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The Lincoln Near-Earth Asteroid Research (LINEAR) Program

January 1, 1998
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Journal Article
Author:
Grant H. Stokes
…
Published in:
Lincoln Laboratory Journal, Vol. 11, No. 1, 1998, pp. 27-40.
Topic:
asteroids
R&D area:
R&D group:

Summary

Lincoln Laboratory has been developing electro-optical space-surveillance technology to detect, characterize, and catalog satellites for more than forty years. Recent advances in highly sensitive, large-format charge-coupled devices (CCDs) allow this technology to be applied to detecting and cataloging asteroids, including near-Earth objects (NEOs). When equipped with a new Lincoln Laboratory focal-plane camera and signal processing technology, the 1-m U.S. Air Force ground-based electro-optical deep-space surveillance (GEODSS) telescopes can conduct sensitive large-coverage searches for Earth-crossing and main-belt asteroids. Field measurements indicate that these enhanced telescopes can achieve a limiting magnitude of 22 over a 2-deg2 field of view with less than 100 sec of integration. This sensitivity rivals that of much larger telescopes equipped with commercial cameras. Working two years under U.S. Air Force sponsorship, we have developed technology for asteroid search operations at the Lincoln Laboratory Experimental Test Site near Socorro, New Mexico. By using a new large-format 2560 X 1960-pixel frame-transfer CCD camera, we have discovered over 10,000 asteroids, including 53 NEOs and 4 comets as designated by the Minor Planet Center (MPC). In March 1998, the Lincoln Near-Earth Asteroid Research (LINEAR) program provided over 150,000 observations of asteroids--nearly 90% of the world's asteroid observations that month--to the MPC, which resulted in the discovery of 13 NEOs and 1 comet. The MPC indicates that the LINEAR program outperforms all asteroid search programs operated to date.
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Summary

Lincoln Laboratory has been developing electro-optical space-surveillance technology to detect, characterize, and catalog satellites for more than forty years. Recent advances in highly sensitive, large-format charge-coupled devices (CCDs) allow this technology to be applied to detecting and cataloging asteroids, including near-Earth objects (NEOs). When equipped with a new Lincoln Laboratory...

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The Lincoln Near-Earth Asteroid Research (LINEAR) Program

AM-FM separation using auditory-motivated filters

January 9, 1997
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Journal Article
Author:
Thomas F. Quatieri
…
Published in:
IEEE Trans. Speech Audio Process., Vol. 5, No. 5, September 1997, pp. 465-480.
Topic:
signal processing
R&D area:
R&D group:

Summary

An approach to the joint estimation of sine-wave amplitude modulation (AM) and frequency modulation (FM) is described based on the transduction of frequency modulation into amplitude modulation by linear filters, being motivated by the hypothesis that the auditory system uses a similar transduction mechanism in measuring sine-wave FM. An AM-FM estimation is described that uses the amplitude envelope of the output of two transduction filters of piecewise-linear spectral shape. The piecewise-linear constraint is then relaxed, allowing a wider class of transduction-filter pairs for AM-FM separation under a monotonicity constraint of the filters' quotient. The particular case of Gaussian filters, and measured auditory filters, although not leading to a solution in closed form, provide for iterative AM-FM estimation. Solution stability analysis and error evaluation are performed and the FM transduction method is compared with the energy separation algorithm, based on the Teager energy operator, and the Hilbert transform method for AM-FM estimation. Finally, a generalization to two-dimensional (2-D) filters is described.
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Summary

An approach to the joint estimation of sine-wave amplitude modulation (AM) and frequency modulation (FM) is described based on the transduction of frequency modulation into amplitude modulation by linear filters, being motivated by the hypothesis that the auditory system uses a similar transduction mechanism in measuring sine-wave FM. An AM-FM...

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AM-FM separation using auditory-motivated filters

Description of radar correlation and interpolation algorithms for the ASR-9 Processor Augmentation Card (9-PAC)

October 27, 1995
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Project Report
Author:
Oliver J. Newell
…
Published in:
MIT Lincoln Laboratory Report ATC-236
Topic:
signal processing
R&D area:
R&D group:

Summary

MIT Lincoln Laboratory, under sponsorship from the Federal Aviation Adminstration (FAA), is conducting a program to replace/upgrade the existing ASR-9 array signal processor (ASP) and associated algorithms to improve performance and future maintainability. The ASR-9 processor augmentation card (9-PAC) replaces the ASP four-board set with a single card containing three TMS320c40 processors and 32 Megabytes of memory. The resulting increase in both processing speed and memory size allows more sophisticated beacon and radar processing algorithms to be implemented. The majority of the improvement to the radar correlation and interpolation (C&I) function lies in the area of geocensoring and adaptive thresholding, where the larger memory capacity of the 9-PAC allows more detailed maps to be maintained. A dynamic road map mechanism has been implemented to reduce the need for manual tuning of the system when the radars are first installed or when new road construction occurs. The map is twice the resolution of the original geocensormap, resulting in a decrease in total area desensitized to radar-only targets. In addition, the new geocensor mechanism makes use of target amplitude information, allowing aircraft with amplitudes significantly greater than the road traffic returns at a particular cell to pass through uncensored. The adaptive thresholding cell geometry has been modified so that adaptive map cells now overlap one another, eliminating the false target breakthrough that occurs in the present system when regions of false alarms due to birds or weather transition from one cell to the next. The entire C & I function has been recorded in a high-level language (ANSI-C), allowing it to be easily ported between platforms and better facilitating off-line analysis.
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Summary

MIT Lincoln Laboratory, under sponsorship from the Federal Aviation Adminstration (FAA), is conducting a program to replace/upgrade the existing ASR-9 array signal processor (ASP) and associated algorithms to improve performance and future maintainability. The ASR-9 processor augmentation card (9-PAC) replaces the ASP four-board set with a single card containing three...

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Description of radar correlation and interpolation algorithms for the ASR-9 Processor Augmentation Card (9-PAC)

Digital signal processing applications in cochlear-implant research

March 1, 1994
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Journal Article
Author:
Joseph Tierney
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Published in:
Lincoln Laboratory Journal, Vol. 7, No. 1, Spring 1994, pp. 31-62.
Topic:
signal processing
R&D area:
R&D group:

Summary

We have developed a facility that enables scientists to investigate a wide range of sound-processing schemes for human subjects with cochlear implants. This digital signal processing (DSP) facility-named the Programmable Interactive System for Cochlear Implant Electrode Stimulation (PISCES)-was designed, built, and tested at Lincoln Laboratory and then installed at the Cochlear Implant Research Laboratory (CIRL) of the Massachusetts Eye and Ear Infirmary (MEEI). New stimulator algorithms that we designed and ran on PISCES have resulted in speech-reception improvements for implant subjects relative to commercial implant stimulators. These improvements were obtained as a result of interactive algorithm adjustment in the clinic, thus demonstrating the importance of a flexible signal-processing facility. Research has continued in the development of a laboratory-based, sohare-controlled, real-time, speech processing system; the exploration of new sound-processing algorithms for improved electrode stimulation; and the design of wearable stimulators that will allow subjects full-time use of stimulator algorithms developed and tested in a laboratory setting.
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Summary

We have developed a facility that enables scientists to investigate a wide range of sound-processing schemes for human subjects with cochlear implants. This digital signal processing (DSP) facility-named the Programmable Interactive System for Cochlear Implant Electrode Stimulation (PISCES)-was designed, built, and tested at Lincoln Laboratory and then installed at the...

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Digital signal processing applications in cochlear-implant research

Energy separation in signal modulations with application to speech analysis

October 1, 1993
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Journal Article
Author:
Petros Maragos
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Published in:
IEEE Trans. Signal Process., Vol. 41, No. 10, October 1993, pp. 3024-3051.
Topic:
signal processing
R&D area:
R&D group:

Summary

Oscillatory signals that have both an amplitude-modulation (AM) and a frequency-modulation (FM) structure are encountered in almost all communication systems. We have also used these structures recently for modeling speech resonances, being motivated by previous work on investigating fluid dynamics phenomena during speech production that provide evidence for the existence of modulations in speech signals. In this paper, we use a nonlinear differential operator that can detect modulations in AM-FM signals by estimating the product of their time-varying amplitude and frequency. This operator essentially tracks the energy needed by a source to produce the oscillatory signal. To solve the fundamental problem of estimating both the amplitude envelope and instantaneous frequency of an AM-FM signal we develop a novel approach that uses nonlinear combinations of instantaneous signal outputs from the energy operator to separate its output energy product into its amplitude modulation and frequency modulation components. The theoretical analysis is done first for continuous-time signals. Then several efficient algorithms are developed and compared for estimating the amplitude envelope and instantaneous frequency of discrete-time AM-FM signals. These energy separation algorithms are then applied to search for modulations in speech resonances, which we model using AM-FM signals to account for time-varying amplitude envelopes and instantaneous frequencies. Our experimental results provide evidence that bandpass filtered speech signals around speech formants contain amplitude and frequency modulations within a pitch period. Overall, the energy separation algorithms, due to their very low computational complexity and instantaneously-adapting nature, are very useful in detecting modulation patterns in speech and other time-varying signals.
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Summary

Oscillatory signals that have both an amplitude-modulation (AM) and a frequency-modulation (FM) structure are encountered in almost all communication systems. We have also used these structures recently for modeling speech resonances, being motivated by previous work on investigating fluid dynamics phenomena during speech production that provide evidence for the existence...

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Energy separation in signal modulations with application to speech analysis

Detection of transient signals using the energy operator

April 27, 1993
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Conference Paper
Author:
Robert B. Dunn
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Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, Vol. 3, ICASSP, 27-30 April 1993, pp. 145-148.
Topic:
signal processing
R&D area:
R&D group:

Summary

A function of the Teager-Kaiser energy operator is introduced as a method for detecting transient signals in the presence of amplitude-modulated and frequency-modulated tonal interference. This function has excellent time resolution and is robust in the presence of white noise. The output of the detection function is also independent of the interference-to-transient ratio when that ratio is large. It is demonstrated that the detection function can be applied to interference signals with multiple amplitude-modulated and frequency-modulated tonal components.
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Summary

A function of the Teager-Kaiser energy operator is introduced as a method for detecting transient signals in the presence of amplitude-modulated and frequency-modulated tonal interference. This function has excellent time resolution and is robust in the presence of white noise. The output of the detection function is also independent of...

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Detection of transient signals using the energy operator

Two-talker pitch tracking for co-channel talker interference suppression

April 30, 1992
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Technical Report
Author:
Marc A. Zissman
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Published in:
MIT Lincoln Laboratory Report TR-951
Topic:
signal processing
R&D area:
R&D group:

Summary

Almost all co-channel talker interference suppression systems use the difference in the pitches of the target and jammer speakers to suppress the jammer and enhance the target. While joint pitch estimators outputting two pitch estimates as a function of time have been proposed, the task of proper assignment of pitch to speaker (two-talker pitch tracking) has proven difficult. This report describes several approaches to the two-talker pitch tracking problem including algorithms for pitch track interpolation, spectral envelope tracking, and spectral envelope classification. When evaluated on an all-voiced two-talker database, the best of these new tracking systems correctly assigned pitch 87% of the time given perfect joint pitch estimation.
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Summary

Almost all co-channel talker interference suppression systems use the difference in the pitches of the target and jammer speakers to suppress the jammer and enhance the target. While joint pitch estimators outputting two pitch estimates as a function of time have been proposed, the task of proper assignment of pitch...

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Two-talker pitch tracking for co-channel talker interference suppression

Initialization for improved IIR filter performance

March 1, 1992
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Journal Article
Author:
Edward S. Chornoboy
Published in:
IEEE Trans. Signal Process., Vol. 40, No. 3, March 1992, pp. 543-550.
Topic:
signal processing
R&D area:
R&D group:

Summary

A new method for initializing the memory registers of IIR filters is introduced. In addition to providing improved performance as compared to other methods of initialization, this method is unique in that it makes no a priori assumptions regarding the input-signal content. Therefore, this method applies equally well to a variety of IIR filter designs and applications. The method is best suited for signal-processing applications in which "batch" processing of the data is used. However, sequential processing can be accommodated when delays at the beginning of a processing segment can be tolerated.
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Summary

A new method for initializing the memory registers of IIR filters is introduced. In addition to providing improved performance as compared to other methods of initialization, this method is unique in that it makes no a priori assumptions regarding the input-signal content. Therefore, this method applies equally well to a...

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Initialization for improved IIR filter performance

Speech nonlinearities, modulations, and energy operators

May 14, 1991
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Conference Paper
Author:
Petros Maragos
…
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, 14-17 May 1991, pp. 421-424.
Topic:
signal processing
R&D area:
R&D group:

Summary

In this paper, we investigate an AM-FM model for representing modulations in speech resonances. Specifically, we propose a frequency modulation (FM) model for the time-varying formants whose amplitude varies as the envelope of an amplitude-modulated (AM) signal. To detect the modulations we apply the energy operator (psi)(x) = (x)^2 - xx and its discrete counterpart. We found that psi can approximately track the envelope of AM signals, the instantaneous frequency of FM signals, and the product of these two functions in the general case of AM-FM signals. Several experiments are reported on the applications of this AM-FM modeling to speech signals, bandpass filtered via Gabor filtering.
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Summary

In this paper, we investigate an AM-FM model for representing modulations in speech resonances. Specifically, we propose a frequency modulation (FM) model for the time-varying formants whose amplitude varies as the envelope of an amplitude-modulated (AM) signal. To detect the modulations we apply the energy operator (psi)(x) = (x)^2 -...

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Speech nonlinearities, modulations, and energy operators

Short-time signal representation by nonlinear difference equations

April 6, 1990
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Conference Paper
Author:
Thomas F. Quatieri
…
Published in:
Proc. IEEE Int. Conf. on Acoustics, Speech and Signal Processing, ICASSP, Vol. 3, Digital Signal Processing, 3-6 April 1990, pp. 1551-1554.
Topic:
signal processing
R&D area:
R&D group:

Summary

The solution of a nonlinear difference equation can take on complicated deterministic behavior which appears to be random for certain values of the equation's coefficients. Due to the sensitivities to initial conditions of the output of such "chaotic" systems, it is difficult to duplicate the waveform structure by parameter analysis and waveform synthesis techniques. In this paper, methods are investigated for short-time analysis and synthesis of signals from a class of second-order difference equations with a cubic nonlinearity. In analysis, two methods are explored for estimating equation coefficients: (1) prediction error minimization (a linear estimation problem) and (2) waveform error minimization (a nonlinear estimation problem). In the latter case, which improves on the prediction error solution, an iterative analysis-by-synthesis method is derived which allows as free variables initial conditions, as well as equation coefficients. Parameter estimates from these techniques are used in sequential short-time synthesis procedures. Possible application to modeling "quasi-periodic" behavior in speech waveforms is discussed.
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Summary

The solution of a nonlinear difference equation can take on complicated deterministic behavior which appears to be random for certain values of the equation's coefficients. Due to the sensitivities to initial conditions of the output of such "chaotic" systems, it is difficult to duplicate the waveform structure by parameter analysis...

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Short-time signal representation by nonlinear difference equations
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