This plugin aims to reduce undesirable background noise and hum in otherwise silent passages. 

When the signal's instantaneous amplitude exceeds the opening threshold, a gate is opened allowing the input signal to pass through.  The time it takes for the gate to open fully can be set with the attack control.  As soon as the signal's RMS power level drops below the closing threshold, the gate closes.  Closing the gate takes a fixed time of 20 ms; the closed gate attenuates the signal by 60 dB. 

To cope with powerline hum as often present in signals from electric guitars, a notch filter can be activated by setting the mains frequency control to a non-zero value.  The filter will prevent this frequency from contributing to the signal power measurement.  While this by itself doesn't eliminate any hum in the signal, it allows the closing threshold to be lowered further without mains hum keeping the gate open unduly.  The default mains setting is 50 Hz. 

This compressor has been designed primarily to create natural-sounding sustain for the electric guitar without sacrificing its brightly percussive character.  However, it appears to apply well to a variety of other sound sources, and with CompressX2 a stereo version is available as well. 

To be able produce strong compression and keep a natural-sounding attack, the design catches attack-phase power spikes with a soft saturation circuit, turning peaks above 0 dB into additional harmonic content while keeping the output level ceiling untouched.  This is the default setting of the mode control, using an approximated atan saturation function.  Three oversampling options are available, 2x with 32-tap filters and 4x with 64 and 128 taps. 

The measure control select which indicator of loudness to base calculations on: peak – instantaneous amplitude – measurement allows the unit to react very quickly, while rms – root mean square power – is of a gentler kind. 

Compression amount is controlled through the strength knob, from 0 effectively disabling the effect, up to a maximum ratio of 16:1.  The attack and release controls map higher values to slower reactions. 

This stereo version of Compress applies uniform compression to both channels in proportion to their combined power. 

This emulation of the tone stack of a traditional Fender-design instrument amplifier has been devised and implemented by David T. Yeh^[yeh06] (the hard part), with subsequent expansion to include more amplifier models by Tim Goetze (the easy part). 

Due to the nature of the original circuit, the bass, mid and treble controls are not operating independently as in a modern three-way equaliser. 

All but the last model are using the procedural implementation with continuously updated direct form II filters and sample rate independency.  It must be noted that the "DC 30" preset has been included despite the slight difference in layout between the British original and the Fender circuit. 

The "5F6-A LT" model is using the lattice filter implementation mentioned in the paper, operating on precomputed simulation data for 44.1 kHz. 

Tracing the stages of an idealised circuit, this plugin aims to recreate those features of traditional guitar amplification electronics that have proved musically useful, and to provide them with the most musical rather than the most authentic ranges of adjustment and character.  CabinetIV provides matching recreations of loudspeaker cabinets. 

The processor consists – with some interconnections – of a configurable lowcut input filter, a ToneStack circuit of the procedural variant, a saturating 'preamp' stage with adjustable gain and variable distortion asymmetry followed by the bright filter, compression characteristics determined by the attack and squash controls and finally a 'power amp' stage with the amount of saturation depending on both gain and power settings. 

Sound quality and computational load can be balanced with the over control affording a choice of 2x or 4x oversampling with 32-tap filters, or 8x with 64 taps.  Lower quality settings will sound slightly grittier and less transparent, and at high gain aliasing may become audible. 

This plugin applies an acoustic instrument body modeling technique^[bank07] to recreate the timbre-shaping of an electric instrument amplifier's speaker cabinet. 

A selection of several hundred response shapes automatically created in the likeness of classic cabinets has been narrowed down to a handful of idealised tones.  As with AmpVTS, which provides a matching recreation of traditional guitar amplification, the design and selection process has been ruled by musicality over fidelity. 

The filter banks implemented are 64 2^nd order IIR and one 128-tap FIR in parallel.  Their parameter presets are shared between the 44.1 and 48 kHz sample rates, the higher rate implying that timbre brightens up.  Higher sample rates produce the same tones by rate conversion, up to 192 kHz. 

Despite the complexity, computational load is very modest thanks to vector arithmetic if a hardware implementation is available – if not, however, the load will be easily an order of magnitude higher, and possibly found to be prohibitive on less powerful hardware. 

This reverb processor is an adaptation of the design discussed by Jon Dattorro^[dat97a] who attributes it to David Griesinger.  A smooth reverb response is achieved by a network of twelve delay lines, two of which are modulated very subtly, in a chorus-like fashion. 

The bandwidth control reduces high-frequency content before it enters the 'tank', while damping controls how quickly the reverberating tail darkens. 

This version of the Plate reverberator comes with stereo inputs. 

Please note that plugin embodies a very basic building block of audio DSP, not an elaborate effect that will be pleasing to hear right away.  To turn saturation into a musically useful effect it is usually combined with some sort of filtering and dynamics modulation. 

The mode control chooses from a selection of clipping functions of varying character.  Even-order harmonics can be added with the bias setting.  Towards the maximum, sound will start to get scratchy and eventually starve away. 

The plugin is 8x oversampled with 64-tap polyphase filters, effectively suppressing aliasing noise for most musical applications.  Changes to the bias control induce short-lived energy at DC in the output.  In order to reduce the computational load incurred when evaluating transcendental functions at eight times the nominal sample rate, these are approximated roughly, using Chebyshev polynomials whose coefficients depend on the amplitude's floating point representation exponent. 

This effect plugin aims to enhance clarity of perception of thin or muddy sound.  It can also be very useful when more bass register definition or more treble presence is called for and generic equalisation does not work without noticeably raising signal and/or noise level.  A stereo version is available as SpiceX2. 

Bass and treble portions of the signal are isolated using two 24 dB/octave Linkwitz-Riley crossover networks^[lr76] to ensure a flat frequency response at zero effect intensity (controlled through the .gain settings).  After compression, a polynomial waveshaper synthesises the first three overtones of the bass register.  This enhances the perception of the fundamental frequency, being the difference tone of these harmonics.  Treble band processing applies an analog-style saturation function without compression.  Synthesised harmonic content is shaped through bandpass and highpass filters and mixed back into the crossover sum signal. 

Stereo version of Spice.  Bass compression is governed by the sum of both channels, as in CompressX2.  Nevertheless, the amount of harmonic generation differing between the two channels can have subtle effects on the stereo image. 

A standard mono chorus with optional feedback.  The parameter range suits subtle effects as well as all-out flanging. 

Modifying the delay time t when feedback is active will cause audible 'zipper' noise. 

This take on the classic effect features two modulation choices, traditional sine-based periodicity or smoothened fractal oscillation. 

Very high resonance settings can cause self-oscillation peaking in excess of 0 dB. 

A versatile selection of filters of varying character in band- and lowpass configuration.  The cutoff frequency can be modulated by both the input signal envelope and by a Lorenz attractor.  The default settings provide some sort of an automatic wah effect. 

The extent of filter modulation is set through the range parameter.  The shape of the modulation is mixed from the attractor and the envelope according to the lfo/env balance.  The attractor signal itself changes with the x/z parameter. 

Filter stage gain can be used to add inter-stage saturation.  To prevent this from causing audible aliasing, the plugin can be run in oversampled mode, at ratios selectable through the over control. 

At very high Q and f combined, the filter stability may become compromised.  Computational load varies greatly with the over and filter settings. 

A stereo delay with resonant filters and fractally modulated panning. 

The delay times are set through the bpm control and the divider adjustment.  Triplet and sixteenth settings create a dotted rhythm.  With every beat, the filter resonance frequencies are retuned to random steps on an equal-tempered chromatic scale, to the reference set through the tune control. 

A classic octave-band biquad-filter design, basically a direct digital translation of the analog original.  There's also a stereo version (Eq10X2). 

Frequency bands centered above Nyquist are automatically disabled. 

The controls of this stereo version of Eq apply to both channels. 

Four adjustable biquad filters in series, in a vector arithmetic implementation.  The default setup is an identity filter with a mode configuration of lowshelve, band, band, hishelve, all at zero gain. 

The Q control value maps non-linearly to actual filter Q: a zero control value results in filter Q of ½, a value of 0.3 corresponds to a Butterworth-equivalent Q of ½√₂, and the maximum control setting of 1 results in a filter Q of 50. 

Parallelisation of the serial filter topology causes its response to lag by three samples. 

Control response is smoothened by crossfading between two filter banks. 

In addition to provoding a basic panorama control, a perception of stereo width is created using complementary filters on the two output channels. 

The output channels always sum to a flat frequency response. 

The design of this plugin owes to the Orban 245F Stereo Synthesizer^[orban93]. 

This plugin reduces the width of a stereophonic signal.  Its primary use is for headphone listening to music that was panned very creatively. 

Mid/side processing tends to sound more transparent for moderate strength settings.  However, it will more strongly attenuate signals that are panned to the far sides of the stereo image (rarely encountered in contemporary music production anymore but quite common, for example, on early Beatles recordings). 

The old friend, indispensable for testing and tuning. 

Mostly white pseudonoise, mixed and filtered from the output of two Dattorro multibit generators^[dat02]. 

This plugin turns the oscillating state of a fractal attractor into an audio stream.  The result is something that most would without much hesitation classify as noise. 

The Lorenz attractor^[lorenz63] is one of the earliest models of deterministic chaos discovered deriving from the Navier-Stokes equations^wp. 

The Rössler system^[roessler76] is similar but contains only one non-linearity. 

The x, y and z controls set the amplitude of the respective variables of the attractor state in the output audio signal. 

The attractor state variables are scaled and translated to stay mostly within the [-1,1] range and not contain a DC offset.  Nevertheless, due to the unpredictable nature of the systems, peak limits cannot be guaranteed.  In addition, some energy near DC may be produced; therefore a configurable high-pass filter is part of the circuit.  It can be turned off by setting the hp parameter to zero. 

The output signal varies with the sample rate. 

A sample-accurate metronome.  Two simplistic modal synthesis models are available for the click: box is a small wooden box struck with a soft wooden mallet, stick a scratchy stick hit.  In addition, there's also a very synthetic beep, and finally dirac, a very nasty single-sample pulse of 0 dB amplitude and little immediate musical use. 

All click sounds are synthesised once when the plugin is loaded and then played back from memory. 

The Chief Executive Oscillator forever calls for more profit. 

Sound data created with the flite^[flite] application.