scispace - formally typeset
Journal ArticleDOI

Exponentially weighted stepsize NLMS adaptive filter based on the statistics of a room impulse response

Shoji Makino, +2 more
- 01 Jan 1993 - 
- Vol. 1, Iss: 1, pp 101-108
TLDR
A normalized least-mean-squares (NLMS) adaptive algorithm with double the convergence speed, at the same computational load, of the conventional NLMS for an acoustic echo canceller is proposed and its fast convergence is demonstrated.
Abstract
A normalized least-mean-squares (NLMS) adaptive algorithm with double the convergence speed, at the same computational load, of the conventional NLMS for an acoustic echo canceller is proposed. This algorithm, called the ES (exponentially weighted stepsize) algorithm, uses a different stepsize (feedback constant) for each weight of an adaptive transversal filter. These stepsizes are time-invariant and weighted proportionally to the expected variation of a room impulse response. The algorithm adjusts coefficients with large errors in large steps, and coefficients with small errors in small steps. A transition formula is derived for the mean-squared coefficient error of the algorithm. The mean stepsize determines the convergence condition, the convergence speed, and the final excess mean-squared error. Modified for a practical multiple DSP structure, the algorithm requires only the same amount of computation as the conventional NLMS. The algorithm is implemented in a commercial acoustic echo canceller, and its fast convergence is demonstrated. >

read more

Citations
More filters
Journal ArticleDOI

Proportionate normalized least-mean-squares adaptation in echo cancelers

TL;DR: On typical echo paths, the proportionate normalized least-mean-squares (PNLMS) adaptation algorithm converges significantly faster than the normalized at-a-glance-time (NLMS) algorithm generally used in echo cancelers to date.
Journal ArticleDOI

Acoustic echo control. An application of very-high-order adaptive filters

TL;DR: The application of high-order adaptive filters to the problem of acoustical echo cancellation with particular application to hands free telephone systems is discussed and a means to achieve robust performance is described.
Journal ArticleDOI

A new class of gradient adaptive step-size LMS algorithms

TL;DR: A simplification to a class of the studied algorithms is proposed and it is shown that this leads to a new class of VSLMS algorithms with reduced complexity but with no observable loss in performance.
Book

Sparse Adaptive Filters for Echo Cancellation

TL;DR: This book presents the most important sparse adaptive filters developed for echo cancellation and proposes some new solutions for further performance improvement, e.g., variable step-size versions and novel proportionate-type affine projection algorithms.
Journal ArticleDOI

Modified LMS algorithms for speech processing with an adaptive noise canceller

TL;DR: In this paper, two modified least mean squares (LMS) algorithms, the weighted sum and sum methods, were proposed to solve the problem by reducing the size of the steps in the weight update equation when the desired signal is strong.
References
More filters
Book

Adaptive Filter Theory

Simon Haykin
TL;DR: In this paper, the authors propose a recursive least square adaptive filter (RLF) based on the Kalman filter, which is used as the unifying base for RLS Filters.
Book

Adaptive Signal Processing

TL;DR: This chapter discusses Adaptive Arrays and Adaptive Beamforming, as well as other Adaptive Algorithms and Structures, and discusses the Z-Transform in Adaptive Signal Processing.
Journal ArticleDOI

Adaptive noise cancelling: Principles and applications

TL;DR: It is shown that in treating periodic interference the adaptive noise canceller acts as a notch filter with narrow bandwidth, infinite null, and the capability of tracking the exact frequency of the interference; in this case the canceller behaves as a linear, time-invariant system, with the adaptive filter converging on a dynamic rather than a static solution.
Journal ArticleDOI

New Method of Measuring Reverberation Time

TL;DR: In this article, a new method of measuring reverberation time was described, which uses tone bursts (or filtered pistol shots) to excite the enclosure and a simple integral over the toneburst response of the enclosure yields, in a single measurement, the ensemble average of the decay curves that would be obtained with bandpass filtered noise as an excitation signal.
Journal ArticleDOI

A learning method for system identification

TL;DR: In this paper, a method for system identification is proposed which is based on the error-correcting training procedure in learning machines, and is referred to as learning identification, which is applicable to cases where the input signal is random and nonstationary, and can be completed within a short time, so that it may be used to identify linear quasi-time-invariant systems in which some parameters vary slowly in comparison with the time required for identification.