SlowShade is a powerful standalone audio tool for creative sound design and detailed analysis. It allows you to: Adjust Playback Speed: Slow down or speed up audio with pitch following speed. Drive: Add rich, analogue warmth and […]
VelorX is a high-fidelity Algorithmic Reverb Processor designed to provide total spatial control through advanced visual feedback. Built around a precision dual-layer FFT engine and a real-time stereo correlation monitor, VelorX eliminates the guesswork of […]
Grav-LF is a precision Bass Enhancement Processor built around a specialized dual-layer FFT analysis engine that monitors the critical 20 Hz to 1000 Hz frequency range in real time. Instead of adjusting the low-end blindly, […]
Krunx-Q is a high-precision bitcrusher and sample rate reducer built around a dual-layer FFT analysis engine that monitors both input and output signals in real time. Instead of treating digital degradation as a simple destructive […]
Clipper 7 is a precision digital clipper built around a dual-layer FFT analysis engine that monitors both input and output signals in real time. Instead of relying solely on waveform-based shaping, Clipper 7 allows users […]
SlowWave is a creative slow-playback and reverb transformation tool that lets you stretch audio between 0.5x and 1.5x while shaping ambience with size and mix controls. It includes LR peak meters, stereo correlation, and dual-layer […]
Balance 5 is a high-precision 5-band parametric equalizer designed for accurate tonal balancing, critical listening, and real-time visual feedback. The standalone version allows you to load and analyse audio files without the need for a […]
Spectra 99 is designed for modern high-definition audio workflows and is fully optimized for sample rates up to 96 kHz, delivering precise and stable spectral analysis in both standard and high-resolution sessions. The standalone application […]
In digital audio processing, one of the most common tasks is reducing background noise while keeping the original signal as clear as possible. This balance is typically expressed as maximizing the Signal-to-Noise Ratio (SNR) — […]
Key Features: Done with FMOD API, diagram shows class dependency and interactive class state structure State containers and segments can be set by order, loop, and randomness Interactive music class uses an open nested loop […]
When mixing in stereo, one of the most overlooked but critical checks is stereo correlation — how “in phase” your left and right channels are. This single number tells you whether your mix will hold […]
The Laplace transform and z-transform are mathematical ways of breaking an impulse response into sinusoids and decaying exponentials. This is done by expressing the system’s characteristics as one complex polynomial divided by another complex polynomial. […]
This is an experimental Schroeder reverberator implementation. I applied three combo filters and then two all pass filters without any buffer. Reverb tail was added to the end of input signal in a simple way, […]
I implemented a recursive moving average filter to reduce background noise in some test audio files. It works in the time domain, which makes it faster than FFT-based algorithms when the number of points is […]
This is my implementation for some of windows used in audio applications. I chose Hamming, Blackman, and Blackman-Harris window functions. Instead of using FFT pack, I directly applied DFT since the DFT size is not […]
This is my DFT implementation to test a wav file. It only accepts mono wav files with 44100 Hz sample rate. I used 32 floating point (with max 23 mantissa) in the range -1 and […]