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src/sound/munt/LA32FloatWaveGenerator.cpp
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359
src/sound/munt/LA32FloatWaveGenerator.cpp
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/* Copyright (C) 2003, 2004, 2005, 2006, 2008, 2009 Dean Beeler, Jerome Fisher
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* Copyright (C) 2011-2017 Dean Beeler, Jerome Fisher, Sergey V. Mikayev
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*
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* This program is free software: you can redistribute it and/or modify
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* it under the terms of the GNU Lesser General Public License as published by
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* the Free Software Foundation, either version 2.1 of the License, or
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* (at your option) any later version.
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*
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* This program is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with this program. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <cstddef>
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#include "internals.h"
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#include "LA32FloatWaveGenerator.h"
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#include "mmath.h"
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#include "Tables.h"
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namespace MT32Emu {
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static const float MIDDLE_CUTOFF_VALUE = 128.0f;
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static const float RESONANCE_DECAY_THRESHOLD_CUTOFF_VALUE = 144.0f;
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static const float MAX_CUTOFF_VALUE = 240.0f;
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float LA32FloatWaveGenerator::getPCMSample(unsigned int position) {
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if (position >= pcmWaveLength) {
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if (!pcmWaveLooped) {
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return 0;
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}
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position = position % pcmWaveLength;
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}
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Bit16s pcmSample = pcmWaveAddress[position];
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float sampleValue = EXP2F(((pcmSample & 32767) - 32787.0f) / 2048.0f);
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return ((pcmSample & 32768) == 0) ? sampleValue : -sampleValue;
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}
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void LA32FloatWaveGenerator::initSynth(const bool useSawtoothWaveform, const Bit8u usePulseWidth, const Bit8u useResonance) {
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sawtoothWaveform = useSawtoothWaveform;
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pulseWidth = usePulseWidth;
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resonance = useResonance;
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wavePos = 0.0f;
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lastFreq = 0.0f;
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pcmWaveAddress = NULL;
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active = true;
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}
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void LA32FloatWaveGenerator::initPCM(const Bit16s * const usePCMWaveAddress, const Bit32u usePCMWaveLength, const bool usePCMWaveLooped, const bool usePCMWaveInterpolated) {
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pcmWaveAddress = usePCMWaveAddress;
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pcmWaveLength = usePCMWaveLength;
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pcmWaveLooped = usePCMWaveLooped;
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pcmWaveInterpolated = usePCMWaveInterpolated;
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pcmPosition = 0.0f;
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active = true;
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}
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// ampVal - Logarithmic amp of the wave generator
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// pitch - Logarithmic frequency of the resulting wave
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// cutoffRampVal - Composed of the base cutoff in range [78..178] left-shifted by 18 bits and the TVF modifier
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float LA32FloatWaveGenerator::generateNextSample(const Bit32u ampVal, const Bit16u pitch, const Bit32u cutoffRampVal) {
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if (!active) {
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return 0.0f;
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}
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float sample = 0.0f;
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// SEMI-CONFIRMED: From sample analysis:
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// (1) Tested with a single partial playing PCM wave 77 with pitchCoarse 36 and no keyfollow, velocity follow, etc.
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// This gives results within +/- 2 at the output (before any DAC bitshifting)
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// when sustaining at levels 156 - 255 with no modifiers.
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// (2) Tested with a special square wave partial (internal capture ID tva5) at TVA envelope levels 155-255.
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// This gives deltas between -1 and 0 compared to the real output. Note that this special partial only produces
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// positive amps, so negative still needs to be explored, as well as lower levels.
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//
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// Also still partially unconfirmed is the behaviour when ramping between levels, as well as the timing.
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float amp = EXP2F(ampVal / -1024.0f / 4096.0f);
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float freq = EXP2F(pitch / 4096.0f - 16.0f) * SAMPLE_RATE;
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if (isPCMWave()) {
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// Render PCM waveform
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int len = pcmWaveLength;
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int intPCMPosition = int(pcmPosition);
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if (intPCMPosition >= len && !pcmWaveLooped) {
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// We're now past the end of a non-looping PCM waveform so it's time to die.
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deactivate();
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return 0.0f;
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}
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float positionDelta = freq * 2048.0f / SAMPLE_RATE;
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// Linear interpolation
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float firstSample = getPCMSample(intPCMPosition);
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// We observe that for partial structures with ring modulation the interpolation is not applied to the slave PCM partial.
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// It's assumed that the multiplication circuitry intended to perform the interpolation on the slave PCM partial
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// is borrowed by the ring modulation circuit (or the LA32 chip has a similar lack of resources assigned to each partial pair).
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if (pcmWaveInterpolated) {
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sample = firstSample + (getPCMSample(intPCMPosition + 1) - firstSample) * (pcmPosition - intPCMPosition);
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} else {
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sample = firstSample;
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}
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float newPCMPosition = pcmPosition + positionDelta;
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if (pcmWaveLooped) {
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newPCMPosition = fmod(newPCMPosition, float(pcmWaveLength));
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}
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pcmPosition = newPCMPosition;
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} else {
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// Render synthesised waveform
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wavePos *= lastFreq / freq;
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lastFreq = freq;
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float resAmp = EXP2F(1.0f - (32 - resonance) / 4.0f);
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{
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//static const float resAmpFactor = EXP2F(-7);
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//resAmp = EXP2I(resonance << 10) * resAmpFactor;
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}
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// The cutoffModifier may not be supposed to be directly added to the cutoff -
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// it may for example need to be multiplied in some way.
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// The 240 cutoffVal limit was determined via sample analysis (internal Munt capture IDs: glop3, glop4).
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// More research is needed to be sure that this is correct, however.
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float cutoffVal = cutoffRampVal / 262144.0f;
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if (cutoffVal > MAX_CUTOFF_VALUE) {
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cutoffVal = MAX_CUTOFF_VALUE;
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}
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// Wave length in samples
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float waveLen = SAMPLE_RATE / freq;
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// Init cosineLen
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float cosineLen = 0.5f * waveLen;
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if (cutoffVal > MIDDLE_CUTOFF_VALUE) {
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cosineLen *= EXP2F((cutoffVal - MIDDLE_CUTOFF_VALUE) / -16.0f); // found from sample analysis
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}
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// Start playing in center of first cosine segment
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// relWavePos is shifted by a half of cosineLen
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float relWavePos = wavePos + 0.5f * cosineLen;
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if (relWavePos > waveLen) {
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relWavePos -= waveLen;
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}
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// Ratio of positive segment to wave length
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float pulseLen = 0.5f;
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if (pulseWidth > 128) {
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pulseLen = EXP2F((64 - pulseWidth) / 64.0f);
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//static const float pulseLenFactor = EXP2F(-192 / 64);
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//pulseLen = EXP2I((256 - pulseWidthVal) << 6) * pulseLenFactor;
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}
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pulseLen *= waveLen;
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float hLen = pulseLen - cosineLen;
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// Ignore pulsewidths too high for given freq
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if (hLen < 0.0f) {
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hLen = 0.0f;
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}
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// Correct resAmp for cutoff in range 50..66
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if ((cutoffVal >= MIDDLE_CUTOFF_VALUE) && (cutoffVal < RESONANCE_DECAY_THRESHOLD_CUTOFF_VALUE)) {
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resAmp *= sin(FLOAT_PI * (cutoffVal - MIDDLE_CUTOFF_VALUE) / 32.0f);
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}
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// Produce filtered square wave with 2 cosine waves on slopes
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// 1st cosine segment
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if (relWavePos < cosineLen) {
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sample = -cos(FLOAT_PI * relWavePos / cosineLen);
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} else
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// high linear segment
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if (relWavePos < (cosineLen + hLen)) {
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sample = 1.f;
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} else
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// 2nd cosine segment
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if (relWavePos < (2 * cosineLen + hLen)) {
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sample = cos(FLOAT_PI * (relWavePos - (cosineLen + hLen)) / cosineLen);
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} else {
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// low linear segment
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sample = -1.f;
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}
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if (cutoffVal < MIDDLE_CUTOFF_VALUE) {
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// Attenuate samples below cutoff 50
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// Found by sample analysis
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sample *= EXP2F(-0.125f * (MIDDLE_CUTOFF_VALUE - cutoffVal));
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} else {
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// Add resonance sine. Effective for cutoff > 50 only
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float resSample = 1.0f;
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// Resonance decay speed factor
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float resAmpDecayFactor = Tables::getInstance().resAmpDecayFactor[resonance >> 2];
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// Now relWavePos counts from the middle of first cosine
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relWavePos = wavePos;
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// negative segments
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if (!(relWavePos < (cosineLen + hLen))) {
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resSample = -resSample;
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relWavePos -= cosineLen + hLen;
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// From the digital captures, the decaying speed of the resonance sine is found a bit different for the positive and the negative segments
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resAmpDecayFactor += 0.25f;
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}
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// Resonance sine WG
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resSample *= sin(FLOAT_PI * relWavePos / cosineLen);
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// Resonance sine amp
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float resAmpFadeLog2 = -0.125f * resAmpDecayFactor * (relWavePos / cosineLen); // seems to be exact
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float resAmpFade = EXP2F(resAmpFadeLog2);
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// Now relWavePos set negative to the left from center of any cosine
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relWavePos = wavePos;
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// negative segment
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if (!(wavePos < (waveLen - 0.5f * cosineLen))) {
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relWavePos -= waveLen;
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} else
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// positive segment
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if (!(wavePos < (hLen + 0.5f * cosineLen))) {
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relWavePos -= cosineLen + hLen;
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}
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// To ensure the output wave has no breaks, two different windows are appied to the beginning and the ending of the resonance sine segment
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if (relWavePos < 0.5f * cosineLen) {
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float syncSine = sin(FLOAT_PI * relWavePos / cosineLen);
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if (relWavePos < 0.0f) {
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// The window is synchronous square sine here
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resAmpFade *= syncSine * syncSine;
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} else {
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// The window is synchronous sine here
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resAmpFade *= syncSine;
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}
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}
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sample += resSample * resAmp * resAmpFade;
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}
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// sawtooth waves
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if (sawtoothWaveform) {
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sample *= cos(FLOAT_2PI * wavePos / waveLen);
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}
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wavePos++;
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// wavePos isn't supposed to be > waveLen
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if (wavePos > waveLen) {
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wavePos -= waveLen;
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}
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}
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// Multiply sample with current TVA value
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sample *= amp;
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return sample;
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}
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void LA32FloatWaveGenerator::deactivate() {
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active = false;
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}
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bool LA32FloatWaveGenerator::isActive() const {
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return active;
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}
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bool LA32FloatWaveGenerator::isPCMWave() const {
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return pcmWaveAddress != NULL;
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}
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void LA32FloatPartialPair::init(const bool useRingModulated, const bool useMixed) {
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ringModulated = useRingModulated;
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mixed = useMixed;
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masterOutputSample = 0.0f;
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slaveOutputSample = 0.0f;
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}
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void LA32FloatPartialPair::initSynth(const PairType useMaster, const bool sawtoothWaveform, const Bit8u pulseWidth, const Bit8u resonance) {
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if (useMaster == MASTER) {
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master.initSynth(sawtoothWaveform, pulseWidth, resonance);
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} else {
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slave.initSynth(sawtoothWaveform, pulseWidth, resonance);
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}
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}
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void LA32FloatPartialPair::initPCM(const PairType useMaster, const Bit16s *pcmWaveAddress, const Bit32u pcmWaveLength, const bool pcmWaveLooped) {
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if (useMaster == MASTER) {
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master.initPCM(pcmWaveAddress, pcmWaveLength, pcmWaveLooped, true);
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} else {
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slave.initPCM(pcmWaveAddress, pcmWaveLength, pcmWaveLooped, !ringModulated);
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}
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}
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void LA32FloatPartialPair::generateNextSample(const PairType useMaster, const Bit32u amp, const Bit16u pitch, const Bit32u cutoff) {
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if (useMaster == MASTER) {
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masterOutputSample = master.generateNextSample(amp, pitch, cutoff);
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} else {
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slaveOutputSample = slave.generateNextSample(amp, pitch, cutoff);
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}
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}
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static inline float produceDistortedSample(float sample) {
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if (sample < -1.0f) {
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return sample + 2.0f;
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} else if (1.0f < sample) {
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return sample - 2.0f;
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}
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return sample;
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}
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float LA32FloatPartialPair::nextOutSample() {
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// Note, LA32FloatWaveGenerator produces each sample normalised in terms of a single playing partial,
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// so the unity sample corresponds to the internal LA32 logarithmic fixed-point unity sample.
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// However, each logarithmic sample is then unlogged to a 14-bit signed integer value, i.e. the max absolute value is 8192.
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// Thus, considering that samples are further mapped to a 16-bit signed integer,
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// we apply a conversion factor 0.25 to produce properly normalised float samples.
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if (!ringModulated) {
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return 0.25f * (masterOutputSample + slaveOutputSample);
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}
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/*
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* SEMI-CONFIRMED: Ring modulation model derived from sample analysis of specially constructed patches which exploit distortion.
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* LA32 ring modulator found to produce distorted output in case if the absolute value of maximal amplitude of one of the input partials exceeds 8191.
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* This is easy to reproduce using synth partials with resonance values close to the maximum. It looks like an integer overflow happens in this case.
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* As the distortion is strictly bound to the amplitude of the complete mixed square + resonance wave in the linear space,
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* it is reasonable to assume the ring modulation is performed also in the linear space by sample multiplication.
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* Most probably the overflow is caused by limited precision of the multiplication circuit as the very similar distortion occurs with panning.
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*/
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float ringModulatedSample = produceDistortedSample(masterOutputSample) * produceDistortedSample(slaveOutputSample);
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return 0.25f * (mixed ? masterOutputSample + ringModulatedSample : ringModulatedSample);
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}
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void LA32FloatPartialPair::deactivate(const PairType useMaster) {
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if (useMaster == MASTER) {
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master.deactivate();
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masterOutputSample = 0.0f;
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} else {
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slave.deactivate();
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slaveOutputSample = 0.0f;
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}
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}
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bool LA32FloatPartialPair::isActive(const PairType useMaster) const {
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return useMaster == MASTER ? master.isActive() : slave.isActive();
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}
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} // namespace MT32Emu
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