Seyed Mohammad Ahadi

According to our database1, Seyed Mohammad Ahadi authored at least 61 papers between 1999 and 2023.

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

Timeline

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Links

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Bibliography

2023
New sparsity measure based on energy distribution.
Displays, December, 2023

A Novel Jointly Optimized Cooperative DAE-DNN Approach Based on a New Multi-Target Step-Wise Learning for Speech Enhancement.
IEEE Access, 2023

2022
Correction: N-dimensional N-microphone sound source localization.
EURASIP J. Audio Speech Music. Process., 2022

2019
Multi-target tracking using CNN-based features: CNNMTT.
Multim. Tools Appl., 2019

2018
Robust Recognition of Noisy Speech Through Partial Imputation of Missing Data.
Circuits Syst. Signal Process., 2018

2017
Spectro-temporal Power Spectrum Features for Noise Robust ASR.
Circuits Syst. Signal Process., 2017

Speech enhancement using sparse dictionary learning in wavelet packet transform domain.
Comput. Speech Lang., 2017

2016
A novel speech enhancement method by learnable sparse and low-rank decomposition and domain adaptation.
Speech Commun., 2016

Designing efficient discriminant functions for multi-category classification using evolutionary methods.
Neurocomputing, 2016

Robust color image segmentation using fuzzy c-means with weighted hue and intensity.
Digit. Signal Process., 2016

New features for speech enhancement using bivariate shrinkage based on redundant wavelet filter-banks.
Comput. Speech Lang., 2016

Bounded cepstral marginalization of missing data for robust speech recognition.
Comput. Speech Lang., 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
On the distribution of Mel-filtered log-spectrum of speech in additive noise.
Speech Commun., 2015

Robust voice activity detection directed by noise classification.
Signal Image Video Process., 2015

Modified coherence-based dictionary learning method for speech enhancement.
IET Signal Process., 2015

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

N-dimensional N-microphone sound source localization.
EURASIP J. Audio Speech Music. Process., 2013

On the Importance of Pre-emphasis and Window Shape in Phase-Based Speech Recognition.
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

A new phase-based feature representation for robust speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2013

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

Real Time High Accuracy 3-D PHAT-Based Sound Source Localization Using a Simple 4-Microphone Arrangement.
IEEE Syst. J., 2012

2011
Phase-Only Speech Reconstruction Using Very Short Frames.
Proceedings of the INTERSPEECH 2011, 2011

A new group delay-based feature for robust speech recognition.
Proceedings of the 2011 IEEE International Conference on Multimedia and Expo, 2011

TDE-ILD-HRTF-Based 3D entire-space sound source localization using only three microphones and source counting.
Proceedings of the International Conference on Electrical Engineering and Informatics, 2011

Spectrally correlated sound source localization in a noisy envirnoment based on a new mask for DUET.
Proceedings of the 8th International Conference on Information, 2011

Environmental sound classification using spectral dynamic features.
Proceedings of the 8th International Conference on Information, 2011

2010
A Logarithmic Quantization Index Modulation for Perceptually Better Data Hiding.
IEEE Trans. Image Process., 2010

A Robust Image Watermarking in the Ridgelet Domain Using Universally Optimum Decoder.
IEEE Trans. Circuits Syst. Video Technol., 2010

An Approach to Detection of High Impedance Fault Using Discrete Wavelet Transform and Artificial Neural Networks.
Simul., 2010

A New Subband-Weighted MVDR-Based Front-End for Robust Speech Recognition.
IEICE Trans. Inf. Syst., 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

Objective evaluation of phase and magnitude only reconstructed speech: New considerations.
Proceedings of the 10th International Conference on Information Sciences, 2010

Reconstructing missing speech spectral components using both temporal and statistical correlations.
Proceedings of the 10th International Conference on Information Sciences, 2010

2009
Robust Multiplicative Patchwork Method for Audio Watermarking.
IEEE Trans. Speech Audio Process., 2009

Bernoulli versus Markov: Investigation of state transition regime in switching-state acoustic models.
Signal Process., 2009

Hybrid statistical pronunciation models designed to be trained by a medium-size corpus.
Comput. Speech Lang., 2009

A new approach for spoken language identification based on sequence kernel SVMs.
Proceedings of the 16th International Conference on Digital Signal Processing, 2009

Logarithmic Quantization Index Modulation: A perceptually better way to embed data within a cover signal.
Proceedings of the IEEE International Conference on Acoustics, 2009

2008
Stressed speech recognition using a warped frequency scale.
IEICE Electron. Express, 2008

Kernel-Based Speaker Clustering for Rapid Speaker Adaptation.
Proceedings of the Fifth International Conference on Information Technology: New Generations (ITNG 2008), 2008

Speaker diarization in a multi-speaker environment using particle swarm optimization and mutual information.
Proceedings of the 2008 IEEE International Conference on Multimedia and Expo, 2008

A universally optimum decoder for multiplicative audio watermarking.
Proceedings of the 2008 IEEE International Conference on Multimedia and Expo, 2008

Vector Quantization Index Modulation watermarking using concentric hyperspherical codebooks.
Proceedings of the IEEE International Conference on Acoustics, 2008

Time-inhomogeneous hidden Bernoulli model: An alternative to hidden Markov model for automatic speech recognition.
Proceedings of the IEEE International Conference on Acoustics, 2008

Robust weighted fuzzy c-means clustering.
Proceedings of the FUZZ-IEEE 2008, 2008

2007
Features based on filtering and spectral peaks in autocorrelation domain for robust speech recognition.
Comput. Speech Lang., 2007

Adaptive ARMA filtering and energy normalization for robust speech recognition.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

Consideration of correlation between noise and clean speech signals in autocorrelation-based robust speech recognition.
Proceedings of the 9th International Symposium on Signal Processing and Its Applications, 2007

Front-end post-processing using histogram equalization combined with ARMA filtering for noise robust speech recognition.
Proceedings of the 15th European Signal Processing Conference, 2007

Improved Autocorrelation-based Noise robust speech recognition using kernel-based cross correlation and overestimation parameters.
Proceedings of the 15th European Signal Processing Conference, 2007

2006
Robust Feature Extraction of Speech Via Noise Reduction in Autocorrelation Domain.
Proceedings of the Multimedia Content Representation, 2006

Robust feature extraction based on spectral peaks of group delay and autocorrelation function and phase domain analysis.
Proceedings of the INTERSPEECH 2006, 2006

Use of Spectral Peaks in Autocorrelation and Group Delay Domains for Robust Speech Recognition.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

2004
Use of formants in stressed and unstressed continuous speech recognition.
Proceedings of the INTERSPEECH 2004, 2004

An energy normalization scheme for improved robustness in speech recognition.
Proceedings of the INTERSPEECH 2004, 2004

Evaluation of the effect of stress on formants in Farsi vowels.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

2003
Statistical evaluation of the influence of stress on pitch frequency and phoneme durations in farsi language.
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

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


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