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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()
+{
+ m_data = &space(0);
+ // Find our direct access
+ m_cache = m_data->cache<0, 0, ENDIANNESS_LITTLE>();
+
+ m_channel = machine().sound().stream_alloc(*this, 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)
+{
+ m_channel->set_output_gain(0, volume / 100.0);
+ m_channel->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 - handle a stream update
+//-------------------------------------------------
+
+void s_dsp_device::sound_stream_update(sound_stream &stream, stream_sample_t **inputs, stream_sample_t **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];
+ }
+}