Application de la transformée en nombres entiers à l'étude et au développement d'un codeur de parole pour transmission sur réseaux IP

Notre étude s'inscrit dans le domaine de la compression des signaux vocaux pour la transmission de voix par réseau Internet (VoIP : Voice over Internet Protocol). Les perspectives à moyen terme étant la mise en place d'une application de téléphonie IP, nos travaux fournissent les premiers éléments pour le fonctionnement en réel d'un système de codage de parole et son intégration à un processeur DSP. Ils se sont concentrés sur le codeur de parole G.729 de type CS-ACELP (Conjugate Structure Algebraic Code-Excited Linear Prediction), retenu parmi les recommandations de l'Union Internationale des Télécommunications (UIT) et déjà reconnu pour sa faible complexité d'implantation. L'étude principale a été d'améliorer ses performances et diminuer sa charge de calcul tout en maintenant un équilibre entre la qualité de codage et la complexité de calcul engendrée. Pour réduire le coût de calcul de ce codeur, nous avons approfondi les bases mathématiques de la Transformée en Nombres Entiers (NTT : Number Theoretic Transform) qui est amenée à trouver des applications de plus en plus diverses en traitement du signal. Nous avons introduit plus particulièrement la Transformée en Nombres de Fermat (FNT : Fermat Number Transform) qui est la plus adaptée aux opérations de traitement numérique. Nous avons constaté que son application à certains algorithmes de codage permet une réduction importante de la complexité de calcul. Ainsi, le développement de nouveaux algorithmes performants, pour la Prédiction Linéaire (LP : Linear Prediction) du signal et la modélisation de l'excitation, a permis une modification du codeur G.729 en vue de son implantation sur un processeur à virgule fixe. De plus, une nouvelle fonction de détection d'activité de parole (VAD : Voice Activity Detection) a permis la mise en place d'une procédure de compression des silences plus efficace et la réduction du débit de transmissionOur study considers the vocal signals compression for the transmission of Voice over Internet Protocol (VoIP). The prospects being the implementation of a telephony IP application, the work provides the first elements for a real-time speech coding system and its integration to a DSP. They are concentrated on the speech CS-ACELP (Conjugate Structure- Algebraic Code-Excited Linear Prediction) G.729 coder, retained among the International Telecommunications Union (ITU) recommendations and already recognized for its low implementation complexity. The main aspect was to improve its performances and to decrease its computational cost, while maintaining the compromise between the coding quality and the required complexity. To reduce the computational cost of this coder, we looked further into the mathematical bases of the Number Theoretic Transform (NTT) which is brought to find more and more various applications in signal processing. We introduced more particularly the Fermat Number Transform (FNT) which is well suited for digital processing operations. Its application to different coding algorithms allows an important reduction of the computational complexity. Thus, the development of new efficient algorithms, for the Linear Prediction (LP) of the speech signal and the excitation modeling, has allowed a modification of the G.729 coder and his implementation on a fixed-point processor. Moreover, a new function of Voice Activity Detection (VAD) has carried out the implementation of one more efficient procedure for silences compression and the reduction of the transmission rate.BREST-BU Droit-Sciences-Sports (290192103) / SudocSudocFranceF

La situation monopolistique est-elle forcément à risques ? : Case study

International PurchasingCe case study traite du sujet de la situation monopolistique fournisseur face à laquelle bon nombre d’entreprises doivent faire face. Cela est d’autant plus vrai dans des secteurs d’activités où des normes contraignantes existent et où une solution sur-mesure est souvent nécessaire pour satisfaire les besoins des clients internes et externes. Dans ce dossier, nous tâcherons d’apporter des solutions aux trois entreprises dans lesquelles nous réalisons notre année d’alternance. A savoir Siemens, PunchPowerglide et Lilly France qui font toutes les trois face à cette problématique. Une question se pose alors : une situation monopolistique est-elle forcément à risques ? Ne peut-on apprendre à vivre avec et en tirer des avantages

Fast pitch modelling for CS-ACELP coder using Fermat Number Transforms

International audienceThis paper presents a double improvements to reduce the speech coding complexity of the pitch prediction in a code-excited linear prediction (CELP) coder. First, the pitch analysis structure is modified. A new fast pitch modelling by linear-filtering (PMF) procedure will determine the adaptive and stochastic codebook contributions of the excitation signal. Afterwards, an efficient implementation of the PMF-CELP coding is proposed by using number theoretic transforms which can significantly reduce the algorithm computation complexity

Realization of Multi-Delay Filter using Fermat Number Transforms

International audienceThis paper is about an efficient implementation of adaptive filtering for echo cancelers. The first objective of this paper is to propose a simplified method of the flexible block Multi-Delay Filter (MDF) algorithm in the time-domain. Then, we will derive a new method for the step-size adaptation coefficient. The second objective is about the realization of a Block Proportionate Normalized Least Mean Squares (BPNLMS++) with the simplified MDF (SMDF) implementation. Using the new step-size method and the smaller block dimension proposed by SMDF, we achieve a faster convergence of the adaptive process with a limited computational cost. Then, an efficient implementation of the new procedure (SMDF-BPNLMS++) block filtering is proposed using Fermat Number Transform, which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor

Realization of Block Robust Adaptive Filters using Generalized Sliding Fermat Number Transform

International audienceThis paper is about an efficient implementation of adaptive filtering for echo cancelers. First, a realization of an improved Block Proportionate Normalized Least Mean Squares (BPNLMS++) using Generalized Sliding Fermat Number Transform (GSFNT) is presented. Unfortunately, during the double-talk mode, the echo cancelers often diverge. We can cope with this problem by employing a double-talk detector formed by two Voice Activity Detectors (VAD's) . We propose a general system based on the Robust-Block-PNLMS ++ (RBPNLMS++) adaptive filter combined with a post-filter. The general system was implemented with GSFNT which can significantly reduce the computation complexity of the filter implantation on Digital Signal Processing (DSP)

Design of a variable rate algorithm for CS-ACELP coder

International audienceThis paper is about the reduction of the computational complexity of the CS-ACELP codec, described in ITU recommendation G.729, and used for the transmission of voice over IP. A voice activity detection module is proposed to replace the G.729 Annex B algorithm. The new procedure was developed to allow its implementation with number theoretic transforms. The use of Fermat number transforms can reduce the cost of variable rate algorithm implementation on digital signal processor (DSP)

Realization of block adaptive filters using Fermat number transforms

International audienceThis paper is about an efficient implementation of adaptive filtering for echo cancelers. First, a realization of an improved Proportionate Normalized Least Mean Squares (PNLMS+ +) adaptive filter using block structure is presented. Then, an efficient implementation of the block filtering process is proposed using Number Theoretic Transforms (NTT) which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor (DSP)