diff options
Diffstat (limited to 'plugins/gme/game-music-emu-0.5.5/gme/Spc_Dsp.cpp')
| -rw-r--r-- | plugins/gme/game-music-emu-0.5.5/gme/Spc_Dsp.cpp | 666 |
1 files changed, 0 insertions, 666 deletions
diff --git a/plugins/gme/game-music-emu-0.5.5/gme/Spc_Dsp.cpp b/plugins/gme/game-music-emu-0.5.5/gme/Spc_Dsp.cpp deleted file mode 100644 index 814ff2bd..00000000 --- a/plugins/gme/game-music-emu-0.5.5/gme/Spc_Dsp.cpp +++ /dev/null @@ -1,666 +0,0 @@ -// Game_Music_Emu 0.5.5. 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 ); - - assert( offsetof (globals_t,unused9 [2]) == register_count ); - assert( 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, -}; 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