A novel sign adaptation scheme for convex combination of two adaptive filters

Lu, Lu; Zhao, Haiquan; He, Zhengyou; Chen, Badong

doi:10.1016/j.aeue.2015.07.009

Cited by 25 publications

(15 citation statements)

References 26 publications

(43 reference statements)

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“…Namely, a large t reduces the steady-state error and overall computational complexity of the proposed algorithms while slows the convergence rate; conversely, a small t improve the convergence rate of the proposed algorithms while increases their steady-state error and overall computational complexity. Nevertheless, based on our extensive simulations, it is found that t ∈ [1,4] for the SM-INSAF and SM-IP-INSAF algorithms, and t ∈ [0.7, 0.9] for the SSM-INSAF and SSM-IP-INSAF algorithms can achieve good trade-off performance. Remark 4: For a set-membership algorithm, the computational cost consists of two parts: C up and C nup , where C up >> C nup , C up accounts for the operations when an update is performed, and C nup accounts for the operations when no update is performed.…”

Section: Discussion Remarkmentioning

confidence: 95%

Set‐membership improved normalised subband adaptive filter algorithms for acoustic echo cancellation

Zhao

Chen

2018

IET signal process.

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In order to improve the performances of recently-presented improved normalized subband adaptive filter (INSAF) and proportionate INSAF algorithms for highly noisy system, this paper proposes their set-membership versions by exploiting the theory of set-membership filtering. Apart from obtaining smaller steady-state error, the proposed algorithms significantly reduce the overall computational complexity. In addition, to further improve the steady-state performance for the algorithms, their smooth variants are developed by using the smoothed absolute subband output errors to update the step sizes. Simulation results in the context of acoustic echo cancellation have demonstrated the superiority of the proposed algorithms.Keywords. Acoustic echo cancellation, improved normalized subband adaptive filter, low signal-noise-ratio, setmembership filtering, sparse system IntroductionIn modern communication network, hands-free telephony and teleconference, the acoustic echo is a common problem that must be eliminated to improve the call quality. In recent decades, therefore, acoustic echo cancellation (AEC) based on adaptive filter has obtained significant attention [1]- [3]. Naturally, AEC is also a system identification problem (i.e., identifying the impulse response of the echo path), but it imposes several characteristics on the adaptive filter. Namely, 1) the input is the speech signal which is nonstationary and highly colored, and 2) the impulse response of the echo path is long and sparse. Owing to its simplicity and ease of implementation, the normalized least mean square (NLMS) algorithm is often used in AEC. However, the NLMS algorithm will suffer from slow convergence when the input signal is colored.To overcome this drawback, the affine projection (AP) family uses K past input signal vectors for updating the filter' coefficients, at each iteration, where K is the projection order [4], [5]. The convergence rate of AP increases as K increases, but meanwhile the steady-state error and computational cost of that become large. To reduce the computational complexity, several AP algorithms with variable projection order [6] as well as fast AP algorithms [7] were opened.

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Section: Discussion Remarkmentioning

confidence: 95%

Set‐membership improved normalised subband adaptive filter algorithms for acoustic echo cancellation

Zhao

Chen

2018

IET signal process.

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“…where B is the wordlength of coefficients. By substituting (12) into (1) and after simplification, we arrive at…”

Section: Mathematical Formulationmentioning

confidence: 99%

“…In addition, it also needs six-multipliers to compute the combined filter output and update of auxiliary variable. However, this has been reduced to three multipliers in [12] by a sign-adaptation scheme. While the proposed algorithm does not involve auxiliary variable and mixing parameter, but still provides the instantaneous transfer of coefficients.…”

Section: Computational Complexitiesmentioning

confidence: 99%

“…8 a that the convergence of the proposed filter inthe fixed‐point implementation is similar to the actual implementation of theCLMS ADF. This is because the proposed filter does not utilise theapproximations made in [11, 12], rather it depends on the correlationof adjacent errors

e_{n - m}

for

m \in false[1, 2 false]

. Moreover, it is found to be stable andshowed similar numerical properties as that of its floating‐pointimplementation.…”

Section: Performance Comparisonmentioning

confidence: 99%

“…Later, Nascimento , et al [11] proposed a new low‐cost strategy for the transfer of filter coefficients from one LMS unit to other. Based on [11], Lu , et al [12] found a new sign‐adaptation scheme for the transfer of filter coefficients. However, the complexity of the LMS filter units is still higher, since it contains a finite‐impulse‐response (FIR) filter, which is made of multiply‐and‐accumulate (MAC) units.…”

Section: Introductionmentioning

confidence: 99%

See 2 more Smart Citations

Improved convergent distributed arithmetic based low complexity pipelined least‐mean‐square filter

Khan¹,

Ahamed²,

Prakash³

2018

IET Circuits, Devices & Systems

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This study presents an improved convergent distributed arithmetic (DA)-based low complexity pipelined least-meansquare filter. The concept is based on a convex combination of two adaptive filters (ADFs) where the convergence performance of the combined filter is adjusted by the step-sizes of ADFs. The proposed technique replaced two ADF units by a single unit of the DA-based ADF. Further reduction in hardware complexity is achieved by sharing the filter partial products. Moreover, a bitlevel coefficient update unit is employed to minimise its hardware complexity. In addition, a novel low-cost strategy is presented to improve the convergence performance of the proposed filter by comparing the time-window corresponding to the maximum correlation of delayed error signals e(n − m) with a pre-defined window with n being time instant and m ∈ [1, 2]. Compared with the best existing scheme, the proposed design offers 46.42% fewer adders, 36.69% fewer registers and 18.75% fewer multiplexers for a 64th-order filter. Application specific integrated circuit synthesis results show that the proposed design occupies 37.10% less chip-area and consumes 24.79% less power. In addition, the proposed design provides 20.35% less area-delay-product and 4.76% less energy-per-sample for 64th order with the fourth-order base unit over the best existing scheme.

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Adaptive Combination of Proportionate NSAF with Individual Activation Factors

Dong

Zhao

2016

Circuits Syst Signal Process

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A novel sign adaptation scheme for convex combination of two adaptive filters

Cited by 25 publications

References 26 publications

Set‐membership improved normalised subband adaptive filter algorithms for acoustic echo cancellation

Set‐membership improved normalised subband adaptive filter algorithms for acoustic echo cancellation

Improved convergent distributed arithmetic based low complexity pipelined least‐mean‐square filter

Adaptive Combination of Proportionate NSAF with Individual Activation Factors

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