Robust Anomaly Detection with Applications to Acoustics and Graphs

Our goal is to develop a robust anomaly detector that can be incorporated into pattern recognition systems that may need to learn, but will never be shunned for making egregious errors. The ability to know what we do not know is a concept often overlooked when developing classifiers to discriminate between different types of normal data in controlled experiments. We believe that an anomaly detector should be used to produce warnings in real applications when operating conditions change dramatically, especially when other classifiers only have a fixed set of bad candidates from which to choose. Our approach to distributional anomaly detection is to gather local information using features tailored to the domain, aggregate all such evidence to form a global density estimate, and then compare it to a model of normal data. A good match to a recognizable distribution is not required. By design, this process can detect the "unknown unknowns" [1] and properly react to the "black swan events" [2] that can have devastating effects on other systems. We demonstrate that our system is robust to anomalies that may not be well-defined or well-understood even if they have contaminated the training data that is assumed to be non-anomalous. In order to develop a more robust speech activity detector, we reformulate the problem to include acoustic anomaly detection and demonstrate state-of-the-art performance using simple distribution modeling techniques that can be used at incredibly high speed. We begin by demonstrating our approach when training on purely normal conversational speech and then remove all annotation from our training data and demonstrate that our techniques can robustly accommodate anomalous training data contamination. When comparing continuous distributions in higher dimensions, we develop a novel method of discarding portions of a semi-parametric model to form a robust estimate of the Kullback-Leibler divergence. Finally, we demonstrate the generality of our approach by using the divergence between distributions of vertex invariants as a graph distance metric and achieve state-of-the-art performance when detecting graph anomalies with neighborhoods of excessive or negligible connectivity. [1] D. Rumsfeld. (2002) Transcript: DoD news briefing - Secretary Rumsfeld and Gen. Myers. [2] N. N. Taleb, The Black Swan: The Impact of the Highly Improbable. Random House, 2007

Phonetic Event-based Whole-Word Modeling Approaches for Speech Recognition

Speech is composed of basic speech sounds called phonemes, and these subword units are the foundation of most speech recognition systems. While detailed acoustic models of phones (and phone sequences) are common, most recognizers model words themselves as a simple concatenation of phonemes and do not closely model the temporal relationships between phonemes within words. Human speech production is constrained by the movement of speech articulators, and there is abundant evidence to indicate that human speech recognition is inextricably linked to the temporal patterns of speech sounds. Structures such as the hidden Markov model (HMM) have proved extremely useful and effective because they offer a convenient framework for combining acoustic modeling of phones with powerful probabilistic language models. However, this convenience masks deficiencies in temporal modeling. Additionally, robust recognition requires complex automatic speech recognition (ASR) systems and entails non-trivial computational costs. As an alternative, we extend previous work on the point process model (PPM) for keyword spotting, an approach to speech recognition expressly based on whole-word modeling of the temporal relations of phonetic events. In our research, we have investigated and advanced a number of major components of this system. First, we have considered alternate methods of determining phonetic events from phone posteriorgrams. We have introduced several parametric approaches to modeling intra-word phonetic timing distributions which allow us to cope with data sparsity issues. We have substantially improved algorithms used to compute keyword detections, capitalizing on the sparse nature of the phonetic input which permits the system to be scaled to large data sets. We have considered enhanced CART-based modeling of phonetic timing distributions based on related text-to-speech synthesis work. Lastly, we have developed a point process based spoken term detection system and applied it to the conversational telephone speech task of the 2006 NIST Spoken Term Detection evaluation. We demonstrate the PPM system to be competitive with state-of-the-art phonetic search systems while requiring significantly fewer computational resources

Comparison of modulation features for phoneme recognition

In this paper, we compare several approaches for the extraction of modulation frequency features from speech signal using a phoneme recognition system. The general framework in these approaches is to decompose the speech signal into a set of sub-bands. Amplitude modulations (AM) in the sub-band signal are used to derive features for automatic speech recognition (ASR). Then, we propose a feature extraction technique which uses autoregressive models (AR) of sub-band Hilbert envelopes in relatively long segments of speech signal. AR models of Hilbert envelopes are derived using frequency domain linear prediction (FDLP). Features are formed by converting the FDLP envelopes into static and dynamic modulation frequency components. In the phoneme recognition experiments using the TIMIT database, the FDLP based modulation frequency features provide significant improvements compared to other techniques (average relative improvement of 7.5 % over the base-line features). Furthermore, a detailed analysis is performed to determine the relative contribution of various processing stages in the proposed technique. Index Terms — Frequency domain linear prediction (FDLP), Modulations, Feature Extraction, Phoneme recognition. 1