Informacja

Drogi użytkowniku, aplikacja do prawidłowego działania wymaga obsługi JavaScript. Proszę włącz obsługę JavaScript w Twojej przeglądarce.

Wyszukujesz frazę "signal filtering" wg kryterium: Temat


Wyświetlanie 1-7 z 7
Tytuł:
Recognizing the Sequences of Code Manipulated Short LFM Signals
Autorzy:
Leszczyński, T.
Powiązania:
https://bibliotekanauki.pl/articles/176689.pdf
Data publikacji:
2012
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
sonar
LFM signal
matched filtering
Opis:
Noise-like binary sequences combined with signals with linear frequency modulation might be suc- cessfully used to increase the reliability of the recognition of both probe and communication signals in the presence of natural and artificial interference. To identify such formed sequences the usage of the two-step matched filtering was suggested and the probabilistic model of the recognition of noise-like code sequences transferred by LFM signals was developed.
Źródło:
Archives of Acoustics; 2012, 37, 3; 295-300
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Optimization of Short Probe Linear Frequency Modulated Signal Parameters
Autorzy:
Pogribny, W.
Leszczyński, T.
Powiązania:
https://bibliotekanauki.pl/articles/176953.pdf
Data publikacji:
2011
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
sonar
LFM signal
matched filtering
Opis:
In many physical experiments, linear frequency modulated (LFM) signals are widely used to probe objects in different environments, from outer-space to under- water. These signals allow a significant improvement in measurement resolution, even when the observation distance is great. For example, using LFM probe signals in underwater investigations enables discovery of even small objects covered by bottom sediments. Recognition of LFM (chirp) signals depends on their compression based on matched filtering. This work presents two simple solutions to improve the resolution of the short chirp signals recognition. These methods are effective only if synchronization between the signal and matched filter (MF) is obtained. This work describes both the aforementioned methods and a method of minimizing the effects of the lack of synchronization. The proposed matched filtering method, with the use of n parallel MFs and other techniques, allows only one sample to be obtained in the main lobe and to accurately locate its position in the appropriate sampling period Ts with accuracy Ts/n. These approaches are appropriate for use in probe signal processing.
Źródło:
Archives of Acoustics; 2011, 36, 4; 861-871
0137-5075
Pojawia się w:
Archives of Acoustics
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Filtering Property of Signal Sampling in General and Under-Sampling as a Specific Operation of Filtering Connected with Signal Shaping at the Same Time
Autorzy:
Borys, Andrzej
Powiązania:
https://bibliotekanauki.pl/articles/226441.pdf
Data publikacji:
2020
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
signal sampling
filtering
discrete-time Fourier transform
Opis:
In this paper, we show that signal sampling operation can be considered as a kind of all-pass filtering in the time domain, when the Nyquist frequency is larger or equal to the maximal frequency in the spectrum of a signal sampled. We demonstrate that this seemingly obvious observation has wide-ranging implications. They are discussed here in detail. Furthermore, we discuss also signal shaping effects that occur in the case of signal under-sampling. That is, when the Nyquist frequency is smaller than the maximal frequency in the spectrum of a signal sampled. Further, we explain the mechanism of a specific signal distortion that arises under these circumstances. We call it the signal shaping, not the signal aliasing, because of many reasons discussed throughout this paper. Mainly however because of the fact that the operation behind it, called also the signal shaping here, is not a filtering in a usual sense. And, it is shown that this kind of shaping depends upon the sampling phase. Furthermore, formulated in other words, this operation can be viewed as a one which shapes the signal and performs the low-pass filtering of it at the same time. Also, an interesting relation connecting the Fourier transform of a signal filtered with the use of an ideal low-pass filter having the cut frequency lying in the region of under-sampling with the Fourier transforms of its two under-sampled versions is derived. This relation is presented in the time domain, too.
Źródło:
International Journal of Electronics and Telecommunications; 2020, 66, 3; 589-594
2300-1933
Pojawia się w:
International Journal of Electronics and Telecommunications
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Analysis of a novel FPGA-based system for filtering audio signals using a finite impulse response filters
Autorzy:
Lipowski, Adrian
Majewski, Paweł
Pluta, Sławomir
Powiązania:
https://bibliotekanauki.pl/articles/2055238.pdf
Data publikacji:
2022
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
audio filtering
FIR filter
FPGA
signal processing
Opis:
In this article, an analysis of an innovative system for filtering signals in the audible range (16 Hz - 20 kHz) on programmable logic devices using a filters with a finite impulse response, is presented. Mentioned system was neat combination of software and hardware platform, where in the program layer a multiple programming languages including VHDL, JavaScript, Matlab or HTML were used to create completely useful application. To determine the coefficients of polynomial filters the Matlab Filter Design & Analysis Tool was used. Thanks to the developed graphic layer, a user-friendly interface was created, which allows easily transfer the required coefficients from the computer to the executive system. The practical implementation made on the FPGA platform, specifically on the Altera DE2- 115 development kit with the FPGA Cyclone IV, was compared with simulation realization of Matlab FIR filters. The performed research confirm the effectiveness of filtration in real time with up to 128th order of the filter for both audio channels simultaneously in FPGA-based system.
Źródło:
International Journal of Electronics and Telecommunications; 2022, 68, 1; 19--26
2300-1933
Pojawia się w:
International Journal of Electronics and Telecommunications
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Short-term positioning accuracy based on mems sensors for smart city solutions
Autorzy:
Grzechca, Damian
Paszek, Krzysztof
Powiązania:
https://bibliotekanauki.pl/articles/220713.pdf
Data publikacji:
2019
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
electro-mechanical system
received signal strength indicator
positioning
filtering
Opis:
The paper presents a method of obtaining short-term positioning accuracy based on micro electro-mechanical system (MEMS) sensors and analysis of the results. A high-accuracy and fast-positioning algorithm must be included due to the high risk of accidents in cities in the future, especially when autonomous objects are taken into account. High-level positioning systems should consider a number of sub-systems such as global positioning system (GPS), CCTV – video analysis, a system based on analysis of signal strength of access points (AP), etc. Short-term positioning means that there are other locating systems with a sufficiently high degree of accuracy based on, e.g. a video camera, but the located object can disappear when it is hidden by other objects, e.g. people, things, shelves etc. In such a case, MEMS sensors can be employed as a positioning system. The paper examines typical movement profiles of a radio-controlled (RC) model and fundamental filtering methods in respect of position accuracy. The authors evaluate the complexity and delay of the filter and the accuracy of the positioning in respect of the current speed and phase of movement (positive acceleration, constant) of the object. It is necessary to know whether and how the length of the filter changes the position accuracy. It has been shown that the use of fundamental filters, which provide solutions in a short time, enables to locate objects with a small error in a limited time.
Źródło:
Metrology and Measurement Systems; 2019, 26, 1; 95-107
0860-8229
Pojawia się w:
Metrology and Measurement Systems
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
A robust fixed-lag smoothing algorithm for dynamic systems with correlated sensor malfunctions
Autorzy:
Grishin, Y. P.
Janczak, D.
Powiązania:
https://bibliotekanauki.pl/articles/199833.pdf
Data publikacji:
2014
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
outliers
robust signal processing
nonlinear filtering
fault detection
systems with changing structures
Opis:
A new robust fixed-lag smoothing algorithm for fault-tolerant signal processing in stochastic dynamic systems in the presence of correlated sensor malfunctions has been developed. The algorithm is developed using a state vector augmentation method and the Gaussian approximation of the current estimate probability density function. The algorithm can be used in the real-time fault-tolerant control systems as well as in radar tracking systems working in the complex interference environment. The performance of the developed algorithm are evaluated by simulations and compared with smoothing and nonlinear filtering algorithms.
Źródło:
Bulletin of the Polish Academy of Sciences. Technical Sciences; 2014, 62, 3; 517-523
0239-7528
Pojawia się w:
Bulletin of the Polish Academy of Sciences. Technical Sciences
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Comparison of multiband filtering, empirical mode decomposition and short-time fourier transform used to extract physiological components from long-term heart rate variability
Autorzy:
Adamczyk, Krzysztof
Polak, Adam G.
Powiązania:
https://bibliotekanauki.pl/articles/2052173.pdf
Data publikacji:
2021
Wydawca:
Polska Akademia Nauk. Czytelnia Czasopism PAN
Tematy:
heart rate variability
nonstationary signal analysis
multiband filtering
empirical mode decomposition
short-time Fourier transform
Hilbert transform
Opis:
Heart rate is constantly changing under the influence of many control signals, as manifested by heart rate variability (HRV). HRV is a nonstationary, irregularly sampled signal, the spectrum of which reveals distinct bands of high, low, very low and ultra-low frequencies (HF, LF, VLF, ULF). VLF and ULF components are the least understood, and their analysis requires HRV records lasting many hours. Moreover, there are still no well-established methods for the reliable extraction of these components. The aim of this work was to select, implement and compare methods which can solve this problem. The performance of multiband filtering (MBF), empirical mode decomposition and the short-time Fourier transform was tested, using synthetic HRV as the ground truth for methods evaluation as well as real data of three patients selected from 25 polysomnographic records with a clear HF component in their spectrograms. The study provided new insights into the components of long-term HRV, including the character of its amplitude and frequency modulation obtained with the Hilbert transform. In addition, the reliability of the extracted HF, LF, VLF and ULF waveforms was demonstrated, and MBF turned out to be the most accurate method, though the signal is strongly nonstationary. The possibility of isolating such waveforms is of great importance both in physiology and pathophysiology, as well as in the automation of medical diagnostics based on HRV.
Źródło:
Metrology and Measurement Systems; 2021, 28, 4; 643-660
0860-8229
Pojawia się w:
Metrology and Measurement Systems
Dostawca treści:
Biblioteka Nauki
Artykuł
    Wyświetlanie 1-7 z 7

    Ta witryna wykorzystuje pliki cookies do przechowywania informacji na Twoim komputerze. Pliki cookies stosujemy w celu świadczenia usług na najwyższym poziomie, w tym w sposób dostosowany do indywidualnych potrzeb. Korzystanie z witryny bez zmiany ustawień dotyczących cookies oznacza, że będą one zamieszczane w Twoim komputerze. W każdym momencie możesz dokonać zmiany ustawień dotyczących cookies