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path: root/src/devices/sound/s_dsp.cpp
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// license:LGPL-2.1+
// copyright-holders:R. Belmont, Brad Martin
/***************************************************************************

  s_dsp.cpp

  File to handle the S-DSP emulation used in Nintendo Super NES.

  By R. Belmont, adapted from OpenSPC 0.3.99 by Brad Martin with permission.
  Thanks to Brad and also to Charles Bilyu? of SNeESe.

  OpenSPC's license terms (the LGPL) follow:

 ---------------------------------------------------------------------------

  Copyright Brad Martin.

  OpenSPC is free software; you can redistribute it and/or modify
  it under the terms of the GNU Lesser General Public License as published by
  the Free Software Foundation; either version 2 of the License, or
  (at your option) any later version.

  OpenSPC is distributed in the hope that it will be useful,
  but WITHOUT ANY WARRANTY; without even the implied warranty of
  MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
  GNU Lesser General Public License for more details.

***************************************************************************/

#include "emu.h"
#include "s_dsp.h"

/***************************************************************************
 CONSTANTS AND MACROS
***************************************************************************/

static const int gauss[]=
{
	0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000,
	0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000,
	0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001,
	0x001, 0x001, 0x001, 0x002, 0x002, 0x002, 0x002, 0x002,
	0x002, 0x002, 0x003, 0x003, 0x003, 0x003, 0x003, 0x004,
	0x004, 0x004, 0x004, 0x004, 0x005, 0x005, 0x005, 0x005,
	0x006, 0x006, 0x006, 0x006, 0x007, 0x007, 0x007, 0x008,
	0x008, 0x008, 0x009, 0x009, 0x009, 0x00A, 0x00A, 0x00A,
	0x00B, 0x00B, 0x00B, 0x00C, 0x00C, 0x00D, 0x00D, 0x00E,
	0x00E, 0x00F, 0x00F, 0x00F, 0x010, 0x010, 0x011, 0x011,
	0x012, 0x013, 0x013, 0x014, 0x014, 0x015, 0x015, 0x016,
	0x017, 0x017, 0x018, 0x018, 0x019, 0x01A, 0x01B, 0x01B,
	0x01C, 0x01D, 0x01D, 0x01E, 0x01F, 0x020, 0x020, 0x021,
	0x022, 0x023, 0x024, 0x024, 0x025, 0x026, 0x027, 0x028,
	0x029, 0x02A, 0x02B, 0x02C, 0x02D, 0x02E, 0x02F, 0x030,
	0x031, 0x032, 0x033, 0x034, 0x035, 0x036, 0x037, 0x038,
	0x03A, 0x03B, 0x03C, 0x03D, 0x03E, 0x040, 0x041, 0x042,
	0x043, 0x045, 0x046, 0x047, 0x049, 0x04A, 0x04C, 0x04D,
	0x04E, 0x050, 0x051, 0x053, 0x054, 0x056, 0x057, 0x059,
	0x05A, 0x05C, 0x05E, 0x05F, 0x061, 0x063, 0x064, 0x066,
	0x068, 0x06A, 0x06B, 0x06D, 0x06F, 0x071, 0x073, 0x075,
	0x076, 0x078, 0x07A, 0x07C, 0x07E, 0x080, 0x082, 0x084,
	0x086, 0x089, 0x08B, 0x08D, 0x08F, 0x091, 0x093, 0x096,
	0x098, 0x09A, 0x09C, 0x09F, 0x0A1, 0x0A3, 0x0A6, 0x0A8,
	0x0AB, 0x0AD, 0x0AF, 0x0B2, 0x0B4, 0x0B7, 0x0BA, 0x0BC,
	0x0BF, 0x0C1, 0x0C4, 0x0C7, 0x0C9, 0x0CC, 0x0CF, 0x0D2,
	0x0D4, 0x0D7, 0x0DA, 0x0DD, 0x0E0, 0x0E3, 0x0E6, 0x0E9,
	0x0EC, 0x0EF, 0x0F2, 0x0F5, 0x0F8, 0x0FB, 0x0FE, 0x101,
	0x104, 0x107, 0x10B, 0x10E, 0x111, 0x114, 0x118, 0x11B,
	0x11E, 0x122, 0x125, 0x129, 0x12C, 0x130, 0x133, 0x137,
	0x13A, 0x13E, 0x141, 0x145, 0x148, 0x14C, 0x150, 0x153,
	0x157, 0x15B, 0x15F, 0x162, 0x166, 0x16A, 0x16E, 0x172,
	0x176, 0x17A, 0x17D, 0x181, 0x185, 0x189, 0x18D, 0x191,
	0x195, 0x19A, 0x19E, 0x1A2, 0x1A6, 0x1AA, 0x1AE, 0x1B2,
	0x1B7, 0x1BB, 0x1BF, 0x1C3, 0x1C8, 0x1CC, 0x1D0, 0x1D5,
	0x1D9, 0x1DD, 0x1E2, 0x1E6, 0x1EB, 0x1EF, 0x1F3, 0x1F8,
	0x1FC, 0x201, 0x205, 0x20A, 0x20F, 0x213, 0x218, 0x21C,
	0x221, 0x226, 0x22A, 0x22F, 0x233, 0x238, 0x23D, 0x241,
	0x246, 0x24B, 0x250, 0x254, 0x259, 0x25E, 0x263, 0x267,
	0x26C, 0x271, 0x276, 0x27B, 0x280, 0x284, 0x289, 0x28E,
	0x293, 0x298, 0x29D, 0x2A2, 0x2A6, 0x2AB, 0x2B0, 0x2B5,
	0x2BA, 0x2BF, 0x2C4, 0x2C9, 0x2CE, 0x2D3, 0x2D8, 0x2DC,
	0x2E1, 0x2E6, 0x2EB, 0x2F0, 0x2F5, 0x2FA, 0x2FF, 0x304,
	0x309, 0x30E, 0x313, 0x318, 0x31D, 0x322, 0x326, 0x32B,
	0x330, 0x335, 0x33A, 0x33F, 0x344, 0x349, 0x34E, 0x353,
	0x357, 0x35C, 0x361, 0x366, 0x36B, 0x370, 0x374, 0x379,
	0x37E, 0x383, 0x388, 0x38C, 0x391, 0x396, 0x39B, 0x39F,
	0x3A4, 0x3A9, 0x3AD, 0x3B2, 0x3B7, 0x3BB, 0x3C0, 0x3C5,
	0x3C9, 0x3CE, 0x3D2, 0x3D7, 0x3DC, 0x3E0, 0x3E5, 0x3E9,
	0x3ED, 0x3F2, 0x3F6, 0x3FB, 0x3FF, 0x403, 0x408, 0x40C,
	0x410, 0x415, 0x419, 0x41D, 0x421, 0x425, 0x42A, 0x42E,
	0x432, 0x436, 0x43A, 0x43E, 0x442, 0x446, 0x44A, 0x44E,
	0x452, 0x455, 0x459, 0x45D, 0x461, 0x465, 0x468, 0x46C,
	0x470, 0x473, 0x477, 0x47A, 0x47E, 0x481, 0x485, 0x488,
	0x48C, 0x48F, 0x492, 0x496, 0x499, 0x49C, 0x49F, 0x4A2,
	0x4A6, 0x4A9, 0x4AC, 0x4AF, 0x4B2, 0x4B5, 0x4B7, 0x4BA,
	0x4BD, 0x4C0, 0x4C3, 0x4C5, 0x4C8, 0x4CB, 0x4CD, 0x4D0,
	0x4D2, 0x4D5, 0x4D7, 0x4D9, 0x4DC, 0x4DE, 0x4E0, 0x4E3,
	0x4E5, 0x4E7, 0x4E9, 0x4EB, 0x4ED, 0x4EF, 0x4F1, 0x4F3,
	0x4F5, 0x4F6, 0x4F8, 0x4FA, 0x4FB, 0x4FD, 0x4FF, 0x500,
	0x502, 0x503, 0x504, 0x506, 0x507, 0x508, 0x50A, 0x50B,
	0x50C, 0x50D, 0x50E, 0x50F, 0x510, 0x511, 0x511, 0x512,
	0x513, 0x514, 0x514, 0x515, 0x516, 0x516, 0x517, 0x517,
	0x517, 0x518, 0x518, 0x518, 0x518, 0x518, 0x519, 0x519,
	0x519, 0x519, 0x519, 0x519, 0x519, 0x519, 0x519, 0x519
};

#undef DEBUG
#undef DBG_KEY
#undef DBG_ENV
#undef DBG_PMOD
#undef DBG_BRR
#undef DBG_ECHO
#undef DBG_INTRP

#undef NO_PMOD
#undef NO_ECHO

/* Ptrs to Gaussian table */
static const int *const G1 = &gauss[256];
static const int *const G2 = &gauss[512];
static const int *const G3 = &gauss[255];
static const int *const G4 = &gauss[0];

static const int        mask = 0xFF;

/* This table is for envelope timing.  It represents the number of counts
   that should be subtracted from the counter each sample period (32kHz).
   The counter starts at 30720 (0x7800). */
static const int CNT_INIT = 0x7800;
static const int ENVCNT[0x20]
	= {
	0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C,
	0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180,
	0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00,
	0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800
	};


ALLOW_SAVE_TYPE(s_dsp_device::env_state_t32);


DEFINE_DEVICE_TYPE(S_DSP, s_dsp_device, "s_dsp", "Nintendo/Sony S-DSP")


s_dsp_device::s_dsp_device(const machine_config &mconfig, const char *tag, device_t *owner, u32 clock)
	: device_t(mconfig, S_DSP, tag, owner, clock)
	, device_sound_interface(mconfig, *this)
	, device_memory_interface(mconfig, *this)
	, m_data_config("data", ENDIANNESS_LITTLE, 8, 16)
{
}


//-------------------------------------------------
//  device_start - device-specific startup
//-------------------------------------------------

void s_dsp_device::device_start()
{
	// Find our direct access
	space().cache(m_cache);
	space().specific(m_data);

	m_channel = stream_alloc_legacy(0, 2, clock() / 64);

	state_register();
}

//-------------------------------------------------
//  device_reset - device-specific reset
//-------------------------------------------------

void s_dsp_device::device_reset()
{
	dsp_reset();
}

//-------------------------------------------------
//  device_clock_changed - called if the clock
//  changes
//-------------------------------------------------

void s_dsp_device::device_clock_changed()
{
	m_channel->set_sample_rate(clock() / 64);
}

//-------------------------------------------------
//  memory_space_config - return a description of
//  any address spaces owned by this device
//-------------------------------------------------

device_memory_interface::space_config_vector s_dsp_device::memory_space_config() const
{
	return space_config_vector{ std::make_pair(0, &m_data_config) };
}


/*****************************************************************************
    IMPLEMENTATION
*****************************************************************************/


/*-------------------------------------------------
 dsp_reset

 Reset emulated DSP
-------------------------------------------------*/

void s_dsp_device::dsp_reset()
{
#ifdef MAME_DEBUG
	logerror("dsp_reset\n");
#endif

	for (int i = 0; i < 8; i++)
	{
#ifndef NO_ECHO
		m_fir_lbuf[i] = 0;
		m_fir_rbuf[i] = 0;
#endif
		m_voice_state[i].on_cnt = 0;
	}

#ifndef NO_ECHO
	m_fir_ptr   = 0;
	m_echo_ptr = 0;
#endif

	m_keys      = 0;
	m_keyed_on  = 0;
	m_noise_cnt = 0;
	m_noise_lev = 0x4000;
	m_dsp_regs[0x6c] |= 0xe0;
	m_dsp_regs[0x4c] = 0;
	m_dsp_regs[0x5c] = 0;
}


/*-------------------------------------------------
 dsp_update

 Mix one sample of audio. sound_ptr is a pointer
 to mix audio into
-------------------------------------------------*/

void s_dsp_device::dsp_update( s16 *sound_ptr )
{
	int V;

	int envx;
	int m;
	int v;
	int vl;
	voice_state_type * vp;
	int vr;

	const u16 sd = m_dsp_regs[0x5d];

	/* Check for reset */
	if (m_dsp_regs[0x6c] & 0x80)
		dsp_reset();

	/* Here we check for keys on/off.  Docs say that successive writes to KON/KOF
	must be separated by at least 2 Ts periods or risk being neglected.
	Therefore DSP only looks at these during an update, and not at the time of
	the write.  Only need to do this once however, since the regs haven't
	changed over the whole period we need to catch up with. */
#ifdef DBG_KEY
	m_dsp_regs[0x4c] &= mask;
#endif

	/* Keying on a voice resets that bit in ENDX */
	m_dsp_regs[0x7c] &= ~m_dsp_regs[0x4c];

	/* Question: what is the expected behavior when pitch modulation is enabled on
	voice 0?  Jurassic Park 2 does this.  For now, using outx of zero for first
	voice. */
	s32 outx = 0;       /* Smpl height (must be signed) */

	/* Same table for noise and envelope */
	m_noise_cnt -= ENVCNT[m_dsp_regs[0x6c] & 0x1f];
	if (m_noise_cnt <= 0)
	{
		m_noise_cnt = CNT_INIT;
		m_noise_lev = (((m_noise_lev << 13) ^ (m_noise_lev << 14)) & 0x4000) | (m_noise_lev >> 1);
	}

	int outl  = 0;
	int outr  = 0;

#ifndef NO_ECHO
	int echol = 0;
	int echor = 0;
#endif

	for (v = 0, m = 1, V = 0; v < 8; v++, V += 16, m <<= 1)
	{
		vp = &m_voice_state[v];

		if (vp->on_cnt && (--vp->on_cnt == 0))
		{
			/* Voice was keyed on */
			m_keys       |= m;
			m_keyed_on   |= m;
#ifdef DBG_KEY
			vl          = m_dsp_regs[(v << 4) + 4];
#endif
			vp->samp_id = (vptr(sd, V) << 16) | lptr(sd, V);
			vp->mem_ptr = vptr(sd, V);

#ifdef DBG_KEY
			logerror("Keying on voice %d, samp=0x%04X (0x%02X)\n", v, vp->mem_ptr, vl);
#endif

			vp->header_cnt = 0;
			vp->half       = 0;
			vp->envx       = 0;
			vp->end        = 0;
			vp->sampptr    = 0;
			vp->mixfrac    = 3 * 4096;

			/* NOTE: Real SNES does *not* appear to initialize the envelope
			counter to anything in particular.  The first cycle always seems to
			come at a random time sooner than expected; as yet, I have been
			unable to find any pattern.  I doubt it will matter though, so
			we'll go ahead and do the full time for now. */
			vp->envcnt   = CNT_INIT;
			vp->envstate = env_state_t32::ATTACK;
		}

		if (m_dsp_regs[0x4c] & m & ~m_dsp_regs[0x5c])
		{
			/* Voice doesn't come on if key off is set */
			m_dsp_regs[0x4c] &= ~m;
			vp->on_cnt       = 8;

#ifdef DBG_KEY
			logerror("Key on set for voice %d\n", v);
#endif
		}

		if (m_keys & m_dsp_regs[0x5c] & m)
		{
			/* Voice was keyed off */
			vp->envstate = env_state_t32::RELEASE;
			vp->on_cnt   = 0;

#ifdef DBG_KEY
			logerror("Keying off voice %d\n", v);
#endif
		}

		if (!(m_keys & m & mask) || ((envx = advance_envelope(v)) < 0))
		{
			m_dsp_regs[V + 8] = 0;
			m_dsp_regs[V + 9] = 0;
			outx             = 0;
			continue;
		}

		vp->pitch = pitch(V);

#ifndef NO_PMOD
		/* Pitch mod uses OUTX from last voice for this one.  Luckily we haven't
		modified OUTX since it was used for last voice. */
		if (m_dsp_regs[0x2d] & m)
		{
#ifdef DBG_PMOD
			logerror("Pitch Modulating voice %d, outx=%ld, old pitch=%d, ", v, outx, vp->pitch);
#endif
			vp->pitch = (vp->pitch * (outx + 32768)) >> 15;
			}
#endif

#ifdef DBG_PMOD
			logerror("pitch=%d\n", vp->pitch);
#endif

			for ( ; vp->mixfrac >= 0; vp->mixfrac -= 4096)
			{
				/* This part performs the BRR decode 'on-the-fly'.  This is more
				correct than the old way, which could be fooled if the data and/or
				the loop point changed while the sample was playing, or if the BRR
				decode didn't produce the same result every loop because of the
				filters.  The event interface still has no chance of keeping up
				with those kinds of tricks, though. */
				if (!vp->header_cnt)
				{
					if (vp->end & 1)
					{
						/* Docs say ENDX bit is set when decode of block with source
						end flag set is done.  Does this apply to looping samples?
						Some info I've seen suggests yes. */
						m_dsp_regs[0x7c] |= m;
						if (vp->end & 2)
						{
							vp->mem_ptr = lptr(sd, V);

#ifdef DBG_BRR
							logerror("BRR looping to 0x%04X\n", vp->mem_ptr);
#endif
						}
						else
						{
#ifdef DBG_KEY
							logerror("BRR decode end, voice %d\n", v);
#endif

							m_keys &= ~m;
							m_dsp_regs[V + 8] = 0;
							vp->envx         = 0;
							while (vp->mixfrac >= 0)
							{
								vp->sampbuf[vp->sampptr] = 0;
								outx         = 0;
								vp->sampptr  = (vp->sampptr + 1) & 3;
								vp->mixfrac -= 4096;
							}
							break;
						}
					}

					vp->header_cnt = 8;
					vl = (u8)read_byte(vp->mem_ptr++);
					vp->range  = vl >> 4;
					vp->end    = vl & 3;
					vp->filter = (vl & 12) >> 2;

#ifdef DBG_BRR
					logerror("V%d: header read, range=%d, end=%d, filter=%d\n", v, vp->range, vp->end, vp->filter);
#endif
				}

				if (vp->half == 0)
				{
					vp->half = 1;
					outx     = ((s8)read_byte(vp->mem_ptr)) >> 4;
				}
				else
				{
					vp->half = 0;
					/* Funkiness to get 4-bit signed to carry through */
					outx   = (s8)(read_byte(vp->mem_ptr++) << 4);
					outx >>= 4;
					vp->header_cnt--;
				}

#ifdef DBG_BRR
				logerror("V%d: nybble=%X, ptr=%04X, smp1=%d, smp2=%d\n", v, outx & 0x0f, vp->mem_ptr, vp->smp1, vp->smp2);
#endif

				/* For invalid ranges (D,E,F): if the nybble is negative, the result
				is F000.  If positive, 0000.  Nothing else like previous range,
				etc. seems to have any effect.  If range is valid, do the shift
				normally.  Note these are both shifted right once to do the filters
				properly, but the output will be shifted back again at the end. */
				if (vp->range <= 0xc)
				{
					outx = (outx << vp->range) >> 1;
				}
				else
				{
					outx &= ~0x7ff;

#ifdef DBG_BRR
					logerror("V%d: invalid range! (%X)\n", v, vp->range);
#endif
				}

#ifdef DBG_BRR
				logerror("V%d: shifted delta=%04X\n", v, (u16)outx);
#endif

				switch (vp->filter)
				{
				case 0:
					break;

				case 1:
					outx += (vp->smp1 >> 1) + ((-vp->smp1) >> 5);
					break;

				case 2:
					outx += vp->smp1 + ((-(vp->smp1 + (vp->smp1 >> 1))) >> 5) - (vp->smp2 >> 1) + (vp->smp2 >> 5);
					break;

				case 3:
					outx += vp->smp1 + ((-(vp->smp1 + (vp->smp1 << 2) + (vp->smp1 << 3))) >> 7)
							- (vp->smp2 >> 1) + ((vp->smp2 + (vp->smp2 >> 1)) >> 4);
					break;
				}

				if (outx < (s16)0x8000)
				{
					outx = (s16)0x8000;
				}
				else if (outx > (s16)0x7fff)
				{
					outx = (s16)0x7fff;
				}

#ifdef DBG_BRR
				logerror("V%d: filter + delta=%04X\n", v, (u16)outx);
#endif

				vp->smp2 = (s16)vp->smp1;
				vp->smp1 = (s16)(outx << 1);
				vp->sampbuf[vp->sampptr] = vp->smp1;

#ifdef DBG_BRR
				logerror("V%d: final output: %04X\n", v, vp->sampbuf[vp->sampptr]);
#endif

				vp->sampptr = (vp->sampptr + 1) & 3;
			}

			if (m_dsp_regs[0x3d] & m)
			{
#ifdef DBG_PMOD
				logerror("Noise enabled, voice %d\n", v);
#endif
				outx = (s16)(m_noise_lev << 1);
			}
			else
			{
			/* Perform 4-Point Gaussian interpolation.  Take an approximation of a
			Gaussian bell-curve, and move it through the sample data at a rate
			determined by the pitch.  The sample output at any given time is
			the sum of the products of each input sample point with the value
			of the bell-curve corresponding to that point. */
			vl  = vp->mixfrac >> 4;
			vr  = ((G4[-vl-1] * vp->sampbuf[vp->sampptr]) >> 11 ) & ~1;
			vr += ((G3[-vl] * vp->sampbuf[(vp->sampptr + 1) & 3]) >> 11) & ~1;
			vr += ((G2[vl] * vp->sampbuf[(vp->sampptr + 2) & 3]) >> 11 ) & ~1;

			/* This is to do the wrapping properly.  Based on my tests with the
			SNES, it appears clipping is done only if it is the fourth addition
			that would cause a wrap.  If it has already wrapped before the
			fourth addition, it is not clipped. */
			vr  = (s16)vr;
			vr += ((G1[vl] * vp->sampbuf[(vp->sampptr + 3) & 3]) >> 11) & ~1;

			if (vr > 32767)
				vr = 32767;
			else if (vr < -32768)
				vr = -32768;

			outx = (s16)vr;

#ifdef DBG_INTRP
			logerror("V%d: mixfrac=%d: [%d]*%d + [%d]*%d + [%d]*%d + [%d]*%d = %d\n", v, vl,
				G1[vl],
				vp->sampbuf[(vp->sampptr + 3) & 3],
				G2[vl],
				vp->sampbuf[(vp->sampptr + 2) & 3],
				G3[-vl],
				vp->sampbuf[(vp->sampptr + 1) & 3],
				G4[-vl-1],
				vp->sampbuf[vp->sampptr],
				outx);
#endif
		}

		/* Advance the sample position for next update. */
		vp->mixfrac += vp->pitch;

		outx = ((outx * envx) >> 11) & ~1;
		m_dsp_regs[V + 9] = outx >> 8;

		vl = (((int)(s8)m_dsp_regs[V    ]) * outx) >> 7;
		vr = (((int)(s8)m_dsp_regs[V + 1]) * outx) >> 7;
		outl += vl;
		outr += vr;

		if (m_dsp_regs[0x4d] & m)
		{
#ifndef NO_ECHO
			echol += vl;
			echor += vr;
#endif
		}
	}

	outl = (outl * (s8)m_dsp_regs[0x0c]) >> 7;
	outr = (outr * (s8)m_dsp_regs[0x1c]) >> 7;

#ifndef NO_ECHO
	/* Perform echo.  First, read mem at current location, and put those samples
	into the FIR filter queue. */
#ifdef DBG_ECHO
	logerror("Echo delay=%dms, feedback=%d%%\n", m_dsp_regs[0x7d] * 16,
		((s8)m_dsp_regs[0x0d] * 100) / 0x7f);
#endif

	int echo_base = ((m_dsp_regs[0x6d] << 8) + m_echo_ptr) & 0xffff;
	m_fir_lbuf[m_fir_ptr] = (s16)read_word(echo_base);
	m_fir_rbuf[m_fir_ptr] = (s16)read_word(echo_base + sizeof(s16));

	/* Now, evaluate the FIR filter, and add the results into the final output. */
	vl = m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x7f];
	vr = m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x7f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x6f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x6f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x5f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x5f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x4f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x4f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x3f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x3f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x2f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x2f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x1f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x1f];
	m_fir_ptr = (m_fir_ptr + 1) & 7;
	vl += m_fir_lbuf[m_fir_ptr] * (s8)m_dsp_regs[0x0f];
	vr += m_fir_rbuf[m_fir_ptr] * (s8)m_dsp_regs[0x0f];

#ifdef DBG_ECHO
	logerror("FIR Coefficients: %02X %02X %02X %02X %02X %02X %02X %02X\n",
		m_dsp_regs[0x0f],
		m_dsp_regs[0x1f],
		m_dsp_regs[0x2f],
		m_dsp_regs[0x3f],
		m_dsp_regs[0x4f],
		m_dsp_regs[0x5f],
		m_dsp_regs[0x6f],
		m_dsp_regs[0x7f]);
#endif

	/* FIR_ptr is left in the position of the oldest sample, the one that will be replaced next update. */
	outl += vl * (s8)m_dsp_regs[0x2c] >> 14;
	outr += vr * (s8)m_dsp_regs[0x3c] >> 14;

	if (!(m_dsp_regs[0x6c] & 0x20))
	{
		/* Add the echo feedback back into the original result, and save that into memory for use later. */
		echol += vl * (s8)m_dsp_regs[0x0d] >> 14;

		if (echol > 32767)
			echol = 32767;
		else if (echol < -32768)
			echol = -32768;

		echor += vr * (s8)m_dsp_regs[0x0D ] >> 14;

		if (echor > 32767)
			echor = 32767;
		else if (echor < -32768)
			echor = -32768;

#ifdef DBG_ECHO
		logerror("Echo: Writing %04X,%04X at location %04X\n", (u16)echol, (u16)echor, echo_base);
#endif

		write_word(echo_base, (u16)echol);
		write_word(echo_base + sizeof(s16), (u16)echor);
	}

	m_echo_ptr += 2 * sizeof(s16);

	if (m_echo_ptr >= ((m_dsp_regs[0x7d] & 0x0f) << 11))
	{
		m_echo_ptr = 0;
	}
#endif                              /* !defined( NO_ECHO ) */

	if (sound_ptr != nullptr)
	{
		if (m_dsp_regs[0x6c] & 0x40)
		{
			/* MUTE */
#ifdef MAME_DEBUG
			logerror("MUTED!\n");
#endif

			*sound_ptr = 0;
			sound_ptr++;
			*sound_ptr = 0;
			sound_ptr++;
		}
		else
		{
			if (outl > 32767)
					*sound_ptr = 32767;
			else if (outl < -32768)
				*sound_ptr = -32768;
			else
				*sound_ptr = outl;

			sound_ptr++;

			if (outr > 32767)
					*sound_ptr = 32767;
			else if (outr < -32768)
				*sound_ptr = -32768;
			else
				*sound_ptr = outr;

			sound_ptr++;
		}
	}
}


/*-------------------------------------------------
 advance_envelope

 Run envelope step & return ENVX. v is the voice
 to process envelope for.
-------------------------------------------------*/

int s_dsp_device::advance_envelope( int v )
{
	int t;

	int envx = m_voice_state[v].envx;

	if (m_voice_state[v].envstate == env_state_t32::RELEASE)
	{
		/* Docs: "When in the state of "key off". the "click" sound is prevented
		by the addition of the fixed value 1/256"  WTF???  Alright, I'm going
		to choose to interpret that this way:  When a note is keyed off, start
		the RELEASE state, which subtracts 1/256th each sample period (32kHz).
		Note there's no need for a count because it always happens every
		update. */
		envx -= 0x8;                    /* 0x8 / 0x800 = 1/256th        */
		if (envx <= 0)
		{
			envx = 0;
			m_keys &= ~(1 << v);
			return -1;
		}

		m_voice_state[v].envx = envx;
		m_dsp_regs[(v << 4) + 8] = envx >> 8;

#ifdef DBG_ENV
		logerror("ENV voice %d: envx=%03X, state=RELEASE\n", v, envx);
#endif

		return envx;
	}

	int cnt = m_voice_state[v].envcnt;
	int adsr1 = m_dsp_regs[(v << 4) + 5];

	if (adsr1 & 0x80)
	{
		switch (m_voice_state[v].envstate)
		{
		case env_state_t32::ATTACK:
			/* Docs are very confusing.  "AR is multiplied by the fixed value
			1/64..."  I believe it means to add 1/64th to ENVX once every
			time ATTACK is updated, and that's what I'm going to implement. */
			t = adsr1 & 0x0f;

				if (t == 0x0f)
			{
#ifdef DBG_ENV
				logerror("ENV voice %d: instant attack\n", v);
#endif

				envx += 0x400;
			}
			else
			{
				cnt -= ENVCNT[(t << 1) + 1];

				if (cnt > 0)
					break;

				envx += 0x20;       /* 0x020 / 0x800 = 1/64         */
				cnt   = CNT_INIT;
			}

			if (envx > 0x7ff)
			{
				envx = 0x7ff;
				m_voice_state[v].envstate = env_state_t32::DECAY;
			}

#ifdef DBG_ENV
			logerror("ENV voice %d: envx=%03X, state=ATTACK\n", v, envx);
#endif

			m_voice_state[v].envx = envx;
			break;

		case env_state_t32::DECAY:
			/* Docs: "DR... [is multiplied] by the fixed value 1-1/256."
			Well, at least that makes some sense.  Multiplying ENVX by
			255/256 every time DECAY is updated. */
			cnt -= ENVCNT[((adsr1 >> 3) & 0x0e) + 0x10];

			if (cnt <= 0)
			{
				cnt   = CNT_INIT;
				envx -= ((envx - 1) >> 8) + 1;
				m_voice_state[v].envx = envx;
			}

			if (envx <= 0x100 * (SL(v) + 1))
				m_voice_state[v].envstate = env_state_t32::SUSTAIN;

#ifdef DBG_ENV
			logerror("ENV voice %d: envx=%03X, state=DECAY\n", v, envx);
#endif

			break;

		case env_state_t32::SUSTAIN:
			/* Docs: "SR [is multiplied] by the fixed value 1-1/256."
			Multiplying ENVX by 255/256 every time SUSTAIN is updated. */
#ifdef DBG_ENV
			if (ENVCNT[SR(v)] == 0)
				logerror("ENV voice %d: envx=%03X, state=SUSTAIN, zero rate\n", v, envx);
#endif

			cnt -= ENVCNT[SR(v)];
			if (cnt > 0)
				break;

			cnt   = CNT_INIT;
			envx -= ((envx - 1) >> 8) + 1;

#ifdef DBG_ENV
			logerror("ENV voice %d: envx=%03X, state=SUSTAIN\n", v, envx);
#endif

			m_voice_state[v].envx = envx;

			/* Note: no way out of this state except by explicit KEY OFF (or switch to GAIN). */
			break;

		case env_state_t32::RELEASE:   /* Handled earlier to prevent GAIN mode from stopping KEY OFF events */
			break;
		}
	}
	else
	{
		/* GAIN mode is set
		Note: if the game switches between ADSR and GAIN modes partway
		through, should the count be reset, or should it continue from
		where it was?  Does the DSP actually watch for that bit to
		change, or does it just go along with whatever it sees when it
		performs the update?  I'm going to assume the latter and not
		update the count, unless I see a game that obviously wants the
		other behavior.  The effect would be pretty subtle, in any case.
		*/
		t = m_dsp_regs[(v << 4) + 7];

		if (t < 0x80)
		{
			envx = t << 4;
			m_voice_state[v].envx = envx;

#ifdef DBG_ENV
			logerror("ENV voice %d: envx=%03X, state=DIRECT\n", v, envx);
#endif
		}
		else
		{
			switch (t >> 5)
			{
			case 4:
				/* Docs: "Decrease (linear): Subtraction of the fixed value 1/64." */
				cnt -= ENVCNT[t & 0x1f];

				if (cnt > 0)
					break;

				cnt   = CNT_INIT;
				envx -= 0x020;      /* 0x020 / 0x800 = 1/64th       */

				if (envx < 0)
					envx = 0;

#ifdef DBG_ENV
				logerror("ENV voice %d: envx=%03X, state=DECREASE\n", v, envx);
#endif

				m_voice_state[v].envx = envx;
				break;

			case 5:
				/* Docs: "Drecrease <sic> (exponential): Multiplication by the fixed value 1-1/256." */
				cnt -= ENVCNT[t & 0x1f];

				if (cnt > 0)
					break;

				cnt = CNT_INIT;
				envx -= ((envx - 1) >> 8) + 1;

#ifdef DBG_ENV
				logerror("ENV voice %d: envx=%03X, state=EXP\n", v, envx);
#endif

				m_voice_state[v].envx = envx;
				break;

			case 6:
				/* Docs: "Increase (linear): Addition of the fixed value 1/64." */
				cnt -= ENVCNT[t & 0x1f];

				if (cnt > 0)
					break;

				cnt = CNT_INIT;
				envx += 0x020;      /* 0x020 / 0x800 = 1/64th   */
				if (envx > 0x7ff)
					envx = 0x7ff;

#ifdef DBG_ENV
				logerror("ENV voice %d: envx=%03X, state=INCREASE\n", v, envx);
#endif

				m_voice_state[v].envx = envx;
				break;

			case 7:
				/* Docs: "Increase (bent line): Addition of the constant
				     1/64 up to .75 of the constaint <sic> 1/256 from .75 to 1." */
				cnt -= ENVCNT[t & 0x1f];

				if (cnt > 0)
					break;

				cnt = CNT_INIT;

				if (envx < 0x600)  /* 0x600 / 0x800 = .75          */
					envx += 0x020;  /* 0x020 / 0x800 = 1/64         */
				else
					envx += 0x008;  /* 0x008 / 0x800 = 1/256        */

				if (envx > 0x7ff)
					envx=0x7ff;

#ifdef DBG_ENV
				logerror("ENV voice %d: envx=%03X, state=INCREASE\n", v, envx);
#endif

				m_voice_state[v].envx = envx;
				break;
			}
		}
	}

	m_voice_state[v].envcnt = cnt;
	m_dsp_regs[(v << 4) + 8] = envx >> 4;

	return envx;
}

/*-------------------------------------------------
    set_volume - sets S-DSP volume level
    for both speakers, used for fade in/out effects
-------------------------------------------------*/

void s_dsp_device::set_volume(int volume)
{
	set_output_gain(0, volume / 100.0);
	set_output_gain(1, volume / 100.0);
}


/***************************
         I/O for DSP
 ***************************/

u8 s_dsp_device::dsp_io_r(offs_t offset)
{
	m_channel->update();

#ifdef NO_ENVX
	if (!machine().side_effects_disabled())
		if (8 == (m_dsp_addr & 0x0f))
			m_dsp_regs[m_dsp_addr] = 0;
#endif

	/* All reads simply return the contents of the addressed register. */
	if (offset & 1)
		return m_dsp_regs[m_dsp_addr & 0x7f];

	return m_dsp_addr;
}

void s_dsp_device::dsp_io_w(offs_t offset, u8 data)
{
	m_channel->update();

	if (offset & 1)
	{
		if (!(m_dsp_addr & 0x80))
		{
			offset = m_dsp_addr & 0x7f;
			if (offset == 0x7c)
			{
				/* Writes to register 0x7c (ENDX) clear ALL bits no matter which value is written */
				m_dsp_regs[offset & 0x7f] = 0;
			}
			else
			{
				/* All other writes store the value in the addressed register as expected. */
				m_dsp_regs[offset & 0x7f] = data;
			}
		}
	}
	else
		m_dsp_addr = data;
}

/*****************************************************************************
    DEVICE INTERFACE
*****************************************************************************/

void s_dsp_device::state_register()
{
	save_item(NAME(m_dsp_addr));
	save_item(NAME(m_dsp_regs));

	save_item(NAME(m_keyed_on));
	save_item(NAME(m_keys));

	save_item(NAME(m_noise_cnt));
	save_item(NAME(m_noise_lev));

#ifndef NO_ECHO
	save_item(NAME(m_fir_lbuf));
	save_item(NAME(m_fir_rbuf));
	save_item(NAME(m_fir_ptr));
	save_item(NAME(m_echo_ptr));
#endif

	for (int v = 0; v < 8; v++)
	{
		save_item(NAME(m_voice_state[v].mem_ptr), v);
		save_item(NAME(m_voice_state[v].end), v);
		save_item(NAME(m_voice_state[v].envcnt), v);
		save_item(NAME(m_voice_state[v].envstate), v);
		save_item(NAME(m_voice_state[v].envx), v);
		save_item(NAME(m_voice_state[v].filter), v);
		save_item(NAME(m_voice_state[v].half), v);
		save_item(NAME(m_voice_state[v].header_cnt), v);
		save_item(NAME(m_voice_state[v].mixfrac), v);
		save_item(NAME(m_voice_state[v].on_cnt), v);
		save_item(NAME(m_voice_state[v].pitch), v);
		save_item(NAME(m_voice_state[v].range), v);
		save_item(NAME(m_voice_state[v].samp_id), v);
		save_item(NAME(m_voice_state[v].sampptr), v);
		save_item(NAME(m_voice_state[v].smp1), v);
		save_item(NAME(m_voice_state[v].smp2), v);
		save_item(NAME(m_voice_state[v].sampbuf), v);
	}
}

//-------------------------------------------------
//  sound_stream_update_legacy - handle a stream update
//-------------------------------------------------

void s_dsp_device::sound_stream_update_legacy(sound_stream &stream, stream_sample_t const * const *inputs, stream_sample_t * const *outputs, int samples)
{
	s16 mix[2];

	for (int i = 0; i < samples; i++)
	{
		mix[0] = mix[1] = 0;
		dsp_update(mix);

		/* Update the buffers */
		outputs[0][i] = (stream_sample_t)mix[0];
		outputs[1][i] = (stream_sample_t)mix[1];
	}
}