home | interaction | teaching | inventions | music | plugins

Max for Live Plugins

Almost a hundred of max for live devices are maintained and shared on maxforlive.com. Currently they total at about 30.000 downloads. Developed devices are mainly aimed at algorithmic composition, audio signal processing and API facilitated application control. You can find the plugins by searching for their 'zb.' prefix or through this link.

AU Distortion

Subharmonic is a plugin based on phase inverting wavesets and it is capable of creating different subharmonics controlled by the modulo parameter. The subharmonics are very clean on synthetic sounds but quasi-harmonic and noisy signals my become heavily distorted. The ratio determines the offset of phase inverted wavesets and is functional only with higher ratio. The exponent defines the decay of the wavesets and is also mainly functional with higher modulos. The plugin contains a resonant low pass filter to keep the rumble and loose the distortion. This thing can actually make the violin play sub-bass frequencies. Much recommended for adding low end to pretty much anything.

Valve implements a quadratic transfer that has two parameters used to specify the input and the output ranges. Curve will zoom into the parabola near value one to yield almost linear transfer at the default value of zero. Offset will tilt the transfer up to yield asymetrical output and therefore more even harmonics. Next it contains asymetrical zero-crossing distortion that uses a low-pass filtered control signal such that the brightness can be controlled. Finally an exponential transfer allows for raw compression with exponent values below 1 and expansion for values higher than one. This is a versatile non-linear component, in order to have a more real sound add some resonances to simulate bodies.

MBDistortion uses the exponential transfer to invoke distortion on three frequency bands. Exponent values above one will produce expansion and below one compression as well as plenty of distortion in both cases. For subtle effects parallel processing and filtering before and after the pluginis highly recommended. Running on higher sample rates is likely to reduce the digital processing artefact called aliasing.

EvenHarmonics allows for fine tuning harmonic distortion of the first four even components. Chebyshev plynomials are used to produce the appropriate output values with 32-bit resolution and no interpolation. The input signal is by default filtered at a frequency to avoid aliasing for the specific component but it can be readjusted. Each harmonic uses a band-pass filter to enable customization. Sine waves produce remarkably clean harmonics but complex signals do not. There might be a significan amount of inter-modulation distortion but this has yet to be researched.

AU Panning

PsychoPan implements panning of monophonic signals based on various psychoacoustic foundings. It enables one to explore how contradictory cues overrule one another. Panning is achieved by amplitude change, tiny delays, fine decorrelation with allpass filtering, coarse decorrelation with time dispersion and head shadow simulation with low pass filtering.

PsychoBal implements balancing of stereophonics signals based on various psychoacoustic foundings. It enables one to explore how contradictory cues overrule one another. Panning is achieved by amplitude change, tiny delays, fine decorrelation with allpass filtering, coarse decorrelation with time dispersion and head shadow simulation with low pass filtering.

PanMod uses a triangle wave to move the input signal in a specified range in the stereo image based on equal-power panning law. The speed of the movement can be specified to be audible or produce amplitude modulation. Left and right input signals are shuffled out of phase. Aside from controlling the source width this plug-in produces unwanted artefacts and needs further development and research.

Width is the simplest plug-in in this collection. It has a single parameter and is based on mid-side conversion and attenuation of the side signal.

AU Adaptive effects

ADFilter achieves sweeps of a band-pass filters cutoff frequency, bandwidth and stereo balancing. All three sweeps are produced by scaling the amplitude-follower values. Each has a specified multiplier - range that can also be used for inversion. Variable attack and decay - lag up and lag down times are available for modifying the control signal for each sweep.

BPFeedback allows for using amplitude following to modulate a delay line - producing pitch shifting effects and the modulation frequency of a band-pass filter. The modulation is done using a sine wave. Feedback parameter is allowed values greater than one to compensate for band-pass filter attenuation and can also cause severe blow-ups.

ADKing is the (stupid) king of the amplitude dependant processing as it uses amplitude following to scale filter cutoff frequency, delay time, pitch-shoftong ratio and ring modulator frequency. The control signal can be smoothed with the ampVariation parameter. Processes within this plug-in are difficult to exactly specify, impossible to justify and therefore not made public.

AU Pitch processing

Arpeggiator modulates an incoming audio signal by pitch shifting. It uses quantization of a triangle wave to control the pitch shift and the cutoff frequency of a band-pass filter. It is fairly inaccurate and experimental implementation

TenBandShift splits the incoming audio signal into ten frequency bands and implements pitch shifting on each band that can be specified in semitones. Useful for harmonizing material.