A stable and distortion-free IIR echo and howling canceler

The authors propose a configuration for an infinite impulse response (IIR) adaptive echo and howling canceller, which consists of a two-channel maximum entropy lattice filter and its inverse filter. The echo is canceled by the adaptive lattice filter, while the signal distortion is eliminated by the inverse lattice. With stability guaranteed without the necessity of testing, the structure costs O (N) multiplications per sampling period. The algorithm can also be greatly simplified for white input cases     

A multirate acoustic echo canceler structure

A new subband echo canceler (SBEC) structure is proposed to reduce the transmission delay introduced by conventional SBEC structures, without distorting the near-end signal. The proposed structure is based on computing two output errors, one for using during single-talk and the other one for using during double-talk periods. With the SBEC structure we propose a double-talk detector with a subband configuration which allows a fast and accurate detection of double-talk periods, enabling the SBEC algorithm to track changes in the echo path impulse response when the near-end signal is absent. Computer simulations using actual speech signals, and subjective evaluation tests are given to show the convergence performance, tracking and double-talk detection ability, of the proposed scheme     

An acoustic echo canceler

The authors describe an experimental 1000-tap single-chip adaptive AEC (acoustic echo canceler) occupying 28 mm² of die area in 3-μm, double-metal, p-well, CMOS technology. A floating-point format and power-of-two multiplications are chosen to simplify the hardware. To exploit pipelining and parallelism, interleaved data storage and multibank memory sharing the same addresses are designed. Hardware minimization is considered from both the system and the architecture perspective. In a loudspeaker telephone application, 27 dB of echo reduction is achieved after 1 s of convergence time     

A stereo echo canceler with correct echo-path identification basedon an input-sliding technique

A new stereo echo canceler with correct echo-path identification based on an input-sliding technique is proposed. A time-varying filter located in one of the two channels periodically delays the input signal. By this input sliding, the correct echo-path identification is achieved. Aliasing components and audible clicks by input-sliding are made inaudible by selecting appropriate parameter values for the time-varying filter. Simulations with the NLMS algorithm and a white Gaussian signal confirm the correct echo-path identification. The subjective quality of the input signal with slides is 4.38 based on the ITU-R five-grade impairment scale. Experimental results based on an implementation by 32-bit floating-point digital signal processors show that ERLE is not degraded by talker changes in the remote room. The mean opinion score is as much as 0.55-point higher than the conventional stereo echo canceler for different round-trip delays     

Improved tracking adaptive noise canceler for nonstationaryenvironments

The performance of multiple reference adaptive noise cancelers is investigated and a new filter structure is proposed that provides better tracking in the multipath, multisource, nonstationary automobile noise environment studied. The filter uses the least mean square (LMS) algorithm with multiple filtering stages and subbanded sections to improve the overall tracking performance while maintaining filter stability     

Cascaded power symmetric IIR filter banks and continuityconstrained adaptive algorithms for acoustic echo cancellation insubbands

Teleconferencing systems and hands-free mobile terminals use acoustic echo cancellation (AEC) for high-quality full-duplex speech communication. The problem of aliasing in subband AEC is addressed. Filter banks with implicit notch filtering are derived from cascaded power symmetric-infinite impulse response (CFS-IIR) filters. It is shown that adaptive filters used with these filter banks must be coupled via continuity constraints to reduce the aliasing in the residual echo. A continuity constrained NLMS algorithm is therefore proposed and evaluated     

A new method for adaptive IIR filter design based on tabu search algorithm

Adaptive digital filters have proven their worth in a wide range of applications such as channel equalisation, noise reduction, echo cancelling, and system identification. These filters can be broadly classified into two groups: finite impulse–response (FIR) and infinite impulse–response (IIR) filters. IIR filters have become the target of increasing interest because these filters can reduce the filter order significantly as compared to FIR filters. Tabu search is a heuristic optimisation algorithm which has been originally developed for combinatorial optimisation problems. It simulates the general rules of intelligent problem solving and has the ability of discovering the global minima in a multi-modal search space. In this work, a novel method based on tabu search is described for the design of adaptive IIR filters.     

A HOMOTOPY CONTINUATION ADAPTIVE IIR FILTERING ALGORITHM

A homotopy continuation adaptive HR filtering algorithm is proposed in this paper.The novel algorithm introduces the homotopy continuation method into the adaptive filtering soas to provide a high stability for adaptive HR filter without any forms of stability monitoringattached.     

The gamma-filter-a new class of adaptive IIR filters withrestricted feedback

Generalized feedforward filters, a class of adaptive filters that combines attractive properties of finite impulse response (FIR) filters with some of the power of infinite impulse response (IIR) filters, are described. A particular case, the gamma filter, generalizes Widrow's adaptive transversal filter (adaline) to an infinite impulse response filter. Yet, the stability condition for the gamma filter is trivial, and LMS adaptation is of the same computational complexity as the conventional transversal filter structure. Preliminary results indicate that the gamma filter is more efficient than the adaptive transversal filter. The authors extend the Wiener-Kopf equation to the gamma filter and develop some analysis tools     

Local echo canceler with optimal input for true full-duplex speechscrambling system

The purpose of this paper is to focus on the local echo-canceling problem of full-duplex scrambled speech communications over a two-wire telephone network when the scrambling transformation is located between the handset and body of a telephone. Such a design makes possible very efficient protection against electromagnetic compromising emanation, which in turn substantially enhances the overall security of a protected communication. We propose a new adaptive FIR filter algorithm for local echo cancellation in such applications. The proposed algorithm differs from the conventional one by the construction of input signals in an optimal way using the D-optimal experiment design. In this way, at each step, we generate a new sample of the D-optimal pilot sequence for the filter parameter estimation. Consequently, the adaptation of the local echo canceler is defined as an initialization process in the first phase of each protected telephone call. The advantage in using the proposed adaptive FIR echo canceler is demonstrated through simulation results     

New orthogonalised IIR adaptive filter

A new IIR-ADF algorithm is proposed in which the discrete cosine transform, a fairly good approach of KLT for a large number of signal classes, is used to split the input signal into a set of nearly orthogonal transform coefficients. Subsequently these coefficients are filtered by a bank of IIR-ADFs whose coefficients are independently updated to minimise a common error. Computer simulations show that the proposed algorithm provides a faster convergence rate than the Gauss-Newton algorithm with a much reduced computational load     

Cascade lattice IIR adaptive filters

The feedback lattice filter forms, including the two-multiplier form and the normalized form, are examined with respect to their relationships to the feedback direct form filter. Specifically, the transformation matrix between the lattice forms and the direct form is derived; parameter and state relationships between the lattice forms and the direct form are therefore obtained. An IIR filter structure-the cascade lattice IIR structure-is constructed. Based on this structure, three IIR adaptive filtering algorithms in the two-multiplier form can then be developed following the gradient approach, the Steiglitz-McBride approach and the hyperstability approach. Convergence of these algorithms is theoretically analyzed using either the ODE approach or the hyperstability theorem. These algorithms are then simplified into forms computationally as efficient as their corresponding direct form algorithms. Relationships of the simplified algorithms to the direct form algorithms are also studied, which disclose a consistency in algorithm structure regardless of the filter form. Three normalized lattice algorithms can also be derived from the two-multiplier lattice algorithms. Experimental results show much improved performance of the normalized lattice algorithms over the two-multiplier lattice algorithms and the direct form algorithms     

A new adaptive recursive RLS-based fast-array IIR filter for active noise and vibration control systems

Infinite impulse response filters have not been used extensively in active noise and vibration control applications. The problems are mainly due to the multimodal error surface and instability of adaptive IIR filters used in such applications. Considering these, in this paper a new adaptive recursive RLS-based fast-array IIR filter for active noise and vibration control applications is proposed. At first an RLS-based adaptive IIR filter with computational complexity of order O(n²) is derived, and a sufficient condition for its stability is proposed by applying passivity theorem on the equivalent feedback representation of this adaptive algorithm. In the second step, to reduce the computational complexity of the algorithm to the order of O(n) as well as to improve its numerical stability, a fast array implementation of this adaptive IIR filter is derived. This is accomplished by extending the existing results of fast-array implementation of adaptive FIR filters to adaptive IIR filters. Comparison of the performance of the fast-array algorithm with that of Erikson’s FuLMS and SHARF algorithms confirms that the proposed algorithm has faster convergence rate and ability to reach a lower minimum mean square error which is of great importance in active noise and vibration control applications.     

A mildly weaker sufficient condition in IIR adaptive filtering

The cross-covariance matrix of two stable autoregressive (AR) sequences is considered. A mildly weaker condition is identified that ensures the nonsingularity of this matrix. As one consequence of this result, a weaker sufficient condition is obtained that would guarantee the unimodality of the mean-square output error surface of an IIR adaptive filter with white noise excitation     

一种新NLMS自适应滤波算法及其在多路回波消除中的应用

提出一种NLMS改进算法并对其收敛性进行了证明。该算法计算复杂度低于Sankaran(1997)所提出的带有正交改正因子的归一化算法(NLMS-OCF)和仿射投影算法(APA),并具有易于实现等特点。仿真结果表明,以单路语音信号作输入时,新算法具有比NLMS-OCF算法更好的收敛速度和精度,而在收敛速度和精度相当的情况下,新算法比APA算法所占用的CPU时间少。将新算法扩展成两路算法后,扩展算法仍然保持了这些特点,与Sankaran(1999)两路NLMS-OCF及Benesty(1996)所提多路仿射算法(APA-MC)相比,新算法更适合于应用到多路回波消除等实时性要求高的场合。1     

Continuously adaptive echo cancellers

Several solutions to the problem of echo cancellation on voice channels, having characteristics which may vary with a frequency of several hertz, are presented. The results of an analysis which lends itself to several original realisations of a device to achieve the required performance are summarised.     

High-speed adaptive echo canceller

To solve the problem of echo cancellation in satellite communication links, an easily calculated algorithm with high-speed tracking properties is proposed.     

A length adaptive,transversal data echo cancelor

This paper deals with the estimation problem of echo cancelor length as applied to LMS data echo cancellation. To this end a Stochastic Gradient transversal data echo cancelor will be proposed which, over simulated channels, was able to adapt to the length of the returned echo in such a way as to limit the cancelor length and, as a consequence, enable the cancelor to converge faster. It will be shown that although final convergence is not improved, the fact that this cancelor can adapt more quickly can be valuable in cases where the echo cancelor has to be trained in a finite number of iterations.     

The feedback adaptive line enhancer: a constrained IIR adaptivefilter

A new adaptive line enhancer (ALE) structure, called the Feedback ALE (FALE), is presented and is shown to be a constrained W adaptive filter. Extensive simulations show that the FALE gives a higher sine-to-broadband ratio (SBR) gain and smaller sine estimation error than does an equal-order ALE; conversely, the order of the FALE can be much low er than the ALE to achieve equivalent performance