// license:LGPL-2.1+ // copyright-holders:R. Belmont, Brad Martin /*************************************************************************** snes_snd.cpp File to handle the sound emulation of the 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 "audio/snes_snd.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 }; /* Make reading the ADSR code easier */ #define SL( v ) (m_dsp_regs[((v) << 4) + 6] >> 5) /* Returns SUSTAIN level */ #define SR( v ) (m_dsp_regs[((v) << 4) + 6] & 0x1f) /* Returns SUSTAIN rate */ /* Handle endianness */ #define LEtoME16( x ) little_endianize_int16(x) #define MEtoLE16( x ) little_endianize_int16(x) ALLOW_SAVE_TYPE(snes_sound_device::env_state_t32); DEFINE_DEVICE_TYPE(SNES_SOUND, snes_sound_device, "snes_sound", "SNES Custom DSP (SPC700)") snes_sound_device::snes_sound_device(const machine_config &mconfig, const char *tag, device_t *owner, u32 clock) : device_t(mconfig, SNES_SOUND, tag, owner, clock) , device_sound_interface(mconfig, *this) { } //------------------------------------------------- // device_start - device-specific startup //------------------------------------------------- void snes_sound_device::device_start() { m_channel = machine().sound().stream_alloc(*this, 0, 2, clock() / 64); m_ram = make_unique_clear(SNES_SPCRAM_SIZE); /* put IPL image at the top of RAM */ memcpy(m_ipl_region, machine().root_device().memregion("sound_ipl")->base(), 64); m_tick_timer = timer_alloc(TIMER_TICK_ID); state_register(); save_pointer(NAME(m_ram), SNES_SPCRAM_SIZE); } //------------------------------------------------- // device_reset - device-specific reset //------------------------------------------------- void snes_sound_device::device_reset() { int i; /* default to ROM visible */ m_ram[0xf1] = 0x80; /* Sort out the ports */ for (i = 0; i < 4; i++) { m_port_in[i] = 0; m_port_out[i] = 0; } for (i = 0; i < 3; i++) { m_timer_enabled[i] = false; m_TnDIV[i] = 256; m_counter[i] = 0; m_subcounter[i] = 0; } attotime period = attotime::from_ticks(32, clock()); m_tick_timer->adjust(period, 0, period); dsp_reset(); } //------------------------------------------------- // device_clock_changed - called if the clock // changes //------------------------------------------------- void snes_sound_device::device_clock_changed() { m_channel->set_sample_rate(clock() / 64); attotime period = attotime::from_ticks(32, clock()); m_tick_timer->adjust(period, 0, period); } inline void snes_sound_device::update_timer_tick(u8 which) { if (m_timer_enabled[which] == false) return; m_subcounter[which]++; // if timer channel is 0 or 1 we update at 64000/8 if (m_subcounter[which] >= 8 || which == 2) { m_subcounter[which] = 0; m_counter[which]++; if (m_counter[which] >= m_TnDIV[which] ) // minus = { m_counter[which] = 0; m_ram[0xfd + which]++; m_ram[0xfd + which] &= 0x0f; } } } void snes_sound_device::device_timer(emu_timer &timer, device_timer_id id, int param, void *ptr) { if (id != TIMER_TICK_ID) throw emu_fatalerror("Unknown id in snes_sound_device::device_timer"); for (int ch = 0; ch < 3; ch++) update_timer_tick(ch); } /***************************************************************************** IMPLEMENTATION *****************************************************************************/ /*------------------------------------------------- dsp_reset Reset emulated DSP -------------------------------------------------*/ void snes_sound_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 snes_sound_device::dsp_update( s16 *sound_ptr ) { int V; int envx; int m; src_dir_type * sd; int v; int vl; voice_state_type * vp; int vr; sd = (src_dir_type *) &m_ram[(int) m_dsp_regs[0x5d] << 8]; /* 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; vl = m_dsp_regs[(v << 4) + 4]; vp->samp_id = *( u32 * )&sd[vl]; vp->mem_ptr = LEtoME16(sd[vl].vptr); #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 = LEtoME16(*((u16 *)&m_dsp_regs[V + 2])) & 0x3fff; #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 = LEtoME16(sd[m_dsp_regs[V + 4]].lptr); #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)m_ram[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)m_ram[vp->mem_ptr]) >> 4; } else { vp->half = 0; /* Funkiness to get 4-bit signed to carry through */ outx = (s8)(m_ram[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)LEtoME16(*(u16 *)&m_ram[echo_base]); m_fir_rbuf[m_fir_ptr] = (s16)LEtoME16(*(u16 *)&m_ram[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 *(u16 *)&m_ram[echo_base] = MEtoLE16((u16)echol); *(u16 *)&m_ram[echo_base + sizeof(s16)] = MEtoLE16((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 snes_sound_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 (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 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; } /*------------------------------------------------- spc700_set_volume - sets SPC700 volume level for both speakers, used for fade in/out effects -------------------------------------------------*/ void snes_sound_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 snes_sound_device::dsp_io_r(offs_t offset) { m_channel->update(); #ifdef NO_ENVX if (8 == (m_ram[0xf2] & 0x0f)) m_dsp_regs[m_ram[0xf2]] = 0; #endif /* All reads simply return the contents of the addressed register. */ return m_dsp_regs[offset & 0x7f]; } void snes_sound_device::dsp_io_w(offs_t offset, u8 data) { m_channel->update(); if (offset == 0x7c) { /* Writes to register 0x7c (ENDX) clear ALL bits no matter which value is written */ m_dsp_regs[offset] = 0; } else { /* All other writes store the value in the addressed register as expected. */ m_dsp_regs[offset] = data; } } /*************************** I/O for SPC700 ***************************/ u8 snes_sound_device::spc_io_r(offs_t offset) { switch (offset) /* Offset is from 0x00f0 */ { case 0x0: //FIXME: Super Bomberman PBW reads from there, is it really write-only? return 0; case 0x1: return 0; //Super Kick Boxing reads port 1 and wants it to be zero. case 0x2: /* Register address */ return m_ram[0xf2]; case 0x3: /* Register data */ return dsp_io_r(m_ram[0xf2]); case 0x4: /* Port 0 */ case 0x5: /* Port 1 */ case 0x6: /* Port 2 */ case 0x7: /* Port 3 */ // osd_printf_debug("%s SPC: rd %02x @ %d\n", machine().describe_context(), m_port_in[offset - 4], offset - 4); return m_port_in[offset - 4]; case 0x8: //normal RAM, can be read even if the ram disabled flag ($f0 bit 1) is active case 0x9: return m_ram[0xf0 + offset]; case 0xa: /* Timer 0 */ case 0xb: /* Timer 1 */ case 0xc: /* Timer 2 */ break; case 0xd: /* Counter 0 */ case 0xe: /* Counter 1 */ case 0xf: /* Counter 2 */ { u8 value = m_ram[0xf0 + offset] & 0x0f; m_ram[0xf0 + offset] = 0; return value; } } return 0; } void snes_sound_device::spc_io_w(offs_t offset, u8 data) { switch (offset) /* Offset is from 0x00f0 */ { case 0x0: logerror("Warning: write to SOUND TEST register with data %02x!\n", data); break; case 0x1: /* Control */ for (int i = 0; i < 3; i++) { if (BIT(data, i) && m_timer_enabled[i] == false) { m_subcounter[i] = 0; m_counter[i] = 0; m_ram[0xfd + i] = 0; } m_timer_enabled[i] = BIT(data, i); //m_timer[i]->enable(m_timer_enabled[i]); } if (BIT(data, 4)) { m_port_in[0] = 0; m_port_in[1] = 0; } if (BIT(data, 5)) { m_port_in[2] = 0; m_port_in[3] = 0; } /* bit 7 = IPL ROM enable */ break; case 0x2: /* Register address */ break; case 0x3: /* Register data - 0x80-0xff is a read-only mirror of 0x00-0x7f */ if (!(m_ram[0xf2] & 0x80)) dsp_io_w(m_ram[0xf2] & 0x7f, data); break; case 0x4: /* Port 0 */ case 0x5: /* Port 1 */ case 0x6: /* Port 2 */ case 0x7: /* Port 3 */ // osd_printf_debug("%s SPC: %02x to APU @ %d\n", machine().describe_context(), data, offset & 3); m_port_out[offset - 4] = data; // Unneeded, we already run at perfect_interleave // machine().scheduler().boost_interleave(attotime::zero, attotime::from_usec(20)); break; case 0xa: /* Timer 0 */ case 0xb: /* Timer 1 */ case 0xc: /* Timer 2 */ // if 0 then TnDiv is divided by 256, otherwise it's divided by 1 to 255 if (data == 0) m_TnDIV[offset - 0xa] = 256; else m_TnDIV[offset - 0xa] = data; break; case 0xd: /* Counter 0 */ case 0xe: /* Counter 1 */ case 0xf: /* Counter 2 */ return; } m_ram[0xf0 + offset] = data; } u8 snes_sound_device::spc_ram_r(offs_t offset) { /* IPL ROM enabled */ if (offset >= 0xffc0 && m_ram[0xf1] & 0x80) return m_ipl_region[offset & 0x3f]; return m_ram[offset]; } void snes_sound_device::spc_ram_w(offs_t offset, u8 data) { m_ram[offset] = data; } u8 snes_sound_device::spc_port_out(offs_t offset) { assert(offset < 4); return m_port_out[offset]; } void snes_sound_device::spc_port_in(offs_t offset, u8 data) { assert(offset < 4); m_port_in[offset] = data; } /***************************************************************************** DEVICE INTERFACE *****************************************************************************/ void snes_sound_device::state_register() { save_item(NAME(m_dsp_regs)); save_item(NAME(m_ipl_region)); 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 save_item(NAME(m_timer_enabled)); save_item(NAME(m_subcounter)); save_item(NAME(m_counter)); save_item(NAME(m_port_in)); save_item(NAME(m_port_out)); save_item(NAME(m_TnDIV)); 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 snes_sound_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]; } }