redquark-amiberry-rb/src/audio.cpp

 /*
  * 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 "audio.h"
#include "sounddep/sound.h"
#include "savestate.h"
#include "gui.h"
#include "xwin.h"
#include "threaddep/thread.h"

#include <math.h>

#define PERIOD_MIN 4
#define PERIOD_MIN_NONCE 60

int audio_channel_mask = 15;
volatile bool cd_audio_mode_changed;

STATIC_INLINE bool isaudio (void)
{
	return currprefs.produce_sound != 0;
}

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.in"

typedef struct {
	int time, output;
} sinc_queue_t;

struct audio_channel_data2
{
	int current_sample, last_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_channel_data
{
	unsigned int evtime;
	bool dmaenstore;
	bool intreq2;
	bool dr;
	bool dsr;
	bool pbufldl;
	bool dat_written;
	uaecptr lc, pt;
	int state;
	int per;
	int len, wlen;
	uae_u16 dat, dat2;
	struct audio_channel_data2 data;
	/* too fast cpu fixes */
	uaecptr ptx;
	bool ptx_written;
	bool ptx_tofetch;
	int dmaofftime_active;
};

static struct audio_channel_data audio_channel[AUDIO_CHANNELS_PAULA];
static struct audio_channel_data2 *audio_data[AUDIO_CHANNELS_PAULA];
int sound_available = 0;
void (*sample_handler) (void);
static void (*sample_prehandler) (unsigned long best_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].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;

/* 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 = 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_mono_func)(uae_u32 w);
static void (*put_sound_word_stereo_func)(uae_u32 left, uae_u32 right);

static void put_sound_word_stereo_func_filter_mixed(uae_u32 lnew, uae_u32 rnew)
{
	uae_u32 rold, lold, tmp;

	lnew = filter(lnew, &sound_filter_state[0]);
	rnew = filter(rnew, &sound_filter_state[1]);

	left_word_saved[saved_ptr] = lnew;
	right_word_saved[saved_ptr] = rnew;

	saved_ptr = (saved_ptr + 1) & mixed_stereo_size;

	lold = left_word_saved[saved_ptr];
	tmp = (rnew * mixed_mul2 + lold * mixed_mul1) / MIXED_STEREO_SCALE;

	rold = right_word_saved[saved_ptr];
	lnew = (lnew * mixed_mul2 + rold * mixed_mul1) / MIXED_STEREO_SCALE;

	PUT_SOUND_WORD_STEREO(lnew, tmp);
}

static void put_sound_word_stereo_func_filter_notmixed(uae_u32 left, uae_u32 right)
{
	left = filter(left, &sound_filter_state[0]);
	right = filter(right, &sound_filter_state[1]);
	PUT_SOUND_WORD_STEREO(left, right);
}

static void put_sound_word_stereo_func_nofilter_mixed(uae_u32 lnew, uae_u32 rnew)
{
	uae_u32 rold, lold, tmp;

	left_word_saved[saved_ptr] = lnew;
	right_word_saved[saved_ptr] = rnew;

	saved_ptr = (saved_ptr + 1) & mixed_stereo_size;

	lold = left_word_saved[saved_ptr];
	tmp = (rnew * mixed_mul2 + lold * mixed_mul1) / MIXED_STEREO_SCALE;

	rold = right_word_saved[saved_ptr];
	lnew = (lnew * mixed_mul2 + rold * mixed_mul1) / MIXED_STEREO_SCALE;

	PUT_SOUND_WORD_STEREO(lnew, tmp);
}

static void put_sound_word_stereo_func_nofilter_notmixed(uae_u32 left, uae_u32 right)
{
	PUT_SOUND_WORD_STEREO(left, right);
}

static void put_sound_word_mono_func_filter(uae_u32 data)
{
	data = filter(data, &sound_filter_state[0]);
	PUT_SOUND_WORD(data);
}

static void put_sound_word_mono_func_nofilter(uae_u32 data)
{
	PUT_SOUND_WORD(data);
}

static void anti_prehandler(unsigned long best_evtime)
{
	int i, output;
	struct audio_channel_data2 *acd;

	/* Handle accumulator antialiasiation */
	for (i = 0; i < AUDIO_CHANNELS_PAULA; 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)
{
	for (auto i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
		auto acd = audio_data[i];
		datasp[i] = 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 n;

	if (sound_use_filter_sinc) {
		n = (sound_use_filter_sinc == FILTER_MODEL_A500) ? 0 : 2;
		if (led_filter_on)
			n += 1;
	} else {
		n = 4;
	}
	auto winsinc = winsinc_integral[n];

	for (auto i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
		auto acd = audio_data[i];
		/* The sum rings with harmonic components up to infinity... */
		auto sum = acd->sinc_output_state << 17;
		/* ...but we cancel them through mixing in BLEPs instead */
		auto offsetpos = acd->sinc_queue_head & (SINC_QUEUE_LENGTH - 1);
		for (int j = 0; j < SINC_QUEUE_LENGTH; j += 1) {
			auto 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);
		}
		auto v = sum >> 15;
		if (v > 32767)
			v = 32767;
		else if (v < -32768)
			v = -32768;
		datasp[i] = v;
	}
}

static void sample16i_sinc_handler(void)
{
	int datas[AUDIO_CHANNELS_PAULA], data1;

	samplexx_sinc_handler(datas);
	data1 = datas[0] + datas[3] + datas[1] + datas[2];
	data1 = FINISH_DATA(data1, 18, 0);

	put_sound_word_mono_func(data1);
	check_sound_buffers();
}

void sample16_handler(void)
{
	int data;
	if (audio_channel[0].data.adk_mask)
    data = audio_channel[0].data.current_sample * audio_channel[0].data.mixvol;
	else
		data = 0;
	if (audio_channel[1].data.adk_mask)
    data += audio_channel[1].data.current_sample * audio_channel[1].data.mixvol;
	if (audio_channel[2].data.adk_mask)
    data += audio_channel[2].data.current_sample * audio_channel[2].data.mixvol;
	if (audio_channel[3].data.adk_mask)
    data += audio_channel[3].data.current_sample * audio_channel[3].data.mixvol;

	data = FINISH_DATA(data, 16, 0);

	put_sound_word_mono_func(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];

	samplexx_anti_handler(datas);
	auto data1 = datas[0] + datas[3] + datas[1] + datas[2];
	data1 = FINISH_DATA(data1, 16, 0);

	put_sound_word_mono_func(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 = data0;
	data = FINISH_DATA(data, 16, 0);

	put_sound_word_mono_func(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;
		uae_u32 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 = data0;
	data = FINISH_DATA(data, 16, 0);

	put_sound_word_mono_func(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 sample16si_anti_handler(void)
{
	int datas[AUDIO_CHANNELS_PAULA];

	samplexx_anti_handler(datas);
	auto data1 = datas[0] + datas[3];
	auto data2 = datas[1] + datas[2];
	data1 = FINISH_DATA(data1, 15, 0);
	data2 = FINISH_DATA(data2, 15, 1);

	put_sound_word_stereo_func(data1, data2);
	check_sound_buffers();
}

static void sample16si_sinc_handler(void)
{
	int datas[AUDIO_CHANNELS_PAULA];

	samplexx_sinc_handler(datas);
	auto data1 = datas[0] + datas[3];
	auto data2 = datas[1] + datas[2];
	data1 = FINISH_DATA(data1, 17, 0);
	data2 = FINISH_DATA(data2, 17, 1);

	put_sound_word_stereo_func(data1, data2);
	check_sound_buffers();
}

void sample16s_handler(void)
{
  int data_l = audio_channel[0].data.adk_mask ? audio_channel[0].data.current_sample * audio_channel[0].data.mixvol : 0;
  int data_r = audio_channel[1].data.adk_mask ? audio_channel[1].data.current_sample * audio_channel[1].data.mixvol : 0;
	if (audio_channel[2].data.adk_mask)
    data_r += audio_channel[2].data.current_sample * audio_channel[2].data.mixvol;
	if (audio_channel[3].data.adk_mask)
    data_l += audio_channel[3].data.current_sample * audio_channel[3].data.mixvol;
	data_l = FINISH_DATA(data_l, 15, 0);
	data_r = FINISH_DATA(data_r, 15, 1);

	put_sound_word_stereo_func(data_l, data_r);
	check_sound_buffers();
}

static void sample16si_crux_handler(void)
{
	auto data0 = audio_channel[0].data.current_sample;
	auto data1 = audio_channel[1].data.current_sample;
	auto data2 = audio_channel[2].data.current_sample;
	auto data3 = audio_channel[3].data.current_sample;
	auto data0p = audio_channel[0].data.last_sample;
	auto data1p = audio_channel[1].data.last_sample;
	auto data2p = audio_channel[2].data.last_sample;
	auto data3p = audio_channel[3].data.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].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;
		uae_u32 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 = FINISH_DATA(data0, 15, 0);
	data1 = FINISH_DATA(data1, 15, 1);

	put_sound_word_stereo_func(data0, data1);
	check_sound_buffers();
}

static void sample16si_rh_handler (void)
{
	auto data0 = audio_channel[0].data.current_sample;
	auto data1 = audio_channel[1].data.current_sample;
	auto data2 = audio_channel[2].data.current_sample;
	auto data3 = audio_channel[3].data.current_sample;
	auto data0p = audio_channel[0].data.last_sample;
	auto data1p = audio_channel[1].data.last_sample;
	auto data2p = audio_channel[2].data.last_sample;
	auto data3p = audio_channel[3].data.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].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... */
	unsigned long delta = audio_channel[0].per;
	unsigned long 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;
	data0 = FINISH_DATA(data0, 15, 0);
	data1 = FINISH_DATA(data1, 15, 1);

	put_sound_word_stereo_func(data0, data1);
	check_sound_buffers();
}

static int audio_work_to_do;

static void zerostate (int nr)
{
	auto cdp = audio_channel + nr;
	cdp->state = 0;
	cdp->evtime = MAX_EV;
	cdp->intreq2 = false;
	cdp->dmaenstore = false;
	cdp->dmaofftime_active = 0;
}

static void schedule_audio (void)
{
	unsigned long best = MAX_EV;

	eventtab[ev_audio].active = 0;
	for (auto i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
		auto 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)
{
	last_cycles = get_cycles();
	next_sample_evtime = scaled_sample_evtime;
	if (!isrestore()) {
		for (auto i = 0; i < AUDIO_CHANNELS_PAULA; i++)
			zerostate(i);
	}
	schedule_audio();
	events_schedule();
}

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)
{
	auto 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;
}

static void update_volume(int nr, uae_u16 v)
{
	auto cdp = audio_channel + nr;
	// 7 bit register in Paula.
	v &= 127;
	if (v > 64)
		v = 64;
	cdp->data.audvol = v;
	cdp->data.mixvol = v;
}

uae_u16 audio_dmal(void)
{
	uae_u16 dmal = 0;
	for (auto nr = 0; nr < AUDIO_CHANNELS_PAULA; nr++) {
		auto 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)
{
	INTREQ_0(0x8000 | (0x80 << nr));
}

static void newsample(int nr, sample8_t sample)
{
	auto cdp = audio_channel + nr;
	cdp->data.last_sample = cdp->data.current_sample;
	cdp->data.current_sample = sample;
}

STATIC_INLINE void setdr(int nr)
{
	auto cdp = audio_channel + nr;
	cdp->dr = true;
	if (cdp->wlen == 1) {
		cdp->dsr = true;
	}
}

static void loaddat(int nr, bool modper)
{
	auto cdp = audio_channel + nr;
	auto audav = adkcon & (0x01 << nr);
	auto 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) {
			update_volume(nr + 1, cdp->dat);
		}
	}
	else {
		cdp->dat2 = cdp->dat;
	}
}
static void loaddat(int nr)
{
	loaddat(nr, false);
}

static void loadper (int nr)
{
	auto 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)
{
	auto cdp = audio_channel + nr;
	auto chan_ena = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));
	auto old_dma = cdp->dmaenstore;
	auto audav = adkcon & (0x01 << nr);
	auto audap = adkcon & (0x10 << nr);
	int napnav = (!audav && !audap) || audav;
	auto 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.
			newsample(nr, (cdp->dat2 >> 0) & 0xff);
			zerostate(nr);
		}
	}

	switch (cdp->state)
	{
	case 0:
		if (chan_ena) {
			cdp->evtime = MAX_EV;
			cdp->state = 1;
			cdp->dr = true;
			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;
		}
		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;
		setirq(nr, 10);
		setdr(nr);
		if (cdp->wlen != 1)
			cdp->wlen = (cdp->wlen - 1) & 0xffff;
		cdp->state = 5;
		break;
	case 5:
		cdp->evtime = MAX_EV;
		if (!chan_ena) {
			zerostate(nr);
			return;
		}
		if (!cdp->dat_written)
			return;
		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) {
			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) {
			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)) {
				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_channel(int nr, bool perfin)
{
	auto cdp = audio_channel + nr;
	audio_state_channel2(nr, perfin);
	cdp->dat_written = false;
}

void audio_state_machine(void)
{
	update_audio();
	for (auto nr = 0; nr < AUDIO_CHANNELS_PAULA; nr++) {
		auto cdp = audio_channel + nr;
		audio_state_channel2(nr, false);
		cdp->dat_written = false;
	}
	schedule_audio();
	events_schedule();
}

void audio_reset(void)
{
	reset_sound();
	memset(sound_filter_state, 0, sizeof sound_filter_state);
	if (!isrestore()) {
		for (auto& i : audio_channel)
		{
			auto cdp = &i;
			memset(cdp, 0, sizeof *audio_channel);
			cdp->per = int(PERIOD_MAX - 1);
			cdp->data.mixvol = 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.sound_stereo != currprefs.sound_stereo
		|| changed_prefs.sound_freq != currprefs.sound_freq)
		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_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)
{
	/* The BLT correction formula below blows up if the cutoff is above nyquist. */
	if (cutoff_freq >= sample_rate / 2)
		return 1.0;

	float 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 / 2.0) * 2.0;
	const float out = 1.0 / (1.0 + 1.0 / omega);
	return out;
}

void check_prefs_changed_audio (void)
{
	if (sound_available) {
		auto ch = sound_prefs_changed();
		if (ch > 0) {
			clear_sound_buffers();
		}
		if (ch) {
			set_audio();
			audio_activate();
		}
	}
}

void set_audio (void)
{
	const auto old_mixed_size = mixed_stereo_size;

	const auto ch = sound_prefs_changed();
	if (ch >= 0)
		close_sound();

	currprefs.produce_sound = changed_prefs.produce_sound;
	currprefs.sound_stereo = changed_prefs.sound_stereo;
	currprefs.sound_freq = changed_prefs.sound_freq;

	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_cd = changed_prefs.sound_volume_cd;

	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;

	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);
	}

	auto sep = (currprefs.sound_stereo_separation = changed_prefs.sound_stereo_separation) * 3 / 2;
	if (sep >= 15)
		sep = 16;
	const auto 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);
	memset(sound_filter_state, 0, sizeof sound_filter_state);
	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);
	}
	sample_prehandler = NULL;
	if (sample_handler == sample16si_sinc_handler || sample_handler == sample16i_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_prehandler = anti_prehandler;
	}
	for (int i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
		struct audio_channel_data *cdp = audio_channel + i;
		audio_data[i] = &cdp->data;
		cdp->data.mixvol = cdp->data.audvol;
	}

	if (currprefs.sound_stereo) {
		if (currprefs.sound_filter) {
			if (mixed_on)
				put_sound_word_stereo_func = put_sound_word_stereo_func_filter_mixed;
			else
				put_sound_word_stereo_func = put_sound_word_stereo_func_filter_notmixed;
		}
		else {
			if (mixed_on)
				put_sound_word_stereo_func = put_sound_word_stereo_func_nofilter_mixed;
			else
				put_sound_word_stereo_func = put_sound_word_stereo_func_nofilter_notmixed;
		}
	}
	else {
		if (currprefs.sound_filter) {
			put_sound_word_mono_func = put_sound_word_mono_func_filter;
		}
		else {
			put_sound_word_mono_func = put_sound_word_mono_func_nofilter;
		}
	}

	if (currprefs.produce_sound == 0) {
		eventtab[ev_audio].active = false;
		events_schedule();
	}
	else {
		audio_activate();
		schedule_audio();
		events_schedule();
	}
	set_config_changed();
	cd_audio_mode_changed = true;
}

void update_audio(void)
{
	unsigned long int n_cycles = 0;

	if (!isaudio())
		goto end;
	if (isrestore())
		goto end;
	if (!is_audio_active())
		goto end;

	n_cycles = get_cycles() - last_cycles;
	while (n_cycles > 0) {
		auto best_evtime = n_cycles + 1;
		uae_u32 rounded;
		int i;

		for (i = 0; i < AUDIO_CHANNELS_PAULA; 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 (sample_prehandler && (currprefs.produce_sound > 1)) {
			sample_prehandler(best_evtime / CYCLE_UNIT);
		}

		for (i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
			if (audio_channel[i].evtime != MAX_EV)
				audio_channel[i].evtime -= best_evtime;
		}

		n_cycles -= best_evtime;

		/* Test if new sample needs to be outputted */
		if ((rounded == best_evtime) && (currprefs.produce_sound > 1)) {
			/* Before the following addition, next_sample_evtime is in range [-0.5, 0.5) */
			next_sample_evtime += scaled_sample_evtime;
			(*sample_handler) ();
		}

		for (i = 0; i < AUDIO_CHANNELS_PAULA; i++) {
			if (audio_channel[i].evtime == 0) {
				audio_state_channel(i, true);
			}
		}
	}
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) {
		audio_work_to_do--;
		if (audio_work_to_do == 0)
			audio_deactivate();
	}
	update_audio();
}

void AUDxDAT(int nr, uae_u16 v)
{
	const auto cdp = audio_channel + nr;
	const int chan_ena = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));

	cdp->dat = v;
	cdp->dat_written = true;
	if (cdp->state == 2 || cdp->state == 3) {
		if (chan_ena) {
			if (cdp->wlen == 1) {
				cdp->wlen = cdp->len;
				cdp->intreq2 = true;
			}
			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;
}

uaecptr audio_getpt(int nr, bool reset)
{
	auto cdp = audio_channel + nr;
	const auto 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)
{
	const auto 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;
	}
	else {
		cdp->lc = (cdp->lc & 0xffff) | ((uae_u32)v << 16);
	}
}

void AUDxLCL(int nr, uae_u16 v)
{
	const auto 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;
	}
	else {
		cdp->lc = (cdp->lc & ~0xffff) | (v & 0xFFFE);
	}
}

void AUDxPER(int nr, uae_u16 v)
{
	const auto cdp = audio_channel + nr;

	audio_activate();
	update_audio();

	unsigned long 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 && 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();
		}
	}
	cdp->per = per;
}

void AUDxLEN(int nr, uae_u16 v)
{
	auto cdp = audio_channel + nr;
	audio_activate();
	update_audio();
	cdp->len = v;
}

void AUDxVOL(int nr, uae_u16 v)
{
	auto cdp = audio_channel + nr;

	audio_activate();
	update_audio();
	update_volume(nr, v);
}

void audio_update_adkmasks(void)
{
	static auto prevcon = -1;
	const unsigned long t = adkcon | (adkcon >> 4);

	audio_channel[0].data.adk_mask = (((t >> 0) & 1) - 1);
	audio_channel[1].data.adk_mask = (((t >> 1) & 1) - 1);
	audio_channel[2].data.adk_mask = (((t >> 2) & 1) - 1);
	audio_channel[3].data.adk_mask = (((t >> 3) & 1) - 1);
	if ((prevcon & 0xff) != (adkcon & 0xff)) {
		audio_activate();
		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 restore_audio_finish(void)
{
	last_cycles = get_cycles();
	schedule_audio();
	events_schedule();
}

uae_u8 *restore_audio(int nr, uae_u8 *src)
{
	const auto acd = audio_channel + nr;

	zerostate(nr);
	acd->state = restore_u8();
	acd->data.audvol = restore_u8();
	acd->intreq2 = restore_u8() != 0;
	const auto 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();
	const auto 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();
	acd->dmaenstore = (dmacon & DMA_MASTER) && (dmacon & (1 << nr));
  acd->data.mixvol = acd->data.audvol;
	return src;
}

uae_u8 *save_audio(int nr, int *len, uae_u8 *dstptr)
{
	const auto 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->data.audvol);
	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);
	*len = dst - dstbak;
	return dstbak;
}