// Game_Music_Emu https://bitbucket.org/mpyne/game-music-emu/ #include "Spc_Dsp.h" #include "blargg_endian.h" #include <string.h> /* Copyright (C) 2007 Shay Green. This module 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 2.1 of the License, or (at your option) any later version. This module 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. You should have received a copy of the GNU Lesser General Public License along with this module; if not, write to the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "blargg_source.h" #ifdef BLARGG_ENABLE_OPTIMIZER #include BLARGG_ENABLE_OPTIMIZER #endif #if INT_MAX < 0x7FFFFFFF #error "Requires that int type have at least 32 bits" #endif // TODO: add to blargg_endian.h #define GET_LE16SA( addr ) ((int16_t) GET_LE16( addr )) #define GET_LE16A( addr ) GET_LE16( addr ) #define SET_LE16A( addr, data ) SET_LE16( addr, data ) static uint8_t const initial_regs [Spc_Dsp::register_count] = { 0x45,0x8B,0x5A,0x9A,0xE4,0x82,0x1B,0x78,0x00,0x00,0xAA,0x96,0x89,0x0E,0xE0,0x80, 0x2A,0x49,0x3D,0xBA,0x14,0xA0,0xAC,0xC5,0x00,0x00,0x51,0xBB,0x9C,0x4E,0x7B,0xFF, 0xF4,0xFD,0x57,0x32,0x37,0xD9,0x42,0x22,0x00,0x00,0x5B,0x3C,0x9F,0x1B,0x87,0x9A, 0x6F,0x27,0xAF,0x7B,0xE5,0x68,0x0A,0xD9,0x00,0x00,0x9A,0xC5,0x9C,0x4E,0x7B,0xFF, 0xEA,0x21,0x78,0x4F,0xDD,0xED,0x24,0x14,0x00,0x00,0x77,0xB1,0xD1,0x36,0xC1,0x67, 0x52,0x57,0x46,0x3D,0x59,0xF4,0x87,0xA4,0x00,0x00,0x7E,0x44,0x9C,0x4E,0x7B,0xFF, 0x75,0xF5,0x06,0x97,0x10,0xC3,0x24,0xBB,0x00,0x00,0x7B,0x7A,0xE0,0x60,0x12,0x0F, 0xF7,0x74,0x1C,0xE5,0x39,0x3D,0x73,0xC1,0x00,0x00,0x7A,0xB3,0xFF,0x4E,0x7B,0xFF }; // if ( io < -32768 ) io = -32768; // if ( io > 32767 ) io = 32767; #define CLAMP16( io )\ {\ if ( (int16_t) io != io )\ io = (io >> 31) ^ 0x7FFF;\ } // Access global DSP register #define REG(n) m.regs [r_##n] // Access voice DSP register #define VREG(r,n) r [v_##n] #define WRITE_SAMPLES( l, r, out ) \ {\ out [0] = l;\ out [1] = r;\ out += 2;\ if ( out >= m.out_end )\ {\ check( out == m.out_end );\ check( m.out_end != &m.extra [extra_size] || \ (m.extra <= m.out_begin && m.extra < &m.extra [extra_size]) );\ out = m.extra;\ m.out_end = &m.extra [extra_size];\ }\ }\ void Spc_Dsp::set_output( sample_t* out, int size ) { require( (size & 1) == 0 ); // must be even if ( !out ) { out = m.extra; size = extra_size; } m.out_begin = out; m.out = out; m.out_end = out + size; } // Volume registers and efb are signed! Easy to forget int8_t cast. // Prefixes are to avoid accidental use of locals with same names. // Interleved gauss table (to improve cache coherency) // interleved_gauss [i] = gauss [(i & 1) * 256 + 255 - (i >> 1 & 0xFF)] static short const interleved_gauss [512] = { 370,1305, 366,1305, 362,1304, 358,1304, 354,1304, 351,1304, 347,1304, 343,1303, 339,1303, 336,1303, 332,1302, 328,1302, 325,1301, 321,1300, 318,1300, 314,1299, 311,1298, 307,1297, 304,1297, 300,1296, 297,1295, 293,1294, 290,1293, 286,1292, 283,1291, 280,1290, 276,1288, 273,1287, 270,1286, 267,1284, 263,1283, 260,1282, 257,1280, 254,1279, 251,1277, 248,1275, 245,1274, 242,1272, 239,1270, 236,1269, 233,1267, 230,1265, 227,1263, 224,1261, 221,1259, 218,1257, 215,1255, 212,1253, 210,1251, 207,1248, 204,1246, 201,1244, 199,1241, 196,1239, 193,1237, 191,1234, 188,1232, 186,1229, 183,1227, 180,1224, 178,1221, 175,1219, 173,1216, 171,1213, 168,1210, 166,1207, 163,1205, 161,1202, 159,1199, 156,1196, 154,1193, 152,1190, 150,1186, 147,1183, 145,1180, 143,1177, 141,1174, 139,1170, 137,1167, 134,1164, 132,1160, 130,1157, 128,1153, 126,1150, 124,1146, 122,1143, 120,1139, 118,1136, 117,1132, 115,1128, 113,1125, 111,1121, 109,1117, 107,1113, 106,1109, 104,1106, 102,1102, 100,1098, 99,1094, 97,1090, 95,1086, 94,1082, 92,1078, 90,1074, 89,1070, 87,1066, 86,1061, 84,1057, 83,1053, 81,1049, 80,1045, 78,1040, 77,1036, 76,1032, 74,1027, 73,1023, 71,1019, 70,1014, 69,1010, 67,1005, 66,1001, 65, 997, 64, 992, 62, 988, 61, 983, 60, 978, 59, 974, 58, 969, 56, 965, 55, 960, 54, 955, 53, 951, 52, 946, 51, 941, 50, 937, 49, 932, 48, 927, 47, 923, 46, 918, 45, 913, 44, 908, 43, 904, 42, 899, 41, 894, 40, 889, 39, 884, 38, 880, 37, 875, 36, 870, 36, 865, 35, 860, 34, 855, 33, 851, 32, 846, 32, 841, 31, 836, 30, 831, 29, 826, 29, 821, 28, 816, 27, 811, 27, 806, 26, 802, 25, 797, 24, 792, 24, 787, 23, 782, 23, 777, 22, 772, 21, 767, 21, 762, 20, 757, 20, 752, 19, 747, 19, 742, 18, 737, 17, 732, 17, 728, 16, 723, 16, 718, 15, 713, 15, 708, 15, 703, 14, 698, 14, 693, 13, 688, 13, 683, 12, 678, 12, 674, 11, 669, 11, 664, 11, 659, 10, 654, 10, 649, 10, 644, 9, 640, 9, 635, 9, 630, 8, 625, 8, 620, 8, 615, 7, 611, 7, 606, 7, 601, 6, 596, 6, 592, 6, 587, 6, 582, 5, 577, 5, 573, 5, 568, 5, 563, 4, 559, 4, 554, 4, 550, 4, 545, 4, 540, 3, 536, 3, 531, 3, 527, 3, 522, 3, 517, 2, 513, 2, 508, 2, 504, 2, 499, 2, 495, 2, 491, 2, 486, 1, 482, 1, 477, 1, 473, 1, 469, 1, 464, 1, 460, 1, 456, 1, 451, 1, 447, 1, 443, 1, 439, 0, 434, 0, 430, 0, 426, 0, 422, 0, 418, 0, 414, 0, 410, 0, 405, 0, 401, 0, 397, 0, 393, 0, 389, 0, 385, 0, 381, 0, 378, 0, 374, }; //// Counters #define RATE( rate, div )\ (rate >= div ? rate / div * 8 - 1 : rate - 1) static unsigned const counter_mask [32] = { RATE( 2,2), RATE(2048,4), RATE(1536,3), RATE(1280,5), RATE(1024,4), RATE( 768,3), RATE( 640,5), RATE( 512,4), RATE( 384,3), RATE( 320,5), RATE( 256,4), RATE( 192,3), RATE( 160,5), RATE( 128,4), RATE( 96,3), RATE( 80,5), RATE( 64,4), RATE( 48,3), RATE( 40,5), RATE( 32,4), RATE( 24,3), RATE( 20,5), RATE( 16,4), RATE( 12,3), RATE( 10,5), RATE( 8,4), RATE( 6,3), RATE( 5,5), RATE( 4,4), RATE( 3,3), RATE( 2,4), RATE( 1,4) }; #undef RATE inline void Spc_Dsp::init_counter() { // counters start out with this synchronization m.counters [0] = 1; m.counters [1] = 0; m.counters [2] = -0x20u; m.counters [3] = 0x0B; int n = 2; for ( int i = 1; i < 32; i++ ) { m.counter_select [i] = &m.counters [n]; if ( !--n ) n = 3; } m.counter_select [ 0] = &m.counters [0]; m.counter_select [30] = &m.counters [2]; } inline void Spc_Dsp::run_counter( int i ) { int n = m.counters [i]; if ( !(n-- & 7) ) n -= 6 - i; m.counters [i] = n; } #define READ_COUNTER( rate )\ (*m.counter_select [rate] & counter_mask [rate]) //// Emulation void Spc_Dsp::run( int clock_count ) { int new_phase = m.phase + clock_count; int count = new_phase >> 5; m.phase = new_phase & 31; if ( !count ) return; uint8_t* const ram = m.ram; uint8_t const* const dir = &ram [REG(dir) * 0x100]; int const slow_gaussian = (REG(pmon) >> 1) | REG(non); int const noise_rate = REG(flg) & 0x1F; // Global volume int mvoll = (int8_t) REG(mvoll); int mvolr = (int8_t) REG(mvolr); if ( mvoll * mvolr < m.surround_threshold ) mvoll = -mvoll; // eliminate surround do { // KON/KOFF reading if ( (m.every_other_sample ^= 1) != 0 ) { m.new_kon &= ~m.kon; m.kon = m.new_kon; m.t_koff = REG(koff); } run_counter( 1 ); run_counter( 2 ); run_counter( 3 ); // Noise if ( !READ_COUNTER( noise_rate ) ) { int feedback = (m.noise << 13) ^ (m.noise << 14); m.noise = (feedback & 0x4000) ^ (m.noise >> 1); } // Voices int pmon_input = 0; int main_out_l = 0; int main_out_r = 0; int echo_out_l = 0; int echo_out_r = 0; voice_t* v = m.voices; uint8_t* v_regs = m.regs; int vbit = 1; do { #define SAMPLE_PTR(i) GET_LE16A( &dir [VREG(v_regs,srcn) * 4 + i * 2] ) int brr_header = ram [v->brr_addr]; int kon_delay = v->kon_delay; // Pitch int pitch = GET_LE16A( &VREG(v_regs,pitchl) ) & 0x3FFF; if ( REG(pmon) & vbit ) pitch += ((pmon_input >> 5) * pitch) >> 10; // KON phases if ( --kon_delay >= 0 ) { v->kon_delay = kon_delay; // Get ready to start BRR decoding on next sample if ( kon_delay == 4 ) { v->brr_addr = SAMPLE_PTR( 0 ); v->brr_offset = 1; v->buf_pos = v->buf; brr_header = 0; // header is ignored on this sample } // Envelope is never run during KON v->env = 0; v->hidden_env = 0; // Disable BRR decoding until last three samples v->interp_pos = (kon_delay & 3 ? 0x4000 : 0); // Pitch is never added during KON pitch = 0; } int env = v->env; // Gaussian interpolation { int output = 0; VREG(v_regs,envx) = (uint8_t) (env >> 4); if ( env ) { // Make pointers into gaussian based on fractional position between samples int offset = (unsigned) v->interp_pos >> 3 & 0x1FE; short const* fwd = interleved_gauss + offset; short const* rev = interleved_gauss + 510 - offset; // mirror left half of gaussian int const* in = &v->buf_pos [(unsigned) v->interp_pos >> 12]; if ( !(slow_gaussian & vbit) ) // 99% { // Faster approximation when exact sample value isn't necessary for pitch mod output = (fwd [0] * in [0] + fwd [1] * in [1] + rev [1] * in [2] + rev [0] * in [3]) >> 11; output = (output * env) >> 11; } else { output = (int16_t) (m.noise * 2); if ( !(REG(non) & vbit) ) { output = (fwd [0] * in [0]) >> 11; output += (fwd [1] * in [1]) >> 11; output += (rev [1] * in [2]) >> 11; output = (int16_t) output; output += (rev [0] * in [3]) >> 11; CLAMP16( output ); output &= ~1; } output = (output * env) >> 11 & ~1; } // Output int l = output * v->volume [0]; int r = output * v->volume [1]; main_out_l += l; main_out_r += r; if ( REG(eon) & vbit ) { echo_out_l += l; echo_out_r += r; } } pmon_input = output; VREG(v_regs,outx) = (uint8_t) (output >> 8); } // Soft reset or end of sample if ( REG(flg) & 0x80 || (brr_header & 3) == 1 ) { v->env_mode = env_release; env = 0; } if ( m.every_other_sample ) { // KOFF if ( m.t_koff & vbit ) v->env_mode = env_release; // KON if ( m.kon & vbit ) { v->kon_delay = 5; v->env_mode = env_attack; REG(endx) &= ~vbit; } } // Envelope if ( !v->kon_delay ) { if ( v->env_mode == env_release ) // 97% { env -= 0x8; v->env = env; if ( env <= 0 ) { v->env = 0; goto skip_brr; // no BRR decoding for you! } } else // 3% { int rate; int const adsr0 = VREG(v_regs,adsr0); int env_data = VREG(v_regs,adsr1); if ( adsr0 >= 0x80 ) // 97% ADSR { if ( v->env_mode > env_decay ) // 89% { env--; env -= env >> 8; rate = env_data & 0x1F; // optimized handling v->hidden_env = env; if ( READ_COUNTER( rate ) ) goto exit_env; v->env = env; goto exit_env; } else if ( v->env_mode == env_decay ) { env--; env -= env >> 8; rate = (adsr0 >> 3 & 0x0E) + 0x10; } else // env_attack { rate = (adsr0 & 0x0F) * 2 + 1; env += rate < 31 ? 0x20 : 0x400; } } else // GAIN { int mode; env_data = VREG(v_regs,gain); mode = env_data >> 5; if ( mode < 4 ) // direct { env = env_data * 0x10; rate = 31; } else { rate = env_data & 0x1F; if ( mode == 4 ) // 4: linear decrease { env -= 0x20; } else if ( mode < 6 ) // 5: exponential decrease { env--; env -= env >> 8; } else // 6,7: linear increase { env += 0x20; if ( mode > 6 && (unsigned) v->hidden_env >= 0x600 ) env += 0x8 - 0x20; // 7: two-slope linear increase } } } // Sustain level if ( (env >> 8) == (env_data >> 5) && v->env_mode == env_decay ) v->env_mode = env_sustain; v->hidden_env = env; // unsigned cast because linear decrease going negative also triggers this if ( (unsigned) env > 0x7FF ) { env = (env < 0 ? 0 : 0x7FF); if ( v->env_mode == env_attack ) v->env_mode = env_decay; } if ( !READ_COUNTER( rate ) ) v->env = env; // nothing else is controlled by the counter } } exit_env: { // Apply pitch int old_pos = v->interp_pos; int interp_pos = (old_pos & 0x3FFF) + pitch; if ( interp_pos > 0x7FFF ) interp_pos = 0x7FFF; v->interp_pos = interp_pos; // BRR decode if necessary if ( old_pos >= 0x4000 ) { // Arrange the four input nybbles in 0xABCD order for easy decoding int nybbles = ram [(v->brr_addr + v->brr_offset) & 0xFFFF] * 0x100 + ram [(v->brr_addr + v->brr_offset + 1) & 0xFFFF]; // Advance read position int const brr_block_size = 9; int brr_offset = v->brr_offset; if ( (brr_offset += 2) >= brr_block_size ) { // Next BRR block int brr_addr = (v->brr_addr + brr_block_size) & 0xFFFF; assert( brr_offset == brr_block_size ); if ( brr_header & 1 ) { brr_addr = SAMPLE_PTR( 1 ); if ( !v->kon_delay ) REG(endx) |= vbit; } v->brr_addr = brr_addr; brr_offset = 1; } v->brr_offset = brr_offset; // Decode // 0: >>1 1: <<0 2: <<1 ... 12: <<11 13-15: >>4 <<11 static unsigned char const shifts [16 * 2] = { 13,12,12,12,12,12,12,12,12,12,12, 12, 12, 16, 16, 16, 0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11 }; int const scale = brr_header >> 4; int const right_shift = shifts [scale]; int const left_shift = shifts [scale + 16]; // Write to next four samples in circular buffer int* pos = v->buf_pos; int* end; // Decode four samples for ( end = pos + 4; pos < end; pos++, nybbles <<= 4 ) { // Extract upper nybble and scale appropriately. Every cast is // necessary to maintain correctness and avoid undef behavior int s = int16_t(uint16_t((int16_t) nybbles >> right_shift) << left_shift); // Apply IIR filter (8 is the most commonly used) int const filter = brr_header & 0x0C; int const p1 = pos [brr_buf_size - 1]; int const p2 = pos [brr_buf_size - 2] >> 1; if ( filter >= 8 ) { s += p1; s -= p2; if ( filter == 8 ) // s += p1 * 0.953125 - p2 * 0.46875 { s += p2 >> 4; s += (p1 * -3) >> 6; } else // s += p1 * 0.8984375 - p2 * 0.40625 { s += (p1 * -13) >> 7; s += (p2 * 3) >> 4; } } else if ( filter ) // s += p1 * 0.46875 { s += p1 >> 1; s += (-p1) >> 5; } // Adjust and write sample CLAMP16( s ); s = (int16_t) (s * 2); pos [brr_buf_size] = pos [0] = s; // second copy simplifies wrap-around } if ( pos >= &v->buf [brr_buf_size] ) pos = v->buf; v->buf_pos = pos; } } skip_brr: // Next voice vbit <<= 1; v_regs += 0x10; v++; } while ( vbit < 0x100 ); // Echo position int echo_offset = m.echo_offset; uint8_t* const echo_ptr = &ram [(REG(esa) * 0x100 + echo_offset) & 0xFFFF]; if ( !echo_offset ) m.echo_length = (REG(edl) & 0x0F) * 0x800; echo_offset += 4; if ( echo_offset >= m.echo_length ) echo_offset = 0; m.echo_offset = echo_offset; // FIR int echo_in_l = GET_LE16SA( echo_ptr + 0 ); int echo_in_r = GET_LE16SA( echo_ptr + 2 ); int (*echo_hist_pos) [2] = m.echo_hist_pos; if ( ++echo_hist_pos >= &m.echo_hist [echo_hist_size] ) echo_hist_pos = m.echo_hist; m.echo_hist_pos = echo_hist_pos; echo_hist_pos [0] [0] = echo_hist_pos [8] [0] = echo_in_l; echo_hist_pos [0] [1] = echo_hist_pos [8] [1] = echo_in_r; #define CALC_FIR_( i, in ) ((in) * (int8_t) REG(fir + i * 0x10)) echo_in_l = CALC_FIR_( 7, echo_in_l ); echo_in_r = CALC_FIR_( 7, echo_in_r ); #define CALC_FIR( i, ch ) CALC_FIR_( i, echo_hist_pos [i + 1] [ch] ) #define DO_FIR( i )\ echo_in_l += CALC_FIR( i, 0 );\ echo_in_r += CALC_FIR( i, 1 ); DO_FIR( 0 ); DO_FIR( 1 ); DO_FIR( 2 ); #if defined (__MWERKS__) && __MWERKS__ < 0x3200 __eieio(); // keeps compiler from stupidly "caching" things in memory #endif DO_FIR( 3 ); DO_FIR( 4 ); DO_FIR( 5 ); DO_FIR( 6 ); // Echo out if ( !(REG(flg) & 0x20) ) { int l = (echo_out_l >> 7) + ((echo_in_l * (int8_t) REG(efb)) >> 14); int r = (echo_out_r >> 7) + ((echo_in_r * (int8_t) REG(efb)) >> 14); // just to help pass more validation tests #if SPC_MORE_ACCURACY l &= ~1; r &= ~1; #endif CLAMP16( l ); CLAMP16( r ); SET_LE16A( echo_ptr + 0, l ); SET_LE16A( echo_ptr + 2, r ); } // Sound out int l = (main_out_l * mvoll + echo_in_l * (int8_t) REG(evoll)) >> 14; int r = (main_out_r * mvolr + echo_in_r * (int8_t) REG(evolr)) >> 14; CLAMP16( l ); CLAMP16( r ); if ( (REG(flg) & 0x40) ) { l = 0; r = 0; } sample_t* out = m.out; WRITE_SAMPLES( l, r, out ); m.out = out; } while ( --count ); } //// Setup void Spc_Dsp::mute_voices( int mask ) { m.mute_mask = mask; for ( int i = 0; i < voice_count; i++ ) { m.voices [i].enabled = (mask >> i & 1) - 1; update_voice_vol( i * 0x10 ); } } void Spc_Dsp::init( void* ram_64k ) { m.ram = (uint8_t*) ram_64k; mute_voices( 0 ); disable_surround( false ); set_output( 0, 0 ); reset(); #ifndef NDEBUG // be sure this sign-extends assert( (int16_t) 0x8000 == -0x8000 ); // be sure right shift preserves sign assert( (-1 >> 1) == -1 ); // check clamp macro int i; i = +0x8000; CLAMP16( i ); assert( i == +0x7FFF ); i = -0x8001; CLAMP16( i ); assert( i == -0x8000 ); blargg_verify_byte_order(); #endif } void Spc_Dsp::soft_reset_common() { require( m.ram ); // init() must have been called already m.noise = 0x4000; m.echo_hist_pos = m.echo_hist; m.every_other_sample = 1; m.echo_offset = 0; m.phase = 0; init_counter(); } void Spc_Dsp::soft_reset() { REG(flg) = 0xE0; soft_reset_common(); } void Spc_Dsp::load( uint8_t const regs [register_count] ) { memcpy( m.regs, regs, sizeof m.regs ); memset( &m.regs [register_count], 0, offsetof (state_t,ram) - register_count ); // Internal state int i; for ( i = voice_count; --i >= 0; ) { voice_t& v = m.voices [i]; v.brr_offset = 1; v.buf_pos = v.buf; } m.new_kon = REG(kon); mute_voices( m.mute_mask ); soft_reset_common(); } void Spc_Dsp::reset() { load( initial_regs ); }