Create your audio products with our DSP modules

Minimize development costs

Employ immediately available code to reduce development time and cost, make more accurate estimates, and use less specialist skills.

Dependable technology

Our DSP modules are written in C, with optional C++ wrappers, do not rely on external dependencies (not even the standard math library), are well tested, thouroughly optimised, and easy to integrate. Their minimal and abstraction-free architecture is geared towards maximum reuse in hardware and software products.

Reduce time-to-market

Our solution allows you to meet aggressive deadlines and hence increase your profits. Furthermore, the flexibility of our modules and the peculiar licensing conditions enable you to develop more projects and enrich your catalog.

Try a simple example application

(Some of the audio samples were provided by Hexagonlab Recording Studios. Thank you Edoardo! ❤️)

This autofilter effect was built by simply putting together the following modules in our collection.

Ladder filter

Full nonlinear virtual analog model of the 4-pole (24 dB/oct) lowpass ladder filter. The model has optional passband gain compensation and output polarity adjustment, and can generate/sustain self-oscillation.

The original filter circuit sculpts the sound of several classic synthesizers from the '70s and a few effect units as well.

Wah

Linear virtual analog model of a classic wah pedal, faithfully recreating its vocal-like characteristic sound.

The original device can be heard on countless recordings starting from the '60s.

Waveshaper

Simple static waveshaper distortion algorithm with mild antialiasing built-in (does not require oversampling).

It can be used to implement saturation and distortion effects in pretty much any project.

Oscillator

Basic oscillator algorithm implementing several waveforms (sine, saw, square - with variable pulse width, triangle - with variable slope, white noise, noise + S&H, random ramps) without antialiasing.

Perfect for LFOs in synthesizers and more generally to implement modulation sources.

One pole filter

One pole (6 dB/oct) lowpass filter algorithm with separate attack and decay time constants and a sticky target-reach threshold option.

This module is essential to implement envelope followers and exponential parameter smoothing.

Peak programme meter

This algorithm implements a fairly standard digital audio peak level meter.

A fundamental component of digital mixers and audio software in general.

Developing music DSP software and hardware projects often poses difficult technical challenges since many different and highly specialized skills are usually involved. There can be multiple paths to reach the same goal but the end result is almost never guaranteed to be satisfactory or convenient despite the time and money invested.

Our library of DSP modules was first conceived for us to avoid continuously reinventing the wheel. It is therefore the result of years of research, development, refinements, usage, and adaptation to different commercial projects. We also strive to make it as usable as possible in different contexts by employing a minimalistic architectural approach.

We offer licenses for each individual module and provide you with full source code access. Each license is valid for an unlimited number of projects and developers within your organisation. Furthermore, we also offer custom development services and, if you prefer, we can take full charge of the development of the audio DSP part of your projects.

If you are interested and/or for further information, e.g. on licensing terms, please just drop us an email at info@orastron.com.