Hamid Sheikhzadeh

Orcid: 0000-0002-9262-4165

According to our database1, Hamid Sheikhzadeh authored at least 80 papers between 1994 and 2024.

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

Timeline

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Bibliography

2024
A proposed method to improve the WER of an ASR system in the noisy reverberant room.
J. Frankl. Inst., 2024

2023
Parallel and Limited Data Voice Conversion Using Stochastic Variational Deep Kernel Learning.
CoRR, 2023

Fast Classification with Sequential Feature Selection in Test Phase.
CoRR, 2023

2022
Speech improvement in noisy reverberant environments using virtual microphones along with proposed array geometry.
EURASIP J. Adv. Signal Process., 2022

Parallel voice conversion with limited training data using stochastic variational deep kernel learning.
Eng. Appl. Artif. Intell., 2022

2021
Progressive Transmission using Recurrent Neural Networks.
CoRR, 2021

Detection of Transformer Winding Axial Displacement by Kirchhoff and Delay and sum Radar Imaging Algorithms.
CoRR, 2021

2020
Pilot Pattern Design for Deep Learning-Based Channel Estimation in OFDM Systems.
IEEE Wirel. Commun. Lett., 2020

Deep feature selection using a teacher-student network.
Neurocomputing, 2020

Bayesian Reinforcement Learning for Link-Level Throughput Maximization.
IEEE Commun. Lett., 2020

2019
Deep Learning-Based Channel Estimation.
IEEE Commun. Lett., 2019

Propagation Channel Modeling by Deep learning Techniques.
CoRR, 2019

2017
Vowel detection using a perceptually-enhanced spectrum matching conditioned to phonetic context and speaker identity.
Speech Commun., 2017

Variational Relevant Sample-Feature Machine: A fully Bayesian approach for embedded feature selection.
Neurocomputing, 2017

2016
Relevance Vector Machine for Survival Analysis.
IEEE Trans. Neural Networks Learn. Syst., 2016

Incremental relevance sample-feature machine: A fast marginal likelihood maximization approach for joint feature selection and classification.
Pattern Recognit., 2016

Sparse Bayesian mixed-effects extreme learning machine, an approach for unobserved clustered heterogeneity.
Neurocomputing, 2016

A double-layer ELM with added feature selection ability using a sparse Bayesian approach.
Neurocomputing, 2016

A Novel and Fast Algorithm for Solving Permutation in Convolutive BSS, Based on Real and Imaginary Decomposition.
Circuits Syst. Signal Process., 2016

Reverberation time estimation based on a model for the power spectral density of reverberant speech.
Proceedings of the 24th European Signal Processing Conference, 2016

2015
Gaussian Kernel Width Optimization for Sparse Bayesian Learning.
IEEE Trans. Neural Networks Learn. Syst., 2015

Voice conversion based on feature combination with limited training data.
Speech Commun., 2015

2013
The Relevance Sample-Feature Machine: A Sparse Bayesian Learning Approach to Joint Feature-Sample Selection.
IEEE Trans. Cybern., 2013

Sequential method for speech segmentation based on Random Matrix Theory.
IET Signal Process., 2013

Semi-spatiotemporal fMRI Brain Decoding.
Proceedings of the International Workshop on Pattern Recognition in Neuroimaging, 2013

Reduced Search Space Frame Alignment Based on Kullback-Leibler Divergence for Voice Conversion.
Proceedings of the Advances in Nonlinear Speech Processing - 6th International Conference, 2013

A Fast Semi-blind Reverberation Time Estimation Using Non-linear Least Squares Method.
Proceedings of the Advances in Nonlinear Speech Processing - 6th International Conference, 2013

Subband blind source separation for convolutive mixture of speech signals based on dynamic modeling.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2013

A frequency domain MVDR beamformer for UWB microwave breast cancer imaging in dispersive mediums.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2013

Speech analysis/synthesis by Gaussian mixture approximation of the speech spectrum for voice conversion.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2013

Voice conversion based on State Space Model and considering global variance.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2013

2012
Single-Microphone LP Residual Skewness-Based Inverse Filtering of the Room Impulse Response.
IEEE Trans. Speech Audio Process., 2012

Efficient Frequency Domain Implementation of Noncausal Multichannel Blind Deconvolution for Convolutive Mixtures of Speech.
IEEE Trans. Speech Audio Process., 2012

Single channel speech separation in modulation frequency domain based on a novel pitch range estimation method.
EURASIP J. Adv. Signal Process., 2012

Binaural speech separation based on the time-frequency binary mask.
Proceedings of the 6th International Symposium on Telecommunications, 2012

A hybrid coherent-incoherent method of modulation filtering for Single Channel Speech Separation.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

2011
Split vector quantization for sinusoidal amplitude and frequency.
J. Zhejiang Univ. Sci. C, 2011

Phase-Only Speech Reconstruction Using Very Short Frames.
Proceedings of the 12th Annual Conference of the International Speech Communication Association, 2011

Quality Improvement of Voice Conversion Systems Based on Trellis Structured Vector Quantization.
Proceedings of the 12th Annual Conference of the International Speech Communication Association, 2011

Convolutive blind source separation based on GDFT filterbanks and pre-determined subband whitening.
Proceedings of the 19th European Signal Processing Conference, 2011

Examination of convolutive blind source separation algorithms based on information theoretic criterion and second-order statistics for cell-phone application.
Proceedings of the 24th Canadian Conference on Electrical and Computer Engineering, 2011

2010
Evaluating single-channel speech separation performance in transform-domain.
J. Zhejiang Univ. Sci. C, 2010

Blind Source Separation in nonminimum-phase systems based on filter decomposition.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2010

Speech segmentation using a hypothesis test based on Random Matrix Theory.
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2010

2009
FDMSM robust signal representation for speech mixtures and noise corrupted audio signals.
IEICE Electron. Express, 2009

2007
Real-Time Cardiac Arrhythmia Detection Using WOLA Filterbank Analysis of EGM Signals.
EURASIP J. Adv. Signal Process., 2007

2006
Cardiac Rhythm Detection and Classification by WOLAFilterbank Analysis of EGM Signals.
Proceedings of the 28th International Conference of the IEEE Engineering in Medicine and Biology Society, 2006

2005
Complexity reduction of partial update oversampled subband adaptive algorithms by selective pruning of polyphase components.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

Partial update subband implementation of complex pseudo-affine projection algorithm on oversampled filterbanks.
Proceedings of the 2005 IEEE International Conference on Acoustics, 2005

Low-resource delayless subband adaptive filter using weighted overlap-add.
Proceedings of the 13th European Signal Processing Conference, 2005

Low-power implementation of a subband Fast Affine Projection algorithm for acoustic echo cancellation.
Proceedings of the 13th European Signal Processing Conference, 2005

2004
A hybrid subband adaptive system for speech enhancement in diffuse noise fields.
IEEE Signal Process. Lett., 2004

An acoustic shock limiting algorithm using time and frequency domain speech features.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

An energy normalization scheme for improved robustness in speech recognition.
Proceedings of the 8th International Conference on Spoken Language Processing, 2004

Low-power implementation of the Bluetooth subband audio codec.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Complexity reduction and regularization of a fast affine projection algorithm for oversampled subband adaptive filters.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

Near-end distortion in over-sampled subband adaptive implementation of affine projection algorithm.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

Sequential LMS for low-resource subband adaptive filtering: Oversampled implementation and polyphase analysis.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

A polyphase model for Fast Affine Projection with Partial Filter Update.
Proceedings of the 2004 12th European Signal Processing Conference, 2004

2003
Subband-based acoustic shock limiting algorithm on a low-resource DSP system.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

Convergence improvement for oversampled subband adaptive noise and echo cancellation.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

An efficient front-end for automatic speech recognition.
Proceedings of the 2003 10th IEEE International Conference on Electronics, 2003

ETSI AMR-2 VAD: evaluation and ultra low-resource implementation.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

Affine projection algorithm for oversampled subband adaptive filters.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
Highly oversampled subband adaptive filters for noise cancellation on a low-resource DSP system.
Proceedings of the 7th International Conference on Spoken Language Processing, ICSLP2002, 2002

A low-resource, miniature implementation of the ETSI distributed speech recognition front-end.
Proceedings of the 7th International Conference on Spoken Language Processing, ICSLP2002, 2002

Real-time speech synthesis on an ultra low-resource, programmable DSP system.
Proceedings of the IEEE International Conference on Acoustics, 2002

An ultra low power, ultra miniature voice command system based on Hidden Markov Models.
Proceedings of the IEEE International Conference on Acoustics, 2002

A subband beamformer on an ultra low-power miniature DSP platform.
Proceedings of the IEEE International Conference on Acoustics, 2002

2001
An improved wavelet-based speech enhancement system.
Proceedings of the EUROSPEECH 2001 Scandinavia, 2001

2000
Objective long-term assessment of speech quality changes in pre-lingual cochlear implant children.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

A rule-based approach to farsi language text-to-phoneme conversion.
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000

A hybrid speech enhancement system based on HMM and spectral subtraction.
Proceedings of the IEEE International Conference on Acoustics, 2000

1999
A layered neural network interfaced with a cochlear model for the study of speech encoding in the auditory system.
Comput. Speech Lang., 1999

Farsi language prosodic structure, research and implementation using a speech synthesizer.
Proceedings of the Sixth European Conference on Speech Communication and Technology, 1999

1998
Speech analysis and recognition using interval statistics generated from a composite auditory model.
IEEE Trans. Speech Audio Process., 1998

HMM-based strategies for enhancement of speech signals embedded in nonstationary noise.
IEEE Trans. Speech Audio Process., 1998

1995
Real-time implementation of HMM-based MMSE algorithm for speech enhancement in hearing aid applications.
Proceedings of the 1995 International Conference on Acoustics, 1995

1994
Waveform-based speech recognition using hidden filter models: parameter selection and sensitivity to power normalization.
IEEE Trans. Speech Audio Process., 1994

Comparative performance of spectral subtraction and HMM-based speech enhancement strategies with application to hearing and design.
Proceedings of ICASSP '94: IEEE International Conference on Acoustics, 1994


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