A simplified variable step-size LMS algorithm for Fourier analysis and its statistical properties☆
Introduction
Adaptive Fourier analysis finds applications in a lot of fields, such as digital communications, power engineering, speech signal processing, music signal processing, biomedical engineering, and control systems, see, e.g., [1], [2], [3], [4], [5], [6], [7], [8], [9]. It enjoys two major advantages over the DFT and its variants [7]. First, it is able to deal with arbitrary intended frequency that does not need to be an integer multiple of the DFT fundamental frequency. Second, it is of recursive nature, inherently capable of tracking nonstationary signals.
So far, various adaptive Fourier analyzers have been proposed and applied in real-life applications, see [2], [3], [4], [5], [6], [7], [8], [9] and references therein. The LMS algorithm was applied in [2], [3], [4] to make the DFT adaptive and useful for nonstationary signals. The same LMS algorithm was discussed and analyzed in [5], [6] for real-valued signals with arbitrary frequencies like music signals generated by electronic keyboards. A set of gradient-based algorithms was proposed based on a p-power error criterion, and their statistical analysis was performed in [7]. This p-power algorithm reduces exactly to the LMS algorithm when p is set to 2. The above-mentioned gradient algorithms provide reasonably good discrete Fourier coefficient (DFC) estimates and require a small number of multiplications [2], [3], [4], [5], [6], [7]. However, their performance may become inadequate in applications that fast convergence and small estimation error are simultaneously imposed.
On the other hand, from 1990s a lot of variable step-size LMS (VSS-LMS) algorithms have been developed in the context of system identification, see, e.g., [10], [11], [12], [13], [14], [15] for some typical ones. Those algorithms enjoy fast convergence in the early stage of adaptation and present less misadjustment in the steady state. Furthermore, computational cost increase in those algorithms is quite moderate, rendering them very attractive for real applications. The use of VSS has been extended to many other signal processing applications, such as frequency estimation [16], adaptive notch filtering [17], and active noise control [18], [19]. To the best of our knowledge, there are only some preliminary results regarding the use of VSS in the realm of adaptive Fourier analysis [20], [21]. The trial presented in [20] is a simple extension of the VSS-LMS algorithm proposed in [10]. The VSS-LMS algorithm derived in [21] suffers from the risk of becoming divergent when the user parameters are not given properly. Further investigation is thus needed into the use of VSS in the context of adaptive Fourier analysis.
In this paper, first, several typical VSS-LMS algorithms derived for system identification or FIR filtering are applied to adaptive Fourier analysis of noisy sinusoidal signals. Their performance as well as computational efficiency of VSS update is compared in detail. Second, a simplified VSS-LMS (SVSS-LMS) algorithm is proposed to perform the Fourier analysis, which presents comparable or even improved performance as compared with those typical VSS-LMS algorithms, and requires less multiplications as well as user parameters. Third, in-depth performance analysis of the proposed algorithm is conducted. Difference equations that describe its dynamics and elegant closed-form expressions for its steady-state performance are derived in detail. Finally, extensive simulations are conducted to clarify the validity of the analytical findings.
The rest of the paper is organized as follows. Section 2 introduces seven (7) typical VSS-LMS algorithms, six (6) developed for system identification and one (1) derived for adaptive Fourier analysis. After comparing their computational complexity and performance, we propose a new SVSS-LMS algorithm based on the insight gained from extensive simulations. In Section 3, performance analysis of the proposed algorithm is conducted, and detailed simulations are provided to show the validity of the analytical results. Section 4 concludes the paper.
Section snippets
A simplified variable step-size LMS algorithm
The noisy sinusoidal signal of interest has discrete frequencies and is expressed bywhere q is the number of frequency components residing in the target signal, ωi is the frequency of the ith component, are the DFCs of frequency components, v(n) is a zero-mean additive white Gaussian noise with variance σ2v. The purpose of an adaptive Fourier analyzer is to estimate the DFCs of all frequency
Performance analysis
Generally speaking, performance analysis of an adaptive algorithm, if tractable and performable, will provide us with much insightful and conclusive information on its statistical properties. In-depth analysis of the SVSS-LMS algorithm will be given in this section from this point of view.
Now, we make some preparations for the analysis. First, we define the DFC estimation errors asUsing these definitions and (1) in (6), one may express the error signal by
Conclusions
A SVSS-LMS algorithm has been proposed for adaptive Fourier analysis of sinusoidal signals in additive noise. It enjoys attractive computational efficiency and very promising estimation and tracking capabilities in both stationary and nonstationary situations, in comparison with the existing VSS-LMS algorithms. Extensive simulations have been conducted to confirm the performance of the proposed algorithm. A complete set of difference equations and elegant steady-state closed-form expressions
References (22)
- et al.
LMS-based notch filter for the estimation of sinusoidal signals in noise
Signal Process.
(1995) - et al.
Real-valued LMS Fourier analyzer for sinusoidal signals in additive noise
Signal Process.
(1998) - et al.
Variable step-size LMS algorithm with a quotient form
Signal Process.
(2009) - et al.
A variable step-size adaptive algorithm for direct frequency estimation
Signal Process.
(2010) - International Conference on Harmonics in Power Systems (Power System Engineering Series), Institute of Science and...
The Adaptive Spectrum Analyzer
(1984)- et al.
Adaptive Fourier estimation of time-varying evoked potentials
IEEE Trans. Biomed. Eng.
(1989) - et al.
The adaptive spectrum analysis for transcription
Trans. Soc. Instrum. Control Eng.
(1992) - et al.
Adaptive algorithm based on least mean p-power error criterion for Fourier analysis in additive noise
IEEE Trans. Signal Process.
(1999) - et al.
Notch Fourier transform
IEEE Trans. Acoust. Speech Signal Process.
(1987)