sample-rate

Convert audio sample rates with anti-aliasing and interpolation methods.

1|Updated Nov 24, 2025
One-click install
npx skills add https://github.com/SpiralCloudOmega/DevTeam6 --skill sample-rate
Or copy as Structured Prompt for Agent
Please help me install this Agent Skill.
Skill: sample-rate
Source: https://github.com/SpiralCloudOmega/DevTeam6/tree/main/.github/skills/audio-dsp/sample-rate
Command: npx skills add https://github.com/SpiralCloudOmega/DevTeam6 --skill sample-rate

SYSTEM DOCUMENTATION & REQUIREMENTS

What problem does it solve?

The skill clarifies Nyquist requirements and provides methods for accurate sample-rate conversion, reducing aliasing and artifacts in digital audio workflows.

Core Features & Use Cases

  • Explains Nyquist-Shannon theorem, anti-aliasing filters, and interpolation methods for reliable sample-rate conversion.
  • Outlines preferred techniques (linear, cubic, Sinc, and polyphase) and considerations for oversampling in nonlinear processing.
  • Use Case: A producer needs to convert project material from 96 kHz to 44.1 kHz with minimal artifacts.

Quick Start

Provide a target sample rate and an interpolation method, then run the conversion on your audio data.

Frequently Asked Questions about sample-rate

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

FAQPage Schema
How do I convert audio sample rate from 96 kHz to 44.1 kHz without aliasing?

An anti-aliasing filter removes frequencies above the Nyquist limit before downsampling. This skill provides accurate sample-rate conversion using interpolation methods like linear, cubic, or sinc to prevent aliasing and preserve audio quality.

What is the best interpolation method for digital audio sample-rate conversion?

The best interpolation method for sample-rate conversion depends on your fidelity needs. This skill outlines techniques including zero-order hold, linear, cubic, and sinc interpolation, with polyphase methods preferred for high-quality mastering workflows.

Why do I need oversampling during nonlinear audio processing?

Oversampling increases the Nyquist frequency to prevent aliasing generated by nonlinear processing. This skill provides oversampling guidance to reduce artifacts, ensuring accurate digital audio processing across DAWs and plugins.

How does the Nyquist-Shannon theorem apply to anti-aliasing filters in audio DSP?

The Nyquist-Shannon theorem dictates that anti-aliasing filters must remove frequencies above half the target sample rate. This skill clarifies these requirements to ensure reliable sample-rate conversion and prevent digital audio artifacts.

Can I use this sample-rate conversion approach for real-time audio processing in my DAW?

Yes, this sample-rate conversion approach applies to real-time audio processing workflows across DAWs and plugins. It supports accurate format conversion and anti-aliasing filtering required for live audio manipulation.