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Digital Equalization for Cancellation of Noise-Like Interferences in Adaptive Spatial Filtering

CIRCUITS SYSTEMS AND SIGNAL PROCESSING 2017年第2期36卷 675-702页

作者： Mandal, Amritakar Mishra, Rajesh Gautam Buddha Univ Sch Informat & Commun Technol Greater Noida 201308 Uttar Pradesh India

Adaptive equalizer is an indispensable component of many radar systems for the cancellation of interference which arises due to mismatches in radar channels. This paper presents a reformulated lms algorithm-based pipelined architecture of adaptive transversal equalizer that uses CORDIC as a main processing element to update angles instead of filter coefficients. In this architecture, hardware complexity has been reduced to a larger extent by utilizing pipeline method throughout the architecture. The lacuna of real-time system remains in the speed at which multiplication can be executed. We have proposed a modified version of pipelined Booth multiplier which uses partial product reduction scheme using 4-2 adder to reduce the computational time significantly. The hardware realization of the design reveals that critical path delay, area utilization and power consumption are reasonably low and therefore proposed design can be suitably be accommodated in modern-day radar system. The efficacy of the proposed interference canceller has been justified with extensive MATLAB simulations as well as hardware synthesis result for mitigation of interference in multi-channel radar systems.

关键词： Interference canceller lms algorithm CORDIC algorithm Transversal equalizer Spatial filtering Radar

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An optimum algorithm for adaptive filtering on acoustic echo cancellation using TMS320C6713 DSP

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DIGITAL SIGNAL PROCESSING 2010年第1期20卷 133-148页

作者： Ozbay, Yueksel Kavsaoglu, Ahmet Resit Selcuk Univ Dept Elect & Elect Engn TR-42075 Konya Turkey Sinop Univ Ayancik Vocat Sch TR-57400 Ayancik Sinop Turkey

In this Study, it is aimed to enhance the intelligibility of speech by canceling out the echo noise. For this purpose, the data transfer software, which is necessary for real time processing of voice signals, and the adaptive filtering algorithm software for the application of acoustic echo cancellation have been developed. An algorithm has been proposed for the determination of optimum adaptation rate (mu) for the least-mean-square (lms) adaptation algorithm that is used in the adaptive filter. The effectiveness of our optimum mu value determination algorithm was demonstrated on a single direction voice conference application with one speaker. In this study, we used a DSP card (TMS320C6713), a Laptop computer, an amplifier, a loudspeaker and two microphones in two applications. In the first application, two microphones were placed close to the loudspeaker, while in the other application, one microphone was placed close to loudspeaker and speech trial was implemented in the far-end microphone. Output of the adaptive filter was observed for mu values of 0, 0.1, 100 and optimum (a value between 0.01 and 100). The best results in the adaptive filter were obtained from optimum mu value. (C) 2009 Elsevier Inc. All rights reserved.

关键词： Echo cancellation Adaptive filtering Adaptation rate lms algorithm FIR filter DSP

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Convergence analysis of a twin-reference complex least-mean-squares algorithm

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IEEE TRANSACTIONS ON SPEECH AND AUDIO PROCESSING 2002年第4期10卷 213-221页

作者： Johansson, S Nordebo, S Claesson, I Blekinge Inst Technol Dept Telecommun & Signal Proc S-37225 Ronneby Sweden

In many noise control applications, the noise is dominated by low frequencies and generated by several independent periodic sources. In such situations the tonal noise may be suppressed by using a narrowband multiple-reference feedforward controller. The performance characteristics of the control system, e.g., the convergence behavior and noise reduction are directly related to the controller adaptation rate as well as the frequency separation of the tonal components in the noise, i.e., the beat frequency. This paper treats the convergence performance of a complex least-mean-squares (lms) algorithm using two reference signals. An analysis of its convergence behavior is presented as well as the results from computer simulations validating the convergence behavior. The convergence of the filter weights and the decrease rate of the squared error (the learning curve) for noise control applications are also discussed.

关键词： active noise control complex algorithm convergence performance feedforward control lms algorithm

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Skin-Induced Motion Artifact Removal for Ambulatory Electrocardiography

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IEEE SENSORS JOURNAL 2022年第15期22卷 15033-15043页

作者： He, Zhuobiao Min, Hao Fudan Univ Inst Microelect Shanghai 200433 Peoples R China

Skin-induced motion artifact ( SIMA) is a troublesome problem for ambulatory electrocardiography (ECG) signal acquisition by e-skin dry sensor. Reference based algorithm is a promising way to tackle it. A highly correlated reference signal is the key for this solution. In this article, by using a new synthesized reference signal for lms algorithm, which combines both skin induced impedance behavior and skin "slope" and recovery "time constant", the SIMA related noise is greatly reduced. Firstly, we show the phenomenon that the SIMA "slope" and "recovery time constant" have some relationshipswith skin induced impedance, its derivatives and specific tissue characteristics. Secondly, We do the offline "slope" and "time constant" calibrationwork first, then measure the real time skin induced impedance, and then combine the above information together, to form a virtual reference signal, then we show the new reference has high correlation with actual SIMA noise. Finally, considering that SIMA potential consists of fast changing part and slow changing part, we use an improved lms algorithm to handle the contradiction between speed and accuracy, which includes fast tracking part and low convergence error part. We set up the SIMA removal system in Matlab, and use actual motion artifact contaminated ECG signal as the input, and then do offline signal processing. Results show that SIMA is improved around 7 similar to 15dB typically on average. Integrating skin-induced impedancewith human body related "slope" and "time constant" as a reference signal seems a feasible method to remove skin-inducedmotion artifact.

关键词： ECG electrode-tissue impedance lms algorithm motion artifacts reference signal skin impedance skin-induced motion artifact time constant

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BROAD-BAND POWER INVERSION ARRAY WITH MAXIMALLY FLAT RESPONSE AT NULL DIRECTIONS

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IEE PROCEEDINGS-F RADAR AND SIGNAL PROCESSING 1989年第4期136卷 161-167页

作者： KO, CC Department of Electrical Engineering National University of Singapore Singapore Singapore

The paper presents and investigates a new lms-based algorithm for a broadband power inversion array using tapped delay line processing. In this algorithm, the signals from eachL-tap delay line are first transformed and separated intoL/2 real-and-quadrature signals, with thelth signal determining the (l−1)th derivative of the frequency response of the array. Rejection of jammers is then obtained by using the lms algorithm to update the processing for the first transformed signals from all the array elements, while the processing for the other signals is adjusted to constrain the frequency response of the array to be maximally flat at the null directions. Since this latter adjustment needs to be performed only sparingly and the signal transformation is simple and straightforward, the complexity of the new algorithm is comparable to that when the lms algorithm is used directly. However, because there is now only one group of real-and-quadrature signals that needs to be adaptively processed, the new algorithm can be significantly faster than the lms algorithm.

关键词： least squares approximations transformed signals antenna phased arrays lms algorithm broadband power inversion array Signal processing and detection maximally flat response Interpolation and function approximation (numerical analysis) jammers rejection signal processing null directions Antenna arrays Information theory frequency response lms-based algorithm tapped delay line processing

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Lattice PFBlms: Fast converging structure for efficient implementation of frequency-domain adaptive filters

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SIGNAL PROCESSING 1999年第1期78卷 79-89页

作者： Chan, KS Farhang-Boroujeny, B Natl Univ Singapore Dept Elect Engn Singapore 119260 Singapore

It has been demonstrated that the unconstrained partitioned fast block lms (PFBlms) algorithm suffers from an inherently slow convergence rate. A solution to this problem has previously been proposed by the second author of this paper, which reduces the correlation by reducing the overlap between partitions. A drawback of this solution is that it requires more memory than the conventional PFBlms algorithm. In this paper, we propose an alternative solution that does not require any additional memory. A lattice structure is used instead to remove the correlation and increase the convergence rate. (C) 1999 Elsevier Science B.V. All rights reserved.

关键词： frequency-domain adaptive filter lms algorithm partitioning

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Experimental Validation of ADALINE Least Mean Square algorithm in a Three-Phase Four-Wire DSTATCOM to Enhance Power Quality

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ELECTRIC POWER COMPONENTS AND SYSTEMS 2020年第8期48卷 769-780页

作者： Penthia, Trilochan Panda, Anup Kumar Mangaraj, Mrutyunjaya Zhejiang Univ Coll Elect Engn Hangzhou Peoples R China Natl Inst Technol Dept Elect Engn Rourkela India Lendi Inst Engn & Technol Elect & Elect Engn Vizianagaram India

The requirement of a robust and adaptive control algorithm in active filters is inevitable nowadays. In this paper, an ADAptive LINEar neuron (ADALINE) based Least Mean Square (lms) control algorithm is proposed for a three-phase-four-wire (3P4W) distribution power system under unbalanced loading conditions. The proposed controller has a unique inbuilt characteristic to extract the fundamental active positive sequence component from an unbalanced load current. Particularly, the control algorithm is intended to calculate an adaptive and suitable amplitude for the three-phase reference source currents which are used for the switching pulse generation in the subsequent steps. Moreover, the Distribution STATic COMpensator (DSTATCOM) is performed to eliminate harmonic contents of supply current and harmful excessive neutral current in the system. Further, the DC-link voltage is regulated in order to reduce the power loss across the DSTATCOM, and to attain a balanced, high-quality power availability at the Point of Common Coupling (PCC). The proposed system is accomplished in the MATLAB2014a/Simulink, and a low power rated 3P4WDSTATCOM prototype incorporating with the dSPACE1104 real-time environment to verify the proposed control algorithm.

关键词： ADALINE lms algorithm power quality custom power device (CPD) THD DSTATCOM harmonics reduction neutral current elimination

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Design of Complex Non-Linear Adaptive Equalizer in Mitigating Severe Intersymbol Interferences

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JOURNAL OF SIGNAL PROCESSING SYSTEMS FOR SIGNAL IMAGE AND VIDEO TECHNOLOGY 2016年第2期84卷 225-236页

作者： Mandal, A. Mishra, R. Gautam Buddha Univ Sch Informat & Commun Technol Yamuna Expressway 201308 Greater Noida U India

In this paper, we propose an efficient architectural design for Adaptive Decision Feedback Equalizer (ADFE) for high-speed reliable data transmission through wired as well as wireless communication channels to minimize Inter Symbol Interference (ISI). High hardware cost of the system makes an urge to redesign it in order to minimize overall circuit complexity and cost. Realization of CORDIC based pipelined ADFE architecture using reformulated lms algorithm has not only reduced multipliers in its feed forward path, the flexibility and modularity of the components makes it possible for easy implementation also. The realization results area efficient, high speed and throughput with good convergence. The efficacy of the proposed equalizer is corroborated with MATLAB simulations for mitigation of severe channel distortion arise due to ISI in high speed communication systems.

关键词： Inter symbol interference Adaptive decision feedback equalizer CORDIC lms algorithm Pipelined multiplier

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Riemann-Liouville operator-based fractional normalised least mean square algorithm with application to decision feedback equalisation of multipath channels

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IET SIGNAL PROCESSING 2016年第6期10卷 575-582页

作者： Shah, Syed Muslim Capital Univ Sci & Technol Dept Elect Engn Islamabad Pakistan

In this study, the authors modify the mapping function by introducing a non-linear term based on Riemann-Liouville definition of fractional derivatives and its application to the mean squared error in addition to the first-order partial derivatives;thus create fractional variants of the least mean square (lms) algorithm and its normalised version. The introduction of fractional term helps increase the convergence rate through the non-linear update term which depends on the fractional order and the in-process lms weights;for steep gradient of the error measure, large changes are made to the weights which help the equaliser filter to better track the effects of multipath fading channels. They verify and validate the working of the proposed technique in the decision feedback equalisation of multipath fading channels for higher-order quadrature amplitude modulations;comparative results are shown in terms of performance metrics as symbol-error rate for various fractional orders and step sizes, combined equaliser and channel responses for different number of training symbols;simulations show that the proposed approach outperforms the conventional counterpart.

关键词： multipath channels fading channels least mean squares methods decision feedback equalisers gradient methods filtering theory quadrature amplitude modulation error statistics Riemann-Liouville operator fractional normalised least mean square error algorithm decision feedback equalisation mapping function modification first-order partial derivative lms algorithm steep gradient equaliser filter multipath fading channel higher-order quadrature amplitude modulation symbol-error rate

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Analog IC-achievable lineariser for power amplifiers based on adaptive feedforward

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ANALOG INTEGRATED CIRCUITS AND SIGNAL PROCESSING 2002年第3期33卷 231-238页

作者： Albertí, EB Sahad, AJZ Univ Politecn Cataluna Dept Signal Theory & Commun Barcelona 08034 Spain

A compensation technique for the amplitude and phase imbalances of a feedforward lineariser using the lms algorithm, implemented entirely in the analog domain, is presented. The lineariser is suited to RF power amplifiers, and maybe entirely developed with analog IC technologies. From simulated two-tone tests, the circuit leads to an intermodulation (IMD) reduction superior to 35 dB for a 64-QAM signal, being cheaper and competitive in performance with usual DSP-based adaptation circuits. Different implementations are evaluated, and the application of the circuit to RF power amplifiers in the DVB-T and in the LMDS frequency bands is considered.

关键词： feedforward lineariser power amplifiers lms algorithm analog signal processing analog integrated circuits

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