This Letter proposes an optimal gain filter for the perceptual acoustic echo suppressor. We designed an optimally-modified log-spectral amplitude estimation algorithm for the gain filter in order to achieve robust suppression of echo and noise. A new parameter including information about interferences (echo and noise) of single-talk duration is statistically analyzed, and then the speech absence probability and the a posteriori SNR are judiciously estimated to determine the optimal solution. The experiments show that the proposed gain filter attains a significantly improved reduction of echo and noise with less speech distortion.

This paper proposes an adaptive algorithm for identifying unknown systems containing nonlinear amplitude characteristics. Usually, the nonlinearity is so small as to be negligible. However, in low cost systems, such as acoustic echo canceller using a small loudspeaker, the nonlinearity deteriorates the performance of the identification. Several methods preventing the deterioration, polynomial or Volterra series approximations, have been hence proposed and studied. However, the conventional methods require high processing cost. In this paper, we propose a method approximating the nonlinear characteristics with a piecewise linear curve and show using computer simulations that the performance can be extremely improved. The proposed method can also reduce the processing cost to only about twice that of the linear adaptive filter system.

A residual acoustic echo cancellation method that employs the masking property is proposed to enhance the speech quality of hands-free communication devices in an automobile environment. The conventional masking property is employed for speech enhancement using the masking threshold of the desired clean speech signal. In this Letter, either the near-end speech or residual noise is selected as the desired signal according to the double-talk detector. Then, the residual echo signal is masked by the desired signal (masker). Experiments confirm the effectiveness of the proposed method by deriving the echo return loss enhancement and by examining speech waveforms and spectrograms.

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.

Broadband access service, including FTTH, is now in widespread use in Japan. More than half of the households that have broadband Internet access construct local area networks (home networks) in their homes. In addition, information appliances such as personal computers, networked audio, and visual devices and game machines are connected to home networks, and many novel service applications are provided via the Internet. However, it is still difficult to install and incorporate these devices and services because networked devices have been developed in different communities. I briefly explain the current status of information appliances and home networking technologies and services and discuss some of the problems in this and their solutions.

A C-band polarimetric radar on Okinawa Island successfully observed large-scale bird migrations over the western Pacific Ocean. The birds generated interesting polarimetric signatures. This paper describes the signatures and speculates bird behavior.

We propose a new robust frequency domain acoustic echo cancellation filter that employs a normalized residual echo enhancement. By interpreting the conventional robust step-size control approaches as a statistical-model-based residual echo enhancement problem, the optimal step-size introduced in the most of conventional approaches is regarded as optimal only on the assumption that both the residual echo and the outlier in the error output signal are described by Gaussian distributions. However, the Gaussian-Gaussian mixture assumption does not always hold well, especially when both the residual echo and the outlier are speech signals (known as a double-talk situation). The proposed filtering scheme is based on the Gaussian-Laplacian mixture assumption for the signals normalized by the reference input signal amplitude. By comparing the performances of the proposed and conventional approaches through the simulations, we show that the Gaussian-Laplacian mixture assumption for the normalized signals can provide a better control scheme for the acoustic echo cancellation.

In the acoustic echo canceller (AEC), the step-size parameter of the adaptive filter must be varied according to the situation if double talk occurs and/or the echo path changes. We propose an AEC that uses a sub-adaptive filter. The proposed AEC can control the step-size parameter according to the situation. Moreover, it offers superior convergence compared to the conventional AEC even when the double talk and the echo path change occur simultaneously. Simulations demonstrate that the proposed AEC can achieve higher ERLE and faster convergence than the conventional AEC. The computational complexity of the proposed AEC can be reduced by reducing the number of taps of the sub-adaptive filter.

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).

Owing to the large amount of speckle noise and ill-defined edges present in echocardiographic images, computer-based boundary detection of the left ventricle has proved to be a challenging problem. In this paper, a Markovian level set method for boundary detection in long-axis echocardiographic images is proposed. It combines Markov random field (MRF) model, which makes use of local statistics with level set method that handles topological changes, to detect a continuous and smooth boundary. Experimental results show that higher accuracy can be achieved with the proposed method compared with two related MRF-based methods.

We propose a gradient-limited affine projection algorithm (GL-APA), which can achieve fast and double-talk-robust convergence in acoustic echo cancellation. GL-APA is derived from the M-estimation-based nonlinear cost function extended for evaluating multiple error signals dealt with in the affine projection algorithm (APA). By considering the nonlinearity of the gradient, we carefully formulate an update equation consistent with multiple input-output relationships, which the conventional APA inherently satisfies to achieve fast convergence. We also newly introduce a scaling rule for the nonlinearity, so we can easily implement GL-APA by using a predetermined primary function as a basis of scaling with any projection order. This guarantees a linkage between GL-APA and the gradient-limited normalized least-mean-squares algorithm (GL-NLMS), which is a conventional algorithm that corresponds to the GL-APA of the first order. The performance of GL-APA is demonstrated with simulation results.

This paper proposes a new echo suppressor based on spectral correlation between the residual echo and the echo replica in an ordinary echo canceller. First, it is revealed by experiments that there is a significant correlation between the spectral amplitudes of the residual echo and the echo replica, and a new model for nonlinear-echo suppression based on the correlation is derived. Next, a new echo suppressor controlling the gain in each frequency bin to suppress the residual echo based on the new model is developed. Simulation results with speech data recorded by a hands-free cellphone show that the proposed echo suppressor reduces the residual echo to an almost inaudible level.

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.

This letter proposes the thinnest hybrid EM wave absorber using a composite magnetic material, which can be applied to the 3 m semi anechoic chambers. We experimentally designed a new hybrid EM wave absorber of the wedge shape, which was made from the ferrite powder, the inorganic fiber and binder. As a result, the length of this absorber could be realized only 6 cm, which was ascertained having the nonflammable. The 3 m semi anechoic chamber is constructed in the size of L9 mW6 mH5.7 m using this absorber, and then the site attenuation is measured according to ANSI C63.4 in the frequency range of 30 MHz-1 GHz. As a result, the measured normalized site attenuation is obtained within 3 dB to the theoretical normalized site attenuation.

This paper investigates the 10 m semi anechoic chamber using a new type hybrid EM wave absorber consisted of the grid-ferrite and the open-top hollow pyramidal EM wave absorber. We designed a new type hybrid EM wave absorber, which length could be slightly realized 65 cm. The 10 m semi anechoic chamber was constructed in the size of L21.5 mW13.5 mH8.9 m as the result of the ray-tracing simulation using this absorber. Then, the site attenuation in the constructed anechoic chamber was measured by using the broadband calculable dipole antennas. As the result, the maximum deviations between the measured site attenuation and theoretical calculated one were obtained within 3.6 dB in the frequency range of 30 MHz to 300 MHz. It was confirmed the validity of a new type hybrid EM wave absorber. Moreover, it was confirmed that the measured results agree with the ray-tracing simulation results, in which the differences are about 1.5 dB.

The site attenuation is an important parameter to evaluate an anechoic chamber. The ray-tracing method has been applied to analyze it. However, the lowest applicable frequency has not been cleared. In this paper, the FDTD method has been applied to analyze the site attenuation of a compact anechoic chamber from 30 MHz to 250 MHz, and this has been compared with the calculated one by the ray-tracing method to evaluate the lowest frequency where the ray-tracing method could be applied. The compact anechoic chamber, where the absorbers are placed on the all walls, has been used for the calculation. For FDTD analysis, the dipole antenna and the absorber have been modeled by using the large cell, whose size is larger than the diameter of the antenna element. For verification, the site attenuation of a compact anechoic chamber has been measured and compared with the calculated values by the FDTD method and the ray-tracing method. As the results, the calculated values by the ray-tracing method have larger deviation than the ones by the FDTD method when the frequency is less than 180 MHz.

Recently, automatic speech recognition in a car has practical uses for applications like car-navigation and hands-free telephone dialers. For noise robustness, the current successes are based on the assumption that there is only a stationary cruising noise. Therefore, the recognition rate is greatly reduced when there is music or news coming from a radio or a CD player in the car. Since reference signals are available from such in-vehicle units, there is great hope that echo cancellers can eliminate the echo component in the observed noisy signals. However, previous research reported that the performance of an echo canceller is degraded in very noisy conditions. This implies it is desirable to combine the processes of echo cancellation and noise reduction. In this paper, we propose a system that uses echo cancellation and spectral subtraction simultaneously. A stationary noise component for spectral subtraction is estimated through the adaptation of an echo canceller. In our experiments, this system significantly reduced the errors in automatic speech recognition compared with the conventional combination of echo cancellation and spectral subtraction.

This letter proposes a novel nonlinear distortion for the unique identification of receiving room impulses in stereo acoustic echo cancellation when applying the frequency-domain adaptive filtering technique. This nonlinear distortion is effective in reducing the coherence between the two incoming audio channels and its influence on audio quality is inaudible.

An adaptive interpolated FIR (IFIR) echo canceller was recently proposed for xDSL applications. This canceller consists of an FIR filter, an IFIR filter, and a tap-weight overlapping and nulling scheme. The FIR filter is used to cancel the short and rapidly changing head echo while the IFIR filter is used to cancel the long and slowly decaying tail echo. This echo canceller, which inherits the stable characteristics of the conventional FIR filter, requires low computational complexity. It is well known that the interpolation filter for an IFIR filter has great influence on the interpolated result. In this paper, a least-squares method is proposed to obtain optimal interpolation filters such that the performance of the IFIR echo canceller can be further improved. Simulations with a wide variety of loop topologies show that the optimal IFIR echo canceller can effectively cancel the echo up to 73.0 dB (for an SHDSL system). About 57% complexity reduction can be achieved compared to a conventional FIR filter.

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.