claude

audio-dsp-engineering

Audio DSP engineering team — agents (audio-dsp-architect, dsp-implementation-engineer) for the layer answering 'what processing architecture, at what latency, on what platform — and how do we make the audio callback real-time-safe and fast?': processing model (block vs sample-by-sample), latency & buffer budget, fixed vs float + denormals, algorithm approach (IIR biquads / FIR, FFT/STFT overlap-add, oversampling/anti-aliasing), plugin formats & frameworks (JUCE, VST3, AU/AUv3, AAX, CLAP, LV2), audio I/O (CoreAudio, ASIO, WASAPI, ALSA/JACK, Web Audio), lock-free parameter passing, spatial/binaural (HRTF), embedded DSP (CMSIS-DSP), SIMD, and measurement (null tests, THD+N, frequency response). skills, a knowledge bank (decision tree + 2026 patterns), and templates. Distinct from streaming-media-engineering (codecs/delivery), conversational-ai-voice-engineering (voice AI agents), and embedded-iot-engineering (general firmware) — this is the real-time signal-processing layer. Needs ravenclaude-core.

9 on GitHubby Matt Corbettv0.1.2MITupdated today
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Plugin audio-dsp-engineering from marketplace ravenclaude in mcorbett51090/RavenClaude. Commands read from the repo's manifest — today.

What it installs

Skills (3)

  • choose-audio-dsp-architecture

    Pick the right audio-DSP architecture for a described product by traversing the audio-DSP architecture decision tree (latency tolerance → processing model → time-vs-frequency domain → fixed-vs-float + denormals → platform/plugin format + audio I/O), then return the recommended processing model, late

  • design-signal-processing-chain

    From an effect or product goal and its architecture, derive the signal-processing chain — the block diagram (stage order), the per-stage algorithm (IIR biquad / FIR / FFT-STFT / delay / dynamics), the per-stage and total latency, the sample rate and gain-staging/headroom, the oversampling plan for a

  • implement-and-optimize-realtime-audio

    Implement a DSP stage as real-time-safe code in the audio callback (no locks, no allocation, no syscalls, no unbounded work — everything pre-allocated at prepare time), handle denormals with flush-to-zero, pass parameters lock-free (atomic / SPSC FIFO) with per-sample smoothing, then optimize the pr

Subagents (2)

  • audio-dsp-architect

    Use to choose the audio-DSP ARCHITECTURE — block/sample processing, latency/buffer budget, plugin format & audio I/O (JUCE/VST3/AU/CLAP; CoreAudio/ASIO/Web Audio), fixed vs float, algorithm approach (IIR/FIR, FFT/STFT, oversampling). NOT for streaming codecs/delivery → streaming-media-engineering.

  • dsp-implementation-engineer

    Use to BUILD & OPTIMIZE real-time audio — real-time-safe DSP in the callback (no locks/alloc/syscalls), biquads/FIR/FFT, denormals/flush-to-zero, SIMD (SSE/NEON), lock-free params, measurement (null/THD+N/impulse response). NOT for conversational voice AI → conversational-ai-voice-engineering.

View source on GitHub mcorbett51090.github.io

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