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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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Estimating and evaluating confidence for forensic speaker recognition

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, Vol. 1, 19-23 March 2005, pp. I-717 - I-720.

Summary

Estimating and evaluating confidence has become a key aspect of the speaker recognition problem because of the increased use of this technology in forensic applications. We discuss evaluation measures for speaker recognition and some of their properties. We then propose a framework for confidence estimation based upon scores and metainformation, such as utterance duration, channel type, and SNR. The framework uses regression techniques with multilayer perceptrons to estimate confidence with a data-driven methodology. As an application, we show the use of the framework in a speaker comparison task drawn from the NIST 2000 evaluation. A relative comparison of different types of meta-information is given. We demonstrate that the new framework can give substantial improvements over standard distribution methods of estimating confidence.
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Summary

Estimating and evaluating confidence has become a key aspect of the speaker recognition problem because of the increased use of this technology in forensic applications. We discuss evaluation measures for speaker recognition and some of their properties. We then propose a framework for confidence estimation based upon scores and metainformation...

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The MIT Lincoln Laboratory RT-04F diarization systems: applications to broadcast audio and telephone conversations

Published in:
NIST Rich Transcription Workshop, 8-11 November 2004.

Summary

Audio diarization is the process of annotating an input audio channel with information that attributes (possibly overlapping) temporal regions of signal energy to their specific sources. These sources can include particular speakers, music, background noise sources, and other signal source/channel characteristics. Diarization has utility in making automatic transcripts more readable and in searching and indexing audio archives. In this paper we describe the systems developed by MITLL and used in DARPA EARS Rich Transcription Fall 2004 (RT-04F) speaker diarization evaluation. The primary system is based on a new proxy speaker model approach and the secondary system follows a more standard BIC based clustering approach. We present experiments analyzing performance of the systems and present a cross-cluster recombination approach that significantly improves performance. In addition, we also present results applying our system to a telephone speech, summed channel speaker detection task.
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Summary

Audio diarization is the process of annotating an input audio channel with information that attributes (possibly overlapping) temporal regions of signal energy to their specific sources. These sources can include particular speakers, music, background noise sources, and other signal source/channel characteristics. Diarization has utility in making automatic transcripts more readable...

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Channel compensation for SVM speaker recognition

Published in:
Odyssey, The Speaker and Language Recognition Workshop, 31 May - 3 June 2004.

Summary

One of the major remaining challenges to improving accuracy in state-of-the-art speaker recognition algorithms is reducing the impact of channel and handset variations on system performance. For Gaussian Mixture Model based speaker recognition systems, a variety of channel-adaptation techniques are known and available for adapting models between different channel conditions, but for the much more recent Support Vector Machine (SVM) based approaches to this problem, much less is known about the best way to handle this issue. In this paper we explore techniques that are specific to the SVM framework in order to derive fully non-linear channel compensations. The result is a system that is less sensitive to specific kinds of labeled channel variations observed in training.
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Summary

One of the major remaining challenges to improving accuracy in state-of-the-art speaker recognition algorithms is reducing the impact of channel and handset variations on system performance. For Gaussian Mixture Model based speaker recognition systems, a variety of channel-adaptation techniques are known and available for adapting models between different channel conditions...

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Fusing discriminative and generative methods for speaker recognition: experiments on switchboard and NFI/TNO field data

Published in:
ODYSSEY 2004, Speaker and Language Recognition Workshop, 31 May - 3 June 2004.

Summary

Discriminatively trained support vector machines have recently been introduced as a novel approach to speaker recognition. Support vector machines (SVMs) have a distinctly different modeling strategy in the speaker recognition problem. The standard Gaussian mixture model (GMM) approach focuses on modeling the probability density of the speaker and the background (a generative approach). In contrast, the SVM models the boundary between the classes. Another interesting aspect of the SVM is that it does not directly produce probabilistic scores. This poses a challenge for combining results with a GMM. We therefore propose strategies for fusing the two approaches. We show that the SVM and GMM are complementary technologies. Recent evaluations by NIST (telephone data) and NFI/TNO (forensic data) give a unique opportunity to test the robustness and viability of fusing GMM and SVM methods. We show that fusion produces a system which can have relative error rates 23% lower than individual systems.
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Summary

Discriminatively trained support vector machines have recently been introduced as a novel approach to speaker recognition. Support vector machines (SVMs) have a distinctly different modeling strategy in the speaker recognition problem. The standard Gaussian mixture model (GMM) approach focuses on modeling the probability density of the speaker and the background...

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Speaker diarisation for broadcast news

Published in:
Odyssey 2004, 31 May - 4 June 2004.

Summary

It is often important to be able to automatically label 'who spoke when' during some audio data. This paper describes two systems for audio segmentation developed at CUED and MIT-LL and evaluates their performance using the speaker diarisation score defined in the 2003 Rich Transcription Evaluation. A new clustering procedure and BIC-based stopping criterion for the CUED system is introduced which improves both performance and robustness to changes in segmentation. Finally a hybrid 'Plug and Play' system is built which combines different parts of the CUED and MIT-LL systems to produce a single system which outperforms both the individual systems.
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Summary

It is often important to be able to automatically label 'who spoke when' during some audio data. This paper describes two systems for audio segmentation developed at CUED and MIT-LL and evaluates their performance using the speaker diarisation score defined in the 2003 Rich Transcription Evaluation. A new clustering procedure...

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The MMSR bilingual and crosschannel corpora for speaker recognition research and evaluation

Summary

We describe efforts to create corpora to support and evaluate systems that meet the challenge of speaker recognition in the face of both channel and language variation. In addition to addressing ongoing evaluation of speaker recognition systems, these corpora are aimed at the bilingual and crosschannel dimensions. We report on specific data collection efforts at the Linguistic Data Consortium, the 2004 speaker recognition evaluation program organized by the National Institute of Standards and Technology (NIST), and the research ongoing at the US Federal Bureau of Investigation and MIT Lincoln Laboratory. We cover the design and requirements, the collections and evaluation integrating discussions of the data preparation, research, technology development and evaluation on a grand scale.
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Summary

We describe efforts to create corpora to support and evaluate systems that meet the challenge of speaker recognition in the face of both channel and language variation. In addition to addressing ongoing evaluation of speaker recognition systems, these corpora are aimed at the bilingual and crosschannel dimensions. We report on...

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The mixer corpus of multilingual, multichannel speaker recognition data

Published in:
Proc. Language Resource Evaluation Conf., LREC, 24-30 May 2004, pp. 627-630.

Summary

This paper describes efforts to create corpora to support and evaluate systems that perform speaker recognition where channel and language may vary. Beyond the ongoing evaluation of speaker recognition systems, these corpora are aimed at the bilingual and cross channel dimensions. We report on specific data collection efforts at the Linguistic Data Consortium and the research ongoing at the US Federal Bureau of Investigation and MIT Lincoln Laboratories. We cover the design and requirements, the collections and final properties of the corpus integrating discussions of the data preparation, research, technology development and evaluation on a grand scale.
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Summary

This paper describes efforts to create corpora to support and evaluate systems that perform speaker recognition where channel and language may vary. Beyond the ongoing evaluation of speaker recognition systems, these corpora are aimed at the bilingual and cross channel dimensions. We report on specific data collection efforts at the...

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Conversational telephone speech corpus collection for the NIST speaker recognition evaluation 2004

Published in:
Proc. Language Resource Evaluation Conf., LREC, 24-30 May 2004, pp. 587-590.

Summary

This paper discusses some of the factors that should be considered when designing a speech corpus collection to be used for text independent speaker recognition evaluation. The factors include telephone handset type, telephone transmission type, language, and (non-telephone) microphone type. The paper describes the design of the new corpus collection being undertaken by the Linguistic Data Consortium (LDC) to support the 2004 and subsequent NIST speech recognition evaluations. Some preliminary information on the resulting 2004 evaluation test set is offered.
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Summary

This paper discusses some of the factors that should be considered when designing a speech corpus collection to be used for text independent speaker recognition evaluation. The factors include telephone handset type, telephone transmission type, language, and (non-telephone) microphone type. The paper describes the design of the new corpus collection...

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Multisensor MELPE using parameter substitution

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 1, 17-21 May 2004, pp. I-477 - I-480.

Summary

The estimation of speech parameters and the intelligibility of speech transmitted through low-rate coders, such as MELP, are severely degraded when there are high levels of acoustic noise in the speaking environment. The application of nonacoustic and nontraditional sensors, which are less sensitive to acoustic noise than the standard microphone, is being investigated as a means to address this problem. Sensors being investigated include the General Electromagnetic Motion Sensor (GEMS) and the Physiological Microphone (P-mic). As an initial effort in this direction, a multisensor MELPe coder using parameter substitution has been developed, where pitch and voicing parameters are obtained from GEMS and PMic sensors, respectively, and the remaining parameters are obtained as usual from a standard acoustic microphone. This parameter substitution technique is shown to produce significant and promising DRT intelligibility improvements over the standard 2400 bps MELPe coder in several high-noise military environments. Further work is in progress aimed at utilizing the nontraditional sensors for additional intelligibility improvements and for more effective lower rate coding in noise.
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Summary

The estimation of speech parameters and the intelligibility of speech transmitted through low-rate coders, such as MELP, are severely degraded when there are high levels of acoustic noise in the speaking environment. The application of nonacoustic and nontraditional sensors, which are less sensitive to acoustic noise than the standard microphone...

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