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See Creative coding, Audio


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  • https://github.com/afms135/signl - an audio effects processor developed to provide an open-source (modifiable and extendable), real-time (no noticable playing latency), low-cost (Raspberry Pi based) standalone unit for musicians. Development started at the University of Glasgow for ENG5220: Real Time Embedded Programming 2019-2020.


  • StudioRack - An open-source audio plugin ecosystem Our tools are built using GitHub and open-source libraries, ensuring you will always be able to access them.
  • https://github.com/studiorack/studiorack-app - Audio plugin app, searchable list of plugins to install and share

  • KVR Audio is a community and news site for popular Audio Plug-in formats and related subjects, such as sample libraries and mobile apps. Our mission is to supply up to date news to VST, AU, RTAS, DX and DSSI/LADSPA plug-in and iOS and Android App users in a friendly, up-front and timely manner.



  • Alsaequal - a real-time adjustable equalizer plugin for ALSA. It can be adjusted using an ALSA compatible mixer, like alsamixergui or alsamixer. Alsaequal uses the Eq CAPS LADSPA Plugin as it's default equalizer but you can change it to use almost any LADSPA plugin, like mbeq from the swh-plugin package.


  • jackEQ - intended to provide an accessible method for tweaking the treble, mid and bass of any JACK aware applications output. Designed specifically for live performance, it is modelled on various DJ mixing consoles which the main author Patrick Shirkey (aka DJ Kotau) has worked with live. LADSPA.


  • x42-eq - a 4 band parametric equalizer with additional low+high shelf filters, Low and High-pass, as well as an optional, custom GUI displaying the transfer function and realtime signal spectrum or spectrogram. It is available as LV2 plugin and standalone JACK-application.


  • Lv2fil - Stereo and mono LV2 plugins, four-band parametric equalisers


  • EQ10Q - an audio plugin bundle over the LV2 standard (http://lv2plug.in) implementing a powerful and flexible parametric equalizer and more (gate)


  • https://github.com/QuentinFAIDIDE/FutureEqualizer - Equalizer with a funny harmonic boost function. This eq computes harmonics from the 3rd peaking eq frequency with 4 Range Modes and let you tweak the gain and resonance as you want. Four modes are available using the range knob, first quarter: replicate 3rd filter one octave above, second quarter: replicate 3rd filter with a 3-notes major chord and one octave above, third quarter: replicate 3rd filter with a 3-notes minor chord and one octave above, last quarter: replicate 3rd filter one and two octave above.


  • Luftikus - a digital adaptation of an analog EQ with fixed half-octave bands and additional high frequency boost. As an improvement to the hardware it allows deeper cuts and supports a keep-gain mode where overall gain changes are avoided.


The Pilgrim


  • NotNotchFilter - a performance-oriented filter designed to replace the mid-EQ found in a standard 3-band DJ mixer. The key advantage of this filter is that it cleanly cuts out single voices or instruments in a track, whereas a standard 3-band filter dampens them. This is because NotNotchFilter, as the name suggests, is not actually a notch filter. Rather, it is a combination of a hipass and lopass filter which work on opposite sides of the target frequency.


  • HiLoFilter - a simple hipass and lopass filter which can be easily controlled with a single knob. It is loosely inspired by the same type of filter found on some Pioneer DJM mixers, and also the The Pilgrim, another great plugin which provides roughly the same functionality.

Frequency Response Correction

  • Frequency Response Correction - Java. The program provides different filter functions to equalize the source signal in a way that it is linear in its frequency response after passing the room and its characteristics.



  • https://github.com/ashafq/a5eq.lv2 - an LV2 plugin featuring a 5-band equalizer that includes a Low Shelf, three Peaking filters, and a High Shelf. It's crafted to assist both amateur and professional audio engineers in precise sound sculpting and can be employed across various audio engineering tasks such as studio production, mastering, or live sound tweaking.

ZL Equalizer

  • ZL Equalizer - a minimum-phase equalizer plugin with the following key features: Multiple Filter Settings: Supports 16 frequency bands, 8 filter types, 5 stereo modes, and 7 variable slopes. High-Quality Sound: With 64-bit floating-point processing and de-cramping technique, outstanding performance is ensured in both low-end and high-end. Adjustable Dynamics: Adjustable threshold, attack, release, and side-chain frequency, etc. Carefully Designed Interface: Interactive spectrum graph, smart collision detection, and smooth animations


Audio Mastering Suite

  • Audio Mastering Suite - Java, works for Windows, Mac and Linux. Features: - Native signal processing by using the resolution and sampling frequency of the source material. - The supported audio format is wav (PCM) 16/24Bit with 44.1kHz, 48kHz, 88.2kHz, 176.4kHz, 96kHz or 192kHz sampling frequency. - High quality resampling of any audio format into 16Bit/44.1kHz by applying "Bandlimited Interpolation". - 64Bit floating point signal processing. - The signal processing chain (order of components) is freely configurable. - Audio monitoring switchable between input and output signal.


  • JAMin - the JACK Audio Connection Kit (JACK) Audio Mastering interface. JAMin is an open source application designed to perform professional audio mastering of stereo input streams. It uses LADSPA for digital signal processing (DSP)


XO Wave

  • XO Wave - a digital audio workstation designed to meet the needs of audio and video professionals, with a focus on CD Mastering and audio for video work. XO Wave provides professional-grade capabilities for manipulating audio, with a familiar and elegant interface. Support for direct recording from any Core Audio device, importing and exporting files in a variety of formats (including songs from iTunes playlists), CD burning, multi-track editing, support for many Audio Units, and QuickTime synchronization and export, make XO Wave a great tool for all kinds of audio work, including podcasting, vodcasting, CD Mastering, mixing and more! Java.



  • matchering - a novel Containerized Web Application and Python Library for audio matching and mastering.It follows a simple idea - you take TWO audio files and feed them into Matchering: TARGET (the track you want to master, you want it to sound like the reference) REFERENCE (another track, like some kind of "wet" popular song, you want your target to sound like it). Our algorithm matches both of these tracks and provides you the mastered TARGET track with the same RMS, FR, peak amplitude and stereo width as the REFERENCE track has.




  • https://github.com/robbert-vdh/diopser - lets you rotate the phase of a signal around a specific frequency without affecting its spectral content. This effect can be used to emphasize transients and other parts of a sound that in a way that isn't possible with regular equalizers or dynamics processors, especially when applied to low pitched or wide band sounds. More extreme settings will make everything sound like a cartoon laser beam, or a psytrance kickdrum. If you are experimenting with those kinds of settings, then you may want to consider temporarily placing a peak limiter after the plugin in case loud resonances start building up.

Windows VST

Windows / Mac

Compression / limiting


  • VLevel - a tool to amplify the soft parts of music so you don't have to fiddle with the volume control. It looks ahead a few seconds, so it can change the volume gradually without ever clipping. Because the volume is changed gradually, "dynamic contrast" is preserved.


  • DPL1 - a look-ahead digital peak limiter, the kind you would use as the final step to avoid clipping when mastering or mixing. It can be used as an effect on individual instrument tracks as well. Latency is 1.2 ms rounded up to the nearest multiple of 8, 16 or 32 samples depending on sampling frequency. This amounts to 56 samples at 44.1 kHz, 64 samples at 48 kHz, and twice those values for 88.2 or 96 kHz.

Radium Compressor

  • http://users.notam02.no/~kjetism/radium/compressor_plugin.php - The system compressor in Radium is also available as an independent audio plugin. The unique interface: Accurately visualize the sound compression. Rapidly helps you find good compressor settings. With the Radium Compressor, you spend less time listening and finetuning. The interface quickly makes you find the sound you want. Other Features: Top class audio quality. The DSP code is implemented by Julius O. Smith III. Julius O. Smith III is a professor in Music and EE at Stanford University, and he is one of the legends in audio research. The compressor is based on code Julius has written for Faust. OSS/$

Visual Compressor

Molot Lite


  • https://github.com/magnetophon/CharacterCompressor - A compressor with character. A bit experimental: It works and sounds wonderfull, but has too many parameters, so is a bit fiddly to use. Also; I have no idea what to name the parameters, or how to explain a lot of them.

x42-limiter / dpl.lv2

  • x42 Digital Peak Limiter - (aka dpl.lv2) is a look-ahead digital peak limiter based on Fons Adriaensen's DPL-1. It is intended to be used as final step to avoid clipping when mixing or mastering, but can also be used on individual tracks as well. It comes in mono and stereo variants, the stereo version applies the same gain reduction to both channels. Latency is 1.2 ms, rounded up to the nearest multiple of 8, 16 or 32 samples depending on sampling frequency. This amounts to 56 samples at 44.1 kHz, 64 samples at 48 kHz, and twice those values for 88.2 or 96 kHz.


  • https://github.com/x42/darc.lv2 - a general purpose audio signal compressor.The compression gain is controlled by threshold and ratio only: Makeup gain is automatically set to maintain equal loudness at -10 dBFS/RMS with a soft knee. It is available as LV2 plugin and standalone JACK-application.




  • https://github.com/magnetophon/CompBus - A group of compressors mixed into a bus, sidechained from that mix bus.It can have any number of compressors with any number of channels per compressor.

Audio Compressor & Limiter



Xhip Compressor

  • Xhip Compressor - designed to loosely model a circuit I designed myself. It is like a pedal or what you might find in a channel-strip. It isn't designed to function as a limiter and it doesn't have instant response, look-ahead, low distortion or any of that stuff. That is exactly what makes it useful. VST

Xhip Limiter

  • Xhip Limiter - A counterpart to the compressor for limiting with the most simple design possible. This limiter is designed for maximal sustain and minimal distortion. It aims to be as transparent as possible in terms of timbre without using any advanced techniques. VST


  • RoughRider3 - one of the most popular dynamics processors on the planet, with well over a half a million downloads over its decade-plus lifespan, and is in heavy daily use by producers the world over. With Audio Damage's RoughRider3, the original is to include an external sidechain input, the ability to turn off the built-in "warming" filter (the FULL BANDWIDTH button), and much more accurate metering.






Windows VST


  • nova67p - a parallel parametric equalizer plugin combined with a compressor. The compressor can optionally operate in frequency dependent and split-band modes. In this case the plugin operates as a parallel dynamic equalizer.


  • https://delamanchavst.wordpress.com/2013/11/20/sixtyfive-compressor-dbx-165a-style-is-now-free/ sixtyfive is a vintage-style, RMS Compressor, inspired by the dbx® 165A, a classic 1970’s compressor found in many studios, but it also adds a couple of new twists. It’s a soft-knee RMS compressor with a vintage flavour. The RMS detection and soft-knee help to give a smooth and musical compression experience, the non-linear response imparts character and the gentle saturation brings colour and warmth. As well as all of the originals features, this version adds parallel compression, extended knob ranges and peak metering


  • GTA - a vintage style ‘character’ compressor, designed like its muscle car name-sake for brute power, pure speed and to make a loud noise. It is stripped down for ease of use and has a unique vintage colouring on top.

Rough Rider 2 Compressor

  • Rough Rider Compressor - a modern compressor with a bit of "vintage" style bite and a uniquely warm sound. Perfect for adding compression effects to your drum buss, it also sounds great with synth bass, clean guitar, and backing vocals. Definitely not an all-purpose compressor, Rough Rider is at its best when used to add pump to rhythmic tracks. Rough Rider is available as both 32- and 64-bit VSTs for Windows, and as a Universal Binary AU/VST for OSX.

Xfer OTT

  • OTT - a free re-creation of a popular aggressive multiband upwards/downwards compressor set-up used by many dubstep and electro producers, by Xfer Records


W1 Limiter

  • W1 Limiter - a clone of Waves L1, with identical output, as well as an approximation of Waves L2.




  • abGate - a LV2 noise gate plugin in the LV2 format to manage noise. A noise gate is a component which attenuates an audio signal when it falls below a set threshold, so it can be applied to an audio track which has one or more periods of silence where no noise should be apparent.







  • GT10QM / GT10QS - Versatile noise-gate plugin with mono (M) and stereo (S) versions. This plugin provides the usual controls in a dynamic gate processor: threshold, range, attack, release and hold. But also features some extra parameters to allow an accurate adjustment. The gain control is useful to bring the input sound level in the correct VU meter range to be able to tune the threshold right. Even though the internal processing is performed in floating point at 32 bits, it’s helpful to adjust the input gain in a standard range to achieve an optimum level visualization. Together with VU meter, there is also a gain reduction meter that aids to set up the controls. Besides the power of dynamic section controls (DYN), there is also a filtered side chain section (SC). Here it is possible to adjust a low pass filter (LPF) and a high pass filter (HPF) and listen the actual side chain signal after filtering with the “Key” button. This side chain flexibility is very interesting to avoid unwanted noises to start triggering the gate.

VeJa Noise-Gate

Xhip Gate

  • Xhip Gate - A gate effect including side-chain input and a few other additional features. Ideally the trigger/threshold detector and envelope generator would be separate plug-ins, although this is not so practical in the majority of cases. Instead a general-purpose AhR envelope is used.



  • moot - a flexible audio mute plugin with a number of additional features. The mute switch (Hit Me) can be assigned to a single midi keyboard key and act in 3 ways: as a latch (one press mute, release stays muted, next press unmute, release stays unmuted), in default mode (hold down to mute, release to play audio), in invert mode (hold down to play audio, release to mute). Windows VST2.
  • gator - creates a random steppy, gated effect, triggering volume between 2 adjustable values by a tempo-sync'd probability-based step sequencer. An LFO can modulate the volume when triggered and trigger pulse length, attack and release can shape the sound. Windows VST2
  • StormGate1 - an unique and innovative amplitude rhythmic gating effect which lets you draw the gating patterns freely by hand or with the aid of powerful drawing tools. PC VST format and MAC VST and AU formats.

  • A1TriggerGate - "There are many sequenced / rhythmic gates out there, but honestly nothing really satisfied me yet. So I decided to write my own plugin for this task..." - Windows/Mac



  • https://github.com/zwaba/MIRRORZ - Plugin lv2 build with Max Gen~ for the Mod DwarfAudio IN is permanently recorded in a 32 seconds buffer (normal speed) the TRIGGER function of MIRRORZ allows you to play backward the last chords/ melody you just played, same for the LOOP function but looped.


Calf Vintage Delay

  • Calf Vintage Delay - based on bpm-oriented delay time settings. Additionally the delayed signal is processed by a filter to simulate old tape-machine based delay effects. Some options of the stereo distribution of the delayed signal makes it very flexible and wide-ranged in sound.


  • TAL-Dub - a vintage style delay effect. It can be used for a wide range of delay effects from clean to extreme distorted, resonating never ending delays :-)

TAL-DUB-II is an extended version of TAL-Dub-I with a completely new sound engine. A 4x oversampled distortion stage allows to add vintage distortion to the delayd signal, but its also possible to make clean delays. A sinus LFO has the possibility to modulate delay time and low pass filter cutoff. Adjustable LFO stereo width is also included. An analog sounding 6dB low pass filter with resonance and a 3dB high cut filter are also parts of TAL-DUB-II. Different routing options open a wide range of possibilities.

TAL-DUB-III is an easy to use delay device with some special features. Its no tape delay emulation and has its own sound. It has an alias free saturation stage, a non-linear 6dB low pass and a 3dB high pass filter that are included in the feedback path of the device. An input drive knob allows to adjust the saturation level. Pop-up menues show the current values of volume, delay-time and feedback knobs. A tab button allowing to adjust the delay time for live sessions.

Xhip mDelay

  • Xhip mDelay - dual delay with stereo, cross and ping-pong modes. It has an LFO and can be used to create anything from flangers and chorus to ordinary delays and strange detuned echos all the way up to extreme pitch modulation effects.


  • https://github.com/qbroquetas/IV-XDelay - a free and open source delay VST effect modelled after a renowned vintage tape recorder. Its aim is to provide music producers with a fun and useful tool that sounds and works like vintage tape echo boxes, as well as to serve as an educational tool for anyone interested in audio development and digital signal processing.


  • https://github.com/MrBollie/bolliedelayxt.lv2 - The evolution of my bolliedelay.lv2, features full control over two seperate channels, HPF/LPF on both, the delay path and feedback path, fractional delay, that enables modulation

Cocoa Delay

  • https://github.com/tesselode/cocoa-delay - a free, open-source delay plugin in VST format. It focuses on clean design, easy operation, and a warm, lively sound.Notable features: Delay time drift, for giving the wet signal a slight wow/flutter effect; Adjustable wet level ducking based on the input signal; Static, ping-pong, and circular pan modes; Drive section based on open-source Airwindows saturation code

Pitched Delay


  • BOWECHO - Quad Modular Delay. Use the key R7P1R64720175164548606433 to unlock the demo version. OS X, Windows and Linux.






  • Tapiir - a simple and flexible audio effects processor, inspired on the classical magnetic tape delay systems used since the early days of electro-acoustic music composition. It provides a graphical user interface consisting of six delay lines, or "taps", which can introduce an almost arbitrarily big or small delay to their inputs and can be feed back to each other. A wide set of effects can be easily achieved by properly configuring and connecting the delay lines: complex echo patterns, resonances, filtering, etc. Delays, interconnections and gains can all be controlled in real time.


  • Rechoir - a delay effect where all the echoes can either harmonize with the source signal or be pitch shifted manually.Harmonic changes can be synced to the tempo and time signature of your sequencer, keeping both the echoes as well as their pitch modulations strictly timed to your composition.



  • https://github.com/ptablas/M-S-Echottage - a Cross-platform VST3 audio plugin compatible with most Digital Audio Workstations (DAWs). It has been developed using the JUCE framework, given its extensive DSP library, GUI capabilities, and efficient API.


  • Tom Pong is a ping pong delay VST Plugin. Windows only. A ping pong delay is a delay which alternates from one speaker to another. If the balance is set to 0.5, a stereo signal will invert itself each time the delay buffer is played back. If the balance is set to either side of 0.5, the signal bounces back and forth between the speakers. Tom Pong can be synced to the host sequencer, or delay times can be set by sample count. Feedback and output level are also adjustable.

  • combover - a comb delay, and four more comb delays, and a step sequencer that does odd things, and a pool of prime numbers that make stuff happen, and some mix n match pitch detection. It also has an xy pad and wet / dry controls that let some stuff through and other stuff not. It does crazy stuff, you should stop reading these words and find out for yourself.

  • Regrader - provides a delay effect in which the repeats degrade in various ways to provide a malevolent twist on the ears.Echoes disintegrate, are decimated, piercingly oscillate and are finally mangled into a sound reminiscent of a jet engine processing gravel. Each of the modules within Regrader have different routing possibilities to finely tune your carefully sculpted noise. Think industrial or dark sound design to find a purpose for the plugin or see what others say.

  • Shoegaze VST - Features: reverse delay, a staple of shoegaze, feedback, minimal interface, no DRM, For Windows only.


See also #Convolution

  • Spin Semiconductor FV-1 - a complete reverb solution in a single IC. With integrated stereo ADC and DACs, the FV-1 can be treated like any other analog component in your products signal path. The FV-1 can access a total of 16 programs, 8 are built in to the internal ROM and the designer may elect to connect a serial EEPROM with 8 additional programs. By using an external EEPROM, the designer can distinguish his product from others by creating a custom program set without the need for a microprocessor in the system.With 3 potentiometer inputs, programs may have real time variable parameters such as decay time in a reverb, rate and depth in a chorus or frequency in a filter. These inputs are available as coefficients to your program and may be used independently of each other.The rich instruction set allows users to program effects of all kinds. With instructions like LOG and EXP, users can easily program audio expansion and compression routines. Integrated digital LFOs and ramp generators allow for programming chorus, flange and pitch shift.

Dragonfly Reverb



  • Multiverb is an audio plug-in that produces a reverb effect on a monaural or stereo audio track. It is available as both a VST plug-in for Windows (Win32 and Win64) and an LV2 plug-in for Linux. Multiverb implements an acoustic system that is an interconnection of multi-port acoustic elements. For a full technical description of the Multiverb algorithm please see my paper in the Journal of Multidisciplinary Engineering Science and Technology (JMEST).


  • REV1 - a reworked version of the reverb originally developed for Aeolus. Its character is more 'hall' than 'plate', but it can be used on a wide variety of instruments or voices. It is not a spatialiser - the early reflections are different for the L and R inputs, but do not correspond to any real room. They have been tuned to match left and right sources to some extent.

DF Zita Rev1


  • mverb - Studio quality, open-source reverb. Its release was intended to provide a practical demonstration of Dattorro’s figure-of-eight reverb structure and provide the open source community with a high quality reverb.




  • Protoverb - an experimental reverb based on the idea of a "room simulator". Most algorithmic reverbs try to avoid resonances or model the reflections of sound from a rooms walls. Protoverb does the opposite. It builds up as many room resonances as possible, modeling the body of air in the room. It therefore does not need to modulate or colour the signal. The result is a very natural sounding reverbration with some interesting features: Long standing frequencies resonate louder, as if the air takes some time to get excited. Multiple instruments don't mash into a diffuse mud, they stay distinct. If you play a short melody, the room seems to repeat a ghost echo of that melody. Those properties are indeed found in churches and large halls, but they're rarely found in conventional algorithmic reverbs.


Xhip Reverb

  • Xhip Reverb - A reverb effect designed to avoid an ultra-smooth fade-to-noise decay.

Teufelsberg Reverb





  • https://github.com/nhthn/nh-ugens - an open source algorithmic reverb unit in a single C++11 header file (src/core/nh_hall.hpp). Features: Allpass loop topology with random modulation for a lush 90's IDM sound; True stereo signal path with controllable spread; Infinite hold support; Respectable CPU use; Permissive MIT license; No dependencies outside the C++ standard library; Bring your own real-time safe memory allocator (no unwanted malloc calls!); Sample rate independence, with SR specified at run time, not compile time; Clean, readable source code for easy modification

(Schroeder) reverb


KR-Reverb FS

  • KR-Reverb FS - an easy to use Reverb processor based on features found on our commercial product KR_Space. KR-Reverb FS is designed for ease of use by adjusting internally the equalization and damping controls to optimal levels for producing a warm reverb sound suitable for a wide range of applications.






  • Room Reverb - an audio plugin that lets you add reverberation to your recordings in your Digital Audio Workstation (DAW). Room Reverb uses the implementation of Moorer's early reflection model and Jon Dattorro's Progenitor Reverb from the Freeverb3 signal processing library.


Simple Reverb


Sound retainer

Sostenuto, "infinite sustain"


Infamous Stuck

  • Infamous Plugins: Stuck - a clone of the electro-harmonix freeze. It drones the note being played when the "Stick It!" port is set to 1 (or the CV port input goes above 1), causing the note to be "stuck". Once the port falls below 1 the drone is released with a decay set in seconds. The drone is added to the dry signal (so original signal is passed through at all times un-processed). This plugin is pretty useless except in live situations, though I'd love someone to creatively prove me wrong.




  • Richter - a two LFO tremolo, the interaction of which creates far more complex volume oscillations than can be created with a single LFO. The rate and depth controls of each LFO can each be modulated by two more LFOs to create variations in frequency and depth, and tempo sync is available on all oscillators. VST, VST3, and Audio Unit, Windows, Mac, and Linux.


Xhip Tremolo

  • Xhip Tremolo - a standard tremolo with the addition of a couple features. One is that it has phase adjustment for the LFO, allowing it to produce a "wide" tremolo panning effect. Second is that the waveform can be adjusted between pulse-like, linear and saturated which produce different characters mimicking various old tremolo effects.


Windows VST


  • Flutterbird - a free, open-source effect plugin for adding pitch and volume fluctuation to incoming audio. It can be used for traditional wow/flutter effects as well as more chaotic, extreme modulation. Flutterbird has four separate oscillators with adjustable speeds that can be mapped to either pitch or volume.




  • https://github.com/jkbd/scannervibrato - A LV2 plug-in for the Chorus or Vibrato modulation effect used in different tonewheel organs. Based on a physical informed model implemented in Faust.


Xhip Phaser

  • Xhip Phaser - standard phaser effect with a variable number of stages. This phaser doubles as a frequency dependent delay, chorus effect and vibrato because it allows you to use up to 128 stages.

Stone phaser




  • vm.lv2 - various plugins with a programmable stack-based virtual machine to modify up to 8 LV2 Control, CV, Audio, Atom and MIDI signals. To be used when that specific simple filter plugin you desperately need does not yet exist.


  • https://github.com/jd-13/Songbird-Formant-Filter - Songbird is a modulated vowel/formant filter. Select two vowels and then modulate between them using either the manual slider or the LFO. There are five vowel sounds and two modulation modes available to choose from. The "freq" modulation mode allows you to modulate a single filter between the two vowels chosen, to create a powerful vocal sound. The "blend" mode provides a more subtle effect, modulating the mix between two parallel filters. Songbird is ideal for creating both vocal bass sounds and subtle filter sweeps.

Xhip Vocal

  • Xhip Vocal - A simple phoneme synthesizing filter made from several parallel formant filters. This filter provides AEIOU formants with control over center frequency and Q with smooth interpolation between formant sets with the phoneme control.

Wolf LFO







Speech to birdsong conversion


  • https://github.com/varlen/randstepfilter - My first VST plugin. Created this project to refresh my C++ skills, recall some signal processing techniques and learn VST plugin development with iPlug2 framework




  • https://github.com/claytonotey/paukn - the pitched audio knife - VST effect with various midi note controlled filters (lowpass, hipass, comb, decimator, granulator, digital waveguide)



Windows VST / Mac

  • sfilter - creates a stepped filtered sequence, to create gating, sweeps or rhythmic modulation of filter cut-off. It uses a variable state filter, varying between 2 adjustable cut-off values according to a tempo-sync’d step sequencer. An LFO can also modulate the filter for extra movement

  • pfilter - creates a steppy, gated effect, but using filter cut-off instead of volume to give a wider range of possibilities. It uses a variable state filter, triggered between 2 adjustable cut-off values by a tempo-sync’d probability-based step sequencer. An LFO can modulate the filter when triggered and trigger pulse length, attack and release can shape the sound

  • sumo - an effect plugin to make any sound as fat as you like. It’s good for fattening up leads and basses, adding some weight to pads, making your vocals chubby and your drums obese

  • DtBlkFx - a freeware Fast-Fourier-Transform (FFT) based Multi effect VST plug-in for Windows and Mac. Precision parametric equalizing with sharp-roll off, adjust individual harmonics of a sound. Harmonic based (or comb) filtering, including active harmonic tracking. Various types of noise control, change contrast between loud and soft frequencies, clip frequencies or apply sound smearing. Frequency shifting, harmonic and non-harmonic shifting, including active harmonic repitch.

  • DvnKeyFilter - Give more expressiveness in your monophonic melodic lines with DvnKeyFilter!! It's a low pass filter with the cutoff frequency controlled by key tracking and a second parameter "Expression" assignable to any midi cc (or disable it and use your DAW envelope).

  • Soundemote: Flower Child Filter - aptly named for its goal to have a clean and resonant sound. There is also a switch for an aggressive growling version that is sure to cater to your more destructive side.


  • You Wa Shock ! - VST/Winamp effect to brighten up and maximize any truck you hand it.



TF Chorus

  • https://github.com/Umcaruje/tfchorus - an LV2 and LADSPA port of TF Chorus by TraumFlug. The great, unique chorus effect was found in the Armstrong package of an old PPA. Any information about this plugin seems to have disappeared from the internets. There was no license, except that it is implied to be "free, open source software" (see src/tfcho_orig.cpp, the original source) and that the rest of Armstrong is released under GPLv2. For now let's say this port is under the same license as the original, whatever it may be.

JPC Ensemble Chorus

YK Chorus

  • https://github.com/SpotlightKid/ykchorus - A chorus audio effect plugin based on DSP code by Togu Audio Line (TAL), inspired by the one found in certain well-known Japanese vintage analog synthesizers (You Know which).



Ring modulation


Windows VST

  • https://www.meldaproduction.com/MRingModulator - performs classic ring-modulation effects using one or two oscillators. With a clean interface that gives easy access to more advanced controls like our adjustable phase difference and shape features, including editable custom waveforms and harmonics.

  • ring thing - a multi-flavour ring modulator, with frequency and mix level controlled by an XY pad and each axis modulated by it’s own tempo-sync LFO. The modulation in both axes is shown graphically on the XY pad.


Wolf Shaper




  • https://github.com/vvvar/PeakEater - a free, easy to use waveshaping plugin. PeakEater lets you choose between different waveshaping functions to clip everything above ceiling level. Therefore, you can boost overall volume of your track safely without worrying that some nasty peak would go above maximum allowed volume level of your DAW. It supports various features such as multiple clipping types, oversampling and visualizations of clipping process that aims to make clipping easy and fun.



Windows VST

  • https://www.meldaproduction.com/MWaveShaper - goes beyond a traditional wave-shaping plugin. Unlike the conventional approach of providing a few predefined patterns, MWaveShaper lets you construct your own shape creating a much greater range and control over your sound.

  • https://www.meldaproduction.com/MComb - an extremely powerful multi-comb filter plugin. Using its 2 extremely versatile modulators it becomes a powerful processor, which can follow a simple LFO, react to input levels, MIDI note, input pitch...




  • SDRR - built to satisfy almost all of your saturation desires. It provides a comprehensive set of controls to manipulate the character of the saturation to make it fit exactly. SDRR offers four different main modes: TUBE, DIGI, FUZZ, DESK and reacts dynamically to the input signal. Each mode has its unique crosstalk behavior, which can be switched off or exaggerated. A unique RMS level difference metering mode makes level matching an easy task. SDRR can be different things: a saturation, a compressor, an EQ, a bit-crusher, a subtle stereo widener, or simply add some movement to your tracks with the DRIFT control. Add warmth, depth and character to your tracks with SDRR.


  • IVGI - can deliver very soft and subtle saturation, that feels at home on the master buss. It is equally capable of very dense and dirty distortion effects to spice up single tracks. IVGI's base sound is comparable to the DESK mode in the big brother SDRR. Windows/Mac VST.



  • https://github.com/jpcima/Bass21 - VST3 is a virtual-analog simulation of the famous Bass Driver DI Version 1 preamp pedal from Tech21. The discretization, while far from being exact, is moderately close to the original.

Make It Crispy

Rotating speaker

  • http://x42-plugins.com/x42/x42-whirl - x42-whirl is a designed to imitate the sound and properties of the electromechanical rotating speaker device that brought world-wide fame to the name and products Don Leslie. It is a standalone version of the effect that came to be with the setBfree synth. Rather than simulating the net effect of the electromechanical device, x42-whirl physically models the properties, which results in very accurate representation of the sound of the real device. Since all individual parts are modelled, x42-whirl not only provides advanced control, but also facilitates customizations, some of which are not feasible in the mechanical device.



  • Arcangel - a jack effect for arctan distortion. Sounds nice and grungy without clipping at high levels, and sounds nice at lower levels.


  • https://github.com/moddevices/mod-distortion - Analog distortion emulation developed by mod team (lv2). The effects were developed suposing you have a -15dB input signal (measured with digital peak meter) when you play loud, so its recommended that you ajust your input gain to this level. We recommend that you use only the stable plugins: -DS1 -Big Muff Pi






  • Jk-drive - The plugin does basic wave shaping distortion with 2x oversampling. It produces both even and odd order hamonic distortion.


  • deteriorate-lv2 - A set of plugins to deteriorate the sound quality of live inputs. The set contains two plugins: A basic granulator, a basic downsampler


  • Deathcrush is a distorsion plugin, made up of some raw effects (such as bitcrusher and compressor) to really ruin your gentle sounds.

Bit Crusher






  • WubFlip - It sort of flips high or low values beyond a threshold making a dirty distorted mess of the sound that might be useful for people making wanting big dirty breaks or synths or something like that. Play around with the sliders. The upper threshold slider needs to be higher (to the right of) than the lower threshold slider otherwise you'll get no sound. The difference between them effects the sound. Then the multiplier slider effects how much "flipping" gets done. LADSPA.


  • Carve - a wave shaping distortion with several available waveforms and two distortion units, which can be configured in serial, parallel or stereo, and blended with the dry sound. VST, VST3, and Audio Unit, Windows, Mac, and Linux



  • Misstortion VST by Nimble Tools - Does its best to be like Logic's Clip Distortion, which makes it very useful for the Hardstyle genre. Hard clip: Simple hard clip distortion. Soft clip: Hyperbolic soft clip distortion. Symmetry: Make your waveform asymmetrical for more finetuned distortion.



  • https://github.com/mzuther/Screamer - Version 1.0 is the accurate Faust reproduction of a VST plug-in I wrote in 2003. In later versions, I have modified the parameter ranges and added some new functionality. This signal mangler features a weird kind of overdrive (see source code for exact formula), a hard-clip distortion, LFO-modulated bit reduction and the possibility of calculating the modulo of a signal. I have never heard anything like it, which is why I finally release it to the wild.













  • https://github.com/sevagh/audio-degradation-toolbox - Python 3 implementation of the MATLAB Audio Degradation Toolbox. The license is GPL due to the original code being GPL. The aim is full feature parity with: Original MATLAB toolbox (with ISMIR2013 additions), A similar tool, audio_degrader. This tool can read non-WAV files as input, but only outputs single-channel WAV files - this is because I find that WAV is the most universal format with friendly-licensed libraries in any language.


  • https://github.com/aleksrutins/dynamite - A customizable LV2 distortion plugin.Screenshot of the Clipping panel. Dynamite has two modes: Clipping and Distortion. Clipping works by amplifying the audio signal (the "Drive" setting), cutting it off at a certain frequency (the "Threshold" setting), and then applying makeup gain (the "Gain" setting). Transmogrifier works by rounding the wave to a square wave, cutting out amplitudes below the "noise threshold", and applying makeup gain (generally negative, to prevent hearing loss). *The UI is GTK2-based so it may not work with all DAWs.*


  • https://github.com/MeijisIrlnd/Transfer - a waveshaper based distortion, except the transfer function is JIT compiled from text inputted by the user. Starting as a tool for testing different transfer functions in other projects, it has evolved into a pretty cool and unique plugin (disclaimer it might not be that unique I have done absolutely no market research whatsoever). It also has a built in gate, and an emphasis filter, which applies the filters to the signal pre waveshaping, then applies the exact opposite filter post waveshaping. Due to popular demand (by popular I mean that one guy on reddit), it now also has has 16x oversampling and you can't turn it off sorry I don't make the rules.


  • MetalTone is modeled after the renowned BOSS MT-2 Metal Zone(*), a high gain distortion pedal with an advanced EQ section.
  • CollisionDrive is modeled after the Horizon Devices Precision Drive(*), a modern overdrive pedal with a built-in noise gate.
  • The Rumor is modeled after the Devi Ever Ruiner(*), a pure, nasty growling Bass Fuzz pedal, with bold out-front presence, and cutting articulation.
  • TubeScreamer is a analog emulation of the classic Ibanez TS-9(*)
  • The ValveCaster is a famous DIY guitar pedal schematic first put together in 2007 by Matsumin
  • The BoobTube is a more versatile variation of the ValveCaster, a famous DIY guitar pedal schematic first put together in 2007 by Matsumin.

Bedroom Studio

Current plugins list:

  • Metalluga - the hard and crips distortion with five controls to customize the effect for your needs: Drive, Level, Weigth, Resonance and Warmth. The main control here is Level, all other builds around it. If don't touch too much the distortion stuff, you can use Metalluga in more soft genres such as blues.
  • Bronza - the plain fuzz pedal with two parameters - Level and Fuzz. Sounds like in sixties
  • Grelka Overdrive - the classic overdrive, has Drive, Level, Lows and Treble parameters to define the sound.
  • Charm - the saturation effect, makes sound more "analog".



Windows VST

  • https://github.com/creativeintent/temper - a digital distortion audio plugin targeting VST, VST3, and AU for OS X and Windows. It builds upon traditional waveshaping techniques using modulated filter coefficients to produce a unique phase distortion. The primary signal processing loop is written with Faust, and compiled with JUCE for the various build targets.

  • Gorgon - Distortion. Glitchmachines Subvert is the evolution of Gorgon. The download contains the unlocked installers. OS X and Windows.

  • https://www.meldaproduction.com/MBitFun - a serious tool for extreme distortion lovers. It converts the audio into limited fixed-point precision form, from a 1 single bit up to 16 bits per sample, and lets you access each bit, applying several operations.

  • FuzzPlus3 - vintage fuzz pedal model, plus a new filter, self-feedback, and a modern procedural user interface. Windows/Mac

  • thrummaschine - a 3-band distortion effect with independant, LFO-driven filters. Make your bass, mid and high frequencies oscillate at different speeds, shapes and pan, with whatever flavour and level of distortion you dial in for each band

  • Imperfection - an effect plugin to put some lofi back into your pristine 64bit audio. Who wants hi fidelity reproduction when you can reduce the quality, take out some of that bottom end, add a smear of saturation and bring your noise floor back up. Hmmm, perfectly imperfect

  • bent - a circuit-bent resynthesis effect. It will recreate the incoming audio into an approximation of itself using a waveform-morphing audio oscillator. Depending on the volume and pitch of the audio, it will gate, stutter and morph the output in sync with your host tempo

  • freq show - screws around with your audio and ouputs an unholy version of whatever you fed it

  • GClip - Free VST wave-shaping signal clipper. Clip peaks off audio with abrupt or smooth wave-shaping. Graph and waveform displays assist in setting the clip level according to the source material. Oversampling can be enabled to reduce aliasing.

  • DvnBitCrusher - It's a bit crusher effect that offers sample rate reduction down to 200Hz, bit depth reduction, distortion unit and low pass filter, dry/wet mix and gain stages at the input and output.





Paul's Extreme Sound Stretch

2xautoconvolution / 2xphases

  • https://github.com/paulnasca/2xphases - This repository contains two programs: "2xautoconvolution" which uses autoconvolution to process audio and an older program "2xphases" which uses long-term FFT audio processing.


  • https://github.com/spluta/TimeStretch - Implements a phase randomized Real FFT time stretch algorithm, the NessStretch, which splits the original sound file into 9 discrete frequency bands, and uses a decreasing frame size to correspond to increasing frequency. Starting with a largest frame of 65536, the algorithm will use the following frequency band/frame size breakdown (assuming 44100 Hz input):0-86 Hz : 65536 frames, 86-172 : 32768, 172-344 : 16384, 344-689 : 8192, 689-1378 : 4096, 1378-2756 : 2048, 2756-5512 : 1024, 5512-11025 : 512, 11025-22050 : 256.The NessStretch is a refinement of Paul Nasca's excellent PaulStretch algorithm. PaulStretch uses a single frame size throughout the entire frequency range. The NessStretch's layered analysis bands are a better match for human frequency perception, and do a better job of resolving shorter, noisier high-frequency sounds (sibilance, snares, etc.).


  • tcStretch - a Windows VST 2.4 plug-in for time stretching, pitch shifting, and blurring. Time stretch can be up to 1 million times slower. Pitch shift is plus or minus one octave. Blurring blends nearby spectral material to make the output less static. Playback is sensitive to transients in the source material. Playback rate and blur amount are automatically adjusted according to the transient contour of the material being stretched. Playing transients at a faster rate than non-transients tends to make the output sound less obviously stretched. Playing transients more slowly than non-transients emphasizes the stretchiness [good when playing in reverse mode with highly transient material]. Adding blur brings in some subtle (or not so subtle) randomness which helps to keep the output less static.

Rubber Band

  • Rubber Band Library is a high quality software library for audio time-stretching and pitch-shifting. It permits you to change the tempo and pitch of an audio stream or recording dynamically and independently of one another. Rubber Band Library is intended for use by developers creating their own application programs rather than directly by end users, although it does also include a simple (free) command-line utility program that you can use for fixed adjustments to the speed and pitch of existing audio files.

Play it Slowly

  • Play it Slowly - software to play back audio files at a different speed or pitch. It does also allow you to loop over a certain part of a file. It's intended to help you learn or transcribe songs. It can also play videos thanks to gstreamer. Play it slowly is intended to be used on a GNU/Linux system like Ubuntu.


  • StretchPlayer is an audio file player that allows you to change the speed of the song without changing the pitch. It will also allow you to transpose the song to another key (while also changing the speed). This is a very powerful tool for musicians who are learning to play a pre-recorded song.


  • PitchTempoPlayer (PTPlayer) is an audio player for Linux that allows to change pitch and speed (tempo) of the sound independently of each other. Fine tuning (less than half tone) is also possible, as well recording, exporting the modified audio file and managing a playlist.



  • SoundTouch - an open-source audio processing library for changing the Tempo, Pitch and Playback Rates of audio streams or audio files. The library additionally supports estimating stable beats-per-minute rates for audio tracks. The SoundTouch library is intended for application developers writing sound processing tools that require tempo/pitch control functionality, or just for playing around with the sound effects.

The SoundTouch library source kit includes also an example utility SoundStretch for processing .wav audio files from command-line interface.

Signalsmith Stretch



  • https://github.com/waywardgeek/sonic - a simple algorithm for speeding up or slowing down speech. However,it's optimized for speed ups of over 2X, unlike previous algorithms for changing speech rate. The Sonic library is a very simple ANSI C library that is designed to easily be integrated into streaming voice applications, like TTS back ends.


  • https://github.com/usdivad/Groovinator - an audio plugin, available in VST and AU format, that performs real-time playhead-aware rhythm modification. Currently supported modification modes are: Classic Stretch, Sample 'n' Shift, and Step Repeater. Developed using the JUCE framework in addition to Olli Parviainen's SoundTouch library and Sergio Castro's implementation of Bjorklund's algorithm.


  • https://github.com/usdivad/Melodrumatic - an audio plugin that lets you "pitch-shift" via delay (i.e. the Doppler effect) to turn unpitched audio into melodies. Controllable via MIDI or mouse :)



  • https://github.com/claytonotey/libsbsms - a library for high quality time and pitch scale modification. It uses octave subband sinusoidal modeling.The API is found in sbsms.h. sbsms_create is called and supplied a callback which feeds sbsms_process samples. The pitch_create and pitch_process functions are called only if pitch shifting is required. It simply sticks a resampler on the end of the FIFO.




  • https://crates.io/crates/ness_stretch - Implements a phase randomized Real FFT time stretch algorithm, the NessStretch, which splits the original sound file into 1-10 discrete frequency bands, and uses a decreasing frame size to correspond to increasing frequency, thus doing a excellent job of resolving shorter, noisier high-frequency sounds and creating rich, lush low frequency pads.


Pitch shifting


  • Autotalent began as the result of a week of recreational signal processing in May 2009. It's a real-time pitch correction plugin. You specify the notes that a singer is allowed to hit, and Autotalent makes sure that they do. You can also use Autotalent for more exotic effects, like the Cher / T-Pain effect, making your voice sound like a chiptune, adding artificial vibrato, or messing with your formants. Autotalent can also be used as a harmonizer that knows how to sing in the scale with you. Or, you can use Autotalent to change the scale of a melody between major and minor or to change the musical mode. LADSPA.


  • talentledhack - an LV2 port of Tom Baran's Autotalent, with added features and improved performance.

Zita AT1

  • AT1 - an 'autotuner', normally used to correct the pitch of a voice singing (slightly) out of tune. Compared to 'Autotalent' it provides an improved pitch estimation algorithm, and much cleaner resampling. AT1 does not include formant correction, so it should be used to correct small errors only and not to really transpose a song. The 'expected' pitch can be controlled by Midi (via Jack only), or be a fixed set of notes. AT1 can probably be used on some instruments as well, but is primarily designed to cover the vocal range. It's also usable as a quick and dirty guitar tuner.

Zita Retuner


  • x42-autotune - aka fat1.lv2, is an auto-tuner based on Fons Adriaensen's zita-at1. The main differences to zita-at1 are that the LV2 plugin version reports its latency to the host, saves the state with the session and the MIDI input has sidechain semantics.



  • TAL-Vocoder is a vintage vocoder emulation with 11 bands that emulates the sound of vocoders from the early 80’s. It includes analog modeled components in combination with digital algorithms such as the SFFT (Short-Time Fast Fourier Transform). This vocoder does not make a direct convolution of the carrier and modulation signal as other digital vocoders maybe do. It includes an envelope follower for every of the eleven bands. This vocoder is optimized for voice processing and includes some algorithms for consonants to make the voice more intelligible. The carrier signal is a VCO (Voltage Controlled Oscillator) with a Pulse, Saw, Noise and SubOsc. But it’s also possible to use the left stereo input as carrier. This way every sound source can be used as carrier signal.


  • vocoder (JACK standalone). It's a complete rewrite of the old vocoder, now done in C++ using FLTK.


  • Vocoder by borsboom - a free software channel vocoder, which imposes vocal effects on a waveform. It can be used to make your voice sound "robot-like", to create a singing synthesizer, to disguise your voice, and many other fun things. This effect was made popular by artists such as Kraftwerk and Laurie Anderson, and Daft Punk (using other hardware/software). This particular software works on pre-recorded files, rather than being a live effect.


  • VocProc - an LV2 plugin for pitch shifting (with or without formant correction), vocoding, automatic pitch correction and harmonizing of singing voice.




  • https://github.com/magnetophon/VoiceOfFaust - Turns any monophonic sound into a synthesizer, preserving the pitch and spectral dynamics of the input. The name was chosen because I use it mostly to turn my voice into a singing robot, and it's made in Faust.


  • PitchBox - a small software application for transforming your voice or music in real-time. PitchBox was developed for shows and exhibitions, and was hence tuned for that purpose. As such, most of the efforts have been done in proposing ear-catching and attractive audio effects, and on reducing the latency as much as possible. On the other hand, fewer efforts have been done in making it easy to install and setup, or in the amount of features it provides. If you want to use the audio effects of PitchBox within an audio editor, most of them are available as LADSPA and VST plugins.



  • https://github.com/lmnt-com/wavegrad - a fast, high-quality neural vocoder designed by the folks at Google Brain. The architecture is described in WaveGrad: Estimating Gradients for Waveform Generation. In short, this model takes a log-scaled Mel spectrogram and converts it to a waveform via iterative refinement.


  • https://github.com/lmnt-com/diffwave - fast, high-quality neural vocoder and waveform synthesizer. It starts with Gaussian noise and converts it into speech via iterative refinement. The speech can be controlled by providing a conditioning signal (e.g. log-scaled Mel spectrogram). The model and architecture details are described in DiffWave: A Versatile Diffusion Model for Audio Synthesis.


  • https://github.com/mindslab-ai/univnet - Unofficial PyTorch Implementation of UnivNet Vocoder. A Neural Vocoder with Multi-Resolution Spectrogram Discriminators for High-Fidelity Waveform Generation.


Pitch Controllable DDSP Vocoders





  • dasp - Differentiable audio signal processors in PyTorch Includes reverberation, distortion, dynamic range processing, equalization, stereo processing and more to come. Enable applications in virtual analog modeling, blind parameter estimation, automated DSP, and style transfer. Supports batching and operation on both CPU and GPU accelerators for fast training and reduced bottlenecks. Open source and free to use for academic and commercial applications under the permissive Apache 2.0 license.

Windows VST

  • Vintage Vocoder - real-time audio effect - VST and DXI plug-in for PC/MAC. Originally a commercial product published by Sonicism Digital Audio Solutions in 2002. This software was used for the robot voices and sound effects in the computer game Freelancer.

  • GVST - GSnap - Free VST pitch-correction. Use with subtle settings to nudge off-key vocals back in line. Extreme settings can create sounds like the famous T-Pain or Cher effect. MIDI control mode allows you to fit a recording to a new melody.

  • La Voz Cantante - a 512 channel vocoder. The modulator input - usually a sung or simpjy spoken voice - is analyzed with respect to its spectral content, which is then applied to the other sound source. The latter may be any externally supplied signal ranging from pink noise, synth pads, guitar or even drums. Alternatively, there is an internal, MIDI driven synth which is optimized for best speech reproduction fidelity. You can blend the high frequencies with noise for more natural sounding plosives and fricatives. Threre is also a noise gate, a compressor and a stereo reverb on board.

Phase vocoder

  • https://en.wikipedia.org/wiki/Phase_vocoder - a type of vocoder which can scale both the frequency and time domains of audio signals by using phase information. The computer algorithm allows frequency-domain modifications to a digital sound file (typically time expansion/compression and pitch shifting). At the heart of the phase vocoder is the short-time Fourier transform (STFT), typically coded using fast Fourier transforms. The STFT converts a time domain representation of sound into a time-frequency representation (the "analysis" phase), allowing modifications to the amplitudes or phases of specific frequency components of the sound, before resynthesis of the time-frequency domain representation into the time domain by the inverse STFT. The time evolution of the resynthesized sound can be changed by means of modifying the time position of the STFT frames prior to the resynthesis operation allowing for time-scale modification of the original sound file.


  • pvc - PVC is a collection of phase vocoder signal processing routines and accompanying shell scripts for use in the transformation and manipulation of sounds. It is written in C and designed to be used in a UNIX environment.



  • pv in WaoN project is yet another phase vocoder implementation for my understanding of the process behind WaoN and others. Here is what you can do: time streching/shrinking without pitch changing (by rate option) and pitch shifting without time streching (by pitch option)


  • pvoc is a collection of LADSPA units and a command line tool for time compression/expansion of sound data making use of the phase-vocoding technique.


  • https://github.com/knector01/pvc - a phase vocoder implementation written in Python. It can be used to time-stretch audio files. This program also implements independent pitch-shifting of audio in the frequency domain, with optional formant correction.Dependencies: numpy, scipy, SoundFileTo use the JACK application you must also install tkinter and JACK-Client.


  • https://github.com/stekyne/PhaseVocoder - A C++ based phase vocoder example that allows pitch and timescale modifications of audio files. Built using the Juce framework.The plan is modernize the code base with current best practices, both C++ and Juce wise. Ideally I'd like to extend it to use higher quality phase locking algorithms for improved sound quality. Also to include more effects than just pitch and timescale modifications. Basically a spectral effect tool would be the end goal, possibly creating a plugin out of the result.This was initially written back in 2010 for my thesis project to showcase the PV algorithm and it's uses.


  • https://github.com/Stenzel/SimpleVocoder - focus of this vocoder is simplicity, the code is not meant to be production ready or good style. It uses a leaky autocorrelation for pitch synchronous synthesis. Although mathematically wrong, it produces quite intelligible results, perhaps better than many classic channel vocoders.

Pulse model

  • https://github.com/gillesdegottex/pulsemodel - Pulse model analysis and synthesis. It is basically the vocoder described in: G. Degottex, P. Lanchantin and M. Gales, "A Log Domain Pulse Model for Parametric Speech Synthesis", IEEE Transactions on Audio, Speech, and Language Processing, 26(1):57-70, 2018.

Frequency Shifter

  • Frequency Shifter - a VST™2.4 software effect for Microsoft Windows® written in native C++ code. Frequency shifting up to ±5000 Hz. Optional LFO with five waveforms. Four frequency ranges, three mix modes.


  • https://www.meldaproduction.com/MFreqShifter - an extremely versatile frequency shifter. Unlike pitch-shifters it doesn't keep harmonic relationships and can provide everything from mild stereo expansion to complete sonic destruction.


  • https://github.com/jurihock/stftPitchShift - a Short-Time Fourier Transform (STFT) based pitch and timbre shifting algorithm implementation, originally inspired by the Stephan M. Bernsee's smbPitchShift.cpp. This repository features two analogical algorithm implementations, C++ and Python. Both contain several function blocks of the same name (but different file extension, of course). In addition to the basic pitch shifting algorithm, it also features spectral multi pitch shifting and cepstral formant preservation extensions. Both sources contain a ready-to-use command line tool as well as a library for custom needs. See more details in the build section. Feel free to check out some demos as well.





Windows / VST

MIDI Choir

  • MIDI Choir - will take a single-pitched audio source and transpose it in real time according to the supplied MIDI notes. My main motivation to create MIDI Choir was to be able to sing harmonies live, however the product may also be used for studio work

Noise reduction




  • https://github.com/josh-richardson/cadmus - a graphical application which allows you to remove background noise from audio in real-time in any communication app. Cadmus adds a notification icon to your shell which allows you to easily select a microphone as a source, and subsequently creates a PulseAudio output which removes all recorded background noise (typing, ambient noise, etc).



  • https://github.com/lawl/NoiseTorch - an easy to use open source application for Linux with PulseAudio. It creates a virtual microphone that suppresses noise, in any application. Use whichever conferencing or VOIP application you like and simply select the NoiseTorch Virtual Microphone as input to torch the sound of your mechanical keyboard, computer fans, trains and the likes.

facebook denoiser

  • https://github.com/facebookresearch/denoiser - Real Time Speech Enhancement in the Waveform Domain (Interspeech 2020)We provide a PyTorch implementation of the paper Real Time Speech Enhancement in the Waveform Domain. In which, we present a causal speech enhancement model working on the raw waveform that runs in real-time on a laptop CPU.


  • https://github.com/rrbluke/CNBF - contains python/tensorflow code to reproduce the experiments presented in our paper Deep Complex-valued Neural Beamformers.


Echo reduction



Response and convolution

kinda messy section

  • https://en.wikipedia.org/wiki/Frequency_response - the quantitative measure of the output spectrum of a system or device in response to a stimulus, and is used to characterize the dynamics of the system. It is a measure of magnitude and phase of the output as a function of frequency, in comparison to the input. In simplest terms, if a sine wave is injected into a system at a given frequency, a linear system will respond at that same frequency with a certain magnitude and a certain phase angle relative to the input. Also for a linear system, doubling the amplitude of the input will double the amplitude of the output. In addition, if the system is time-invariant (so LTI), then the frequency response also will not vary with time. Thus for LTI systems, the frequency response can be seen as applying the system's transfer function to a purely imaginary number argument representing the frequency of the sinusoidal excitation. Two applications of frequency response analysis are related but have different objectives.

For an audio system, the objective may be to reproduce the input signal with no distortion. That would require a uniform (flat) magnitude of response up to the bandwidth limitation of the system, with the signal delayed by precisely the same amount of time at all frequencies. That amount of time could be seconds, or weeks or months in the case of recorded media. In contrast, for a feedback apparatus used to control a dynamic system, the objective is to give the closed-loop system improved response as compared to the uncompensated system. The feedback generally needs to respond to system dynamics within a very small number of cycles of oscillation (usually less than one full cycle), and with a definite phase angle relative to the commanded control input. For feedback of sufficient amplification, getting the phase angle wrong can lead to instability for an open-loop stable system, or failure to stabilize a system that is open-loop unstable. Digital filters may be used for both audio systems and feedback control systems, but since the objectives are different, generally the phase characteristics of the filters will be significantly different for the two applications.

  • https://en.wikipedia.org/wiki/Transient_response - the response of a system to a change from an equilibrium or a steady state. The transient response is not necessarily tied to abrupt events but to any event that affects the equilibrium of the system. The impulse response and step response are transient responses to a specific input (an impulse and a step, respectively).
  • https://en.wikipedia.org/wiki/Impulse_response - impulse response function (IRF), of a dynamic system is its output when presented with a brief input signal, called an impulse. More generally, an impulse response refers to the reaction of any dynamic system in response to some external change. In both cases, the impulse response describes the reaction of the system as a function of time (or possibly as a function of some other independent variable that parameterizes the dynamic behavior of the system). In all these cases, the dynamic system and its impulse response may be actual physical objects, or may be mathematical systems of equations describing such objects.

  • Aliki - an integrated system for Impulse Response measurements, using the logaritmic sweep method developed by Prof. Angelo Farina.

  • DRC - a program used to generate correction filters for acoustic compensation of HiFi and audio systems in general, including listening room compensation. DRC generates just the FIR correction filters, which can be used with a real time or offline convolver to provide real time or offline correction. DRC doesn't provide convolution features, and provides only some simplified, although really accurate, measuring tools.

  • REW - free room acoustics analysis software for measuring and analysing room and loudspeaker responses. The audio analysis features of REW help you optimise the acoustics of your listening room, studio or home theater and find the best locations for your speakers, subwoofers and listening position. It includes tools for generating audio test signals; measuring SPL and impedance; measuring frequency and impulse responses; measuring distortion; generating phase, group delay and spectral decay plots, waterfalls, spectrograms and energy-time curves; generating real time analyser (RTA) plots; calculating reverberation times; calculating Thiele-Small parameters; determining the frequencies and decay times of modal resonances; displaying equaliser responses and automatically adjusting the settings of parametric equalisers to counter the effects of room modes and adjust responses to match a target curve.

  • https://en.wikipedia.org/wiki/Convolution - In acoustics, reverberation is the convolution of the original sound with echoes from objects surrounding the sound source. In digital signal processing, convolution is used to map the impulse response of a real room on a digital audio signal.

In electronic music convolution is the imposition of a spectral or rhythmic structure on a sound. Often this envelope or structure is taken from another sound. The convolution of two signals is the filtering of one through the other.

  • deconvolv - convolution and deconvolution of WAV files. Supported signal processing functions: correlation, convolution, de-convolution, convolution with Hilbert transformation, de-convolution with Hilbert transformation.

  • The HISSTools Impulse Response Toolbox: Convolution for the Masses - this paper introduces the HISSTools project, and its first release, the HISSTools Impulse Response Toolbox (HIRT); a set of tools for solving problems relating to convolution and impulse responses (IRs). Primarily, the aims and de- sign criteria for the HISSTools project are discussed. The elements of the HIRT are then outlined, along with motivating factors for its development, underlying technologies, design considerations and potential applications.
    • https://github.com/HISSTools/HISSTools_Impulse_Response_Toolbox - HISSTools first release is a set of tools for working with convolution and impulse responses in Max. This set of object addresses various tasks, including measuring impulse responses, spectral display from realtime data/ buffers, and buffer-based convolution, deconvolution and inversion.

  • ExpoChirpToolbox - an impulse response (IR) measurement tool chain in Pure Data, available for Windows, OSX and Linux. It implements the Exponential Sine Sweep method which has been so succesfully advocated by Angelo Farina. The toolbox is in developement, and has functionality for the generation of test signals, recording test responses, IR editing and basic IR analysis. The edited IR can be applied as a convolution filter in the toolbox.

  • BruteFIR - a software convolution engine, a program for applying long FIR filters to multi-channel digital audio, either offline or in realtime. Its basic operation is specified through a configuration file, and filters, attenuation and delay can be changed in runtime through a simple command line interface. The FIR filter algorithm used is an optimised frequency domain algorithm, partly implemented in hand-coded assembler, thus throughput is extremely high. In realtime, a standard computer can typically run more than 10 channels with more than 60000 filter taps each.

  • IR - a no-latency/low-latency, realtime, high performance signal convolver especially for creating reverb effects. Supports impulse responses with 1, 2 or 4 channels, in any soundfile format supported by libsndfile. [4]

  • x42 Zero Config Convolver - A preset-only IR Reverb/Convolver. No parameters, no file-loading, just presets. This convolution effect allows processing with variable buffersize, up to a the nominal block-size. It is intended for use with Ardour 6, a LV2 host that provides realtime-priority information to this plugin for efficient background processing.

  • https://github.com/johnflynnjohnflynn/jLibrary - FFT, convolution and general Pure Data library. jTabConv: Convolve two tables by multiplication in the frequency domain using FFT. jTabLookup; Fill a table with values processed by cross-connected object(s).

  • https://github.com/LCAV/pyroomacoustics - a software package aimed at the rapid development and testing of audio array processing algorithms. "Suppose, for example, you wanted to produce a radio crime drama, and it so happens that, according to the scriptwriter, the story line absolutely must culminate in a satanic mass that quickly degenerates into a violent shootout, all taking place right around the altar of the highly reverberant acoustic environment of Oxford's Christ Church cathedral. To ensure that it sounds authentic, you asked the Dean of Christ Church for permission to record the final scene inside the cathedral, but somehow he fails to be convinced of the artistic merit of your production, and declines to give you permission. But recorded in a conventional studio, the scene sounds flat. So what do you do?"


  • SpecMatch - can be used to adapt the sound produced by a Guitarix setting to another recorded sound. It can also be used independently of Guitarix (cf. specmatch --help). Then you will need another convolver like the LV2 Convolution Reverb to use the produced filter. You can also use just the Python modules (e.g. from specmatch import SmoothedIR).

  • Jconvolver - a Convolution Engine for JACK, based on FFT convolution and using non-uniform partition sizes: small ones at the start of the IR and building up to the most efficient size further on. It can perform zero-delay processing with moderate CPU load. Jconvolver uses the convolution engine designed for Aella, a convolution application for reverberation processing (to be announced later). This distributes the calculation over up to five threads, one for each partition size, running at priorities just below the the one of JACK's processing thread. This engine is a separate library that will be documented as soon as I can find the time.
  • https://github.com/brugal/gtrfx/tree/master/jconv - a Convolution Engine for JACK using FFT-based partitioned convolution with multiple partition sizes. It's a command line version of what will be the core of the Aella reverb processor, but without the special reverb feautures, preset management, reverb envelope editing etc. that Aella will have.

  • HybridReverb2 - a convolution-based reverberation effect which combines the superior sound quality of a convolution reverb with the tuning capability of a feedback delay network. The sound quality of a convolution reverb depends on the quality of the used room impulse responses. HybridReverb2 comes with a set of room impulse responses which were synthesized with tinyAVE, an auralization software which was developed at the Institute of Communication Acoustics, Ruhr-Universität Bochum (Borß and Martin, 2009; Borß, 2009a). These room impulse responses are designed for a speaker setup with two front and two rear speakers (Borß, 2009b). For a full surround sound effect, you will need two plugins, one plugin which uses a "front" preset for the front channels and a second plugin which uses the corresponding "rear" preset for the rear channels.

  • https://github.com/hzeller/folve - a FUSE filesystem that convolves audio files on-the-fly including gapless support. Need to have precision-filtered audio adapted for your speakers and room (or just for effects), but your audio system doesn't have plug-ins for Finite-Impulse-Response filters and just can read files ? Folve is for you!

  • https://github.com/edward-ly/GeneticReverb - A VST 2 audio effect plugin written in MATLAB that uses a genetic algorithm to generate a random impulse response describing the reverberation of an artificial room, and uses the impulse response to apply convolution reverb to an audio signal in real-time. A MATLAB script version (in scripts/main.m) is also available, which accepts a WAV audio file as input instead. The input is combined with the impulse response via convolution, applying the reverb effect to the pre-recorded audio.

  • Voxengo Deconvolver - offers a very convenient environment in which to deconvolve large sets of recorded files for use with convolution plug-ins that support only a small subset of available bit-depths. Windows $.

  • keFIR - provides music producers and sound engineers with a zero-latency FIR filter effect designed to help them enhance their tracks and generate astonishing sounds. Windows VST.

  • https://github.com/GAMMA-UMD/TS-RIR - official implementation of TS-RIRGAN. We started our implementation from WaveGAN. TS-RIRGAN is a one-dimensional CycleGAN that takes synthetic RIRs as raw waveform audio and translates it into real RIRs. Our network architecture is shown below.


  • https://github.com/ehabets/RIR-Generator - The image method, proposed by Allen and Berkley in 1979, is probably one of the most frequently used methods in the acoustic signal processing community to create synthetic room impulse responses. A mex-function, which can be used in MATLAB, was developed to generate multi-channel room impulse responses using the image method. This function enables the user to control the reflection order, room dimension, and microphone directivity.This repository includes a tutorial, MATLAB examples, and the source code of the mex-function.


  • https://github.com/anton-jeran/FAST-RIR -the official implementation of our neural-network-based fast diffuse room impulse response generator (FAST-RIR) for generating room impulse responses (RIRs) for a given acoustic environment.


  • Broom - Use the power of the Web Audio API to capture Room Impulse Responses... in the browser! This web app uses a sine sweep to excite the room. The microphone captures the response of the room you want to measure. This Sweep Response is then processed to an Impulse Response, ready for you to download.

Impulse responses / IR

  • https://github.com/rirnet/rirnet - This thesis work discuss the possibility to extract the impulse response function needed to construct a reverberant speech signal given that there exist an anechoic speech signal obscured by system influence, using deep learning. We propose two frameworks for combining deep autoencoder neural networks (for learning impulse response function features using point cloud representations) with convolutional neural networks that extract latent representations of the impulse response functions, either consuming time-series directly or a MFCC representation of the signal. We also present a framework that approximate the impulse response function from MFCC input directly. An emphasis is put on how to represent and obtain simulated data. Finally, we present results that show that the impulse response function can be extracted from reverberant signals with some accuracy.

  • https://github.com/jonashaag/RealRIRs - A collection of loaders for real (recorded) impulse response databases, because apparently people cannot to stick to a single way of storing audio data.

  • https://github.com/sh01k/MeshRIR - a dataset of acoustic room impulse responses (RIRs) at finely meshed grid points. Two subdatasets are currently available: one consists of IRs in a 3D cuboidal region from a single source, and the other consists of IRs in a 2D square region from an array of 32 sources. This dataset is suitable for evaluating sound field analysis and synthesis methods.

Cabinet emulation



  • https://github.com/olegkapitonov/spiceAmp - Non-realtime high realistic software guitar processor. Works with *.wav files as input and output. It uses ngspice for electric circuit simulation and FFT convolver with Impulse Response *.wav file for cabinet simulation.

Neural Cab

  • https://github.com/Thiagohgl/neural-cab-audio-plugin - a FIR guitar cabinet simulator which generates its transfer functions by means of a Variational Auto-Encoder (VAE) trained with an additional adversarial loss and a very simple Boundary Element Method (BEM) simulation to consider the microphone position. It is coded using JUCE and Onnx for the Machine Learning part. You can see a video with the usage of the plugin below:




  • https://github.com/omnp/mnamp - LV2, Amplifier plugin based on gathered principles and rules of thumb more than any kind of direct emulation of anything

Machine learning


  • https://github.com/keyth72/SmartGuitarAmp - uses a WaveNet model to recreate the sound of real world hardware. The current version models a small tube amp, with the ability to add more options in the future. There is a clean/lead channel, which is equivalent to the amp's clean and full drive settings. Gain and EQ knobs were added to modulate the modeled sound.


  • https://github.com/keyth72/SmartGuitarPedal - uses a WaveNet model to recreate the sound of real world hardware, such as a TS9 Tubescreamer or Blues Jr amp. Drive and Level are used for simple ways to control the sound. The WaveNet model is effective at emulating distortion style effects or tube amplifiers.

HEAR 2021 Baseline

  • https://github.com/neuralaudio/hear-baseline -A simple DSP-based audio embedding consisting of a Mel-frequency spectrogram followed by a random projection. Serves as the naive baseline model for the HEAR 2021 and implements the common API required by the competition evaluation.


Neural Amp Modeler




  • https://github.com/magenta/ddsp - a library of differentiable versions of common DSP functions (such as synthesizers, waveshapers, and filters). This allows these interpretable elements to be used as part of an deep learning model, especially as the output layers for audio generation.

Aida DSP LV2

  • https://github.com/AidaDSP/aidadsp-lv2 - Aida DSP lv2 plugin bundle, intended to be used with MOD Audio's products and derivatives. rt-neural-generic.lv2, leverages RTNeural to model pedals or amps. Play realistic Amps or Pedals captured with cutting-edge ML technology. Full featured 5-band EQ with adjustable Q, frequencies and pre/post switch. Input and Output Volume Controls


  • https://github.com/audacitorch/HARP - an ARA plug-in that allows for hosted, asynchronous, remote processing with deep learning models by routing audio from a digital audio workstation (DAW, through Gradio endpoints. Because Gradio apps can be hosted locally or in the cloud (e.g., HuggingFace Spaces), HARP lets DAW users access large state-of-the-art models with GPU compute from the cloud without breaking their within-DAW workflow.
  • https://github.com/audacitorch/pyharp - a Python library designed to embed Gradio apps for audio processing in a Digital Audio Workstation (DAW, through the HARP plugin. PyHARP creates hosted, asynchronous, remote processing (HARP) endpoints in DAWs, facilitating the integration of deep learning audio models into DAW environments through Gradio. HARP is designed for processing audio in a DAW with deep learning models that are too large to run in real-time and/or on a user's CPU, or otherwise require a large offline context for processing. There are many examples of these kinds of models, like MusicGen or VampNet.

Other solutions for realtime processing with neural networks in DAWs (e.g. NeuTone), require highly optimized models that can run on a local CPU and rely on the model code to be traced/scripted into a JIT representation, which can be a challenge for the model developer. PyHARP, on the other hand, relies on Gradio, which allows for the use of any Python function as a processing endpoint. PyHARP doesn't require for your deep learning code to be optimized or JIT compiled. PyHARP lets you process audio in a DAW using your deep learning library of choice. Tensorflow? PyTorch? Jax? Librosa? You pick. It doesn't even have to be deep learning code! Any arbitrary Python function can be used as a HARP processing endpoint, as long as it can be wrapped in a Gradio interface.


  • -2309.07314- AudioSR: Versatile Audio Super-resolution at Scale - Audio super-resolution is a fundamental task that predicts high-frequency components for low-resolution audio, enhancing audio quality in digital applications. Previous methods have limitations such as the limited scope of audio types (e.g., music, speech) and specific bandwidth settings they can handle (e.g., 4kHz to 8kHz). In this paper, we introduce a diffusion-based generative model, AudioSR, that is capable of performing robust audio super-resolution on versatile audio types, including sound effects, music, and speech. Specifically, AudioSR can upsample any input audio signal within the bandwidth range of 2kHz to 16kHz to a high-resolution audio signal at 24kHz bandwidth with a sampling rate of 48kHz. Extensive objective evaluation on various audio super-resolution benchmarks demonstrates the strong result achieved by the proposed model. In addition, our subjective evaluation shows that AudioSR can acts as a plug-and-play module to enhance the generation quality of a wide range of audio generative models, including AudioLDM, Fastspeech2, and MusicGen.




Amplio 2.0

  • Amplio 2.0 - (VST Effect) is a way to enhance boring and dull sound. It's three band Equalizer with adjustable bands and additional multiband effect modules. Plugin was originally designed to enhance drum patterns, but version 2 is powerfull enough to be used on any type of sound.



  • Fracture - features a buffer effect, a multimode filter, three LFOs and a delay. The order of the effects in the processing chain can also be reconfigured. This plugin is geared toward adding glitchy articulations and abstract textures to your projects. Use it on anything from drums and percussion to synth lines and sound effects. Fracture’s intuitive interface and diverse features make it simple to give your projects a unique technical edge.


  • ExEf - Extreme Effect, is an extremely powerful and flexible Real Time effect engine running on a PC under LINUX. It is designed to work with guitars, microphones and other instruments. It can run both in X Window System and command line. It supports both recording an post-processing. For easy start, try some presets! Requirepments include PC - Pentium MMX or above (Alpha may work as well), Full duplex soundcard.

Disto:Fx Free

  • Disto:Fx Free (Dirty Sound Destructor) - a multi-fx with distortion/dynamic shaper/saturation/filter/ring modulator/phaser/EQ and Input-Output control units. macOS AU/VST, Win VST 32-bit/64-bit.


  • MixMaxTrix by ArtV - Mixer Plugin VST3 LV2 - channel routing tool, multi-fx, crossover, multiband-fx, transient shaper, etc. disguised as a channel mixer. Bundles some own and some open source FX.This is a swiss-army knife FX to complement your existing ones, helping to keep your mixes tidy. It has a lot of tricks on its sleeve. Seriously, this is not a conventional plugin.



Windows VST

  • truc is a multi-effect VST plug-in, with 4 banks of effects controlled by the movement of 2 pucks. The top puck controls the level of each effect bank, the bottom puck modulates any 4 of 13 parameters within the effect banks. This allows continuous morphing of the sound by moving the pucks to vary the impact of each effect bank. As well as manually controlling each puck (either by mouse or midi controller) you can lock the pucks together and/or set them to move automatically, either randomly or by a configurable LFO, all in sync with your project tempo. Windows VST.
  • truc2 is a multi-effect plug-in with 4 different effect modules and two automated XY pads to modulate their levels and parameters. It is designed to add variation and movement, anywhere along the scale of subtle to overkill and is suitable for any material. The 4 effect modules are DIRT, GRAIN, RING and DELAY. The first XY pad modulates the volume/mix level of each module, whilst the second XY pad modulates any 4 of the 15 automatable parameters. Both XY pads can be moved manually and additionally automated with a variety of LFO shapes, speeds and depths. Windows VST.

  • SynthTrack - an effect plugin. Applied to audio tracks, this plug-in applies Filter ADSR and LFO effects to synths on flat chords or moving sound waves. With this effect you can create chopping effects to your favorite sounds. It's ideal for creating the typical rhythmic gated pad sounds. With the envelope controlled step sequencer it's even possible to turn your pad sound into a powerful arp-like sequence. This plugin synchronizes to the host sequencer / DAW tempo. Windows/Mac VST.

Rack based


  • guitarix is a virtual guitar amplifier for Linux running on Jack Audio Connection Kit. It is free as in speech and free as in beer. The available sourcecode allows to build it on other UNIX-like systems, too, namely for BSD and for MacOSX.

  • https://bitbucket.org/doughammond/guitarix-foot-remote/src/master - intergrates with Guitarix's JSONRPC remote control protocol, to enable bank, preset and unit switching via hardware buttons on Raspberry Pi. It is designed to be run under i3 tiling window manager, with Guitarix in the upper 85% of the screen space and this app in the lower 15% screen space, so that the app buttons line up with the hardware buttons on the unit.


  • Rakarrack is a richly featured multi-effects processor emulating a guitar effects pedalboard. Effects include compressor, expander, noise gate, graphic equalizer, parametric equalizer, exciter, shuffle, convolotron, valve, flanger, dual flange, chorus, musicaldelay, arpie, echo with reverse playback, musical delay, reverb, digital phaser, analogic phaser, synthfilter, varyband, ring, wah-wah, alien-wah, mutromojo, harmonizer, looper and four flexible distortion modules including sub-octave modulation and dirty octave up. Most of the effects engine is built from modules found in the excellent software synthesizer ZynAddSubFX. Presets and user interface are optimized for guitar, but Rakarrack processes signals in stereo while it does not apply internal band-limiting filtering, and thus is well suited to all musical instruments and vocals. Rakarrack is designed for Linux distributions with Jack Audio Connection Kit.


  • GNUitar is guitar effects software that allows you to use your PC as guitar processor. It includes the following effects: wah-wah, sustain, distortion, reverberator, echo, delay, tremolo, vibrato, and chorus/flanger.



  • CP-GFX is simply a Cross Platform Guitar Effect Processor. The aim of the project is to create an extensible and easy to use program which is easy to port to different platforms an operating systems. Currently in development are Linux x86 and Win32 builds.


  • RedFX - FX Processor (for guitar mainly) Effects: Noise Filter, Compressor, Wah, Distortion, Tremolo, Phaser, Flanger-Vibrato, Pitch Shifter, Delay, Reverb, EQ.


  • Ecamegapedal is real-time effects processor software with a graphical user interface for controlling the effect parameters. It is meant to be used as a virtual guitar-fx or studio effects box. In addition to real-time operation, it also supports reading from and writing to audio files. All audio object and effect plugin types provided by the Ecasound libraries are supported. This includes ALSA, JACK, OSS, aRts, over 20 file formats, over 30 effect types, LADSPA plugins, and multi-operator effect presets. The implementation is based on the Ecasound and Qt libraries.


  • https://github.com/brugal/gtrfx - Guitar effects program with plug-in in support allowing you to play guitar through your computer. Includes custom plug-ins for distortion, gain, recording, piped audio input, pitch tuner, etc. Supports alsa, jack, lasdpa audio plug-ins, and jconv convolution engine for guitar cabinet simulation.


  • rack - ncurses jack ladspa-rack


  • https://github.com/forart/ayemux - a program that patches LADSPA audio plugins using Jack to create a real time audio fx system similar to Guitar Rig. Patches of plugins can be saved as racks. Each rack can have multiple states. The states and racks can be switch via a midi foot controller.


  • ToneLib-GFX - a collection of guitar amps, speaker cabinets and effects implemented in software for use with your computer, Windows PC or MAC. The software comes as a Standalone Application and as plug-ins for use with the most popular DAW (digital audio workstation) recording software. The Tonelib-GFX also supports 3rd party impulse responses (IRs), offering even greater tonal flexibility.


  • https://github.com/andrepxx/go-dsp-guitar - implements a cross-platform multichannel multi-effects processor for electric guitars and other instruments, based upon concepts and algorithms originating from the field of circuit simulation. The software takes the signals from N audio input channels, processes them and provides N + 3 audio output channels. The user may, for example, connect the signal from individual instruments to separate input channels of his / her sound card / audio interface. The input signals are then taken and put through dedicated signal chains for processing. The software provides one dedicated signal chain for each input. The output from the last processing element in each chain is then sent to one of the output channels, providing one output channel for each of the input channels. The remaining three output channels include a dedicated metronome, which creates a monophonic click track, as well as a pair of "master output" channels providing a stereo mixdown of all processed signals, say for monitoring purposes.To manipulate the signal, the user may choose from a variety of highly customizable signal processing units, including the following.

Various collections



  • TAP-plugins is short for Tom's Audio Processing plugins. It is a bunch of LADSPA plugins for digital audio processing, intended for use in a professional DAW environment such as Ardour. These plugins should compile and run on any recent (that is, not seriously outdated) GNU/Linux system. They don't require any special libraries besides the standard GNU C and math libraries, which are expected to be provided on the machine used for compiling.


  • CAPS is a collection of audio plugins comprising basic virtual guitar amplification and a small range of classic effects, signal processors and generators of mostly elementary and occasionally exotic nature. LADSPA.



  • LSP (Linux Studio Plugins) is a collection of open-source plugins currently compatible with LADSPA and LV2 formats. Phase Detector, Delay Compensator Mono, Delay Compensator Stereo, Delay Compensator X2 Stereo.


  • Infamous Plugins is a collection of open-source LV2 plugins. It hopefully helps fill some holes, supplying non-existing plugins for linux audio. There is little interest in creating ANOTHER compressor, or ANOTHER EQ when myriad other excellent lv2 versions of such already exist. At least until I become interested in making one of those things and feel I can do something different...


  • ArtyFX - a plugin bundle of artistic real-time audio effects. The aim of this plugin collection is to allow the designing of your sound just as you desired using a fast, efficient workflow.


  • Calf Studio Gear - available exclusively for LINUX-based operating systems and runs as a stand-alone effect rack connectable through Jack sound server or as plug-ins in every audio host that is able to fire up LV2 compilant devices, e.g. the highly recommended Ardour Audio Workstation. Play your SF2 sample banks, create filthy organs, fatten your sounds with phasers, delays, reverbs and other FX, process your recordings with gates, compressors, deesser and finally master your stuff with multiband dynamics - for free!


  • mda-vst - Windows VST, including Bandisto - Multi-band distortion. BeatBox - Drum replacer, Combo - Amp & speaker simulator, De-ess - High frequency dynamics processor, Degrade - Sample quality reduction, Delay - Simple stereo delay with feedback tone control, Detune - Simple up/down pitch shifting thickener, Dither - Range of dither types including noise shaping, DubDelay - Delay with feedback saturation and time/pitch modulation, Dynamics - Compressor / Limiter / Gate, Envelope - Envelope follower / VCA, Image - Stereo image adjustment and M-S matrix, Leslie - Rotary speaker simulator, Limiter - Opto-electronic style, limiter, Loudness - Equal loudness contours for bass EQ and mix correction , Multiband - Multi-band compressor with M-S processing modes, Overdrive - Soft distortion, Re-Psycho! - Drum loop pitch changer, RezFilter - Resonant filter with LFO and envelope follower, Round Panner - 3D panner, Shepard - Continuously rising/falling tone generator, Splitter - Frequency / level crossover for setting up dynamic processing, Stereo Simulator - Haas delay and comb filtering, Sub-Bass Synthesizer - Several low frequency enhancement methods, Talkbox - High resolution vocoder, TestTone - Signal generator with pink and white noise, impulses and sweeps, Thru-Zero Flanger - Classic tape-flanging simulation, Tracker - Pitch tracking oscillator, or pitch tracking EQ, Vocoder - Switchable 8 or 16 band vocoder, VocInput - Pitch tracking oscillator for generating vocoder carrier input

  • MDA-LV2 is an LV2 port of the MDA plugins by Paul Kellett. It contains 36 high-quality plugins for a variety of tasks. This is a more or less faithful port of both the effects and instrument plugins. The only functional difference in code is to support LV2-style toggle ports (> 0.0 is on, rather than 0.5). All the plugins have been tested, and thanks to several bug fixes this collection should be more reliable than the original.



  • SAFE Plug-ins (SAFE stands for Semantic Audio Feature Extraction) are a series of DAW plug-ins that allow the user to provide timbral descriptions of the audio they are processing. The plug-in then analyses the audio and saves the anonymous data to our server. This data is collected from all users and analysed to give a general synopsis of the types of sound that a given descriptor is used for. All this information can then be used to create a series of ‘semantic plug-in settings’. Users will be able to load plug-in settings by typing in descriptive words regarding the timbre of the sound being processed. The more people who upload descriptors to the server the more perceptually representative the downloaded plug-in settings will get.



  • x42-plugins - professional audio processing units available as LV2-plugins and JACK-applications

DISTRHO Mini-Series

DISTRHO OneKnob-Series




DPF Ports


  • Russolo Suite - a collection of LV2 plugins (and in the future, hopefully, VST) developed by Valerio Orlandini and named after the Futurist musician Luigi Russolo. For the moment, the attention is focused on the first part of this project: a sufficiently crazy synthesizer, called (what a surprise) Crazynth, and a do-it-all effect, called Omnifono.






  • https://github.com/pjotrompet/Freaked - So far this contains- A pre-delay which can be very long and blurry- A Reverb Tail, can be very long. Together these two makes a nice Reverb- A Distortion, based on Filter techniques. a awesome real-time granulator programmed by Mayank Sanganeria.


  • Computer Music Toolkit (CMT) is a collection of LADSPA plugins for use with software synthesis and recording packages on Linux. See the license before use.





  • ELE - the Excellent Low-latency Effects


  • ExEf - Extreme Effect, is an extremely powerful and flexible Real Time effect engine running on a PC under LINUX. It is designed to work with guitars, microphones and other instruments. It can run both in X Window System and command line.


  • https://github.com/laudrup/Creox4 - a real-time sound processor. You can plug your electric guitar or any other musical instrument directly to the PC's sound card and start experimenting with various sound effects. Creox has a nice user-friendly GUI, a preset support, a low-latency DSP engine and each effect parameter can be altered "on the fly".


  • Louderbox - a complete 8 band audio processor. Louderbox is intended to be used with software stereo and R[B]DS generators (but perfectly usable for other things (such as web "radio") using the jack audio connection kit under Linux (and possibly other systems but this is untested). LADSPA.


  • Mustajuuri - an audio signal processing application and toolkit. It is designed to meet wide range of needs. The first and foremost is real-time effects processing. Mustajuuri can process guitar, vocals or any instrument with ease. It is also useful if you have a virtual reality system with more than 10 loudspeakers and you wonder how to control them all :-)


  • BetabugsAudio (archived) - here you will find the plug-ins that we have available for download. These will have download buttons beneath them. Any GUIs without a download button are currently in development and not presently available. All other completed GUIs that are currently in need of a caring and affectionate programmer are available for viewing on the "job ads" page.




  • Airwindows Cheatsheet - This site is meant to be a quick reference for the constantly growing list of Airwindows Plugins.Info originally collected by Tiki Horea with help from the Airwindows Audiophiles Facebook Group.


  • https://github.com/juandagilc/Audio-Effects - Collection of audio effects plugins implemented from the explanations in the book "Audio Effects: Theory, Implementation and Application" by Joshua D. Reiss and Andrew P. McPherson.


  • ScorchCrafter Guitar FX DAW Plug-ins - A group of audio DAW plug-ins targeting Windows (VST), Mac (VST/AU), and Linux, mostly for Guitar Amplifier simulation, with the C-120 being the flagship product (which started off long ago as a closed-source VST). Open source, mostly under GPL.


Robot Audio Plugins



  • https://github.com/theotherjenkutler/sononaut - a set of eight open source VST/AU/LV2 audio processing plugins that mimics and re-imagines the way that sounds could be heard on other planets in the solar system by Jen Kutler and Stefana Fratila.


Windows VST / macOS AU


  • GVST - several free VST effects and instruments for Windows. For the main part they are designed to be simple, light-weight and efficient, although some are more ambitious and some more experimental. Effects; GBand - Band-pass filter. GChorus - Chorus effect. GClip - Wave-shaping signal clipper. GComp - Compressor. GComp2 - Compressor. GDelay - Delay effect. GDuckDly - Ducking delay effect. GFader - Signal gain (-100 to 0 dB). GGain - Signal gain (-12 to 12 dB). GGate - Gate. GGrain - Granular resynthesis. GHi - High-pass filter. GLow - Low-pass filter. GLFO - Triple LFO effect. GMax - Limiter. GMonoBass - Bass stereo imaging effect. GMulti - Multi-band compressor and stereo enhancer. GNormal - Noise generator for avoiding denormal problems. GRevDly - Reverse delay effect. GSnap - Pitch-correction. GTune - Chromatic tuner.

ReaPlugs VST

  • ReaPlugs VST FX Suite - Want to use some of the comprehensive FX plug-ins that REAPER provides, but stuck in another host? Haven't made the switch yet? Fear not -- you can download ReaPlugs, a package of FX that includes many of the plug-ins that come with REAPER, for free!


Anarchy Effects is a cross-platform bundle consisting of 5 audio plugins, each of which does a different novel form of frequency domain processing. VST versions now comply to VST 2.4 standard and plugins in different formats are available for both Mac and PC. Features: Parameter automation using MIDI controllers or VST automation. Complies with VST 2.4 standard. 32-bit & 64-bit versions for Mac (VST/AU) & PC (VST).

  • SpectralAutopan – assigns different pan positions to the different component pitches in the input signal. The effect pitch has on pan position is controlled by control points, which can change in pan position and pitch according to LFOs. This adds stereo depth and motion to sounds.
  • Corkscrew – mixes together multiple pitch-shifts of the input signal, increasing or decreasing their pitches in parallel, and fading them in/out at the extremes of their range. This creates the illusion of a sound that seems to continually rise or fall, but doesn’t actually change in average pitch.
  • HarmonicAdder – creates harmonic resonances by pitch shifting the dominant frequencies in your input signal by the various intervals in the harmonic series (octave, octave+fifth, two octaves, two octaves+major third etc). These harmonics can be mixed with the dry input signal to make it more resonant, or used on their own as a new sound.
  • LengthSeparator – bisects the input signal according to the lengths of its component frequencies. Short sounds become the ‘transient’ part, long sounds become the ‘stable’ part. These parts can be isolated (ie the other part removed), or assigned different pan positions to create stereo movement.
  • Convoluter – applies a convolution matrix to the spectral representation of the input signal. This bends the sound along the continuum between pure sine tones and pure noise.

  • GeoSynth - a vst instrument made several years ago, but never released – I wasn’t as excited as I’d hoped about the sounds it made. But it’s here now so you can judge for yourself. There’s no doubting that it’s a great idea, whether the sounds light your candle or not.

  • SwarmSynth - a vst instrument which uses a flocking algorithm to control a bank of oscillators as they move through an envelope-constrained 5 dimensional parametric hyperspace.


  • Svep Phaser - Flanger - Chorus (VST + AU + AAX) - Svep is a stereo modulation filter effect suitable for any sound. All parameters are easily editable in one screen and the clean and responsive user interface encourages creativity. Tweak it to produce anything from old-school phasers to subtle choruses.


  • SyndtSphere - VST + AU. basically a sphere version of the polyphonic synthesizer Syndt. With a minimalistic approach, it features a unique experience of ”surfing” between presets. All parameters are morphed according to the proximity of the different presets. By rotating a sphere that consists of more than 70 professionally created states, anyone can dial in the perfect sound without having to deal with specific parameters. In addition to the sphere, a ping-pong delay and a few more global settings are available.

to sort

  • https://github.com/JFichtl/noiseworks - intended to provide both working examples of simple digital audio effects and simple, readable code that can be used to learn just how these things work. Every single effect should consist of one or two functions or classes with the algorithms spelled out in a clear, concise way

  • Destroy FX - Free VST plugins, Free Audio Units, Windows/Mac VST/Au
    • https://github.com/sophiapoirier/destroyfx - These are some of the plugins that are part of the Destroy FX plugin pack. We support two plugin formats: VST and Audio Unit (AU). VST versions should be (nearly) source-portable to any platform which supports that. Audio Unit is a format exclusive to macOS and iOS. Some advanced features may not work on all platforms or in all hosts.

  • Tweakbench - free VST instruments and free VST effects