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WinUAE/audio.cpp
Frode Solheim add7f46b1e trivial changes in several modules
2015-09-04 22:36:11 +02:00

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/*
* UAE - The Un*x Amiga Emulator
*
* Paula audio emulation
*
* Copyright 1995, 1996, 1997 Bernd Schmidt
* Copyright 1996 Marcus Sundberg
* Copyright 1996 Manfred Thole
* Copyright 2006 Toni Wilen
*
* new filter algorithm and anti&sinc interpolators by Antti S. Lankila
*
*/
#include "sysconfig.h"
#include "sysdeps.h"
#include "options.h"
#include "memory.h"
#include "custom.h"
#include "newcpu.h"
#include "autoconf.h"
#include "gensound.h"
#include "audio.h"
#include "sounddep/sound.h"
#include "events.h"
#include "savestate.h"
#include "driveclick.h"
#include "zfile.h"
#include "uae.h"
#include "gui.h"
#include "xwin.h"
#include "debug.h"
#include "sndboard.h"
#ifdef AVIOUTPUT
#include "avioutput.h"
#endif
#ifdef AHI
#include "traps.h"
#include "ahidsound.h"
#include "ahidsound_new.h"
#endif
#include "threaddep/thread.h"
#include <math.h>
#define DEBUG_AUDIO 0
#define DEBUG_AUDIO_HACK 0
#define DEBUG_CHANNEL_MASK 15
#define TEST_AUDIO 0
#define PERIOD_MIN 4
#define PERIOD_MIN_NONCE 60
int audio_channel_mask = 15;
STATIC_INLINE bool isaudio (void)
{
return currprefs.produce_sound != 0;
}
#if DEBUG_AUDIO > 0 || DEBUG_AUDIO_HACK > 0
static bool debugchannel (int ch)
{
return ((1 << ch) & DEBUG_CHANNEL_MASK) != 0;
}
#endif
STATIC_INLINE bool usehacks(void)
{
return currprefs.cpu_model >= 68020 || currprefs.m68k_speed != 0 || (currprefs.cs_hacks & 4);
}
#define SINC_QUEUE_MAX_AGE 2048
/* Queue length 256 implies minimum emulated period of 8. This should be
* sufficient for all imaginable purposes. This must be power of two. */
#define SINC_QUEUE_LENGTH 256
#include "sinctable.cpp"
typedef struct {
int time, output;
} sinc_queue_t;
struct audio_channel_data {
unsigned int adk_mask;
unsigned int evtime;
bool dmaenstore;
bool intreq2;
bool dr;
bool dsr;
bool pbufldl;
int drhpos;
bool dat_written;
uaecptr lc, pt;
int current_sample, last_sample;
int state;
int per;
int vol;
int len, wlen;
uae_u16 dat, dat2;
int sample_accum, sample_accum_time;
int sinc_output_state;
sinc_queue_t sinc_queue[SINC_QUEUE_LENGTH];
int sinc_queue_time;
int sinc_queue_head;
#if TEST_AUDIO > 0
bool hisample, losample;
bool have_dat;
int per_original;
#endif
/* too fast cpu fixes */
uaecptr ptx;
bool ptx_written;
bool ptx_tofetch;
int dmaofftime_active;
};
static int audio_channel_count = AUDIO_CHANNELS_PAULA;
static bool audio_extra_channels[2];
static int samplecnt;
#if SOUNDSTUFF > 0
static int extrasamples, outputsample, doublesample;
#endif
int sampleripper_enabled;
struct ripped_sample
{
struct ripped_sample *next;
uae_u8 *sample;
int len, per, changed;
};
static struct ripped_sample *ripped_samples;
void write_wavheader (struct zfile *wavfile, uae_u32 size, uae_u32 freq)
{
uae_u16 tw;
uae_u32 tl;
int bits = 8, channels = 1;
zfile_fseek (wavfile, 0, SEEK_SET);
zfile_fwrite ("RIFF", 1, 4, wavfile);
tl = 0;
if (size)
tl = size - 8;
zfile_fwrite (&tl, 1, 4, wavfile);
zfile_fwrite ("WAVEfmt ", 1, 8, wavfile);
tl = 16;
zfile_fwrite (&tl, 1, 4, wavfile);
tw = 1;
zfile_fwrite (&tw, 1, 2, wavfile);
tw = channels;
zfile_fwrite (&tw, 1, 2, wavfile);
tl = freq;
zfile_fwrite (&tl, 1, 4, wavfile);
tl = freq * channels * bits / 8;
zfile_fwrite (&tl, 1, 4, wavfile);
tw = channels * bits / 8;
zfile_fwrite (&tw, 1, 2, wavfile);
tw = bits;
zfile_fwrite (&tw, 1, 2, wavfile);
zfile_fwrite ("data", 1, 4, wavfile);
tl = 0;
if (size)
tl = size - 44;
zfile_fwrite (&tl, 1, 4, wavfile);
}
static void convertsample (uae_u8 *sample, int len)
{
int i;
for (i = 0; i < len; i++)
sample[i] += 0x80;
}
static void namesplit (TCHAR *s)
{
int l;
l = _tcslen (s) - 1;
while (l >= 0) {
if (s[l] == '.')
s[l] = 0;
if (s[l] == '\\' || s[l] == '/' || s[l] == ':' || s[l] == '?') {
l++;
break;
}
l--;
}
if (l > 0)
memmove (s, s + l, (_tcslen (s + l) + 1) * sizeof (TCHAR));
}
void audio_sampleripper (int mode)
{
struct ripped_sample *rs = ripped_samples;
int cnt = 1;
TCHAR path[MAX_DPATH], name[MAX_DPATH], filename[MAX_DPATH];
TCHAR underline[] = _T("_");
TCHAR extension[4];
struct zfile *wavfile;
if (mode < 0) {
while (rs) {
struct ripped_sample *next = rs->next;
xfree(rs);
rs = next;
}
ripped_samples = NULL;
return;
}
while (rs) {
if (rs->changed) {
rs->changed = 0;
fetch_ripperpath (path, sizeof (path) / sizeof (TCHAR));
name[0] = 0;
if (currprefs.floppyslots[0].dfxtype >= 0)
_tcscpy (name, currprefs.floppyslots[0].df);
else if (currprefs.cdslots[0].inuse)
_tcscpy (name, currprefs.cdslots[0].name);
if (!name[0])
underline[0] = 0;
namesplit (name);
_tcscpy (extension, _T("wav"));
_stprintf (filename, _T("%s%s%s%03d.%s"), path, name, underline, cnt, extension);
wavfile = zfile_fopen (filename, _T("wb"), 0);
if (wavfile) {
int freq = rs->per > 0 ? (currprefs.ntscmode ? 3579545 : 3546895 / rs->per) : 8000;
write_wavheader (wavfile, 0, 0);
convertsample (rs->sample, rs->len);
zfile_fwrite (rs->sample, rs->len, 1, wavfile);
convertsample (rs->sample, rs->len);
write_wavheader (wavfile, zfile_ftell(wavfile), freq);
zfile_fclose (wavfile);
write_log (_T("SAMPLERIPPER: %d: %dHz %d bytes\n"), cnt, freq, rs->len);
} else {
write_log (_T("SAMPLERIPPER: failed to open '%s'\n"), filename);
}
}
cnt++;
rs = rs->next;
}
}
static void do_samplerip (struct audio_channel_data *adp)
{
struct ripped_sample *rs = ripped_samples, *prev;
int len = adp->wlen * 2;
uae_u8 *smp = chipmem_xlate_indirect (adp->pt);
int cnt = 0, i;
if (!smp || !chipmem_check_indirect (adp->pt, len))
return;
for (i = 0; i < len; i++) {
if (smp[i] != 0)
break;
}
if (i == len || len <= 2)
return;
prev = NULL;
while(rs) {
if (rs->sample) {
if (len == rs->len && !memcmp (rs->sample, smp, len))
break;
/* replace old identical but shorter sample */
if (len > rs->len && !memcmp (rs->sample, smp, rs->len)) {
xfree (rs->sample);
rs->sample = xmalloc (uae_u8, len);
memcpy (rs->sample, smp, len);
write_log (_T("SAMPLERIPPER: replaced sample %d (%d -> %d)\n"), cnt, rs->len, len);
rs->len = len;
rs->per = adp->per / CYCLE_UNIT;
rs->changed = 1;
audio_sampleripper (0);
return;
}
}
prev = rs;
rs = rs->next;
cnt++;
}
if (rs || cnt > 100)
return;
rs = xmalloc (struct ripped_sample ,1);
if (prev)
prev->next = rs;
else
ripped_samples = rs;
rs->len = len;
rs->per = adp->per / CYCLE_UNIT;
rs->sample = xmalloc (uae_u8, len);
memcpy (rs->sample, smp, len);
rs->next = NULL;
rs->changed = 1;
write_log (_T("SAMPLERIPPER: sample added (%06X, %d bytes), total %d samples\n"), adp->pt, len, ++cnt);
audio_sampleripper (0);
}
static struct audio_channel_data audio_channel[AUDIO_CHANNELS_MAX];
int sound_available = 0;
void (*sample_handler) (void);
static void(*sample_prehandler) (unsigned long best_evtime);
static void(*extra_sample_prehandler) (unsigned long best_evtime);
float sample_evtime;
float scaled_sample_evtime;
int sound_cd_volume[2];
int sound_paula_volume[2];
static unsigned long last_cycles;
static float next_sample_evtime;
typedef uae_s8 sample8_t;
#define DO_CHANNEL_1(v, c) do { (v) *= audio_channel[c].vol; } while (0)
#define SBASEVAL16(logn) ((logn) == 1 ? SOUND16_BASE_VAL >> 1 : SOUND16_BASE_VAL)
STATIC_INLINE int FINISH_DATA (int data, int bits, int ch)
{
if (bits == 16) {
return data;
} else if (bits - 16 > 0) {
data >>= bits - 16;
} else {
int shift = 16 - bits;
data <<= shift;
}
data = data * sound_paula_volume[ch] / 32768;
return data;
}
static uae_u32 right_word_saved[SOUND_MAX_DELAY_BUFFER];
static uae_u32 left_word_saved[SOUND_MAX_DELAY_BUFFER];
static uae_u32 right2_word_saved[SOUND_MAX_DELAY_BUFFER];
static uae_u32 left2_word_saved[SOUND_MAX_DELAY_BUFFER];
static int saved_ptr, saved_ptr2;
static int mixed_on, mixed_stereo_size, mixed_mul1, mixed_mul2;
static int led_filter_forced, sound_use_filter, sound_use_filter_sinc, led_filter_on;
/* denormals are very small floating point numbers that force FPUs into slow
mode. All lowpass filters using floats are suspectible to denormals unless
a small offset is added to avoid very small floating point numbers. */
#define DENORMAL_OFFSET (1E-10)
static struct filter_state {
float rc1, rc2, rc3, rc4, rc5;
} sound_filter_state[AUDIO_CHANNELS_PAULA];
static float a500e_filter1_a0;
static float a500e_filter2_a0;
static float filter_a0; /* a500 and a1200 use the same */
enum {
FILTER_NONE = 0,
FILTER_MODEL_A500,
FILTER_MODEL_A1200
};
/* Amiga has two separate filtering circuits per channel, a static RC filter
* on A500 and the LED filter. This code emulates both.
*
* The Amiga filtering circuitry depends on Amiga model. Older Amigas seem
* to have a 6 dB/oct RC filter with cutoff frequency such that the -6 dB
* point for filter is reached at 6 kHz, while newer Amigas have no filtering.
*
* The LED filter is complicated, and we are modelling it with a pair of
* RC filters, the other providing a highboost. The LED starts to cut
* into signal somewhere around 5-6 kHz, and there's some kind of highboost
* in effect above 12 kHz. Better measurements are required.
*
* The current filtering should be accurate to 2 dB with the filter on,
* and to 1 dB with the filter off.
*/
static int filter (int input, struct filter_state *fs)
{
int o;
float normal_output, led_output;
input = (uae_s16)input;
switch (sound_use_filter) {
case FILTER_MODEL_A500:
fs->rc1 = a500e_filter1_a0 * input + (1 - a500e_filter1_a0) * fs->rc1 + DENORMAL_OFFSET;
fs->rc2 = a500e_filter2_a0 * fs->rc1 + (1-a500e_filter2_a0) * fs->rc2;
normal_output = fs->rc2;
fs->rc3 = filter_a0 * normal_output + (1 - filter_a0) * fs->rc3;
fs->rc4 = filter_a0 * fs->rc3 + (1 - filter_a0) * fs->rc4;
fs->rc5 = filter_a0 * fs->rc4 + (1 - filter_a0) * fs->rc5;
led_output = fs->rc5;
break;
case FILTER_MODEL_A1200:
normal_output = input;
fs->rc2 = filter_a0 * normal_output + (1 - filter_a0) * fs->rc2 + DENORMAL_OFFSET;
fs->rc3 = filter_a0 * fs->rc2 + (1 - filter_a0) * fs->rc3;
fs->rc4 = filter_a0 * fs->rc3 + (1 - filter_a0) * fs->rc4;
led_output = fs->rc4;
break;
case FILTER_NONE:
default:
return input;
}
if (led_filter_on)
o = led_output;
else
o = normal_output;
if (o > 32767)
o = 32767;
else if (o < -32768)
o = -32768;
return o;
}
/* Always put the right word before the left word. */
static void put_sound_word_right (uae_u32 w)
{
if (mixed_on) {
right_word_saved[saved_ptr] = w;
return;
}
PUT_SOUND_WORD (w);
}
static void put_sound_word_left (uae_u32 w)
{
if (mixed_on) {
uae_u32 rold, lold, rnew, lnew, tmp;
left_word_saved[saved_ptr] = w;
lnew = w - SOUND16_BASE_VAL;
rnew = right_word_saved[saved_ptr] - SOUND16_BASE_VAL;
saved_ptr = (saved_ptr + 1) & mixed_stereo_size;
lold = left_word_saved[saved_ptr] - SOUND16_BASE_VAL;
tmp = (rnew * mixed_mul2 + lold * mixed_mul1) / MIXED_STEREO_SCALE;
tmp += SOUND16_BASE_VAL;
rold = right_word_saved[saved_ptr] - SOUND16_BASE_VAL;
w = (lnew * mixed_mul2 + rold * mixed_mul1) / MIXED_STEREO_SCALE;
PUT_SOUND_WORD (w);
PUT_SOUND_WORD (tmp);
} else {
PUT_SOUND_WORD (w);
}
}
static void put_sound_word_right2 (uae_u32 w)
{
if (mixed_on) {
right2_word_saved[saved_ptr2] = w;
return;
}
PUT_SOUND_WORD (w);
}
static void put_sound_word_left2 (uae_u32 w)
{
if (mixed_on) {
uae_u32 rold, lold, rnew, lnew, tmp;
left2_word_saved[saved_ptr2] = w;
lnew = w - SOUND16_BASE_VAL;
rnew = right2_word_saved[saved_ptr2] - SOUND16_BASE_VAL;
saved_ptr2 = (saved_ptr2 + 1) & mixed_stereo_size;
lold = left2_word_saved[saved_ptr2] - SOUND16_BASE_VAL;
tmp = (rnew * mixed_mul2 + lold * mixed_mul1) / MIXED_STEREO_SCALE;
tmp += SOUND16_BASE_VAL;
rold = right2_word_saved[saved_ptr2] - SOUND16_BASE_VAL;
w = (lnew * mixed_mul2 + rold * mixed_mul1) / MIXED_STEREO_SCALE;
PUT_SOUND_WORD (w);
PUT_SOUND_WORD (tmp);
} else {
PUT_SOUND_WORD (w);
}
}
static void anti_prehandler (unsigned long best_evtime)
{
int i, output;
struct audio_channel_data *acd;
/* Handle accumulator antialiasiation */
for (i = 0; i < audio_channel_count; i++) {
acd = &audio_channel[i];
output = (acd->current_sample * acd->vol) & acd->adk_mask;
acd->sample_accum += output * best_evtime;
acd->sample_accum_time += best_evtime;
}
}
static void samplexx_anti_handler (int *datasp, int ch_start, int ch_num)
{
int i, j;
for (i = ch_start, j = 0; j < ch_num; i++, j++) {
datasp[j] = audio_channel[i].sample_accum_time ? (audio_channel[i].sample_accum / audio_channel[i].sample_accum_time) : 0;
audio_channel[i].sample_accum = 0;
audio_channel[i].sample_accum_time = 0;
}
}
static void sinc_prehandler_paula (unsigned long best_evtime)
{
int i, output;
struct audio_channel_data *acd;
for (i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
acd = &audio_channel[i];
int vol = acd->vol;
output = (acd->current_sample * vol) & acd->adk_mask;
/* if output state changes, record the state change and also
* write data into sinc queue for mixing in the BLEP */
if (acd->sinc_output_state != output) {
acd->sinc_queue_head = (acd->sinc_queue_head - 1) & (SINC_QUEUE_LENGTH - 1);
acd->sinc_queue[acd->sinc_queue_head].time = acd->sinc_queue_time;
acd->sinc_queue[acd->sinc_queue_head].output = output - acd->sinc_output_state;
acd->sinc_output_state = output;
}
acd->sinc_queue_time += best_evtime;
}
}
/* this interpolator performs BLEP mixing (bleps are shaped like integrated sinc
* functions) with a type of BLEP that matches the filtering configuration. */
static void samplexx_sinc_handler (int *datasp, int ch_start, int ch_num)
{
int n, i, k;
int const *winsinc;
if (sound_use_filter_sinc && ch_start == 0) {
n = (sound_use_filter_sinc == FILTER_MODEL_A500) ? 0 : 2;
if (led_filter_on)
n += 1;
} else {
n = 4;
}
winsinc = winsinc_integral[n];
for (i = ch_start, k = 0; k < ch_num; i++, k++) {
int j, v;
struct audio_channel_data *acd = &audio_channel[i];
/* The sum rings with harmonic components up to infinity... */
int sum = acd->sinc_output_state << 17;
/* ...but we cancel them through mixing in BLEPs instead */
int offsetpos = acd->sinc_queue_head & (SINC_QUEUE_LENGTH - 1);
for (j = 0; j < SINC_QUEUE_LENGTH; j += 1) {
int age = acd->sinc_queue_time - acd->sinc_queue[offsetpos].time;
if (age >= SINC_QUEUE_MAX_AGE || age < 0)
break;
sum -= winsinc[age] * acd->sinc_queue[offsetpos].output;
offsetpos = (offsetpos + 1) & (SINC_QUEUE_LENGTH - 1);
}
v = sum >> 15;
if (v > 32767)
v = 32767;
else if (v < -32768)
v = -32768;
datasp[k] = v;
}
}
static void do_filter(int *data, int num)
{
if (currprefs.sound_filter)
*data = filter(*data, &sound_filter_state[num]);
}
static void get_extra_channels(int *data1, int *data2, int sample1, int sample2)
{
int d1 = *data1 + sample1;
int d2 = *data2 + sample2;
if (d1 < -32768)
d1 = -32768;
if (d1 > 32767)
d1 = 32767;
if (d2 < -32768)
d2 = -32768;
if (d2 > 32767)
d2 = 32767;
int needswap = currprefs.sound_stereo_swap_paula ^ currprefs.sound_stereo_swap_ahi;
if (needswap) {
*data1 = d2;
*data2 = d1;
} else {
*data1 = d1;
*data2 = d2;
}
}
static void get_extra_channels_sample(int *data1, int *data2, int mode)
{
if (audio_extra_channels[0]) {
int datas[2];
samplexx_anti_handler(datas, AUDIO_CHANNEL_SNDBOARD_LEFT, 2);
get_extra_channels(data1, data2, datas[0], datas[1]);
}
if (audio_extra_channels[1]) {
int datas[2];
samplexx_anti_handler(datas, AUDIO_CHANNEL_CDA_LEFT, 2);
get_extra_channels(data1, data2, datas[0], datas[1]);
}
}
static void get_extra_channels_sample_mono(int *data1, int mode)
{
if (audio_extra_channels[0]) {
int datas[1];
samplexx_anti_handler(datas, AUDIO_CHANNEL_SNDBOARD_LEFT, 1);
int d1 = *data1 + datas[0];
if (d1 < -32768)
d1 = -32768;
if (d1 > 32767)
d1 = 32767;
*data1 = d1;
}
if (audio_extra_channels[1]) {
int datas[1];
samplexx_anti_handler(datas, AUDIO_CHANNEL_CDA_LEFT, 1);
int d1 = *data1 + datas[0];
if (d1 < -32768)
d1 = -32768;
if (d1 > 32767)
d1 = 32767;
*data1 = d1;
}
}
static void sample16i_sinc_handler (void)
{
int datas[AUDIO_CHANNELS_PAULA], data1;
samplexx_sinc_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data1 = datas[0] + datas[3] + datas[1] + datas[2];
data1 = FINISH_DATA (data1, 18, 0);
do_filter(&data1, 0);
get_extra_channels_sample_mono(&data1, 2);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data1);
check_sound_buffers ();
}
void sample16_handler (void)
{
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
int data;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
data0 &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data0 += data1;
data0 += data2;
data0 += data3;
data = SBASEVAL16(2) + data0;
data = FINISH_DATA (data, 16, 0);
do_filter(&data, 0);
get_extra_channels_sample_mono(&data, 0);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data);
check_sound_buffers ();
}
/* This interpolator examines sample points when Paula switches the output
* voltage and computes the average of Paula's output */
static void sample16i_anti_handler (void)
{
int datas[AUDIO_CHANNELS_PAULA], data1;
samplexx_anti_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data1 = datas[0] + datas[3] + datas[1] + datas[2];
data1 = FINISH_DATA (data1, 16, 0);
do_filter(&data1, 0);
get_extra_channels_sample_mono(&data1, 1);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data1);
check_sound_buffers ();
}
static void sample16i_rh_handler (void)
{
unsigned long delta, ratio;
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
int data0p = audio_channel[0].last_sample;
int data1p = audio_channel[1].last_sample;
int data2p = audio_channel[2].last_sample;
int data3p = audio_channel[3].last_sample;
int data;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
/* linear interpolation and summing up... */
delta = audio_channel[0].per;
ratio = ((audio_channel[0].evtime % delta) << 8) / delta;
data0 = (data0 * (256 - ratio) + data0p * ratio) >> 8;
delta = audio_channel[1].per;
ratio = ((audio_channel[1].evtime % delta) << 8) / delta;
data0 += (data1 * (256 - ratio) + data1p * ratio) >> 8;
delta = audio_channel[2].per;
ratio = ((audio_channel[2].evtime % delta) << 8) / delta;
data0 += (data2 * (256 - ratio) + data2p * ratio) >> 8;
delta = audio_channel[3].per;
ratio = ((audio_channel[3].evtime % delta) << 8) / delta;
data0 += (data3 * (256 - ratio) + data3p * ratio) >> 8;
data = SBASEVAL16(2) + data0;
data = FINISH_DATA (data, 16, 0);
do_filter(&data, 0);
get_extra_channels_sample_mono(&data, 0);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data);
check_sound_buffers ();
}
static void sample16i_crux_handler (void)
{
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
int data0p = audio_channel[0].last_sample;
int data1p = audio_channel[1].last_sample;
int data2p = audio_channel[2].last_sample;
int data3p = audio_channel[3].last_sample;
int data;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
{
struct audio_channel_data *cdp;
unsigned long ratio, ratio1;
#define INTERVAL (scaled_sample_evtime * 3)
cdp = audio_channel + 0;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data0 = (data0 * ratio + data0p * (4096 - ratio)) >> 12;
cdp = audio_channel + 1;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data1 = (data1 * ratio + data1p * (4096 - ratio)) >> 12;
cdp = audio_channel + 2;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data2 = (data2 * ratio + data2p * (4096 - ratio)) >> 12;
cdp = audio_channel + 3;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data3 = (data3 * ratio + data3p * (4096 - ratio)) >> 12;
}
data1 += data2;
data0 += data3;
data0 += data1;
data = SBASEVAL16(2) + data0;
data = FINISH_DATA (data, 16, 0);
do_filter(&data, 0);
get_extra_channels_sample_mono(&data, 0);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data);
check_sound_buffers ();
}
#ifdef HAVE_STEREO_SUPPORT
STATIC_INLINE void make6ch (uae_s32 d0, uae_s32 d1, uae_s32 d2, uae_s32 d3)
{
uae_s32 sum = d0 + d1 + d2 + d3;
sum /= 8;
PUT_SOUND_WORD (sum);
PUT_SOUND_WORD (sum);
}
void sample16ss_handler (void)
{
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
data0 &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data0 = FINISH_DATA (data0, 14, 0);
data1 = FINISH_DATA (data1, 14, 0);
data2 = FINISH_DATA (data2, 14, 1);
data3 = FINISH_DATA (data3, 14, 1);
do_filter(&data0, 0);
do_filter(&data1, 1);
do_filter(&data2, 3);
do_filter(&data3, 2);
get_extra_channels_sample(&data0, &data1, 0);
get_extra_channels_sample(&data2, &data3, 0);
set_sound_buffers ();
put_sound_word_left (data0);
put_sound_word_right (data1);
if (currprefs.sound_stereo == SND_6CH)
make6ch (data0, data1, data2, data3);
put_sound_word_left2 (data3);
put_sound_word_right2 (data2);
check_sound_buffers ();
}
/* This interpolator examines sample points when Paula switches the output
* voltage and computes the average of Paula's output */
static void sample16ss_anti_handler (void)
{
int data0, data1, data2, data3;
int datas[AUDIO_CHANNELS_PAULA];
samplexx_anti_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data0 = FINISH_DATA (datas[0], 14, 0);
data1 = FINISH_DATA (datas[1], 14, 0);
data2 = FINISH_DATA (datas[2], 14, 1);
data3 = FINISH_DATA (datas[3], 14, 1);
do_filter(&data0, 0);
do_filter(&data1, 1);
do_filter(&data2, 3);
do_filter(&data3, 2);
get_extra_channels_sample(&data0, &data1, 1);
get_extra_channels_sample(&data3, &data2, 1);
set_sound_buffers ();
put_sound_word_left (data0);
put_sound_word_right (data1);
if (currprefs.sound_stereo == SND_6CH)
make6ch (data0, data1, data2, data3);
put_sound_word_left2 (data3);
put_sound_word_right2 (data2);
check_sound_buffers ();
}
static void sample16si_anti_handler (void)
{
int datas[AUDIO_CHANNELS_PAULA], data1, data2;
samplexx_anti_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data1 = datas[0] + datas[3];
data2 = datas[1] + datas[2];
data1 = FINISH_DATA (data1, 15, 0);
data2 = FINISH_DATA (data2, 15, 1);
do_filter(&data1, 0);
do_filter(&data2, 1);
get_extra_channels_sample(&data1, &data2, 1);
set_sound_buffers ();
put_sound_word_left (data1);
put_sound_word_right (data2);
check_sound_buffers ();
}
static void sample16ss_sinc_handler (void)
{
int data0, data1, data2, data3;
int datas[AUDIO_CHANNELS_PAULA];
samplexx_sinc_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data0 = FINISH_DATA (datas[0], 16, 0);
data1 = FINISH_DATA (datas[1], 16, 0);
data2 = FINISH_DATA (datas[2], 16, 1);
data3 = FINISH_DATA (datas[3], 16, 1);
do_filter(&data0, 0);
do_filter(&data1, 1);
do_filter(&data2, 3);
do_filter(&data3, 2);
get_extra_channels_sample(&data0, &data1, 2);
get_extra_channels_sample(&data3, &data2, 2);
set_sound_buffers ();
put_sound_word_left (data0);
put_sound_word_right (data1);
if (currprefs.sound_stereo == SND_6CH)
make6ch (data0, data1, data2, data3);
put_sound_word_left2 (data3);
put_sound_word_right2 (data2);
check_sound_buffers ();
}
static void sample16si_sinc_handler (void)
{
int datas[AUDIO_CHANNELS_PAULA], data1, data2;
samplexx_sinc_handler (datas, 0, AUDIO_CHANNELS_PAULA);
data1 = datas[0] + datas[3];
data2 = datas[1] + datas[2];
data1 = FINISH_DATA (data1, 17, 0);
data2 = FINISH_DATA (data2, 17, 1);
do_filter(&data1, 0);
do_filter(&data2, 1);
get_extra_channels_sample(&data1, &data2, 2);
set_sound_buffers ();
put_sound_word_left (data1);
put_sound_word_right (data2);
check_sound_buffers ();
}
void sample16s_handler (void)
{
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
data0 &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data0 += data3;
data1 += data2;
data2 = SBASEVAL16(1) + data0;
data2 = FINISH_DATA (data2, 15, 0);
data3 = SBASEVAL16(1) + data1;
data3 = FINISH_DATA (data3, 15, 1);
do_filter(&data2, 0);
do_filter(&data3, 1);
get_extra_channels_sample(&data2, &data3, 0);
set_sound_buffers ();
put_sound_word_left (data2);
put_sound_word_right (data3);
check_sound_buffers ();
}
static void sample16si_crux_handler (void)
{
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
int data0p = audio_channel[0].last_sample;
int data1p = audio_channel[1].last_sample;
int data2p = audio_channel[2].last_sample;
int data3p = audio_channel[3].last_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
{
struct audio_channel_data *cdp;
unsigned long ratio, ratio1;
#define INTERVAL (scaled_sample_evtime * 3)
cdp = audio_channel + 0;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data0 = (data0 * ratio + data0p * (4096 - ratio)) >> 12;
cdp = audio_channel + 1;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data1 = (data1 * ratio + data1p * (4096 - ratio)) >> 12;
cdp = audio_channel + 2;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data2 = (data2 * ratio + data2p * (4096 - ratio)) >> 12;
cdp = audio_channel + 3;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data3 = (data3 * ratio + data3p * (4096 - ratio)) >> 12;
}
data1 += data2;
data0 += data3;
data2 = SBASEVAL16(1) + data0;
data2 = FINISH_DATA (data2, 15, 0);
data3 = SBASEVAL16(1) + data1;
data3 = FINISH_DATA (data3, 15, 1);
do_filter(&data2, 0);
do_filter(&data3, 1);
get_extra_channels_sample(&data2, &data3, 0);
set_sound_buffers ();
put_sound_word_left (data2);
put_sound_word_right (data3);
check_sound_buffers ();
}
static void sample16si_rh_handler (void)
{
unsigned long delta, ratio;
int data0 = audio_channel[0].current_sample;
int data1 = audio_channel[1].current_sample;
int data2 = audio_channel[2].current_sample;
int data3 = audio_channel[3].current_sample;
int data0p = audio_channel[0].last_sample;
int data1p = audio_channel[1].last_sample;
int data2p = audio_channel[2].last_sample;
int data3p = audio_channel[3].last_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
/* linear interpolation and summing up... */
delta = audio_channel[0].per;
ratio = ((audio_channel[0].evtime % delta) << 8) / delta;
data0 = (data0 * (256 - ratio) + data0p * ratio) >> 8;
delta = audio_channel[1].per;
ratio = ((audio_channel[1].evtime % delta) << 8) / delta;
data1 = (data1 * (256 - ratio) + data1p * ratio) >> 8;
delta = audio_channel[2].per;
ratio = ((audio_channel[2].evtime % delta) << 8) / delta;
data1 += (data2 * (256 - ratio) + data2p * ratio) >> 8;
delta = audio_channel[3].per;
ratio = ((audio_channel[3].evtime % delta) << 8) / delta;
data0 += (data3 * (256 - ratio) + data3p * ratio) >> 8;
data2 = SBASEVAL16(1) + data0;
data2 = FINISH_DATA (data2, 15, 0);
data3 = SBASEVAL16(1) + data1;
data3 = FINISH_DATA (data3, 15, 1);
do_filter(&data2, 0);
do_filter(&data3, 1);
get_extra_channels_sample(&data2, &data3, 0);
set_sound_buffers ();
put_sound_word_left (data2);
put_sound_word_right (data3);
check_sound_buffers ();
}
#else
void sample16s_handler (void)
{
sample16_handler ();
}
static void sample16si_crux_handler (void)
{
sample16i_crux_handler ();
}
static void sample16si_rh_handler (void)
{
sample16i_rh_handler ();
}
#endif
static int audio_work_to_do;
static void zerostate (int nr)
{
struct audio_channel_data *cdp = audio_channel + nr;
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("%d: ZEROSTATE\n"), nr);
#endif
cdp->state = 0;
cdp->evtime = MAX_EV;
cdp->intreq2 = 0;
cdp->dmaenstore = false;
cdp->dmaofftime_active = 0;
#if TEST_AUDIO > 0
cdp->have_dat = false;
#endif
}
static void schedule_audio (void)
{
unsigned long best = MAX_EV;
int i;
eventtab[ev_audio].active = 0;
eventtab[ev_audio].oldcycles = get_cycles ();
for (i = 0; i < audio_channel_count; i++) {
struct audio_channel_data *cdp = audio_channel + i;
if (cdp->evtime != MAX_EV) {
if (best > cdp->evtime) {
best = cdp->evtime;
eventtab[ev_audio].active = 1;
}
}
}
eventtab[ev_audio].evtime = get_cycles () + best;
}
static void audio_event_reset (void)
{
int i;
last_cycles = get_cycles ();
next_sample_evtime = scaled_sample_evtime;
if (!isrestore ()) {
for (i = 0; i < AUDIO_CHANNELS_PAULA; i++)
zerostate (i);
}
schedule_audio ();
events_schedule ();
samplecnt = 0;
extrasamples = 0;
outputsample = 1;
doublesample = 0;
}
void audio_deactivate (void)
{
gui_data.sndbuf_status = 3;
gui_data.sndbuf = 0;
audio_work_to_do = 0;
pause_sound_buffer ();
clear_sound_buffers ();
audio_event_reset ();
}
int audio_activate (void)
{
int ret = 0;
if (!audio_work_to_do) {
restart_sound_buffer ();
ret = 1;
audio_event_reset ();
}
audio_work_to_do = 4 * maxvpos_nom * 50;
return ret;
}
STATIC_INLINE int is_audio_active (void)
{
return audio_work_to_do;
}
uae_u16 audio_dmal (void)
{
uae_u16 dmal = 0;
for (int nr = 0; nr < AUDIO_CHANNELS_PAULA; nr++) {
struct audio_channel_data *cdp = audio_channel + nr;
if (cdp->dr)
dmal |= 1 << (nr * 2);
if (cdp->dsr)
dmal |= 1 << (nr * 2 + 1);
cdp->dr = cdp->dsr = false;
}
return dmal;
}
static int isirq (int nr)
{
return INTREQR () & (0x80 << nr);
}
static void setirq (int nr, int which)
{
#if DEBUG_AUDIO > 0
struct audio_channel_data *cdp = audio_channel + nr;
if (debugchannel (nr) && cdp->wlen > 1)
write_log (_T("SETIRQ%d (%d,%d) PC=%08X\n"), nr, which, isirq (nr) ? 1 : 0, M68K_GETPC);
#endif
INTREQ_0 (0x8000 | (0x80 << nr));
}
static void newsample (int nr, sample8_t sample)
{
struct audio_channel_data *cdp = audio_channel + nr;
#if DEBUG_AUDIO > 0
if (!debugchannel (nr))
sample = 0;
#endif
#if DEBUG_AUDIO > 2
if (debugchannel (nr))
write_log (_T("SAMPLE%d: %02x\n"), nr, sample & 0xff);
#endif
if (!(audio_channel_mask & (1 << nr)))
sample = 0;
cdp->last_sample = cdp->current_sample;
cdp->current_sample = sample;
}
STATIC_INLINE void setdr (int nr)
{
struct audio_channel_data *cdp = audio_channel + nr;
#if TEST_AUDIO > 0
if (debugchannel (nr) && cdp->dr)
write_log (_T("%d: DR already active (STATE=%d)\n"), nr, cdp->state);
#endif
cdp->drhpos = current_hpos ();
cdp->dr = true;
if (cdp->wlen == 1) {
cdp->dsr = true;
#if DEBUG_AUDIO > 0
if (debugchannel (nr) && cdp->wlen > 1)
write_log (_T("DSR%d PT=%08X PC=%08X\n"), nr, cdp->pt, M68K_GETPC);
#endif
}
}
static void loaddat (int nr, bool modper)
{
struct audio_channel_data *cdp = audio_channel + nr;
int audav = adkcon & (0x01 << nr);
int audap = adkcon & (0x10 << nr);
if (audav || (modper && audap)) {
if (nr >= 3)
return;
if (modper && audap) {
if (cdp->dat == 0)
cdp[1].per = 65536 * CYCLE_UNIT;
else if (cdp->dat > PERIOD_MIN)
cdp[1].per = cdp->dat * CYCLE_UNIT;
else
cdp[1].per = PERIOD_MIN * CYCLE_UNIT;
} else if (audav) {
cdp[1].vol = cdp->dat;
cdp[1].vol &= 127;
if (cdp[1].vol > 64)
cdp[1].vol = 64;
}
} else {
#if TEST_AUDIO > 0
if (debugchannel (nr)) {
if (cdp->hisample || cdp->losample)
write_log (_T("%d: high or low sample not used\n"), nr);
if (!cdp->have_dat)
write_log (_T("%d: dat not updated. STATE=%d 1=%04x 2=%04x\n"), nr, cdp->state, cdp->dat, cdp->dat2);
}
cdp->hisample = cdp->losample = true;
cdp->have_dat = false;
#endif
#if DEBUG_AUDIO > 2
if (debugchannel (nr))
write_log (_T("LOAD%dDAT: New:%04x, Old:%04x\n"), nr, cdp->dat, cdp->dat2);
#endif
cdp->dat2 = cdp->dat;
}
}
static void loaddat (int nr)
{
loaddat (nr, false);
}
STATIC_INLINE void loadper (int nr)
{
struct audio_channel_data *cdp = audio_channel + nr;
cdp->evtime = cdp->per;
if (cdp->evtime < CYCLE_UNIT)
write_log (_T("LOADPER%d bug %d\n"), nr, cdp->evtime);
}
static void audio_state_channel2 (int nr, bool perfin)
{
struct audio_channel_data *cdp = audio_channel + nr;
bool chan_ena = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));
bool old_dma = cdp->dmaenstore;
int audav = adkcon & (0x01 << nr);
int audap = adkcon & (0x10 << nr);
int napnav = (!audav && !audap) || audav;
int hpos = current_hpos ();
cdp->dmaenstore = chan_ena;
if (currprefs.produce_sound == 0) {
zerostate (nr);
return;
}
audio_activate ();
if ((cdp->state == 2 || cdp->state == 3) && usehacks()) {
if (!chan_ena && old_dma) {
// DMA switched off, state=2/3 and "too fast CPU": set flag
cdp->dmaofftime_active = true;
}
if (cdp->dmaofftime_active && !old_dma && chan_ena) {
// We are still in state=2/3 and program is going to re-enable
// DMA. Force state to zero to prevent CPU timed DMA wait
// routines in common tracker players to lose notes.
#if DEBUG_AUDIO_HACK > 0
if (debugchannel (nr))
write_log (_T("%d: INSTADMAOFF\n"), nr, M68K_GETPC);
#endif
newsample (nr, (cdp->dat2 >> 0) & 0xff);
// if (napnav)
// setirq (nr, 91);
zerostate (nr);
}
}
#if DEBUG_AUDIO > 0
if (debugchannel (nr) && old_dma != chan_ena) {
write_log (_T("%d:DMA=%d IRQ=%d PC=%08x\n"), nr, chan_ena, isirq (nr) ? 1 : 0, M68K_GETPC);
}
#endif
switch (cdp->state)
{
case 0:
if (chan_ena) {
cdp->evtime = MAX_EV;
cdp->state = 1;
cdp->dr = true;
cdp->drhpos = hpos;
cdp->wlen = cdp->len;
cdp->ptx_written = false;
/* Some programs first start short empty sample and then later switch to
* real sample, we must not enable the hack in this case
*/
if (cdp->wlen > 2)
cdp->ptx_tofetch = true;
cdp->dsr = true;
#if TEST_AUDIO > 0
cdp->have_dat = false;
#endif
#if DEBUG_AUDIO > 0
if (debugchannel (nr)) {
write_log (_T("%d:0>1: LEN=%d PC=%08x\n"), nr, cdp->wlen, M68K_GETPC);
}
#endif
} else if (cdp->dat_written && !isirq (nr)) {
cdp->state = 2;
setirq (nr, 0);
loaddat (nr);
if (usehacks() && cdp->per < 10 * CYCLE_UNIT) {
// make sure audio.device AUDxDAT startup returns to idle state before DMA is enabled
newsample (nr, (cdp->dat2 >> 0) & 0xff);
zerostate (nr);
} else {
cdp->pbufldl = true;
audio_state_channel2 (nr, false);
}
} else {
zerostate (nr);
}
break;
case 1:
cdp->evtime = MAX_EV;
if (!chan_ena) {
zerostate (nr);
return;
}
if (!cdp->dat_written)
return;
#if TEST_AUDIO > 0
if (debugchannel (nr) && !cdp->have_dat)
write_log (_T("%d: state 1 but no have_dat\n"), nr);
cdp->have_dat = false;
cdp->losample = cdp->hisample = false;
#endif
setirq (nr, 10);
setdr (nr);
if (cdp->wlen != 1)
cdp->wlen = (cdp->wlen - 1) & 0xffff;
cdp->state = 5;
if (sampleripper_enabled)
do_samplerip (cdp);
break;
case 5:
cdp->evtime = MAX_EV;
if (!chan_ena) {
zerostate (nr);
return;
}
if (!cdp->dat_written)
return;
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("%d:>5: LEN=%d PT=%08X PC=%08X\n"), nr, cdp->wlen, cdp->pt, M68K_GETPC);
#endif
if (cdp->ptx_written) {
cdp->ptx_written = 0;
cdp->lc = cdp->ptx;
}
loaddat (nr);
if (napnav)
setdr (nr);
cdp->state = 2;
loadper (nr);
cdp->pbufldl = true;
cdp->intreq2 = 0;
audio_state_channel2 (nr, false);
break;
case 2:
if (cdp->pbufldl) {
#if TEST_AUDIO > 0
if (debugchannel (nr) && cdp->hisample == false)
write_log (_T("%d: high sample used twice\n"), nr);
cdp->hisample = false;
#endif
newsample (nr, (cdp->dat2 >> 8) & 0xff);
loadper (nr);
cdp->pbufldl = false;
}
if (!perfin)
return;
if (audap)
loaddat (nr, true);
if (chan_ena) {
if (audap)
setdr (nr);
if (cdp->intreq2 && audap)
setirq (nr, 21);
} else {
if (audap)
setirq (nr, 22);
}
cdp->pbufldl = true;
cdp->state = 3;
audio_state_channel2 (nr, false);
break;
case 3:
if (cdp->pbufldl) {
#if TEST_AUDIO > 0
if (debugchannel (nr) && cdp->losample == false)
write_log (_T("%d: low sample used twice\n"), nr);
cdp->losample = false;
#endif
newsample (nr, (cdp->dat2 >> 0) & 0xff);
loadper (nr);
cdp->pbufldl = false;
}
if (!perfin)
return;
if (chan_ena) {
loaddat (nr);
if (cdp->intreq2 && napnav)
setirq (nr, 31);
if (napnav)
setdr (nr);
} else {
if (isirq (nr)) {
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("%d: IDLE\n"), nr);
#endif
zerostate (nr);
return;
}
loaddat (nr);
if (napnav)
setirq (nr, 32);
}
cdp->intreq2 = 0;
cdp->pbufldl = true;
cdp->state = 2;
audio_state_channel2 (nr, false);
break;
}
}
static void audio_state_cda(void);
static void audio_state_channel (int nr, bool perfin)
{
struct audio_channel_data *cdp = audio_channel + nr;
if (nr < AUDIO_CHANNELS_PAULA) {
audio_state_channel2 (nr, perfin);
cdp->dat_written = false;
} else if (nr == AUDIO_CHANNEL_SNDBOARD_LEFT || nr == AUDIO_CHANNEL_SNDBOARD_RIGHT) {
audio_state_sndboard(nr - AUDIO_CHANNEL_SNDBOARD_LEFT);
} else if (nr == AUDIO_CHANNEL_CDA_LEFT) {
audio_state_cda();
}
}
void audio_state_machine (void)
{
update_audio ();
for (int nr = 0; nr < AUDIO_CHANNELS_PAULA; nr++) {
struct audio_channel_data *cdp = audio_channel + nr;
audio_state_channel2 (nr, false);
cdp->dat_written = false;
}
schedule_audio ();
events_schedule ();
}
void audio_reset (void)
{
int i;
struct audio_channel_data *cdp;
#ifdef AHI
ahi_close_sound ();
free_ahi_v2 ();
#endif
reset_sound ();
memset (sound_filter_state, 0, sizeof sound_filter_state);
if (!isrestore ()) {
for (i = 0; i < AUDIO_CHANNELS_MAX; i++) {
cdp = &audio_channel[i];
memset (cdp, 0, sizeof *audio_channel);
cdp->per = PERIOD_MAX - 1;
cdp->vol = 0;
cdp->evtime = MAX_EV;
}
}
last_cycles = get_cycles ();
next_sample_evtime = scaled_sample_evtime;
schedule_audio ();
events_schedule ();
}
static int sound_prefs_changed (void)
{
if (!config_changed)
return 0;
if (changed_prefs.produce_sound != currprefs.produce_sound
|| changed_prefs.win32_soundcard != currprefs.win32_soundcard
|| changed_prefs.sound_stereo != currprefs.sound_stereo
|| changed_prefs.sound_maxbsiz != currprefs.sound_maxbsiz
|| changed_prefs.sound_freq != currprefs.sound_freq
|| changed_prefs.sound_auto != currprefs.sound_auto)
return 1;
if (changed_prefs.sound_stereo_separation != currprefs.sound_stereo_separation
|| changed_prefs.sound_mixed_stereo_delay != currprefs.sound_mixed_stereo_delay
|| changed_prefs.sound_interpol != currprefs.sound_interpol
|| changed_prefs.sound_volume_paula != currprefs.sound_volume_paula
|| changed_prefs.sound_volume_cd != currprefs.sound_volume_cd
|| changed_prefs.sound_volume_master != currprefs.sound_volume_master
|| changed_prefs.sound_volume_board != currprefs.sound_volume_board
|| changed_prefs.sound_stereo_swap_paula != currprefs.sound_stereo_swap_paula
|| changed_prefs.sound_stereo_swap_ahi != currprefs.sound_stereo_swap_ahi
|| changed_prefs.sound_cdaudio != currprefs.sound_cdaudio
|| changed_prefs.sound_filter != currprefs.sound_filter
|| changed_prefs.sound_filter_type != currprefs.sound_filter_type)
return -1;
return 0;
}
/* This computes the 1st order low-pass filter term b0.
* The a1 term is 1.0 - b0. The center frequency marks the -3 dB point. */
#ifndef M_PI
#define M_PI 3.14159265358979323846
#endif
static float rc_calculate_a0 (int sample_rate, int cutoff_freq)
{
float omega;
/* The BLT correction formula below blows up if the cutoff is above nyquist. */
if (cutoff_freq >= sample_rate / 2)
return 1.0;
omega = 2 * M_PI * cutoff_freq / sample_rate;
/* Compensate for the bilinear transformation. This allows us to specify the
* stop frequency more exactly, but the filter becomes less steep further
* from stopband. */
omega = tan (omega / 2) * 2;
return 1 / (1 + 1 / omega);
}
void check_prefs_changed_audio (void)
{
int ch;
if (sound_available) {
ch = sound_prefs_changed ();
if (ch > 0) {
#ifdef AVIOUTPUT
AVIOutput_Restart ();
#endif
clear_sound_buffers ();
}
if (ch) {
set_audio ();
audio_activate ();
}
}
#ifdef DRIVESOUND
driveclick_check_prefs ();
#endif
}
static void set_extra_prehandler(void)
{
if (audio_channel_count > AUDIO_CHANNELS_PAULA && sample_prehandler != anti_prehandler) {
extra_sample_prehandler = anti_prehandler;
} else {
extra_sample_prehandler = NULL;
}
}
void set_audio (void)
{
int old_mixed_size = mixed_stereo_size;
int sep, delay;
int ch;
ch = sound_prefs_changed ();
if (ch >= 0)
close_sound ();
currprefs.produce_sound = changed_prefs.produce_sound;
currprefs.win32_soundcard = changed_prefs.win32_soundcard;
currprefs.sound_stereo = changed_prefs.sound_stereo;
currprefs.sound_auto = changed_prefs.sound_auto;
currprefs.sound_freq = changed_prefs.sound_freq;
currprefs.sound_maxbsiz = changed_prefs.sound_maxbsiz;
currprefs.sound_stereo_separation = changed_prefs.sound_stereo_separation;
currprefs.sound_mixed_stereo_delay = changed_prefs.sound_mixed_stereo_delay;
currprefs.sound_interpol = changed_prefs.sound_interpol;
currprefs.sound_filter = changed_prefs.sound_filter;
currprefs.sound_filter_type = changed_prefs.sound_filter_type;
currprefs.sound_volume_paula = changed_prefs.sound_volume_paula;
currprefs.sound_volume_master = changed_prefs.sound_volume_master;
currprefs.sound_volume_board = changed_prefs.sound_volume_board;
currprefs.sound_volume_cd = changed_prefs.sound_volume_cd;
currprefs.sound_cdaudio = changed_prefs.sound_cdaudio;
currprefs.sound_stereo_swap_paula = changed_prefs.sound_stereo_swap_paula;
currprefs.sound_stereo_swap_ahi = changed_prefs.sound_stereo_swap_ahi;
sound_cd_volume[0] = sound_cd_volume[1] = (100 - (currprefs.sound_volume_cd < 0 ? 0 : currprefs.sound_volume_cd)) * 32768 / 100;
sound_paula_volume[0] = sound_paula_volume[1] = (100 - currprefs.sound_volume_paula) * 32768 / 100;
sndboard_ext_volume();
if (ch >= 0) {
if (currprefs.produce_sound >= 2) {
if (!init_audio ()) {
if (! sound_available) {
write_log (_T("Sound is not supported.\n"));
} else {
write_log (_T("Sorry, can't initialize sound.\n"));
currprefs.produce_sound = 1;
/* So we don't do this every frame */
changed_prefs.produce_sound = 1;
}
}
}
next_sample_evtime = scaled_sample_evtime;
last_cycles = get_cycles ();
compute_vsynctime ();
} else {
sound_volume (0);
}
sep = (currprefs.sound_stereo_separation = changed_prefs.sound_stereo_separation) * 3 / 2;
if (sep >= 15)
sep = 16;
delay = currprefs.sound_mixed_stereo_delay = changed_prefs.sound_mixed_stereo_delay;
mixed_mul1 = MIXED_STEREO_SCALE / 2 - sep;
mixed_mul2 = MIXED_STEREO_SCALE / 2 + sep;
mixed_stereo_size = delay > 0 ? (1 << delay) - 1 : 0;
mixed_on = sep < MIXED_STEREO_MAX || mixed_stereo_size > 0;
if (mixed_on && old_mixed_size != mixed_stereo_size) {
saved_ptr = 0;
memset (right_word_saved, 0, sizeof right_word_saved);
}
led_filter_forced = -1; // always off
sound_use_filter = sound_use_filter_sinc = 0;
if (currprefs.sound_filter) {
if (currprefs.sound_filter == FILTER_SOUND_ON)
led_filter_forced = 1;
if (currprefs.sound_filter == FILTER_SOUND_EMUL)
led_filter_forced = 0;
if (currprefs.sound_filter_type == FILTER_SOUND_TYPE_A500)
sound_use_filter = FILTER_MODEL_A500;
else if (currprefs.sound_filter_type == FILTER_SOUND_TYPE_A1200)
sound_use_filter = FILTER_MODEL_A1200;
}
a500e_filter1_a0 = rc_calculate_a0 (currprefs.sound_freq, 6200);
a500e_filter2_a0 = rc_calculate_a0 (currprefs.sound_freq, 20000);
filter_a0 = rc_calculate_a0 (currprefs.sound_freq, 7000);
led_filter_audio ();
/* Select the right interpolation method. */
if (sample_handler == sample16_handler
|| sample_handler == sample16i_crux_handler
|| sample_handler == sample16i_rh_handler
|| sample_handler == sample16i_sinc_handler
|| sample_handler == sample16i_anti_handler)
{
sample_handler = (currprefs.sound_interpol == 0 ? sample16_handler
: currprefs.sound_interpol == 3 ? sample16i_rh_handler
: currprefs.sound_interpol == 4 ? sample16i_crux_handler
: currprefs.sound_interpol == 2 ? sample16i_sinc_handler
: sample16i_anti_handler);
} else if (sample_handler == sample16s_handler
|| sample_handler == sample16si_crux_handler
|| sample_handler == sample16si_rh_handler
|| sample_handler == sample16si_sinc_handler
|| sample_handler == sample16si_anti_handler)
{
sample_handler = (currprefs.sound_interpol == 0 ? sample16s_handler
: currprefs.sound_interpol == 3 ? sample16si_rh_handler
: currprefs.sound_interpol == 4 ? sample16si_crux_handler
: currprefs.sound_interpol == 2 ? sample16si_sinc_handler
: sample16si_anti_handler);
} else if (sample_handler == sample16ss_handler
|| sample_handler == sample16ss_sinc_handler
|| sample_handler == sample16ss_anti_handler)
{
sample_handler = (currprefs.sound_interpol == 0 ? sample16ss_handler
: currprefs.sound_interpol == 3 ? sample16ss_handler
: currprefs.sound_interpol == 4 ? sample16ss_handler
: currprefs.sound_interpol == 2 ? sample16ss_sinc_handler
: sample16ss_anti_handler);
}
sample_prehandler = NULL;
if (sample_handler == sample16si_sinc_handler || sample_handler == sample16i_sinc_handler || sample_handler == sample16ss_sinc_handler) {
sample_prehandler = sinc_prehandler_paula;
sound_use_filter_sinc = sound_use_filter;
sound_use_filter = 0;
} else if (sample_handler == sample16si_anti_handler || sample_handler == sample16i_anti_handler || sample_handler == sample16ss_anti_handler) {
sample_prehandler = anti_prehandler;
}
set_extra_prehandler();
if (currprefs.produce_sound == 0) {
eventtab[ev_audio].active = 0;
events_schedule ();
} else {
audio_activate ();
schedule_audio ();
events_schedule ();
}
set_config_changed ();
}
void update_audio (void)
{
unsigned long int n_cycles = 0;
#if SOUNDSTUFF > 1
static int samplecounter;
#endif
if (!isaudio ())
goto end;
if (isrestore ())
goto end;
if (!is_audio_active ())
goto end;
n_cycles = get_cycles () - last_cycles;
while (n_cycles > 0) {
unsigned long int best_evtime = n_cycles + 1;
unsigned long rounded;
int i;
for (i = 0; i < audio_channel_count; i++) {
if (audio_channel[i].evtime != MAX_EV && best_evtime > audio_channel[i].evtime)
best_evtime = audio_channel[i].evtime;
}
/* next_sample_evtime >= 0 so floor() behaves as expected */
rounded = floorf (next_sample_evtime);
if ((next_sample_evtime - rounded) >= 0.5)
rounded++;
if (currprefs.produce_sound > 1 && best_evtime > rounded)
best_evtime = rounded;
if (best_evtime > n_cycles)
best_evtime = n_cycles;
/* Decrease time-to-wait counters */
next_sample_evtime -= best_evtime;
if (currprefs.produce_sound > 1) {
if (sample_prehandler)
sample_prehandler (best_evtime / CYCLE_UNIT);
if (extra_sample_prehandler)
extra_sample_prehandler(best_evtime / CYCLE_UNIT);
}
for (i = 0; i < audio_channel_count; i++) {
if (audio_channel[i].evtime != MAX_EV)
audio_channel[i].evtime -= best_evtime;
}
n_cycles -= best_evtime;
if (currprefs.produce_sound > 1) {
/* Test if new sample needs to be outputted */
if (rounded == best_evtime) {
/* Before the following addition, next_sample_evtime is in range [-0.5, 0.5) */
next_sample_evtime += scaled_sample_evtime;
#if SOUNDSTUFF > 1
next_sample_evtime -= extrasamples * 15;
doublesample = 0;
if (--samplecounter <= 0) {
samplecounter = currprefs.sound_freq / 1000;
if (extrasamples > 0) {
outputsample = 1;
doublesample = 1;
extrasamples--;
} else if (extrasamples < 0) {
outputsample = 0;
doublesample = 0;
extrasamples++;
}
}
#endif
(*sample_handler) ();
#if SOUNDSTUFF > 1
if (outputsample == 0)
outputsample = -1;
else if (outputsample < 0)
outputsample = 1;
#endif
}
}
for (i = 0; i < audio_channel_count; i++) {
if (audio_channel[i].evtime == 0) {
audio_state_channel (i, true);
if (audio_channel[i].evtime == 0) {
write_log (_T("evtime==0 sound bug channel %d\n"), i);
audio_channel[i].evtime = MAX_EV;
}
}
}
}
end:
last_cycles = get_cycles () - n_cycles;
}
void audio_evhandler (void)
{
update_audio ();
schedule_audio ();
}
void audio_hsync (void)
{
if (!currprefs.produce_sound)
return;
if (!isaudio ())
return;
if (audio_work_to_do > 0 && currprefs.sound_auto && !audio_extra_channels[0] && !audio_extra_channels[1]
#ifdef AVIOUTPUT
&& !avioutput_enabled
#endif
) {
audio_work_to_do--;
if (audio_work_to_do == 0)
audio_deactivate ();
}
update_audio ();
}
void AUDxDAT (int nr, uae_u16 v, uaecptr addr)
{
struct audio_channel_data *cdp = audio_channel + nr;
int chan_ena = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));
#if DEBUG_AUDIO > 0
if (debugchannel (nr) && (DEBUG_AUDIO > 1 || (!chan_ena || addr == 0xffffffff || (cdp->state != 2 && cdp->state != 3)))) {
write_log (_T("AUD%dDAT: %04X ADDR=%08X LEN=%d/%d %d,%d,%d %06X\n"), nr,
v, addr, cdp->wlen, cdp->len, cdp->state, chan_ena, isirq (nr) ? 1 : 0, M68K_GETPC);
}
#endif
cdp->dat = v;
cdp->dat_written = true;
#if TEST_AUDIO > 0
if (debugchannel (nr) && cdp->have_dat)
write_log (_T("%d: audxdat 1=%04x 2=%04x but old dat not yet used\n"), nr, cdp->dat, cdp->dat2);
cdp->have_dat = true;
#endif
if (cdp->state == 2 || cdp->state == 3) {
if (chan_ena) {
if (cdp->wlen == 1) {
cdp->wlen = cdp->len;
cdp->intreq2 = true;
if (sampleripper_enabled)
do_samplerip (cdp);
#if DEBUG_AUDIO > 0
if (debugchannel (nr) && cdp->wlen > 1)
write_log (_T("AUD%d looped, IRQ=%d, LC=%08X LEN=%d\n"), nr, isirq (nr) ? 1 : 0, cdp->pt, cdp->wlen);
#endif
} else {
cdp->wlen = (cdp->wlen - 1) & 0xffff;
}
}
} else {
audio_activate ();
update_audio ();
audio_state_channel (nr, false);
schedule_audio ();
events_schedule ();
}
cdp->dat_written = false;
}
void AUDxDAT (int nr, uae_u16 v)
{
AUDxDAT (nr, v, 0xffffffff);
}
uaecptr audio_getpt (int nr, bool reset)
{
struct audio_channel_data *cdp = audio_channel + nr;
uaecptr p = cdp->pt;
cdp->pt += 2;
if (reset)
cdp->pt = cdp->lc;
cdp->ptx_tofetch = false;
return p;
}
void AUDxLCH (int nr, uae_u16 v)
{
struct audio_channel_data *cdp = audio_channel + nr;
audio_activate ();
update_audio ();
// someone wants to update PT but DSR has not yet been processed.
// too fast CPU and some tracker players: enable DMA, CPU delay, update AUDxPT with loop position
if (usehacks() && ((cdp->ptx_tofetch && cdp->state == 1) || cdp->ptx_written)) {
cdp->ptx = cdp->lc;
cdp->ptx_written = true;
#if DEBUG_AUDIO_HACK > 0
if (debugchannel (nr))
write_log (_T("AUD%dLCH HACK: %04X %08X (%d) (%d %d %08x)\n"), nr, v, M68K_GETPC, cdp->state, cdp->dsr, cdp->ptx_written, cdp->ptx);
#endif
} else {
cdp->lc = (cdp->lc & 0xffff) | ((uae_u32)v << 16);
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("AUD%dLCH: %04X %08X (%d) (%d %d %08x)\n"), nr, v, M68K_GETPC, cdp->state, cdp->dsr, cdp->ptx_written, cdp->ptx);
#endif
}
}
void AUDxLCL (int nr, uae_u16 v)
{
struct audio_channel_data *cdp = audio_channel + nr;
audio_activate ();
update_audio ();
if (usehacks() && ((cdp->ptx_tofetch && cdp->state == 1) || cdp->ptx_written)) {
cdp->ptx = cdp->lc;
cdp->ptx_written = true;
#if DEBUG_AUDIO_HACK > 0
if (debugchannel (nr))
write_log (_T("AUD%dLCL HACK: %04X %08X (%d) (%d %d %08x)\n"), nr, v, M68K_GETPC, cdp->state, cdp->dsr, cdp->ptx_written, cdp->ptx);
#endif
} else {
cdp->lc = (cdp->lc & ~0xffff) | (v & 0xFFFE);
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("AUD%dLCL: %04X %08X (%d) (%d %d %08x)\n"), nr, v, M68K_GETPC, cdp->state, cdp->dsr, cdp->ptx_written, cdp->ptx);
#endif
}
}
void AUDxPER (int nr, uae_u16 v)
{
struct audio_channel_data *cdp = audio_channel + nr;
unsigned long per;
audio_activate ();
update_audio ();
per = v * CYCLE_UNIT;
if (per == 0)
per = PERIOD_MAX - 1;
if (per < PERIOD_MIN * CYCLE_UNIT) {
/* smaller values would cause extremely high cpu usage */
per = PERIOD_MIN * CYCLE_UNIT;
}
if (per < PERIOD_MIN_NONCE * CYCLE_UNIT && !currprefs.cpu_cycle_exact && cdp->dmaenstore) {
/* DMAL emulation and low period can cause very very high cpu usage on slow performance PCs
* Only do this hack if audio DMA is active.
*/
per = PERIOD_MIN_NONCE * CYCLE_UNIT;
}
if (cdp->per == PERIOD_MAX - 1 && per != PERIOD_MAX - 1) {
cdp->evtime = CYCLE_UNIT;
if (isaudio ()) {
schedule_audio ();
events_schedule ();
}
}
#if TEST_AUDIO > 0
cdp->per_original = v;
#endif
cdp->per = per;
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("AUD%dPER: %d %08X\n"), nr, v, M68K_GETPC);
#endif
}
void AUDxLEN (int nr, uae_u16 v)
{
struct audio_channel_data *cdp = audio_channel + nr;
audio_activate ();
update_audio ();
cdp->len = v;
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("AUD%dLEN: %d %08X\n"), nr, v, M68K_GETPC);
#endif
}
void AUDxVOL (int nr, uae_u16 v)
{
struct audio_channel_data *cdp = audio_channel + nr;
// 7 bit register in Paula.
v &= 127;
if (v > 64)
v = 64;
audio_activate ();
update_audio ();
cdp->vol = v;
#if DEBUG_AUDIO > 0
if (debugchannel (nr))
write_log (_T("AUD%dVOL: %d %08X\n"), nr, v, M68K_GETPC);
#endif
}
void audio_update_adkmasks (void)
{
static int prevcon = -1;
unsigned long t = adkcon | (adkcon >> 4);
audio_channel[0].adk_mask = (((t >> 0) & 1) - 1);
audio_channel[1].adk_mask = (((t >> 1) & 1) - 1);
audio_channel[2].adk_mask = (((t >> 2) & 1) - 1);
audio_channel[3].adk_mask = (((t >> 3) & 1) - 1);
if ((prevcon & 0xff) != (adkcon & 0xff)) {
audio_activate ();
#if DEBUG_AUDIO > 0
write_log (_T("ADKCON=%02x %08X\n"), adkcon & 0xff, M68K_GETPC);
#endif
prevcon = adkcon;
}
}
int init_audio (void)
{
return init_sound ();
}
void led_filter_audio (void)
{
led_filter_on = 0;
if (led_filter_forced > 0 || (gui_data.powerled && led_filter_forced >= 0))
led_filter_on = 1;
}
void audio_vsync (void)
{
#if 0
#if SOUNDSTUFF > 0
int max, min;
int vsync = isvsync ();
static int lastdir;
if (1 || !vsync) {
extrasamples = 0;
return;
}
min = -10 * 10;
max = vsync ? 10 * 10 : 20 * 10;
extrasamples = 0;
if (gui_data.sndbuf < min) { // +1
extrasamples = (min - gui_data.sndbuf) / 10;
lastdir = 1;
} else if (gui_data.sndbuf > max) { // -1
extrasamples = (max - gui_data.sndbuf) / 10;
} else if (gui_data.sndbuf > 1 * 50 && lastdir < 0) {
extrasamples--;
} else if (gui_data.sndbuf < -1 * 50 && lastdir > 0) {
extrasamples++;
} else {
lastdir = 0;
}
if (extrasamples > 99)
extrasamples = 99;
if (extrasamples < -99)
extrasamples = -99;
#endif
#endif
}
void restore_audio_finish (void)
{
last_cycles = get_cycles ();
schedule_audio ();
events_schedule ();
}
uae_u8 *restore_audio (int nr, uae_u8 *src)
{
struct audio_channel_data *acd = audio_channel + nr;
zerostate (nr);
acd->state = restore_u8 ();
acd->vol = restore_u8 ();
acd->intreq2 = restore_u8 () ? true : false;
uae_u8 flags = restore_u8 ();
acd->dr = acd->dsr = false;
if (flags & 1)
acd->dr = true;
if (flags & 2)
acd->dsr = true;
acd->len = restore_u16 ();
acd->wlen = restore_u16 ();
uae_u16 p = restore_u16 ();
acd->per = p ? p * CYCLE_UNIT : PERIOD_MAX;
acd->dat = acd->dat2 = restore_u16 ();
acd->lc = restore_u32 ();
acd->pt = restore_u32 ();
acd->evtime = restore_u32 ();
if (flags & 0x80)
acd->drhpos = restore_u8 ();
else
acd->drhpos = 1;
acd->dmaenstore = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));
return src;
}
uae_u8 *save_audio (int nr, int *len, uae_u8 *dstptr)
{
struct audio_channel_data *acd = audio_channel + nr;
uae_u8 *dst, *dstbak;
if (dstptr)
dstbak = dst = dstptr;
else
dstbak = dst = xmalloc (uae_u8, 100);
save_u8 (acd->state);
save_u8 (acd->vol);
save_u8 (acd->intreq2);
save_u8 ((acd->dr ? 1 : 0) | (acd->dsr ? 2 : 0) | 0x80);
save_u16 (acd->len);
save_u16 (acd->wlen);
save_u16 (acd->per == PERIOD_MAX ? 0 : acd->per / CYCLE_UNIT);
save_u16 (acd->dat);
save_u32 (acd->lc);
save_u32 (acd->pt);
save_u32 (acd->evtime);
save_u8 (acd->drhpos);
*len = dst - dstbak;
return dstbak;
}
static void audio_set_extra_channels(void)
{
audio_channel_count = audio_extra_channels[1] ? AUDIO_CHANNELS_PAULA + 4 : (audio_extra_channels[0] ? AUDIO_CHANNELS_PAULA + 2 : AUDIO_CHANNELS_PAULA);
set_extra_prehandler();
}
void audio_enable_sndboard(bool enable)
{
struct audio_channel_data *acd = audio_channel + AUDIO_CHANNEL_SNDBOARD_LEFT;
if (!enable) {
audio_extra_channels[0] = false;
acd[1].evtime = acd->evtime = MAX_EV;
} else {
audio_extra_channels[0] = true;
acd[1].evtime = acd->evtime = CYCLE_UNIT;
acd[1].adk_mask = acd->adk_mask = 0xffffffff;
acd[1].vol = acd->vol = 1;
audio_activate();
}
audio_set_extra_channels();
}
static void audio_enable_cda(bool enable)
{
struct audio_channel_data *acd = audio_channel + AUDIO_CHANNEL_CDA_LEFT;
if (!enable) {
audio_extra_channels[1] = false;
acd[1].evtime = acd->evtime = MAX_EV;
} else {
audio_extra_channels[1] = true;
acd[1].evtime = acd->evtime = CYCLE_UNIT;
acd[1].adk_mask = acd->adk_mask = 0xffffffff;
acd[1].vol = acd->vol = 1;
audio_activate();
}
audio_set_extra_channels();
}
void audio_state_sndboard_state(int ch, int sample, unsigned int evt)
{
struct audio_channel_data *acd = audio_channel + AUDIO_CHANNEL_SNDBOARD_LEFT + ch;
acd->last_sample = acd->current_sample;
acd->current_sample = sample;
acd->evtime = evt;
}
static uae_s16 *cda_bufptr;
static int cda_length, cda_userdata;
static unsigned int cda_evt;
static uae_s16 dummy_buffer[4] = { 0 };
static CDA_CALLBACK cda_next_cd_audio_buffer_callback;
static int cda_volume[2];
void update_cda_sound(double clk)
{
cda_evt = clk * CYCLE_UNIT / 44100;
}
void audio_cda_volume(int left, int right)
{
for (int j = 0; j < 2; j++) {
cda_volume[j] = j == 0 ? left : right;
cda_volume[j] = sound_cd_volume[j] * cda_volume[j] / 32768;
if (cda_volume[j])
cda_volume[j]++;
if (cda_volume[j] >= 32768)
cda_volume[j] = 32768;
}
}
void audio_cda_new_buffer(uae_s16 *buffer, int length, int userdata, CDA_CALLBACK next_cd_audio_buffer_callback)
{
if (!buffer) {
cda_bufptr = dummy_buffer;
cda_length = 0;
} else {
cda_bufptr = buffer;
cda_length = length;
cda_userdata = userdata;
if (!audio_extra_channels[1])
audio_enable_cda(true);
}
cda_next_cd_audio_buffer_callback = next_cd_audio_buffer_callback;
audio_activate();
}
static void audio_state_cda(void)
{
if (cda_bufptr >= dummy_buffer && cda_bufptr <= dummy_buffer + 4) {
audio_enable_cda(false);
return;
}
struct audio_channel_data *acd = audio_channel + AUDIO_CHANNEL_CDA_LEFT;
acd->last_sample = acd->current_sample;
acd->current_sample = cda_bufptr[0] * cda_volume[0] / 32768;
acd->evtime = cda_evt;
acd++;
acd->last_sample = acd->current_sample;
acd->current_sample = cda_bufptr[1] * cda_volume[1] / 32768;
acd->evtime = cda_evt;
cda_bufptr += 2;
cda_length--;
if (cda_length <= 0 && cda_next_cd_audio_buffer_callback) {
cda_next_cd_audio_buffer_callback(cda_userdata);
}
}
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