comparison src/console/Spc_Dsp.cxx @ 316:fb513e10174e trunk

[svn] - merge libconsole-blargg into mainline libconsole: + obsoletes plugins-ugly:sapplug

315:2294f3a6f136

// Game_Music_Emu 0.3.0. http://www.slack.net/~ant/

// Based on Brad Martin's OpenSPC DSP emulator

#include "Spc_Dsp.h"

#include <string.h>

#include "blargg_endian.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_BEGIN

Spc_Dsp::Spc_Dsp( uint8_t* ram_ ) : ram( ram_ )
{
	set_gain( 1.0 );
	mute_voices( 0 );
	disable_surround( false );
	
	BOOST_STATIC_ASSERT( sizeof (g) == register_count && sizeof (voice) == register_count );
}

void Spc_Dsp::mute_voices( int mask )
{
	for ( int i = 0; i < voice_count; i++ )

{
	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 signs of volumes differ
			if ( left * right < surround_threshold )
			{
				if ( left < 0 )
					volume [0] = -left;
				else

			}
			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 const short 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

			
			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;

			{
				noise_count = env_rate_init;
				
				noise_amp = BOOST::int16_t (noise * 2);
				
				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.
		long prev_outx = 0;
		
		int echol = 0;
		int echor = 0;
		int left = 0;
		int right = 0;

				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

					delta = (delta >> 14) & ~0x7FF;
				
				// One, two and three point IIR filters
				int smp1 = voice.interp0;
				int smp2 = voice.interp1;
				switch ( (voice.block_header >> 2) & 3 )
				{
					case 0:
						break;
					
					case 1:
						delta += smp1 >> 1;
						delta += (-smp1) >> 5;
						break;
					
					case 2:
						delta += smp1;
						delta += (-(smp1 + (smp1 >> 1))) >> 5;
						delta -= smp2 >> 1;
						delta += smp2 >> 5;
						break;
					
					case 3:
						delta += smp1;
						delta += (-(smp1 + (smp1 << 2) + (smp1 << 3))) >> 7;
						delta -= smp2 >> 1;
						delta += (smp2 + (smp2 >> 1)) >> 4;
						break;
				}
				
				voice.interp3 = voice.interp2;
				voice.interp2 = smp2;
				voice.interp1 = smp1;

				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*) ((char*) gauss + index);
			const BOOST::int16_t* table2 = (BOOST::int16_t*) ((char*) gauss + (255*4 - index));
			int s = ((table  [0] * voice.interp3) >> 12) +
					((table  [1] * voice.interp2) >> 12);
			s +=    ((table2 [1] * voice.interp1) >> 12) +
			// to do: should clamp here
					((table2 [0] * voice.interp0) >> 12);
			int output = noise_amp; // noise is rarely used
			if ( !(g.noise_enables & vbit) )
				output = clamp_16( s * 2 );
			
			// 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 = output >> 8;
			if ( g.echo_ons & vbit )
			{
				echol += l;
				echor += r;
			}
			left += l;
			right += r;
		}
		// end of channel loop
		
		// main volume control

		
		// 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] = fb_left;
		fir_pos [0] [1] = fb_right;
		fir_pos [8] [0] = fb_left; // duplicate at +8 eliminates wrap checking below
		fir_pos [8] [1] = fb_right;
		
		// FIR
		fb_left =       fb_left * fir_coeff [7] +
				fir_pos [1] [0] * fir_coeff [6] +
				fir_pos [2] [0] * fir_coeff [5] +

			left  = clamp_16( left  );
			right = clamp_16( right );
			
			int mute = g.flags & 0x40;
			
			out_buf [0] = left;
			out_buf [1] = right;
			out_buf += 2;
			
			// muting
			if ( mute )
			{

   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,
};

316:fb513e10174e

// Game_Music_Emu 0.5.1. 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 );
	
	BOOST_STATIC_ASSERT( sizeof (g) == register_count && sizeof (voice) == register_count );
	blargg_verify_byte_order();
}

void Spc_Dsp::mute_voices( int mask )
{
	for ( int i = 0; i < voice_count; i++ )

{
	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

			}
			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

			
			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;

			{
				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;

				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

					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;

				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

		
		// 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] +

			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 )
			{

   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,
};