Wyniki wyszukiwania - Biblioteka Nauki

1

Kodowanie i odtwarzanie sygnałów losowych z wykorzystaniem ekonomicznych modulacji różnicowych

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Pogribny W. , Rajs A.

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2002

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tom Vol. 48, z. 1

7-25

PL

Praca jest poświęcona zbadaniu algorytmów różnicowego kodowania z równomiernym próbkowaniem sygnałów losowych. Uwagę skupiono na algorytmach koderów z krokami kwantowania proporcjonalnymi do potęgi liczby 2. Algorytmy te umożliwiają uproszczenie operacji mnożenia, sprowadzając je do zwykłego przesunięcie kodu, co jest szczególnie ważne przy cyfrowej obróbce sygnałów. Celem pracy jest opracowanie i zbadanie dokładności ekonomicznych i efektywnych algorytmów kodowania sygnałów losowych z wyżej wymienionymi krokami na podstawie różnicowej PCM (DPCM) oraz jednobitowej adaptacyjnej modulacji delta (ADM) ze zmienną stromością w dziedzinach czasowej i częstotliwościowej oraz propozycja struktur koderów i dekoderów z wykorzystaniem opracowanych algorytmów. Jako sygnały wejściowe wybrano szum różowy oraz sygnał mowy (słowa w języku polskim według podziału na fonemy). Wybór algorytmów opierał się na minimalizacji błędów średniokwadratowych w dziedzinach czasowej i częstotliwościowej, oraz ocenie subiektywnej.

EN

This paper is devoted to study of differencial alghoritms to random signals encoding. We were considered alghoritms of encoders with multiplication factor equal to power 2, which make possible very simple mathematical operations (code shift only). This is esspecially important to signals procesing. Purpose of this paper is work out and testing accuracy economical encoders algorithms to random signals which based on one-bit adaptative delta modulation (ADM) and differencial PCM (DPCM) in time and frequency domains, and we propose encoder and decoder structures. The exploration were conducted for pink noise and voice signal (according to division on polish phonemes) which was a man, women and child voice. Choice of the algorithms was carried out by means of minimising the mean square error quantization in time and frequency domains, and subjective estimate

2

Influence of chirp signal parameters on windowed matching filtration resolution

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Pogribny W. , Leszczyński T.

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tom Vol. 10

157-162

EN

In this work we studied the proposed effective methods of increasing of a resolution of a matched filtration in time domain of the short broadband signals with the use of linear modulation of frequency (the so-called chirp signals) which product BT does not exceed 50 and initial frequency is not zero. The choice of matched filter pulse response parameters as smoothing window and matched to it an initial phase and a sampling rate as well as the influence of non linear operations on the matching filtration results have been investigated in order to essentially narrow the output signal main lobe as well as increase the chirp compression. The best results of the recognition resolution and compression improvement occur when at the same time there are used the non linear operations on the convolutions and the pulse response parameters are matched to the chosen window.

3

Recognition improvement of short chirp signals

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Pogribny W. , Leszczyński T.

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2008

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tom Vol. 54, No 2

113-125

EN

Broadband signals with the use of linear frequency modulation LMF (so-called chirp signals) are widely used in localization. Recognition of the chirp signals depends mainly on their compression based on the matched filtration. This work presents two simple solutions of the essential improvement of the recognition resolution of the short chirp signals, which product BT does not exceed 100. The first of them is connected with matching chirp and impulse response (IR) parameters to a specific window so that the form of the filter frequency response (FR) would be the nearest to the form of the window amplitude spectrum. The second one is based on removing the negative convolutions of the matched filtration in the time domain. The authors have found that the best result is achieved when both methods are used simultaneously. These ways were checked out for different windows and chirps with bigger BT too. Apart from that, the structure of matched filter (MF) with the use of above-mentioned methods is proposed.

4

Różnicowa DCT w zagadnieniach przetwarzania sygnałów

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Pogribny W. , Drechny M.

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tom Vol. 52, z. 4

493-509

PL

Dyskretna transformata kosinusowa (DCT) jest szeroko stosowana do analizy i kompresji sygnałów. Klasyczne algorytmy DCT bazują na obliczeniach wykonywanych w wielobitowym formacie PCM (Pulse Code Modulation). Ograniczeniem zastosowania formatu PCM w DCT jest użycie wielobitowych operacji mnożenia, co często utrudnia jego wykorzystanie w systemach czasu rzeczywistego. Z drugiej strony, użycie niektórych niskobitowych rodzajów modulacji delta (DM) zapewnia taką samą dokładność obliczeń jak w przypadku użycia wielobitowego formatu PCM, lecz takie podejście nie zostało wystarczająco zbadane. Celem pracy było opracowanie i zbadanie efektywnych różnicowych algorytmów i struktur procesorów DCT pracujących w czasie rzeczywistym. W pracy tej przedstawiono nowe szybkie różnicowe algorytmy DCT. Szczególną uwagę skupiono na możliwości wyeliminowania w zaproponowanych algorytmach czasochłonnych operacji mnożenia i zastąpienia ich szybkimi operacjami bitowego przesunięcia SHIFT. Zastosowanie przesunięć umożliwiają kroki kwantowania będące potęgą liczby 2. Do zbadania dokładności zaproponowanych algorytmów wykorzystano sygnały szumu różowego oraz mowy. Zaproponowane zostały szybkie i efektywne struktury procesorów specjalizowanych realizujące opracowane algorytmy. Wyniki symulacji komputerowych potwierdziły efektywność opracowanych podejść.

EN

Discrete Cosine Transform (DCT) is widely used in analysis and signal compression. The classical DCT algorithms are based on calculations in multibit PCM (Pulse Code Modulation) format. The restriction of using PCM format in DCT is the application of multibit multiplication operations, which often limits the usage of PCM in real time systems. On the other hand, using some low bit kinds of delta modulation (DM) ensures the same calculation accuracy as in case of the multibit PCM format, however this approach has not been examined sufficiently yet. The purpose of the work was working out and studying the effective differential DCT algorithms and processors structures working in real time. In this work are presented new fast differential DCT algorithms. The particular attention was paid to the possibility of eliminating the multiplication operations in the proposed algorithms and replacing them by fast SHIFT operations. The application of shifts is done on the basis of the quantization steps, which are the power of number 2. To study the accuracy of proposed algorithms pink noise and voice signals were used. The authors proposed fast and effective structures of specialized processors realizing the worked out algorithms. The results of computer simulations confirm the efficiency of the presented approach.

5

Studying the exactitucle of digital matching filtration of widepass location signals

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Pogribny W. , Rozhankivsky I. , Milewski A. , Lozynsky V.

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tom Vol. 4

209--212

EN

The resolution of matching digital filtration of wide pass location signals in time domain is studied in this work. The advantages of this kind of processing in comparison to the processing on fast convolutions in frequency domain are demonstrated. A choice of parameters of the long and short chirp signals and smoothing window have been studied to increase the ratio of the main lobe to side lobes of convolutions. This choice has carried out with the use of computer simulation and result has shown in this paper.

6

Fuzzy Extreme Analysis in Time- and Frequency Domains

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Pogribny W. , Drzycimski M. , Drzycimski Z.

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tom Vol. 57, No. 4

511-518

EN

Very often, we are interested in the shape of a signal or an envelope of the signal or its spectrum. Classical extreme analysis (CEA) produces too many minor details of a signal shape. This fact, in the context of telecommunications, does not allow us to ensure a significant coefficient of a signal compression. Another option for signal analysis, based on the Delta Modulation (DM), lacks sufficient dynamic range at a relatively low sampling rate. One more "true envelope" manner is based on the cepstral analysis, and requires too many operations. Other methods are used for the definition of a signal envelope. The discrete Hilbert transform (DHT) is only expedient to capture an envelope of a narrowband signal and, moreover, requires too many mathematical operations. Other manners based on decimation, as well as the use of a signal rectifying followed by low-pass filtering do not always ensure sufficient accuracy and signal compression coefficient. Fuzzy EA (FEA) is free from similar drawbacks. Its first and second differences are compared with no zero limits, and that allows us to take into consideration only major details of the signal or spectrum shape. Consequently we obtain both an envelope of wideband signal, and a signal significant compression in real time. This article focuses on FEA features connected with the aforementioned tasks. Apart from the FEA algorithm, the article outlines some methods of signal reconstruction after FEA in both domains, and the structure of the FEA specializedprocessor. FEA application in both domains is demonstrated through examples.