Ramdas Kumaresan

Orcid: 0000-0002-8897-4568

Affiliations:
  • University of Rhode Island, Kingston, RI, USA


According to our database1, Ramdas Kumaresan authored at least 53 papers between 1980 and 2020.

Collaborative distances:
  • Dijkstra number2 of four.
  • Erdős number3 of four.

Awards

IEEE Fellow

IEEE Fellow 1993, "For contributions to the development of subspace methods for modeling and identifying low-rank signals and systems.".

Timeline

Legend:

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Links

Online presence:

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Bibliography

2020
Improved Auditory-Inspired Signal Processing Algorithm Design for Tracking Multiple Frequency Components.
SN Comput. Sci., 2020

2018
Harmonic sum-based method for heart rate estimation using PPG signals affected with motion artifacts.
J. Ambient Intell. Humaniz. Comput., 2018

2016
Bandpass phase shifter and analytic signal generator.
Signal Process., 2016

2014
Auditory-inspired pitch extraction using a Synchrony Capture Filterbank and phase alignment.
Proceedings of the IEEE International Conference on Acoustics, 2014

2012
Synchrony capture filterbank (SCFB): An auditory periphery inspired method for tracking sinusoids.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

2011
Multiple pitch identification using cochlear-like frequency capture and harmonic grouping.
Proceedings of the IEEE International Conference on Acoustics, 2011

2010
Encoding Bandpass Signals Using Zero/Level Crossings: A Model-Based Approach.
IEEE Trans. Speech Audio Process., 2010

2005
Comprehensive modulation representation for automatic speech recognition.
Proceedings of the INTERSPEECH 2005, 2005

Adaptive Filterbanks Inspired By the Auditory System for Speech Feature Extraction.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

2003
Average instantaneous frequency (AIF) and average log-envelopes (ALE) for ASR with the Aurora 2 database.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

2001
On the duality between line-spectral frequencies and zero-crossings of signals.
IEEE Trans. Speech Audio Process., 2001

2000
On minimum/maximum/all-pass decompositions in time and frequency domains.
IEEE Trans. Signal Process., 2000

On decomposing speech into modulated components.
IEEE Trans. Speech Audio Process., 2000

A new real-zero conversion algorithm.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
On designing stable allpass filters using AR modeling.
IEEE Trans. Signal Process., 1999

1998
A parametric modeling approach to Hilbert transformation.
IEEE Signal Process. Lett., 1998

An inverse signal approach to computing the envelope of a real valued signal.
IEEE Signal Process. Lett., 1998

Unique Positive FM-AM Decomposition of Signals.
Multidimens. Syst. Signal Process., 1998

Toeplitz and Hankel matrix approximation using structured approach.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

Algorithm for decomposing an analytic signal into AM and positive FM components.
Proceedings of the 1998 IEEE International Conference on Acoustics, 1998

1996
Separation of cochannel signals using the parametric constant modulus algorithm.
Proceedings of the 1996 IEEE International Conference on Acoustics, 1996

A variable frame pitch estimator and test results.
Proceedings of the 1996 IEEE International Conference on Acoustics, 1996

1995
Dynamic tracking filters for decomposing nonstationary sinusoidal signals.
Proceedings of the 1995 International Conference on Acoustics, 1995

Effect of model mismatch on polynomial envelope and phase modeling of nonstationary sinusoids.
Proceedings of the 1995 International Conference on Acoustics, 1995

1994
Resolving ambiguities in estimating spatial frequencies in sparse linear array.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

Voiced-speech analysis based on the residual interfering signal canceler (RISC) algorithm.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994

1993
On the estimation of rational transfer functions from samples of the power spectrum.
IEEE Trans. Signal Process., 1993

Formant tones: weak components at formant frequencies actively generated in the vocal tract.
Proceedings of the IEEE International Conference on Acoustics, 1993

Segmentation of textures with different roughness using the model of isotropic two-dimensional fractional Brownian motion.
Proceedings of the IEEE International Conference on Acoustics, 1993

1992
A low rank weighted matrix approximation method for robust estimation of sinusoid parameters.
Proceedings of the 1992 IEEE International Conference on Acoustics, 1992

1991
FIR prefiltering improves Prony's method.
IEEE Trans. Signal Process., 1991

New pole-by-pole model fitting approach for signal parameter estimation.
Proceedings of the 1991 International Conference on Acoustics, 1991

Fitting a pole-zero filter model to arbitrary frequency response samples.
Proceedings of the 1991 International Conference on Acoustics, 1991

Adaptive FIR-filter for control systems with periodic disturbance.
Proceedings of the 1991 International Conference on Acoustics, 1991

1990
On a frequency domain analog of Prony's method.
IEEE Trans. Acoust. Speech Signal Process., 1990

1989
A fast and accurate RNS scaling technique for high speed signal processing.
IEEE Trans. Acoust. Speech Signal Process., 1989

Fast Base Extension Using a Redundant Modulus in RNS.
IEEE Trans. Computers, 1989

1988
Binary multiplication with PN sequences.
IEEE Trans. Acoust. Speech Signal Process., 1988

VLSI implementation of neural networks based on PN sequences.
Neural Networks, 1988

Some structured matrix approximation problems.
Proceedings of the IEEE International Conference on Acoustics, 1988

1987
An accurate scaling technique in improved residue number system arithmetic.
Proceedings of the IEEE International Conference on Acoustics, 1987

Estimation of angles of arrivals of broadband signals.
Proceedings of the IEEE International Conference on Acoustics, 1987

Array signal processing with interconnected Neuron-like elements.
Proceedings of the IEEE International Conference on Acoustics, 1987

1986
An algorithm for pole-zero modeling and spectral analysis.
IEEE Trans. Acoust. Speech Signal Process., 1986

An exact least squares fitting technique for two-dimensional frequency wavenumber estimation.
Proc. IEEE, 1986

Vector-radix algorithm for a 2-D discrete Hartley transform.
Proc. IEEE, 1986

1985
A prime factor FFT algorithm with real valued arithmetic.
Proc. IEEE, 1985

High resolution bearing estimation without eigen decomposition.
Proceedings of the IEEE International Conference on Acoustics, 1985

Efficient architectures for implementing the prime-factor Fourier transform modules.
Proceedings of the IEEE International Conference on Acoustics, 1985

1984
Accuracy of frequency estimation and its relation to prediction filter order.
Proceedings of the IEEE International Conference on Acoustics, 1984

Multiprocessor system for speech processing and telecommunications.
Proceedings of the IEEE International Conference on Acoustics, 1984

1982
Accurate parameter estimation of noisy speech-like signals.
Proceedings of the IEEE International Conference on Acoustics, 1982

1980
Improved spectral resolution II.
Proceedings of the IEEE International Conference on Acoustics, 1980


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