On The Use of High Order Derivatives for High Performance Alphabet Recognition

Colloque avec actes et comité de lecture. internationale.International audienceIn this paper I propose new feature vectors for automatic speech recognition. They are based on Mel-cepstrum vectors augmented by derivatives. In the literature, many systems using just two derivatives ---delta and delta delta--- are described. But none explores the use of higher order derivatives. This paper presents alphabet recognition results on the Isolet database, using feature vectors containing up to the fifth-order derivatives. For this paper I did not use the HTK toolkit proposed by Cambridge University. I developed my own HMM system. I show that with vectors incorporating all the derivatives up to the fifth one, 97.54% mean recognition accuracy was achieved, result which is comparable to the best published one on this database (97.6%), if the recognition accuracy confidence interval concerning this task (approximately 0.3\%) is taken into account. It is important to note that this result was obtained without segmenting the speech files by an endpoint detection algorithm. This is an unfavourable experimental condition compared to previous published research works. As a consequence, my system is one of the most powerful systems ever implemented for alphabet recognition

Levels of heavy metals in wetland and marine vascular plants and their biomonitoring potential : a comparative assessment

The present study investigated the levels of As, Cd, Cr, Cu, Hg, Mn, Ni, Pb and Zn in the seagrasses Posidonia oceanica and Cymodocea nodosa, and in the wetland macrophytes Phragmites australis, Arundo donax, Typha domingensis, Apium nodiflorum, and Nasturtium officinale. Results showed that the bioaccumulation capacity from sediments, translocation, total levels in plant tissues, and bioindication of metals in sediments, are generally species-specific. In particular, the patterns of metals in the aquatic plants studied were overall independent of ecology (coasts vs wetlands), biomass, anatomy (rhizomatous vs non rhizomatous plants), and life form (hemicrytophytes vs hydrophytes). However, marine phanerogams and wetland macrophytes shared some characteristics such as high levels of heavy metals in their below-ground organs, similar capacity of element translocation in the rhizosphere, compartmentalization of metals in the different plant organs, and potential as bioindicators of Cu, Mn and Zn levels in the substratum. In particular, the present findings indicate that, despite ecological and morphological similarities, different plant species tend to respond differently to exposure to heavy metals. Furthermore, this seems to result from the species individual ability to accumulate and detoxify the various metals rather than being attributed to differences in their ecological and morpho-anatomical characteristics.peer-reviewe

Heat shock stimulation of a tilapia heat shock protein 70 promoter is mediated by a distal element

peer reviewedWe reported previously that a tilapia (Oreochromis mossambicus) heat shock protein 70 (HSP70) promoter is able to confer heat shock response on a reporter gene after transient expression both in cell culture and in microinjected zebrafish embryos. Here we present the first functional analysis of a fish HSP70 promoter, the tiHSP70 promoter. Using transient expression experiments in carp EPC (epithelioma papulosum cyprini) cells and in microinjected zebrafish embryos, we show that a distal heat shock response element (HSE1) at approx. -800 is predominantly responsible for the heat shock response of the tiHSP70 promoter. This element specifically binds an inducible transcription factor, most probably heat shock factor, and a constitutive factor. The constitutive complex is not observed with the non-functional, proximal HSE3 sequence, suggesting that both factors are required for the heat shock response mediated by HSE1

Un Algorithme de Réduction de la Réverbération de Signaux Issus du Vocoder de Phase

Colloque avec actes et comité de lecture. nationale.National audienceTime-Scale modifications of speech signals, based on frequency-domain techniques are hampered by an important artifact called phasiness. This artifact corresponds to the destruction of the shape of the original signal, i.e. the de-synchronisation between the phases of frequency components. This paper describes an algorithm that preserves the shape invariance of speech signals in the context of the phase vocoder. At ICASSP'2001 we presented a first version of this work where phases were corrected at the onsets of the voiced portions of the speech signals. In this study, we extended the previous work by allowing the algorithm to synchronize and correct the phases at regular intervals of the voiced segments of speech signals. Due to our algorithm, modified signals, even for large expansion factors, are of high quality and almost exempt of phasiness. A demonstration is proposed at the web page: http://www.loria.fr/\~jdm/PhaseVocoder/index.html where several audio files can be down-loaded

An Efficient F0 Determination Algorithm Based on the Implicit Calculation of the Autocorrelation of the Temporal Excitation Signal

Colloque avec actes et comité de lecture.In this paper we are presenting a new formulation for determining the fundamental frequency. The evaluation of the pitch is a very difficult problem mainly because of the great variability and irregularity of the speech signals. The algorithm we are presenting is original so far as it relies on the implicit calculation of the autocorrelation of the temporal excitation signal. We have tested our algorithm on the Bagshaw database, created at the Centre for Speech Technology Research at Edinburgh, which is primarily dedicated to the evaluation of algorithms estimating the fundamental frequency of speech. The results of our experiments show that our approach is very reliable

Vers la transformation de la parole oesophagienne en voix laryngée à l'aide de techniques de conversion vocale

National audienceCe travail concerne le développement d'un système de conversion de voix oesophagienne dans le but est de rendre plus intelligible celle-ci. La conversion de voix est une technique de transformation d'un signal de parole d'un locuteur source, de manière à ce qu'il semble, à l'écoute, être prononcé par un locuteur cible. Etant donnée la spécificité de la voix oesophagienne, nous proposons dans cette étude d'appliquer une nouvelle technique de conversion vocale en tenant compte de la particularité de l'appareil vocal des patients qui ont subi une ablation de larynx. En effet, l'ablation des cordes vocales perturbe profondément le signal glottique et par conséquent la voix oesophagienne acquise par le patient laryngectomisé est difficile à comprendre, rauque et faible en intensité. Dans la littérature, plusieurs techniques de conversion des voix ont été proposées, parmi lesquelles, la technique du codage linéaire prédictif pour la conversion vocale [1] et la régression linéaire multi-variée [2] qui vise à réduire la discontinuité et la distorsion spectrale

Enhancement of esophageal speech using voice conversion techniques

International audienceThis paper presents a novel approach for enhancing esophageal speech using voice conversion techniques. Esophageal speech (ES) is an alternative voice that allows a patient with no vocal cords to produce sounds after total laryngectomy: this voice has a poor degree of intelligibility and a poor quality. To address this issue, we propose a speaking-aid system enhancing ES in order to clarify and make it more natural. Given the specificity of ES, in this study we propose to apply a new voice conversion technique taking into account the particularity of the pathological vocal apparatus. We trained deep neural networks (DNNs) and Gaussian mixture models (GMMs) to predict " laryngeal " vocal tract features from esophageal speech. The converted vectors are then used to estimate the excitation cepstral coefficients and phase by a search in the target training space previously encoded as a binary tree. The voice resynthesized sounds like a laryngeal voice i.e., is more natural than the original ES, with an effective reconstruction of the prosodic information while retaining , and this is the highlight of our study, the characteristics of the vocal tract inherent to the source speaker. The results of voice conversion evaluated using objective and subjective experiments , validate the proposed approach

On the Use of Wavelets and Cepstrum Excitation for Pitch Determination in Real-Time

International audienceIn the current paper, we propose a new pitch tracking technique based on a wavelet transform in the temporal domain. Our algorithm is designed to determine the pitch frequency of the speech signal using a simple voicing decision algorithm. The pitch period is extracted from the cepstrum excitation signal processed by a wavelet transform; then the pitch contour is refined by thresholding and correction algorithms without any post-processing. The results obtained show that the proposed algorithm provides very good pitch contours compared to those furnished by the Bagshaw database

Une nouvelle méthodologie prédictive fondée sur un modèle séquence à séquence utilisé pour la transformation de la parole œsophagienne en voix laryngée

La situation sanitaire ne permettant pas d’organiser les 9èmes Journées de Phonétique Clinique dans les meilleures conditions (à savoir en présentiel), le comité de programme a décidé d’annuler cette édition 2021 et d’organiser à la place une journée dédiée à la présentation des contributions acceptées le 27 mai 2021.National audienc

Real-Time Signal Reconstruction from Short-Time Fourier Transform Magnitude Spectra Using FPGAs

International audienceReal Time implementation of new frequential domain synthesizer algorithm based on the Nawab approach for signal reconstruction from short-time Fourier transform magnitude spectra is proposed in this paper. A register-transferlevel (RTL) synthesizer based on our algorithm was designed and simulated using VHDL as the hardware description language in respecting real time delay. The implemented RTL model was verified by comparing its performances with those obtained from a Python language implementation of the same synthesizer. We prove in particular that the proposed algorithm can be implemented in real time with a sampling frequency up to 60 kHz. Finally a real time implementation using an Altera DE2 development kit, generating good quality audio signals, was implemented