Back to Researchers List

M. AHCENE Abed

MCA

Directory of teachers

Department

BASE COMMON ST Departement ST

Research Interests

Automatic Speech Processing SIW technology applications in telecommunication systems

Contact Info

University of M'Sila, Algeria

Email : abed.ahcene@univ-msila.dz

On the Web:

Google Scholar N/A
ResearchGate N/A
ORCID N/A
SC
Scopus N/A

Recent Publications

2024-09-29

New Low-Complexity Selected Active Coefficients Adaptive Sparse Algorithm for Teleconferencing Systems and Identification

Addressing the challenge of sparse acoustic channel is important in communication systems such as tele- and video-conferencing systems. Sparse impulse response plays an efficient role in acoustic identification systems, particularly in long acoustic rooms. This article presents an enhanced version of the Improved Proportionate Normalized Least-Mean-Squares (IP-NLMS) algorithm. It adapts only the active coefficients, thereby reducing the computational complexity of IP-NLMS algorithm. The selected active coefficients of the IP-NLMS algorithm (SAC-IP-NLMS) are proposed exactly in sparse impulse response (SIR) in order to reduce the computational complexity with faster convergence rate. Several simulations conducted across various sparse environments - based on the time evolution of error signals, the mean square error, the echo return loss enhancement and the computational complexity - validate the effectiveness of the proposed algorithm.

Citation

M. AHCENE Abed, (2024-09-29), "New Low-Complexity Selected Active Coefficients Adaptive Sparse Algorithm for Teleconferencing Systems and Identification", [national] Jordan Journal of Electrical Engineering , Scientific Research and Innovation Support Fund (SRISF)

Read Full Paper

2024-06-26

New Variable Selected Coefficients Adaptive Sparse Algorithm for Acoustic System Identification

In communication systems as public switched telephone networks and tele-and-vision-conferencing system, addressing the challenge of sparse acoustic echo is of paramount importance. The sparse impulse response identification is very essential in acoustic echo cancellation systems (AEC) exactly in sparse acoustic environments. This paper introduces an enhanced improved proportionate normalized-least-mean-square (IP-NLMS) algorithm, utilizing efficient variable step-size parameters and adapting only the active coefficients based on selection bloc for reducing the computational complexity. The proposed Variable Selection Coefficients IP-NLMS algorithm (VSC-IP-NLMS) focuses on adapting the selected active coefficients of the sparse impulse response (SIR), in order to both accuracy and convergence speed. Extensive simulations conducted under various sparse environments confirm the efficacy of the proposed algorithm. As important characteristic of this proposed VSC-IP-NLMS, it achieves these remarkable results with significantly reduced computational complexity compared to sparse and variable adaptive filtering algorithms, offering a promising avenue for improving the quality of communication systems.

Citation

M. AHCENE Abed, (2024-06-26), "New Variable Selected Coefficients Adaptive Sparse Algorithm for Acoustic System Identification", [national] Traitement du Signal , International Information and Engineering Technology Association (IIETA)

Read Full Paper

2024-06-01

Improved Proportionate Symmetric Backward Adaptive Speech Enhancement Approach

This research focuses on the development of speech enhancement techniques for two-channel audio systems. Specifically, we explore the utilization of an efficient sparseness recursive algorithm to tackle this challenge. The algorithm is designed to identify and attenuate noise components present in the audio signals, with the aim of improving the overall audio quality. In this investigation, we propose innovative approaches and enhancements to the sparseness recursive normalized least mean square (NLMS) algorithm, denoted Backward µ-law Proportionate NLMS (BMPNLMS), making it more suitable and effective for two-channel speech enhancement. By capitalizing on the sparsity properties of the audio signals, techniques proposed in this paper aim to enhance the desired audio while suppressing unwanted noise. Performance of the presented algorithm was examined by rigorous experiments based on several criteria. The obtained results thoroughly confirm the effectiveness of the proposed approach in real-world situations.

Citation

M. AHCENE Abed, (2024-06-01), "Improved Proportionate Symmetric Backward Adaptive Speech Enhancement Approach", [national] Jordan Journal of Electrical Engineering , Scientific Research and Innovation Support Fund (SRISF)

Read Full Paper

2024-04-21

SIW structure optimization for 5G waveguide design

Modern 5G telecommunications systems tend to use higher frequencies. The design of such devices is becoming increasingly miniaturized for more powerful applications and technologies. In this context, Substrate Integrated Waveguides (SIW) replace conventional waveguides. Their design is mainly based on mathematical models. However, the resulting model presents a frequency shift with respect to the working frequency. In this paper we present an optimization method for ensuring the good performance of waveguides, based on SIW technology. These waveguides are easily integrated into 5G telecommunication systems. We'll then look at the different results obtained after changing the variables asiw and h). The direct waveguide had the lowest frequency shift (|Δf|=0.11GHz) at asiw =3.6mm, whereas the indirect waveguide had the best frequency shift (|Δf|=0.01GHz) at asiw =3.25mm.

Citation

M. AHCENE Abed, (2024-04-21), "SIW structure optimization for 5G waveguide design", [international] 2024 8th International Conference on Image and Signal Processing and their Applications (ISPA) , University Mohamed Khider of Biskra

Read Full Paper

2024-02-13

Design of low Profile, High Bandwidth Circular Patch Antenna at 28 GHz for Fifth Generation Wireless Technology

This article introduces a novel technique to develop a circular patch antenna specifically tailored for 5G wireless technology operating in the 28 GHz frequency range. By thoughtfully incorporating a rectangular slot into the design on a Rogers substrate, not only does the antenna's performance get enhanced, but its bandwidth also gets expanded. This expansion in bandwidth allows the antenna to effectively cover a broader frequency range, which holds immense significance for practical applications. The compact design occupies only 8.3×8.5 mm 2 of space. Through simulations and measurements, it is demonstrated that the antenna achieves a 10 dB higher gain, ensuring a significant improvement in signal power.It delivers a bandwidth of 2.23 GHz, extending across the spectrum from 27.77 GHz to 30 GHz. The resonance analysis reveals a good return loss of-34.3 dB, indicating excellent signal matching. Furthermore, the antenna achieves a voltage standing wave ratio (VSWR) of 1.11, facilitating the effective transfer of radio frequency energy. The results indicate satisfactory performance in terms of matching, losses, and efficiency, positioning this design as a promising candidate for efficient communication in 5G networks.

Citation

M. AHCENE Abed, (2024-02-13), "Design of low Profile, High Bandwidth Circular Patch Antenna at 28 GHz for Fifth Generation Wireless Technology", [international] International Conference on Electronics, Energy and Measurement , University of Medea

Read Full Paper

2024-02-05

New Multi-band MIMO Antennas Based on SIW Technology : Design and Performance Analysis

In telecommunication systems, the antenna is an important element for transmitting information between transmitter system and receiver system. To improve the performance of antennas in terms of coverage area, they are usually grouped into arrays. Generally, if the spacing between arrays elements is less than half a wavelength, a mutual coupling usually occurs, which reduces the adaptability of the system. It is therefore very difficult to install these elements in mobile units of limited size. In this paper, we propose new design of multi-band MIMO antenna based on Substrate Integrated Waveguide (SIW) technology. The model system is tested by simulation in HFSS environment with the conducting copper is produced on substrate of FR4-Epoxy, a dielectric of 4.4 at frequency 2-5 GHz. This paper is devoted to the presentation and analysis of the proposed antenna using SIW technology, and study theirs characteristics exactly in the frequency band [2 − 5GHz]. The simulations are performed using the HFSS simulation software.

Citation

M. AHCENE Abed, (2024-02-05), "New Multi-band MIMO Antennas Based on SIW Technology : Design and Performance Analysis", [international] Second International Conference on Energy Transition and Security (ICETS) , University of Adrar

Read Full Paper

2023-12-12

A Robust Adaptive Filtering Algorithm with Double Talk Detection in-car Communication System

This study focus on solving the acoustic Echo Cancelation (AEC) using adaptive filtering algorithm. The paper introduces as approach to incorporate a double Talk Detection (DTD) mechanism based on Normalized Cross Correlation (NCC) into the Fast Normalized Least Mean Square (FNLMS) algorithm. This integrated algorithm exhibits enhanced convergence speed. The results from simulations validate the effectiveness of this approach in mitigating acoustic echo, particularly in scenarios involving real impulse responses, such as a car cabin impulse response.

Citation

M. AHCENE Abed, HASSANI Islam, BENDOUMIA Rédha, ZOULIKHA Meriem, KOUDDED Elhachemi, , (2023-12-12), "A Robust Adaptive Filtering Algorithm with Double Talk Detection in-car Communication System", [international] The Second International Conference on Energy Transition and Security , Adrar, Algeria

Read Full Paper

2023-06-01

LOW-COST C-BAND SIW BANDPASS FILTER USING FR4-EPOXY SUBSTRATE

This paper describes a substrate integrated waveguide (SIW) filter built on an Fr4-Epoxy substrate with a dielectric constant 4.4 and a height of h=1.6mm. SIW-based devices have piqued the interest of researchers in recent years due to their low loss, small size, and low cost. The goal is to simulate and realize a SIW filter for C-band applications. The designed filter is analyzed using the reflection coefficient S11 and electric field distributions. We used the HFSS simulator. According to the findings, there is a high degree of agreement between the simulated and realized filters. The results also show that the filter has a very good response. It also shows a bandwidth of 6.3GHz around the C-band frequencies. This filter's pass-band ranges from 5.39 to 6.83GHz, with an insertion loss of 4.2dB and a return loss of 45.49dB.

Citation

M. AHCENE Abed, (2023-06-01), "LOW-COST C-BAND SIW BANDPASS FILTER USING FR4-EPOXY SUBSTRATE", [national] Jordanian Journal of Computers and Information Technology , Princess Sumaya University for Technology (PSUT)

Read Full Paper

2023-05-15

Reconfigurable Sierpinski Triangle Fractal Antenna Research and Development

This paper proposes a frequency-configurable (Sierpenski Triangle) fractal antenna. The antenna design is modeled and simulated using HFSS software, and the Sierpenski fractal antenna's reconfigurability is achieved by adding a PIN diode. We investigated four states for the first Reconfigurable order Sierpenski Fractal antenna and five states for the second Reconfigurable order Sierpenski Fractal antenna based on the diode's ON/OFF configuration states. Based on the coefficient comparison, the results show that the proposed Fractal Reconfigurable Antenna and the two proposed structures can be reconfigured in frequency. Furthermore, the antenna S1C3 with a frequency of 𝒇 = 𝟏𝟏.𝟓𝟕𝟗𝟑𝑮𝑯𝒛 adapts best to the first structure, while the antenna S2C5 with a frequency of 𝒇 = 𝟒.𝟗𝟎𝟖𝟐𝑮𝑯𝒛 adapts best to the second structure.

Citation

M. AHCENE Abed, Aissa Amrouche, Ahmed Bouchekhlal, Abdelmalek Birane, Abderrahim Si Saber, , (2023-05-15), "Reconfigurable Sierpinski Triangle Fractal Antenna Research and Development", [international] International Conference on Advances in Electrical and Computer Engineering (ICAECE'2023) , University Larbi Tebessi of Tebessa

Read Full Paper

2022-12-21

Real-Time Detection of Vehicle License Plates Numbers

Object Detection (OD) techniques have emerged as the key to dealing with the most complex computer vision problems in recent years. Vehicle License Plate Detection (VLPD) is the most important stage of any vehicle license plate recognition system (VLPR) because changes in its size, orientation, color, and background, contrast, and resolution have a direct impact on the system's robustness and accuracy. The purpose of this paper is to present an object detector for detecting vehicle license plates in real-world scenes. We developed a new dataset of vehicle license plate numbers and used it to train our custom model. In YOLO-v3 layers, we decreased the number of classes to one in order to improve the detector. When we evaluated the system, we achieved precision, recall, and overall accuracy metrics of 0.95, 0.96, and 92.83 percent, respectively.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Nabil Hezil, Youssouf Bentrcia, , (2022-12-21), "Real-Time Detection of Vehicle License Plates Numbers", [international] 2022 2nd International Conference on New Technologies of Information and Communication (NTIC) , Mila, Algeria

Read Full Paper

2022-11-22

Detection and Localization of Arabic Text in Natural Scene Images

Text identification in natural scenes has emerged as a major research area in computer vision., with several applications including video retrieval and quick translation systems. Text identification is difficult in natural scene images due to changes in text size, color, orientation, background, contrast, and resolution. Text detection in complex backgrounds is more important to us. To address this issue, this work provides a detection and verification approach based on You Only Look Once Yolo (YOLO) algorithm. To get more accurate end findings, we use the YOLO-v4-Tiny approach to develop our custom text detection model. To evaluate the system, we constructed a ground verification system. Our method obtained 94.85% precision for experimental data.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Youssouf Bentrcia, Nabil Hezil, Khadidja Nesrine Boubakeur, Khadidja Ghribi, , (2022-11-22), "Detection and Localization of Arabic Text in Natural Scene Images", [international] International Conference on Computer Communications and Intelligent Systems (I3CIS) , Jijel, Algeria

Read Full Paper

2022-05-08

Vehicle Detection and Tracking in Real-time using YOLOv4-tiny

Vehicle detection and tracking is a popular research topic in Intelligent Transportation Systems. The goal of this paper is to detect, identify, and track vehicles in surveillance camera footage in order for them to be extracted efficiently and accurately. You Only Look Once version 4 (YOLOv4) algorithm is used in this paper to propose a real-time vehicle detection and tracking system. The suggested system has been evaluated using a variety of measures, including accuracy, precision, and recognition recall. For the experimental data, the system attained an accuracy of 96.30 percent and an overall accuracy of 94.17 percent. The results reveal that the suggested system successfully tracks vehicles in the scene.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Youssouf Bentrcia, Nabil Hezil, , (2022-05-08), "Vehicle Detection and Tracking in Real-time using YOLOv4-tiny", [international] International Conference on Image and Signal Processing and their Applications (ISPA) , Mostaganem, Algeria

Read Full Paper

2022-03-29

DNN-Based Arabic Speech Synthesis

This article discusses a Deep Neural Network-based Text-to-Speech synthesis for the Arabic language. Subjective and objective tests were used to evaluate the system. We used the Mean Opinion Score (MOS) for subjective evaluation, and the Diagnostic Rhyme Test (DRT) to test the intelligibility of some consonants and vowels. We use the Perceptual Evaluation of Speech Quality (PESQ) for objective evaluation. The results have a mean of 3.92/5, 3.88/5 for the MOS and DRT tests, respectively, and 3.17/5 for the PESQ test; the majority of words and sentences were recognized, and the system's overall evaluation quality was satisfactory. Furthermore, the results show a significant improvement in the quality of synthesized speech for DNN-based TTS when compared to its HMM-based counterpart.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Youssouf Bentrcia, Khadidja Nesrine Boubakeur, , (2022-03-29), "DNN-Based Arabic Speech Synthesis", [international] International Conference on Electrical and Electronics Engineering (ICEEE) , Istanbul (Turkey)

Read Full Paper

2021-09-01

Balanced Arabic corpus design for speech synthesis

This paper aims to design and validate a phonetically balanced speech corpus for Arabic language. Designing and developing a rich and phonetically balanced corpus in optimal context is one of the key issues in building high quality of text-to-speech synthesis systems. The rich characteristic is in the sense that it must contain all the possible phonemes on the right and left
context, whereas the balanced characteristic is in the sense that it respects the phonetic distribution in the language. We propose a new methodology for designing and implementing such corpus for speech synthesis purposes. The paper explains the whole creation process of this corpus, beginning with the design stage, corpus creation, recording phases, and finally the segmentation of the speech corpus. The speech corpus contains 202 sentences with 6174 phonemes. In order to validate the speech corpus, an Arabic speech synthesis system using Hidden Markov Models has been developed.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Kamel Ferrat, Khadidja Nesrine Boubakeur, Youssouf Bentrcia, Leila Falek, , (2021-09-01), "Balanced Arabic corpus design for speech synthesis", [national] International Journal of Speech Technology , Springer Nature

Read Full Paper

2021-04-09

Voice Quality Evaluation Platform for the Arabic Language

The voice quality evaluation of communication systems is necessary for technical and commercial reasons for the expansion of digital networks, mobile or VoIP and speech synthesis systems. Voice quality can be evaluated using two types of methods: subjective MOS (Mean Opinion Score) and objective PESQ (Perceptual Evaluation Speech Quality). In this work, we propose a study of voice quality evaluation methods; in which a subjective and objective evaluation platform of voice quality is provided. We applied different tests used in the assessment of the speech signal quality to speech synthesis signal for the Arabic Language (GArabic : Generic Arabic). The listeners recognized the majority of changes of consonants and vowels in words and sentences with respective percentages of 98.26%, 92.97%. The comparison between MOS and PESQ tests gave a good correlation coefficient of 0.922.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Khadidja Nesrine Boubakeur, Youssouf Bentrcia, Kamel Ferrat, Leila Falek, , (2021-04-09), "Voice Quality Evaluation Platform for the Arabic Language", [international] International Conference on Electrical and Electronics Engineering (ICEEE) , Istanbul (Turkey)

Read Full Paper

2019-04-16

Arabic Speech Synthesis System Based on HMM

The work presented in this paper is about Text-to-Speech (TTS) synthesis and, more particularly, about statistical speech synthesis using the Hidden Markov Models (HMM). The main objective of this work is to study the functioning of the HMM-based speech synthesis system (HTS) and the implementation of this method to create a system that produces understandable speech output for a given Arabic text. We have done a brief description of the statistical parametric speech synthesis based on HMM, the steps followed to implement this method for Arabic language. Finally, for the evaluations of the system are based on subjective mean opinion score and objective tests. Regarding the intelligibility, naturalness aspects (listening) and the quality (Perceptual Evaluation of Speech Quality (PESQ)).

Citation

M. AHCENE Abed, (2019-04-16), "Arabic Speech Synthesis System Based on HMM", [international] International Conference on Electrical and Electronics Engineering (ICEEE) , Istanbul (Turkey)

Read Full Paper

2017-05-17

New Method for Stemming of Arabic Language Text

Because of its complex morphology, the Arabic language has a very different and difficult structure than other languages. Several stemming approaches that are applied to Arabic language, but a complete stemmer for this language is not available. The existing stem-based stemmers for stemming Arabic text have a poor performance in terms of accuracy and error rates. The aim of this study is to build an effective stemmer that answer the problems of Information Retrieval (IR), and presents new way to build electronic Arabic lexicon by using the most frequency roots as the input of lexicons.

Citation

M. AHCENE Abed, AMROUCHE Aissa, Boubakeur Khadidja Nesrine, , (2017-05-17), "New Method for Stemming of Arabic Language Text", [international] International Conference on Engineering Research & Applications (ICERA-17) , Istanbul (Turkey)

Read Full Paper

Investigation of HTK for Arabic Phonemes Boundary Detection

In this paper we propose an automatic Arabic phonemes boundary detection system. This system is mainly used to perform an automatic speech corpus labeling. Because the manual labeling is a hard task and consumes time. We have used the HTK (Hidden Markov Tools Kit) model to solve this problem. The Hidden Markov Models implementation is used to detect phonemes boundaries with the textual information given by the transcription file. The final system improves a Correct Classification Rate of 89.5%, obtained by 5-HMM of 8 Gaussian components. Keywords: HTK, HMM, Arabic

Citation

M. AHCENE Abed, Aissa Amrouche, Khadidja Nesrine Boubakeur, , (2017-05-17), "Investigation of HTK for Arabic Phonemes Boundary Detection", [international] International Conference on Engineering Research & Applications (ICERA-17) , Istanbul (Turkey)

Read Full Paper

2016-10-26

Automatic segmentation of Arabic speech signals by HMM and ANN

In this paper, we propose an automatic segmentation system of speech into phonemes for the Arabic language. This segmentation is based on two different techniques : Hidden Markov Models (HMM) and Artificial Neural Networks (ANN). Both systems were used to classify the speech signals, extracted from ALGASD corpus (ALGerian Arabic Speech Database), into five classes : fricatives, plosives, nasals, liquids and vowels. These methods achieve important performances with advantage of ANN.

Citation

M. AHCENE Abed, (2016-10-26), "Automatic segmentation of Arabic speech signals by HMM and ANN", [international] 2016 International Conference on Electrical Sciences and Technologies in Maghreb (CISTEM) , Morocco

Read Full Paper

2016-07-01

HMM/GMM Classification for ArticulationDisorder Correction among Algerian Children

In this paper, we propose an automatic classification for Arabic phonemic substitution using a Hidden MarkovModel/Gaussian Mixture Model (HMM/GMM) systems. The main objective is to help Algerian children in the correction ofarticulation problems. Five cases are analyzed in the experiments, 20 Arabic words are recorded by a 20 Algerian children,with age range between 4 and 6 years old. Signals are recorded and stored as wave format with 16kHz as sampling rate, 12Mel Frequency Cepstral Coefficients (MFCC), with their first and second derivates, respectively ∆ and ∆∆ are extracted fromeach signal and used to the training and recognition phases. The proposed system achieved its best accuracy recognition85.73%, with 5-stats HMM when the output function is modelled by a GMM with 8Gaussian components.

Citation

M. AHCENE Abed, (2016-07-01), "HMM/GMM Classification for ArticulationDisorder Correction among Algerian Children", [national] The International Arab Journal of Information Technology , Zarqa University

Read Full Paper

← Back to Researchers List