Studio clinico e realizzazione di uno stetoscopio digitale indossabile

Quando si pensa alla figura di un medico, non lo si può immaginare senza uno stetoscopio alle sue spalle. Lo stetoscopio è uno strumento molto semplice ma molto importante, inventato in Francia nel 1816 da René Théophile Hyacinthe Laennec, permise di poter auscultare ed analizzare i rumori generati dai battiti cardiaci ottenendo suoni più puliti e rendendo meno invasiva la procedura, precedentemente svolta posizionando l'orecchio sul petto del paziente. Nel 1970 David Littman migliorò notevolmente la struttura dello stetoscopio precedente, creando lo strumento che noi tutti conosciamo. Grazie alla rapida evoluzione dell'elettronica negli ultimi anni si hanno oggi a disposizione una moltitudine di nuovi ed efficaci sensori, alcuni dei quali trovano impiego nell'ambito dell'ingegneria biomedica. In particolare, si è pensato di utilizzare queste nuove tecnologie per la progettazione e lo sviluppo di uno stetoscopio elettronico. Il forte vantaggio di uno stetoscopio elettronico, rispetto al classico biomeccanico, risiede nell'elaborazione, nel filtraggio e nell'amplificazione del segnale che si può applicare all'auscultazione, oltreché nella possibilità di poter registrare il segnale acquisito, rendendo questo dispositivo uno strumento potenzialmente efficace per la telemedicina. In questo elaborato verrà trattato lo sviluppo di uno stetoscopio elettronico, evidenziando tutti i problemi e le criticità riscontrate durante il progetto in ogni punto di vista, software, hardware e signal processing. Tutto ciò non prima di una consistente introduzione necessaria per comprendere la tipologia di segnale d'interesse e l'analisi di alcuni tipi di dispositivi presenti in commercio e non, da cui si è tratta ispirazione. Seguirà un analisi critica dei risultati ottenuti, confrontando i segnali ottenuti dal dispositivo progettato con quelli ottenuti in reparto utilizzando due dispositivi commerciali, analizzando le differenze, punti di forza e di debolezza.

Numerical comparison between conventional dispersion compensating fibers and photonic crystal fibers as lumped Raman amplifiers

In this paper we discuss the use of photonic crystal fibers (PCFs) as discrete devices for simultaneous wideband dispersion compensation and Raman amplification. The performance of the PCFs in terms of gain, ripple, optical signal-to-noise ratio (OSNR) and required fiber length for complete dispersion compensation is compared with conventional dispersion compensating fibers (DCFs). The main goal is to determine the minimum PCF loss beyond which its performance surpasses a state-of-the-art DCF and justifies practical use in telecommunication systems. (C) 2009 Optical Society of America

Study of phase transition in (Pb,Ba)TiO(3) thin films

Dielectric and Raman scattering experiments were performed on polycrystalline Pb(1-x)Ba(x)TiO(3) thin films (x=0.40 and 0.60) as a function of temperature. The dielectric study on single phase compositions revealed that a diffuse-type phase transition occurred upon transformation of the cubic paraelectric to the tetragonal ferroelectric phase in all thin films, which showed a broadening of the dielectric peak. Diffusivity was found to increase with increasing barium contents in the composition range under study. In addition, the temperature dependence of Raman scattering spectra was investigated through the ferroelectric phase transition. The temperature dependence of the phonon frequencies was used to characterize the phase transitions. Raman modes persisted above the tetragonal to cubic phase transition temperature, although all optical modes should be Raman inactive. The origin of these modes was interpreted as a breakdown of the local cubic symmetry by chemical disorder. The lack of a well-defined transition temperature and the presence of broadbands in some temperature intervals above the paraferroelectric phase transition temperature suggest a diffuse-type phase transition. (C) 2008 American Institute of Physics.

A small area 8 bits 50 MHz CMOS DAC for Bluetooth transmitter

This paper presents a small-area CMOS current-steering segmented digital-to-analog converter (DAC) design intended for RF transmitters in 2.45 GHz Bluetooth applications. The current-source design strategy is based on an iterative scheme whose variables are adjusted in a simple way, minimizing the area and the power consumption, and meeting the design specifications. A theoretical analysis of static-dynamic requirements and a new layout strategy to attain a small-area current-steering DAC are included. The DAC was designed and implemented in 0.35 mu m CMOS technology, requiring an active area of just 200 mu m x 200 mu m. Experimental results, with a full-scale output current of 700 mu A and a 3.3 V power supply, showed a spurious-free dynamic range of 58 dB for a 1 MHz output sine wave and sampling frequency of 50 MHz, with differential and integral nonlinearity of 0.3 and 0.37 LSB, respectively.

FWM Constraints Management for Lightpath Establishment in GMPLS Networks

In all-optical networks, management of physical layer restrictions should collaborate in lightpath establishment. Label-Switched Path validation in Generalized MultiProtocol Label Switching on Dense Wavelength Division Multiplexing network requires the treatment of the physical impairment-related parameters along the provisioned route. In this paper we propose, for the first time in our view, the generation of an optical layer database by simulation that specifically characterizes the dynamic FWM impairments for the lightpaths provisioned in a GMPLS/DWDM network.

An Array Recursive Least-Squares Algorithm With Generic Nonfading Regularization Matrix

We present a novel array RLS algorithm with forgetting factor that circumvents the problem of fading regularization, inherent to the standard exponentially-weighted RLS, by allowing for time-varying regularization matrices with generic structure. Simulations in finite precision show the algorithm`s superiority as compared to alternative algorithms in the context of adaptive beamforming.

Transient and Steady-State Analysis of the Affine Combination of Two Adaptive Filters

In this paper, we propose an approach to the transient and steady-state analysis of the affine combination of one fast and one slow adaptive filters. The theoretical models are based on expressions for the excess mean-square error (EMSE) and cross-EMSE of the component filters, which allows their application to different combinations of algorithms, such as least mean-squares (LMS), normalized LMS (NLMS), and constant modulus algorithm (CMA), considering white or colored inputs and stationary or nonstationary environments. Since the desired universal behavior of the combination depends on the correct estimation of the mixing parameter at every instant, its adaptation is also taken into account in the transient analysis. Furthermore, we propose normalized algorithms for the adaptation of the mixing parameter that exhibit good performance. Good agreement between analysis and simulation results is always observed.

Distributed Estimation Over an Adaptive Incremental Network Based on the Affine Projection Algorithm

We study the problem of distributed estimation based on the affine projection algorithm (APA), which is developed from Newton`s method for minimizing a cost function. The proposed solution is formulated to ameliorate the limited convergence properties of least-mean-square (LMS) type distributed adaptive filters with colored inputs. The analysis of transient and steady-state performances at each individual node within the network is developed by using a weighted spatial-temporal energy conservation relation and confirmed by computer simulations. The simulation results also verify that the proposed algorithm provides not only a faster convergence rate but also an improved steady-state performance as compared to an LMS-based scheme. In addition, the new approach attains an acceptable misadjustment performance with lower computational and memory cost, provided the number of regressor vectors and filter length parameters are appropriately chosen, as compared to a distributed recursive-least-squares (RLS) based method.

MODELING THE UNVOICED COMPONENT IN THE CANONICAL REPRESENTATION OF SPEECH

The canonical representation of speech constitutes a perfect reconstruction (PR) analysis-synthesis system. Its parameters are the autoregressive (AR) model coefficients, the pitch period and the voiced and unvoiced components of the excitation represented as transform coefficients. Each set of parameters may be operated on independently. A time-frequency unvoiced excitation (TFUNEX) model is proposed that has high time resolution and selective frequency resolution. Improved time-frequency fit is obtained by using for antialiasing cancellation the clustering of pitch-synchronous transform tracks defined in the modulation transform domain. The TFUNEX model delivers high-quality speech while compressing the unvoiced excitation representation about 13 times over its raw transform coefficient representation for wideband speech.

Stochastic stability analysis for the constant-modulus algorithm

We derive an easy-to-compute approximate bound for the range of step-sizes for which the constant-modulus algorithm (CMA) will remain stable if initialized close to a minimum of the CM cost function. Our model highlights the influence, of the signal constellation used in the transmission system: for smaller variation in the modulus of the transmitted symbols, the algorithm will be more robust, and the steady-state misadjustment will be smaller. The theoretical results are validated through several simulations, for long and short filters and channels.

Improving the tracking capability of adaptive filters via convex combination

As is well known, Hessian-based adaptive filters (such as the recursive-least squares algorithm (RLS) for supervised adaptive filtering, or the Shalvi-Weinstein algorithm (SWA) for blind equalization) converge much faster than gradient-based algorithms [such as the least-mean-squares algorithm (LMS) or the constant-modulus algorithm (CMA)]. However, when the problem is tracking a time-variant filter, the issue is not so clear-cut: there are environments for which each family presents better performance. Given this, we propose the use of a convex combination of algorithms of different families to obtain an algorithm with superior tracking capability. We show the potential of this combination and provide a unified theoretical model for the steady-state excess mean-square error for convex combinations of gradient- and Hessian-based algorithms, assuming a random-walk model for the parameter variations. The proposed model is valid for algorithms of the same or different families, and for supervised (LMS and RLS) or blind (CMA and SWA) algorithms.

A Class of Channels Resulting in Ill-Convergence for CMA in Decision Feedback Equalizers

This paper analyzes the convergence of the constant modulus algorithm (CMA) in a decision feedback equalizer using only a feedback filter. Several works had already observed that the CMA presented a better performance than decision directed algorithm in the adaptation of the decision feedback equalizer, but theoretical analysis always showed to be difficult specially due to the analytical difficulties presented by the constant modulus criterion. In this paper, we surmount such obstacle by using a recent result concerning the CM analysis, first obtained in a linear finite impulse response context with the objective of comparing its solutions to the ones obtained through the Wiener criterion. The theoretical analysis presented here confirms the robustness of the CMA when applied to the adaptation of the decision feedback equalizer and also defines a class of channels for which the algorithm will suffer from ill-convergence when initialized at the origin.

Avoiding Divergence in the Shalvi-Weinstein Algorithm

The most popular algorithms for blind equalization are the constant-modulus algorithm (CMA) and the Shalvi-Weinstein algorithm (SWA). It is well-known that SWA presents a higher convergence rate than CMA. at the expense of higher computational complexity. If the forgetting factor is not sufficiently close to one, if the initialization is distant from the optimal solution, or if the signal-to-noise ratio is low, SWA can converge to undesirable local minima or even diverge. In this paper, we show that divergence can be caused by an inconsistency in the nonlinear estimate of the transmitted signal. or (when the algorithm is implemented in finite precision) by the loss of positiveness of the estimate of the autocorrelation matrix, or by a combination of both. In order to avoid the first cause of divergence, we propose a dual-mode SWA. In the first mode of operation. the new algorithm works as SWA; in the second mode, it rejects inconsistent estimates of the transmitted signal. Assuming the persistence of excitation condition, we present a deterministic stability analysis of the new algorithm. To avoid the second cause of divergence, we propose a dual-mode lattice SWA, which is stable even in finite-precision arithmetic, and has a computational complexity that increases linearly with the number of adjustable equalizer coefficients. The good performance of the proposed algorithms is confirmed through numerical simulations.

Turbulent Time and Length Scale Measurements in High-Velocity Open Channel Flows

In high-velocity open channel flows, the measurements of air-water flow properties are complicated by the strong interactions between the flow turbulence and the entrained air. In the present study, an advanced signal processing of traditional single- and dual-tip conductivity probe signals is developed to provide further details on the air-water turbulent level, time and length scales. The technique is applied to turbulent open channel flows on a stepped chute conducted in a large-size facility with flow Reynolds numbers ranging from 3.8 E+5 to 7.1 E+5. The air water flow properties presented some basic characteristics that were qualitatively and quantitatively similar to previous skimming flow studies. Some self-similar relationships were observed systematically at both macroscopic and microscopic levels. These included the distributions of void fraction, bubble count rate, interfacial velocity and turbulence level at a macroscopic scale, and the auto- and cross-correlation functions at the microscopic level. New correlation analyses yielded a characterisation of the large eddies advecting the bubbles. Basic results included the integral turbulent length and time scales. The turbulent length scales characterised some measure of the size of large vortical structures advecting air bubbles in the skimming flows, and the data were closely related to the characteristic air-water depth Y90. In the spray region, present results highlighted the existence of an upper spray region for C > 0.95 to 0.97 in which the distributions of droplet chord sizes and integral advection scales presented some marked differences with the rest of the flow.

Scalable Visualization of Galaxies, Oceans, and Brains

One of the challenges in scientific visualization is to generate software libraries suitable for the large-scale data emerging from tera-scale simulations and instruments. We describe the efforts currently under way at SDSC and NPACI to address these challenges. The scope of the SDSC project spans data handling, graphics, visualization, and scientific application domains. Components of the research focus on the following areas: intelligent data storage, layout and handling, using an associated “Floor-Plan” (meta data); performance optimization on parallel architectures; extension of SDSC’s scalable, parallel, direct volume renderer to allow perspective viewing; and interactive rendering of fractional images (“imagelets”), which facilitates the examination of large datasets. These concepts are coordinated within a data-visualization pipeline, which operates on component data blocks sized to fit within the available computing resources. A key feature of the scheme is that the meta data, which tag the data blocks, can be propagated and applied consistently. This is possible at the disk level, in distributing the computations across parallel processors; in “imagelet” composition; and in feature tagging. The work reflects the emerging challenges and opportunities presented by the ongoing progress in high-performance computing (HPC) and the deployment of the data, computational, and visualization Grids.