This paper is about an efficient implementation of adaptive filtering for echo cancelers. The first objective of this paper is to propose a simplified method of the flexible block Multi-Delay Filter (MDF) algorithm in the time-domain. Then, we will derive a new method for the step-size adaptation coefficient. The second objective is about the realization of a Block Proportionate Normalized Least Mean Squares (BPNLMS++) with the simplified MDF (SMDF) implementation. Using the new step-size method and the smaller block dimension proposed by SMDF, we achieve a faster convergence of the adaptive process with a limited computational cost. Then, an efficient implementation of the new procedure (SMDF-BPNLMS++) block filtering is proposed using Fermat Number Transform, which can significantly reduce the computation complexity of filter implantation on Digital Signal Processor.

This paper presents a novel stereophonic acoustic echo canceling scheme without preprocessing. To accurately estimate echo path keeping the high level of performance in echo erasing, this scheme uses two filters, of which one filter is utilized as a guideline which does not erases echo but helps updating of the other filter, which actually erases echo. In addition, we propose a new filter dividing technique to apply to the filter divide scheme, and utilize this as the guideline. Numerical examples demonstrate that the proposed scheme improves the convergence behavior compared to conventional methods both in system mismatch (i.e., normalized coefficients error) and Echo Return Loss Enhancement (ERLE).

A new stereo echo canceler with input slides and counter-lateralization is proposed. Convergence of filter coefficients to the correct echo paths is obtained by pre-processing which delays the input signal periodically by one sample in one of the two channels. The time difference between the two stereo components of the input signals causes a shift of the sound image. This shift is compensated for by presenting the delayed component of the stereo signals to a loudspeaker at a higher intensity, and the other component at a lower intensity. Correct echo-path identification is analytically shown in a more general form than in the preceding literatures. A subjective listening test shows that this method is more effective for vocal musics. The processed signals are scored 0.45 lower than the original input signals, using the ITU-R five-grade impairment scale.

In this paper, we propose two adaptive filtering schemes for Stereophonic Acoustic Echo Cancellation (SAEC), which are based on the adaptive projected subgradient method (Yamada et al., 2003). To overcome the so-called non-uniqueness problem, the schemes utilize a certain preprocessing technique which generates two different states of input signals. The first one simultaneously uses, for fast convergence, data from two states of inputs, meanwhile the other selects, for stability, data based on a simple min-max criteria. In addition to the above difference, the proposed schemes commonly enjoy (i) robustness against noise by introducing the stochastic property sets, and (ii) only linear computational complexity, since it is free from solving systems of linear equations. Numerical examples demonstrate that the proposed schemes achieve, even in noisy situations, compared with the conventional technique, (i) much faster and more stable convergence in the learning process as well as (ii) lower level mis-identification of echo paths and higher level Echo Return Loss Enhancement (ERLE) around the steady state.

This paper proposes a scalable multiecho cancellation system based on multiple autonomic and distributed echo canceler units. The proposed system does not have any common control section. Distributed control sections are equipped with in multiple echo cancelers operating autonomically. Necessary information is transferred from one unit to the next one. When the number of echoes to be canceled is changed, the necessary number of echo canceler units, each of which may be realized on a single chip, are simply plugged in or unplugged. The proposed system also provides fast convergence thanks to the novel coefficient location algorithm which consists of flat-delay estimation and constrained tap-position control. The input signal is evaluated at each tap to determine when to terminate flat-delay estimation. The number of exchanged taps is selected larger in flat-delay estimation than in constrained tap-position control. The convergence time with a colored-signal input is reduced by approximately 50% over STWQ, and 80% over full-tap NLMS algorithm. With a real speech input, the proposed system cancels the echo by about 20 dB. Tap-positions have been shown to be controlled correctly.

This paper proposes a fast timing recovery method with a decision feedback equalizer for baudrate sampling. The proposed method features two special techniques. The first one is for coarse estimation of the sampling phase. Internal signals of the oversampled analog-to-digital converter at different phases are directly taken out for parallel evaluation. The second technique provides fine tuning with a phase-modification stepsize which is adaptively controlled by the residual intersymbol interference. Simulation results by a full-duplex digital transmission system with a multilevel line code show superiority of the proposed method. The coarse timing estimation and the fine tuning reduce 75% and 40% of the time required by the conventional method,respectively. The overall saving in timing recovery is almost 60% over the conventional method. The proposed method could easily be extended to other applications with a decision feedback equalizer.

This paper presents a new structure for noise and echo cancelers based on a combined fast abaptive algorithm. The main purpose of the new structure is to detect both the double-talk and the unknown path change. This goal is accomplished by using two adaptive filters. A main adaptive filter Fn, adjusted only in the non-double-talk period by the normalized LMS algorithm, is used for providing the canceler output. An auxiliary adaptive filter Ff, adjusted by the fast RLS algorithm, is used for detecting the double-talk and obtaining a near optimum tap-weight vector for Fn in the initialization period and whenever the unknown path has a sudden or fast change. The proposed structure is examined through computer simulation on a noise cancellation problem. Good cancellation performance and stable operation are obtained when signal is a speech corrupted by a white noise, a colored noise and another speech signal. Simulation results also show that the proposed structure is capable of distinguishing the near-end signal from the noise path change and quickly tracking this change.