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The MIT-LL/IBM 2006 speaker recognition system: high-performance reduced-complexity recognition

Published in:
Proc. 32nd IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, April 2007, pp. IV-217 - IV-220.

Summary

Many powerful methods for speaker recognition have been introduced in recent years--high-level features, novel classifiers, and channel compensation methods. A common arena for evaluating these methods has been the NIST speaker recognition evaluation (SRE). In the NIST SRE from 2002-2005, a popular approach was to fuse multiple systems based upon cepstral features and different linguistic tiers of high-level features. With enough enrollment data, this approach produced dramatic error rate reductions and showed conceptually that better performance was attainable. A drawback in this approach is that many high-level systems were being run independently requiring significant computational complexity and resources. In 2006, MIT Lincoln Laboratory focused on a new system architecture which emphasized reduced complexity. This system was a carefully selected mixture of high-level techniques, new classifier methods, and novel channel compensation techniques. This new system has excellent accuracy and has substantially reduced complexity. The performance and computational aspects of the system are detailed on a NIST 2006 SRE task.
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Summary

Many powerful methods for speaker recognition have been introduced in recent years--high-level features, novel classifiers, and channel compensation methods. A common arena for evaluating these methods has been the NIST speaker recognition evaluation (SRE). In the NIST SRE from 2002-2005, a popular approach was to fuse multiple systems based upon...

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SVM based speaker verification using a GMM supervector kernel and NAP variability compensation

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, Speech and Language Processing, ICASSP, Vol. 1, 14-19 May 2006, pp. 97-100.

Summary

Gaussian mixture models with universal backgrounds (UBMs) have become the standard method for speaker recognition. Typically, a speaker model is constructed by MAP adaptation of the means of the UBM. A GMM supervector is constructed by stacking the means of the adapted mixture components. A recent discovery is that latent factor analysis of this GMM supervector is an effective method for variability compensation. We consider this GMM supervector in the context of support vector machines. We construct a support vector machine kernel using the GMM supervector. We show similarities based on this kernel between the method of SVM nuisance attribute projection (NAP) and the recent results in latent factor analysis. Experiments on a NIST SRE 2005 corpus demonstrate the effectiveness of the new technique.
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Summary

Gaussian mixture models with universal backgrounds (UBMs) have become the standard method for speaker recognition. Typically, a speaker model is constructed by MAP adaptation of the means of the UBM. A GMM supervector is constructed by stacking the means of the adapted mixture components. A recent discovery is that latent...

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Support vector machines using GMM supervectors for speaker verification

Published in:
IEEE Signal Process. Lett., Vol. 13, No. 5, May 2006, pp. 308-311.

Summary

Gaussian mixture models (GMMs) have proven extremely successful for text-independent speaker recognition. The standard training method for GMMmodels is to use MAP adaptation of the means of the mixture components based on speech from a target speaker. Recent methods in compensation for speaker and channel variability have proposed the idea of stacking the means of the GMM model to form a GMM mean supervector. We examine the idea of using the GMM supervector in a support vector machine (SVM) classifier. We propose two new SVM kernels based on distance metrics between GMM models. We show that these SVM kernels produce excellent classification accuracy in a NIST speaker recognition evaluation task.
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Summary

Gaussian mixture models (GMMs) have proven extremely successful for text-independent speaker recognition. The standard training method for GMMmodels is to use MAP adaptation of the means of the mixture components based on speech from a target speaker. Recent methods in compensation for speaker and channel variability have proposed the idea...

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Speaker adaptive cohort selection for Tnorm in text-independent speaker verification

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 19-23 March 2005, pp. I-741 - I-744.

Summary

In this paper we discuss an extension to the widely used score normalization technique of test normalization (Tnorm) for text-independent speaker verification. A new method of speaker Adaptive-Tnorm that offers advantages over the standard Tnorm by adjusting the speaker set to the target model is presented. Examples of this improvement using the 2004 NIST SRE data are also presented.
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Summary

In this paper we discuss an extension to the widely used score normalization technique of test normalization (Tnorm) for text-independent speaker verification. A new method of speaker Adaptive-Tnorm that offers advantages over the standard Tnorm by adjusting the speaker set to the target model is presented. Examples of this improvement...

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The 2004 MIT Lincoln Laboratory speaker recognition system

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 19-23 March 2005, pp. I-177 - I-180.

Summary

The MIT Lincoln Laboratory submission for the 2004 NIST Speaker Recognition Evaluation (SRE) was built upon seven core systems using speaker information from short-term acoustics, pitch and duration prosodic behavior, and phoneme and word usage. These different levels of information were modeled and classified using Gaussian Mixture Models, Support Vector Machines and N-gram language models and were combined using a single layer perception fuser. The 2004 SRE used a new multi-lingual, multi-channel speech corpus that provided a challenging speaker detection task for the above systems. In this paper we describe the core systems used and provide an overview of their performance on the 2004 SRE detection tasks.
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Summary

The MIT Lincoln Laboratory submission for the 2004 NIST Speaker Recognition Evaluation (SRE) was built upon seven core systems using speaker information from short-term acoustics, pitch and duration prosodic behavior, and phoneme and word usage. These different levels of information were modeled and classified using Gaussian Mixture Models, Support Vector...

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Automatic dysphonia recognition using biologically-inspired amplitude-modulation features

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 19-23 March 2005, pp. I-873 - I-876.

Summary

A dysphonia, or disorder of the mechanisms of phonation in the larynx, can create time-varying amplitude fluctuations in the voice. A model for band-dependent analysis of this amplitude modulation (AM) phenomenon in dysphonic speech is developed from a traditional communications engineering perspective. This perspective challenges current dysphonia analysis methods that analyze AM in the time-domain signal. An automatic dysphonia recognition system is designed to exploit AM in voice using a biologically-inspired model of the inferior colliculus. This system, built upon a Gaussian-mixture-model (GMM) classification backend, recognizes the presence of dysphonia in the voice signal. Recognition experiments using data obtained from the Kay Elemetrics Voice Disorders Database suggest that the system provides complementary information to state-of-the-art mel-cepstral features. We present dysphonia recognition as an approach to developing features that capture glottal source differences in normal speech.
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Summary

A dysphonia, or disorder of the mechanisms of phonation in the larynx, can create time-varying amplitude fluctuations in the voice. A model for band-dependent analysis of this amplitude modulation (AM) phenomenon in dysphonic speech is developed from a traditional communications engineering perspective. This perspective challenges current dysphonia analysis methods that...

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Beyond cepstra: exploiting high-level information in speaker recognition

Summary

Traditionally speaker recognition techniques have focused on using short-term, low-level acoustic information such as cepstra features extracted over 20-30 ms windows of speech. But speech is a complex behavior conveying more information about the speaker than merely the sounds that are characteristic of his vocal apparatus. This higher-level information includes speaker-specific prosodics, pronunciations, word usage and conversational style. In this paper, we review some of the techniques to extract and apply these sources of high-level information with results from the NIST 2003 Extended Data Task.
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Summary

Traditionally speaker recognition techniques have focused on using short-term, low-level acoustic information such as cepstra features extracted over 20-30 ms windows of speech. But speech is a complex behavior conveying more information about the speaker than merely the sounds that are characteristic of his vocal apparatus. This higher-level information includes...

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Auditory signal processing as a basis for speaker recognition

Published in:
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 19-22 October, 2003, pp. 111-114.

Summary

In this paper, we exploit models of auditory signal processing at different levels along the auditory pathway for use in speaker recognition. A low-level nonlinear model, at the cochlea, provides accentuated signal dynamics, while a a high-level model, at the inferior colliculus, provides frequency analysis of modulation components that reveals additional temporal structure. A variety of features are derived from the low-level dynamic and high-level modulation signals. Fusion of likelihood scores from feature sets at different auditory levels with scores from standard mel-cepstral features provides an encouraging speaker recognition performance gain over use of the mel-cepstrum alone with corpora from land-line and cellular telephone communications.
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Summary

In this paper, we exploit models of auditory signal processing at different levels along the auditory pathway for use in speaker recognition. A low-level nonlinear model, at the cochlea, provides accentuated signal dynamics, while a a high-level model, at the inferior colliculus, provides frequency analysis of modulation components that reveals...

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Speaker verification using text-constrained Gaussian mixture models

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. I, 13-17 May 2002, pp. I-677 - I-680.

Summary

In this paper we present an approach to close the gap between text-dependent and text-independent speaker verification performance. Text-constrained GMM-UBM systems are created using word segmentations produced by a LVCSR system on conversational speech allowing the system to focus on speaker differences over a constrained set of acoustic units. Results on the 2001 NiST extended data task show this approach can be used to produce an equal error rate of < 1%.
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Summary

In this paper we present an approach to close the gap between text-dependent and text-independent speaker verification performance. Text-constrained GMM-UBM systems are created using word segmentations produced by a LVCSR system on conversational speech allowing the system to focus on speaker differences over a constrained set of acoustic units. Results...

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Speaker indexing in large audio databases using anchor models

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 7-11 May 2001, pp. 429-432.

Summary

This paper introduces the technique of anchor modeling in the applications of speaker detection and speaker indexing. The anchor modeling algorithm is refined by pruning the number of models needed. The system is applied to the speaker detection problem where its performance is shown to fall short of the state-of-the-art Gaussian Mixture Model with Universal Background Model (GMM-UBM) system. However, it is further shown that its computational efficiency lends itself to speaker indexing for searching large audio databases for desired speakers. Here, excessive computation may prohibit the use of the GMM-UBM recognition system. Finally, the paper presents a method for cascading anchor model and GMM-UBM detectors for speaker indexing. This approach benefits from the efficiency of anchor modeling and high accuracy of GMM-UBM recognition.
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Summary

This paper introduces the technique of anchor modeling in the applications of speaker detection and speaker indexing. The anchor modeling algorithm is refined by pruning the number of models needed. The system is applied to the speaker detection problem where its performance is shown to fall short of the state-of-the-art...

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