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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"
#ifdef DRIVESOUND
#include "driveclick.h"
#endif
#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_AUDIO2 0
#define DEBUG_AUDIO_HACK 0
#define DEBUG_CHANNEL_MASK 15
#define TEST_AUDIO 0
#define TEST_MISSED_DMA 0
#define TEST_MANUAL_AUDIO 0
#define PERIOD_MIN 1
#define PERIOD_MIN_NONCE 60
#define PERIOD_MIN_LOOP 16
#define PERIOD_MIN_LOOP_COUNT 64
#define PERIOD_LOW 124
int audio_channel_mask = 15;
volatile bool cd_audio_mode_changed;
STATIC_INLINE bool isaudio (void)
{
return currprefs.produce_sound != 0;
}
#if DEBUG_AUDIO > 0 || DEBUG_AUDIO_HACK > 0 || DEBUG_AUDIO2 > 0
static bool debugchannel (int ch)
{
return ((1 << ch) & DEBUG_CHANNEL_MASK) != 0;
}
#endif
STATIC_INLINE bool usehacks(void)
{
return !(currprefs.cs_hacks & 8) && (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_data2
{
int current_sample, last_sample;
uae_u8 new_sample;
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;
int audvol;
int mixvol;
unsigned int adk_mask;
};
struct audio_stream_data
{
bool active;
unsigned int evtime;
struct audio_channel_data2 data[AUDIO_CHANNEL_MAX_STREAM_CH];
SOUND_STREAM_CALLBACK cb;
void *cb_data;
};
#define FIR_WIDTH 512
#define VOLCNT_BUFFER_SIZE 4096
union sIntFlt { uae_u32 U32; float F32; };
static float firmem[2 * FIR_WIDTH + 1];
struct audio_channel_data
{
uae_u32 evtime;
bool dmaenstore;
bool intreq2;
int irqcheck;
bool dr;
bool dsr;
bool pbufldl;
int drhpos;
bool dat_written;
#if TEST_MISSED_DMA
bool dat_loaded;
#endif
#if TEST_MANUAL_AUDIO
bool mdat_loaded;
#endif
uaecptr lc, pt;
int state;
int per;
int len, wlen;
int volcnt;
uae_u16 dat, dat2;
struct audio_channel_data2 data;
#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;
int dmaofftime_cpu_cnt;
uaecptr dmaofftime_pc;
int minperloop;
int volcntbufcnt;
float volcntbuf[VOLCNT_BUFFER_SIZE];
};
static int audio_extra_streams[AUDIO_CHANNEL_STREAMS];
static int audio_total_extra_streams;
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;
uaecptr pt;
int len, per, changed;
};
static struct ripped_sample *ripped_samples;
void write_wavheader (struct zfile *wavfile, size_t size, uae_u32 freq)
{
uae_u16 tw;
size_t 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 = uaetcslen(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) {
int type = -1;
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);
type = PATH_FLOPPY;
} else if (currprefs.cdslots[0].inuse) {
_tcscpy(name, currprefs.cdslots[0].name);
type = PATH_CD;
}
if (!name[0])
underline[0] = 0;
if (type >= 0)
cfgfile_resolve_path_load(name, sizeof(name) / sizeof(TCHAR), type);
namesplit (name);
_tcscpy (extension, _T("wav"));
_stprintf (filename, _T("%s%s%s_%08x_%06x_%03d.%s"), path, name, underline, rs->pt, rs->len, 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_ftell32(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->pt = adp->pt;
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->pt = adp->pt;
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_PAULA];
static struct audio_stream_data audio_stream[AUDIO_CHANNEL_STREAMS];
static struct audio_channel_data2 *audio_data[AUDIO_CHANNELS_PAULA + AUDIO_CHANNEL_STREAMS * AUDIO_CHANNEL_MAX_STREAM_CH];
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 evt_t last_cycles;
static float next_sample_evtime;
static int previous_volcnt_update;
typedef uae_s8 sample8_t;
#define DO_CHANNEL_1(v, c) do { (v) *= audio_channel[c].data.mixvol; } 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) {
int shift = 16 - bits;
data <<= shift;
} else {
int shift = bits - 16;
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;
#define PAULARATE 3740000
static float Sinc(float x)
{
return x ? sinf(x) / x : 1;
}
static float Hamming(float x)
{
float pi = 4 * atanf(1);
float v;
if (x > -1 && x < 1)
v = cosf(x * pi / 2);
else
v = 0;
return v * v;
}
static void makefir(void)
{
float pi = 4 * atanf(1);
float *FIRTable = firmem + FIR_WIDTH;
float yscale = float(currprefs.sound_freq) / float(PAULARATE);
float xscale = pi * yscale;
for (int i = -FIR_WIDTH; i <= FIR_WIDTH; i++)
FIRTable[i] = yscale * Sinc(float(i) * xscale) * Hamming(float(i) / float(FIR_WIDTH - 1));
}
/* 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,
FILTER_MODEL_A500_FIXEDONLY
};
/* 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 = (float)(a500e_filter1_a0 * input + (1.0f - a500e_filter1_a0) * fs->rc1 + DENORMAL_OFFSET);
fs->rc2 = a500e_filter2_a0 * fs->rc1 + (1.0f - 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_A500_FIXEDONLY:
fs->rc1 = (float)(a500e_filter1_a0 * input + (1.0f - a500e_filter1_a0) * fs->rc1 + DENORMAL_OFFSET);
fs->rc2 = a500e_filter2_a0 * fs->rc1 + (1.0f - a500e_filter2_a0) * fs->rc2;
normal_output = fs->rc2;
led_output = fs->rc2;
break;
case FILTER_MODEL_A1200:
normal_output = (float)input;
fs->rc2 = (float)(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 = (int)led_output;
else
o = (int)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(tmp);
PUT_SOUND_WORD(w);
} 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(tmp);
PUT_SOUND_WORD(w);
} else {
PUT_SOUND_WORD(w);
}
}
static void anti_prehandler (unsigned long best_evtime)
{
int i, output;
struct audio_channel_data2 *acd;
/* Handle accumulator antialiasiation */
for (i = 0; audio_data[i]; i++) {
acd = audio_data[i];
output = (acd->current_sample * acd->mixvol) & 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++) {
struct audio_channel_data2 *acd = audio_data[i];
datasp[j] = acd->sample_accum_time ? (acd->sample_accum / acd->sample_accum_time) : 0;
acd->sample_accum = 0;
acd->sample_accum_time = 0;
}
}
static void sinc_prehandler_paula (unsigned long best_evtime)
{
int i, output;
struct audio_channel_data2 *acd;
for (i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
acd = audio_data[i];
int vol = acd->mixvol;
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 || sound_use_filter_sinc == FILTER_MODEL_A500_FIXEDONLY) ? 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_data2 *acd = audio_data[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 ? *data2 : 0) + 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;
if (data2)
*data2 = d1;
} else {
*data1 = d1;
if (data2)
*data2 = d2;
}
}
static void do_extra_channels(int idx, int ch, int *data1, int *data2, int *data3, int *data4, int *data5, int *data6)
{
idx += AUDIO_CHANNELS_PAULA;
if (ch == 2) {
int datas[2];
samplexx_anti_handler(datas, idx, 2);
get_extra_channels(data1, data2, datas[0], datas[1]);
} else if (ch == 1) {
int datas[1];
samplexx_anti_handler(datas, idx, 1);
int d1 = *data1 + datas[0];
if (d1 < -32768)
d1 = -32768;
if (d1 > 32767)
d1 = 32767;
*data1 = d1;
if (data2)
*data2 = d1;
} else if (ch > 2) {
int datas[AUDIO_CHANNEL_MAX_STREAM_CH];
samplexx_anti_handler(datas, idx, 6);
get_extra_channels(data1, data2, datas[0], datas[1]);
if (data3 && data4)
get_extra_channels(data3, data4, datas[2], datas[3]);
if (data5 && data6)
get_extra_channels(data5, data6, datas[4], datas[5]);
}
}
static void get_extra_channels_sample2(int *data1, int *data2, int mode)
{
if (!audio_total_extra_streams)
return;
int idx = 0;
for (int i = 0; i < AUDIO_CHANNEL_STREAMS; i++) {
int ch = audio_extra_streams[i];
if (ch) {
do_extra_channels(idx, ch, data1, data2, NULL, NULL, NULL, NULL);
idx += ch;
}
}
}
static void get_extra_channels_sample6(int *data1, int *data2, int *data3, int *data4, int *data5, int *data6, int mode)
{
if (!audio_total_extra_streams)
return;
int idx = 0;
for (int i = 0; i < AUDIO_CHANNEL_STREAMS; i++) {
int ch = audio_extra_streams[i];
if (ch) {
do_extra_channels(idx, ch, data1, data2, data3, data4, data5, data6);
idx += ch;
}
}
}
static void set_sound_buffers(void)
{
#if SOUNDSTUFF > 1
paula_sndbufpt_prev = paula_sndbufpt_start;
paula_sndbufpt_start = paula_sndbufpt;
#endif
}
static void clear_sound_buffers(void)
{
memset(paula_sndbuffer, 0, paula_sndbufsize);
paula_sndbufpt = paula_sndbuffer;
}
static void check_sound_buffers(void)
{
#if SOUNDSTUFF > 1
int len;
#endif
if (active_sound_stereo == SND_4CH_CLONEDSTEREO) {
((uae_u16 *)paula_sndbufpt)[0] = ((uae_u16 *)paula_sndbufpt)[-2];
((uae_u16 *)paula_sndbufpt)[1] = ((uae_u16 *)paula_sndbufpt)[-1];
paula_sndbufpt = (uae_u16 *)(((uae_u8 *)paula_sndbufpt) + 2 * 2);
} else if (active_sound_stereo == SND_6CH_CLONEDSTEREO) {
uae_s16 *p = ((uae_s16 *)paula_sndbufpt);
uae_s32 sum;
p[2] = p[-2];
p[3] = p[-1];
sum = (uae_s32)(p[-2]) + (uae_s32)(p[-1]) + (uae_s32)(p[2]) + (uae_s32)(p[3]);
p[0] = sum / 8;
p[1] = sum / 8;
paula_sndbufpt = (uae_u16 *)(((uae_u8 *)paula_sndbufpt) + 4 * 2);
} else if (active_sound_stereo == SND_8CH_CLONEDSTEREO) {
uae_s16 *p = ((uae_s16 *)paula_sndbufpt);
uae_s32 sum;
p[2] = p[-2];
p[3] = p[-1];
p[4] = p[-2];
p[5] = p[-1];
sum = (uae_s32)(p[-2]) + (uae_s32)(p[-1]) + (uae_s32)(p[2]) + (uae_s32)(p[3]);
p[0] = sum / 8;
p[1] = sum / 8;
paula_sndbufpt = (uae_u16 *)(((uae_u8 *)paula_sndbufpt) + 6 * 2);
}
#if SOUNDSTUFF > 1
if (outputsample == 0)
return;
len = paula_sndbufpt - paula_sndbufpt_start;
if (outputsample < 0) {
int i;
uae_s16 *p1 = (uae_s16 *)paula_sndbufpt_prev;
uae_s16 *p2 = (uae_s16 *)paula_sndbufpt_start;
for (i = 0; i < len; i++) {
*p1 = (*p1 + *p2) / 2;
}
paula_sndbufpt = paula_sndbufpt_start;
}
#endif
if ((uae_u8 *)paula_sndbufpt - (uae_u8 *)paula_sndbuffer >= paula_sndbufsize) {
finish_sound_buffer();
}
#if SOUNDSTUFF > 1
while (doublesample-- > 0) {
memcpy(paula_sndbufpt, paula_sndbufpt_start, len * 2);
paula_sndbufpt += len;
if ((uae_u8 *)paula_sndbufpt - (uae_u8 *)paula_sndbuffer >= paula_sndbufsize) {
finish_sound_buffer();
paula_sndbufpt = paula_sndbuffer;
}
}
#endif
}
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_sample2(&data1, NULL, 2);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data1);
check_sound_buffers ();
}
void sample16_handler (void)
{
int data0 = audio_channel[0].data.current_sample;
int data1 = audio_channel[1].data.current_sample;
int data2 = audio_channel[2].data.current_sample;
int data3 = audio_channel[3].data.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].data.adk_mask;
data1 &= audio_channel[1].data.adk_mask;
data2 &= audio_channel[2].data.adk_mask;
data3 &= audio_channel[3].data.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_sample2(&data, NULL, 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_sample2(&data1, NULL, 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].data.current_sample;
int data1 = audio_channel[1].data.current_sample;
int data2 = audio_channel[2].data.current_sample;
int data3 = audio_channel[3].data.current_sample;
int data0p = audio_channel[0].data.last_sample;
int data1p = audio_channel[1].data.last_sample;
int data2p = audio_channel[2].data.last_sample;
int data3p = audio_channel[3].data.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].data.adk_mask;
data0p &= audio_channel[0].data.adk_mask;
data1 &= audio_channel[1].data.adk_mask;
data1p &= audio_channel[1].data.adk_mask;
data2 &= audio_channel[2].data.adk_mask;
data2p &= audio_channel[2].data.adk_mask;
data3 &= audio_channel[3].data.adk_mask;
data3p &= audio_channel[3].data.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_sample2(&data, NULL, 0);
set_sound_buffers ();
PUT_SOUND_WORD_MONO (data);
check_sound_buffers ();
}
static void sample16i_crux_handler (void)
{
int data0 = audio_channel[0].data.current_sample;
int data1 = audio_channel[1].data.current_sample;
int data2 = audio_channel[2].data.current_sample;
int data3 = audio_channel[3].data.current_sample;
int data0p = audio_channel[0].data.last_sample;
int data1p = audio_channel[1].data.last_sample;
int data2p = audio_channel[2].data.last_sample;
int data3p = audio_channel[3].data.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].data.adk_mask;
data0p &= audio_channel[0].data.adk_mask;
data1 &= audio_channel[1].data.adk_mask;
data1p &= audio_channel[1].data.adk_mask;
data2 &= audio_channel[2].data.adk_mask;
data2p &= audio_channel[2].data.adk_mask;
data3 &= audio_channel[3].data.adk_mask;
data3p &= audio_channel[3].data.adk_mask;
{
struct audio_channel_data *cdp;
unsigned long ratio, ratio1;
#define INTERVAL ((int)(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;