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rockbox/apps/codecs/libspc/spc_dsp.c
Nils Wallménius d502086e7f Fix yellow by making the ifdef hell slightly worse 
git-svn-id: svn://svn.rockbox.org/rockbox/trunk@23790 a1c6a512-1295-4272-9138-f99709370657
2009-11-29 21:22:23 +00:00

1276 lines
50 KiB
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/***************************************************************************
* __________ __ ___.
* Open \______ \ ____ ____ | | _\_ |__ _______ ___
* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
* \/ \/ \/ \/ \/
* $Id$
*
* Copyright (C) 2007-2008 Michael Sevakis (jhMikeS)
* Copyright (C) 2006-2007 Adam Gashlin (hcs)
* Copyright (C) 2004-2007 Shay Green (blargg)
* Copyright (C) 2002 Brad Martin
*
* This program is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public License
* as published by the Free Software Foundation; either version 2
* of the License, or (at your option) any later version.
*
* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
* KIND, either express or implied.
*
****************************************************************************/
/* The DSP portion (awe!) */
#include "codeclib.h"
#include "spc_codec.h"
#include "spc_profiler.h"
#if defined(CPU_COLDFIRE) || defined (CPU_ARM)
int32_t fir_buf[FIR_BUF_CNT]
__attribute__ ((aligned (FIR_BUF_ALIGN*1))) IBSS_ATTR;
#endif
#if SPC_BRRCACHE
/* a little extra for samples that go past end */
int16_t BRRcache [BRR_CACHE_SIZE] CACHEALIGN_ATTR;
#endif
void DSP_write( struct Spc_Dsp* this, int i, int data )
{
assert( (unsigned) i < REGISTER_COUNT );
this->r.reg [i] = data;
int high = i >> 4;
int low = i & 0x0F;
if ( low < 2 ) /* voice volumes */
{
int left = *(int8_t const*) &this->r.reg [i & ~1];
int right = *(int8_t const*) &this->r.reg [i | 1];
struct voice_t* v = this->voice_state + high;
v->volume [0] = left;
v->volume [1] = right;
}
else if ( low == 0x0F ) /* fir coefficients */
{
this->fir_coeff [7 - high] = (int8_t) data; /* sign-extend */
}
}
/* if ( n < -32768 ) out = -32768; */
/* if ( n > 32767 ) out = 32767; */
#define CLAMP16( n ) \
({ \
if ( (int16_t) n != n ) \
n = 0x7FFF ^ (n >> 31); \
n; \
})
#if SPC_BRRCACHE
static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
struct voice_t* voice,
struct raw_voice_t const* const raw_voice ) ICODE_ATTR;
static void decode_brr( struct Spc_Dsp* this, unsigned start_addr,
struct voice_t* voice,
struct raw_voice_t const* const raw_voice )
{
/* setup same variables as where decode_brr() is called from */
#undef RAM
#define RAM ram.ram
struct src_dir const* const sd =
&ram.sd[this->r.g.wave_page * 0x100/sizeof(struct src_dir)];
struct cache_entry_t* const wave_entry =
&this->wave_entry [raw_voice->waveform];
/* the following block can be put in place of the call to
decode_brr() below
*/
{
DEBUGF( "decode at %08x (wave #%d)\n",
start_addr, raw_voice->waveform );
/* see if in cache */
int i;
for ( i = 0; i < this->oldsize; i++ )
{
struct cache_entry_t* e = &this->wave_entry_old [i];
if ( e->start_addr == start_addr )
{
DEBUGF( "found in wave_entry_old (oldsize=%d)\n",
this->oldsize );
*wave_entry = *e;
goto wave_in_cache;
}
}
wave_entry->start_addr = start_addr;
uint8_t const* const loop_ptr =
RAM + letoh16(sd[raw_voice->waveform].loop);
short* loop_start = 0;
short* out = BRRcache + start_addr * 2;
wave_entry->samples = out;
*out++ = 0;
int smp1 = 0;
int smp2 = 0;
uint8_t const* addr = RAM + start_addr;
int block_header;
do
{
if ( addr == loop_ptr )
{
loop_start = out;
DEBUGF( "loop at %08lx (wave #%d)\n",
(unsigned long)(addr - RAM), raw_voice->waveform );
}
/* header */
block_header = *addr;
addr += 9;
voice->addr = addr;
int const filter = (block_header & 0x0C) - 0x08;
/* scaling
(invalid scaling gives -4096 for neg nybble, 0 for pos) */
static unsigned char const right_shifts [16] = {
5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29,
};
static unsigned char const left_shifts [16] = {
0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11
};
int const scale = block_header >> 4;
int const right_shift = right_shifts [scale];
int const left_shift = left_shifts [scale];
/* output position */
out += BRR_BLOCK_SIZE;
int offset = -BRR_BLOCK_SIZE << 2;
do /* decode and filter 16 samples */
{
/* Get nybble, sign-extend, then scale
get byte, select which nybble, sign-extend, then shift based
on scaling. also handles invalid scaling values. */
int delta = (int) (int8_t) (addr [offset >> 3] << (offset & 4))
>> right_shift << left_shift;
out [offset >> 2] = smp2;
if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */
{
delta -= smp2 >> 1;
delta += smp2 >> 5;
smp2 = smp1;
delta += smp1;
delta += (-smp1 - (smp1 >> 1)) >> 5;
}
else
{
if ( filter == -4 ) /* mode 0x04 */
{
delta += smp1 >> 1;
delta += (-smp1) >> 5;
}
else if ( filter > -4 ) /* mode 0x0C */
{
delta -= smp2 >> 1;
delta += (smp2 + (smp2 >> 1)) >> 4;
delta += smp1;
delta += (-smp1 * 13) >> 7;
}
smp2 = smp1;
}
delta = CLAMP16( delta );
smp1 = (int16_t) (delta * 2); /* sign-extend */
}
while ( (offset += 4) != 0 );
/* if next block has end flag set, this block ends early */
/* (verified) */
if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
{
/* skip last 9 samples */
out -= 9;
goto early_end;
}
}
while ( !(block_header & 1) && addr < RAM + 0x10000 );
out [0] = smp2;
out [1] = smp1;
early_end:
wave_entry->end = (out - 1 - wave_entry->samples) << 12;
wave_entry->loop = 0;
if ( (block_header & 2) )
{
if ( loop_start )
{
int loop = out - loop_start;
wave_entry->loop = loop;
wave_entry->end += 0x3000;
out [2] = loop_start [2];
out [3] = loop_start [3];
out [4] = loop_start [4];
}
else
{
DEBUGF( "loop point outside initial wave\n" );
}
}
DEBUGF( "end at %08lx (wave #%d)\n",
(unsigned long)(addr - RAM), raw_voice->waveform );
/* add to cache */
this->wave_entry_old [this->oldsize++] = *wave_entry;
wave_in_cache:;
}
}
#endif
static void 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) ICODE_ATTR;
static void 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) {
#undef RAM
#define RAM ram.ram
int const env_rate_init = 0x7800;
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? */
unsigned start_addr = letoh16(sd[raw_voice->waveform].start);
#if !SPC_BRRCACHE
{
voice->addr = RAM + start_addr;
/* BRR filter uses previous samples */
voice->samples [BRR_BLOCK_SIZE + 1] = 0;
voice->samples [BRR_BLOCK_SIZE + 2] = 0;
/* decode three samples immediately */
voice->position = (BRR_BLOCK_SIZE + 3) * 0x1000 - 1;
voice->block_header = 0; /* "previous" BRR header */
}
#else
{
voice->position = 3 * 0x1000 - 1;
struct cache_entry_t* const wave_entry =
&this->wave_entry [raw_voice->waveform];
/* predecode BRR if not already */
if ( wave_entry->start_addr != start_addr )
{
/* the following line can be replaced by the indicated block
in decode_brr() */
decode_brr( this, start_addr, voice, raw_voice );
}
voice->samples = wave_entry->samples;
voice->wave_end = wave_entry->end;
voice->wave_loop = wave_entry->loop;
}
#endif
}
}
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
#if 0
EXIT_TIMER(cpu);
ENTER_TIMER(dsp);
#endif
/* 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)];
#ifdef ROCKBOX_BIG_ENDIAN
/* Convert endiannesses before entering loops - these
get used alot */
const uint32_t rates[VOICE_COUNT] =
{
GET_LE16A( this->r.voice[0].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[1].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[2].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[3].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[4].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[5].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[6].rate ) & 0x3FFF,
GET_LE16A( this->r.voice[7].rate ) & 0x3FFF,
};
#define VOICE_RATE(x) *(x)
#define IF_RBE(...) __VA_ARGS__
#ifdef CPU_COLDFIRE
/* Initialize mask register with the buffer address mask */
asm volatile ("move.l %[m], %%mask" : : [m]"i"(FIR_BUF_MASK));
const int echo_wrap = (this->r.g.echo_delay & 15) * 0x800;
const int echo_start = this->r.g.echo_page * 0x100;
#endif /* CPU_COLDFIRE */
#else
#define VOICE_RATE(x) (GET_LE16(raw_voice->rate) & 0x3FFF)
#define IF_RBE(...)
#endif /* ROCKBOX_BIG_ENDIAN */
#if !SPC_NOINTERP
int const slow_gaussian = (this->r.g.pitch_mods >> 1) |
this->r.g.noise_enables;
#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;
/* each rate divides exactly into 0x7800 without remainder */
int const env_rate_init = 0x7800;
static unsigned short const env_rates [0x20] ICONST_ATTR =
{
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
};
do /* one pair of output samples per iteration */
{
/* Noise */
if ( this->r.g.noise_enables )
{
if ( (this->noise_count -=
env_rates [this->r.g.flags & 0x1F]) <= 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;
int echo_1 = 0;
#endif
long prev_outx = 0; /* TODO: correct value for first channel? */
int chans_0 = 0;
int 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;
int vbit = 1;
IF_RBE( const uint32_t* vr = rates; )
for ( ; vbit < 0x100; vbit <<= 1, ++voice, ++raw_voice IF_RBE( , ++vr ) )
{
/* pregen involves checking keyon, etc */
#if 0
ENTER_TIMER(dsp_pregen);
#endif
/* Key on events are delayed */
int key_on_delay = voice->key_on_delay;
if ( --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 ( env_mode != state_release ) /* 99% of the time */
{
env_timer = voice->env_timer;
if ( adsr0 & 0x80 ) /* 79% of the time */
{
int adsr1 = raw_voice->adsr [1];
if ( 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:;
}
#if 0
EXIT_TIMER(dsp_pregen);
ENTER_TIMER(dsp_gen);
#endif
#if !SPC_BRRCACHE
/* Decode BRR block */
if ( voice->position >= BRR_BLOCK_SIZE * 0x1000 )
{
voice->position -= BRR_BLOCK_SIZE * 0x1000;
uint8_t const* addr = voice->addr;
if ( addr >= RAM + 0x10000 )
addr -= 0x10000;
/* action based on previous block's header */
if ( voice->block_header & 1 )
{
addr = RAM + letoh16(sd[raw_voice->waveform].loop);
this->r.g.wave_ended |= vbit;
if ( !(voice->block_header & 2) ) /* 1% of the time */
{
/* first block was end block;
don't play anything (verified) */
/* 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;
goto skip_decode;
}
}
/* header */
int const block_header = *addr;
addr += 9;
voice->addr = addr;
voice->block_header = block_header;
int const filter = (block_header & 0x0C) - 0x08;
/* scaling (invalid scaling gives -4096 for neg nybble,
0 for pos) */
static unsigned char const right_shifts [16] = {
5, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 29, 29, 29,
};
static unsigned char const left_shifts [16] = {
0, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 11, 11, 11
};
int const scale = block_header >> 4;
int const right_shift = right_shifts [scale];
int const left_shift = left_shifts [scale];
/* previous samples */
int smp2 = voice->samples [BRR_BLOCK_SIZE + 1];
int smp1 = voice->samples [BRR_BLOCK_SIZE + 2];
voice->samples [0] = voice->samples [BRR_BLOCK_SIZE];
/* output position */
short* out = voice->samples + (1 + BRR_BLOCK_SIZE);
int offset = -BRR_BLOCK_SIZE << 2;
/* if next block has end flag set,
this block ends early (verified) */
if ( (block_header & 3) != 3 && (*addr & 3) == 1 )
{
/* arrange for last 9 samples to be skipped */
int const skip = 9;
out += (skip & 1);
voice->samples [skip] = voice->samples [BRR_BLOCK_SIZE];
voice->position += skip * 0x1000;
offset = (-BRR_BLOCK_SIZE + (skip & ~1)) << 2;
addr -= skip / 2;
/* force sample to end on next decode */
voice->block_header = 1;
}
do /* decode and filter 16 samples */
{
/* Get nybble, sign-extend, then scale
get byte, select which nybble, sign-extend, then shift
based on scaling. also handles invalid scaling values.*/
int delta = (int) (int8_t) (addr [offset >> 3] <<
(offset & 4)) >> right_shift << left_shift;
out [offset >> 2] = smp2;
if ( filter == 0 ) /* mode 0x08 (30-90% of the time) */
{
delta -= smp2 >> 1;
delta += smp2 >> 5;
smp2 = smp1;
delta += smp1;
delta += (-smp1 - (smp1 >> 1)) >> 5;
}
else
{
if ( filter == -4 ) /* mode 0x04 */
{
delta += smp1 >> 1;
delta += (-smp1) >> 5;
}
else if ( filter > -4 ) /* mode 0x0C */
{
delta -= smp2 >> 1;
delta += (smp2 + (smp2 >> 1)) >> 4;
delta += smp1;
delta += (-smp1 * 13) >> 7;
}
smp2 = smp1;
}
delta = CLAMP16( delta );
smp1 = (int16_t) (delta * 2); /* sign-extend */
}
while ( (offset += 4) != 0 );
out [0] = smp2;
out [1] = smp1;
skip_decode:;
}
#endif
/* Get rate (with possible modulation) */
int rate = VOICE_RATE(vr);
if ( this->r.g.pitch_mods & vbit )
rate = (rate * (prev_outx + 32768)) >> 15;
#if !SPC_NOINTERP
/* Interleved gauss table (to improve cache coherency). */
/* gauss [i * 2 + j] = normal_gauss [(1 - j) * 256 + i] */
static short const 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,
};
/* Gaussian interpolation using most recent 4 samples */
long position = voice->position;
voice->position += rate;
short const* interp = voice->samples + (position >> 12);
int offset = position >> 4 & 0xFF;
/* Only left half of gaussian kernel is in table, so we must mirror
for right half */
short const* fwd = gauss + offset * 2;
short const* rev = gauss + 510 - offset * 2;
/* Use faster gaussian interpolation when exact result isn't needed
by pitch modulator of next channel */
int amp_0, amp_1;
if ( !(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. */
int output = (((fwd [0] * interp [0] +
fwd [1] * interp [1] +
rev [1] * interp [2] +
rev [0] * interp [3] ) >> 11) * voice->envx) >> 11;
/* duplicated here to give compiler more to run in parallel */
amp_0 = voice->volume [0] * output;
amp_1 = voice->volume [1] * output;
raw_voice->outx = output >> 8;
}
else
{
int output = *(int16_t*) &this->noise;
if ( !(this->r.g.noise_enables & vbit) )
{
output = (fwd [0] * interp [0]) & ~0xFFF;
output = (output + fwd [1] * interp [1]) & ~0xFFF;
output = (output + rev [1] * interp [2]) >> 12;
output = (int16_t) (output * 2);
output += ((rev [0] * interp [3]) >> 12) * 2;
output = CLAMP16( output );
}
output = (output * voice->envx) >> 11 & ~1;
/* duplicated here to give compiler more to run in parallel */
amp_0 = voice->volume [0] * output;
amp_1 = voice->volume [1] * output;
prev_outx = output;
raw_voice->outx = (int8_t) (output >> 8);
}
#else /* SPCNOINTERP */
/* two-point linear interpolation */
#ifdef CPU_COLDFIRE
int amp_0 = (int16_t)this->noise;
int amp_1;
if ( (this->r.g.noise_enables & vbit) == 0 )
{
uint32_t f = voice->position;
int32_t y0;
/**
* Formula (fastest found so far of MANY):
* output = y0 + f*y1 - f*y0
*/
asm volatile (
/* separate fractional and whole parts */
"move.l %[f], %[y1] \r\n"
"and.l #0xfff, %[f] \r\n"
"lsr.l %[sh], %[y1] \r\n"
/* load samples y0 (upper) & y1 (lower) */
"move.l 2(%[s], %[y1].l*2), %[y1] \r\n"
/* %acc0 = f*y1 */
"mac.w %[f]l, %[y1]l, %%acc0 \r\n"
/* %acc0 -= f*y0 */
"msac.w %[f]l, %[y1]u, %%acc0 \r\n"
/* separate out y0 and sign extend */
"swap %[y1] \r\n"
"movea.w %[y1], %[y0] \r\n"
/* fetch result, scale down and add y0 */
"movclr.l %%acc0, %[y1] \r\n"
/* output = y0 + (result >> 12) */
"asr.l %[sh], %[y1] \r\n"
"add.l %[y0], %[y1] \r\n"
: [f]"+d"(f), [y0]"=&a"(y0), [y1]"=&d"(amp_0)
: [s]"a"(voice->samples), [sh]"d"(12)
);
}
/* apply voice envelope to output */
asm volatile (
"mac.w %[output]l, %[envx]l, %%acc0 \r\n"
:
: [output]"r"(amp_0), [envx]"r"(voice->envx)
);
/* advance voice position */
voice->position += rate;
/* fetch output, scale and apply left and right
voice volume */
asm volatile (
"movclr.l %%acc0, %[output] \r\n"
"asr.l %[sh], %[output] \r\n"
"mac.l %[vvol_0], %[output], %%acc0 \r\n"
"mac.l %[vvol_1], %[output], %%acc1 \r\n"
: [output]"=&d"(amp_0)
: [vvol_0]"r"((int)voice->volume[0]),
[vvol_1]"r"((int)voice->volume[1]),
[sh]"d"(11)
);
/* save this output into previous, scale and save in
output register */
prev_outx = amp_0;
raw_voice->outx = amp_0 >> 8;
/* fetch final voice output */
asm volatile (
"movclr.l %%acc0, %[amp_0] \r\n"
"movclr.l %%acc1, %[amp_1] \r\n"
: [amp_0]"=r"(amp_0), [amp_1]"=r"(amp_1)
);
#elif defined (CPU_ARM)
int amp_0, amp_1;
if ( (this->r.g.noise_enables & vbit) != 0 ) {
amp_0 = *(int16_t *)&this->noise;
} else {
uint32_t f = voice->position;
amp_0 = (uint32_t)voice->samples;
asm volatile(
"mov %[y1], %[f], lsr #12 \r\n"
"eor %[f], %[f], %[y1], lsl #12 \r\n"
"add %[y1], %[y0], %[y1], lsl #1 \r\n"
"ldrsh %[y0], [%[y1], #2] \r\n"
"ldrsh %[y1], [%[y1], #4] \r\n"
"sub %[y1], %[y1], %[y0] \r\n"
"mul %[f], %[y1], %[f] \r\n"
"add %[y0], %[y0], %[f], asr #12 \r\n"
: [f]"+r"(f), [y0]"+r"(amp_0), [y1]"=&r"(amp_1)
);
}
voice->position += rate;
asm volatile(
"mul %[amp_1], %[amp_0], %[envx] \r\n"
"mov %[amp_0], %[amp_1], asr #11 \r\n"
"mov %[amp_1], %[amp_0], asr #8 \r\n"
: [amp_0]"+r"(amp_0), [amp_1]"=&r"(amp_1)
: [envx]"r"(voice->envx)
);
prev_outx = amp_0;
raw_voice->outx = (int8_t)amp_1;
asm volatile(
"mul %[amp_1], %[amp_0], %[vol_1] \r\n"
"mul %[amp_0], %[vol_0], %[amp_0] \r\n"
: [amp_0]"+r"(amp_0), [amp_1]"+r"(amp_1)
: [vol_0]"r"((int)voice->volume[0]),
[vol_1]"r"((int)voice->volume[1])
);
#else /* Unoptimized CPU */
int output;
if ( (this->r.g.noise_enables & vbit) == 0 )
{
int const fraction = voice->position & 0xfff;
short const* const pos = (voice->samples + (voice->position >> 12)) + 1;
output = pos[0] + ((fraction * (pos[1] - pos[0])) >> 12);
} else {
output = *(int16_t *)&this->noise;
}
voice->position += rate;
output = (output * voice->envx) >> 11;
/* duplicated here to give compiler more to run in parallel */
int amp_0 = voice->volume [0] * output;
int amp_1 = voice->volume [1] * output;
prev_outx = output;
raw_voice->outx = (int8_t) (output >> 8);
#endif /* CPU_* */
#endif /* SPCNOINTERP */
#if SPC_BRRCACHE
if ( voice->position >= voice->wave_end )
{
long loop_len = voice->wave_loop << 12;
voice->position -= loop_len;
this->r.g.wave_ended |= vbit;
if ( !loop_len )
{
this->keys_down ^= vbit;
raw_voice->envx = 0;
voice->envx = 0;
}
}
#endif
#if 0
EXIT_TIMER(dsp_gen);
ENTER_TIMER(dsp_mix);
#endif
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
#if 0
EXIT_TIMER(dsp_mix);
#endif
}
/* end of voice loop */
#if !SPC_NOECHO
#ifdef CPU_COLDFIRE
/* 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_wrap )
echo_pos = 0;
this->echo_pos = echo_pos;
int fb = swap_odd_even32(*(int32_t *)echo_ptr);
int out_0, out_1;
/* Keep last 8 samples */
*this->last_fir_ptr = fb;
this->last_fir_ptr = this->fir_ptr;
/* Apply echo FIR filter to output samples read from echo buffer -
circular buffer is hardware incremented and masked; FIR
coefficients and buffer history are loaded in parallel with
multiply accumulate operations. Shift left by one here and once
again when calculating feedback to have sample values justified
to bit 31 in the output to ease endian swap, interleaving and
clamping before placing result in the program's echo buffer. */
int _0, _1, _2;
asm volatile (
"move.l (%[fir_c]) , %[_2] \r\n"
"mac.w %[fb]u, %[_2]u, <<, (%[fir_p])+&, %[_0], %%acc0 \r\n"
"mac.w %[fb]l, %[_2]u, <<, (%[fir_p])& , %[_1], %%acc1 \r\n"
"mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
"mac.w %[_0]l, %[_2]l, <<, 4(%[fir_c]) , %[_2], %%acc1 \r\n"
"mac.w %[_1]u, %[_2]u, <<, 4(%[fir_p])& , %[_0], %%acc0 \r\n"
"mac.w %[_1]l, %[_2]u, <<, 8(%[fir_p])& , %[_1], %%acc1 \r\n"
"mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
"mac.w %[_0]l, %[_2]l, <<, 8(%[fir_c]) , %[_2], %%acc1 \r\n"
"mac.w %[_1]u, %[_2]u, <<, 12(%[fir_p])& , %[_0], %%acc0 \r\n"
"mac.w %[_1]l, %[_2]u, <<, 16(%[fir_p])& , %[_1], %%acc1 \r\n"
"mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
"mac.w %[_0]l, %[_2]l, <<, 12(%[fir_c]) , %[_2], %%acc1 \r\n"
"mac.w %[_1]u, %[_2]u, <<, 20(%[fir_p])& , %[_0], %%acc0 \r\n"
"mac.w %[_1]l, %[_2]u, << , %%acc1 \r\n"
"mac.w %[_0]u, %[_2]l, << , %%acc0 \r\n"
"mac.w %[_0]l, %[_2]l, << , %%acc1 \r\n"
: [_0]"=&r"(_0), [_1]"=&r"(_1), [_2]"=&r"(_2),
[fir_p]"+a"(this->fir_ptr)
: [fir_c]"a"(this->fir_coeff), [fb]"r"(fb)
);
/* Generate output */
asm volatile (
/* fetch filter results _after_ gcc loads asm
block parameters to eliminate emac stalls */
"movclr.l %%acc0, %[out_0] \r\n"
"movclr.l %%acc1, %[out_1] \r\n"
/* apply global volume */
"mac.l %[chans_0], %[gv_0] , %%acc2 \r\n"
"mac.l %[chans_1], %[gv_1] , %%acc3 \r\n"
/* apply echo volume and add to final output */
"mac.l %[ev_0], %[out_0], >>, %%acc2 \r\n"
"mac.l %[ev_1], %[out_1], >>, %%acc3 \r\n"
: [out_0]"=&r"(out_0), [out_1]"=&r"(out_1)
: [chans_0]"r"(chans_0), [gv_0]"r"(global_vol_0),
[ev_0]"r"((int)this->r.g.echo_volume_0),
[chans_1]"r"(chans_1), [gv_1]"r"(global_vol_1),
[ev_1]"r"((int)this->r.g.echo_volume_1)
);
/* Feedback into echo buffer */
if ( !(this->r.g.flags & 0x20) )
{
asm volatile (
/* scale echo voices; saturate if overflow */
"mac.l %[sh], %[e1] , %%acc1 \r\n"
"mac.l %[sh], %[e0] , %%acc0 \r\n"
/* add scaled output from FIR filter */
"mac.l %[out_1], %[ef], <<, %%acc1 \r\n"
"mac.l %[out_0], %[ef], <<, %%acc0 \r\n"
/* swap and fetch feedback results - simply
swap_odd_even32 mixed in between macs and
movclrs to mitigate stall issues */
"move.l #0x00ff00ff, %[sh] \r\n"
"movclr.l %%acc1, %[e1] \r\n"
"swap %[e1] \r\n"
"movclr.l %%acc0, %[e0] \r\n"
"move.w %[e1], %[e0] \r\n"
"and.l %[e0], %[sh] \r\n"
"eor.l %[sh], %[e0] \r\n"
"lsl.l #8, %[sh] \r\n"
"lsr.l #8, %[e0] \r\n"
"or.l %[sh], %[e0] \r\n"
/* save final feedback into echo buffer */
"move.l %[e0], (%[echo_ptr]) \r\n"
: [e0]"+d"(echo_0), [e1]"+d"(echo_1)
: [out_0]"r"(out_0), [out_1]"r"(out_1),
[ef]"r"((int)this->r.g.echo_feedback),
[echo_ptr]"a"((int32_t *)echo_ptr),
[sh]"d"(1 << 9)
);
}
/* Output final samples */
asm volatile (
/* fetch output saved in %acc2 and %acc3 */
"movclr.l %%acc2, %[out_0] \r\n"
"movclr.l %%acc3, %[out_1] \r\n"
/* scale right by global_muting shift */
"asr.l %[gm], %[out_0] \r\n"
"asr.l %[gm], %[out_1] \r\n"
: [out_0]"=&d"(out_0), [out_1]"=&d"(out_1)
: [gm]"d"(global_muting)
);
out_buf [ 0] = out_0;
out_buf [WAV_CHUNK_SIZE] = out_1;
out_buf ++;
#elif defined (CPU_ARM)
/* Read feedback from echo buffer */
int echo_pos = this->echo_pos;
uint8_t* const echo_ptr = RAM +
((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF);
echo_pos += 4;
if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 )
echo_pos = 0;
this->echo_pos = echo_pos;
int fb_0 = GET_LE16SA( echo_ptr );
int fb_1 = GET_LE16SA( echo_ptr + 2 );
/* Keep last 8 samples */
int32_t *fir_ptr = this->fir_ptr;
/* Apply FIR */
asm volatile (
"str %[fb_0], [%[fir_p]], #4 \r\n"
"str %[fb_1], [%[fir_p]], #4 \r\n"
/* duplicate at +8 eliminates wrap checking below */
"str %[fb_0], [%[fir_p], #56] \r\n"
"str %[fb_1], [%[fir_p], #60] \r\n"
: [fir_p]"+r"(fir_ptr)
: [fb_0]"r"(fb_0), [fb_1]"r"(fb_1)
);
this->fir_ptr = (int32_t *)((intptr_t)fir_ptr & FIR_BUF_MASK);
int32_t *fir_coeff = this->fir_coeff;
asm volatile (
"ldmia %[fir_c]!, { r0-r1 } \r\n"
"ldmia %[fir_p]!, { r4-r5 } \r\n"
"mul %[fb_0], r0, %[fb_0] \r\n"
"mul %[fb_1], r0, %[fb_1] \r\n"
"mla %[fb_0], r4, r1, %[fb_0] \r\n"
"mla %[fb_1], r5, r1, %[fb_1] \r\n"
"ldmia %[fir_c]!, { r0-r1 } \r\n"
"ldmia %[fir_p]!, { r2-r5 } \r\n"
"mla %[fb_0], r2, r0, %[fb_0] \r\n"
"mla %[fb_1], r3, r0, %[fb_1] \r\n"
"mla %[fb_0], r4, r1, %[fb_0] \r\n"
"mla %[fb_1], r5, r1, %[fb_1] \r\n"
"ldmia %[fir_c]!, { r0-r1 } \r\n"
"ldmia %[fir_p]!, { r2-r5 } \r\n"
"mla %[fb_0], r2, r0, %[fb_0] \r\n"
"mla %[fb_1], r3, r0, %[fb_1] \r\n"
"mla %[fb_0], r4, r1, %[fb_0] \r\n"
"mla %[fb_1], r5, r1, %[fb_1] \r\n"
"ldmia %[fir_c]!, { r0-r1 } \r\n"
"ldmia %[fir_p]!, { r2-r5 } \r\n"
"mla %[fb_0], r2, r0, %[fb_0] \r\n"
"mla %[fb_1], r3, r0, %[fb_1] \r\n"
"mla %[fb_0], r4, r1, %[fb_0] \r\n"
"mla %[fb_1], r5, r1, %[fb_1] \r\n"
: [fb_0]"+r"(fb_0), [fb_1]"+r"(fb_1),
[fir_p]"+r"(fir_ptr), [fir_c]"+r"(fir_coeff)
:
: "r0", "r1", "r2", "r3", "r4", "r5"
);
/* Generate output */
int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
>> global_muting;
int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
>> global_muting;
out_buf [ 0] = amp_0;
out_buf [WAV_CHUNK_SIZE] = amp_1;
out_buf ++;
if ( !(this->r.g.flags & 0x20) )
{
/* Feedback into echo buffer */
int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
e0 = CLAMP16( e0 );
SET_LE16A( echo_ptr , e0 );
e1 = CLAMP16( e1 );
SET_LE16A( echo_ptr + 2, e1 );
}
#else /* Unoptimized CPU */
/* Read feedback from echo buffer */
int echo_pos = this->echo_pos;
uint8_t* const echo_ptr = RAM +
((this->r.g.echo_page * 0x100 + echo_pos) & 0xFFFF);
echo_pos += 4;
if ( echo_pos >= (this->r.g.echo_delay & 15) * 0x800 )
echo_pos = 0;
this->echo_pos = echo_pos;
int fb_0 = GET_LE16SA( echo_ptr );
int fb_1 = GET_LE16SA( echo_ptr + 2 );
/* Keep last 8 samples */
int (* const fir_ptr) [2] = this->fir_buf + this->fir_pos;
this->fir_pos = (this->fir_pos + 1) & (FIR_BUF_HALF - 1);
fir_ptr [ 0] [0] = fb_0;
fir_ptr [ 0] [1] = fb_1;
/* duplicate at +8 eliminates wrap checking below */
fir_ptr [FIR_BUF_HALF] [0] = fb_0;
fir_ptr [FIR_BUF_HALF] [1] = fb_1;
/* Apply FIR */
fb_0 *= this->fir_coeff [0];
fb_1 *= this->fir_coeff [0];
#define DO_PT( i )\
fb_0 += fir_ptr [i] [0] * this->fir_coeff [i];\
fb_1 += fir_ptr [i] [1] * this->fir_coeff [i];
DO_PT( 1 )
DO_PT( 2 )
DO_PT( 3 )
DO_PT( 4 )
DO_PT( 5 )
DO_PT( 6 )
DO_PT( 7 )
/* Generate output */
int amp_0 = (chans_0 * global_vol_0 + fb_0 * this->r.g.echo_volume_0)
>> global_muting;
int amp_1 = (chans_1 * global_vol_1 + fb_1 * this->r.g.echo_volume_1)
>> global_muting;
out_buf [ 0] = amp_0;
out_buf [WAV_CHUNK_SIZE] = amp_1;
out_buf ++;
if ( !(this->r.g.flags & 0x20) )
{
/* Feedback into echo buffer */
int e0 = (echo_0 >> 7) + ((fb_0 * this->r.g.echo_feedback) >> 14);
int e1 = (echo_1 >> 7) + ((fb_1 * this->r.g.echo_feedback) >> 14);
e0 = CLAMP16( e0 );
SET_LE16A( echo_ptr , e0 );
e1 = CLAMP16( e1 );
SET_LE16A( echo_ptr + 2, e1 );
}
#endif /* CPU_* */
#else /* SPCNOECHO == 1*/
/* Generate output */
int amp_0 = (chans_0 * global_vol_0) >> global_muting;
int amp_1 = (chans_1 * global_vol_1) >> global_muting;
out_buf [ 0] = amp_0;
out_buf [WAV_CHUNK_SIZE] = amp_1;
out_buf ++;
#endif /* SPCNOECHO */
}
while ( --count );
#if 0
EXIT_TIMER(dsp);
ENTER_TIMER(cpu);
#endif
}
void DSP_reset( struct Spc_Dsp* this )
{
this->keys_down = 0;
this->echo_pos = 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 );
int i;
for ( i = VOICE_COUNT; --i >= 0; )
{
struct voice_t* v = this->voice_state + i;
v->env_mode = state_release;
v->addr = ram.ram;
}
#if SPC_BRRCACHE
this->oldsize = 0;
for ( i = 0; i < 256; i++ )
this->wave_entry [i].start_addr = -1;
#endif
#if defined(CPU_COLDFIRE)
this->fir_ptr = fir_buf;
this->last_fir_ptr = &fir_buf [7];
ci->memset( fir_buf, 0, sizeof fir_buf );
#elif defined (CPU_ARM)
this->fir_ptr = fir_buf;
ci->memset( fir_buf, 0, sizeof fir_buf );
#else
this->fir_pos = 0;
ci->memset( this->fir_buf, 0, sizeof this->fir_buf );
#endif
assert( offsetof (struct globals_t,unused9 [2]) == REGISTER_COUNT );
assert( sizeof (this->r.voice) == REGISTER_COUNT );
}
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