Wyniki wyszukiwania - BazTech

1

Tandem repeats detection in DNA sequences using Kaiser window based adaptive S-transform

Sharma S. D., Saxena R., Sharma S. N.

Bio-Algorithms and Med-Systems

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2017

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Vol. 13, no. 3

167--173

EN

In computational biology the development of algorithms for the identification of tandem repeats in DNA sequences is a challenging problem. Tandem repeats identification is helpful in gene annotation, forensics, and the study of human evolution. In this work a signal processing algorithm based on adaptive S-transform, with Kaiser window, has been proposed for the exact and approximate tandem repeats detection. Usage of Kaiser window helped in identifying short as well as long tandem repeats. Thus, the limitation of earlier S-transform based algorithm that identified only microsatellites has been alleviated by this more versatile algorithm. The superiority of this algorithm has been established by comparative simulation studies with other reported methods.

2

Design and Implementation of Dual-Microphone Adaptive-Active Noise Cancellation System for De-Noising Speech Signal

Sharifi-Tehrani O., Ghafarioun E.

Przegląd Elektrotechniczny

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2012

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R. 88, nr 2

258-261

EN

Active Noise Control (ANC) has become an important field of research in recent decades. Noise Control in industrial environments and conference halls as well as in communication systems has been studied under the title adaptive-active noise cancellation-control (AANCC) and the results of these studies have been used in practical applications. Reducing time dependent noise is one of the ways recommended for speech enhancement. Here we have introduced an artificial neural network called ADALINE as a smart dual microphone active noise control system. This artificial neural network identifies sources of noise and interference during its training phase and adjusts accordingly. In this way the system reduces the input signal noise. Tests and implementations presented here are based on speech in Persian language and cumulative white Gaussian noise and the interference is assumed to be of the cosine type.

PL

Przedstawiono analizę szumów w otoczeniu przemysłowym i w salach konferencyjnych a następnie przedstawiono metody adaptacyjnych metod redukcji szumów. Szczególną uwagę zwrócono na szum zależny od czasu. Zastosowano metodę podwójnego mikrofonu i wykorzystano sieci neuronowe. Sieć identyfikuje źródło szumu i zakłóceń. Metodę sprawdzono doświadczalnie.

3

Novel hardware-efficient design of LMS-based adaptive FIR filter utilizing Finite State Machine and Block-RAM

Sharifi-Tehrani O.

Przegląd Elektrotechniczny

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2011

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R. 87, nr 7

240-244

EN

A novel pure-hardware design of LMS-based adaptive FIR filter core is proposed which is highly efficient in FPGA area/resource utilization and speed. Unlike HW/SW co-design and other pure-hardware methods, the required area/resource is reduced while keeping the speed in an appropriate level by taking advantage of finite state machine (FSM) and using internal block-rams (BRAM). This model because of being completely general (device independent), gives the ability of implementation on different FPGA brands and thus, is suitable for embedded systems, system-onprogrammable- chip (SoPC) and network-on-chip (NoC) applications.

PL

Opisano projekt filtru adapatacyjnego SOI który może byc wykorzystany w technice FPGA. Dla zapewnienia odpowiedniej szybkości zastosowano metodę FSM (finite state machine) i wewnętrzny RAM. Układ może być wykorzystany w systemach typu SoPC (system on programmable chip).

4

Adaptacyjne układy sterowania tworzące strefy ciszy wokół poruszającego się mikrofonu

Michalczyk M. I.

Pomiary Automatyka Kontrola

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2011

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R. 57, nr 1

9-11

PL

Przedstawiono wyniki badań nad adaptacyjnymi układami aktywnej redukcji hałasu wykorzystywanymi do tworzenia stref ciszy wokół poruszającego się w pomieszczeniu mikrofonu. Pokazano, że w części przypadków zastosowanie adaptacyjnego algorytmu FX-LMS zapewnia poprawne działanie układu, czyli śledzenie przez wytworzoną strefę ciszy zmian położenia mikrofonu. W przypadku złożonych zakłóceń lub szerszego zakresu ruchu mikrofonu, koniecznym jest zastosowanie dodatkowych algorytmów bieżącej estymacji modelu obiektu sterowania.

EN

The paper summarizes the novel research on a special class of digital sound processing systems with time varying sound sensor position, conducted in the Institute of Automatic Control, Silesian University of Technology, Gliwice, Poland. These are adaptive active noise control (ANC) systems creating zones of quiet around a microphone, position of which is being changed in enclosure. They use the LMS-based adaptive control algorithm which is parameterized with a model of the so called secondary path. The problem arises when the microphone position is changed during the ANC system operation, because the secondary path modelling errors occur . They, in some cases, can be coped by the adaptive control algorithm itself, however, in other cases an additional, on-line identification routine has to be used to update the secondary path model. These two situations are the subject of the presented research. The results of real world experiments showed that the zone of quiet can adaptively track the movement of the microphone if the LMS-based adaptive control algorithm is applied. However, if the noise is random and the spatial range of the microphone movement is wider, then the more complex algorithms have to be used, which is shown on the basis of simulation experiments, verified in the ANC laboratory.

5

Adaptive Algorithms for Enhancement of Speech Subject to a High-Level Noise

Latos M., Pawełczyk M.

Archives of Acoustics

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2010

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Vol. 35, No. 2

203-212

EN

There are many industrial environments which are exposed to a high-level noise, sometimes much higher than the level of speech. Verbal communication is then practically unfeasible. In order to increase the speech intelligibility, appropriate speech enhancement algorithms can be used. It is impossible to filter off the noise completely from the acquired signal by using a conventional filter, because of two reasons. First, the speech and the noise frequency contents are overlapping. Second, the noise properties are subject to change. The adaptive realisation of the Wienerbased approach can be, however, applied. Two structures are possible. One is the line enhancer, where the predictive realisation of the Wiener approach is used. The benefit of using this structure it that it does not require additional apparatus. The second structure takes advantage of the high level of noise. Under such condition, placing another microphone, even close to the primary one, can provide a reference signal well correlated with the noise disturbing the speech and lacking the information about the speech. Then, the classical Wiener filter can be used, to produce an estimate of the noise based on the reference signal. That noise estimate can be then subtracted from the disturbed speech. Both algorithms are verified, based on the data obtained from the real industrial environment. For laboratory experiments the G.R.A.S. artificial head and two microphones, one at back side of an earplug and another at the mouth are used.

6

Active noise control with moving error microphone

Michalczyk M. I.

Mechanics / AGH University of Science and Technology

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2008

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Vol. 27, no. 3

113-119

EN

Electro-acoustic plants controlled by active noise control (ANC) systems in enclosures are usually time-varying. Their changes can be caused by movements of an error microphone, around which zones of quiet are created. Then, the time variations of the electro-acoustic plant can be fast and adaptive ANC algorithms for tracking variations of error microphone position should be used. The preliminary research results showed that the created zone of quiet can move tracking the movement of the error microphone, when filtered-x least mean squares (FX-LMS) algorithm is applied. The paper presents further results of real-world experiments conducted on a special laboratory stand, which enabled to move the error microphone round the circle trajectory with constant rota- tional speed. Two modifications of FX-LMS algorithm were applied in order to accelerate control algorithm convergence and improve tracking properties of the ANC system: normalized FX-LMS and modified FX-LMS algorithm. It was shown, that the zone of quiet can track movement of the error microphone, however, there was no significant difference in control algorithms performance. In all cases the zone of quiet tracked the movement of the error microphone, even for the error microphone velocity as high as 3.7 m/s, and high attenuation was obtained - from 16 dB for the fast microphone movement up to 28 dB for the slowest microphone movement.

PL

Obiekty elektroakustyczne w układach aktywnego tłumienia hałasu (ATH) są zazwyczaj zmienne w czasie. Ich zmiany mogą być spowodowane m.in. ruchem mikrofonu błędu, wokół którego tworzone są przestrzenne strefy ciszy. Zastosowanie adaptacyjnych algorytmów ATH do śledzenia szybkich zmian pozycji mikrofonu błędu jest słabo zbadane. Wstępne badania pokazały, że strefa ciszy może poruszać się za mikrofonem błędu, w przypadku gdy algorytmem sterowania jest algorytm LMS z filtracją sygnału odniesienia (FX-LMS). Artykuł prezentuje wyniki badań w układzie ATH na specjalnym stanowisku laboratoryjnym umożliwiającym ruch mikrofonu po okręgu z zadaną szybkością. Aby zwiększyć szybkość zbieżności algorytmu FX-LMS, zastosowano dwie jego literaturowe modyfikacje: znormalizowany algorytm FX-LMS i zmodyfikowany algorytm FX-LMS. Pokazano, że strefa ciszy nadąża za ruchem mikrofonu, jednak nie ma znaczącej różnicy pomiędzy działaniem układu z zastosowaniem poszczególnych badanych algorytmów. We wszystkich przypadkach, strefa ciszy śledziła mikrofon, nawet dla ruchu z prędkością linową 3,7 m/s, i uzyskano wysokie tłumienie zakłócenia - od 16 dB dla największej prędkości mikrofonu do 28 dB w przypadku wolnego ruchu mikrofonu.

7

Application of genetic algorithmbin an active noise control system

Makarewicz G.

Archives of Acoustics

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2007

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Vol. 32, No. 4

839--849

EN

An active noise control (ANC) system utilizing a genetic algorithm for reduction of noise in duct is described. A continuous genetic algorithm with a heuristic crossover method was applied in the controller of the system. The ANC system was tested on a laboratory stand. Measurements of the efficiency of the system related to basic parameters of the genetic algorithm were performed. A comparison of the effectiveness of ANC system based on the genetic algorithm to the system based on the LMS algorithm is presented.

8

Pipelined architectures for the LMS adaptive Volterra filter

Glentis G.-O., Georgoulakis K.

Foundations of Computing and Decision Sciences

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2006

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Vol. 31, No. 2

135-155

EN

In this paper, efficient pipelined architectures for Least Mean Square (LMS) adaptive filtering and system identification of discrete-time Volterra models is presented. First, the multichannel embedding is adopted for the transformation of the discrete-time Volterra model to an equivalent multi-input single output format. Then, the LMS algorithm with delayed coefficients adaptation is applied, for the identification of the model parameters. The adaptation delay introduced in the computational flow of the adaptive scheme, allows for a pipelined implementation, however, the convergence and tracking properties of the algorithm are affected. Proper correction terms are subsequently introduced that compensate the adaptation delay and give results identical to the original LMS algorithm, subject to a latency delay.

9

Estimation of uncertainty propagation in the least squares algorithms for impedance components measurement

Augustyn J.

Zeszyty Naukowe Politechniki Świętokrzyskiej. Elektryka

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2005

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Z. 42

9-22

EN

The properties of some algorithms based on digital signal processing for the impedance components evaluation in circuits with sampling sensor have been analysed. It is supposed that the voltage and current are sampled synchronously to the fundamental frequency of the generated sinusoidal signal. Two groups of fitting sine wave algorithms, which are based on the least mean square (LMS) technique, have been described. The first one reconstructs indirect measurement method. The second group of algorithms estimates the unknown impedance components by direct method. In all these algorithms to simplify the calculations one can use different form of input matrix. In order to verify the performance of the considered algorithms (e.g., accuracy, estimator bias and convergence) the Monte Carlo simulations are realised in MATLAB.

PL

Wśród wielu algorytmów do pomiaru składowych impedancji można wyróżnić algorytmy dopasowania próbek do przebiegu sinusoidalnego opisanego modelem (4). W artykule przeanalizowano dwie grupy algorytmów. Algorytm, który oblicza wartości składowych impedancji z zależności (7) na podstawie wyznaczonych metodą najmniejszych kwadratów (6) składowych ortogo-nalnych amplitud napięcia i prądu, realizuje pośrednią metodę pomiaru. W drugim poszukiwane war-tości składowych otrzymuje się bezpośrednio w wyniku zastosowania metody najmniejszych kwadra-tów (6) do sygnału postaci (9). Dla uproszczenia obliczeń zaproponowano modyfikację tego algoryt-mu przez zastosowanie metody zmiennych instrumentalnych (IVM) (12) z macierzą zmiennych in-strumentalnych, zawierającą w kolumnach odpowiednio wartości nieparzystej i parzystej funkcji Walsha pierwszego rzędu (13). Dzięki temu uzyskano znaczną redukcję operacji mnożenia wymaga-nych do wyznaczenia estymat składowych impedancji. W celu weryfikacji właściwości zaproponowa-nych algorytmów przeprowadzono badania symulacyjne w środowisku programu MATLAB. Zbadano wpływ parametrów toru przetwarzania analogowo-cyfrowego na wartość niepewności przetwarzania poszczególnych algorytmów. Przeprowadzone badania symulacyjne wykazały, że algorytm IVM z wykorzystaniem funkcji Walsha zapewnia znaczne uproszczenie wymaganych obliczeń bez istotnego pogorszenia wyników pomiaru składowych w porównaniu z pozostałymi analizowanymi metodami.

10

Implementation of Adaptive Generalized Sidelobe Cancellers Using Efficient Complex Valued Arithmetic

Glentis G. O.

International Journal of Applied Mathematics and Computer Science

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2003

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Vol. 13, no 4

549-566

EN

Low complexity realizations of Least Mean Squared (LMS) error, Generalized Sidelobe Cancellers (GSCs) applied to adaptive beamforming are considered. The GSC method provides a simple way for implementing adaptive Linear Constraint Minimum Variance (LCMV) beamformers. Low complexity realizations of adaptive GSCs are of great importance for the design of high sampling rate, and/or small size and low power adaptive beamforming systems. The LMS algorithm and its Transform Domain (TD-LMS) counterpart are considered for the adaptive processing task involved in the design of optimum GSC systems. Since all input signals are represented by complex variables, complex valued arithmetic is utilized for the realization of GSC algorithms, either on general purpose computers, or on dedicated VLSI ASICs. Using algorithmic strength reduction (SR) techniques, two novel algorithms are developed for efficient realizations of both LMS GSCs and TD-LMS GSC schemes. Both of the proposed algorithms are implemented using real valued arithmetic only, whilst reducing the number of multipliers by 25% and 20%, respectively. When VLSI implementation aspects are considered, both the proposed algorithms result in reduced power dissipation and silicon area realizations. The performance of the proposed realizations of the LMS based GSC methods is illustrated in the context of typical beamforming applications.