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snes9x/apu/SPC_DSP.cpp
OV2 9f3b69f575 Fix Dual Orb 2 sound glitch (byuu)
2010-09-25 20:54:49 +02:00

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// snes_spc 0.9.0. 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,0x44,0x00,0x4E,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.
// Gaussian interpolation
static short const gauss [512] =
{
0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0,
1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2,
2, 2, 3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 5, 5, 5, 5,
6, 6, 6, 6, 7, 7, 7, 8, 8, 8, 9, 9, 9, 10, 10, 10,
11, 11, 11, 12, 12, 13, 13, 14, 14, 15, 15, 15, 16, 16, 17, 17,
18, 19, 19, 20, 20, 21, 21, 22, 23, 23, 24, 24, 25, 26, 27, 27,
28, 29, 29, 30, 31, 32, 32, 33, 34, 35, 36, 36, 37, 38, 39, 40,
41, 42, 43, 44, 45, 46, 47, 48, 49, 50, 51, 52, 53, 54, 55, 56,
58, 59, 60, 61, 62, 64, 65, 66, 67, 69, 70, 71, 73, 74, 76, 77,
78, 80, 81, 83, 84, 86, 87, 89, 90, 92, 94, 95, 97, 99, 100, 102,
104, 106, 107, 109, 111, 113, 115, 117, 118, 120, 122, 124, 126, 128, 130, 132,
134, 137, 139, 141, 143, 145, 147, 150, 152, 154, 156, 159, 161, 163, 166, 168,
171, 173, 175, 178, 180, 183, 186, 188, 191, 193, 196, 199, 201, 204, 207, 210,
212, 215, 218, 221, 224, 227, 230, 233, 236, 239, 242, 245, 248, 251, 254, 257,
260, 263, 267, 270, 273, 276, 280, 283, 286, 290, 293, 297, 300, 304, 307, 311,
314, 318, 321, 325, 328, 332, 336, 339, 343, 347, 351, 354, 358, 362, 366, 370,
374, 378, 381, 385, 389, 393, 397, 401, 405, 410, 414, 418, 422, 426, 430, 434,
439, 443, 447, 451, 456, 460, 464, 469, 473, 477, 482, 486, 491, 495, 499, 504,
508, 513, 517, 522, 527, 531, 536, 540, 545, 550, 554, 559, 563, 568, 573, 577,
582, 587, 592, 596, 601, 606, 611, 615, 620, 625, 630, 635, 640, 644, 649, 654,
659, 664, 669, 674, 678, 683, 688, 693, 698, 703, 708, 713, 718, 723, 728, 732,
737, 742, 747, 752, 757, 762, 767, 772, 777, 782, 787, 792, 797, 802, 806, 811,
816, 821, 826, 831, 836, 841, 846, 851, 855, 860, 865, 870, 875, 880, 884, 889,
894, 899, 904, 908, 913, 918, 923, 927, 932, 937, 941, 946, 951, 955, 960, 965,
969, 974, 978, 983, 988, 992, 997,1001,1005,1010,1014,1019,1023,1027,1032,1036,
1040,1045,1049,1053,1057,1061,1066,1070,1074,1078,1082,1086,1090,1094,1098,1102,
1106,1109,1113,1117,1121,1125,1128,1132,1136,1139,1143,1146,1150,1153,1157,1160,
1164,1167,1170,1174,1177,1180,1183,1186,1190,1193,1196,1199,1202,1205,1207,1210,
1213,1216,1219,1221,1224,1227,1229,1232,1234,1237,1239,1241,1244,1246,1248,1251,
1253,1255,1257,1259,1261,1263,1265,1267,1269,1270,1272,1274,1275,1277,1279,1280,
1282,1283,1284,1286,1287,1288,1290,1291,1292,1293,1294,1295,1296,1297,1297,1298,
1299,1300,1300,1301,1302,1302,1303,1303,1303,1304,1304,1304,1304,1304,1305,1305,
};
inline int SPC_DSP::interpolate( voice_t const* v )
{
// Make pointers into gaussian based on fractional position between samples
int offset = v->interp_pos >> 4 & 0xFF;
short const* fwd = gauss + 255 - offset;
short const* rev = gauss + offset; // mirror left half of gaussian
int const* in = &v->buf [(v->interp_pos >> 12) + v->buf_pos];
int out;
out = (fwd [ 0] * in [0]) >> 11;
out += (fwd [256] * in [1]) >> 11;
out += (rev [256] * in [2]) >> 11;
out = (int16_t) out;
out += (rev [ 0] * in [3]) >> 11;
CLAMP16( out );
out &= ~1;
return out;
}
//// Counters
int const simple_counter_range = 2048 * 5 * 3; // 30720
static unsigned const counter_rates [32] =
{
simple_counter_range + 1, // never fires
2048, 1536,
1280, 1024, 768,
640, 512, 384,
320, 256, 192,
160, 128, 96,
80, 64, 48,
40, 32, 24,
20, 16, 12,
10, 8, 6,
5, 4, 3,
2,
1
};
static unsigned const counter_offsets [32] =
{
1, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
536, 0, 1040,
0,
0
};
inline void SPC_DSP::init_counter()
{
m.counter = 0;
}
inline void SPC_DSP::run_counters()
{
if ( --m.counter < 0 )
m.counter = simple_counter_range - 1;
}
inline unsigned SPC_DSP::read_counter( int rate )
{
return ((unsigned) m.counter + counter_offsets [rate]) % counter_rates [rate];
}
//// Envelope
inline void SPC_DSP::run_envelope( voice_t* const v )
{
int env = v->env;
if ( v->env_mode == env_release ) // 60%
{
if ( (env -= 0x8) < 0 )
env = 0;
v->env = env;
}
else
{
int rate;
int env_data = VREG(v->regs,adsr1);
if ( m.t_adsr0 & 0x80 ) // 99% ADSR
{
if ( v->env_mode >= env_decay ) // 99%
{
env--;
env -= env >> 8;
rate = env_data & 0x1F;
if ( v->env_mode == env_decay ) // 1%
rate = (m.t_adsr0 >> 3 & 0x0E) + 0x10;
}
else // env_attack
{
rate = (m.t_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
}
}
//// BRR Decoding
inline void SPC_DSP::decode_brr( voice_t* v )
{
// Arrange the four input nybbles in 0xABCD order for easy decoding
int nybbles = m.t_brr_byte * 0x100 + m.ram [(v->brr_addr + v->brr_offset + 1) & 0xFFFF];
int const header = m.t_brr_header;
// Write to next four samples in circular buffer
int* pos = &v->buf [v->buf_pos];
int* end;
if ( (v->buf_pos += 4) >= brr_buf_size )
v->buf_pos = 0;
// Decode four samples
for ( end = pos + 4; pos < end; pos++, nybbles <<= 4 )
{
// Extract nybble and sign-extend
int s = (int16_t) nybbles >> 12;
// Shift sample based on header
int const shift = header >> 4;
s = (s << shift) >> 1;
if ( shift >= 0xD ) // handle invalid range
s = (s >> 25) << 11; // same as: s = (s < 0 ? -0x800 : 0)
// Apply IIR filter (8 is the most commonly used)
int const filter = 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
}
}
//// Misc
#define MISC_CLOCK( n ) inline void SPC_DSP::misc_##n()
MISC_CLOCK( 27 )
{
m.t_pmon = REG(pmon) & 0xFE; // voice 0 doesn't support PMON
}
MISC_CLOCK( 28 )
{
m.t_non = REG(non);
m.t_eon = REG(eon);
m.t_dir = REG(dir);
}
MISC_CLOCK( 29 )
{
if ( (m.every_other_sample ^= 1) != 0 )
m.new_kon &= ~m.kon; // clears KON 63 clocks after it was last read
}
MISC_CLOCK( 30 )
{
if ( m.every_other_sample )
{
m.kon = m.new_kon;
m.t_koff = REG(koff) | m.mute_mask;
}
run_counters();
// Noise
if ( !read_counter( REG(flg) & 0x1F ) )
{
int feedback = (m.noise << 13) ^ (m.noise << 14);
m.noise = (feedback & 0x4000) ^ (m.noise >> 1);
}
}
//// Voices
#define VOICE_CLOCK( n ) void SPC_DSP::voice_##n( voice_t* const v )
inline VOICE_CLOCK( V1 )
{
m.t_dir_addr = m.t_dir * 0x100 + m.t_srcn * 4;
m.t_srcn = VREG(v->regs,srcn);
}
inline VOICE_CLOCK( V2 )
{
// Read sample pointer (ignored if not needed)
uint8_t const* entry = &m.ram [m.t_dir_addr];
if ( !v->kon_delay )
entry += 2;
m.t_brr_next_addr = GET_LE16A( entry );
m.t_adsr0 = VREG(v->regs,adsr0);
// Read pitch, spread over two clocks
m.t_pitch = VREG(v->regs,pitchl);
}
inline VOICE_CLOCK( V3a )
{
m.t_pitch += (VREG(v->regs,pitchh) & 0x3F) << 8;
}
inline VOICE_CLOCK( V3b )
{
// Read BRR header and byte
m.t_brr_byte = m.ram [(v->brr_addr + v->brr_offset) & 0xFFFF];
m.t_brr_header = m.ram [v->brr_addr]; // brr_addr doesn't need masking
}
VOICE_CLOCK( V3c )
{
// Pitch modulation using previous voice's output
if ( m.t_pmon & v->vbit )
m.t_pitch += ((m.t_output >> 5) * m.t_pitch) >> 10;
if ( v->kon_delay )
{
// Get ready to start BRR decoding on next sample
if ( v->kon_delay == 5 )
{
v->brr_addr = m.t_brr_next_addr;
v->brr_offset = 1;
v->buf_pos = 0;
m.t_brr_header = 0; // header is ignored on this sample
m.kon_check = true;
if (take_spc_snapshot)
{
take_spc_snapshot = 0;
if (spc_snapshot_callback)
spc_snapshot_callback();
}
}
// Envelope is never run during KON
v->env = 0;
v->hidden_env = 0;
// Disable BRR decoding until last three samples
v->interp_pos = 0;
if ( --v->kon_delay & 3 )
v->interp_pos = 0x4000;
// Pitch is never added during KON
m.t_pitch = 0;
}
// Gaussian interpolation
{
int output = interpolate( v );
// Noise
if ( m.t_non & v->vbit )
output = (int16_t) (m.noise * 2);
// Apply envelope
m.t_output = (output * v->env) >> 11 & ~1;
v->t_envx_out = (uint8_t) (v->env >> 4);
}
// Immediate silence due to end of sample or soft reset
if ( REG(flg) & 0x80 || (m.t_brr_header & 3) == 1 )
{
v->env_mode = env_release;
v->env = 0;
}
if ( m.every_other_sample )
{
// KOFF
if ( m.t_koff & v->vbit )
v->env_mode = env_release;
// KON
if ( m.kon & v->vbit )
{
v->kon_delay = 5;
v->env_mode = env_attack;
}
}
// Run envelope for next sample
if ( !v->kon_delay )
run_envelope( v );
}
inline void SPC_DSP::voice_output( voice_t const* v, int ch )
{
// Apply left/right volume
int amp = (m.t_output * (int8_t) VREG(v->regs,voll + ch)) >> 7;
amp *= ((stereo_switch & (1 << (v->voice_number + ch * voice_count))) ? 1 : 0);
// Add to output total
m.t_main_out [ch] += amp;
CLAMP16( m.t_main_out [ch] );
// Optionally add to echo total
if ( m.t_eon & v->vbit )
{
m.t_echo_out [ch] += amp;
CLAMP16( m.t_echo_out [ch] );
}
}
VOICE_CLOCK( V4 )
{
// Decode BRR
m.t_looped = 0;
if ( v->interp_pos >= 0x4000 )
{
decode_brr( v );
if ( (v->brr_offset += 2) >= brr_block_size )
{
// Start decoding next BRR block
assert( v->brr_offset == brr_block_size );
v->brr_addr = (v->brr_addr + brr_block_size) & 0xFFFF;
if ( m.t_brr_header & 1 )
{
v->brr_addr = m.t_brr_next_addr;
m.t_looped = v->vbit;
}
v->brr_offset = 1;
}
}
// Apply pitch
v->interp_pos = (v->interp_pos & 0x3FFF) + m.t_pitch;
// Keep from getting too far ahead (when using pitch modulation)
if ( v->interp_pos > 0x7FFF )
v->interp_pos = 0x7FFF;
// Output left
voice_output( v, 0 );
}
inline VOICE_CLOCK( V5 )
{
// Output right
voice_output( v, 1 );
// ENDX, OUTX, and ENVX won't update if you wrote to them 1-2 clocks earlier
int endx_buf = REG(endx) | m.t_looped;
// Clear bit in ENDX if KON just began
if ( v->kon_delay == 5 )
endx_buf &= ~v->vbit;
m.endx_buf = (uint8_t) endx_buf;
}
inline VOICE_CLOCK( V6 )
{
(void) v; // avoid compiler warning about unused v
m.outx_buf = (uint8_t) (m.t_output >> 8);
}
inline VOICE_CLOCK( V7 )
{
// Update ENDX
REG(endx) = m.endx_buf;
m.envx_buf = v->t_envx_out;
}
inline VOICE_CLOCK( V8 )
{
// Update OUTX
VREG(v->regs,outx) = m.outx_buf;
}
inline VOICE_CLOCK( V9 )
{
// Update ENVX
VREG(v->regs,envx) = m.envx_buf;
}
// Most voices do all these in one clock, so make a handy composite
inline VOICE_CLOCK( V3 )
{
voice_V3a( v );
voice_V3b( v );
voice_V3c( v );
}
// Common combinations of voice steps on different voices. This greatly reduces
// code size and allows everything to be inlined in these functions.
VOICE_CLOCK(V7_V4_V1) { voice_V7(v); voice_V1(v+3); voice_V4(v+1); }
VOICE_CLOCK(V8_V5_V2) { voice_V8(v); voice_V5(v+1); voice_V2(v+2); }
VOICE_CLOCK(V9_V6_V3) { voice_V9(v); voice_V6(v+1); voice_V3(v+2); }
//// Echo
// Current echo buffer pointer for left/right channel
#define ECHO_PTR( ch ) (&m.ram [m.t_echo_ptr + ch * 2])
// Sample in echo history buffer, where 0 is the oldest
#define ECHO_FIR( i ) (m.echo_hist_pos [i])
// Calculate FIR point for left/right channel
#define CALC_FIR( i, ch ) ((ECHO_FIR( i + 1 ) [ch] * (int8_t) REG(fir + i * 0x10)) >> 6)
#define ECHO_CLOCK( n ) inline void SPC_DSP::echo_##n()
inline void SPC_DSP::echo_read( int ch )
{
int s = GET_LE16SA( ECHO_PTR( ch ) );
// second copy simplifies wrap-around handling
ECHO_FIR( 0 ) [ch] = ECHO_FIR( 8 ) [ch] = s >> 1;
}
ECHO_CLOCK( 22 )
{
// History
if ( ++m.echo_hist_pos >= &m.echo_hist [echo_hist_size] )
m.echo_hist_pos = m.echo_hist;
m.t_echo_ptr = (m.t_esa * 0x100 + m.echo_offset) & 0xFFFF;
echo_read( 0 );
// FIR (using l and r temporaries below helps compiler optimize)
int l = CALC_FIR( 0, 0 );
int r = CALC_FIR( 0, 1 );
m.t_echo_in [0] = l;
m.t_echo_in [1] = r;
}
ECHO_CLOCK( 23 )
{
int l = CALC_FIR( 1, 0 ) + CALC_FIR( 2, 0 );
int r = CALC_FIR( 1, 1 ) + CALC_FIR( 2, 1 );
m.t_echo_in [0] += l;
m.t_echo_in [1] += r;
echo_read( 1 );
}
ECHO_CLOCK( 24 )
{
int l = CALC_FIR( 3, 0 ) + CALC_FIR( 4, 0 ) + CALC_FIR( 5, 0 );
int r = CALC_FIR( 3, 1 ) + CALC_FIR( 4, 1 ) + CALC_FIR( 5, 1 );
m.t_echo_in [0] += l;
m.t_echo_in [1] += r;
}
ECHO_CLOCK( 25 )
{
int l = m.t_echo_in [0] + CALC_FIR( 6, 0 );
int r = m.t_echo_in [1] + CALC_FIR( 6, 1 );
l = (int16_t) l;
r = (int16_t) r;
l += (int16_t) CALC_FIR( 7, 0 );
r += (int16_t) CALC_FIR( 7, 1 );
CLAMP16( l );
CLAMP16( r );
m.t_echo_in [0] = l & ~1;
m.t_echo_in [1] = r & ~1;
}
inline int SPC_DSP::echo_output( int ch )
{
int out = (int16_t) ((m.t_main_out [ch] * (int8_t) REG(mvoll + ch * 0x10)) >> 7) +
(int16_t) ((m.t_echo_in [ch] * (int8_t) REG(evoll + ch * 0x10)) >> 7);
CLAMP16( out );
return out;
}
ECHO_CLOCK( 26 )
{
// Left output volumes
// (save sample for next clock so we can output both together)
m.t_main_out [0] = echo_output( 0 );
// Echo feedback
int l = m.t_echo_out [0] + (int16_t) ((m.t_echo_in [0] * (int8_t) REG(efb)) >> 7);
int r = m.t_echo_out [1] + (int16_t) ((m.t_echo_in [1] * (int8_t) REG(efb)) >> 7);
CLAMP16( l );
CLAMP16( r );
m.t_echo_out [0] = l & ~1;
m.t_echo_out [1] = r & ~1;
}
ECHO_CLOCK( 27 )
{
// Output
int l = m.t_main_out [0];
int r = echo_output( 1 );
m.t_main_out [0] = 0;
m.t_main_out [1] = 0;
// TODO: global muting isn't this simple (turns DAC on and off
// or something, causing small ~37-sample pulse when first muted)
if ( REG(flg) & 0x40 )
{
l = 0;
r = 0;
}
// Output sample to DAC
#ifdef SPC_DSP_OUT_HOOK
SPC_DSP_OUT_HOOK( l, r );
#else
sample_t* out = m.out;
WRITE_SAMPLES( l, r, out );
m.out = out;
#endif
}
ECHO_CLOCK( 28 )
{
m.t_echo_enabled = REG(flg);
}
inline void SPC_DSP::echo_write( int ch )
{
if ( !(m.t_echo_enabled & 0x20) )
SET_LE16A( ECHO_PTR( ch ), m.t_echo_out [ch] );
m.t_echo_out [ch] = 0;
}
ECHO_CLOCK( 29 )
{
m.t_esa = REG(esa);
if ( !m.echo_offset )
m.echo_length = (REG(edl) & 0x0F) * 0x800;
m.echo_offset += 4;
if ( m.echo_offset >= m.echo_length )
m.echo_offset = 0;
// Write left echo
echo_write( 0 );
m.t_echo_enabled = REG(flg);
}
ECHO_CLOCK( 30 )
{
// Write right echo
echo_write( 1 );
}
//// Timing
// Execute clock for a particular voice
#define V( clock, voice ) voice_##clock( &m.voices [voice] );
/* The most common sequence of clocks uses composite operations
for efficiency. For example, the following are equivalent to the
individual steps on the right:
V(V7_V4_V1,2) -> V(V7,2) V(V4,3) V(V1,5)
V(V8_V5_V2,2) -> V(V8,2) V(V5,3) V(V2,4)
V(V9_V6_V3,2) -> V(V9,2) V(V6,3) V(V3,4) */
// Voice 0 1 2 3 4 5 6 7
#define GEN_DSP_TIMING \
PHASE( 0) V(V5,0)V(V2,1)\
PHASE( 1) V(V6,0)V(V3,1)\
PHASE( 2) V(V7_V4_V1,0)\
PHASE( 3) V(V8_V5_V2,0)\
PHASE( 4) V(V9_V6_V3,0)\
PHASE( 5) V(V7_V4_V1,1)\
PHASE( 6) V(V8_V5_V2,1)\
PHASE( 7) V(V9_V6_V3,1)\
PHASE( 8) V(V7_V4_V1,2)\
PHASE( 9) V(V8_V5_V2,2)\
PHASE(10) V(V9_V6_V3,2)\
PHASE(11) V(V7_V4_V1,3)\
PHASE(12) V(V8_V5_V2,3)\
PHASE(13) V(V9_V6_V3,3)\
PHASE(14) V(V7_V4_V1,4)\
PHASE(15) V(V8_V5_V2,4)\
PHASE(16) V(V9_V6_V3,4)\
PHASE(17) V(V1,0) V(V7,5)V(V4,6)\
PHASE(18) V(V8_V5_V2,5)\
PHASE(19) V(V9_V6_V3,5)\
PHASE(20) V(V1,1) V(V7,6)V(V4,7)\
PHASE(21) V(V8,6)V(V5,7) V(V2,0) /* t_brr_next_addr order dependency */\
PHASE(22) V(V3a,0) V(V9,6)V(V6,7) echo_22();\
PHASE(23) V(V7,7) echo_23();\
PHASE(24) V(V8,7) echo_24();\
PHASE(25) V(V3b,0) V(V9,7) echo_25();\
PHASE(26) echo_26();\
PHASE(27) misc_27(); echo_27();\
PHASE(28) misc_28(); echo_28();\
PHASE(29) misc_29(); echo_29();\
PHASE(30) misc_30();V(V3c,0) echo_30();\
PHASE(31) V(V4,0) V(V1,2)\
#if !SPC_DSP_CUSTOM_RUN
void SPC_DSP::run( int clocks_remain )
{
require( clocks_remain > 0 );
int const phase = m.phase;
m.phase = (phase + clocks_remain) & 31;
switch ( phase )
{
loop:
#define PHASE( n ) if ( n && !--clocks_remain ) break; case n:
GEN_DSP_TIMING
#undef PHASE
if ( --clocks_remain )
goto loop;
}
}
#endif
//// Setup
void SPC_DSP::init( void* ram_64k )
{
m.ram = (uint8_t*) ram_64k;
mute_voices( 0 );
disable_surround( false );
set_output( 0, 0 );
reset();
stereo_switch = 0xffff;
take_spc_snapshot = 0;
spc_snapshot_callback = 0;
#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();
for (int i = 0; i < voice_count; i++)
m.voices[i].voice_number = i;
}
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
for ( int i = voice_count; --i >= 0; )
{
voice_t* v = &m.voices [i];
v->brr_offset = 1;
v->vbit = 1 << i;
v->regs = &m.regs [i * 0x10];
}
m.new_kon = REG(kon);
m.t_dir = REG(dir);
m.t_esa = REG(esa);
soft_reset_common();
}
void SPC_DSP::reset() { load( initial_regs ); }
//// State save/load
#if !SPC_NO_COPY_STATE_FUNCS
void SPC_State_Copier::copy( void* state, size_t size )
{
func( buf, state, size );
}
int SPC_State_Copier::copy_int( int state, int size )
{
BOOST::uint8_t s [2];
SET_LE16( s, state );
func( buf, &s, size );
return GET_LE16( s );
}
void SPC_State_Copier::skip( int count )
{
if ( count > 0 )
{
char temp [64];
memset( temp, 0, sizeof temp );
do
{
int n = sizeof temp;
if ( n > count )
n = count;
count -= n;
func( buf, temp, n );
}
while ( count );
}
}
void SPC_State_Copier::extra()
{
int n = 0;
SPC_State_Copier& copier = *this;
SPC_COPY( uint8_t, n );
skip( n );
}
void SPC_DSP::copy_state( unsigned char** io, copy_func_t copy )
{
SPC_State_Copier copier( io, copy );
// DSP registers
copier.copy( m.regs, register_count );
// Internal state
// Voices
int i;
for ( i = 0; i < voice_count; i++ )
{
voice_t* v = &m.voices [i];
// BRR buffer
int i;
for ( i = 0; i < brr_buf_size; i++ )
{
int s = v->buf [i];
SPC_COPY( int16_t, s );
v->buf [i] = v->buf [i + brr_buf_size] = s;
}
SPC_COPY( uint16_t, v->interp_pos );
SPC_COPY( uint16_t, v->brr_addr );
SPC_COPY( uint16_t, v->env );
SPC_COPY( int16_t, v->hidden_env );
SPC_COPY( uint8_t, v->buf_pos );
SPC_COPY( uint8_t, v->brr_offset );
SPC_COPY( uint8_t, v->kon_delay );
{
int m = v->env_mode;
SPC_COPY( uint8_t, m );
v->env_mode = (enum env_mode_t) m;
}
SPC_COPY( uint8_t, v->t_envx_out );
copier.extra();
}
// Echo history
for ( i = 0; i < echo_hist_size; i++ )
{
int j;
for ( j = 0; j < 2; j++ )
{
int s = m.echo_hist_pos [i] [j];
SPC_COPY( int16_t, s );
m.echo_hist [i] [j] = s; // write back at offset 0
}
}
m.echo_hist_pos = m.echo_hist;
memcpy( &m.echo_hist [echo_hist_size], m.echo_hist, echo_hist_size * sizeof m.echo_hist [0] );
// Misc
SPC_COPY( uint8_t, m.every_other_sample );
SPC_COPY( uint8_t, m.kon );
SPC_COPY( uint16_t, m.noise );
SPC_COPY( uint16_t, m.counter );
SPC_COPY( uint16_t, m.echo_offset );
SPC_COPY( uint16_t, m.echo_length );
SPC_COPY( uint8_t, m.phase );
SPC_COPY( uint8_t, m.new_kon );
SPC_COPY( uint8_t, m.endx_buf );
SPC_COPY( uint8_t, m.envx_buf );
SPC_COPY( uint8_t, m.outx_buf );
SPC_COPY( uint8_t, m.t_pmon );
SPC_COPY( uint8_t, m.t_non );
SPC_COPY( uint8_t, m.t_eon );
SPC_COPY( uint8_t, m.t_dir );
SPC_COPY( uint8_t, m.t_koff );
SPC_COPY( uint16_t, m.t_brr_next_addr );
SPC_COPY( uint8_t, m.t_adsr0 );
SPC_COPY( uint8_t, m.t_brr_header );
SPC_COPY( uint8_t, m.t_brr_byte );
SPC_COPY( uint8_t, m.t_srcn );
SPC_COPY( uint8_t, m.t_esa );
SPC_COPY( uint8_t, m.t_echo_enabled );
SPC_COPY( int16_t, m.t_main_out [0] );
SPC_COPY( int16_t, m.t_main_out [1] );
SPC_COPY( int16_t, m.t_echo_out [0] );
SPC_COPY( int16_t, m.t_echo_out [1] );
SPC_COPY( int16_t, m.t_echo_in [0] );
SPC_COPY( int16_t, m.t_echo_in [1] );
SPC_COPY( uint16_t, m.t_dir_addr );
SPC_COPY( uint16_t, m.t_pitch );
SPC_COPY( int16_t, m.t_output );
SPC_COPY( uint16_t, m.t_echo_ptr );
SPC_COPY( uint8_t, m.t_looped );
copier.extra();
}
#endif
//// Snes9x Accessor
void SPC_DSP::set_spc_snapshot_callback( void (*callback) (void) )
{
spc_snapshot_callback = callback;
}
void SPC_DSP::dump_spc_snapshot( void )
{
take_spc_snapshot = 1;
}
void SPC_DSP::set_stereo_switch( int value )
{
stereo_switch = value;
}
SPC_DSP::uint8_t SPC_DSP::reg_value( int ch, int addr )
{
return m.voices[ch].regs[addr];
}
int SPC_DSP::envx_value( int ch )
{
return m.voices[ch].env;
}
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