cpp-dsp

Implement C++ digital signal processing algorithms for real-time audio applications.

1|Updated Feb 24, 2026
One-click install
npx skills add https://github.com/sebasdv/my-skills --skill cpp-dsp
Or copy as Structured Prompt for Agent
Please help me install this Agent Skill.
Skill: cpp-dsp
Source: https://github.com/sebasdv/my-skills/tree/main/cpp-dsp
Command: npx skills add https://github.com/sebasdv/my-skills --skill cpp-dsp

SYSTEM DOCUMENTATION & REQUIREMENTS

💡 This Skill includes references (resource) components.

What problem does it solve?

This Skill provides essential C++ building blocks for digital signal processing in audio applications, enabling developers to implement complex audio effects and synthesizers efficiently and safely within real-time constraints.

Core Features & Use Cases

  • Real-time DSP Primitives: Offers optimized C++ code for filters, oscillators, envelopes, and more, suitable for audio threads.
  • Performance & Safety: Focuses on avoiding heap allocations, branches in hot paths, and blocking operations for stable, high-performance audio processing.
  • Use Case: Implement a custom synthesizer voice by combining the WavetableOsc and Adsr classes, then process its output through a BiquadFilter and softClip for a unique sound.

Quick Start

Use the cpp-dsp skill to implement a biquad filter with a cutoff frequency of 1000 Hz, a Q of 0.707, and a sample rate of 44100 Hz.

Frequently Asked Questions about cpp-dsp

High-intent search queries and answers about installing and using this skill.

FAQPage Schema
How do I implement real-time DSP algorithms in C++ without causing audio thread dropouts?

Real-time DSP in C++ requires avoiding heap allocations, hot-path branches, and blocking operations on the audio thread. This approach provides optimized primitives like filters and oscillators designed specifically for stable, high-performance audio processing.

How do I build a custom synthesizer voice using C++ DSP components?

Build a synthesizer voice by combining wavetable oscillators with ADSR envelopes to shape the sound. You can then process the resulting audio output through a biquad filter and apply soft clipping to create a unique effect.

What is the best way to implement a biquad filter for real-time audio in C++?

The best way to implement a biquad filter is using optimized C++ classes designed for real-time audio constraints. You can instantiate it with specific parameters like a 1000 Hz cutoff, 0.707 Q, and 44100 Hz sample rate.

Can I use fixed-point arithmetic instead of floating-point for DSP algorithms in C++?

Yes, fixed-point considerations are supported when designing DSP algorithms in C++. This approach accommodates specific hardware or performance requirements while maintaining the necessary real-time audio processing constraints.

What C++ DSP primitives are needed for audio effects and parameter smoothing?

Essential C++ DSP primitives for audio effects include biquad filters, wavetable oscillators, ADSR envelopes, delay lines, soft clipping, and parameter smoothing. These building blocks enable complex audio effect and synthesizer development.

Why should I avoid heap allocation when writing C++ audio effects for real-time applications?

Avoiding heap allocation in real-time C++ audio effects prevents unpredictable memory management delays that cause audio thread dropouts. This practice ensures stable, high-performance digital signal processing without blocking operations.