ref: 3f1b1d5a61e954eb7690887eaf43ab7e549491fb
dir: /snes/dsp.c/
#include <stdio.h> #include <stdlib.h> #include <string.h> #include <stdint.h> #include <stdbool.h> #include <stddef.h> #include <limits.h> #include "dsp_regs.h" #include "dsp.h" #define MY_CHANGES 1 static const int rateValues[32] = { 0, 2048, 1536, 1280, 1024, 768, 640, 512, 384, 320, 256, 192, 160, 128, 96, 80, 64, 48, 40, 32, 24, 20, 16, 12, 10, 8, 6, 5, 4, 3, 2, 1 }; static const int gaussValues[512] = { 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x000, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x001, 0x002, 0x002, 0x002, 0x002, 0x002, 0x002, 0x002, 0x003, 0x003, 0x003, 0x003, 0x003, 0x004, 0x004, 0x004, 0x004, 0x004, 0x005, 0x005, 0x005, 0x005, 0x006, 0x006, 0x006, 0x006, 0x007, 0x007, 0x007, 0x008, 0x008, 0x008, 0x009, 0x009, 0x009, 0x00A, 0x00A, 0x00A, 0x00B, 0x00B, 0x00B, 0x00C, 0x00C, 0x00D, 0x00D, 0x00E, 0x00E, 0x00F, 0x00F, 0x00F, 0x010, 0x010, 0x011, 0x011, 0x012, 0x013, 0x013, 0x014, 0x014, 0x015, 0x015, 0x016, 0x017, 0x017, 0x018, 0x018, 0x019, 0x01A, 0x01B, 0x01B, 0x01C, 0x01D, 0x01D, 0x01E, 0x01F, 0x020, 0x020, 0x021, 0x022, 0x023, 0x024, 0x024, 0x025, 0x026, 0x027, 0x028, 0x029, 0x02A, 0x02B, 0x02C, 0x02D, 0x02E, 0x02F, 0x030, 0x031, 0x032, 0x033, 0x034, 0x035, 0x036, 0x037, 0x038, 0x03A, 0x03B, 0x03C, 0x03D, 0x03E, 0x040, 0x041, 0x042, 0x043, 0x045, 0x046, 0x047, 0x049, 0x04A, 0x04C, 0x04D, 0x04E, 0x050, 0x051, 0x053, 0x054, 0x056, 0x057, 0x059, 0x05A, 0x05C, 0x05E, 0x05F, 0x061, 0x063, 0x064, 0x066, 0x068, 0x06A, 0x06B, 0x06D, 0x06F, 0x071, 0x073, 0x075, 0x076, 0x078, 0x07A, 0x07C, 0x07E, 0x080, 0x082, 0x084, 0x086, 0x089, 0x08B, 0x08D, 0x08F, 0x091, 0x093, 0x096, 0x098, 0x09A, 0x09C, 0x09F, 0x0A1, 0x0A3, 0x0A6, 0x0A8, 0x0AB, 0x0AD, 0x0AF, 0x0B2, 0x0B4, 0x0B7, 0x0BA, 0x0BC, 0x0BF, 0x0C1, 0x0C4, 0x0C7, 0x0C9, 0x0CC, 0x0CF, 0x0D2, 0x0D4, 0x0D7, 0x0DA, 0x0DD, 0x0E0, 0x0E3, 0x0E6, 0x0E9, 0x0EC, 0x0EF, 0x0F2, 0x0F5, 0x0F8, 0x0FB, 0x0FE, 0x101, 0x104, 0x107, 0x10B, 0x10E, 0x111, 0x114, 0x118, 0x11B, 0x11E, 0x122, 0x125, 0x129, 0x12C, 0x130, 0x133, 0x137, 0x13A, 0x13E, 0x141, 0x145, 0x148, 0x14C, 0x150, 0x153, 0x157, 0x15B, 0x15F, 0x162, 0x166, 0x16A, 0x16E, 0x172, 0x176, 0x17A, 0x17D, 0x181, 0x185, 0x189, 0x18D, 0x191, 0x195, 0x19A, 0x19E, 0x1A2, 0x1A6, 0x1AA, 0x1AE, 0x1B2, 0x1B7, 0x1BB, 0x1BF, 0x1C3, 0x1C8, 0x1CC, 0x1D0, 0x1D5, 0x1D9, 0x1DD, 0x1E2, 0x1E6, 0x1EB, 0x1EF, 0x1F3, 0x1F8, 0x1FC, 0x201, 0x205, 0x20A, 0x20F, 0x213, 0x218, 0x21C, 0x221, 0x226, 0x22A, 0x22F, 0x233, 0x238, 0x23D, 0x241, 0x246, 0x24B, 0x250, 0x254, 0x259, 0x25E, 0x263, 0x267, 0x26C, 0x271, 0x276, 0x27B, 0x280, 0x284, 0x289, 0x28E, 0x293, 0x298, 0x29D, 0x2A2, 0x2A6, 0x2AB, 0x2B0, 0x2B5, 0x2BA, 0x2BF, 0x2C4, 0x2C9, 0x2CE, 0x2D3, 0x2D8, 0x2DC, 0x2E1, 0x2E6, 0x2EB, 0x2F0, 0x2F5, 0x2FA, 0x2FF, 0x304, 0x309, 0x30E, 0x313, 0x318, 0x31D, 0x322, 0x326, 0x32B, 0x330, 0x335, 0x33A, 0x33F, 0x344, 0x349, 0x34E, 0x353, 0x357, 0x35C, 0x361, 0x366, 0x36B, 0x370, 0x374, 0x379, 0x37E, 0x383, 0x388, 0x38C, 0x391, 0x396, 0x39B, 0x39F, 0x3A4, 0x3A9, 0x3AD, 0x3B2, 0x3B7, 0x3BB, 0x3C0, 0x3C5, 0x3C9, 0x3CE, 0x3D2, 0x3D7, 0x3DC, 0x3E0, 0x3E5, 0x3E9, 0x3ED, 0x3F2, 0x3F6, 0x3FB, 0x3FF, 0x403, 0x408, 0x40C, 0x410, 0x415, 0x419, 0x41D, 0x421, 0x425, 0x42A, 0x42E, 0x432, 0x436, 0x43A, 0x43E, 0x442, 0x446, 0x44A, 0x44E, 0x452, 0x455, 0x459, 0x45D, 0x461, 0x465, 0x468, 0x46C, 0x470, 0x473, 0x477, 0x47A, 0x47E, 0x481, 0x485, 0x488, 0x48C, 0x48F, 0x492, 0x496, 0x499, 0x49C, 0x49F, 0x4A2, 0x4A6, 0x4A9, 0x4AC, 0x4AF, 0x4B2, 0x4B5, 0x4B7, 0x4BA, 0x4BD, 0x4C0, 0x4C3, 0x4C5, 0x4C8, 0x4CB, 0x4CD, 0x4D0, 0x4D2, 0x4D5, 0x4D7, 0x4D9, 0x4DC, 0x4DE, 0x4E0, 0x4E3, 0x4E5, 0x4E7, 0x4E9, 0x4EB, 0x4ED, 0x4EF, 0x4F1, 0x4F3, 0x4F5, 0x4F6, 0x4F8, 0x4FA, 0x4FB, 0x4FD, 0x4FF, 0x500, 0x502, 0x503, 0x504, 0x506, 0x507, 0x508, 0x50A, 0x50B, 0x50C, 0x50D, 0x50E, 0x50F, 0x510, 0x511, 0x511, 0x512, 0x513, 0x514, 0x514, 0x515, 0x516, 0x516, 0x517, 0x517, 0x517, 0x518, 0x518, 0x518, 0x518, 0x518, 0x519, 0x519 }; static void dsp_cycleChannel(Dsp* dsp, int ch); static void dsp_handleEcho(Dsp* dsp, int* outputL, int* outputR); static void dsp_handleGain(Dsp* dsp, int ch); static void dsp_decodeBrr(Dsp* dsp, int ch); static int16_t dsp_getSample(Dsp* dsp, int ch, int sampleNum, int offset); static void dsp_handleNoise(Dsp* dsp); Dsp* dsp_init(uint8_t *apu_ram) { Dsp* dsp = (Dsp*)malloc(sizeof(Dsp)); dsp->apu_ram = apu_ram; return dsp; } void dsp_free(Dsp* dsp) { free(dsp); } void dsp_reset(Dsp* dsp) { memset(dsp->ram, 0, sizeof(dsp->ram)); dsp->ram[ENDX] = 0xff; // set ENDX bit for all channels for(int i = 0; i < 8; i++) { dsp->channel[i].pitch = 0; dsp->channel[i].pitchCounter = 0; dsp->channel[i].pitchModulation = false; memset(dsp->channel[i].decodeBuffer, 0, sizeof(dsp->channel[i].decodeBuffer)); dsp->channel[i].srcn = 0; dsp->channel[i].decodeOffset = 0; dsp->channel[i].previousFlags = 0; dsp->channel[i].old = 0; dsp->channel[i].older = 0; dsp->channel[i].useNoise = false; memset(dsp->channel[i].adsrRates, 0, sizeof(dsp->channel[i].adsrRates)); dsp->channel[i].rateCounter = 0; dsp->channel[i].adsrState = 0; dsp->channel[i].sustainLevel = 0; dsp->channel[i].useGain = false; dsp->channel[i].gainMode = 0; dsp->channel[i].directGain = false; dsp->channel[i].gainValue = 0; dsp->channel[i].gain = 0; dsp->channel[i].keyOn = false; dsp->channel[i].keyOff = false; dsp->channel[i].sampleOut = 0; dsp->channel[i].volumeL = 0; dsp->channel[i].volumeR = 0; dsp->channel[i].echoEnable = false; } dsp->dirPage = 0; dsp->evenCycle = false; dsp->mute = true; dsp->reset = true; dsp->masterVolumeL = 0; dsp->masterVolumeR = 0; dsp->noiseSample = -0x4000; dsp->noiseRate = 0; dsp->noiseCounter = 0; dsp->echoWrites = false; dsp->echoVolumeL = 0; dsp->echoVolumeR = 0; dsp->feedbackVolume = 0; dsp->echoBufferAdr = 0; dsp->echoDelay = 1; dsp->echoRemain = 1; dsp->echoBufferIndex = 0; dsp->firBufferIndex = 0; memset(dsp->firValues, 0, sizeof(dsp->firValues)); memset(dsp->firBufferL, 0, sizeof(dsp->firBufferL)); memset(dsp->firBufferR, 0, sizeof(dsp->firBufferR)); memset(dsp->sampleBuffer, 0, sizeof(dsp->sampleBuffer)); dsp->sampleOffset = 0; } void dsp_saveload(Dsp *dsp, SaveLoadFunc *func, void *ctx) { func(ctx, &dsp->ram, sizeof(Dsp) - offsetof(Dsp, ram)); } void dsp_cycle(Dsp* dsp) { int totalL = 0; int totalR = 0; for(int i = 0; i < 8; i++) { dsp_cycleChannel(dsp, i); totalL += (dsp->channel[i].sampleOut * dsp->channel[i].volumeL) >> 6; totalR += (dsp->channel[i].sampleOut * dsp->channel[i].volumeR) >> 6; totalL = totalL < -0x8000 ? -0x8000 : (totalL > 0x7fff ? 0x7fff : totalL); // clamp 16-bit totalR = totalR < -0x8000 ? -0x8000 : (totalR > 0x7fff ? 0x7fff : totalR); // clamp 16-bit } totalL = (totalL * dsp->masterVolumeL) >> 7; totalR = (totalR * dsp->masterVolumeR) >> 7; totalL = totalL < -0x8000 ? -0x8000 : (totalL > 0x7fff ? 0x7fff : totalL); // clamp 16-bit totalR = totalR < -0x8000 ? -0x8000 : (totalR > 0x7fff ? 0x7fff : totalR); // clamp 16-bit dsp_handleEcho(dsp, &totalL, &totalR); if(dsp->mute) { totalL = 0; totalR = 0; } dsp_handleNoise(dsp); // put it in the samplebuffer, if space if (dsp->sampleOffset < 534) { dsp->sampleBuffer[dsp->sampleOffset * 2] = totalL; dsp->sampleBuffer[dsp->sampleOffset * 2 + 1] = totalR; dsp->sampleOffset++; } dsp->evenCycle = !dsp->evenCycle; } static void dsp_handleEcho(Dsp* dsp, int* outputL, int* outputR) { // get value out of ram uint16_t adr = dsp->echoBufferAdr + dsp->echoBufferIndex * 4; dsp->firBufferL[dsp->firBufferIndex] = ( dsp->apu_ram[adr] + (dsp->apu_ram[(adr + 1) & 0xffff] << 8) ); dsp->firBufferL[dsp->firBufferIndex] >>= 1; dsp->firBufferR[dsp->firBufferIndex] = ( dsp->apu_ram[(adr + 2) & 0xffff] + (dsp->apu_ram[(adr + 3) & 0xffff] << 8) ); dsp->firBufferR[dsp->firBufferIndex] >>= 1; // calculate FIR-sum int sumL = 0, sumR = 0; for(int i = 0; i < 8; i++) { sumL += (dsp->firBufferL[(dsp->firBufferIndex + i + 1) & 0x7] * dsp->firValues[i]) >> 6; sumR += (dsp->firBufferR[(dsp->firBufferIndex + i + 1) & 0x7] * dsp->firValues[i]) >> 6; if(i == 6) { // clip to 16-bit before last addition sumL = ((int16_t) (sumL & 0xffff)); // clip 16-bit sumR = ((int16_t) (sumR & 0xffff)); // clip 16-bit } } sumL = sumL < -0x8000 ? -0x8000 : (sumL > 0x7fff ? 0x7fff : sumL); // clamp 16-bit sumR = sumR < -0x8000 ? -0x8000 : (sumR > 0x7fff ? 0x7fff : sumR); // clamp 16-bit // modify output with sum int outL = *outputL + ((sumL * dsp->echoVolumeL) >> 7); int outR = *outputR + ((sumR * dsp->echoVolumeR) >> 7); *outputL = outL < -0x8000 ? -0x8000 : (outL > 0x7fff ? 0x7fff : outL); // clamp 16-bit *outputR = outR < -0x8000 ? -0x8000 : (outR > 0x7fff ? 0x7fff : outR); // clamp 16-bit // get echo input int inL = 0, inR = 0; for(int i = 0; i < 8; i++) { if(dsp->channel[i].echoEnable) { inL += (dsp->channel[i].sampleOut * dsp->channel[i].volumeL) >> 6; inR += (dsp->channel[i].sampleOut * dsp->channel[i].volumeR) >> 6; inL = inL < -0x8000 ? -0x8000 : (inL > 0x7fff ? 0x7fff : inL); // clamp 16-bit inR = inR < -0x8000 ? -0x8000 : (inR > 0x7fff ? 0x7fff : inR); // clamp 16-bit } } // write this to ram inL += (sumL * dsp->feedbackVolume) >> 7; inR += (sumR * dsp->feedbackVolume) >> 7; inL = inL < -0x8000 ? -0x8000 : (inL > 0x7fff ? 0x7fff : inL); // clamp 16-bit inR = inR < -0x8000 ? -0x8000 : (inR > 0x7fff ? 0x7fff : inR); // clamp 16-bit inL &= 0xfffe; inR &= 0xfffe; if(dsp->echoWrites) { dsp->apu_ram[adr] = inL & 0xff; dsp->apu_ram[(adr + 1) & 0xffff] = inL >> 8; dsp->apu_ram[(adr + 2) & 0xffff] = inR & 0xff; dsp->apu_ram[(adr + 3) & 0xffff] = inR >> 8; } // handle indexes dsp->firBufferIndex++; dsp->firBufferIndex &= 7; dsp->echoBufferIndex++; dsp->echoRemain--; if(dsp->echoRemain == 0) { dsp->echoRemain = dsp->echoDelay; dsp->echoBufferIndex = 0; } } static void dsp_cycleChannel(Dsp* dsp, int ch) { // handle pitch counter uint16_t pitch = dsp->channel[ch].pitch; if(ch > 0 && dsp->channel[ch].pitchModulation) { int factor = (dsp->channel[ch - 1].sampleOut >> 4) + 0x400; pitch = (pitch * factor) >> 10; if(pitch > 0x3fff) pitch = 0x3fff; } int newCounter = dsp->channel[ch].pitchCounter + pitch; if(newCounter > 0xffff) { // next sample dsp_decodeBrr(dsp, ch); } dsp->channel[ch].pitchCounter = newCounter; int16_t sample = 0; if(dsp->channel[ch].useNoise) { sample = dsp->noiseSample; } else { sample = dsp_getSample(dsp, ch, dsp->channel[ch].pitchCounter >> 12, (dsp->channel[ch].pitchCounter >> 4) & 0xff); } #if !MY_CHANGES if(dsp->evenCycle) { // handle keyon/off (every other cycle) if(dsp->channel[ch].keyOff) { // go to release dsp->channel[ch].adsrState = 4; } else if(dsp->channel[ch].keyOn) { dsp->channel[ch].keyOn = false; // restart current sample dsp->channel[ch].previousFlags = 0; uint16_t samplePointer = dsp->dirPage + 4 * dsp->channel[ch].srcn; dsp->channel[ch].decodeOffset = dsp->apu_ram[samplePointer]; dsp->channel[ch].decodeOffset |= dsp->apu_ram[(samplePointer + 1) & 0xffff] << 8; memset(dsp->channel[ch].decodeBuffer, 0, sizeof(dsp->channel[ch].decodeBuffer)); dsp->channel[ch].gain = 0; dsp->channel[ch].adsrState = dsp->channel[ch].useGain ? 3 : 0; } } #endif // handle reset if(dsp->reset) { dsp->channel[ch].adsrState = 4; dsp->channel[ch].gain = 0; } // handle envelope/adsr bool doingDirectGain = dsp->channel[ch].adsrState != 4 && dsp->channel[ch].useGain && dsp->channel[ch].directGain; uint16_t rate = dsp->channel[ch].adsrState == 4 ? 0 : dsp->channel[ch].adsrRates[dsp->channel[ch].adsrState]; if(dsp->channel[ch].adsrState != 4 && !doingDirectGain && rate != 0) { dsp->channel[ch].rateCounter++; } if(dsp->channel[ch].adsrState == 4 || (!doingDirectGain && dsp->channel[ch].rateCounter >= rate && rate != 0)) { if(dsp->channel[ch].adsrState != 4) dsp->channel[ch].rateCounter = 0; dsp_handleGain(dsp, ch); } if(doingDirectGain) dsp->channel[ch].gain = dsp->channel[ch].gainValue; // set outputs dsp->ram[(ch << 4) | 8] = dsp->channel[ch].gain >> 4; sample = (sample * dsp->channel[ch].gain) >> 11; dsp->ram[(ch << 4) | 9] = sample >> 7; dsp->channel[ch].sampleOut = sample; } static void dsp_handleGain(Dsp* dsp, int ch) { switch(dsp->channel[ch].adsrState) { case 0: { // attack uint16_t rate = dsp->channel[ch].adsrRates[dsp->channel[ch].adsrState]; dsp->channel[ch].gain += rate == 1 ? 1024 : 32; if(dsp->channel[ch].gain >= 0x7e0) dsp->channel[ch].adsrState = 1; if(dsp->channel[ch].gain > 0x7ff) dsp->channel[ch].gain = 0x7ff; break; } case 1: { // decay dsp->channel[ch].gain -= ((dsp->channel[ch].gain - 1) >> 8) + 1; if(dsp->channel[ch].gain < dsp->channel[ch].sustainLevel) dsp->channel[ch].adsrState = 2; break; } case 2: { // sustain dsp->channel[ch].gain -= ((dsp->channel[ch].gain - 1) >> 8) + 1; break; } case 3: { // gain switch(dsp->channel[ch].gainMode) { case 0: { // linear decrease dsp->channel[ch].gain -= 32; // decreasing below 0 will underflow to above 0x7ff if(dsp->channel[ch].gain > 0x7ff) dsp->channel[ch].gain = 0; break; } case 1: { // exponential decrease dsp->channel[ch].gain -= ((dsp->channel[ch].gain - 1) >> 8) + 1; break; } case 2: { // linear increase dsp->channel[ch].gain += 32; if(dsp->channel[ch].gain > 0x7ff) dsp->channel[ch].gain = 0; break; } case 3: { // bent increase dsp->channel[ch].gain += dsp->channel[ch].gain < 0x600 ? 32 : 8; if(dsp->channel[ch].gain > 0x7ff) dsp->channel[ch].gain = 0; break; } } break; } case 4: { // release dsp->channel[ch].gain -= 8; // decreasing below 0 will underflow to above 0x7ff if(dsp->channel[ch].gain > 0x7ff) dsp->channel[ch].gain = 0; break; } } } static int16_t dsp_getSample(Dsp* dsp, int ch, int sampleNum, int offset) { int16_t news = dsp->channel[ch].decodeBuffer[sampleNum + 3]; int16_t olds = dsp->channel[ch].decodeBuffer[sampleNum + 2]; int16_t olders = dsp->channel[ch].decodeBuffer[sampleNum + 1]; int16_t oldests = dsp->channel[ch].decodeBuffer[sampleNum]; int out = (gaussValues[0xff - offset] * oldests) >> 10; out += (gaussValues[0x1ff - offset] * olders) >> 10; out += (gaussValues[0x100 + offset] * olds) >> 10; out = ((int16_t) (out & 0xffff)); // clip 16-bit out += (gaussValues[offset] * news) >> 10; out = out < -0x8000 ? -0x8000 : (out > 0x7fff ? 0x7fff : out); // clamp 16-bit return out >> 1; } static void dsp_decodeBrr(Dsp* dsp, int ch) { // copy last 3 samples (16-18) to first 3 for interpolation dsp->channel[ch].decodeBuffer[0] = dsp->channel[ch].decodeBuffer[16]; dsp->channel[ch].decodeBuffer[1] = dsp->channel[ch].decodeBuffer[17]; dsp->channel[ch].decodeBuffer[2] = dsp->channel[ch].decodeBuffer[18]; // handle flags from previous block if(dsp->channel[ch].previousFlags == 1 || dsp->channel[ch].previousFlags == 3) { // loop sample uint16_t samplePointer = dsp->dirPage + 4 * dsp->channel[ch].srcn; dsp->channel[ch].decodeOffset = dsp->apu_ram[(samplePointer + 2) & 0xffff]; dsp->channel[ch].decodeOffset |= (dsp->apu_ram[(samplePointer + 3) & 0xffff]) << 8; if(dsp->channel[ch].previousFlags == 1) { // also release and clear gain dsp->channel[ch].adsrState = 4; dsp->channel[ch].gain = 0; } dsp->ram[ENDX] |= 1 << ch; // set ENDX bit for channel } uint8_t header = dsp->apu_ram[dsp->channel[ch].decodeOffset++]; int shift = header >> 4; int filter = (header & 0xc) >> 2; dsp->channel[ch].previousFlags = header & 0x3; uint8_t curByte = 0; int old = dsp->channel[ch].old; int older = dsp->channel[ch].older; for(int i = 0; i < 16; i++) { int s = 0; if(i & 1) { s = curByte & 0xf; } else { curByte = dsp->apu_ram[dsp->channel[ch].decodeOffset++]; s = curByte >> 4; } if(s > 7) s -= 16; if(shift <= 0xc) { s = (s << shift) >> 1; } else { s = (s >> 3) << 12; } switch(filter) { case 1: s += old + (-old >> 4); break; case 2: s += 2 * old + ((3 * -old) >> 5) - older + (older >> 4); break; case 3: s += 2 * old + ((13 * -old) >> 6) - older + ((3 * older) >> 4); break; } s = s < -0x8000 ? -0x8000 : (s > 0x7fff ? 0x7fff : s); // clamp 16-bit s = ((int16_t) ((s & 0x7fff) << 1)) >> 1; // clip 15-bit older = old; old = s; dsp->channel[ch].decodeBuffer[i + 3] = s; } dsp->channel[ch].older = older; dsp->channel[ch].old = old; } static void dsp_handleNoise(Dsp* dsp) { if(dsp->noiseRate != 0) { dsp->noiseCounter++; } if(dsp->noiseCounter >= dsp->noiseRate && dsp->noiseRate != 0) { int bit = (dsp->noiseSample & 1) ^ ((dsp->noiseSample >> 1) & 1); dsp->noiseSample = ((dsp->noiseSample >> 1) & 0x3fff) | (bit << 14); dsp->noiseSample = ((int16_t) ((dsp->noiseSample & 0x7fff) << 1)) >> 1; dsp->noiseCounter = 0; } } uint8_t dsp_read(Dsp* dsp, uint8_t adr) { return dsp->ram[adr]; } void dsp_write(Dsp *dsp, uint8_t adr, uint8_t val) { int ch = adr >> 4; switch (adr) { case V0VOLL: case V1VOLL: case V2VOLL: case V3VOLL: case V4VOLL: case V5VOLL: case V6VOLL: case V7VOLL: { dsp->channel[ch].volumeL = val; break; } case V0VOLR: case V1VOLR: case V2VOLR: case V3VOLR: case V4VOLR: case V5VOLR: case V6VOLR: case V7VOLR: { dsp->channel[ch].volumeR = val; break; } case V0PITCHL: case V1PL: case V2PL: case V3PL: case V4PL: case V5PL: case V6PL: case V7PL: { dsp->channel[ch].pitch = (dsp->channel[ch].pitch & 0x3f00) | val; break; } case V0PITCHH: case V1PH: case V2PH: case V3PH: case V4PH: case V5PH: case V6PH: case V7PH: { dsp->channel[ch].pitch = ((dsp->channel[ch].pitch & 0x00ff) | (val << 8)) & 0x3fff; break; } case V0SRCN: case V1SRCN: case V2SRCN: case V3SRCN: case V4SRCN: case V5SRCN: case V6SRCN: case V7SRCN: { dsp->channel[ch].srcn = val; break; } case V0ADSR1: case V1ADSR1: case V2ADSR1: case V3ADSR1: case V4ADSR1: case V5ADSR1: case V6ADSR1: case V7ADSR1: { dsp->channel[ch].adsrRates[0] = rateValues[(val & 0xf) * 2 + 1]; dsp->channel[ch].adsrRates[1] = rateValues[((val & 0x70) >> 4) * 2 + 16]; dsp->channel[ch].useGain = (val & 0x80) == 0; break; } case V0ADSR2: case V1ADSR2: case V2ADSR2: case V3ADSR2: case V4ADSR2: case V5ADSR2: case V6ADSR2: case V7ADSR2: { dsp->channel[ch].adsrRates[2] = rateValues[val & 0x1f]; dsp->channel[ch].sustainLevel = (((val & 0xe0) >> 5) + 1) * 0x100; break; } case V0GAIN: case V1GAIN: case V2GAIN: case V3GAIN: case V4GAIN: case V5GAIN: case V6GAIN: case V7GAIN: { dsp->channel[ch].directGain = (val & 0x80) == 0; if (val & 0x80) { dsp->channel[ch].gainMode = (val & 0x60) >> 5; dsp->channel[ch].adsrRates[3] = rateValues[val & 0x1f]; } else { dsp->channel[ch].gainValue = (val & 0x7f) * 16; } break; } case MVOLL: { dsp->masterVolumeL = val; break; } case MVOLR: { dsp->masterVolumeR = val; break; } case EVOLL: { dsp->echoVolumeL = val; break; } case EVOLR: { dsp->echoVolumeR = val; break; } case KON: { for (int ch = 0; ch < 8; ch++) { dsp->channel[ch].keyOn = val & (1 << ch); #if MY_CHANGES if (dsp->channel[ch].keyOn) { dsp->channel[ch].keyOn = false; // restart current sample dsp->channel[ch].previousFlags = 0; uint16_t samplePointer = dsp->dirPage + 4 * dsp->channel[ch].srcn; dsp->channel[ch].decodeOffset = dsp->apu_ram[samplePointer]; dsp->channel[ch].decodeOffset |= dsp->apu_ram[(samplePointer + 1) & 0xffff] << 8; memset(dsp->channel[ch].decodeBuffer, 0, sizeof(dsp->channel[ch].decodeBuffer)); dsp->channel[ch].gain = 0; dsp->channel[ch].adsrState = dsp->channel[ch].useGain ? 3 : 0; } #endif } break; } case KOF: { for (int ch = 0; ch < 8; ch++) { dsp->channel[ch].keyOff = val & (1 << ch); #if MY_CHANGES if (dsp->channel[ch].keyOff) { // go to release dsp->channel[ch].adsrState = 4; } #endif } break; } case FLG: { dsp->reset = val & 0x80; dsp->mute = val & 0x40; dsp->echoWrites = (val & 0x20) == 0; dsp->noiseRate = rateValues[val & 0x1f]; break; } case ENDX: { val = 0; // any write clears ENDx break; } case EFB: { dsp->feedbackVolume = val; break; } case PMON: { for (int i = 0; i < 8; i++) { dsp->channel[i].pitchModulation = val & (1 << i); } break; } case NON: { for (int i = 0; i < 8; i++) { dsp->channel[i].useNoise = val & (1 << i); } break; } case EON: { for (int i = 0; i < 8; i++) { dsp->channel[i].echoEnable = val & (1 << i); } break; } case DIR: { dsp->dirPage = val << 8; break; } case ESA: { dsp->echoBufferAdr = val << 8; break; } case EDL: { dsp->echoDelay = (val & 0xf) * 512; // 2048-byte steps, stereo sample is 4 bytes if (dsp->echoDelay == 0) dsp->echoDelay = 1; break; } case FIR0: case FIR1: case FIR2: case FIR3: case FIR4: case FIR5: case FIR6: case FIR7: { dsp->firValues[ch] = val; break; } } dsp->ram[adr] = val; } void dsp_getSamples(Dsp* dsp, int16_t* sampleData, int samplesPerFrame, int numChannels) { // resample from 534 samples per frame to wanted value float adder = 534.0f / samplesPerFrame; float location = 0.0f; if (numChannels == 1) { for (int i = 0; i < samplesPerFrame; i++) { int sampleL = dsp->sampleBuffer[((int)location) * 2]; int sampleR = dsp->sampleBuffer[((int)location) * 2 + 1]; sampleData[i] = (sampleL + sampleR) >> 1; location += adder; } } else { for (int i = 0; i < samplesPerFrame; i++) { int sampleL = dsp->sampleBuffer[((int)location) * 2]; int sampleR = dsp->sampleBuffer[((int)location) * 2 + 1]; sampleData[i * 2] = sampleL; sampleData[i * 2 + 1] = sampleR; location += adder; } } dsp->sampleOffset = 0; }