rt_lpc ~ Realtime Linear Predictive Coding


rt_lpc is a light-weight application that performs real-time LPC analysis and synthesis. It features the following:

  • real-time LPC analysis
  • real-time LPC synthesis
  • visualization of original, predicted, and error waveforms
  • visualization of vocal tract shape from LPC coefficients
  • adjustable LPC analysis order
  • adjustable synthesis pitch shift
  • MIDI controlled pitch (hit ‘m’)
  • lots of other choices (pitch pulse source selection, emphasis filter)
  • STFT plot
  • modular LPC library
  • available on MacOS X, Linux, and Windows under GPL
  • part of the sndtools distribution

soundlab.cs.princeton.edu/software/rt_lpc/
en.wikipedia.org/wiki/Linear_predictive_coding
github.com/lewark/lpc.lv2

sndpeek ~ Realtime Audio Visualizer


sndpeek is just what it sounds (and looks) like:

  • real-time 3D animated display/playback
  • can use mic-input or wav/aiff/snd/raw/mat file (with playback)
  • time-domain waveform
  • FFT magnitude spectrum
  • 3D waterfall plot
  • lissajous! (interchannel correlation)
  • rotatable and scalable display
  • freeze frame! (for didactic purposes)
  • real-time spectral feature extraction (centroid, rms, flux, rolloff)
  • available on MacOS X, Linux, and Windows under GPL
  • part of the sndtools distribution.

www.gewang.com/software/sndpeek
soundlab.cs.princeton.edu/software/sndpeek
www.cs.princeton.edu/sound/software/sndpeek/look

Cava ~ Cross-platform Audio Visualizer


Cava is a bar spectrum audio visualizer for terminal or desktop (SDL).

Cava works on:

  • Linux
  • FreeBSD
  • macOS
  • Windows

This program is not intended for scientific use. It’s written to look responsive and aesthetic when used to visualize music.

github.com/karlstav/cava
Cavalier ~ Visualize Audio With CAVA

FRKB ~ Rapid Audio Organization Tool


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.
  • Waveform Visualization: Provides audio waveform display.
  • BPM Analysis: Displays BPM information.

github.com/coderDJing/FRKB_Rapid-Audio-Organization-Tool

Power Tab Editor ~ Guitar Tablature Editor


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
  • Complete customization of keyboard shortcuts
  • Importing of Guitar Pro tabs

github.com/powertab/powertabeditor

Filtron ~ Variable Filter


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
  • Self-oscillating resonance
  • Fat Internal saturation
  • Post overdrive with Cold and Hot modes
  • Stereo Modulation via CV
  • Integrates with Gatekeeper
  • Optimized for audio rate modulation
  • State of the art zero delay filter design
  • Internally oversampled
  • 100% FREE

polyversemusic.com/products/filtron

RTL Utility ~ Measure Round Trip Latency


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.

oblique-audio.com/rtl-utility

Tube Preamp ~  Tube Preamp Emulation


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.

ddmf.eu/phi-L-Audio-Tube-Preamp

UVR ~ Ultimate Vocal Remover GUI


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).

github.com/Anjok07/ultimatevocalremovergui

Foobar2000 ~ Installing Analog VU Meters


Adding an analog VU meter to the Foobar2000 2.x Default User Interface

Required component: foo_vis_vumeter

Directions:

  1. Download foo_vis_vumeter
  2. Click on foo_vis_vumeter which should install the component
  3. Click Apply and then Ok to restart Foobar
  4. Copy some .bin skin files to the …\AppData\Roaming\foobar2000-v2\vumeter folder (Shift+File>Browse configuration folder)
  5. Select View > Layout > Enable layout editing mode
  6. Right click the tab area in a panel and select Add New Tab
  7. Right click the new tab and rename it “Analog VU Meter” or whatever
  8. Right click the new tab display area and select Add New UI Element
  9. Select ‘Analog VU Meter (DirectX 12)’ from the list (Playback Visualization)
  10. Disable layout editing mode

Parameters:

Right mouse click on the meter display:

  • submenus – Layout / Mode / Levels / Decay / Tuning / Options
  • Fullscreen
  • Skin Selection

Mouse wheel (hover over meters)

  • Adjusts current Tuning selection

Resources:
Analog VU Meter Skin Gallery
Foobar2000 1.x ~ Installing Analog VU Meters

Sources:
www.hydrogenaud.io/forums/index.php?showtopic=33939
www.hydrogenaud.io/forums/index.php?showtopic=106793
www.head-fi.org/t/616963/the-foobar2000-resource-thread/45#post_9382013
foobar2000.ru/forum/viewtopic.php?t=5012 (Russian)

CamillaDSP ~ IIR & FIR Engine For Crossovers & Room Correction


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.

henquist.github.io
github.com/HEnquist/camilladsp

Polyphone ~ Cross-platform Soundfont Editor


Polyphone is an open-source soundfont editor for creating musical instruments, available for Windows, Mac OS X and Linux.

Features:

  • editing of sf2, sf3, sfz and sfArk file formats
  • compatible with jack and asio audio servers
  • built-in synthesizer, controlled by a virtual keyboard or midi signals
  • automatic recognition of root keys
  • automatic loop of samples
  • simultaneous editing of parameters
  • specific tools for musical instrument creation
  • recorder to keep a trace of what is played in a .wav file
  • soundfont browser connected to the online repository
www.polyphone.io
github.com/davy7125/polyphone
community.linuxmint.com/software/view/polyphone