Speaker recognition performance degrades substantially in case of vocal effort mismatch (e.g. shouted vs. normal speech) between test and enrollment utterances. Such a mismatch is often encountered, for example, in forensic speaker recognition. This paper introduces a novel spectral mapping method which, when employed jointly with a statistical mapping technique, converts the Mel-frequency band energies of normal speech towards their counterparts in shouted speech. The aim is to obtain more robust performance in speaker recognition by tackling vocal effort mismatch between enrollment and test utterances. The processing is performed on the speech signal before feature extraction. The proposed approach was evaluated by testing the performance of a state-of-the-art i-vector-based speaker recognition system with and without applying the spectral mapping processing to the enrollment data. The results show that pre-processing with the proposed approach results in considerable improvement in correct identification rates.

18th Annual Conference of the International Speech Communication Association, INTERSPEECH 2017, Stockholm, Sweden, 20-24 August 2017

A linear predictive spectral estimation method based on higher-lag autocorrelation coefficients is proposed for the noise-robust feature extraction from speech. The method, called higher-lag linear prediction, is derived from a signal prediction model that is optimized in the mean square sense using a cost function that has two prediction error terms, the first of which is similar to that of conventional linear prediction and the second of which is a delayed version introducing an integer delay of M samples. This basic form is developed further into the combined higher-lag linear prediction (CHLLP) model by simultaneously taking advantage of the zero-lag and higher-lag predictions. The CHLLP model was used in the computation of mel-frequency cepstral coefficients and compared with several reference feature extraction methods in speaker recognition. The experiments were conducted by using a modern i-vector-based system. Noise-corruption was done using both additive car, babble, and factory noise in different signal-to-noise ratio conditions as well as speech recordings from real noisy conditions. The results indicate that CHLLP outperformed the reference feature extraction methods in almost all the comparisons in the noise-corrupted conditions and the performance of CHLLP was only slightly inferior to the nonparametric FFT-based spectral modeling in the clean condition.

Linear discriminant analysis (LDA) is a powerful technique in pattern recognition to reduce the dimensionality of data vectors. It maximizes discriminability by retaining only those directions that minimize the ratio of within-class and between-class variance. In this paper, using the same principles as for conventional LDA, we propose to employ uncertainties of the noisy or distorted input data in order to estimate maximally discriminant directions. We demonstrate the efficiency of the proposed uncertain LDA on two applications using state-of-the-art techniques. First, we experiment with an automatic speech recognition task, in which the uncertainty of observations is imposed by real-world additive noise. Next, we examine a full-scale speaker recognition system, considering the utterance duration as the source of uncertainty in authenticating a speaker. The experimental results show that when employing an appropriate uncertainty estimation algorithm, uncertain LDA outperforms its conventional LDA counterpart.

Linear prediction is one of the most established techniques in signal estimation, and it is widely utilized in speech signal processing. It has been long understood that the nerve firing rate of human auditory system can be approximated by power law non-linearity, and this has been the motivation behind using perceptual linear prediction in extracting acoustic features in a variety of speech processing applications. In this paper, we revisit the application of power law non-linearity in speech spectrum estimation by compressing/expanding power spectrum in autocorrelation-based linear prediction. The development of so-called LP- α is motivated by a desire to obtain spectral features that present less mismatch than conventionally used spectrum estimation methods when speech of normal loudness is compared to speech under vocal effort. The effectiveness of the proposed approach is demonstrated in a speaker recognition task conducted under severe vocal effort mismatch comparing shouted versus normal speech mode.

Non-negative matrix factorisations are used in several branches of signal processing and data analysis for separation and classification. Sparsity constraints are commonly set on the model to promote discovery of a small number of dominant patterns. In group sparse models, atoms considered to belong to a consistent group are permitted to activate together, while activations across groups are suppressed, reducing the number of simultaneously active sources or other structures. Whereas most group sparse models require explicit division of atoms into separate groups without addressing their mutual relations, we propose a constraint that permits dynamic relationships between atoms or groups, based on any defined distance measure. The resulting solutions promote approximation with components considered similar to each other. Evaluation results are shown for speech enhancement and noise robust speech and speaker recognition.

The vast majority of speaker recognition cross-entropy evaluations are focused on score domain. By examining the generalized relative distance between genuine and impostor sub-spaces, biometric characteristics become comparable to other authentication approaches. In this paper we demonstrate that the i-vector feature space's biometric information measured by relative entropy is comparable to e.g., knowledge-based mechanisms or face recognition. Examining NIST SRE 2004-2010 corpora, short samples of e.g, 5 seconds duration, comprise already 127 bits in a text-independent scenario. Further, the vast majority of short samples does not fall below 50% of the biometric information of samples having a duration of more than 40 seconds. The generalized i-vector feature space entropy of long samples corresponds to 182.1 bits, and the highest lower entropy bound of a subject was observed at 471.6 bits.

Recognition and classification of speech content in everyday environments is challenging due to the large diversity of real-world noise sources, which may also include competing speech. At signal-to-noise ratios below 0 dB, a majority of features may become corrupted, severely degrading the performance of clas-sifiers built upon clean observations of a target class. As the energy and complexity of competing sources increase, their ex-plicit modelling becomes integral for successful detection and classification of target speech. We have previously demon-strated how non-negative compositional modelling in a spec-trogram space is suitable for robust recognition of speech and speakers even at low SNRs. In this work, the sparse coding approach is extended to cover the whole separation and clas-sification chain to recognise the speaker of short utterances in difficult noise environments. A convolutive matrix factorisation and coding system is evaluated on 2nd CHiME Track 1 data. Over 98 % average speaker recognition accuracy is achieved for shorter than three second utterances at +9...-6 dB SNR, illus-trating the system’s performance in challenging conditions. Index Terms: speaker recognition, noise robustness, composi-tional models, sparse coding, non-negative matrix factorization

One of the biggest challenges in speaker recognition is incom-plete observations in test phase caused by availability of only short duration utterances. The problem with short utterances is that speaker recognition needs to be handled by having in-formation from only limited amount of acoustic classes. By considering limited observations from a test speaker, the re-sulting i-vector as a representative of short utterance will be uncertain; the shorter the duration, the higher the uncertainty. In recent studies, an uncertainty decoding technique has been employed in probabilistic linear discriminant analysis (PLDA) modeling in order to account for uncertain i-vectors. In this paper, we propose to extend uncertainty handling using simpli-fied PLDA scoring and modified imputation. We experiment with a state-of-the-art speaker recognition system focusing on uncertainty caused by controlled utterance duration. The uncer-tainties after i-vector extraction are being propagated through pre-processing steps and both uncertainty decoding and modi-fied imputation are considered. Our experimental results indi-cate improved equal error rate and detection cost attained by us-ing uncertainty-of-observation techniques in dealing with short duration utterances. Index Terms: speaker verification, duration, uncertainty de-coding, modified imputation

Speech under face cover constitute a case that is increasingly met by forensic speech experts. Wearing face cover mostly hap-pens when an individual strives to conceal his or her identity. Based on the material of face cover and the level of contact with speech production organs, speech production becomes affected by face mask and a part of speech energy gets absorbed in the mask. There has been little research on how speech acoustics is affected by different face masks and how face covers might affect performance of automatic speaker recognition systems. In the present paper, we have collected speech under face mask with the aim of studying the effects of wearing different masks on state-of-the-art text-independent automatic speaker recogni-tion system. The preliminary speaker recognition rates along with mask identification experiments are presented in this pa-per.

Assessing the impact of ageing on biometric systems is an important challenge. In this paper, an i-vector speaker verifi-cation framework is used to evaluate the impact of long-term ageing on state-of-the-art speaker verification. Using the Trin-ity College Dublin Speaker Ageing (TCDSA) database, it is ob-served that the performance of the i-vector system, in terms of both discrimination and calibration, degrades progressively as the absolute age difference between training and testing sam-ples increases. In the case of male speakers, the equal error rate (EER) increases from 4.61 % at an ageing difference of 0–1 years to 32.74 % at an age difference of 51–60 years. The performance of a Gaussian Mixture Model- Universal Back-ground Model (GMM-UBM) system is presented for compari-son. It is shown that while the i-vector system outperforms the GMM-UBM system, as absolute age difference increases, the performance of both degrades at a similar rate. It is concluded that long-term ageing variability is distinct from everyday inter-session variability, and therefore must be dealt with via dedi-cated compensation strategies.

Abstract: We attack the problem of trajectory similarity by approximating the trajectories using a geographical grid based on the MGRS 2D coordinate system. We propose a spatial similarity measure which is computationally feasible for big data collections. The proposed measure is based on cell matching with a similarity metric drawn from Jaccard index. We equip the proposed method with interpolation and dilation to overcome the problems missing data and different sampling frequencies when comparing two trajectories. The proposed measure is implemented online in the framework of Mopsia. acs.uef.fi/mopsi 1

In this paper, three utterance modelling approaches, namely Gaussian Mean Supervector (GMS), i-vector and Gaussian Posterior Probability Supervector (GPPS), are applied to the accent recognition problem. For each utterance modeling method, three different classifiers, namely the Support Vector Machine (SVM), the Naive Bayesian Classifier (NBC) and the Sparse Representation Classifier (SRC), are employed to find out suitable matches between the utterance modelling schemes and the classifiers. The evaluation database is formed by using English utterances of speakers whose native languages are Russian, Hindi, American English, Thai, Vietnamese and Cantonese. These utterances are drawn from the National Institute of Standards and Technology (NIST) 2008 Speaker Recognition Evaluation (SRE) database. The study results show that GPPS and i-vector are more effective than GMS in this accent recognition task. It is also concluded that among the employed classifiers, the best matches for i-vector and GPPS are SVM and SRC, respectively.

Text-independent speaker identification is studied using neutral and shouted speech in Finnish to analyze the effect of vocal mode mismatch between training and test utterances. Standard mel-frequency cepstral coefficient (MFCC) features with Gaussian mixture model (GMM) recognizer are used for speaker identification. The results indicate that speaker identification accuracy reduces from perfect (100 %) to 8.71 % under vocal mode mismatch. Because of this dramatic degradation in recognition accuracy, we propose to use a joint density GMM mapping technique for compensating the MFCC features. This mapping is trained on a disjoint emotional speech corpus to create a completely speaker- and speech mode independent emotion-neutralizing mapping. As a result of the compensation, the 8.71 % identification accuracy increases to 32.00 % without degrading the non-mismatched train-test conditions much.

Speaker recognition systems trained on long duration utterances are known to perform significantly worse when short test segments are encountered. To address this mismatch, we analyze the effect of duration variability on phoneme distributions of speech utterances and i-vector length. We demonstrate that, as utterance duration is decreased, number of detected unique phonemes and i-vector length approaches zero in a logarithmic and non-linear fashion, respectively. Assuming duration variability as an additive noise in the i-vector space, we propose three different strategies for its compensation: i) multi-duration training in Probabilistic Linear Discriminant Analysis (PLDA) model, ii) score calibration using log duration as a Quality Measure Function (QMF), and iii) multi-duration PLDA training with synthesized short duration i-vectors. Experiments are designed based on the 2012 National Institute of Standards and Technology (NIST) Speaker Recognition Evaluation (SRE) protocol with varying test utterance duration. Experimental results demonstrate the effectiveness of the proposed schemes on short duration test conditions, especially with the QMF calibration approach.

Inspired by the NIST SRE-2012 evaluation conditions we train the PLDA classifier in an i-vector speaker recognition system with different speaker populations, either including or excluding the target speakers in the evaluation. Including the target speakers in the PLDA training is always beneficial w.r.t. completely excluding them-which is the normal situation in pre-2012 SRE protocols-even in the Pknown= 0 evaluation condition. However, adding other speakers than just the targets speakers can slightly increase performance. We also investigated the effect of adding i-vectors extracted from segments with added noise in the PLDA training. This generally makes the system more robust to noise in the test segments, and doesn't hurt performance in the clean condition. The paper further details the 'simple to compound' log-likelihood-ratio conversion necessary for SRE-2012 style calibration.

In this paper, we present a novel system for enhancing a target speech corrupted in a non-stationary real-life noise scenario. The proposed system consists of one spatial beamformer based on GCC-PHAT-estimated time-delay of arrival followed by three postfilters applied in a sequential way, namely: Wiener filter, minimum mean square error estimator (MMSE) of the log-amplitude, and a model-driven postfilter (MDP) that relies on particular speech signal statistics captured by target speaker Gaussian mixture model. The beamformer accounts for the directional interferences while the MMSE speech enhancement suppresses the stationary background noise, and MDP contributes to suppress the non-stationary sources from the binaural mixture. In our evaluation, multiple objective quality metrics are used to report the speech enhancement and separation performance, averaged on the CHiME development set. The proposed system performs better than standard state-of-the-art techniques and shows comparable performance with other systems submitted to the CHiME challenge. More precisely, it is successful in suppressing the non-stationary interfering sources at different SNR levels supported by the relatively high scores for signal-to-interference-ratio.

In this paper, we study the impact of exploiting the spectral phase information to further improve the speech quality of the single-channel speech enhancement algorithms. In particular, we focus on the two required steps in a typical single-channel speech enhancement system, namely: parameter estimation solved by a minimum mean square error (MMSE) estimator of the speech spectral amplitude, followed by signal reconstruction stage, where the observed noisy phase is often used. For the parameter estimation stage, in contrast to conventional Wiener filter, a new MMSE estimator is derived which takes into account the clean phase information as a prior information. In our experiments, we show that by including the phase information in the two steps, it is possible to improve the perceived signal quality of the enhanced signal significantly with respect to the methods that do not employ the phase information.

\n Contains fulltext :\n 116325.pdf (author's version ) (Open Access)\n

I4U is a joint entry of nine research Institutes and Universities across 4 continents to NIST SRE 2012. It started with a brief discussion during the Odyssey 2012 workshop in Singapore. An online discussion group was soon set up, providing a discussion platform for different issues surrounding NIST SRE’12. Noisy test segments, uneven multi-session training, variable enrollment duration, and the issue of open-set identification were actively discussed leading to various solutions integrated to the I4U submission. The joint submission and several of its 17 sub-systems were among top-performing systems. We summarize the lessons learnt from this large-scale effort.

Automatic speech recognition (ASR) performance suffers severely from non-stationary noise, precluding widespread use of ASR in natural environments. Recently, so-termed uncertainty-of-observation techniques have helped to recover good performance. These consider the clean speech features as a hidden variable, of which the observable features are only an imperfect estimate. An estimated error variance of features is therefore used to further guide recognition. Based on the same idea, we introduce a new strategy: Reducing the speech feature dimensionality for optimal discriminance under observation uncertainty can yield significantly improved recognition performance, and is derived easily via Fisher's criterion of discriminant analysis.

This paper investigates the effect of utterance duration to the calibration of a modern i-vector speaker recognition system with probabilistic linear discriminant analysis (PLDA) modeling. A calibration approach to deal with these effects using quality measure functions (QMFs) is proposed to include duration in the calibration transformation. Extensive experiments are performed in order to evaluate the robustness of the proposed calibration approach for unseen conditions in the training of calibration parameters. Using the latest NIST corpora for evaluation, results highlight the importance of considering the quality metrics like duration in calibrating the scores for automatic speaker recognition systems.

Different short-term spectrum estimators for speaker verification under additive noise are considered. Conventionally, mel-frequency cepstral coefficients (MFCCs) are computed from discrete Fourier transform (DFT) spectra of windowed speech frames. Recently, linear prediction (LP) and its temporally weighted variants have been substituted as the spectrum analysis method in speech and speaker recognition. In this paper, 12 different short-term spectrum estimation methods are compared for speaker verification under additive noise contamination. Experimental results conducted on NIST 2002 SRE show that the spectrum estimation method has a large effect on recognition performance and stabilized weighted LP (SWLP) and minimum variance distortionless response (MVDR) methods yield approximately 7 % and 8 % relative improvements over the standard DFT method at −10 dB SNR level of factory and babble noises, respectively in terms of equal error rate (EER).

Previous studies on performance evaluation of single-channel speech separation (SCSS) algorithms mostly focused on automatic speech recognition (ASR) accuracy as their performance measure. Assessing the separated signals by different metrics other than this has the benefit that the results are expected to carry on to other applications beyond ASR. In this paper, in addition to conventional speech quality metrics (PESQ and SNRloss), we also evaluate the separation systems output using different source separation metrics: blind source separation evaluation (BSS EVAL) and perceptual evaluation methods for audio source separation (PEASS) measures. In our experiments, we apply these measures on the separated signals obtained by two well-known systems in the SCSS challenge to assess the objective and subjective quality of their output signals. Comparing subjective and objective measurements shows that PESQ and PEASS quality metrics predict well the subjective quality of separated signals obtained by the separation systems. From the results it is observed that the short-time objective intelligibility (STOI) measure predict the speech intelligibility results.

Spectrogram factorisation using a dictionary of spectro-temporal atoms has been successfully employed to separate a mixed audio signal into its source components. When atoms from multiple sources are included in a combined dictionary, the relative weights of activated atoms reveal likely sources as well as the content of each source. Enforcing sparsity on the activation weights produces solutions, where only a small number of atoms are active at a time. In this paper we pro-pose using group sparsity to restrict simultaneous activation of sources, allowing us to discover the identity of an unknown speaker from multiple candidates, and further to recognise the phonetic content more reliably with a narrowed down subset of atoms belonging to the most likely speakers. An evalua-tion on the CHiME corpus shows that the use of group sparsity improves the results of noise robust speaker identification and speech recognition using speaker-dependent models. Index Terms: group sparsity, speech recognition, speaker iden-tification, spectrogram factorization

Single-channel speech separation algorithms frequently ignore the issue of accurate phase estimation while reconstructing the enhanced signal. Instead, they directly employ the mixed-signal phase for signal reconstruction which leads to undesired traces of the interfering source in the target signal. In this paper, assuming a given knowledge of signal spectrum amplitude, we present a solution to estimate the phase information for signal reconstruction of the sources from a single-channel mixture observation. We first investigate the effectiveness of the proposed phase estimation method employing known magnitude spectra of sources as an ideal case. We further relax the ideal signal spectra assumption by perturbing the clean signal spectra via Gaussian noise. The results show that for both scenarios, ideal and noisy magnitude signal spectra, the proposed phase estimation approach offers improved signal reconstruction accuracy, segmental SNR and PESQ compared to benchmark methods, and those neglecting the phase information.

Regularization of linear prediction based mel-frequency cepstral coefficient (MFCC) extraction in speaker verification is considered. Commonly, MFCCs are extracted from the discrete Fourier transform (DFT) spectrum of speech frames. In this paper, DFT spectrum estimate is replaced with the recently proposed regularized linear prediction (RLP) method. Regularization of temporally weighted variants, weighted LP (WLP) and stabilized WLP (SWLP) which have earlier shown success in speech and speaker recognition, is also introduced. A novel type of double autocorrelation (DAC) lag windowing is also proposed to enhance robustness. Experiments on the NIST 2002 corpus indicate that regularized all-pole methods (RLP, RWLP and RSWLP) yield large improvement on recognition accuracy under additive factory and babble noise conditions in terms of both equal error rate (EER) and minimum detection cost function (MinDCF).

In speech and audio applications, short-term signal spectrum is often represented using mel-frequency cepstral coefficients (MFCCs) computed from a windowed discrete Fourier transform (DFT). Windowing reduces spectral leakage but variance of the spectrum estimate remains high. An elegant extension to windowed DFT is the so-called multitaper method which uses multiple time-domain windows (tapers) with frequency-domain averaging. Multitapers have received little attention in speech processing even though they produce low-variance features. In this paper, we propose the multitaper method for MFCC extraction with a practical focus. We provide, first, detailed statistical analysis of MFCC bias and variance using autoregressive process simulations on the TIMIT corpus. For speaker verification experiments on the NIST 2002 and 2008 SRE corpora, we consider three Gaussian mixture model based classifiers with universal background model (GMM-UBM), support vector machine (GMM-SVM) and joint factor analysis (GMM-JFA). Multitapers improve MinDCF over the baseline windowed DFT by relative 20.4% (GMM-SVM) and 13.7% (GMM-JFA) on the interview-interview condition in NIST 2008. The GMM-JFA system further reduces MinDCF by 18.7% on the telephone data. With these improvements and generally noncritical parameter selection, multitaper MFCCs are a viable candidate for replacing the conventional MFCCs.

In this paper, we present a novel system for joint speaker identification and speech separation. For speaker identification a single-channel speaker identification algorithm is proposed which provides an estimate of signal-to-signal ratio (SSR) as a by-product. For speech separation, we propose a sinusoidal model-based algorithm. The speech separation algorithm consists of a double-talk/single-talk detector followed by a minimum mean square error estimator of sinusoidal parameters for finding optimal codevectors from pre-trained speaker codebooks. In evaluating the proposed system, we start from a situation where we have prior information of codebook indices, speaker identities and SSR-level, and then, by relaxing these assumptions one by one, we demonstrate the efficiency of the proposed fully blind system. In contrast to previous studies that mostly focus on automatic speech recognition (ASR) accuracy, here, we report the objective and subjective results as well. The results show that the proposed system performs as well as the best of the state-of-the-art in terms of perceived quality while its performance in terms of speaker identification and automatic speech recognition results are generally lower. It outperforms the state-of-the-art in terms of intelligibility showing that the ASR results are not conclusive. The proposed method achieves on average, 52.3% ASR accuracy, 41.2 points in MUSHRA and 85.9% in speech intelligibility.

Best language recognition performance is commonly obtained by fusing the scores of several heterogeneous systems. Regardless the fusion approach, it is assumed that different systems may contribute complementary information, either because they are developed on different datasets, or because they use different features or different modeling approaches. Most authors apply fusion as a final resource for improving performance based on an existing set of systems. Though relative performance gains decrease as larger sets of systems are considered, best performance is usually attained by fusing all the available systems, which may lead to high computational costs. In this paper, we aim to discover which technologies combine the best through fusion and to analyse the factors (data, features, modeling methodologies, etc.) that may explain such a good performance. Results are presented and discussed for a number of systems provided by the participating sites and the organizing team of the Albayzin 2010 Language Recognition Evaluation. We hope the conclusions of this work help research groups make better decisions in developing language recognition technology.

Most of the single-channel speech separation (SCSS) systems use the short-time Fourier transform as their parametric features. Recent studies have shown that employing sinusoidal features for the SCSS application results in a high perceived speech quality. In this paper, we make a systematic study on automatic speech recognition results for a SCSS system that uses sinusoidal features composed of amplitude and frequency. We compare the speech recognition results with those already reported by other participants in the single-channel speech separation and recognition challenge. Our results show that a newly proposed system achieves an overall recognition accuracy of 52.3%, ranges at the median over all other participants in the challenge. Index Terms: sinusoidal modeling, single-channel speech separation and recognition challenge.

In this paper, we propose a closed loop system to improve the performance of single-channel speech separation in a speaker independent scenario. The system is composed of two interconnected blocks: a separation block and a speaker identification block. The improvement is accomplished by incorporating the speaker identities found by the speaker identification block as additional information for the separation block, which converts the speaker-independent separation problem to a speaker-dependent one where the speaker codebooks are known. Simulation results show that the closed loop system enhances the quality of the separated output signals. To assess the improvements, the results are reported in terms of PESQ for both target and masked signals.

Gaussian selection is a technique applied in the GMM-UBM framework to accelerate score calculation. We have recently introduced a novel Gaussian selection method known as sorted GMM (SGMM). SGMM uses scalar-indexing of the universal background model mean vectors to achieve fast search of the top-scoring Gaussians. In the present work we extend this method by using 2-dimensional indexing, which leads to simultaneous frame and Gaussian selection. Our results on the NIST 2002 speaker recognition evaluation corpus indicate that both the 1- and 2- dimensional SGMMs outperform frame decimation and temporal tracking of top-scoring Gaussians by a wide margin (in terms of Gaussian computations relative to GMM-UBM as baseline).

In this paper, we consider speaker identification for the co-channel scenario in which speech mixture from speakers is recorded by one microphone only. The goal is to identify both of the speakers from their mixed signal. High recognition accuracies have already been reported when an accurately estimated signal-to-signal ratio (SSR) is available. In this paper, we approach the problem without estimating SSR. We show that a simple method based on fusion of adapted Gaussian mixture models and Kullback-Leibler divergence calculated between models, achieves an accuracy of 97% and 93% when the two target speakers enlisted as three and two most probable speakers, respectively.

Usually the mel-frequency cepstral coefficients (MFCCs) are derived via Hamming windowed DFT spectrum. In this paper, we advocate to use a so-called multitaper method instead. Mul-titaper methods form a spectrum estimate using multiple win-dow functions and frequency-domain averaging. Multitapers provide a robust spectrum estimate but have not received much attention in speech processing. Our speaker recognition exper-iment on NIST 2002 yields equal error rates (EERs) of 9.66 % (clean data) and 16.41 % (-10 dB SNR) for the conventional Hamming method and 8.13 % (clean data) and 14.63 % (-10 dB SNR) using multitapers. Multitapering is a simple and robust alternative to the Hamming window method. Index Terms: speaker verification, multiple window method 1.

This paper introduces a generalized formulation of linear prediction (LP), including both conventional and temporally weighted LP analysis methods as special cases. The temporally weighted methods have recently been successfully applied to noise robust spectrum analysis in speech and speaker recognition applications. In comparison to those earlier methods, the new generalized approach allows more versatility in weighting different parts of the data in the LP analysis. Two such weighted methods are evaluated and compared to the conventional spectrum modeling methods FFT and LP, as well as the temporally weighted methods WLP and SWLP, by substituting each of them in turn as the spectrum estimation method of the MFCC feature extraction stage of a GMM-UBM based speaker verification system. The new methods are shown to lead to performance improvement in several cases involving channel distortion and additive noise mismatch between the training and recognition conditions. Index Terms: linear prediction, speaker verification, mel frequency cepstral coefficients

This paper describes a novel approach to improve monoaural speaker identification where two speakers are present in a single-microphone recording. The goal is to identify both of the underlying speakers in the given mixture. The proposed approach is composed of a double-talk detector (DTD) as a preprocessor and speaker identification back-end. We demonstrate that including the double-talk detector improves the speaker identification accuracy. Experiments on GRID corpus show that including the DTD improves average recognition accuracy from 96.53% to 97.43%.

Text-independent speaker verification under additive noise corruption is considered. In the popular mel-frequency cepstral coefficient (MFCC) front-end, the conventional Fourier-based spectrum estimation is substituted with weighted linear predictive methods, which have earlier shown success in noise-robust speech recognition. Two temporally weighted variants of linear predictive modeling are introduced to speaker verification and they are compared to FFT, which is normally used in computing MFCCs, and to conventional linear prediction. The effect of speech enhancement (spectral subtraction) on the system performance with each of the four feature representations is also investigated. Experiments by the authors on the NIST 2002 SRE corpus indicate that the accuracy of the conventional and proposed features are close to each other on clean data. For factory noise at 0 dB SNR level, baseline FFT and the better of the proposed features give EERs of 17.4% and 15.6%, respectively. These accuracies improve to 11.6% and 11.2%, respectively, when spectral subtraction is included as a preprocessing method. The new features hold a promise for noise-robust speaker verification.

Usually the mel-frequency cepstral coefficients are estimated either from a periodogram or from a windowed periodogram. We state a general estimator which also includes multitaper estimators. We propose approximations of the variance and bias of the estimate of each coefficient. By using Monte Carlo computations, we demonstrate that the approximations are accurate. Using the proposed formulas, the peak matched multitaper estimator is shown to have low mean square error (squared bias + variance) on speech-like processes. It is also shown to perform slightly better in the NIST 2006 speaker verification task as compared to the Hamming window conventionally used in this context.

Recently, we introduced the sorted Gaussian mixture models (SGMMs) algorithm providing the means to tradeoff performance for operational speed and thus permitting the speed-up of GMM-based classification schemes. The performance of the SGMM algorithm depends on the proper choice of the sorting function, and the proper adjustment of its parameters. In the present work, we employ particle swarm optimization (PSO) and an appropriate fitness function to find the most advantageous parameters of the sorting function. We evaluate the practical significance of our approach on the text-independent speaker verification task utilizing the NIST 2002 speaker recognition evaluation (SRE) database while following the NIST SRE experimental protocol. The experimental results demonstrate a superior performance of the SGMM algorithm using PSO when compared to the original SGMM. For comprehensiveness we also compared these results with those from a baseline Gaussian mixture model–universal background model (GMM-UBM) system. The experimental results suggest that the performance loss due to speed-up is partially mitigated using PSO-derived weights in a sorted GMM-based scheme.

In this paper a new inter-frame fast scoring scheme is proposed for Gaussian mixture model universal background model (GMM-UBM) speaker verification systems. It is combined with a recently introduced intra-frame efficient scoring method called the sorted Gaussian mixture model (SGMM) classifier which itself uses a sorted UBM known as the sorted background model (SBM). To enhance the performance of the system a GMM identifier is applied as a post-processing block. Experimental results show that the performance of this combined method compares favorably with the baseline GMM-UBM system, while the computational load of the proposed system is greatly less than that of the baseline system.

In this paper we propose a new segmentation algorithm called delta MFCC based speech segmentation (DMFCC-SS), with application to speaker recognition systems. We show that DMFCC-SS can separate the regions of speech that result from similar likelihood scores using models such as a Gaussian mixture model (GMM), and can therefore be used to identify the regions of speech between two transitional states in a speech signal. By combining this segmentation algorithm with the discriminative power of transient frames in speaker recognition, we can investigate the tradeoff in speed-up rates that result from DMFCC-SS, with speaker verification equal error rates that result from representatives of each segment. We use a universal background model Gaussian mixture model (UBM-GMM) as a baseline system. The proposed speed-up algorithm, working in the pre-processing stage, performs well while having no computational load compared to the main GMM system. Experimental results show the superior performance of this pre-processing method in comparison with other algorithms working in the pre-processing stage of a UBM-GMM system.

In this paper a Gaussian mixture model (GMM) classifier, called GMM identifier, is proposed as an efficient post-processing method to enhance the performance of a GMM-based speaker verification system; such as Gaussian mixture model universal background model (GMM-UBM) and structural Gaussian mixture models with structural background model (SGMM-SBM) speaker verification schemes. The proposed classifier shows good performance while its computational load is almost negligible compared to the main GMM system. Experimental results show the superior performance of this post-processing method in comparison with a neural-network post-processor for such applications