VOICE BIOMETRICS UNDER MISMATCHED NOISE CONDITIONS

This thesis describes research into effective voice biometrics (speaker recognition) under mismatched noise conditions. Over the last two decades, this class of biometrics has been the subject of considerable research due to its various applications in such areas as telephone banking, remote access control and surveillance. One of the main challenges associated with the deployment of voice biometrics in practice is that of undesired variations in speech characteristics caused by environmental noise. Such variations can in turn lead to a mismatch between the corresponding test and reference material from the same speaker. This is found to adversely affect the performance of speaker recognition in terms of accuracy. To address the above problem, a novel approach is introduced and investigated. The proposed method is based on minimising the noise mismatch between reference speaker models and the given test utterance, and involves a new form of Test-Normalisation (T-Norm) for further enhancing matching scores under the aforementioned adverse operating conditions. Through experimental investigations, based on the two main classes of speaker recognition (i.e. verification/ open-set identification), it is shown that the proposed approach can significantly improve the performance accuracy under mismatched noise conditions. In order to further improve the recognition accuracy in severe mismatch conditions, an approach to enhancing the above stated method is proposed. This, which involves providing a closer adjustment of the reference speaker models to the noise condition in the test utterance, is shown to considerably increase the accuracy in extreme cases of noisy test data. Moreover, to tackle the computational burden associated with the use of the enhanced approach with open-set identification, an efficient algorithm for its realisation in this context is introduced and evaluated. The thesis presents a detailed description of the research undertaken, describes the experimental investigations and provides a thorough analysis of the outcomes

Combined-channel instantaneous frequency analysis for audio source separation based on comodulation

Thesis (Ph. D.)--Harvard-MIT Division of Health Sciences and Technology, 2008.Includes bibliographical references (p. 295-303).Normal human listeners have a remarkable ability to focus on a single sound or speaker of interest and to block out competing sound sources. Individuals with hearing impairments, on the other hand, often experience great difficulty in noisy environments. The goal of our research is to develop novel signal processing methods inspired by neural auditory processing that can improve current speech separation systems. These could potentially be of use as assistive devices for the hearing impaired, and in many other communications applications. Our focus is the monaural case where spatial information is not available. Much perceptual evidence indicates that detecting common amplitude and frequency variation in acoustic signals plays an important role in the separation process. The physical mechanisms of sound generation in many sources cause common onsets/offsets and correlated increases/decreases in both amplitude and frequency among the spectral components of an individual source, which can potentially serve as a distinct signature. However, harnessing these common modulation patterns is difficult because when spectral components of competing sources overlap within the bandwidth of a single auditory filter, the modulation envelope of the resultant waveform resembles that of neither source. To overcome this, for the coherent, constant-frequency AM case, we derive a set of matrix equations which describes the mixture, and we prove that there exists a unique factorization under certain constraints. These constraints provide insight into the importance of onset cues in source separation. We develop algorithms for solving the system in those cases in which a unique solution exists. This work has direct bearing on the general theory of non-negative matrix factorization which has recently been applied to various problems in biology and learning. For the general, incoherent, AM and FM case, the situation is far more complex because constructive and destructive interference between sources causes amplitude fluctuations within channels that obscures the modulation patterns of individual sources.(cont.) Motivated by the importance of temporal processing in the auditory system, and specifically, the use of extrema, we explore novel methods for estimating instantaneous amplitude, frequency, and phase of mixtures of sinusoids by comparing the location of local maxima of waveforms from various frequency channels. By using an overlapping exponential filter bank model with properties resembling the cochlea, and combining information from multiple frequency bands, we are able to achieve extremely high frequency and time resolution. This allows us to isolate and track the behavior of individual spectral components which can be compared and grouped with others of like type. Our work includes both computational and analytic approaches to the general problem. Two suites of tests were performed. The first were comparative evaluations of three filter-bank-based algorithms on sets of harmonic-like signals with constant frequencies. One of these algorithms was selected for further performance tests on more complex waveforms, including AM and FM signals of various types, harmonic sets in noise, and actual recordings of male and female speakers, both individual and mixed. For the frequency-varying case, initial results of signal analysis with our methods appear to resolve individual sidebands of single harmonics on short time scales, and raise interesting conceptual questions on how to define, use and interpret the concept of instantaneous frequency. Based on our results, we revisit a number of questions in current auditory research, including the need for both rate and place coding, the asymmetrical shapes of auditory filters, and a possible explanation for the deficit of the hearing impaired in noise.by Barry David Jacobson.Ph.D