Anthropomorphic Coding of Speech and Audio: A Model Inversion Approach

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 paper is to provide a tutorial on perceptual speech and audio coding using an invertible auditory model. In this approach, the audio signal is converted into an auditory representation using an invertible auditory model. The auditory representation is quantized and coded. Upon decoding, it is then transformed back into the acoustic domain. This transformation converts a complex distortion criterion into a simple one, thus facilitating quantization with low complexity. We briefly review past work on auditory models and describe in more detail the components of our invertible model and its inversion procedure, that is, the method to reconstruct the signal from the output of the auditory model. We summarize attempts to use the auditory representation for low-bit-rate coding. Our approach also allows the exploitation of the inherent redundancy of the human auditory system for the purpose of multiple description (joint source-channel) coding

An analog electronic cochlea

An analog electronic cochlea has been built in CMOS VLSI technology using micropower techniques. The key point of the model and circuit is that a cascade of simple, nearly linear, second-order filter stages with controllable Q parameters suffices to capture the physics of the fluid-dynamic traveling-wave system in the cochlea, including the effects of adaptation and active gain involving the outer hair cells. Measurements on the test chip suggest that the circuit matches both the theory and observations from real cochleas

An Analog Electronic Cochlea

An engineered system that hears, such as a speech recognizer, can be designed by modeling the cochlea, or inner ear, and higher levels of the auditory nervous system. To be useful in such a system, a model of the cochlea should incorporate a variety of known effects, such as an asymmetric low-pass/bandpass response at each output channel, a short ringing time, and active adaptation to a wide range of input signal levels. An analog electronic cochlea has been built in CMOS VLSI technology using micropower techniques to achieve this goal of usefulness via realism. The key point of the model and circuit is that a cascade of simple, nearly linear, second-order filter stages with controllable Q parameters suffices to capture the physics of the fluid-dynamic traveling-wave system in the cochlea, including the effects of adaptation and active gain involving the outer hair cells. Measurements on the test chip suggest that the circuit matches both the theory and observations from real cochleas

Détection de la double parole dans le contexte de radiotéléphone main-libre en véhicule

L'environnement très bruité, le couplage du haut parleur (HP) avec le microphone, ainsi que le problème de gain du HP dans un contexte de radio mobile en véhicule font l'objet de plusieurs travaux en télécommunications. Des algorithmes pour réduire le bruit et pour annuler l'écho (A.E) ont été proposés dans la littérature scientifique. En général, tous les algorithmes d'annulation d'écho sont basés sur des filtres à coefficients adaptatifs qui fonctionnent assez bien. Cependant, la façon d'adapter les coefficients influence terriblement les performances. Nous proposons ici une technique qui permet de mieux détecter les moments de mises à jour des coefficients des filtres (paramètres). Normalement, ces filtres ne doivent pas être adaptés lorsque le locuteur local parle (locuteurs installés en véhicule). On a généralement recourt à des algorithmes à base d'énergie afin de séparer la voix du locuteur local de celle du correspondant lointain. Nous proposons une technique, qui au lieu de l'énergie, utilise un détecteur de hauteur tonale (D.H.T) et qui est basé sur un modèle auditif (Rouat et al., Speech Comm. Jour., 1997). Ce DHT est introduit en cascade avec le filtre auto-regressif (A.R.) déjà inclus dans le système. Conjointement, le DHT et le filtre A.R. nous ont permis d'annuler le fondamental, les composantes harmoniques et la contribution vocale du locuteur lointain sur le canal microphone