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Merge pull request #200 from infrasonicaudio/alt-ladder-filter
Add alternate MIT-licensed ladder filter module
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/* Ported from Audio Library for Teensy, Ladder Filter | ||
* Copyright (c) 2021, Richard van Hoesel | ||
* Copyright (c) 2024, Infrasonic Audio LLC | ||
* | ||
* Permission is hereby granted, free of charge, to any person obtaining a copy | ||
* of this software and associated documentation files (the "Software"), to deal | ||
* in the Software without restriction, including without limitation the rights | ||
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell | ||
* copies of the Software, and to permit persons to whom the Software is | ||
* furnished to do so, subject to the following conditions: | ||
* | ||
* The above copyright notice, development funding notice, and this permission | ||
* notice shall be included in all copies or substantial portions of the Software. | ||
* | ||
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR | ||
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, | ||
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE | ||
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER | ||
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, | ||
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN | ||
* THE SOFTWARE. | ||
*/ | ||
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//----------------------------------------------------------- | ||
// Huovilainen New Moog (HNM) model as per CMJ jun 2006 | ||
// Richard van Hoesel, v. 1.03, Feb. 14 2021 | ||
// v1.7 (Infrasonic/Daisy) add configurable filter mode | ||
// v1.6 (Infrasonic/Daisy) removes polyphase FIR, uses 2x linear | ||
// oversampling for performance reasons | ||
// v1.5 adds polyphase FIR or Linear interpolation | ||
// v1.4 FC extended to 18.7kHz, max res to 1.8, 4x oversampling, | ||
// and a minor Q-tuning adjustment | ||
// v.1.03 adds oversampling, extended resonance, | ||
// and exposes parameters input_drive and passband_gain | ||
// v.1.02 now includes both cutoff and resonance "CV" modulation inputs | ||
// please retain this header if you use this code. | ||
//----------------------------------------------------------- | ||
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#include "ladder.h" | ||
#include "Utility/dsp.h" | ||
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using namespace daisysp; | ||
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static inline float fast_tanh(float x) | ||
{ | ||
if(x > 3.0f) | ||
return 1.0f; | ||
if(x < -3.0f) | ||
return -1.0f; | ||
float x2 = x * x; | ||
return x * (27.0f + x2) / (27.0f + 9.0f * x2); | ||
} | ||
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void LadderFilter::Init(float sample_rate) | ||
{ | ||
sample_rate_ = sample_rate; | ||
sr_int_recip_ = 1.0f / (sample_rate * kInterpolation); | ||
alpha_ = 1.0f; | ||
K_ = 1.0f; | ||
Fbase_ = 1000.0f; | ||
Qadjust_ = 1.0f; | ||
oldinput_ = 0.f; | ||
mode_ = FilterMode::LP24; | ||
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SetPassbandGain(0.5f); | ||
SetInputDrive(0.5f); | ||
SetFreq(5000.f); | ||
SetRes(0.2f); | ||
} | ||
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float LadderFilter::Process(float in) | ||
{ | ||
float input = in * drive_; | ||
float total = 0.0f; | ||
float interp = 0.0f; | ||
for(size_t os = 0; os < kInterpolation; os++) | ||
{ | ||
float u = (interp * oldinput_ + (1.0f - interp) * input) | ||
- (z1_[3] - pbg_ * input) * K_ * Qadjust_; | ||
u = fast_tanh(u); | ||
float stage1 = LPF(u, 0); | ||
float stage2 = LPF(stage1, 1); | ||
float stage3 = LPF(stage2, 2); | ||
float stage4 = LPF(stage3, 3); | ||
total += weightedSumForCurrentMode( | ||
{input, stage1, stage2, stage3, stage4}) | ||
* kInterpolationRecip; | ||
interp += kInterpolationRecip; | ||
} | ||
oldinput_ = input; | ||
return total; | ||
} | ||
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__attribute__((optimize("unroll-loops"))) void | ||
LadderFilter::ProcessBlock(float* buf, size_t size) | ||
{ | ||
for(size_t i = 0; i < size; i++) | ||
{ | ||
buf[i] = Process(buf[i]); | ||
} | ||
} | ||
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void LadderFilter::SetFreq(float freq) | ||
{ | ||
Fbase_ = freq; | ||
compute_coeffs(freq); | ||
} | ||
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void LadderFilter::SetRes(float res) | ||
{ | ||
// maps resonance = 0->1 to K = 0 -> 4 | ||
res = daisysp::fclamp(res, 0.0f, kMaxResonance); | ||
K_ = 4.0f * res; | ||
} | ||
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void LadderFilter::SetPassbandGain(float pbg) | ||
{ | ||
pbg_ = daisysp::fclamp(pbg, 0.0f, 0.5f); | ||
SetInputDrive(drive_); | ||
} | ||
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void LadderFilter::SetInputDrive(float odrv) | ||
{ | ||
drive_ = daisysp::fmax(odrv, 0.0f); | ||
if(drive_ > 1.0f) | ||
{ | ||
drive_ = fmin(drive_, 4.0f); | ||
// max is 4 when pbg = 0, and 2.5 when pbg is 0.5 | ||
drive_scaled_ = 1.0f + (drive_ - 1.0f) * (1.0f - pbg_); | ||
} | ||
else | ||
{ | ||
drive_scaled_ = drive_; | ||
} | ||
} | ||
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float LadderFilter::LPF(float s, int i) | ||
{ | ||
// (1.0 / 1.3) (0.3 / 1.3) | ||
float ft = s * 0.76923077f + 0.23076923f * z0_[i] - z1_[i]; | ||
ft = ft * alpha_ + z1_[i]; | ||
z1_[i] = ft; | ||
z0_[i] = s; | ||
return ft; | ||
} | ||
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void LadderFilter::compute_coeffs(float freq) | ||
{ | ||
freq = daisysp::fclamp(freq, 5.0f, sample_rate_ * 0.425f); | ||
float wc = freq * 2.0f * PI_F * sr_int_recip_; | ||
float wc2 = wc * wc; | ||
alpha_ = 0.9892f * wc - 0.4324f * wc2 + 0.1381f * wc * wc2 | ||
- 0.0202f * wc2 * wc2; | ||
//Qadjust = 1.0029f + 0.0526f * wc - 0.0926 * wc2 + 0.0218* wc * wc2; | ||
Qadjust_ = 1.006f + 0.0536f * wc - 0.095f * wc2 - 0.05f * wc2 * wc2; | ||
// revised hfQ (rvh - feb 14 2021) | ||
} | ||
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float LadderFilter::weightedSumForCurrentMode( | ||
const std::array<float, 5>& stage_outs) | ||
{ | ||
// Weighted filter stage mixing to achieve selected response | ||
// as described in "Oscillator and Filter Algorithms for Virtual Analog Synthesis" | ||
// Välimäki and Huovilainen, Computer Music Journal, vol 60, 2006 | ||
switch(mode_) | ||
{ | ||
case FilterMode::LP24: return stage_outs[4]; | ||
case FilterMode::LP12: return stage_outs[2]; | ||
case FilterMode::BP24: | ||
return (stage_outs[2] + stage_outs[4]) * 4.0f | ||
- stage_outs[3] * 8.0f; | ||
case FilterMode::BP12: return (stage_outs[1] - stage_outs[2]) * 2.0f; | ||
case FilterMode::HP24: | ||
return stage_outs[0] + stage_outs[4] | ||
- ((stage_outs[1] + stage_outs[3]) * 4.0f) | ||
+ stage_outs[2] * 6.0f; | ||
case FilterMode::HP12: | ||
return stage_outs[0] + stage_outs[2] - stage_outs[1] * 2.0f; | ||
default: return 0.0f; | ||
} | ||
} |
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