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IEEE International Conference on Communications (ICC)

作者： L.J. Faber S.H. Ardalan Center of Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

The general-order multichannel fast transversal filter (FTF) algorithm is a generalization of the multichannel FTF algorithm in which the orders of the input channels can be independently specified. A normalized version of this algorithm is derived and applied to a special case of a two-input-channel, joint-process estimator: one of the input channels is defined as the unit-delayed joint-process signal. The configuration is an equation error form for least-squares pole-zero (ARMA) system identification; it is also used in decision-directed equalization and echo cancellation. A summary of the algorithm is given in proper execution order, with an operations count for each step.< >

关键词： signal processing algorithms Finite impulse response filter IIR filters Transversal filters signal processing Equations System identification Autoregressive processes Delay estimation Decision feedback equalizers

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On the sensitivity of transversal RLS algorithms to random perturbations in the filter coefficients

On the sensitivity of transversal RLS algorithms to random p...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S. Ardalan Center for Communications and Signal Processing Department of Electrical and Computer Engineering North Carolina State University Raleigh NC USA

Transversal recursive least-squares (RLS) algorithms estimate filter coefficients which minimize the accumulated sum of the square of the error residuals, termed the error power. The sensitivity of this error power to random perturbations about the optimum filter coefficients is investigated. Expressions are derived for the mean and variance of the deviation from the optimum error power. It is shown that for the prewindowed growing-memory RLS algorithm ( lambda =1), the mean value of the deviation increases linearly with the number of iterations. The variance of the deviation increases in proportion to the square of the number of iterations. Expressions are derived for the variance of correlated signals. These expressions show that the variance of the deviation for correlated signals increases compared to uncorrelated white signals by a term related to the sum of the square of the off-diagonal elements of the sample autocorrelation matrix. Expressions are also derived for the mean and variance of the deviation for the exponentially windowed RLS algorithm ( lambda <1). In this case, the deviations are bounded and inversely proportional to 1- lambda .< >

关键词： Resonance light scattering Transversal filters signal processing algorithms Vectors computer errors Roundoff errors Kalman filters Least squares methods Fluctuations signal analysis

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Analysis of delta-sigma modulators with bandlimited Gaussian inputs

Analysis of delta-sigma modulators with bandlimited Gaussian...

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： S.H. Ardalan Department of Electrical & Computer Engineering Center for Communications & Signal Processing North Carolina State University Raleigh NC USA

Delta-sigma modulation is gaining widespread use as a monolithic analog-to-digital and digital-to-analog conversion technique in a wide variety of applications. The behavior of delta-sigma modulators depends on the statistics and nature of the input signal. The behavior of the circuit is input-amplitude-dependent, due to the nonlinearity (two-level quantizer). The nonlinear behavior of high-order delta-sigma modulators with bandlimited Gaussian inputs is analyzed.< >

关键词： Delta modulation signal to noise ratio Delta-sigma modulation Circuits Feeds Linear systems Nonlinear distortion Quantization Additive noise Bandwidth

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Speech training aids for hearing-impaired individuals: II. Configuration of the Johns Hopkins aids

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Journal of Rehabilitation Research and Development 1988年第4期25卷 63-68页

作者： Ferguson, J.B. Bernstein, L.E. Goldstein Jr., M.H. Speech Processing Laboratory Department of Electrical and Computer Engineering Johns Hopkins University Baltimore MD 21218 United States

Two interrelated personal computer (PC) based speech training aids have been developed: one for use in a school or clinic, the Speech Training Station (STS);and the other for the deaf child's home, the Speech Practice Station (SPS). The STS monitors speech production by microphone, electroglottograph, and pneumotachograph. The SPS system uses only the microphone input. Both systems utilize commercially available board-level hardware and a custom analog preprocessor board for the analysis of the acoustic and/or physiologic inputs. The school system has been used by speech therapists for diagnosis, training by game playing, and specification of exercises for the SPS. The home system provides directed speech practice between therapy sessions.

关键词： HUMAN REHABILITATION engineering

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Pulsewidth-constrained signaling and trellis coded modulation on the direct-detection optical channel

Pulsewidth-constrained signaling and trellis coded modulatio...

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IEEE Global Telecommunications Conference and Exhibition. Communications for the Information Age

作者： G.L. Bechtel J.W. Modestino Center of Communications and Digital Signal Processing Electrical and Computer Engineering Department Northeastern University Boston MA USA Electrical Computer and System Engineering Department Rensselaer Polytechnic Institute Troy NY USA

Pulsewidth-constrained signals are developed for use on the degraded direct-detection optical channel. A two-dimensional representation for these signals is obtained and used to design a trellis-coded-modulation (TCM) system for use on the optical channel. The theoretical advantage of optical TCM is shown, through use of the cutoff rate, to range from 7.5 to 9 dB, depending on the throughput rate of the system. The practical potential is shown, through the actual design of a TCM system, to be as high as 6.7 dB.< >

关键词： Pulse modulation Optical pulses Modulation coding Optical modulation Bandwidth Optical signal processing Optical noise Optical receivers AWGN Energy efficiency

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signal processing For Ocean Tomography With Moving Ships

Signal Processing For Ocean Tomography With Moving Ships

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Asilomar Conference on signals, Systems & computers

作者： T.G. Birdsall K. Metzger R.C. Spindel Communication and Signal Processing Laboratory Department of Electrical Engineering and Computer Science University of Michigan Ann Arbor MI USA Applied Physics Laboratory University of Washington Seattle WA USA

In ocean acoustic tomography, maximal length binary shift- register sequences, m-sequences, are used to modulate acoustic carriers to achieve high average power and good time and Doppler resolution. To date, the underwater transmitters and receivers have been in fixed positions, and signal processing has consisted of demodulation followed by factor inverse filtering. Ocean tomography now is being extended to include the use of moving, ship-towed, transmitters and receivers, where signal processing must account for Doppler time and frequency resealing. This paper describes the signal demodulation and processing methods developed for moving ship tomography and presents illustrative results.

关键词： signal processing Oceans Tomography Marine vehicles Transmitters Spatial resolution signal resolution Filtering Acoustic signal processing Demodulation

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A geometric approach to real time pitch detection

A geometric approach to real time pitch detection

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International Conference on Acoustics, Speech, and signal processing (ICASSP)

作者： T.D. Nguyen J.B. Ferguson T.W. McNamara Speech Processing Laboratory Department of Electrical and Computer Engineering Johns Hopkins University Baltimore MD USA

An approach to time-domain pitch detection based on the concepts of center of mass is presented and evaluated. Excursions, or bumps, in speech signals are treated as geometric areas and replaced by their total mass lumped at the centers of mass. Intervals between masses are grouped into candidate classes. Coincidence and coherence indices of these classes are computed to determine the most likely pitch estimate. Postprocessing consists of a simple error-correction and silence-detection scheme. This algorithm compares favourably in performance with the autocorrelation method, using pitch contours from electroglottograph signals as a reference. The algorithm is tested in noisy environments simulated by uniformly distributed white noise and multitalker babble noise. Results show that the algorithm is robust and accurate. The implementation of this algorithm for a vibrotactile device to aid lipreading is described.< >

关键词： Time domain analysis Speech processing Testing Speech synthesis Speech analysis White noise Detection algorithms Energy states Laboratories Error correction

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Short channel effects in submicron self-aligned gate heterostructure field effect transistors

Short channel effects in submicron self-aligned gate heteros...

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International Electron Devices Meeting (IEDM)

作者： C.J. Han P.P. Ruden D. Grider A. Fraasch K. Newstrom P. Joslyn M. Shur Honeywell Sensors and Signal Processing Laboratory Bloomington MN USA Department of Electrical Engineering University of Minnesota Minneapolis MN USA

AlGaAs/GaAs self-aligned gate heterostructure FETs with gate lengths varying from 0.3 to 1.5 mu m were fabricated to study short-channel effects. Peak extrinsic transconductance as high as 360 mS/mm was achieved with this process. Short-channel effects such as increases in the output conductance, increases in the subthreshold current, and shifts in the threshold voltage are reported for temperatures ranging from 30 degrees C to 100 degrees C. The observations follow closely predictions from a simple model which attributes the effects to space-charge-limited electron injection into the GaAs buffer layer beneath the actual two-dimensional electron gas channel.< >

关键词： HEMTs MODFETs Gallium arsenide Electrons Transconductance Subthreshold current Threshold voltage Temperature distribution Predictive models Buffer layers

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Adaptive Filtering via Cumulants and LMS Algorithm

Adaptive Filtering via Cumulants and LMS Algorithm

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American Control Conference (ACC)

作者： Hsing-Hsing Chiang Chrysostomos L. Nikias Communications and Digital Signal Processing (CDSP) Center for Research and Graduate Studies Department of Electrical and Computer Engineering Northeastern University Boston MA USA

A new adaptive identification scheme is introduced for a non-Gaussian white noise driven linear, nonminimum phase FIR system. The adaptive scheme is based on non-causal auto-regressive (AR) modeling of higher-order cumulants of the system output. In particular, the matnitude and phase response estimates at each iteration are expressed in terms of the updated parameters of the non-causal AR model. The set of updated AR parameters is obtained by employing the LMS algorithm and by using higher-order cumulants instead of time samples of the output signal. It is demonstrated by means of standard examples that the new adaptive scheme works well and as expected outperform the modified (autocorrelation-based) LMS algorithm for nonminimum phase system identification.

关键词： Adaptive filters Least squares approximation Filtering algorithms signal processing algorithms Finite impulse response filter Autocorrelation Digital signal processing Geophysical measurements White noise Adaptive systems

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Cumulant techniques for the estimation and cancellation of intersymbol interference

Cumulant techniques for the estimation and cancellation of i...

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Annual ASSP Workshop on Spectrum Estimation and Modeling

作者： D. Hatzinakos C.L. Nikias Communications and Digital Signal Processing (CDSP) Center of Research and Graduate Studies Department of Electrical and Computer Engineering Northeastern University Boston MA USA

A discrete-time method is proposed for the estimation and cancellation of intersymbol interference in a digital communication channel. The received signal is first demodulated and sampled and then the fourth-order cumulants of the resulting discrete-time sequence are estimated. The method estimates the channel impulse response from the complex cepstrum of the aforementioned fourth-order cumulants (i.e., tricepstrum). As such, the proposed method depends only on the second- and fourth-order statistics of the transmitted sequence and is capable of reconstructing nonminimum-phase impulse responses. Monte Carlo simulation results demonstrate the effectiveness of the method, its low sensitivity to observation noise, and its improved performance in terms of probability of error of the reconstructed transmitted sequence. Performance comparisons are also given using existing equalization techniques.< >

关键词： Interference cancellation Intersymbol interference signal processing algorithms Decision feedback equalizers Maximum likelihood estimation Digital communication Adaptive equalizers Digital signal processing Cepstrum Statistics

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