Hamid Sheikhzadeh
Orcid: 0000-0002-9262-4165
According to our database1,
Hamid Sheikhzadeh
authored at least 80 papers
between 1994 and 2024.
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Bibliography
2024
J. Frankl. Inst., 2024
2023
Parallel and Limited Data Voice Conversion Using Stochastic Variational Deep Kernel Learning.
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
Detection of Transformer Winding Axial Displacement by Kirchhoff and Delay and sum Radar Imaging Algorithms.
CoRR, 2021
2020
IEEE Wirel. Commun. Lett., 2020
IEEE Commun. Lett., 2020
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
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
IEEE Trans. Neural Networks Learn. Syst., 2015
Speech Commun., 2015
2013
The Relevance Sample-Feature Machine: A Sparse Bayesian Learning Approach to Joint Feature-Sample Selection.
IEEE Trans. Cybern., 2013
IET Signal Process., 2013
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
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
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
J. Zhejiang Univ. Sci. C, 2011
Proceedings of the INTERSPEECH 2011, 2011
Quality Improvement of Voice Conversion Systems Based on Trellis Structured Vector Quantization.
Proceedings of the INTERSPEECH 2011, 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
J. Zhejiang Univ. Sci. C, 2010
Proceedings of the IEEE International Symposium on Signal Processing and Information Technology, 2010
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
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
IEEE Signal Process. Lett., 2004
An acoustic shock limiting algorithm using time and frequency domain speech features.
Proceedings of the INTERSPEECH 2004, 2004
Proceedings of the INTERSPEECH 2004, 2004
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
Proceedings of the 2004 12th European Signal Processing Conference, 2004
2003
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
Proceedings of the 2003 10th IEEE International Conference on Electronics, 2003
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003
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
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
Proceedings of the IEEE International Conference on Acoustics, 2002
2001
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
Proceedings of the Sixth International Conference on Spoken Language Processing, 2000
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