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Cryogenic YB3+-doped solid-state lasers

Published in:
IEEE J. Sel. Topics in Quantum Electron., Vol. 13, No. 3, May/June 2007, pp. 448-459.

Summary

Cryogenically cooled solid-state lasers promise a revolution in power scalability while maintaining a good beam quality because of significant improvements in efficiency and thermo-optic properties. This is particularly true forYb3+ lasers because of their relatively lowquantum defect and relatively broadband absorption even at cryogenic temperatures. Thermo-optic properties of host materials, including thermal conductivity, thermal expansion, and refractive index at low temperature, are reviewed and data presented for YAG (ceramic and single crystal), GGG, GdVO4, and Y2O3. Spectroscopic properties of Yb:YAG and Yb:LiYF4 (YLF) including absorption cross sections, emission cross sections, and fluorescence lifetimes at cryogenic temperatures are characterized. Recent experiments have pushed the power from an end-pumped cryogenically cooled Yb:YAG laser to 455-W continuous-wave output power from 640-W incident pump power at anM2 of 1.4.
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Summary

Cryogenically cooled solid-state lasers promise a revolution in power scalability while maintaining a good beam quality because of significant improvements in efficiency and thermo-optic properties. This is particularly true forYb3+ lasers because of their relatively lowquantum defect and relatively broadband absorption even at cryogenic temperatures. Thermo-optic properties of host materials...

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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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Surveillance improvement algorithms for Airport Surface Detection Equipment Model X (ASDE-X) at Dallas-Fort Worth Airport

Published in:
MIT Lincoln Laboratory Report ATC-333

Summary

Operational testing of the Runway Status Lights (RWSL) system at the Dallas-Fort Worth (DFW) airport has detected a number of cases where faults in the ASDE-X/DFW surveillance data have led to erroneous operation of the status lights. Among the surveillance problems noted during testing at DFW were: (a) false tracks, (b) track positional jumps to false locations, (c) Mode S track splits, (d) ATCRBS track splits, (e) invalid Mode C altitudes, (f) invalid track velocities, and (g) spurious Mode 3/a 06078 code tracks. The RWSL surveillance improvement algorithms package in this document is placed between the ASDE-X/DFW surveillance data source and the RESL safety logic. The surveillance improvement algorithms perform a variety of reasonableness and consistency checks on the input data and set validity flags and report status values for each input report which are then passed on to the RWSL safety logic. These flags and status values allow the RWSL to ignore erroneous reports and to avoid using questionable report components in the subsequent RWSL logic. This document illustrates the performance of the RWSL surveillance improvement algorithms package with examples from DFW analysis. It is shown that the RWSL surveillance improvement algorithms package substantially reduces the impact of the known ASDE-X/DFW surveillance anomalies on the performance of the RWSL safety logic. The RWSL surveillance improvement algorithms package may also host future algorithms necessary to mitigate further problems that might be detected in the surveillance data.
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Summary

Operational testing of the Runway Status Lights (RWSL) system at the Dallas-Fort Worth (DFW) airport has detected a number of cases where faults in the ASDE-X/DFW surveillance data have led to erroneous operation of the status lights. Among the surveillance problems noted during testing at DFW were: (a) false tracks...

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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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