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“Digital
Signal Processing” No. 4-2024 |
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Analysis of the channel capacity of the MIMO information transmission system in a turbulent environment with mutual coupling
M.V. Grachev, e-mail: grachev.m.v@rsreu.ru
Yu. N. Parshin, e-mail: parshin.y.n@rsreu.ru
Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: MIMO communication system; inhomogeneous continuous medium; matrix of channel coefficients.
Abstract
To develop algorithms for spatial encoding and decoding of signals it is necessary to take into account the properties of the propagation medium. For this purpose information about the properties and values of the signal transmission coefficients from each transmitting antenna to each receiving antenna is used. The features of electromagnetic wave propagation are significant attenuation of the signal in the medium and multipath caused by signal reflections from local inhomogeneities of the medium, which causes a difference in propagation conditions for each pair of transmitting and receiving antennas.
A model of an inhomogeneous continuous medium, which is a layered structure with varying dielectric permittivity, is considered. The matrix of channel coefficients of the MIMO system is modeled in a quasi-continuously layered inhomogeneous continuous medium with turbulence. Electromagnetic waves hitting the aperture of the receiving antenna are determined on the basis of the beam theory. In continuous media with small electrical dimensions of the antenna system and close proximity of the elements, the effect of mutual coupling is observed, which changes the field distribution and radiation characteristics of the antennas. The mutual coupling of the antenna system’s elements in the transmitter and receiver affects the channel matrix together with the characteristics of wave propagation in inhomogeneous continuous media. The channel capacity of a MIMO system is calculated with uniform distribution of signal power, optimal power distribution by water filling and orthogonal spatial coding. The influence of the properties of a continuous medium on the channel capacity of the considered MIMO systems is analyzed. The presence of inhomogeneity and turbulence of the continuous medium significantly reduces the received signal power.
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Effectiveness evaluation of the signal processing algorithm at the background of low-frequency noise based on experimental data
A.Yu. Parshin, e-mail: parshin.a.y@rsreu.ru
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: low-frequency noise, flicker noise, «popcorn» noise, non-Gaussian model, optimal processing, signal-to-noise ratio.
Abstract
The widespread use of communication systems with low capacity leads to the development of methods for organizing communication channels using ultra-narrowband signals. A special feature of the signal processing is the significant influence of low-frequency noise. One of the most dangerous are the noises of active devices such as flicker noise and "popcorn" noise, which have a non-Gaussian properties. It is necessary to develop algorithms for processing ultra-narrow-band signals at the background of low-frequency noises, taking into account the non-Gaussian properties.
This article is devoted to adaptive signal processing at the background of low-frequency noises of various nature. A mixture of thermal, flicker noise and "popcorn" noise is considered as an interference. It is required to analyze the effectiveness of adaptive algorithms for processing ultra-narrowband signals at the background of low-frequency noise. The features of the application of various low-frequency noise models are considered: Leeson model, Du Pre-van der Zyl model, a model based on a system of differential equations. The research uses a model based on a system of differential equations and describes a method for estimating the variance, kurtosis coefficient and effective width of the flicker noise spectrum from the observed mixture of low-frequency and thermal noise.
The adaptation of the model parameters is performed both by modeling low-frequency noise in the form of a Gaussian value and using measured data from a flicker noise generator in the form of an operational amplifier. This approach makes it possible to adapt the signal processing algorithm when evaluating noise parameters in the absence of a useful signal. The application of the algorithm will reduce the influence of low-frequency noise of active devices operating both in transmitter and receiver. In order to evaluate the efficiency of the processing algorithm in real conditions, a sequence of samples of the noise process was experimentally obtained from a prototype of a low-frequency noise generator on an operational amplifier developed by the author.
The results of modeling the processing algorithms show the overall advantage of the estimation-correlation-compensation algorithm. The simulation is performed for different signal durations, which allows us to take into account the influence of low-frequency interference over a larger frequency range. Thus, over long durations of the signal in question, the width of its spectrum decreases, which increases the negative effect of interference. A check of the algorithm's operability on experimentally obtained implementations of a mixture of flicker noise and popcorn noise confirmed the effectiveness of using a low-frequency noise model based on a system of differential equations, as well as an algorithm for estimated correlation and compensation processing.
References
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Implementation of fast Fourier transform of non-equidistant pulse sequences on FPGA
V.I. Koshelev, e-mail: koshelev.v.i@rsreu.ru
N.H. Trinh., e-mail: ngochieu.radioscientist@mail.ru
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: multi-channel frequency filter, non-equidistant pulse sequence, fast Fourier transform, FPGA, Al-tera Cyclone IV.
Abstract
The Fast Fourier Transform (FFT) processor is widely used to perform a variety of radio engineering tasks, in particular multi-channel frequency filtering (MFF) of signals with an unknown frequency. Multichannel frequency filtering is widely used in radio engineering systems to measure the Doppler effect using a coherent pulse train.
Using a pulse train with a constant pulse repetition rate causes blind velocities and blind ranges. To avoid these unexpected effects, a non-equidistant pulse sequence (NPS) is used. However, the use of the classical FFT for NPS processing does not ensure optimal processing according to the energy criterion of the signal-to-noise ratio. The authors proposed a modified fast Fourier transform algorithm to improve the signal-to-noise ratio at the output of the FFT.
This study is devoted to the implementation of the modified fast Fourier transform (MFFT) algorithm on field programmable gate arrays (FPGA) that provide the maximum signal-to-noise ratio when processing NPI. This algorithm is based on changing the FFT structure by introducing additional phase rotation blocks that take into account the wobble of the NPI pulse repetition period. The architecture of the MFFT algorithm is designed to minimize the use of memory and built-in multipliers. In this regard, the only radix-2 calculation module is used in the calculation process. The MFFT algorithm is described in Verilog HDL and tested on the Altera Cyclone IV FPGA board.
The aim of the work is to implement the FFT and MFFT algorithms in FPGAs for processing non-equidistant pulse sequences. A comparative analysis of the obtained experimental results of calculating the FFT and MFFT with the results of computer calculations is carried out.
References
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14. Koshelev V. I., Trinh N.H. Optimization of the weight processing algorithm in multichannel Doppler filtering // Bulletin of the Higher Educational Institutions of Russia. Radio Electronics. 2024. Vol. 27. no. 2. pp. 93–104. https://doi.org/10.32603/1993-8985-2024-27-2-93-104.
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16. Koshelev V.I., Trinh N.H. Algorithm for the fast Fourier transform of nonequidistant pulse sequences // Bulletin of the Russian State Radiotechnical University. 2023. no. 85. pp. 3–13.
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Investigation of time-frequency signal processing in a system of Doppler
receivers
V.K. Klochko, e-mail: klochkovk@mail.ru
B.H. Vu, e-mail: ronando2441996@gmail.com
Keywords: signal sources, time-frequency processing, receiver positioning system, signal detection, angular coordinate estimates.
Abstract
The potential possibility of increasing the resolution of reflection signals from several moving sources of a different physical nature by signal processing in a semi-active receiver system is being investigated. The criterion for increasing the resolution of signals is the probability of detecting all signals. For mobile sources, the difference in signals is manifested both in signal delay time and frequency. The task is to detect signals by determining their number and parameter estimates, which make it possible to find the parameters of the sources themselves during the observation period.
As a prototype, a method for processing signals in the frequency domain is taken, based on the allocation of spectral components in the frequency spectrum of several spatial channels, measuring the phases of spectral components and estimating angular coordinates by the phase difference method.
To investigate the possibility of achieving better results than in the frequency domain, a method for processing signals in the time-frequency domain has been developed. For the received mixture of signals, a sequence of their time counts is formed, which is subjected to filtering operations with the detection of the time points of the appearance of each signal and the estimation of frequency and phase, and the next signal is allocated by subtracting from the smoothed mixture of signals extrapolated estimates of the signals detected earlier. Processing in the time and frequency domains is carried out in parallel, the processing results are combined, and estimates of the number of signals and their parameters are given.
To investigate the potential possibility of obtaining even better results, a method for signal proc essing in a receiver system consisting of a transceiver station and two auxiliary receivers has been developed. In this case, the ranges to each source and the coordinates of the velocity vector of each of them are estimated during one observation period.
Based on the simulation results, it is shown that due to time-frequency processing in the receiver system, it is potentially possible to increase the probability of detecting all signals from 0.87 to 0.96 (based on the results of modeling three signals) in comparison with time-frequency processing in one transceiver station, as well as to determine the parameters and dynamic properties of the sources.
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6. Klochko V. K., Vu B. H. Chastotno-vremennaya obrabotka signalov v doplerovskom radiopriemnike // Cifrovaya obrabotka signalov. 2023. no. 2. pp. 15 - 21.
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The approach to correction of autonomous navigation system
V.K. Klochko, e-mail: klochkovk@mail.ru
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: autonomous navigation, aircraft, reference and current images, reference points, airborne radar, optical station.
Abstract
An approach to correction of an autonomous navigation system when accompanied by a moving object is proposed. The approach is based on the classification of singular points of the reference and current image of the terrain to correct the movement of the object and differs in the presence of a classification criterion for the minimum square of the norm of the error vector of conjugation pairs of singular points. The aim of the work is to develop a method for analyzing reference points obtained on the basis of optical and radar images of the terrain in an autonomous navigation system.
For comparison, an autonomous navigation method based on comparing current and reference terrain images and calculating corrections for horizontal object displacements in the presence of altitude measurements is considered. The disadvantages of this method are noted: calculating the correlation function based on terrain images requires large computational costs; the spatial orientation of the object is not taken into account; there is a lack of information about the direction of movement of the object based on the results of its position correction; In the case of radar use, the station's operation is carried out continuously in the active mode of radiation of the probing signal and is subject to external radio interference.
The approach is proposed to eliminate these disadvantages. The proposed approach for correcting the movement of an object allows using fast matrix operations for solving linear algebraic equations instead of the correlation function, taking into account the spatial orientation of the object, determining the vector of the direction of movement of the object to an even position at current moments along an extrapolated trajectory and combining the operation of an onboard radar with the operation of an optical station.
The approach can be used in navigation systems for tracking moving objects of different physical nature.
References
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4. Bakulev P.A., Sosnovsky A.A. Radio navigation systems. Textbook for universities. Moscow: Radio Engineering, 2005. 224 p.
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7. Methods of automatic detection and tracking of objects. Image processing and control / B.A. Alpatov, P.V. Babayan, O.E. Balashov, A.I. Stepashkin. Moscow: Radiotechnika, 2008. 176 p.
8. Zhou S.B., Li Y. The SIFT image feature matching based on the Plural Differential // Advanced Materials Research (Volumes 268 – 270). 2011 pp. 2172 - 2184.
9. Bastanlar Y., Temizel A., Yardimci Y. Improved SIFT Matching for Image Pairs with a Scale Difference // IET Electronics. – Volume 46, Issue 5, 2010. P. 346 - 352.
10. Wu Z., Wang J., Li C., Yan Y., Chu C. Bidirectional Matching Algorithm for Target Tracking Based on SIFT //Advanced Research on Computer Education, Simulation and Modeling Communications in Computer and Information Science. Volume 175, 2011. P. 253 - 258.
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FMCW MIMO radar design based on smoothed fast phase coded shift keying
Y.M. Meleshin, e-mail: kykymberr@gmail.com
M.S. Khasanov, e-mail: mkhaes@gmail.com
V.N. Karpov, e-mail: vadimkarpov@icloud.com
National Research University of Electronic Technology (MIET), Moscow, Russia
Keywords: FMCW, MIMO Radar, phase shift keying, Radar, DAC.
Abstract
The article is devoted to MIMO radars based on linear frequency modulated signals (FMCW). It is known that classical FMCW is not suitable for use in MIMO, and requires additional actions to form N different orthogonal signals. The article examines various methods for ensuring this orthogonality.
The options for separating signals by means of time-division multiplexing (TDM) and frequency-division multiplexing (FDM) are considered, by means of which it was established that the most promising way to separate signals is the use of chirp signals with additional phase-code manipulation in accordance with orthogonal binary pseudo-random sequences. In addition, two main methods are compared that allow achieving this: the fast manipulation method and the slow manipulation method.
This article presents studies of FMCW MIMO radars with fast phase-code manipulation, since this approach is quite new, and research in this area is relevant. The results of the presented studies show the main limitation in terms of high side lobe level in the frequency domain for medium and long range radar operation.
Based on the results of the study, a new approach to the implementation of phase manipulation smoothing was proposed, which made it possible to implement manipulation smoothing without hardware complications of the radar.
References
1. Meleshin Y.M., Romanova E.O., Lyalin K.S., Khasanov M.S., Dovgal T.A. BPSK-based MIMO Radar Energy Efficiency Analysis // 2021 IEEE Conference of Russian Young Researchers in Electrical and Electronic Engineering (ElConRus), St. Petersburg, Moscow, Russia, pp. 2538-2540, 2021.
2. Pereverzev A.L., Lyalin K.S., Meleshin Y.M., Karpov V.N., Kolesnikov R.N. Development of a MIMO radar based on FKM signals, (in Russian), Nanoindustry, vol. 15, no. 113, pp. 54-58, 2022.
3. Sun S., Petropulu A.P., Poor H.V. MIMO radar for advanced driver-assistance systems and autonomous driving: advantages and challenges, IEEE Signal Processing Magazine, vol. 37, no. 4, pp. 98- 117, July 2020.
4. Sit Y.L., Li G., Manchala S., Afrasiabi H., Sturm C., Lubbert U. BPSK-based MIMO FMCW automotive-radar concept for 3D position measurement, 2018 15th European Radar Conference (EuRAD), Madrid, Spain, pp. 289-292, 2018.
5. Patrick D., Huang Y., Brennan P.V. FMCW based MIMO imaging radar, ARMMS, 2014.
6. Pan M., Chen B. MIMO high frequency surface wave radar using sparse frequency FMCW signals, International Journal of Antennas and Propagation, pp. 1-16, 2017.
7. Hinz J., Zolzer U. A MIMO FMCW radar approach to hfswr, Advances in Radio Science, vol. 9, pp. 159-163, 2011.
8. Kumbul U., Petrov N., Vaucher C. S., Yarovoy A. Phase-coded FMCW for coherent MIMO radar, IEEE Transactions on Microwave Theory and Techniques, vol. 71, no. 6, pp. 2721-2733, June 2023.
9. Khasanov M.S., Meleshin Y.M., Karpov V.N. Investigation into FMCW MIMO Radar Design Based on Fast Phase Coded Waveforms, 2024 26th International Conference on Digital Signal Processing and its Applications (DSPA), Moscow, Russian Federation, pp. 1-4, 2024.
10. Jeon S. –Y. et al. W-band FMCW MIMO radar system for highresolution multimode imaging with time- and frequency-division multiplexing, IEEE Transactions on Geoscience and Remote Sensing, vol. 58, no. 7, pp. 5042-5057, July 2020.
A method for creating artificial reverberation based on the use of dynamic convolution
Sergey V. Perelygin, e-mail: sergey.perelygin@gmail.com
Sergey L. Kiselev, e-mail: serg1135@yandex.ru
St Petersburg State University of Film and TV, St Petersburg, Russia
Keywords: artificial reverberation, algorithmic (modelling) and convolutional reverb, impulse response, dynamic convolution, Fourier transform, modulation, sound reflection.
Abstract
Artificial reverberation makes it possible to give the sound a diffuse color, which is inherent in acoustic processes in any enclosed space (room), and to form a subjective feeling of direct presence in this space for the listener. The operation of algorithmic, or modeling, reverberators is based on a model that takes into account the propagation of a sound wave in space and its reflection from obstacles (multi-tap delay line, or FIR-filter). Calculating an absolutely accurate reverberation response with the ability to move from point to point in space is still a computationally expensive operation.
Another type of reverberator – convolutional – forms a reverberation response based on the convolution of the input signal and a pre-measured (recorded) impulse response of the room. The use of impulse responses of real rooms as FIR-filter coefficients in a convolution reverberator is its advantage, but the output of such a device will be a time-invariant stationary reverberation. The natural sound of reverberation in a room is caused by the influence of factors such as the movement of air masses, rapid changes in their temperature, humidity, etc. These factors continuously change the impulse response of the room. In this case, the reverberation model of the room will be considered as non-stationary, and the convolution of the input signal with the time-varying impulse response of the room will thus be dynamic.
The difficulty of simulating the change in impulse response over time is that, in essence, it is necessary to implement modulation that will be applied within such limits and according to such laws that it will not be perceived as modulation, but will only add naturalness to the sound of the signal processed by the reverberator. The authors of the article propose an algorithm for creating such a modulation by continuously processing not the pulse characteristic of the filter, but the processed signal itself. In this case, during the processing, the signal is modulated by amplitude and frequency. These modulations, within insignificant limits set by the user, change the instantaneous amplitude and instantaneous frequency of the signal, respectively. Next, the convolution of the modulated signal with the stationary (recorded) impulse response of the room is made.
The method developed by the authors was implemented as a program in the Pure Data graphical programming environment and can be used to produce artificial reverberation with realistic sound, as well as to further improve dynamic convolution algorithms.
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A review of modifications of Cole's importance sampling method applied to LDPC
A.R. Nektov, e-mail: nektov.a.r@labsphera.ru
A.A. Ovinnikov, e-mail: ovinnikov.a.a@tor.rsreu.ru
Sphera Lab Limited Liability Company
The Ryazan State Radio Engineering University (RSREU), Russia, Ryazan
Keywords: LDPC, trapping sets, importance sampling, Cole’s method.
Abstract
The paper describes the main parameters of LDPC, a decoder based on a belief propagation algorithm, as well as a classification of sets of pseudo-codewords that have a significant impact on the performance of these codes in the range of large values of SNR. Further description in the paper is given specifically for a trapping sets.
The Cole’s method is described, based on the use of importance sampling, which is used to improve the convergence of the modeling process for estimating the noise immunity of LDPC. This method consists of three steps:
1) Searching for trapping sets and low-weight code words. to do this, part of the known input data of the decoder is distorted in a certain way;
2) Assessing the degree of influence of the found elements on the calculation of the characteristics of the code by determining the amount of noise at which an error occurs, determined by the action of a particular element;
3) The use of importance sampling to estimate the frequency of packet and bit errors, the idea of which is based on the fact that some values of a random variable in the modeling process are of great importance for the estimated function.
At the end of the paper, inaccuracies in the Cole’s method that require clarification or additional research are considered, and various modifications of this method for evaluating the performance of LDPC are presented, which are aimed at improving the accuracy of error probability estimation and reducing computational costs compared to the original Cole’s method.
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2. Bashirreza Karimi, Amir.H Banihashemi. Construction of QC-LDPC Codes with Low Error Floor by Efficient Systematic Search and Elimination of Trapping Sets. Available at: arxiv.org/abs/1902.07332v3. (accessed: 25.10.2024)
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4. Usatuk V.S., Egorov S.I. Construction of LDPC codes using a modified Cole significance sampling method // Proceedings of the Southwestern State University, 2023, vol. 27, no. 1, pp. 92–110.
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Analysis of neural network algorithms for segmentation
of water areas in SAR images
A.V. Sennikov, e-mail: alexeysennikov76@yandex.ru
R.V. Larionov, e-mail: rv.larionov@yandex.ru
V.V. Khryashchev, e-mail: v.khryashchev@uniyar.ac.ru
A.L. Priorov, e-mail: pri@uniyar.ac.ru
P.G. Demidov Yaroslavl State University (YARSU), Yaroslavl, Russia
Keywords: satellite SAR images, segmentation, deep learning, neural networks, speckle noise, water areas.
Abstract
The article proposes an algorithm for water area segmentation using satellite SAR images. The study used a set of 27 satellite images with a spatial size of approximately 200 by 300 kilo-meters with a resolution of 10 meters per pixel. Three models with the ResNet 34+U-Net, Seg-Former_b5 and SegNeXt_l architectures are used as neural network models. All models were trained using three policies and its combination: random batch, balanced batch, augmentation in-variance. The corresponding algorithm accepts patches with 2 channels - VV and VH - as input and produces a binary segmentation mask at the output. To evaluate the performance of the mod-els, such metrics as Dice, F measure, accuracy and recall were used. The highest value of Dice was equal 0.90 using balanced batch and augmentation invariance. However, all models have difficul-ties in accurate segmentation of images at the boundaries of water surfaces, which leads to a large number of false positives.
Also, within the framework of this study, an assessment of the impact of speckle noise on the quality of the neural network model was carried out, which showed that even with a noticeable increase in noise, measured by the PSNR metric, dropping to values of 9.65-9.86 dB, the model does not lose accuracy. Both for the original set and for the noisy one, the value of the Dice metric remains within 0.96-0.97, the F1 metric – within 0.81-0.82 and the Recall metric – within 0.97-0.98.
The proposed algorithm can be used in water segmentation tasks.
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