Mercurial > audlegacy-plugins
view src/console/Spc_Dsp.cxx @ 316:fb513e10174e trunk
[svn] - merge libconsole-blargg into mainline libconsole:
+ obsoletes plugins-ugly:sapplug
| author | nenolod |
|---|---|
| date | Thu, 30 Nov 2006 19:54:33 -0800 |
| parents | 3da1b8942b8b |
| children | 986f098da058 |
line wrap: on
line source
// Game_Music_Emu 0.5.1. http://www.slack.net/~ant/ // Based on Brad Martin's OpenSPC DSP emulator #include "Spc_Dsp.h" #include "blargg_endian.h" #include <string.h> /* Copyright (C) 2002 Brad Martin */ /* Copyright (C) 2004-2006 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 Spc_Dsp::Spc_Dsp( uint8_t* ram_ ) : ram( ram_ ) { set_gain( 1.0 ); mute_voices( 0 ); disable_surround( false ); BOOST_STATIC_ASSERT( sizeof (g) == register_count && sizeof (voice) == register_count ); blargg_verify_byte_order(); } void Spc_Dsp::mute_voices( int mask ) { for ( int i = 0; i < voice_count; i++ ) voice_state [i].enabled = (mask >> i & 1) ? 31 : 7; } void Spc_Dsp::reset() { keys = 0; echo_ptr = 0; noise_count = 0; noise = 1; fir_offset = 0; g.flags = 0xE0; // reset, mute, echo off g.key_ons = 0; for ( int i = 0; i < voice_count; i++ ) { voice_t& v = voice_state [i]; v.on_cnt = 0; v.volume [0] = 0; v.volume [1] = 0; v.envstate = state_release; } memset( fir_buf, 0, sizeof fir_buf ); } void Spc_Dsp::write( int i, int data ) { require( (unsigned) i < register_count ); reg [i] = data; int high = i >> 4; switch ( i & 0x0F ) { // voice volume case 0: case 1: { short* volume = voice_state [high].volume; int left = (int8_t) reg [i & ~1]; int right = (int8_t) reg [i | 1]; volume [0] = left; volume [1] = right; // kill surround only if enabled and signs of volumes differ if ( left * right < surround_threshold ) { if ( left < 0 ) volume [0] = -left; else volume [1] = -right; } break; } // fir coefficients case 0x0F: fir_coeff [high] = (int8_t) data; // sign-extend break; } } // 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). Each count divides exactly into // 0x7800 without remainder. const int env_rate_init = 0x7800; static short const env_rates [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 }; const int env_range = 0x800; inline int Spc_Dsp::clock_envelope( int v ) { /* Return value is current * ENVX */ raw_voice_t& raw_voice = this->voice [v]; voice_t& voice = voice_state [v]; int envx = voice.envx; if ( voice.envstate == state_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 -= env_range / 256; if ( envx <= 0 ) { envx = 0; keys &= ~(1 << v); return -1; } voice.envx = envx; raw_voice.envx = envx >> 8; return envx; } int cnt = voice.envcnt; int adsr1 = raw_voice.adsr [0]; if ( adsr1 & 0x80 ) { switch ( voice.envstate ) { case state_attack: { // increase envelope by 1/64 each step int t = adsr1 & 15; if ( t == 15 ) { envx += env_range / 2; } else { cnt -= env_rates [t * 2 + 1]; if ( cnt > 0 ) break; envx += env_range / 64; cnt = env_rate_init; } if ( envx >= env_range ) { envx = env_range - 1; voice.envstate = state_decay; } voice.envx = envx; break; } case state_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 -= env_rates [((adsr1 >> 3) & 0xE) + 0x10]; if ( cnt <= 0 ) { cnt = env_rate_init; envx -= ((envx - 1) >> 8) + 1; voice.envx = envx; } int sustain_level = raw_voice.adsr [1] >> 5; if ( envx <= (sustain_level + 1) * 0x100 ) voice.envstate = state_sustain; break; } case state_sustain: // Docs: "SR [is multiplied] by the fixed value 1-1/256." // Multiplying ENVX by 255/256 every time SUSTAIN is // updated. cnt -= env_rates [raw_voice.adsr [1] & 0x1F]; if ( cnt <= 0 ) { cnt = env_rate_init; envx -= ((envx - 1) >> 8) + 1; voice.envx = envx; } break; case state_release: // handled above 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. */ int t = raw_voice.gain; if (t < 0x80) { envx = voice.envx = t << 4; } else switch (t >> 5) { case 4: /* Docs: "Decrease (linear): Subtraction * of the fixed value 1/64." */ cnt -= env_rates [t & 0x1F]; if (cnt > 0) break; cnt = env_rate_init; envx -= env_range / 64; if ( envx < 0 ) { envx = 0; if ( voice.envstate == state_attack ) voice.envstate = state_decay; } voice.envx = envx; break; case 5: /* Docs: "Drecrease <sic> (exponential): * Multiplication by the fixed value * 1-1/256." */ cnt -= env_rates [t & 0x1F]; if (cnt > 0) break; cnt = env_rate_init; envx -= ((envx - 1) >> 8) + 1; if ( envx < 0 ) { envx = 0; if ( voice.envstate == state_attack ) voice.envstate = state_decay; } voice.envx = envx; break; case 6: /* Docs: "Increase (linear): Addition of * the fixed value 1/64." */ cnt -= env_rates [t & 0x1F]; if (cnt > 0) break; cnt = env_rate_init; envx += env_range / 64; if ( envx >= env_range ) envx = env_range - 1; voice.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 -= env_rates [t & 0x1F]; if (cnt > 0) break; cnt = env_rate_init; if ( envx < env_range * 3 / 4 ) envx += env_range / 64; else envx += env_range / 256; if ( envx >= env_range ) envx = env_range - 1; voice.envx = envx; break; } } voice.envcnt = cnt; raw_voice.envx = envx >> 4; return envx; } // Clamp n into range -32768 <= n <= 32767 inline int clamp_16( int n ) { if ( (BOOST::int16_t) n != n ) n = BOOST::int16_t (0x7FFF - (n >> 31)); return n; } void Spc_Dsp::run( long count, short* out_buf ) { // to do: make clock_envelope() inline so that this becomes a leaf function? // Should we just fill the buffer with silence? Flags won't be cleared // during this run so it seems it should keep resetting every sample. if ( g.flags & 0x80 ) reset(); struct src_dir { char start [2]; char loop [2]; }; const src_dir* const sd = (src_dir*) &ram [g.wave_page * 0x100]; int left_volume = g.left_volume; int right_volume = g.right_volume; if ( left_volume * right_volume < surround_threshold ) right_volume = -right_volume; // kill global surround left_volume *= emu_gain; right_volume *= emu_gain; while ( --count >= 0 ) { // 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. g.wave_ended &= ~g.key_ons; // Keying on a voice resets that bit in ENDX. if ( g.noise_enables ) { noise_count -= env_rates [g.flags & 0x1F]; if ( noise_count <= 0 ) { noise_count = env_rate_init; noise_amp = BOOST::int16_t (noise * 2); // TODO: switch to Galios style int feedback = (noise << 13) ^ (noise << 14); noise = (feedback & 0x4000) | (noise >> 1); } } // What is the expected behavior when pitch modulation is enabled on // voice 0? Jurassic Park 2 does this. Assume 0 for now. blargg_long prev_outx = 0; int echol = 0; int echor = 0; int left = 0; int right = 0; for ( int vidx = 0; vidx < voice_count; vidx++ ) { const int vbit = 1 << vidx; raw_voice_t& raw_voice = voice [vidx]; voice_t& voice = voice_state [vidx]; if ( voice.on_cnt && !--voice.on_cnt ) { // key on keys |= vbit; voice.addr = GET_LE16( sd [raw_voice.waveform].start ); voice.block_remain = 1; voice.envx = 0; voice.block_header = 0; voice.fraction = 0x3FFF; // decode three samples immediately voice.interp0 = 0; // BRR decoder filter uses previous two samples voice.interp1 = 0; // 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. voice.envcnt = env_rate_init; voice.envstate = state_attack; } if ( g.key_ons & vbit & ~g.key_offs ) { // voice doesn't come on if key off is set g.key_ons &= ~vbit; voice.on_cnt = 8; } if ( keys & g.key_offs & vbit ) { // key off voice.envstate = state_release; voice.on_cnt = 0; } int envx; if ( !(keys & vbit) || (envx = clock_envelope( vidx )) < 0 ) { raw_voice.envx = 0; raw_voice.outx = 0; prev_outx = 0; continue; } // Decode samples when fraction >= 1.0 (0x1000) for ( int n = voice.fraction >> 12; --n >= 0; ) { if ( !--voice.block_remain ) { if ( voice.block_header & 1 ) { g.wave_ended |= vbit; if ( voice.block_header & 2 ) { // verified (played endless looping sample and ENDX was set) voice.addr = GET_LE16( sd [raw_voice.waveform].loop ); } else { // first block was end block; don't play anything (verified) goto sample_ended; // to do: find alternative to goto } } voice.block_header = ram [voice.addr++]; voice.block_remain = 16; // nybbles } // if next block has end flag set, *this* block ends *early* (verified) if ( voice.block_remain == 9 && (ram [voice.addr + 5] & 3) == 1 && (voice.block_header & 3) != 3 ) { sample_ended: g.wave_ended |= vbit; keys &= ~vbit; raw_voice.envx = 0; voice.envx = 0; // add silence samples to interpolation buffer do { voice.interp3 = voice.interp2; voice.interp2 = voice.interp1; voice.interp1 = voice.interp0; voice.interp0 = 0; } while ( --n >= 0 ); break; } int delta = ram [voice.addr]; if ( voice.block_remain & 1 ) { delta <<= 4; // use lower nybble voice.addr++; } // Use sign-extended upper nybble delta = int8_t (delta) >> 4; // 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. int shift = voice.block_header >> 4; delta = (delta << shift) >> 1; if ( shift > 0x0C ) delta = (delta >> 14) & ~0x7FF; // One, two and three point IIR filters int smp1 = voice.interp0; int smp2 = voice.interp1; if ( voice.block_header & 8 ) { delta += smp1; delta -= smp2 >> 1; if ( !(voice.block_header & 4) ) { delta += (-smp1 - (smp1 >> 1)) >> 5; delta += smp2 >> 5; } else { delta += (-smp1 * 13) >> 7; delta += (smp2 + (smp2 >> 1)) >> 4; } } else if ( voice.block_header & 4 ) { delta += smp1 >> 1; delta += (-smp1) >> 5; } voice.interp3 = voice.interp2; voice.interp2 = smp2; voice.interp1 = smp1; voice.interp0 = BOOST::int16_t (clamp_16( delta ) * 2); // sign-extend } // rate (with possible modulation) int rate = GET_LE16( raw_voice.rate ) & 0x3FFF; if ( g.pitch_mods & vbit ) rate = (rate * (prev_outx + 32768)) >> 15; // Gaussian interpolation using most recent 4 samples int index = voice.fraction >> 2 & 0x3FC; voice.fraction = (voice.fraction & 0x0FFF) + rate; const BOOST::int16_t* table = (BOOST::int16_t const*) ((char const*) gauss + index); const BOOST::int16_t* table2 = (BOOST::int16_t const*) ((char const*) gauss + (255*4 - index)); int s = ((table [0] * voice.interp3) >> 12) + ((table [1] * voice.interp2) >> 12) + ((table2 [1] * voice.interp1) >> 12); s = (BOOST::int16_t) (s * 2); s += (table2 [0] * voice.interp0) >> 11 & ~1; int output = clamp_16( s ); if ( g.noise_enables & vbit ) output = noise_amp; // scale output and set outx values output = (output * envx) >> 11 & ~1; // output and apply muting (by setting voice.enabled to 31) // if voice is externally disabled (not a SNES feature) int l = (voice.volume [0] * output) >> voice.enabled; int r = (voice.volume [1] * output) >> voice.enabled; prev_outx = output; raw_voice.outx = int8_t (output >> 8); if ( g.echo_ons & vbit ) { echol += l; echor += r; } left += l; right += r; } // end of channel loop // main volume control left = (left * left_volume ) >> (7 + emu_gain_bits); right = (right * right_volume) >> (7 + emu_gain_bits); // Echo FIR filter // read feedback from echo buffer int echo_ptr = this->echo_ptr; uint8_t* echo_buf = &ram [(g.echo_page * 0x100 + echo_ptr) & 0xFFFF]; echo_ptr += 4; if ( echo_ptr >= (g.echo_delay & 15) * 0x800 ) echo_ptr = 0; int fb_left = (BOOST::int16_t) GET_LE16( echo_buf ); // sign-extend int fb_right = (BOOST::int16_t) GET_LE16( echo_buf + 2 ); // sign-extend this->echo_ptr = echo_ptr; // put samples in history ring buffer const int fir_offset = this->fir_offset; short (*fir_pos) [2] = &fir_buf [fir_offset]; this->fir_offset = (fir_offset + 7) & 7; // move backwards one step fir_pos [0] [0] = (short) fb_left; fir_pos [0] [1] = (short) fb_right; fir_pos [8] [0] = (short) fb_left; // duplicate at +8 eliminates wrap checking below fir_pos [8] [1] = (short) fb_right; // FIR fb_left = fb_left * fir_coeff [7] + fir_pos [1] [0] * fir_coeff [6] + fir_pos [2] [0] * fir_coeff [5] + fir_pos [3] [0] * fir_coeff [4] + fir_pos [4] [0] * fir_coeff [3] + fir_pos [5] [0] * fir_coeff [2] + fir_pos [6] [0] * fir_coeff [1] + fir_pos [7] [0] * fir_coeff [0]; fb_right = fb_right * fir_coeff [7] + fir_pos [1] [1] * fir_coeff [6] + fir_pos [2] [1] * fir_coeff [5] + fir_pos [3] [1] * fir_coeff [4] + fir_pos [4] [1] * fir_coeff [3] + fir_pos [5] [1] * fir_coeff [2] + fir_pos [6] [1] * fir_coeff [1] + fir_pos [7] [1] * fir_coeff [0]; left += (fb_left * g.left_echo_volume ) >> 14; right += (fb_right * g.right_echo_volume) >> 14; // echo buffer feedback if ( !(g.flags & 0x20) ) { echol += (fb_left * g.echo_feedback) >> 14; echor += (fb_right * g.echo_feedback) >> 14; SET_LE16( echo_buf , clamp_16( echol ) ); SET_LE16( echo_buf + 2, clamp_16( echor ) ); } if ( out_buf ) { // write final samples left = clamp_16( left ); right = clamp_16( right ); int mute = g.flags & 0x40; out_buf [0] = (short) left; out_buf [1] = (short) right; out_buf += 2; // muting if ( mute ) { out_buf [-2] = 0; out_buf [-1] = 0; } } } } // Base normal_gauss table is almost exactly (with an error of 0 or -1 for each entry): // int normal_gauss [512]; // normal_gauss [i] = exp((i-511)*(i-511)*-9.975e-6)*pow(sin(0.00307096*i),1.7358)*1304.45 // Interleved gauss table (to improve cache coherency). // gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] const BOOST::int16_t Spc_Dsp::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, };