Digital Signal Processing

Scientific & Technical

“Digital Signal Processing” No. 2-2017

19 International Scientific Research Conference “Digital Signal Processing and its Application - DSPA-2017”

In the issue:

- multirate signal processing
- synthesis of the integer IIR filters
- radar signal processing
- forming radar image
- LFMCW synthetic aperture radar
- pitch frequency
- energy efficiency of telecommunication systems
- HF data modems

- neural network signal processing

Theoretical Fundamentals of Design of Telecommunication Systems with High Energy Efficiency
M.A. Bykhovskiy
Moscow Technical University of Communications and Informatics (MTUCI), Russia, Moscow, e

Keywords: energy efficiency of telecommunication systems, Shannon’s border, multidimensional signalensemble, noise immunity code, telecommunication systems design.

The author determines the borders of telecommunication systems’ energy efficiency, which allow for comparison of values of the main parameters of specific telecommunication systems (marginal speed of message transmission and signal-to-noise per bit ratio) with potentially possible values of these parameters. It is shown that when designing a telecommunication system with high energy efficiency, it is advisable to choose optimal modulation methods for message transmission – multidimensional volume-spherical or surface-spherical signal ensembles. The article discusses the issues of parameter selection of such signal ensembles that allow to approach Shannon’s border the spectral efficiency of a system being designed. In addition, the work discusses the issues of such noise immunity code selection for a designed system, that allow simplification of the communication system through the usage of the simplest, in the technical sense, modulator/demodulator, as well as coder and decoder.


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5. V.A. Varguasin, I.A. Cikin. Methods of Increase of Power and Spectral Efficiency of a Digital Radio Communication. SPb.: BHB-Peterburg, 2013, p. 352.

6. Bykhovskiy M.A. Veroiatnost oshibki dlia optimalnih mnogomernih kodov v gaussovskom kanale sviazi i ih osnovnie harakteristiki (Bit error probability for optimal codes in a gaussian communication ñhannel and their key characteristics.) // Electrosvyaz, 2016, no. 2, pp. 55-61

7. Bykhovskiy M.A. Pomehoustojchivost priema optimalnih signalov raspologenih na poverhnosti N-mernoi sferi (Noise immunity of the reception of optimal signals located on a surface of an N-dimensional sphere). // Electrosvyaz, 2016, no. 3, pp. 40-46

8. Bykhovskiy M.A. Propusknaia sposobnost kanala sviazi pri peredatche signalov ogranitchenoi dlitelnosti (Bandwidth capacity of telecommunication channel during transmission of signals of limited duration.) // Electrosvyaz, 2016, no. 8, pp. 37-42

Energy Efficiency of Telecommunication Systems that use Multidimensional Signal Ensembles and Cascade Source Coding
M.A. Bykhovskiy, e-mail:
Moscow Technical University of Communications and Informatics (MTUCI), Russia, Moscow

Keywords: cascade source coding, multidimensional signal ensemble, noise immunity of signal reception, Reed-Solomon codes, complexity realization of demodulators and decoders.


The author investigates a telecommunication system that uses cascade source message coding and where the signals from multidimensional surface-spherical (SS) signal ensemble; such signals create the inner code without contributing excess symbols) are used for message transmission via continuous telecommunication channel; Reed Solomon (RS) code is used as an outer code.

The article contains formulas allowing to determine the probability of error reception of the signals in the investigated telecommunication system; such formulas determine the impact of the SS signals’ length and Reed-Solomon codes’ length on the noise immunity of the signal reception.

It is shown that in such a system it is possible to provide energy efficiency of message transmission that is fairly close to energy efficiency of an “ideal”, according to Shannon, telecommunication system while using rather simple demodulators of the received signals and decoders of the RS codes. It is also noted that cascade codes are more energy efficient than modern turbocodes.

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2. W. Wesley Peterson, E.J. Weldon, Jr., Error Correcting Codes. The MIT Press Cambridge, Massachhusetts and London, England, 1972, p. 594.

3. Bykhovskiy M.A. Pomehoustojchivost priema optimalnih signalov raspologenih na poverhnosti N-mernoi sferi (Noise immunity of the reception of optimal signals located on a surface of an N-dimensional sphere). // Electrosvyaz, 2016, no. 3, pp. 40-46

4. V.A. Varguasin, I.A. Cikin. Methods of Increase of Power and Spectral Efficiency of a Digital Radio Communication. SPb.: BHB-Peterburg, 2013, p. 352.

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Application of Guard Interval in Single-tone HF Data Modems
Maslakov M.L., e-mail:
PJSC «Russian institute for power radiobuilding», Russia, Saint-Petersburg

Keywords: intersymbol interference, adaptive correction, impulse response of the channel, integral convolution type equation, ill-posed problem, guard interval, bit error rate.

It is known that in modern single-tone or serial HF data modems the methods of adaptive signal correction are used for compensation of intersymbol interference (ISI) [2, 3]. Transmitted signal is provided with test symbols sequence for the calculation of correction filter (CF) or equalizer coefficients.

The work is intended to decrease bit error rate (BER) of a single-tone HF data modem by introducing guard intervals (GI) between the test and information signals, provided that the data rate is maintained unchanged. The influence of different types of GI on the calculation accuracy of communication channel impulse response (IR) is illustrated in comparison with the one specified in the simulation.

The author proposes to use GI as a cyclic prefix of the test sequence, and also a new method of GI compensation is considered. It is shown that GI application makes it possible to increase the calculation accuracy of channel IR and, accordingly, CF IR. The form of the GI and the DFT interval should be taken into account.

Moreover, the simulations showed that application of GI in single-tone HF data modems makes it possible to decrease BER in a fading channel maintaining the data rate unchanged. In addition, under certain conditions, it becomes possible to increase the data rate of the modem.

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10. Maslakov M.L. Pat. ¹ 2573270. RU. H04L 1/20. Method of adaptive correction with guard interval compensation / Egorov V.V., Katanovich A.A., Lobov S.A., Maslakov M.L., Mingalev A.N., Smal M.S., Timofeev A.E. 2016.

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Mismatched filter synthesis, minimizing integrated sidelobe level at the segment of time axis for phase-coded pulse

G.V. Zaytsev, e-mail:
N.S. Kondranina, e-mail:
D.M. Litvinov, e-mail:
PSC SIC "Almaz"

Keywords: phase-coded pulse, mismatched filtering, minimization of integrated sidelobe level, low correlation zone.

Sidelobe suppression of a phase-coded radar pulse at the receiver filter output is a classical problem which also arises in many other technical domains. In practical situations it is often sufficient to suppress only a part of these sidelobes. The paper presents method of mismatched filter synthesis, minimizing integrated sidelobe level at the segment of time axis for binary phase-coded pulse. The method described is a generalization of the known algorithm [3, 5]. The paper investigates filter output characteristics which may be achieved using this method for a binary modulation sequences and a zero Doppler shift.

Let s0 be modulation binary sequence, expressed by row vector, with n elements ±1, let s0 denote the sequence of length N=n+2m containing s0 a central part with m proceeding and m subsequent zeros, and let x be a real row vector of length N of the filter pulse response (reference sequence). Filter output is a crosscorrelation function of sequences x and s:

where k is the index of nonzero output sample: k∈ Ω ={-(n+m-1),...,-1,0,1,...,n+m-1}. The segment of sidelobe suppression is given by successive indexes k∈ Ψ={a,a+1,...,b},M∉ Ψ, where index M of he main lobe is excluded.

Integrated sidelobe level for the problem described is determined by expression

and calculated in decibels, where wk is a positive weight equal to unity in the segment and equal to some small value outside the segment. It is shown in the paper that optimal vector x minimizing ISL for a fixed s, is given by the formula x = sR-1, where R is a certain N×N matrix, which is a function of s.

The paper describes the following properties of filter output for this solution:
• for segments with width z < N sidelobes are suppressed practically completely;
• segment width may be expended by increasing the value N;
• for the case n=N loss in signal-to-noise ratio is over the range 0,5-1 dB for segment width z < (0,5-0,8)n and sequences s0 with high merit factor.

Average losses in signal-to-noise ratio are calculated as a function of segment width for several families of modulation sequences s0: M sequences [8], Legendre sequences [9], and sequences up to length 271 with best known integrated sidelobe level of autocorrelation function [7].

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Multirate signal processing for narrowband noise rejection
V.V. Vityazev, e-mail:
P.B. Nikishkin
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan

multi-rate, processing, rejection, noise, narrowband, frequency selection, modeling.

The classic problem of digital frequency selection of signals is rejection narrowband low-pass components. The order of FIR-filter will be increased proportion to sampling frequency to the rejection bandwidth. Rejection bandwidth is constituted a small part of the general operating frequency band.

Narrowband rejection filter is implemented method of compensation for noise with figure 1a. The two-stage implementation is proposed based on figure 1b. This system is differed using decimation and interpolation effects impulse response when implementing narrowband filter.

The method is used for multiple reduction in computing costs. The degree of rejection noise is determined accuracy of approximation desired frequency response of a narrowband filter in the pass-band.

The synthesis of transfer function narrowband filter in the band are made special requirements. Requirements in accuracy of approximation in the pass-band are weakened. Two methods of effective implementation are possible. The first method is suggested using structure of quadrature modulation. The second method is implemented using complex impulse characteristics.

The quadrature modulation is increased computational costs. The impulse response of comb and anti-aliasing filters are taken real value, when ω0=2πk/ν. The computation costs are becoming commensurate with costs on rejection filter.

The total computational efficiency of narrowband rejection filter are determined order of rejection filter in interference compensation circuits.

Narrowband rejection filter are realized alternative way with use decimation and interpolation the converted signal. The comparative analysis of characteristic are shown achieve specified properties of selectivity at substantially less computational costs.

The multistage realization of narrowband filter are reduced computing costs.

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Synthesis of the integer IIR filters with short coefficient wordlength
V.N. Bugrov, e-mail:

N.I. Lobachevsky state university of Nizhni Novgorod (NNSU) , Russia, N.Novgorod

Keywords: the integer IIR-filter, integer nonlinear programming, multifunctional synthesis, criterion function.

The main factors determining the speed of the IIR filter include calculation arithmetic and data representation in the digital filtering algorithm. At present, indirect design of recursive filters by analog prototype prevails with the use of the bilinear transformation method when using real arithmetic algorithms in digital filtering algorithms, usually in a floating-point format. The real data representation format forces them to quantize their values, which leads to distortion of frequency characteristics, the appearance of quantization noise, the need to scale the real coefficients of the filter, and the appearance of small limit cycles in the quantization of the results of internal computations. However, the possibility of direct synthesis of digital IIR filters directly in the integer state space can be provided by the methodology of integer nonlinear programming (INP). The INP ideology makes it possible to efficiently design integer filters with a given bit of data representation and maximum performance of the requirements for the set of frequency characteristics of the filter for an arbitrary form of their assignment. Currently, in the overall nomenclature of commercial digital platforms, a significant proportion is occupied by 8-bit digital platforms with integer computational arithmetic. When implementing high-speed integer digital filters on specialized platforms or on a chip, the data representation can be even lower (up to 4 or even up to 3 bits).

In this article, we present several typical examples of solving the problems of synthesizing integer IIR of minimum data representation capacity, examples illustrating the principal possibilities of this approach to the multifunctional design of digital systems. The stability of the solution for integer IIR filters is guaranteed by the priority implementation of the functional stability conditions in the process of the INP-synthesis. It is possible to set the required maximum pole radius of the transfer function in the synthesis, which allows to effectively control the quality factor of the designed filter in the event of the occurrence of limit cycles of one kind or another. During the synthesis of the cascade integer filter, the necessary scaling of the signal can be provided. There is no need to use indirect scaling techniques, for example, using Lp-norm.

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Measurement method of pitch with inter period accumulation of speech signal
V.V. Savchenko, e-mail:
The Nizhniy Novgorod State Linguistic University (NNSLU), Russia, Nizhniy Novgorod

Keywords: speech, speech signal, pitch frequency, automatic speech processing, speech technology.

In the article a new method for measuring the pitch frequency of a speech signal in conditions of increased noise is proposed. The effectiveness of the method is achieved due to the effect of interperiodic accumulation.

The specified effect is realized in a multichannel frequency measurement system using several parallel connected signal recirculator with adjustable delay periods in the feedback circuits. The effectiveness of the method has been studied theoretically and experimentally.

The accuracy and sensitivity of the method are estimated as a function of the interference intensity at the input of the measurer. It is shown that for a signal-to-interference ratio of 20 dB or more, the error of the new method does not exceed (1 ... 2) % of the nominal value of the frequency. It is agrees well with the potentially achievable accuracy under the conditions in question. In conditions of high noise the gain the threshold value of the signal-to-interference ratio in comparison with the world analogs is 4-5 dB or more.

The received results and conclusions drawn allow to recommend this method for practical application in systems of the automatic speech processing in the conditions of action of the intensive acoustic noise.

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A method of validating Analog-to-Digital Conversion requirements in Frequency-Modulated Continuous-Wave Synthetic-Aperture Radars
L.B. Ryazantsev, e-mail:
I.F. Kupryashkin, e-mail:
V.P. Likhachev, e-mail:
The Military Educational-Research Centre of Air Force "Air Force Academy named after pro-fessor N.E.Zhukovsky and Y.A.Gagarin", Russia, Voronezh

Keywords: FMCW SAR, dynamic range, ADC resolution, unmanned aerial vehicle, oversampling.

Due to their small weight, small size and relatively simple design, frequency-modulated continuous-wave synthetic-aperture radars (FMCW SARs) are currently one of the most promising types of radar systems for small unmanned aerial vehicles (UAVs).

When developing a FMCW SAR, the ADC resolution is chosen based on the dynamic range of echo signals output by the receiver. Compared with impulse systems, here the dynamic range is characterised by larger values, because its upper boundary is determined not by the total strength of a separate distance range being resolved, but by the total strength of the echo signal of the whole surface being mapped within the antenna illumination footprint. In addition, in small-size FMCW SARs, it is not possible to implement the traditional techniques of dynamic-range extension used in impulse radars (sensitivity-time control and cosecant antenna patterns), because distances are meas-ured based on the frequency principle and stringent requirements are set for the weight- and size-related characteristics of the antenna systems. As a result, the dynamic range of the signals may reach around 70–80 dB, which suggests ADC resolutions of 12–14 bits and sampling rates of up to tenths of MHz.

Studies aimed at reducing the amount of information recorded (to enable its transmission via a radio channel from the UAV to the ground station) have shown that a progressive reduction of the ADC resolution from 16 bits to 8, 4, 2 and 1 bits does not lead to a noticeable visual decrease in the quality of radar images created and in their deciphering qualities, due to the oversampling effect.

In this paper, we present a method of validating requirements for the ADC resolution in FMCW SARs (given the conditions under which the surface survey must be performed) that takes into account the oversampling effect. The results obtained show that the ADC resolution can be significantly reduced (down to binary quantization) without any loss of the quality of radar images created. This reduction in the amount of information recorded by a FMCW SAR allows us to lower, proportionately to the reduction in the ADC resolution, computing requirements when synthesizing radar images (including on the carrier) and radio-channel throughput requirements when transmit-ting recorded signals to the ground control station.


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Combined method of heterogeneous images imposition in aviation computer vision system
A.I. Novikov, e-mail:
A.A. Loginov, e-mail:
D.A. Kolchaev, e-mail:
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan

Keywords: real and synthesized images, correlation-extreme algorithm, linear prediction, combination of images, quality of combining.

A combined algorithm for imposition of a real image with an image synthesized from a digital terrain map is presented. The algorithm works in two modes. In the basic mode, the images are imposed on the basis of the predicted values of the navigation parameters according to their actual values, with the correction of the latter to the correction values. Corrections are calculated as a result of comparing the actual values of the parameters with the values obtained by the correlation-extreme algorithm. Corrections are calculated once during the transition to the auxiliary mode, in which the images are imposed using a correlation-extreme algorithm.

The prediction of the navigation parameter values is carried out for each frame of the video sequence according to the linear models in the finite memory scheme. The optimal estimates of navigational parameters by the correlation-extreme algorithm are calculated once when the difference between the forecast and actual values of at least one navigation parameter exceeds the specified error.

The comparative tests of the algorithm in comparison with known algorithms of the same type that confirmed the correspondence of the qualitative parameters of the proposed algorithm to the requirements of the corresponding documents were carried out.


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Estimation of direction-of-arrival by means of maximum likelihood artificial neural networks
T.Ya. Shevgunov., e-mail:
E.N. Efimov., e-mail:
D.V. Filimonova, e-mail:
D.I. Voskresenskiy, e-mail:
Moscow Aviation Institute (National Research University), Russia, Moscow

Keywords: artificial neural networks, maximum likelihood, parametric estimation, directon-of-arrival method, circular antenna array.

The use of feedforward artificial neural network (ANN) resembling multilayer perceptron architecture is proposed for synthesis of the direction-of-arrival (DOA) maximum likelihood estimator. Two learning approaches are considered for building up ANN which outputs are able to approximate the vector of parameters originally obtained with estimator based on maximum likelihood concept. The first one is conventional minimization of the squared errors between ANN output and solution found via numerical optimization technique. The second approach is based on the maximization of the target function evaluated by substituting the measured data and the parameter vector estimated by the ANN. A brief theoretical foundation of the proposed approach is presented in the assumption that the model of the antenna system used for the observation is known as well as the values of its vector of parameters.

As a practical example, the implementation of direction-of-arrival (DOA) estimator in a passive radar system assembled for position location equipped with active ring array antenna is considered. Modified ANN topology based on multilayer perceptron architecture is used to obtain DOA estimator directly from the output of the network. The results of numerical simulation carried out for the wide range of angles are compared to the optimal numerical solution and the Cramer-Rao Lower Bound. The results indicate that there is no significant dependency of the accuracy of estimator on true parameter value and the standard deviation of estimator increases no more than 10 percent while the consumed computational time decreases no less than 12 times.


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