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1

Fuzzy Extreme Analysis in Time- and Frequency Domains

Pogribny W., Drzycimski M., Drzycimski Z.

International Journal of Electronics and Telecommunications

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2011

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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.

2

Optimization of Short Probe Linear Frequency Modulated Signal Parameters

Pogribny W., Leszczyński T.

Archives of Acoustics

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2011

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Vol. 36, No. 4

861-871

EN

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.

3

Modified sliding wiener-khintchin trans form

Pogribny W., Surma D.

Zeszyty Naukowe. Telekomunikacja i Elektronika / Uniwersytet Technologiczno-Przyrodniczy w Bydgoszczy

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2010

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z. 13

31--41

EN

The article presents the new approach to increase a speed of Sliding Discrete Wiener-Khintchin Transform (SDW-KT) algorithm on the basis of recurrent correlation analysis (CA) algorithm using. In this case it is not necessary to calculate the whole correlation function by each analyzing window position. There is taken note of analyzing window parameters selecting for sliding DW-KT, FFT, and Sliding Periodogram too. Worked out approach predominance over SFFT and Periodogram is demonstrated on examples of short noisy signal recognition.

PL

W artykule przedstawiono opracowany nowy szybki rekurencyjny algorytm, ślizgającej dyskretnej analizy korelacyjnej (SDW-KT- Sliding Discrete Wiener-Khitnchin Transform), pozwalający na istotne zmniejszenie ilości operacji przy kolejnych przemieszczeniach okna analizy, co prowadzi do zwiększenia szybkości przetwarzania SDW-KT. Zwrócono także uwagę na dobranie parametrów okien dla ślizgających przetwarzań DW-KT, FFT (Fast Fourier Transform) i PERIODOGRAMU. Przewagi opracowanych podejść nad SFFT i ślizgającym periodogramem przedstawiono na przykładach wykrywania krótkotrwałych zaszumionych sygnałów.

4

Implementation of the direct windowed DHT parallel algorithm

Pogribny W., Sulima M.

Zeszyty Naukowe. Telekomunikacja i Elektronika / Uniwersytet Technologiczno-Przyrodniczy w Bydgoszczy

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2010

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z. 13

93--102

EN

Hardware realizations of an algorithm of discrete Hilbert transform (DHT) are made mainly with using the FFT because of its higher resolution [1] [4] in comparison to the direct DHT based on unwindowed impulse response (IR). On the other hand, as it is shown previously in [5] [6] the appropriate use of weight windows for IR provides a higher resolution of the direct DHT in comparison to the DHT method with FFT using. In this work a parallel algorithm and appropriate filter structure for the direct windowed DHT which are faster than the algorithms and structures based on the FFT are presented. The work of the algorithm and filter is illustrated by examples of phase transitions capturing of noisy signals.

PL

Sprzętowe realizacje algorytmu dyskretnego przekształcenia Hilberta (DHT) w przewadze wykonywane są przy użyciu DHT opartego na FFT, ze względu na jego wyższą rozdzielczość w porównaniu z bezpośrednim DHT z nieokienkowaną charakterystyką impulsową. Z drugiej strony, jak pokazano prędzej w [5] [6], odpowiednie użycie okien wagowych dla charakterystyki impulsowej pozwala otrzymać wyższą rozdzielczość bezpośredniego DHT w porównaniu z metodą DHT opartą na FFT. W tej pracy zostały zaprezentowane algorytm równoległy oraz odpowiednia struktura filtru dla bezpośredniego okienkowanego DHT, które są szybsze w porównaniu z algorytmami i strukturami opartymi na FFT. Działanie zaprezentowanych algorytmu i filtru zostało zademonstrowane na przykładach przechwytywania zmian fazy sygnałów zaszumionych.

5

Recognition improvement of short chirp signals

Pogribny W., Leszczyński T.

Electronics and Telecommunications Quarterly

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2008

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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.

6

Influence of chirp signal parameters on windowed matching filtration resolution

Pogribny W., Leszczyński T.

Hydroacoustics

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2007

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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.

7

The increase of DFT and DCT computation rate and accuracy with the use of parallel operations

Pogribny W., Drechny M.

Electronics and Telecommunications Quarterly

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2007

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Vol. 53, No 2

155-176

EN

The increase of DFT and DCT DFT (Discrete Fourier Transform) and its derivative DCT (Discrete Cosine Transform) are the transforms most often used in DSP (Digital Signal Processing), especially in data communications for signal compression [1, 2, 3, 4]. DFT and DCT algorithms have been modified and their rate and accuracy optimized for many years [3, 4]. Most of them are calculated in multibit PCM (Pulse Code Modulation) format. The differential DPCM (Differential Pulse Code Modulation) format, used in this work can be an alternative for PCM format applied in DFT and DCT. It ensures higher accuracy of computation with code word length shorter than PCM code word. When we modify DPCM format (Section 3) in such a way that the quantization steps are set of the numbers with a base 2 and exponent belonging to a natural numbers set, the multiplication operation rate, as one of the most often used operation in DSP, increases. It is possible because multiplication operations can be replaced with fast shift bit logical operations. The parallel combination of some MDPCM (Modified Differential Pulse Code Modulations) codes creates SDPCM (Synthesized Differential Pulse Code Modulation) code (Section 3), which has high computational accuracy, equal to the DPCM accuracy, however it does not require multiplications. In most cases, parallel computations lead to their rate increase in comparison to computation rate of sequentially operations. These calculations, apart from using appropriate and accurate algorithms require applying the systems which enable the effective work of the parallel methods. Thus, for this purpose the programmable FPGA devices (Fields Programmable Gates Array) have been the most commonly used recently. Their main advantages are high speed of operations, the possibility of programming every computational structure and their low price. In this work, apart from fast parallel DFT and DCT algorithms, we presented the structures of processing DFT and DCT systems (specialized processors) working in parallel way. The processing systems presented in Section 5 allow fast and accurate calculations without time-consuming multiplications. With reference to the article [5] presenting fast differential DCT algorithms, in this work the authors proposed another way of increasing the rate and accuracy of DCT computations, which consists in the modifications of a partV

8

Różnicowa DCT w zagadnieniach przetwarzania sygnałów

Pogribny W., Drechny M.

Kwartalnik Elektroniki i Telekomunikacji

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2006

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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.

9

Realizacja splotów w formatach mieszanych z wykorzystaniem operacji przesunięcia

Pogribny W., Marciniak B.

Kwartalnik Elektroniki i Telekomunikacji

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2005

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Vol. 51, z. 4

555--570

PL

Wysoka rozdzielczość i szybkość filtracji mogą być osiągnięte dzięki zastosowaniu w operacjach splotów formatów mieszanych DM-PCM, w których sygnał wejściowy jest przedstawiony w formacie modulacji delta (DM) a współczynniki wagowe charakterystyki impulsowej w formacie PCM. Wiadomo, że wykorzystanie modulacji DPCM czy adaptacyjnej ADPCM, jako popularnych rodzajów DM, w filtracji cyfrowej pozwala na uzyskanie znacznie lepszej rozdzielczości w porównaniu z innymi rodzajami DM, ale wymaga wykonania operacji na liczbach wielobitowych. Aby uniknąć tej niedogodności w pracy zaproponowano aby kroki modulacji DPCM syntezować z kroków zmodyfikowanych DPCM (MDPCM), w których przyjęto, że kolejne kroki kwantyzacji są proporcjonalne do potęg liczby 2. Takie rozwiązanie pozwala na całkowite zastąpienie operacji mnożenia operacją przesunięcia bitów odpowiednich słów kodowych, co prowadzi do zwiększenia szybkości filtracji i pozwala na obliczenie poszczególnych składników splotu w czasie jednego taktu pracy procesora. Metoda ta może być efektywnie zrealizowana za pomocą układów programowalnych (np. FPGA - ang. Field Programmable Gate Array), pozwalających na uproszczenie struktury projektowanego filtru, zwiększenie jego niezawodności, obniżenie kosztów wytwarzania oraz gabarytów. Celem pracy jest zwiększenie rozdzielczości i jednocześnie szybkości filtracji w oparciu o sploty w formatach mieszanych np. syntetyczna DPCM wraz z PCM (SDPCM-PCM) i zaproponowanie odpowiednich struktur procesorów specjalizowanych. Otrzymane wyniki badań symulacyjnych wskazują na celowość wykorzystania takich algorytmów w procesorach specjalizowanych do filtracji cyfrowej.

EN

High resolution and speed of filtration can be obtained by the use age of mixed DM-PCM formats in convolutions when input signal is represented in delta modulation (DM) format and weight factors of impulse response are represented in PCM format. It is known, that use of differential modulation DPCM or adaptive ADPCM in digital filtration allows to obtain higher resolution in comparison with other kinds of DM, but requires to perform operations on multibit numbers. In order to avoid the inconvenience, in this work we propose to synthesize the DPCM quantization steps on base steps of modified DPCM (MDPCM), the steps of which are proportional to the powers of 2. This solution allows to change multiplication operations into shift operations on bits of appropriate code words. It allows to increase filtration speed as well as to compute the respective components of convolution during one tact of processor. In this article we have shown 3 ways of SDPCM step synthesis. In the 1st option, the synthesis was carried out on the basis of steps of 2 MDPCM coders. However, with the lengthening of code words the SNR value increased as well. It is turned out that for the lengths of code words above 5 bits, the missing values of coder SDPCM steps strongly affect the result (Table 1). In order to avoid such situations, in the 2nd option, for synthesis of the SDPCM coder's steps 3 MDPCM coders were used, giving better SNR values (around 10 dB) at the same time. The 3rd option, which is a modification of option 2, allows to shorten the code words in the respective MDPCM coders. Purpose of this work is to improve the resolution and at the same time the fast-acting of filtration based on the convolution in mixed formats e.g. synthetic DPCM together with PCM (SDPCM-PCM) as well as working-out the suitable specialized processor structures. Obtained simulation results indicate efficiency of these algorithms destined for specialized processors to digital filtration.

10

Analysis of noisy signals on base of differential correlation algorithms

Pogribny W., Rozhankivsky I., Leszczyński T., Lutowski Z.

Hydroacoustics

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2005

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Vol. 8

155-162

EN

The study was conducted in Lake Lansk (surface area - 1042 ha, maximum depth - 53 m), in which the former dominant was vendace. Commercial catch results from the past decade indicate that there has been a seventy-two-fold decreased in vendace stocks. Hydroacoustic estimates indicate that there are 7.9 million fish in the pelagic zone. Approximately 93% of these inhabit the epilimnion (6.8 thousand fishźha-1), which is 7.6 times larger than that in the hypolimnion. There were 2.5 times fewer fish in a similarly sized southern region of the lake in 2004 than in 2001. Using the numbers, species structure, and individual weight, the biomass of the fish inhabiting the pelagic zone was calculated at 194.7 kgźha-1. The largest share was of vendace at 45.7% (89.0 kgźha-1), followed by roach at 18% (35.0 kgźha-1) and bleak at 13.1% (25.5 kgźha-1). The share of 0+ age group vendace was determined to be 70%. This might explain the decrease in commercial vendace catches caused by poor environmental conditions and overfishing. It also forecasts an improvement in resources of this species in the coming years.

11

Kodowanie i odtwarzanie sygnałów losowych z wykorzystaniem ekonomicznych modulacji różnicowych

Pogribny W., Rajs A.

Kwartalnik Elektroniki i Telekomunikacji

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2002

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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

12

Studying the exactitucle of digital matching filtration of widepass location signals

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

Hydroacoustics

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2001

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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.

13

The use of fast convolution and correlation analysis to increase the resistance of digital location

Pogribny W., Rożankiwsky I., Milewski A., Drzycimski Z., Leszczyński T.

Hydroacoustics

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1999

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Vol. 2

167--170

EN

The method of analog location with the use of devices with SAW that is now being used in our country is fast, however it is nof accurate enough and has insufficient resistance to bofh natural and artificial noise. In the paper is proposed to increase the resistance of the digital location system by means of using a location system manipulated with noise-like code, and additional signal compression, filtration based on fast convolution and correlation analysis to isolate noise-like codes. The system proposed here easily adjusts to the changes in the code of the signal generated by the radar, and it also works in real time.