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Speech quality assessment methods are necessary for evaluating and documenting treatment outcomes of patients suffering from degraded speech due to Parkinson's disease, stroke, or other disease processes. Subj...
Measuring speech quality by machines overcomes two major drawbacks of subjective listening tests, their low speed and high cost. Real-time, accurate, and economical objective measurement of speech quality open...
A new articulatory synthesizer (SAPWindows), with a modular and flexible design, is described. A comprehensive acoustic model and a new interactive glottal source were implemented. Perceptual tests and simulat...
Auditory modeling is a well-established methodology that provides insight into human perception and that facilitates the extraction of signal features that are most relevant to the listener. The aim of this pa...
Parametric-stereo coding is a technique to efficiently code a stereo audio signal as a monaural signal plus a small amount of parametric overhead to describe the stereo image. The stereo properties are analyze...
We explore the use of physiologically inspired auditory features with both physiologically motivated and statistical audio classification methods. We use features derived from a biophysically defensible model ...
The paper introduces a new objective method for speech quality assessment called Fourier-lapped multilayer perceptron (FLMLP). This method uses an overcomplete transform based on the discrete Fourier transform...
We analyse the properties of the physiological model of the adaptive behaviour of the chemical synapse between inner hair cells (IHC) and auditory neurons. On the basis of the performed analysis, we propose eq...
The acoustic wave received at the ears is processed by the human auditory system to separate different sounds along the intensity, pitch, and timbre dimensions. Conventional Fourier-based signal processing, wh...
We present an example of an anthropomorphic approach, in which auditory-based cues are combined with temporal correlation to implement a source separation system. The auditory features are based on spectral am...
Hidden Markov model (HMM) has been a popular mathematical approach for sequence classification such as speech recognition since 1980s. In this paper, a novel two-channel training strategy is proposed for discr...
Psychoacoustical models have been used extensively within audio coding applications over the past decades. Recently, parametric coding techniques have been applied to general audio and this has created the nee...
Recently, a new signal security system called TDCEA (two-dimensional circulation encryption algorithm) was proposed for real-time multimedia data transmission. This paper gives a comprehensive analysis on the ...
We propose a beamforming scheme for ultrasound imaging leading to the generation of two sets of images, one with oscillations only in the axial direction and one with oscillations only in the lateral direction...
Any digital signal processing algorithm or processor can be reasonably described as a digital filter. The main advantage of an infinite impulse response (IIR) filter is that it can provide a much better perfor...
Textural features are one of the most important types of useful information contained in images. In practice, these features are commonly masked by noise. Relatively little attention has been paid to texture p...
This paper presents efficient algorithms for the analysis of nonstationary multicomponent signals based on modified local polynomial time-frequency transform. The signals to be analyzed are divided into a numb...
A new variation to the RLS algorithm is presented. In the clipped RLS algorithm (CRLS), proposed in updating the filter weights and computation of the inverse correlation matrix, the input signal is quantized ...
Zero crossings or extrema of a wavelet transform constitute important signatures for signal analysis with the advantage of great simplicity. In this paper, we introduce a fast frequency-estimation method based...
This paper presents a method based on fitting a sum-of-exponentials model to the nonuniformly sampled data, for clustering the time series of gene expression data. The structure of the model is estimated by us...
A new algorithm is proposed for updating the weights of an adaptive filter. The proposed algorithm is a modification of an existing method, namely, the clipped LMS, and uses a three-level quantization (
Multiwavelets, wavelets with several scaling functions, offer simultaneous orthogonality, symmetry, and short support, which is not possible with ordinary (scalar) wavelets. These properties make multiwavelets...
Voice morphing is the process of producing intermediate or hybrid voices between the utterances of two speakers. It can also be defined as the process of gradually transforming the voice of one speaker to that...
Error-feedback, normalized, and array-based recursions represent equivalent RLS lattice adaptive filters which are known to offer better numerical properties under finite-precision implementations. This is the...
The first aim of this work is to propose the design of a system-on-chip (SoC) platform dedicated to digital image and signal processing, which is tuned to implement efficiently multiply-and-accumulate (MAC) ve...
We propose a new multiway filtering based on fourth-order cumulants for the denoising of noisy data tensor with correlated Gaussian noise. The classical multiway filtering is based on the TUCKALS3 algorithm th...
Machine perception is a difficult problem both from a practical or implementation point of view as well as from a theoretical or algorithm point of view. Machine perception systems based on biological percepti...
We are concerned with the design, implementation, and validation of a perception SoC based on an ultrasonic array of sensors. The proposed SoC is dedicated to ultrasonic echography applications. A rapid protot...
This contribution presents two spectral amplitude estimators for acoustical background noise suppression based on maximum a posteriori estimation and super-Gaussian statistical modelling of the speech DFT ampl...
A low-power, CMOS retina with real-time, pixel-level processing capabilities is presented. Features extraction and edge-enhancement are implemented with fully programmable 1D Gabor convolutions. An equivalent ...
This paper presents an FPGA-based system for detecting people from video. The system is designed to use JPEG-compressed frames from a network camera. Unlike previous approaches that use techniques such as back...
Image processing requires more computational power and data throughput than most conventional processors can provide. Designing specific hardware can improve execution time and achieve better performance per u...
This paper describes prototyping of a neuroadaptive smart antenna beamforming algorithm using hardware-software implemented RBF neural network and FPGA system-on-programmable-chip (SoPC) approach. The aim is t...
This paper presents an adaptive peak frequency estimation method using a database that stores PARCOR coefficients as key attributes and the corresponding peak frequencies as nonkey attributes. The least-square...
SKiPPER is a SKeleton-based Parallel Programming EnviRonment being developed since 1996 and running at LASMEA Laboratory, the Blaise-Pascal University, France. The main goal of the project was to demonstrate t...
We propose a method and a tool for automatic generation of hardware implementation of a decision rule based on the Adaboost algorithm. We review the principles of the classification method and we evaluate its ...
Manual pocket depth probing has been widely used as a retrospective diagnosis method in periodontics. However, numerous studies have questioned its ability to accurately measure the anatomic pocket depth. In t...
A new scheme of concatenating the block turbo code (BTC) with the space-time block code (STBC) for an OFDM system in dispersive fading channels is investigated in this paper. The good error correcting capabili...
As many coded systems operate at very low signal-to-noise ratios, synchronization becomes a very difficult task. In many cases, conventional algorithms will either require long training sequences or result in ...
In this paper, we derive the Cramér-Rao bound (CRB) for joint carrier phase, carrier frequency, and timing estimation from a noisy linearly modulated signal with encoded data symbols. We obtain a closed-form expr...
We study the equalization procedure in discrete multitone (DMT)-based systems, in particular, in DMT-based ADSL systems. Traditionally, equalization is performed in the time domain by means of a channel shorte...
The geometric interpretation of turbo decoding has founded a framework, and provided tools for the analysis of parallel-concatenated codes decoding. In this paper, we extend this analytical basis for the decod...
We propose an iterative multiuser detector for turbo-coded synchronous and asynchronous direct-sequence CDMA (DS-CDMA) systems. The receiver is derived from the maximum a posteriori (MAP) estimation of the sin...
Whenever variable-length entropy codes are used in the presence of a noisy channel, any channel errors will propagate and cause significant harm. Despite using channel codes, some residual errors always remain...
The demand for low-cost and low-power decoder chips has resulted in renewed interest in low-complexity decoding algorithms. In this paper, a novel theoretical framework for improving the performance of turbo d...
The parallel and serial concatenation of codes is well established as a practical means of achieving excellent performance. In this paper, we introduce the parallel and serial concatenation of single parity-ch...
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Citation Impact 2023
Journal Impact Factor: 1.7
5-year Journal Impact Factor: 1.6
Source Normalized Impact per Paper (SNIP): 0.727
SCImago Journal Rank (SJR): 0.477
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Submission to first editorial decision (median days): 13
Submission to acceptance (median days): 134
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