// Commander X16 Emulator

// Copyright (c) 2020 Frank van den Hoef

// All rights reserved. License: 2-clause BSD

#include "audio.h"

#include "glue.h"

#include "vera_psg.h"

#include "vera_pcm.h"

#include "wav_recorder.h"

#include "ymglue.h"

#include "midi.h"

#include <stdint.h>

#include <stdio.h>

#include <string.h>

#include <stdlib.h>

/* RRDC (control_backend.c): tee each mixed stereo pair into the /audio ring. */

void x16_control_push_audio(int16_t l, int16_t r);

#ifdef __EMSCRIPTEN__

	#define SAMPLES_PER_BUFFER (1024)

	#define SAMP_POS_FRAC_BITS (22)

#else

	#define SAMPLES_PER_BUFFER (256)

	#define SAMP_POS_FRAC_BITS (24)

#endif

#define VERA_SAMP_CLKS_PER_CPU_CLK ((25000000ULL << SAMP_POS_FRAC_BITS) / 512 / MHZ / 1000000)

#define YM_SAMP_CLKS_PER_CPU_CLK ((3579545ULL << SAMP_POS_FRAC_BITS) / 64 / MHZ / 1000000)

#define FS_SAMP_CLKS_PER_CPU_CLK VERA_SAMP_CLKS_PER_CPU_CLK

#define SAMPLE_BYTES (2 * sizeof(int16_t))

#define SAMP_POS_MASK (SAMPLES_PER_BUFFER - 1)

#define SAMP_POS_MASK_FRAC (((uint32_t)SAMPLES_PER_BUFFER << SAMP_POS_FRAC_BITS) - 1)

// windowed sinc

static const int16_t filter[512] = {

	32767,32765,32761,32755,32746,32736,32723,32707,32690,32670,32649,32625,32598,32570,32539,32507,

	32472,32435,32395,32354,32310,32265,32217,32167,32115,32061,32004,31946,31885,31823,31758,31691,

	31623,31552,31479,31404,31327,31248,31168,31085,31000,30913,30825,30734,30642,30547,30451,30353,

	30253,30151,30048,29943,29835,29726,29616,29503,29389,29273,29156,29037,28916,28793,28669,28544,

	28416,28288,28157,28025,27892,27757,27621,27483,27344,27204,27062,26918,26774,26628,26481,26332,

	26182,26031,25879,25726,25571,25416,25259,25101,24942,24782,24621,24459,24296,24132,23967,23801,

	23634,23466,23298,23129,22959,22788,22616,22444,22271,22097,21923,21748,21572,21396,21219,21042,

	20864,20686,20507,20328,20148,19968,19788,19607,19426,19245,19063,18881,18699,18517,18334,18152,

	17969,17786,17603,17420,17237,17054,16871,16688,16505,16322,16139,15957,15774,15592,15409,15227,

	15046,14864,14683,14502,14321,14141,13961,13781,13602,13423,13245,13067,12890,12713,12536,12360,

	12185,12010,11836,11663,11490,11317,11146,10975,10804,10635,10466,10298,10131, 9964, 9799, 9634,

	 9470, 9306, 9144, 8983, 8822, 8662, 8504, 8346, 8189, 8033, 7878, 7724, 7571, 7419, 7268, 7118,

	 6969, 6822, 6675, 6529, 6385, 6241, 6099, 5958, 5818, 5679, 5541, 5405, 5269, 5135, 5002, 4870,

	 4739, 4610, 4482, 4355, 4229, 4104, 3981, 3859, 3738, 3619, 3500, 3383, 3268, 3153, 3040, 2928,

	 2817, 2708, 2600, 2493, 2388, 2284, 2181, 2079, 1979, 1880, 1783, 1686, 1591, 1498, 1405, 1314,

	 1225, 1136, 1049,  963,  879,  795,  714,  633,  554,  476,  399,  323,  249,  176,  105,   34,

	  -34, -102, -168, -234, -298, -361, -422, -482, -542, -599, -656, -712, -766, -819, -871, -922,

	 -971,-1020,-1067,-1113,-1158,-1202,-1244,-1286,-1326,-1366,-1404,-1441,-1477,-1512,-1546,-1579,

	-1611,-1642,-1671,-1700,-1728,-1755,-1781,-1806,-1830,-1852,-1874,-1896,-1916,-1935,-1953,-1971,

	-1987,-2003,-2018,-2032,-2045,-2058,-2069,-2080,-2090,-2099,-2108,-2116,-2123,-2129,-2134,-2139,

	-2143,-2147,-2150,-2152,-2153,-2154,-2154,-2154,-2153,-2151,-2149,-2146,-2143,-2139,-2135,-2130,

	-2124,-2118,-2112,-2105,-2098,-2090,-2082,-2073,-2064,-2054,-2045,-2034,-2024,-2012,-2001,-1989,

	-1977,-1965,-1952,-1939,-1926,-1912,-1898,-1884,-1870,-1855,-1840,-1825,-1810,-1794,-1778,-1762,

	-1746,-1730,-1714,-1697,-1680,-1663,-1646,-1629,-1612,-1595,-1577,-1560,-1542,-1525,-1507,-1489, 

	-1471,-1453,-1435,-1418,-1400,-1382,-1364,-1346,-1328,-1310,-1292,-1274,-1256,-1238,-1220,-1203,

	-1185,-1167,-1150,-1132,-1115,-1097,-1080,-1063,-1046,-1029,-1012, -995, -978, -962, -945, -929,

     -912, -896, -880, -864, -849, -833, -817, -802, -787, -772, -757, -742, -727, -713, -699, -684,

	 -670, -656, -643, -629, -616, -603, -589, -577, -564, -551, -539, -526, -514, -502, -491, -479,

	 -468, -456, -445, -434, -423, -413, -402, -392, -381, -371, -361, -352, -342, -333, -323, -314,

	 -305, -296, -288, -279, -270, -262, -254, -246, -238, -230, -222, -215, -207, -200, -193, -186,

	 -179, -172, -165, -158, -152, -145, -139, -133, -127, -120, -114, -108, -103,  -97,  -91,  -85,

	  -80,  -74,  -69,  -63,  -58,  -53,  -47,  -42,  -37,  -32,  -27,  -22,  -17,  -12,   -7,   -2

};

static SDL_AudioDeviceID audio_dev;

static int16_t * buffer;

static uint32_t buffer_size = 0;

static uint32_t rdidx = 0;

static uint32_t wridx = 0;

static uint32_t buffer_written = 0;

static uint32_t vera_samp_pos_rd = 0;

static uint32_t vera_samp_pos_wr = 0;

static uint32_t vera_samp_pos_hd = 0;

static uint32_t ym_samp_pos_rd = 0;

static uint32_t ym_samp_pos_wr = 0;

static uint32_t ym_samp_pos_hd = 0;

static uint32_t fs_samp_pos_rd = 0;

static uint32_t fs_samp_pos_wr = 0;

static uint32_t fs_samp_pos_hd = 0;

static uint32_t vera_samps_per_host_samps = 0;

static uint32_t ym_samps_per_host_samps = 0;

static uint32_t fs_samps_per_host_samps = 0;

// The *_SAMP_CLKS_PER_CPU_CLK ratios, worked out once in audio_init():

// MHZ is a variable, so the macros divide at run time, and audio_step()

// runs after every instruction.

static uint64_t vera_samp_clks_per_cpu_clk = 0;

static uint64_t ym_samp_clks_per_cpu_clk = 0;

static uint64_t fs_samp_clks_per_cpu_clk = 0;

static uint32_t limiter_amp = 0;

static int16_t psg_buf[2 * SAMPLES_PER_BUFFER];

static int16_t pcm_buf[2 * SAMPLES_PER_BUFFER];

static int16_t ym_buf[2 * SAMPLES_PER_BUFFER];

static int16_t fs_buf[2 * SAMPLES_PER_BUFFER];

uint32_t host_sample_rate = 0;

static void

audio_callback(void *userdata, Uint8 *stream, int len)

{

	if (audio_dev == 0) {

		return;

	}

	int expected = SAMPLES_PER_BUFFER * SAMPLE_BYTES;

	if (len != expected) {

		printf("Audio buffer size mismatch! (expected: %d, got: %d)\n", expected, len);

		return;

	}

	uint32_t spos = 0;

	if (rdidx > wridx) {

		uint32_t actual_len = SDL_min(len / SAMPLE_BYTES, (buffer_size - rdidx) / 2);

		if (actual_len > 0) {

			memcpy(&stream[spos], &buffer[rdidx], actual_len * SAMPLE_BYTES);

			spos += actual_len * SAMPLE_BYTES;

			len -= actual_len * SAMPLE_BYTES;

			rdidx = (rdidx + actual_len * 2) % buffer_size;

			buffer_written -= actual_len * 2;

		}

	}

	uint32_t actual_len = SDL_min(len / SAMPLE_BYTES, (wridx - rdidx) / 2);

	if (actual_len > 0) {

		memcpy(&stream[spos], &buffer[rdidx], actual_len * SAMPLE_BYTES);

		spos += actual_len * SAMPLE_BYTES;

		len -= actual_len * SAMPLE_BYTES;

		rdidx = (rdidx + actual_len * 2) % buffer_size;

		buffer_written -= actual_len * 2;

	}

	if (len > 0) memset(&stream[spos], 0, len);

}

void

audio_init(const char *dev_name, int num_audio_buffers)

{

	if (audio_dev > 0) {

		audio_close();

	}

	if (dev_name) {

		if (!strcmp("none", dev_name)) {

			return;

		}

	}

	// Set number of buffers

	int num_bufs = num_audio_buffers;

	if (num_bufs < 3) {

		num_bufs = 3;

	}

	if (num_bufs > 1024) {

		num_bufs = 1024;

	}

	buffer_size = SAMPLES_PER_BUFFER * num_bufs * 2;

	// Allocate audio buffer

	buffer = malloc(buffer_size * sizeof(int16_t));

	rdidx = 0;

	wridx = 0;

	buffer_written = 0;

	SDL_AudioSpec desired;

	SDL_AudioSpec obtained;

	// Setup SDL audio

	memset(&desired, 0, sizeof(desired));

	desired.freq     = AUDIO_SAMPLERATE;

	desired.format   = AUDIO_S16SYS;

	desired.samples  = SAMPLES_PER_BUFFER;

	desired.channels = 2;

	desired.callback = audio_callback;

	audio_dev = SDL_OpenAudioDevice(dev_name, 0, &desired, &obtained, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE);

	if (audio_dev <= 0) {

		fprintf(stderr, "SDL_OpenAudioDevice failed: %s\n", SDL_GetError());

		if (dev_name != NULL) {

			audio_usage();

		}

		exit(-1);

	}

	if (obtained.freq <= 0 || (AUDIO_SAMPLERATE / obtained.freq) > SAMPLES_PER_BUFFER) {

		fprintf(stderr, "Obtained sample rate is too low");

		audio_close();

		return;

	}

	// Init YM2151 emulation. 3.579545 MHz clock

	YM_Create(3579545);

	YM_init(3579545/64, 60);

	host_sample_rate = obtained.freq;

	vera_samps_per_host_samps = ((25000000ULL << SAMP_POS_FRAC_BITS) / 512 / host_sample_rate);

	ym_samps_per_host_samps = ((3579545ULL << SAMP_POS_FRAC_BITS) / 64 / host_sample_rate);

	fs_samps_per_host_samps = vera_samps_per_host_samps;

	vera_samp_clks_per_cpu_clk = VERA_SAMP_CLKS_PER_CPU_CLK;

	ym_samp_clks_per_cpu_clk = YM_SAMP_CLKS_PER_CPU_CLK;

	fs_samp_clks_per_cpu_clk = FS_SAMP_CLKS_PER_CPU_CLK;

	vera_samp_pos_rd = 0;

	vera_samp_pos_wr = 0;

	vera_samp_pos_hd = 0;

	ym_samp_pos_rd = 0;

	ym_samp_pos_wr = 0;

	ym_samp_pos_hd = 0;

	fs_samp_pos_rd = 0;

	fs_samp_pos_wr = 0;

	fs_samp_pos_hd = 0;

	limiter_amp = (1 << 16);

	psg_buf[0] = psg_buf[1] = 0;

	pcm_buf[0] = pcm_buf[1] = 0;

	ym_buf[0] = ym_buf[1] = 0;

	fs_buf[0] = fs_buf[1] = 0;

	// Start playback

	SDL_PauseAudioDevice(audio_dev, 0);

}

void

audio_close(void)

{

	if (audio_dev == 0) {

		return;

	}

	SDL_CloseAudioDevice(audio_dev);

	audio_dev = 0;

	// Free audio buffers

	if (buffer != NULL) {

		free(buffer);

		buffer = NULL;

	}

}

void

audio_step(int cpu_clocks)

{

	// Accumulate how many samples each source have to render

	if (audio_dev == 0) {

		return;

	}

	while (cpu_clocks > 0) {

		// Only the source with the higest sample rate (YM2151) is needed for calculation

		uint32_t ym_room = (ym_samp_pos_rd - ym_samp_pos_hd - (1 << SAMP_POS_FRAC_BITS)) & SAMP_POS_MASK_FRAC;

		// min(cpu_clocks, ym_room / ratio), dividing only when the buffer

		// is nearly full rather than after every instruction

		uint32_t max_cpu_clks = (uint64_t)cpu_clocks * ym_samp_clks_per_cpu_clk <= ym_room ? (uint32_t)cpu_clocks : (uint32_t)(ym_room / ym_samp_clks_per_cpu_clk);

		vera_samp_pos_hd = (vera_samp_pos_hd + max_cpu_clks * vera_samp_clks_per_cpu_clk) & SAMP_POS_MASK_FRAC;

		ym_samp_pos_hd = (ym_samp_pos_hd + max_cpu_clks * ym_samp_clks_per_cpu_clk) & SAMP_POS_MASK_FRAC;

		fs_samp_pos_hd = (fs_samp_pos_hd + max_cpu_clks * fs_samp_clks_per_cpu_clk) & SAMP_POS_MASK_FRAC;

		cpu_clocks -= max_cpu_clks;

		if (cpu_clocks > 0) audio_render();

	}

}

// How many CPU clocks audio_step() can take in one call and still do

// exactly what that many one-clock calls would: up to and including the

// clock on which it next has to render, which is when the CPU can see a

// change (the PCM FIFO, VERA's AFLOW interrupt, the YM2151's state).

uint32_t

audio_clocks_to_render(void)

{

	if (audio_dev == 0) {

		return UINT32_MAX;

	}

	uint32_t ym_room = (ym_samp_pos_rd - ym_samp_pos_hd - (1 << SAMP_POS_FRAC_BITS)) & SAMP_POS_MASK_FRAC;

	return (uint32_t)(ym_room / ym_samp_clks_per_cpu_clk) + 1;

}

void

audio_render()

{

	// Render all audio sources until read and write positions catch up

	// This happens when there's a change to sound registers or one of the

	// sources' sample buffer head position is too far

	if (audio_dev == 0) {

		return;

	}

	uint32_t pos, len;

	pos = (vera_samp_pos_wr + 1) & SAMP_POS_MASK;

	len = ((vera_samp_pos_hd >> SAMP_POS_FRAC_BITS) - vera_samp_pos_wr) & SAMP_POS_MASK;

	vera_samp_pos_wr = vera_samp_pos_hd >> SAMP_POS_FRAC_BITS;

	if (pos + len > SAMPLES_PER_BUFFER) {

		psg_render(&psg_buf[pos * 2], SAMPLES_PER_BUFFER - pos);

		pcm_render(&pcm_buf[pos * 2], SAMPLES_PER_BUFFER - pos);

		len -= SAMPLES_PER_BUFFER - pos;

		pos = 0;

	}

	if (len > 0) {

		psg_render(&psg_buf[pos * 2], len);

		pcm_render(&pcm_buf[pos * 2], len);

	}

	pos = (ym_samp_pos_wr + 1) & SAMP_POS_MASK;

	len = ((ym_samp_pos_hd >> SAMP_POS_FRAC_BITS) - ym_samp_pos_wr) & SAMP_POS_MASK;

	ym_samp_pos_wr = ym_samp_pos_hd >> SAMP_POS_FRAC_BITS;

	if ((pos + len) > SAMPLES_PER_BUFFER) {

		YM_stream_update((uint16_t *)&ym_buf[pos * 2], SAMPLES_PER_BUFFER - pos);

		len -= SAMPLES_PER_BUFFER - pos;

		pos = 0;

	}

	if (len > 0) {

		YM_stream_update((uint16_t *)&ym_buf[pos * 2], len);

	}

	pos = (fs_samp_pos_wr + 1) & SAMP_POS_MASK;

	len = ((fs_samp_pos_hd >> SAMP_POS_FRAC_BITS) - fs_samp_pos_wr) & SAMP_POS_MASK;

	fs_samp_pos_wr = fs_samp_pos_hd >> SAMP_POS_FRAC_BITS;

	if ((pos + len) > SAMPLES_PER_BUFFER) {

		midi_synth_render(&fs_buf[pos * 2], SAMPLES_PER_BUFFER - pos);

		len -= SAMPLES_PER_BUFFER - pos;

		pos = 0;

	}

	if (len > 0) {

		midi_synth_render(&fs_buf[pos * 2], len);

	}

	uint32_t wridx_old = wridx;

	uint32_t len_vera = (vera_samp_pos_hd - vera_samp_pos_rd) & SAMP_POS_MASK_FRAC;

	uint32_t len_ym = (ym_samp_pos_hd - ym_samp_pos_rd) & SAMP_POS_MASK_FRAC;

	uint32_t len_fs = (fs_samp_pos_hd - fs_samp_pos_rd) & SAMP_POS_MASK_FRAC;

	if (len_vera < (4 << SAMP_POS_FRAC_BITS) || len_ym < (4 << SAMP_POS_FRAC_BITS) || len_fs < (4 << SAMP_POS_FRAC_BITS)) {

		// not enough samples yet, at least 4 are needed for the filter

		return;

	}

	len_vera = (len_vera - (4 << SAMP_POS_FRAC_BITS)) / vera_samps_per_host_samps;

	len_ym = (len_ym - (4 << SAMP_POS_FRAC_BITS)) / ym_samps_per_host_samps;

	len_fs = (len_fs - (4 << SAMP_POS_FRAC_BITS)) / fs_samps_per_host_samps;

	len = SDL_min(len_vera, len_ym);

	len = SDL_min(len, len_fs);

	SDL_LockAudioDevice(audio_dev);

	for (int i = 0; i < len; i++) {

		int32_t samp[8];

		int32_t filter_idx = 0;

		int32_t vera_out_l = 0;

		int32_t vera_out_r = 0;

		int32_t ym_out_l = 0;

		int32_t ym_out_r = 0;

		int32_t fs_out_l = 0;

		int32_t fs_out_r = 0;

		// Don't resample VERA outputs if the host sample rate is as desired

		if (host_sample_rate == AUDIO_SAMPLERATE) {

			pos = (vera_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;

			vera_out_l = ((uint32_t)psg_buf[pos] + pcm_buf[pos]) << 14;

			vera_out_r = ((uint32_t)psg_buf[pos + 1] + pcm_buf[pos + 1]) << 14;

		} else {

			filter_idx = (vera_samp_pos_rd >> (SAMP_POS_FRAC_BITS - 8)) & 0xff;

			pos = (vera_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;

			for (int j = 0; j < 8; j += 2) {

				samp[j] = (int32_t)psg_buf[pos] + pcm_buf[pos];

				samp[j + 1] = (int32_t)psg_buf[pos + 1] + pcm_buf[pos + 1];

				pos = (pos + 2) & (SAMP_POS_MASK * 2);

			}

			vera_out_l += samp[0] * filter[256 + filter_idx];

			vera_out_r += samp[1] * filter[256 + filter_idx];

			vera_out_l += samp[2] * filter[  0 + filter_idx];

			vera_out_r += samp[3] * filter[  0 + filter_idx];

			vera_out_l += samp[4] * filter[255 - filter_idx];

			vera_out_r += samp[5] * filter[255 - filter_idx];

			vera_out_l += samp[6] * filter[511 - filter_idx];

			vera_out_r += samp[7] * filter[511 - filter_idx];

		}

		filter_idx = (ym_samp_pos_rd >> (SAMP_POS_FRAC_BITS - 8)) & 0xff;

		pos = (ym_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;

		for (int j = 0; j < 8; j += 2) {

			samp[j] = ym_buf[pos];

			samp[j + 1] = ym_buf[pos + 1];

			pos = (pos + 2) & (SAMP_POS_MASK * 2);

		}

		ym_out_l += samp[0] * filter[256 + filter_idx];

		ym_out_r += samp[1] * filter[256 + filter_idx];

		ym_out_l += samp[2] * filter[  0 + filter_idx];

		ym_out_r += samp[3] * filter[  0 + filter_idx];

		ym_out_l += samp[4] * filter[255 - filter_idx];

		ym_out_r += samp[5] * filter[255 - filter_idx];

		ym_out_l += samp[6] * filter[511 - filter_idx];

		ym_out_r += samp[7] * filter[511 - filter_idx];

		// Don't resample MIDI synth outputs if the host sample rate matches it

		if (host_sample_rate == AUDIO_SAMPLERATE) {

			pos = (fs_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;

			fs_out_l = (uint32_t)fs_buf[pos] << 14;

			fs_out_r = (uint32_t)fs_buf[pos + 1] << 14;

		} else {

			filter_idx = (fs_samp_pos_rd >> (SAMP_POS_FRAC_BITS - 8)) & 0xff;

			pos = (fs_samp_pos_rd >> SAMP_POS_FRAC_BITS) * 2;

			for (int j = 0; j < 8; j += 2) {

				samp[j] = fs_buf[pos];

				samp[j + 1] = fs_buf[pos + 1];

				pos = (pos + 2) & (SAMP_POS_MASK * 2);

			}

			fs_out_l += samp[0] * filter[256 + filter_idx];

			fs_out_r += samp[1] * filter[256 + filter_idx];

			fs_out_l += samp[2] * filter[  0 + filter_idx];

			fs_out_r += samp[3] * filter[  0 + filter_idx];

			fs_out_l += samp[4] * filter[255 - filter_idx];

			fs_out_r += samp[5] * filter[255 - filter_idx];

			fs_out_l += samp[6] * filter[511 - filter_idx];

			fs_out_r += samp[7] * filter[511 - filter_idx];

		}

		// VERA+YM mixing is according to the Developer Board

		// Loudest single PSG channel is 1/8 times the max output

		// mix = (psg + pcm) * 2 + ym + fs * 4

		int32_t mix_l = (vera_out_l >> 13) + (ym_out_l >> 15) + (fs_out_l >> 12);

		int32_t mix_r = (vera_out_r >> 13) + (ym_out_r >> 15) + (fs_out_r >> 12);

		uint32_t amp = SDL_max(SDL_abs(mix_l), SDL_abs(mix_r));

		if (amp > 32767) {

			uint32_t limiter_amp_new = (32767 << 16) / amp;

			limiter_amp = SDL_min(limiter_amp_new, limiter_amp);

		}

		buffer[wridx++] = (int16_t)((mix_l * limiter_amp) >> 16);

		buffer[wridx++] = (int16_t)((mix_r * limiter_amp) >> 16);

		x16_control_push_audio(buffer[wridx - 2], buffer[wridx - 1]);  /* RRDC /audio tee */

		if (limiter_amp < (1 << 16)) limiter_amp++;

		vera_samp_pos_rd = (vera_samp_pos_rd + vera_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

		ym_samp_pos_rd = (ym_samp_pos_rd + ym_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

		fs_samp_pos_rd = (fs_samp_pos_rd + fs_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

		if (wridx == buffer_size) {

			wav_recorder_process(&buffer[wridx_old], (buffer_size - wridx_old) / 2);

			wridx = 0;

			wridx_old = 0;

		}

	}

	if ((wridx - wridx_old) > 0) {

		wav_recorder_process(&buffer[wridx_old], (wridx - wridx_old) / 2);

	}

	buffer_written += len * 2;

	if (buffer_written > buffer_size) {

		// Prevent the buffer from overflowing by skipping the read pointer ahead.

		uint32_t buffer_skip_amount = (buffer_written / buffer_size) * SAMPLES_PER_BUFFER * 2;

		rdidx = (rdidx + buffer_skip_amount) % buffer_size;

		buffer_written -= buffer_skip_amount;

	}

	SDL_UnlockAudioDevice(audio_dev);

	// catch up all buffers if they are too far behind

	uint32_t skip = len_vera - len;

	if (skip > 1) {

		vera_samp_pos_rd = (vera_samp_pos_rd + vera_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

	}

	skip = len_ym - len;

	if (skip > 1) {

		ym_samp_pos_rd = (ym_samp_pos_rd + ym_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

	}

	skip = len_fs - len;

	if (skip > 1) {

		fs_samp_pos_rd = (fs_samp_pos_rd + fs_samps_per_host_samps) & SAMP_POS_MASK_FRAC;

	}

}

void

audio_usage(void)

{

	// SDL_GetAudioDeviceName doesn't work if audio isn't initialized.

	// Since argument parsing happens before initializing SDL, ensure the

	// audio subsystem is initialized before printing audio device names.

	SDL_InitSubSystem(SDL_INIT_AUDIO);

	// List all available sound devices

	printf("The following sound output devices are available:\n");

	const int sounds = SDL_GetNumAudioDevices(0);

	for (int i = 0; i < sounds; ++i) {

		printf("\t%s\n", SDL_GetAudioDeviceName(i, 0));

	}

	SDL_Quit();

	exit(1);

}