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qzdoom/game-music-emu/gme/Spc_Dsp.cpp
Randy Heit fbddfbe576 Upgrade game-music-emu to version 0.6.0
2014-04-03 09:19:51 -05:00

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// Game_Music_Emu 0.6.0. http://www.slack.net/~ant/
#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 ) ((BOOST::int16_t) GET_LE16( addr ))
#define GET_LE16A( addr ) GET_LE16( addr )
#define SET_LE16A( addr, data ) SET_LE16( addr, data )
static BOOST::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
int s = ((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 ); }
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