Publications

Refine Results

(Filters Applied) Clear All

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.
READ LESS

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.

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE

Triage framework for resource conservation in a speaker identification system

Published in:
Proc. 32nd IEEE Int. Conf. on Acoustics, Speech, and Signal Processing, ICASSP, April 2007, pp. IV-69 - IV-72.

Summary

We present a novel framework for triaging (prioritizing and discarding) data to conserve resources for a speaker identification (SID) system. Our work is motivated by applications that require a SID system to process an overwhelming volume of audio data. We design a triage filter whose goal is to conserve recognizer resources while preserving relevant content. We propose triage methods that use signal quality assessment tools, a scaled-down version of the main recognizer itself, and a fusion of these measures. We define a new precision-based measure of effectiveness for our triage framework. Our experimental results with the 35-speaker tactical SID corpus bear out the validity of our approach.
READ LESS

Summary

We present a novel framework for triaging (prioritizing and discarding) data to conserve resources for a speaker identification (SID) system. Our work is motivated by applications that require a SID system to process an overwhelming volume of audio data. We design a triage filter whose goal is to conserve recognizer...

READ MORE

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.
READ LESS

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...

READ MORE

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.
READ LESS

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...

READ MORE