A model for adaptive reduced-dimensionality equalisation

We present a method for mapping between the input space of a parametric equaliser and a lower-dimensional representation, whilst preserving the effect’s dependency on the incoming audio signal. The model consists of a parameter weighting stage in which the parameters are scaled to spectral features of the audio signal, followed by a mapping process, in which the equaliser’s 13 inputs are converted to (x, y) coordinates. The model is trained with parameter space data representing two timbral adjectives (warm and bright), measured across a range of musical instrument samples, allowing users to impose a semantically-meaningful timbral modification using the lower-dimensional interface. We test 10 mapping techniques, comprising of dimensionality reduction and reconstruction methods, and show that a stacked autoencoder algorithm exhibits the lowest parameter reconstruction variance, thus providing an accurate map between the input and output space. We demonstrate that the model provides an intuitive method for controlling the audio effect’s parameter space, whilst accurately reconstructing the trajectories of each parameter and adapting to the incoming audio spectrum.