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- #include <../base.hpp>
- #define ADSP_CPP
- #include "adsp_tables.cpp"
- void aDSP::enter() { loop:
- run();
- goto loop;
- }
- uint8 aDSP::readb(uint16 addr) {
- return spcram[addr];
- }
- void aDSP::writeb(uint16 addr, uint8 data) {
- spcram[addr] = data;
- }
- uint16 aDSP::readw(uint16 addr) {
- return (readb(addr + 0)) | (readb(addr + 1) << 8);
- }
- void aDSP::writew(uint16 addr, uint16 data) {
- writeb(addr + 0, data);
- writeb(addr + 1, data >> 8);
- }
- uint8 aDSP::read(uint8 addr) {
- addr &= 127;
- int v = addr >> 4;
- int n = addr & 15;
- switch(addr) {
- case 0x00: case 0x10: case 0x20: case 0x30:
- case 0x40: case 0x50: case 0x60: case 0x70:
- return voice[v].VOLL;
- case 0x01: case 0x11: case 0x21: case 0x31:
- case 0x41: case 0x51: case 0x61: case 0x71:
- return voice[v].VOLR;
- case 0x02: case 0x12: case 0x22: case 0x32:
- case 0x42: case 0x52: case 0x62: case 0x72:
- return voice[v].PITCH;
- case 0x03: case 0x13: case 0x23: case 0x33:
- case 0x43: case 0x53: case 0x63: case 0x73:
- return voice[v].PITCH >> 8;
- case 0x04: case 0x14: case 0x24: case 0x34:
- case 0x44: case 0x54: case 0x64: case 0x74:
- return voice[v].SRCN;
- case 0x05: case 0x15: case 0x25: case 0x35:
- case 0x45: case 0x55: case 0x65: case 0x75:
- return voice[v].ADSR1;
- case 0x06: case 0x16: case 0x26: case 0x36:
- case 0x46: case 0x56: case 0x66: case 0x76:
- return voice[v].ADSR2;
- case 0x07: case 0x17: case 0x27: case 0x37:
- case 0x47: case 0x57: case 0x67: case 0x77:
- return voice[v].GAIN;
- case 0x08: case 0x18: case 0x28: case 0x38:
- case 0x48: case 0x58: case 0x68: case 0x78:
- return voice[v].ENVX;
- case 0x09: case 0x19: case 0x29: case 0x39:
- case 0x49: case 0x59: case 0x69: case 0x79:
- return voice[v].OUTX;
- case 0x0f: case 0x1f: case 0x2f: case 0x3f:
- case 0x4f: case 0x5f: case 0x6f: case 0x7f:
- return status.FIR[v];
- case 0x0c: return status.MVOLL;
- case 0x1c: return status.MVOLR;
- case 0x2c: return status.EVOLL;
- case 0x3c: return status.EVOLR;
- case 0x4c: return status.KON;
- case 0x5c: return status.KOFF;
- case 0x6c: return status.FLG;
- case 0x7c: return status.ENDX;
- case 0x0d: return status.EFB;
- case 0x2d: return status.PMON;
- case 0x3d: return status.NON;
- case 0x4d: return status.EON;
- case 0x5d: return status.DIR;
- case 0x6d: return status.ESA;
- case 0x7d: return status.EDL;
- }
- return dspram[addr];
- }
- void aDSP::write(uint8 addr, uint8 data) {
- //0x80-0xff is a read-only mirror of 0x00-0x7f
- if(addr & 0x80)return;
- int v = addr >> 4;
- int n = addr & 15;
- switch(addr) {
- case 0x00: case 0x10: case 0x20: case 0x30:
- case 0x40: case 0x50: case 0x60: case 0x70:
- voice[v].VOLL = data;
- break;
- case 0x01: case 0x11: case 0x21: case 0x31:
- case 0x41: case 0x51: case 0x61: case 0x71:
- voice[v].VOLR = data;
- break;
- case 0x02: case 0x12: case 0x22: case 0x32:
- case 0x42: case 0x52: case 0x62: case 0x72:
- voice[v].PITCH &= 0xff00;
- voice[v].PITCH |= data;
- break;
- case 0x03: case 0x13: case 0x23: case 0x33:
- case 0x43: case 0x53: case 0x63: case 0x73:
- voice[v].PITCH &= 0x00ff;
- voice[v].PITCH |= data << 8;
- break;
- case 0x04: case 0x14: case 0x24: case 0x34:
- case 0x44: case 0x54: case 0x64: case 0x74:
- voice[v].SRCN = data;
- break;
- case 0x05: case 0x15: case 0x25: case 0x35:
- case 0x45: case 0x55: case 0x65: case 0x75:
- voice[v].ADSR1 = data;
- voice[v].AdjustEnvelope();
- break;
- case 0x06: case 0x16: case 0x26: case 0x36:
- case 0x46: case 0x56: case 0x66: case 0x76:
- voice[v].ADSR2 = data;
- //sustain_level = 0-7, 7 is a special case handled by ATTACK envx mode
- voice[v].env_sustain = (voice[v].ADSR_sus_level() + 1) << 8;
- voice[v].AdjustEnvelope();
- break;
- case 0x07: case 0x17: case 0x27: case 0x37:
- case 0x47: case 0x57: case 0x67: case 0x77:
- voice[v].GAIN = data;
- voice[v].AdjustEnvelope();
- break;
- case 0x08: case 0x18: case 0x28: case 0x38:
- case 0x48: case 0x58: case 0x68: case 0x78:
- voice[v].ENVX = data;
- break;
- case 0x09: case 0x19: case 0x29: case 0x39:
- case 0x49: case 0x59: case 0x69: case 0x79:
- voice[v].OUTX = data;
- break;
- case 0x0f: case 0x1f: case 0x2f: case 0x3f:
- case 0x4f: case 0x5f: case 0x6f: case 0x7f:
- status.FIR[v] = data;
- break;
- case 0x0c: status.MVOLL = data; break;
- case 0x1c: status.MVOLR = data; break;
- case 0x2c: status.EVOLL = data; break;
- case 0x3c: status.EVOLR = data; break;
- case 0x4c:
- status.KON = data;
- status.kon = data;
- break;
- case 0x5c:
- status.KOFF = data;
- break;
- case 0x6c:
- status.FLG = data;
- status.noise_rate = rate_table[data & 0x1f];
- break;
- case 0x7c:
- //read-only register, writes clear all bits of ENDX
- status.ENDX = 0;
- break;
- case 0x0d: status.EFB = data; break;
- case 0x2d: status.PMON = data; break;
- case 0x3d: status.NON = data; break;
- case 0x4d: status.EON = data; break;
- case 0x5d: status.DIR = data; break;
- case 0x6d: status.ESA = data; break;
- case 0x7d: status.EDL = data; break;
- }
- dspram[addr] = data;
- }
- void aDSP::power() {
- spcram = r_smp->get_spcram_handle();
- memset(dspram, 0x00, 128);
- for(int v = 0; v < 8; v++) {
- voice[v].VOLL = 0;
- voice[v].VOLR = 0;
- voice[v].PITCH = 0;
- voice[v].SRCN = 0;
- voice[v].ADSR1 = 0;
- voice[v].ADSR2 = 0;
- voice[v].GAIN = 0;
- status.FIR[v] = 0;
- }
- status.FLG = 0xe0;
- status.MVOLL = status.MVOLR = 0;
- status.EVOLL = status.EVOLR = 0;
- status.ENDX = 0;
- status.EFB = 0;
- status.PMON = 0;
- status.NON = 0;
- status.EON = 0;
- status.DIR = 0;
- status.ESA = 0;
- status.EDL = 0;
- status.echo_length = 0;
- reset();
- }
- void aDSP::reset() {
- status.KON = 0x00;
- status.KOFF = 0x00;
- status.FLG |= 0xe0;
- status.kon = 0x00;
- status.esa = 0x00;
- status.noise_ctr = 0;
- status.noise_rate = 0;
- status.noise_sample = 0x4000;
- status.echo_index = 0;
- status.fir_buffer_index = 0;
- for(int v = 0; v < 8; v++) {
- voice[v].ENVX = 0;
- voice[v].OUTX = 0;
- voice[v].pitch_ctr = 0;
- voice[v].brr_index = 0;
- voice[v].brr_ptr = readw((status.DIR << 8) + (voice[v].SRCN << 2));
- voice[v].brr_looped = false;
- voice[v].brr_data[0] = 0;
- voice[v].brr_data[1] = 0;
- voice[v].brr_data[2] = 0;
- voice[v].brr_data[3] = 0;
- voice[v].brr_data_index = 0;
- voice[v].envx = 0;
- voice[v].env_ctr = 0;
- voice[v].env_rate = 0;
- voice[v].env_state = SILENCE;
- voice[v].env_mode = DIRECT;
- status.fir_buffer[0][v] = 0;
- status.fir_buffer[1][v] = 0;
- }
- dsp_counter = 0;
- }
- void aDSP::run() {
- uint8 pmon;
- int32 sample;
- int32 msamplel, msampler;
- int32 esamplel, esampler;
- int32 fir_samplel, fir_sampler;
- pmon = status.PMON & ~status.NON & ~1;
- if((dsp_counter++ & 1) == 0) {
- for(uint v = 0; v < 8; v++) {
- if(status.soft_reset()) {
- if(voice[v].env_state != SILENCE) {
- voice[v].env_state = SILENCE;
- voice[v].AdjustEnvelope();
- }
- }
- if(status.KOFF & (1 << v)) {
- if(voice[v].env_state != SILENCE && voice[v].env_state != RELEASE) {
- voice[v].env_state = RELEASE;
- voice[v].AdjustEnvelope();
- }
- }
- if(status.kon & (1 << v)) {
- voice[v].brr_ptr = readw((status.DIR << 8) + (voice[v].SRCN << 2));
- voice[v].brr_index = -9;
- voice[v].brr_looped = false;
- voice[v].brr_data[0] = 0;
- voice[v].brr_data[1] = 0;
- voice[v].brr_data[2] = 0;
- voice[v].brr_data[3] = 0;
- voice[v].envx = 0;
- voice[v].env_state = ATTACK;
- voice[v].AdjustEnvelope();
- }
- }
- status.ENDX &= ~status.kon;
- status.kon = 0;
- }
- /*****
- * update noise
- *****/
- status.noise_ctr += status.noise_rate;
- if(status.noise_ctr >= 0x7800) {
- status.noise_ctr -= 0x7800;
- status.noise_sample = (status.noise_sample >> 1) | (((status.noise_sample << 14) ^ (status.noise_sample << 13)) & 0x4000);
- }
- msamplel = msampler = 0;
- esamplel = esampler = 0;
- /*****
- * process voice channels
- *****/
- for(int v = 0; v < 8; v++) {
- if(voice[v].brr_index < -1) {
- voice[v].brr_index++;
- voice[v].OUTX = voice[v].outx = 0;
- voice[v].ENVX = 0;
- continue;
- }
- if(voice[v].brr_index >= 0) {
- if(pmon & (1 << v)) {
- voice[v].pitch_ctr += (voice[v].pitch_rate() * (voice[v - 1].outx + 0x8000)) >> 15;
- } else {
- voice[v].pitch_ctr += voice[v].pitch_rate();
- }
- } else {
- voice[v].pitch_ctr = 0x3000;
- voice[v].brr_index = 0;
- }
- /*****
- * decode BRR samples
- *****/
- while(voice[v].pitch_ctr >= 0) {
- voice[v].pitch_ctr -= 0x1000;
- voice[v].brr_data_index++;
- voice[v].brr_data_index &= 3;
- if(voice[v].brr_index == 0) {
- voice[v].brr_header = readb(voice[v].brr_ptr);
- if(voice[v].brr_header_flags() == BRR_END) {
- status.ENDX |= (1 << v);
- voice[v].env_state = SILENCE;
- voice[v].AdjustEnvelope();
- }
- }
- #define S(x) voice[v].brr_data[(voice[v].brr_data_index + (x)) & 3]
- if(voice[v].env_state != SILENCE) {
- sample = readb(voice[v].brr_ptr + 1 + (voice[v].brr_index >> 1));
- if(voice[v].brr_index & 1) {
- sample = sclip<4>(sample);
- } else {
- sample = sclip<4>(sample >> 4);
- }
- if(voice[v].brr_header_shift() <= 12) {
- sample = (sample << voice[v].brr_header_shift() >> 1);
- } else {
- sample &= ~0x7ff;
- }
- switch(voice[v].brr_header_filter()) {
- case 0: //direct
- break;
- case 1: //15/16
- sample += S(-1) + ((-S(-1)) >> 4);
- break;
- case 2: //61/32 - 15/16
- sample += (S(-1) << 1) + ((-((S(-1) << 1) + S(-1))) >> 5)
- - S(-2) + (S(-2) >> 4);
- break;
- case 3: //115/64 - 13/16
- sample += (S(-1) << 1) + ((-(S(-1) + (S(-1) << 2) + (S(-1) << 3))) >> 6)
- - S(-2) + (((S(-2) << 1) + S(-2)) >> 4);
- break;
- }
- S(0) = sample = sclip<15>(sclamp<16>(sample));
- } else {
- S(0) = sample = 0;
- }
- if(++voice[v].brr_index > 15) {
- voice[v].brr_index = 0;
- if(voice[v].brr_header_flags() & BRR_END) {
- if(voice[v].brr_header_flags() & BRR_LOOP) {
- status.ENDX |= (1 << v);
- }
- voice[v].brr_ptr = readw((status.DIR << 8) + (voice[v].SRCN << 2) + 2);
- voice[v].brr_looped = true;
- } else {
- voice[v].brr_ptr += 9;
- }
- }
- }
- /*****
- * volume envelope adjust
- *****/
- voice[v].env_ctr += voice[v].env_rate;
- if(voice[v].env_ctr >= 0x7800) {
- voice[v].env_ctr -= 0x7800;
- switch(voice[v].env_mode) {
- case DIRECT:
- voice[v].env_rate = 0;
- break;
- case LINEAR_DEC:
- voice[v].envx -= 32;
- if(voice[v].envx <= 0) {
- voice[v].envx = 0;
- voice[v].env_rate = 0;
- voice[v].env_mode = DIRECT;
- }
- break;
- case LINEAR_INC:
- voice[v].envx += 32;
- if(voice[v].envx >= 0x7ff) {
- voice[v].envx = 0x7ff;
- voice[v].env_rate = 0;
- voice[v].env_mode = DIRECT;
- if(voice[v].ADSR_enabled() && voice[v].env_state == ATTACK) {
- voice[v].env_state = ((voice[v].env_sustain == 0x800) ? SUSTAIN : DECAY);
- voice[v].AdjustEnvelope();
- }
- }
- break;
- case EXP_DEC:
- //multiply by 255/256ths
- voice[v].envx -= ((voice[v].envx - 1) >> 8) + 1;
- if(voice[v].ADSR_enabled() && voice[v].env_state == DECAY && voice[v].envx <= voice[v].env_sustain) {
- voice[v].env_state = SUSTAIN;
- voice[v].AdjustEnvelope();
- } else if(voice[v].envx <= 0) {
- voice[v].envx = 0;
- voice[v].env_rate = 0;
- voice[v].env_mode = DIRECT;
- }
- break;
- case BENT_INC:
- if(voice[v].envx < 0x600) {
- voice[v].envx += 32;
- } else {
- voice[v].envx += 8;
- if(voice[v].envx >= 0x7ff) {
- voice[v].envx = 0x7ff;
- voice[v].env_rate = 0;
- voice[v].env_mode = DIRECT;
- }
- }
- break;
- case FAST_ATTACK:
- voice[v].envx += 0x400;
- if(voice[v].envx >= 0x7ff) {
- voice[v].envx = 0x7ff;
- //attack raises to max envx. if sustain is also set to max envx, skip decay phase
- voice[v].env_state = ((voice[v].env_sustain == 0x800) ? SUSTAIN : DECAY);
- voice[v].AdjustEnvelope();
- }
- break;
- case RELEASE_DEC:
- voice[v].envx -= 8;
- if(voice[v].envx <= 0) {
- voice[v].env_state = SILENCE;
- voice[v].AdjustEnvelope();
- }
- break;
- }
- }
- voice[v].ENVX = voice[v].envx >> 4;
- /*****
- * gaussian interpolation / noise
- *****/
- if(status.NON & (1 << v)) {
- sample = sclip<15>(status.noise_sample);
- } else {
- int32 d = voice[v].pitch_ctr >> 4; //-256 <= sample <= -1
- sample = ((gaussian_table[ -1 - d] * S(-3)) >> 11);
- sample += ((gaussian_table[255 - d] * S(-2)) >> 11);
- sample += ((gaussian_table[512 + d] * S(-1)) >> 11);
- sample = sclip <15>(sample);
- sample += ((gaussian_table[256 + d] * S( 0)) >> 11);
- sample = sclamp<15>(sample);
- }
- #undef S
- /*****
- * envelope / volume adjust
- *****/
- sample = (sample * voice[v].envx) >> 11;
- voice[v].outx = sample << 1;
- voice[v].OUTX = sample >> 7;
- if(!status.mute()) {
- msamplel += ((sample * voice[v].VOLL) >> 7) << 1;
- msampler += ((sample * voice[v].VOLR) >> 7) << 1;
- }
- if((status.EON & (1 << v)) && status.echo_write()) {
- esamplel += ((sample * voice[v].VOLL) >> 7) << 1;
- esampler += ((sample * voice[v].VOLR) >> 7) << 1;
- }
- }
- /*****
- * echo (FIR) adjust
- *****/
- #define F(c,x) status.fir_buffer[c][(status.fir_buffer_index + (x)) & 7]
- status.fir_buffer_index++;
- F(0,0) = readw((status.esa << 8) + status.echo_index + 0);
- F(1,0) = readw((status.esa << 8) + status.echo_index + 2);
- fir_samplel = (F(0,-0) * status.FIR[7] +
- F(0,-1) * status.FIR[6] +
- F(0,-2) * status.FIR[5] +
- F(0,-3) * status.FIR[4] +
- F(0,-4) * status.FIR[3] +
- F(0,-5) * status.FIR[2] +
- F(0,-6) * status.FIR[1] +
- F(0,-7) * status.FIR[0]);
- fir_sampler = (F(1,-0) * status.FIR[7] +
- F(1,-1) * status.FIR[6] +
- F(1,-2) * status.FIR[5] +
- F(1,-3) * status.FIR[4] +
- F(1,-4) * status.FIR[3] +
- F(1,-5) * status.FIR[2] +
- F(1,-6) * status.FIR[1] +
- F(1,-7) * status.FIR[0]);
- #undef F
- /*****
- * update echo buffer
- *****/
- if(status.echo_write()) {
- esamplel += (fir_samplel * status.EFB) >> 14;
- esampler += (fir_sampler * status.EFB) >> 14;
- esamplel = sclamp<16>(esamplel);
- esampler = sclamp<16>(esampler);
- writew((status.esa << 8) + status.echo_index + 0, esamplel);
- writew((status.esa << 8) + status.echo_index + 2, esampler);
- }
- status.echo_index += 4;
- if(status.echo_index >= status.echo_length) {
- status.echo_index = 0;
- status.echo_length = (status.EDL & 0x0f) << 11;
- }
- //ESA read occurs at roughly 22/32th sample
- //ESA fetch occurs at roughly 29/32th sample
- //as this is not a subsample-level S-DSP emulator,
- //simulate ~25/32th delay by caching ESA for one
- //complete sample ...
- status.esa = status.ESA;
- /*****
- * main output adjust
- *****/
- if(!status.mute()) {
- msamplel = (msamplel * status.MVOLL) >> 7;
- msampler = (msampler * status.MVOLR) >> 7;
- msamplel += (fir_samplel * status.EVOLL) >> 14;
- msampler += (fir_sampler * status.EVOLR) >> 14;
- msamplel = sclamp<16>(msamplel);
- msampler = sclamp<16>(msampler);
- }
- snes.audio.update(msamplel, msampler);
- scheduler.addclocks_dsp(32 * 3 * 8);
- }
- aDSP::aDSP() {}
- aDSP::~aDSP() {}
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