2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)最新文献

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Analysis of a multi-fiber emitter field for free space optics transmitters 自由空间光发射机多光纤发射场分析
V. I. Voronov, Fanil N. Nuramov, Amdjed A. Alquliah
{"title":"Analysis of a multi-fiber emitter field for free space optics transmitters","authors":"V. I. Voronov, Fanil N. Nuramov, Amdjed A. Alquliah","doi":"10.1109/SINKHROINFO.2017.7997512","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997512","url":null,"abstract":"This report is devoted to the analysis of field characteristics in the far zone of a multi-fiber laser emitter for free space optical communication line, containing in each optical fiber phase modulators providing the required amplitude-phase distribution.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"34 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116988098","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
Adaptation of Psychoacoustic analysis to wavelet domain in lossy audio coding 有损音频编码中心理声学分析对小波域的适应
G. Rogozinsky, Daniel R. Fadeyev, D. Podolsky
{"title":"Adaptation of Psychoacoustic analysis to wavelet domain in lossy audio coding","authors":"G. Rogozinsky, Daniel R. Fadeyev, D. Podolsky","doi":"10.1109/SINKHROINFO.2017.7997551","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997551","url":null,"abstract":"The paper presents the authors' current research in adaptation of the MPEG AAC Psychoacoustic Model to the Discrete Wavelet Packet coding algorithm. We discovered the complex distribution of wavelet quantization noise across the audio spectrum due to significant cross-band aliasing. We also obtained the wavelet quantization noise function related to the analysis bands of MPEG AAC Psychoacoustic Model. The described approach provided the quantization matrix, which can be used for exact estimation of overall quantization noise during lossy coding.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"6 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121756566","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Adaptive blind equalizer for HF channels 高频信道自适应盲均衡器
N. Miroshnikova
{"title":"Adaptive blind equalizer for HF channels","authors":"N. Miroshnikova","doi":"10.1109/SINKHROINFO.2017.7997541","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997541","url":null,"abstract":"In this paper, the problem of blind adaptive equalization of HF channel is considered using a state space approach. HF channel is multipath propagation channel due to multiple reflections on ionospheric layers, which causing intersymbol interference (ISI) at the receiver output. The problem of ISI suppression is referred to as deconvolution or channel equalization. At present, a family of methods for equalizing and estimating the channel based on the criterion of minimum mean-square error (MMSE) has become widespread in engineering practice. These methods assume the use of a training or pilot symbols known by the receiver to estimate the channel and “train” the equalizer. However, in order to increase the bandwidth efficiency blind deconvolution or channel equalization methods have been developing recently more actively. The essence of these methods is the task of equalization and estimation of the channel from sensors outputs without any a priory knowledge of the original signals. The classical criterion for blind equalization is the constant modulus (CM) criterion, which is an extension of the Godart algorithms family. However, this method has slow convergence, and is not applicable to all modulation methods used in the HF systems. The paper considers a learning algorithm based on information theory and natural gradient is used to solve the optimization problem. To effectively use the algorithm in a changing channel environment, it is suggested to use an additional optimization of the algorithm step size. The channel model is state-space description of a dynamic system. The main advantage of state-space model is that is flexible and can be used for internal description of system. Based on developed state-space model and measurement models, an adaptive optimum-size blind equalization algorithm is proposed to track the HF channel variation in time. Proposed algorithm is compared to CMA and classical stochastic gradient descent algorithms for blind deconvolution. In numerical simulations, it is observed that the proposed approach can track the channel variations with good performance. During computer simulations under good, moderate and poor HF ionospheric channel conditions, it is observed, that proposed adaptive equalization algorithm with adaptive step-size for blind deconvolution provides reliable equalization error and can track the variation of the channel in time with high accuracy","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"86 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116734376","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 2
Measurement of radio signal energy parameters 测量无线电信号能量参数
M. Zhodzishsky, V. Veitsel, R. Kurynin
{"title":"Measurement of radio signal energy parameters","authors":"M. Zhodzishsky, V. Veitsel, R. Kurynin","doi":"10.1109/SINKHROINFO.2017.7997523","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997523","url":null,"abstract":"The present paper discusses different techniques of measuring Carrier-to-Noise density ratio (C/N0) in navigation and information-communication systems. C/N0 means the ratio of the received signal power to the noise spectral density N0 at receiver's input (at the output of an antenna with a low-noise amplifier (LNA)). Correlation processing is performed in the receiver to obtain in-phase (I) and quadra-phase (Q) components of the input signal. Three estimation algorithms of C/N0 [Hz] have been analyzed in the paper. The first algorithm is robust: it requires minimal a priori information about signals, noise and receiver configuration. This algorithm is based on estimation in real time both expected value and the variance of in-phase component I. The second algorithm is called I-noise-immune, since it provides much more accurate estimates for C/N0, than the robust one but it needs a priori data about variance of component I. This requirement can be satisfied if value N0 is mainly determined by internal receiver noise (more precisely, - LNA noise), but not by received external interference. Both algorithms are applicable when Phase locked loop (PLL) operates quite precisely in the receiver. The third algorithm called Z-noise-immune, unlike the two first algorithms, does not need precise operation of PLL or even PLL availability in the receiver at all, but similar to the second algorithm is based on the assumption that value N0 is determined by internal noise rather than external interference. There were shown the third algorithm produces unbiased estimate of C/N0.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"56 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"116910477","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 1
Comparison second order based blind signal separation with classical adaptive interference cancellation methods in the case of ill-conditioned statistics 比较了二阶盲信号分离与经典自适应干扰消除方法在病态统计情况下的区别
S. S. Adjemov, A. A. Kuchumov, N. Y. Liberovskiy, V. Priputin
{"title":"Comparison second order based blind signal separation with classical adaptive interference cancellation methods in the case of ill-conditioned statistics","authors":"S. S. Adjemov, A. A. Kuchumov, N. Y. Liberovskiy, V. Priputin","doi":"10.1109/SINKHROINFO.2017.7997494","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997494","url":null,"abstract":"In the last years blind source separation methods increasingly frequently use in digital signal processing. Their advantage is that we haven't to know any additional information about the source signals. The BSS method uses two fundamental presumptions. The first one is that the observation signals are linearly dependent on source signals. The second presumption is that the source signals must be independent from each other. The possibility of source separation using just observe signals let to decrease systematical error which correlate with the wrong data of antenna array. The purpose of this paper is comparison the BSS method with another one and efficiency of modification the BSS method with the Tikhonov regularization. The MVDR and Timegate methods were chosen for the comparison with BSS method. The experiment was run in the Matlab. Two sinusoidal mutually spaced signals fall into uniform linear array. The maximum signal-noise ratio was chosen as the criterion. The experiment shows that BSS method better separate the signals that the other ones. In the second part of the paper BSS method was analyzed in the case of ill-conditioned statistics. This situation is possible when the number of antenna elements is larger than the number of source signals. An experiment was run in the Matlab where the rate of off-diagonal elements of the statistics was calculated after the diagonalization. The experiment shows that the Tikhonov regularization essentially decreases the summar off-diagonal elements rate and improves source separation in case of ill-conditioned statistics.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"1 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"121546547","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 3
Applying of the optimal combining algorithm in a high-frequency communicatiom system 最优组合算法在高频通信系统中的应用
V. Magsumov, V. Priputin
{"title":"Applying of the optimal combining algorithm in a high-frequency communicatiom system","authors":"V. Magsumov, V. Priputin","doi":"10.1109/SINKHROINFO.2017.7997538","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997538","url":null,"abstract":"The paper is devoted to the implementation on the FPGA of the algorithm for the optimal combining of signals. Described the algorithm, the architecture of the optimal combining block, implemented to a four-element antenna system. The signal-to-noise ratio chosen as a criterion reflecting the efficiency of interference suppression in the reception band (increase of noise immunity). The effectiveness of the proposed approach was demonstrated experimentally on high-frequency signals.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"60 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"126580000","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
Modeling of efficiency UHF class-D power amplifier with bandpass sigma-delta modulation 带通σ - δ调制的高效UHF d类功率放大器建模
O. Varlamov, I. Chugunov
{"title":"Modeling of efficiency UHF class-D power amplifier with bandpass sigma-delta modulation","authors":"O. Varlamov, I. Chugunov","doi":"10.1109/SINKHROINFO.2017.7997508","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997508","url":null,"abstract":"Sigma-delta modulation is a method of constructing the radio frequency linear power amplifier with the using non-linear components. It can significantly simplify the structure of the power amplification path, which in this case shall operate as one-bit DACs. The preliminary modeling of potential different variants of the output stage in classes E, F, D (voltage switching and current switching) and the pulsed gate modulation was done. It is shown that the required linearity of amplitude characteristic can be realized only in the class D amplifier with voltage switching. Such an amplifier can be constructed in a bridge or half-bridge circuits.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"40 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"134218468","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 16
Development of an intra-group command-information interaction system using cluster approach 利用集群方法开发组内命令-信息交互系统
A. Krenev, L. Kazakov, A. Tsarev
{"title":"Development of an intra-group command-information interaction system using cluster approach","authors":"A. Krenev, L. Kazakov, A. Tsarev","doi":"10.1109/SINKHROINFO.2017.7997534","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997534","url":null,"abstract":"The paper presents an approach to development of an intra-group interaction systems using clusters. It is shown that fully connected networks are an effective way to communication reliability enhancement. This effect is achieved by means of simultaneous use of frequency, time and special division. Simulation results show a good convergence to the maximum achievable performance.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"70 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"133721885","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 0
Centralized control of traffic flows in wireless LANs based on the SDN concept 基于SDN概念的无线局域网流量集中控制
Aleksey Amelyanovich, Mikhail Shpakov, A. Muthanna, M. Buinevich, A. Vladyko
{"title":"Centralized control of traffic flows in wireless LANs based on the SDN concept","authors":"Aleksey Amelyanovich, Mikhail Shpakov, A. Muthanna, M. Buinevich, A. Vladyko","doi":"10.1109/SINKHROINFO.2017.7997495","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997495","url":null,"abstract":"The paper proposes to use a solution based on software-defined networks (SDN) to control traffic flows in wireless local area networks (Wi-Fi). The paper examines the testing methods for a software-configurable wireless local area network based on the model network of the St. Petersburg State University of Telecommunications. Classify traffic based on DSCP values in the IP-headers of packets. After classification, traffic will be given priority over the OpenFlow port of the switch. This method allows for end-to-end quality support and support for traffic classification when interworking. In the course of practical experiment, the interaction of access points (AP), which are emulated in Mininet with the SDN controller, is considered. When using the OpenDaylight controller, the network topology can be controlled through the user-friendly graphical interface (GUI) of the SDN. As a result, methods were examined using DSCP values for QoS management in SDN networks.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"21 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"115249256","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 5
Algorithm elaboration of iterative processing of M-CPFSK signals in the ionospheric channel with selective interferences 选择性干扰下电离层信道M-CPFSK信号迭代处理的算法阐述
M. M. Tolkachev, E. M. Lobov, N. A. Kandaurov
{"title":"Algorithm elaboration of iterative processing of M-CPFSK signals in the ionospheric channel with selective interferences","authors":"M. M. Tolkachev, E. M. Lobov, N. A. Kandaurov","doi":"10.1109/SINKHROINFO.2017.7997564","DOIUrl":"https://doi.org/10.1109/SINKHROINFO.2017.7997564","url":null,"abstract":"In the report signal and code constructions (SCC) on the basis of continuous phase frequency shift key (CPFSK-64, CPFSK-256, CPFSK-1024) using low-speed turbo codes are provided. In the course of simulation program models of the modulator, demodulator, coder and decoder on a high-level language C++ were developed for application in system of simulation modeling of the digital communication systems Spectr-2. Codes were selected with the following code relations 5/16, 6/32, 7/64. The modulator and the demodulator has the user window from which it is supposed to select a type of modulation, modulation index and code rate. Sequences of synchronization was used during transmission of an information block in model of the system M-FSK of signal`s synchronization. By transmission of a sequence of synchronization, it is supposed to transfer the sectional M-sequences. When carrying out the analysis it was supposed to research SCC in Watterson's channel with additive white Gaussian noise, and in the presence of selective interferences. Built-in models of the generator of white Gaussian noise, model of the channel of a Waterson and measuring units for data output were used in the diagram of the program complex Spectr-2. In this paper, the signal occupied bandwidth of the channel 40000 kHz. Selective interferences made 25 percent from bandwidth of all signal. Processing of SKK includes an algorithm of normalization of output responses of filters for reduction of influence of selective interferences in case of detection and restoration of a signal. With growth of number of frequencies in CPFSK, there was a difficulty of processing of a signal in a receiving part. For the solution of this problem of using frequency shift-keying 256 is change of the reference generator in the demodulator, which makes calculation of values of a complex signal. It allowed to reduce the number of the operations calculated for 1 clock period of the system Spectr-2. However, for modulation of CPFSK-1024 it was required to completely, change digital processing of the accepted signal. The solution was found in application digital the filter of bank. As a prototype the filter with band pass range of 150 Hz and a band of a barrage of 300 Hz with suppression 70 dB was selected. The order of the filter prototype is 32768. Characteristics of bit error rate and frame error rate in different conditions were received because of simulation.","PeriodicalId":372303,"journal":{"name":"2017 Systems of Signal Synchronization, Generating and Processing in Telecommunications (SINKHROINFO)","volume":"43 1","pages":"0"},"PeriodicalIF":0.0,"publicationDate":"2017-07-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":null,"resultStr":null,"platform":"Semanticscholar","paperid":"128417524","PeriodicalName":null,"FirstCategoryId":null,"ListUrlMain":null,"RegionNum":0,"RegionCategory":"","ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":"","EPubDate":null,"PubModel":null,"JCR":null,"JCRName":null,"Score":null,"Total":0}
引用次数: 3
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