Informacja

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

Wyszukujesz frazę "signal filtering" wg kryterium: Temat


Tytuł:
Zastosowanie transformaty falkowej do filtrowania szybkozmiennych składowych krzywej natężenia promieniowania słonecznego
Employing a wavelet transform to filter quick change components of solar radiation intensity curve
Autorzy:
Kapica, J.
Powiązania:
https://bibliotekanauki.pl/articles/290500.pdf
Data publikacji:
2010
Wydawca:
Polskie Towarzystwo Inżynierii Rolniczej
Tematy:
dekompozycja falkowa
filtracja sygnału
promieniowanie słoneczne
wavelet decomposition
signal filtering
solar radiation
Opis:
Artykuł przedstawia zastosowanie transformaty falkowej do filtrowania szybkozmiennych składowych krzywej natężenia promieniowania słonecznego. Dokonano pomiarów natężenia promieniowania słonecznego dla kilku charakterystycznych pod względem zmienności dni. Pomiary były dokonywane z częstotliwością 2 próbki na sekundę co jest wartością dużą, jak dla pomiarów nasłonecznienia. Następnie tak uzyskane dane zostały poddane ponownemu próbkowaniu z mniejszymi częstotliwościami bezpośrednio oraz po odfiltrowaniu składowych szybkozmiennych przy użyciu transformaty falkowej. Dla każdego z przypadków wyznaczona została dzienna ilość energii słonecznej.
The article presents the issue of using a wavelet transform to filter quick change components of solar radiation intensity curve. The research involved measuring solar radiation for several days, which were distinctive as regards insolation variability. The measurements were carried out at the rate of 2 samples per second, which is high for insolation measurements. Subsequently, the obtained data was subjected to another sampling at lower frequencies, directly and after quick change components were filtered out using the wavelet transform. Daily solar energy amount was determined for each of the cases.
Źródło:
Inżynieria Rolnicza; 2010, R. 14, nr 7, 7; 87-92
1429-7264
Pojawia się w:
Inżynieria Rolnicza
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Mathematical methods of signal analysis applied in medical diagnostic
Autorzy:
Ciecierski, Konrad A.
Powiązania:
https://bibliotekanauki.pl/articles/331189.pdf
Data publikacji:
2020
Wydawca:
Uniwersytet Zielonogórski. Oficyna Wydawnicza
Tematy:
decision support system
signal filtering
data fusion
temporal analysis
system wspomagania decyzji
filtrowanie sygnału
fuzja danych
Opis:
Digital signal processing, such as filtering, information extraction, and fusion of various results, is currently an integral part of advanced medical therapies. It is especially important in neurosurgery during deep-brain stimulation procedures. In such procedures, the surgical target is accessed using special electrodes while not being directly visible. This requires very precise identification of brain structures in 3D space throughout the surgery. In the case of deep-brain stimulation surgery for Parkinson’s disease (PD), the target area—the subthalamic nucleus (STN)—is located deep within the brain. It is also very small (just a few millimetres across), which makes this procedure even more difficult. For this reason, various signals are acquired, filtered, and finally fused, to provide the neurosurgeon with the exact location of the target. These signals come from preoperative medical imaging (such as MRI and CT), and from recordings of brain activity carried out during surgery using special brain-implanted electrodes. Using the method described in this paper, it is possible to construct a decision-support system that, during surgery, analyses signals recorded within the patient’s brain and classifies them as recorded within the STN or not. The constructed classifier discriminates signals with a sensitivity of 0.97 and a specificity of 0.96. The described algorithm is currently used for deep-brain stimulation surgeries among PD patients.
Źródło:
International Journal of Applied Mathematics and Computer Science; 2020, 30, 3; 449-462
1641-876X
2083-8492
Pojawia się w:
International Journal of Applied Mathematics and Computer Science
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Wybrane metody rekonstrukcji zakłóceń utrwalonych w dowodowych nagraniach dźwiękowych
Selected methods of noise reduction included in evidential audio recordings
Autorzy:
Michałek, Marcin
Powiązania:
https://bibliotekanauki.pl/articles/499883.pdf
Data publikacji:
2013
Wydawca:
Centralne Laboratorium Kryminalistyczne Policji
Tematy:
fonoskopia
filtracja sygnałów
redukcja zakłóceń
korekcja nagrania
forensic audio analysis
signal filtering
noise reduction
improving recording quality
Opis:
Celem niniejszego artykułu jest przedstawienie problematyki redukcji zakłóceń w nagraniach dźwiękowych z uwzględnieniem badań kryminalistycznych. Spisanie treści wypowiedzi z nagrań jest przedmiotem niemal każdej ekspertyzy fonoskopijnej. W dowodowych nagraniach dźwiękowych bardzo często występują liczne zakłócenia, charakteryzujące się dużą amplitudą oraz zakresem częstotliwościowym skutecznie maskującym sygnał mowy. Zakłócenia te mogą wynikać ze sposobu nagrywania, z warunków akustycznych podczas zdarzenia jak i użytej techniki rejestracji. Wiąże się to z koniecznością wykonywania korekcji nagrań w celu poprawy jakości sygnału mowy. W artykule opisano najważniejsze terminy odnoszące się do redukcji zakłóceń w nagraniach dźwiękowych oraz zastosowanie filtrów cyfrowych typu IIR, FIR oraz selektywnych peak i notch. Oprócz klasycznej filtracji sygnałów duże znaczenie w tej dziedzinie mają metody do adaptacyjnej redukcji zakłóceń. Praca przedstawia również właściwości filtrów adaptacyjnych LMS i RLS, algorytmu do estymacji amplitudy sygnału mowy w oparciu o estymator MMSE STSA, jak również metodę odejmowania widma. Oprócz technik wykorzystujących analizę częstotliwościową w artykule zaprezentowano sposoby redukcji zakłóceń w dziedzinie czasu. Wybrane metody zostały opisane teoretycznie. Artykuł zawiera także przykłady praktycznego zastosowania zaprezentowanych filtrów i algorytmów.
The aim of the paper is to present the issue of noise reduction in audio recordings for forensic research. Speech to text transcription is a subject almost every forensic audio analysis. In evidence audio recordings very often there are numerous noises characterized by a high volume and frequency which effectively mask the speech signal. These noises may result from the way of the recording, acoustic conditions during the event and the technique of registration that was used. It is connected with necessity to make correction of the recordings to improve quality of speech signal. The paper describes the most important technical terms related to noise reduction in audio recordings and the use of digital filters like IIR, FIR, peak and notch. In addition to the classical signal filtering methods adaptive noise reduction algorithms play an important part in this case. The article presents some features of LMS and RLS adaptive filters, algorithm for speech amplitude estimation based on MMSE STSA estimator and spectral subtraction method as well. Apart from methods using frequency analysis the article contains some details of noise reduction in time domain. Selected methods were described theoretically. The article also contains examples of the practical application of the presented filters and algorithms.
Źródło:
Problemy Kryminalistyki; 2013, 281; 28-35
0552-2153
Pojawia się w:
Problemy Kryminalistyki
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Separation of groups of free radicals from noised EPR spectrum using genetic algorithm and gradient method
Autorzy:
Bernas, M.
Ramos, P.
Powiązania:
https://bibliotekanauki.pl/articles/951652.pdf
Data publikacji:
2013
Wydawca:
Uniwersytet Śląski. Wydział Informatyki i Nauki o Materiałach. Instytut Informatyki. Zakład Systemów Komputerowych
Tematy:
signal filtering
genetic algorithms
spectra analysis
free radicals
EPR spectroscopy
function approximation
filtrowanie sygnału
algorytmy genetyczne
analiza widma
wolne rodniki
spektroskopia EPR
aproksymacja funkcji
Opis:
Different groups of free radicals exist in biological material like animal tissues or plants parts. The processes like heating or cooling creates additional types of free radicals groups in this organic matter, due to changes in chemical bonds. The paper proposes a method to determine types and concentrations of different groups of free radicals in the matter processed in various temperatures. The method extracts the spectrum of free radicals using electron paramagnetic resonance with the microwave power of 2.2 mW. Then an automatic method to find a best possible fit using limited number of theoretical mathematical functions is proposed. The match is found using spectrum filtration, and a genetic algorithm implementation supported by a Gradient Method. The obtained results were compared against the samples prepared by an expert. Finally, some remarks were given and new possibilities for future research were proposed.
Źródło:
Journal of Medical Informatics & Technologies; 2013, 22; 117-123
1642-6037
Pojawia się w:
Journal of Medical Informatics & Technologies
Dostawca treści:
Biblioteka Nauki
Artykuł
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:
A Comparative Study of Various Edge Detection Techniques for Underwater Images
Autorzy:
Awalludin, Ezmahamrul Afreen
Arsad, Tengku Noorfarahana T.
Yussof, Wan Nural Jawahir Hj Wan
Bachok, Zainudin
Hitam, Muhammad Suzuri
Powiązania:
https://bibliotekanauki.pl/articles/2058499.pdf
Data publikacji:
2022
Wydawca:
Instytut Łączności - Państwowy Instytut Badawczy
Tematy:
edge detection
mean square error
median filtering
peak signal to noise
wiener filtering
Opis:
Nowadays, underwater image identification is a challenging task for many researchers focusing on various ap plications, such as tracking fish species, monitoring coral reef species, and counting marine species. Because underwater im ages frequently suffer from distortion and light attenuation, pre-processing steps are required in order to enhance their quality. In this paper, we used multiple edge detection techniques to determine the edges of the underwater images. The pictures were pre-processed with the use of specific techniques, such as enhancement processing, Wiener filtering, median filtering and thresholding. Coral reef pictures were used as a dataset of underwater images to test the efficiency of each edge detection method used in the experiment. All coral reef image datasets were captured using an underwater GoPro camera. The performance of each edge detection technique was evaluated using mean square error (MSE) and peak signal to noise ratio (PSNR). The lowest MSE value and the highest PSNR value represent the best quality of underwater images. The results of the experiment showed that the Canny edge detection technique outperformed other approaches used in the course of the project.
Źródło:
Journal of Telecommunications and Information Technology; 2022, 1; 23--33
1509-4553
1899-8852
Pojawia się w:
Journal of Telecommunications and Information Technology
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
Filtration of digital signals using wavelets
Autorzy:
Łuczak, D.
Skrzypiński, M.
Powiązania:
https://bibliotekanauki.pl/articles/376402.pdf
Data publikacji:
2016
Wydawca:
Politechnika Poznańska. Wydawnictwo Politechniki Poznańskiej
Tematy:
wavelet transform
on-line filtering
Gaussian noise
digital signal
Opis:
The article constitutes a set of research referring to the removal of the noise of Gaussian distribution of signal using wavelet analysis. The thesis has been launched of presentation of existing analysis algorithms, starting of discrete transform related to Mallat’s algorithm, and ending with the selected methods on inverse wavelet transform. The task was carried out by using the simulation scripts of package Wavelet Toolbox in MATLAB. The studies of using on-line filtering algorithms has been undergone. The delay value, which was implemented by the various filtration methods, was determined through simulations. The inclusion of the comparison of results allows to evaluate the quality of filtration.
Źródło:
Poznan University of Technology Academic Journals. Electrical Engineering; 2016, 85; 187-195
1897-0737
Pojawia się w:
Poznan University of Technology Academic Journals. Electrical Engineering
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
Tytuł:
Basic solutions for a program system for dynamic systems simulation
Autorzy:
Timofeev, A. O.
Powiązania:
https://bibliotekanauki.pl/articles/92971.pdf
Data publikacji:
2006
Wydawca:
Uniwersytet Przyrodniczo-Humanistyczny w Siedlcach
Tematy:
dynamic system simulation
creating interactive applications
Amethyst program system
calculating the component state
decentralized processing the signal delays
filtering
analog objects simulating
Opis:
The article presents basic solutions of the Amethyst program system. The system generates interactive applications for simulating complex dynamic systems. The method of presenting the information about the state of the simulated objects and method of calculating the component state are proposed. Method of decentralized processing the signal delays and filtering is discussed. Method of analog objects simulating is proposed. The article presents examples of projects produced for C++ Builder, Visual C++ and Visual Basic 2005 environment projects. The executable times of simulation are compared for the above mentioned environment applications.
Źródło:
Studia Informatica : systems and information technology; 2006, 1(7); 255-263
1731-2264
Pojawia się w:
Studia Informatica : systems and information technology
Dostawca treści:
Biblioteka Nauki
Artykuł
Tytuł:
The design of digital audio filter system used in tomatis method stimulation
Autorzy:
Jóźwiak, Krzysztof
Bujacz, Michał
Królak, Aleksandra
Powiązania:
https://bibliotekanauki.pl/articles/384341.pdf
Data publikacji:
2019
Wydawca:
Sieć Badawcza Łukasiewicz - Przemysłowy Instytut Automatyki i Pomiarów
Tematy:
sound filtering
digital signal processing
shelving filter
FIR filter
Electronic Ear stimulator
Tomatis Method
filtrowanie dźwięku
przetwarzanie sygnału cyfrowego
filtr FIR
metoda Tomatisa
Opis:
The Tomatis Method is a rehabilitation technique used in psychology, the main aim of which is stimulating the cochlea in the inner ear by filtered air-conducted and bone-conducted sounds. The system of electronic filters and amplifiers used for this therapy is called the Electronic Ear. Commonly, it is a commercial analog device that is expensive and after a few years its functionality declines. In this paper, we propose a digital Electronic Ear system using an STM32F4 family micro-controller and ADC/ DAC integrated circuits. The design of the digital sound filters allows to adjust more parameters and overcomes some of the constraints of analog systems. In this paper, we provide a short review of the Tomatis Method, the main functions of the Electronic Ear and we describe the designed system with comparison measurements to the analog original.
Źródło:
Journal of Automation Mobile Robotics and Intelligent Systems; 2019, 13, 1; 73-78
1897-8649
2080-2145
Pojawia się w:
Journal of Automation Mobile Robotics and Intelligent Systems
Dostawca treści:
Biblioteka Nauki
Artykuł

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