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663
Source/DSP/UniversalEngine.cpp
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663
Source/DSP/UniversalEngine.cpp
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#include "UniversalEngine.h"
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namespace FDNReverb {
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namespace {
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static bool isMathPrime(int n) noexcept {
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if (n < 2) return false;
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if (n == 2) return true;
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if (n % 2 == 0) return false;
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for (int i = 3; i * i <= n; i += 2)
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if (n % i == 0) return false;
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return true;
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}
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static int findNearestUniquePrime(int target,
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const std::array<int, 16>& usedPrimes,
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int usedCount) noexcept {
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target = std::max(target, 2);
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for (int offset = 0; offset < 100000; ++offset) {
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int hi = target + offset;
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if (isMathPrime(hi)) {
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bool used = false;
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for (int k = 0; k < usedCount; ++k)
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if (usedPrimes[k] == hi) { used = true; break; }
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if (!used) return hi;
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}
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int lo = target - offset;
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if (offset > 0 && lo >= 2 && isMathPrime(lo)) {
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bool used = false;
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for (int k = 0; k < usedCount; ++k)
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if (usedPrimes[k] == lo) { used = true; break; }
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if (!used) return lo;
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}
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}
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return target;
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}
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} // anonymous namespace
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UniversalEngine::UniversalEngine() {
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fbVec.fill(0.0f);
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constexpr float phi = 1.6180339887f;
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for (int i = 0; i < FDN_ORDER; ++i) {
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lfos[i].state = 12345u + static_cast<uint32_t>(i) * 9876u;
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lfos[i].smoothed = 0.0f;
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const float angle = static_cast<float>(i) * phi;
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const float frac = angle - std::floor(angle);
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lfos[i].rateMultiplier = 0.80f + frac * 0.40f;
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// * LFO: noise LFO offset
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const float cAngle = static_cast<float>(i + 5) * phi;
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chorusLFOs[i].phase = cAngle - std::floor(cAngle);
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const float cRateAngle = static_cast<float>(i + 11) * phi;
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chorusLFOs[i].rateScale = 0.30f + (cRateAngle - std::floor(cRateAngle)) * 0.50f;
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}
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}
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void UniversalEngine::prepare(double sampleRate, int /*maxBlockSize*/) {
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fs = sampleRate;
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#if AMBIVALENCE_USE_STAGE2_ABSORPTION
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MagnitudeResponseFitter::precomputeInteractionMatrix(sampleRate);
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#endif
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auto getPow2 = [](size_t s) -> size_t {
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size_t p = 1;
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while (p < s) p *= 2;
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return p;
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};
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size_t totalMemoryNeeded =
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getPow2(static_cast<size_t>(fs * 0.5)) // * preDelay (max 500ms)
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+ getPow2(static_cast<size_t>(fs * 1.0))
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+ getPow2(static_cast<size_t>(fs * 0.05)) * 4
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+ getPow2(static_cast<size_t>(fs * 0.5)) * FDN_ORDER
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+ getPow2(static_cast<size_t>(fs * 0.05)) * FDN_ORDER * SERIAL_APF_STAGES;
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memoryPool.allocate(totalMemoryNeeded);
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int mask = 0;
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float* ptr = nullptr;
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// * PreDelay (max 500ms)
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ptr = memoryPool.requestMemory(static_cast<size_t>(fs * 0.5), mask);
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preDelayLine.init(ptr, mask);
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ptr = memoryPool.requestMemory(static_cast<size_t>(fs * 1.0), mask);
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erDelay.init(ptr, mask);
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for (int i = 0; i < 4; ++i) {
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ptr = memoryPool.requestMemory(static_cast<size_t>(fs * 0.05), mask);
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inputDiffusers[i].init(ptr, mask);
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}
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for (int i = 0; i < FDN_ORDER; ++i) {
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ptr = memoryPool.requestMemory(static_cast<size_t>(fs * 0.5), mask);
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fdnDelays[i].init(ptr, mask);
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for (int s = 0; s < SERIAL_APF_STAGES; ++s) {
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ptr = memoryPool.requestMemory(static_cast<size_t>(fs * 0.05), mask);
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nestedAllpassDelays[i][s].init(ptr, mask);
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}
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}
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acousticMetrics.prepare(sampleRate, 2000.0f);
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currentERTapCount = 0;
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currentERDelaySamples.fill(0.0f);
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currentERGains.fill(0.0f);
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outputLimiter.prepare(sampleRate);
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outputEQ.prepare(sampleRate);
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duckingAttackCoeff = 1.0f - std::exp(-1.0f / (static_cast<float>(fs) * 0.010f));
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duckingReleaseCoeff = 1.0f - std::exp(-1.0f / (static_cast<float>(fs) * 0.200f));
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duckingEnvelope = 0.0f;
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// * DC coefficient : fc ~ 5Hz 1HPF
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dcBlockerCoeff = 1.0f - (6.28318530718f * 5.0f / static_cast<float>(fs));
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dcX1.fill(0.0f);
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dcY1.fill(0.0f);
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// * Soft-knee: RMS envelope coefficient (~3ms)
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fdnRmsEnv.fill(0.0f);
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rmsCoeff = 1.0f - std::exp(-1.0f / (static_cast<float>(fs) * 0.003f));
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reset();
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}
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void UniversalEngine::reset() {
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memoryPool.clear();
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fbVec.fill(0.0f);
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#if AMBIVALENCE_USE_STAGE2_ABSORPTION
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for (auto& lineFilters : absorptionFiltersS2)
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for (auto& f : lineFilters) f.reset();
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#else
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for (auto& f : absorptionFilters) f.reset();
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#endif
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acousticMetrics.reset();
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saturatorL.reset();
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saturatorR.reset();
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outputLimiter.reset();
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outputEQ.reset();
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duckingEnvelope = 0.0f;
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dcX1.fill(0.0f);
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dcY1.fill(0.0f);
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fdnRmsEnv.fill(0.0f);
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for (auto& dl : fdnDelays) dl.resetState(); // * Thiran allpass state
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for (auto& lfo : lfos) lfo.smoothed = 0.0f;
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}
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void UniversalEngine::setParams(const DSPParams& p) {
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activeParams = p;
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switch (p.algorithmIndex) {
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case 0: case 1: currentTopology = ReverbTopology::Room; break;
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case 2: case 3: currentTopology = ReverbTopology::Hall; break;
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case 4: currentTopology = ReverbTopology::Plate; break;
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case 5: currentTopology = ReverbTopology::Spring; break;
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case 6: currentTopology = ReverbTopology::Goldfoil; break;
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}
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const float attMs = juce::jmax(0.1f, p.duckingAttackMs);
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const float relMs = juce::jmax(0.1f, p.duckingRelMs);
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duckingAttackCoeff = 1.0f - std::exp(-1.0f / (static_cast<float>(fs) * attMs * 0.001f));
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duckingReleaseCoeff = 1.0f - std::exp(-1.0f / (static_cast<float>(fs) * relMs * 0.001f));
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// * PreDelay: ms -> sample count
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preDelaySamples = p.preDelayMs * 0.001f * static_cast<float>(fs);
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outputEQ.setLoCutHz(p.loCutHz);
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outputEQ.setHiCutHz(p.hiCutHz);
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updateTopologyAndRouting();
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}
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void UniversalEngine::calculatePrimePowerDelays() {
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const float fsf = static_cast<float>(fs);
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const float sizeCoeff = juce::jlimit(0.5f, 2.0f, activeParams.roomSizeScale + 1.0f);
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const float minDelayMs = 15.0f + sizeCoeff * 7.5f;
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const float maxDelayMs = 50.0f + sizeCoeff * 75.0f;
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const int minDelaySamples = std::max(11, static_cast<int>(minDelayMs * 0.001f * fsf));
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const int maxDelaySamples = static_cast<int>(maxDelayMs * 0.001f * fsf);
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const float logMin = std::log(static_cast<float>(minDelaySamples));
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const float logMax = std::log(static_cast<float>(maxDelaySamples));
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std::array<int, FDN_ORDER> usedPrimes;
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usedPrimes.fill(0);
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for (int i = 0; i < FDN_ORDER; ++i) {
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const float t = static_cast<float>(i) / static_cast<float>(FDN_ORDER - 1);
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const float logTgt = logMin + t * (logMax - logMin);
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const int target = static_cast<int>(std::round(std::exp(logTgt)));
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const int prime = findNearestUniquePrime(target, usedPrimes, i);
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usedPrimes[i] = prime;
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fdnBaseDelaySamples[i] = static_cast<float>(prime);
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}
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}
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void UniversalEngine::updateTopologyAndRouting() {
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calculatePrimePowerDelays();
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auto& preset = *ALL_PRESETS[activeParams.algorithmIndex];
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std::array<float, NUM_BANDS> scaledRT60 = preset.acoustics.rt60;
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for (auto& v : scaledRT60) v *= activeParams.decayScale;
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// -------------------------------------------------------------------------
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// * 2) fix : proMode always Tilt / band apply
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// -------------------------------------------------------------------------
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// old implementation : if (activeParams.proMode) { ... }
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// when ProMode is OFF, the Tilt / band coefficients were not applied,
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// so the RT60 graph kept the preset's original curve.
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//
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// new implementation: always apply; the coefficients default to 1.0f,
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// so changing them scales the RT60 graph,
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// and reset to 1.0f when loadPresetDefaults() is called.
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//
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// -------------------------------------------------------------------------
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scaledRT60[0] *= activeParams.tiltLow;
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scaledRT60[1] *= activeParams.tiltLow;
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scaledRT60[2] *= activeParams.tiltLow;
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scaledRT60[3] *= activeParams.tiltMid;
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scaledRT60[4] *= activeParams.tiltMid;
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scaledRT60[5] *= activeParams.tiltMid;
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scaledRT60[6] *= activeParams.tiltMid;
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scaledRT60[7] *= activeParams.tiltHigh;
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scaledRT60[8] *= activeParams.tiltHigh;
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scaledRT60[9] *= activeParams.tiltHigh;
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for (int b = 0; b < NUM_BANDS; ++b)
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scaledRT60[b] *= activeParams.rtBands[b];
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#if AMBIVALENCE_USE_STAGE2_ABSORPTION
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std::array<float, NUM_BANDS> targetDbAccum;
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targetDbAccum.fill(0.0f);
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for (int i = 0; i < FDN_ORDER; ++i) {
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auto s2 = MagnitudeResponseFitter::designStage2(
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static_cast<int>(fdnBaseDelaySamples[i]), fs, scaledRT60,
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activeParams.hfDamping, activeParams.lfAbsorption);
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for (int b = 0; b < NUM_BANDS; ++b) {
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currentAbsorptionCoeffsS2[i][b] = s2.geqStages[b];
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targetDbAccum[b] += s2.targetDb[b];
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}
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}
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const float representativeDelay = fdnBaseDelaySamples[FDN_ORDER / 2];
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for (int b = 0; b < NUM_BANDS; ++b) {
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const float avgTargetDb = targetDbAccum[b] / static_cast<float>(FDN_ORDER);
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if (avgTargetDb < -0.001f) {
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effectiveRT60[b] = -60.0f * representativeDelay
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/ (static_cast<float>(fs) * avgTargetDb);
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}
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else {
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effectiveRT60[b] = scaledRT60[b];
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}
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effectiveRT60[b] = juce::jlimit(0.05f, 30.0f, effectiveRT60[b]);
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}
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#else
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effectiveRT60 = scaledRT60;
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for (int i = 0; i < FDN_ORDER; ++i) {
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auto absoStages = FilterDesign::designAbsorption(
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static_cast<int>(fdnBaseDelaySamples[i]), fs, scaledRT60,
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activeParams.hfDamping, activeParams.lfAbsorption);
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currentAbsorptionCoeffs[i] = absoStages[0];
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}
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#endif
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// -------------------------------------------------------------------------
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// * EDT fix : band average LF/HF correction
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// -------------------------------------------------------------------------
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// old implementation : effectiveRT60[4] (500Hz) band use
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// -> HF Damping high band below EDT
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// -> LF Absorption low band below EDT
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//
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// new implementation : mid-band band (125Hz~4kHz = band 2~7) average value use
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// -> band LF/HF correction influence
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// -> (31Hz, 63Hz, 8kHz, 16kHz) ( psychoacoustically EDT
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// , value unstable )
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// -------------------------------------------------------------------------
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float rt60Mid = 0.0f;
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for (int b = 2; b <= 7; ++b)
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rt60Mid += effectiveRT60[b];
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rt60Mid = std::max(0.1f, rt60Mid / 6.0f);
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// -------------------------------------------------------------------------
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// * metallic sound (1): Decay depends on saturation
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// -------------------------------------------------------------------------
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// each FDN loop pass runs processMicroSaturation(), and reverberation
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// nonlinear distortion accumulates in the reverb and shifts the filter
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// response, producing metallic ringing.
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//
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// policy: not applied below a 2.0 s mid-band RT60 average, scaled between 2.0 s and 6.0 s,
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// and fully bypassed above 6.0 s.
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// -------------------------------------------------------------------------
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microSatBlend = juce::jlimit(0.0f, 1.0f, 1.0f - (rt60Mid - 2.0f) / 4.0f);
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// -------------------------------------------------------------------------
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// * metallic sound (2): Decay depends on modulation
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// -------------------------------------------------------------------------
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// longer reverb tails require deeper modulation at the filter peaks.
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// as used by Lexicon / Strymon.
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//
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// * modulation depth (scaled down for short reverbs)
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// RT60 <= 1.0 s -> 1.0x (min)
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// RT60 = 3.0s -> 2.0x
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// RT60 >= 5.0 s -> 3.0x (max)
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// -------------------------------------------------------------------------
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modDepthScale = 1.0f + juce::jlimit(0.0f, 2.0f, (rt60Mid - 1.0f) * 0.5f);
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constexpr float baseDB = 16.0f;
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float decayCompDB = 7.0f * std::log10(rt60Mid);
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static constexpr std::array<float, 7> algorithmOffsetDB = {
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+0.8f, +0.9f, +0.5f, +0.5f, +1.5f, +0.6f, +0.6f
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};
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float algoOffset = algorithmOffsetDB[juce::jlimit(0, 6, activeParams.algorithmIndex)];
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switch (currentTopology) {
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case ReverbTopology::Room:
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bypassER = false; bypassInputDiffusers = false;
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apfGain = 0.3f; diffusionSensitivity = 1.0f;
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break;
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case ReverbTopology::Hall:
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bypassER = false; bypassInputDiffusers = false;
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apfGain = 0.618f; diffusionSensitivity = 1.0f;
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break;
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case ReverbTopology::Plate:
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bypassER = true; bypassInputDiffusers = false;
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apfGain = 0.7f; diffusionSensitivity = 0.7f;
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break;
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case ReverbTopology::Spring:
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bypassER = true; bypassInputDiffusers = false;
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apfGain = 0.5f; diffusionSensitivity = 0.5f;
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break;
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case ReverbTopology::Goldfoil:
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bypassER = true; bypassInputDiffusers = false;
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apfGain = 0.75f; diffusionSensitivity = 0.8f;
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break;
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}
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const auto& erPattern = PRESET_ER_PATTERNS[
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juce::jlimit(0, 6, activeParams.algorithmIndex)];
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currentERTapCount = erPattern.numTaps;
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float erSizeScale = 0.5f + activeParams.roomSizeScale;
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for (int i = 0; i < erPattern.numTaps; ++i) {
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currentERDelaySamples[i] = erPattern.taps[i].delayMs * 0.001f
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* static_cast<float>(fs) * erSizeScale;
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currentERGains[i] = erPattern.taps[i].gain;
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}
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if (erPattern.numTaps == 0) bypassER = true;
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float edtCoeff = 0.7f;
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switch (currentTopology) {
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case ReverbTopology::Room: edtCoeff = 0.70f; break;
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case ReverbTopology::Hall: edtCoeff = 0.95f; break;
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case ReverbTopology::Plate: edtCoeff = 0.60f; break;
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case ReverbTopology::Spring: edtCoeff = 0.50f; break;
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case ReverbTopology::Goldfoil: edtCoeff = 0.85f; break;
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}
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theoreticalEDT = rt60Mid * edtCoeff;
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float satMultiplier = 1.0f;
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switch (currentTopology) {
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case ReverbTopology::Room: satMultiplier = 0.90f; break;
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case ReverbTopology::Hall: satMultiplier = 0.93f; break;
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case ReverbTopology::Plate: satMultiplier = 1.00f; break;
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case ReverbTopology::Spring: satMultiplier = 1.05f; break;
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case ReverbTopology::Goldfoil: satMultiplier = 1.02f; break;
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}
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float effectiveSatAmount = juce::jlimit(0.0f, 1.0f,
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activeParams.saturation * satMultiplier);
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saturatorL.setAmount(effectiveSatAmount);
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saturatorR.setAmount(effectiveSatAmount);
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saturatorL.setMode(activeParams.satTypeIdx);
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saturatorR.setMode(activeParams.satTypeIdx);
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lateMakeupGainLinear = juce::Decibels::decibelsToGain(baseDB + decayCompDB + algoOffset);
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}
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inline void UniversalEngine::fastWalshHadamardTransform(
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std::array<float, 16>& v) noexcept
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{
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for (int h = 1; h < 16; h *= 2) {
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for (int i = 0; i < 16; i += h * 2) {
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for (int j = i; j < i + h; ++j) {
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float x = v[j], y = v[j + h];
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v[j] = x + y;
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v[j + h] = x - y;
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}
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}
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}
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for (int i = 0; i < 16; ++i) v[i] *= 0.25f;
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}
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inline void UniversalEngine::applySignFlipping(
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std::array<float, 16>& v) noexcept
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{
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static constexpr std::array<float, 16> flip = {
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1.f, -1.f, 1.f, -1.f, -1.f, 1.f, -1.f, 1.f,
|
||||
1.f, 1.f, -1.f, -1.f, -1.f, -1.f, 1.f, 1.f
|
||||
};
|
||||
for (int i = 0; i < 16; ++i) v[i] *= flip[i];
|
||||
}
|
||||
|
||||
void UniversalEngine::processBlock(const float* inL, const float* inR,
|
||||
float* outL, float* outR,
|
||||
int numSamples) noexcept
|
||||
{
|
||||
// * CPU: fs float (processBlock throughout use )
|
||||
const float fsf = static_cast<float>(fs);
|
||||
|
||||
// * modulation : squared curve + coefficient suppress
|
||||
// modAmount^2 low band gradually , 0.001f entire
|
||||
// : modAmt=0.5 -> 48smp(1ms) / : modAmt=0.5 -> 12smp(0.25ms)
|
||||
const float modAmtCurved = activeParams.modAmount * activeParams.modAmount;
|
||||
const float depthSamples = modAmtCurved * 0.001f * fsf * modDepthScale;
|
||||
const float wetGain = juce::Decibels::decibelsToGain(activeParams.wetDB);
|
||||
const float stereoWidth = activeParams.stereoWidth;
|
||||
const float erLevel = activeParams.erLevel;
|
||||
const float lateLevel = activeParams.lateLevel;
|
||||
const bool erSolo = activeParams.erSolo;
|
||||
const float duckThreshLin = juce::Decibels::decibelsToGain(activeParams.duckingThreshDB);
|
||||
const float duckAmountDB = activeParams.duckingAmount;
|
||||
|
||||
const float effectiveDiffusion = activeParams.diffusion * diffusionSensitivity;
|
||||
const float diffuserGain = 0.25f + effectiveDiffusion * 0.55f;
|
||||
const float effectiveApfGain = apfGain * (0.60f + effectiveDiffusion * 0.40f);
|
||||
|
||||
const float sideBoost = stereoWidth * 1.5f;
|
||||
const float erLeakage = (1.0f - stereoWidth) * 0.7f;
|
||||
|
||||
// * CPU: apfGainStage loop -> before compute
|
||||
const float apfGainStage = effectiveApfGain * 0.78f;
|
||||
|
||||
// * CPU: freqModScale before compute (16ch)
|
||||
std::array<float, FDN_ORDER> freqModScales;
|
||||
constexpr float invFdnM1 = 1.0f / static_cast<float>(FDN_ORDER - 1);
|
||||
for (int i = 0; i < FDN_ORDER; ++i)
|
||||
freqModScales[i] = 0.5f + (1.0f - static_cast<float>(i) * invFdnM1) * 1.0f;
|
||||
|
||||
// * CPU: input diffuser time before compute
|
||||
std::array<float, 4> diffuserDelaySmp;
|
||||
for (int i = 0; i < 4; ++i)
|
||||
diffuserDelaySmp[i] = (3.0f + i * 2.0f) * 0.001f * fsf;
|
||||
|
||||
// * CPU: Allpass before compute (16ch x 3)
|
||||
constexpr float apfBaseMs[SERIAL_APF_STAGES] = { 1.5f, 2.3f, 3.7f };
|
||||
constexpr float apfSpreadMs[SERIAL_APF_STAGES] = { 0.30f, 0.37f, 0.47f };
|
||||
constexpr float apfModFrac[SERIAL_APF_STAGES] = { 0.15f, 0.10f, 0.07f };
|
||||
const float msToSmp = 0.001f * fsf;
|
||||
std::array<std::array<float, SERIAL_APF_STAGES>, FDN_ORDER> apfBaseDelaySmp;
|
||||
for (int i = 0; i < FDN_ORDER; ++i)
|
||||
for (int s = 0; s < SERIAL_APF_STAGES; ++s)
|
||||
apfBaseDelaySmp[i][s] = (apfBaseMs[s] + i * apfSpreadMs[s]) * msToSmp;
|
||||
|
||||
// * CPU: ER tapGain * 0.5f before compute
|
||||
std::array<float, MAX_ER_TAPS> erTapGainsHalf;
|
||||
for (int t = 0; t < currentERTapCount; ++t)
|
||||
erTapGainsHalf[t] = currentERGains[t] * 0.5f;
|
||||
|
||||
// * CPU: soft-knee threshold squared before compute (sqrt avoid )
|
||||
constexpr float compThresh = 0.35f;
|
||||
constexpr float compThreshSq = compThresh * compThresh;
|
||||
|
||||
std::array<float, FDN_ORDER> lfoCoeffs;
|
||||
{
|
||||
constexpr float twoPi = 6.28318530718f;
|
||||
for (int i = 0; i < FDN_ORDER; ++i) {
|
||||
const float fc = activeParams.modRate * lfos[i].rateMultiplier;
|
||||
lfoCoeffs[i] = juce::jlimit(0.0001f, 0.9999f,
|
||||
1.0f - std::exp(-twoPi * fc / fsf));
|
||||
// * LFO update
|
||||
chorusLFOs[i].phaseInc = activeParams.modRate * chorusLFOs[i].rateScale / fsf;
|
||||
}
|
||||
}
|
||||
|
||||
for (int n = 0; n < numSamples; ++n) {
|
||||
const float leftIn = inL[n];
|
||||
const float rightIn = inR[n];
|
||||
const float midIn = (leftIn + rightIn) * 0.5f;
|
||||
const float sideIn = (leftIn - rightIn) * 0.5f;
|
||||
float erOutL = 0.0f, erOutR = 0.0f;
|
||||
|
||||
// * PreDelay: dry time
|
||||
// ERFDN input .
|
||||
// dry attack after ,
|
||||
// clarity (D50/C50) significantly above .
|
||||
preDelayLine.write(midIn);
|
||||
const float delayedMid = (preDelaySamples > 0.5f)
|
||||
? preDelayLine.read(preDelaySamples)
|
||||
: midIn;
|
||||
|
||||
const float inputPeak = juce::jmax(std::abs(leftIn), std::abs(rightIn));
|
||||
const float envCoeff = (inputPeak > duckingEnvelope)
|
||||
? duckingAttackCoeff : duckingReleaseCoeff;
|
||||
duckingEnvelope += (inputPeak - duckingEnvelope) * envCoeff;
|
||||
|
||||
float duckGainLinear = 1.0f;
|
||||
if (duckAmountDB > 0.001f && duckingEnvelope > duckThreshLin) {
|
||||
const float envDB = 20.0f * std::log10(juce::jmax(duckingEnvelope, 1e-6f));
|
||||
const float overDB = envDB - activeParams.duckingThreshDB;
|
||||
const float gainRedDB = -juce::jmin(overDB, duckAmountDB);
|
||||
duckGainLinear = juce::Decibels::decibelsToGain(gainRedDB);
|
||||
}
|
||||
|
||||
float fdnInputMid = delayedMid;
|
||||
if (!bypassInputDiffusers) {
|
||||
for (int i = 0; i < 4; ++i) {
|
||||
float d = inputDiffusers[i].read(diffuserDelaySmp[i]);
|
||||
float w = fdnInputMid + diffuserGain * d;
|
||||
inputDiffusers[i].write(w);
|
||||
fdnInputMid = d - diffuserGain * w;
|
||||
}
|
||||
}
|
||||
|
||||
if (!bypassER) {
|
||||
erDelay.write(delayedMid);
|
||||
float erTotalL = 0.0f, erTotalR = 0.0f;
|
||||
for (int t = 0; t < currentERTapCount; ++t) {
|
||||
const float tapValue = erDelay.read(currentERDelaySamples[t]);
|
||||
const float tapGain = erTapGainsHalf[t];
|
||||
const float tg = tapValue * tapGain;
|
||||
const float tgLeak = tg * erLeakage;
|
||||
if (t % 2 == 0) {
|
||||
erTotalL += tg;
|
||||
erTotalR += tgLeak;
|
||||
}
|
||||
else {
|
||||
erTotalR += tg;
|
||||
erTotalL += tgLeak;
|
||||
}
|
||||
}
|
||||
erOutL = erTotalL;
|
||||
erOutR = erTotalR;
|
||||
}
|
||||
|
||||
// * ER -> Late: feed the ER output into the FDN input
|
||||
// the early reflections are wall-surface reflections that seed the Late Reverb,
|
||||
// making the ER-to-Late transition natural and smooth.
|
||||
if (!bypassER) {
|
||||
fdnInputMid += (erOutL + erOutR) * 0.5f * 0.15f;
|
||||
}
|
||||
|
||||
std::array<float, 16> currentFb = fbVec;
|
||||
fastWalshHadamardTransform(currentFb);
|
||||
applySignFlipping(currentFb);
|
||||
|
||||
float fdnOutL = 0.0f, fdnOutR = 0.0f;
|
||||
std::array<float, 16> nextFb;
|
||||
|
||||
for (int i = 0; i < FDN_ORDER; ++i) {
|
||||
const float lfoVal = lfos[i].tick(lfoCoeffs[i]);
|
||||
// * modulation: sine-wave LFO + noise LFO
|
||||
// noise = random (suppresses metallic ringing)
|
||||
// chorus = smoothly accumulated (rich tail)
|
||||
const float chorusVal = chorusLFOs[i].tick();
|
||||
const float combinedLfo = lfoVal + chorusVal * 0.6f;
|
||||
|
||||
// * frequency-dependent modulation: high bands modulate less than low bands
|
||||
const float freqModScale = freqModScales[i];
|
||||
const float delaySmp = fdnBaseDelaySamples[i]
|
||||
+ combinedLfo * depthSamples * freqModScale;
|
||||
float d = fdnDelays[i].read(delaySmp);
|
||||
|
||||
#if AMBIVALENCE_USE_STAGE2_ABSORPTION
|
||||
for (int s = 0; s < ABSO_STAGES_S2; ++s)
|
||||
d = absorptionFiltersS2[i][s].tick(d, currentAbsorptionCoeffsS2[i][s]);
|
||||
#else
|
||||
d = absorptionFilters[i].tick(d, currentAbsorptionCoeffs[i]);
|
||||
#endif
|
||||
|
||||
// * metallic sound (3): DC blocker (1st-order HPF, fc ~ 5 Hz)
|
||||
// saturation in the FDN loop absorption filters can
|
||||
// accumulate DC; blocking it prevents low-band asymmetric distortion.
|
||||
{
|
||||
const float dcIn = d;
|
||||
const float dcOut = dcIn - dcX1[i] + dcBlockerCoeff * dcY1[i];
|
||||
dcX1[i] = dcIn;
|
||||
dcY1[i] = dcOut;
|
||||
d = dcOut;
|
||||
}
|
||||
|
||||
// * soft-knee compression (in the FDN feedback loop)
|
||||
// an RMS envelope over the threshold triggers compression.
|
||||
// * CPU: sqrt only runs above threshold (compare on squared values)
|
||||
{
|
||||
fdnRmsEnv[i] += (d * d - fdnRmsEnv[i]) * rmsCoeff;
|
||||
if (fdnRmsEnv[i] > compThreshSq) {
|
||||
const float env = std::sqrt(fdnRmsEnv[i]);
|
||||
const float over = env - compThresh;
|
||||
d *= compThresh / (compThresh + over * 0.65f);
|
||||
}
|
||||
}
|
||||
|
||||
// * metallic sound (1): Decay depends on saturation
|
||||
// microSatBlend=1.0 -> applied (into the reverb loop)
|
||||
// microSatBlend=0.0 -> fully bypassed
|
||||
if (microSatBlend > 0.001f) {
|
||||
const float sat = processMicroSaturation(d);
|
||||
d = d + (sat - d) * microSatBlend;
|
||||
}
|
||||
|
||||
// * 3 nested allpass filters (echo density)
|
||||
// * CPU: apfGainStage precomputed per block
|
||||
float apfOut = d;
|
||||
{
|
||||
for (int s = 0; s < SERIAL_APF_STAGES; ++s) {
|
||||
const float apfModDepth = depthSamples * apfModFrac[s];
|
||||
const float apfDelaySmp = apfBaseDelaySmp[i][s]
|
||||
+ combinedLfo * apfModDepth * freqModScale;
|
||||
float apfD = nestedAllpassDelays[i][s].read(apfDelaySmp);
|
||||
float apfW = apfOut + apfGainStage * apfD;
|
||||
nestedAllpassDelays[i][s].write(apfW);
|
||||
apfOut = apfD - apfGainStage * apfW;
|
||||
}
|
||||
}
|
||||
|
||||
nextFb[i] = apfOut;
|
||||
|
||||
const float sideForCh = (i % 2 == 0 ? +sideIn : -sideIn) * sideBoost;
|
||||
const float fdnInputForThisCh = (fdnInputMid + sideForCh) * 0.25f;
|
||||
fdnDelays[i].write(fdnInputForThisCh + currentFb[i]);
|
||||
|
||||
const float crossLeak = 1.0f - stereoWidth;
|
||||
if (i % 2 == 0) {
|
||||
fdnOutL += apfOut;
|
||||
fdnOutR += apfOut * crossLeak;
|
||||
}
|
||||
else {
|
||||
fdnOutR += apfOut;
|
||||
fdnOutL += apfOut * crossLeak;
|
||||
}
|
||||
}
|
||||
|
||||
fdnOutL *= 0.125f;
|
||||
fdnOutR *= 0.125f;
|
||||
fbVec = nextFb;
|
||||
|
||||
const float erMixL = bypassER ? 0.0f : erOutL * erLevel;
|
||||
const float erMixR = bypassER ? 0.0f : erOutR * erLevel;
|
||||
const float lateMixL = fdnOutL * lateMakeupGainLinear * lateLevel;
|
||||
const float lateMixR = fdnOutR * lateMakeupGainLinear * lateLevel;
|
||||
|
||||
acousticMetrics.processSample((lateMixL + lateMixR) * 0.5f);
|
||||
|
||||
float satL = saturatorL.processSample(lateMixL);
|
||||
float satR = saturatorR.processSample(lateMixR);
|
||||
|
||||
if (erSolo) { satL = 0.0f; satR = 0.0f; }
|
||||
|
||||
float wetL = erMixL + satL;
|
||||
float wetR = erMixR + satR;
|
||||
outputEQ.process(wetL, wetR);
|
||||
|
||||
const float finalWetGain = wetGain * duckGainLinear;
|
||||
outL[n] = wetL * finalWetGain;
|
||||
outR[n] = wetR * finalWetGain;
|
||||
|
||||
outputLimiter.process(outL[n], outR[n]);
|
||||
}
|
||||
}
|
||||
|
||||
} // namespace FDNReverb
|
||||
Loading…
Add table
Add a link
Reference in a new issue