qzdoom/snes_spc/fast_dsp/SPC_DSP.h
Randy Heit 3bfcc5c09c - Removed lots of spc_* cvars that are no longer meaningful and changed
spc_amp from a x.4 fixed point number to a normal float.
- Switched SPC playback from the external SNESAPU.DLL to Blargg's LGPL
  snes_spc library. I've compiled it with the fast DSP rather than the
  highly accurate one, since I didn't notice a meaningful difference between
  the two in my limited testing. In short: SPC playback is now built in to
  ZDoom. You don't need to download anything extra to make it work, and it
  also works on Linux as well as Windows (though building with Linux is
  currently untested).
- Fixed: Stereo separation was calculated very wrongly when in 2D sound mode.


SVN r794 (trunk)
2008-03-11 22:17:57 +00:00

212 lines
5.5 KiB
C++

// Fast SNES SPC-700 DSP emulator (about 3x speed of accurate one)
// snes_spc 0.9.0
#ifndef SPC_DSP_H
#define SPC_DSP_H
#include "blargg_common.h"
struct SPC_DSP {
public:
typedef BOOST::uint8_t uint8_t;
// Setup
// Initializes DSP and has it use the 64K RAM provided
void init( void* ram_64k );
// Sets destination for output samples. If out is NULL or out_size is 0,
// doesn't generate any.
typedef short sample_t;
void set_output( sample_t* out, int out_size );
// Number of samples written to output since it was last set, always
// a multiple of 2. Undefined if more samples were generated than
// output buffer could hold.
int sample_count() const;
// Emulation
// Resets DSP to power-on state
void reset();
// Emulates pressing reset switch on SNES
void soft_reset();
// Reads/writes DSP registers. For accuracy, you must first call spc_run_dsp()
// to catch the DSP up to present.
int read ( int addr ) const;
void write( int addr, int data );
// Runs DSP for specified number of clocks (~1024000 per second). Every 32 clocks
// a pair of samples is be generated.
void run( int clock_count );
// Sound control
// Mutes voices corresponding to non-zero bits in mask (overrides VxVOL with 0).
// Reduces emulation accuracy.
enum { voice_count = 8 };
void mute_voices( int mask );
// If true, prevents channels and global volumes from being phase-negated
void disable_surround( bool disable = true );
// State
// Resets DSP and uses supplied values to initialize registers
enum { register_count = 128 };
void load( uint8_t const regs [register_count] );
// DSP register addresses
// Global registers
enum {
r_mvoll = 0x0C, r_mvolr = 0x1C,
r_evoll = 0x2C, r_evolr = 0x3C,
r_kon = 0x4C, r_koff = 0x5C,
r_flg = 0x6C, r_endx = 0x7C,
r_efb = 0x0D, r_pmon = 0x2D,
r_non = 0x3D, r_eon = 0x4D,
r_dir = 0x5D, r_esa = 0x6D,
r_edl = 0x7D,
r_fir = 0x0F // 8 coefficients at 0x0F, 0x1F ... 0x7F
};
// Voice registers
enum {
v_voll = 0x00, v_volr = 0x01,
v_pitchl = 0x02, v_pitchh = 0x03,
v_srcn = 0x04, v_adsr0 = 0x05,
v_adsr1 = 0x06, v_gain = 0x07,
v_envx = 0x08, v_outx = 0x09
};
public:
enum { extra_size = 16 };
sample_t* extra() { return m.extra; }
sample_t const* out_pos() const { return m.out; }
public:
BLARGG_DISABLE_NOTHROW
typedef BOOST::int8_t int8_t;
typedef BOOST::int16_t int16_t;
enum { echo_hist_size = 8 };
enum env_mode_t { env_release, env_attack, env_decay, env_sustain };
enum { brr_buf_size = 12 };
struct voice_t
{
int buf [brr_buf_size*2];// decoded samples (twice the size to simplify wrap handling)
int* buf_pos; // place in buffer where next samples will be decoded
int interp_pos; // relative fractional position in sample (0x1000 = 1.0)
int brr_addr; // address of current BRR block
int brr_offset; // current decoding offset in BRR block
int kon_delay; // KON delay/current setup phase
env_mode_t env_mode;
int env; // current envelope level
int hidden_env; // used by GAIN mode 7, very obscure quirk
int volume [2]; // copy of volume from DSP registers, with surround disabled
int enabled; // -1 if enabled, 0 if muted
};
private:
struct state_t
{
uint8_t regs [register_count];
// Echo history keeps most recent 8 samples (twice the size to simplify wrap handling)
int echo_hist [echo_hist_size * 2] [2];
int (*echo_hist_pos) [2]; // &echo_hist [0 to 7]
int every_other_sample; // toggles every sample
int kon; // KON value when last checked
int noise;
int echo_offset; // offset from ESA in echo buffer
int echo_length; // number of bytes that echo_offset will stop at
int phase; // next clock cycle to run (0-31)
unsigned counters [4];
int new_kon;
int t_koff;
voice_t voices [voice_count];
unsigned* counter_select [32];
// non-emulation state
uint8_t* ram; // 64K shared RAM between DSP and SMP
int mute_mask;
int surround_threshold;
sample_t* out;
sample_t* out_end;
sample_t* out_begin;
sample_t extra [extra_size];
};
state_t m;
void init_counter();
void run_counter( int );
void soft_reset_common();
void write_outline( int addr, int data );
void update_voice_vol( int addr );
};
#include <assert.h>
inline int SPC_DSP::sample_count() const { return int(m.out - m.out_begin); }
inline int SPC_DSP::read( int addr ) const
{
assert( (unsigned) addr < register_count );
return m.regs [addr];
}
inline void SPC_DSP::update_voice_vol( int addr )
{
int l = (int8_t) m.regs [addr + v_voll];
int r = (int8_t) m.regs [addr + v_volr];
if ( l * r < m.surround_threshold )
{
// signs differ, so negate those that are negative
l ^= l >> 7;
r ^= r >> 7;
}
voice_t& v = m.voices [addr >> 4];
int enabled = v.enabled;
v.volume [0] = l & enabled;
v.volume [1] = r & enabled;
}
inline void SPC_DSP::write( int addr, int data )
{
assert( (unsigned) addr < register_count );
m.regs [addr] = (uint8_t) data;
int low = addr & 0x0F;
if ( low < 0x2 ) // voice volumes
{
update_voice_vol( low ^ addr );
}
else if ( low == 0xC )
{
if ( addr == r_kon )
m.new_kon = (uint8_t) data;
if ( addr == r_endx ) // always cleared, regardless of data written
m.regs [r_endx] = 0;
}
}
inline void SPC_DSP::disable_surround( bool disable )
{
m.surround_threshold = disable ? 0 : -0x4000;
}
#define SPC_NO_COPY_STATE_FUNCS 1
#define SPC_LESS_ACCURATE 1
#endif