsound_chip *chip)
{
chip->ready_flag = 0;
chip->out[0] = chip->out[1] = 0;
chip->state = 0;
chip->state_counter = 0;
}
uint8_t qsound_stream_update(struct qsound_chip *chip, int16_t **outputs, int samples)
{
// Clear the buffers
memset(outputs[0], 0, samples * sizeof(*outputs[0]));
memset(outputs[1], 0, samples * sizeof(*outputs[1]));
for (int i = 0; i < samples; i ++)
{
qsound_update(chip);
outputs[0][i] = chip->out[0];
outputs[1][i] = chip->out[1];
}
return 0;
}
uint8_t qsound_w(struct qsound_chip *chip, uint8_t offset, uint8_t data)
{
switch (offset)
{
case 0:
chip->data_latch = (chip->data_latch & 0x00ff) | (data << 8);
break;
case 1:
chip->data_latch = (chip->data_latch & 0xff00) | data;
break;
case 2:
qsound_write_data(chip, data, chip->data_latch);
break;
default:
break;
}
return 0;
}
uint8_t qsound_r(struct qsound_chip *chip)
{
// ready bit (0x00 = busy, 0x80 == ready)
return chip->ready_flag;
}
void qsound_write_data(struct qsound_chip *chip, uint8_t address, uint16_t data)
{
uint16_t *destination = chip->register_map[address];
if(destination)
*destination = data;
chip->ready_flag = 0;
}
uint16_t qsound_read_data(struct qsound_chip *chip, uint8_t address)
{
uint16_t data = 0;
uint16_t *source = chip->register_map[address];
if(source)
data = *source;
return data;
}
// ============================================================================
static void init_pan_tables(struct qsound_chip *chip)
{
int i;
for(i=0;i<33;i++)
{
// dry mixing levels
chip->pan_tables[PANTBL_LEFT][PANTBL_DRY][i] = qsound_dry_mix_table[i];
chip->pan_tables[PANTBL_RIGHT][PANTBL_DRY][i] = qsound_dry_mix_table[32-i];
// wet mixing levels
chip->pan_tables[PANTBL_LEFT][PANTBL_WET][i] = qsound_wet_mix_table[i];
chip->pan_tables[PANTBL_RIGHT][PANTBL_WET][i] = qsound_wet_mix_table[32-i];
// linear panning, only for dry component. wet component is muted.
chip->pan_tables[PANTBL_LEFT][PANTBL_DRY][i+0x30] = qsound_linear_mix_table[i];
chip->pan_tables[PANTBL_RIGHT][PANTBL_DRY][i+0x30] = qsound_linear_mix_table[32-i];
}
}
static void init_register_map(struct qsound_chip *chip)
{
int i;
// unused registers
for(i=0;i<256;i++)
chip->register_map[i] = NULL;
// PCM registers
for(i=0;i<16;i++) // PCM voices
{
chip->register_map[(i<<3)+0] = (uint16_t*)&chip->voice[(i+1)%16].bank; // Bank applies to the next channel
chip->register_map[(i<<3)+1] = (uint16_t*)&chip->voice[i].addr; // Current sample position and start position.
chip->register_map[(i<<3)+2] = (uint16_t*)&chip->voice[i].rate; // 4.12 fixed point decimal.
chip->register_map[(i<<3)+3] = (uint16_t*)&chip->voice[i].phase;
chip->register_map[(i<<3)+4] = (uint16_t*)&chip->voice[i].loop_len;
chip->register_map[(i<<3)+5] = (uint16_t*)&chip->voice[i].end_addr;
chip->register_map[(i<<3)+6] = (uint16_t*)&chip->voice[i].volume;
chip->register_map[(i<<3)+7] = NULL; // unused
chip->register_map[i+0x80] = (uint16_t*)&chip->voice_pan[i];
chip->register_map[i+0xba] = (uint16_t*)&chip->voice[i].echo;
}
// ADPCM registers
for(i=0;i<3;i++) // ADPCM voices
{
// ADPCM sample rate is fixed to 8khz. (one channel is updated every third sample)
chip->register_map[(i<<2)+0xca] = (uint16_t*)&chip->adpcm[i].start_addr;
chip->register_map[(i<<2)+0xcb] = (uint16_t*)&chip->adpcm[i].end_addr;
chip->register_map[(i<<2)+0xcc] = (uint16_t*)&chip->adpcm[i].bank;
chip->register_map[(i<<2)+0xcd] = (uint16_t*)&chip->adpcm[i].volume;
chip->register_map[i+0xd6] = (uint16_t*)&chip->adpcm[i].flag; // non-zero to start ADPCM playback
chip->register_map[i+0x90] = (uint16_t*)&chip->voice_pan[16+i];
}
// QSound registers
chip->register_map[0x93] = (uint16_t*)&chip->echo.feedback;
chip->register_map[0xd9] = (uint16_t*)&chip->echo.end_pos;
chip->register_map[0xe2] = (uint16_t*)&chip->delay_update; // non-zero to update delays
chip->register_map[0xe3] = (uint16_t*)&chip->next_state;
for(i=0;i<2;i++) // left, right
{
// Wet
chip->register_map[(i<<1)+0xda] = (uint16_t*)&chip->filter[i].table_pos;
chip->register_map[(i<<1)+0xde] = (uint16_t*)&chip->wet[i].delay;
chip->register_map[(i<<1)+0xe4] = (uint16_t*)&chip->wet[i].volume;
// Dry
chip->register_map[(i<<1)+0xdb] = (uint16_t*)&chip->alt_filter[i].table_pos;
chip->register_map[(i<<1)+0xdf] = (uint16_t*)&chip->dry[i].delay;
chip->register_map[(i<<1)+0xe5] = (uint16_t*)&chip->dry[i].volume;
}
}
static inline int16_t get_sample(struct qsound_chip *chip, uint16_t bank,uint16_t address)
{
uint32_t rom_addr;
uint8_t sample_data;
if (! chip->rom_mask)
return 0; // no ROM loaded
if (! (bank & 0x8000))
return 0; // ignore attempts to read from DSP program ROM
bank &= 0x7FFF;
rom_addr = (bank << 16) | (address << 0);
sample_data = chip->rom_data[rom_addr];
return (int16_t)((sample_data << 8) | (sample_data << 0)); // MAME currently expands the 8 bit ROM data to 16 bits this way.
}
static inline int16_t* get_filter_table(struct qsound_chip *chip, uint16_t offset)
{
int index;
if (offset >= 0xf2e && offset < 0xfff)
return (int16_t*)&qsound_filter_data2[offset-0xf2e]; // overlapping filter data
index = (offset-0xd53)/95;
if(index >= 0 && index < 5)
return (int16_t*)&qsound_filter_data[index]; // normal tables
return NULL; // no filter found.
}
/********************************************************************/
// updates one DSP sample
void qsound_update(struct qsound_chip *chip)
{
switch(chip->state)
{
default:
case STATE_INIT1:
case STATE_INIT2:
state_init(chip); return;
case STATE_REFRESH1:
state_refresh_filter_1(chip); return;
case STATE_REFRESH2:
state_refresh_filter_2(chip); return;
case STATE_NORMAL1:
case STATE_NORMAL2:
state_normal_update(chip); return;
}
}
// Initialization routine
static void state_init(struct qsound_chip *chip)
{
int mode = (chip->state == STATE_INIT2) ? 1 : 0;
int i;
// we're busy for 4 samples, including the filter refresh.
if(chip->state_counter >= 2)
{
chip->state_counter = 0;
chip->state = chip->next_state;
return;
}
else if(chip->state_counter == 1)
{
chip->state_counter++;
return;
}
memset(chip->voice, 0, sizeof(chip->voice));
memset(chip->adpcm, 0, sizeof(chip->adpcm));
memset(chip->filter, 0, sizeof(chip->filter));
memset(chip->alt_filter, 0, sizeof(chip->alt_filter));
memset(chip->wet, 0, sizeof(chip->wet));
memset(chip->dry, 0, sizeof(chip->dry));
memset(&chip->echo, 0, sizeof(chip->echo));
for(i=0;i<19;i++)
{
chip->voice_pan[i] = 0x120;
chip->voice_output[i] = 0;
}
for(i=0;i<16;i++)
chip->voice[i].bank = 0x8000;
for(i=0;i<3;i++)
chip->adpcm[i].bank = 0x8000;
if(mode == 0)
{
// mode 1
chip->wet[0].delay = 0;
chip->dry[0].delay = 46;
chip->wet[1].delay = 0;
chip->dry[1].delay = 48;
chip->filter[0].table_pos = 0xdb2;
chip->filter[1].table_pos = 0xe11;
chip->echo.end_pos = 0x554 + 6;
chip->next_state = STATE_REFRESH1;
}
else
{
// mode 2
chip->wet[0].delay = 1;
chip->dry[0].delay = 0;
chip->wet[1].delay = 0;
chip->dry[1].delay = 0;
chip->filter[0].table_pos = 0xf73;
chip->filter[1].table_pos = 0xfa4;
chip->alt_filter[0].table_pos = 0xf73;
chip->alt_filter[1].table_pos = 0xfa4;
chip->echo.end_pos = 0x53c + 6;
chip->next_state = STATE_REFRESH2;
}
chip->wet[0].volume = 0x3fff;
chip->dry[0].volume = 0x3fff;
chip->wet[1].volume = 0x3fff;
chip->dry[1].volume = 0x3fff;
chip->delay_update = 1;
chip->ready_flag = 0;
chip->state_counter = 1;
}
// Updates filter parameters for mode 1
static void state_refresh_filter_1(struct qsound_chip *chip)
{
const int16_t *table;
for(int ch=0; ch<2; ch++)
{
chip->filter[ch].delay_pos = 0;
chip->filter[ch].tap_count = 95;
table = get_filter_table(chip,chip->filter[ch].table_pos);
if (table != NULL)
memcpy(chip->filter[ch].taps, table, 95 * sizeof(int16_t));
}
chip->state = chip->next_state = STATE_NORMAL1;
}
// Updates filter parameters for mode 2
static void state_refresh_filter_2(struct qsound_chip *chip)
{
const int16_t *table;
for(int ch=0; ch<2; ch++)
{
chip->filter[ch].delay_pos = 0;
chip->filter[ch].tap_count = 45;
table = get_filter_table(chip,chip->filter[ch].table_pos);
if (table != NULL)
memcpy(chip->filter[ch].taps, table, 45 * sizeof(int16_t));
chip->alt_filter[ch].delay_pos = 0;
chip->alt_filter[ch].tap_count = 44;
table = get_filter_table(chip,chip->alt_filter[ch].table_pos);
if (table != NULL)
memcpy(chip->alt_filter[ch].taps, table, 44 * sizeof(int16_t));
}
chip->state = chip->next_state = STATE_NORMAL2;
}
// Updates a PCM voice. There are 16 voices, each are updated every sample
// with full rate and volume control.
static inline int16_t pcm_update(struct qsound_chip *chip, int voice_no, int32_t *echo_out)
{
struct qsound_voice* v = &chip->voice[voice_no];
int32_t new_phase;
int16_t output = 0;
if(!(chip->mute_mask & (1<<voice_no)))
{
// Read sample from rom and apply volume
output = (v->volume * get_sample(chip, v->bank, v->addr))>>14;
*echo_out += (output * v->echo)<<2;
}
// Add delta to the phase and loop back if required
new_phase = v->rate + ((v->addr<<12) | (v->phase>>4));
if((new_phase>>12) >= v->end_addr)
new_phase -= (v->loop_len<<12);
new_phase = CLAMP(new_phase, -0x8000000, 0x7FFFFFF);
v->addr = new_phase>>12;
v->phase = (new_phase<<4)&0xffff;
return output;
}
// Updates an ADPCM voice. There are 3 voices, one is updated every sample
// (effectively making the ADPCM rate 1/3 of the master sample rate), and
// volume is set when starting samples only.
// The ADPCM algorithm is supposedly similar to Yamaha ADPCM. It also seems
// like Capcom never used it, so this was not emulated in the earlier QSound
// emulators.
static inline void adpcm_update(struct qsound_chip *chip, int voice_no, int nibble)
{
struct qsound_adpcm *v = &chip->adpcm[voice_no];
int32_t delta;
int8_t step;
if(!nibble)
{
// Mute voice when it reaches the end address.
if(v->cur_addr == v->end_addr)
v->cur_vol = 0;
// Playback start flag
if(v->flag)
{
chip->voice_output[16+voice_no] = 0;
v->flag = 0;
v->step_size = 10;
v->cur_vol = v->volume;
v->cur_addr = v->start_addr;
}
// get top nibble
step = get_sample(chip, v->bank, v->cur_addr) >> 8;
}
else
{
// get bottom nibble
step = get_sample(chip, v->bank, v->cur_addr++) >> 4;
}
// shift with sign extend
step >>= 4;
// delta = (0.5 + abs(v->step)) * v->step_size
delta = ((1+abs(step<<1)) * v->step_size)>>1;
if(step <= 0)
delta = -delta;
delta += chip->voice_output[16+voice_no];
delta = CLAMP(delta,-32768,32767);
if(chip->mute_mask & (1<<(16+voice_no)))
chip->voice_output[16+voice_no] = 0;
else
chip->voice_output[16+voice_no] = (delta * v->cur_vol)>>16;
v->step_size = (adpcm_step_table[8+step] * v->step_size) >> 6;
v->step_size = CLAMP(v->step_size, 1, 2000);
}
// The echo effect is pretty simple. A moving average filter is used on
// the output from the delay line to smooth samples over time.
static inline int16_t echo(struct qsound_echo *r,int32_t input)
{
// get average of last 2 samples from the delay line
int32_t new_sample;
int32_t old_sample = r->delay_line[r->delay_pos];
int32_t last_sample = r->last_sample;
r->last_sample = old_sample;
old_sample = (old_sample+last_sample) >> 1;
// add current sample to the delay line
new_sample = input + ((old_sample * r->feedback)<<2);
r->delay_line[r->delay_pos++] = new_sample>>16;
if(r->delay_pos >= r->length)
r->delay_pos = 0;
return old_sample;
}
// Process a sample update
static void state_normal_update(struct qsound_chip *chip)
{
int v, ch;
int32_t echo_input = 0;
int16_t echo_output;
chip->ready_flag = 0x80;
// recalculate echo length
if(chip->state == STATE_NORMAL2)
chip->echo.length = chip->echo.end_pos - 0x53c;
else
chip->echo.length = chip->echo.end_pos - 0x554;
chip->echo.length = CLAMP(chip->echo.length, 0, 1024);
// update PCM voices
for(v=0; v<16; v++)
{
chip->voice_output[v] = pcm_update(chip, v, &echo_input);
}
// update ADPCM voices (one every third sample)
adpcm_update(chip, chip->state_counter % 3, chip->state_counter / 3);
echo_output = echo(&chip->echo,echo_input);
// now, we do the magic stuff
for(ch=0; ch<2; ch++)
{
// Echo is output on the unfiltered component of the left channel and
// the filtered component of the right channel.
int32_t wet = (ch == 1) ? echo_output<<16 : 0;
int32_t dry = (ch == 0) ? echo_output<<16 : 0;
int32_t output = 0;
for(int v=0; v<19; v++)
{
uint16_t pan_index = chip->voice_pan[v]-0x110;
if(pan_index > 97)
pan_index = 97;
// Apply different volume tables on the dry and wet inputs.
dry -= (chip->voice_output[v] * chip->pan_tables[ch][PANTBL_DRY][pan_index])<<2;
wet -= (chip->voice_output[v] * chip->pan_tables[ch][PANTBL_WET][pan_index])<<2;
}
// Apply FIR filter on 'wet' input
wet = fir(&chip->filter[ch], wet >> 16);
// in mode 2, we do this on the 'dry' input too
if(chip->state == STATE_NORMAL2)
dry = fir(&chip->alt_filter[ch], dry >> 16);
// output goes through a delay line and attenuation
output = (delay(&chip->wet[ch], wet) + delay(&chip->dry[ch], dry));
// DSP round function
output = (output + 0x2000) >> 14;
chip->out[ch] = CLAMP(output, -0x7fff, 0x7fff);
if(chip->delay_update)
{
delay_update(&chip->wet[ch]);
delay_update(&chip->dry[ch]);
}
}
chip->delay_update = 0;
// after 6 samples, the next state is executed.
chip->state_counter++;
if(chip->state_counter > 5)
{
chip->state_counter = 0;
chip->state = chip->next_state;
}
}
// Apply the FIR filter used as the Q1 transfer function
static inline int32_t fir(struct qsound_fir *f, int16_t input)
{
int32_t output = 0, tap = 0;
for(; tap < (f->tap_count-1); tap++)
{
output -= (f->taps[tap] * f->delay_line[f->delay_pos++])<<2;
if(f->delay_pos >= f->tap_count-1)
f->delay_pos = 0;
}
output -= (f->taps[tap] * input)<<2;
f->delay_line[f->delay_pos++] = input;
if(f->delay_pos >= f->tap_count-1)
f->delay_pos = 0;
return output;
}
// Apply delay line and component volume
static inline int32_t delay(struct qsound_delay *d, int32_t input)
{
int32_t output;
d->delay_line[d->write_pos++] = input>>16;
if(d->write_pos >= 51)
d->write_pos = 0;
output = d->delay_line[d->read_pos++]*d->volume;
if(d->read_pos >= 51)
d->read_pos = 0;
return output;
}
// Update the delay read position to match new delay length
static inline void delay_update(struct qsound_delay *d)
{
int16_t new_read_pos = (d->write_pos - d->delay) % 51;
if(new_read_pos < 0)
new_read_pos += 51;
d->read_pos = new_read_pos;
}

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src/engine/platform/sound/qsound.h Normal file
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#ifndef QSOUND_H
#define QSOUND_H
#ifdef __cplusplus
extern "C" {
#endif
/*
Capcom DL-1425 QSound emulator
==============================
by superctr (Ian Karlsson)
with thanks to Valley Bell
2018-05-12 - 2018-05-15
*/
#include <stdint.h>
struct qsound_voice {
uint16_t bank;
int16_t addr; // top word is the sample address
uint16_t phase;
uint16_t rate;
int16_t loop_len;
int16_t end_addr;
int16_t volume;
int16_t echo;
};
struct qsound_adpcm {
uint16_t start_addr;
uint16_t end_addr;
uint16_t bank;
int16_t volume;
uint16_t flag;
int16_t cur_vol;
int16_t step_size;
uint16_t cur_addr;
};
// Q1 Filter
struct qsound_fir {
int tap_count; // usually 95
int delay_pos;
int16_t table_pos;
int16_t taps[95];
int16_t delay_line[95];
};
// Delay line
struct qsound_delay {
int16_t delay;
int16_t volume;
int16_t write_pos;
int16_t read_pos;
int16_t delay_line[51];
};
struct qsound_echo {
uint16_t end_pos;
int16_t feedback;
int16_t length;
int16_t last_sample;
int16_t delay_line[1024];
int16_t delay_pos;
};
struct qsound_chip {
unsigned long rom_mask;
uint8_t *rom_data;
uint32_t mute_mask;
uint16_t data_latch;
int16_t out[2];
int16_t pan_tables[2][2][98];
struct qsound_voice voice[16];
struct qsound_adpcm adpcm[3];
uint16_t voice_pan[16+3];
int16_t voice_output[16+3];
struct qsound_echo echo;
struct qsound_fir filter[2];
struct qsound_fir alt_filter[2];
struct qsound_delay wet[2];
struct qsound_delay dry[2];
uint16_t state;
uint16_t next_state;
uint16_t delay_update;
int state_counter;
int ready_flag;
uint16_t *register_map[256];
};
long qsound_start(struct qsound_chip *chip, int clock);
void qsound_reset(struct qsound_chip *chip);
void qsound_update(struct qsound_chip *chip);
uint8_t qsound_stream_update(struct qsound_chip *chip, int16_t **outputs, int samples);
uint8_t qsound_w(struct qsound_chip *chip, uint8_t offset, uint8_t data);
uint8_t qsound_r(struct qsound_chip *chip);
void qsound_write_data(struct qsound_chip *chip, uint8_t address, uint16_t data);
uint16_t qsound_read_data(struct qsound_chip *chip, uint8_t address);
#ifdef __cplusplus
};
#endif
#endif