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“Digital
Signal Processing” No. 4-2019 |
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Analysis of quantized FIR filters
Mingazin A.T.
, e-mail: alexmin@radis.ru
RADIS Ltd, Russia, Moscow
Keywords: optimum FIR filters, quantized coefficients, variation of initial parameters, analysis of direct, cascade and bound real structures.
Abstract
For analysis/synthesis of FIR filters with quantized coefficients (quantized FIR filters), a direct method is often used. Due to its simplicity has found widespread use in many publications and books on digital signal processing. As a rule the degree of influence of the coefficient quantization on the change of the filter magnitude response is estimated for only one set of initial parameters. At the same time the coefficient wordlength can be specified or selected iteratively in order to find its minimum value without violation of requirements to a magnitude response. However, by using one or even several sets of parameters it is not always correct to judge about the advantage of an magnitude response approximation or filter structure. With the help of variation of initial parameters (VIP) it is possible to improve results of direct method of synthesis of quantized FIR filters and even to find solutions comparable to obtained by methods of variation of coefficients on discrete set of their values. In addition, plotted curves of the controlled parameters of the FIR filter from a selected initial parameter can be directly used for analysis (VIP analysis) of coefficient quantization effect, similar to that previously applied to Zolotarev-Cauer IIR filters.
In this paper, VIP analysis is applied to optimal low-pass FIR filters with symmetrical impulse response synthesized on the basis of the Remez - Parks - McClellan algorithm (see, for example, the function cremez (...) in MATLAB). The description of controlled and initial parameters is given. Dependences of controlled parameters on initial parameters for continued and quantized coefficients îf low-pass filters are presented. The piecewise constant nature of the plotted curves is explained. The analysis of similar curves for direct structure narrowband and wideband filters at big and small relations of ripple levels of magnitude response in passband and stopband is carried out. Dependences of the maximum relative error of the magnitude response on the initial ratio of ripple levels for direct and cascade and also for direct and bound real filter structures are compared.
References
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The discrete structure of the zeros and poles location in the Z-plane of the recursive digital filters with a finite word length
Lesnikov V.A., e-mail: vladislav.lesnikov.ru@ieee.org
Naumovich T.V.
Chastikov A.V.
Keywords: IIR digital filters, resolved positions of the zeros and poles, finite wordlength, discretized z-plane, plane algebraic curves.
Abstract
It is known that the resolved positions of the zeros and poles of the second order IIR digital filters with finite wordlength form a discrete structure called in this paper the topography of a discretized z-plane. In the publications of the authors, the z-plane topography was described in detail. This article is devoted to the definition of equations describing plane algebraic curves on which the allowed positions for the zeros and poles of filters of arbitrary order are located.
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Adaptive Antenna Array for Operation in Interference and Multipath Conditions
Djigan V.I., Doctor of Sciences, a Chief Scientific Officer of the Institute for Design Problems in Microelectronics of Russian Academy of Sciences, Moscow, Russia, and a Professor of the National Research University of Electronic Technology, Moscow, Russia., e-mail: djigan@yandex.ru
Keywords: adaptive array, RLS algorithm, pilot signal, circular array.
Abstract
Today adaptive signal processing is widely used in radio system equipment in the part of antenna arrays and channel equalizers. This became possible due to the digital antenna array technology, which provides an access to antenna signals in base-band, because the signals are required for the processing in adaptive filters, which adaptive arrays and equalizers are based on.
Adaptive arrays use different criteria of operation, among which is a criterion of the Mean Square Error (MSE) between pilot and array output signals. This criterion is widely used in communication systems. The solution allows to suppress interferences in array output signal and to steer the main beam of the array towards the desired signal source, if there is no multipath of the signal.
However, if there is a multipath, the adaptive algorithm cannot minimize MSE. An equalizer, which has to suppress the multipath, cannot do its job well due to interference signals, which are not suppressed completely at array output. Such independent cascade solution (adaptive array and equalizer) is not effective, that was demonstrated in a diversity of publication on the topic.
The given paper proposes a solution of the problem, which is based on a simultaneous using of an adaptive antenna array and a feedback equalizer for signal receiving, if there are the sources of the interference and the desired signal multipath.
The paper presents the architecture of such array and equalizer, adaptive algorithm (Recursive Least Squares, RLS) and simulation results, confirming the efficiency of the solution in a circular array.
The simulation are conducted for the circular array with 3 antennas, which receives PSK-8 signal via two-rays channel in the presence of two interference sources with 30 dB Signal-to-Interference Ratio (SIR) each.
The solution with symbol-rate equalizer demonstrates the ability to steer of the main beam of array towards the information signal source, to suppress interference signals creating dips in radiation pattern of about -100 dB, and equalize the multipath channels with 0.05 …. 0.5 dB final frequency response ripple dependently Signal-to-Noise Ratio (SNR) in array receivers (-30 … -10 dB).
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Doppler frequency shift estimation of single-tone phase-shift-keying signal
Egorov V.V.1,2
Lobov S.A.1,2
Maslakov M.L.1, e-mail: maslakovml@gmail.com
Mingalev A.N.1,2, e-mail: ming@list.ru
1JSC «Russian institute for power radioengineering», Russia, Saint-Petersburg,
2Saint-Petersburg State University of Aerospace Instrumentation, Russia, Saint-Petersburg
Keywords: Doppler shift, estimation, phase, phase-shift-keying.
Abstract
The operation of mobile radio communication systems, in cases where one or both subscribers are in motion, a Doppler frequency shift occurs. In addition, the Doppler frequency shift can be caused by the movement of the surface reflecting the radio signal (for example, a high frequency radio communication is characterized by a change in the height of the ionosphere layer), as well as a possible discrepancy in the frequencies of the reference generators in the transmitting and / or receiving devices. The Doppler frequency shift leads to a significant increase the bit error rate of data transmission systems.
The Doppler frequency shift estimation problem occurs in almost all radio communication and radar systems, and is especially relevant in aviation and satellite communication systems, as well as in positioning systems.
In this work, the authors propose a method for the Doppler frequency shift estimation from information single-tone signals with phase-shift-keying modulation based on the symbols phase values.
The proposed method allows real-time estimation of the Doppler frequency shift of the received signal. A feature of this method is that there is no need to transmit a special test signal. Estimation is carried out directly by the information signal based analyses the symbols phases. The information bit sequence is unknown. Also note that relatively small sequences of information BPSK symbols are required to obtain an estimation.
Addition, the methods of correct processing on the phase circle are proposed.
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Detection of impairment of speech function in patients with ALS based on analysis of voice signal
M.I. Vashkevich, e-mail: vashkevich@bsuir.by
Yu.N. Rushkevich , e-mail: rushkevich@tut.by
Belarusian State University of Informatics and Radioelectronics (BSUIR), Belarus, Minsk
Keywords: amyotrophic lateral sclerosis, acoustic analysis, jitter, shimmer, classification.
Abstract
The paper presents an approach to the detection impairment of speech function in patients with amyotrophic lateral sclerosis (ALS) based on acoustic analysis of a voice signal. The aim of the work was to check the suitability of the sustain vowel /a/ phonation test for the automatic detection of patients with ALS. There are four groups of voice signal parameters obtained using fundamental frequency (f0) analysis have been considered: 1) jitters (4 variants); 2) shimmers (5 variants); 3) statistical parameters of (f0) (standard deviation, phonatory frequency range, directional perturbation factor); 4) time-frequency parameters of (f0) contour (pitch period entropy, pathology vibrato index). The total number of parameters extracted from voice signal was 14.
The voice recordings used in this study were collected in the Republican Scientific and Clinical Center for Neurology and Neurosurgery (Minsk, Belarus). A total of 54 people were recorded, including 39 healthy (23 men, 16 women) and 15 ALS patients with signs of bulbar dysfunction (6 men, 9 women). The average age in the healthy group was 41.9 years (SD 16.3), and the average age in the ALS group was 57.7 years (SD 9.0). For discriminating between the two classes of normal and pathological cases two widely spread machine learning algorithms: linear discriminant analysis (LDA) and k-Nearest Neighbors (k-NN) were used. For determining classification accuracy k-fold cross-validation method (with k=4) was used. Since the dimensionality of the feature sets was low we have performed exhaustive search through all possible feature subset to find the best one.
The experiments showed that the use of the proposed methods of acoustic voice analysis and classification according to the method of k nearest neighbors made it possible to obtain the detection system with accuracy of 95.7% (sensitivity 91.5% and specificity of 97,4%).
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Noise immunity of a modem based on dynamic chaos according to Veshkurtsev law in a channel with Gaussian noise
Yu.M. Veshkurtsev, e-mail: vym1940@mail.ru
Institute radiolektroniki, service and diagnostics (IRSID), Russia, Omsk
Keywords: dynamic chaos, Gaussian noise, characteristic function, Veshkurtsev law, modem, mathematical expectation, noise immunity, error probability, signal-to-noise ratio.
Abstract
Digital technologies extend the limits of the use of mathematics in the development of new signal processing algorithms. It becomes possible to apply random quantities and even random processes and thus move to dynamic chaos. A modem for data transmission is considered by modulating a characteristic function of a signal with large-scale chaotic, instantaneous values of which are distributed according to Veshkurtsev`s law. This law was first proposed by the author in his publications describing the modem. It also shows that in a channel with "white" noise the modem has noise immunity by 27 dB higher than the same characteristics of its nearest analogues at the same probability of errors.
Analysis of noise immunity of the proposed modem in a channel with Gaussian noise other than "white" noise was performed. It has been found that the noise immunity of a modem depends on the mathematical expectation of noise in a complex manner. It can be increased by 10 dB at a fixed error probability or reduced to an error probability of 0.5 in the range of 40 dB signal-to-noise power ratios with a lower range limit of minus 10 dB.
For the wired communication channel, recommendations have been developed to combat the interfering factor in the form of mathematical noise expectation. In radio channel, compensation of mathematical noise expectation at modem input is performed automatically due to operation of antenna-feeder system. Thus, the noise stability of the modem remains the same with the probability of mistakes at the level of 1•10^(-45).
References
1. Gulyaev Yu.V. Information technologies on the basis of dynamic chaos for transfer, processing, storage and information security / Yu.V. Gulyaev, R.V. Belyaev, G.N. Vorontsov, N.N. Zologin, V.I. Kalinin, E.E. Kalyanov, V.V. Kislov, V.Ya. Kislov, V.V. Kolesov, E.A. Myasin, E.P. Chigin//Radio electronics. Nanosystems. Information technologies, 2018. Volume 10. -- ¹ 2. pp. 279 - 312.
2. Veshkurtsev Yu.M. Noise immunity of modem during reception of signal with distribution of instant values according to Tikhonov law//Digital signal processing, 2019. ¹ 2. pp. 49 - 53.
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Formation of radar images in the automotive industrial monitoring system with the combination of autofocus and motion compensation
V.V. Androsov
S.V. Vityazev, e-mail: vityazev.s.v@rsreu.ru
D.V. Valuyskiy, e-mail: valuyskiy.d.v@tor.rsreu.ru
V.V. Vityazev, e-mail: vityazev.v.v@rsreu.ru
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: radar imaging, PGA, autofocus, automotive radar imaging, motion compensation, resolution enhancement.
Abstract
The article deals with the problem of formation of radar images in order to map and monitor the state of industrial areas with the use of a car as a carrier of a radar station. The features of radio vision conditions in such systems are described. The solution of trajectory instabilities compensation problem and joining of partial frames into a full image frame by methods of PGA autofocus and motion compensation are presented.
The results of modeling and field experiments illustrating the effectiveness of the proposed solutions are given. Additionally, the use of sharpness optimization autofocus as an alternative to PGA, as well as the use of a stripmap mode of imaging radar operation are considered.
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Investigation of the passage of a quantized primary timing sequence (PSS) in LTE OFDMA technology through a fading channel model
Kiseleva T. P., e-mail: golzev2011@yandex.ru
Keywords: quantization, the number of quantization levels, the sequence Zadoff-Chu (ZC),primary synchronization signal (PSS), the technology of LTE, OFDMA, cross-correlation function (VCF), a communication channel with Rayleigh fading.
Abstract
In this article, we continue to study the quantized Zadov-Chu (ZC) sequence used as the primary synchronization sequence (PSS-Primary Synchronization Signal) when constructing the data frame of the downlink (DL) technology LTE OFDMA (Long Term Evolution Orthogonal Frequency Division Multiple Access). A model of the communication channel with Rayleigh attenuations is presented and the characteristics of the channel for studying the passage of the non-quantized and quantized ZC sequence with a quantization step of 1/16, 1/32, 1/64 are determined. Simulation mathematical modeling of the passage channel with Rayleigh fading primary synchronization sequence PSS built on the ZC sequence(of 25.62) in the transmitted frame of information did not reveal a significant difference in the quality of synchronization in the case of using unquantized sequence or ZC sequence is quantized with the given quantization step. The result of the research is the obtained area of uncertainty of the intercorrelation function (VCF) of the test and accepted sequence ZC(25.62), which passed the Rayleigh communication channel with a different number of paths of reflected signals. The results of the VCF calculation are averaged over 50 tests for each number of paths, both in terms of the maximum value of the normalized VCF and in terms of the time delay relative to the time position of the maximum peak of the normalized autocorrelation function (ACF) of the ZC test sequence(25.62). An algorithm for estimating the delay spread of reflected signals is proposed to infer the possibility of not using the equalizer for the Rayleigh channel model under consideration at the stage of user synchronization with the base cellular station.
Quantization of the ZC sequence both when used as a PSS in the DL direction, and as a preamble and probing and demodulated sequences in the UL transmission direction reduces the computational load and hardware complexity and, as a result, the cost of mobile systems.
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