looper/backends/playback/zsm/x16emu/audio.c
2024-09-28 10:31:18 -07:00

312 lines
12 KiB
C

// Commander X16 Emulator
// Copyright (c) 2020 Frank van den Hoef
// All rights reserved. License: 2-clause BSD
#define IN_AUDIO
#include "audio.h"
#include "glue.h"
#include "vera_psg.h"
#include "vera_pcm.h"
#include "ymglue.h"
#include <stdint.h>
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
// windowed sinc
static const int16_t filter[512] = {
32767,32765,32761,32755,32746,32736,32723,32707,32690,32670,32649,32625,32598,32570,32539,32507,
32472,32435,32395,32354,32310,32265,32217,32167,32115,32061,32004,31946,31885,31823,31758,31691,
31623,31552,31479,31404,31327,31248,31168,31085,31000,30913,30825,30734,30642,30547,30451,30353,
30253,30151,30048,29943,29835,29726,29616,29503,29389,29273,29156,29037,28916,28793,28669,28544,
28416,28288,28157,28025,27892,27757,27621,27483,27344,27204,27062,26918,26774,26628,26481,26332,
26182,26031,25879,25726,25571,25416,25259,25101,24942,24782,24621,24459,24296,24132,23967,23801,
23634,23466,23298,23129,22959,22788,22616,22444,22271,22097,21923,21748,21572,21396,21219,21042,
20864,20686,20507,20328,20148,19968,19788,19607,19426,19245,19063,18881,18699,18517,18334,18152,
17969,17786,17603,17420,17237,17054,16871,16688,16505,16322,16139,15957,15774,15592,15409,15227,
15046,14864,14683,14502,14321,14141,13961,13781,13602,13423,13245,13067,12890,12713,12536,12360,
12185,12010,11836,11663,11490,11317,11146,10975,10804,10635,10466,10298,10131, 9964, 9799, 9634,
9470, 9306, 9144, 8983, 8822, 8662, 8504, 8346, 8189, 8033, 7878, 7724, 7571, 7419, 7268, 7118,
6969, 6822, 6675, 6529, 6385, 6241, 6099, 5958, 5818, 5679, 5541, 5405, 5269, 5135, 5002, 4870,
4739, 4610, 4482, 4355, 4229, 4104, 3981, 3859, 3738, 3619, 3500, 3383, 3268, 3153, 3040, 2928,
2817, 2708, 2600, 2493, 2388, 2284, 2181, 2079, 1979, 1880, 1783, 1686, 1591, 1498, 1405, 1314,
1225, 1136, 1049, 963, 879, 795, 714, 633, 554, 476, 399, 323, 249, 176, 105, 34,
-34, -102, -168, -234, -298, -361, -422, -482, -542, -599, -656, -712, -766, -819, -871, -922,
-971,-1020,-1067,-1113,-1158,-1202,-1244,-1286,-1326,-1366,-1404,-1441,-1477,-1512,-1546,-1579,
-1611,-1642,-1671,-1700,-1728,-1755,-1781,-1806,-1830,-1852,-1874,-1896,-1916,-1935,-1953,-1971,
-1987,-2003,-2018,-2032,-2045,-2058,-2069,-2080,-2090,-2099,-2108,-2116,-2123,-2129,-2134,-2139,
-2143,-2147,-2150,-2152,-2153,-2154,-2154,-2154,-2153,-2151,-2149,-2146,-2143,-2139,-2135,-2130,
-2124,-2118,-2112,-2105,-2098,-2090,-2082,-2073,-2064,-2054,-2045,-2034,-2024,-2012,-2001,-1989,
-1977,-1965,-1952,-1939,-1926,-1912,-1898,-1884,-1870,-1855,-1840,-1825,-1810,-1794,-1778,-1762,
-1746,-1730,-1714,-1697,-1680,-1663,-1646,-1629,-1612,-1595,-1577,-1560,-1542,-1525,-1507,-1489,
-1471,-1453,-1435,-1418,-1400,-1382,-1364,-1346,-1328,-1310,-1292,-1274,-1256,-1238,-1220,-1203,
-1185,-1167,-1150,-1132,-1115,-1097,-1080,-1063,-1046,-1029,-1012, -995, -978, -962, -945, -929,
-912, -896, -880, -864, -849, -833, -817, -802, -787, -772, -757, -742, -727, -713, -699, -684,
-670, -656, -643, -629, -616, -603, -589, -577, -564, -551, -539, -526, -514, -502, -491, -479,
-468, -456, -445, -434, -423, -413, -402, -392, -381, -371, -361, -352, -342, -333, -323, -314,
-305, -296, -288, -279, -270, -262, -254, -246, -238, -230, -222, -215, -207, -200, -193, -186,
-179, -172, -165, -158, -152, -145, -139, -133, -127, -120, -114, -108, -103, -97, -91, -85,
-80, -74, -69, -63, -58, -53, -47, -42, -37, -32, -27, -22, -17, -12, -7, -2
};
static int16_t * buffer;
uint32_t buffer_size = 0;
static uint32_t rdidx = 0;
static uint32_t wridx = 0;
static uint32_t buffer_written = 0;
static uint32_t vera_samp_pos_rd = 0;
static uint32_t vera_samp_pos_wr = 0;
static uint32_t vera_samp_pos_hd = 0;
static uint32_t ym_samp_pos_rd = 0;
static uint32_t ym_samp_pos_wr = 0;
static uint32_t ym_samp_pos_hd = 0;
static uint32_t vera_samps_per_host_samps = 0;
static uint32_t ym_samps_per_host_samps = 0;
static uint32_t limiter_amp = 0;
static int16_t psg_buf[2 * SAMPLES_PER_BUFFER];
static int16_t pcm_buf[2 * SAMPLES_PER_BUFFER];
static int16_t ym_buf[2 * SAMPLES_PER_BUFFER];
uint32_t host_sample_rate = 0;
void
audio_callback(void *userdata, Uint8 *stream, int len)
{
int expected = SAMPLES_PER_BUFFER * SAMPLE_BYTES;
if (len != expected) {
printf("Audio buffer size mismatch! (expected: %d, got: %d)\n", expected, len);
return;
}
uint32_t spos = 0;
if (rdidx > wridx) {
uint32_t actual_len = SDL_min(len / SAMPLE_BYTES, (buffer_size - rdidx) / 2);
if (actual_len > 0) {
memcpy(&stream[spos], &buffer[rdidx], actual_len * SAMPLE_BYTES);
spos += actual_len * SAMPLE_BYTES;
len -= actual_len * SAMPLE_BYTES;
rdidx = (rdidx + actual_len * 2) % buffer_size;
buffer_written -= actual_len * 2;
}
}
uint32_t actual_len = SDL_min(len / SAMPLE_BYTES, (wridx - rdidx) / 2);
if (actual_len > 0) {
memcpy(&stream[spos], &buffer[rdidx], actual_len * SAMPLE_BYTES);
spos += actual_len * SAMPLE_BYTES;
len -= actual_len * SAMPLE_BYTES;
rdidx = (rdidx + actual_len * 2) % buffer_size;
buffer_written -= actual_len * 2;
}
if (len > 0) memset(&stream[spos], 0, len);
}
SDL_AudioSpec obtained;
void
audio_init(int num_audio_buffers)
{
// Set number of buffers
int num_bufs = num_audio_buffers;
if (num_bufs < 3) {
num_bufs = 3;
}
if (num_bufs > 1024) {
num_bufs = 1024;
}
buffer_size = SAMPLES_PER_BUFFER * num_bufs * 2;
// Allocate audio buffer
buffer = malloc(buffer_size * sizeof(int16_t));
rdidx = 0;
wridx = 0;
buffer_written = 0;
SDL_AudioSpec desired;
// Setup SDL audio
memset(&desired, 0, sizeof(desired));
desired.freq = AUDIO_SAMPLERATE;
desired.format = AUDIO_S16SYS;
desired.samples = SAMPLES_PER_BUFFER;
desired.channels = 2;
obtained = desired;
if (obtained.freq <= 0 || (AUDIO_SAMPLERATE / obtained.freq) > SAMPLES_PER_BUFFER) {
fprintf(stderr, "Obtained sample rate is too low");
audio_close();
return;
}
// Init YM2151 emulation. 3.579545 MHz clock
YM_Create(3579545);
YM_init(3579545/64, 60);
host_sample_rate = obtained.freq;
vera_samps_per_host_samps = ((25000000ULL << SAMP_POS_FRAC_BITS) / 512 / host_sample_rate);
ym_samps_per_host_samps = ((3579545ULL << SAMP_POS_FRAC_BITS) / 64 / host_sample_rate);
vera_samp_pos_rd = 0;
vera_samp_pos_wr = 0;
vera_samp_pos_hd = 0;
ym_samp_pos_rd = 0;
ym_samp_pos_wr = 0;
ym_samp_pos_hd = 0;
limiter_amp = (1 << 16);
psg_buf[0] = psg_buf[1] = 0;
pcm_buf[0] = pcm_buf[1] = 0;
ym_buf[0] = ym_buf[1] = 0;
}
void
audio_close(void)
{
// Free audio buffers
if (buffer != NULL) {
free(buffer);
buffer = NULL;
}
}
void
audio_step(int cpu_clocks)
{
while (cpu_clocks > 0) {
// Only the source with the higest sample rate (YM2151) is needed for calculation
uint32_t max_cpu_clks_ym = ((ym_samp_pos_rd - ym_samp_pos_hd - (1 << SAMP_POS_FRAC_BITS)) & SAMP_POS_MASK_FRAC) / YM_SAMP_CLKS_PER_CPU_CLK;
uint32_t max_cpu_clks = SDL_min(cpu_clocks, max_cpu_clks_ym);
vera_samp_pos_hd = (vera_samp_pos_hd + max_cpu_clks * VERA_SAMP_CLKS_PER_CPU_CLK) & SAMP_POS_MASK_FRAC;
ym_samp_pos_hd = (ym_samp_pos_hd + max_cpu_clks * YM_SAMP_CLKS_PER_CPU_CLK) & SAMP_POS_MASK_FRAC;
cpu_clocks -= max_cpu_clks;
if (cpu_clocks > 0) audio_render();
}
}
void
audio_render()
{
// Render all audio sources until read and write positions catch up
// This happens when there's a change to sound registers or one of the
// sources' sample buffer head position is too far
uint32_t pos, len;
pos = (vera_samp_pos_wr + 1) & SAMP_POS_MASK;
len = ((vera_samp_pos_hd >> SAMP_POS_FRAC_BITS) - vera_samp_pos_wr) & SAMP_POS_MASK;
vera_samp_pos_wr = vera_samp_pos_hd >> SAMP_POS_FRAC_BITS;
if (pos + len > SAMPLES_PER_BUFFER) {
psg_render(&psg_buf[pos * 2], SAMPLES_PER_BUFFER - pos);
pcm_render(&pcm_buf[pos * 2], SAMPLES_PER_BUFFER - pos);
len -= SAMPLES_PER_BUFFER - pos;
pos = 0;
}
if (len > 0) {
psg_render(&psg_buf[pos * 2], len);
pcm_render(&pcm_buf[pos * 2], len);
}
pos = (ym_samp_pos_wr + 1) & SAMP_POS_MASK;
len = ((ym_samp_pos_hd >> SAMP_POS_FRAC_BITS) - ym_samp_pos_wr) & SAMP_POS_MASK;
ym_samp_pos_wr = ym_samp_pos_hd >> SAMP_POS_FRAC_BITS;
if ((pos + len) > SAMPLES_PER_BUFFER) {
YM_stream_update((uint16_t *)&ym_buf[pos * 2], SAMPLES_PER_BUFFER - pos);
len -= SAMPLES_PER_BUFFER - pos;
pos = 0;
}
if (len > 0) {
YM_stream_update((uint16_t *)&ym_buf[pos * 2], len);
}
uint32_t wridx_old = wridx;
uint32_t len_vera = (vera_samp_pos_hd - vera_samp_pos_rd) & SAMP_POS_MASK_FRAC;
uint32_t len_ym = (ym_samp_pos_hd - ym_samp_pos_rd) & SAMP_POS_MASK_FRAC;
if (len_vera < (4 << SAMP_POS_FRAC_BITS) || len_ym < (4 << SAMP_POS_FRAC_BITS)) {
// not enough samples yet, at least 4 are needed for the filter
return;
}
len_vera = (len_vera - (4 << SAMP_POS_FRAC_BITS)) / vera_samps_per_host_samps;
len_ym = (len_ym - (4 << SAMP_POS_FRAC_BITS)) / ym_samps_per_host_samps;
len = SDL_min(len_vera, len_ym);
for (int i = 0; i < len; i++) {
int32_t samp[8];
int32_t filter_idx = 0;
int32_t vera_out_l = 0;
int32_t vera_out_r = 0;
int32_t ym_out_l = 0;
int32_t ym_out_r = 0;
// Don't resample VERA outputs if the host sample rate is as desired
if (host_sample_rate == AUDIO_SAMPLERATE) {
pos = (vera_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;
vera_out_l = ((int32_t)psg_buf[pos] + pcm_buf[pos]) << 14;
vera_out_r = ((int32_t)psg_buf[pos + 1] + pcm_buf[pos + 1]) << 14;
} else {
filter_idx = (vera_samp_pos_rd >> (SAMP_POS_FRAC_BITS - 8)) & 0xff;
pos = (vera_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;
for (int j = 0; j < 8; j += 2) {
samp[j] = (int32_t)psg_buf[pos] + pcm_buf[pos];
samp[j + 1] = (int32_t)psg_buf[pos + 1] + pcm_buf[pos + 1];
pos = (pos + 2) & (SAMP_POS_MASK * 2);
}
vera_out_l += samp[0] * filter[256 + filter_idx];
vera_out_r += samp[1] * filter[256 + filter_idx];
vera_out_l += samp[2] * filter[ 0 + filter_idx];
vera_out_r += samp[3] * filter[ 0 + filter_idx];
vera_out_l += samp[4] * filter[255 - filter_idx];
vera_out_r += samp[5] * filter[255 - filter_idx];
vera_out_l += samp[6] * filter[511 - filter_idx];
vera_out_r += samp[7] * filter[511 - filter_idx];
}
filter_idx = (ym_samp_pos_rd >> (SAMP_POS_FRAC_BITS - 8)) & 0xff;
pos = (ym_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;
for (int j = 0; j < 8; j += 2) {
samp[j] = ym_buf[pos];
samp[j + 1] = ym_buf[pos + 1];
pos = (pos + 2) & (SAMP_POS_MASK * 2);
}
ym_out_l += samp[0] * filter[256 + filter_idx];
ym_out_r += samp[1] * filter[256 + filter_idx];
ym_out_l += samp[2] * filter[ 0 + filter_idx];
ym_out_r += samp[3] * filter[ 0 + filter_idx];
ym_out_l += samp[4] * filter[255 - filter_idx];
ym_out_r += samp[5] * filter[255 - filter_idx];
ym_out_l += samp[6] * filter[511 - filter_idx];
ym_out_r += samp[7] * filter[511 - filter_idx];
// Mixing is according to the Developer Board
// Loudest single PSG channel is 1/8 times the max output
// mix = (psg + pcm) * 2 + ym
int32_t mix_l = (vera_out_l >> 13) + (ym_out_l >> 15);
int32_t mix_r = (vera_out_r >> 13) + (ym_out_r >> 15);
uint32_t amp = SDL_max(SDL_abs(mix_l), SDL_abs(mix_r));
if (amp > 32767) {
uint32_t limiter_amp_new = (32767 << 16) / amp;
limiter_amp = SDL_min(limiter_amp_new, limiter_amp);
}
buffer[wridx++] = (int16_t)((mix_l * limiter_amp) >> 16);
buffer[wridx++] = (int16_t)((mix_r * limiter_amp) >> 16);
wridx %= buffer_size;
if (limiter_amp < (1 << 16)) limiter_amp++;
vera_samp_pos_rd = (vera_samp_pos_rd + vera_samps_per_host_samps) & SAMP_POS_MASK_FRAC;
ym_samp_pos_rd = (ym_samp_pos_rd + ym_samps_per_host_samps) & SAMP_POS_MASK_FRAC;
}
buffer_written += len * 2;
if (buffer_written > buffer_size) {
// Prevent the buffer from overflowing by skipping the read pointer ahead.
uint32_t buffer_skip_amount = (buffer_written / buffer_size) * SAMPLES_PER_BUFFER * 2;
rdidx = (rdidx + buffer_skip_amount) % buffer_size;
buffer_written -= buffer_skip_amount;
}
// catch up all buffers if they are too far behind
uint32_t skip = len_vera - len;
if (skip > 1) {
vera_samp_pos_rd = (vera_samp_pos_rd + vera_samps_per_host_samps) & SAMP_POS_MASK_FRAC;
}
skip = len_ym - len;
if (skip > 1) {
ym_samp_pos_rd = (ym_samp_pos_rd + ym_samps_per_host_samps) & SAMP_POS_MASK_FRAC;
}
}