We propose a new framework that generalizes the parameters of neural network models to $C^*$-algebra-valued ones. $C^*$-algebra is a generalization of the space of complex numbers. A typical example is the space of continuous functions on a compact space. This generalization enables us to combine multiple models continuously and use tools for functions such as regression and integration. Consequently, we can learn features of data efficiently and adapt the models to problems continuously. We apply our framework to practical problems such as density estimation and few-shot learning and show that our framework enables us to learn features of data even with a limited number of samples. Our new framework highlights the potential possibility of applying the theory of $C^*$-algebra to general neural network models.

This study attempts to simulate evaluation flow after the flooding in the Kanto-Tohoku heavy rain in September 2015, using a mobile GPS data. A simple Monte Carlo approach is introduced to simulate micro-scale people flow from spatially sparse GPS information. The simulation result shows that GPS data effectively recovers people flow after flooding even if the simulation approach is very simple.

Urban heatwave is increasingly severe as the global warming advances. Even worse, in Tokyo, an increasing proportion of residences are vulnerable against heats under the aging society. Today, heatwave monitoring is an emergent task toward climate adaptive urban development. Our final goal is developing a system to monitor real-time and micro-scale heatwave risk in Tokyo. To achieve it, we performed the following observation experiments: airborne monitoring, monitoring from the Tokyo Sky Tree, and micro-scale monitoring censoring inside and outside comforts. A method to combine these multi-scale information is developed to estimate micro-scale spatiotemporal behavior on heatwave risks. Based on the result, it is discussed how we can achieve the real-time and micro-scale heatwave monitoring, and make Tokyo more risk adaptive.

Many studies have shown that articulatory features can significantly improve the performance of automatic speech recognition systems. Unfortunately, such features are not available at recognition time. There are two main approaches to solve this problem: a feature-based approach, the most popular example of which is the acoustic-to-articulatory inversion, where the missing articulatory features are generated from the speech signal, and a model-based approach, where articulatory information is embedded in the model structure and parameters in a way that allows recognition using only acoustic features. In this paper, we propose two new methods to integrate articulatory information into a phoneme recognition system. One of them is feature based, and the other is model based. In both cases, the underlying acoustic model (AM) is a deep neural networks-hidden Markov model (DNN-HMM) hybrid. In the feature-based method, the articulatory inversion DNN and the acoustic model DNN are trained jointly using a linear combination of their loss functions. In the model-based method, we utilize the generalized distillation framework to train the AM DNN. In this case, first, a teacher DNN is trained on both the acoustic and articulatory features, and then its outputs are used as additional targets during the AM DNN training with acoustic features only. A 7-fold cross-validation experiments using 42 speakers from the XRMB database showed that both the proposed methods provide about 22% to 25% performance improvement with respect to the DNN acoustic model trained with acoustic features only.

Modelling and forecasting of asset volatility and covariance is of prime importance in the construction of portfolios. In this paper, we present a generalised multi-factor model that incorporates heteroskedasticity and dependence in the idiosyncratic error terms. We apply this model to forecasting the time-varying covariances in a basket of high interest rate and a basket of low interest rate carry trade currencies and then utilise these forecasts for portfolio optimisation. We compare traditional Markowitz portfolio optimisation to the more recently popular risk-based portfolio optimisation. Our model is shown to provide superior risk-adjusted returns for a currency carry trade strategy over the period 1999 – 2014.

We develop a new algorithm for spatial field reconstruction in heterogeneous (mixed analog & digital sensors) wireless sensor networks (WSNs). We consider spatial physical phenomena which are observed by a heterogeneous WSN, meaning that it is partially consists of sparsely deployed high-quality sensors and partially of low-quality sensors. The high-quality sensors transmit their (continuous) noisy observations to the Fusion Centre (FC), while the low-quality sensors first perform a simple thresholding operation and then transmit their binary values over imperfect wireless channels to the FC. We develop a novel algorithm that is based on a multivariate series expansion approach resulting in a Saddle-point type approximation. We then present comprehensive study of the performance gain that can be obtained by augmenting the high-quality sensors with low-quality sensors using real data of insurance storm surge database known as the Extreme Wind Storms Catalogue.

This paper proposes a novel countermeasure framework to detect spoofing attacks to reduce the vulnerability of automatic speaker verification (ASV) systems. Recently, ASV systems have reached equivalent performances equivalent to those of other biometric modalities. However, spoofing techniques against these systems have also progressed drastically. Experimentation using advanced speech synthesis and voice conversion techniques has showed unacceptable false acceptance rates and several new countermeasure algorithms have been explored to detect spoofing materials accurately. However, the countermeasures proposed so far are based on the acoustic differences between natural speech signals and artificial speech signals, expected to become gradually smaller in the near future. In this paper, we focus on voice liveness detection, which aims to validate whether the presented speech signals originated from a live human. We use the phenomenon of pop noise, which is a distortion that happens when human breath reaches a microphone,as liveness evidence. This paper proposes pop noise detection algorithms and shows through an experimental study that they can be used to discriminate live voice signals from artificial ones generated by means of speech synthesis techniques.

We propose a novel application of a family of non-parametric statistical models to estimate head-related transfer functions (HRTFs) using spatial-temporal Gaussian processes (GPs). In this approach, we model the head-related impulse response (HRIR) utilizing non-parametric regression via a GP. The challenge posed by this problem involves accurate modeling of the spatial correlation structure jointly with the temporal correlation structure at each spatial location for the HRIR. We solve this problem by constructing a joint spatial-temporal kernel characterizing the GP regression model. To perform inference, we estimate the hyper-parameters of the GP regression kernel via maximum signal-to-deviation-ratio estimation on the basis of a real experimental setup in which we collected observations of the HRIR using two head-and-torso simulators (HATSs): KEMAR and B&K. We also perform cross validation of the model by training on the KEMAR system and assessing the generalization of our model and its out-of-sample predictive power for HRIRs at any locations that we predict by the model assessed on the B&K system. The corresponding HRTFs are obtained as the Fourier transform of the HRIRs. In the experiments, we show that our method is robust against variation in the azimuth interval needed to perform high-accuracy interpolation and has the expressive power to handle the individual characteristics of each HATS.

We present a novel generative model for audio event transcription that recognizes “events” on audio signals including multiple kinds of overlapping sounds. In the proposed model, firstly, the overlapping audio events are modeled based on nonnegative matrix factorization into which Bayesian nonparametric approaches: the Markov Indian buffet process and the Chinese restaurant process, are incorporated. This approach allows us to automatically transcribe the events while avoiding the model selection problem by assuming a countably infinite number of possible audio events in the input signal. Then, Bayesian logistic regression annotates the audio frames with the multiple event labels in a semi-supervised learning setup. Experimental results show that our model can better annotate an audio signal in comparison with a baseline method. Additionally, we verify that our infinite generative model is also able to detect unknown audio events that are not included in the training data.

Availability of large amounts of raw unlabeled data has sparked the recent surge in semi-supervised learning research. In most works, however, it is assumed that labeled and unlabeled data come from the same distribution. This restriction is removed in the self-taught learning approach where unlabeled data can be different, but nevertheless have similar structure. First, a representation is learned from the unlabeled data via sparse coding and then it is applied to the labeled data used for classification. In this work, we implemented this method for the music genre classification task using two different databases: one as unlabeled data pool and the other for supervised classifier training. Music pieces come from 10 and 6 genres for each database respectively, while only one genre is common for both of them. Results from wide variety of experimental settings show that the self-taught learning method improves the classification rate when the amount of labeled data is small and, more interestingly, that consistent improvement can be achieved for a wide range of unlabeled data sizes.

INTERSPEECH 2012: The 13th Annual Conference of the International Speech Communication Association, September 9-13, 2012, Portland, Oregon, USA.

The sequence kernel has been shown to be a promising kernel function for learning from sequential data such as speech and DNA. However, it is not scalable to massive datasets due to its high computational cost. In this paper, we propose a method of approximating the sequence kernel that is shown to be computationally very efficient. More specifically, we formulate the problem of approximating the sequence kernel as the problem of obtaining a pre-image in a reproducing kernel Hilbert space. The effectiveness of the proposed approximation is demonstrated in text-independent speaker identification experiments with 10 male speakers-our approach provides significant reduction in computation time with limited performance degradation. Based on the proposed method, we develop a real-time kernel-based speaker identification system using Virtual Studio Technology (VST).

Spoken Document Retrieval (SDR) and Spoken Term Detection (STD) have been two of the most intensively investigated topics in spoken document processing research according to the establishment of the SDR and STD test collections by the Text REtrieval Conference (TREC) and NIST. Because Japanese spoken document processing researchers also requires such test collections for SDR and STD, we have established a working group to develop these collections in Special Interest Group-Spoken Language Processing (SIG-SLP) of the Information Processing Society of Japan. The working group has constructed and made available a test collection for SDR, and is now constructing new test collections for STD that will be open to researchers. The present paper introduces the policies, outline, and schedule of the new test collections. Then, the new test collections are compared with the NIST STD test collections. Index Terms: spoken term detection, test collection 1.

INTERSPEECH2010: 11th Annual Conference of the International Speech Communication Association, September 26-30, 2010, Chiba, Japan.

Hidden Markov models (HMMs) are powerful generative models for sequential data that have been used in automatic speech recognition for more than two decades. Despite their popularity, HMMs make inaccurate assumptions about speech signals, thereby limiting the achievable performance of the conventional speech recognizer. Penalized logistic regression (PLR) is a well-founded discriminative classifier with long roots in the history of statistics. Its classification performance is often compared with that of the popular support vector machine (SVM). However, for speech classification, only limited success with PLR has been reported, partially due to the difficulty with sequential data. In this paper, we present an elegant way of incorporating HMMs in the PLR framework. This leads to a powerful discriminative classifier that naturally handles sequential data. In this approach, speech classification is done using affine combinations of HMM log-likelihoods. We believe that such combinations of HMMs lead to a more accurate classifier than the conventional HMM-based classifier. Unlike similar approaches, we jointly estimate the HMM parameters and the PLR parameters using a single training criterion. The extension to continuous speech recognition is done via rescoring of N-best lists or lattices.

We investigate a method using support vector machines (SVMs) with walk-based graph kernels for high-level feature extraction from images. In this method, each image is first segmented into a finite set of homogeneous segments and then represented as a segmentation graph where each vertex is a segment and edges connect adjacent segments. Given a set of features associated with each segment, we then obtain a positive definite kernel between images by comparing walks in the respective segmentation graphs, and image classification is carried out with an SVM based on this kernel. In a benchmark experiment on the MediaMill challenge problem, the mean average precision increased from 0.216 (baseline) to 0.341 when our method was utilized.

In this paper, we propose a novel semisupervised speaker identification method that can alleviate the influence of non-stationarity such as session dependent variation, the recording environment change, and physical condition/emotion. We assume that the utterance variation follows the covariate shift model, where only the utterance sample distribution changes in the training and test phases. Our method consists of weighted versions of kernel logistic regression and cross-validation and is theoretically shown to have the capability of alleviating the influence of covariate shift. We experimentally show through text-independent speaker identification simulations that the proposed method is promising in dealing with variations in session dependent utterance variation.

We investigate a novel method for speaker verification with nonaudible murmur (NAM) segments. NAM is recorded using a special microphone placed on the neck and is hard for other people to hear. We have already reported a method based on a support vector machine (SVM) using NAM segments to use a keyword phrase effectively. To further exploit keyword-specific features, we introduce a global alignment (GA) kernel and penalized logistic regression machine (PLRM). In the experiments using NAM from 55 speakers, our method achieved an error reduction rate of roughly 60% compared with the SVM-based method using a polynomial kernel.

State-of-the-art pattern recognition approaches like neural networks or kernel methods have only had limited success in speech recognition. The difficulties often encountered include the varying lengths of speech signals as well as how to deal with sequences of labels (e.g., digit strings) and unknown segmentation. In this paper we present a combined hidden Markov model (HMM) and penalized logistic regression machine (PLRM) approach to continuous speech recognition that can cope with both of these difficulties. The key ingredients of our approach are N-best rescoring and PLRM with garbage class. Experiments on the Aurora2 connected digits database show significant increase in recognition accuracy relative to a purely HMM-based system.

We propose in this paper a new family of kernels to handle time series, notably speech data, within the framework of kernel methods which includes popular algorithms such as the support vector machine. These kernels elaborate on the well known dynamic time warping (DTW) family of distances by considering the same set of elementary operations, namely substitutions and repetitions of tokens, to map a sequence onto another. Associating to each of these operations a given score, DTW algorithms use dynamic programming techniques to compute an optimal sequence of operations with high overall score, in this paper we consider instead the score spanned by all possible alignments, take a smoothed version of their maximum and derive a kernel out of this formulation. We prove that this kernel is positive definite under favorable conditions and show how it can be tuned effectively for practical applications as we report encouraging results on a speech recognition task.

We investigated a speaker verification method that uses non-audible murmur (NAM) segments using newly collected data and obtained several findings that will be useful when speaker verification systems are made in practice. NAM is recorded using a special microphone placed on the surface of the body, so it includes almost no external noise and is hard for other people to hear. By utilizing these properties, we have already reported a text-dependent method using NAM segments that can use a keyword phrase safely. This paper extends the examination with newly collected data consisting of NAM uttered by 18 male and 9 female imposter speakers and by 18 male and 10 female customer speakers. Experiments with various numbers of training utterances and sessions show that it is effective to use data recorded in multiple sessions. We also investigated the minimum number of training utterances needed in our method.

One significant problem for spoken language systems is how to cope with users' out-of-domain (OOD) utterances which cannot be handled by the back-end application system. In this paper, we propose a novel OOD detection framework, which makes use of the classification confidence scores of multiple topics and applies a linear discriminant model to perform in-domain verification. The verification model is trained using a combination of deleted interpolation of the in-domain data and minimum-classification-error training, and does not require actual OOD data during the training process, thus realizing high portability. When applied to the "phrasebook" system, a single utterance read-style speech task, the proposed approach achieves an absolute reduction in OOD detection errors of up to 8.1 points (40% relative) compared to a baseline method based on the maximum topic classification score. Furthermore, the proposed approach realizes comparable performance to an equivalent system trained on both in-domain and OOD data, while requiring no OOD data during training. We also apply this framework to the "machine-aided-dialogue" corpus, a spontaneous dialogue speech task, and extend the framework in two manners. First, we introduce topic clustering which enables reliable topic confidence scores to be generated even for indistinct utterances, and second, we implement methods to effectively incorporate dialogue context. Integration of these two methods into the proposed framework significantly improves OOD detection performance, achieving a further reduction in equal error rate (EER) of 7.9 points

We propose a new approach to isolated-word speech recognition based on penalized logistic regression machines (PLRMs). With this approach we combine the hidden Markov model (HMM) with multiclass logistic regression resulting in a powerful speech recognizer which provides us with the posterior probability for each word. Experiments on the English E-set show significant improvements compared to conventional HMM-based speech recognition

We propose a speaker verification method using non-audible murmur (NAM) segments, which are different from normal speech and hard for other people to catch them. To use NAM, we therefore take a text-dependent verification strategy in which each user utters her/his own keyword phrase and utilize not only speaker-specific but also keyword-specific acoustic information. We expect this strategy to yield a relatively high performance. NAM segments, which consist of multiple short-term feature vectors, are used as input vectors to capture keyword-specific acoustic information well. To handle segments with a large number of dimensions, we use the support vector machine (SVM). In experiments using NAM data of 19 male and 10 female speakers recorded in three different sessions, we achieved equal error rates of 0.04% (male) and 1.1% (female) when using 145-ms-long NAM segments. These rates are half or less those obtained with 25-ms-long input vectors.

In this work we investigate the use of a greedy training algorithm for use with the dual Penalized Logistic Regression Machine (dPLRM), and our target application is detection of broad class phonetic features. The use of a greedy training algorithm is meant to alleviate the infeasible memory and computational demands that arises during the learning phase when the amount of training data increases. We show that using only a subset of the training data, chosen in a greedy manner, we can achieve as good as or better performance as when using the full training set. We can also train dPLRMs using data sets that are significantly larger than what our current computational resources can accommodate when using non-greedy approaches.

The SAGE (space-alternating generalized expectation-maximization) algorithm (Celeux, G. et al., 2001) is one of the most elegant and popular extensions of the EM (expectation maximization) algorithm for performing ML (maximum likelihood) or MAP (maximum a posteriori) parameter estimation. This algorithm updates parameter components by subblocks by alternating missing data spaces. Its efficiency has been reported in numerous simulation studies. We propose here an MCMC (Markov chain Monte Carlo) strategy named SADA (space-alternating data augmentation) which relies on the same principle in order to sample efficiently from (posterior) distributions and we discuss its application to finite mixtures of Gaussians. For this model, we also present an original implementation of the SAGE algorithm. In Monte Carlo simulations and in an application for speaker recognition, these methods, which are straightforward modifications of the standard EM and DA (data augmentation) algorithms, consistently outperform them.

One significant problem for spoken language systems is how to cope with users' OOD (out-of-domain) utterances which cannot be handled by the back-end system. In this paper, we propose a novel OOD detection framework, which makes use of classification confidence scores of multiple topics and trains a linear discriminant in-domain verifier using gradient probabilistic descent (GPD). Training is based on deleted interpolation of the in-domain data, and thus does not require actual OOD data, providing high portability. Three topic classification schemes of word N-gram models, latent semantic analysis (LSA), and support vector machines (SVM) are evaluated, and SVM is shown to have the greatest discriminative ability. In an OOD detection task, the proposed approach achieves an absolute reduction in equal error rate (EER) of 6.5% compared to a baseline method based on a simple combination of multiple-topic classifications. Furthermore, comparison with a system trained using OOD data demonstrates that the proposed training scheme realizes comparable performance while requiring no knowledge of the OOD data set.

An efficient, scalable speech recognition architecture is proposed for multidomain dialog systems by combining topic detection and topic-dependent language modeling. The inferred domain is automatically detected from the user's utterance, and speech recognition is then performed with an appropriate domain-dependent language model. The architecture improves accuracy and efficiency over current approaches and is scaleable to a large number of domains. In this paper, a novel framework using a multilayer hierarchy of language models is introduced in order to improve robustness against topic detection errors. The proposed system provides a relative reduction in WER of 10.5% over a single language model system. Furthermore it achieves an accuracy that is comparable to using multiple language models in parallel while using only a fraction of the computational cost.

A speech recognition architecture combining topic detection and topic-dependent language modeling is proposed. In this architecture, a hierarchical back-off mechanism is introduced to improve system robustness. Detailed topic models are applied when topic detection is confident, and wider models that cover multiple topics are applied in cases of uncertainty. In this paper, two topic detection methods are evaluated for the architecture: unigram likelihood and SVM (Support Vector Machine). On the ATR Basic Travel Expression corpus, both topic detection methods provide a comparable reduction in WER of 10.0% and 11.1 % respectively over a single language model system. Finally the proposed re-decoding approach is compared with an equivalent system based on re-scoring. It is shown that redecoding is vital to provide optimal recognition performance. 1.

This paper investigates a 2-class classifier approach with the aim of improving the word verification performance. The classifier operates on a discriminant function which is a linear combination of the smoothed likelihood ratios for the N-best candidates and the background (BG) and out-of-vocabulary (OOV) filler models, and is optimized using discriminative training to minimize the classification error. This paper discusses several strategies involving the likelihood ratio based formulation and the use of N-best candidates and the BG and OOV models in the classifier. In word verification experiments using a connected-digit database containing utterances recorded in a moving car with a hands-free microphone, the likelihood ratio based formulation achieved a relative error reduction of 35% in comparison with a likelihood based formulation. In addition, we observed that the use of N-best candidates and the BG and OOV models improved the performance with a relative error reduction of roughly 10%.

This paper introduces a Japanese spontaneous speech database of 3,771 speakers with wide regional and age distributions. This database is designed to capture Japanese spontaneous speech characteristics and is used to develop a speaker-independent (SI) speech recognition system. This paper describes the data collection and transcription. Moreover, we show preliminary analyses through SI speech recognition experiments with the database.

This paper investigates a new method for creating speaker models that are robust against utterance variation in continuous distribution hidden Markov model-based speaker verification. In this method, the distribution of the session-independent features for each speaker is estimated by separately modeling the session-to-session utterance variation as two distinct variations: one session-dependent and the other session-independent. In practice, joint normalization of the session-dependent utterance variation and estimation of the parameters of speaker models is performed based on a speaker adaptive training algorithm. The resulting speaker models more accurately represent session-independent speaker characteristics, and the discriminatory capabilities of these models increases. In text-independent speaker verification experiments using data uttered by 20 speakers in 7 sessions over 16 months, we show that the proposed method achieves a 15% reduction in the error rate.

This paper investigates a method of creating robust speaker models that are not sensitive to session-dependent (SD) utterance-variation and handset-dependent (HD) distortion for hidden Markov model (HMM)-based speaker veri cation systems in a real telephone network. We recently reported a method of creating session-independent (SI) speaker-HMMs that are not sensitive to SD utterance-variation. In that method, the distortion function that transforms SI speaker-HMMs to SD speaker-HMMs is introduced, and the parameters in the function and the speaker-HMM parameters are jointly estimated using a speaker adaptive training algorithm. This paper proposes a new method that is less sensitive to SD utterance-variation and HD distortion than the previous method. This new idea focuses on di erent di culties in estimating parameters in distortion functions for SD utterance-variation and HD distortion. In text-independent veri cation experiments with recorded data from di erent handsets, the error reduction rate of the improved method compared with that of the conventional method of cepstral mean normalization is 24% when each speaker-HMM is recreated using data uttered in ve sessions.

Smoothed estimation and utterance verification are introduced into the N-best-based speaker adaptation method. That method is effective even for speakers whose decodings using speaker-independent (SI) models are error-prone, that is, for speakers for whom adaptation techniques are truly needed. The smoothed estimation improves the performance for such speakers, and the utterance verification reduces the required amount of calculation. Performance evaluation using connected-digit (four-digit strings) recognition experiments performed over actual telephone lines showed a reduction of 36.4% in the error rates for speakers whose decodings using SI models are error-prone. To try and find an effective model-transformation for speaker adaptation, we discuss replacing mixture-mean bias estimation by the widely used mixture-mean linear-regression-matrix estimation.

We describe a method of updating a hidden Markov model (HMM) for speaker verification using a small amount of new data for each speaker. The HMM is updated by adapting the model parameters to the new data by maximum a posteriori (MAP) estimation. The initial values of the a priori parameters in MAP estimation are set using training speech used for first creating a speaker HMM. We also present a method of resetting the a priori threshold as the updating of the model proceeds. Evaluation of the performance of the two methods using 10 male speakers showed that the verification error rate was about 42% of that without updating.

An instantaneous speaker adaptation method is proposed that uses N-best decoding for continuous mixture-density hidden-Markov-model based speech recognition systems.An N-best paradigm of multiple-pass search strategies is used that makes this method eective even for speakers whose decodings using speaker-independent models are error-prone.To cope with an insucient amount of data, our method uses constrained maximum a posteriori estimation, in which the parameter vector space is clustered, and a mixturemean bias is estimated for each cluster.Moreover, to maintain continuity between clusters, a bias for each mixturemean is calculated as the weighted sum of the estimated biases.Performance evaluation using connected-digit (fourdigit strings) recognition experiments performed over actual telephone lines showed more than a 20% reduction in the error rates, even for speakers whose decodings using speakerindependent models were error-prone.

Abstract This paper investigates a speaker recognition method that is robust against background noise. In noisy environments, one important issue is how to create a model for each speaker so as to compensate for noise. The method described here is based on hidden Markov model (HMM) composition, which combines a speaker HMM and a noise-source HMM into a noise-added speaker HMM with a particular signal-to-noise ratio (SNR). Since it is difficult to measure the SNR of input speech with non-stationary noise exactly, this method creates several noise-added speaker HMMs with various SNRs. The HMM that has the highest likelihood value for the input speech is selected, and a speaker decision is made using this likelihood value. Experimental application of this method to text-independent speaker identification and verification in various kinds of noisy environments demonstrated considerable improvement in speaker recognition for speech utterances of male speakers.

For text-independent speaker identification and verification based on the multivariate auto-regression (MAR) model, the authors consider two distance measures: the log likelihood ratio (LLR) and the symmetrized likelihood ratio (SLR) measure, which is a symmetric version of the likelihood ratio measure. The results of experiments indicate that the LLR gives better performance than the SLR for longer training data of 5 or more sentences, and the SLR measure is better for shorter training data. When 10 sentences are used for training, identification and verification rates (after likelihood normalization) are almost the same as those obtained by an HMM-based method. The optimum order of the MAR model is 2 or 3, and the optimum frame period is 16 ms.>

Speaker adaptation methods for tied-mixture-based phoneme models are investigated for text-prompted speaker recognition. For this type of speaker recognition, speaker-specific phoneme models are essential for verifying both the key text and the speaker. This paper proposes a new method of creating speaker-specific phoneme models. This uses speaker-independent (universal) phoneme models consisting of tied-mixture HMMs and adapts the feature space of the tied-mixtures to that of the speaker through phoneme-dependent/independent iterative training. Therefore, it can adapt models of phonemes that have a small amount of training data to the speaker. The proposed method was tested using 15 speakers' voices recorded over 10 months and achieved a speaker and text verification rate of 99.4% even when both the voices of different speakers and different texts uttered by the true speaker were to be rejected.>

This paper compares a VQ (vector quantization)-distortion-based speaker recognition method and discrete/continuous ergodic HMM (hidden Markov model)-based ones, especially from the viewpoint of robustness against utterance variations. The authors show that a continuous ergodic HMM is as robust as a VQ-distortion method when enough data is available and that a continuous ergodic HMM is far superior to a discrete ergodic HMM. They also show that the information on transitions between different states is ineffective for text-independent speaker recognition. Therefore, the speaker recognition rates using a continuous ergodic HMM are strongly correlated with the total number of mixtures irrespective of the number of states.>

A VQ (vector quantization)-distortion-based speaker recognition method and discrete/continuous ergodic HMM (hidden Markov model)-based ones are compared, especially from the viewpoint of robustness against utterance variations. It is shown that a continuous ergodic HMM is far superior to a discrete ergodic HMM. It is also shown that the information on transitions between different states is ineffective for text-independent speaker recognition. Therefore, the speaker identification rates using a continuous ergodic HMM are strongly correlated with the total number of mixtures irrespective of the number of states. It is also found that, for continuous ergodic HMM-based speaker recognition, the distortion-intersection measure (DIM), which was introduced as a VQ-distortion measure to increase the robustness against utterance variations, is effective.>

The authors describe a VQ (vector-quantization)-based text-independent speaker recognition method which is robust against utterance variations. Three techniques are introduced to cope with temporal and text-dependent spectral variations. First, either an ergodic hidden Markov model or a voiced/unvoiced decision is used to classify input speech into broad phonetic classes. Second, a new distance measure, the distortion-intersection measure (DIM), is introduced for calculating VQ distortion of input speech compared to speaker-independent codebooks. Third, a normalization method, talker variability normalization (TVN), is introduced. TVN normalizes parameter variation taking both inter- and intra-speaker variability into consideration. The system was tested using utterances of nine speakers recorded over three years. The combination of the three techniques achieves high speaker identification accuracies of 98.5% using only vocal tract information and 99.0% using both vocal tract and pitch information.>