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audio.cpp
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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"
#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 MAX_EV ~0ul
//#define DEBUG_AUDIO
#define DEBUG_CHANNEL_MASK 15
//#define TEST_AUDIO
#define PERIOD_MIN 4
int audio_channel_mask = 15;
STATIC_INLINE bool isaudio (void)
{
return currprefs.produce_sound != 0;
}
static bool debugchannel (int ch)
{
return ((1 << ch) & DEBUG_CHANNEL_MASK) != 0;
}
STATIC_INLINE bool usehacks (void)
{
return currprefs.cpu_model >= 68020 || currprefs.m68k_speed != 0;
}
#define SINC_QUEUE_MAX_AGE 2048
/* Queue length 128 implies minimum emulated period of 16. I add a few extra
* entries so that CPU updates during minimum period can be played back. */
#define SINC_QUEUE_LENGTH (SINC_QUEUE_MAX_AGE / 16 + 2)
#include "sinctable.cpp"
typedef struct {
int age, output;
} sinc_queue_t;
struct audio_channel_data {
unsigned long adk_mask;
unsigned long 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_length;
#ifdef TEST_AUDIO
bool hisample, losample;
bool have_dat;
int per_original;
#endif
};
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[] = L"_";
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));
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, L"wav");
_stprintf (filename, L"%s%s%s%03.3d.%s", path, name, underline, cnt, extension);
wavfile = zfile_fopen (filename, L"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 (L"SAMPLERIPPER: %d: %dHz %d bytes\n", cnt, freq, rs->len);
} else {
write_log (L"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->len * 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 (L"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 (L"SAMPLERIPPER: sample added (%06X, %d bytes), total %d samples\n", adp->pt, len, ++cnt);
audio_sampleripper (0);
}
static struct audio_channel_data audio_channel[4];
int sound_available = 0;
void (*sample_handler) (void);
static void (*sample_prehandler) (unsigned long best_evtime);
static float sample_evtime;
float scaled_sample_evtime;
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 logn)
{
if (14 - bits + logn > 0) {
data >>= 14 - bits + logn;
} else {
int shift = bits - 14 - logn;
int right = data & ((1 << shift) - 1);
data <<= shift;
data |= right;
}
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[4];
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_NONE:
return input;
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;
}
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_INLINE void put_sound_word_right (uae_u32 w)
{
if (mixed_on) {
right_word_saved[saved_ptr] = w;
return;
}
PUT_SOUND_WORD_RIGHT (w);
}
STATIC_INLINE 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_LEFT (w);
PUT_SOUND_WORD_RIGHT (tmp);
} else {
PUT_SOUND_WORD_LEFT (w);
}
}
STATIC_INLINE void put_sound_word_right2 (uae_u32 w)
{
if (mixed_on) {
right2_word_saved[saved_ptr2] = w;
return;
}
PUT_SOUND_WORD_RIGHT2 (w);
}
STATIC_INLINE 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_LEFT2 (w);
PUT_SOUND_WORD_RIGHT2 (tmp);
} else {
PUT_SOUND_WORD_LEFT2 (w);
}
}
#define DO_CHANNEL(v, c) do { (v) &= audio_channel[c].adk_mask; data += v; } while (0);
static void anti_prehandler (unsigned long best_evtime)
{
int i, output;
struct audio_channel_data *acd;
/* Handle accumulator antialiasiation */
for (i = 0; i < 4; 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_INLINE void samplexx_anti_handler (int *datasp)
{
int i;
for (i = 0; i < 4; i++) {
datasp[i] = 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 (unsigned long best_evtime)
{
int i, j, output;
struct audio_channel_data *acd;
for (i = 0; i < 4; i++) {
acd = &audio_channel[i];
output = (acd->current_sample * acd->vol) & acd->adk_mask;
/* age the sinc queue and truncate it when necessary */
for (j = 0; j < acd->sinc_queue_length; j += 1) {
acd->sinc_queue[j].age += best_evtime;
if (acd->sinc_queue[j].age >= SINC_QUEUE_MAX_AGE) {
acd->sinc_queue_length = j;
break;
}
}
/* 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) {
if (acd->sinc_queue_length > SINC_QUEUE_LENGTH - 1) {
//write_log (L"warning: sinc queue truncated. Last age: %d.\n", acd->sinc_queue[SINC_QUEUE_LENGTH-1].age);
acd->sinc_queue_length = SINC_QUEUE_LENGTH - 1;
}
/* make room for new and add the new value */
memmove (&acd->sinc_queue[1], &acd->sinc_queue[0],
sizeof (acd->sinc_queue[0]) * acd->sinc_queue_length);
acd->sinc_queue_length += 1;
acd->sinc_queue[0].age = best_evtime;
acd->sinc_queue[0].output = output - acd->sinc_output_state;
acd->sinc_output_state = output;
}
}
}
/* this interpolator performs BLEP mixing (bleps are shaped like integrated sinc
* functions) with a type of BLEP that matches the filtering configuration. */
STATIC_INLINE void samplexx_sinc_handler (int *datasp)
{
int i, n;
int const *winsinc;
if (sound_use_filter_sinc) {
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 = 0; i < 4; i += 1) {
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 */
for (j = 0; j < acd->sinc_queue_length; j += 1)
sum -= winsinc[acd->sinc_queue[j].age] * acd->sinc_queue[j].output;
v = sum >> 17;
if (v > 32767)
v = 32767;
else if (v < -32768)
v = -32768;
datasp[i] = v;
}
}
static void sample16i_sinc_handler (void)
{
int datas[4], data1;
samplexx_sinc_handler (datas);
data1 = datas[0] + datas[3] + datas[1] + datas[2];
data1 = FINISH_DATA (data1, 16, 2);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data1);
check_sound_buffers ();
}
void sample16_handler (void)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 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, 2);
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[4], data1;
samplexx_anti_handler (datas);
data1 = datas[0] + datas[3] + datas[1] + datas[2];
data1 = FINISH_DATA (data1, 16, 2);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data1);
check_sound_buffers ();
}
static void sample16i_rh_handler (void)
{
unsigned long delta, ratio;
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 data3p = audio_channel[3].last_sample;
uae_u32 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, 2);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data);
check_sound_buffers ();
}
static void sample16i_crux_handler (void)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 data3p = audio_channel[3].last_sample;
uae_u32 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, 2);
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)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 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, 16, 0);
data1 = FINISH_DATA (data1, 16, 0);
data2 = FINISH_DATA (data2, 16, 0);
data3 = FINISH_DATA (data3, 16, 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 */
void sample16ss_anti_handler (void)
{
int data0, data1, data2, data3;
int datas[4];
samplexx_anti_handler (datas);
data0 = FINISH_DATA (datas[0], 16, 0);
data1 = FINISH_DATA (datas[1], 16, 0);
data2 = FINISH_DATA (datas[2], 16, 0);
data3 = FINISH_DATA (datas[3], 16, 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 ();
}
static void sample16si_anti_handler (void)
{
int datas[4], data1, data2;
samplexx_anti_handler (datas);
data1 = datas[0] + datas[3];
data2 = datas[1] + datas[2];
data1 = FINISH_DATA (data1, 16, 1);
data2 = FINISH_DATA (data2, 16, 1);
set_sound_buffers ();
put_sound_word_left (data1);
put_sound_word_right (data2);
check_sound_buffers ();
}
void sample16ss_sinc_handler (void)
{
int data0, data1, data2, data3;
int datas[4];
samplexx_sinc_handler (datas);
data0 = FINISH_DATA (datas[0], 16, 0);
data1 = FINISH_DATA (datas[1], 16, 0);
data2 = FINISH_DATA (datas[2], 16, 0);
data3 = FINISH_DATA (datas[3], 16, 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 ();
}
static void sample16si_sinc_handler (void)
{
int datas[4], data1, data2;
samplexx_sinc_handler (datas);
data1 = datas[0] + datas[3];
data2 = datas[1] + datas[2];
data1 = FINISH_DATA (data1, 16, 1);
data2 = FINISH_DATA (data2, 16, 1);
set_sound_buffers ();
put_sound_word_left (data1);
put_sound_word_right (data2);
check_sound_buffers ();
}
void sample16s_handler (void)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 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, 16, 1);
data3 = SBASEVAL16(1) + data1;
data3 = FINISH_DATA (data3, 16, 1);
set_sound_buffers ();
put_sound_word_left (data2);
put_sound_word_right (data3);
check_sound_buffers ();
}
static void sample16si_crux_handler (void)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 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, 16, 1);
data3 = SBASEVAL16(1) + data1;
data3 = FINISH_DATA (data3, 16, 1);
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;
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 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;