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/*
Copyright 2013 Michael Pavone
This file is part of BlastEm.
BlastEm is free software distributed under the terms of the GNU General Public License version 3 or greater. See COPYING for full license text.
*/
#include "psg.h"
#include "render.h"
#include "blastem.h"
#include <string.h>
#include <stdlib.h>
#include <stdio.h>
#include <math.h>
void psg_init(psg_context * context, uint32_t sample_rate, uint32_t master_clock, uint32_t clock_div, uint32_t samples_frame, uint32_t lowpass_cutoff)
{
memset(context, 0, sizeof(*context));
context->audio_buffer = malloc(sizeof(*context->audio_buffer) * samples_frame);
context->back_buffer = malloc(sizeof(*context->audio_buffer) * samples_frame);
context->clock_inc = clock_div;
context->sample_rate = sample_rate;
context->samples_frame = samples_frame;
double rc = (1.0 / (double)lowpass_cutoff) / (2.0 * M_PI);
double dt = 1.0 / ((double)master_clock / (double)clock_div);
double alpha = dt / (dt + rc);
context->lowpass_alpha = (int32_t)(((double)0x10000) * alpha);
psg_adjust_master_clock(context, master_clock);
for (int i = 0; i < 4; i++) {
context->volume[i] = 0xF;
}
}
void psg_free(psg_context *context)
{
free(context->audio_buffer);
//TODO: Figure out how to make this 100% safe
//audio thread could still be using this
free(context->back_buffer);
free(context);
}
#define BUFFER_INC_RES 0x40000000UL
void psg_adjust_master_clock(psg_context * context, uint32_t master_clock)
{
uint64_t old_inc = context->buffer_inc;
context->buffer_inc = ((BUFFER_INC_RES * (uint64_t)context->sample_rate) / (uint64_t)master_clock) * (uint64_t)context->clock_inc;
}
void psg_write(psg_context * context, uint8_t value)
{
if (value & 0x80) {
context->latch = value & 0x70;
uint8_t channel = value >> 5 & 0x3;
if (value & 0x10) {
context->volume[channel] = value & 0xF;
} else {
if (channel == 3) {
switch(value & 0x3)
{
case 0:
case 1:
case 2:
context->counter_load[3] = 0x10 << (value & 0x3);
context->noise_use_tone = 0;
break;
default:
context->counter_load[3] = context->counter_load[2];
context->noise_use_tone = 1;
}
context->noise_type = value & 0x4;
context->lsfr = 0x8000;
} else {
context->counter_load[channel] = (context->counter_load[channel] & 0x3F0) | (value & 0xF);
if (channel == 2 && context->noise_use_tone) {
context->counter_load[3] = context->counter_load[2];
}
}
}
} else {
if (!(context->latch & 0x10)) {
uint8_t channel = context->latch >> 5 & 0x3;
if (channel != 3) {
context->counter_load[channel] = (value << 4 & 0x3F0) | (context->counter_load[channel] & 0xF);
if (channel == 2 && context->noise_use_tone) {
context->counter_load[3] = context->counter_load[2];
}
}
}
}
}
#define PSG_VOL_DIV 14
//table shamelessly swiped from PSG doc from smspower.org
int16_t volume_table[16] = {
32767/PSG_VOL_DIV, 26028/PSG_VOL_DIV, 20675/PSG_VOL_DIV, 16422/PSG_VOL_DIV, 13045/PSG_VOL_DIV, 10362/PSG_VOL_DIV,
8231/PSG_VOL_DIV, 6568/PSG_VOL_DIV, 5193/PSG_VOL_DIV, 4125/PSG_VOL_DIV, 3277/PSG_VOL_DIV, 2603/PSG_VOL_DIV,
2067/PSG_VOL_DIV, 1642/PSG_VOL_DIV, 1304/PSG_VOL_DIV, 0
};
void psg_run(psg_context * context, uint32_t cycles)
{
while (context->cycles < cycles) {
for (int i = 0; i < 4; i++) {
if (context->counters[i]) {
context->counters[i] -= 1;
}
if (!context->counters[i]) {
context->counters[i] = context->counter_load[i];
context->output_state[i] = !context->output_state[i];
if (i == 3 && context->output_state[i]) {
context->noise_out = context->lsfr & 1;
context->lsfr = (context->lsfr >> 1) | (context->lsfr << 15);
if (context->noise_type) {
//white noise
if (context->lsfr & 0x40) {
context->lsfr ^= 0x8000;
}
}
}
}
}
context->last_sample = context->accum;
context->accum = 0;
for (int i = 0; i < 3; i++) {
if (context->output_state[i]) {
context->accum += volume_table[context->volume[i]];
}
}
if (context->noise_out) {
context->accum += volume_table[context->volume[3]];
}
int32_t tmp = context->accum * context->lowpass_alpha + context->last_sample * (0x10000 - context->lowpass_alpha);
context->accum = tmp >> 16;
context->buffer_fraction += context->buffer_inc;
if (context->buffer_fraction >= BUFFER_INC_RES) {
context->buffer_fraction -= BUFFER_INC_RES;
int32_t tmp = context->last_sample * ((context->buffer_fraction << 16) / context->buffer_inc);
tmp += context->accum * (0x10000 - ((context->buffer_fraction << 16) / context->buffer_inc));
context->audio_buffer[context->buffer_pos++] = tmp >> 16;
if (context->buffer_pos == context->samples_frame) {
if (!headless) {
render_wait_psg(context);
}
}
}
context->cycles += context->clock_inc;
}
}
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