FRKB is a cross-platform desktop application designed for audio professionals (such as DJs). The current beta version is compatible with Windows and will be adapted for macOS once stable. It is still under active development.
Core Features:
Portable: Easily transfer the database to mobile devices for on-the-go use.
Audio Fingerprint Deduplication: Identify and exclude duplicate tracks using audio fingerprint technology, providing prompts during import to keep your music collection clean and efficient.
Ergonomic Shortcuts: Ergonomically designed shortcuts that allow most operations to be performed with the left hand, making the organization process smoother and more efficient.
Direct File Management: When adding tracks, FRKB directly manages the audio files themselves, ensuring that the organization results are immediately reflected in the computer’s folders, achieving a “what you see is what you get” effect.
Power Tab Editor 2.0 – A powerful cross platform guitar tablature viewer and editor inspired by the ceased development and missing source code from the original Power Tab Editor. This project is open-source and written from scratch so that your favorite tabbing platform can continuously grow with your needs.
Key Features:
Cross platform – Windows, Mac, & Linux
Tabbed layout for opening multiple files at the same time
Mixer interface for adjusting volumes during playback
Filtron is a 12dB state variable filter that can smoothly transition between lowpass, bandpass, and highpass. It is capable of self-oscillation with resonance levels that can reach up to 11! Filtron also features a fat sounding internal saturation algorithm and a sizzly post overdrive with two modes to choose from: cold and hot.
12db state variable filter
Transition smoothly between Lowpass, Bandpass, Highpass
RTL Utility is a tool for measuring the Round Trip Latency of your Digital Audio Workstation (DAW) and audio interface. The utility is used for low latency performance testing by system builders, reviewers, device manufacturers and at dawbench.com.
When your DAW sends data to your audio interface for playback, it doesn’t send a continuous stream of data one bit at a time. What it does is fill up a section of RAM called a buffer and sends that in a single message when it is ready. Before sending the next message it has to fill the buffer again. This wait time introduces a latency, or delay, between something happening in your DAW and when you actually hear it.
While you are recording, the audio interface buffers and sends data to your DAW in a similar fashion. This introduces latency into your recordings.
If you send a signal from your DAW, out through the audio interface and back in via a loopback patch, then there will be a round trip latency which is the sum of the output and input delays. This is the RTL.
The plugin is very unique in the sense that it gives you full control over all aspects of the modelled circuit. All values of all the capacitors, resistors etc. can be tuned by the user.
This is a free plugin. It comes in all the usual plugin formats (VST/VST3/AU/AAX), for 32 and 64 bit hosts, and for Mac Intel & Silicon.
This application uses state-of-the-art source separation models to remove vocals from audio files. UVR’s core developers trained all of the models provided in this package (except for the Demucs v3 and v4 4-stem models).
A tool to create audio processing pipelines for applications such as active crossovers or room correction. It is written in Rust to benefit from the safety and elegant handling of threading that this language provides. Supported platforms: Linux, macOS, Windows.
Audio data is captured from a capture device and sent to a playback device. Alsa, PulseAudio, Jack, Wasapi and CoreAudio are currently supported for both capture and playback.
The processing pipeline consists of any number of filters and mixers. Mixers are used to route audio between channels and to change the number of channels in the stream. Filters can be both IIR and FIR. IIR filters are implemented as biquads, while FIR use convolution via FFT/IFFT. A filter can be applied to any number of channels. All processing is done in chunks of a fixed number of samples. A small number of samples gives a small in-out latency while a larger number is required for long FIR filters. The full configuration is given in a YAML file.
The Audio File 