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// Game_Music_Emu 0.5.2. 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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