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Diffstat (limited to 'src/devices/sound/s_dsp.cpp')
-rw-r--r-- | src/devices/sound/s_dsp.cpp | 1080 |
1 files changed, 1080 insertions, 0 deletions
diff --git a/src/devices/sound/s_dsp.cpp b/src/devices/sound/s_dsp.cpp new file mode 100644 index 00000000000..976fb33f5a2 --- /dev/null +++ b/src/devices/sound/s_dsp.cpp @@ -0,0 +1,1080 @@ +// 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]; + } +} |