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Automatic language identification

Published in:
Wiley Encyclopedia of Electrical and Electronics Engineering, Vol. 2, pp. 104-9, 2007.

Summary

Automatic language identification is the process by which the language of digitized spoken words is recognized by a computer. It is one of several processes in which information is extracted automatically from a speech signal.
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Summary

Automatic language identification is the process by which the language of digitized spoken words is recognized by a computer. It is one of several processes in which information is extracted automatically from a speech signal.

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Low-bit-rate speech coding

Author:
Published in:
Chapter 16 in Springer Handbook of Speech Processing and Communication, 2007, pp. 331-50.

Summary

Low-bit-rate speech coding, at rates below 4 kb/s, is needed for both communication and voice storage applications. At such low rates, full encoding of the speech waveform is not possible; therefore, low-rate coders rely instead on parametric models to represent only the most perceptually relevant aspects of speech. While there are a number of different approaches for this modeling, all can be related to the basic linear model of speech production, where an excitation signal drives a vocal-tract filter. The basic properties of the speech signal and of human speech perception can explain the principles of parametric speech coding as applied in early vocoders. Current speech modeling approaches, such as mixed excitation linear prediction, sinusoidal coding, and waveform interpolation, use more-sophisticated versions of these same concepts. Modern techniques for encoding the model parameters, in particular using the theory of vector quantization, allow the encoding of the model information with very few bits per speech frame. Successful standardization of low-rate coders has enabled their widespread use for both military and satellite communications, at rates from 4 kb/s all the way down to 600 b/s. However, the goal of toll-quality low-rate coding continues to provide a research challenge.
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Summary

Low-bit-rate speech coding, at rates below 4 kb/s, is needed for both communication and voice storage applications. At such low rates, full encoding of the speech waveform is not possible; therefore, low-rate coders rely instead on parametric models to represent only the most perceptually relevant aspects of speech. While there...

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Nuisance attribute projection

Published in:
Chapter in Speech Communication, May 2007.

Summary

Cross-channel degradation is one of the significant challenges facing speaker recognition systems. We study this problem in the support vector machine (SVM) context and nuisance variable compensation in high-dimensional spaces more generally. We present an approach to nuisance variable compensation by removing nuisance attribute-related dimensions in the SVM expansion space via projections. Training to remove these dimensions is accomplished via an eigenvalue problem. The eigenvalue problem attempts to reduce multisession variation for the same speaker, reduce different channel effects, and increase "distance" between different speakers. Experiments show significant improvement in performance for the cross-channel case.
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Summary

Cross-channel degradation is one of the significant challenges facing speaker recognition systems. We study this problem in the support vector machine (SVM) context and nuisance variable compensation in high-dimensional spaces more generally. We present an approach to nuisance variable compensation by removing nuisance attribute-related dimensions in the SVM expansion space...

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Text-independent speaker recognition

Published in:
Springer Handbook of Speech Processing and Communication, 2007, pp. 763-81.

Summary

In this chapter, we focus on the area of text-independent speaker verification, with an emphasis on unconstrained telephone conversational speech. We begin by providing a general likelihood ratio detection task framework to describe the various components in modern text-independent speaker verification systems. We next describe the general hierarchy of speaker information conveyed in the speech signal and the issues involved in reliably exploiting these levels of information for practical speaker verification systems. We then describe specific implementations of state-of-the-art text-independent speaker verification systems utilizing low-level spectral information and high-level token sequence information with generative and discriminative modeling techniques. Finally, we provide a performance assessment of these systems using the National Institute of Standards and Technology (NIST) speaker recognition evaluation telephone corpora.
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Summary

In this chapter, we focus on the area of text-independent speaker verification, with an emphasis on unconstrained telephone conversational speech. We begin by providing a general likelihood ratio detection task framework to describe the various components in modern text-independent speaker verification systems. We next describe the general hierarchy of speaker...

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ILR-based MT comprehension test with multi-level questions

Published in:
Human Language Technology, North American Chapter of the Association for Computational Linguistics, HLT/NAACL, 22-27 April 2007.

Summary

We present results from a new Interagency Language Roundtable (ILR) based comprehension test. This new test design presents questions at multiple ILR difficulty levels within each document. We incorporated Arabic machine translation (MT) output from three independent research sites, arbitrarily merging these materials into one MT condition. We contrast the MT condition, for both text and audio data types, with high quality human reference Gold Standard (GS) translations. Overall, subjects achieved 95% comprehension for GS and 74% for MT, across all genres and difficulty levels. Interestingly, comprehension rates do not correlate highly with translation error rates, suggesting that we are measuring an additional dimension of MT quality.
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Summary

We present results from a new Interagency Language Roundtable (ILR) based comprehension test. This new test design presents questions at multiple ILR difficulty levels within each document. We incorporated Arabic machine translation (MT) output from three independent research sites, arbitrarily merging these materials into one MT condition. We contrast the...

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A new approach to achieving high-performance power amplifier linearization

Published in:
IEEE Radar Conf., 17-20 April 2007. doi: 10.1109/RADAR.2007.374329

Summary

Digital baseband predistortion (DBP) is not particularly well suited to linearizing wideband power amplifiers (PAs); this is due to the exorbitant price paid in computational complexity. One of the underlying reasons for the computational complexity of DBP is the inherent inefficiency of using a sufficiently deep memory and a high enough polynomial order to span the multidimensional signal space needed to mitigate PA-induced nonlinear distortion. Therefore we have developed a new mathematical method to efficiently search for and localize those regions in the multidimensional signal space that enable us to invert PA nonlinearities with a significant reduction in computational complexity. Using a wideband code division multiple access (CDMA) signal we demonstrate and compare the PA linearization performance and computational complexity of our algorithm to that of conventional DBP techniques using measured results.
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Summary

Digital baseband predistortion (DBP) is not particularly well suited to linearizing wideband power amplifiers (PAs); this is due to the exorbitant price paid in computational complexity. One of the underlying reasons for the computational complexity of DBP is the inherent inefficiency of using a sufficiently deep memory and a high...

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Language recognition with word lattices and support vector machines

Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, 15-20 April 2007, Vol. IV, pp. 989-992.

Summary

Language recognition is typically performed with methods that exploit phonotactics--a phone recognition language modeling (PRLM) system. A PRLM system converts speech to a lattice of phones and then scores a language model. A standard extension to this scheme is to use multiple parallel phone recognizers (PPRLM). In this paper, we modify this approach in two distinct ways. First, we replace the phone tokenizer by a powerful speech-to-text system. Second, we use a discriminative support vector machine for language modeling. Our goals are twofold. First, we explore the ability of a single speech-to-text system to distinguish multiple languages. Second, we fuse the new system with an SVM PRLM system to see if it complements current approaches. Experiments on the 2005 NIST language recognition corpus show the new word system accomplishes these goals and has significant potential for language recognition.
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Summary

Language recognition is typically performed with methods that exploit phonotactics--a phone recognition language modeling (PRLM) system. A PRLM system converts speech to a lattice of phones and then scores a language model. A standard extension to this scheme is to use multiple parallel phone recognizers (PPRLM). In this paper, we...

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An evaluation of audio-visual person recognition on the XM2VTS corpus using the Lausanne protocols

Published in:
Proc. 32nd IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, April 2007, pp. IV-237 - 240.

Summary

A multimodal person recognition architecture has been developed for the purpose of improving overall recognition performance and for addressing channel-specific performance shortfalls. This multimodal architecture includes the fusion of a face recognition system with the MIT/LLGMM/UBM speaker recognition architecture. This architecture exploits the complementary and redundant nature of the face and speech modalities. The resulting multimodal architecture has been evaluated on theXM2VTS corpus using the Lausanne open set verification protocols, and demonstrates excellent recognition performance. The multimodal architecture also exhibits strong recognition performance gains over the performance of the individual modalities.
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Summary

A multimodal person recognition architecture has been developed for the purpose of improving overall recognition performance and for addressing channel-specific performance shortfalls. This multimodal architecture includes the fusion of a face recognition system with the MIT/LLGMM/UBM speaker recognition architecture. This architecture exploits the complementary and redundant nature of the face...

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Robust speaker recognition with cross-channel data: MIT-LL results on the 2006 NIST SRE auxiliary microphone task

Published in:
Proc. 32nd IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, April 2007, pp. IV-49 - IV-52.

Summary

One particularly difficult challenge for cross-channel speaker verification is the auxiliary microphone task introduced in the 2005 and 2006 NIST Speaker Recognition Evaluations, where training uses telephone speech and verification uses speech from multiple auxiliary microphones. This paper presents two approaches to compensate for the effects of auxiliary microphones on the speech signal. The first compensation method mitigates session effects through Latent Factor Analysis (LFA) and Nuisance Attribute Projection (NAP). The second approach operates directly on the recorded signal with noise reduction techniques. Results are presented that show a reduction in the performance gap between telephone and auxiliary microphone data.
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Summary

One particularly difficult challenge for cross-channel speaker verification is the auxiliary microphone task introduced in the 2005 and 2006 NIST Speaker Recognition Evaluations, where training uses telephone speech and verification uses speech from multiple auxiliary microphones. This paper presents two approaches to compensate for the effects of auxiliary microphones on...

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Multisensor dynamic waveform fusion

Published in:
Proc. 32nd Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, April 2007, pp. IV-577 - IV-580.

Summary

Speech communication is significantly more difficult in severe acoustic background noise environments, especially when low-rate speech coders are used. Non-acoustic sensors, such as radar sensors, vibrometers, and bone-conduction microphones, offer significant potential in these situations. We extend previous work on fixed waveform fusion from multiple sensors to an optimal dynamic waveform fusion algorithm that minimizes both additive noise and signal distortion in the estimated speech signal. We show that a minimum mean squared error (MMSE) waveform matching criterion results in a generalized multichannel Wiener filter, and that this filter will simultaneously perform waveform fusion, noise suppression, and crosschannel noise cancellation. Formal intelligibility and quality testing demonstrate significant improvement from this approach.
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Summary

Speech communication is significantly more difficult in severe acoustic background noise environments, especially when low-rate speech coders are used. Non-acoustic sensors, such as radar sensors, vibrometers, and bone-conduction microphones, offer significant potential in these situations. We extend previous work on fixed waveform fusion from multiple sensors to an optimal dynamic...

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