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19 results on '"Djeradi, Amar."'

5. Behavior of speech acoustics parameters and compensatory strategies adopted in noised environment

Author

Falek, Leila, Teffahi, Hocine, Djeradi, Amar, System, HAL, and Société Française d'Acoustique

Subjects
[SPI.ACOU] Engineering Sciences [physics]/Acoustics [physics.class-ph]
Abstract

We acoustically analyzed behavior of speech signal produced in noisy constraint by four speakers, when noise is sent by a helmet to speaker and when noise is sent by high speaker. The goal is to find speech signal acoustic parameters which are most sensitive to noise and the compensatory strategies adopted by the speakers to counter this constraint. Results obtained in both cases show a great constraint influence over timing, energy, fundamental frequency F0, formants (F1, F2) of the analyzed units. They can be summarized such as: (1) In temporal field: We noted an increase in analyzed units duration. We can conclude that speaker tends to slow down his elocution rhythm with constraint in order to better render comprehensible itself; (2) In energy field: there is an increase in intensity. Thus the speaker speaks extremely in noise constraint; (3) In frequential field: (a) We noted an increase in formants F0 and F1 but F2 stay stable in spite of noise constraint. Its meaning that the speakers, while speaking extremely, increase their aperture but the vowels anteriority does not change. (b) Vocalic spaces are reduced with centralization. Its meaning that the speakers modify their formants to avoid constraint while remaining understandable.

Published
2012

6. Labial constraint effects on the speech signal acoustic parameters

Author

Falek, Leila, Teffahi, Hocine, Djeradi, Amar, System, HAL, and Société Française d'Acoustique

Subjects
[SPI.ACOU] Engineering Sciences [physics]/Acoustics [physics.class-ph]
Abstract

We studied in this work the influence of labial constraint in speech production. We acoustically analyzed vowels behavior in constraint, then the constraint influence on the coarticulation on 4 speakers. Results obtained show a great influence of the constraint on the acoustic parameters in particular on the frequential parameters which are key parameters in synthesis, recognition and coding systems of the speech and in orthophonics reeducation programmes for handicapped persons. These results can be summarized such as: In the temporal field: A temporal reduction in the analyzed units, therefore the speaker tends to accelerate his rate/rhythm of elocution with the constraint. In the frequential field: (1) Spaces vocalic F2 (F1) are reduced but without centralization of the vowels. Thus the speakers reach with difficulty their acoustic targets with the constraint to see not at all; (2) there is variation of fundamental frequency according to the formants F0 (F1), F0 (F2) and F0 (F3). However direction of variation depends on speaker. Thus constraint varies the intonation; (3) locus equation slope obtained increase all with constraint. Thus labial constraint increases coarticulation degree.

Published
2012

7. Audiovisual Speaker Identification Based on Lip and Speech Modalities.

Author

Chelali, Fatma and Djeradi, Amar

Subjects
VOICEPRINTS, AUDIOVISUAL materials, DATA fusion (Statistics), DISCRETE cosine transforms, LINEAR predictive coding
Abstract

In this article, we present a bimodal speaker identification method, which integrates both acoustic and visual features and where the two audiovisual stream modalities are processed in parallel. We also propose a fusion technique that combines the two modalities to make the final recognition decision. Experiments are conducted on an audiovisual dataset containing the 28 Arabic syllables pronounced by ten speakers. Results show the importance of the visual information that is provided by Discrete Cosine Transform (DCT) and Discrete Wavelet Transform (DWT) in addition to the audio information corresponding to the Mel Frequency Cepstral Coefficients (MFCC) and Perceptual Linear Predictive (PLP). Furthermore, some artificial neural networks such as Multilayer Perceptron (MLP) and Radial Basis Function (RBF) were investigated and tested successfully in this dataset by presenting good recognition performances with serial concatenation for the acoustic and visual vectors. [ABSTRACT FROM AUTHOR]

Published
2017

15. A study method of noise effects on the acoustic parameters of speech signal.

Author

Falek, Leila, Teffahi, Hocine, and Djeradi, Amar

Abstract

This study is part of adaptive mechanisms research in speech production. We were interested to acoustic variations of the voice message when the speaker is placed in a noisy environment. We propose a methodology to highlight the articulatory strategies adopted by the speaker to adapt to the noisy environment. This methodology consists of two parts: a noise effect analysis on the timing in the one hand and on the frequency envelope of the speech signal in the other. In addition, we also highlighted the strategies adopted in the event of significant decrease of auditory feedback in the presence of noise [1]. The results are considered good because they rejoin those found in literature[1][22]. Furthermore, the methodology proposed in this study can be used in rehabilitation for the deaf (test the degree of hearing loss before and after treatment), in communication systems (level estimation of the signal alteration by noise before its transmission, in coding speech, recognition, synthesis etc…) [ABSTRACT FROM PUBLISHER]

Published
2012
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16. MFCC and vector quantization for Arabic fricatives speech/speaker recognition.

Author

Chelali, Fatma Zohra. and Djeradi, Amar.

Abstract

This article develops a speaker-dependent Arabic phonemes recognition system using MFCC analysis and the VQ-LBG algorithm. The system is examined with and without vector quantization in order to analyze the effect of compression in an acoustic parameterization phase. Our experimental results show that vector quantization using a codebook of size 16 achieves good results compared to the system without quantization for a majority of the phonemes studied. [ABSTRACT FROM PUBLISHER]

Published
2012
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18. Speaker Identification System based on PLP Coefficients and Artificial Neural Network.

Author

Chelali, Fatma zohra, Djeradi, Amar, and Djeradi, Rachida

Subjects
SPEECH perception, ARTIFICIAL neural networks, FEATURE extraction, PHONEME (Linguistics), LINGUISTICS
Abstract

Speaker recognition is the process of automatically recognizing who is speaking on the basis of individual information included in speech waves. Feature extraction for speech recognition is a subject of a major interest today; different features have been investigated in speech recognition systems. The perceptual linear predictive PLP: this technique uses three concepts from the psychophysics of hearing to derive an estimate of the auditory spectrum: (1) the critical-band spectral resolution, (2) the equal loudness curve, and (3) the intensity-loudness power law. This paper discusses the development of a speaker identification and phoneme classification system. In particular, we develop an artificial neural network: multilayer perceptron MLP using PLP coefficients of voice signal. The performance of the system has been tested in experiments using 14 Arabic phonemes, specifically the Arabic fricatives uttered by 4 Algerian native speakers. Our results demonstrates the efficiency of the PLP-MLP algorithm, a good recognition rate was obtained. [ABSTRACT FROM AUTHOR]

Published
2011

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