Meriem Jaïdane

Orcid: 0000-0002-9096-6161

According to our database1, Meriem Jaïdane authored at least 65 papers between 1988 and 2019.

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

Timeline

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Bibliography

2019
Random forest-based approach for physiological functional variable selection for driver's stress level classification.
Stat. Methods Appl., 2019

Self-similarity analysis of vehicle driver's electrodermal activity.
Qual. Reliab. Eng. Int., 2019

2018
Perceptually Controlled Reshaping of Sound Histograms.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

Watermark-Driven Acoustic Echo Cancellation.
IEEE ACM Trans. Audio Speech Lang. Process., 2018

Enhancing speech intelligibility in reverberant spaces by a speech features distributions dependent pre-processing.
Int. J. Speech Technol., 2018

AffectiveROAD system and database to assess driver's attention.
Proceedings of the 33rd Annual ACM Symposium on Applied Computing, 2018

2017
Audio texturedness indicator based on a direct and reverse short listening time analysis.
Multim. Tools Appl., 2017

2016
Late pre-dereverberation for speech intelligibility enhancement in public address systems.
Proceedings of the International Symposium on Signal, Image, Video and Communications, 2016

Texturedness decision time for audio texturedness indicator.
Proceedings of the International Symposium on Signal, Image, Video and Communications, 2016

Dynamic range variability of sound energy decay for reverberation time estimation in railway noisy environments.
Proceedings of the 2nd International Conference on Advanced Technologies for Signal and Image Processing, 2016

2015
Selection of a Closed-Form Expression Polynomial Orthogonal Basis for Robust Nonlinear System Identification.
J. Signal Process. Syst., 2015

Intelligibility enhancement of vocal announcements for public address systems: a design for all through a presbycusis pre-compensation filter.
Proceedings of the INTERSPEECH 2015, 2015

Temporal entropy-based texturedness indicator for audio signals.
Proceedings of the 2015 IEEE International Conference on Acoustics, 2015

2014
Non-Negative Blind Source Separation Algorithm Based on Minimum Aperture Simplicial Cone.
IEEE Trans. Signal Process., 2014

Nonlinear Audio Systems Identification Through Audio Input Gaussianization.
IEEE ACM Trans. Audio Speech Lang. Process., 2014

2012
Arabic adaptation of Phonology and Memory test using entropy-based analysis of word complexity.
Proceedings of the 11th International Conference on Information Science, 2012

Simplicial Cone Shrinking Algorithm for Unmixing Nonnegative Sources.
Proceedings of the 2012 IEEE International Conference on Acoustics, 2012

Geometrical Method Using Simplicial Cones for Overdetermined Nonnegative Blind Source Separation: Application to Real PET Images.
Proceedings of the Latent Variable Analysis and Signal Separation, 2012

2011
Trend Extraction for seasonal Time Series Using Ensemble Empirical Mode Decomposition.
Adv. Data Sci. Adapt. Anal., 2011

Presbyacousis and stress evaluation in urban settings.
Proceedings of the 4th International Symposium on Applied Sciences in Biomedical and Communication Technologies, 2011

Regularized Gradient algorithm for Non-Negative Independent Component Analysis.
Proceedings of the IEEE International Conference on Acoustics, 2011

An improved scheme of audio watermarking based on turbo codes and channel effect modeling.
Proceedings of the IEEE International Conference on Acoustics, 2011

How does the clipped LMS outperform the LMS?
Proceedings of the 19th European Signal Processing Conference, 2011

2010
Sub-quantization/orthogonalization and optimization of algorithm-architecture adequacy for optimal polynomial filtering.
Proceedings of the 18th European Signal Processing Conference, 2010

2009
Revisiting quantization theorem through audiowatermarking.
Proceedings of the IEEE International Conference on Acoustics, 2009

Generalized Gaussian mixture model.
Proceedings of the 17th European Signal Processing Conference, 2009

Experimental mappings and validation of the dependence on the language of objective speech quality scores in actual GSM network conditions.
Proceedings of the 17th European Signal Processing Conference, 2009

2008
Finite alphabet generator with parameterized Markov chain transition matrix.
Proceedings of the 15th IEEE International Conference on Electronics, Circuits and Systems, 2008

Weighted criteria for RF power amplifiers identification in wide-band context.
Proceedings of the 15th IEEE International Conference on Electronics, Circuits and Systems, 2008

2007
Turbo code based detection for audio watermarking: the generalized Gaussian noise channel model.
Proceedings of the 14th IEEE International Conference on Electronics, 2007

Adaptive Decision Feedback Equalizer for SUI channel models.
Proceedings of the 15th European Signal Processing Conference, 2007

An hybrid approach of low frequency room equalization: Notch filters based on common acoustical pole modeling.
Proceedings of the 15th European Signal Processing Conference, 2007

"Gaussianization" Method for identification of memoryless nonlinear audio systems.
Proceedings of the 15th European Signal Processing Conference, 2007

Visually-based audio texture segmentation for audio scene analysis.
Proceedings of the 15th European Signal Processing Conference, 2007

2006
A new nonstationary LMS algorithm for tracking Markovian time varying systems.
Signal Process., 2006

Ultrasound Backscatter Characterization by Using Markov Random Field Model.
Proceedings of the 2006 IEEE International Conference on Acoustics Speech and Signal Processing, 2006

Speech processing in the watermarked domain: Application in adaptive Acoustic Echo Cancellation.
Proceedings of the 14th European Signal Processing Conference, 2006

Adaptive subband Notch Filter for RFI cancellation in low interference to signal ratio.
Proceedings of the 14th European Signal Processing Conference, 2006

2005
Audio watermarking: a way to stationnarize audio signals.
IEEE Trans. Signal Process., 2005

A non stationary/infinite precision system analogy for fixed-point digital adaptive filter analysis.
Proceedings of the 12th IEEE International Conference on Electronics, 2005

A robust non-uniform LUT indexing method in digital predistortion linearization of RF power amplifiers.
Proceedings of the 13th European Signal Processing Conference, 2005

2004
Performance Analysis of Adaptive Volterra Filters in the Finite-Alphabet Input Case.
EURASIP J. Adv. Signal Process., 2004

A new Wiener filtering based detection scheme for time domain perceptual audio watermarking.
Proceedings of the 2004 IEEE International Conference on Acoustics, 2004

2003
Introduction of the CELP structure of the GSM coder in the acoustic echo canceller for the GSM network.
Proceedings of the 8th European Conference on Speech Communication and Technology, EUROSPEECH 2003, 2003

Watermarking influence on the stationarity of audio signals.
Proceedings of the 2003 IEEE International Conference on Acoustics, 2003

2002
Audiowatermark detection for all-pass pirat attack: Hybrid blind equalization/Wiener deconvolution approach.
Proceedings of the 11th European Signal Processing Conference, 2002

Combined acoustic echo canceller for the GSM network.
Proceedings of the 11th European Signal Processing Conference, 2002

2001
Exact performances analysis of a selective coefficient adaptive algorithm in acoustic echo cancellation.
Proceedings of the IEEE International Conference on Acoustics, 2001

2000
On exact convergence results of adaptive filters: the finite alphabet case.
Signal Process., 2000

On exact performances of adaptive Volterra filters: the finite alphabet case.
Proceedings of the IEEE International Symposium on Circuits and Systems, 2000

Design of blind decision feedback equalizers for Markovian time varying channels.
Proceedings of the IEEE International Conference on Acoustics, 2000

Does white input give better convergence speed in adaptive filtering?
Proceedings of the 10th European Signal Processing Conference, 2000

On the robustness of adaptive predictive scheme for tracking randomly time-varying channels.
Proceedings of the 10th European Signal Processing Conference, 2000

1999
Exact convergence analysis of affine projection algorithm: the finite alphabet inputs case.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

A non-stationary RLS algorithm for adaptive tracking of Markov time varying channel.
Proceedings of the 1999 IEEE International Conference on Acoustics, 1999

1998
A finite memory non stationary LMS algorithm for adaptive tracking Markovian time-varying channel.
Proceedings of the 5th IEEE International Conference on Electronics, Circuits and Systems, 1998

Comparison of based adaptive predictive schemes for improvement of tracking randomly time-varying systems.
Proceedings of the 5th IEEE International Conference on Electronics, Circuits and Systems, 1998

Exact analysis of the tracking capability of time-varying channels: the finite alphabet inputs case.
Proceedings of the 5th IEEE International Conference on Electronics, Circuits and Systems, 1998

1997
Best input for optimal tracking randomly time-varying systems: justification of adaptive predictive structure.
Proceedings of the 1997 IEEE International Conference on Acoustics, 1997

1996
A non stationary LMS algorithm for adaptive tracking of a Markov time-varying system.
Proceedings of the 8th European Signal Processing Conference, 1996

Soft decision solution to ill convergence of blind decision feedback equalizers.
Proceedings of the 8th European Signal Processing Conference, 1996

1995
Blind cancellation of intersymbol interference in decision feedback equalizers.
Proceedings of the 1995 International Conference on Acoustics, 1995

1990
Theoretical analysis of the ADPCM CCITT algorithm.
IEEE Trans. Commun., 1990

1989
Mistracking in successive PCM/ADPCM transcoders.
IEEE Trans. Commun., 1989

1988
Stability of adaptive recursive filters.
Proceedings of the IEEE International Conference on Acoustics, 1988


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