[Free Dsp] Analog Filter (24dB/oct)

[d.healey]

@griffinboy said in Free Effect: Analog Filter for scriptnode (24dB/oct):

Now restart hise, and you will be able to use the Node:

I think the process has changed with the latest version of HISE?

[griffinboy]

@d-healey

Did it! I'm not on the latest version of hise, what is different? I should update

[d.healey]

@griffinboy I'm basing my assumption on this thread - https://forum.hise.audio/topic/11731/scriptnode-compilation-workflow-and-what-to-do-about-it

[Christoph Hart]

Now restart hise, and you will be able to use the Node:

Yes. it should read now:

Now press OK and you will be able to use the node

It's the small details :)

[griffinboy]

@Christoph-Hart

Nice update!

[DabDab]

@griffinboy Waiting for your Youtube Tutorials.

[ustk]

@griffinboy Nice one!

[DabDab]

I have tried to make it a High Pass Filter from the Griffin_LadderFilter.h provided by @griffinboy
I am not a good C++ DSP programmer. Test it whether it is working or not.

/*
Copyright (c) 2024 griffinboy

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 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.
*/

#pragma once
#include <JuceHeader.h>

namespace project
{
    using namespace juce;
    using namespace hise;
    using namespace scriptnode;

    template <typename T>
    class ScopedValue
    {
    public:
        explicit ScopedValue(T& val) : value(val), ref(val) {}
        ~ScopedValue() { ref = value; }
        T& get() { return value; }

    private:
        T value;
        T& ref;
        ScopedValue(const ScopedValue&) = delete;
        ScopedValue& operator=(const ScopedValue&) = delete;
    };

    template <int NV>
    struct Griffin_HighPassFilter : public data::base
    {
        SNEX_NODE(Griffin_HighPassFilter);

        struct MetadataClass
        {
            SN_NODE_ID("Griffin_HighPassFilter");
        };

        static constexpr bool isModNode() { return false; }
        static constexpr bool isPolyphonic() { return NV > 1; }
        static constexpr bool hasTail() { return true; }
        static constexpr bool isSuspendedOnSilence() { return false; }
        static constexpr int getFixChannelAmount() { return 2; }

        static constexpr int NumTables = 0;
        static constexpr int NumSliderPacks = 0;
        static constexpr int NumAudioFiles = 0;
        static constexpr int NumFilters = 0;
        static constexpr int NumDisplayBuffers = 0;

        void prepare(PrepareSpecs specs)
        {
            filtersLeft.prepare(specs);
            filtersRight.prepare(specs);

            for (auto& filter : filtersLeft)
                filter.prepare(specs.sampleRate);

            for (auto& filter : filtersRight)
                filter.prepare(specs.sampleRate);
        }

        void reset()
        {
            for (auto& filter : filtersLeft)
                filter.reset();

            for (auto& filter : filtersRight)
                filter.reset();
        }

        template <typename ProcessDataType>
        void process(ProcessDataType& data)
        {
            auto& fixData = data.template as<ProcessData<getFixChannelAmount()>>();
            auto audioBlock = fixData.toAudioBlock();

            auto* leftChannelData = audioBlock.getChannelPointer(0);
            auto* rightChannelData = audioBlock.getChannelPointer(1);
            int numSamples = (int)data.getNumSamples();

            for (auto& leftFilter : filtersLeft)
            {
                leftFilter.process(leftChannelData, numSamples);
            }

            for (auto& rightFilter : filtersRight)
            {
                rightFilter.process(rightChannelData, numSamples);
            }
        }

        class AudioEffect
        {
        public:
            AudioEffect() = default;

            void prepare(float sampleRate)
            {
                fs = sampleRate;
                reset();
            }

            void reset()
            {
                x1 = x2 = y1 = y2 = 0.0f;
            }

            void updateCoefficients(float fc, float q)
            {
                float omega = 2.0f * MathConstants<float>::pi * fc / fs;
                float alpha = std::sin(omega) / (2.0f * q);
                float cosOmega = std::cos(omega);

                float a0 = 1.0f + alpha;
                b0 = (1.0f + cosOmega) / (2.0f * a0);
                b1 = -(1.0f + cosOmega) / a0;
                b2 = (1.0f + cosOmega) / (2.0f * a0);
                a1 = (-2.0f * cosOmega) / a0;
                a2 = (1.0f - alpha) / a0;
            }

            void process(float* samples, int numSamples)
            {
                for (int i = 0; i < numSamples; ++i)
                {
                    samples[i] = processSample(samples[i]);
                }
            }

        private:
            float fs = 44100.0f;
            float b0 = 0.0f, b1 = 0.0f, b2 = 0.0f;
            float a1 = 0.0f, a2 = 0.0f;
            float x1 = 0.0f, x2 = 0.0f;
            float y1 = 0.0f, y2 = 0.0f;

            inline float processSample(float input)
            {
                float output = b0 * input + b1 * x1 + b2 * x2 - a1 * y1 - a2 * y2;
                
                x2 = x1;
                x1 = input;
                y2 = y1;
                y1 = output;
                
                return output;
            }
        };

        template <int P>
        void setParameter(double value)
        {
            if (P == 0) // Cutoff Frequency
            {
                cutoffFrequency = static_cast<float>(value);
                for (auto& filter : filtersLeft)
                    filter.updateCoefficients(cutoffFrequency, q);

                for (auto& filter : filtersRight)
                    filter.updateCoefficients(cutoffFrequency, q);
            }
            else if (P == 1) // Q Factor
            {
                q = static_cast<float>(value);
                for (auto& filter : filtersLeft)
                    filter.updateCoefficients(cutoffFrequency, q);

                for (auto& filter : filtersRight)
                    filter.updateCoefficients(cutoffFrequency, q);
            }
        }

        void createParameters(ParameterDataList& data)
        {
            {
                parameter::data p("Cutoff Frequency", { 20.0, 20000.0, 1.0 });
                registerCallback<0>(p);
                p.setDefaultValue(1000.0);
                data.add(std::move(p));
            }
            {
                parameter::data p("Q Factor", { 0.1, 10.0, 0.01 });
                registerCallback<1>(p);
                p.setDefaultValue(0.707);
                data.add(std::move(p));
            }
        }

        void setExternalData(const ExternalData& data, int index) {}
        void handleHiseEvent(HiseEvent& e) {}

        template <typename FrameDataType>
        void processFrame(FrameDataType& data) {}

    private:
        PolyData<AudioEffect, NV> filtersLeft;
        PolyData<AudioEffect, NV> filtersRight;
        float cutoffFrequency = 1000.0f;
        float q = 0.707f;
    };
}

[HISEnberg]

@griffinboy Just curious, have you figured out how to incorporate the filter display in external C++ nodes?

[griffinboy]

@HISEnberg

You know what, I've never tried it but it should be possible.

[HISEnberg]

@griffinboy Ya I haven't tried either, I was just be lazy and reaching out for a solution first before trying it myself. I'll look through some of the stock HISE filters and post here once I find the solution.

[griffinboy]

@HISEnberg

Make a new post if you find anything

[Orvillain]

@griffinboy @HISEnberg I'd love to know this too. I would imagine it is similar to the thing I wanted to do where I wanted to send out modified audio buffers to a display. I gave up in the end coz I couldn't figure it out.

[HISEnberg]

@Orvillain Ah yes I meant to revisit that. I did figure it out but the solution was admittedly more AI-influenced than I care to admit, so I was waiting until I have a more proper understanding of it to share.