33f3af2b8d
Why? Why not? Cuts a few MHz. Change-Id: Ied5c70b1aedd255cbe5d42b7d3028bbe47aad01d
977 lines
32 KiB
C
977 lines
32 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2007-2008 Michael Sevakis (jhMikeS)
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* Copyright (C) 2006-2007 Adam Gashlin (hcs)
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* Copyright (C) 2004-2007 Shay Green (blargg)
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* Copyright (C) 2002 Brad Martin
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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/* The DSP portion (awe!) */
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#include "codeclib.h"
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#include "spc_codec.h"
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#include "spc_profiler.h"
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#define CLAMP16( n ) clip_sample_16( n )
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#if defined(CPU_ARM)
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#if ARM_ARCH >= 6
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#include "cpu/spc_dsp_armv6.c"
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#elif ARM_ARCH >= 5
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#include "cpu/spc_dsp_armv5.c"
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#else
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#include "cpu/spc_dsp_armv4.c"
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#endif
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#elif defined (CPU_COLDFIRE)
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#include "cpu/spc_dsp_coldfire.c"
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#endif
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/* Above may still use generic implementations. Also defines final
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function names. */
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#include "spc_dsp_generic.c"
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/* each rate divides exactly into 0x7800 without remainder */
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static unsigned short const env_rates [0x20] ICONST_ATTR_SPC =
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{
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0x0000, 0x000F, 0x0014, 0x0018, 0x001E, 0x0028, 0x0030, 0x003C,
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0x0050, 0x0060, 0x0078, 0x00A0, 0x00C0, 0x00F0, 0x0140, 0x0180,
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0x01E0, 0x0280, 0x0300, 0x03C0, 0x0500, 0x0600, 0x0780, 0x0A00,
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0x0C00, 0x0F00, 0x1400, 0x1800, 0x1E00, 0x2800, 0x3C00, 0x7800
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};
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#if !SPC_NOINTERP
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/* Interleved gauss table (to improve cache coherency). */
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/* gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] */
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static int16_t gauss_table [512] IDATA_ATTR_SPC MEM_ALIGN_ATTR =
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{
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370,1305, 366,1305, 362,1304, 358,1304,
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354,1304, 351,1304, 347,1304, 343,1303,
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339,1303, 336,1303, 332,1302, 328,1302,
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325,1301, 321,1300, 318,1300, 314,1299,
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311,1298, 307,1297, 304,1297, 300,1296,
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297,1295, 293,1294, 290,1293, 286,1292,
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283,1291, 280,1290, 276,1288, 273,1287,
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270,1286, 267,1284, 263,1283, 260,1282,
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257,1280, 254,1279, 251,1277, 248,1275,
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245,1274, 242,1272, 239,1270, 236,1269,
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233,1267, 230,1265, 227,1263, 224,1261,
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221,1259, 218,1257, 215,1255, 212,1253,
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210,1251, 207,1248, 204,1246, 201,1244,
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199,1241, 196,1239, 193,1237, 191,1234,
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188,1232, 186,1229, 183,1227, 180,1224,
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178,1221, 175,1219, 173,1216, 171,1213,
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168,1210, 166,1207, 163,1205, 161,1202,
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159,1199, 156,1196, 154,1193, 152,1190,
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150,1186, 147,1183, 145,1180, 143,1177,
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141,1174, 139,1170, 137,1167, 134,1164,
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132,1160, 130,1157, 128,1153, 126,1150,
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124,1146, 122,1143, 120,1139, 118,1136,
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117,1132, 115,1128, 113,1125, 111,1121,
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109,1117, 107,1113, 106,1109, 104,1106,
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102,1102, 100,1098, 99,1094, 97,1090,
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95,1086, 94,1082, 92,1078, 90,1074,
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89,1070, 87,1066, 86,1061, 84,1057,
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83,1053, 81,1049, 80,1045, 78,1040,
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77,1036, 76,1032, 74,1027, 73,1023,
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71,1019, 70,1014, 69,1010, 67,1005,
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66,1001, 65, 997, 64, 992, 62, 988,
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61, 983, 60, 978, 59, 974, 58, 969,
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56, 965, 55, 960, 54, 955, 53, 951,
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52, 946, 51, 941, 50, 937, 49, 932,
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48, 927, 47, 923, 46, 918, 45, 913,
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44, 908, 43, 904, 42, 899, 41, 894,
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40, 889, 39, 884, 38, 880, 37, 875,
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36, 870, 36, 865, 35, 860, 34, 855,
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33, 851, 32, 846, 32, 841, 31, 836,
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30, 831, 29, 826, 29, 821, 28, 816,
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27, 811, 27, 806, 26, 802, 25, 797,
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24, 792, 24, 787, 23, 782, 23, 777,
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22, 772, 21, 767, 21, 762, 20, 757,
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20, 752, 19, 747, 19, 742, 18, 737,
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17, 732, 17, 728, 16, 723, 16, 718,
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15, 713, 15, 708, 15, 703, 14, 698,
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14, 693, 13, 688, 13, 683, 12, 678,
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12, 674, 11, 669, 11, 664, 11, 659,
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10, 654, 10, 649, 10, 644, 9, 640,
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9, 635, 9, 630, 8, 625, 8, 620,
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8, 615, 7, 611, 7, 606, 7, 601,
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6, 596, 6, 592, 6, 587, 6, 582,
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5, 577, 5, 573, 5, 568, 5, 563,
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4, 559, 4, 554, 4, 550, 4, 545,
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4, 540, 3, 536, 3, 531, 3, 527,
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3, 522, 3, 517, 2, 513, 2, 508,
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2, 504, 2, 499, 2, 495, 2, 491,
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2, 486, 1, 482, 1, 477, 1, 473,
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1, 469, 1, 464, 1, 460, 1, 456,
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1, 451, 1, 447, 1, 443, 1, 439,
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0, 434, 0, 430, 0, 426, 0, 422,
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0, 418, 0, 414, 0, 410, 0, 405,
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0, 401, 0, 397, 0, 393, 0, 389,
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0, 385, 0, 381, 0, 378, 0, 374,
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};
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#endif /* !SPC_NOINTERP */
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void DSP_write( struct Spc_Dsp* this, int i, int data )
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{
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assert( (unsigned) i < REGISTER_COUNT );
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this->r.reg [i] = data;
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int high = i >> 4;
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int low = i & 0x0F;
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if ( low < 2 ) /* voice volumes */
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{
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int left = *(int8_t const*) &this->r.reg [i & ~1];
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int right = *(int8_t const*) &this->r.reg [i | 1];
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struct voice_t* v = this->voice_state + high;
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v->volume [0] = left;
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v->volume [1] = right;
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}
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else if ( low < 4 ) /* voice rates */
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{
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struct voice_t* v = this->voice_state + high;
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v->rate = GET_LE16A( this->r.voice[high].rate ) & 0x3fff;
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}
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#if !SPC_NOECHO
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else if ( low == 0x0F ) /* fir coefficients */
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{
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this->fir.coeff [7 - high] = (int8_t) data; /* sign-extend */
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}
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#endif /* !SPC_NOECHO */
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}
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/* Decode BRR block */
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static inline void
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decode_brr_block( struct voice_t* voice, unsigned start_addr, int16_t* out )
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{
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/* header */
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start_addr += 9; /* point to next header */
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uint8_t const* addr = ram.ram + start_addr;
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unsigned block_header = addr[-9];
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voice->wave.block_header = block_header;
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voice->wave.start_addr = start_addr;
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/* previous samples */
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int smp2 = out [0];
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int smp1 = out [1];
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int offset = -BRR_BLOCK_SIZE * 4;
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#if !SPC_BRRCACHE
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out [-(BRR_BLOCK_SIZE + 1)] = out [-1];
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/* if next block has end flag set,
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this block ends early (verified) */
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if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
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{
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/* arrange for last 9 samples to be skipped */
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int const skip = 9;
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out [skip - (BRR_BLOCK_SIZE + 1)] = out [-1];
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out += (skip & 1);
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voice->wave.position += skip * 0x1000;
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offset = (-BRR_BLOCK_SIZE + (skip & ~1)) * 4;
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addr -= skip / 2;
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/* force sample to end on next decode */
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voice->wave.block_header = 1;
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}
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#endif /* !SPC_BRRCACHE */
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int const filter = block_header & 0x0c;
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int const scale = block_header >> 4;
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if ( filter == 0x08 ) /* filter 2 (30-90% of the time) */
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{
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/* y[n] = x[n] + 61/32 * y[n-1] - 15/16 * y[n-2] */
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do /* decode and filter 16 samples */
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{
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/* Get nybble, sign-extend, then scale
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get byte, select which nybble, sign-extend, then shift
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based on scaling. */
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int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
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delta = (delta << scale) >> 1;
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if (scale > 0xc)
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delta = (delta >> 17) << 11;
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out [offset >> 2] = smp2;
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delta -= smp2 >> 1;
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delta += smp2 >> 5;
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delta += smp1;
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delta += (-smp1 - (smp1 >> 1)) >> 5;
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delta = CLAMP16( delta );
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smp2 = smp1;
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smp1 = (int16_t) (delta * 2); /* sign-extend */
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}
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while ( (offset += 4) != 0 );
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}
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else if ( filter == 0x04 ) /* filter 1 */
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{
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/* y[n] = x[n] + 15/16 * y[n-1] */
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do /* decode and filter 16 samples */
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{
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/* Get nybble, sign-extend, then scale
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get byte, select which nybble, sign-extend, then shift
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based on scaling. */
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int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
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delta = (delta << scale) >> 1;
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if (scale > 0xc)
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delta = (delta >> 17) << 11;
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out [offset >> 2] = smp2;
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delta += smp1 >> 1;
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delta += (-smp1) >> 5;
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delta = CLAMP16( delta );
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smp2 = smp1;
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smp1 = (int16_t) (delta * 2); /* sign-extend */
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}
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while ( (offset += 4) != 0 );
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}
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else if ( filter == 0x0c ) /* filter 3 */
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{
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/* y[n] = x[n] + 115/64 * y[n-1] - 13/16 * y[n-2] */
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do /* decode and filter 16 samples */
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{
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/* Get nybble, sign-extend, then scale
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get byte, select which nybble, sign-extend, then shift
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based on scaling. */
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int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
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delta = (delta << scale) >> 1;
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if (scale > 0xc)
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delta = (delta >> 17) << 11;
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out [offset >> 2] = smp2;
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delta -= smp2 >> 1;
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delta += (smp2 + (smp2 >> 1)) >> 4;
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delta += smp1;
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delta += (-smp1 * 13) >> 7;
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delta = CLAMP16( delta );
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smp2 = smp1;
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smp1 = (int16_t) (delta * 2); /* sign-extend */
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}
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while ( (offset += 4) != 0 );
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}
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else /* filter 0 */
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{
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/* y[n] = x[n] */
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do /* decode and filter 16 samples */
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{
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/* Get nybble, sign-extend, then scale
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get byte, select which nybble, sign-extend, then shift
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based on scaling. */
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int delta = (int8_t)(addr [offset >> 3] << (offset & 4)) >> 4;
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delta = (delta << scale) >> 1;
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if (scale > 0xc)
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delta = (delta >> 17) << 11;
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out [offset >> 2] = smp2;
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smp2 = smp1;
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smp1 = delta * 2;
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}
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while ( (offset += 4) != 0 );
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}
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#if SPC_BRRCACHE
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if ( !(block_header & 1) )
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{
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/* save to end of next block (for next call) */
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out [BRR_BLOCK_SIZE ] = smp2;
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out [BRR_BLOCK_SIZE + 1] = smp1;
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}
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else
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#endif /* SPC_BRRCACHE */
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{
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/* save to end of this block */
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out [0] = smp2;
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out [1] = smp1;
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}
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}
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#if SPC_BRRCACHE
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static void NO_INLINE ICODE_ATTR_SPC
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brr_decode_cache( struct Spc_Dsp* this, struct src_dir const* sd,
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unsigned start_addr, struct voice_t* voice,
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unsigned waveform, bool initial_point )
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{
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/* a little extra for samples that go past end */
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static int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR;
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DEBUGF( "decode at %04x (wave #%u)\n", start_addr, waveform );
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struct cache_entry_t* const wave_entry = &this->wave_entry [waveform];
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wave_entry->start_addr = start_addr;
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unsigned const loop_addr = letoh16( sd [waveform].loop );
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int16_t* loop_start = NULL;
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DEBUGF( "loop addr at %04x (wave #%u)\n", loop_addr, waveform );
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int16_t* out = BRRcache + start_addr * 2;
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wave_entry->samples = out;
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/* BRR filter uses previous samples */
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if ( initial_point )
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{
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/* initialize filters */
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out [BRR_BLOCK_SIZE + 1] = 0;
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out [BRR_BLOCK_SIZE + 2] = 0;
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}
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*out++ = 0;
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unsigned block_header;
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do
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{
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if ( start_addr == loop_addr )
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{
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loop_start = out;
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DEBUGF( "loop found at %04x (wave #%u)\n", start_addr, waveform );
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}
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/* output position - preincrement */
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out += BRR_BLOCK_SIZE;
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decode_brr_block( voice, start_addr, out );
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block_header = voice->wave.block_header;
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start_addr = voice->wave.start_addr;
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/* if next block has end flag set, this block ends early */
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/* (verified) */
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if ( (block_header & 3) != 3 && (start_addr[ram.ram] & 3) == 1 )
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{
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/* skip last 9 samples */
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DEBUGF( "block early end (wave #%u)\n", waveform );
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out -= 9;
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break;
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}
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}
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while ( !(block_header & 1) && start_addr < 0x10000 );
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wave_entry->end = (out - 1 - wave_entry->samples) << 12;
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wave_entry->loop = 0;
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wave_entry->loop_addr = 0;
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wave_entry->block_header = block_header;
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if ( (block_header & 2) )
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{
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if ( loop_start )
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{
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wave_entry->loop = out - loop_start;
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wave_entry->end += 0x3000;
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out [2] = loop_start [2];
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out [3] = loop_start [3];
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out [4] = loop_start [4];
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}
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else
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{
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DEBUGF( "loop point outside initial wave\n" );
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/* Plan filter init for later decoding at loop point */
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int16_t* next = BRRcache + loop_addr * 2;
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next [BRR_BLOCK_SIZE + 1] = out [0];
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next [BRR_BLOCK_SIZE + 2] = out [1];
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}
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wave_entry->loop_addr = loop_addr;
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}
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DEBUGF( "end at %04x (wave #%u)\n\n", start_addr, waveform );
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/* add to cache */
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this->wave_entry_old [this->oldsize++] = *wave_entry;
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}
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/* see if in cache */
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static inline bool
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brr_probe_cache( struct Spc_Dsp* this, unsigned start_addr,
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struct cache_entry_t* wave_entry )
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{
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if ( wave_entry->start_addr == start_addr )
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return true;
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for ( unsigned i = 0; i < this->oldsize; i++ )
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{
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struct cache_entry_t* e = &this->wave_entry_old [i];
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if ( e->start_addr == start_addr )
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{
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#if 0 /* do NOT want to see all the key down stuff for cached waves */
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DEBUGF( "found in wave_entry_old (oldsize=%u)\n",
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this->oldsize );
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#endif
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*wave_entry = *e;
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return true; /* Wave in cache */
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}
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}
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|
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return false;
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}
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|
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static NO_INLINE ICODE_ATTR_SPC void
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brr_key_on( struct Spc_Dsp* this, struct src_dir const* sd,
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struct voice_t* voice, struct raw_voice_t const* raw_voice,
|
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unsigned start_addr )
|
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{
|
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bool initial_point = false;
|
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unsigned waveform = raw_voice->waveform;
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|
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if (start_addr == (unsigned)-1)
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{
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initial_point = true;
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start_addr = letoh16( sd [waveform].start );
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}
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|
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struct cache_entry_t* const wave_entry = &this->wave_entry [waveform];
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|
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/* predecode BRR if not already */
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if ( !brr_probe_cache( this, start_addr, wave_entry ) )
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{
|
|
/* actually decode it */
|
|
brr_decode_cache( this, sd, start_addr, voice, waveform,
|
|
initial_point );
|
|
}
|
|
|
|
voice->wave.position = 3 * 0x1000 - 1; /* 0x2fff */
|
|
voice->wave.samples = wave_entry->samples;
|
|
voice->wave.end = wave_entry->end;
|
|
voice->wave.loop = wave_entry->loop;
|
|
voice->wave.start_addr = wave_entry->start_addr;
|
|
voice->wave.loop_addr = wave_entry->loop_addr;
|
|
voice->wave.block_header = wave_entry->block_header;
|
|
}
|
|
|
|
static inline int brr_decode( struct Spc_Dsp* this, struct src_dir const* sd,
|
|
struct voice_t* voice,
|
|
struct raw_voice_t const* raw_voice )
|
|
{
|
|
if ( voice->wave.position < voice->wave.end )
|
|
return 0;
|
|
|
|
long loop_len = voice->wave.loop << 12;
|
|
|
|
if ( !loop_len )
|
|
{
|
|
if ( !(voice->wave.block_header & 2 ) )
|
|
return 2;
|
|
|
|
/* "Loop" is outside initial waveform */
|
|
brr_key_on( this, sd, voice, raw_voice, voice->wave.loop_addr );
|
|
}
|
|
|
|
voice->wave.position -= loop_len;
|
|
return 1;
|
|
|
|
(void)sd; (void)raw_voice;
|
|
}
|
|
|
|
#else /* !SPC_BRRCACHE */
|
|
|
|
static inline void
|
|
brr_key_on( struct Spc_Dsp* this, struct src_dir const* sd,
|
|
struct voice_t* voice, struct raw_voice_t const* raw_voice,
|
|
unsigned start_addr )
|
|
{
|
|
voice->wave.start_addr = letoh16( sd [raw_voice->waveform].start );
|
|
/* BRR filter uses previous samples */
|
|
voice->wave.samples [BRR_BLOCK_SIZE + 1] = 0;
|
|
voice->wave.samples [BRR_BLOCK_SIZE + 2] = 0;
|
|
/* force decode on next brr_decode call */
|
|
voice->wave.position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1; /* 0x12fff */
|
|
voice->wave.block_header = 0; /* "previous" BRR header */
|
|
(void)this; (void)start_addr;
|
|
}
|
|
|
|
static inline int brr_decode( struct Spc_Dsp* this, struct src_dir const* sd,
|
|
struct voice_t* voice,
|
|
struct raw_voice_t const* raw_voice )
|
|
{
|
|
if ( voice->wave.position < BRR_BLOCK_SIZE * 0x1000 )
|
|
return 0;
|
|
|
|
voice->wave.position -= BRR_BLOCK_SIZE * 0x1000;
|
|
|
|
unsigned start_addr = voice->wave.start_addr;
|
|
|
|
if ( start_addr >= 0x10000 )
|
|
start_addr -= 0x10000;
|
|
|
|
unsigned block_header = voice->wave.block_header;
|
|
|
|
/* action based on previous block's header */
|
|
int dec = 0;
|
|
|
|
if ( block_header & 1 )
|
|
{
|
|
start_addr = letoh16( sd [raw_voice->waveform].loop );
|
|
dec = 1;
|
|
|
|
if ( !(block_header & 2) ) /* 1% of the time */
|
|
{
|
|
/* first block was end block;
|
|
don't play anything (verified) */
|
|
return 2;
|
|
}
|
|
}
|
|
|
|
decode_brr_block( voice, start_addr,
|
|
&voice->wave.samples [1 + BRR_BLOCK_SIZE] );
|
|
|
|
return dec;
|
|
(void)this;
|
|
}
|
|
#endif /* SPC_BRRCACHE */
|
|
|
|
static void NO_INLINE ICODE_ATTR_SPC
|
|
key_on( struct Spc_Dsp* const this, struct voice_t* const voice,
|
|
struct src_dir const* const sd,
|
|
struct raw_voice_t const* const raw_voice,
|
|
const int key_on_delay, const int vbit )
|
|
{
|
|
voice->key_on_delay = key_on_delay;
|
|
|
|
if ( key_on_delay == 0 )
|
|
{
|
|
this->keys_down |= vbit;
|
|
voice->envx = 0;
|
|
voice->env_mode = state_attack;
|
|
voice->env_timer = ENV_RATE_INIT; /* TODO: inaccurate? */
|
|
brr_key_on( this, sd, voice, raw_voice, -1 );
|
|
}
|
|
}
|
|
|
|
void DSP_run_( struct Spc_Dsp* this, long count, int32_t* out_buf )
|
|
{
|
|
#undef RAM
|
|
#if defined(CPU_ARM) && !SPC_BRRCACHE
|
|
uint8_t* const ram_ = ram.ram;
|
|
#define RAM ram_
|
|
#else
|
|
#define RAM ram.ram
|
|
#endif
|
|
EXIT_TIMER(cpu);
|
|
ENTER_TIMER(dsp);
|
|
|
|
/* 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. */
|
|
|
|
{
|
|
int key_ons = this->r.g.key_ons;
|
|
int key_offs = this->r.g.key_offs;
|
|
/* keying on a voice resets that bit in ENDX */
|
|
this->r.g.wave_ended &= ~key_ons;
|
|
/* key_off bits prevent key_on from being acknowledged */
|
|
this->r.g.key_ons = key_ons & key_offs;
|
|
|
|
/* process key events outside loop, since they won't re-occur */
|
|
struct voice_t* voice = this->voice_state + 8;
|
|
int vbit = 0x80;
|
|
do
|
|
{
|
|
--voice;
|
|
if ( key_offs & vbit )
|
|
{
|
|
voice->env_mode = state_release;
|
|
voice->key_on_delay = 0;
|
|
}
|
|
else if ( key_ons & vbit )
|
|
{
|
|
voice->key_on_delay = 8;
|
|
}
|
|
}
|
|
while ( (vbit >>= 1) != 0 );
|
|
}
|
|
|
|
struct src_dir const* const sd =
|
|
&ram.sd [this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
|
|
|
|
#if !SPC_NOINTERP
|
|
int const slow_gaussian = (this->r.g.pitch_mods >> 1) |
|
|
this->r.g.noise_enables;
|
|
#endif
|
|
#if !SPC_NOECHO
|
|
int const echo_start = this->r.g.echo_page * 0x100;
|
|
int const echo_delay = (this->r.g.echo_delay & 15) * 0x800;
|
|
#endif
|
|
/* (g.flags & 0x40) ? 30 : 14 */
|
|
int const global_muting = ((this->r.g.flags & 0x40) >> 2) + 14 - 8;
|
|
int const global_vol_0 = this->r.g.volume_0;
|
|
int const global_vol_1 = this->r.g.volume_1;
|
|
|
|
do /* one pair of output samples per iteration */
|
|
{
|
|
/* Noise */
|
|
if ( this->r.g.noise_enables )
|
|
{
|
|
this->noise_count -= env_rates [this->r.g.flags & 0x1F];
|
|
|
|
if ( this->noise_count <= 0 )
|
|
{
|
|
this->noise_count = ENV_RATE_INIT;
|
|
int feedback = (this->noise << 13) ^ (this->noise << 14);
|
|
this->noise = (feedback & 0x8000) ^ (this->noise >> 1 & ~1);
|
|
}
|
|
}
|
|
|
|
#if !SPC_NOECHO
|
|
int echo_0 = 0, echo_1 = 0;
|
|
#endif /* !SPC_NOECHO */
|
|
long prev_outx = 0; /* TODO: correct value for first channel? */
|
|
int chans_0 = 0, chans_1 = 0;
|
|
|
|
/* TODO: put raw_voice pointer in voice_t? */
|
|
struct raw_voice_t * raw_voice = this->r.voice;
|
|
struct voice_t* voice = this->voice_state;
|
|
|
|
for (int vbit = 1; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice )
|
|
{
|
|
/* pregen involves checking keyon, etc */
|
|
ENTER_TIMER(dsp_pregen);
|
|
|
|
/* Key on events are delayed */
|
|
int key_on_delay = voice->key_on_delay;
|
|
|
|
if ( UNLIKELY ( --key_on_delay >= 0 ) ) /* <1% of the time */
|
|
key_on( this, voice, sd, raw_voice, key_on_delay, vbit );
|
|
|
|
if ( !(this->keys_down & vbit) ) /* Silent channel */
|
|
{
|
|
silent_chan:
|
|
raw_voice->envx = 0;
|
|
raw_voice->outx = 0;
|
|
prev_outx = 0;
|
|
continue;
|
|
}
|
|
|
|
/* Envelope */
|
|
{
|
|
int const ENV_RANGE = 0x800;
|
|
int env_mode = voice->env_mode;
|
|
int adsr0 = raw_voice->adsr [0];
|
|
int env_timer;
|
|
if ( LIKELY ( env_mode != state_release ) ) /* 99% of the time */
|
|
{
|
|
env_timer = voice->env_timer;
|
|
if ( LIKELY ( adsr0 & 0x80 ) ) /* 79% of the time */
|
|
{
|
|
int adsr1 = raw_voice->adsr [1];
|
|
if ( LIKELY ( env_mode == state_sustain ) ) /* 74% of the time */
|
|
{
|
|
if ( (env_timer -= env_rates [adsr1 & 0x1F]) > 0 )
|
|
goto write_env_timer;
|
|
|
|
int envx = voice->envx;
|
|
envx--; /* envx *= 255 / 256 */
|
|
envx -= envx >> 8;
|
|
voice->envx = envx;
|
|
/* TODO: should this be 8? */
|
|
raw_voice->envx = envx >> 4;
|
|
goto init_env_timer;
|
|
}
|
|
else if ( env_mode < 0 ) /* 25% state_decay */
|
|
{
|
|
int envx = voice->envx;
|
|
if ( (env_timer -=
|
|
env_rates [(adsr0 >> 3 & 0x0E) + 0x10]) <= 0 )
|
|
{
|
|
envx--; /* envx *= 255 / 256 */
|
|
envx -= envx >> 8;
|
|
voice->envx = envx;
|
|
/* TODO: should this be 8? */
|
|
raw_voice->envx = envx >> 4;
|
|
env_timer = ENV_RATE_INIT;
|
|
}
|
|
|
|
int sustain_level = adsr1 >> 5;
|
|
if ( envx <= (sustain_level + 1) * 0x100 )
|
|
voice->env_mode = state_sustain;
|
|
|
|
goto write_env_timer;
|
|
}
|
|
else /* state_attack */
|
|
{
|
|
int t = adsr0 & 0x0F;
|
|
if ( (env_timer -= env_rates [t * 2 + 1]) > 0 )
|
|
goto write_env_timer;
|
|
|
|
int envx = voice->envx;
|
|
|
|
int const step = ENV_RANGE / 64;
|
|
envx += step;
|
|
if ( t == 15 )
|
|
envx += ENV_RANGE / 2 - step;
|
|
|
|
if ( envx >= ENV_RANGE )
|
|
{
|
|
envx = ENV_RANGE - 1;
|
|
voice->env_mode = state_decay;
|
|
}
|
|
voice->envx = envx;
|
|
/* TODO: should this be 8? */
|
|
raw_voice->envx = envx >> 4;
|
|
goto init_env_timer;
|
|
}
|
|
}
|
|
else /* gain mode */
|
|
{
|
|
int t = raw_voice->gain;
|
|
if ( t < 0x80 )
|
|
{
|
|
raw_voice->envx = t;
|
|
voice->envx = t << 4;
|
|
goto env_end;
|
|
}
|
|
else
|
|
{
|
|
if ( (env_timer -= env_rates [t & 0x1F]) > 0 )
|
|
goto write_env_timer;
|
|
|
|
int envx = voice->envx;
|
|
int mode = t >> 5;
|
|
if ( mode <= 5 ) /* decay */
|
|
{
|
|
int step = ENV_RANGE / 64;
|
|
if ( mode == 5 ) /* exponential */
|
|
{
|
|
envx--; /* envx *= 255 / 256 */
|
|
step = envx >> 8;
|
|
}
|
|
if ( (envx -= step) < 0 )
|
|
{
|
|
envx = 0;
|
|
if ( voice->env_mode == state_attack )
|
|
voice->env_mode = state_decay;
|
|
}
|
|
}
|
|
else /* attack */
|
|
{
|
|
int const step = ENV_RANGE / 64;
|
|
envx += step;
|
|
if ( mode == 7 &&
|
|
envx >= ENV_RANGE * 3 / 4 + step )
|
|
envx += ENV_RANGE / 256 - step;
|
|
|
|
if ( envx >= ENV_RANGE )
|
|
envx = ENV_RANGE - 1;
|
|
}
|
|
voice->envx = envx;
|
|
/* TODO: should this be 8? */
|
|
raw_voice->envx = envx >> 4;
|
|
goto init_env_timer;
|
|
}
|
|
}
|
|
}
|
|
else /* state_release */
|
|
{
|
|
int envx = voice->envx;
|
|
if ( (envx -= ENV_RANGE / 256) > 0 )
|
|
{
|
|
voice->envx = envx;
|
|
raw_voice->envx = envx >> 8;
|
|
goto env_end;
|
|
}
|
|
else
|
|
{
|
|
/* bit was set, so this clears it */
|
|
this->keys_down ^= vbit;
|
|
voice->envx = 0;
|
|
goto silent_chan;
|
|
}
|
|
}
|
|
init_env_timer:
|
|
env_timer = ENV_RATE_INIT;
|
|
write_env_timer:
|
|
voice->env_timer = env_timer;
|
|
env_end:;
|
|
}
|
|
|
|
EXIT_TIMER(dsp_pregen);
|
|
|
|
ENTER_TIMER(dsp_gen);
|
|
|
|
switch ( brr_decode( this, sd, voice, raw_voice ) )
|
|
{
|
|
case 2:
|
|
/* bit was set, so this clears it */
|
|
this->keys_down ^= vbit;
|
|
|
|
/* since voice->envx is 0,
|
|
samples and position don't matter */
|
|
raw_voice->envx = 0;
|
|
voice->envx = 0;
|
|
case 1:
|
|
this->r.g.wave_ended |= vbit;
|
|
}
|
|
|
|
/* Get rate (with possible modulation) */
|
|
int rate = voice->rate;
|
|
if ( this->r.g.pitch_mods & vbit )
|
|
rate = (rate * (prev_outx + 32768)) >> 15;
|
|
|
|
uint32_t position = voice->wave.position;
|
|
voice->wave.position += rate;
|
|
|
|
int output;
|
|
int amp_0, amp_1;
|
|
|
|
#if !SPC_NOINTERP
|
|
/* Gaussian interpolation using most recent 4 samples */
|
|
|
|
/* Only left half of gaussian kernel is in table, so we must mirror
|
|
for right half */
|
|
int offset = ( position >> 4 ) & 0xFF;
|
|
int16_t const* fwd = gauss_table + offset * 2;
|
|
int16_t const* rev = gauss_table + 510 - offset * 2;
|
|
|
|
/* Use faster gaussian interpolation when exact result isn't needed
|
|
by pitch modulator of next channel */
|
|
if ( LIKELY ( !(slow_gaussian & vbit) ) ) /* 99% of the time */
|
|
{
|
|
/* Main optimization is lack of clamping. Not a problem since
|
|
output never goes more than +/- 16 outside 16-bit range and
|
|
things are clamped later anyway. Other optimization is to
|
|
preserve fractional accuracy, eliminating several masks. */
|
|
output = gaussian_fast_interp( voice->wave.samples, position,
|
|
fwd, rev );
|
|
output = gaussian_fast_amp( voice, output, &_0, &_1 );
|
|
}
|
|
else /* slow gaussian */
|
|
#endif /* !SPC_NOINTERP (else two-point linear interpolation) */
|
|
{
|
|
output = *(int16_t *)&this->noise;
|
|
|
|
if ( !(this->r.g.noise_enables & vbit) )
|
|
output = interp( voice->wave.samples, position, fwd, rev );
|
|
|
|
/* Apply envelope and volume */
|
|
output = apply_amp( voice, output, &_0, &_1 );
|
|
}
|
|
|
|
prev_outx = output;
|
|
raw_voice->outx = output >> 8;
|
|
|
|
EXIT_TIMER(dsp_gen);
|
|
|
|
ENTER_TIMER(dsp_mix);
|
|
|
|
chans_0 += amp_0;
|
|
chans_1 += amp_1;
|
|
#if !SPC_NOECHO
|
|
if ( this->r.g.echo_ons & vbit )
|
|
{
|
|
echo_0 += amp_0;
|
|
echo_1 += amp_1;
|
|
}
|
|
#endif /* !SPC_NOECHO */
|
|
|
|
EXIT_TIMER(dsp_mix);
|
|
}
|
|
/* end of voice loop */
|
|
|
|
/* Generate output */
|
|
int amp_0, amp_1;
|
|
#if !SPC_NOECHO
|
|
/* Read feedback from echo buffer */
|
|
int echo_pos = this->echo_pos;
|
|
uint8_t* const echo_ptr = RAM + ((echo_start + echo_pos) & 0xFFFF);
|
|
|
|
echo_pos += 4;
|
|
|
|
if ( echo_pos >= echo_delay )
|
|
echo_pos = 0;
|
|
|
|
this->echo_pos = echo_pos;
|
|
|
|
/* Apply FIR */
|
|
int fb_0, fb_1;
|
|
echo_apply( this, echo_ptr, &fb_0, &fb_1 );
|
|
|
|
if ( !(this->r.g.flags & 0x20) )
|
|
{
|
|
/* Feedback into echo buffer */
|
|
echo_feedback( this, echo_ptr, echo_0, echo_1, fb_0, fb_1 );
|
|
}
|
|
#endif /* !SPC_NOECHO */
|
|
|
|
mix_output( this, global_muting, global_vol_0, global_vol_1,
|
|
chans_0, chans_1, fb_0, fb_1, &_0, &_1 );
|
|
|
|
out_buf [ 0] = amp_0;
|
|
out_buf [WAV_CHUNK_SIZE] = amp_1;
|
|
out_buf ++;
|
|
}
|
|
while ( --count );
|
|
|
|
EXIT_TIMER(dsp);
|
|
ENTER_TIMER(cpu);
|
|
}
|
|
|
|
void DSP_reset( struct Spc_Dsp* this )
|
|
{
|
|
this->keys_down = 0;
|
|
this->noise_count = 0;
|
|
this->noise = 2;
|
|
|
|
this->r.g.flags = 0xE0; /* reset, mute, echo off */
|
|
this->r.g.key_ons = 0;
|
|
|
|
ci->memset( this->voice_state, 0, sizeof this->voice_state );
|
|
|
|
for ( int i = VOICE_COUNT; --i >= 0; )
|
|
{
|
|
struct voice_t* v = this->voice_state + i;
|
|
v->env_mode = state_release;
|
|
v->wave.start_addr = 0;
|
|
}
|
|
|
|
#if SPC_BRRCACHE
|
|
this->oldsize = 0;
|
|
for ( int i = 0; i < 256; i++ )
|
|
this->wave_entry [i].start_addr = 0xffff;
|
|
#endif /* SPC_BRRCACHE */
|
|
|
|
#if !SPC_NOINTERP && GAUSS_TABLE_SCALE
|
|
if (gauss_table[0] == 370)
|
|
{
|
|
/* Not yet scaled */
|
|
for ( int i = 0; i < 512; i++)
|
|
gauss_table[i] <<= GAUSS_TABLE_SCALE;
|
|
}
|
|
#endif /* !SPC_NOINTERP && GAUSS_TABLE_SCALE */
|
|
|
|
#if !SPC_NOECHO
|
|
this->echo_pos = 0;
|
|
echo_init(this);
|
|
#endif /* SPC_NOECHO */
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assert( offsetof (struct globals_t,unused9 [2]) == REGISTER_COUNT );
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assert( sizeof (this->r.voice) == REGISTER_COUNT );
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}
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