DSP & Analog Circuit Modeling (WDF, Biquads, Polyphase OS)

    1

    Design real-time audio algorithms, analog circuit emulations, and SIMD-optimized processing pipelines.

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    Works with the AI tools you already use

    Claude CodeClaude CodeCursorCursorCodex CLICodex CLIMuseMuseDotsDots+22 more

    DSP & Analog Circuit Modeling (WDF, Biquads, Polyphase OS)

    Example session with this skill installed

    Generate a C++ Biquad low-pass filter using a Direct Form II topology. Optimize the processing loop for AVX2 and ensure it is real-time safe with denormal prevention.

    • Read your context and instructions
    • Compiled the dsp & analog
    // Biquad DFII with AVX2 Optimization
    void process(float* data, int count) {
        const __m256 v_dc_offset = _mm256_set1_ps(1e-25f);
        // ... SIMD coefficients and state management ...
        for (int i = 0; i < count; i += 8) {
            __m256 x = _mm256_loadu_ps(&data[i]);
            x = _mm256_add_ps(x, v_dc_offset); // Denormal prevention
            // ... vectorized DFII logic ...
        }
    }
    

    Connects securely to your tools. The creator never sees your data.

    What you get

    Model non-linear analog circuits using WDF structures.Implement polyphase FIR engines for aliasing-free distortion.Accelerate audio loops with AVX2 and ARM NEON intrinsics.Apply real-time safety patterns to prevent denormal-related CPU spikes.

    About this skill

    The problem

    Designing audio DSP algorithms requires balancing mathematical precision with strict real-time performance. Manual implementation often leads to denormal issues, CPU spikes from dynamic allocation, and inefficient processing loops that fail to use SIMD hardware.

    What it does

    • Generates Wave Digital Filter (WDF) structures for non-linear analog circuit emulation like diodes and tubes.
    • Designs discrete-time filter topologies including Biquads and State Variable Filters (SVF).
    • Implements polyphase FIR oversampling chains to eliminate aliasing in saturators.
    • Produces SIMD-optimized C++ code targeting AVX2, AVX-512, and ARM NEON.

    Frameworks & tools

    C++ (C++17 or later), JUCE, SIMD (AVX/NEON), and standard DSP libraries.

    Why this beats prompting it yourself

    General LLMs often produce code with hidden "real-time safety" violations like memory allocation in the audio thread. This skill enforces strict audio processing constraints, including denormal prevention and branchless non-linearities, which are critical for stable plugin performance.

    Use cases

    • Emulate the transfer function and impedance of a vintage tube preamp.
    • Build a high-performance 8x oversampling engine for a distortion effect.
    • Vectorize an existing gain or filter loop using AVX2 intrinsics.
    • Implement stable, high-order filters without the risk of numerical explosion.

    Known limitations

    Does not handle GUI/UX code or VST3/AU wrapper boilerplate. Focuses strictly on the signal processing core and mathematical modeling.

    How to install

    Works the same in every agent - Claude, Cursor, Codex, Copilot and 20+ more.

    ~30 seconds
    1. 1

      Download the ZIP

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    2. 2

      Unzip into your skills folder

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    3. 3

      Ask your agent to use it

      Restart the agent if it was already running. It picks the skill up automatically - no config needed.

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    Recently published to Agensi

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    Security scanned

    Verified clean 1 month ago

    • Passed all security checks, Safe to install

    Listed1 month ago

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