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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. | ||
| */ | ||
|
|
||
| //----------------------------------------------------------- | ||
| // 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. | ||
| //----------------------------------------------------------- | ||
|
|
||
| #include "ladder.h" | ||
| #include "Utility/dsp.h" | ||
|
|
||
| using namespace daisysp; | ||
|
|
||
| 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); | ||
| } | ||
|
|
||
| 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; | ||
|
|
||
| SetPassbandGain(0.5f); | ||
| SetInputDrive(0.5f); | ||
| SetFreq(5000.f); | ||
| SetRes(0.2f); | ||
| } | ||
|
|
||
| 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; | ||
| } | ||
|
|
||
| __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]); | ||
| } | ||
| } | ||
|
|
||
| void LadderFilter::SetFreq(float freq) | ||
| { | ||
| Fbase_ = freq; | ||
| compute_coeffs(freq); | ||
| } | ||
|
|
||
| void LadderFilter::SetRes(float res) | ||
| { | ||
| // maps resonance = 0->1 to K = 0 -> 4 | ||
| res = daisysp::fclamp(res, 0.0f, kMaxResonance); | ||
| K_ = 4.0f * res; | ||
| } | ||
|
|
||
| void LadderFilter::SetPassbandGain(float pbg) | ||
| { | ||
| pbg_ = daisysp::fclamp(pbg, 0.0f, 0.5f); | ||
| SetInputDrive(drive_); | ||
| } | ||
|
|
||
| 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_; | ||
| } | ||
| } | ||
|
|
||
| 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; | ||
| } | ||
|
|
||
| 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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