Calibration of sound source localisation for robots using multiple adaptive filter models of the cerebellum

The aim of this research was to investigate the calibration of Sound Source Localisation (SSL) for robots using the adaptive filter model of the cerebellum and how this could be automatically adapted for multiple acoustic environments. The role of the cerebellum has mainly been identified in the context of motor control, and only in recent years has it been recognised that it has a wider role to play in the senses and cognition. The adaptive filter model of the cerebellum has been successfully applied to a number of robotics applications but so far none involving auditory sense. Multiple models frameworks such as MOdular Selection And Identification for Control (MOSAIC) have also been developed in the context of motor control, and this has been the inspiration for adaptation of audio calibration in multiple acoustic environments; again, application of this approach in the area of auditory sense is completely new. The thesis showed that it was possible to calibrate the output of an SSL algorithm using the adaptive filter model of the cerebellum, improving the performance compared to the uncalibrated SSL. Using an adaptation of the MOSAIC framework, and specifically using responsibility estimation, a system was developed that was able to select an appropriate set of cerebellar calibration models and to combine their outputs in proportion to how well each was able to calibrate, to improve the SSL estimate in multiple acoustic contexts, including novel contexts. The thesis also developed a responsibility predictor, also part of the MOSAIC framework, and this improved the robustness of the system to abrupt changes in context which could otherwise have resulted in a large performance error. Responsibility prediction also improved robustness to missing ground truth, which could occur in challenging environments where sensory feedback of ground truth may become impaired, which has not been addressed in the MOSAIC literature, adding to the novelty of the thesis. The utility of the so-called cerebellar chip has been further demonstrated through the development of a responsibility predictor that is based on the adaptive filter model of the cerebellum, rather than the more conventional function fitting neural network used in the literature. Lastly, it was demonstrated that the multiple cerebellar calibration architecture is capable of limited self-organising from a de-novo state, with a predetermined number of models. It was also demonstrated that the responsibility predictor could learn against its model after self-organisation, and to a limited extent, during self-organisation. The thesis addresses an important question of how a robot could improve its ability to listen in multiple, challenging acoustic environments, and recommends future work to develop this ability

Array signal processing algorithms for localization and equalization in complex acoustic channels

The reproduction of realistic soundscapes in consumer electronic applications has been a driving force behind the development of spatial audio signal processing techniques. In order to accurately reproduce or decompose a particular spatial sound field, being able to exploit or estimate the effects of the acoustic environment becomes essential. This requires both an understanding of the source of the complexity in the acoustic channel (the acoustic path between a source and a receiver) and the ability to characterize its spatial attributes. In this thesis, we explore how to exploit or overcome the effects of the acoustic channel for sound source localization and sound field reproduction. The behaviour of a typical acoustic channel can be visualized as a transformation of its free field behaviour, due to scattering and reflections off the measurement apparatus and the surfaces in a room. These spatial effects can be modelled using the solutions to the acoustic wave equation, yet the physical nature of these scatterers typically results in complex behaviour with frequency. The first half of this thesis explores how to exploit this diversity in the frequency-domain for sound source localization, a concept that has not been considered previously. We first extract down-converted subband signals from the broadband audio signal, and collate these signals, such that the spatial diversity is retained. A signal model is then developed to exploit the channel's spatial information using a signal subspace approach. We show that this concept can be applied to multi-sensor arrays on complex-shaped rigid bodies as well as the special case of binaural localization. In both c! ases, an improvement in the closely spaced source resolution is demonstrated over traditional techniques, through simulations and experiments using a KEMAR manikin. The binaural analysis further indicates that the human localization performance in certain spatial regions is limited by the lack of spatial diversity, as suggested in perceptual experiments in the literature. Finally, the possibility of exploiting known inter-subband correlated sources (e.g., speech) for localization in under-determined systems is demonstrated. The second half of this thesis considers reverberation control, where reverberation is modelled as a superposition of sound fields created by a number of spatially distributed sources. We consider the mode/wave-domain description of the sound field, and propose modelling the reverberant modes as linear transformations of the desired sound field modes. This is a novel concept, as we consider each mode transformation to be independent of other modes. This model is then extended to sound field control, and used to derive the compensation signals required at the loudspeakers to equalize the reverberation. We show that estimating the reverberant channel and controlling the sound field now becomes a single adaptive filtering problem in the mode-domain, where the modes can be adapted independently. The performance of the proposed method is compared with existing adaptive and non-adaptive sound field control techniques through simulations. Finally, it is shown that an order of magnitude reduction in the computational complexity can be achieved, while maintaining comparable performance to existing adaptive control techniques